August 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: SOF: ipc4-topology: Add module parameter to ignore the CPC value
by Peter Ujfalusi 22 Aug '23

22 Aug '23

Add a new module parameter ipc4_ignore_cpc which can be used to force the kernel to ignore the queried CPC value for all firmware modules and use 0 instead. The CPC lookup is still done to report missing configurations and the debug print is going to be different to be explicit that the CPC is ignored and what was the value we would have used. The CPC value is sent to the firmware with the MOD_INIT_INSTANCE message and it is used by the firmware as a parameter for clock scaling. The flag is intended to be used only when there is a need to validate the firmware behavior regarding to clock scaling since the 0 CPC value will force the DSP to run in full speed, disabling the scaling and provides additional counter point to rule out clock management related issues. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 26 +++++++++++++++++++++++--- 1 file changed, 23 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 633f6040d712..733e5bff59a1 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -17,6 +17,19 @@ #include "ipc4-topology.h" #include "ops.h" +/* + * The ignore_cpc flag can be used to ignore the CPC value for all modules by + * using 0 instead. + * The CPC is sent to the firmware along with the SOF_IPC4_MOD_INIT_INSTANCE + * message and it is used for clock scaling. + * 0 as CPC value will instruct the firmware to use maximum frequency, thus + * deactivating the clock scaling. + */ +static bool ignore_cpc; +module_param_named(ipc4_ignore_cpc, ignore_cpc, bool, 0444); +MODULE_PARM_DESC(ipc4_ignore_cpc, + "Ignore CPC values. This option will disable clock scaling in firmware."); + #define SOF_IPC4_GAIN_PARAM_ID 0 #define SOF_IPC4_TPLG_ABI_SIZE 6 #define SOF_IPC4_CHAIN_DMA_BUF_SIZE_MS 2 @@ -970,9 +983,16 @@ sof_ipc4_update_resource_usage(struct snd_sof_dev *sdev, struct snd_sof_widget * /* Update base_config->cpc from the module manifest */ sof_ipc4_update_cpc_from_manifest(sdev, fw_module, base_config); - dev_dbg(sdev->dev, "%s: ibs / obs / cpc: %u / %u / %u\n", - swidget->widget->name, base_config->ibs, base_config->obs, - base_config->cpc); + if (ignore_cpc) { + dev_dbg(sdev->dev, "%s: ibs / obs: %u / %u, forcing cpc to 0 from %u\n", + swidget->widget->name, base_config->ibs, base_config->obs, + base_config->cpc); + base_config->cpc = 0; + } else { + dev_dbg(sdev->dev, "%s: ibs / obs / cpc: %u / %u / %u\n", + swidget->widget->name, base_config->ibs, base_config->obs, + base_config->cpc); + } } static int sof_ipc4_widget_assign_instance_id(struct snd_sof_dev *sdev, -- 2.41.0

2 1

[PATCH 1/3] ASoC: SOF: amd: Add sof support for vangogh platform
by Venkata Prasad Potturu 22 Aug '23

