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

[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

2 28

[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

2 18

[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

3 2

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

3 8

[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

2 1

[PATCH v5 00/16] ALSA/ASoC: hda: Address format selection limitations and ambiguity
by Cezary Rojewski 27 Nov '23

27 Nov '23

Patchset aims to address format selection restrictions present currently in the HDAudio library. Formats which we are concerned about are 20 and 24 valid bits per sample within 32 bit depth container. One may identify them as S20_LE and S24_LE except that those, according to comments found in include/uapi/sound/asound.h, are for LSB-aligned scenarios. HDAudio streams expect MSB-aligned data, no matter if we are speaking of HOST (SDxFMT) or LINK (PPLCxFMT) side - chapter 4.5.1 of the public HDAudio specification. In short, S20_LE and S24_LE are invalid options. Right now, given the implementation of snd_hdac_query_supported_pcm() within sound/hda/hdac_device.c, even if a codec responds with: "I support all possible formats specified within HDAudio specification", there will be no option to open a 20/32 or 24/32 stream. The kernel will force the stream to be opened using the highest available bit depth. After discussing subject initially with Jaroslav and Takashi, suggestion was made to utilize 'subformat' option to address the problem. The eye-opening discussion begun much earlier though, in 2019 [1]. Paired with PRs for alsa-utils [2] and alsa-lib [3]. Flow of changes: The very first patch adds MSBITS subformat options to allow for granular 20/32, 24/32 and 32/32 format selection. The next two make sure subformat is actually honored during runtime. Most of that code is based on format-related API. Follow up is upgrade to the hda stream-format interface - several functions are added to make the granular format selection simple in the HDAudio world. Core of the implementation is based on the existing snd_hdac_calc_stream_format(). The next ten patches are straightforward switch from one interface to another with cleanup of now-unsed function as a finishing touch. Last but not least - the avs-driver, on which the problem analyzed and debugged, is updated to no longer acknowledge S24_LE as a valid format option. Results with skylake-driver and snd_hda_intel show status quo on our RVPs. PR filed on SOF github shows promising results too [4]. Changes in v5: - reworded 'bps' to 'bits' in all occurrences. - fixed an issue of MSBITS_MAX not being reported for S32_LE as reported by Jaroslav - snd_pcm_subformat_width() has been inlined into its only user: snd_pcm_hw_params_bits() as suggested by Jaroslav. - updated commit messege for patch 01 as it was out of date given the recent updates. Changes in v4: - fixed compilation issues in sof-driver, patch 12/16, reported by ikp - fixed sparse warnings in patch 01/16, reported by ikp - updated commit message for patch 03: "ASoC pcm: Honor subformat when configuring runtime", as snd_pcm_hw_copy() is gone since revision v3. Changes in v3: - merged the first two patches as suggested by Jaroslav - re-authored patch 01 to Jaroslav, added my Co-developed-by. - added Jaroslav' Co-developed-by to patch 02. - 'subformats' field now S32_LE-specific. Given the fact that it is the only format currently requiring subformat-intervention, functionality is narrowed to reduce amount of memory allocations and cleanup. Suggested by Jaroslav. - note to the above: the hdaudio part converted 1:1 as requested, patch 02/16 - note #2: alsa part converted to S32_LE-specific yet without addition of the chicken bit. Instead, struct snd_pcm_hardware is updated with u32 subformat mask to do the job. - ALSA-core additions in form of snd_pcm_subformat_width() and snd_pcm_hw_params_bps() relocated from 01/16 to the user, patch 05. Changes in v2: - patch 01/17, introduced struct snd_pcm_subformat which task is to represent subformat-mask on per format basis. Expectation is that manipulated arrays of subformats always end with a sentinel entry - patch 01/17, added snd_pcm_hw_copy() as the copying snd_pcm_hardware becomes non-trivial - patch 02/17, added hw_rule that produces final subformat mask based on provided formats as suggested by Jaroslav - patch 04/17, soc_pcm_hw_update_subformat() refactored as the subformat intersection becomes non-trivial - relevant functions releasing resources occupied by hda_pcm and snd_pcm_runtime updated to also kfree() subformats - except for 16/17, no changes to patches past 04/17, retaining acks for these Changes in v1: - fixed UBSAN due to missing snd_pcm_subformat_names[] entries for new subformats - as HDMI stream capabilities are assigned on PCM open, patch 16/17 has been updated to ignore such codecs for now. A separate patchset will take care of this case - params_bps() reworded to snd_pcm_hw_params_bps() - fixed compilation issues in sof-driver, patch 13/17 [1]: https://lore.kernel.org/alsa-devel/20190905053302.9262-1-pawel.harlozinski@… [2]: https://github.com/alsa-project/alsa-utils/pull/228 [3]: https://github.com/alsa-project/alsa-lib/pull/342 [4]: https://github.com/thesofproject/linux/pull/4539 Cezary Rojewski (15): ALSA: hda: Honor subformat when querying PCMs ASoC: pcm: Honor subformat when configuring runtime ALSA: hda: Upgrade stream-format infrastructure ALSA: hda: Switch to new stream-format interface ALSA: hda/hdmi: Switch to new stream-format interface ALSA: hda/ca0132: Switch to new stream-format interface ASoC: codecs: hda: Switch to new stream-format interface ASoC: codecs: hdac_hda: Switch to new stream-format interface ASoC: codecs: hdac_hdmi: Switch to new stream-format interface ASoC: Intel Skylake: Switch to new stream-format interface ASoC: SOF: Intel: Switch to new stream-format interface ASoC: Intel: avs: Switch to new stream-format interface ALSA: hda: Drop snd_hdac_calc_stream_format() ASoC: Intel: avs: Kill S24_LE format ASoC: Intel: avs: Unhardcode HDAudio BE DAI drivers description Jaroslav Kysela (1): ALSA: pcm: Introduce MSBITS subformat interface include/sound/hda_codec.h | 5 +- include/sound/hdaudio.h | 13 +-- include/sound/pcm.h | 7 ++ include/sound/pcm_params.h | 2 + include/sound/soc.h | 1 + include/uapi/sound/asound.h | 7 +- sound/core/pcm.c | 3 + sound/core/pcm_lib.c | 34 +++++++ sound/core/pcm_native.c | 55 ++++++++++- sound/hda/hdac_device.c | 156 +++++++++++++++++++++--------- sound/pci/hda/hda_codec.c | 2 + sound/pci/hda/hda_controller.c | 10 +- sound/pci/hda/patch_ca0132.c | 3 +- sound/pci/hda/patch_hdmi.c | 6 +- sound/soc/codecs/hda-dai.c | 7 +- sound/soc/codecs/hda.c | 2 + sound/soc/codecs/hdac_hda.c | 8 +- sound/soc/codecs/hdac_hdmi.c | 10 +- sound/soc/intel/avs/loader.c | 4 +- sound/soc/intel/avs/path.c | 2 +- sound/soc/intel/avs/pcm.c | 58 +++++++++-- sound/soc/intel/avs/probes.c | 3 +- sound/soc/intel/avs/topology.c | 13 ++- sound/soc/intel/skylake/skl-pcm.c | 13 ++- sound/soc/soc-pcm.c | 10 ++ sound/soc/sof/intel/hda-dai-ops.c | 21 ++-- tools/include/uapi/sound/asound.h | 7 +- 27 files changed, 352 insertions(+), 110 deletions(-) -- 2.25.1

