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

[PATCH v1] slimbus: qcom-ngd-ctrl: Add timeout for wait operation
by Viken Dadhaniya 06 Mar '24

06 Mar '24

In current driver qcom_slim_ngd_up_worker() indefinitely waiting for ctrl->qmi_up completion object. This is resulting in workqueue lockup on Kthread. Added wait_for_completion_interruptible_timeout to allow the thread to wait for specific timeout period and bail out instead waiting infinitely. Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index efeba8275a66..c27076d8b7db 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -83,6 +83,7 @@ #define QCOM_SLIM_NGD_AUTOSUSPEND MSEC_PER_SEC #define SLIM_RX_MSGQ_TIMEOUT_VAL 0x10000 +#define SLIM_QMI_TIMEOUT_MS 1000 #define SLIM_LA_MGR 0xFF #define SLIM_ROOT_FREQ 24576000 @@ -1451,7 +1452,11 @@ static void qcom_slim_ngd_up_worker(struct work_struct *work) ctrl = container_of(work, struct qcom_slim_ngd_ctrl, ngd_up_work); /* Make sure qmi service is up before continuing */ - wait_for_completion_interruptible(&ctrl->qmi_up); + if (!wait_for_completion_interruptible_timeout(&ctrl->qmi_up, + msecs_to_jiffies(SLIM_QMI_TIMEOUT_MS))) { + dev_err(ctrl->dev, "QMI wait timeout\n"); + return; + } mutex_lock(&ctrl->ssr_lock); qcom_slim_ngd_enable(ctrl, true); -- QUALCOMM INDIA, on behalf of Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

2 2

[PATCH] ASoC: da7219: read fmw property to get mclk for non-dts systems
by cujomalainey＠chromium.org 06 Mar '24

06 Mar '24

From: Akshu Agrawal <akshu.agrawal(a)amd.com> Non-dts based systems can use ACPI DSDT to pass on the mclk to da7219. This enables da7219 mclk to be linked to system clock. Enable/Disable of the mclk is already handled in the codec so platform drivers don't have to explicitly do handling of mclk. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Karthik Ramasubramanian <kramasub(a)google.com> --- include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219.c | 8 +++++++- 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/include/sound/da7219.h b/include/sound/da7219.h index dde4542b084f3..4cf4c52be3f7f 100644 --- a/include/sound/da7219.h +++ b/include/sound/da7219.h @@ -40,6 +40,8 @@ struct da7219_pdata { const char *dai_clk_names[DA7219_DAI_NUM_CLKS]; + const char *mclk_name; + /* Mic */ enum da7219_micbias_voltage micbias_lvl; enum da7219_mic_amp_in_sel mic_amp_in_sel; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 311ea7918b312..800f61d993043 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1782,6 +1782,8 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct device *dev) pdata->dai_clk_names[DA7219_DAI_WCLK_IDX], pdata->dai_clk_names[DA7219_DAI_BCLK_IDX]); + device_property_read_string(dev, "dlg,mclk-name", &pdata->mclk_name); + if (device_property_read_u32(dev, "dlg,micbias-lvl", &of_val32) >= 0) pdata->micbias_lvl = da7219_fw_micbias_lvl(dev, of_val32); else @@ -2521,7 +2523,11 @@ static int da7219_probe(struct snd_soc_component *component) da7219_handle_pdata(component); /* Check if MCLK provided */ - da7219->mclk = clk_get(component->dev, "mclk"); + if (da7219->pdata->mclk_name) + da7219->mclk = clk_get(NULL, da7219->pdata->mclk_name); + if (!da7219->mclk) + da7219->mclk = clk_get(component->dev, "mclk"); + if (IS_ERR(da7219->mclk)) { if (PTR_ERR(da7219->mclk) != -ENOENT) { ret = PTR_ERR(da7219->mclk); -- 2.44.0.rc1.240.g4c46232300-goog

3 4

Dmic nhlt update
by GitHub pull_request - opened 05 Mar '24

05 Mar '24

alsa-project/alsa-utils pull request #254 was opened from jsarha: Driver version 4 update from for DMIC NHLT generator, and some error print improvements. Request URL : https://github.com/alsa-project/alsa-utils/pull/254 Patch URL : https://github.com/alsa-project/alsa-utils/pull/254.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v3] ASoC: soc-core.c: Prefer to return dai->driver->name in snd_soc_dai_name_get()
by Chancel Liu 05 Mar '24

05 Mar '24

ASoC machine driver can use snd_soc_{of_}get_dlc() (A) to get DAI name for dlc (snd_soc_dai_link_component). In this function call dlc->dai_name is parsed via snd_soc_dai_name_get() (B). (A) int snd_soc_get_dlc(...) { ... (B) dlc->dai_name = snd_soc_dai_name_get(dai); ... } (B) has a priority to return dai->name as dlc->dai_name. In most cases card can probe successfully. However it has an issue that ASoC tries to rebind card. Here is a simplified flow for example: | a) Card probes successfully at first | b) One of the component bound to this card is removed for some | reason the component->dev is released | c) That component is re-registered v d) ASoC calls snd_soc_try_rebind_card() a) points dlc->dai_name to dai->name. b) releases all resource of the old DAI. c) creates new DAI structure. In result d) can not use dlc->dai_name to add new created DAI. So it's reasonable that prefer to return dai->driver->name in snd_soc_dai_name_get() because dai->driver is a pre-defined global variable. Also update snd_soc_is_matching_dai() for alignment. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- changes in v2: - update snd_soc_is_matching_dai() for alignment changes in v3: - resolve conflicts when applying patch, no others changed sound/soc/soc-core.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 507cd3015ff4..1e94edba12eb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -283,13 +283,13 @@ static int snd_soc_is_matching_dai(const struct snd_soc_dai_link_component *dlc, /* see snd_soc_dai_name_get() */ - if (strcmp(dlc->dai_name, dai->name) == 0) - return 1; - if (dai->driver->name && strcmp(dlc->dai_name, dai->driver->name) == 0) return 1; + if (strcmp(dlc->dai_name, dai->name) == 0) + return 1; + if (dai->component->name && strcmp(dlc->dai_name, dai->component->name) == 0) return 1; @@ -300,12 +300,12 @@ static int snd_soc_is_matching_dai(const struct snd_soc_dai_link_component *dlc, const char *snd_soc_dai_name_get(struct snd_soc_dai *dai) { /* see snd_soc_is_matching_dai() */ - if (dai->name) - return dai->name; - if (dai->driver->name) return dai->driver->name; + if (dai->name) + return dai->name; + if (dai->component->name) return dai->component->name; -- 2.43.0

2 1

[PATCH v3] ASoC: cs-amp-lib: Add KUnit test for calibration helpers
by Richard Fitzgerald 05 Mar '24