22 Aug '23

Add pci driver and platform driver to enable SOF support on ACP5x architecture based Vangogh platform. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/sof/amd/Kconfig | 12 ++- sound/soc/sof/amd/Makefile | 4 +- sound/soc/sof/amd/acp-dsp-offset.h | 8 +- sound/soc/sof/amd/acp.h | 9 +- sound/soc/sof/amd/pci-vangogh.c | 105 +++++++++++++++++++ sound/soc/sof/amd/vangogh.c | 156 +++++++++++++++++++++++++++++ 6 files changed, 290 insertions(+), 4 deletions(-) create mode 100644 sound/soc/sof/amd/pci-vangogh.c create mode 100644 sound/soc/sof/amd/vangogh.c diff --git a/sound/soc/sof/amd/Kconfig b/sound/soc/sof/amd/Kconfig index 7dbc8df5cfe6..f2faa08f0c0e 100644 --- a/sound/soc/sof/amd/Kconfig +++ b/sound/soc/sof/amd/Kconfig @@ -2,7 +2,7 @@ # This file is provided under a dual BSD/GPLv2 license. When using or # redistributing this file, you may do so under either license. # -# Copyright(c) 2021 Advanced Micro Devices, Inc. All rights reserved. +# Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. All rights reserved. config SND_SOC_SOF_AMD_TOPLEVEL tristate "SOF support for AMD audio DSPs" @@ -34,6 +34,16 @@ config SND_SOC_SOF_AMD_RENOIR help Select this option for SOF support on AMD Renoir platform +config SND_SOC_SOF_AMD_VANGOGH + tristate "SOF support for VANGOGH" + depends on SND_SOC_SOF_PCI + select SND_SOC_SOF_AMD_COMMON + help + Select this option for SOF support + on AMD Vangogh platform. + Say Y if you want to enable SOF on Vangogh. + If unsure select "N". + config SND_SOC_SOF_AMD_REMBRANDT tristate "SOF support for REMBRANDT" depends on SND_SOC_SOF_PCI diff --git a/sound/soc/sof/amd/Makefile b/sound/soc/sof/amd/Makefile index ef9f7df4e379..f3b375e67a6f 100644 --- a/sound/soc/sof/amd/Makefile +++ b/sound/soc/sof/amd/Makefile @@ -2,13 +2,15 @@ # This file is provided under a dual BSD/GPLv2 license. When using or # redistributing this file, you may do so under either license. # -# Copyright(c) 2021 Advanced Micro Devices, Inc. All rights reserved. +# Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. All rights reserved. snd-sof-amd-acp-objs := acp.o acp-loader.o acp-ipc.o acp-pcm.o acp-stream.o acp-trace.o acp-common.o snd-sof-amd-acp-$(CONFIG_SND_SOC_SOF_ACP_PROBES) = acp-probes.o snd-sof-amd-renoir-objs := pci-rn.o renoir.o snd-sof-amd-rembrandt-objs := pci-rmb.o rembrandt.o +snd-sof-amd-vangogh-objs := pci-vangogh.o vangogh.o obj-$(CONFIG_SND_SOC_SOF_AMD_COMMON) += snd-sof-amd-acp.o obj-$(CONFIG_SND_SOC_SOF_AMD_RENOIR) +=snd-sof-amd-renoir.o obj-$(CONFIG_SND_SOC_SOF_AMD_REMBRANDT) +=snd-sof-amd-rembrandt.o +obj-$(CONFIG_SND_SOC_SOF_AMD_VANGOGH) +=snd-sof-amd-vangogh.o diff --git a/sound/soc/sof/amd/acp-dsp-offset.h b/sound/soc/sof/amd/acp-dsp-offset.h index 920155dee819..19ef2b4c46d6 100644 --- a/sound/soc/sof/amd/acp-dsp-offset.h +++ b/sound/soc/sof/amd/acp-dsp-offset.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2021 Advanced Micro Devices, Inc. All rights reserved. + * Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. All rights reserved. * * Author: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> */ @@ -49,27 +49,33 @@ #define ACP_CONTROL 0x1004 #define ACP3X_I2S_PIN_CONFIG 0x1400 +#define ACP5X_I2S_PIN_CONFIG 0x1400 #define ACP6X_I2S_PIN_CONFIG 0x1440 /* Registers offsets from ACP_PGFSM block */ #define ACP3X_PGFSM_BASE 0x141C +#define ACP5X_PGFSM_BASE 0x1424 #define ACP6X_PGFSM_BASE 0x1024 #define PGFSM_CONTROL_OFFSET 0x0 #define PGFSM_STATUS_OFFSET 0x4 #define ACP3X_CLKMUX_SEL 0x1424 +#define ACP5X_CLKMUX_SEL 0x142C #define ACP6X_CLKMUX_SEL 0x102C /* Registers from ACP_INTR block */ #define ACP3X_EXT_INTR_STAT 0x1808 +#define ACP5X_EXT_INTR_STAT 0x1808 #define ACP6X_EXT_INTR_STAT 0x1A0C #define ACP3X_DSP_SW_INTR_BASE 0x1814 +#define ACP5X_DSP_SW_INTR_BASE 0x1814 #define ACP6X_DSP_SW_INTR_BASE 0x1808 #define DSP_SW_INTR_CNTL_OFFSET 0x0 #define DSP_SW_INTR_STAT_OFFSET 0x4 #define DSP_SW_INTR_TRIG_OFFSET 0x8 #define ACP_ERROR_STATUS 0x18C4 #define ACP3X_AXI2DAGB_SEM_0 0x1880 +#define ACP5X_AXI2DAGB_SEM_0 0x1884 #define ACP6X_AXI2DAGB_SEM_0 0x1874 /* Registers from ACP_SHA block */ diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 72fa0af971f0..73c3e1373d42 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2021 Advanced Micro Devices, Inc. All rights reserved. + * Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. All rights reserved. * * Author: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> */ @@ -32,6 +32,7 @@ #define ACP_DSP_INTR_EN_MASK 0x00000001 #define ACP3X_SRAM_PTE_OFFSET 0x02050000 +#define ACP5X_SRAM_PTE_OFFSET 0x02050000 #define ACP6X_SRAM_PTE_OFFSET 0x03800000 #define PAGE_SIZE_4K_ENABLE 0x2 #define ACP_PAGE_SIZE 0x1000 @@ -56,9 +57,11 @@ #define ACP_DSP_TO_HOST_IRQ 0x04 #define ACP_RN_PCI_ID 0x01 +#define ACP_VANGOGH_PCI_ID 0x50 #define ACP_RMB_PCI_ID 0x6F #define HOST_BRIDGE_CZN 0x1630 +#define HOST_BRIDGE_VGH 0x1645 #define HOST_BRIDGE_RMB 0x14B5 #define ACP_SHA_STAT 0x8000 #define ACP_PSP_TIMEOUT_US 1000000 @@ -163,6 +166,7 @@ struct acp_dsp_stream { struct sof_amd_acp_desc { unsigned int rev; + const char *name; unsigned int host_bridge_id; u32 pgfsm_base; u32 ext_intr_stat; @@ -253,6 +257,8 @@ extern struct snd_sof_dsp_ops sof_acp_common_ops; extern struct snd_sof_dsp_ops sof_renoir_ops; int sof_renoir_ops_init(struct snd_sof_dev *sdev); +extern struct snd_sof_dsp_ops sof_vangogh_ops; +int sof_vangogh_ops_init(struct snd_sof_dev *sdev); extern struct snd_sof_dsp_ops sof_rembrandt_ops; int sof_rembrandt_ops_init(struct snd_sof_dev *sdev); @@ -282,4 +288,5 @@ static inline const struct sof_amd_acp_desc *get_chip_info(struct snd_sof_pdata int acp_probes_register(struct snd_sof_dev *sdev); void acp_probes_unregister(struct snd_sof_dev *sdev); +extern struct snd_soc_acpi_mach snd_soc_acpi_amd_vangogh_sof_machines[]; #endif diff --git a/sound/soc/sof/amd/pci-vangogh.c b/sound/soc/sof/amd/pci-vangogh.c new file mode 100644 index 000000000000..d8be42fbcb6d --- /dev/null +++ b/sound/soc/sof/amd/pci-vangogh.c @@ -0,0 +1,105 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2023 Advanced Micro Devices, Inc. All rights reserved. +// +// Authors: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> + +/*. + * PCI interface for Vangogh ACP device + */ + +#include <linux/module.h> +#include <linux/pci.h> +#include <linux/platform_device.h> +#include <sound/sof.h> +#include <sound/soc-acpi.h> + +#include "../ops.h" +#include "../sof-pci-dev.h" +#include "../../amd/mach-config.h" +#include "acp.h" +#include "acp-dsp-offset.h" + +#define ACP5X_FUTURE_REG_ACLK_0 0x1864 + +static const struct sof_amd_acp_desc vangogh_chip_info = { + .rev = 5, + .name = "vangogh", + .host_bridge_id = HOST_BRIDGE_VGH, + .pgfsm_base = ACP5X_PGFSM_BASE, + .ext_intr_stat = ACP5X_EXT_INTR_STAT, + .dsp_intr_base = ACP5X_DSP_SW_INTR_BASE, + .sram_pte_offset = ACP5X_SRAM_PTE_OFFSET, + .hw_semaphore_offset = ACP5X_AXI2DAGB_SEM_0, + .acp_clkmux_sel = ACP5X_CLKMUX_SEL, + .probe_reg_offset = ACP5X_FUTURE_REG_ACLK_0, +}; + +static const struct sof_dev_desc vangogh_desc = { + .machines = snd_soc_acpi_amd_vangogh_sof_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .chip_info = &vangogh_chip_info, + .ipc_supported_mask = BIT(SOF_IPC), + .ipc_default = SOF_IPC, + .default_fw_path = { + [SOF_IPC] = "amd/sof", + }, + .default_tplg_path = { + [SOF_IPC] = "amd/sof-tplg", + }, + .default_fw_filename = { + [SOF_IPC] = "sof-vangogh.ri", + }, + .nocodec_tplg_filename = "sof-acp.tplg", + .ops = &sof_vangogh_ops, + .ops_init = sof_vangogh_ops_init, +}; + +static int acp_pci_vgh_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) +{ + unsigned int flag; + + if (pci->revision != ACP_VANGOGH_PCI_ID) + return -ENODEV; + + flag = snd_amd_acp_find_config(pci); + if (flag != FLAG_AMD_SOF && flag != FLAG_AMD_SOF_ONLY_DMIC) + return -ENODEV; + + return sof_pci_probe(pci, pci_id); +}; + +static void acp_pci_vgh_remove(struct pci_dev *pci) +{ + sof_pci_remove(pci); +} + +/* PCI IDs */ +static const struct pci_device_id vgh_pci_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_AMD, ACP_PCI_DEV_ID), + .driver_data = (unsigned long)&vangogh_desc}, + { 0, } +}; +MODULE_DEVICE_TABLE(pci, vgh_pci_ids); + +/* pci_driver definition */ +static struct pci_driver snd_sof_pci_amd_vgh_driver = { + .name = KBUILD_MODNAME, + .id_table = vgh_pci_ids, + .probe = acp_pci_vgh_probe, + .remove = acp_pci_vgh_remove, + .driver = { + .pm = &sof_pci_pm, + }, +}; +module_pci_driver(snd_sof_pci_amd_vgh_driver); + +MODULE_LICENSE("Dual BSD/GPL"); +MODULE_IMPORT_NS(SND_SOC_SOF_AMD_COMMON); +MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/amd/vangogh.c b/sound/soc/sof/amd/vangogh.c new file mode 100644 index 000000000000..f3f6bd790788 --- /dev/null +++ b/sound/soc/sof/amd/vangogh.c @@ -0,0 +1,156 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2023 Advanced Micro Devices, Inc. +// +// Authors: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> + +/* + * Hardware interface for Audio DSP on Vangogh platform + */ + +#include <linux/platform_device.h> +#include <linux/module.h> + +#include "../ops.h" +#include "../sof-audio.h" +#include "acp.h" +#include "acp-dsp-offset.h" + +#define I2S_HS_INSTANCE 0 +#define I2S_BT_INSTANCE 1 +#define I2S_SP_INSTANCE 2 +#define PDM_DMIC_INSTANCE 3 +#define I2S_HS_VIRTUAL_INSTANCE 4 + +static struct snd_soc_dai_driver vangogh_sof_dai[] = { + [I2S_HS_INSTANCE] = { + .id = I2S_HS_INSTANCE, + .name = "acp-sof-hs", + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + /* Supporting only stereo for I2S HS controller capture */ + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + }, + + [I2S_BT_INSTANCE] = { + .id = I2S_BT_INSTANCE, + .name = "acp-sof-bt", + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + /* Supporting only stereo for I2S BT controller capture */ + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + }, + + [I2S_SP_INSTANCE] = { + .id = I2S_SP_INSTANCE, + .name = "acp-sof-sp", + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + /* Supporting only stereo for I2S SP controller capture */ + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + }, + + [PDM_DMIC_INSTANCE] = { + .id = PDM_DMIC_INSTANCE, + .name = "acp-sof-dmic", + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 4, + .rate_min = 8000, + .rate_max = 48000, + }, + }, + + [I2S_HS_VIRTUAL_INSTANCE] = { + .id = I2S_HS_VIRTUAL_INSTANCE, + .name = "acp-sof-hs-virtual", + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + /* Supporting only stereo for I2S HS-Virtual controller capture */ + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + }, +}; + +/* Vangogh ops */ +struct snd_sof_dsp_ops sof_vangogh_ops; +EXPORT_SYMBOL_NS(sof_vangogh_ops, SND_SOC_SOF_AMD_COMMON); + +int sof_vangogh_ops_init(struct snd_sof_dev *sdev) +{ + /* common defaults */ + memcpy(&sof_vangogh_ops, &sof_acp_common_ops, sizeof(struct snd_sof_dsp_ops)); + + sof_vangogh_ops.drv = vangogh_sof_dai; + sof_vangogh_ops.num_drv = ARRAY_SIZE(vangogh_sof_dai); + + return 0; +} + +MODULE_IMPORT_NS(SND_SOC_SOF_AMD_COMMON); +MODULE_DESCRIPTION("VANGOGH SOF Driver"); +MODULE_LICENSE("Dual BSD/GPL"); -- 2.25.1

