February 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: qcom: Fix uninitialized pointer dmactl
by Colin Ian King 21 Feb '24

21 Feb '24

In the case where __lpass_get_dmactl_handle is called and the driver id dai_id is invalid the pointer dmactl is not being assigned a value, and dmactl contains a garbage value since it has not been initialized and so the null check may not work. Fix this to initialize dmactl to NULL. One could argue that modern compilers will set this to zero, but it is useful to keep this initialized as per the same way in functions __lpass_platform_codec_intf_init and lpass_cdc_dma_daiops_hw_params. Cleans up clang scan build warning: sound/soc/qcom/lpass-cdc-dma.c:275:7: warning: Branch condition evaluates to a garbage value [core.uninitialized.Branch] Fixes: b81af585ea54 ("ASoC: qcom: Add lpass CPU driver for codec dma control") Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/qcom/lpass-cdc-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cdc-dma.c b/sound/soc/qcom/lpass-cdc-dma.c index 48b03e60e3a3..8106c586f68a 100644 --- a/sound/soc/qcom/lpass-cdc-dma.c +++ b/sound/soc/qcom/lpass-cdc-dma.c @@ -259,7 +259,7 @@ static int lpass_cdc_dma_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *soc_runtime = snd_soc_substream_to_rtd(substream); - struct lpaif_dmactl *dmactl; + struct lpaif_dmactl *dmactl = NULL; int ret = 0, id; switch (cmd) { -- 2.39.2

2 1

[PATCH 00/10] ASoC: Intel: avs: Fixes and new platforms support
by Cezary Rojewski 21 Feb '24

21 Feb '24

The avs-driver continues to be utilized on more recent Intel machines. As TGL-based (cAVS 2.5) e.g.: RPL, inherit most of the functionality from previous platforms: SKL <- APL <- CNL <- ICL <- TGL rather than putting everything into a single file, the platform-specific bits are split into cnl/icl/tgl.c files instead. Makes the division clear and code easier to maintain. Layout of the patchset: First are two changes combined together address the sound-clipping problem, present when only one stream is running - specifically one CAPTURE stream. Follow up is naming-scheme adjustment for some of the existing functions what improves code incohesiveness. As existing IPC/IRQ code operates solely on cAVS 1.5 architecture, it needs no abstraction. The situation changes when newer platforms come into the picture. Thus the next two patches abstract the existing IPC/IRQ handlers so that majority of the common code can be re-used. The ICCMAX change stands out a bit - the AudioDSP firmware loading procedure differs on ICL-based platforms (and onwards) and having a separate commit makes the situation clear to the developers who are going to support the solution from LTS perspective. For that reason I decided not to merge it into the commit introducing the icl.c file. Cezary Rojewski (10): ASoC: Intel: avs: L1SEN reference counted ASoC: Intel: avs: Fix sound clipping in single capture scenario ASoC: Intel: avs: Prefix SKL/APL-specific members ASoC: Intel: avs: Abstract IPC handling ASoC: Intel: avs: Abstract IRQ handling ASoC: Intel: avs: CNL-based platforms support ASoC: Intel: avs: ICL-based platforms support ASoC: Intel: avs: TGL-based platforms support ASoC: Intel: avs: ICCMAX recommendations for ICL+ platforms ASoC: Intel: avs: Populate board selection with new I2S entries include/sound/hda_register.h | 2 + sound/soc/intel/avs/Makefile | 2 +- sound/soc/intel/avs/apl.c | 58 ++++---- sound/soc/intel/avs/avs.h | 66 ++++++--- sound/soc/intel/avs/board_selection.c | 85 +++++++++++ sound/soc/intel/avs/cnl.c | 61 ++++++++ sound/soc/intel/avs/core.c | 160 ++++++++++++++++++--- sound/soc/intel/avs/icl.c | 197 ++++++++++++++++++++++++++ sound/soc/intel/avs/ipc.c | 66 +++------ sound/soc/intel/avs/loader.c | 2 +- sound/soc/intel/avs/messages.c | 1 + sound/soc/intel/avs/messages.h | 38 ++++- sound/soc/intel/avs/pcm.c | 77 +++++++++- sound/soc/intel/avs/registers.h | 21 ++- sound/soc/intel/avs/skl.c | 59 +++++--- sound/soc/intel/avs/tgl.c | 54 +++++++ 16 files changed, 807 insertions(+), 142 deletions(-) create mode 100644 sound/soc/intel/avs/cnl.c create mode 100644 sound/soc/intel/avs/icl.c create mode 100644 sound/soc/intel/avs/tgl.c -- 2.25.1

2 11

[RESEND PATCH v6 v6 1/2] ASoC: dt-bindings: qcom,wsa8840: Add reset-gpios for shared line
by Krzysztof Kozlowski 21 Feb '24

