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[PATCH] ASoC: cs35l56: Remove default from IRQ1_CFG register
by Richard Fitzgerald 09 Feb '24

09 Feb '24

The driver never uses the IRQ1_CFG register so there's no need to provide a default value. It's set as a readable register only for debugging through the regmap registers file. A system-specific firmware could overwrite this register with a non-default value. Therefore the driver can't hardcode what the initial value actually is. As the register is only for debugging the value can be left unknown until someone wants to read it through debugfs. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56-shared.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 02fba4bc0a14..995d979b6d87 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -51,7 +51,6 @@ static const struct reg_default cs35l56_reg_defaults[] = { { CS35L56_SWIRE_DP3_CH2_INPUT, 0x00000019 }, { CS35L56_SWIRE_DP3_CH3_INPUT, 0x00000029 }, { CS35L56_SWIRE_DP3_CH4_INPUT, 0x00000028 }, - { CS35L56_IRQ1_CFG, 0x00000000 }, { CS35L56_IRQ1_MASK_1, 0x83ffffff }, { CS35L56_IRQ1_MASK_2, 0xffff7fff }, { CS35L56_IRQ1_MASK_4, 0xe0ffffff }, -- 2.30.2

2 1

[PATCH v4] ASoC: tas2781: remove unused acpi_subysystem_id
by Gergo Koteles 09 Feb '24

09 Feb '24

The acpi_subysystem_id is only written and freed, not read, so unnecessary. Signed-off-by: Gergo Koteles <soyer(a)irl.hu> --- Changes since v3: - remove Fixes and CC tags from commit message - remove In-Reply-To from header - rebase onto for-next - link to v3: https://lore.kernel.org/lkml/7f056a4148fec176812ff6cc490860bf565b161c.17072… Changes since v2: remove sub from tas2781_read_acpi Changes since v1: remove physdev from tas2781_read_acpi --- include/sound/tas2781.h | 1 - sound/pci/hda/tas2781_hda_i2c.c | 12 ------------ sound/soc/codecs/tas2781-comlib.c | 1 - 3 files changed, 14 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 9aff384941de..99ca3e401fd1 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -103,7 +103,6 @@ struct tasdevice_priv { struct tm tm; enum device_catlog_id catlog_id; - const char *acpi_subsystem_id; unsigned char cal_binaryname[TASDEVICE_MAX_CHANNELS][64]; unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; unsigned char coef_binaryname[64]; diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 1bfb00102a77..4c9a788c3501 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -111,9 +111,7 @@ static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) static int tas2781_read_acpi(struct tasdevice_priv *p, const char *hid) { struct acpi_device *adev; - struct device *physdev; LIST_HEAD(resources); - const char *sub; int ret; adev = acpi_dev_get_first_match_dev(hid, NULL, -1); @@ -129,18 +127,8 @@ static int tas2781_read_acpi(struct tasdevice_priv *p, const char *hid) acpi_dev_free_resource_list(&resources); strscpy(p->dev_name, hid, sizeof(p->dev_name)); - physdev = get_device(acpi_get_first_physical_node(adev)); acpi_dev_put(adev); - /* No side-effect to the playback even if subsystem_id is NULL*/ - sub = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); - if (IS_ERR(sub)) - sub = NULL; - - p->acpi_subsystem_id = sub; - - put_device(physdev); - return 0; err: diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 5d0e5348b361..3aa81514dad7 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -408,7 +408,6 @@ void tasdevice_remove(struct tasdevice_priv *tas_priv) { if (gpio_is_valid(tas_priv->irq_info.irq_gpio)) gpio_free(tas_priv->irq_info.irq_gpio); - kfree(tas_priv->acpi_subsystem_id); mutex_destroy(&tas_priv->codec_lock); } EXPORT_SYMBOL_GPL(tasdevice_remove); base-commit: aa0dc84513e94a6ee2df360cf5e3b207abf734e0 -- 2.43.0