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[PATCH] sound/soc/codecs/rt5640.c: fix typos
by Senhong Liu 22 Aug '23

22 Aug '23

I noticed typos and i fixed them. Signed-off-by: Senhong Liu <liusenhong2022(a)email.szu.edu.cn> --- sound/soc/codecs/rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 8920726c38e8..15e1a62b9e57 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2570,7 +2570,7 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5640", rt5640); if (ret) { - dev_warn(component->dev, "Failed to reguest IRQ %d: %d\n", rt5640->irq, ret); + dev_warn(component->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret); rt5640_disable_jack_detect(component); return; } @@ -2625,7 +2625,7 @@ static void rt5640_enable_hda_jack_detect( NULL, rt5640_irq, IRQF_TRIGGER_RISING | IRQF_ONESHOT, "rt5640", rt5640); if (ret) { - dev_warn(component->dev, "Failed to reguest IRQ %d: %d\n", rt5640->irq, ret); + dev_warn(component->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret); rt5640->irq = -ENXIO; return; } -- 2.25.1

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[PATCH 0/4] ASoC: SOF: ipc4-topology: Fixes for pipelines with SRC
by Peter Ujfalusi 22 Aug '23

22 Aug '23

Hi, the SRC component in a pipeline provides flexibility on the sampling rate which was not handled previously. This series will improve the kernel side with the needed logic to be able to deal with the SRC type of components in pipelines. Regards, Peter --- Ranjani Sridharan (4): ASoC: SOF: ipc4-topology: Modify pipeline params based on SRC output format ASoC: SOF: ipc4-topology: Fix the output reference params for SRC ASoC: SOF: ipc4-topology: Fix pipeline params at the output of copier ASoC: SOF: ipc4-topology: Modify the reference output valid_bits for copier sound/soc/sof/ipc4-topology.c | 134 ++++++++++++++++++++++------------ 1 file changed, 89 insertions(+), 45 deletions(-) -- 2.41.0

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[PATCH] soundwire: intel_ace2x: add DAI hw_params/prepare/hw_free callbacks
by Bard Liao 22 Aug '23