21 Feb '24

On newer Qualcomm platforms, like X1E80100-CRD, the WSA884x speakers share SD_N GPIOs between two speakers, thus a coordinated assertion is needed. Linux supports handling shared GPIO lines through "reset-gpios" property, thus allow specifying either powerdown or reset GPIOs (these are the same). Cc: Bartosz Golaszewski <brgl(a)bgdev.pl> Cc: Sean Anderson <sean.anderson(a)seco.com> Acked-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Resending because reset part of the patchset was applied, so this can go independently via ASoC. https://lore.kernel.org/all/38fda6619769da7240517982adfe734cb653ff5e.camel@… Changes in v6 ============= 1. Add Rb/Ack tags. For full changelog, see: https://lore.kernel.org/all/20240129115216.96479-1-krzysztof.kozlowski@lina… --- .../devicetree/bindings/sound/qcom,wsa8840.yaml | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml index d717017b0fdb..22798d22d981 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml @@ -28,6 +28,10 @@ properties: description: Powerdown/Shutdown line to use (pin SD_N) maxItems: 1 + reset-gpios: + description: Powerdown/Shutdown line to use (pin SD_N) + maxItems: 1 + '#sound-dai-cells': const: 0 @@ -37,11 +41,16 @@ properties: required: - compatible - reg - - powerdown-gpios - '#sound-dai-cells' - vdd-1p8-supply - vdd-io-supply +oneOf: + - required: + - powerdown-gpios + - required: + - reset-gpios + unevaluatedProperties: false examples: -- 2.34.1

1 1

[PATCH] slimbus: core: Remove usage of the deprecated ida_simple_xx() API
by Christophe JAILLET 21 Feb '24

21 Feb '24

ida_alloc() and ida_free() should be preferred to the deprecated ida_simple_get() and ida_simple_remove(). Note that the upper limit of ida_simple_get() is exclusive, but the one of ida_alloc_range() is inclusive. So change this change allows one more device. Previously address 0xFE was never used. Fixes: 46a2bb5a7f7e ("slimbus: core: Add slim controllers support") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- /!\ Untested /!\ The change of behavior is motivated by the comment in drivers/slimbus/slimbus.h stating that /* Manager's logical address is set to 0xFF per spec */, so all other values, including 0xFE should be valid, IMHO. So considering this as a bug-fix, I added a Fixes tag. --- drivers/slimbus/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/slimbus/core.c b/drivers/slimbus/core.c index d43873bb5fe6..01cbd4621981 100644 --- a/drivers/slimbus/core.c +++ b/drivers/slimbus/core.c @@ -436,8 +436,8 @@ static int slim_device_alloc_laddr(struct slim_device *sbdev, if (ret < 0) goto err; } else if (report_present) { - ret = ida_simple_get(&ctrl->laddr_ida, - 0, SLIM_LA_MANAGER - 1, GFP_KERNEL); + ret = ida_alloc_max(&ctrl->laddr_ida, + SLIM_LA_MANAGER - 1, GFP_KERNEL); if (ret < 0) goto err; -- 2.34.1

2 2

[PATCH] slimbus: core: make slimbus_bus const
by Ricardo B. Marliere 21 Feb '24

21 Feb '24

Since commit d492cc2573a0 ("driver core: device.h: make struct bus_type a const *"), the driver core can properly handle constant struct bus_type, move the slimbus_bus variable to be a constant structure as well, placing it into read-only memory which can not be modified at runtime. Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Suggested-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Ricardo B. Marliere <ricardo(a)marliere.net> --- This patch is based on https://git.kernel.org/pub/scm/linux/kernel/git/srini/slimbus.git/, should this tree be listed under "SERIAL LOW-POWER INTER-CHIP MEDIA BUS (SLIMbus)" in the MAINTAINERS file? --- drivers/slimbus/core.c | 2 +- include/linux/slimbus.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/slimbus/core.c b/drivers/slimbus/core.c index d43873bb5fe6..375218e02e28 100644 --- a/drivers/slimbus/core.c +++ b/drivers/slimbus/core.c @@ -100,7 +100,7 @@ static int slim_device_uevent(const struct device *dev, struct kobj_uevent_env * return add_uevent_var(env, "MODALIAS=slim:%s", dev_name(&sbdev->dev)); } -struct bus_type slimbus_bus = { +const struct bus_type slimbus_bus = { .name = "slimbus", .match = slim_device_match, .probe = slim_device_probe, diff --git a/include/linux/slimbus.h b/include/linux/slimbus.h index 12c9719b2a55..3042385b7b40 100644 --- a/include/linux/slimbus.h +++ b/include/linux/slimbus.h @@ -10,7 +10,7 @@ #include <linux/completion.h> #include <linux/mod_devicetable.h> -extern struct bus_type slimbus_bus; +extern const struct bus_type slimbus_bus; /** * struct slim_eaddr - Enumeration address for a SLIMbus device --- base-commit: 04b945e4cf81a12365f8207a4d34dbc81ba17413 change-id: 20240213-bus_cleanup-slimbus-e23b9e976fd2 Best regards, -- Ricardo B. Marliere <ricardo(a)marliere.net>

3 2

[PATCH] ASoC: cs35l56: Must clear HALO_STATE before issuing SYSTEM_RESET
by Richard Fitzgerald 21 Feb '24

21 Feb '24

The driver must write 0 to HALO_STATE before sending the SYSTEM_RESET command to the firmware. HALO_STATE is in DSP memory, which is preserved across a soft reset. The SYSTEM_RESET command does not change the value of HALO_STATE. There is period of time while the CS35L56 is resetting, before the firmware has started to boot, where a read of HALO_STATE will return the value it had before the SYSTEM_RESET. If the driver does not clear HALO_STATE, this would return BOOT_DONE status even though the firmware has not booted. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 8a731fd37f8b ("ASoC: cs35l56: Move utility functions to shared file") --- sound/soc/codecs/cs35l56-shared.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 995d979b6d87..cb4e83126b08 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -335,6 +335,7 @@ void cs35l56_wait_min_reset_pulse(void) EXPORT_SYMBOL_NS_GPL(cs35l56_wait_min_reset_pulse, SND_SOC_CS35L56_SHARED); static const struct reg_sequence cs35l56_system_reset_seq[] = { + REG_SEQ0(CS35L56_DSP1_HALO_STATE, 0), REG_SEQ0(CS35L56_DSP_VIRTUAL1_MBOX_1, CS35L56_MBOX_CMD_SYSTEM_RESET), }; -- 2.30.2

