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[PATCH v2] ASoC: Intel: add sof-nau8825 machine driver
by Mac Chiang 08 Nov '21

08 Nov '21

From: David Lin <CTLIN0(a)nuvoton.com> The machine driver is a generic machine driver for SOF with nau8825 codec w or w/o speaker additionally. Depending on the SOC HDMI, DMIC, Bluetooth offload support are added dynamically. Only add information related to SOF since the machine driver was only tested with SOF. There are currently 4 i2s machine variants of ADL. This supports the headphone NUA8825(SSP0) alone or with smart or dumb speakers. Board 2,3,4 use SSP2 for Bluetooth offload support except board 1. Board 1 : NAU8825 + RT1019P(SSP2) Board 2 : NAU8825 + MAX98373(SSP1) Board 3 : NAU8825 + MAX98360A(SSP1) Board 4 : NAU8825 Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> Co-developed-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- 1. remove hdac_hdmi relevant codes support 2. remove duplicated max_98360a_dai_link calls. Instead of checking no_amp condition 3. alter module license name to "GPL" 4. due to rt1019 non-i2c mode, so change to SND_SOC_RT1015P in KConfig --- sound/soc/intel/boards/Kconfig | 19 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/sof_nau8825.c | 651 ++++++++++++++++++ .../intel/common/soc-acpi-intel-adl-match.c | 35 + 4 files changed, 707 insertions(+) create mode 100644 sound/soc/intel/boards/sof_nau8825.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5a225a4ce604..31d2f10672fd 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -526,6 +526,25 @@ config SND_SOC_INTEL_SOF_ES8336_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_SOF_NAU8825_MACH + tristate "SOF with nau8825 codec in I2S Mode" + depends on I2C && ACPI && GPIOLIB + depends on ((SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC) &&\ + (MFD_INTEL_LPSS || COMPILE_TEST)) + select SND_SOC_NAU8825 + select SND_SOC_RT1015P + select SND_SOC_MAX98373_I2C + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_SOC_INTEL_HDA_DSP_COMMON + select SND_SOC_INTEL_SOF_MAXIM_COMMON + help + This adds support for ASoC machine driver for SOF platforms + with nau8825 codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL if (SND_SOC_SOF_COMETLAKE && SND_SOC_SOF_HDA_LINK) diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 9ee8ed864f5d..3ea273d27168 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -22,6 +22,7 @@ snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o snd-soc-sof_rt5682-objs := sof_rt5682.o sof_realtek_common.o snd-soc-sof_cs42l42-objs := sof_cs42l42.o snd-soc-sof_es8336-objs := sof_es8336.o +snd-soc-sof_nau8825-objs := sof_nau8825.o sof_realtek_common.o snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o @@ -44,6 +45,7 @@ snd-soc-sof-sdw-objs += sof_sdw.o \ obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_SOF_CS42L42_MACH) += snd-soc-sof_cs42l42.o obj-$(CONFIG_SND_SOC_INTEL_SOF_ES8336_MACH) += snd-soc-sof_es8336.o +obj-$(CONFIG_SND_SOC_INTEL_SOF_NAU8825_MACH) += snd-soc-sof_nau8825.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c new file mode 100644 index 000000000000..33de043b66c6 --- /dev/null +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -0,0 +1,651 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright(c) 2021 Intel Corporation. +// Copyright(c) 2021 Nuvoton Corporation. + +/* + * Intel SOF Machine Driver with Nuvoton headphone codec NAU8825 + * and speaker codec RT1019P MAX98360a or MAX98373 + */ +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/dmi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/sof.h> +#include <sound/soc-acpi.h> +#include "../../codecs/nau8825.h" +#include "../common/soc-intel-quirks.h" +#include "hda_dsp_common.h" +#include "sof_realtek_common.h" +#include "sof_maxim_common.h" + +#define NAME_SIZE 32 + +#define SOF_NAU8825_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) +#define SOF_NAU8825_SSP_CODEC_MASK (GENMASK(2, 0)) +#define SOF_SPEAKER_AMP_PRESENT BIT(3) +#define SOF_NAU8825_SSP_AMP_SHIFT 4 +#define SOF_NAU8825_SSP_AMP_MASK (GENMASK(6, 4)) +#define SOF_NAU8825_SSP_AMP(quirk) \ + (((quirk) << SOF_NAU8825_SSP_AMP_SHIFT) & SOF_NAU8825_SSP_AMP_MASK) +#define SOF_NAU8825_NUM_HDMIDEV_SHIFT 7 +#define SOF_NAU8825_NUM_HDMIDEV_MASK (GENMASK(9, 7)) +#define SOF_NAU8825_NUM_HDMIDEV(quirk) \ + (((quirk) << SOF_NAU8825_NUM_HDMIDEV_SHIFT) & SOF_NAU8825_NUM_HDMIDEV_MASK) + +/* BT audio offload: reserve 3 bits for future */ +#define SOF_BT_OFFLOAD_SSP_SHIFT 10 +#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(12, 10)) +#define SOF_BT_OFFLOAD_SSP(quirk) \ + (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) +#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(13) +#define SOF_RT1019P_SPEAKER_AMP_PRESENT BIT(14) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(15) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(16) + +static unsigned long sof_nau8825_quirk = SOF_NAU8825_SSP_CODEC(0); + +struct sof_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct sof_card_private { + struct clk *mclk; + struct snd_soc_jack sof_headset; + struct list_head hdmi_pcm_list; +}; + +static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); + struct sof_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + /* dai_link id is 1:1 mapped to the PCM device */ + pcm->device = rtd->dai_link->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int sof_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + + struct snd_soc_jack *jack; + int ret; + + /* + * 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, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->sof_headset; + + 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(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +}; + +static void sof_nau8825_codec_exit(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + + snd_soc_component_set_jack(component, NULL, NULL); +} + +static int sof_nau8825_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 clk_freq, ret; + + clk_freq = sof_dai_get_bclk(rtd); /* BCLK freq */ + + if (clk_freq <= 0) { + dev_err(rtd->dev, "get bclk freq failed: %d\n", clk_freq); + return -EINVAL; + } + + /* Configure clock for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_BLK, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set BCLK clock %d\n", ret); + return ret; + } + + /* Configure pll for codec */ + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, clk_freq, + params_rate(params) * 256); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set BCLK: %d\n", ret); + return ret; + } + + return ret; +} + +static struct snd_soc_ops sof_nau8825_ops = { + .hw_params = sof_nau8825_hw_params, +}; + +static struct snd_soc_dai_link_component platform_component[] = { + { + /* name might be overridden during probe */ + .name = "0000:00:1f.3" + } +}; + +static int sof_card_late_probe(struct snd_soc_card *card) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_dapm_context *dapm = &card->dapm; + 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"); + snd_soc_dapm_disable_pin(dapm, "Right Spk"); + err = snd_soc_dapm_sync(dapm); + if (err < 0) + return err; + } + + return hda_dsp_hdmi_build_controls(card, pcm->codec_dai->component); +} + +static const struct snd_kcontrol_new sof_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_kcontrol_new speaker_controls[] = { + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget sof_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + +static const struct snd_soc_dapm_widget speaker_widgets[] = { + SND_SOC_DAPM_SPK("Spk", NULL), +}; + +static const struct snd_soc_dapm_widget dmic_widgets[] = { + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route sof_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* other jacks */ + { "MIC", NULL, "Headset Mic" }, +}; + +static const struct snd_soc_dapm_route speaker_map[] = { + /* speaker */ + { "Spk", NULL, "Speaker" }, +}; + +static const struct snd_soc_dapm_route dmic_map[] = { + /* digital mics */ + {"DMic", NULL, "SoC DMIC"}, +}; + +static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, speaker_widgets, + ARRAY_SIZE(speaker_widgets)); + if (ret) { + dev_err(rtd->dev, "unable to add dapm controls, ret %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_add_card_controls(card, speaker_controls, + ARRAY_SIZE(speaker_controls)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, speaker_map, + ARRAY_SIZE(speaker_map)); + + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +static int dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets, + ARRAY_SIZE(dmic_widgets)); + if (ret) { + dev_err(card->dev, "DMic widget addition failed: %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map, + ARRAY_SIZE(dmic_map)); + + if (ret) + dev_err(card->dev, "DMic map addition failed: %d\n", ret); + + return ret; +} + +/* sof audio machine driver for nau8825 codec */ +static struct snd_soc_card sof_audio_card_nau8825 = { + .name = "nau8825", /* the sof- prefix is added by the core */ + .owner = THIS_MODULE, + .controls = sof_controls, + .num_controls = ARRAY_SIZE(sof_controls), + .dapm_widgets = sof_widgets, + .num_dapm_widgets = ARRAY_SIZE(sof_widgets), + .dapm_routes = sof_map, + .num_dapm_routes = ARRAY_SIZE(sof_map), + .fully_routed = true, + .late_probe = sof_card_late_probe, +}; + +static struct snd_soc_dai_link_component nau8825_component[] = { + { + .name = "i2c-10508825:00", + .dai_name = "nau8825-hifi", + } +}; + +static struct snd_soc_dai_link_component dmic_component[] = { + { + .name = "dmic-codec", + .dai_name = "dmic-hifi", + } +}; + +static struct snd_soc_dai_link_component rt1019p_component[] = { + { + .name = "RTL1019:00", + .dai_name = "HiFi", + } +}; + +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + +static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, + int ssp_codec, + int ssp_amp, + int dmic_be_num, + int hdmi_num) +{ + struct snd_soc_dai_link_component *idisp_components; + struct snd_soc_dai_link_component *cpus; + struct snd_soc_dai_link *links; + int i, id = 0; + + links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) * + sof_audio_card_nau8825.num_links, GFP_KERNEL); + cpus = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component) * + sof_audio_card_nau8825.num_links, GFP_KERNEL); + if (!links || !cpus) + goto devm_err; + + /* codec SSP */ + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", ssp_codec); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].codecs = nau8825_component; + links[id].num_codecs = ARRAY_SIZE(nau8825_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_nau8825_codec_init; + links[id].exit = sof_nau8825_codec_exit; + links[id].ops = &sof_nau8825_ops; + 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_codec); + if (!links[id].cpus->dai_name) + goto devm_err; + + id++; + + /* dmic */ + if (dmic_be_num > 0) { + /* at least we have dmic01 */ + links[id].name = "dmic01"; + links[id].cpus = &cpus[id]; + links[id].cpus->dai_name = "DMIC01 Pin"; + links[id].init = dmic_init; + if (dmic_be_num > 1) { + /* set up 2 BE links at most */ + links[id + 1].name = "dmic16k"; + links[id + 1].cpus = &cpus[id + 1]; + links[id + 1].cpus->dai_name = "DMIC16k Pin"; + dmic_be_num = 2; + } + } + + for (i = 0; i < dmic_be_num; i++) { + links[id].id = id; + links[id].num_cpus = 1; + links[id].codecs = dmic_component; + links[id].num_codecs = ARRAY_SIZE(dmic_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].ignore_suspend = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + id++; + } + + /* HDMI */ + if (hdmi_num > 0) { + idisp_components = devm_kzalloc(dev, + sizeof(struct snd_soc_dai_link_component) * + hdmi_num, GFP_KERNEL); + if (!idisp_components) + goto devm_err; + } + for (i = 1; i <= hdmi_num; i++) { + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d", i); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d Pin", i); + if (!links[id].cpus->dai_name) + goto devm_err; + + idisp_components[i - 1].name = "ehdaudio0D2"; + idisp_components[i - 1].dai_name = devm_kasprintf(dev, + GFP_KERNEL, + "intel-hdmi-hifi%d", + i); + if (!idisp_components[i - 1].dai_name) + goto devm_err; + + links[id].codecs = &idisp_components[i - 1]; + links[id].num_codecs = 1; + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_hdmi_init; + links[id].dpcm_playback = 1; + links[id].no_pcm = 1; + id++; + } + + /* speaker amp */ + if (sof_nau8825_quirk & SOF_SPEAKER_AMP_PRESENT) { + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", ssp_amp); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + if (sof_nau8825_quirk & SOF_RT1019P_SPEAKER_AMP_PRESENT) { + links[id].codecs = rt1019p_component; + links[id].num_codecs = ARRAY_SIZE(rt1019p_component); + links[id].init = speaker_codec_init; + } else if (sof_nau8825_quirk & + SOF_MAX98373_SPEAKER_AMP_PRESENT) { + links[id].codecs = max_98373_components; + links[id].num_codecs = ARRAY_SIZE(max_98373_components); + links[id].init = max_98373_spk_codec_init; + links[id].ops = &max_98373_ops; + /* feedback stream */ + links[id].dpcm_capture = 1; + } else if (sof_nau8825_quirk & + SOF_MAX98360A_SPEAKER_AMP_PRESENT) { + max_98360a_dai_link(&links[id]); + } else { + goto devm_err; + } + + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].dpcm_playback = 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_amp); + if (!links[id].cpus->dai_name) + goto devm_err; + id++; + } + + /* BT audio offload */ + if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { + int port = (sof_nau8825_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> + SOF_BT_OFFLOAD_SSP_SHIFT; + + links[id].id = id; + 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) + goto devm_err; + links[id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port); + if (!links[id].name) + goto devm_err; + 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].num_cpus = 1; + } + + return links; +devm_err: + return NULL; +} + +static int sof_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link *dai_links; + struct snd_soc_acpi_mach *mach; + struct sof_card_private *ctx; + int dmic_be_num, hdmi_num; + int ret, ssp_amp, ssp_codec; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + if (pdev->id_entry && pdev->id_entry->driver_data) + sof_nau8825_quirk = (unsigned long)pdev->id_entry->driver_data; + + mach = pdev->dev.platform_data; + + /* A speaker amp might not be present when the quirk claims one is. + * Detect this via whether the machine driver match includes quirk_data. + */ + if ((sof_nau8825_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data) + sof_nau8825_quirk &= ~SOF_SPEAKER_AMP_PRESENT; + + dev_dbg(&pdev->dev, "sof_nau8825_quirk = %lx\n", sof_nau8825_quirk); + + /* default number of DMIC DAI's */ + dmic_be_num = 2; + hdmi_num = (sof_nau8825_quirk & SOF_NAU8825_NUM_HDMIDEV_MASK) >> + SOF_NAU8825_NUM_HDMIDEV_SHIFT; + /* default number of HDMI DAI's */ + if (!hdmi_num) + hdmi_num = 3; + + ssp_amp = (sof_nau8825_quirk & SOF_NAU8825_SSP_AMP_MASK) >> + SOF_NAU8825_SSP_AMP_SHIFT; + + ssp_codec = sof_nau8825_quirk & SOF_NAU8825_SSP_CODEC_MASK; + + /* compute number of dai links */ + sof_audio_card_nau8825.num_links = 1 + dmic_be_num + hdmi_num; + + if (sof_nau8825_quirk & SOF_SPEAKER_AMP_PRESENT) + sof_audio_card_nau8825.num_links++; + + if (sof_nau8825_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) + max_98373_set_codec_conf(&sof_audio_card_nau8825); + + if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) + sof_audio_card_nau8825.num_links++; + + dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, + dmic_be_num, hdmi_num); + if (!dai_links) + return -ENOMEM; + + sof_audio_card_nau8825.dai_link = dai_links; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + sof_audio_card_nau8825.dev = &pdev->dev; + + /* set platform name for each dailink */ + ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_nau8825, + mach->mach_params.platform); + if (ret) + return ret; + + snd_soc_card_set_drvdata(&sof_audio_card_nau8825, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &sof_audio_card_nau8825); +} + +static const struct platform_device_id board_ids[] = { + { + .name = "sof_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, + { + .name = "adl_rt1019p_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1019P_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(2) | + SOF_NAU8825_NUM_HDMIDEV(4)), + }, + { + .name = "adl_max98373_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98373_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(1) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, + { + /* The limitation of length of char array, shorten the name */ + .name = "adl_mx98360a_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(1) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, + { } +}; +MODULE_DEVICE_TABLE(platform, board_ids); + +static struct platform_driver sof_audio = { + .probe = sof_audio_probe, + .driver = { + .name = "sof_nau8825", + .pm = &snd_soc_pm_ops, + }, + .id_table = board_ids, +}; +module_platform_driver(sof_audio) + +/* Module information */ +MODULE_DESCRIPTION("SOF Audio Machine driver for NAU8825"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); +MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); 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 f5b21a95d222..98dc41b682cf 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -285,6 +285,11 @@ static const struct snd_soc_acpi_codecs adl_max98360a_amp = { .codecs = {"MX98360A"} }; +static const struct snd_soc_acpi_codecs adl_rt1019p_amp = { + .num_codecs = 1, + .codecs = {"RTL1019"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -310,6 +315,36 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", }, + { + .id = "10508825", + .drv_name = "adl_rt1019p_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_rt1019p_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-rt1019-nau8825.tplg", + }, + { + .id = "10508825", + .drv_name = "adl_max98373_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98373_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98373-nau8825.tplg", + }, + { + .id = "10508825", + .drv_name = "adl_mx98360a_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-mx98360a-nau8825.tplg", + }, + { + .id = "10508825", + .drv_name = "sof_nau8825", + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-nau8825.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.20.1