22 Aug '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The code is fork-lifted from intel.c and is mostly similar *except* for the SHIM configuration which cannot be done here with the introduction of HDAudio Extended links. The ACE2.x SOF side also requires the hw_free and trigger callbacks to be implemented for HDaudio DMA management Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/intel_ace2x.c | 283 ++++++++++++++++++++++++++++++++ 1 file changed, 283 insertions(+) diff --git a/drivers/soundwire/intel_ace2x.c b/drivers/soundwire/intel_ace2x.c index 1be0bea5f40f..a9d25ae0b73f 100644 --- a/drivers/soundwire/intel_ace2x.c +++ b/drivers/soundwire/intel_ace2x.c @@ -10,6 +10,7 @@ #include <linux/soundwire/sdw_registers.h> #include <linux/soundwire/sdw.h> #include <linux/soundwire/sdw_intel.h> +#include <sound/pcm_params.h> #include <sound/hda-mlink.h> #include "cadence_master.h" #include "bus.h" @@ -191,10 +192,292 @@ static bool intel_check_cmdsync_unlocked(struct sdw_intel *sdw) return hdac_bus_eml_sdw_check_cmdsync_unlocked(sdw->link_res->hbus); } +/* DAI callbacks */ +static int intel_params_stream(struct sdw_intel *sdw, + struct snd_pcm_substream *substream, + struct snd_soc_dai *dai, + struct snd_pcm_hw_params *hw_params, + int link_id, int alh_stream_id) +{ + struct sdw_intel_link_res *res = sdw->link_res; + struct sdw_intel_stream_params_data params_data; + + params_data.substream = substream; + params_data.dai = dai; + params_data.hw_params = hw_params; + params_data.link_id = link_id; + params_data.alh_stream_id = alh_stream_id; + + if (res->ops && res->ops->params_stream && res->dev) + return res->ops->params_stream(res->dev, + &params_data); + return -EIO; +} + +static int intel_free_stream(struct sdw_intel *sdw, + struct snd_pcm_substream *substream, + struct snd_soc_dai *dai, + int link_id) + +{ + struct sdw_intel_link_res *res = sdw->link_res; + struct sdw_intel_stream_free_data free_data; + + free_data.substream = substream; + free_data.dai = dai; + free_data.link_id = link_id; + + if (res->ops && res->ops->free_stream && res->dev) + return res->ops->free_stream(res->dev, + &free_data); + + return 0; +} + /* * DAI operations */ +static int intel_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); + struct sdw_intel *sdw = cdns_to_intel(cdns); + struct sdw_cdns_dai_runtime *dai_runtime; + struct sdw_cdns_pdi *pdi; + struct sdw_stream_config sconfig; + struct sdw_port_config *pconfig; + int ch, dir; + int ret; + + dai_runtime = cdns->dai_runtime_array[dai->id]; + if (!dai_runtime) + return -EIO; + + ch = params_channels(params); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + dir = SDW_DATA_DIR_RX; + else + dir = SDW_DATA_DIR_TX; + + pdi = sdw_cdns_alloc_pdi(cdns, &cdns->pcm, ch, dir, dai->id); + + if (!pdi) { + ret = -EINVAL; + goto error; + } + + /* the SHIM will be configured in the callback functions */ + + sdw_cdns_config_stream(cdns, ch, dir, pdi); + + /* store pdi and state, may be needed in prepare step */ + dai_runtime->paused = false; + dai_runtime->suspended = false; + dai_runtime->pdi = pdi; + + /* Inform DSP about PDI stream number */ + ret = intel_params_stream(sdw, substream, dai, params, + sdw->instance, + pdi->intel_alh_id); + if (ret) + goto error; + + sconfig.direction = dir; + sconfig.ch_count = ch; + sconfig.frame_rate = params_rate(params); + sconfig.type = dai_runtime->stream_type; + + sconfig.bps = snd_pcm_format_width(params_format(params)); + + /* Port configuration */ + pconfig = kzalloc(sizeof(*pconfig), GFP_KERNEL); + if (!pconfig) { + ret = -ENOMEM; + goto error; + } + + pconfig->num = pdi->num; + pconfig->ch_mask = (1 << ch) - 1; + + ret = sdw_stream_add_master(&cdns->bus, &sconfig, + pconfig, 1, dai_runtime->stream); + if (ret) + dev_err(cdns->dev, "add master to stream failed:%d\n", ret); + + kfree(pconfig); +error: + return ret; +} + +static int intel_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); + struct sdw_intel *sdw = cdns_to_intel(cdns); + struct sdw_cdns_dai_runtime *dai_runtime; + int ch, dir; + int ret = 0; + + dai_runtime = cdns->dai_runtime_array[dai->id]; + if (!dai_runtime) { + dev_err(dai->dev, "failed to get dai runtime in %s\n", + __func__); + return -EIO; + } + + if (dai_runtime->suspended) { + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_pcm_hw_params *hw_params; + + hw_params = &rtd->dpcm[substream->stream].hw_params; + + dai_runtime->suspended = false; + + /* + * .prepare() is called after system resume, where we + * need to reinitialize the SHIM/ALH/Cadence IP. + * .prepare() is also called to deal with underflows, + * but in those cases we cannot touch ALH/SHIM + * registers + */ + + /* configure stream */ + ch = params_channels(hw_params); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + dir = SDW_DATA_DIR_RX; + else + dir = SDW_DATA_DIR_TX; + + /* the SHIM will be configured in the callback functions */ + + sdw_cdns_config_stream(cdns, ch, dir, dai_runtime->pdi); + + /* Inform DSP about PDI stream number */ + ret = intel_params_stream(sdw, substream, dai, + hw_params, + sdw->instance, + dai_runtime->pdi->intel_alh_id); + } + + return ret; +} + +static int +intel_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); + struct sdw_intel *sdw = cdns_to_intel(cdns); + struct sdw_cdns_dai_runtime *dai_runtime; + int ret; + + dai_runtime = cdns->dai_runtime_array[dai->id]; + if (!dai_runtime) + return -EIO; + + /* + * The sdw stream state will transition to RELEASED when stream-> + * master_list is empty. So the stream state will transition to + * DEPREPARED for the first cpu-dai and to RELEASED for the last + * cpu-dai. + */ + ret = sdw_stream_remove_master(&cdns->bus, dai_runtime->stream); + if (ret < 0) { + dev_err(dai->dev, "remove master from stream %s failed: %d\n", + dai_runtime->stream->name, ret); + return ret; + } + + ret = intel_free_stream(sdw, substream, dai, sdw->instance); + if (ret < 0) { + dev_err(dai->dev, "intel_free_stream: failed %d\n", ret); + return ret; + } + + dai_runtime->pdi = NULL; + + return 0; +} + +static int intel_pcm_set_sdw_stream(struct snd_soc_dai *dai, + void *stream, int direction) +{ + return cdns_set_sdw_stream(dai, stream, direction); +} + +static void *intel_get_sdw_stream(struct snd_soc_dai *dai, + int direction) +{ + struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); + struct sdw_cdns_dai_runtime *dai_runtime; + + dai_runtime = cdns->dai_runtime_array[dai->id]; + if (!dai_runtime) + return ERR_PTR(-EINVAL); + + return dai_runtime->stream; +} + +static int intel_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) +{ + struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); + struct sdw_intel *sdw = cdns_to_intel(cdns); + struct sdw_intel_link_res *res = sdw->link_res; + struct sdw_cdns_dai_runtime *dai_runtime; + int ret = 0; + + /* + * The .trigger callback is used to program HDaudio DMA and send required IPC to audio + * firmware. + */ + if (res->ops && res->ops->trigger) { + ret = res->ops->trigger(substream, cmd, dai); + if (ret < 0) + return ret; + } + + dai_runtime = cdns->dai_runtime_array[dai->id]; + if (!dai_runtime) { + dev_err(dai->dev, "failed to get dai runtime in %s\n", + __func__); + return -EIO; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_SUSPEND: + + /* + * The .prepare callback is used to deal with xruns and resume operations. + * In the case of xruns, the DMAs and SHIM registers cannot be touched, + * but for resume operations the DMAs and SHIM registers need to be initialized. + * the .trigger callback is used to track the suspend case only. + */ + + dai_runtime->suspended = true; + + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dai_runtime->paused = true; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + dai_runtime->paused = false; + break; + default: + break; + } + + return ret; +} + static const struct snd_soc_dai_ops intel_pcm_dai_ops = { + .hw_params = intel_hw_params, + .prepare = intel_prepare, + .hw_free = intel_hw_free, + .trigger = intel_trigger, + .set_stream = intel_pcm_set_sdw_stream, + .get_stream = intel_get_sdw_stream, }; static const struct snd_soc_component_driver dai_component = { -- 2.25.1

