diff options
Diffstat (limited to 'sound/soc/intel')
80 files changed, 4526 insertions, 790 deletions
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 7c85d1bb9c12..ded903f95b67 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -216,7 +216,7 @@ config SND_SOC_INTEL_AVS depends on COMMON_CLK select SND_SOC_ACPI if ACPI select SND_SOC_TOPOLOGY - select SND_HDA + select SND_SOC_HDA select SND_HDA_EXT_CORE select SND_HDA_DSP_LOADER select SND_INTEL_DSP_CONFIG @@ -226,5 +226,8 @@ config SND_SOC_INTEL_AVS capabilities. This includes Skylake, Kabylake, Amberlake and Apollolake. +# Machine board drivers +source "sound/soc/intel/avs/boards/Kconfig" + # ASoC codec drivers source "sound/soc/intel/boards/Kconfig" diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 335c32732994..fd59b35a62ba 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -831,9 +831,9 @@ static int sst_get_ssp_mode(struct snd_soc_dai *dai, unsigned int fmt) dev_dbg(dai->dev, "Enter:%s, format=%x\n", __func__, format); switch (format) { - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: return SSP_MODE_PROVIDER; - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: return SSP_MODE_CONSUMER; default: dev_err(dai->dev, "Invalid ssp protocol: %d\n", format); @@ -1328,7 +1328,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) { struct sst_data *drv = snd_soc_dai_get_drvdata(dai); struct snd_soc_dapm_widget *w; - struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dapm_path *p; dev_dbg(dai->dev, "enter, dai-name=%s dir=%d\n", dai->name, stream); @@ -1392,7 +1392,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) static int sst_fill_module_list(struct snd_kcontrol *kctl, struct snd_soc_dapm_widget *w, int type) { - struct sst_module *module = NULL; + struct sst_module *module; struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); struct sst_ids *ids = w->priv; int ret = 0; diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index 3a42d68c0247..160b50f479fb 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -114,7 +114,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) static irqreturn_t intel_sst_irq_thread_mrfld(int irq, void *context) { struct intel_sst_drv *drv = (struct intel_sst_drv *) context; - struct ipc_post *__msg, *msg = NULL; + struct ipc_post *__msg, *msg; unsigned long irq_flags; spin_lock_irqsave(&drv->rx_msg_lock, irq_flags); diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 4e8382097e61..4e039c7173d8 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -28,7 +28,7 @@ struct sst_block *sst_create_block(struct intel_sst_drv *ctx, u32 msg_id, u32 drv_id) { - struct sst_block *msg = NULL; + struct sst_block *msg; dev_dbg(ctx->dev, "Enter\n"); msg = kzalloc(sizeof(*msg), GFP_KERNEL); @@ -63,7 +63,7 @@ struct sst_block *sst_create_block(struct intel_sst_drv *ctx, int sst_wake_up_block(struct intel_sst_drv *ctx, int result, u32 drv_id, u32 ipc, void *data, u32 size) { - struct sst_block *block = NULL; + struct sst_block *block; dev_dbg(ctx->dev, "Enter\n"); @@ -91,7 +91,7 @@ int sst_wake_up_block(struct intel_sst_drv *ctx, int result, int sst_free_block(struct intel_sst_drv *ctx, struct sst_block *freed) { - struct sst_block *block = NULL, *__block; + struct sst_block *block, *__block; dev_dbg(ctx->dev, "Enter\n"); spin_lock_bh(&ctx->block_lock); @@ -341,7 +341,7 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, } /* FW sent short error response for an IPC */ - if (msg_high.part.result && drv_id && !msg_high.part.large) { + if (msg_high.part.result && !msg_high.part.large) { /* 32-bit FW error code in msg_low */ dev_err(sst_drv_ctx->dev, "FW sent error response 0x%x", msg_low); sst_wake_up_block(sst_drv_ctx, msg_high.part.result, diff --git a/sound/soc/intel/avs/Makefile b/sound/soc/intel/avs/Makefile index b6b93ae80304..919212825f21 100644 --- a/sound/soc/intel/avs/Makefile +++ b/sound/soc/intel/avs/Makefile @@ -10,3 +10,6 @@ snd-soc-avs-objs += trace.o CFLAGS_trace.o := -I$(src) obj-$(CONFIG_SND_SOC_INTEL_AVS) += snd-soc-avs.o + +# Machine support +obj-$(CONFIG_SND_SOC) += boards/ diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig new file mode 100644 index 000000000000..4d68e3ef992b --- /dev/null +++ b/sound/soc/intel/avs/boards/Kconfig @@ -0,0 +1,121 @@ +# SPDX-License-Identifier: GPL-2.0-only +menu "Intel AVS Machine drivers" + depends on SND_SOC_INTEL_AVS + +comment "Available DSP configurations" + +config SND_SOC_INTEL_AVS_MACH_DA7219 + tristate "da7219 I2S board" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_DA7219 + help + This adds support for AVS with DA7219 I2S codec configuration. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +config SND_SOC_INTEL_AVS_MACH_DMIC + tristate "DMIC generic board" + select SND_SOC_DMIC + help + This adds support for AVS with Digital Mic array configuration. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +config SND_SOC_INTEL_AVS_MACH_HDAUDIO + tristate "HD-Audio generic board" + select SND_SOC_HDA + help + This adds support for AVS with HDAudio codec configuration. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +config SND_SOC_INTEL_AVS_MACH_I2S_TEST + tristate "I2S test board" + help + This adds support for I2S test-board which can be used to verify + transfer over I2S interface with SSP loopback scenarios. + +config SND_SOC_INTEL_AVS_MACH_MAX98357A + tristate "max98357A I2S board" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_MAX98357A + help + This adds support for AVS with MAX98357A I2S codec configuration. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +config SND_SOC_INTEL_AVS_MACH_MAX98373 + tristate "max98373 I2S board" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_MAX98373 + help + This adds support for AVS with MAX98373 I2S codec configuration. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +config SND_SOC_INTEL_AVS_MACH_NAU8825 + tristate "nau8825 I2S board" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_NAU8825 + help + This adds support for ASoC machine driver with NAU8825 I2S audio codec. + It is meant to be used with AVS driver. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +config SND_SOC_INTEL_AVS_MACH_RT274 + tristate "rt274 in I2S mode" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT274 + help + This adds support for ASoC machine driver with RT274 I2S audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +config SND_SOC_INTEL_AVS_MACH_RT286 + tristate "rt286 in I2S mode" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT286 + help + This adds support for ASoC machine driver with RT286 I2S audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +config SND_SOC_INTEL_AVS_MACH_RT298 + tristate "rt298 in I2S mode" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT298 + help + This adds support for ASoC machine driver with RT298 I2S audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +config SND_SOC_INTEL_AVS_MACH_RT5682 + tristate "rt5682 in I2S mode" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT5682_I2C + help + This adds support for ASoC machine driver with RT5682 I2S audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +config SND_SOC_INTEL_AVS_MACH_SSM4567 + tristate "ssm4567 I2S board" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_SSM4567 + help + This adds support for ASoC machine driver with SSM4567 I2S audio codec. + It is meant to be used with AVS driver. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +endmenu diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile new file mode 100644 index 000000000000..bc75376d58c2 --- /dev/null +++ b/sound/soc/intel/avs/boards/Makefile @@ -0,0 +1,27 @@ +# SPDX-License-Identifier: GPL-2.0-only + +snd-soc-avs-da7219-objs := da7219.o +snd-soc-avs-dmic-objs := dmic.o +snd-soc-avs-hdaudio-objs := hdaudio.o +snd-soc-avs-i2s-test-objs := i2s_test.o +snd-soc-avs-max98357a-objs := max98357a.o +snd-soc-avs-max98373-objs := max98373.o +snd-soc-avs-nau8825-objs := nau8825.o +snd-soc-avs-rt274-objs := rt274.o +snd-soc-avs-rt286-objs := rt286.o +snd-soc-avs-rt298-objs := rt298.o +snd-soc-avs-rt5682-objs := rt5682.o +snd-soc-avs-ssm4567-objs := ssm4567.o + +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DA7219) += snd-soc-avs-da7219.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_I2S_TEST) += snd-soc-avs-i2s-test.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_MAX98357A) += snd-soc-avs-max98357a.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_MAX98373) += snd-soc-avs-max98373.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_NAU8825) += snd-soc-avs-nau8825.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT298) += snd-soc-avs-rt298.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT5682) += snd-soc-avs-rt5682.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_SSM4567) += snd-soc-avs-ssm4567.o diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c new file mode 100644 index 000000000000..02ae542ad779 --- /dev/null +++ b/sound/soc/intel/avs/boards/da7219.c @@ -0,0 +1,282 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski <cezary.rojewski@intel.com> +// + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <sound/soc-dapm.h> +#include <uapi/linux/input-event-codes.h> +#include "../../../codecs/da7219.h" +#include "../../../codecs/da7219-aad.h" + +#define DA7219_DAI_NAME "da7219-hifi" + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +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 = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found. Unable to set/unset codec pll\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop PLL: %d\n", ret); + } else if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL_SRM, + 0, DA7219_PLL_FREQ_OUT_98304); + if (ret) + dev_err(card->dev, "failed to start PLL: %d\n", ret); + } + + return ret; +} + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + /* HP jack connectors - unknown if we have jack detection */ + {"Headphone Jack", NULL, "HPL"}, + {"Headphone Jack", NULL, "HPR"}, + + {"MIC", NULL, "Headset Mic"}, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, +}; + +static int avs_da7219_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + struct snd_soc_card *card = runtime->card; + struct snd_soc_jack *jack; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); + int clk_freq; + int ret; + + jack = snd_soc_card_get_drvdata(card); + clk_freq = 19200000; + + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, clk_freq, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(card->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3 | SND_JACK_LINEOUT, jack); + if (ret) { + dev_err(card->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + da7219_aad_jack_det(component, jack); + + return 0; +} + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-DLGS7219:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, DA7219_DAI_NAME); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_da7219_codec_init; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_da7219_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_da7219"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_da7219_driver = { + .probe = avs_da7219_probe, + .driver = { + .name = "avs_da7219", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_da7219_driver); + +MODULE_AUTHOR("Cezary Rojewski <cezary.rojewski@intel.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_da7219"); diff --git a/sound/soc/intel/avs/boards/dmic.c b/sound/soc/intel/avs/boards/dmic.c new file mode 100644 index 000000000000..90a921638572 --- /dev/null +++ b/sound/soc/intel/avs/boards/dmic.c @@ -0,0 +1,93 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski <cezary.rojewski@intel.com> +// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com> +// + +#include <linux/device.h> +#include <linux/module.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> + +SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC Pin"))); +SND_SOC_DAILINK_DEF(dmic_wov_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC WoV Pin"))); +SND_SOC_DAILINK_DEF(dmic_codec, DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); +/* Name overridden on probe */ +SND_SOC_DAILINK_DEF(platform, DAILINK_COMP_ARRAY(COMP_PLATFORM(""))); + +static struct snd_soc_dai_link card_dai_links[] = { + /* Back ends */ + { + .name = "DMIC", + .id = 0, + .dpcm_capture = 1, + .nonatomic = 1, + .no_pcm = 1, + SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), + }, + { + .name = "DMIC WoV", + .id = 1, + .dpcm_capture = 1, + .nonatomic = 1, + .no_pcm = 1, + .ignore_suspend = 1, + SND_SOC_DAILINK_REG(dmic_wov_pin, dmic_codec, platform), + }, +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route card_routes[] = { + {"DMic", NULL, "SoC DMIC"}, + {"DMIC Rx", NULL, "Capture"}, + {"DMIC WoV Rx", NULL, "Capture"}, +}; + +static int avs_dmic_probe(struct platform_device *pdev) +{ + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + int ret; + + mach = dev_get_platdata(dev); + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = "avs_dmic"; + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = card_dai_links; + card->num_links = ARRAY_SIZE(card_dai_links); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = card_routes; + card->num_dapm_routes = ARRAY_SIZE(card_routes); + card->fully_routed = true; + + ret = snd_soc_fixup_dai_links_platform_name(card, mach->mach_params.platform); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_dmic_driver = { + .probe = avs_dmic_probe, + .driver = { + .name = "avs_dmic", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_dmic_driver); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_dmic"); diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c new file mode 100644 index 000000000000..d2fc41d39448 --- /dev/null +++ b/sound/soc/intel/avs/boards/hdaudio.c @@ -0,0 +1,294 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski <cezary.rojewski@intel.com> +// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com> +// + +#include <linux/platform_device.h> +#include <sound/hda_codec.h> +#include <sound/hda_i915.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../../codecs/hda.h" + +static int avs_create_dai_links(struct device *dev, struct hda_codec *codec, int pcm_count, + const char *platform_name, struct snd_soc_dai_link **links) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + struct hda_pcm *pcm; + const char *cname = dev_name(&codec->core.dev); + int i; + + dl = devm_kcalloc(dev, pcm_count, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + pcm = list_first_entry(&codec->pcm_list_head, struct hda_pcm, list); + + for (i = 0; i < pcm_count; i++, pcm = list_next_entry(pcm, list)) { + dl[i].name = devm_kasprintf(dev, GFP_KERNEL, "%s link%d", cname, i); + if (!dl[i].name) + return -ENOMEM; + + dl[i].id = i; + dl[i].nonatomic = 1; + dl[i].no_pcm = 1; + dl[i].dpcm_playback = 1; + dl[i].dpcm_capture = 1; + dl[i].platforms = platform; + dl[i].num_platforms = 1; + + dl[i].codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + dl[i].cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + if (!dl[i].codecs || !dl[i].cpus) + return -ENOMEM; + + dl[i].