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[PATCH v2] ASoC: Intel: avs: Expose FW version with sysfs
by Amadeusz Sławiński 09 Feb '24

09 Feb '24

From: Cezary Rojewski <cezary.rojewski(a)intel.com> Add functionality to read version of loaded FW from sysfs. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- Changes in v2: - Adjust path in Documentation to correct one --- .../ABI/testing/sysfs-bus-pci-devices-avs | 8 +++++ sound/soc/intel/avs/Makefile | 3 +- sound/soc/intel/avs/avs.h | 4 +++ sound/soc/intel/avs/core.c | 1 + sound/soc/intel/avs/sysfs.c | 35 +++++++++++++++++++ 5 files changed, 50 insertions(+), 1 deletion(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-pci-devices-avs create mode 100644 sound/soc/intel/avs/sysfs.c diff --git a/Documentation/ABI/testing/sysfs-bus-pci-devices-avs b/Documentation/ABI/testing/sysfs-bus-pci-devices-avs new file mode 100644 index 0000000000000..ebff3fa12055d --- /dev/null +++ b/Documentation/ABI/testing/sysfs-bus-pci-devices-avs @@ -0,0 +1,8 @@ +What: /sys/devices/pci0000:00/<dev>/avs/fw_version +Date: February 2024 +Contact: Cezary Rojewski <cezary.rojewski(a)intel.com> +Description: + Version of AudioDSP firmware ASoC avs driver is communicating + with. + + Format: %d.%d.%d.%d, type:major:minor:build. diff --git a/sound/soc/intel/avs/Makefile b/sound/soc/intel/avs/Makefile index 460ee6599daf0..a3fad926d0fb0 100644 --- a/sound/soc/intel/avs/Makefile +++ b/sound/soc/intel/avs/Makefile @@ -1,7 +1,8 @@ # SPDX-License-Identifier: GPL-2.0-only snd-soc-avs-objs := dsp.o ipc.o messages.o utils.o core.o loader.o \ - topology.o path.o pcm.o board_selection.o control.o + topology.o path.o pcm.o board_selection.o control.o \ + sysfs.o snd-soc-avs-objs += cldma.o snd-soc-avs-objs += skl.o apl.o diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index d694e08e44e18..69c912feb8a7e 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -392,4 +392,8 @@ static inline void avs_debugfs_init(struct avs_dev *adev) { } static inline void avs_debugfs_exit(struct avs_dev *adev) { } #endif +/* Filesystems integration */ + +extern const struct attribute_group *avs_attr_groups[]; + #endif /* __SOUND_SOC_INTEL_AVS_H */ diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index db78eb2f01080..273a902168562 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -776,6 +776,7 @@ static struct pci_driver avs_pci_driver = { .probe = avs_pci_probe, .remove = avs_pci_remove, .shutdown = avs_pci_shutdown, + .dev_groups = avs_attr_groups, .driver = { .pm = &avs_dev_pm, }, diff --git a/sound/soc/intel/avs/sysfs.c b/sound/soc/intel/avs/sysfs.c new file mode 100644 index 0000000000000..cce21636fbc02 --- /dev/null +++ b/sound/soc/intel/avs/sysfs.c @@ -0,0 +1,35 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2024 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski <cezary.rojewski(a)intel.com> +// Amadeusz Slawinski <amadeuszx.slawinski(a)linux.intel.com> +// + +#include <linux/sysfs.h> +#include "avs.h" + +static ssize_t fw_version_show(struct device *dev, struct device_attribute *attr, char *buf) +{ + struct avs_dev *adev = to_avs_dev(dev); + struct avs_fw_version *fw_version = &adev->fw_cfg.fw_version; + + return sysfs_emit(buf, "%d.%d.%d.%d\n", fw_version->major, fw_version->minor, + fw_version->hotfix, fw_version->build); +} +static DEVICE_ATTR_RO(fw_version); + +static struct attribute *avs_fw_attrs[] = { + &dev_attr_fw_version.attr, + NULL +}; + +static const struct attribute_group avs_attr_group = { + .name = "avs", + .attrs = avs_fw_attrs, +}; + +const struct attribute_group *avs_attr_groups[] = { + &avs_attr_group, + NULL +}; -- 2.34.1