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[PATCH 1/2] ASoC: dt-bindings: nau8821: Add single-ended input feature
by Seven Lee 22 Aug '23

22 Aug '23

Add input with single-ended control. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nuvoton,nau8821.yaml | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml index fc2f4ce4db88..3e54abd4ca74 100644 --- a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -89,6 +89,12 @@ properties: $ref: /schemas/types.yaml#/definitions/uint32 default: 3072000 + nuvoton,left-input-single-end: + description: Enable left input with single-ended settings if set. + For the headset mic application, the single-ended control is + just limited to the left adc for design demand. + type: boolean + '#sound-dai-cells': const: 0 @@ -114,6 +120,7 @@ examples: nuvoton,jkdet-pull-enable; nuvoton,jkdet-pull-up; nuvoton,key-enable; + nuvoton,left-input-single-end; nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; nuvoton,micbias-voltage = <6>; nuvoton,vref-impedance = <2>; -- 2.25.1

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Add a config for the Allen & Heath Zedi 10 mixer.
by GitHub pull_request - opened 21 Aug '23

21 Aug '23

alsa-project/alsa-ucm-conf pull request #341 was opened from pololevu: [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12402535… The mixer has an embedded 4x4 USB interface, this config surfaces the mixer channels as 4 mono inputs, 2 stereo inputs and 2 stereo outputs. Config based on the Behringer Flow8 config. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/341 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/341.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: linux-next: Tree for Aug 21 (sound/pci/hda/tas2781_hda_i2c.o)
by Randy Dunlap 21 Aug '23

21 Aug '23

On 8/20/23 23:46, Stephen Rothwell wrote: > Hi all, > > Changes since 20230818: on x86_64: ld: sound/pci/hda/tas2781_hda_i2c.o: in function `tas2781_save_calibration.isra.0': tas2781_hda_i2c.c:(.text+0x1075): undefined reference to `efi' ld: tas2781_hda_i2c.c:(.text+0x108b): undefined reference to `efi' ld: tas2781_hda_i2c.c:(.text+0x119b): undefined reference to `efi' ld: tas2781_hda_i2c.c:(.text+0x11b1): undefined reference to `efi' Full randconfig file is attached. -- ~Randy