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "%s-cpu%d", cname, i); + if (!dl[i].cpus->dai_name) + return -ENOMEM; + + dl[i].codecs->name = devm_kstrdup(dev, cname, GFP_KERNEL); + dl[i].codecs->dai_name = pcm->name; + dl[i].num_codecs = 1; + dl[i].num_cpus = 1; + } + + *links = dl; + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, struct hda_codec *codec, int pcm_count, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + struct hda_pcm *pcm; + const char *cname = dev_name(&codec->core.dev); + int i, n = 0; + + /* at max twice the number of pcms */ + dr = devm_kcalloc(dev, pcm_count * 2, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + pcm = list_first_entry(&codec->pcm_list_head, struct hda_pcm, list); + + for (i = 0; i < pcm_count; i++, pcm = list_next_entry(pcm, list)) { + struct hda_pcm_stream *stream; + int dir; + + dir = SNDRV_PCM_STREAM_PLAYBACK; + stream = &pcm->stream[dir]; + if (!stream->substreams) + goto capture_routes; + + dr[n].sink = devm_kasprintf(dev, GFP_KERNEL, "%s %s", pcm->name, + snd_pcm_direction_name(dir)); + dr[n].source = devm_kasprintf(dev, GFP_KERNEL, "%s-cpu%d Tx", cname, i); + if (!dr[n].sink || !dr[n].source) + return -ENOMEM; + n++; + +capture_routes: + dir = SNDRV_PCM_STREAM_CAPTURE; + stream = &pcm->stream[dir]; + if (!stream->substreams) + continue; + + dr[n].sink = devm_kasprintf(dev, GFP_KERNEL, "%s-cpu%d Rx", cname, i); + dr[n].source = devm_kasprintf(dev, GFP_KERNEL, "%s %s", pcm->name, + snd_pcm_direction_name(dir)); + if (!dr[n].sink || !dr[n].source) + return -ENOMEM; + n++; + } + + *routes = dr; + *num_routes = n; + return 0; +} + +/* Should be aligned with SectionPCM's name from topology */ +#define FEDAI_NAME_PREFIX "HDMI" + +static struct snd_pcm * +avs_card_hdmi_pcm_at(struct snd_soc_card *card, int hdmi_idx) +{ + struct snd_soc_pcm_runtime *rtd; + int dir = SNDRV_PCM_STREAM_PLAYBACK; + + for_each_card_rtds(card, rtd) { + struct snd_pcm *spcm; + int ret, n; + + spcm = rtd->pcm ? rtd->pcm->streams[dir].pcm : NULL; + if (!spcm || !strstr(spcm->id, FEDAI_NAME_PREFIX)) + continue; + + ret = sscanf(spcm->id, FEDAI_NAME_PREFIX "%d", &n); + if (ret != 1) + continue; + if (n == hdmi_idx) + return rtd->pcm; + } + + return NULL; +} + +static int avs_card_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_acpi_mach *mach = dev_get_platdata(card->dev); + struct hda_codec *codec = mach->pdata; + struct hda_pcm *hpcm; + /* Topology pcm indexing is 1-based */ + int i = 1; + + list_for_each_entry(hpcm, &codec->pcm_list_head, list) { + struct snd_pcm *spcm; + + spcm = avs_card_hdmi_pcm_at(card, i); + if (spcm) { + hpcm->pcm = spcm; + hpcm->device = spcm->device; + dev_info(card->dev, "%s: mapping HDMI converter %d to PCM %d (%p)\n", + __func__, i, hpcm->device, spcm); + } else { + hpcm->pcm = NULL; + hpcm->device = SNDRV_PCM_INVALID_DEVICE; + dev_warn(card->dev, "%s: no PCM in topology for HDMI converter %d\n", + __func__, i); + } + i++; + } + + return hda_codec_probe_complete(codec); +} + +static int avs_probing_link_init(struct snd_soc_pcm_runtime *rtm) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_acpi_mach *mach; + struct snd_soc_dai_link *links = NULL; + struct snd_soc_card *card = rtm->card; + struct hda_codec *codec; + struct hda_pcm *pcm; + int ret, n, pcm_count = 0; + + mach = dev_get_platdata(card->dev); + codec = mach->pdata; + + if (list_empty(&codec->pcm_list_head)) + return -EINVAL; + list_for_each_entry(pcm, &codec->pcm_list_head, list) + pcm_count++; + + ret = avs_create_dai_links(card->dev, codec, pcm_count, mach->mach_params.platform, &links); + if (ret < 0) { + dev_err(card->dev, "create links failed: %d\n", ret); + return ret; + } + + for (n = 0; n < pcm_count; n++) { + ret = snd_soc_add_pcm_runtime(card, &links[n]); + if (ret < 0) { + dev_err(card->dev, "add links failed: %d\n", ret); + return ret; + } + } + + ret = avs_create_dapm_routes(card->dev, codec, pcm_count, &routes, &n); + if (ret < 0) { + dev_err(card->dev, "create routes failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, routes, n); + if (ret < 0) { + dev_err(card->dev, "add routes failed: %d\n", ret); + return ret; + } + + return 0; +} + +SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); + +static struct snd_soc_dai_link probing_link = { + .name = "probing-LINK", + .id = -1, + .nonatomic = 1, + .no_pcm = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .cpus = dummy, + .num_cpus = ARRAY_SIZE(dummy), + .init = avs_probing_link_init, +}; + +static int avs_hdaudio_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link *binder; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + struct hda_codec *codec; + + mach = dev_get_platdata(dev); + codec = mach->pdata; + + /* codec may be unloaded before card's probe() fires */ + if (!device_is_registered(&codec->core.dev)) + return -ENODEV; + + binder = devm_kmemdup(dev, &probing_link, sizeof(probing_link), GFP_KERNEL); + if (!binder) + return -ENOMEM; + + binder->platforms = devm_kzalloc(dev, sizeof(*binder->platforms), GFP_KERNEL); + binder->codecs = devm_kzalloc(dev, sizeof(*binder->codecs), GFP_KERNEL); + if (!binder->platforms || !binder->codecs) + return -ENOMEM; + + binder->codecs->name = devm_kstrdup(dev, dev_name(&codec->core.dev), GFP_KERNEL); + if (!binder->codecs->name) + return -ENOMEM; + + binder->platforms->name = mach->mach_params.platform; + binder->num_platforms = 1; + binder->codecs->dai_name = "codec-probing-DAI"; + binder->num_codecs = 1; + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = binder->codecs->name; + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = binder; + card->num_links = 1; + card->fully_routed = true; + if (hda_codec_is_display(codec)) + card->late_probe = avs_card_late_probe; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_hdaudio_driver = { + .probe = avs_hdaudio_probe, + .driver = { + .name = "avs_hdaudio", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_hdaudio_driver) + +MODULE_DESCRIPTION("Intel HD-Audio machine driver"); +MODULE_AUTHOR("Cezary Rojewski <cezary.rojewski@intel.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_hdaudio"); diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c new file mode 100644 index 000000000000..8f0fd87bc866 --- /dev/null +++ b/sound/soc/intel/avs/boards/i2s_test.c @@ -0,0 +1,180 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski <cezary.rojewski@intel.com> +// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com> +// + +#include <linux/module.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <sound/soc-dapm.h> + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "snd-soc-dummy"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "snd-soc-dummy-dai"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_dr = 2; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + dr[0].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%dpb", ssp_port); + dr[0].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[0].sink || !dr[0].source) + return -ENOMEM; + + dr[1].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[1].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%dcp", ssp_port); + if (!dr[1].sink || !dr[1].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_create_dapm_widgets(struct device *dev, int ssp_port, + struct snd_soc_dapm_widget **widgets, int *num_widgets) +{ + struct snd_soc_dapm_widget *dw; + const int num_dw = 2; + + dw = devm_kcalloc(dev, num_dw, sizeof(*dw), GFP_KERNEL); + if (!dw) + return -ENOMEM; + + dw[0].id = snd_soc_dapm_hp; + dw[0].reg = SND_SOC_NOPM; + dw[0].name = devm_kasprintf(dev, GFP_KERNEL, "ssp%dpb", ssp_port); + if (!dw[0].name) + return -ENOMEM; + + dw[1].id = snd_soc_dapm_mic; + dw[1].reg = SND_SOC_NOPM; + dw[1].name = devm_kasprintf(dev, GFP_KERNEL, "ssp%dcp", ssp_port); + if (!dw[1].name) + return -ENOMEM; + + *widgets = dw; + *num_widgets = num_dw; + + return 0; +} + +static int avs_i2s_test_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_widget *widgets; + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, num_widgets; + int ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = devm_kasprintf(dev, GFP_KERNEL, "ssp%d-loopback", ssp_port); + if (!card->name) + return -ENOMEM; + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d\n", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d\n", ret); + return ret; + } + + ret = avs_create_dapm_widgets(dev, ssp_port, &widgets, &num_widgets); + if (ret) { + dev_err(dev, "Failed to create dapm widgets: %d\n", ret); + return ret; + } + + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = dai_link; + card->num_links = 1; + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->dapm_widgets = widgets; + card->num_dapm_widgets = num_widgets; + card->fully_routed = true; + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_i2s_test_driver = { + .probe = avs_i2s_test_probe, + .driver = { + .name = "avs_i2s_test", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_i2s_test_driver); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_i2s_test"); diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c new file mode 100644 index 000000000000..921f42caf7e0 --- /dev/null +++ b/sound/soc/intel/avs/boards/max98357a.c @@ -0,0 +1,154 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski <cezary.rojewski@intel.com> +// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com> +// + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <sound/soc-dapm.h> + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_SPK("Spk", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + { "Spk", NULL, "Speaker" }, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "MX98357A:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "HiFi"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 1; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "HiFi Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_max98357a_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = "avs_max98357a"; + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_max98357a_driver = { + .probe = avs_max98357a_probe, + .driver = { + .name = "avs_max98357a", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_max98357a_driver) + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_max98357a"); diff --git a/sound/soc/intel/avs/boards/max98373.c b/sound/soc/intel/avs/boards/max98373.c new file mode 100644 index 000000000000..0fa8f5606385 --- /dev/null +++ b/sound/soc/intel/avs/boards/max98373.c @@ -0,0 +1,239 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski <cezary.rojewski@intel.com> +// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com> +// + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <sound/soc-dapm.h> + +#define MAX98373_DEV0_NAME "i2c-MX98373:00" +#define MAX98373_DEV1_NAME "i2c-MX98373:01" +#define MAX98373_CODEC_NAME "max98373-aif1" + +static struct snd_soc_codec_conf card_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX98373_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX98373_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, +}; + +static int +avs_max98373_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int avs_max98373_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *runtime = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int ret, i; + + for_each_rtd_codec_dais(runtime, i, codec_dai) { + if (!strcmp(codec_dai->component->name, MAX98373_DEV0_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret); + return ret; + } + } + if (!strcmp(codec_dai->component->name, MAX98373_DEV1_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret); + return ret; + } + } + } + + return 0; +} + +static const struct snd_soc_ops avs_max98373_ops = { + .hw_params = avs_max98373_hw_params, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs) * 2, GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs[0].name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_DEV0_NAME); + dl->codecs[0].dai_name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_CODEC_NAME); + dl->codecs[1].name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_DEV1_NAME); + dl->codecs[1].dai_name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_CODEC_NAME); + if (!dl->cpus->dai_name || !dl->codecs[0].name || !dl->codecs[0].dai_name || + !dl->codecs[1].name || !dl->codecs[1].dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 2; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; + dl->be_hw_params_fixup = avs_max98373_be_fixup; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + dl->ignore_pmdown_time = 1; + dl->ops = &avs_max98373_ops; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Left HiFi Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Right HiFi Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_max98373_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = "avs_max98373"; + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = dai_link; + card->num_links = 1; + card->codec_conf = card_codec_conf; + card->num_configs = ARRAY_SIZE(card_codec_conf); + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_max98373_driver = { + .probe = avs_max98373_probe, + .driver = { + .name = "avs_max98373", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_max98373_driver) + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_max98373"); diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c new file mode 100644 index 000000000000..f76909e9f990 --- /dev/null +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -0,0 +1,353 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski <cezary.rojewski@intel.com> +// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com> +// + +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../../codecs/nau8825.h" + +#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" + +static int +avs_nau8825_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, 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 = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EINVAL; + } + + if (!SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "set sysclk err = %d\n", ret); + return ret; + } + } + + return 0; +} + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, avs_nau8825_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + { "MIC", NULL, "Headset Mic" }, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, +}; + +static struct snd_soc_jack_pin card_headset_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int avs_nau8825_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_jack_pin *pins; + struct snd_soc_jack *jack; + struct snd_soc_card *card = runtime->card; + int num_pins, ret; + + jack = snd_soc_card_get_drvdata(card); + num_pins = ARRAY_SIZE(card_headset_pins); + + pins = devm_kmemdup(card->dev, card_headset_pins, sizeof(*pins) * num_pins, GFP_KERNEL); + if (!pins) + return -ENOMEM; + + /* + * 4 buttons here map to the google Reference headset. + * The use of these buttons can be decided by the user space. + */ + ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, + jack, pins, num_pins); + if (ret) + return ret; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + //snd_soc_component_set_jack(component, jack, NULL); + // TODO: Fix nau8825 codec to use .set_jack, like everyone else + nau8825_enable_jack_detect(component, jack); + + return 0; +} + +static int +avs_nau8825_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int