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[PATCH 00/24] ASoC: Intel: Machine driver updates for 6.9
by Pierre-Louis Bossart 09 Feb '24

09 Feb '24

A fairly large update this time, with Brent Lu's continued cleanup of Chromebook cards, and updates for new SoundWire configurations. Bard Liao (11): ASoC: Intel: sof_sdw: use single rtd_init for rt_amps ASoC: Intel: add get_codec_dai_by_name helper function ASoC: Intel: sof_sdw_rt_sdca_jack_common: use helper to get codec dai by name ASoC: Intel: sof_sdw_rt711: use helper to get codec dai by name ASoC: Intel: sof_sdw_rt712_sdca: use helper to get codec dai by name ASoC: Intel: sof_sdw_rt700: use helper to get codec dai by name ASoC: Intel: sof_sdw_cs42l42: use helper to get codec dai by name ASoC: Intel: sof_sdw_rt5682: use helper to get codec dai by name ASoC: Intel: sof_sdw: add common sdw dai link init ASoC: Intel: sof_sdw: remove .init callbacks ASoC: Intel: sof_sdw: starts non sdw BE id with the highest sdw BE id Brent Lu (8): ASoC: Intel: sof_rt5682: board id cleanup for jsl boards ASoC: Intel: sof_rt5682: board id cleanup for tgl boards ASoC: Intel: sof_rt5682: board id cleanup for adl boards ASoC: Intel: sof_rt5682: board id cleanup for rpl boards ASoC: Intel: sof_rt5682: board id cleanup for mtl boards ASoC: Intel: sof_rt5682: dmi quirk cleanup for mtl boards ASoC: Intel: board_helpers: support DAI link order customization ASoC: Intel: sof_cs42l42: use common module for DAI link generation Chao Song (2): ASoC: Intel: soc-acpi: add RT712 support for LNL ASoC: Intel: soc-acpi-intel-lnl-match: Add rt722 support Charles Keepax (2): ASoC: Intel: ssp-common: Add stub for sof_ssp_get_codec_name ASoC: Intel: sof_sdw: Remove unused function prototypes mosomate (1): ASoC: Intel: common: DMI remap for rebranded Intel NUC M15 (LAPRC710) laptops drivers/soundwire/dmi-quirks.c | 8 + sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_board_helpers.c | 225 ++++++++++++------ sound/soc/intel/boards/sof_board_helpers.h | 29 +++ sound/soc/intel/boards/sof_cs42l42.c | 224 +++++------------ sound/soc/intel/boards/sof_rt5682.c | 151 +----------- sound/soc/intel/boards/sof_sdw.c | 106 +++++++-- sound/soc/intel/boards/sof_sdw_common.h | 82 ++----- sound/soc/intel/boards/sof_sdw_cs42l42.c | 35 ++- sound/soc/intel/boards/sof_sdw_cs42l43.c | 25 +- sound/soc/intel/boards/sof_sdw_cs_amp.c | 3 +- sound/soc/intel/boards/sof_sdw_maxim.c | 4 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 35 ++- sound/soc/intel/boards/sof_sdw_rt700.c | 35 ++- sound/soc/intel/boards/sof_sdw_rt711.c | 19 +- sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 39 ++- sound/soc/intel/boards/sof_sdw_rt715.c | 12 +- sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 12 +- sound/soc/intel/boards/sof_sdw_rt_amp.c | 53 +---- .../boards/sof_sdw_rt_sdca_jack_common.c | 19 +- sound/soc/intel/boards/sof_ssp_common.h | 8 + .../intel/common/soc-acpi-intel-adl-match.c | 12 +- .../intel/common/soc-acpi-intel-jsl-match.c | 10 +- .../intel/common/soc-acpi-intel-lnl-match.c | 102 ++++++++ .../intel/common/soc-acpi-intel-mtl-match.c | 4 +- .../intel/common/soc-acpi-intel-rpl-match.c | 4 +- .../intel/common/soc-acpi-intel-tgl-match.c | 6 +- 27 files changed, 585 insertions(+), 678 deletions(-) -- 2.40.1

