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

tlv320aic3x: Failed to init class D
by Marc Kleine-Budde 28 Nov '23

28 Nov '23

Hello, we are using an I2C connected ti,tlv320aic3007. During probe, regmap_register_patch() returns -6, which leads to the error message: "tlv320aic3x 2-0018: Failed to init class D: -6". I think this is because the codec is still in reset and does not respond to the I2C transfers. How to get rid of the error message? Power on the codec, just to write the register sequence or don't print the error message on "-6"? regards, Marc -- Pengutronix e.K. | Marc Kleine-Budde | Embedded Linux | https://www.pengutronix.de | Vertretung Nürnberg | Phone: +49-5121-206917-129 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-9 |

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[PATCH v2] ASoC: hdac_hda: Conditionally register dais for HDMI and Analog
by Peter Ujfalusi 28 Nov '23

28 Nov '23

The current driver is registering the same dais for each hdev found in the system which results duplicated widgets to be registered and the kernel log contains similar prints: snd_hda_codec_realtek ehdaudio0D0: ASoC: sink widget AIF1TX overwritten snd_hda_codec_realtek ehdaudio0D0: ASoC: source widget AIF1RX overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi3 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi2 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi1 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Codec Output Pin1 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Codec Input Pin1 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Analog Codec Playback overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Digital Codec Playback overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Alt Analog Codec Playback overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Analog Codec Capture overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Digital Codec Capture overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Alt Analog Codec Capture overwritten To avoid such issue, split the dai array into HDMI and non HDMI array and register them conditionally: for HDMI hdev only register the dais needed for HDMI for non HDMI hdev do not register the HDMI dais. Depends-on: 3d1dc8b1030d ("ASoC: Intel: skl_hda_dsp_generic: Drop HDMI routes when HDMI is not available") Link: https://github.com/thesofproject/linux/issues/4509 Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi, Changes since v1: - Drop the patch for patch_hdmi to export a funtion to match the device - Use the struct hdac_hda_priv.need_display_power boolean as indication that the device is a HDMI/DP audio codec Regards, Peter sound/soc/codecs/hdac_hda.c | 23 ++++++++++++++++++++--- 1 file changed, 20 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 355f30779a34..b075689db2dc 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -132,6 +132,9 @@ static struct snd_soc_dai_driver hdac_hda_dais[] = { .sig_bits = 24, }, }, +}; + +static struct snd_soc_dai_driver hdac_hda_hdmi_dais[] = { { .id = HDAC_HDMI_0_DAI_ID, .name = "intel-hdmi-hifi1", @@ -607,8 +610,16 @@ static const struct snd_soc_component_driver hdac_hda_codec = { .endianness = 1, }; +static const struct snd_soc_component_driver hdac_hda_hdmi_codec = { + .probe = hdac_hda_codec_probe, + .remove = hdac_hda_codec_remove, + .idle_bias_on = false, + .endianness = 1, +}; + static int hdac_hda_dev_probe(struct hdac_device *hdev) { + struct hdac_hda_priv *hda_pvt = dev_get_drvdata(&hdev->dev); struct hdac_ext_link *hlink; int ret; @@ -621,9 +632,15 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) snd_hdac_ext_bus_link_get(hdev->bus, hlink); /* ASoC specific initialization */ - ret = devm_snd_soc_register_component(&hdev->dev, - &hdac_hda_codec, hdac_hda_dais, - ARRAY_SIZE(hdac_hda_dais)); + if (hda_pvt->need_display_power) + ret = devm_snd_soc_register_component(&hdev->dev, + &hdac_hda_hdmi_codec, hdac_hda_hdmi_dais, + ARRAY_SIZE(hdac_hda_hdmi_dais)); + else + ret = devm_snd_soc_register_component(&hdev->dev, + &hdac_hda_codec, hdac_hda_dais, + ARRAY_SIZE(hdac_hda_dais)); + if (ret < 0) { dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret); return ret; -- 2.43.0

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[PATCH] ASoC: SOF: acp: Add prevent against NULL
by Kamil Duljas 28 Nov '23