4 3

[PATCH] ALSA: firewire-motu: add support for MOTU Track 16
by Takashi Sakamoto 08 Nov '21

08 Nov '21

Mark of the Unicorn designed Track 16 2011 as one of models in third generation of its FireWire series. The model is already discontinued. It consists of below ICs: * Texas Instruments TSB41AB1 * Microchip (SMSC) USB3300 * Xilinx Spartan-3A FPGA, XC3S700A * Texas Instruments TMS320C6722 * Microchip (Atmel) AT91SAM SAM7S512 It supports sampling transfer frequency up to 192.0 kHz. The packet format differs depending on both of current sampling transfer frequency and the type of signal in optical interfaces. The model supports transmission of PCM frames as well as MIDI messages. The model supports command mechanism to configure internal DSP. Hardware meter information is available in the first 2 chunks of each data block of tx packet. This commit adds support for it. $ cd linux-firewire-tools/src $ python crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 04107d95 bus_info_length 4, crc_length 16, crc 32149 404 31333934 bus_name "1394" 408 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 40c 0001f200 company_id 0001f2 | 410 000a83c4 device_id 00000a83c4 | EUI-64 0001f200000a83c4 root directory ----------------------------------------------------------------- 414 0004ef04 directory_length 4, crc 61188 418 030001f2 vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 d1000002 --> unit directory at 428 424 8d000005 --> eui-64 leaf at 438 unit directory at 428 ----------------------------------------------------------------- 428 00035b04 directory_length 3, crc 23300 42c 120001f2 specifier id 430 13000039 version 434 17102800 model eui-64 leaf at 438 ----------------------------------------------------------------- 438 0002b25f leaf_length 2, crc 45663 43c 0001f200 company_id 0001f2 | 440 000a83c4 device_id 00000a83c4 | EUI-64 0001f200000a83c4 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 3 +++ sound/firewire/motu/motu-protocol-v3.c | 17 ++++++++++++++--- sound/firewire/motu/motu.c | 1 + sound/firewire/motu/motu.h | 1 + 4 files changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index fd109bea4c53..22b6c779682a 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -169,6 +169,7 @@ config SND_FIREWIRE_MOTU * 828 * 896 * 828mk2 + * 896hd * Traveler * Ultralite * 8pre @@ -176,7 +177,9 @@ config SND_FIREWIRE_MOTU * 828mk3 (Hybrid) * Ultralite mk3 (FireWire only) * Ultralite mk3 (Hybrid) + * Traveler mk3 * Audio Express + * Track 16 * 4pre To compile this driver as a module, choose M here: the module diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index 1371f38505d0..8a0426920a76 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -189,7 +189,7 @@ int snd_motu_protocol_v3_switch_fetching_mode(struct snd_motu *motu, sizeof(reg)); } -static int detect_packet_formats_dual_opt_iface(struct snd_motu *motu, u32 data) +static int detect_packet_formats_with_opt_ifaces(struct snd_motu *motu, u32 data) { if (data & V3_ENABLE_OPT_IN_IFACE_A) { if (data & V3_NO_ADAT_OPT_IN_IFACE_A) { @@ -261,8 +261,9 @@ int snd_motu_protocol_v3_cache_packet_formats(struct snd_motu *motu) if (motu->spec == &snd_motu_spec_828mk3_fw || motu->spec == &snd_motu_spec_828mk3_hybrid || - motu->spec == &snd_motu_spec_traveler_mk3) - return detect_packet_formats_dual_opt_iface(motu, data); + motu->spec == &snd_motu_spec_traveler_mk3 || + motu->spec == &snd_motu_spec_track16) + return detect_packet_formats_with_opt_ifaces(motu, data); else return 0; } @@ -317,6 +318,16 @@ const struct snd_motu_spec snd_motu_spec_audio_express = { .rx_fixed_pcm_chunks = {10, 10, 0}, }; +const struct snd_motu_spec snd_motu_spec_track16 = { + .name = "Track16", + .protocol_version = SND_MOTU_PROTOCOL_V3, + .flags = SND_MOTU_SPEC_RX_MIDI_3RD_Q | + SND_MOTU_SPEC_TX_MIDI_3RD_Q | + SND_MOTU_SPEC_COMMAND_DSP, + .tx_fixed_pcm_chunks = {14, 14, 14}, + .rx_fixed_pcm_chunks = {6, 6, 6}, +}; + const struct snd_motu_spec snd_motu_spec_4pre = { .name = "4pre", .protocol_version = SND_MOTU_PROTOCOL_V3, diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 080728dd278f..f8b7fe38751c 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -173,6 +173,7 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x000030, &snd_motu_spec_ultralite_mk3), // Hybrid. SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3_hybrid), // Hybrid. SND_MOTU_DEV_ENTRY(0x000033, &snd_motu_spec_audio_express), + SND_MOTU_DEV_ENTRY(0x000039, &snd_motu_spec_track16), SND_MOTU_DEV_ENTRY(0x000045, &snd_motu_spec_4pre), { } }; diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 6ad1feab478b..4189f2192284 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -141,6 +141,7 @@ extern const struct snd_motu_spec snd_motu_spec_828mk3_hybrid; extern const struct snd_motu_spec snd_motu_spec_traveler_mk3; extern const struct snd_motu_spec snd_motu_spec_ultralite_mk3; extern const struct snd_motu_spec snd_motu_spec_audio_express; +extern const struct snd_motu_spec snd_motu_spec_track16; extern const struct snd_motu_spec snd_motu_spec_4pre; int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit, -- 2.32.0

2 1

[PATCH v0 23/42] ASoC: soc-jack: Check notifier registration return value
by Borislav Petkov 08 Nov '21

08 Nov '21

From: Borislav Petkov <bp(a)suse.de> Avoid homegrown notifier registration checks. No functional changes. Signed-off-by: Borislav Petkov <bp(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/soc/soc-jack.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index d798765d168c..0875f5022066 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -181,7 +181,8 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins); void snd_soc_jack_notifier_register(struct snd_soc_jack *jack, struct notifier_block *nb) { - blocking_notifier_chain_register(&jack->notifier, nb); + if (blocking_notifier_chain_register(&jack->notifier, nb)) + pr_warn("Jack status notifier already registered\n"); } EXPORT_SYMBOL_GPL(snd_soc_jack_notifier_register); -- 2.29.2

1 0

Support TC Helicon GoXLR
by Anthony Bourguignon 08 Nov '21

08 Nov '21

Hi everyone, I’m the owner of a GoXLR : https://www.tc-helicon.com/product.html?modelCode=P0CQK . It is a usb sound mixer that is not supported officially with Linux. The main issue is that the GoXLR is a multi inputs / multi outputs device, but it is not seen as one. It works great under Windows thanks to a driver. With Linux, it is seen as a single usb sound device with a single multichannel output. Right now, I managed with other people to separate those channels with pulseaudio or jack or even pipewire. For exemple, Front Left and Front Right are the stereo output named "System". Rear left and rear right are another stereo output name Game. There are other outputs, and it is the same mechanism for the inputs. But I’ve seen that a few usb devices do have quirks for them to behave as expected, without the need of configuration with pulse or pipewire. But from what I understand, it’s only to disable subdevices or reorder outputs. Am I wrong ? Is it possible to add a quirk for a device like the GoXLR and change the multichannel output into multiple subdevices with alsa ? Thanks a lot if you can help me and the others owners. Anthony

1 0

ucm2: Add support for RT5650 on MediaTek SoCs
by GitHub pull_request - opened 08 Nov '21

08 Nov '21

alsa-project/alsa-ucm-conf pull request #117 was opened from kholk: At least one MT8173 machine uses a RT5650 codec and that's Google Elm: add a UCM2 configuration for this codec. This was tested on Acer Chromebook R13 (MT8173 Elm). Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/117 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/117.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: PCM: Fix NULL dereference at mmap checks
by Takashi Iwai 07 Nov '21

07 Nov '21

The recent refactoring of mmap handling caused Oops on some devices that don't use the standard memory allocations. This patch addresses it by allowing snd_dma_buffer_mmap() helper to receive the NULL pointer dmab argument (and return an error appropriately). Fixes: a202bd1ad86d ("ALSA: core: Move mmap handler into memalloc ops") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 0145dabda56d..de7e9d2df39d 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -183,8 +183,11 @@ EXPORT_SYMBOL_GPL(snd_devm_alloc_dir_pages); int snd_dma_buffer_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { - const struct snd_malloc_ops *ops = snd_dma_get_ops(dmab); + const struct snd_malloc_ops *ops; + if (!dmab) + return -ENOENT; + ops = snd_dma_get_ops(dmab); if (ops && ops->mmap) return ops->mmap(dmab, area); else -- 2.31.1

1 0

[Bug][5.16-rc0] Between commits dcd68326d29b and 7ddb58cb0eca, was broken play sound via the DisplayPort and HDMI.
by Mikhail Gavrilov 07 Nov '21