2 25

ALSA-LIB: Support for format IEC958_SUBFRAME_LE in the plug plugin?
by Pavel Hofman 09 Feb '24

09 Feb '24

Hi, RPi has recently moved to the VC4 driver which accepts only IEC958_SUBFRAME_LE format https://github.com/torvalds/linux/blob/ecb1b8288dc7ccbdcb3b9df005fa1c0e0c03… . Since then people started to have issues with their previous configs which use only the plug plugin (plughw:X). Wrapping the device with the hdmi plugin solves the problem, as it should. But e.g. mainline java support for alsa offers only hw cards wrapped with the plug plugin (hardcoded, not possible to specify the device name directly). Stock Java apps then do not work with RPi HDMI. I was wondering if it made sense to add support for the IEC958 formats (using the iec958 plugin methods) to the plug plugin. It may be complicated with the hdmi_mode hint though, I do not know if there is any API to recognize HDMI vs. SPDIF. Maybe a different format specifically for HDMI could have been perhaps useful but it may be too late for that. Thank you very much for consideration. With regards, Pavel.

2 11

[PATCH] ALSA: hda/realtek: Add "Intel Reference board" SSID in the ALC256.
by Pierre-Louis Bossart 09 Feb '24

09 Feb '24

From: "Sayed, Karimuddin" <karimuddin.sayed(a)intel.com> Add "Intel Reference board" SSID in the alc256. Enable "power saving mode" and Enable "headset jack mode". Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Sayed, Karimuddin <karimuddin.sayed(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f6f16622f9cc..0052a654204d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10081,6 +10081,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x1252, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x12cc, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x10ec, 0x12f6, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP), -- 2.40.1

2 1

[PATCH] ALSA: HDA: intel-sdw-acpi: add kernel parameter to select alternate controller
by Pierre-Louis Bossart 09 Feb '24

09 Feb '24

Existing DSDT or SSDT platforms hard-code clock and frame shape configurations. For validation, we'd like to use alternate configurations. It's not always possible to generate new tables due to missing symbols, and modifying existing objects usually leads to AE_OBJECT_EXIST errors. The mechanism suggested in this patch is to add a NEW ACPI controller device with a different _ADR value. e.g. Scope (_SB_.PC00.RP08.PXSX.HDAS) { Device (SDWP) { Name (_ADR, 0x40000001) // _ADR: Address The desired _ADR can be passed as a parameter with options snd-intel-sdw-acpi sdw_ctrl_addr=0x40000001 This solution leads to minimal tables with just what the developers or validation engineers need, and without overriding any of the existing firmware definitions. It's consistent with the recommendation to extend ACPI definitions and not redefine them with a risk of conflict. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/hda/intel-sdw-acpi.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/hda/intel-sdw-acpi.c b/sound/hda/intel-sdw-acpi.c index b57d72ea4503..5f60658c6051 100644 --- a/sound/hda/intel-sdw-acpi.c +++ b/sound/hda/intel-sdw-acpi.c @@ -23,6 +23,10 @@ static int ctrl_link_mask; module_param_named(sdw_link_mask, ctrl_link_mask, int, 0444); MODULE_PARM_DESC(sdw_link_mask, "Intel link mask (one bit per link)"); +static ulong ctrl_addr = 0x40000000; +module_param_named(sdw_ctrl_addr, ctrl_addr, ulong, 0444); +MODULE_PARM_DESC(sdw_ctrl_addr, "Intel SoundWire Controller _ADR"); + static bool is_link_enabled(struct fwnode_handle *fw_node, u8 idx) { struct fwnode_handle *link; @@ -141,6 +145,9 @@ static acpi_status sdw_intel_acpi_cb(acpi_handle handle, u32 level, if (FIELD_GET(GENMASK(31, 28), adr) != SDW_LINK_TYPE) return AE_OK; /* keep going */ + if (adr != ctrl_addr) + return AE_OK; /* keep going */ + /* found the correct SoundWire controller */ info->handle = handle; -- 2.40.1