2 1

[PATCH] ASoC: cs42l42: Remove redundant delays in suspend().
by Vitaly Rodionov 21 Feb '24

21 Feb '24

This patch will remove redundant delay and minimise total suspend() function call time. Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- include/sound/cs42l42.h | 5 ++--- sound/soc/codecs/cs42l42.c | 1 - 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/include/sound/cs42l42.h b/include/sound/cs42l42.h index 3994e933db19..1bd8eee54f66 100644 --- a/include/sound/cs42l42.h +++ b/include/sound/cs42l42.h @@ -809,8 +809,7 @@ #define CS42L42_PLL_LOCK_TIMEOUT_US 1250 #define CS42L42_HP_ADC_EN_TIME_US 20000 #define CS42L42_PDN_DONE_POLL_US 1000 -#define CS42L42_PDN_DONE_TIMEOUT_US 200000 -#define CS42L42_PDN_DONE_TIME_MS 100 -#define CS42L42_FILT_DISCHARGE_TIME_MS 46 +#define CS42L42_PDN_DONE_TIMEOUT_US 235000 +#define CS42L42_PDN_DONE_TIME_MS 65 #endif /* __CS42L42_H */ diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 2d11c5125f73..60d366e53526 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -2195,7 +2195,6 @@ int cs42l42_suspend(struct device *dev) /* Discharge FILT+ */ regmap_update_bits(cs42l42->regmap, CS42L42_PWR_CTL2, CS42L42_DISCHARGE_FILT_MASK, CS42L42_DISCHARGE_FILT_MASK); - msleep(CS42L42_FILT_DISCHARGE_TIME_MS); regcache_cache_only(cs42l42->regmap, true); gpiod_set_value_cansleep(cs42l42->reset_gpio, 0); -- 2.40.1

2 1

[PATCH 1/5] ASoC: amd: ps: refactor acp device configuration read logic
by Vijendar Mukunda 21 Feb '24