avs_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtm = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtm, 0); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + break; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + } + + return ret; +} + + +static const struct snd_soc_ops avs_nau8825_ops = { + .trigger = avs_nau8825_trigger, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-10508825:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, SKL_NUVOTON_CODEC_DAI); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_nau8825_codec_init; + dl->be_hw_params_fixup = avs_nau8825_be_fixup; + dl->ops = &avs_nau8825_ops; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EINVAL; + } + + if (codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] && + codec_dai->playback_widget->active) + snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, SND_SOC_CLOCK_IN); + + return avs_card_set_jack(card, jack); +} + +static int avs_nau8825_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_nau8825"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_nau8825_driver = { + .probe = avs_nau8825_probe, + .driver = { + .name = "avs_nau8825", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_nau8825_driver) + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_nau8825"); diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c new file mode 100644 index 000000000000..afef5a3ca60b --- /dev/null +++ b/sound/soc/intel/avs/boards/rt274.c @@ -0,0 +1,310 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski <cezary.rojewski@intel.com> +// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com> +// + +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../../codecs/rt274.h" + +#define AVS_RT274_FREQ_OUT 24000000 +#define AVS_RT274_BE_FIXUP_RATE 48000 +#define RT274_CODEC_DAI "rt274-aif1" + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static int +avs_rt274_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, 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 = snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI); + if (!codec_dai) + return -EINVAL; + + /* Codec needs clock for Jack detection and button press */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2, AVS_RT274_FREQ_OUT, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret); + return ret; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + int ratio = 100; + + snd_soc_dai_set_bclk_ratio(codec_dai, ratio); + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK, + AVS_RT274_BE_FIXUP_RATE * ratio, AVS_RT274_FREQ_OUT); + if (ret) { + dev_err(codec_dai->dev, "failed to enable PLL2: %d\n", ret); + return ret; + } + } + + return 0; +} + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, avs_rt274_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC", NULL, "Mic Jack"}, + + {"Headphone Jack", NULL, "Platform Clock"}, + {"MIC", NULL, "Platform Clock"}, +}; + +static struct snd_soc_jack_pin card_headset_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int avs_rt274_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_jack_pin *pins; + struct snd_soc_jack *jack; + struct snd_soc_card *card = runtime->card; + int num_pins, ret; + + jack = snd_soc_card_get_drvdata(card); + num_pins = ARRAY_SIZE(card_headset_pins); + + pins = devm_kmemdup(card->dev, card_headset_pins, sizeof(*pins) * num_pins, GFP_KERNEL); + if (!pins) + return -ENOMEM; + + ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET, jack, pins, num_pins); + if (ret) + return ret; + + snd_soc_component_set_jack(component, jack, NULL); + + /* 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(card->dev, "can't set codec pcm format %d\n", ret); + return ret; + } + + card->dapm.idle_bias_off = true; + + return 0; +} + +static int avs_rt274_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = AVS_RT274_BE_FIXUP_RATE; + channels->min = channels->max = 2; + + /* set SSPN to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-INT34C2:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "rt274-aif1"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_rt274_codec_init; + dl->be_hw_params_fixup = avs_rt274_be_fixup; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_rt274_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_rt274"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_rt274_driver = { + .probe = avs_rt274_probe, + .driver = { + .name = "avs_rt274", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_rt274_driver); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_rt274"); diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c new file mode 100644 index 000000000000..e51d4e181274 --- /dev/null +++ b/sound/soc/intel/avs/boards/rt286.c @@ -0,0 +1,281 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski <cezary.rojewski@intel.com> +// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com> +// + +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../../codecs/rt286.h" + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + /* HP jack connectors - unknown if we have jack detect */ + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC1", NULL, "Mic Jack"}, + + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, +}; + +static struct snd_soc_jack_pin card_headset_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int avs_rt286_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + struct snd_soc_jack_pin *pins; + struct snd_soc_jack *jack; + struct snd_soc_card *card = runtime->card; + int num_pins, ret; + + jack = snd_soc_card_get_drvdata(card); + num_pins = ARRAY_SIZE(card_headset_pins); + + pins = devm_kmemdup(card->dev, card_headset_pins, sizeof(*pins) * num_pins, GFP_KERNEL); + if (!pins) + return -ENOMEM; + + ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, + pins, num_pins); + if (ret) + return ret; + + snd_soc_component_set_jack(component, jack, NULL); + + return 0; +} + +static int avs_rt286_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int +avs_rt286_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *runtime = substream->private_data; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 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(runtime->dev, "Set codec sysclk failed: %d\n", ret); + + return ret; +} + +static const struct snd_soc_ops avs_rt286_ops = { + .hw_params = avs_rt286_hw_params, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-INT343A:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "rt286-aif1"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_rt286_codec_init; + dl->be_hw_params_fixup = avs_rt286_be_fixup; + dl->ops = &avs_rt286_ops; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_rt286_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_rt286"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_rt286_driver = { + .probe = avs_rt286_probe, + .driver = { + .name = "avs_rt286", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_rt286_driver); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_rt286"); diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c new file mode 100644 index 000000000000..b28d36872dcb --- /dev/null +++ b/sound/soc/intel/avs/boards/rt298.c @@ -0,0 +1,281 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski <cezary.rojewski@intel.com> +// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com> +// + +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../../codecs/rt298.h" + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + /* HP jack connectors - unknown if we have jack detect */ + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC1", NULL, "Mic Jack"}, + + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, +}; + +static struct snd_soc_jack_pin card_headset_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int avs_rt298_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + struct snd_soc_jack_pin *pins; + struct snd_soc_jack *jack; + struct snd_soc_card *card = runtime->card; + int num_pins, ret; + + jack = snd_soc_card_get_drvdata(card); + num_pins = ARRAY_SIZE(card_headset_pins); + + pins = devm_kmemdup(card->dev, card_headset_pins, sizeof(*pins) * num_pins, GFP_KERNEL); + if (!pins) + return -ENOMEM; + + ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, + pins, num_pins); + if (ret) + return ret; + + snd_soc_component_set_jack(component, jack, NULL); + + return 0; +} + +static int avs_rt298_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int +avs_rt298_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 = asoc_rtd_to_codec(rtd, 0); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT298_SCLK_S_PLL, 19200000, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "Set codec sysclk failed: %d\n", ret); + + return ret; +} + +static const struct snd_soc_ops avs_rt298_ops = { + .hw_params = avs_rt298_hw_params, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-INT343A:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "rt298-aif1"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_rt298_codec_init; + dl->be_hw_params_fixup = avs_rt298_be_fixup; + dl->ops = &avs_rt298_ops; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_rt298_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_rt298"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_rt298_driver = { + .probe = avs_rt298_probe, + .driver = { + .name = "avs_rt298", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_rt298_driver); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_rt298"); diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c new file mode 100644 index 000000000000..01f9b9f0c12b --- /dev/null +++ b/sound/soc/intel/avs/boards/rt5682.c @@ -0,0 +1,340 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski <cezary.rojewski@intel.com> +// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com> +// + +#include <linux/clk.h> +#include <linux/dmi.h> +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/rt5682.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../common/soc-intel-quirks.h" +#include "../../../codecs/rt5682.h" + +#define AVS_RT5682_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) +#define AVS_RT5682_SSP_CODEC_MASK (GENMASK(2, 0)) +#define AVS_RT5682_MCLK_EN BIT(3) +#define AVS_RT5682_MCLK_24MHZ BIT(4) + +/* Default: MCLK on, MCLK 19.2M, SSP0 */ +static unsigned long avs_rt5682_quirk = AVS_RT5682_MCLK_EN | AVS_RT5682_SSP_CODEC(0); + +static int avs_rt5682_quirk_cb(const struct dmi_system_id *id) +{ + avs_rt5682_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id avs_rt5682_quirk_table[] = { + { + .callback = avs_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "WhiskeyLake Client"), + }, + .driver_data = (void *)(AVS_RT5682_MCLK_EN | + AVS_RT5682_MCLK_24MHZ | + AVS_RT5682_SSP_CODEC(1)), + }, + { + .callback = avs_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Ice Lake Client"), + }, + .driver_data = (void *)(AVS_RT5682_MCLK_EN | + AVS_RT5682_SSP_CODEC(0)), + }, + {} +}; + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + /* HP jack connectors - unknown if we have jack detect */ + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* other jacks */ + { "IN1P", NULL, "Headset Mic" }, +}; + +static int avs_rt5682_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + struct snd_soc_jack *jack; + struct snd_soc_card *card = runtime->card; + int ret; + + jack = snd_soc_card_get_drvdata(card); + + /* Need to enable ASRC function for 24MHz mclk rate */ + if ((avs_rt5682_quirk & AVS_RT5682_MCLK_EN) && + (avs_rt5682_quirk & AVS_RT5682_MCLK_24MHZ)) { + rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER | + RT5682_AD_STEREO1_FILTER, RT5682_CLK_SEL_I2S1_ASRC); + } + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, jack); + if (ret) { + dev_err(card->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + ret = snd_soc_component_set_jack(component, jack, NULL); + if (ret) { + dev_err(card->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return 0; +}; + +static int +avs_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *runtime = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); + int clk_id, clk_freq; + int pll_out, ret; + + if (avs_rt5682_quirk & AVS_RT5682_MCLK_EN) { + clk_id = RT5682_PLL1_S_MCLK; + if (avs_rt5682_quirk & AVS_RT5682_MCLK_24MHZ) + clk_freq = 24000000; + else + clk_freq = 19200000; + } else { + clk_id = RT5682_PLL1_S_BCLK1; + clk_freq = params_rate(params) * 50; + } + + pll_out = params_rate(params) * 512; + + ret = snd_soc_dai_set_pll(codec_dai, 0, clk_id, clk_freq, pll_out); + if (ret < 0) + dev_err(runtime->dev, "snd_soc_dai_set_pll err = %d\n", ret); + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, pll_out, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(runtime->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + /* slot_width should equal or large than data length, set them be the same */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, params_width(params)); + if (ret < 0) { + dev_err(runtime->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_ops avs_rt5682_ops = { + .hw_params = avs_rt5682_hw_params, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-10EC5682:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "rt5682-aif1"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->init = avs_rt5682_codec_init; + dl->ops = &avs_rt5682_ops; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_rt5682_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + if (pdev->id_entry && pdev->id_entry->driver_data) + avs_rt5682_quirk = (unsigned long)pdev->id_entry->driver_data; + + dmi_check_system(avs_rt5682_quirk_table); + dev_dbg(dev, "avs_rt5682_quirk = %lx\n", avs_rt5682_quirk); + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_rt5682"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_rt5682_driver = { + .probe = avs_rt5682_probe, + .driver = { + .name = "avs_rt5682", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_rt5682_driver) + +MODULE_AUTHOR("Cezary Rojewski <cezary.rojewski@intel.