28 Nov '23

When kasprintf function fail, then NULL is returned. The callers dereference them without null checked. Signed-off-by: Kamil Duljas <kamil.duljas(a)gmail.com> --- sound/soc/sof/amd/acp.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 603ea5fc0d0d..c60a3e28b5f3 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -548,11 +548,15 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) dmi_id = dmi_first_match(acp_sof_quirk_table); if (dmi_id && dmi_id->driver_data) { adata->fw_code_bin = kasprintf(GFP_KERNEL, "%s/sof-%s-code.bin", - plat_data->fw_filename_prefix, - chip->name); + plat_data->fw_filename_prefix, + chip->name); + if (!adata->fw_code_bin) + return -ENOMEM; adata->fw_data_bin = kasprintf(GFP_KERNEL, "%s/sof-%s-data.bin", - plat_data->fw_filename_prefix, - chip->name); + plat_data->fw_filename_prefix, + chip->name); + if (!adata->fw_data_bin) + return -ENOMEM; adata->signed_fw_image = dmi_id->driver_data; dev_dbg(sdev->dev, "fw_code_bin:%s, fw_data_bin:%s\n", adata->fw_code_bin, -- 2.42.0.windows.2

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[PATCH 00/27] ASoC: Intel: Link handling rework and fixes
by Peter Ujfalusi 28 Nov '23

28 Nov '23

Hi, SOF board updates for 6.8 including few small fix and the majority is to add generic helpers for codec, amp, BT offload, HDMI-In and DAI link generation among various machine drivers. Regards, Peter --- Bard Liao (2): ASoC: Intel: sof_maxim_common: add else between 2 if test ASoC: Intel: sof_maxim_common: check return value Brent Lu (24): ASoC: Intel: sof_ssp_amp: remove dead code ASoC: Intel: ssp-common: get codec name function ASoC: Intel: board_helpers: support codec link initialization ASoC: Intel: sof_cs42l42: use common module for codec link ASoC: Intel: sof_nau8825: use common module for codec link ASoC: Intel: sof_rt5682: use common module for codec link ASoC: Intel: board_helpers: support amp link initialization ASoC: Intel: sof_cs42l42: use common module for amp link ASoC: Intel: sof_nau8825: use common module for amp link ASoC: Intel: sof_rt5682: use common module for amp link ASoC: Intel: sof_ssp_amp: use common module for amp link ASoC: Intel: sof_ssp_amp: rename function parameter ASoC: Intel: board_helpers: support BT offload link initialization ASoC: Intel: sof_cs42l42: use common module for BT offload link ASoC: Intel: sof_nau8825: use common module for BT offload link ASoC: Intel: sof_rt5682: use common module for BT offload link ASoC: Intel: sof_ssp_amp: use common module for BT offload link ASoC: Intel: sof_ssp_amp: simplify HDMI-In quirks ASoC: Intel: board_helpers: support HDMI-In link initialization ASoC: Intel: sof_rt5682: use common module for HDMI-In link ASoC: Intel: sof_ssp_amp: use common module for HDMI-In link ASoC: Intel: board_helpers: support DAI link array generation ASoC: Intel: sof_nau8825: use common module for DAI link generation ASoC: Intel: sof_rt5682: use common module for DAI link generation Chao Song (1): ASoC: Intel: cht_bsw_rt5672: check return value sound/soc/intel/boards/cht_bsw_rt5672.c | 8 +- sound/soc/intel/boards/sof_board_helpers.c | 330 +++++++++++++++++++++ sound/soc/intel/boards/sof_board_helpers.h | 31 ++ sound/soc/intel/boards/sof_cs42l42.c | 229 ++++---------- sound/soc/intel/boards/sof_maxim_common.c | 13 +- sound/soc/intel/boards/sof_nau8825.c | 224 ++++---------- sound/soc/intel/boards/sof_rt5682.c | 328 ++++++-------------- sound/soc/intel/boards/sof_ssp_amp.c | 160 ++++------ sound/soc/intel/boards/sof_ssp_common.c | 21 ++ sound/soc/intel/boards/sof_ssp_common.h | 1 + 10 files changed, 653 insertions(+), 692 deletions(-) -- 2.43.0

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[PATCH 0/2] ALSA/ASoC: hdmi/hdac_hda: Conditionally register dais
by Peter Ujfalusi 28 Nov '23