2 1

[PATCH v2] ASoC: cs35l56: Fix deadlock in ASP1 mixer register initialization
by Richard Fitzgerald 08 Feb '24

08 Feb '24

Rewrite the handling of ASP1 TX mixer mux initialization to prevent a deadlock during component_remove(). The firmware can overwrite the ASP1 TX mixer registers with system-specific settings. This is mainly for hardware that uses the ASP as a chip-to-chip link controlled by the firmware. Because of this the driver cannot know the starting state of the ASP1 mixer muxes until the firmware has been downloaded and rebooted. The original workaround for this was to queue a work function from the dsp_work() job. This work then read the register values (populating the regmap cache the first time around) and then called snd_soc_dapm_mux_update_power(). The problem with this is that it was ultimately triggered by cs35l56_component_probe() queueing dsp_work, which meant that it would be running in parallel with the rest of the ASoC component and card initialization. To prevent accessing DAPM before it was fully initialized the work function took the card mutex. But this would deadlock if cs35l56_component_remove() was called before the work job had completed, because ASoC calls component_remove() with the card mutex held. This new version removes the work function. Instead the regmap cache and DAPM mux widgets are initialized the first time any of the associated ALSA controls is read or written. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 07f7d6e7a124 ("ASoC: cs35l56: Fix for initializing ASP1 mixer registers") --- Change from V1: Move the check of asp1_mixer_widgets_initialized into cs35l56_sync_asp1_mixer_widgets_with_firmware(). --- sound/soc/codecs/cs35l56.c | 157 +++++++++++++++++-------------------- sound/soc/codecs/cs35l56.h | 2 +- 2 files changed, 75 insertions(+), 84 deletions(-) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index ebed5ab1245b..98d3957c66e7 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -68,6 +68,66 @@ static const char * const cs35l56_asp1_mux_control_names[] = { "ASP1 TX1 Source", "ASP1 TX2 Source", "ASP1 TX3 Source", "ASP1 TX4 Source" }; +static int cs35l56_sync_asp1_mixer_widgets_with_firmware(struct cs35l56_private *cs35l56) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cs35l56->component); + const char *prefix = cs35l56->component->name_prefix; + char full_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + const char *name; + struct snd_kcontrol *kcontrol; + struct soc_enum *e; + unsigned int val[4]; + int i, item, ret; + + if (cs35l56->asp1_mixer_widgets_initialized) + return 0; + + /* + * Resume so we can read the registers from silicon if the regmap + * cache has not yet been populated. + */ + ret = pm_runtime_resume_and_get(cs35l56->base.dev); + if (ret < 0) + return ret; + + /* Wait for firmware download and reboot */ + cs35l56_wait_dsp_ready(cs35l56); + + ret = regmap_bulk_read(cs35l56->base.regmap, CS35L56_ASP1TX1_INPUT, + val, ARRAY_SIZE(val)); + + pm_runtime_mark_last_busy(cs35l56->base.dev); + pm_runtime_put_autosuspend(cs35l56->base.dev); + + if (ret) { + dev_err(cs35l56->base.dev, "Failed to read ASP1 mixer