21 Feb '24

Refactor acp device configuration read logic and use common function to scan SoundWire devices. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/Kconfig | 17 ++++ sound/soc/amd/ps/acp63.h | 11 +++ sound/soc/amd/ps/pci-ps.c | 176 +++++++++++--------------------------- 3 files changed, 78 insertions(+), 126 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 273688c05317..fa74635cee08 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -132,9 +132,26 @@ config SND_SOC_AMD_RPL_ACP6x Say m if you have such a device. If unsure select "N". +config SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE + tristate + select SOUNDWIRE_AMD if SND_SOC_AMD_SOUNDWIRE != n + select SND_AMD_SOUNDWIRE_ACPI if ACPI + +config SND_SOC_AMD_SOUNDWIRE + tristate "Support for SoundWire based AMD platforms" + default SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE + depends on SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE + depends on ACPI && SOUNDWIRE + depends on !(SOUNDWIRE=m && SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE=y) + help + This adds support for SoundWire for AMD platforms. + Say Y if you want to enable SoundWire links with SOF. + If unsure select "N". + config SND_SOC_AMD_PS tristate "AMD Audio Coprocessor-v6.3 Pink Sardine support" select SND_AMD_ACP_CONFIG + select SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE depends on X86 && PCI && ACPI help This option enables Audio Coprocessor i.e ACP v6.3 support on diff --git a/sound/soc/amd/ps/acp63.h b/sound/soc/amd/ps/acp63.h index 8b853b8d0219..123b9ade69d4 100644 --- a/sound/soc/amd/ps/acp63.h +++ b/sound/soc/amd/ps/acp63.h @@ -5,6 +5,7 @@ * Copyright (C) 2022, 2023 Advanced Micro Devices, Inc. All rights reserved. */ +#include <linux/soundwire/sdw_amd.h> #include <sound/acp63_chip_offset_byte.h> #define ACP_DEVICE_ID 0x15E2 @@ -263,6 +264,11 @@ struct sdw_dma_ring_buf_reg { * @sdw0_dev_index: SoundWire Manager-0 platform device index * @sdw1_dev_index: SoundWire Manager-1 platform device index * @sdw_dma_dev_index: SoundWire DMA controller platform device index + * @info: SoundWire AMD information found in ACPI tables + * @is_sdw_dev: flag set to true when any SoundWire manager instances are available + * @is_pdm_dev: flag set to true when ACP PDM controller exists + * @is_pdm_config: flat set to true when PDM configuration is selected from BIOS + * @is_sdw_config: flag set to true when SDW configuration is selected from BIOS * @sdw0-dma_intr_stat: DMA interrupt status array for SoundWire manager-SW0 instance * @sdw_dma_intr_stat: DMA interrupt status array for SoundWire manager-SW1 instance * @acp_reset: flag set to true when bus reset is applied across all @@ -282,6 +288,11 @@ struct acp63_dev_data { u16 sdw0_dev_index; u16 sdw1_dev_index; u16 sdw_dma_dev_index; + struct sdw_amd_acpi_info info; + bool is_sdw_dev; + bool is_pdm_dev; + bool is_pdm_config; + bool is_sdw_config; u16 sdw0_dma_intr_stat[ACP63_SDW0_DMA_MAX_STREAMS]; u16 sdw1_dma_intr_stat[ACP63_SDW1_DMA_MAX_STREAMS]; bool acp_reset; diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 5927eef04170..c97e418a88ce 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -237,122 +237,51 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) return IRQ_NONE; } -static int sdw_amd_scan_controller(struct device *dev) +#if IS_ENABLED(CONFIG_SND_SOC_AMD_SOUNDWIRE) +static int acp_scan_sdw_devices(struct device *dev, u64 addr) { + struct acpi_device *sdw_dev; struct acp63_dev_data *acp_data; - struct fwnode_handle *link; - char name[32]; - u32 sdw_manager_bitmap; - u8 count = 0; - u32 acp_sdw_power_mode = 0; - int index; - int ret; acp_data = dev_get_drvdata(dev); - /* - * Current implementation is based on MIPI DisCo 2.0 spec. - * Found controller, find links supported. - */ - ret = fwnode_property_read_u32_array((acp_data->sdw_fw_node), "mipi-sdw-manager-list", - &sdw_manager_bitmap, 1); - - if (ret) { - dev_dbg(dev, "Failed to read mipi-sdw-manager-list: %d\n", ret); - return -EINVAL; - } - count = hweight32(sdw_manager_bitmap); - /* Check count is within bounds */ - if (count > AMD_SDW_MAX_MANAGERS) { - dev_err(dev, "Manager count %d exceeds max %d\n", count, AMD_SDW_MAX_MANAGERS); - return -EINVAL; - } + if (!addr) + return -ENODEV; - if (!count) { - dev_dbg(dev, "No SoundWire Managers detected\n"); - return -EINVAL; - } - dev_dbg(dev, "ACPI reports %d SoundWire Manager devices\n", count); - acp_data->sdw_manager_count = count; - for (index = 0; index < count; index++) { - scnprintf(name, sizeof(name), "mipi-sdw-link-%d-subproperties", index); - link = fwnode_get_named_child_node(acp_data->sdw_fw_node, name); - if (!link) { - dev_err(dev, "Manager node %s not found\n", name); - return -EIO; - } + sdw_dev = acpi_find_child_device(ACPI_COMPANION(dev), addr, 0); + if (!sdw_dev) + return -ENODEV; - ret = fwnode_property_read_u32(link, "amd-sdw-power-mode", &acp_sdw_power_mode); - if (ret) - return ret; - /* - * when SoundWire configuration is selected from acp pin config, - * based on manager instances count, acp init/de-init sequence