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_rt5682"); diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c new file mode 100644 index 000000000000..9f84c8ab3447 --- /dev/null +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -0,0 +1,271 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski <cezary.rojewski@intel.com> +// Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com> +// + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../../codecs/nau8825.h" + +#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" +#define SKL_SSM_CODEC_DAI "ssm4567-hifi" + +static struct snd_soc_codec_conf card_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF("i2c-INT343B:00"), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF("i2c-INT343B:01"), + .name_prefix = "Right", + }, +}; + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Speaker"), + SOC_DAPM_PIN_SWITCH("Right Speaker"), +}; + +static int +platform_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, 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 = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EINVAL; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000, + SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(card->dev, "set sysclk err = %d\n", ret); + } else { + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(card->dev, "set sysclk err = %d\n", ret); + } + + return ret; +} + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_SPK("Left Speaker", NULL), + SND_SOC_DAPM_SPK("Right Speaker", NULL), + SND_SOC_DAPM_SPK("DP1", NULL), + SND_SOC_DAPM_SPK("DP2", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + {"Left Speaker", NULL, "Left OUT"}, + {"Right Speaker", NULL, "Right OUT"}, +}; + +static int avs_ssm4567_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + + /* Slot 1 for left */ + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(runtime, 0), 0x01, 0x01, 2, 48); + if (ret < 0) + return ret; + + /* Slot 2 for right */ + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(runtime, 1), 0x02, 0x02, 2, 48); + if (ret < 0) + return ret; + + return 0; +} + +static int +avs_ssm4567_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs) * 2, GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs[0].name = devm_kasprintf(dev, GFP_KERNEL, "i2c-INT343B:00"); + dl->codecs[0].dai_name = devm_kasprintf(dev, GFP_KERNEL, "ssm4567-hifi"); + dl->codecs[1].name = devm_kasprintf(dev, GFP_KERNEL, "i2c-INT343B:01"); + dl->codecs[1].dai_name = devm_kasprintf(dev, GFP_KERNEL, "ssm4567-hifi"); + if (!dl->cpus->dai_name || !dl->codecs[0].name || !dl->codecs[0].dai_name || + !dl->codecs[1].name || !dl->codecs[1].dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 2; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_ssm4567_codec_init; + dl->be_hw_params_fixup = avs_ssm4567_be_fixup; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + dl->ignore_pmdown_time = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 4; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Left Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Right Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "Left Capture Sense"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "Right Capture Sense"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_ssm4567_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = "avs_ssm4567-adi"; + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = dai_link; + card->num_links = 1; + card->codec_conf = card_codec_conf; + card->num_configs = ARRAY_SIZE(card_codec_conf); + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + card->disable_route_checks = true; + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_ssm4567_driver = { + .probe = avs_ssm4567_probe, + .driver = { + .name = "avs_ssm4567", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_ssm4567_driver) + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_ssm4567"); diff --git a/sound/soc/intel/avs/cldma.c b/sound/soc/intel/avs/cldma.c index d100c6ba4d8a..d7a9390b5e48 100644 --- a/sound/soc/intel/avs/cldma.c +++ b/sound/soc/intel/avs/cldma.c @@ -176,17 +176,17 @@ int hda_cldma_reset(struct hda_cldma *cl) return ret; } - snd_hdac_stream_updateb(cl, SD_CTL, 1, 1); - ret = snd_hdac_stream_readb_poll(cl, SD_CTL, reg, (reg & 1), AVS_CL_OP_INTERVAL_US, - AVS_CL_OP_TIMEOUT_US); + snd_hdac_stream_updateb(cl, SD_CTL, SD_CTL_STREAM_RESET, SD_CTL_STREAM_RESET); + ret = snd_hdac_stream_readb_poll(cl, SD_CTL, reg, (reg & SD_CTL_STREAM_RESET), + AVS_CL_OP_INTERVAL_US, AVS_CL_OP_TIMEOUT_US); if (ret < 0) { dev_err(cl->dev, "cldma set SRST failed: %d\n", ret); return ret; } - snd_hdac_stream_updateb(cl, SD_CTL, 1, 0); - ret = snd_hdac_stream_readb_poll(cl, SD_CTL, reg, !(reg & 1), AVS_CL_OP_INTERVAL_US, - AVS_CL_OP_TIMEOUT_US); + snd_hdac_stream_updateb(cl, SD_CTL, SD_CTL_STREAM_RESET, 0); + ret = snd_hdac_stream_readb_poll(cl, SD_CTL, reg, !(reg & SD_CTL_STREAM_RESET), + AVS_CL_OP_INTERVAL_US, AVS_CL_OP_TIMEOUT_US); if (ret < 0) { dev_err(cl->dev, "cldma unset SRST failed: %d\n", ret); return ret; diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 3a0997c3af2b..c50c20fd681a 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -23,6 +23,7 @@ #include <sound/hdaudio_ext.h> #include <sound/intel-dsp-config.h> #include <sound/intel-nhlt.h> +#include "../../codecs/hda.h" #include "avs.h" #include "cldma.h" @@ -356,7 +357,7 @@ static int avs_bus_init(struct avs_dev *adev, struct pci_dev *pci, const struct struct device *dev = &pci->dev; int ret; - ret = snd_hdac_ext_bus_init(&bus->core, dev, NULL, NULL); + ret = snd_hdac_ext_bus_init(&bus->core, dev, NULL, &soc_hda_ext_bus_ops); if (ret < 0) return ret; @@ -439,12 +440,9 @@ static int avs_pci_probe(struct pci_dev *pci, const struct pci_device_id *id) if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); - if (!dma_set_mask(dev, DMA_BIT_MASK(64))) { - dma_set_coherent_mask(dev, DMA_BIT_MASK(64)); - } else { - dma_set_mask(dev, DMA_BIT_MASK(32)); - dma_set_coherent_mask(dev, DMA_BIT_MASK(32)); - } + if (!dma_set_mask_and_coherent(dev, DMA_BIT_MASK(64))) + dma_set_mask_and_coherent(dev, DMA_BIT_MASK(32)); + dma_set_max_seg_size(dev, UINT_MAX); ret = avs_hdac_bus_init_streams(bus); if (ret < 0) { @@ -555,6 +553,7 @@ static int __maybe_unused avs_suspend_common(struct avs_dev *adev) return AVS_IPC_RET(ret); } + avs_ipc_block(adev->ipc); avs_dsp_op(adev, int_control, false); snd_hdac_ext_bus_ppcap_int_enable(bus, false); diff --git a/sound/soc/intel/avs/dsp.c b/sound/soc/intel/avs/dsp.c index 06d2f7af520f..b881100d3e02 100644 --- a/sound/soc/intel/avs/dsp.c +++ b/sound/soc/intel/avs/dsp.c @@ -13,6 +13,7 @@ #define AVS_ADSPCS_INTERVAL_US 500 #define AVS_ADSPCS_TIMEOUT_US 50000 +#define AVS_ADSPCS_DELAY_US 1000 int avs_dsp_core_power(struct avs_dev *adev, u32 core_mask, bool power) { @@ -26,6 +27,8 @@ int avs_dsp_core_power(struct avs_dev *adev, u32 core_mask, bool power) value = power ? mask : 0; snd_hdac_adsp_updatel(adev, AVS_ADSP_REG_ADSPCS, mask, value); + /* Delay the polling to avoid false positives. */ + usleep_range(AVS_ADSPCS_DELAY_US, 2 * AVS_ADSPCS_DELAY_US); mask = AVS_ADSPCS_CPA_MASK(core_mask); value = power ? mask : 0; @@ -82,11 +85,15 @@ int avs_dsp_core_stall(struct avs_dev *adev, u32 core_mask, bool stall) reg, (reg & mask) == value, AVS_ADSPCS_INTERVAL_US, AVS_ADSPCS_TIMEOUT_US); - if (ret) + if (ret) { dev_err(adev->dev, "core_mask %d %sstall failed: %d\n", core_mask, stall ? "" : "un", ret); + return ret; + } - return ret; + /* Give HW time to propagate the change. */ + usleep_range(AVS_ADSPCS_DELAY_US, 2 * AVS_ADSPCS_DELAY_US); + return 0; } int avs_dsp_core_enable(struct avs_dev *adev, u32 core_mask) diff --git a/sound/soc/intel/avs/ipc.c b/sound/soc/intel/avs/ipc.c index d755ba8b8518..020d85c7520d 100644 --- a/sound/soc/intel/avs/ipc.c +++ b/sound/soc/intel/avs/ipc.c @@ -480,6 +480,7 @@ static int avs_dsp_do_send_msg(struct avs_dev *adev, struct avs_ipc_msg *request ret = ipc->rx.rsp.status; if (reply) { reply->header = ipc->rx.header; + reply->size = ipc->rx.size; if (reply->data && ipc->rx.size) memcpy(reply->data, ipc->rx.data, reply->size); } diff --git a/sound/soc/intel/avs/loader.c b/sound/soc/intel/avs/loader.c index 542fd44aa501..9e3f8ff33a87 100644 --- a/sound/soc/intel/avs/loader.c +++ b/sound/soc/intel/avs/loader.c @@ -27,8 +27,8 @@ #define APL_ROM_INIT_RETRIES 3 #define AVS_FW_INIT_POLLING_US 500 -#define AVS_FW_INIT_TIMEOUT_US 3000000 #define AVS_FW_INIT_TIMEOUT_MS 3000 +#define AVS_FW_INIT_TIMEOUT_US (AVS_FW_INIT_TIMEOUT_MS * 1000) #define AVS_CLDMA_START_DELAY_MS 100 diff --git a/sound/soc/intel/avs/messages.c b/sound/soc/intel/avs/messages.c index 6404fce8cde4..d4bcee1aabcf 100644 --- a/sound/soc/intel/avs/messages.c +++ b/sound/soc/intel/avs/messages.c @@ -59,7 +59,7 @@ int avs_ipc_unload_modules(struct avs_dev *adev, u16 *mod_ids, u32 num_mod_ids) request.data = mod_ids; request.size = sizeof(*mod_ids) * num_mod_ids; - ret = avs_dsp_send_msg_timeout(adev, &request, NULL, AVS_CL_TIMEOUT_MS); + ret = avs_dsp_send_msg(adev, &request, NULL); if (ret) avs_ipc_err(adev, &request, "unload multiple modules", ret); @@ -378,7 +378,6 @@ int avs_ipc_get_large_config(struct avs_dev *adev, u16 module_id, u8 instance_id union avs_module_msg msg = AVS_MODULE_REQUEST(LARGE_CONFIG_GET); struct avs_ipc_msg request; struct avs_ipc_msg reply = {{0}}; - size_t size; void *buf; int ret; @@ -406,15 +405,14 @@ int avs_ipc_get_large_config(struct avs_dev *adev, u16 module_id, u8 instance_id return ret; } - size = reply.rsp.ext.large_config.data_off_size; - buf = krealloc(reply.data, size, GFP_KERNEL); + buf = krealloc(reply.data, reply.size, GFP_KERNEL); if (!buf) { kfree(reply.data); return -ENOMEM; } *reply_data = buf; - *reply_size = size; + *reply_size = reply.size; return 0; } @@ -476,6 +474,9 @@ int avs_ipc_get_fw_config(struct avs_dev *adev, struct avs_fw_cfg *cfg) &payload, &payload_size); if (ret) return ret; + /* Non-zero payload expected for FIRMWARE_CONFIG. */ + if (!payload_size) + return -EREMOTEIO; while (offset < payload_size) { tlv = (struct avs_tlv *)(payload + offset); @@ -561,6 +562,7 @@ int avs_ipc_get_fw_config(struct avs_dev *adev, struct avs_fw_cfg *cfg) case AVS_FW_CFG_DMA_BUFFER_CONFIG: case AVS_FW_CFG_SCHEDULER_CONFIG: case AVS_FW_CFG_CLOCKS_CONFIG: + case AVS_FW_CFG_RESERVED: break; default: @@ -589,6 +591,9 @@ int avs_ipc_get_hw_config(struct avs_dev *adev, struct avs_hw_cfg *cfg) &payload, &payload_size); if (ret) return ret; + /* Non-zero payload expected for HARDWARE_CONFIG. */ + if (!payload_size) + return -EREMOTEIO; while (offset < payload_size) { tlv = (struct avs_tlv *)(payload + offset); @@ -672,6 +677,9 @@ int avs_ipc_get_modules_info(struct avs_dev *adev, struct avs_mods_info **info) &payload, &payload_size); if (ret) return ret; + /* Non-zero payload expected for MODULES_INFO. */ + if (!payload_size) + return -EREMOTEIO; *info = (struct avs_mods_info *)payload; return 0; diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index 3d46dd5e5bc4..ce157a8d6552 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -449,35 +449,39 @@ static int avs_modext_create(struct avs_dev *adev, struct avs_path_module *mod) return ret; } +static int avs_probe_create(struct avs_dev *adev, struct avs_path_module *mod) +{ + dev_err(adev->dev, "Probe module can't be instantiated by topology"); + return -EINVAL; +} + +struct avs_module_create { + guid_t *guid; + int (*create)(struct avs_dev *adev, struct avs_path_module *mod); +}; + +static struct avs_module_create avs_module_create[] = { + { &AVS_MIXIN_MOD_UUID, avs_modbase_create }, + { &AVS_MIXOUT_MOD_UUID, avs_modbase_create }, + { &AVS_KPBUFF_MOD_UUID, avs_modbase_create }, + { &AVS_COPIER_MOD_UUID, avs_copier_create }, + { &AVS_MICSEL_MOD_UUID, avs_micsel_create }, + { &AVS_MUX_MOD_UUID, avs_mux_create }, + { &AVS_UPDWMIX_MOD_UUID, avs_updown_mix_create }, + { &AVS_SRCINTC_MOD_UUID, avs_src_create }, + { &AVS_AEC_MOD_UUID, avs_aec_create }, + { &AVS_ASRC_MOD_UUID, avs_asrc_create }, + { &AVS_INTELWOV_MOD_UUID, avs_wov_create }, + { &AVS_PROBE_MOD_UUID, avs_probe_create }, +}; + static int avs_path_module_type_create(struct avs_dev *adev, struct avs_path_module *mod) { const guid_t *type = &mod->template->cfg_ext->type; - if (guid_equal(type, &AVS_MIXIN_MOD_UUID) || - guid_equal(type, &AVS_MIXOUT_MOD_UUID) || - guid_equal(type, &AVS_KPBUFF_MOD_UUID)) - return avs_modbase_create(adev, mod); - if (guid_equal(type, &AVS_COPIER_MOD_UUID)) - return avs_copier_create(adev, mod); - if (guid_equal(type, &AVS_MICSEL_MOD_UUID)) - return avs_micsel_create(adev, mod); - if (guid_equal(type, &AVS_MUX_MOD_UUID)) - return avs_mux_create(adev, mod); - if (guid_equal(type, &AVS_UPDWMIX_MOD_UUID)) - return avs_updown_mix_create(adev, mod); - if (guid_equal(type, &AVS_SRCINTC_MOD_UUID)) - return avs_src_create(adev, mod); - if (guid_equal(type, &AVS_AEC_MOD_UUID)) - return avs_aec_create(adev, mod); - if (guid_equal(type, &AVS_ASRC_MOD_UUID)) - return avs_asrc_create(adev, mod); - if (guid_equal(type, &AVS_INTELWOV_MOD_UUID)) - return avs_wov_create(adev, mod); - - if (guid_equal(type, &AVS_PROBE_MOD_UUID)) { - dev_err(adev->dev, "Probe module can't be instantiated by topology"); - return -EINVAL; - } + for (int i = 0; i < ARRAY_SIZE(avs_module_create); i++) + if (guid_equal(type, avs_module_create[i].guid)) + return avs_module_create[i].create(adev, mod); return avs_modext_create(adev, mod); } diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 668f533578a6..f21b0cdd3206 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -846,7 +846,6 @@ static const struct snd_soc_component_driver avs_component_driver = { .pcm_construct = avs_component_construct, .module_get_upon_open = 1, /* increment refcount when a pcm is opened */ .topology_name_prefix = "intel/avs", - .non_legacy_dai_naming = true, }; static int avs_soc_component_register(struct device *dev, const char *name, @@ -1172,7 +1171,6 @@ static const struct snd_soc_component_driver avs_hda_component_driver = { .remove_order = SND_SOC_COMP_ORDER_EARLY, .module_get_upon_open = 1, .topology_name_prefix = "intel/avs", - .non_legacy_dai_naming = true, }; int avs_hda_platform_register(struct avs_dev *adev, const char *name) diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index 6a06fe387d13..8a9f9fc48938 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -808,6 +808,30 @@ static const struct avs_tplg_token_parser pin_format_parsers[] = { }, }; +static void +assign_copier_gtw_instance(struct snd_soc_component *comp, struct avs_tplg_modcfg_ext *cfg) +{ + struct snd_soc_acpi_mach *mach; + + if (!guid_equal(&cfg->type, &AVS_COPIER_MOD_UUID)) + return; + + /* Only I2S boards assign port instance in ->i2s_link_mask. */ + switch (cfg->copier.dma_type) { + case AVS_DMA_I2S_LINK_OUTPUT: + case AVS_DMA_I2S_LINK_INPUT: + break; + default: + return; + } + + mach = dev_get_platdata(comp->card->dev); + + /* Automatic assignment only when board describes single SSP. */ + if (hweight_long(mach->mach_params.i2s_link_mask) == 1 && !cfg->copier.vindex.i2s.instance) + cfg->copier.vindex.i2s.instance = __ffs(mach->mach_params.i2s_link_mask); +} + static int avs_tplg_parse_modcfg_ext(struct snd_soc_component *comp, struct avs_tplg_modcfg_ext *cfg, struct snd_soc_tplg_vendor_array *tuples, @@ -827,6 +851,9 @@ static int avs_tplg_parse_modcfg_ext(struct snd_soc_component *comp, if (ret) return ret; + /* Update copier gateway based on board's i2s_link_mask. */ + assign_copier_gtw_instance(comp, cfg); + block_size -= esize; /* Parse trailing in/out pin formats if any. */ if (block_size) { diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f3873b5bea87..aa12d7e3dd2f 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -41,7 +41,7 @@ config SND_SOC_INTEL_SOF_CIRRUS_COMMON if SND_SOC_INTEL_CATPT config SND_SOC_INTEL_HASWELL_MACH - tristate "Haswell Lynxpoint" + tristate "Haswell with RT5640 I2S codec" depends on I2C depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST depends on X86_INTEL_LPSS || COMPILE_TEST @@ -85,7 +85,7 @@ config SND_SOC_INTEL_BDW_RT5677_MACH If unsure select "N". config SND_SOC_INTEL_BROADWELL_MACH - tristate "Broadwell Wildcatpoint" + tristate "Broadwell with RT286 I2S codec" depends on I2C depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST depends on X86_INTEL_LPSS || COMPILE_TEST @@ -660,7 +660,6 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES || COMPILE_TEST depends on SOUNDWIRE - depends on SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_MAX98373_I2C select SND_SOC_MAX98373_SDW select SND_SOC_RT700_SDW diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 40c0c3d1c500..eea1e26acfda 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -1,8 +1,8 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-sst-haswell-objs := haswell.o +snd-soc-sst-haswell-objs := hsw_rt5640.o snd-soc-sst-bdw-rt5650-mach-objs := bdw-rt5650.o snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o -snd-soc-sst-broadwell-objs := broadwell.o +snd-soc-sst-broadwell-objs := bdw_rt286.