From 7d3fe292efb637d1f748926390a3a4cc90c4c4e9 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 13 Nov 2024 17:22:19 +0530 Subject: [PATCH 01/16] ASoC: amd: acp: add RT711, RT714 & RT1316 support for acp 6.3 platform This patch add supports for corresponding codecs on acp6.3 platform hardware configuration. SDW0: RT711 Jack SDW0: RT1316 Left Speaker SDW0: RT1316 Right Speaker SDW1: RT714 DMIC Signed-off-by: Vijendar Mukunda Link: https://patch.msgid.link/20241113115223.3274868-3-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/amd-acp63-acpi-match.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/amd/acp/amd-acp63-acpi-match.c b/sound/soc/amd/acp/amd-acp63-acpi-match.c index 5e506c9e3da6..9b6a49c051cd 100644 --- a/sound/soc/amd/acp/amd-acp63-acpi-match.c +++ b/sound/soc/amd/acp/amd-acp63-acpi-match.c @@ -130,6 +130,11 @@ struct snd_soc_acpi_mach snd_soc_acpi_amd_acp63_sdw_machines[] = { .links = acp63_rt722_only, .drv_name = "amd_sdw", }, + { + .link_mask = BIT(0) | BIT(1), + .links = acp63_4_in_1_sdca, + .drv_name = "amd_sdw", + }, {}, }; EXPORT_SYMBOL(snd_soc_acpi_amd_acp63_sdw_machines); -- 2.50.1 From 56d540befd5940dc34b4e22cc9b8ce9bb45946f7 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 13 Nov 2024 17:22:20 +0530 Subject: [PATCH 02/16] ASoC: amd: ps: add soundwire machines for acp6.3 platform Add SoundWire machines for acp 6.3 platform for legacy(No DSP) stack. Signed-off-by: Vijendar Mukunda Link: https://patch.msgid.link/20241113115223.3274868-4-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/ps/pci-ps.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 4365499c8f82..a7583844f5b4 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -598,6 +598,7 @@ static int snd_acp63_probe(struct pci_dev *pci, dev_err(&pci->dev, "ACP platform devices creation failed\n"); goto de_init; } + adata->machines = snd_soc_acpi_amd_acp63_sdw_machines; ret = acp63_machine_register(&pci->dev); if (ret) { dev_err(&pci->dev, "ACP machine register failed\n"); -- 2.50.1 From 393347cc10ea24c9f93b45e8e2f90fcc48ab1d8e Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 13 Nov 2024 17:22:21 +0530 Subject: [PATCH 03/16] ASoC: amd: acp: move get_acp63_cpu_pin_id() to common file get_acp63_cpu_pin_id() is the common SoundWire machine driver helper function will be used for AMD Legacy(No DSP) generic SoundWire machine driver as well. Move get_acp63_cpu_pin_id() function to common place holder. Signed-off-by: Vijendar Mukunda Link: https://patch.msgid.link/20241113115223.3274868-5-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/Kconfig | 7 +++ sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-sdw-mach-common.c | 64 +++++++++++++++++++++++++ sound/soc/amd/acp/acp-sdw-sof-mach.c | 49 +------------------ sound/soc/amd/acp/soc_amd_sdw_common.h | 2 + 5 files changed, 76 insertions(+), 48 deletions(-) create mode 100644 sound/soc/amd/acp/acp-sdw-mach-common.c diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 88391e4c17e3..acd047d558bd 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -119,10 +119,17 @@ config SND_SOC_AMD_SOF_MACH help This option enables SOF sound card support for ACP audio. +config SND_SOC_AMD_SDW_MACH_COMMON + tristate + help + This option enables common SoundWire Machine driver module for + AMD platforms. + config SND_SOC_AMD_SOF_SDW_MACH tristate "AMD SOF Soundwire Machine Driver Support" depends on X86 && PCI && ACPI depends on SOUNDWIRE + select SND_SOC_AMD_SDW_MACH_COMMON select SND_SOC_SDW_UTILS select SND_SOC_DMIC select SND_SOC_RT711_SDW diff --git a/sound/soc/amd/acp/Makefile b/sound/soc/amd/acp/Makefile index 82cf5d180b3a..0e6c4022e7a2 100644 --- a/sound/soc/amd/acp/Makefile +++ b/sound/soc/amd/acp/Makefile @@ -23,6 +23,7 @@ snd-acp-mach-y := acp-mach-common.o snd-acp-legacy-mach-y := acp-legacy-mach.o acp3x-es83xx/acp3x-es83xx.o snd-acp-sof-mach-y := acp-sof-mach.o snd-soc-acpi-amd-match-y := amd-acp63-acpi-match.o +snd-acp-sdw-mach-y := acp-sdw-mach-common.o snd-acp-sdw-sof-mach-y += acp-sdw-sof-mach.o obj-$(CONFIG_SND_SOC_AMD_ACP_PCM) += snd-acp-pcm.o @@ -41,4 +42,5 @@ obj-$(CONFIG_SND_SOC_AMD_MACH_COMMON) += snd-acp-mach.o obj-$(CONFIG_SND_SOC_AMD_LEGACY_MACH) += snd-acp-legacy-mach.o obj-$(CONFIG_SND_SOC_AMD_SOF_MACH) += snd-acp-sof-mach.o obj-$(CONFIG_SND_SOC_ACPI_AMD_MATCH) += snd-soc-acpi-amd-match.o +obj-$(CONFIG_SND_SOC_AMD_SDW_MACH_COMMON) += snd-acp-sdw-mach.o obj-$(CONFIG_SND_SOC_AMD_SOF_SDW_MACH) += snd-acp-sdw-sof-mach.o diff --git a/sound/soc/amd/acp/acp-sdw-mach-common.c b/sound/soc/amd/acp/acp-sdw-mach-common.c new file mode 100644 index 000000000000..d9393cc4a302 --- /dev/null +++ b/sound/soc/amd/acp/acp-sdw-mach-common.c @@ -0,0 +1,64 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright(c) 2024 Advanced Micro Devices, Inc. + +/* + * acp-sdw-mach-common - Common machine driver helper functions for + * legacy(No DSP) stack and SOF stack. + */ + +#include +#include +#include "soc_amd_sdw_common.h" + +int get_acp63_cpu_pin_id(u32 sdw_link_id, int be_id, int *cpu_pin_id, struct device *dev) +{ + switch (sdw_link_id) { + case AMD_SDW0: + switch (be_id) { + case SOC_SDW_JACK_OUT_DAI_ID: + *cpu_pin_id = ACP63_SW0_AUDIO0_TX; + break; + case SOC_SDW_JACK_IN_DAI_ID: + *cpu_pin_id = ACP63_SW0_AUDIO0_RX; + break; + case SOC_SDW_AMP_OUT_DAI_ID: + *cpu_pin_id = ACP63_SW0_AUDIO1_TX; + break; + case SOC_SDW_AMP_IN_DAI_ID: + *cpu_pin_id = ACP63_SW0_AUDIO1_RX; + break; + case SOC_SDW_DMIC_DAI_ID: + *cpu_pin_id = ACP63_SW0_AUDIO2_RX; + break; + default: + dev_err(dev, "Invalid be id:%d\n", be_id); + return -EINVAL; + } + break; + case AMD_SDW1: + switch (be_id) { + case SOC_SDW_JACK_OUT_DAI_ID: + case SOC_SDW_AMP_OUT_DAI_ID: + *cpu_pin_id = ACP63_SW1_AUDIO0_TX; + break; + case SOC_SDW_JACK_IN_DAI_ID: + case SOC_SDW_AMP_IN_DAI_ID: + case SOC_SDW_DMIC_DAI_ID: + *cpu_pin_id = ACP63_SW1_AUDIO0_RX; + break; + default: + dev_err(dev, "invalid be_id:%d\n", be_id); + return -EINVAL; + } + break; + default: + dev_err(dev, "Invalid link id:%d\n", sdw_link_id); + return -EINVAL; + } + return 0; +} +EXPORT_SYMBOL_NS_GPL(get_acp63_cpu_pin_id, SND_SOC_AMD_SDW_MACH); + +MODULE_DESCRIPTION("AMD SoundWire Common Machine