should be - * executed as part of PM ops only when Bus reset is applied for the active - * SoundWire manager instances. - */ - if (acp_sdw_power_mode != AMD_SDW_POWER_OFF_MODE) { - acp_data->acp_reset = false; - return 0; - } - } + acp_data->info.handle = sdw_dev->handle; + acp_data->info.count = AMD_SDW_MAX_MANAGERS; + return amd_sdw_scan_controller(&acp_data->info); +} +#else +static int acp_scan_sdw_devices(struct device *dev, u64 addr) +{ return 0; } +#endif -static int get_acp63_device_config(u32 config, struct pci_dev *pci, struct acp63_dev_data *acp_data) +static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *acp_data) { - struct acpi_device *dmic_dev; - struct acpi_device *sdw_dev; + struct acpi_device *pdm_dev; const union acpi_object *obj; + u32 config; bool is_dmic_dev = false; bool is_sdw_dev = false; int ret; - dmic_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), ACP63_DMIC_ADDR, 0); - if (dmic_dev) { - /* is_dmic_dev flag will be set when ACP PDM controller device exists */ - if (!acpi_dev_get_property(dmic_dev, "acp-audio-device-type", - ACPI_TYPE_INTEGER, &obj) && - obj->integer.value == ACP_DMIC_DEV) - is_dmic_dev = true; - } - - sdw_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), ACP63_SDW_ADDR, 0); - if (sdw_dev) { - acp_data->sdw_fw_node = acpi_fwnode_handle(sdw_dev); - ret = sdw_amd_scan_controller(&pci->dev); - /* is_sdw_dev flag will be set when SoundWire Manager device exists */ - if (!ret) - is_sdw_dev = true; - } - if (!is_dmic_dev && !is_sdw_dev) - return -ENODEV; - dev_dbg(&pci->dev, "Audio Mode %d\n", config); + config = readl(acp_data->acp63_base + ACP_PIN_CONFIG); switch (config) { case ACP_CONFIG_4: case ACP_CONFIG_5: case ACP_CONFIG_10: case ACP_CONFIG_11: - if (is_dmic_dev) { - acp_data->pdev_config = ACP63_PDM_DEV_CONFIG; - acp_data->pdev_count = ACP63_PDM_MODE_DEVS; - } + acp_data->is_pdm_config = true; break; case ACP_CONFIG_2: case ACP_CONFIG_3: - if (is_sdw_dev) { - switch (acp_data->sdw_manager_count) { - case 1: - acp_data->pdev_config = ACP63_SDW_DEV_CONFIG; - acp_data->pdev_count = ACP63_SDW0_MODE_DEVS; - break; - case 2: - acp_data->pdev_config = ACP63_SDW_DEV_CONFIG; - acp_data->pdev_count = ACP63_SDW0_SDW1_MODE_DEVS; - break; - default: - return -EINVAL; - } - } + acp_data->is_sdw_config = true; break; case ACP_CONFIG_6: case ACP_CONFIG_7: @@ -360,40 +289,36 @@ static int get_acp63_device_config(u32 config, struct pci_dev *pci, struct acp63 case ACP_CONFIG_8: case ACP_CONFIG_13: case ACP_CONFIG_14: - if (is_dmic_dev && is_sdw_dev) { - switch (acp_data->sdw_manager_count) { - case 1: - acp_data->pdev_config = ACP63_SDW_PDM_DEV_CONFIG; - acp_data->pdev_count = ACP63_SDW0_PDM_MODE_DEVS; - break; - case 2: - acp_data->pdev_config = ACP63_SDW_PDM_DEV_CONFIG; - acp_data->pdev_count = ACP63_SDW0_SDW1_PDM_MODE_DEVS; - break; - default: - return -EINVAL; - } - } else if (is_dmic_dev) { - acp_data->pdev_config = ACP63_PDM_DEV_CONFIG; - acp_data->pdev_count = ACP63_PDM_MODE_DEVS; - } else if (is_sdw_dev) { - switch (acp_data->sdw_manager_count) { - case 1: - acp_data->pdev_config = ACP63_SDW_DEV_CONFIG; - acp_data->pdev_count = ACP63_SDW0_MODE_DEVS; - break; - case 2: - acp_data->pdev_config = ACP63_SDW_DEV_CONFIG; - acp_data->pdev_count = ACP63_SDW0_SDW1_MODE_DEVS; - break; - default: - return -EINVAL; - } - } + acp_data->is_pdm_config = true; + acp_data->is_sdw_config = true; break; default: break; } + + if (acp_data->is_pdm_config) { + pdm_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), ACP63_DMIC_ADDR, 0); + if (pdm_dev) { + /* is_dmic_dev flag will be set when ACP PDM controller device exists */ + if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type", + ACPI_TYPE_INTEGER, &obj) && + obj->integer.value == ACP_DMIC_DEV) + is_dmic_dev = true; + } + } + + if (acp_data->is_sdw_config) { + ret = acp_scan_sdw_devices(&pci->dev, ACP63_SDW_ADDR); + if (!ret && acp_data->info.link_mask) + is_sdw_dev = true; + } + + acp_data->is_pdm_dev = is_dmic_dev; + acp_data->is_sdw_dev = is_sdw_dev; + if (!is_dmic_dev && !is_sdw_dev) { + dev_dbg(&pci->dev, "No PDM or SoundWire manager devices found\n"); + return -ENODEV; + } return 0; } @@ -576,7 +501,6 @@ static int snd_acp63_probe(struct pci_dev *pci, struct acp63_dev_data *adata; u32 addr; u32 irqflags, flag; - int val; int ret; irqflags = IRQF_SHARED; @@ -637,8 +561,7 @@ static int snd_acp63_probe(struct pci_dev *pci, dev_err(&pci->dev, "ACP PCI IRQ request failed\n"); goto de_init; } - val = readl(adata->acp63_base + ACP_PIN_CONFIG); - ret = get_acp63_device_config(val, pci, adata); + ret = get_acp63_device_config(pci, adata); /* ACP PCI driver probe should be continued even PDM or SoundWire Devices are not found */ if (ret) { dev_dbg(&pci->dev, "get acp device config failed:%d\n", ret); @@ -740,4 +663,5 @@ module_pci_driver(ps_acp63_driver); MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); MODULE_AUTHOR("Syed.SabaKareem(a)amd.com"); MODULE_DESCRIPTION("AMD ACP Pink Sardine PCI driver"); +MODULE_IMPORT_NS(SND_AMD_SOUNDWIRE_ACPI); MODULE_LICENSE("GPL v2"); -- 2.34.1