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o snd-soc-sst-bxt-rt298-objs := bxt_rt298.o snd-soc-sst-sof-pcm512x-objs := sof_pcm512x.o diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c index aae857fdcdb8..67c3f49b924c 100644 --- a/sound/soc/intel/boards/bdw-rt5650.c +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -249,6 +249,7 @@ static struct snd_soc_dai_link bdw_rt5650_dais[] = { /* SSP0 - Codec */ .name = "Codec", .id = 0, + .nonatomic = 1, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC, diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index d0ecbba2febe..31488702768e 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -349,6 +349,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { /* SSP0 - Codec */ .name = "Codec", .id = 0, + .nonatomic = 1, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC, diff --git a/sound/soc/intel/boards/bdw_rt286.c b/sound/soc/intel/boards/bdw_rt286.c new file mode 100644 index 000000000000..6b76df0e7c9b --- /dev/null +++ b/sound/soc/intel/boards/bdw_rt286.c @@ -0,0 +1,280 @@ +// 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 <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#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 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, + .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 void bdw_rt286_disable_jack(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) { + if (!strcmp(component->name, "i2c-INT343A:00")) { + dev_dbg(component->dev, "disabling jack detect before going to suspend.\n"); + snd_soc_component_set_jack(component, NULL, NULL); + break; + } + } +} + +static int bdw_rt286_suspend(struct snd_soc_card *card) +{ + bdw_rt286_disable_jack(card); + + return 0; +} + +static int bdw_rt286_resume(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) { + if (!strcmp(component->name, "i2c-INT343A:00")) { + dev_dbg(component->dev, "enabling jack detect for resume.\n"); + snd_soc_component_set_jack(component, &card_headset, NULL); + break; + } + } + + return 0; +} + +static struct snd_soc_card bdw_rt286_card = { + .owner = THIS_MODULE, + .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, + .suspend_pre = bdw_rt286_suspend, + .resume_post = bdw_rt286_resume, +}; + +/* 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 int bdw_rt286_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + bdw_rt286_disable_jack(card); + + return 0; +} + +static struct platform_driver bdw_rt286_driver = { + .probe = bdw_rt286_probe, + .remove = bdw_rt286_remove, + .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"); diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c deleted file mode 100644 index c30a9dca6801..000000000000 --- a/sound/soc/intel/boards/broadwell.c +++ /dev/null @@ -1,336 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Intel Broadwell Wildcatpoint SST Audio - * - * Copyright (C) 2013, Intel Corporation. All rights reserved. - */ - -#include <linux/module.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/jack.h> -#include <sound/pcm_params.h> -#include <sound/soc-acpi.h> - -#include "../../codecs/rt286.h" - -static struct snd_soc_jack broadwell_headset; -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin broadwell_headset_pins[] = { - { - .pin = "Mic Jack", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, -}; - -static const struct snd_kcontrol_new broadwell_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), - SOC_DAPM_PIN_SWITCH("Headphone Jack"), -}; - -static const struct snd_soc_dapm_widget broadwell_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 broadwell_rt286_map[] = { - - /* speaker */ - {"Speaker", NULL, "SPOR"}, - {"Speaker", NULL, "SPOL"}, - - /* HP jack connectors - unknown if we have jack deteck */ - {"Headphone Jack", NULL, "HPO Pin"}, - - /* other jacks */ - {"MIC1", NULL, "Mic Jack"}, - {"LINE1", NULL, "Line Jack"}, - - /* digital mics */ - {"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 broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; - int ret = 0; - ret = snd_soc_card_jack_new_pins(rtd->card, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset, - broadwell_headset_pins, ARRAY_SIZE(broadwell_headset_pins)); - if (ret) - return ret; - - rt286_mic_detect(component, &broadwell_headset); - return 0; -} - - -static int broadwell_ssp0_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 *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - - /* The ADSP will covert the FE rate to 48k, stereo */ - rate->min = rate->max = 48000; - chan->min = chan->max = 2; - - /* set SSP0 to 16 bit */ - params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); - return 0; -} - -static int broadwell_rt286_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, "can't set codec sysclk configuration\n"); - return ret; - } - - return ret; -} - -static const struct snd_soc_ops broadwell_rt286_ops = { - .hw_params = broadwell_rt286_hw_params, -}; - -static const unsigned int channels[] = { - 2, -}; - -static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, -}; - -static int broadwell_fe_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - /* Board supports stereo configuration only */ - runtime->hw.channels_max = 2; - return snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); -} - -static const struct snd_soc_ops broadwell_fe_ops = { - .startup = broadwell_fe_startup, -}; - -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"))); - -/* broadwell digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link broadwell_rt286_dais[] = { - /* 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}, - .ops = &broadwell_fe_ops, - .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, - .no_pcm = 1, - .init = broadwell_rt286_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = broadwell_ssp0_fixup, - .ops = &broadwell_rt286_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp0_port, codec, platform), - }, -}; - -static int broadwell_disable_jack(struct snd_soc_card *card) -{ - struct snd_soc_component *component; - - for_each_card_components(card, component) { - if (!strcmp(component->name, "i2c-INT343A:00")) { - - dev_dbg(component->dev, "disabling jack detect before going to suspend.\n"); - rt286_mic_detect(component, NULL); - break; - } - } - - return 0; -} - -static int broadwell_suspend(struct snd_soc_card *card) -{ - return broadwell_disable_jack(card); -} - -static int broadwell_resume(struct snd_soc_card *card){ - struct snd_soc_component *component; - - for_each_card_components(card, component) { - if (!strcmp(component->name, "i2c-INT343A:00")) { - - dev_dbg(component->dev, "enabling jack detect for resume.\n"); - rt286_mic_detect(component, &broadwell_headset); - break; - } - } - return 0; -} - -/* 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" -#define DRIVER_NAME NULL /* card name will be used for driver name */ - -/* broadwell audio machine driver for WPT + RT286S */ -static struct snd_soc_card broadwell_rt286 = { - .owner = THIS_MODULE, - .dai_link = broadwell_rt286_dais, - .num_links = ARRAY_SIZE(broadwell_rt286_dais), - .controls = broadwell_controls, - .num_controls = ARRAY_SIZE(broadwell_controls), - .dapm_widgets = broadwell_widgets, - .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), - .dapm_routes = broadwell_rt286_map, - .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map), - .fully_routed = true, - .suspend_pre = broadwell_suspend, - .resume_post = broadwell_resume, -}; - -static int broadwell_audio_probe(struct platform_device *pdev) -{ - struct snd_soc_acpi_mach *mach; - int ret; - - broadwell_rt286.dev = &pdev->dev; - - /* override platform name, if required */ - mach = pdev->dev.platform_data; - ret = snd_soc_fixup_dai_links_platform_name(&broadwell_rt286, - mach->mach_params.platform); - if (ret) - return ret; - - /* set card and driver name */ - if (snd_soc_acpi_sof_parent(&pdev->dev)) { - broadwell_rt286.name = SOF_CARD_NAME; - broadwell_rt286.driver_name = SOF_DRIVER_NAME; - } else { - broadwell_rt286.name = CARD_NAME; - broadwell_rt286.driver_name = DRIVER_NAME; - } - - return devm_snd_soc_register_card(&pdev->dev, &broadwell_rt286); -} - -static int broadwell_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - return broadwell_disable_jack(card); -} - -static struct platform_driver broadwell_audio = { - .probe = broadwell_audio_probe, - .remove = broadwell_audio_remove, - .driver = { - .name = "broadwell-audio", - .pm = &snd_soc_pm_ops - }, -}; - -module_platform_driver(broadwell_audio) - -/* Module information */ -MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); -MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:broadwell-audio"); diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index d98376da425a..7c6c95e99ade 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -186,6 +186,17 @@ static const struct snd_soc_dapm_route gemini_map[] = { {"ssp2 Rx", NULL, "Capture"}, }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -231,10 +242,12 @@ static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, - &broxton_headset); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &broxton_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 75995d17597d..4bd93c3ba377 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -176,7 +176,7 @@ static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - rt298_mic_detect(component, &broxton_headset); + snd_soc_component_set_jack(component, &broxton_headset, NULL); snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index 0eed68a11f7e..ae899866863e 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -126,7 +126,7 @@ static int byt_cht_cx2072x_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC); + SND_SOC_DAIFMT_BP_FP); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index eb19bf16afad..a0c8f1d3f8ce 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -81,7 +81,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC); + SND_SOC_DAIFMT_BP_FP); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index a08507783e44..6432b83f616f 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -265,7 +265,7 @@ static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC + SND_SOC_DAIFMT_BP_FP ); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); diff --git a/sound/soc/intel/boards/bytcht_nocodec.c b/sound/soc/intel/boards/bytcht_nocodec.c index 115c2bcaabd4..7fc03f2efd35 100644 --- a/sound/soc/intel/boards/bytcht_nocodec.c +++ b/sound/soc/intel/boards/bytcht_nocodec.c @@ -61,7 +61,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC); + SND_SOC_DAIFMT_BP_FP); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index ed9fa1728722..fb9d9e271845 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1413,7 +1413,7 @@ static int byt_rt5640_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC); + SND_SOC_DAIFMT_BP_FP); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; @@ -1636,7 +1636,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) * with the codec driver/pdata are non-existent */ - struct acpi_chan_package chan_package; + struct acpi_chan_package chan_package = { 0 }; /* format specified: 2 64-bit integers */ struct acpi_buffer format = {sizeof("NN"), "NN"}; diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index d467fcaa48ea..2beb686768f2 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -706,7 +706,7 @@ static int byt_rt5651_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC + SND_SOC_DAIFMT_BP_FP ); if (ret < 0) { @@ -952,7 +952,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) * with the codec driver/pdata are non-existent */ - struct acpi_chan_package chan_package; + struct acpi_chan_package chan_package = { 0 }; /* format specified: 2 64-bit integers */ struct acpi_buffer format = {sizeof("NN"), "NN"}; diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index 330c0ace1638..45a6805787f5 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -265,7 +265,7 @@ static int byt_wm5102_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC); + SND_SOC_DAIFMT_BP_FP); if (ret) { dev_err(rtd->dev, "Error setting format to I2S: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index a5160f27adea..64eb73525ee3 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -264,8 +264,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBC_CFC; + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_BP_FP; ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt); if (ret < 0) { diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 45c301ea5e00..96501aed8bee 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -362,7 +362,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC + SND_SOC_DAIFMT_BP_FP ); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); @@ -372,7 +372,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC + SND_SOC_DAIFMT_BC_FC ); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); @@ -396,7 +396,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBC_CFC); + SND_SOC_DAIFMT_BC_FC); if (ret < 0) { dev_err(rtd->dev, "can't set format to TDM %d\n", ret); return ret; @@ -603,7 +603,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) * with the codec driver/pdata are non-existent */ - struct acpi_chan_package chan_package; + struct acpi_chan_package chan_package = { 0 }; /* format specified: 2 64-bit integers */ struct acpi_buffer format = {sizeof("NN"), "NN"}; diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index c80324f34b1b..ca47f6476b07 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -300,7 +300,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC); + SND_SOC_DAIFMT_BP_FP); if (ret < 0) { dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index a99f74a15b5f..20da83d9eece 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -121,6 +121,17 @@ static const struct snd_soc_dapm_route cml_rt1011_tt_map[] = { {"TR Ext Spk", NULL, "TR SPO" }, }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -137,11 +148,13 @@ static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3, - &ctx->headset); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 170164baae7d..cf0f89db3e20 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -78,6 +78,17 @@ static const struct snd_soc_dapm_widget geminilake_widgets[] = { SND_SOC_DAPM_SPK("HDMI3", NULL), }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static const struct snd_soc_dapm_route geminilake_map[] = { /* HP jack connectors - unknown if we have jack detection */ { "Headphone Jack", NULL, "HPOL" }, @@ -173,10 +184,12 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, - &ctx->geminilake_headset); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->geminilake_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c deleted file mode 100644 index aa61e101f793..000000000000 --- a/sound/soc/intel/boards/haswell.c +++ /dev/null @@ -1,202 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Intel Haswell Lynxpoint SST Audio - * - * Copyright (C) 2013, Intel Corporation. All rights reserved. - */ - -#include <linux/module.