07 Nov '21

Hi! Between commits dcd68326d29b and 7ddb58cb0eca, was broken play sound via the DisplayPort and HDMI. When selected DisplayPort or HDMI in a gnome control center instead of sound in speakers in the kernel log pops errors: [ 69.228324] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00000 flags=0x0000] [ 69.228672] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00180 flags=0x0000] [ 69.229321] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00280 flags=0x0000] [ 69.229988] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00380 flags=0x0000] [ 69.230653] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00480 flags=0x0000] [ 69.231320] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00580 flags=0x0000] [ 69.231986] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00680 flags=0x0000] [ 69.232651] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00780 flags=0x0000] [ 69.233319] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00880 flags=0x0000] [ 69.233986] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00980 flags=0x0000] [ 77.693031] amd_iommu_report_page_fault: 5527 callbacks suppressed I tried to bisect the bad commit but didn't find the exact commit. $ git bisect log git bisect start # good: [dcd68326d29b62f3039e4f4d23d3e38f24d37360] Merge tag 'devicetree-for-5.16' of git://git.kernel.org/pub/scm/linux/kernel/git/robh/linux git bisect good dcd68326d29b62f3039e4f4d23d3e38f24d37360 # bad: [7ddb58cb0ecae8e8b6181d736a87667cc9ab8389] Merge tag 'clk-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/clk/linux git bisect bad 7ddb58cb0ecae8e8b6181d736a87667cc9ab8389 # bad: [2219b0ceefe835b92a8a74a73fe964aa052742a2] Merge tag 'soc-5.16' of git://git.kernel.org/pub/scm/linux/kernel/git/soc/soc git bisect bad 2219b0ceefe835b92a8a74a73fe964aa052742a2 # good: [2003c44e28ac9759200a78dda20c5f695949e3f4] ASoC: cs42l42: Prevent NULL pointer deref in interrupt handler git bisect good 2003c44e28ac9759200a78dda20c5f695949e3f4 # bad: [ff0700f03609b9f0defacd4ce96d9519d721e0a2] Merge tag 'sound-5.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound git bisect bad ff0700f03609b9f0defacd4ce96d9519d721e0a2 # good: [956ac4f1f53c58f87f38fd79d4c87e9146245c09] Merge series "ASoC: Sanity checks and soc-topology updates" from Cezary Rojewski <cezary.rojewski(a)intel.com>: git bisect good 956ac4f1f53c58f87f38fd79d4c87e9146245c09 # skip: [634ec0b2906efd46f6f57977e172aa3470aca432] ALSA: firewire-motu: notify event for parameter change in register DSP model git bisect skip 634ec0b2906efd46f6f57977e172aa3470aca432 # bad: [407359d44ed33974137b9158da356d53f999dcf2] ALSA: firewire-motu: export meter information to userspace as float value git bisect bad 407359d44ed33974137b9158da356d53f999dcf2 # good: [c18c4966033e6473a472fb65fbd5a6441603fbf7] ALSA: pcm: Unify snd_pcm_delay() and snd_pcm_hwsync() git bisect good c18c4966033e6473a472fb65fbd5a6441603fbf7 # skip: [73325f60e2ed28f04032d43c2828b73776cfefd0] ALSA: memalloc: Support for non-coherent page allocation git bisect skip 73325f60e2ed28f04032d43c2828b73776cfefd0 # skip: [4c9eda8f37f9523f1d2ccbb442ce641e8c981c9f] ALSA: firewire-motu: queue event for parameter change in register DSP model git bisect skip 4c9eda8f37f9523f1d2ccbb442ce641e8c981c9f # skip: [58b62ab7025912ce1be36e3ba19d49620a0161b6] ALSA: firewire-motu: add ioctl command to read cached hardware meter git bisect skip 58b62ab7025912ce1be36e3ba19d49620a0161b6 # bad: [5aec579e08e4f2be7103ae264ac8f34883eb9273] ALSA: uapi: Fix a C++ style comment in asound.h git bisect bad 5aec579e08e4f2be7103ae264ac8f34883eb9273 # skip: [7d843c494a9b69d07bc0588124599e3f665a1496] ALSA: firewire-motu: parse messages for input parameters in register DSP model git bisect skip 7d843c494a9b69d07bc0588124599e3f665a1496 # skip: [ca15a09ccc5bd2731c5fcb667e6ea3bbbf8f5772] ALSA: firewire-motu: add ioctl command to read cached parameters in register DSP model git bisect skip ca15a09ccc5bd2731c5fcb667e6ea3bbbf8f5772 # skip: [dc36a9755a572781903d79f8437d109b72662da5] ALSA: firewire-motu: parse messages for mixer source parameters in register-DSP model git bisect skip dc36a9755a572781903d79f8437d109b72662da5 # skip: [41cc23389f5fc64bdac78b73935a44bd5abc990d] ALSA: firewire-motu: parse messages for line input parameters in register DSP model git bisect skip 41cc23389f5fc64bdac78b73935a44bd5abc990d # skip: [a25684a956468ee8bbbee44649e41e5d447e5adc] ALSA: memalloc: Support for non-contiguous page allocation git bisect skip a25684a956468ee8bbbee44649e41e5d447e5adc # skip: [6ca81d2b6305a884da441fd0281ff01afd5f8c7e] ALSA: firewire-motu: parse messages for output parameters in register DSP model git bisect skip 6ca81d2b6305a884da441fd0281ff01afd5f8c7e # skip: [90b28f3bb85c39b11daf29d473ef21a935c70ec5] ALSA: firewire-motu: add message parser for meter information in command DSP model git bisect skip 90b28f3bb85c39b11daf29d473ef21a935c70ec5 # skip: [3c05f1477e62ea5a0a8797ba6a545b1dc751fb31] ALSA: ISA: not for M68K git bisect skip 3c05f1477e62ea5a0a8797ba6a545b1dc751fb31 # skip: [2d9ea39917a4e4293bc2caea902c7059a330b611] ALSA: memalloc: Convert x86 SG-buffer handling with non-contiguous type git bisect skip 2d9ea39917a4e4293bc2caea902c7059a330b611 # skip: [ce69bed5557b05dd1918556d4e90c293382155ae] ALSA: firewire-motu: parse messages for mixer output parameters in register DSP model git bisect skip ce69bed5557b05dd1918556d4e90c293382155ae # bad: [b15706471abe916a16a38bee4434612998d869d2] ALSA: firewire: Fix C++ style comments in uapi header git bisect bad b15706471abe916a16a38bee4434612998d869d2 # skip: [bea36afa102e37d5e4d9ea519f14d1c92d512e45] ALSA: firewire-motu: add message parser to gather meter information in register DSP model git bisect skip bea36afa102e37d5e4d9ea519f14d1c92d512e45 # only skipped commits left to test # possible first bad commit: [b15706471abe916a16a38bee4434612998d869d2] ALSA: firewire: Fix C++ style comments in uapi header # possible first bad commit: [2d9ea39917a4e4293bc2caea902c7059a330b611] ALSA: memalloc: Convert x86 SG-buffer handling with non-contiguous type # possible first bad commit: [73325f60e2ed28f04032d43c2828b73776cfefd0] ALSA: memalloc: Support for non-coherent page allocation # possible first bad commit: [a25684a956468ee8bbbee44649e41e5d447e5adc] ALSA: memalloc: Support for non-contiguous page allocation # possible first bad commit: [3c05f1477e62ea5a0a8797ba6a545b1dc751fb31] ALSA: ISA: not for M68K # possible first bad commit: [634ec0b2906efd46f6f57977e172aa3470aca432] ALSA: firewire-motu: notify event for parameter change in register DSP model # possible first bad commit: [4c9eda8f37f9523f1d2ccbb442ce641e8c981c9f] ALSA: firewire-motu: queue event for parameter change in register DSP model # possible first bad commit: [ca15a09ccc5bd2731c5fcb667e6ea3bbbf8f5772] ALSA: firewire-motu: add ioctl command to read cached parameters in register DSP model # possible first bad commit: [7d843c494a9b69d07bc0588124599e3f665a1496] ALSA: firewire-motu: parse messages for input parameters in register DSP model # possible first bad commit: [41cc23389f5fc64bdac78b73935a44bd5abc990d] ALSA: firewire-motu: parse messages for line input parameters in register DSP model # possible first bad commit: [6ca81d2b6305a884da441fd0281ff01afd5f8c7e] ALSA: firewire-motu: parse messages for output parameters in register DSP model # possible first bad commit: [ce69bed5557b05dd1918556d4e90c293382155ae] ALSA: firewire-motu: parse messages for mixer output parameters in register DSP model # possible first bad commit: [dc36a9755a572781903d79f8437d109b72662da5] ALSA: firewire-motu: parse messages for mixer source parameters in register-DSP model # possible first bad commit: [58b62ab7025912ce1be36e3ba19d49620a0161b6] ALSA: firewire-motu: add ioctl command to read cached hardware meter # possible first bad commit: [90b28f3bb85c39b11daf29d473ef21a935c70ec5] ALSA: firewire-motu: add message parser for meter information in command DSP model # possible first bad commit: [bea36afa102e37d5e4d9ea519f14d1c92d512e45] ALSA: firewire-motu: add message parser to gather meter information in register DSP model I hope it helps. The full kernel log is attached to this message. -- Best Regards, Mike Gavrilov.

2 1

[PATCH] sound/soc: adds TAS5754M digital input amplifier component driver
by joerg＠hifiberry.com 07 Nov '21