2 5

[PATCH v8] ASoc: tas2783: Add tas2783 codec driver
by Shenghao Ding 20 Feb '24

20 Feb '24

The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp speaker protection algorithm. The integrated speaker voltage and current sense provides for real-time monitoring of lodspeakers. The ASoC component provides the majority of the functionality of the device, all the audio functions. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v8: - simplify tasdevice_set_sdw_stream - fixed some Linux coding style - fixed the spelling mistakes - Select left/right channel based on unique id - a longer description has been added - remove unused crc8 in KCONFIG - Invert of RE here means reciprocal of RE - correct indentation - correct the wrong pow calibrated register - correct comment on RESET register - fixed usually lines don't end with an open parenthesis - fixed tasdevice_sdw_shutdown - use defines for function, entity, etc. for readable - remove unnecessary codec_lock in component_probe() - remove struct sdw_stream_data - remove unused macro definitions - use BIT(2) and BIT(1) instead of 0x2 and 0x1 - implement pm_runtime - reg_start is u8, reg_start + i only adds i byte, it should be reg_start + i * 4, because the calibration values - fixed total_sz in tas2783_calibration - add a define for TASDEVICE_REG(0, 0, 0x0a) - rework macro style in header file - Add select CRC32 - rewrite the mute and the return value handling - Add comments on bin file - Add link_id into the fw name to distinguish the same sdw id on different sdw id. - remove unnecessary mutex_lock - use set_active() in tasdevice_io_init and move all other parts in the probe. - move pm_runtime_get_noresume to the probe - fix the return value of putvol function - add put_noidle in tasdevice_sdw_remove - remove unused header file(of.h) - correct the style of the multi-line comment - Set the readable registers in tas2783_readable_register for debugfs - Use new PM macros - Use default put and get inline SOC_SINGLE_RANGE_TLV and SOC_SINGLE_SX_TLV - Use respective %pt to log the calibrated time - Remove unnecessary struct tm - Add trailing comma tasdevice_dapm_widgets and tasdevice_audio_map - Rewrite the return value handling in tasdevice_sdw_probe - use dev_num instead of id.unique_id - rework the log --- sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 844 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 103 ++++ 4 files changed, 965 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 59f9742e9ff4..6ba3fd0c0811 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -247,6 +247,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1847,6 +1848,21 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + depends on EFI + select REGMAP + select REGMAP_SOUNDWIRE + select CRC32 + select CRC32_SARWATE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + TAS2783 supports the MIPI SoundWire v1.2 compatible interface + for audio and control data. At the end of calibrated data appends + the crc data with crc32_sarwate for error detection. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f53baa2b9565..6afceb126ae3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -284,6 +284,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -664,6 +665,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(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 diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..ebb7102828c5 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,844 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Kevin Lu <kevin-lu(a)ti.com> +// Shenghao Ding <shenghao-ding(a)ti.com> +// + +#include <asm/unaligned.h> +#include <linux/acpi.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/pcm_params.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include <sound/tas2781-tlv.h> +#include "tas2783.h" + +static const unsigned int tas2783_cali_reg[] = { + TAS2783_CALIBRATION_RE, /* Resistance */ + TAS2783_CALIBRATION_RE_LOW, /* Low limitation of RE */ + TAS2783_CALIBRATION_INV_RE, /* Reciprocal of RE */ + TAS2783_CALIBRATION_POW, /* RMS Power */ + TAS2783_CALIBRATION_TLIMIT /* Temperature limitation */ +}; + +static const struct reg_default tas2783_reg_defaults[] = { + /* Default values for ROM mode. Activated. */ + { 0x8002, 0x1a }, /* AMP inactive. */ + { 0x8097, 0xc8 }, + { 0x80b5, 0x74 }, + { 0x8099, 0x20 }, + { 0xfe8d, 0x0d }, + { 0xfebe, 0x4a }, + { 0x8230, 0x00 }, + { 0x8231, 0x00 }, + { 0x8232, 0x00 }, + { 0x8233, 0x01 }, + { 0x8418, 0x00 }, /* Set volume to 0 dB. */ + { 0x8419, 0x00 }, + { 0x841a, 0x00 }, + { 0x841b, 0x00 }, + { 0x8428, 0x40 }, /* Unmute channel */ + { 0x8429, 0x00 }, + { 0x842a, 0x00 }, + { 0x842b, 0x00 }, + { 0x8548, 0x00 }, /* Set volume to 0 dB. */ + { 0x8549, 0x00 }, + { 0x854a, 0x00 }, + { 0x854b, 0x00 }, + { 0x8558, 0x40 }, /* Unmute channel */ + { 0x8559, 0x00 }, + { 0x855a, 0x00 }, + { 0x855b, 0x00 }, + { 0x800a, 0x3a }, /* Enable both channel */ + { 0x800e, 0x44 }, + { 0x800f, 0x40 }, + { 0x805c, 0x99 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_VOLUME, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0x807F: /* Page 0 */ + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + /* + * Only software reset register was volatiled. + * Bit is self clearing. + * 0b = Don't reset + * 1b = reset + */ + case 0x8001: + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x44ffffff, + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, dvc_tlv), + SOC_SINGLE_SX_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, amp_vol_tlv), +}; + +static void tas2783_apply_calib(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + struct regmap *map = tas_dev->regmap; + u8 *cali_start; + u16 dev_num; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + dev_num = clamp(tas_dev->sdw_peripheral->dev_num, 1, 4) - 1; + /* + * The area saving tas2783 calibrated data is specified by its + * dev_num. cali_start is the first address of current tas2783's + * calibrated data. + */ + cali_start = (u8 *)(cali_data + dev_num * sizeof(tas2783_cali_reg)); + for (int i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + ret = regmap_bulk_write(map, tas2783_cali_reg[i], + &cali_start[4 * i], 4); + if (ret) { + dev_err(tas_dev->dev, "Cali failed %x:%d\n", + tas2783_cali_reg[i], ret); + break; + } + } +} + +/* + * Load the calibration data, including speaker impedance, f0, etc. + * Calibration is done by the manufacturer in the factory. The calibration + * data are used by the algorithm for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * The DSP will work with default data values if calibrated data are + * missing or are invalid. + * Layout of calibrated Data in UEFI: + * Calibrated Data of Dev 0 (20 bytes) + * Calibrated Data of Dev 1 (20 bytes) + * Calibrated Data of Dev 2 (20 bytes) + * Calibrated Data of Dev 3 (20 bytes) + * CRC (4 bytes) + */ +static void tas2783_calibration(struct tasdevice_priv *tas_dev) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + unsigned int attr = 0, crc; + unsigned int *tmp_val; + efi_status_t status; + + tas_dev->cali_data.total_sz = 0; + + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_dev->cali_data.total_sz, tas_dev->cali_data.data); + if (status == EFI_BUFFER_TOO_SMALL) { + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_dev->cali_data.total_sz, + tas_dev->cali_data.data); + dev_dbg(tas_dev->dev, "cali get %lx bytes result:%ld\n", + tas_dev->cali_data.total_sz, status); + } + if (status != 0) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return; + } + + tmp_val = (unsigned int *)tas_dev->cali_data.data; + + crc = crc32(~0, tas_dev->cali_data.data, 84) ^ ~0; + + if (crc == tmp_val[21]) { + /* Date and time of when calibration was done. */ + tas2783_apply_calib(tas_dev, tmp_val); + dev_dbg(tas_dev->dev, "%ptTs", &tmp_val[20]); + } else { + dev_dbg(tas_dev->dev, "CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[21]); + tas_dev->cali_data.total_sz = 0; + } +} + +static void tasdevice_dspfw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct tas2783_firmware_node *p; + struct regmap *map = tas_dev->regmap; + unsigned char *buf = NULL; + int offset = 0, img_sz; + int ret; + + if (!fmw || !fmw->data) { + dev_warn(tas_dev->dev, + "%s: failed to read %s: work in bypass-dsp mode.\n", + __func__, tas_dev->dspfw_binaryname); + return; + } + buf = (unsigned char *)fmw->data; + + img_sz = get_unaligned_le32(&buf[offset]); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_warn(tas_dev->dev, "%s: size not matching, %d %u.", + __func__, (int)fmw->size, img_sz); + return; + } + + while (offset < img_sz) { + p = (struct tas2783_firmware_node *)(buf + offset); + if (p->length > 1) { + ret = regmap_bulk_write(map, p->download_addr, + buf + offset + sizeof(unsigned int) * 5, p->length); + } else { + ret = regmap_write(map, p->download_addr, + *(buf + offset + sizeof(unsigned int) * 5)); + } + + if (ret != 0) { + dev_dbg(tas_dev->dev, + "%s: load FW fail: %d, work in bypass.