05 Mar '24

Add a KUnit test for the cs-amp-lib library. This has test cases for cs_amp_get_efi_calibration_data() and cs_amp_write_cal_coeffs(). A KUNIT_STATIC_STUB_REDIRECT() has been added to cs_amp_get_efi_variable() and cs_amp_write_cal_coeff() so that the KUnit test can redirect these to test harness functions. Much of the testing involves invoking the same function with different parameters, i.e. the number of amps and the amp index within the array. This uses parameterization rather than looping. The idea is to avoid looping over configurations within one test case as that has a higher chance of having a bug that doesn't actually test all the expected cases. Having the test run exactly one configuration, and then tear-down, is less prone to accidentally skipped configurations. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- include/sound/cs-amp-lib.h | 14 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs-amp-lib-test.c | 709 +++++++++++++++++++++++++++++ sound/soc/codecs/cs-amp-lib.c | 18 +- 5 files changed, 754 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/cs-amp-lib-test.c diff --git a/include/sound/cs-amp-lib.h b/include/sound/cs-amp-lib.h index 077fe36885b5..f481148735e1 100644 --- a/include/sound/cs-amp-lib.h +++ b/include/sound/cs-amp-lib.h @@ -49,4 +49,18 @@ int cs_amp_write_cal_coeffs(struct cs_dsp *dsp, const struct cirrus_amp_cal_data *data); int cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, int amp_index, struct cirrus_amp_cal_data *out_data); + +struct cs_amp_test_hooks { + efi_status_t (*get_efi_variable)(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf); + + int (*write_cal_coeff)(struct cs_dsp *dsp, + const struct cirrus_amp_cal_controls *controls, + const char *ctl_name, u32 val); +}; + +extern const struct cs_amp_test_hooks * const cs_amp_test_hooks; + #endif /* CS_AMP_LIB_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 15f287784d8b..f78ea2f86fa6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -732,6 +732,19 @@ config SND_SOC_CROS_EC_CODEC config SND_SOC_CS_AMP_LIB tristate +config SND_SOC_CS_AMP_LIB_TEST + tristate "KUnit test for Cirrus Logic cs-amp-lib" + depends on KUNIT + default KUNIT_ALL_TESTS + select SND_SOC_CS_AMP_LIB + help + This builds KUnit tests for the Cirrus Logic common + amplifier library. + For more information on KUnit and unit tests in general, + please refer to the KUnit documentation in + Documentation/dev-tools/kunit/. + If in doubt, say "N". + config SND_SOC_CS35L32 tristate "Cirrus Logic CS35L32 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 0fc40640e5d0..7c075539dc47 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -60,6 +60,7 @@ snd-soc-cpcap-objs := cpcap.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cros-ec-codec-objs := cros_ec_codec.o snd-soc-cs-amp-lib-objs := cs-amp-lib.o +snd-soc-cs-amp-lib-test-objs := cs-amp-lib-test.o snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o @@ -454,6 +455,7 @@ obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CPCAP) += snd-soc-cpcap.o obj-$(CONFIG_SND_SOC_CROS_EC_CODEC) += snd-soc-cros-ec-codec.o obj-$(CONFIG_SND_SOC_CS_AMP_LIB) += snd-soc-cs-amp-lib.o +obj-$(CONFIG_SND_SOC_CS_AMP_LIB_TEST) += snd-soc-cs-amp-lib-test.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o diff --git a/sound/soc/codecs/cs-amp-lib-test.c b/sound/soc/codecs/cs-amp-lib-test.c new file mode 100644 index 000000000000..15f991b2e16e --- /dev/null +++ b/sound/soc/codecs/cs-amp-lib-test.c @@ -0,0 +1,709 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// KUnit test for the Cirrus common amplifier library. +// +// Copyright (C) 2024 Cirrus Logic, Inc. and +// Cirrus Logic International Semiconductor Ltd. + +#include <kunit/test.h> +#include <kunit/static_stub.h> +#include <linux/firmware/cirrus/cs_dsp.h> +#include <linux/firmware/cirrus/wmfw.h> +#include <linux/gpio/driver.h> +#include <linux/list.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/random.h> +#include <sound/cs-amp-lib.h> + +struct cs_amp_lib_test_priv { + struct platform_device amp_pdev; + + struct cirrus_amp_efi_data *cal_blob; + struct list_head ctl_write_list; +}; + +struct cs_amp_lib_test_ctl_write_entry { + struct list_head list; + unsigned int value; + char name[16]; +}; + +struct cs_amp_lib_test_param { + int num_amps; + int amp_index; +}; + +static void cs_amp_lib_test_init_dummy_cal_blob(struct kunit *test, int num_amps) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + unsigned int blob_size; + + blob_size = offsetof(struct cirrus_amp_efi_data, data) + + sizeof(struct cirrus_amp_cal_data) * num_amps; + + priv->cal_blob = kunit_kzalloc(test, blob_size, GFP_KERNEL); + KUNIT_ASSERT_NOT_NULL(test, priv->cal_blob); + + priv->cal_blob->size = blob_size; + priv->cal_blob->count = num_amps; + + get_random_bytes(priv->cal_blob->data, sizeof(struct cirrus_amp_cal_data) * num_amps); +} + +static u64 cs_amp_lib_test_get_target_uid(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + u64 uid; + + uid = priv->cal_blob->data[param->amp_index].calTarget[1]; + uid <<= 32; + uid |= priv->cal_blob->data[param->amp_index].calTarget[0]; + + return uid; +} + +/* Redirected get_efi_variable to simulate that the file is too short */ +static efi_status_t cs_amp_lib_test_get_efi_variable_nohead(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf) +{ + if (!buf) { + *size = offsetof(struct cirrus_amp_efi_data, data) - 1; + return EFI_BUFFER_TOO_SMALL; + } + + return EFI_NOT_FOUND; +} + +/* Should return -EOVERFLOW if the header is larger than the EFI data */ +static void cs_amp_lib_test_cal_data_too_short_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int ret; + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable_nohead); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, 0, &result_data); + KUNIT_EXPECT_EQ(test, ret, -EOVERFLOW); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* Redirected get_efi_variable to simulate that the count is larger than the file */ +static efi_status_t cs_amp_lib_test_get_efi_variable_bad_count(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf) +{ + struct kunit *test = kunit_get_current_test(); + struct cs_amp_lib_test_priv *priv = test->priv; + + if (!buf) { + /* + * Return a size that is shorter than required for the + * declared number of entries. + */ + *size = priv->cal_blob->size - 1; + return EFI_BUFFER_TOO_SMALL; + } + + memcpy(buf, priv->cal_blob, priv->cal_blob->size - 1); + + return EFI_SUCCESS; +} + +/* Should return -EOVERFLOW if the entry count is larger than the EFI data */ +static void cs_amp_lib_test_cal_count_too_big_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable_bad_count); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, 0, &result_data); + KUNIT_EXPECT_EQ(test, ret, -EOVERFLOW); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* Redirected get_efi_variable to simulate that the variable not found */ +static efi_status_t cs_amp_lib_test_get_efi_variable_none(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf) +{ + return EFI_NOT_FOUND; +} + +/* If EFI doesn't contain a cal data variable the result should be -ENOENT */ +static void cs_amp_lib_test_no_cal_data_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int ret; + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable_none); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, 0, &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* Redirected get_efi_variable to simulate reading a cal data blob */ +static efi_status_t cs_amp_lib_test_get_efi_variable(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf) +{ + static const efi_char16_t expected_name[] = L"CirrusSmartAmpCalibrationData"; + static const efi_guid_t expected_guid = + EFI_GUID(0x02f9af02, 0x7734, 0x4233, 0xb4, 0x3d, 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); + struct kunit *test = kunit_get_current_test(); + struct cs_amp_lib_test_priv *priv = test->priv; + + KUNIT_EXPECT_NOT_ERR_OR_NULL(test, name); + KUNIT_EXPECT_NOT_ERR_OR_NULL(test, guid); + KUNIT_EXPECT_NOT_ERR_OR_NULL(test, size); + + KUNIT_EXPECT_MEMEQ(test, name, expected_name, sizeof(expected_name)); + KUNIT_EXPECT_MEMEQ(test, guid, &expected_guid, sizeof(expected_guid)); + + if (!buf) { + *size = priv->cal_blob->size; + return EFI_BUFFER_TOO_SMALL; + } + + KUNIT_ASSERT_GE_MSG(test, ksize(buf), priv->cal_blob->size, "Buffer to small"); + + memcpy(buf, priv->cal_blob, priv->cal_blob->size); + + return EFI_SUCCESS; +} + +/* Get cal data block for a given amp, matched by target UID. */ +static void cs_amp_lib_test_get_efi_cal_by_uid_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + struct cirrus_amp_cal_data result_data; + u64 target_uid; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, param->num_amps); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + target_uid = cs_amp_lib_test_get_target_uid(test); + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, target_uid, -1, &result_data); + KUNIT_EXPECT_EQ(test, ret, 0); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); + + KUNIT_EXPECT_EQ(test, result_data.calTarget[0], target_uid & 0xFFFFFFFFULL); + KUNIT_EXPECT_EQ(test, result_data.calTarget[1], target_uid >> 32); + KUNIT_EXPECT_EQ(test, result_data.calTime[0], + priv->cal_blob->data[param->amp_index].calTime[0]); + KUNIT_EXPECT_EQ(test, result_data.calTime[1], + priv->cal_blob->data[param->amp_index].calTime[1]); + KUNIT_EXPECT_EQ(test, result_data.calAmbient, + priv->cal_blob->data[param->amp_index].calAmbient); + KUNIT_EXPECT_EQ(test, result_data.calStatus, + priv->cal_blob->data[param->amp_index].calStatus); + KUNIT_EXPECT_EQ(test, result_data.calR, + priv->cal_blob->data[param->amp_index].calR); +} + +/* Get cal data block for a given amp index without checking target UID. */ +static void cs_amp_lib_test_get_efi_cal_by_index_unchecked_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + struct cirrus_amp_cal_data result_data; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, param->num_amps); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, + param->amp_index, &result_data); + KUNIT_EXPECT_EQ(test, ret, 0); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); + + KUNIT_EXPECT_EQ(test, result_data.calTime[0], + priv->cal_blob->data[param->amp_index].calTime[0]); + KUNIT_EXPECT_EQ(test, result_data.calTime[1], + priv->cal_blob->data[param->amp_index].calTime[1]); + KUNIT_EXPECT_EQ(test, result_data.calAmbient, + priv->cal_blob->data[param->amp_index].calAmbient); + KUNIT_EXPECT_EQ(test, result_data.calStatus, + priv->cal_blob->data[param->amp_index].calStatus); + KUNIT_EXPECT_EQ(test, result_data.calR, + priv->cal_blob->data[param->amp_index].calR); +} + +/* Get cal data block for a given amp index with checked target UID. */ +static void cs_amp_lib_test_get_efi_cal_by_index_checked_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + struct cirrus_amp_cal_data result_data; + u64 target_uid; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, param->num_amps); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + target_uid = cs_amp_lib_test_get_target_uid(test); + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, target_uid, + param->amp_index, &result_data); + KUNIT_EXPECT_EQ(test, ret, 0); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); + + KUNIT_EXPECT_EQ(test, result_data.calTime[0], + priv->cal_blob->data[param->amp_index].calTime[0]); + KUNIT_EXPECT_EQ(test, result_data.calTime[1], + priv->cal_blob->data[param->amp_index].calTime[1]); + KUNIT_EXPECT_EQ(test, result_data.calAmbient, + priv->cal_blob->data[param->amp_index].calAmbient); + KUNIT_EXPECT_EQ(test, result_data.calStatus, + priv->cal_blob->data[param->amp_index].calStatus); + KUNIT_EXPECT_EQ(test, result_data.calR, + priv->cal_blob->data[param->amp_index].calR); +} + +/* + * Get cal data block for a given amp index with checked target UID. + * The UID does not match so the result should be -ENOENT. + */ +static void cs_amp_lib_test_get_efi_cal_by_index_uid_mismatch_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + struct cirrus_amp_cal_data result_data; + u64 target_uid; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, param->num_amps); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + /* Get a target UID that won't match the entry */ + target_uid = ~cs_amp_lib_test_get_target_uid(test); + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, target_uid, + param->amp_index, &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* + * Get cal data block for a given amp, where the cal data does not + * specify calTarget so the lookup falls back to using the index + */ +static void cs_amp_lib_test_get_efi_cal_by_index_fallback_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + struct cirrus_amp_cal_data result_data; + static const u64 bad_target_uid = 0xBADCA100BABABABAULL; + int i, ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, param->num_amps); + + /* Make all the target values zero so they are ignored */ + for (i = 0; i < priv->cal_blob->count; ++i) { + priv->cal_blob->data[i].calTarget[0] = 0; + priv->cal_blob->data[i].calTarget[1] = 0; + } + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, bad_target_uid, + param->amp_index, &result_data); + KUNIT_EXPECT_EQ(test, ret, 0); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); + + KUNIT_EXPECT_EQ(test, result_data.calTime[0], + priv->cal_blob->data[param->amp_index].calTime[0]); + KUNIT_EXPECT_EQ(test, result_data.calTime[1], + priv->cal_blob->data[param->amp_index].calTime[1]); + KUNIT_EXPECT_EQ(test, result_data.calAmbient, + priv->cal_blob->data[param->amp_index].calAmbient); + KUNIT_EXPECT_EQ(test, result_data.calStatus, + priv->cal_blob->data[param->amp_index].calStatus); + KUNIT_EXPECT_EQ(test, result_data.calR, + priv->cal_blob->data[param->amp_index].calR); +} + +/* + * If the target UID isn't present in the cal data, and there isn't an + * index to fall back do, the result should be -ENOENT. + */ +static void cs_amp_lib_test_get_efi_cal_uid_not_found_noindex_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + static const u64 bad_target_uid = 0xBADCA100BABABABAULL; + int i, ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Make all the target values != bad_target_uid */ + for (i = 0; i < priv->cal_blob->count; ++i) { + priv->cal_blob->data[i].calTarget[0] &= ~(bad_target_uid & 0xFFFFFFFFULL); + priv->cal_blob->data[i].calTarget[1] &= ~(bad_target_uid >> 32); + } + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, bad_target_uid, -1, + &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* + * If the target UID isn't present in the cal data, and the index is + * out of range, the result should be -ENOENT. + */ +static void cs_amp_lib_test_get_efi_cal_uid_not_found_index_not_found_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + static const u64 bad_target_uid = 0xBADCA100BABABABAULL; + int i, ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Make all the target values != bad_target_uid */ + for (i = 0; i < priv->cal_blob->count; ++i) { + priv->cal_blob->data[i].calTarget[0] &= ~(bad_target_uid & 0xFFFFFFFFULL); + priv->cal_blob->data[i].calTarget[1] &= ~(bad_target_uid >> 32); + } + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, bad_target_uid, 99, + &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* + * If the target UID isn't given, and the index is out of range, the + * result should be -ENOENT. + */ +static void cs_amp_lib_test_get_efi_cal_no_uid_index_not_found_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, 99, &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* If neither the target UID or the index is given the result should be -ENOENT. */ +static void cs_amp_lib_test_get_efi_cal_no_uid_no_index_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, -1, &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* + * If the UID is passed as 0 this must not match an entry with an + * unpopulated calTarget + */ +static void cs_amp_lib_test_get_efi_cal_zero_not_matched_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int i, ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Make all the target values zero so they are ignored */ + for (i = 0; i < priv->cal_blob->count; ++i) { + priv->cal_blob->data[i].calTarget[0] = 0; + priv->cal_blob->data[i].calTarget[1] = 0; + } + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, -1, &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +static const struct cirrus_amp_cal_controls cs_amp_lib_test_calibration_controls = { + .alg_id = 0x9f210, + .mem_region = WMFW_ADSP2_YM, + .ambient = "CAL_AMBIENT", + .calr = "CAL_R", + .status = "CAL_STATUS", + .checksum = "CAL_CHECKSUM", +}; + +static int cs_amp_lib_test_write_cal_coeff(struct cs_dsp *dsp, + const struct cirrus_amp_cal_controls *controls, + const char *ctl_name, u32 val) +{ + struct kunit *test = kunit_get_current_test(); + struct cs_amp_lib_test_priv *priv = test->priv; + struct cs_amp_lib_test_ctl_write_entry *entry; + + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, ctl_name); + KUNIT_EXPECT_PTR_EQ(test, controls, &cs_amp_lib_test_calibration_controls); + + entry = kunit_kzalloc(test, sizeof(*entry), GFP_KERNEL); + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, entry); + + INIT_LIST_HEAD(&entry->list); + strscpy(entry->name, ctl_name, sizeof(entry->name)); + entry->value = val; + + list_add_tail(&entry->list, &priv->ctl_write_list); + + return 0; +} + +static void cs_amp_lib_test_write_cal_data_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cs_amp_lib_test_ctl_write_entry *entry; + struct cirrus_amp_cal_data data; + struct cs_dsp *dsp; + int ret; + + dsp = kunit_kzalloc(test, sizeof(*dsp), GFP_KERNEL); + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, dsp); + dsp->dev = &priv->amp_pdev.dev; + + get_random_bytes(&data, sizeof(data)); + + /* Redirect calls to write firmware controls */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->write_cal_coeff, + cs_amp_lib_test_write_cal_coeff); + + ret = cs_amp_write_cal_coeffs(dsp, &cs_amp_lib_test_calibration_controls, &data); + KUNIT_EXPECT_EQ(test, ret, 0); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->write_cal_coeff); + + KUNIT_EXPECT_EQ(test, list_count_nodes(&priv->ctl_write_list), 4); + + /* Checksum control must be written last */ + entry = list_last_entry(&priv->ctl_write_list, typeof(*entry), list); + KUNIT_EXPECT_STREQ(test, entry->name, cs_amp_lib_test_calibration_controls.checksum); + KUNIT_EXPECT_EQ(test, entry->value, data.calR + 1); + list_del(&entry->list); + + entry = list_first_entry(&priv->ctl_write_list, typeof(*entry), list); + KUNIT_EXPECT_STREQ(test, entry->name, cs_amp_lib_test_calibration_controls.ambient); + KUNIT_EXPECT_EQ(test, entry->value, data.calAmbient); + list_del(&entry->list); + + entry = list_first_entry(&priv->ctl_write_list, typeof(*entry), list); + KUNIT_EXPECT_STREQ(test, entry->name, cs_amp_lib_test_calibration_controls.calr); + KUNIT_EXPECT_EQ(test, entry->value, data.calR); + list_del(&entry->list); + + entry = list_first_entry(&priv->ctl_write_list, typeof(*entry), list); + KUNIT_EXPECT_STREQ(test, entry->name, cs_amp_lib_test_calibration_controls.status); + KUNIT_EXPECT_EQ(test, entry->value, data.calStatus); +} + +static void cs_amp_lib_test_dev_release(struct device *dev) +{ +} + +static int