driver"); +MODULE_AUTHOR("Vijendar Mukunda "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index 8fce8cb957c9..0d256c0749c9 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -64,54 +64,6 @@ static const struct snd_soc_ops sdw_ops = { .shutdown = asoc_sdw_shutdown, }; -static int get_acp63_cpu_pin_id(u32 sdw_link_id, int be_id, int *cpu_pin_id, struct device *dev) -{ - switch (sdw_link_id) { - case AMD_SDW0: - switch (be_id) { - case SOC_SDW_JACK_OUT_DAI_ID: - *cpu_pin_id = ACP63_SW0_AUDIO0_TX; - break; - case SOC_SDW_JACK_IN_DAI_ID: - *cpu_pin_id = ACP63_SW0_AUDIO0_RX; - break; - case SOC_SDW_AMP_OUT_DAI_ID: - *cpu_pin_id = ACP63_SW0_AUDIO1_TX; - break; - case SOC_SDW_AMP_IN_DAI_ID: - *cpu_pin_id = ACP63_SW0_AUDIO1_RX; - break; - case SOC_SDW_DMIC_DAI_ID: - *cpu_pin_id = ACP63_SW0_AUDIO2_RX; - break; - default: - dev_err(dev, "Invalid be id:%d\n", be_id); - return -EINVAL; - } - break; - case AMD_SDW1: - switch (be_id) { - case SOC_SDW_JACK_OUT_DAI_ID: - case SOC_SDW_AMP_OUT_DAI_ID: - *cpu_pin_id = ACP63_SW1_AUDIO0_TX; - break; - case SOC_SDW_JACK_IN_DAI_ID: - case SOC_SDW_AMP_IN_DAI_ID: - case SOC_SDW_DMIC_DAI_ID: - *cpu_pin_id = ACP63_SW1_AUDIO0_RX; - break; - default: - dev_err(dev, "invalid be_id:%d\n", be_id); - return -EINVAL; - } - break; - default: - dev_err(dev, "Invalid link id:%d\n", sdw_link_id); - return -EINVAL; - } - return 0; -} - static const char * const type_strings[] = {"SimpleJack", "SmartAmp", "SmartMic"}; static int create_sdw_dailink(struct snd_soc_card *card, @@ -491,3 +443,4 @@ MODULE_DESCRIPTION("ASoC AMD SoundWire Generic Machine driver"); MODULE_AUTHOR("Vijendar Mukunda Date: Wed, 13 Nov 2024 17:22:22 +0530 Subject: [PATCH 04/16] ASoC: amd: acp: add soundwire machine driver for legacy stack Add SoundWire machine driver for legacy(No DSP) stack for ACP6.3 platform. Signed-off-by: Vijendar Mukunda Link: https://patch.msgid.link/20241113115223.3274868-6-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/Kconfig | 22 ++ sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-sdw-legacy-mach.c | 486 ++++++++++++++++++++++++ sound/soc/amd/acp/soc_amd_sdw_common.h | 2 + 4 files changed, 512 insertions(+) create mode 100644 sound/soc/amd/acp/acp-sdw-legacy-mach.c diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index acd047d558bd..03f3fcbba5af 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -144,6 +144,28 @@ config SND_SOC_AMD_SOF_SDW_MACH on AMD platform. If unsure select "N". +config SND_SOC_AMD_LEGACY_SDW_MACH + tristate "AMD Legacy(No DSP) Soundwire Machine Driver Support" + depends on X86 && PCI && ACPI + depends on SOUNDWIRE + select SND_SOC_AMD_SDW_MACH_COMMON + select SND_SOC_SDW_UTILS + select SND_SOC_DMIC + select SND_SOC_RT711_SDW + select SND_SOC_RT711_SDCA_SDW + select SND_SOC_RT712_SDCA_SDW + select SND_SOC_RT712_SDCA_DMIC_SDW + select SND_SOC_RT1316_SDW + select SND_SOC_RT715_SDW + select SND_SOC_RT715_SDCA_SDW + select SND_SOC_RT722_SDCA_SDW + help + This option enables Legacy(No DSP) sound card support for SoundWire + enabled AMD platforms along with ACP PDM controller. + Say Y if you want to enable SoundWire based machine driver support + on AMD platform. + If unsure select "N". + endif # SND_SOC_AMD_ACP_COMMON config SND_AMD_SOUNDWIRE_ACPI diff --git a/sound/soc/amd/acp/Makefile b/sound/soc/amd/acp/Makefile index 0e6c4022e7a2..bb2702036338 100644 --- a/sound/soc/amd/acp/Makefile +++ b/sound/soc/amd/acp/Makefile @@ -25,6 +25,7 @@ snd-acp-sof-mach-y := acp-sof-mach.o snd-soc-acpi-amd-match-y := amd-acp63-acpi-match.o snd-acp-sdw-mach-y := acp-sdw-mach-common.o snd-acp-sdw-sof-mach-y += acp-sdw-sof-mach.o +snd-acp-sdw-legacy-mach-y += acp-sdw-legacy-mach.o obj-$(CONFIG_SND_SOC_AMD_ACP_PCM) += snd-acp-pcm.o obj-$(CONFIG_SND_SOC_AMD_ACP_I2S) += snd-acp-i2s.o @@ -44,3 +45,4 @@ obj-$(CONFIG_SND_SOC_AMD_SOF_MACH) += snd-acp-sof-mach.o obj-$(CONFIG_SND_SOC_ACPI_AMD_MATCH) += snd-soc-acpi-amd-match.o obj-$(CONFIG_SND_SOC_AMD_SDW_MACH_COMMON) += snd-acp-sdw-mach.o obj-$(CONFIG_SND_SOC_AMD_SOF_SDW_MACH) += snd-acp-sdw-sof-mach.o +obj-$(CONFIG_SND_SOC_AMD_LEGACY_SDW_MACH) += snd-acp-sdw-legacy-mach.o diff --git a/sound/soc/amd/acp/acp-sdw-legacy-mach.c b/sound/soc/amd/acp/acp-sdw-legacy-mach.c new file mode 100644 index 000000000000..48952a238946 --- /dev/null +++ b/sound/soc/amd/acp/acp-sdw-legacy-mach.c @@ -0,0 +1,486 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright(c) 2024 Advanced Micro Devices, Inc. + +/* + * acp-sdw-legacy-mach - ASoC legacy Machine driver for AMD SoundWire platforms + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "soc_amd_sdw_common.h" +#include "../../codecs/rt711.h" + +static unsigned long soc_sdw_quirk = RT711_JD1; +static int quirk_override = -1; +module_param_named(quirk, quirk_override, int, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); + +static void log_quirks(struct device *dev) +{ + if (SOC_JACK_JDSRC(soc_sdw_quirk)) + dev_dbg(dev, "quirk realtek,jack-detect-source %ld\n", + SOC_JACK_JDSRC(soc_sdw_quirk)); + if (soc_sdw_quirk & ASOC_SDW_ACP_DMIC) + dev_dbg(dev, "quirk SOC_SDW_ACP_DMIC enabled\n"); +} + +static int soc_sdw_quirk_cb(const struct dmi_system_id *id) +{ + soc_sdw_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id soc_sdw_quirk_table[] = { + { + .callback = soc_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AMD"), + DMI_MATCH(DMI_PRODUCT_NAME, "Birman-PHX"), + }, + .driver_data = (void *)RT711_JD2, + }, + {} +}; + +static const struct snd_soc_ops sdw_ops = { + .startup = asoc_sdw_startup, + .prepare = asoc_sdw_prepare, + .trigger = asoc_sdw_trigger, + .hw_params = asoc_sdw_hw_params, + .hw_free = asoc_sdw_hw_free, + .shutdown = asoc_sdw_shutdown, +}; + +static const char * const type_strings[] = {"SimpleJack", "SmartAmp", "SmartMic"}; + +static int create_sdw_dailink(struct snd_soc_card *card, + struct asoc_sdw_dailink *soc_dai, + struct snd_soc_dai_link **dai_links, + int *be_id, struct snd_soc_codec_conf **codec_conf, + struct snd_soc_dai_link_component *sdw_platform_component) +{ + struct device *dev = card->dev; + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct amd_mc_ctx *amd_ctx = (struct amd_mc_ctx *)ctx->private; + struct asoc_sdw_endpoint *soc_end; + int cpu_pin_id; + int stream; + int ret; + + list_for_each_entry(soc_end, &soc_dai->endpoints, list) { + if (soc_end->name_prefix) { + (*codec_conf)->dlc.name = soc_end->codec_name; + (*codec_conf)->name_prefix = soc_end->name_prefix; + (*codec_conf)++; + } + + if (soc_end->include_sidecar) { + ret = soc_end->codec_info->add_sidecar(card, dai_links, codec_conf); + if (ret) + return ret; + } + } + + for_each_pcm_streams(stream) { + static const char * const sdw_stream_name[] = { + "SDW%d-PIN%d-PLAYBACK", + "SDW%d-PIN%d-CAPTURE", + "SDW%d-PIN%d-PLAYBACK-%s", + "SDW%d-PIN%d-CAPTURE-%s", + }; + struct snd_soc_dai_link_ch_map *codec_maps; + struct snd_soc_dai_link_component *codecs; + struct snd_soc_dai_link_component *cpus; + int num_cpus = hweight32(soc_dai->link_mask[stream]); + int num_codecs = soc_dai->num_devs[stream]; + int playback, capture; + int j = 0; + char *name; + + if (!soc_dai->num_devs[stream]) + continue; + + soc_end = list_first_entry(&soc_dai->endpoints, + struct asoc_sdw_endpoint, list); + + *be_id = soc_end->dai_info->dailink[stream]; + if (*be_id < 0) { + dev_err(dev, "Invalid dailink id %d\n", *be_id); + return -EINVAL; + } + + switch (amd_ctx->acp_rev) { + case ACP63_PCI_REV: + ret = get_acp63_cpu_pin_id(ffs(soc_end->link_mask - 1), + *be_id, &cpu_pin_id, dev); + if (ret) + return ret; + break; + default: + return -EINVAL; + } + /* create stream name according to first link id */ + if (ctx->append_dai_type) { + name = devm_kasprintf(dev, GFP_KERNEL, + sdw_stream_name[stream + 2], + ffs(soc_end->link_mask) - 1, + cpu_pin_id, + type_strings[soc_end->dai_info->dai_type]); + } else { + name = devm_kasprintf(dev, GFP_KERNEL, + sdw_stream_name[stream], + ffs(soc_end->link_mask) - 1, + cpu_pin_id); + } + if (!name) + return -ENOMEM; + + cpus = devm_kcalloc(dev, num_cpus, sizeof(*cpus), GFP_KERNEL); + if (!cpus) + return -ENOMEM; + + codecs = devm_kcalloc(dev, num_codecs, sizeof(*codecs), GFP_KERNEL); + if (!codecs) + return -ENOMEM; + + codec_maps = devm_kcalloc(dev, num_codecs, sizeof(*codec_maps), GFP_KERNEL); + if (!codec_maps) + return -ENOMEM; + + list_for_each_entry(soc_end, &soc_dai->endpoints, list) { + if (!soc_end->dai_info->direction[stream]) + continue; + + int link_num = ffs(soc_end->link_mask) - 1; + + cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SDW%d Pin%d", + link_num, cpu_pin_id); + dev_dbg(dev, "cpu->dai_name:%s\n", cpus->dai_name); + if (!cpus->dai_name) + return -ENOMEM; + + codec_maps[j].cpu = 0; + codec_maps[j].codec = j; + + codecs[j].name = soc_end->codec_name; + codecs[j].dai_name = soc_end->dai_info->dai_name; + j++; + } + + WARN_ON(j != num_codecs); + + playback = (stream == SNDRV_PCM_STREAM_PLAYBACK); + capture = (stream == SNDRV_PCM_STREAM_CAPTURE); + + asoc_sdw_init_dai_link(dev, *dai_links, be_id, name, playback, capture, + cpus, num_cpus, sdw_platform_component, + 1, codecs, num_codecs, + 0, asoc_sdw_rtd_init, &sdw_ops); + /* + * SoundWire DAILINKs use 'stream' functions and Bank Switch operations + * based on wait_for_completion(), tag them as 'nonatomic'. + */ + (*dai_links)->nonatomic = true; + (*dai_links)->ch_maps = codec_maps; + + list_for_each_entry(soc_end, &soc_dai->endpoints, list) { + if (soc_end->dai_info->init) + soc_end->dai_info->init(card, *dai_links, + soc_end->codec_info, + playback); + } + + (*dai_links)++; + } + + return 0; +} + +static int create_sdw_dailinks(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, int *be_id, + struct asoc_sdw_dailink *soc_dais, + struct snd_soc_codec_conf **codec_conf) +{ + struct device *dev = card->dev; + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct amd_mc_ctx *amd_ctx = (struct amd_mc_ctx *)ctx->private; + struct snd_soc_dai_link_component *sdw_platform_component; + int ret; + + sdw_platform_component = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!sdw_platform_component) + return -ENOMEM; + + switch (amd_ctx->acp_rev) { + case ACP63_PCI_REV: + sdw_platform_component->name = "amd_ps_sdw_dma.0"; + break; + default: + return -EINVAL; + } + + /* generate DAI links by each sdw link */ + while (soc_dais->initialised) { + int current_be_id; + + ret = create_sdw_dailink(card, soc_dais, dai_links, + ¤t_be_id, codec_conf, sdw_platform_component); + if (ret) + return ret; + + /* Update the be_id to match the highest ID used for SDW link */ + if (*be_id < current_be_id) + *be_id = current_be_id; + + soc_dais++; + } + + return 0; +} + +static int create_dmic_dailinks(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, int *be_id, int no_pcm) +{ + struct device *dev = card->dev; + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct amd_mc_ctx *amd_ctx = (struct amd_mc_ctx *)ctx->private; + struct snd_soc_dai_link_component *pdm_cpu; + struct snd_soc_dai_link_component *pdm_platform; + int ret; + + pdm_cpu = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); + if (!pdm_cpu) + return -ENOMEM; + + pdm_platform = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); + if (!pdm_platform) + return -ENOMEM; + + switch (amd_ctx->acp_rev) { + case ACP63_PCI_REV: + pdm_cpu->name = "acp_ps_pdm_dma.0"; + pdm_platform->name = "acp_ps_pdm_dma.0"; + break; + default: + return -EINVAL; + } + + *be_id = ACP_DMIC_BE_ID; + ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, "acp-dmic-codec", + 0, 1, // DMIC only supports capture + pdm_cpu->name, pdm_platform->name, 1, + "dmic-codec.0", "dmic-hifi", no_pcm, + asoc_sdw_dmic_init, NULL); + if (ret) + return ret; + + (*dai_links)++; + + return 0; +} + +static int soc_card_dai_links_create(struct snd_soc_card *card) +{ + struct device *dev = card->dev; + struct snd_soc_acpi_mach *mach = dev_get_platdata(card->dev); + int sdw_be_num = 0, dmic_num = 0; + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; + struct asoc_sdw_endpoint *soc_ends __free(kfree) = NULL; + struct asoc_sdw_dailink *soc_dais __free(kfree) = NULL; + struct snd_soc_codec_conf *codec_conf; + struct snd_soc_dai_link *dai_links; + int num_devs = 0; + int num_ends = 0; + int num_links; + int be_id = 0; + int ret; + + ret = asoc_sdw_count_sdw_endpoints(card, &num_devs, &num_ends); + if (ret < 0) { + dev_err(dev, "failed to count devices/endpoints: %d\n", ret); + return ret; + } + + /* One per DAI link, worst case is a DAI link for every endpoint */ + soc_dais = kcalloc(num_ends, sizeof(*soc_dais), GFP_KERNEL); + if (!soc_dais) + return -ENOMEM; + + /* One per endpoint, ie. each DAI on each codec/amp */ + soc_ends = kcalloc(num_ends, sizeof(*soc_ends), GFP_KERNEL); + if (!soc_ends) + return -ENOMEM; + + ret = asoc_sdw_parse_sdw_endpoints(card, soc_dais, soc_ends, &num_devs); + if (ret < 0) + return ret; + + sdw_be_num = ret; + + /* enable dmic */ + if (soc_sdw_quirk & ASOC_SDW_ACP_DMIC || mach_params->dmic_num) + dmic_num = 1; + + dev_dbg(dev, "sdw %d, dmic %d", sdw_be_num, dmic_num); + + codec_conf = devm_kcalloc(dev, num_devs, sizeof(*codec_conf), GFP_KERNEL); + if (!codec_conf) + return -ENOMEM; + + /* allocate BE dailinks */ + num_links = sdw_be_num + dmic_num; + dai_links = devm_kcalloc(dev, num_links, sizeof(*dai_links), GFP_KERNEL); + if (!dai_links) + return -ENOMEM; + + card->codec_conf = codec_conf; + card->num_configs = num_devs; + card->dai_link = dai_links; + card->num_links = num_links; + + /* SDW */ + if (sdw_be_num) { + ret = create_sdw_dailinks(card, &dai_links, &be_id, + soc_dais, &codec_conf); + if (ret) + return ret; + } + + /* dmic */ + if (dmic_num > 0) { + if (ctx->ignore_internal_dmic) { + dev_warn(dev, "Ignoring ACP DMIC\n"); + } else { + ret = create_dmic_dailinks(card, &dai_links, &be_id, 0); + if (ret) + return ret; + } + } + + WARN_ON(codec_conf != card->codec_conf + card->num_configs); + WARN_ON(dai_links != card->dai_link + card->num_links); + + return ret; +} + +static int mc_probe(struct platform_device *pdev) +{ + struct snd_soc_acpi_mach *mach = dev_get_platdata(&pdev->dev); + struct snd_soc_card *card; + struct amd_mc_ctx *amd_ctx; + struct asoc_sdw_mc_private *ctx; + int amp_num = 0, i; + int ret; + + amd_ctx = devm_kzalloc(&pdev->dev, sizeof(*amd_ctx), GFP_KERNEL); + if (!amd_ctx) + return -ENOMEM; + + amd_ctx->acp_rev = mach->mach_params.subsystem_rev; + amd_ctx->max_sdw_links = ACP63_SDW_MAX_LINKS; + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + ctx->codec_info_list_count = asoc_sdw_get_codec_info_list_count(); + ctx->private = amd_ctx; + card = &ctx->card; + card->dev = &pdev->dev; + card->name = "amd-soundwire"; + card->owner = THIS_MODULE; + card->late_probe = asoc_sdw_card_late_probe; + + snd_soc_card_set_drvdata(card, ctx); + + dmi_check_system(soc_sdw_quirk_table); + + if (quirk_override != -1) { + dev_info(card->dev, "Overriding quirk 0x%lx => 0x%x\n", + soc_sdw_quirk, quirk_override); + soc_sdw_quirk = quirk_override; + } + + log_quirks(card->dev); + + ctx->mc_quirk = soc_sdw_quirk; + dev_dbg(card->dev, "legacy quirk 0x%lx\n", ctx->mc_quirk); + /* reset amp_num to ensure amp_num++ starts from 0 in each probe */ + for (i = 0; i < ctx->codec_info_list_count; i++) + codec_info_list[i].amp_num = 0; + + ret = soc_card_dai_links_create(card); + if (ret < 0) + return ret; + + /* + * the default amp_num is zero for each codec and + * amp_num will only be increased for active amp + * codecs on used platform + */ + for (i = 0; i < ctx->codec_info_list_count; i++) + amp_num += codec_info_list[i].amp_num; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + " cfg-amp:%d", amp_num); + if (!card->components) + return -ENOMEM; + if (mach->mach_params.dmic_num) { + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s mic:dmic cfg-mics:%d", + card->components, + mach->mach_params.dmic_num); + if (!card->components) + return -ENOMEM; + } + + /* Register the card */ + ret = devm_snd_soc_register_card(card->dev, card); + if (ret) { + dev_err_probe(card->dev, ret, "snd_soc_register_card failed %d\n", ret); + asoc_sdw_mc_dailink_exit_loop(card); + return ret; + } + + platform_set_drvdata(pdev, card); + + return ret; +} + +static void mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + asoc_sdw_mc_dailink_exit_loop(card); +} + +static const struct platform_device_id mc_id_table[] = { + { "amd_sdw", }, + {} +}; +MODULE_DEVICE_TABLE(platform, mc_id_table); + +static struct platform_driver soc_sdw_driver = { + .driver = { + .name = "amd_sdw", + .pm = &snd_soc_pm_ops, + }, + .probe = mc_probe, + .remove = mc_remove, + .id_table = mc_id_table, +}; + +module_platform_driver(soc_sdw_driver); + +MODULE_DESCRIPTION("ASoC AMD SoundWire Legacy Generic Machine driver"); +MODULE_AUTHOR("Vijendar Mukunda "); +MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS(SND_SOC_SDW_UTILS); +MODULE_IMPORT_NS(SND_SOC_AMD_SDW_MACH); diff --git a/sound/soc/amd/acp/soc_amd_sdw_common.h b/sound/soc/amd/acp/soc_amd_sdw_common.h index eba92cd004d4..b7bae107c13e 100644 --- a/sound/soc/amd/acp/soc_amd_sdw_common.h +++ b/sound/soc/amd/acp/soc_amd_sdw_common.h @@ -36,6 +36,8 @@ #define ACP63_SW1_AUDIO0_TX 0 #define ACP63_SW1_AUDIO0_RX 1 +#define ACP_DMIC_BE_ID 4 + struct amd_mc_ctx { unsigned int acp_rev; unsigned int max_sdw_links; -- 2.50.1 From 76b5a3b2afdce1460dcd06221f7aa8eb2b807b1f Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 13 Nov 2024 17:22:23 +0530 Subject: [PATCH 05/16] ASoC: amd: ps: fix the pcm device numbering for acp 6.3 platform Fixed PCM device numbering is required for defining common alsa ucm changes for generic soundwire machine driver for legacy(No DSP) stack. Ex: For Headphone playback use case, use PCM device number as 0. For Headset mic Capture use case, PCM device number as 1. Set the 'use_dai_pcm_id' flag true in soundwire dma driver for acp 6.3 platform. This will fix the pcm device numbering based on dai_link->id. Signed-off-by: Vijendar Mukunda Link: https://patch.msgid.link/20241113115223.3274868-7-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/ps/ps-sdw-dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/amd/ps/ps-sdw-dma.c b/sound/soc/amd/ps/ps-sdw-dma.c index 3b4b9c6b3171..b602cca92b8b 100644 --- a/sound/soc/amd/ps/ps-sdw-dma.c +++ b/sound/soc/amd/ps/ps-sdw-dma.c @@ -445,6 +445,8 @@ static const struct snd_soc_component_driver acp63_sdw_component = { .trigger = acp63_sdw_dma_trigger, .pointer = acp63_sdw_dma_pointer, .pcm_construct = acp63_sdw_dma_new, + .use_dai_pcm_id = true, + }; static int acp63_sdw_platform_probe(struct platform_device *pdev) -- 2.50.1 From fb5e67c9d03b4a65fd43acc18cbafffff15bd8f9 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 13 Nov 2024 13:08:07 +0000 Subject: [PATCH 06/16] ASoC: SOF: ipc4-topology: remove redundant assignment to variable ret The variable ret is being assigned a zero value however the value is never read because ret is being re-assigned later after the end of the switch statement. The assignment is redundant and can be removed. Signed-off-by: Colin Ian King Link: https://patch.msgid.link/20241113130807.1386754-1-colin.i.king@gmail.