\n", + __func__, ret); + return; + } + offset += sizeof(unsigned int) * 5 + p->length; + } + + tas2783_calibration(tas_dev); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC"), +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"}, +}; + +static int tasdevice_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int tasdevice_sdw_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 tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + int ret; + + dev_dbg(dai->dev, "%s: dai_name %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(tas_priv->sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) { + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); + return ret; + } + + dev_dbg(dai->dev, "%s: format: %i rate: %i\n", __func__, + params_format(params), params_rate(params)); + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tas_priv->sdw_peripheral, sdw_stream); + + return 0; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct regmap *map = tas_dev->regmap; + int ret; + + dev_dbg(tas_dev->dev, "%s: %d.\n", __func__, mute); + + if (mute) { + if (direction == SNDRV_PCM_STREAM_CAPTURE) { + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_INACTIVE); + if (ret) + dev_err(tas_dev->dev, + "%s: Disable AEF failed.\n", __func__); + } else { + /* FU23 mute (0x40400108) */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 1); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 mute failed.\n", __func__); + goto out; + } + /* + * Both playback and echo data will be shutdown in + * playback stream. + */ + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK | + TAS2783_REG_AEF_MASK, + TAS2783_REG_PWR_MODE_ACTIVE | + TAS2783_REG_PWR_MODE_SW_PWD); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR&AEF shutdown failed.\n", + __func__); + goto out; + } + tas_dev->pstream = false; + } + } else { + /* FU23 Unmute, 0x40400108. */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 Unmute failed.\n", __func__); + goto out; + } + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK, + TAS2783_REG_PWR_MODE_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR Unmute failed.\n", __func__); + goto out; + } + tas_dev->pstream = true; + } else { + /* Capture stream is the echo ref data for voice. + * Without playback, it can't be active. + */ + if (tas_dev->pstream == true) { + ret = regmap_update_bits(map, + TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: AEF enable failed.\n", + __func__); + goto out; + } + } else { + dev_err(tas_dev->dev, + "%s: No playback, no AEF!", __func__); + ret = -EINVAL; + } + } + } +out: + if (ret) + dev_err(tas_dev->dev, "Mute or unmute %d failed %d.\n", + mute, ret); + + return ret; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_comp_probe(struct snd_soc_component *comp) +{ + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(comp); + acpi_handle handle = ACPI_HANDLE(tas_dev->dev); + const struct firmware *fw_entry = NULL; + const char *sub = NULL; + int ret, value_sdw; + + if (handle) { + sub = acpi_get_subsystem_id(handle); + if (IS_ERR(sub)) + sub = NULL; + } + + tas_dev->component = comp; + + /* + * Each tas2783 in the system has its own dspfw. + */ + if (comp->name_prefix) { + /* + * name_prefix.bin stores the dsp firmware including speaker + * protection algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied into + * firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, "%s-tas2783.bin", + comp->name_prefix); + } else { + /* Compatible with the previous naming rule */ + if (sub) { + /* + * subsystem_id-link_id[0,1,...,N]-dev_num[1,...,4].bin stores + * the dsp firmware including speaker protection algorithm, + * audio acoustic algorithm, speaker characters and algorithm + * params, it must be copied into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "%s-%d-%d.bin", sub, + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->dev_num); + } else { + /* + * tas2783-link_id[0,1,...,N]-dev_num[1,...,4].bin stores + * the dsp firmware including speaker protection algorithm, + * audio acoustic algorithm, speaker characters and algorithm + * params, it must be copied into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "tas2783-%d-%d.bin", + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->dev_num); + } + } + + ret = request_firmware(&fw_entry, tas_dev->dspfw_binaryname, + tas_dev->dev); + if (ret) { + dev_err(tas_dev->dev, + "%s: request_firmware %x open status: %d.\n", __func__, + tas_dev->sdw_peripheral->id.unique_id, ret); + goto out; + } + + tasdevice_dspfw_ready(fw_entry, tas_dev); + + /* Select left/right channel based on device number. */ + value_sdw = 0x1a; + value_sdw |= (tas_dev->sdw_peripheral->dev_num & BIT(0)) << 4; + dev_dbg(tas_dev->dev, "%s: dev_num = %u", __func__, + tas_dev->sdw_peripheral->dev_num); + regmap_write(tas_dev->regmap, TAS2783_REG_TDM_RX_CFG, value_sdw); + if (ret != 0) + dev_warn(tas_dev->dev, "%s: L/R setting failed: %d.\n", + __func__, ret); + +out: + if (fw_entry) + release_firmware(fw_entry); + return 0; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_comp_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + return ret; + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + pm_runtime_enable(tas_dev->dev); + + return 0; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + struct sdw_dpn_prop *dpn; + unsigned long addr; + int nval, i, j; + u32 bit; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = BIT(2); /* BITMAP: 00000100 */ + prop->sink_ports = BIT(1); /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static int tasdevice_io_init(struct device *dev, + struct sdw_slave *slave) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(tas_priv->regmap, false); + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + * Update count of parent 'active' children + */ + pm_runtime_set_active(&slave->dev); + } + + /* sw reset */ + ret = regmap_write(tas_priv->regmap, TAS2873_REG_SWRESET, + TAS2873_REG_SWRESET_RESET); + if (ret) { + dev_err(tas_priv->dev, "Reset failed.\n"); + goto out; + } + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, false); + regcache_mark_dirty(tas_priv->regmap); + } + + tas_priv->first_hw_init = true; + tas_priv->hw_init = true; + +out: + return ret; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tas_priv->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tas_priv->hw_init = false; + /* + * Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_priv->hw_init || tas_priv->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return -ENOMEM; + + tas_dev->dev = dev; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_peripheral = peripheral; + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) + return dev_err_probe(dev, PTR_ERR(tas_dev->regmap), + "Failed devm_regmap_init_sdw."); + + return tasdevice_init(tas_dev); +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(&peripheral->dev); + + if (tas_dev->first_hw_init) + pm_runtime_disable(tas_dev->dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {} +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static int __maybe_unused tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (!tas_priv->hw_init) + return 0; + + regcache_cache_only(tas_priv->regmap, true); + + return 0; +} + +#define TAS2783_PROBE_TIMEOUT_MS 5000 + +static int __maybe_unused tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long time; + + if (!tas_priv->first_hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT_MS)); + if (!time) { + dev_err(&slave->dev, "Init not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(tas_priv->regmap, false); + regcache_sync(tas_priv->regmap); + + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume) + RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume, NULL) +}; + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = &tas2783_sdca_pm, + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..41bdce1964ff --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,103 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2023 - 2024 Texas Instruments Incorporated + * https://www.ti.com + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + * Shenghao Ding <shenghao-ding(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 + \ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0x00, 0x01) +#define TAS2873_REG_SWRESET_MASK BIT(0) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +#define TAS2873_REG_PWR_CTRL TASDEVICE_REG(0x0, 0x00, 0x02) +#define TAS2783_REG_AEF_MASK GENMASK(4, 3) +#define TAS2783_REG_AEF_ACTIVE 0 +#define TAS2783_REG_AEF_INACTIVE GENMASK(4, 3) +#define TAS2783_REG_PWR_MODE_MASK GENMASK(2, 0) +#define TAS2783_REG_PWR_MODE_ACTIVE 0 +#define TAS2783_REG_PWR_MODE_SW_PWD BIT(1) + +#define TAS2783_REG_TDM_RX_CFG TASDEVICE_REG(0x0, 0x00, 0x0a) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1a) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x13, 0x70) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +/* TAS2783 SDCA Control - function number */ +#define TAS2783_FUNC_TYPE_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_FU_VOLUME 0x02 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 +#define TAS2783_DSPFW_FILENAME_LEN 64 + +#define TAS2783_MAX_CALIDATA_SIZE 252 + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct snd_soc_component *component; + struct calibration_data cali_data; + struct sdw_slave *sdw_peripheral; + enum sdw_slave_status status; + struct sdw_bus_params params; + struct regmap *regmap; + struct device *dev; + unsigned char dspfw_binaryname[TAS2783_DSPFW_FILENAME_LEN]; + unsigned char dev_name[32]; + unsigned int chip_id; + bool pstream; + bool hw_init; + bool first_hw_init; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