cs_amp_lib_test_case_init(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv; + int ret; + + KUNIT_ASSERT_NOT_NULL(test, cs_amp_test_hooks); + + priv = kunit_kzalloc(test, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + test->priv = priv; + INIT_LIST_HEAD(&priv->ctl_write_list); + + /* Create dummy amp driver dev */ + priv->amp_pdev.name = "cs_amp_lib_test_drv"; + priv->amp_pdev.id = -1; + priv->amp_pdev.dev.release = cs_amp_lib_test_dev_release; + ret = platform_device_register(&priv->amp_pdev); + KUNIT_ASSERT_GE_MSG(test, ret, 0, "Failed to register amp platform device\n"); + + return 0; +} + +static void cs_amp_lib_test_case_exit(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + + if (priv->amp_pdev.name) + platform_device_unregister(&priv->amp_pdev); +} + +static const struct cs_amp_lib_test_param cs_amp_lib_test_get_cal_param_cases[] = { + { .num_amps = 2, .amp_index = 0 }, + { .num_amps = 2, .amp_index = 1 }, + + { .num_amps = 3, .amp_index = 0 }, + { .num_amps = 3, .amp_index = 1 }, + { .num_amps = 3, .amp_index = 2 }, + + { .num_amps = 4, .amp_index = 0 }, + { .num_amps = 4, .amp_index = 1 }, + { .num_amps = 4, .amp_index = 2 }, + { .num_amps = 4, .amp_index = 3 }, + + { .num_amps = 5, .amp_index = 0 }, + { .num_amps = 5, .amp_index = 1 }, + { .num_amps = 5, .amp_index = 2 }, + { .num_amps = 5, .amp_index = 3 }, + { .num_amps = 5, .amp_index = 4 }, + + { .num_amps = 6, .amp_index = 0 }, + { .num_amps = 6, .amp_index = 1 }, + { .num_amps = 6, .amp_index = 2 }, + { .num_amps = 6, .amp_index = 3 }, + { .num_amps = 6, .amp_index = 4 }, + { .num_amps = 6, .amp_index = 5 }, + + { .num_amps = 8, .amp_index = 0 }, + { .num_amps = 8, .amp_index = 1 }, + { .num_amps = 8, .amp_index = 2 }, + { .num_amps = 8, .amp_index = 3 }, + { .num_amps = 8, .amp_index = 4 }, + { .num_amps = 8, .amp_index = 5 }, + { .num_amps = 8, .amp_index = 6 }, + { .num_amps = 8, .amp_index = 7 }, +}; + +static void cs_amp_lib_test_get_cal_param_desc(const struct cs_amp_lib_test_param *param, + char *desc) +{ + snprintf(desc, KUNIT_PARAM_DESC_SIZE, "num_amps:%d amp_index:%d", + param->num_amps, param->amp_index); +} + +KUNIT_ARRAY_PARAM(cs_amp_lib_test_get_cal, cs_amp_lib_test_get_cal_param_cases, + cs_amp_lib_test_get_cal_param_desc); + +static struct kunit_case cs_amp_lib_test_cases[] = { + /* Tests for getting calibration data from EFI */ + KUNIT_CASE(cs_amp_lib_test_cal_data_too_short_test), + KUNIT_CASE(cs_amp_lib_test_cal_count_too_big_test), + KUNIT_CASE(cs_amp_lib_test_no_cal_data_test), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_uid_not_found_noindex_test), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_uid_not_found_index_not_found_test), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_no_uid_index_not_found_test), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_no_uid_no_index_test), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_zero_not_matched_test), + KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_uid_test, + cs_amp_lib_test_get_cal_gen_params), + KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_index_unchecked_test, + cs_amp_lib_test_get_cal_gen_params), + KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_index_checked_test, + cs_amp_lib_test_get_cal_gen_params), + KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_index_uid_mismatch_test, + cs_amp_lib_test_get_cal_gen_params), + KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_index_fallback_test, + cs_amp_lib_test_get_cal_gen_params), + + /* Tests for writing calibration data */ + KUNIT_CASE(cs_amp_lib_test_write_cal_data_test), + + { } /* terminator */ +}; + +static struct kunit_suite cs_amp_lib_test_suite = { + .name = "snd-soc-cs-amp-lib-test", + .init = cs_amp_lib_test_case_init, + .exit = cs_amp_lib_test_case_exit, + .test_cases = cs_amp_lib_test_cases, +}; + +kunit_test_suite(cs_amp_lib_test_suite); + +MODULE_IMPORT_NS(SND_SOC_CS_AMP_LIB); +MODULE_DESCRIPTION("KUnit test for Cirrus Logic amplifier library"); +MODULE_AUTHOR("Richard Fitzgerald <rf(a)opensource.cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs-amp-lib.c b/sound/soc/codecs/cs-amp-lib.c index 4e2e5157a73f..01ef4db5407d 100644 --- a/sound/soc/codecs/cs-amp-lib.c +++ b/sound/soc/codecs/cs-amp-lib.c @@ -6,6 +6,7 @@ // Cirrus Logic International Semiconductor Ltd. #include <asm/byteorder.h> +#include <kunit/static_stub.h> #include <linux/dev_printk.h> #include <linux/efi.h> #include <linux/firmware/cirrus/cs_dsp.h> @@ -27,6 +28,8 @@ static int cs_amp_write_cal_coeff(struct cs_dsp *dsp, __be32 beval = cpu_to_be32(val); int ret; + KUNIT_STATIC_STUB_REDIRECT(cs_amp_write_cal_coeff, dsp, controls, ctl_name, val); + if (IS_REACHABLE(CONFIG_FW_CS_DSP)) { mutex_lock(&dsp->pwr_lock); cs_ctl = cs_dsp_get_ctl(dsp, ctl_name, controls->mem_region, controls->alg_id); @@ -84,7 +87,7 @@ int cs_amp_write_cal_coeffs(struct cs_dsp *dsp, const struct cirrus_amp_cal_controls *controls, const struct cirrus_amp_cal_data *data) { - if (IS_REACHABLE(CONFIG_FW_CS_DSP)) + if (IS_REACHABLE(CONFIG_FW_CS_DSP) || IS_ENABLED(CONFIG_SND_SOC_CS_AMP_LIB_TEST)) return _cs_amp_write_cal_coeffs(dsp, controls, data); else return -ENODEV; @@ -98,6 +101,8 @@ static efi_status_t cs_amp_get_efi_variable(efi_char16_t *name, { u32 attr; + KUNIT_STATIC_STUB_REDIRECT(cs_amp_get_efi_variable, name, guid, size, buf); + if (IS_ENABLED(CONFIG_EFI)) return efi.get_variable(name, guid, &attr, size, buf); @@ -250,13 +255,22 @@ static int _cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, int cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, int amp_index, struct cirrus_amp_cal_data *out_data) { - if (IS_ENABLED(CONFIG_EFI)) + if (IS_ENABLED(CONFIG_EFI) || IS_ENABLED(CONFIG_SND_SOC_CS_AMP_LIB_TEST)) return _cs_amp_get_efi_calibration_data(dev, target_uid, amp_index, out_data); else return -ENOENT; } EXPORT_SYMBOL_NS_GPL(cs_amp_get_efi_calibration_data, SND_SOC_CS_AMP_LIB); +static const struct cs_amp_test_hooks cs_amp_test_hook_ptrs = { + .get_efi_variable = cs_amp_get_efi_variable, + .write_cal_coeff = cs_amp_write_cal_coeff, +}; + +const struct cs_amp_test_hooks * const cs_amp_test_hooks = + PTR_IF(IS_ENABLED(CONFIG_SND_SOC_CS_AMP_LIB_TEST), &cs_amp_test_hook_ptrs); +EXPORT_SYMBOL_NS_GPL(cs_amp_test_hooks, SND_SOC_CS_AMP_LIB); + MODULE_DESCRIPTION("Cirrus Logic amplifier library"); MODULE_AUTHOR("Richard Fitzgerald <rf(a)opensource.cirrus.com>"); MODULE_LICENSE("GPL"); -- 2.39.2