regs: %d\n", ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(cs35l56_asp1_mux_control_names); ++i) { + name = cs35l56_asp1_mux_control_names[i]; + + if (prefix) { + snprintf(full_name, sizeof(full_name), "%s %s", prefix, name); + name = full_name; + } + + kcontrol = snd_soc_card_get_kcontrol(dapm->card, name); + if (!kcontrol) { + dev_warn(cs35l56->base.dev, "Could not find control %s\n", name); + continue; + } + + e = (struct soc_enum *)kcontrol->private_value; + item = snd_soc_enum_val_to_item(e, val[i] & CS35L56_ASP_TXn_SRC_MASK); + snd_soc_dapm_mux_update_power(dapm, kcontrol, item, e, NULL); + } + + cs35l56->asp1_mixer_widgets_initialized = true; + + return 0; +} + static int cs35l56_dspwait_asp1tx_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -78,9 +138,9 @@ static int cs35l56_dspwait_asp1tx_get(struct snd_kcontrol *kcontrol, unsigned int addr, val; int ret; - /* Wait for mux to be initialized */ - cs35l56_wait_dsp_ready(cs35l56); - flush_work(&cs35l56->mux_init_work); + ret = cs35l56_sync_asp1_mixer_widgets_with_firmware(cs35l56); + if (ret) + return ret; addr = cs35l56_asp1_mixer_regs[index]; ret = regmap_read(cs35l56->base.regmap, addr, &val); @@ -106,9 +166,9 @@ static int cs35l56_dspwait_asp1tx_put(struct snd_kcontrol *kcontrol, bool changed; int ret; - /* Wait for mux to be initialized */ - cs35l56_wait_dsp_ready(cs35l56); - flush_work(&cs35l56->mux_init_work); + ret = cs35l56_sync_asp1_mixer_widgets_with_firmware(cs35l56); + if (ret) + return ret; addr = cs35l56_asp1_mixer_regs[index]; val = snd_soc_enum_item_to_val(e, item); @@ -124,70 +184,6 @@ static int cs35l56_dspwait_asp1tx_put(struct snd_kcontrol *kcontrol, return changed; } -static void cs35l56_mark_asp1_mixer_widgets_dirty(struct cs35l56_private *cs35l56) -{ - struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cs35l56->component); - const char *prefix = cs35l56->component->name_prefix; - char full_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - const char *name; - struct snd_kcontrol *kcontrol; - struct soc_enum *e; - unsigned int val[4]; - int i, item, ret; - - /* - * Resume so we can read the registers from silicon if the regmap - * cache has not yet been populated. - */ - ret = pm_runtime_resume_and_get(cs35l56->base.dev); - if (ret < 0) - return; - - ret = regmap_bulk_read(cs35l56->base.regmap, CS35L56_ASP1TX1_INPUT, - val, ARRAY_SIZE(val)); - - pm_runtime_mark_last_busy(cs35l56->base.dev); - pm_runtime_put_autosuspend(cs35l56->base.dev); - - if (ret) { - dev_err(cs35l56->base.dev, "Failed to read ASP1 mixer regs: %d\n", ret); - return; - } - - snd_soc_card_mutex_lock(dapm->card); - WARN_ON(!dapm->card->instantiated); - - for (i = 0; i < ARRAY_SIZE(cs35l56_asp1_mux_control_names); ++i) { - name = cs35l56_asp1_mux_control_names[i]; - - if (prefix) { - snprintf(full_name, sizeof(full_name), "%s %s", prefix, name); - name = full_name; - } - - kcontrol = snd_soc_card_get_kcontrol(dapm->card, name); - if (!kcontrol) { - dev_warn(cs35l56->base.dev, "Could not find control %s\n", name); - continue; - } - - e = (struct soc_enum *)kcontrol->private_value; - item = snd_soc_enum_val_to_item(e, val[i] & CS35L56_ASP_TXn_SRC_MASK); - snd_soc_dapm_mux_update_power(dapm, kcontrol, item, e, NULL); - } - - snd_soc_card_mutex_unlock(dapm->card); -} - -static void cs35l56_mux_init_work(struct work_struct *work) -{ - struct cs35l56_private *cs35l56 = container_of(work, - struct cs35l56_private, - mux_init_work); - - cs35l56_mark_asp1_mixer_widgets_dirty(cs35l56); -} - static DECLARE_TLV_DB_SCALE(vol_tlv, -10000, 25, 0); static const struct snd_kcontrol_new cs35l56_controls[] = { @@ -936,14 +932,6 @@ static void cs35l56_dsp_work(struct work_struct *work) else cs35l56_patch(cs35l56, firmware_missing); - - /* - * Set starting value of ASP1 mux widgets. Updating a mux takes - * the DAPM mutex. Post this to a separate job so that DAPM - * power-up can wait for dsp_work to complete without deadlocking - * on the DAPM mutex. - */ - queue_work(cs35l56->dsp_wq, &cs35l56->mux_init_work); err: pm_runtime_mark_last_busy(cs35l56->base.dev); pm_runtime_put_autosuspend(cs35l56->base.dev); @@ -989,6 +977,13 @@ static int cs35l56_component_probe(struct snd_soc_component *component) debugfs_create_bool("can_hibernate", 0444, debugfs_root, &cs35l56->base.can_hibernate); debugfs_create_bool("fw_patched", 0444, debugfs_root, &cs35l56->base.fw_patched); + /* + * The widgets for the ASP1TX mixer can't be initialized + * until the firmware has been downloaded and rebooted. + */ + regcache_drop_region(cs35l56->base.regmap, CS35L56_ASP1TX1_INPUT, CS35L56_ASP1TX4_INPUT); + cs35l56->asp1_mixer_widgets_initialized = false; + queue_work(cs35l56->dsp_wq, &cs35l56->dsp_work); return 0; @@ -999,7 +994,6 @@ static void cs35l56_component_remove(struct snd_soc_component *component) struct cs35l56_private *cs35l56 = snd_soc_component_get_drvdata(component); cancel_work_sync(&cs35l56->dsp_work); - cancel_work_sync(&cs35l56->mux_init_work); if (cs35l56->dsp.cs_dsp.booted) wm_adsp_power_down(&cs35l56->dsp); @@ -1070,10 +1064,8 @@ int cs35l56_system_suspend(struct device *dev) dev_dbg(dev, "system_suspend\n"); - if (cs35l56->component) { + if (cs35l56->component) flush_work(&cs35l56->dsp_work); - cancel_work_sync(&cs35l56->mux_init_work); - } /* * The interrupt line is normally shared, but after we start suspending @@ -1224,7 +1216,6 @@ static int cs35l56_dsp_init(struct cs35l56_private *cs35l56) return -ENOMEM; INIT_WORK(&cs35l56->dsp_work, cs35l56_dsp_work); - INIT_WORK(&cs35l56->mux_init_work, cs35l56_mux_init_work); dsp = &cs35l56->dsp; cs35l56_init_cs_dsp(&cs35l56->base, &dsp->cs_dsp); diff --git a/sound/soc/codecs/cs35l56.h b/sound/soc/codecs/cs35l56.h index 596b141e3f96..b000e7365e40 100644 --- a/sound/soc/codecs/cs35l56.h +++ b/sound/soc/codecs/cs35l56.h @@ -34,7 +34,6 @@ struct cs35l56_private { struct wm_adsp dsp; /* must be first member */ struct cs35l56_base base; struct work_struct dsp_work; - struct work_struct mux_init_work; struct workqueue_struct *dsp_wq; struct snd_soc_component *component; struct regulator_bulk_data supplies[CS35L56_NUM_BULK_SUPPLIES]; @@ -52,6 +51,7 @@ struct cs35l56_private { u8 asp_slot_count; bool tdm_mode; bool sysclk_set; + bool asp1_mixer_widgets_initialized; u8 old_sdw_clock_scale; }; -- 2.30.2