28 Nov '23

Hi, Each instance of the hdac_hda registers DAIs to be used for analog and HDMI audio. When the system have more than one HDA codec - like most devices then the kernel log will include the following warning prints: snd_hda_codec_realtek ehdaudio0D0: ASoC: sink widget AIF1TX overwritten snd_hda_codec_realtek ehdaudio0D0: ASoC: source widget AIF1RX overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi3 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi2 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi1 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Codec Output Pin1 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Codec Input Pin1 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Analog Codec Playback overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Digital Codec Playback overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Alt Analog Codec Playback overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Analog Codec Capture overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Digital Codec Capture overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Alt Analog Codec Capture overwritten These are printed when the second instance of the hdac_hda is probing since it is re-registering the DAIs with the same name. Using some clever way of prefixing might solve the issue but that would need non backwards compatible changes to topology files. Pragmatically the hdac_hda instance fro the normal codec should not register DAIs for HDMI and the hdac_hda instance for the HDMI should not register DAIs for the analog codec - for example on a SDW device where we have HDA-HDMI and sdw codecs. To keep the hdac_hda vendor neutral (altrough it is only used on Intel platforms afaik) add a helper function to answer the question: is this codec HDMI? Use the new helper to decide which sets of DAIs to register for the probing hdac_hda instance. Regards, Peter --- Peter Ujfalusi (2): ALSA: hda/hdmi: Add helper function to check if a device is HDMI codec ASoC: hdac_hda: Conditionally register dais for HDMI and Analog include/sound/hdaudio.h | 10 ++++++++++ sound/pci/hda/patch_hdmi.c | 13 +++++++++++++ sound/soc/codecs/hdac_hda.c | 22 +++++++++++++++++++--- 3 files changed, 42 insertions(+), 3 deletions(-) -- 2.43.0

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[PATCH] soundwire: stream: fix NULL pointer dereference for multi_link
by Krzysztof Kozlowski 28 Nov '23

28 Nov '23

If bus is marked as multi_link, but number of masters in the stream is not higher than bus->hw_sync_min_links (bus->multi_link && m_rt_count >= bus->hw_sync_min_links), bank switching should not happen. The first part of do_bank_switch() code properly takes these conditions into account, but second part (sdw_ml_sync_bank_switch()) relies purely on bus->multi_link property. This is not balanced and leads to NULL pointer dereference: Unable to handle kernel NULL pointer dereference at virtual address 0000000000000000 ... Call trace: wait_for_completion_timeout+0x124/0x1f0 do_bank_switch+0x370/0x6f8 sdw_prepare_stream+0x2d0/0x438 qcom_snd_sdw_prepare+0xa0/0x118 sm8450_snd_prepare+0x128/0x148 snd_soc_link_prepare+0x5c/0xe8 __soc_pcm_prepare+0x28/0x1ec dpcm_be_dai_prepare+0x1e0/0x2c0 dpcm_fe_dai_prepare+0x108/0x28c snd_pcm_do_prepare+0x44/0x68 snd_pcm_action_single+0x54/0xc0 snd_pcm_action_nonatomic+0xe4/0xec snd_pcm_prepare+0xc4/0x114 snd_pcm_common_ioctl+0x1154/0x1cc0 snd_pcm_ioctl+0x54/0x74 Fixes: ce6e74d008ff ("soundwire: Add support for multi link bank switch") Cc: <stable(a)vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/soundwire/stream.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 9dc6399f206a..f9c0adc0738d 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -742,14 +742,15 @@ static int sdw_bank_switch(struct sdw_bus *bus, int m_rt_count) * sdw_ml_sync_bank_switch: Multilink register bank switch * * @bus: SDW bus instance + * @multi_link: whether this is a multi-link stream with hardware-based sync * * Caller function should free the buffers on error */ -static int sdw_ml_sync_bank_switch(struct sdw_bus *bus) +static int sdw_ml_sync_bank_switch(struct sdw_bus *bus, bool multi_link) { unsigned long time_left; - if (!bus->multi_link) + if (!multi_link) return 0; /* Wait for completion of transfer */ @@ -847,7 +848,7 @@ static int do_bank_switch(struct sdw_stream_runtime *stream) bus->bank_switch_timeout = DEFAULT_BANK_SWITCH_TIMEOUT; /* Check if bank switch was successful */ - ret = sdw_ml_sync_bank_switch(bus); + ret = sdw_ml_sync_bank_switch(bus, multi_link); if (ret < 0) { dev_err(bus->dev, "multi link bank switch failed: %d\n", ret); -- 2.34.1

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Set Mic VREF to 100
by Kailang 28 Nov '23

28 Nov '23

Hi Takashi, This platform need to set MIC vref to 100%. BR, Kailang

2 1

[PATCH RFC] ALSA: scarlett2: Add ioctls for user-space access
by Geoffrey D. Bennett 27 Nov '23