2 1

[PATCH v10] The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp speaker protection algorithm. The integrated speaker voltage and current sense provides for real-time monitoring of loudspeakers.
by Shenghao Ding 05 Mar '24

05 Mar '24

The ASoC component provides the majority of the functionality of the device, all the audio functions. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v10: - simplify tasdevice_set_sdw_stream - fixed some Linux coding style - fixed the spelling mistakes - Select left/right channel based on unique id - a longer description has been added - remove unused crc8 in KCONFIG - Invert of RE here means reciprocal of RE - correct indentation - correct the wrong pow calibrated register - correct comment on RESET register - fixed usually lines don't end with an open parenthesis - fixed tasdevice_sdw_shutdown - use defines for function, entity, etc. for readable - remove unnecessary codec_lock in component_probe() - remove struct sdw_stream_data - remove unused macro definitions - use BIT(2) and BIT(1) instead of 0x2 and 0x1 - implement pm_runtime - reg_start is u8, reg_start + i only adds i byte, it should be reg_start + i * 4, because the calibration values - fixed total_sz in tas2783_calibration - add a define for TASDEVICE_REG(0, 0, 0x0a) - rework macro style in header file - Add select CRC32 - rewrite the mute and the return value handling - Add comments on bin file - Add link_id into the fw name to distinguish the same sdw id on different sdw id. - remove unnecessary mutex_lock - use set_active() in tasdevice_io_init and move all other parts in the probe. - move pm_runtime_get_noresume to the probe - fix the return value of putvol function - add put_noidle in tasdevice_sdw_remove - remove unused header file(of.h) - correct the style of the multi-line comment - Set the readable registers in tas2783_readable_register for debugfs - Use new PM macros - Use default put and get inline SOC_SINGLE_RANGE_TLV and SOC_SINGLE_SX_TLV - Use respective %pt to log the calibrated time - Remove unnecessary struct tm - Add trailing comma tasdevice_dapm_widgets and tasdevice_audio_map - Rewrite the return value handling in tasdevice_sdw_probe - use dev_num instead of id.unique_id - rework the log - Add code to check the size of calibrated data in UEFI - Add description of Layout of calibrated Data in UEFI - switch back link_id and unique_id to identify the devcie - unique_id offset as the device index of the calibrated data - update calibrated data layout stored in UEFI, add link_id and unique_id into the calibrated data in case of tas2783 sitting in different SDQW buses. --- sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 932 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 106 ++++ 4 files changed, 1056 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 59f9742e9ff4..6ba3fd0c0811 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -247,6 +247,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1847,6 +1848,21 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + depends on EFI + select REGMAP + select REGMAP_SOUNDWIRE + select CRC32 + select CRC32_SARWATE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + TAS2783 supports the MIPI SoundWire v1.2 compatible interface + for audio and control data. At the end of calibrated data appends + the crc data with crc32_sarwate for error detection. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f53baa2b9565..6afceb126ae3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -284,6 +284,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -664,6 +665,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..7bd151942a83 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,932 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Kevin Lu <kevin-lu(a)ti.com> +// Shenghao Ding <shenghao-ding(a)ti.com> +// + +#include <asm/unaligned.h> +#include <linux/acpi.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/pcm_params.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include <sound/tas2781-tlv.h> +#include "tas2783.h" + +static const unsigned int tas2783_cali_reg[] = { + TAS2783_CALIBRATION_RE, /* Resistance */ + TAS2783_CALIBRATION_RE_LOW, /* Low limitation of RE */ + TAS2783_CALIBRATION_INV_RE, /* Reciprocal of RE */ + TAS2783_CALIBRATION_POW, /* RMS Power */ + TAS2783_CALIBRATION_TLIMIT /* Temperature limitation */ +}; + +static const struct reg_default tas2783_reg_defaults[] = { + /* Default values for ROM mode. Activated. */ + { 0x8002, 0x1a }, /* AMP inactive. */ + { 0x8097, 0xc8 }, + { 0x80b5, 0x74 }, + { 0x8099, 0x20 }, + { 0xfe8d, 0x0d }, + { 0xfebe, 0x4a }, + { 0x8230, 0x00 }, + { 0x8231, 0x00 }, + { 0x8232, 0x00 }, + { 0x8233, 0x01 }, + { 0x8418, 0x00 }, /* Set volume to 0 dB. */ + { 0x8419, 0x00 }, + { 0x841a, 0x00 }, + { 0x841b, 0x00 }, + { 0x8428, 0x40 }, /* Unmute channel */ + { 0x8429, 0x00 }, + { 0x842a, 0x00 }, + { 0x842b, 0x00 }, + { 0x8548, 0x00 }, /* Set volume to 0 dB. */ + { 0x8549, 0x00 }, + { 0x854a, 0x00 }, + { 0x854b, 0x00 }, + { 0x8558, 0x40 }, /* Unmute channel */ + { 0x8559, 0x00 }, + { 0x855a, 0x00 }, + { 0x855b, 0x00 }, + { 0x800a, 0x3a }, /* Enable both channel */ + { 0x800e, 0x44 }, + { 0x800f, 0x40 }, + { 0x805c, 0x99 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_VOLUME, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0x807F: /* Page 0 */ + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + /* + * Only software reset register was volatiled. + * Bit is self clearing. + * 0b = Don't reset + * 1b = reset + */ + case 0x8001: + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x44ffffff, + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, dvc_tlv), + SOC_SINGLE_SX_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, amp_vol_tlv), +}; + +static void tas2783_apply_calibv1(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + struct regmap *map = tas_dev->regmap; + u8 *cali_start; + u16 offset; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + offset = tas_dev->sdw_peripheral->id.unique_id - + TAS2783_UNIQUE_ID_BASE; + if (offset >= TAS2783_MAX_DEV_NUM) { + dev_err(tas_dev->dev, "%s: offset(%u) is out of range.\n", + __func__, offset); + return; + } + + /* + * The area saving tas2783 calibrated data is specified by its + * unique_id offset. cali_start is the first address of current + * tas2783's calibrated data. + */ + cali_start = (u8 *)(cali_data + offset * sizeof(tas2783_cali_reg)); + for (int i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + ret = regmap_bulk_write(map, tas2783_cali_reg[i], + &cali_start[4 * i], 4); + if (ret) { + dev_err(tas_dev->dev, "Cali failed %x:%d\n", + tas2783_cali_reg[i], ret); + break; + } + } +} + +static void tas2783_apply_calibv2(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + const unsigned int arr_size = ARRAY_SIZE(tas2783_cali_reg); + struct regmap *map = tas_dev->regmap; + unsigned int dev_sum = cali_data[1], i, j, k; + u8 *cali_start; + u16 dev_info; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + dev_info = tas_dev->sdw_peripheral->bus->link_id | + tas_dev->sdw_peripheral->id.unique_id << 16; + + /* + * The area saving tas2783 calibrated data is specified by its + * unique_id offset. cali_start is the first address of current + * tas2783's calibrated data. + */ + cali_start = (u8 *)&cali_data[3]; + for (i = 0; i < dev_sum; i++) { + k = i * (arr_size + 1) + 3; + if (dev_info != cali_data[k]) { + for (j = 0; j < arr_size; j++) { + k = 4 * (k + 1 + j); + ret = regmap_bulk_write(map, + tas2783_cali_reg[j], + &cali_start[k], 4); + if (ret) { + dev_err(tas_dev->dev, + "Cali failed %x:%d\n", + tas2783_cali_reg[j], ret); + break; + } + } + break; + } + } +} + +/* + * Load the calibration data, including speaker impedance, f0, etc. + * Calibration is done by the manufacturer in the factory. The calibration + * data are used by the algorithm for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * The DSP will work with default data values if calibrated data are + * missing or are invalid. + * Layout of calibrated Data in UEFI(total 256 bytes): + * V2: + * ChipID (2783, 4 bytes) + * Device-Sum (4 bytes) + * TimeStamp of Calibration (4 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i info () { + * SDW link id (2bytes) + * SDW unique_id (2bytes) + * } + * Calibrated Data of Device #i (20 bytes) + * } + * CRC (4 bytes) + * Reserved (the rest) + * Or + * V1: + * Calibrated Data of Dev 0(unique_id offset 0) (20 bytes) + * Calibrated Data of Dev 1(unique_id offset 1) (20 bytes) + * Calibrated Data of Dev 2(unique_id offset 2) (20 bytes) + * Calibrated Data of Dev 3(unique_id offset 3) (20 bytes) + * Calibrated Data of Dev 4(unique_id offset 4) (20 bytes) + * Calibrated Data of Dev 5(unique_id offset 5) (20 bytes) + * Calibrated Data of Dev 6(unique_id offset 6) (20 bytes) + * Calibrated Data of Dev 7(unique_id offset 7) (20 bytes) + * TimeStamp of Calibration (4 bytes) + * CRC (4 bytes) + * Reserved (88 bytes) + */ +static void tas2783_calibration(struct tasdevice_priv *tas_dev) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct calibration_data cali_data; + unsigned int *tmp_val; + unsigned int crc; + efi_status_t status; + + cali_data.total_sz = 0; + + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, NULL); + if (status == EFI_BUFFER_TOO_SMALL + && cali_data.total_sz < TAS2783_MAX_CALIDATA_SIZE) { + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, + cali_data.data); + dev_dbg(tas_dev->dev, "%s: cali get %lx bytes result:%ld\n", + __func__, cali_data.total_sz, status); + } + if (status != 0) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return; + } + + tmp_val = (unsigned int *)cali_data.data; + + if (tmp_val[0] == 2783) { + /* Calibrated Data V2 */ + unsigned int dev_sum = tmp_val[1]; + + if (dev_sum > TAS2783_MAX_DEV_NUM || + dev_sum == 0) { + dev_dbg(tas_dev->dev, "No dev in calibrated data V2."); + return; + } + crc = crc32(~0, cali_data.data, 12 + dev_sum * 24) ^ ~0; + if (crc == tmp_val[3 + dev_sum * 6]) { + tas2783_apply_calibv2(tas_dev, tmp_val); + dev_dbg(tas_dev->dev, "V2: %ptTs", &tmp_val[40]); + } else { + dev_dbg(tas_dev->dev, + "V2: CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[41]); + } + } else { + /* Calibrated Data V1 */ + /* 8 devs * 20 bytes calibrated data/dev + 4 bytes Timestamp */ + crc = crc32(~0, cali_data.data, 164) ^ ~0; + if (crc == tmp_val[41]) { + /* Date and time of when calibration was done. */ + tas2783_apply_calibv1(tas_dev, tmp_val); + dev_dbg(tas_dev->dev, "V1: %ptTs", &tmp_val[40]); + } else { + dev_dbg(tas_dev->dev, + "V1: CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[41]); + } + } +} + +static void tasdevice_dspfw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct tas2783_firmware_node *p; + struct regmap *map = tas_dev->regmap; + unsigned char *buf = NULL; + int offset = 0, img_sz; + int ret; + + if (!fmw || !fmw->data) { + dev_warn(tas_dev->dev, + "%s: failed to read %s: work in bypass-dsp mode.