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 56427d6e3679..624f52d2183c 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -755,7 +755,6 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) * It is fine to call kfree(ipc4_copier->copier_config) since * ipc4_copier->copier_config is null. */ - ret = 0; break; } -- 2.50.1 From de35b06bf15cb56c96c7a69474e305852cc170e3 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 13 Nov 2024 14:41:33 +0800 Subject: [PATCH 07/16] ASoC: sdca: test adev before calling acpi_dev_for_each_child MIME-Version: 1.0 Content-Type: text/plain; charset=utf8 Content-Transfer-Encoding: 8bit sdca_lookup_functions may be called by a Peripheral that is not listed in the ACPI table. Testing adev is required to avoid kernel NULL pointer dereference. Fixes: 3a513da1ae33 ("ASoC: SDCA: add initial module") Signed-off-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Liam Girdwood Reviewed-by: Ranjani Sridharan Link: https://patch.msgid.link/20241113064133.162501-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sdca/sdca_functions.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/sdca/sdca_functions.c b/sound/soc/sdca/sdca_functions.c index a6ad57430dd4..e6e5629c7054 100644 --- a/sound/soc/sdca/sdca_functions.c +++ b/sound/soc/sdca/sdca_functions.c @@ -165,6 +165,10 @@ void sdca_lookup_functions(struct sdw_slave *slave) struct device *dev = &slave->dev; struct acpi_device *adev = to_acpi_device_node(dev->fwnode); + if (!adev) { + dev_info(dev, "No matching ACPI device found, ignoring peripheral\n"); + return; + } acpi_dev_for_each_child(adev, find_sdca_function, &slave->sdca_data); } EXPORT_SYMBOL_NS(sdca_lookup_functions, SND_SOC_SDCA); -- 2.50.1 From 2b974284aa073d6e2936f9032e8ad7b99480b5b8 Mon Sep 17 00:00:00 2001 From: "Hendrik v. Raven" Date: Thu, 14 Nov 2024 12:01:25 +0100 Subject: [PATCH 08/16] ASoc: simple-mux: add idle-state support So far the mux changes it state immediately, even when not in use. Allow overriding this behaviour by specifying an optional idle-state. This state is used whenever the mux is powered down, only switching to the selected state on power up. If unspecified it defaults to as-is, maintaining the previous behaviour. Signed-off-by: Hendrik v. Raven Link: https://patch.msgid.link/20241114-simple-mux-idle-state-v2-1-a30cb37d2be2@merzmedtech.de Signed-off-by: Mark Brown --- sound/soc/codecs/simple-mux.c | 39 ++++++++++++++++++++++++++++++++++- 1 file changed, 38 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/simple-mux.c b/sound/soc/codecs/simple-mux.c index 240af0563283..390696440155 100644 --- a/sound/soc/codecs/simple-mux.c +++ b/sound/soc/codecs/simple-mux.c @@ -6,6 +6,7 @@ #include #include +#include #include #include @@ -16,6 +17,7 @@ struct simple_mux { struct gpio_desc *gpiod_mux; unsigned int mux; const char *mux_texts[MUX_TEXT_SIZE]; + unsigned int idle_state; struct soc_enum mux_enum; struct snd_kcontrol_new mux_mux; struct snd_soc_dapm_widget mux_widgets[MUX_WIDGET_SIZE]; @@ -57,6 +59,9 @@ static int simple_mux_control_put(struct snd_kcontrol *kcontrol, priv->mux = ucontrol->value.enumerated.item[0]; + if (priv->idle_state != MUX_IDLE_AS_IS && dapm->bias_level < SND_SOC_BIAS_PREPARE) + return 0; + gpiod_set_value_cansleep(priv->gpiod_mux, priv->mux); return snd_soc_dapm_mux_update_power(dapm, kcontrol, @@ -75,10 +80,33 @@ static unsigned int simple_mux_read(struct snd_soc_component *component, static const struct snd_kcontrol_new simple_mux_mux = SOC_DAPM_ENUM_EXT("Muxer", simple_mux_enum, simple_mux_control_get, simple_mux_control_put); +static int simple_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct simple_mux *priv = snd_soc_component_get_drvdata(c); + + if (priv->idle_state != MUX_IDLE_AS_IS) { + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + gpiod_set_value_cansleep(priv->gpiod_mux, priv->mux); + break; + case SND_SOC_DAPM_POST_PMD: + gpiod_set_value_cansleep(priv->gpiod_mux, priv->idle_state); + break; + default: + break; + } + } + + return 0; +} + static const struct snd_soc_dapm_widget simple_mux_dapm_widgets[MUX_WIDGET_SIZE] = { SND_SOC_DAPM_INPUT("IN1"), SND_SOC_DAPM_INPUT("IN2"), - SND_SOC_DAPM_MUX("MUX", SND_SOC_NOPM, 0, 0, &simple_mux_mux), // see simple_mux_probe() + SND_SOC_DAPM_MUX_E("MUX", SND_SOC_NOPM, 0, 0, &simple_mux_mux, // see simple_mux_probe() + simple_mux_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_OUTPUT("OUT"), }; @@ -93,6 +121,7 @@ static int simple_mux_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; struct device_node *np = dev->of_node; struct simple_mux *priv; + int ret; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -121,6 +150,14 @@ static int simple_mux_probe(struct platform_device *pdev) /* Overwrite text ("Input 1", "Input 2") if property exists */ of_property_read_string_array(np, "state-labels", priv->mux_texts, MUX_TEXT_SIZE); + ret = of_property_read_u32(np, "idle-state", &priv->idle_state); + if (ret < 0) { + priv->idle_state = MUX_IDLE_AS_IS; + } else if (priv->idle_state != MUX_IDLE_AS_IS && priv->idle_state >= 2) { + dev_err(dev, "invalid idle-state %u\n", priv->idle_state); + return -EINVAL; + } + /* switch to use priv data instead of default */ priv->mux_enum.texts = priv->mux_texts; priv->mux_mux.private_value = (unsigned long)&priv->mux_enum; -- 2.50.1 From 3b7e11a0116c30848d44429650ad80f9cc3bd963 Mon Sep 17 00:00:00 2001 From: "Hendrik v. Raven" Date: Thu, 14 Nov 2024 12:01:26 +0100 Subject: [PATCH 09/16] ASoC: dt-bindings: simple-mux: add idle-state property simple-mux