07 Nov '21

From: Joerg Schambacher <joerg(a)hifiberry.com> Adds a minimum component driver to run the amplifier in I2S master mode only from standard audio clocks. Therefore, it only allows 44.1, 88.2, 176.4, 48, 96 and 192ksps with 16, 20, 24 and 32 bits sample size. Digital volume control and the -6dB switch are supported. Signed-off-by: Joerg Schambacher <joerg(a)hifiberry.com> --- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5754m.c | 547 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas5754m.h | 259 +++++++++++++++++ 4 files changed, 816 insertions(+) create mode 100644 sound/soc/codecs/tas5754m.c create mode 100644 sound/soc/codecs/tas5754m.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82ee233a269d..cf0584948fcf 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -210,6 +210,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 + imply SND_SOC_TAS5754M imply SND_SOC_TAS6424 imply SND_SOC_TDA7419 imply SND_SOC_TFA9879 @@ -1419,6 +1420,13 @@ config SND_SOC_TAS5720 Enable support for Texas Instruments TAS5720L/M high-efficiency mono Class-D audio power amplifiers. +config SND_SOC_TAS5754M + tristate "Texas Instruments TAS5754M Digital Input Audio amplifier" + depends on I2C + help + Enable support for Texas Instruments TAS5754M digital input + Class-D audio power amplifiers. + config SND_SOC_TAS6424 tristate "Texas Instruments TAS6424 Quad-Channel Audio amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8dcea2c4604a..39984900258a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -227,6 +227,7 @@ snd-soc-sti-sas-objs := sti-sas.o snd-soc-tas5086-objs := tas5086.o snd-soc-tas571x-objs := tas571x.o snd-soc-tas5720-objs := tas5720.o +snd-soc-tas5754m-objs := tas5754m.o snd-soc-tas6424-objs := tas6424.o snd-soc-tda7419-objs := tda7419.o snd-soc-tas2770-objs := tas2770.o @@ -555,6 +556,7 @@ obj-$(CONFIG_SND_SOC_TAS2764) += snd-soc-tas2764.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o +obj-$(CONFIG_SND_SOC_TAS5754M) += snd-soc-tas5754m.o obj-$(CONFIG_SND_SOC_TAS6424) += snd-soc-tas6424.o obj-$(CONFIG_SND_SOC_TDA7419) += snd-soc-tda7419.o obj-$(CONFIG_SND_SOC_TAS2770) += snd-soc-tas2770.o diff --git a/sound/soc/codecs/tas5754m.c b/sound/soc/codecs/tas5754m.c new file mode 100644 index 000000000000..19b16a2c4817 --- /dev/null +++ b/sound/soc/codecs/tas5754m.c @@ -0,0 +1,547 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Driver for the TAS5754M Audio Amplifier in Master mode (only) + * supports only standard audio frequencies 44.1 to 192 ksps + * + * Author: Joerg Schambacher <joerg(a)hifiberry.com> + * with fragments from Andy Liu <andy-liu(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/clk.h> +#include <linux/slab.h> +#include <linux/of.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/delay.h> + +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> + +#include "tas5754m.h" + +#define TAS5754M_RATES (SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_176400) +#define TAS5754M_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + + +static const struct reg_sequence tas5754m_init_sequence[] = { + { TAS5754M_RESET, 0x00 }, + { TAS5754M_MUTE, 0x11 }, + { TAS5754M_POWER, 0x00 }, + { TAS5754M_PLL_EN, 0x00 }, + { TAS5754M_RESET, 0x00 }, + { TAS5754M_GPIO_OUTPUT_3, 0x02 }, + { TAS5754M_GPIO_OUTPUT_4, 0x02 }, + { TAS5754M_GPIO_OUTPUT_6, 0x02 }, + { TAS5754M_GPIO_EN, 0x2c }, + { TAS5754M_GPIO_CONTROL_1, 0x04 }, + { TAS5754M_BCLK_LRCLK_CFG, 0x11 }, + { TAS5754M_MASTER_MODE, 0x7c }, + { TAS5754M_ERROR_DETECT, 0x77 }, + { TAS5754M_PLL_EN, 0x01 }, + { TAS5754M_PLL_REF, 0x00 }, + { TAS5754M_PLL_COEFF_0, 0x03 }, + { TAS5754M_PLL_COEFF_1, 0x0c }, + { TAS5754M_PLL_COEFF_2, 0x00 }, + { TAS5754M_PLL_COEFF_3, 0x00 }, + { TAS5754M_PLL_COEFF_4, 0x00 }, + { TAS5754M_DAC_REF, 0x30 }, + { TAS5754M_DSP_CLKDIV, 0x01 }, + { TAS5754M_DAC_CLKDIV, 0x0f }, + { TAS5754M_NCP_CLKDIV, 0x03 }, + { TAS5754M_OSR_CLKDIV, 0x00 }, + { TAS5754M_FS_SPEED_MODE, 0x00 }, + { TAS5754M_MASTER_CLKDIV_1, 0x0f }, + { TAS5754M_MASTER_CLKDIV_2, 0x1f }, + { TAS5754M_I2S_1, 0x00 }, + { TAS5754M_I2S_2, 0x01 }, + { TAS5754M_PLL_EN, 0x01 }, + { TAS5754M_MASTER_MODE, 0x7f }, + { TAS5754M_MUTE, 0x11 }, +}; + +static const struct reg_default tas5754m_reg_defaults[] = { + { TAS5754M_RESET, 0x00 }, + { TAS5754M_POWER, 0x00 }, + { TAS5754M_MUTE, 0x00 }, + { TAS5754M_DSP, 0x00 }, + { TAS5754M_PLL_REF, 0x00 }, + { TAS5754M_DAC_REF, 0x00 }, + { TAS5754M_DAC_ROUTING, 0x11 }, + { TAS5754M_DSP_PROGRAM, 0x01 }, + { TAS5754M_CLKDET, 0x00 }, + { TAS5754M_AUTO_MUTE, 0x00 }, + { TAS5754M_ERROR_DETECT, 0x00 }, + { TAS5754M_DIGITAL_VOLUME_1, 0x00 }, + { TAS5754M_DIGITAL_VOLUME_2, 0x30 }, + { TAS5754M_DIGITAL_VOLUME_3, 0x30 }, + { TAS5754M_DIGITAL_MUTE_1, 0x22 }, + { TAS5754M_DIGITAL_MUTE_2, 0x00 }, + { TAS5754M_DIGITAL_MUTE_3, 0x07 }, + { TAS5754M_OUTPUT_AMPLITUDE, 0x00 }, + { TAS5754M_ANALOG_GAIN_CTRL, 0x00 }, + { TAS5754M_UNDERVOLTAGE_PROT, 0x00 }, + { TAS5754M_ANALOG_MUTE_CTRL, 0x00 }, + { TAS5754M_ANALOG_GAIN_BOOST, 0x00 }, + { TAS5754M_VCOM_CTRL_1, 0x00 }, + { TAS5754M_VCOM_CTRL_2, 0x01 }, + { TAS5754M_BCLK_LRCLK_CFG, 0x00 }, + { TAS5754M_MASTER_MODE, 0x7c }, + { TAS5754M_GPIO_DACIN, 0x00 }, + { TAS5754M_GPIO_PLLIN, 0x00 }, + { TAS5754M_SYNCHRONIZE, 0x10 }, + { TAS5754M_PLL_COEFF_0, 0x00 }, + { TAS5754M_PLL_COEFF_1, 0x00 }, + { TAS5754M_PLL_COEFF_2, 0x00 }, + { TAS5754M_PLL_COEFF_3, 0x00 }, + { TAS5754M_PLL_COEFF_4, 0x00 }, + { TAS5754M_DSP_CLKDIV, 0x00 }, + { TAS5754M_DAC_CLKDIV, 0x00 }, + { TAS5754M_NCP_CLKDIV, 0x00 }, + { TAS5754M_OSR_CLKDIV, 0x00 }, + { TAS5754M_MASTER_CLKDIV_1, 0x00 }, + { TAS5754M_MASTER_CLKDIV_2, 0x00 }, + { TAS5754M_FS_SPEED_MODE, 0x00 }, + { TAS5754M_IDAC_1, 0x01 }, + { TAS5754M_IDAC_2, 0x00 }, +}; + +static bool tas5754m_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS5754M_RESET: + case TAS5754M_POWER: + case TAS5754M_MUTE: + case TAS5754M_PLL_EN: + case TAS5754M_SPI_MISO_FUNCTION: + case TAS5754M_DSP: + case TAS5754M_GPIO_EN: + case TAS5754M_BCLK_LRCLK_CFG: + case TAS5754M_DSP_GPIO_INPUT: + case TAS5754M_MASTER_MODE: + case TAS5754M_PLL_REF: + case TAS5754M_DAC_REF: + case TAS5754M_GPIO_DACIN: + case TAS5754M_GPIO_PLLIN: + case TAS5754M_SYNCHRONIZE: + case TAS5754M_PLL_COEFF_0: + case TAS5754M_PLL_COEFF_1: + case TAS5754M_PLL_COEFF_2: + case TAS5754M_PLL_COEFF_3: + case TAS5754M_PLL_COEFF_4: + case TAS5754M_DSP_CLKDIV: + case TAS5754M_DAC_CLKDIV: + case TAS5754M_NCP_CLKDIV: + case TAS5754M_OSR_CLKDIV: + case TAS5754M_MASTER_CLKDIV_1: + case TAS5754M_MASTER_CLKDIV_2: + case TAS5754M_FS_SPEED_MODE: + case TAS5754M_IDAC_1: + case TAS5754M_IDAC_2: + case TAS5754M_ERROR_DETECT: + case TAS5754M_I2S_1: + case TAS5754M_I2S_2: + case TAS5754M_DAC_ROUTING: + case