\n", + __func__, tas_dev->dspfw_binaryname); + return; + } + buf = (unsigned char *)fmw->data; + + img_sz = get_unaligned_le32(&buf[offset]); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_warn(tas_dev->dev, "%s: size not matching, %d %u.", + __func__, (int)fmw->size, img_sz); + return; + } + + while (offset < img_sz) { + p = (struct tas2783_firmware_node *)(buf + offset); + if (p->length > 1) { + ret = regmap_bulk_write(map, p->download_addr, + buf + offset + sizeof(unsigned int) * 5, p->length); + } else { + ret = regmap_write(map, p->download_addr, + *(buf + offset + sizeof(unsigned int) * 5)); + } + + if (ret != 0) { + dev_dbg(tas_dev->dev, + "%s: load FW fail: %d, work in bypass.\n", + __func__, ret); + return; + } + offset += sizeof(unsigned int) * 5 + p->length; + } + + tas2783_calibration(tas_dev); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC"), +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"}, +}; + +static int tasdevice_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + int ret; + + dev_dbg(dai->dev, "%s: dai_name %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(tas_priv->sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) { + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); + return ret; + } + + dev_dbg(dai->dev, "%s: format: %i rate: %i\n", __func__, + params_format(params), params_rate(params)); + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tas_priv->sdw_peripheral, sdw_stream); + + return 0; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct regmap *map = tas_dev->regmap; + int ret; + + dev_dbg(tas_dev->dev, "%s: %d.\n", __func__, mute); + + if (mute) { + if (direction == SNDRV_PCM_STREAM_CAPTURE) { + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_INACTIVE); + if (ret) + dev_err(tas_dev->dev, + "%s: Disable AEF failed.\n", __func__); + } else { + /* FU23 mute (0x40400108) */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 1); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 mute failed.\n", __func__); + goto out; + } + /* + * Both playback and echo data will be shutdown in + * playback stream. + */ + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK | + TAS2783_REG_AEF_MASK, + TAS2783_REG_PWR_MODE_ACTIVE | + TAS2783_REG_PWR_MODE_SW_PWD); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR&AEF shutdown failed.\n", + __func__); + goto out; + } + tas_dev->pstream = false; + } + } else { + /* FU23 Unmute, 0x40400108. */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 Unmute failed.\n", __func__); + goto out; + } + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK, + TAS2783_REG_PWR_MODE_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR Unmute failed.\n", __func__); + goto out; + } + tas_dev->pstream = true; + } else { + /* Capture stream is the echo ref data for voice. + * Without playback, it can't be active. + */ + if (tas_dev->pstream == true) { + ret = regmap_update_bits(map, + TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: AEF enable failed.\n", + __func__); + goto out; + } + } else { + dev_err(tas_dev->dev, + "%s: No playback, no AEF!", __func__); + ret = -EINVAL; + } + } + } +out: + if (ret) + dev_err(tas_dev->dev, "Mute or unmute %d failed %d.\n", + mute, ret); + + return ret; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_comp_probe(struct snd_soc_component *comp) +{ + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(comp); + acpi_handle handle = ACPI_HANDLE(tas_dev->dev); + const struct firmware *fw_entry = NULL; + const char *sub = NULL; + int ret; + + if (handle) { + sub = acpi_get_subsystem_id(handle); + if (IS_ERR(sub)) + sub = NULL; + } + + tas_dev->component = comp; + + /* + * Each tas2783 in the system has its own dspfw. + */ + if (comp->name_prefix) { + /* + * name_prefix.bin stores the dsp firmware including speaker + * protection algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied into + * firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, "%s-tas2783.bin", + comp->name_prefix); + } else { + /* Compatible with the previous naming rule */ + if (sub) { + /* + * subsys_id-link_id[0,1,...,N]-unique_id[1,...,4].bin + * stores the dsp firmware including speaker protection + * algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied + * into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "%s-%d-%d.bin", sub, + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } else { + /* + * tas2783-link_id[0,1,...,N]-unique_id[1,...,4].bin + * stores the dsp firmware including speaker protection + * algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied + * into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "tas2783-%d-%d.bin", + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } + } + + ret = request_firmware(&fw_entry, tas_dev->dspfw_binaryname, + tas_dev->dev); + if (ret) { + dev_err(tas_dev->dev, + "%s: request_firmware %x open status: %d.\n", __func__, + tas_dev->sdw_peripheral->id.unique_id, ret); + goto out; + } + + tasdevice_dspfw_ready(fw_entry, tas_dev); + +out: + if (fw_entry) + release_firmware(fw_entry); + return 0; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_comp_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + return ret; + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + pm_runtime_enable(tas_dev->dev); + + return 0; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + struct sdw_dpn_prop *dpn; + unsigned long addr; + int nval, i, j; + u32 bit; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = BIT(2); /* BITMAP: 00000100 */ + prop->sink_ports = BIT(1); /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static int tasdevice_io_init(struct device *dev, + struct sdw_slave *slave) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(tas_priv->regmap, false); + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + * Update count of parent 'active' children. + */ + pm_runtime_set_active(&slave->dev); + } + + /* sw reset */ + ret = regmap_write(tas_priv->regmap, TAS2873_REG_SWRESET, + TAS2873_REG_SWRESET_RESET); + if (ret) { + dev_err(tas_priv->dev, "Reset failed.\n"); + goto out; + } + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, false); + regcache_mark_dirty(tas_priv->regmap); + } + + tas_priv->first_hw_init = true; + tas_priv->hw_init = true; + +out: + return ret; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tas_priv->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tas_priv->hw_init = false; + /* + * Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_priv->hw_init || tas_priv->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return -ENOMEM; + + tas_dev->dev = dev; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_peripheral = peripheral; + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) + return dev_err_probe(dev, PTR_ERR(tas_dev->regmap), + "Failed devm_regmap_init_sdw."); + + return tasdevice_init(tas_dev); +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(&peripheral->dev); + + if (tas_dev->first_hw_init) + pm_runtime_disable(tas_dev->dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {} +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static int __maybe_unused tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (!tas_priv->hw_init) + return 0; + + regcache_cache_only(tas_priv->regmap, true); + + return 0; +} + +#define TAS2783_PROBE_TIMEOUT_MS 5000 + +static int __maybe_unused tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long time; + + if (!tas_priv->first_hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT_MS)); + if (!time) { + dev_err(&slave->dev, "Init not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(tas_priv->regmap, false); + regcache_sync(tas_priv->regmap); + + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume) + RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume, NULL) +}; + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = &tas2783_sdca_pm, + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..631776dcec4c --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,106 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2023 - 2024 Texas Instruments Incorporated + * https://www.ti.com + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + * Shenghao Ding <shenghao-ding(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +/* Unique id start */ +#define TAS2783_UNIQUE_ID_BASE 0x08 +#define TAS2783_MAX_DEV_NUM 8 + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 + \ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0x00, 0x01) +#define TAS2873_REG_SWRESET_MASK BIT(0) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +#define TAS2873_REG_PWR_CTRL TASDEVICE_REG(0x0, 0x00, 0x02) +#define TAS2783_REG_AEF_MASK GENMASK(4, 3) +#define TAS2783_REG_AEF_ACTIVE 0 +#define TAS2783_REG_AEF_INACTIVE GENMASK(4, 3) +#define TAS2783_REG_PWR_MODE_MASK GENMASK(2, 0) +#define TAS2783_REG_PWR_MODE_ACTIVE 0 +#define TAS2783_REG_PWR_MODE_SW_PWD BIT(1) + +#define TAS2783_REG_TDM_RX_CFG TASDEVICE_REG(0x0, 0x00, 0x0a) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1a) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x13, 0x70) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +/* TAS2783 SDCA Control - function number */ +#define TAS2783_FUNC_TYPE_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_FU_VOLUME 0x02 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 +#define TAS2783_DSPFW_FILENAME_LEN 64 + +#define TAS2783_MAX_CALIDATA_SIZE 256 + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct snd_soc_component *component; + struct sdw_slave *sdw_peripheral; + enum sdw_slave_status status; + struct sdw_bus_params params; + struct regmap *regmap; + struct device *dev; + unsigned char dspfw_binaryname[TAS2783_DSPFW_FILENAME_LEN]; + unsigned char dev_name[32]; + unsigned int chip_id; + bool pstream; + bool hw_init; + bool first_hw_init; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