immediately activates the new output, even when it is powered down. This can be undesirable in some cases, for example when a mechanical relais is used. Adds "idle-state" property from the mux controller to select the output state to be used when the mux is powered down. Signed-off-by: Hendrik v. Raven Link: https://patch.msgid.link/20241114-simple-mux-idle-state-v2-2-a30cb37d2be2@merzmedtech.de Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/simple-audio-mux.yaml | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml index 194ac1d4f4f5..9b1bda4852e1 100644 --- a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml +++ b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml @@ -29,6 +29,10 @@ properties: $ref: /schemas/types.yaml#/definitions/string-array maxItems: 2 + idle-state: + description: If present specifies the state when the mux is powered down + $ref: /schemas/mux/mux-controller.yaml#/properties/idle-state + sound-name-prefix: true required: @@ -43,4 +47,5 @@ examples: compatible = "simple-audio-mux"; mux-gpios = <&gpio 3 0>; state-labels = "Label_A", "Label_B"; + idle-state = <0>; }; -- 2.50.1 From c48a4497356f701f94f1951626637ae240af909e Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 13 Nov 2024 18:57:13 +0100 Subject: [PATCH 10/16] ASoC: sma1307: fix uninitialized variable refence When firmware loading is disabled, gcc warns that the local 'fw' variable fails to get initialized: sound/soc/codecs/sma1307.c: In function 'sma1307_setting_loaded.isra': sound/soc/codecs/sma1307.c:1717:12: error: 'fw' is used uninitialized [-Werror=uninitialized] 1717 | if (!fw) { | ^ sound/soc/codecs/sma1307.c:1712:32: note: 'fw' was declared here 1712 | const struct firmware *fw; Check the return code from request_firmware() to ensure that the firmware is correctly set, and drop the incorrect release_firmware() on that uninitialized data. Fixes: 576c57e6b4c1 ("ASoC: sma1307: Add driver for Iron Device SMA1307") Signed-off-by: Arnd Bergmann Link: https://patch.msgid.link/20241113175734.2443315-1-arnd@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/sma1307.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/sma1307.c b/sound/soc/codecs/sma1307.c index 81638768ac12..f2cea6186d98 100644 --- a/sound/soc/codecs/sma1307.c +++ b/sound/soc/codecs/sma1307.c @@ -1711,13 +1711,13 @@ static void sma1307_setting_loaded(struct sma1307_priv *sma1307, const char *fil { const struct firmware *fw; int *data, size, offset, num_mode; + int ret; - request_firmware(&fw, file, sma1307->dev); + ret = request_firmware(&fw, file, sma1307->dev); - if (!fw) { - dev_err(sma1307->dev, "%s: failed to read \"%s\"\n", - __func__, setting_file); - release_firmware(fw); + if (ret) { + dev_err(sma1307->dev, "%s: failed to read \"%s\": %pe\n", + __func__, setting_file, ERR_PTR(ret)); sma1307->set.status = false; return; } else if ((fw->size) < SMA1307_SETTING_HEADER_SIZE) { -- 2.50.1 From ba888450828befb0607219f34c03aa8645625447 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Thu, 14 Nov 2024 11:28:51 +0100 Subject: [PATCH 11/16] ASoC: stm32: dfsdm: change rate upper limits Increase rate upper limit to 192kHz to reflect the rate range actually supported by the STM32 DFSDM peripheral. Signed-off-by: Olivier Moysan Link: https://patch.msgid.link/20241114102851.2497942-1-olivier.moysan@foss.st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_adfsdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 78bd817af839..c914d1c46850 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -142,7 +142,7 @@ static const struct snd_soc_dai_driver stm32_adfsdm_dai = { SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, - .rate_max = 48000, + .rate_max = 192000, }, .ops = &stm32_adfsdm_dai_ops, }; -- 2.50.1 From a59360466a712d416f8cddfa4e52e118c53aa3a3 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 13 Nov 2024 14:44:17 +0800 Subject: [PATCH 12/16] ASoC: Intel: soc-acpi-intel-lnl-match: add rt712_vb + rt1320 support MIME-Version: 1.0 Content-Type: text/plain; charset=utf8 Content-Transfer-Encoding: 8bit Realtek Gen6 AIOC supports rt712_vb on SoundWire link 2 and rt1320 on SoundWire link 1. Signed-off-by: Bard Liao Reviewed-by: Ranjani Sridharan Reviewed-by: Liam Girdwood Reviewed-by: Péter Ujfalusi Link: https://patch.msgid.link/20241113064418.162592-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-lnl-match.c | 65 +++++++++++++++++++ 1 file changed, 65 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c index 094ed4b27cb0..98a9c36d7a4c 100644 --- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c @@ -8,6 +8,7 @@ #include #include +#include "soc-acpi-intel-sdca-quirks.h" #include "soc-acpi-intel-sdw-mockup-match.h" struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_machines[] = { @@ -90,6 +91,30 @@ static const struct snd_soc_acpi_endpoint rt722_endpoints[] = { }, }; +static const struct snd_soc_acpi_endpoint jack_amp_g1_dmic_endpoints_endpoints[] = { + /* Jack Endpoint */ + { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + /* Amp Endpoint, work as spk_l_endpoint */ + { + .num = 1, + .aggregated = 1, + .group_position = 0, + .group_id = 1, + }, + /* DMIC Endpoint */ + { + .num = 2, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, +}; + static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = { { /* Jack Playback Endpoint */ .num = 0, @@ -198,6 +223,15 @@ static const struct snd_soc_acpi_adr_device rt1712_3_single_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt712_vb_2_group1_adr[] = { + { + .adr = 0x000230025D071201ull, + .num_endpoints = ARRAY_SIZE(jack_amp_g1_dmic_endpoints_endpoints), + .endpoints = jack_amp_g1_dmic_endpoints_endpoints, + .name_prefix = "rt712" + } +}; + static const struct snd_soc_acpi_adr_device rt722_0_single_adr[] = { { .adr = 0x000030025d072201ull, @@ -252,6 +286,15 @@ static const struct snd_soc_acpi_adr_device rt1318_2_group1_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt1320_1_group1_adr[] = { + { + .adr = 0x000130025D132001ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "rt1320-1" + } +}; + static const struct snd_soc_acpi_adr_device rt713_0_adr[] = { { .adr = 0x000031025D071301ull, @@ -410,6 +453,21 @@ static const struct snd_soc_acpi_link_adr lnl_sdw_rt713_l0_rt1318_l1[] = { {} }; +static const struct snd_soc_acpi_link_adr lnl_sdw_rt712_vb_l2_rt1320_l1[] = { + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(rt712_vb_2_group1_adr), + .adr_d = rt712_vb_2_group1_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1320_1_group1_adr), + .adr_d = rt1320_1_group1_adr, + }, + {} +}; + +/* this table is used when there is no I2S codec present */ /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { /* mockup tests need to be first */ @@ -485,6 +543,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-lnl-rt713-l0-rt1318-l1.tplg" }, + { + .link_mask = BIT(1) | BIT(2), + .links = lnl_sdw_rt712_vb_l2_rt1320_l1, + .drv_name = "sof_sdw", + .machine_check = snd_soc_acpi_intel_sdca_is_device_rt712_vb, + .sof_tplg_filename = "sof-lnl-rt712-l2-rt1320-l1.tplg" + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_lnl_sdw_machines); -- 2.50.1 From 9b4662d0df9f4433f1828904ba5e8733c1ad5158 Mon Sep 17 00:00:00 2001 From: zhang jiao Date: Thu, 14 Nov 2024 15:58:22 +0800 Subject: [PATCH 13/16] ALSA: ump: Fix the wrong format specifier The format specifier of "unsigned int" in snprintf() should be "%u", not "%d". Signed-off-by: zhang jiao Link: https://patch.msgid.link/20241114075822.41614-1-zhangjiao2@cmss.chinamobile.com Signed-off-by: Takashi Iwai --- sound/core/ump.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/ump.c b/sound/core/ump.c index ab4932dc499f..5d4dd207e5ab 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -387,7 +387,7 @@ int snd_ump_block_new(struct snd_ump_endpoint *ump, unsigned int blk, fb->info.first_group = first_group; fb->info.num_groups = num_groups; /* fill the default name, may be overwritten to a better name */ - snprintf(fb->info.name, sizeof(fb->info.name), "Group %d-%d", + snprintf(fb->info.name, sizeof(fb->info.name), "Group %u-%u", first_group + 1, first_group + num_groups); /* put the entry in the ordered list */ -- 2.50.1 From 82ff5abc2edcfba0c0f1a1be807795e2876f46e9 Mon Sep 17 00:00:00 2001 From: Jonas Karlman Date: Fri, 15 Nov 2024 04:43:44 +0000 Subject: [PATCH 14/16] ASoC: hdmi-codec: reorder channel allocation list The ordering in hdmi_codec_get_ch_alloc_table_idx() results in wrong channel allocation for a number of cases, e.g. when ELD reports FL|FR|LFE|FC|RL|RR or FL|FR|LFE|FC|RL|RR|RC|RLC|RRC: ca_id 0x01 with speaker mask FL|FR|LFE is selected instead of ca_id 0x03 with speaker mask FL|FR|LFE|FC for 4 channels and ca_id 0x04 with speaker mask FL|FR|RC gets selected instead of ca_id 0x0b with speaker mask FL|FR|LFE|FC|RL|RR for 6 channels Fix this by reordering the channel allocation list with most specific speaker masks at the top. Signed-off-by: Jonas Karlman Signed-off-by: Christian Hewitt Link: https://patch.msgid.link/20241115044344.3510979-1-christianshewitt@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 140 +++++++++++++++++++--------------- 1 file changed, 77 insertions(+), 63 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 74caae52e127..d9df29a26f4f 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -185,84 +185,97 @@ static const struct snd_pcm_chmap_elem hdmi_codec_8ch_chmaps[] = { /* * hdmi_codec_channel_alloc: speaker configuration available for CEA * - * This is an ordered list that must match with hdmi_codec_8ch_chmaps struct + * This is an ordered list where ca_id must exist in hdmi_codec_8ch_chmaps * The preceding ones have better chances to be selected by * hdmi_codec_get_ch_alloc_table_idx(). */ static const struct hdmi_codec_cea_spk_alloc hdmi_codec_channel_alloc[] = { { .ca_id = 0x00, .n_ch = 2, - .mask = FL | FR}, - /* 2.1 */ - { .ca_id = 0x01, .n_ch = 4, - .mask = FL | FR | LFE}, - /* Dolby Surround */ + .mask = FL | FR }, + { .ca_id = 0x03, .n_ch = 4, + .mask = FL | FR | LFE | FC }, { .ca_id = 0x02, .n_ch = 4, .mask = FL | FR | FC }, - /* surround51 */ + { .ca_id = 0x01, .n_ch = 4, + .mask = FL | FR | LFE }, { .ca_id = 0x0b, .n_ch = 6, - .mask = FL | FR | LFE | FC | RL | RR}, - /* surround40 */ - { .ca_id = 0x08, .n_ch = 6, - .mask = FL | FR | RL | RR }, - /* surround41 */ - { .ca_id = 0x09, .n_ch = 6, - .mask = FL | FR | LFE | RL | RR }, - /* surround50 */ + .mask = FL | FR | LFE | FC | RL | RR }, { .ca_id = 0x0a, .n_ch = 6, .mask = FL | FR | FC | RL | RR }, - /* 6.1 */ - { .ca_id = 0x0f, .n_ch = 8, - .mask = FL | FR | LFE | FC | RL | RR | RC }, - /* surround71 */ + { .ca_id = 0x09, .n_ch = 6, + .mask = FL | FR | LFE | RL | RR }, + { .ca_id = 0x08, .n_ch = 6, + .mask = FL | FR | RL | RR }, + { .ca_id = 0x07, .n_ch = 6, + .mask = FL | FR | LFE | FC | RC }, + { .ca_id = 0x06, .n_ch = 6, + .mask = FL | FR | FC | RC }, + { .ca_id = 0x05, .n_ch = 6, + .mask = FL | FR | LFE | RC }, + { .ca_id = 0x04, .n_ch = 6, + .mask = FL | FR | RC }, { .ca_id = 0x13, .n_ch = 8, .mask = FL | FR | LFE | FC | RL | RR | RLC | RRC }, - /* others */ - { .ca_id = 0x03, .n_ch = 8, - .mask = FL | FR | LFE | FC }, - { .ca_id = 0x04, .n_ch = 8, - .mask = FL | FR | RC}, - { .ca_id = 0x05, .n_ch = 8, - .mask = FL | FR | LFE | RC }, - { .ca_id = 0x06, .n_ch = 8, - .mask = FL | FR | FC | RC }, - { .ca_id = 0x07, .n_ch = 8, - .mask = FL | FR | LFE | FC | RC }, - { .ca_id = 0x0c, .n_ch = 8, - .mask = FL | FR | RC | RL | RR }, - { .ca_id = 0x0d, .n_ch = 8, - .mask = FL | FR | LFE | RL | RR | RC }, - { .ca_id = 0x0e, .n_ch = 8, - .mask = FL | FR | FC | RL | RR | RC }, - { .ca_id = 0x10, .n_ch = 8, - .mask = FL | FR | RL | RR | RLC | RRC }, - { .ca_id = 0x11, .n_ch = 8, - .mask = FL | FR | LFE | RL | RR | RLC | RRC }, + { .ca_id = 0x1f, .n_ch = 8, + .mask = FL | FR | LFE | FC | RL | RR | FLC | FRC }, { .ca_id = 0x12, .n_ch = 8, .mask = FL | FR | FC | RL | RR | RLC | RRC }, - { .ca_id = 0x14, .n_ch = 8, - .mask = FL | FR | FLC | FRC }, - { .ca_id = 0x15, .n_ch = 8, - .mask = FL | FR | LFE | FLC | FRC }, - { .ca_id = 0x16, .n_ch = 8, - .mask = FL | FR | FC | FLC | FRC }, - { .ca_id = 0x17, .n_ch = 8, - .mask = FL | FR | LFE | FC | FLC | FRC }, - { .ca_id = 0x18, .n_ch = 8, - .mask = FL | FR | RC | FLC | FRC }, - { .ca_id = 0x19, .n_ch = 8, - .mask = FL | FR | LFE | RC | FLC | FRC }, - { .ca_id = 