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-acpi.h> -#include <sound/pcm_params.h> - -#include "../../codecs/rt5640.h" - -/* Haswell ULT platforms have a Headphone and Mic jack */ -static const struct snd_soc_dapm_widget haswell_widgets[] = { - SND_SOC_DAPM_HP("Headphones", NULL), - SND_SOC_DAPM_MIC("Mic", NULL), -}; - -static const struct snd_soc_dapm_route haswell_rt5640_map[] = { - - {"Headphones", NULL, "HPOR"}, - {"Headphones", NULL, "HPOL"}, - {"IN2P", NULL, "Mic"}, - - /* CODEC BE connections */ - {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, - {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, -}; - -static int haswell_ssp0_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 ADSP will covert the FE rate to 48k, 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 haswell_rt5640_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, RT5640_SCLK_S_MCLK, 12288000, - SND_SOC_CLOCK_IN); - - if (ret < 0) { - dev_err(rtd->dev, "can't set codec sysclk configuration\n"); - return ret; - } - - /* set correct codec filter for DAI format and clock config */ - snd_soc_component_update_bits(codec_dai->component, 0x83, 0xffff, 0x8000); - - return ret; -} - -static const struct snd_soc_ops haswell_rt5640_ops = { - .hw_params = haswell_rt5640_hw_params, -}; - -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -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(codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT33CA:00", "rt5640-aif1"))); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); - -SND_SOC_DAILINK_DEF(ssp0_port, - DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); - -static struct snd_soc_dai_link haswell_rt5640_dais[] = { - /* Front End DAI links */ - { - .name = "System", - .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", - .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, - .no_pcm = 1, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = haswell_ssp0_fixup, - .ops = &haswell_rt5640_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp0_port, codec, platform), - }, -}; - -/* audio machine driver for Haswell Lynxpoint DSP + RT5640 */ -static struct snd_soc_card haswell_rt5640 = { - .name = "haswell-rt5640", - .owner = THIS_MODULE, - .dai_link = haswell_rt5640_dais, - .num_links = ARRAY_SIZE(haswell_rt5640_dais), - .dapm_widgets = haswell_widgets, - .num_dapm_widgets = ARRAY_SIZE(haswell_widgets), - .dapm_routes = haswell_rt5640_map, - .num_dapm_routes = ARRAY_SIZE(haswell_rt5640_map), - .fully_routed = true, -}; - -static int haswell_audio_probe(struct platform_device *pdev) -{ - struct snd_soc_acpi_mach *mach; - int ret; - - haswell_rt5640.dev = &pdev->dev; - - /* override platform name, if required */ - mach = pdev->dev.platform_data; - ret = snd_soc_fixup_dai_links_platform_name(&haswell_rt5640, - mach->mach_params.platform); - if (ret) - return ret; - - return devm_snd_soc_register_card(&pdev->dev, &haswell_rt5640); -} - -static struct platform_driver haswell_audio = { - .probe = haswell_audio_probe, - .driver = { - .name = "haswell-audio", - .pm = &snd_soc_pm_ops, - }, -}; - -module_platform_driver(haswell_audio) - -/* Module information */ -MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); -MODULE_DESCRIPTION("Intel SST Audio for Haswell Lynxpoint"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:haswell-audio"); diff --git a/sound/soc/intel/boards/hda_dsp_common.c b/sound/soc/intel/boards/hda_dsp_common.c index 5c31ddc0884a..83c7dfbccd9d 100644 --- a/sound/soc/intel/boards/hda_dsp_common.c +++ b/sound/soc/intel/boards/hda_dsp_common.c @@ -62,8 +62,8 @@ int hda_dsp_hdmi_build_controls(struct snd_soc_card *card, hpcm->pcm = spcm; hpcm->device = spcm->device; dev_dbg(card->dev, - "%s: mapping HDMI converter %d to PCM %d (%p)\n", - __func__, i, hpcm->device, spcm); + "mapping HDMI converter %d to PCM %d (%p)\n", + i, hpcm->device, spcm); } else { hpcm->pcm = NULL; hpcm->device = SNDRV_PCM_INVALID_DEVICE; diff --git a/sound/soc/intel/boards/hsw_rt5640.c b/sound/soc/intel/boards/hsw_rt5640.c new file mode 100644 index 000000000000..050c53ebd6ba --- /dev/null +++ b/sound/soc/intel/boards/hsw_rt5640.c @@ -0,0 +1,177 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Sound card driver for Intel Haswell Lynx Point with Realtek 5640 + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../codecs/rt5640.h" + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route card_routes[] = { + {"Headphones", NULL, "HPOR"}, + {"Headphones", NULL, "HPOL"}, + {"IN2P", NULL, "Mic"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +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 48k, 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, RT5640_SCLK_S_MCLK, 12288000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "set codec sysclk failed: %d\n", ret); + return ret; + } + + /* Set correct codec filter for DAI format and clock config. */ + snd_soc_component_update_bits(codec_dai->component, 0x83, 0xffff, 0x8000); + + 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(codec, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT33CA:00", "rt5640-aif1"))); +SND_SOC_DAILINK_DEF(platform, DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); +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", + .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", + .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, + .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 struct snd_soc_card hsw_rt5640_card = { + .name = "haswell-rt5640", + .owner = THIS_MODULE, + .dai_link = card_dai_links, + .num_links = ARRAY_SIZE(card_dai_links), + .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, +}; + +static int hsw_rt5640_probe(struct platform_device *pdev) +{ + struct snd_soc_acpi_mach *mach; + struct device *dev = &pdev->dev; + int ret; + + hsw_rt5640_card.dev = dev; + mach = dev_get_platdata(dev); + + ret = snd_soc_fixup_dai_links_platform_name(&hsw_rt5640_card, mach->mach_params.platform); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, &hsw_rt5640_card); +} + +static struct platform_driver hsw_rt5640_driver = { + .probe = hsw_rt5640_probe, + .driver = { + .name = "hsw_rt5640", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(hsw_rt5640_driver) + +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Sound card driver for Intel Haswell Lynx Point with Realtek 5640"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:hsw_rt5640"); diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index ceabed85e9da..329457e3e3a2 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -99,6 +99,17 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_POST_PMD), }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static const struct snd_soc_dapm_route kabylake_map[] = { { "Headphone Jack", NULL, "HPL" }, { "Headphone Jack", NULL, "HPR" }, @@ -179,10 +190,12 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(kabylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, - &ctx->kabylake_headset); + ret = snd_soc_card_jack_new_pins(kabylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->kabylake_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 703ccff634b0..362579f25835 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -119,6 +119,17 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_POST_PMD), }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static const struct snd_soc_dapm_route kabylake_map[] = { /* speaker */ { "Left Spk", NULL, "Left BE_OUT" }, @@ -354,10 +365,12 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(kabylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, - &ctx->kabylake_headset); + ret = snd_soc_card_jack_new_pins(kabylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->kabylake_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 8d37b2676a81..2d4224c5b152 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -206,6 +206,17 @@ static const struct snd_soc_dapm_widget kabylake_5663_widgets[] = { SND_SOC_DAPM_POST_PMD), }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static const struct snd_soc_dapm_route kabylake_5663_map[] = { { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, @@ -271,10 +282,12 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(kabylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3, - &ctx->kabylake_headset); + ret = snd_soc_card_jack_new_pins(kabylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &ctx->kabylake_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 564c70a0fbc8..2c79fca57b19 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -145,6 +145,17 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static const struct snd_soc_dapm_route kabylake_map[] = { /* Headphones */ { "Headphone Jack", NULL, "Platform Clock" }, @@ -228,10 +239,12 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(&kabylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3, - &ctx->kabylake_headset); + ret = snd_soc_card_jack_new_pins(&kabylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &ctx->kabylake_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index f4b4eeca3e03..81144efb4b44 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -75,7 +75,7 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); int ret = 0; - dev_dbg(card->dev, "%s: dai link name - %s\n", __func__, link->name); + dev_dbg(card->dev, "dai link name - %s\n", link->name); link->platforms->name = ctx->platform_name; link->nonatomic = 1; @@ -203,7 +203,7 @@ static int skl_hda_audio_probe(struct platform_device *pdev) struct skl_hda_private *ctx; int ret; - dev_dbg(&pdev->dev, "%s: entry\n", __func__); + dev_dbg(&pdev->dev, "entry\n"); ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 8e2d03e36079..8dceb0b02581 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -97,6 +97,17 @@ static const struct snd_soc_dapm_widget skylake_widgets[] = { SND_SOC_DAPM_POST_PMD), }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static const struct snd_soc_dapm_route skylake_map[] = { /* HP jack connectors - unknown if we have jack detection */ { "Headphone Jack", NULL, "HPOL" }, @@ -163,9 +174,11 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(&skylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3, &skylake_headset); + ret = snd_soc_card_jack_new_pins(&skylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, &skylake_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 501f0bbfc404..62c0d46d0086 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -101,6 +101,17 @@ static const struct snd_soc_dapm_widget skylake_widgets[] = { SND_SOC_DAPM_POST_PMD), }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static const struct snd_soc_dapm_route skylake_map[] = { /* HP jack connectors - unknown if we have jack detection */ {"Headphone Jack", NULL, "HPOL"}, @@ -182,9 +193,11 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) * 4 buttons here map to the google Reference headset * The use of these buttons can be decided by the user space. */ - ret = snd_soc_card_jack_new(&skylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3, &skylake_headset); + ret = snd_soc_card_jack_new_pins(&skylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, &skylake_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index e9f9520dcea4..4f3d655e2bfa 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -133,7 +133,7 @@ static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - rt286_mic_detect(component, &skylake_headset); + snd_soc_component_set_jack(component, &skylake_headset, NULL); snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index 6a979c333bc5..85ffd065895d 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -41,8 +41,13 @@ #define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) #define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(25) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_BT_OFFLOAD_PRESENT BIT(25) +#define SOF_CS42L42_SSP_BT_SHIFT 26 +#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) +#define SOF_CS42L42_SSP_BT(quirk) \ + (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(29) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(30) enum { LINK_NONE = 0, @@ -50,6 +55,18 @@ enum { LINK_SPK = 2, LINK_DMIC = 3, LINK_HDMI = 4, + LINK_BT = 5, +}; + +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, }; /* Default: SSP2 */ @@ -98,11 +115,13 @@ static int sof_cs42l42_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3, - jack); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + jack, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; @@ -277,6 +296,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int create_spk_amp_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -466,9 +492,50 @@ devm_err: return -ENOMEM; } +static int create_bt_offload_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_bt) +{ + /* bt offload */ + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) + return 0; + + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ssp_bt); + if (!links[*id].name) + goto devm_err; + + links[*id].id = *id; + links[*id].codecs = dummy_component; + links[*id].num_codecs = ARRAY_SIZE(dummy_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_bt); + if (!links[*id].cpus->dai_name) + goto devm_err; + + (*id)++; + + return 0; + +devm_err: + return -ENOMEM; +} + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, + int ssp_bt, int dmic_be_num, int hdmi_num) { @@ -521,6 +588,14 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; } break; + case LINK_BT: + ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); + if (ret < 0) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + break; case LINK_NONE: /* caught here if it's not used as terminator in macro */ default: @@ -542,7 +617,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; - int ret, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -567,6 +642,9 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + SOF_CS42L42_SSP_BT_SHIFT; + ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; @@ -577,9 +655,11 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_cs42l42.num_links++; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, - dmic_be_num, hdmi_num); + ssp_bt, dmic_be_num, hdmi_num); if (!dai_links) return -ENOMEM; @@ -620,6 +700,17 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1)) | SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), }, + { + .name = "adl_mx98360a_cs4242", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1) | + SOF_CS42L42_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_SSP_BT(2) | + SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index a83f30b687cf..34cf849a8344 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -135,6 +135,17 @@ static const struct snd_soc_dapm_route max98360a_map[] = { {"DMic", NULL, "SoC DMIC"}, }; +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static struct snd_soc_jack headset; static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) @@ -156,11 +167,13 @@ static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3 | SND_JACK_LINEOUT, - &headset); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index 23d03e0f7759..c7f33c89588e 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -28,6 +28,24 @@ #define SOF_ES8336_SSP_CODEC_MASK (GENMASK(3, 0)) #define SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK BIT(4) + +/* HDMI capture*/ +#define SOF_SSP_HDMI_CAPTURE_PRESENT BIT(14) +#define SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT 15 +#define SOF_NO_OF_HDMI_CAPTURE_SSP_MASK (GENMASK(16, 15)) +#define SOF_NO_OF_HDMI_CAPTURE_SSP(quirk) \ + (((quirk) << SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT) & SOF_NO_OF_HDMI_CAPTURE_SSP_MASK) + +#define SOF_HDMI_CAPTURE_1_SSP_SHIFT 7 +#define SOF_HDMI_CAPTURE_1_SSP_MASK (GENMASK(9, 7)) +#define SOF_HDMI_CAPTURE_1_SSP(quirk) \ + (((quirk) << SOF_HDMI_CAPTURE_1_SSP_SHIFT) & SOF_HDMI_CAPTURE_1_SSP_MASK) + +#define SOF_HDMI_CAPTURE_2_SSP_SHIFT 10 +#define SOF_HDMI_CAPTURE_2_SSP_MASK (GENMASK(12, 10)) +#define SOF_HDMI_CAPTURE_2_SSP(quirk) \ + (((quirk) << SOF_HDMI_CAPTURE_2_SSP_SHIFT) & SOF_HDMI_CAPTURE_2_SSP_MASK) + #define SOF_ES8336_ENABLE_DMIC BIT(5) #define SOF_ES8336_JD_INVERTED BIT(6) #define SOF_ES8336_HEADPHONE_GPIO BIT(7) @@ -57,28 +75,26 @@ static const struct acpi_gpio_params enable_gpio0 = { 0, 0, true }; static const struct acpi_gpio_params enable_gpio1 = { 1, 0, true }; static const struct acpi_gpio_mapping acpi_speakers_enable_gpio0[] = { - { "speakers-enable-gpios", &enable_gpio0, 1 }, + { "speakers-enable-gpios", &enable_gpio0, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, { } }; static const struct acpi_gpio_mapping acpi_speakers_enable_gpio1[] = { - { "speakers-enable-gpios", &enable_gpio1, 1 }, + { "speakers-enable-gpios", &enable_gpio1, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, }; static const struct acpi_gpio_mapping acpi_enable_both_gpios[] = { - { "speakers-enable-gpios", &enable_gpio0, 1 }, - { "headphone-enable-gpios", &enable_gpio1, 1 }, + { "speakers-enable-gpios", &enable_gpio0, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, + { "headphone-enable-gpios", &enable_gpio1, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, { } }; static const struct acpi_gpio_mapping acpi_enable_both_gpios_rev_order[] = { - { "speakers-enable-gpios", &enable_gpio1, 1 }, - { "headphone-enable-gpios", &enable_gpio0, 1 }, + { "speakers-enable-gpios", &enable_gpio1, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, + { "headphone-enable-gpios", &enable_gpio0, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, { } }; -static const struct acpi_gpio_mapping *gpio_mapping = acpi_speakers_enable_gpio0; - static void log_quirks(struct device *dev) { dev_info(dev, "quirk mask %#lx\n", quirk); @@ -272,15 +288,6 @@ static int sof_es8336_quirk_cb(const struct dmi_system_id *id) { quirk = (unsigned long)id->driver_data; - if (quirk & SOF_ES8336_HEADPHONE_GPIO) { - if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) - gpio_mapping = acpi_enable_both_gpios; - else - gpio_mapping = acpi_enable_both_gpios_rev_order; - } else if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) { - gpio_mapping = acpi_speakers_enable_gpio1; - } - return 1; } @@ -356,6 +363,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int sof_es8336_late_probe(struct snd_soc_card *card) { struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); @@ -507,6 +521,37 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, id++; } + /* HDMI-In SSP */ + if (quirk & SOF_SSP_HDMI_CAPTURE_PRESENT) { + int num_of_hdmi_ssp = (quirk & SOF_NO_OF_HDMI_CAPTURE_SSP_MASK) >> + SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT; + + for (i = 1; i <= num_of_hdmi_ssp; i++) { + int port = (i == 1 ? (quirk & SOF_HDMI_CAPTURE_1_SSP_MASK) >> + SOF_HDMI_CAPTURE_1_SSP_SHIFT : + (quirk & SOF_HDMI_CAPTURE_2_SSP_MASK) >> + SOF_HDMI_CAPTURE_2_SSP_SHIFT); + + links[id].cpus = &cpus[id]; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", port); + if (!links[id].cpus->dai_name) + return NULL; + links[id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-HDMI", port); + if (!links[id].name) + return NULL; + links[id].id = id + hdmi_id_offset; + links[id].codecs = dummy_component; + links[id].num_codecs = ARRAY_SIZE(dummy_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + links[id].num_cpus = 1; + id++; + } + } + return links; devm_err: @@ -529,6 +574,7 @@ static int sof_es8336_probe(struct platform_device *pdev) struct acpi_device *adev; struct snd_soc_dai_link *dai_links; struct device *codec_dev; + const struct acpi_gpio_mapping *gpio_mapping; unsigned int cnt = 0; int dmic_be_num = 0; int hdmi_num = 3; @@ -541,29 +587,34 @@ static int sof_es8336_probe(struct platform_device *pdev) card = &sof_es8336_card; card->dev = dev; + if (pdev->id_entry && pdev->id_entry->driver_data) + quirk = (unsigned long)pdev->id_entry->driver_data; + /* check GPIO DMI quirks */ dmi_check_system(sof_es8336_quirk_table); - if (!mach->mach_params.i2s_link_mask) { - dev_warn(dev, "No I2S link information provided, using SSP0. This may need to be modified with the quirk module parameter\n"); - } else { - /* - * Set configuration based on platform NHLT. - * In this machine driver, we can only support one SSP for the - * ES8336 link, the else-if below are intentional. - * In some cases multiple SSPs can be reported by NHLT, starting MSB-first - * seems to pick the right connection. - */ - unsigned long ssp = 0; - - if (mach->mach_params.i2s_link_mask & BIT(2)) - ssp = SOF_ES8336_SSP_CODEC(2); - else if (mach->mach_params.i2s_link_mask & BIT(1)) - ssp = SOF_ES8336_SSP_CODEC(1); - else if (mach->mach_params.i2s_link_mask & BIT(0)) - ssp = SOF_ES8336_SSP_CODEC(0); - - quirk |= ssp; + /* Use NHLT configuration only for Non-HDMI capture use case. + * Because more than one SSP will be enabled for HDMI capture hence wrong codec + * SSP will be set. + */ + if (mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER) { + if (!mach->mach_params.i2s_link_mask) { + dev_warn(dev, "No I2S link information provided, using SSP0. This may need to be modified with the quirk module parameter\n"); + } else { + /* + * Set configuration based on platform NHLT. + * In this machine driver, we can only support one SSP for the + * ES8336 link. + * In some cases multiple SSPs can be reported by NHLT, starting MSB-first + * seems to pick the right connection. + */ + unsigned long ssp; + + /* fls returns 1-based results, SSPs indices are 0-based */ + ssp = fls(mach->mach_params.i2s_link_mask) - 1; + + quirk |= ssp; + } } if (mach->mach_params.dmic_num) @@ -579,7 +630,13 @@ static int sof_es8336_probe(struct platform_device *pdev) if (quirk & SOF_ES8336_ENABLE_DMIC) dmic_be_num = 2; - sof_es8336_card.num_links += dmic_be_num + hdmi_num; + /* compute number of dai links */ + sof_es8336_card.num_links = 1 + dmic_be_num + hdmi_num; + + if (quirk & SOF_SSP_HDMI_CAPTURE_PRESENT) + sof_es8336_card.num_links += (quirk & SOF_NO_OF_HDMI_CAPTURE_SSP_MASK) >> + SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT; + dai_links = sof_card_dai_links_create(dev, SOF_ES8336_SSP_CODEC(quirk), dmic_be_num, hdmi_num); @@ -635,6 +692,17 @@ static int sof_es8336_probe(struct platform_device *pdev) } /* get speaker enable GPIO */ + if (quirk & SOF_ES8336_HEADPHONE_GPIO) { + if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) + gpio_mapping = acpi_enable_both_gpios; + else + gpio_mapping = acpi_enable_both_gpios_rev_order; + } else if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) { + gpio_mapping = acpi_speakers_enable_gpio1; + } else { + gpio_mapping = acpi_speakers_enable_gpio0; + } + ret = devm_acpi_dev_add_driver_gpios(codec_dev, gpio_mapping); if (ret) dev_warn(codec_dev, "unable to add GPIO mapping table\n"); @@ -690,6 +758,21 @@ static int sof_es8336_remove(struct platform_device *pdev) return 0; } +static const struct platform_device_id board_ids[] = { + { + .name = "adl_es83x6_c1_h02", + .driver_data = (kernel_ulong_t)(SOF_ES8336_SSP_CODEC(1) | + SOF_NO_OF_HDMI_CAPTURE_SSP(2) | + SOF_HDMI_CAPTURE_1_SSP(0) | + SOF_HDMI_CAPTURE_2_SSP(2) | + SOF_SSP_HDMI_CAPTURE_PRESENT | + SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK | + SOF_ES8336_JD_INVERTED), + }, + { } +}; +MODULE_DEVICE_TABLE(platform, board_ids); + static struct platform_driver sof_es8336_driver = { .driver = { .name = "sof-essx8336", @@ -697,6 +780,7 @@ static struct platform_driver sof_es8336_driver = { }, .probe = sof_es8336_probe, .remove = sof_es8336_remove, + .id_table = board_ids, }; module_platform_driver(sof_es8336_driver); diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index 97dcd204a246..8d7e5ba9e516 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -81,6 +81,17 @@ static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int sof_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) { struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -93,11 +104,13 @@ static int sof_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3, - &ctx->sof_headset); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sof_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; @@ -177,11 +190,6 @@ static int sof_card_late_probe(struct snd_soc_card *card) struct sof_hdmi_pcm *pcm; int err; - if (list_empty(&ctx->hdmi_pcm_list)) - return -EINVAL; - - pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head); - if (sof_nau8825_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { /* Disable Left and Right Spk pin after boot */ snd_soc_dapm_disable_pin(dapm, "Left Spk"); @@ -191,6 +199,11 @@ static int sof_card_late_probe(struct snd_soc_card *card) return err; } + if (list_empty(&ctx->hdmi_pcm_list)) + return -EINVAL; + + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head); + return hda_dsp_hdmi_build_controls(card, pcm->codec_dai->component); } diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c index 6815204e58d5..d4c67d5340a9 100644 --- a/sound/soc/intel/boards/sof_pcm512x.c +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -419,7 +419,7 @@ static int sof_audio_probe(struct platform_device *pdev) static int sof_pcm512x_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - struct snd_soc_component *component = NULL; + struct snd_soc_component *component; for_each_card_components(card, component) { if (!strcmp(component->name, pcm512x_component[0].name)) { diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index 2ab568c1d40b..b9643ca2e2f2 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -463,26 +463,26 @@ EXPORT_SYMBOL_NS(sof_rt1308_dai_link, SND_SOC_INTEL_SOF_REALTEK_COMMON); * 2-amp Configuration for RT1019 */ -static const struct snd_soc_dapm_route rt1019_dapm_routes[] = { +static const struct snd_soc_dapm_route rt1019p_dapm_routes[] = { /* speaker */ { "Left Spk", NULL, "Speaker" }, { "Right Spk", NULL, "Speaker" }, }; -static struct snd_soc_dai_link_component rt1019_components[] = { +static struct snd_soc_dai_link_component rt1019p_components[] = { { - .name = RT1019_DEV0_NAME, - .dai_name = RT1019_CODEC_DAI, + .name = RT1019P_DEV0_NAME, + .dai_name = RT1019P_CODEC_DAI, }, }; -static int rt1019_init(struct snd_soc_pcm_runtime *rtd) +static int rt1019p_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; int ret; - ret = snd_soc_dapm_add_routes(&card->dapm, rt1019_dapm_routes, - ARRAY_SIZE(rt1019_dapm_routes)); + ret = snd_soc_dapm_add_routes(&card->dapm, rt1019p_dapm_routes, + ARRAY_SIZE(rt1019p_dapm_routes)); if (ret) { dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); return ret; @@ -490,13 +490,13 @@ static int rt1019_init(struct snd_soc_pcm_runtime *rtd) return ret; } -void sof_rt1019_dai_link(struct snd_soc_dai_link *link) +void sof_rt1019p_dai_link(struct snd_soc_dai_link *link) { - link->codecs = rt1019_components; - link->num_codecs = ARRAY_SIZE(rt1019_components); - link->init = rt1019_init; + link->codecs = rt1019p_components; + link->num_codecs = ARRAY_SIZE(rt1019p_components); + link->init = rt1019p_init; } -EXPORT_SYMBOL_NS(sof_rt1019_dai_link, SND_SOC_INTEL_SOF_REALTEK_COMMON); +EXPORT_SYMBOL_NS(sof_rt1019p_dai_link, SND_SOC_INTEL_SOF_REALTEK_COMMON); MODULE_DESCRIPTION("ASoC Intel SOF Realtek helpers"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h index ec3eea633e04..778443421090 100644 --- a/sound/soc/intel/boards/sof_realtek_common.h +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -39,9 +39,9 @@ void sof_rt1015_codec_conf(struct snd_soc_card *card); #define RT1308_DEV0_NAME "i2c-10EC1308:00" void sof_rt1308_dai_link(struct snd_soc_dai_link *link); -#define RT1019_CODEC_DAI "HiFi" -#define RT1019_DEV0_NAME "RTL1019:00" +#define RT1019P_CODEC_DAI "HiFi" +#define RT1019P_DEV0_NAME "RTL1019:00" -void sof_rt1019_dai_link(struct snd_soc_dai_link *link); +void sof_rt1019p_dai_link(struct snd_soc_dai_link *link); #endif /* __SOF_REALTEK_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 4a90a0a5d831..045965312245 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -247,6 +247,17 @@ static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -294,11 +305,13 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3, - &ctx->sof_headset); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sof_headset, + jack_pins, + ARRAY_SIZE(jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; @@ -434,6 +447,15 @@ static int sof_card_late_probe(struct snd_soc_card *card) struct sof_hdmi_pcm *pcm; int err; + if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { + /* Disable Left and Right Spk pin after boot */ + snd_soc_dapm_disable_pin(dapm, "Left Spk"); + snd_soc_dapm_disable_pin(dapm, "Right Spk"); + err = snd_soc_dapm_sync(dapm); + if (err < 0) + return err; + } + /* HDMI is not supported by SOF on Baytrail/CherryTrail */ if (is_legacy_cpu || !ctx->idisp_codec) return 0; @@ -464,15 +486,6 @@ static int sof_card_late_probe(struct snd_soc_card *card) return err; } - if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { - /* Disable Left and Right Spk pin after boot */ - snd_soc_dapm_disable_pin(dapm, "Left Spk"); - snd_soc_dapm_disable_pin(dapm, "Right Spk"); - err = snd_soc_dapm_sync(dapm); - if (err < 0) - return err; - } - return hdac_hdmi_jack_port_init(component, &card->dapm); } @@ -731,7 +744,7 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) { sof_rt1015p_dai_link(&links[id]); } else if (sof_rt5682_quirk & SOF_RT1019_SPEAKER_AMP_PRESENT) { - sof_rt1019_dai_link(&links[id]); + sof_rt1019p_dai_link(&links[id]); } else if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { links[id].codecs = max_98373_components; @@ -1079,6 +1092,14 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "mtl_mx98357_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index ad826ad82d51..a49bfaab6b21 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -250,6 +250,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { .callback = sof_sdw_quirk_cb, .