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ASoC: amd: yc: Fix non-functional mic on ASUS E1504FA
by Malcolm Hart 27 Nov '23

27 Nov '23

Like other ASUS models the Asus Vivobook E1504FA requires an entry in the quirk list to enable the internal microphone. Showing with 7 additions and 0 deletions. 7 changes: 7 additions & 0 deletions 7 sound/soc/amd/yc/acp6x-mach.c @@ -283,6 +283,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "M6500RC"), } }, { .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), DMI_MATCH(DMI_PRODUCT_NAME, "E1504FA"), } }, { I have this laptop and I have tested this patch successfully. Malcolm

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[PATCH] ALSA: hda: intel-nhlt: Ignore vbps when looking for DMIC 32 bps format
by Peter Ujfalusi 27 Nov '23

27 Nov '23

When looking up DMIC blob from the NHLT table and the format is 32 bits, ignore the vbps matching for 32 bps for DMIC since some NHLT table have the vbps as 24, some have it as 32. The DMIC hardware supports only one type of 32 bit sample size, which is 24 bit sampling on the MSB side and bits[1:0] is used for indicating the channel number. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(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> --- sound/hda/intel-nhlt.c | 33 +++++++++++++++++++++++++++++---- 1 file changed, 29 insertions(+), 4 deletions(-) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 2c4dfc0b7e34..696a958d93e9 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -238,7 +238,7 @@ EXPORT_SYMBOL(intel_nhlt_ssp_mclk_mask); static struct nhlt_specific_cfg * nhlt_get_specific_cfg(struct device *dev, struct nhlt_fmt *fmt, u8 num_ch, - u32 rate, u8 vbps, u8 bps) + u32 rate, u8 vbps, u8 bps, bool ignore_vbps) { struct nhlt_fmt_cfg *cfg = fmt->fmt_config; struct wav_fmt *wfmt; @@ -255,8 +255,12 @@ nhlt_get_specific_cfg(struct device *dev, struct nhlt_fmt *fmt, u8 num_ch, dev_dbg(dev, "Endpoint format: ch=%d fmt=%d/%d rate=%d\n", wfmt->channels, _vbps, _bps, wfmt->samples_per_sec); + /* + * When looking for exact match of configuration ignore the vbps + * from NHLT table when ignore_vbps is true + */ if (wfmt->channels == num_ch && wfmt->samples_per_sec == rate && - vbps == _vbps && bps == _bps) + (ignore_vbps || vbps == _vbps) && bps == _bps) return &cfg->config; cfg = (struct nhlt_fmt_cfg *)(cfg->config.caps + cfg->config.size); @@ -289,6 +293,7 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, { struct nhlt_specific_cfg *cfg; struct nhlt_endpoint *epnt; + bool ignore_vbps = false; struct nhlt_fmt *fmt; int i; @@ -298,7 +303,26 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, dev_dbg(dev, "Looking for configuration:\n"); dev_dbg(dev, " vbus_id=%d link_type=%d dir=%d, dev_type=%d\n", bus_id, link_type, dir, dev_type); - dev_dbg(dev, " ch=%d fmt=%d/%d rate=%d\n", num_ch, vbps, bps, rate); + if (link_type == NHLT_LINK_DMIC && bps == 32 && (vbps == 24 || vbps == 32)) { + /* + * The DMIC hardware supports only one type of 32 bits sample + * size, which is 24 bit sampling on the MSB side and bits[1:0] + * are used for indicating the channel number. + * It has been observed that some NHLT tables have the vbps + * specified as 32 while some uses 24. + * The format these variations describe are identical, the + * hardware is configured and behaves the same way. + * Note: when the samples assumed to be vbps=32 then the 'noise' + * introduced by the lower two bits (channel number) have no + * real life implication on audio quality. + */ + dev_dbg(dev, + " ch=%d fmt=%d rate=%d (vbps is ignored for DMIC 32bit format)\n", + num_ch, bps, rate); + ignore_vbps = true; + } else { + dev_dbg(dev, " ch=%d fmt=%d/%d rate=%d\n", num_ch, vbps, bps, rate); + } dev_dbg(dev, "Endpoint count=%d\n", nhlt->endpoint_count); epnt = (struct nhlt_endpoint *)nhlt->desc; @@ -307,7 +331,8 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, if (nhlt_check_ep_match(dev, epnt, bus_id, link_type, dir, dev_type)) { fmt = (struct nhlt_fmt *)(epnt->config.caps + epnt->config.size); - cfg = nhlt_get_specific_cfg(dev, fmt, num_ch, rate, vbps, bps); + cfg = nhlt_get_specific_cfg(dev, fmt, num_ch, rate, + vbps, bps, ignore_vbps); if (cfg) return cfg; } -- 2.43.0

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