1 0

[PATCH 0/3] ASoC: wm8782: Allow higher audio rates
by John Watts 21 Aug '23

21 Aug '23

The wm8782 supports higher audio rates than just 48kHz. This is configured by setting the FSAMPEN pin on the codec chip. This patch series introduces the 'max-rate' device tree property to allow indicating which rates are fully supported by the codec without accidentally overclocking. John Watts (3): ASoC: wm8782: Handle maximum audio rate at runtime ASoC: wm8782: Use max-rate device tree property ASoC: dt-bindings: wlf,wm8782: Add max-rate property .../devicetree/bindings/sound/wm8782.txt | 2 + sound/soc/codecs/wm8782.c | 49 ++++++++++++++----- 2 files changed, 39 insertions(+), 12 deletions(-) -- 2.41.0

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[PATCH] [6.4.y] ASoC: SOF: intel: hda: Clean up link DMA for IPC3 during stop
by Jaroslav Kysela 21 Aug '23

21 Aug '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> commit 90219f1bd273055f1dc1d7bdc0965755b992c045 upstream. With IPC3, we reset hw_params during the stop trigger, so we should also clean up the link DMA during the stop trigger. Cc: <stable(a)vger.kernel.org> # 6.4.x Fixes: 1bf83fa6654c ("ASoC: SOF: Intel: hda-dai: Do not perform DMA cleanup during stop") Closes: https://github.com/thesofproject/linux/issues/4455 Closes: https://github.com/thesofproject/linux/issues/4482 Closes: https://bugzilla.kernel.org/show_bug.cgi?id=217673 Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20230808110627.32375-1-peter.ujfalusi@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Note that many recent Intel based laptops are affected. Added missing code for 6.4 kernels to keep the fix simple not depending on the other changes. This commit is present in 6.5 tree already. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/soc/sof/intel/hda-dai-ops.c | 13 ++++++++++++- sound/soc/sof/intel/hda-dai.c | 8 ++++---- sound/soc/sof/intel/hda.h | 2 ++ 3 files changed, 18 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index 4b39cecacd68..5938046f46b2 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -289,16 +289,27 @@ static const struct hda_dai_widget_dma_ops hda_ipc4_chain_dma_ops = { static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, struct snd_pcm_substream *substream, int cmd) { + struct hdac_ext_stream *hext_stream = hda_get_hext_stream(sdev, cpu_dai, substream); struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); switch (cmd) { case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: { struct snd_sof_dai_config_data data = { 0 }; + int ret; data.dai_data = DMA_CHAN_INVALID; - return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_FREE, &data); + ret = hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_FREE, &data); + if (ret < 0) + return ret; + + if (cmd == SNDRV_PCM_TRIGGER_STOP) + return hda_link_dma_cleanup(substream, hext_stream, cpu_dai, codec_dai); + + break; } case SNDRV_PCM_TRIGGER_PAUSE_PUSH: return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_PAUSE, NULL); diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 44a5d94c5050..8a76320c3b99 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -91,10 +91,10 @@ hda_dai_get_ops(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai return sdai->platform_private; } -static int hda_link_dma_cleanup(struct snd_pcm_substream *substream, - struct hdac_ext_stream *hext_stream, - struct snd_soc_dai *cpu_dai, - struct snd_soc_dai *codec_dai) +int hda_link_dma_cleanup(struct snd_pcm_substream *substream, + struct hdac_ext_stream *hext_stream, + struct snd_soc_dai *cpu_dai, + struct snd_soc_dai *codec_dai) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(cpu_dai->component); const struct hda_dai_widget_dma_ops *ops = hda_dai_get_ops(substream, cpu_dai); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index c4befacde23e..94c738eae751 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -942,5 +942,7 @@ const struct hda_dai_widget_dma_ops * hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget); int hda_dai_config(struct snd_soc_dapm_widget *w, unsigned int flags, struct snd_sof_dai_config_data *data); +int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream, + struct snd_soc_dai *cpu_dai, struct snd_soc_dai *codec_dai); #endif -- 2.41.0

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