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0AF0") + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2 | + SOF_SDW_FOUR_SPK), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0AF3"), }, /* No Jack */ @@ -315,6 +325,23 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { RT711_JD2 | SOF_SDW_FOUR_SPK), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "HP"), + DMI_MATCH(DMI_PRODUCT_NAME, "OMEN by HP Gaming Laptop 16-k0xxx"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2), + }, + /* MeteorLake devices */ + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Intel_mtlrvp"), + }, + .driver_data = (void *)(RT711_JD1 | SOF_SDW_TGL_HDMI), + }, {} }; @@ -1127,10 +1154,14 @@ static int sof_card_dai_links_create(struct device *dev, for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) codec_info_list[i].amp_num = 0; - if (sof_sdw_quirk & SOF_SDW_TGL_HDMI) - hdmi_num = SOF_TGL_HDMI_COUNT; - else - hdmi_num = SOF_PRE_TGL_HDMI_COUNT; + if (mach_params->codec_mask & IDISP_CODEC_MASK) { + ctx->idisp_codec = true; + + if (sof_sdw_quirk & SOF_SDW_TGL_HDMI) + hdmi_num = SOF_TGL_HDMI_COUNT; + else + hdmi_num = SOF_PRE_TGL_HDMI_COUNT; + } ssp_mask = SOF_SSP_GET_PORT(sof_sdw_quirk); /* @@ -1150,9 +1181,6 @@ static int sof_card_dai_links_create(struct device *dev, return ret; } - if (mach_params->codec_mask & IDISP_CODEC_MASK) - ctx->idisp_codec = true; - /* enable dmic01 & dmic16k */ dmic_num = (sof_sdw_quirk & SOF_SDW_PCH_DMIC || mach_params->dmic_num) ? 2 : 0; comp_num += dmic_num; @@ -1375,7 +1403,9 @@ HDMI: static int sof_sdw_card_late_probe(struct snd_soc_card *card) { - int i, ret; + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + int ret = 0; + int i; for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { if (!codec_info_list[i].late_probe) @@ -1386,7 +1416,10 @@ static int sof_sdw_card_late_probe(struct snd_soc_card *card) return ret; } - return sof_sdw_hdmi_card_late_probe(card); + if (ctx->idisp_codec) + ret = sof_sdw_hdmi_card_late_probe(card); + + return ret; } /* SoC card */ @@ -1433,7 +1466,7 @@ static int mc_probe(struct platform_device *pdev) int amp_num = 0, i; int ret; - dev_dbg(&pdev->dev, "Entry %s\n", __func__); + dev_dbg(&pdev->dev, "Entry\n"); ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c index 49ff0871e9e7..8291967f23f3 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711.c +++ b/sound/soc/intel/boards/sof_sdw_rt711.c @@ -139,6 +139,9 @@ int sof_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_l { struct mc_private *ctx = snd_soc_card_get_drvdata(card); + if (!ctx->headset_codec_dev) + return 0; + device_remove_software_node(ctx->headset_codec_dev); put_device(ctx->headset_codec_dev); diff --git a/sound/soc/intel/boards/sof_sdw_rt711_sdca.c b/sound/soc/intel/boards/sof_sdw_rt711_sdca.c index b3fc32bacfa8..7f16304d025b 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt711_sdca.c @@ -140,6 +140,9 @@ int sof_sdw_rt711_sdca_exit(struct snd_soc_card *card, struct snd_soc_dai_link * { struct mc_private *ctx = snd_soc_card_get_drvdata(card); + if (!ctx->headset_codec_dev) + return 0; + device_remove_software_node(ctx->headset_codec_dev); put_device(ctx->headset_codec_dev); diff --git a/sound/soc/intel/catpt/device.c b/sound/soc/intel/catpt/device.c index 85a34e37316d..d48a71d2cf1e 100644 --- a/sound/soc/intel/catpt/device.c +++ b/sound/soc/intel/catpt/device.c @@ -254,14 +254,11 @@ static int catpt_acpi_probe(struct platform_device *pdev) return -ENODEV; } - spec = device_get_match_data(dev); - if (!spec) - return -ENODEV; - cdev = devm_kzalloc(dev, sizeof(*cdev), GFP_KERNEL); if (!cdev) return -ENOMEM; + spec = (const struct catpt_spec *)id->driver_data; catpt_dev_init(cdev, dev, spec); /* map DSP bar address */ diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index a26000cd5ceb..30ca5416c9a3 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -667,7 +667,9 @@ static int catpt_dai_pcm_new(struct snd_soc_pcm_runtime *rtm, if (!memcmp(&cdev->devfmt[devfmt.iface], &devfmt, sizeof(devfmt))) return 0; - pm_runtime_get_sync(cdev->dev); + ret = pm_runtime_resume_and_get(cdev->dev); + if (ret < 0 && ret != -EACCES) + return ret; ret = catpt_ipc_set_device_format(cdev, &devfmt); @@ -853,9 +855,12 @@ static int catpt_mixer_volume_get(struct snd_kcontrol *kcontrol, snd_soc_kcontrol_component(kcontrol); struct catpt_dev *cdev = dev_get_drvdata(component->dev); u32 dspvol; + int ret; int i; - pm_runtime_get_sync(cdev->dev); + ret = pm_runtime_resume_and_get(cdev->dev); + if (ret < 0 && ret != -EACCES) + return ret; for (i = 0; i < CATPT_CHANNELS_MAX; i++) { dspvol = catpt_mixer_volume(cdev, &cdev->mixer, i); @@ -876,7 +881,9 @@ static int catpt_mixer_volume_put(struct snd_kcontrol *kcontrol, struct catpt_dev *cdev = dev_get_drvdata(component->dev); int ret; - pm_runtime_get_sync(cdev->dev); + ret = pm_runtime_resume_and_get(cdev->dev); + if (ret < 0 && ret != -EACCES) + return ret; ret = catpt_set_dspvol(cdev, cdev->mixer.mixer_hw_id, ucontrol->value.integer.value); @@ -897,6 +904,7 @@ static int catpt_stream_volume_get(struct snd_kcontrol *kcontrol, struct catpt_dev *cdev = dev_get_drvdata(component->dev); long *ctlvol = (long *)kcontrol->private_value; u32 dspvol; + int ret; int i; stream = catpt_stream_find(cdev, pin_id); @@ -906,7 +914,9 @@ static int catpt_stream_volume_get(struct snd_kcontrol *kcontrol, return 0; } - pm_runtime_get_sync(cdev->dev); + ret = pm_runtime_resume_and_get(cdev->dev); + if (ret < 0 && ret != -EACCES) + return ret; for (i = 0; i < CATPT_CHANNELS_MAX; i++) { dspvol = catpt_stream_volume(cdev, stream, i); @@ -937,7 +947,9 @@ static int catpt_stream_volume_put(struct snd_kcontrol *kcontrol, return 0; } - pm_runtime_get_sync(cdev->dev); + ret = pm_runtime_resume_and_get(cdev->dev); + if (ret < 0 && ret != -EACCES) + return ret; ret = catpt_set_dspvol(cdev, stream->info.stream_hw_id, ucontrol->value.integer.value); @@ -1013,7 +1025,9 @@ static int catpt_loopback_switch_put(struct snd_kcontrol *kcontrol, return 0; } - pm_runtime_get_sync(cdev->dev); + ret = pm_runtime_resume_and_get(cdev->dev); + if (ret < 0 && ret != -EACCES) + return ret; ret = catpt_ipc_mute_loopback(cdev, stream->info.stream_hw_id, mute); diff --git a/sound/soc/intel/catpt/sysfs.c b/sound/soc/intel/catpt/sysfs.c index 9579e233a15d..1bdbcc04dc71 100644 --- a/sound/soc/intel/catpt/sysfs.c +++ b/sound/soc/intel/catpt/sysfs.c @@ -15,7 +15,9 @@ static ssize_t fw_version_show(struct device *dev, struct catpt_fw_version version; int ret; - pm_runtime_get_sync(cdev->dev); + ret = pm_runtime_resume_and_get(cdev->dev); + if (ret < 0 && ret != -EACCES) + return ret; ret = catpt_ipc_get_fw_version(cdev, &version); diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index fef0b2d1de68..8ca8f872ec80 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -9,6 +9,7 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m soc-acpi-intel-cml-match.o soc-acpi-intel-icl-match.o \ soc-acpi-intel-tgl-match.o soc-acpi-intel-ehl-match.o \ soc-acpi-intel-jsl-match.o soc-acpi-intel-adl-match.o \ + soc-acpi-intel-mtl-match.o \ soc-acpi-intel-hda-match.o \ soc-acpi-intel-sdw-mockup-match.o diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index c1385161cdc8..9990d5502d26 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -8,6 +8,11 @@ #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> +static const struct snd_soc_acpi_codecs essx_83x6 = { + .num_codecs = 3, + .codecs = { "ESSX8316", "ESSX8326", "ESSX8336"}, +}; + static const struct snd_soc_acpi_endpoint single_endpoint = { .num = 0, .aggregated = 0, @@ -137,6 +142,15 @@ static const struct snd_soc_acpi_adr_device rt1316_2_single_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt1316_3_single_adr[] = { + { + .adr = 0x000330025D131601ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt1316-1" + } +}; + static const struct snd_soc_acpi_adr_device rt714_0_adr[] = { { .adr = 0x000030025D071401ull, @@ -326,6 +340,20 @@ static const struct snd_soc_acpi_link_adr adl_sdw_rt1316_link2_rt714_link0[] = { {} }; +static const struct snd_soc_acpi_link_adr adl_sdw_rt711_link0_rt1316_link3[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_sdca_0_adr), + .adr_d = rt711_sdca_0_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(rt1316_3_single_adr), + .adr_d = rt1316_3_single_adr, + }, + {} +}; + static const struct snd_soc_acpi_adr_device mx8373_2_adr[] = { { .adr = 0x000223019F837300ull, @@ -412,6 +440,11 @@ static const struct snd_soc_acpi_codecs adl_max98390_amp = { .codecs = {"MX98390"} }; +static const struct snd_soc_acpi_codecs adl_lt6911_hdmi = { + .num_codecs = 1, + .codecs = {"INTC10B0"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .comp_ids = &adl_rt5682_rt5682s_hp, @@ -479,12 +512,34 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .drv_name = "adl_rt5682", .sof_tplg_filename = "sof-adl-rt5682.tplg", }, + { + .id = "10134242", + .drv_name = "adl_mx98360a_cs4242", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_tplg_filename = "sof-adl-max98360a-cs42l42.tplg", + }, /* place amp-only boards in the end of table */ { .id = "CSC3541", .drv_name = "adl_cs35l41", .sof_tplg_filename = "sof-adl-cs35l41.tplg", }, + { + .comp_ids = &essx_83x6, + .drv_name = "adl_es83x6_c1_h02", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_lt6911_hdmi, + .sof_tplg_filename = "sof-adl-es83x6-ssp1-hdmi-ssp02.tplg", + }, + { + .comp_ids = &essx_83x6, + .drv_name = "sof-essx8336", + .sof_tplg_filename = "sof-adl-es83x6", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER | + SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | + SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); @@ -540,6 +595,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[] = { .sof_tplg_filename = "sof-adl-rt1316-l2-mono-rt714-l0.tplg", }, { + .link_mask = 0x9, /* 2 active links required */ + .links = adl_sdw_rt711_link0_rt1316_link3, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-adl-rt711-l0-rt1316-l3.tplg", + }, + { .link_mask = 0x1, /* link0 required */ .links = adl_rvp, .drv_name = "sof_sdw", diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index 0441df97b260..cbcb649604e5 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -12,7 +12,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_haswell_machines[] = { { .id = "INT33CA", - .drv_name = "haswell-audio", + .drv_name = "hsw_rt5640", .fw_filename = "intel/IntcSST1.bin", .sof_tplg_filename = "sof-hsw.tplg", }, @@ -23,7 +23,7 @@ EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_haswell_machines); struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { { .id = "INT343A", - .drv_name = "broadwell-audio", + .drv_name = "bdw_rt286", .fw_filename = "intel/IntcSST2.bin", .sof_tplg_filename = "sof-bdw-rt286.tplg", }, @@ -41,7 +41,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { }, { .id = "INT33CA", - .drv_name = "haswell-audio", + .drv_name = "hsw_rt5640", .fw_filename = "intel/IntcSST2.bin", .sof_tplg_filename = "sof-bdw-rt5640.tplg", }, diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c new file mode 100644 index 000000000000..36c361fb28a4 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -0,0 +1,89 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * soc-acpi-intel-mtl-match.c - tables and support for MTL ACPI enumeration. + * + * Copyright (c) 2022, Intel Corporation. + * + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> +#include "soc-acpi-intel-sdw-mockup-match.h" + +static const struct snd_soc_acpi_codecs mtl_max98357a_amp = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + +static const struct snd_soc_acpi_codecs mtl_rt5682_rt5682s_hp = { + .num_codecs = 2, + .codecs = {"10EC5682", "RTL5682"}, +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { + { + .comp_ids = &mtl_rt5682_rt5682s_hp, + .drv_name = "mtl_mx98357_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &mtl_max98357a_amp, + .sof_tplg_filename = "sof-mtl-max98357a-rt5682.tplg", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_machines); + +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, +}; + +static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { + { + .adr = 0x000030025D071101ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt711" + } +}; + +static const struct snd_soc_acpi_link_adr mtl_rvp[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_sdca_0_adr), + .adr_d = rt711_sdca_0_adr, + }, + {} +}; + +/* this table is used when there is no I2S codec present */ +struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { + /* mockup tests need to be first */ + { + .link_mask = GENMASK(3, 0), + .links = sdw_mockup_headset_2amps_mic, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt711-rt1308-rt715.tplg", + }, + { + .link_mask = BIT(0) | BIT(1) | BIT(3), + .links = sdw_mockup_headset_1amp_mic, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt711-rt1308-mono-rt715.tplg", + }, + { + .link_mask = GENMASK(2, 0), + .links = sdw_mockup_mic_headset_1amp, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt715-rt711-rt1308-mono.tplg", + }, + { + .link_mask = BIT(0), + .links = mtl_rvp, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt711.tplg", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_sdw_machines); diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index a6fb74ba1c42..b4893365d01d 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -388,15 +388,17 @@ static snd_pcm_uframes_t kmb_pcm_pointer(struct snd_soc_component *component, } static const struct snd_soc_component_driver kmb_component = { - .name = "kmb", - .pcm_construct = kmb_platform_pcm_new, - .open = kmb_pcm_open, - .trigger = kmb_pcm_trigger, - .pointer = kmb_pcm_pointer, + .name = "kmb", + .pcm_construct = kmb_platform_pcm_new, + .open = kmb_pcm_open, + .trigger = kmb_pcm_trigger, + .pointer = kmb_pcm_pointer, + .legacy_dai_naming = 1, }; static const struct snd_soc_component_driver kmb_component_dma = { - .name = "kmb", + .name = "kmb", + .legacy_dai_naming = 1, }; static int kmb_probe(struct snd_soc_dai *cpu_dai) @@ -497,11 +499,11 @@ static int kmb_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) int ret; switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_BC_FC: kmb_i2s->clock_provider = false; ret = 0; break; - case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_BP_FP: writel(CLOCK_PROVIDER_MODE, kmb_i2s->pss_base + I2S_GEN_CFG_0); ret = clk_prepare_enable(kmb_i2s->clk_i2s); diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 55f310e91b55..9d72ebd812af 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1380,7 +1380,10 @@ static int skl_platform_soc_probe(struct snd_soc_component *component) const struct skl_dsp_ops *ops; int ret; - pm_runtime_get_sync(component->dev); + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0 && ret != -EACCES) + return ret; + if (bus->ppcap) { skl->component = component; diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 9bdf020a2b64..e06eac592da1 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2950,9 +2950,6 @@ static int skl_tplg_get_pvt_data(struct snd_soc_tplg_dapm_widget *tplg_w, block_size = ret; off += array->size; - array = (struct snd_soc_tplg_vendor_array *) - (tplg_w->priv.data + off); - data = (tplg_w->priv.data + off); if (block_type == SKL_TYPE_TUPLE) { @@ -3599,9 +3596,6 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, block_size = ret; off += array->size; - array = (struct snd_soc_tplg_vendor_array *) - (manifest->priv.data + off); - data = (manifest->priv.data + off); if (block_type == SKL_TYPE_TUPLE) { |