0x1a, .n_ch = 8, - .mask = FL | FR | RC | FC | FLC | FRC }, - { .ca_id = 0x1b, .n_ch = 8, - .mask = FL | FR | LFE | RC | FC | FLC | FRC }, - { .ca_id = 0x1c, .n_ch = 8, - .mask = FL | FR | RL | RR | FLC | FRC }, - { .ca_id = 0x1d, .n_ch = 8, - .mask = FL | FR | LFE | RL | RR | FLC | FRC }, { .ca_id = 0x1e, .n_ch = 8, .mask = FL | FR | FC | RL | RR | FLC | FRC }, - { .ca_id = 0x1f, .n_ch = 8, - .mask = FL | FR | LFE | FC | RL | RR | FLC | FRC }, + { .ca_id = 0x11, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR | RLC | RRC }, + { .ca_id = 0x1d, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR | FLC | FRC }, + { .ca_id = 0x10, .n_ch = 8, + .mask = FL | FR | RL | RR | RLC | RRC }, + { .ca_id = 0x1c, .n_ch = 8, + .mask = FL | FR | RL | RR | FLC | FRC }, + { .ca_id = 0x0f, .n_ch = 8, + .mask = FL | FR | LFE | FC | RL | RR | RC }, + { .ca_id = 0x1b, .n_ch = 8, + .mask = FL | FR | LFE | RC | FC | FLC | FRC }, + { .ca_id = 0x0e, .n_ch = 8, + .mask = FL | FR | FC | RL | RR | RC }, + { .ca_id = 0x1a, .n_ch = 8, + .mask = FL | FR | RC | FC | FLC | FRC }, + { .ca_id = 0x0d, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR | RC }, + { .ca_id = 0x19, .n_ch = 8, + .mask = FL | FR | LFE | RC | FLC | FRC }, + { .ca_id = 0x0c, .n_ch = 8, + .mask = FL | FR | RC | RL | RR }, + { .ca_id = 0x18, .n_ch = 8, + .mask = FL | FR | RC | FLC | FRC }, + { .ca_id = 0x17, .n_ch = 8, + .mask = FL | FR | LFE | FC | FLC | FRC }, + { .ca_id = 0x16, .n_ch = 8, + .mask = FL | FR | FC | FLC | FRC }, + { .ca_id = 0x15, .n_ch = 8, + .mask = FL | FR | LFE | FLC | FRC }, + { .ca_id = 0x14, .n_ch = 8, + .mask = FL | FR | FLC | FRC }, + { .ca_id = 0x0b, .n_ch = 8, + .mask = FL | FR | LFE | FC | RL | RR }, + { .ca_id = 0x0a, .n_ch = 8, + .mask = FL | FR | FC | RL | RR }, + { .ca_id = 0x09, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR }, + { .ca_id = 0x08, .n_ch = 8, + .mask = FL | FR | RL | RR }, + { .ca_id = 0x07, .n_ch = 8, + .mask = FL | FR | LFE | FC | RC }, + { .ca_id = 0x06, .n_ch = 8, + .mask = FL | FR | FC | RC }, + { .ca_id = 0x05, .n_ch = 8, + .mask = FL | FR | LFE | RC }, + { .ca_id = 0x04, .n_ch = 8, + .mask = FL | FR | RC }, + { .ca_id = 0x03, .n_ch = 8, + .mask = FL | FR | LFE | FC }, + { .ca_id = 0x02, .n_ch = 8, + .mask = FL | FR | FC }, + { .ca_id = 0x01, .n_ch = 8, + .mask = FL | FR | LFE }, }; struct hdmi_codec_priv { @@ -371,7 +384,8 @@ static int hdmi_codec_chmap_ctl_get(struct snd_kcontrol *kcontrol, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hdmi_codec_priv *hcp = info->private_data; - map = info->chmap[hcp->chmap_idx].map; + if (hcp->chmap_idx != HDMI_CODEC_CHMAP_IDX_UNKNOWN) + map = info->chmap[hcp->chmap_idx].map; for (i = 0; i < info->max_channels; i++) { if (hcp->chmap_idx == HDMI_CODEC_CHMAP_IDX_UNKNOWN) -- 2.50.1 From e3f8064d8b29036f037fd1ff6000e5d959d84843 Mon Sep 17 00:00:00 2001 From: Huacai Chen Date: Fri, 15 Nov 2024 23:06:53 +0800 Subject: [PATCH 15/16] ALSA: hda: Poll jack events for LS7A HD-Audio LS7A HD-Audio disable interrupts and use polling mode due to hardware drawbacks. As a result, unsolicited jack events are also unusable. If we want to support headphone hotplug, we need to also poll jack events. Here we use 1500ms as the poll interval if no module parameter specify it. Cc: stable@vger.kernel.org Signed-off-by: Huacai Chen Link: https://patch.msgid.link/20241115150653.2819100-1-chenhuacai@loongson.cn Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6e271777feb9..4a62440adfaf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1859,6 +1859,8 @@ static int azx_first_init(struct azx *chip) bus->polling_mode = 1; bus->not_use_interrupts = 1; bus->access_sdnctl_in_dword = 1; + if (!chip->jackpoll_interval) + chip->jackpoll_interval = msecs_to_jiffies(1500); } err = pcim_iomap_regions(pci, 1 << 0, "ICH HD audio"); -- 2.50.1 From 0e84b414ca3778fd9308df52241a3617d82c20d2 Mon Sep 17 00:00:00 2001 From: Pei Xiao Date: Wed, 20 Nov 2024 14:30:19 +0800 Subject: [PATCH 16/16] ALSA: ac97: bus: Fix the mistake in the comment Fix mistake in the comment. sound/ac97/bus.c:192: warning: Function parameter or member 'drv' not described in 'snd_ac97_codec_driver_register' sound/ac97/bus.c:192: warning: Excess function parameter 'dev' description in 'snd_ac97_codec_driver_register' sound/ac97/bus.c:205: warning: Function parameter or member 'drv' not described in 'snd_ac97_codec_driver_unregister' sound/ac97/bus.c:205: warning: Excess function parameter 'dev' description in 'snd_ac97_codec_driver_unregister' sound/ac97/bus.c:351: warning: Function parameter or member 'codecs_pdata' not described in 'snd_ac97_controller_register' Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202411180804.FUfdymYO-lkp@intel.com/ Fixes: 74426fbff66e ("ALSA: ac97: add an ac97 bus") Signed-off-by: Pei Xiao Link: https://patch.msgid.link/3990bfc8cd47637908eaa179802c1d91459d829b.1732083924.git.xiaopei01@kylinos.cn Signed-off-by: Takashi Iwai --- sound/ac97/bus.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 96d4d7eb879f..8dfffdc101a2 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -180,7 +180,7 @@ static int ac97_bus_reset(struct ac97_controller *ac97_ctrl) /** * snd_ac97_codec_driver_register - register an AC97 codec driver - * @dev: AC97 driver codec to register + * @drv: AC97 driver codec to register * * Register an AC97 codec driver to the ac97 bus driver, aka. the AC97 digital * controller. @@ -196,7 +196,7 @@ EXPORT_SYMBOL_GPL(snd_ac97_codec_driver_register); /** * snd_ac97_codec_driver_unregister - unregister an AC97 codec driver - * @dev: AC97 codec driver to unregister + * @drv: AC97 codec driver to unregister * * Unregister a previously registered ac97 codec driver. */ @@ -338,6 +338,7 @@ static int ac97_add_adapter(struct ac97_controller *ac97_ctrl) * @dev: the device providing the ac97 DC function * @slots_available: mask of the ac97 codecs that can be scanned and probed * bit0 => codec 0, bit1 => codec 1 ... bit 3 => codec 3 + * @codecs_pdata: codec platform data * * Register a digital controller which can control up to 4 ac97 codecs. This is * the controller side of the AC97 AC-link, while the slave side are the codecs. -- 2.50.1