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[PATCH v2 0/3] ASoC: Intel: avs: Add support for sending initial module config
by Amadeusz Sławiński 08 Feb '24

08 Feb '24

In some cases it may be desirable to provide default initial configuration for FW module using payload. To facilitate such solution extend topology to contain initial config information, parse it and then send data to FW if present. Amadeusz Sławiński (3): ASoC: Intel: avs: UAPI: Add tokens for initial config feature ASoC: Intel: avs: Add topology parsing support for initial config ASoC: Intel: avs: Send initial config to module if present include/uapi/sound/intel/avs/tokens.h | 9 ++ sound/soc/intel/avs/path.c | 33 ++++++ sound/soc/intel/avs/topology.c | 164 +++++++++++++++++++++++++- sound/soc/intel/avs/topology.h | 13 ++ 4 files changed, 217 insertions(+), 2 deletions(-) -- 2.34.1

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[PATCH v13 00/53] Introduce QC USB SND audio offloading support
by Wesley Cheng 08 Feb '24

08 Feb '24

Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addresed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. USB SND: - Added a platform ops during usb_audio_suspend(). This allows for the USB offload driver to halt the audio stream when system enters PM suspend. This ensures the audio DSP is not issuing transfers on the USB bus. - Do not override platform ops if they are already populated. - Introduce a shared status variable between the USB offload and USB SND layers, to ensure that only one path is active at a time. If the USB bus is occupied, then userspace is notified that the path is busy. Mathias Nyman (13): xhci: fix possible null pointer dereference at secondary interrupter removal xhci: fix off by one check when adding a secondary interrupter. xhci: Add interrupt pending autoclear flag to each interrupter xhci: Add helper to set an interrupters interrupt moderation interval xhci: make isoc_bei_interval variable interrupter specific. xhci: remove unnecessary event_ring_deq parameter from xhci_handle_event() xhci: update event ring dequeue pointer position to controller correctly xhci: move event processing for one interrupter to a separate function xhci: add helper that checks for unhandled events on a event ring xhci: Don't check if the event ring is valid before every event TRB xhci: Decouple handling an event from checking for unhandled events xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (40): usb: host: xhci: Export enable and disable interrupter APIs usb: host: xhci: Repurpose event handler for skipping interrupter events xhci: export XHCI IMOD setting helper for interrupters usb: host: xhci-sideband: Expose a sideband interrupter enable API usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Save UAC sample size information usb: dwc3: Specify maximum number of XHCI interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: Check for support for requested audio format ASoC: usb: Add PCM format check API for USB backend ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: Do not allow USB offload path if PCM device is in use ASoC: dt-bindings: Add Q6USB backend ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ALSA: usb-audio: qcom: Populate PCM and USB chip information ASoC: qcom: qdsp6: Add support to track available USB PCM devices ASoC: Introduce SND kcontrols to select sound card and PCM device ASoC: qcom: qdsp6: Add SOC USB offload select get/put callbacks ASoC: Add SND kcontrol for fetching USB offload status ASoC: qcom: qdsp6: Add PCM ops to track current state ASoC: usb: Create SOC USB SND jack kcontrol ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: usb: Fetch ASoC sound card information ALSA: usb-audio: mixer: Add USB offloading mixer control ALSA: usb-audio: qcom: Use card and PCM index from QMI request ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add ASoC: qcom: Populate SoC components string ASoC: doc: Add documentation for SOC USB .../devicetree/bindings/sound/qcom,q6usb.yaml | 55 + .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 611 ++++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 5 +- drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 53 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 240 ++- drivers/usb/host/xhci-sideband.c | 439 ++++ drivers/usb/host/xhci.c | 97 +- drivers/usb/host/xhci.h | 21 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-sideband.h | 70 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 90 + sound/soc/Makefile | 2 +- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/common.c | 41 + sound/soc/qcom/common.h | 4 +- sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 193 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 401 ++++ sound/soc/qcom/sm8250.c | 14 +- sound/soc/soc-usb.c | 538 +++++ sound/usb/Kconfig | 19 + sound/usb/Makefile | 3 +- sound/usb/card.c | 123 ++ sound/usb/card.h | 24 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer.c | 5 + sound/usb/mixer_usb_offload.c | 82 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 94 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1910 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 49 files changed, 6243 insertions(+), 176 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6usb.yaml create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

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