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[v2 PATCH 0/1] ALSA: virtio: add support for audio controls
by Aiswarya Cyriac 20 Feb '24

20 Feb '24

From: Anton Yakovlev <anton.yakovlev(a)opensynergy.com> Changes in v2 - Fix reporting of incorrect number of items for ENUMERATED controls Implementation of support for audio controls in accordance with the extension of the virtio sound device specification[1] planned for virtio-v1.3-cs01. Design　of virtual audio controls is based on and derived from ALSA audio controls. It allows the driver to perform all standard operations, such as reading and writing audio control value, as well as working with metadata (represented in the TLV form). The driver part was tested on top of the Linux 5.10 kernel. As a device part was used OpenSynergy proprietary implementation. [1] https://lists.oasis-open.org/archives/virtio-comment/202104/msg00013.html Anton Yakovlev (1): ALSA: virtio: add support for audio controls include/uapi/linux/virtio_snd.h | 154 +++++++++++ sound/virtio/Makefile | 1 + sound/virtio/virtio_card.c | 21 ++ sound/virtio/virtio_card.h | 22 ++ sound/virtio/virtio_kctl.c | 464 ++++++++++++++++++++++++++++++++ 5 files changed, 662 insertions(+) create mode 100644 sound/virtio/virtio_kctl.c -- 2.25.1

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