27 Nov '23

In order to support functions such as firmware upgrade from user-space, add ioctls for submitting arbitrary proprietary requests through scarlett2_usb() and requesting/releasing exclusive access. --- Hi Takashi, I recently figured how to update the firmware on Scarlett Gen 2+ devices. I think the best way to implement this is with an ioctl giving access to the scarlett2_usb() function from user-space, plus two ioctls to request/release exclusive access. Does something like this seem reasonable? What ioctl magic/command values should I use? Should lock/unlock be separate, or one ioctl takes an argument for which operation to do? Thanks, Geoffrey. MAINTAINERS | 1 + include/uapi/sound/scarlett_gen2.h | 28 +++++++++ sound/usb/mixer_scarlett_gen2.c | 94 +++++++++++++++++++++++++++++- 3 files changed, 121 insertions(+), 2 deletions(-) create mode 100644 include/uapi/sound/scarlett_gen2.h diff --git a/MAINTAINERS b/MAINTAINERS index 4cc6bf79fdd8..25e5e40f7118 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -8066,6 +8066,7 @@ M: Geoffrey D. Bennett <g(a)b4.vu> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git +F: include/uapi/sound/scarlett_gen2.h F: sound/usb/mixer_scarlett_gen2.c FORCEDETH GIGABIT ETHERNET DRIVER diff --git a/include/uapi/sound/scarlett_gen2.h b/include/uapi/sound/scarlett_gen2.h new file mode 100644 index 000000000000..0b51a9754ba2 --- /dev/null +++ b/include/uapi/sound/scarlett_gen2.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ +/* + * Focusrite Scarlett Gen 2/3 and Clarett USB/Clarett+ Driver for ALSA + * + * Copyright (c) 2023 by Geoffrey D. Bennett <g at b4.vu> + */ +#ifndef __UAPI_SOUND_SCARLETT_GEN2_H +#define __UAPI_SOUND_SCARLETT_GEN2_H + +#include <linux/types.h> +#include <linux/ioctl.h> + +/* Send a proprietary format request to the interface */ +struct snd_scarlett_gen2_usb_cmd { + u32 cmd; + void *req_data; + u16 req_size; + void *resp_data; + u16 resp_size; +}; + +#define SCARLETT2_IOCTL_USB_CMD _IOWR('S', 0x60, struct snd_scarlett_gen2_usb_cmd) + +/* Request/release exclusive access */ +#define SCARLETT2_IOCTL_LOCK _IO('S', 0x61) +#define SCARLETT2_IOCTL_UNLOCK _IO('S', 0x62) + +#endif /* __UAPI_SOUND_SCARLETT_GEN2_H */ diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index ffd398f26d2c..f1ef87b54813 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -144,6 +144,9 @@ #include <sound/control.h> #include <sound/tlv.h> +#include <sound/hwdep.h> + +#include <uapi/sound/scarlett_gen2.h> #include "usbaudio.h" #include "mixer.h" @@ -4262,10 +4265,84 @@ static int snd_scarlett_gen2_controls_create( return 0; } +static int scarlett_gen2_ioctl_usb_cmd(struct snd_hwdep *hw, unsigned long arg) +{ + struct scarlett2_data *private = hw->private_data; + struct snd_scarlett_gen2_usb_cmd usb_cmd; + int err; + void *req_data = NULL; + void *resp_data = NULL; + + // get cmd & req/resp buffers + if (copy_from_user(&usb_cmd, (void __user *)arg, sizeof(usb_cmd))) + return -EFAULT; + + // allocate request buffer, copy data from user + if (usb_cmd.req_size > 0) { + req_data = kmalloc(usb_cmd.req_size, GFP_KERNEL); + if (!req_data) { + err = -ENOMEM; + goto exit; + } + if (copy_from_user(req_data, usb_cmd.req_data, + usb_cmd.req_size)) { + err = -EFAULT; + goto exit; + } + } + + // allocate response buffer + if (usb_cmd.resp_size > 0) { + resp_data = kmalloc(usb_cmd.resp_size, GFP_KERNEL); + if (!resp_data) { + err = -ENOMEM; + goto exit; + } + } + + // send request, get response + err = scarlett2_usb(private->mixer, usb_cmd.cmd, + req_data, usb_cmd.req_size, + resp_data, usb_cmd.resp_size); + if (err < 0) + goto exit; + + // copy response to user + if (usb_cmd.resp_size > 0) + if (copy_to_user(usb_cmd.resp_data, resp_data, + usb_cmd.resp_size)) + err = -EFAULT; + +exit: + kfree(req_data); + kfree(resp_data); + + return err; +} + +static int scarlett_gen2_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, + unsigned int cmd, unsigned long arg) +{ + switch (cmd) { + + case SCARLETT2_IOCTL_USB_CMD: + return scarlett_gen2_ioctl_usb_cmd(hw, arg); + + // TODO + case SCARLETT2_IOCTL_LOCK: + case SCARLETT2_IOCTL_UNLOCK: + return -EINVAL; + + default: + return -ENOIOCTLCMD; + } +} + int snd_scarlett_gen2_init(struct usb_mixer_interface *mixer) { struct snd_usb_audio *chip = mixer->chip; const struct scarlett2_device_entry *entry; + struct snd_hwdep *hw; int err; /* only use UAC_VERSION_2 */ @@ -4302,11 +4379,24 @@ int snd_scarlett_gen2_init(struct usb_mixer_interface *mixer) USB_ID_PRODUCT(chip->usb_id)); err = snd_scarlett_gen2_controls_create(mixer, entry); - if (err < 0) + if (err < 0) { usb_audio_err(mixer->chip, "Error initialising %s Mixer Driver: %d", entry->series_name, err); + return err; + } + + err = snd_hwdep_new(mixer->chip->card, "Focusrite Control", 0, &hw); + if (err < 0) { + usb_audio_err(mixer->chip, + "Error creating hwdep device: %d", + err); + return err; + } - return err; + hw->private_data = mixer->private_data; + hw->ops.ioctl = scarlett_gen2_hwdep_ioctl; + + return 0; } -- 2.41.0