TAS5754M_DSP_PROGRAM: + case TAS5754M_CLKDET: + case TAS5754M_AUTO_MUTE: + case TAS5754M_DIGITAL_VOLUME_1: + case TAS5754M_DIGITAL_VOLUME_2: + case TAS5754M_DIGITAL_VOLUME_3: + case TAS5754M_DIGITAL_MUTE_1: + case TAS5754M_DIGITAL_MUTE_2: + case TAS5754M_DIGITAL_MUTE_3: + case TAS5754M_GPIO_OUTPUT_1: + case TAS5754M_GPIO_OUTPUT_2: + case TAS5754M_GPIO_OUTPUT_3: + case TAS5754M_GPIO_OUTPUT_4: + case TAS5754M_GPIO_OUTPUT_5: + case TAS5754M_GPIO_OUTPUT_6: + case TAS5754M_GPIO_CONTROL_1: + case TAS5754M_GPIO_CONTROL_2: + case TAS5754M_OVERFLOW: + case TAS5754M_RATE_DET_1: + case TAS5754M_RATE_DET_2: + case TAS5754M_RATE_DET_3: + case TAS5754M_RATE_DET_4: + case TAS5754M_CLOCK_STATUS: + case TAS5754M_ANALOG_MUTE_DET: + case TAS5754M_GPIN: + case TAS5754M_DIGITAL_MUTE_DET: + case TAS5754M_OUTPUT_AMPLITUDE: + case TAS5754M_ANALOG_GAIN_CTRL: + case TAS5754M_UNDERVOLTAGE_PROT: + case TAS5754M_ANALOG_MUTE_CTRL: + case TAS5754M_ANALOG_GAIN_BOOST: + case TAS5754M_VCOM_CTRL_1: + case TAS5754M_VCOM_CTRL_2: + case TAS5754M_CRAM_CTRL: + case TAS5754M_FLEX_A: + case TAS5754M_FLEX_B: + return true; + default: + return reg < 0x7f; + } +} + +static bool tas5754m_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS5754M_PLL_EN: + case TAS5754M_OVERFLOW: + case TAS5754M_RATE_DET_1: + case TAS5754M_RATE_DET_2: + case TAS5754M_RATE_DET_3: + case TAS5754M_RATE_DET_4: + case TAS5754M_CLOCK_STATUS: + case TAS5754M_ANALOG_MUTE_DET: + case TAS5754M_GPIN: + case TAS5754M_DIGITAL_MUTE_DET: + case TAS5754M_CRAM_CTRL: + return true; + default: + return reg < 0x7f; + } +} + +struct tas5754m_priv { + struct regmap *regmap; + struct clk *sclk; +}; + +static const struct regmap_range_cfg tas5754m_range = { + .name = "Pages", + .range_min = TAS5754M_VIRT_BASE, + .range_max = TAS5754M_MAX_REGISTER, + .selector_reg = TAS5754M_PAGE, + .selector_mask = 0x7f, + .window_start = 0, + .window_len = 128, +}; + +const struct regmap_config tas5754m_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .ranges = &tas5754m_range, + .num_ranges = 1, + .max_register = TAS5754M_MAX_REGISTER, + + .reg_defaults = tas5754m_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5754m_reg_defaults), + .readable_reg = tas5754m_readable, + .volatile_reg = tas5754m_volatile, + + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(tas5754m_regmap); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1); +static const DECLARE_TLV_DB_SCALE(analog_tlv, -600, 600, 0); + +static const struct snd_kcontrol_new tas5754m_controls[] = { +SOC_DOUBLE_R_TLV("Digital Playback Volume", TAS5754M_DIGITAL_VOLUME_2, + TAS5754M_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), +SOC_DOUBLE_TLV("Analog Playback Volume", TAS5754M_ANALOG_GAIN_CTRL, + TAS5754M_LAGN_SHIFT, TAS5754M_RAGN_SHIFT, 1, 1, analog_tlv), +}; + +static int tas5754m_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct tas5754m_priv *tas5754m = + snd_soc_component_get_drvdata(component); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + ret = regmap_update_bits(tas5754m->regmap, + TAS5754M_POWER, TAS5754M_RQST, 0); + if (ret != 0) { + dev_err(component->dev, + "Failed to remove standby: %d\n", ret); + return ret; + } + break; + + case SND_SOC_BIAS_OFF: + ret = regmap_update_bits(tas5754m->regmap, + TAS5754M_POWER, TAS5754M_RQST, TAS5754M_RQST); + if (ret != 0) { + dev_err(component->dev, + "Failed to request standby: %d\n", ret); + return ret; + } + break; + } + + return 0; +} + +static int tas5754m_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tas5754m_priv *tas5754m = + snd_soc_component_get_drvdata(component); + unsigned long bclk; + unsigned long mclk; + int sample_len; + int bclk_div; + int lrclk_div; + int alen; + int ret; + + switch (params_width(params)) { + case 16: + sample_len = 16; + alen = TAS5754M_ALEN_16; + break; + case 20: + sample_len = 32; + alen = TAS5754M_ALEN_20; + break; + case 24: + sample_len = 32; + alen = TAS5754M_ALEN_24; + break; + case 32: + sample_len = 32; + alen = TAS5754M_ALEN_32; + break; + default: + dev_err(component->dev, "Unsupported sample size: %d\n", + params_width(params)); + return -EINVAL; + } + ret = regmap_update_bits(tas5754m->regmap, TAS5754M_I2S_1, alen, alen); + if (ret != 0) { + dev_err(component->dev, + "Cannot set sample size: %d\n", ret); + return ret; + } + + switch (params_rate(params)) { + case 44100: + case 48000: + ret = regmap_write(tas5754m->regmap, + TAS5754M_FS_SPEED_MODE, TAS5754M_FSSP_48KHZ); + break; + case 88200: + case 96000: + ret = regmap_write(tas5754m->regmap, + TAS5754M_FS_SPEED_MODE, TAS5754M_FSSP_96KHZ); + break; + case 176400: + case 192000: + ret = regmap_write(tas5754m->regmap, + TAS5754M_FS_SPEED_MODE, TAS5754M_FSSP_192KHZ); + break; + default: + dev_err(component->dev, "Sample rate not supported: %d\n", + params_rate(params)); + return -EINVAL; + } + if (ret != 0) { + dev_err(component->dev, "Failed to config PLL\n"); + return ret; + } + + + mclk = clk_get_rate(tas5754m->sclk); + bclk = sample_len * 2 * params_rate(params); + bclk_div = mclk / bclk; + lrclk_div = sample_len * 2; + + // stop LR / SCLK clocks + ret = regmap_write(tas5754m->regmap, TAS5754M_MASTER_MODE, 0x7c); + + // set SCLK divider + ret |= regmap_write(tas5754m->regmap, TAS5754M_MASTER_CLKDIV_1, + bclk_div - 1); + + // set LRCLK divider + ret |= regmap_write(tas5754m->regmap, TAS5754M_MASTER_CLKDIV_2, + lrclk_div - 1); + + // restart LR / SCLK clocks + ret |= regmap_write(tas5754m->regmap, TAS5754M_MASTER_MODE, 0x7f); + if (ret != 0) { + dev_err(component->dev, "Failed to config PLL\n"); + return ret; + } + + return 0; +} + +static const struct snd_soc_component_driver tas5754m_soc_component = { + .set_bias_level = tas5754m_set_bias_level, + .idle_bias_on = true, + .controls = tas5754m_controls, + .num_controls = ARRAY_SIZE(tas5754m_controls), +}; + +static int tas5754m_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_component *component = dai->component; + + if (mute) { + snd_soc_component_write(component, TAS5754M_MUTE, 0x11); + } else { + usleep_range(1000, 2000); + snd_soc_component_write(component, TAS5754M_MUTE, 0x00); + } + return 0; +} + +static const struct snd_soc_dai_ops tas5754m_dai_ops = { + .mute_stream = tas5754m_mute, + .hw_params = tas5754m_hw_params, +}; + +static struct snd_soc_dai_driver tas5754m_dai = { + .name = "tas5754m-amplifier", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TAS5754M_RATES, + .formats = TAS5754M_FORMATS, + }, + .ops = &tas5754m_dai_ops, +}; + +static int tas5754m_probe(struct device *dev, struct regmap *regmap) +{ + struct tas5754m_priv *tas5754m; + int ret; + + tas5754m = devm_kzalloc(dev, sizeof(struct tas5754m_priv), GFP_KERNEL); + if (!tas5754m) + return -ENOMEM; + + dev_set_drvdata(dev, tas5754m); + tas5754m->regmap = regmap; + + ret = regmap_multi_reg_write(regmap, tas5754m_init_sequence, + ARRAY_SIZE(tas5754m_init_sequence)); + + if (ret != 0) { + dev_err(dev, "Failed to initialize TAS5754M: %d\n", ret); + goto err; + } + + tas5754m->sclk = devm_clk_get(dev, NULL); + if (PTR_ERR(tas5754m->sclk) == -EPROBE_DEFER) { + ret = -EPROBE_DEFER; + goto err; + } + if (!IS_ERR(tas5754m->sclk)) { + ret = clk_prepare_enable(tas5754m->sclk); + if (ret != 0) { + dev_err(dev, "Failed to enable SCLK: %d\n", ret); + goto err; + } + } + + ret = snd_soc_register_component(dev, + &tas5754m_soc_component, &tas5754m_dai, 1); + if (ret != 0) { + dev_err(dev, "Failed to register CODEC: %d\n", ret); + goto err; + } + + return 0; + +err: + return ret; + +} + +static int tas5754m_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) +{ + struct regmap *regmap; + struct regmap_config config = tas5754m_regmap; + + /* enable auto-increment mode */ + config.read_flag_mask = 0x80; + config.write_flag_mask = 0x80; + + regmap = devm_regmap_init_i2c(i2c, &config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return tas5754m_probe(&i2c->dev, regmap); +} + +static int tas5754m_remove(struct device *dev) +{ + snd_soc_unregister_component(dev); + + return 0; +} + +static int tas5754m_i2c_remove(struct i2c_client *i2c) +{ + tas5754m_remove(&i2c->dev); + + return 0; +} + +static const struct i2c_device_id tas5754m_i2c_id[] = { + { "tas5754m", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas5754m_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id tas5754m_of_match[] = { + { .compatible = "ti,tas5754m", }, + { .compatible = "ti,tas5756m", }, + { } +}; +MODULE_DEVICE_TABLE(of, tas5754m_of_match); +#endif + +static struct i2c_driver tas5754m_i2c_driver = { + .probe = tas5754m_i2c_probe, + .remove = tas5754m_i2c_remove, + .id_table = tas5754m_i2c_id, + .driver = { + .name = "tas5754m", + .of_match_table = tas5754m_of_match, + }, +}; + +module_i2c_driver(tas5754m_i2c_driver); + +MODULE_AUTHOR("Joerg Schambacher <joerg(a)hifiberry.com>"); +MODULE_DESCRIPTION("TAS5754M Audio Amplifier Driver - Master mode only"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas5754m.h b/sound/soc/codecs/tas5754m.h new file mode 100644 index 000000000000..492b8abede6c --- /dev/null +++ b/sound/soc/codecs/tas5754m.h @@ -0,0 +1,259 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Driver for the TAS5754M DAC+amplifier combo devices + * + * Author: (copied from pcm512x.h) + * Mark Brown <broonie(a)kernel.org> + * Copyright 2014 Linaro Ltd + */ + +#ifndef _SND_SOC_TAS5754M +#define _SND_SOC_TAS5754M + +#include <linux/pm.h> +#include <linux/regmap.h> + +#define TAS5754M_VIRT_BASE 0x000 +#define TAS5754M_PAGE_LEN 0x80 +#define TAS5754M_PAGE_BASE(n) (TAS5754M_VIRT_BASE + (TAS5754M_PAGE_LEN * n)) + +#define TAS5754M_PAGE 0 + +#define TAS5754M_RESET (TAS5754M_PAGE_BASE(0) + 1) +#define TAS5754M_POWER (TAS5754M_PAGE_BASE(0) + 2) +#define TAS5754M_MUTE (TAS5754M_PAGE_BASE(0) + 3) +#define TAS5754M_PLL_EN (TAS5754M_PAGE_BASE(0) + 4) +#define TAS5754M_SPI_MISO_FUNCTION (TAS5754M_PAGE_BASE(0) + 6) +#define TAS5754M_DSP (TAS5754M_PAGE_BASE(0) + 7) +#define TAS5754M_GPIO_EN (TAS5754M_PAGE_BASE(0) + 8) +#define TAS5754M_BCLK_LRCLK_CFG (TAS5754M_PAGE_BASE(0) + 9) +#define TAS5754M_DSP_GPIO_INPUT (TAS5754M_PAGE_BASE(0) + 10) +#define TAS5754M_MASTER_MODE (TAS5754M_PAGE_BASE(0) + 12) +#define TAS5754M_PLL_REF (TAS5754M_PAGE_BASE(0) + 13) +#define TAS5754M_DAC_REF (TAS5754M_PAGE_BASE(0) + 14) +#define TAS5754M_GPIO_DACIN (TAS5754M_PAGE_BASE(0) + 16) +#define TAS5754M_GPIO_PLLIN (TAS5754M_PAGE_BASE(0) + 18) +#define TAS5754M_SYNCHRONIZE (TAS5754M_PAGE_BASE(0) + 19) +#define TAS5754M_PLL_COEFF_0 (TAS5754M_PAGE_BASE(0) + 20) +#define TAS5754M_PLL_COEFF_1 (TAS5754M_PAGE_BASE(0) + 21) +#define TAS5754M_PLL_COEFF_2 (TAS5754M_PAGE_BASE(0) + 22) +#define TAS5754M_PLL_COEFF_3 (TAS5754M_PAGE_BASE(0) + 23) +#define TAS5754M_PLL_COEFF_4 (TAS5754M_PAGE_BASE(0) + 24) +#define TAS5754M_DSP_CLKDIV (TAS5754M_PAGE_BASE(0) + 27) +#define TAS5754M_DAC_CLKDIV (TAS5754M_PAGE_BASE(0) + 28) +#define TAS5754M_NCP_CLKDIV (TAS5754M_PAGE_BASE(0) + 29) +#define TAS5754M_OSR_CLKDIV (TAS5754M_PAGE_BASE(0) + 30) +#define TAS5754M_MASTER_CLKDIV_1 (TAS5754M_PAGE_BASE(0) + 32) +#define TAS5754M_MASTER_CLKDIV_2 (TAS5754M_PAGE_BASE(0) + 33) +#define TAS5754M_FS_SPEED_MODE (TAS5754M_PAGE_BASE(0) + 34) +#define TAS5754M_IDAC_1 (TAS5754M_PAGE_BASE(0) + 35) +#define TAS5754M_IDAC_2 (TAS5754M_PAGE_BASE(0) + 36) +#define TAS5754M_ERROR_DETECT (TAS5754M_PAGE_BASE(0) + 37) +#define TAS5754M_I2S_1 (TAS5754M_PAGE_BASE(0) + 40) +#define TAS5754M_I2S_2 (TAS5754M_PAGE_BASE(0) + 41) +#define TAS5754M_DAC_ROUTING (TAS5754M_PAGE_BASE(0) + 42) +#define TAS5754M_DSP_PROGRAM (TAS5754M_PAGE_BASE(0) + 43) +#define TAS5754M_CLKDET (TAS5754M_PAGE_BASE(0) + 44) +#define TAS5754M_AUTO_MUTE (TAS5754M_PAGE_BASE(0) + 59) +#define TAS5754M_DIGITAL_VOLUME_1 (TAS5754M_PAGE_BASE(0) + 60) +#define TAS5754M_DIGITAL_VOLUME_2 (TAS5754M_PAGE_BASE(0) + 61) +#define TAS5754M_DIGITAL_VOLUME_3 (TAS5754M_PAGE_BASE(0) + 62) +#define TAS5754M_DIGITAL_MUTE_1 (TAS5754M_PAGE_BASE(0) + 63) +#define TAS5754M_DIGITAL_MUTE_2 (TAS5754M_PAGE_BASE(0) + 64) +#define TAS5754M_DIGITAL_MUTE_3 (TAS5754M_PAGE_BASE(0) + 65) +#define TAS5754M_GPIO_OUTPUT_1 (TAS5754M_PAGE_BASE(0) + 80) +#define TAS5754M_GPIO_OUTPUT_2 (TAS5754M_PAGE_BASE(0) + 81) +#define TAS5754M_GPIO_OUTPUT_3 (TAS5754M_PAGE_BASE(0) + 82) +#define TAS5754M_GPIO_OUTPUT_4 (TAS5754M_PAGE_BASE(0) + 83) +#define TAS5754M_GPIO_OUTPUT_5 (TAS5754M_PAGE_BASE(0) + 84) +#define TAS5754M_GPIO_OUTPUT_6 (TAS5754M_PAGE_BASE(0) + 85) +#define TAS5754M_GPIO_CONTROL_1 (TAS5754M_PAGE_BASE(0) + 86) +#define TAS5754M_GPIO_CONTROL_2 (TAS5754M_PAGE_BASE(0) + 87) +#define TAS5754M_OVERFLOW (TAS5754M_PAGE_BASE(0) + 90) +#define TAS5754M_RATE_DET_1 (TAS5754M_PAGE_BASE(0) + 91) +#define TAS5754M_RATE_DET_2 (TAS5754M_PAGE_BASE(0) + 92) +#define TAS5754M_RATE_DET_3 (TAS5754M_PAGE_BASE(0) + 93) +#define TAS5754M_RATE_DET_4 (TAS5754M_PAGE_BASE(0) + 94) +#define TAS5754M_CLOCK_STATUS (TAS5754M_PAGE_BASE(0) + 95) +#define TAS5754M_ANALOG_MUTE_DET (TAS5754M_PAGE_BASE(0) + 108) +#define TAS5754M_GPIN (TAS5754M_PAGE_BASE(0) + 119) +#define TAS5754M_DIGITAL_MUTE_DET (TAS5754M_PAGE_BASE(0) + 120) + +#define TAS5754M_OUTPUT_AMPLITUDE (TAS5754M_PAGE_BASE(1) + 1) +#define TAS5754M_ANALOG_GAIN_CTRL (TAS5754M_PAGE_BASE(1) + 2) +#define TAS5754M_UNDERVOLTAGE_PROT (TAS5754M_PAGE_BASE(1) + 5) +#define TAS5754M_ANALOG_MUTE_CTRL (TAS5754M_PAGE_BASE(1) + 6) +#define TAS5754M_ANALOG_GAIN_BOOST (TAS5754M_PAGE_BASE(1) + 7) +#define TAS5754M_VCOM_CTRL_1 (TAS5754M_PAGE_BASE(1) + 