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[PATCH] ASoC: SOF: Intel: mtl: call dsp dump when boot retry fails
by Peter Ujfalusi 27 Nov '23

27 Nov '23

From: Yong Zhi <yong.zhi(a)intel.com> Call snd_sof_dsp_dbg_dump() with the same flags/dump_msg as used in function hda_loader.c/cl_dsp_init(). Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/mtl.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 254dbbeac1d0..3ef9e5c37028 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -440,7 +440,8 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; const struct sof_intel_dsp_desc *chip = hda->desc; unsigned int status; - u32 ipc_hdr; + u32 ipc_hdr, flags; + char *dump_msg; int ret; /* step 1: purge FW request */ @@ -493,8 +494,18 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) return 0; err: - snd_sof_dsp_dbg_dump(sdev, "MTL DSP init fail", 0); + flags = SOF_DBG_DUMP_PCI | SOF_DBG_DUMP_MBOX | SOF_DBG_DUMP_OPTIONAL; + + /* after max boot attempts make sure that the dump is printed */ + if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) + flags &= ~SOF_DBG_DUMP_OPTIONAL; + + dump_msg = kasprintf(GFP_KERNEL, "Boot iteration failed: %d/%d", + hda->boot_iteration, HDA_FW_BOOT_ATTEMPTS); + snd_sof_dsp_dbg_dump(sdev, dump_msg, flags); mtl_dsp_core_power_down(sdev, SOF_DSP_PRIMARY_CORE); + + kfree(dump_msg); return ret; } -- 2.43.0

2 1

[PATCH] ASoC: SOF: Intel: Use existing helpers to change GPROCEN and PIE bits
by Peter Ujfalusi 27 Nov '23

27 Nov '23

Instead of directly changing the GPROCEN/PIE bits in PPCTL we should use the existing helper hda_dsp_ctrl_ppcap_enable() and hda_dsp_ctrl_ppcap_int_enable() helpers for clarity. Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-loader.c | 5 ++--- sound/soc/sof/intel/hda.c | 6 ++---- 2 files changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 46fb2d1425e9..1805cf754beb 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -510,9 +510,8 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) return chip_info->init_core_mask; /* disable DSP */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, - SOF_HDA_REG_PP_PPCTL, - SOF_HDA_PPCTL_GPROCEN, 0); + hda_dsp_ctrl_ppcap_enable(sdev, false); + return ret; } diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 744c0dd5766d..fe4ae349dad5 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1350,8 +1350,7 @@ void hda_dsp_remove(struct snd_sof_dev *sdev) if (!sdev->dspless_mode_selected) { /* disable DSP IRQ */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, - SOF_HDA_PPCTL_PIE, 0); + hda_dsp_ctrl_ppcap_int_enable(sdev, false); } /* disable CIE and GIE interrupts */ @@ -1366,8 +1365,7 @@ void hda_dsp_remove(struct snd_sof_dev *sdev) chip->power_down_dsp(sdev); /* disable DSP */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, - SOF_HDA_PPCTL_GPROCEN, 0); + hda_dsp_ctrl_ppcap_enable(sdev, false); skip_disable_dsp: free_irq(sdev->ipc_irq, sdev); -- 2.43.0

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