8) +#define TAS5754M_VCOM_CTRL_2 (TAS5754M_PAGE_BASE(1) + 9) + +#define TAS5754M_CRAM_CTRL (TAS5754M_PAGE_BASE(44) + 1) + +#define TAS5754M_FLEX_A (TAS5754M_PAGE_BASE(253) + 63) +#define TAS5754M_FLEX_B (TAS5754M_PAGE_BASE(253) + 64) + +#define TAS5754M_MAX_REGISTER (TAS5754M_PAGE_BASE(253) + 64) + +/* Page 0, Register 1 - reset */ +#define TAS5754M_RSTR (1 << 0) +#define TAS5754M_RSTM (1 << 4) + +/* Page 0, Register 2 - power */ +#define TAS5754M_RQPD (1 << 0) +#define TAS5754M_RQPD_SHIFT 0 +#define TAS5754M_RQST (1 << 4) +#define TAS5754M_RQST_SHIFT 4 + +/* Page 0, Register 3 - mute */ +#define TAS5754M_RQMR (1 << 0) +#define TAS5754M_RQMR_SHIFT 0 +#define TAS5754M_RQML (1 << 4) +#define TAS5754M_RQML_SHIFT 4 + +/* Page 0, Register 4 - PLL */ +#define TAS5754M_PLLE (1 << 0) +#define TAS5754M_PLLE_SHIFT 0 +#define TAS5754M_PLCK (1 << 4) +#define TAS5754M_PLCK_SHIFT 4 + +/* Page 0, Register 7 - DSP */ +#define TAS5754M_SDSL (1 << 0) +#define TAS5754M_SDSL_SHIFT 0 +#define TAS5754M_DEMP (1 << 4) +#define TAS5754M_DEMP_SHIFT 4 + +/* Page 0, Register 8 - GPIO output enable */ +#define TAS5754M_G1OE (1 << 0) +#define TAS5754M_G2OE (1 << 1) +#define TAS5754M_G3OE (1 << 2) +#define TAS5754M_G4OE (1 << 3) +#define TAS5754M_G5OE (1 << 4) +#define TAS5754M_G6OE (1 << 5) + +/* Page 0, Register 9 - BCK, LRCLK configuration */ +#define TAS5754M_LRKO (1 << 0) +#define TAS5754M_LRKO_SHIFT 0 +#define TAS5754M_BCKO (1 << 4) +#define TAS5754M_BCKO_SHIFT 4 +#define TAS5754M_BCKP (1 << 5) +#define TAS5754M_BCKP_SHIFT 5 + +/* Page 0, Register 12 - Master mode BCK, LRCLK reset */ +#define TAS5754M_RLRK (1 << 0) +#define TAS5754M_RLRK_SHIFT 0 +#define TAS5754M_RBCK (1 << 1) +#define TAS5754M_RBCK_SHIFT 1 + +/* Page 0, Register 13 - PLL reference */ +#define TAS5754M_SREF (7 << 4) +#define TAS5754M_SREF_SHIFT 4 +#define TAS5754M_SREF_SCK (0 << 4) +#define TAS5754M_SREF_BCK (1 << 4) +#define TAS5754M_SREF_GPIO (3 << 4) + +/* Page 0, Register 14 - DAC reference */ +#define TAS5754M_SDAC (7 << 4) +#define TAS5754M_SDAC_SHIFT 4 +#define TAS5754M_SDAC_MCK (0 << 4) +#define TAS5754M_SDAC_PLL (1 << 4) +#define TAS5754M_SDAC_SCK (3 << 4) +#define TAS5754M_SDAC_BCK (4 << 4) +#define TAS5754M_SDAC_GPIO (5 << 4) + +/* Page 0, Register 16, 18 - GPIO source for DAC, PLL */ +#define TAS5754M_GREF (7 << 0) +#define TAS5754M_GREF_SHIFT 0 +#define TAS5754M_GREF_GPIO1 (0 << 0) +#define TAS5754M_GREF_GPIO2 (1 << 0) +#define TAS5754M_GREF_GPIO3 (2 << 0) +#define TAS5754M_GREF_GPIO4 (3 << 0) +#define TAS5754M_GREF_GPIO5 (4 << 0) +#define TAS5754M_GREF_GPIO6 (5 << 0) + +/* Page 0, Register 19 - synchronize */ +#define TAS5754M_RQSY (1 << 0) +#define TAS5754M_RQSY_RESUME (0 << 0) +#define TAS5754M_RQSY_HALT (1 << 0) + +/* Page 0, Register 34 - fs speed mode */ +#define TAS5754M_FSSP (3 << 0) +#define TAS5754M_FSSP_SHIFT 0 +#define TAS5754M_FSSP_48KHZ (0 << 0) +#define TAS5754M_FSSP_96KHZ (1 << 0) +#define TAS5754M_FSSP_192KHZ (2 << 0) +#define TAS5754M_FSSP_384KHZ (3 << 0) + +/* Page 0, Register 37 - Error detection */ +#define TAS5754M_IPLK (1 << 0) +#define TAS5754M_DCAS (1 << 1) +#define TAS5754M_IDCM (1 << 2) +#define TAS5754M_IDCH (1 << 3) +#define TAS5754M_IDSK (1 << 4) +#define TAS5754M_IDBK (1 << 5) +#define TAS5754M_IDFS (1 << 6) + +/* Page 0, Register 40 - I2S configuration */ +#define TAS5754M_ALEN (3 << 0) +#define TAS5754M_ALEN_SHIFT 0 +#define TAS5754M_ALEN_16 (0 << 0) +#define TAS5754M_ALEN_20 (1 << 0) +#define TAS5754M_ALEN_24 (2 << 0) +#define TAS5754M_ALEN_32 (3 << 0) +#define TAS5754M_AFMT (3 << 4) +#define TAS5754M_AFMT_SHIFT 4 +#define TAS5754M_AFMT_I2S (0 << 4) +#define TAS5754M_AFMT_DSP (1 << 4) +#define TAS5754M_AFMT_RTJ (2 << 4) +#define TAS5754M_AFMT_LTJ (3 << 4) + +/* Page 0, Register 42 - DAC routing */ +#define TAS5754M_AUPR_SHIFT 0 +#define TAS5754M_AUPL_SHIFT 4 + +/* Page 0, Register 59 - auto mute */ +#define TAS5754M_ATMR_SHIFT 0 +#define TAS5754M_ATML_SHIFT 4 + +/* Page 0, Register 63 - ramp rates */ +#define TAS5754M_VNDF_SHIFT 6 +#define TAS5754M_VNDS_SHIFT 4 +#define TAS5754M_VNUF_SHIFT 2 +#define TAS5754M_VNUS_SHIFT 0 + +/* Page 0, Register 64 - emergency ramp rates */ +#define TAS5754M_VEDF_SHIFT 6 +#define TAS5754M_VEDS_SHIFT 4 + +/* Page 0, Register 65 - Digital mute enables */ +#define TAS5754M_ACTL_SHIFT 2 +#define TAS5754M_AMLE_SHIFT 1 +#define TAS5754M_AMRE_SHIFT 0 + +/* Page 0, Register 80-85, GPIO output selection */ +#define TAS5754M_GxSL (31 << 0) +#define TAS5754M_GxSL_SHIFT 0 +#define TAS5754M_GxSL_OFF (0 << 0) +#define TAS5754M_GxSL_DSP (1 << 0) +#define TAS5754M_GxSL_REG (2 << 0) +#define TAS5754M_GxSL_AMUTB (3 << 0) +#define TAS5754M_GxSL_AMUTL (4 << 0) +#define TAS5754M_GxSL_AMUTR (5 << 0) +#define TAS5754M_GxSL_CLKI (6 << 0) +#define TAS5754M_GxSL_SDOUT (7 << 0) +#define TAS5754M_GxSL_ANMUL (8 << 0) +#define TAS5754M_GxSL_ANMUR (9 << 0) +#define TAS5754M_GxSL_PLLLK (10 << 0) +#define TAS5754M_GxSL_CPCLK (11 << 0) +#define TAS5754M_GxSL_UV0_7 (14 << 0) +#define TAS5754M_GxSL_UV0_3 (15 << 0) +#define TAS5754M_GxSL_PLLCK (16 << 0) + +/* Page 1, Register 2 - analog volume control */ +#define TAS5754M_RAGN_SHIFT 0 +#define TAS5754M_LAGN_SHIFT 4 + +/* Page 1, Register 7 - analog boost control */ +#define TAS5754M_AGBR_SHIFT 0 +#define TAS5754M_AGBL_SHIFT 4 + +#endif -- 2.17.1

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[PATCH] ALSA: hda/realtek: Add quirk for ASUS UX550VE
by Takashi Iwai 07 Nov '21

07 Nov '21

ASUS UX550VE (SSID 1043:1970) requires a similar workaround for managing the routing of the 4 speakers like some other ASUS models. Add a corresponding quirk entry for fixing it. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212641 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3601749f2df6..a6098c942aeb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8698,6 +8698,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE), + SND_PCI_QUIRK(0x1043, 0x1970, "ASUS UX550VE", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1982, "ASUS B1400CEPE", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE), -- 2.26.2

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Re: [Alsa-user] USB audio on riscv64
by Kip Warner 07 Nov '21

07 Nov '21

On Sat, 2021-11-06 at 14:50 -0700, chris hermansen wrote: > To allow remote users to use ALSA, you need to add those users to the > audio group, however this is not recommended by default (see note > below). Thanks Chris. For anyone else having a similar problem, adding myself to the audio group, like so, fixed the problem: $ sudo usermod -a -G audio kip -- Kip Warner -- Senior Software Engineer OpenPGP signed/encrypted mail preferred https://www.thevertigo.com

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