2 1

[PATCH] mfd: cs42l43: Fix wrong GPIO_FN_SEL and SPI_CLK_CONFIG1 defaults
by Maciej Strozek 05 Mar '24

05 Mar '24

Two regs have wrong values in existing fields, change them to match the datasheet. Fixes: ace6d1448138 ("mfd: cs42l43: Add support for cs42l43 core driver") Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- drivers/mfd/cs42l43.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/mfd/cs42l43.c b/drivers/mfd/cs42l43.c index 73c88ee6a866..1cea3f8f467d 100644 --- a/drivers/mfd/cs42l43.c +++ b/drivers/mfd/cs42l43.c @@ -84,7 +84,7 @@ const struct reg_default cs42l43_reg_default[CS42L43_N_DEFAULTS] = { { CS42L43_DRV_CTRL_5, 0x136C00C0 }, { CS42L43_GPIO_CTRL1, 0x00000707 }, { CS42L43_GPIO_CTRL2, 0x00000000 }, - { CS42L43_GPIO_FN_SEL, 0x00000000 }, + { CS42L43_GPIO_FN_SEL, 0x00000004 }, { CS42L43_MCLK_SRC_SEL, 0x00000000 }, { CS42L43_SAMPLE_RATE1, 0x00000003 }, { CS42L43_SAMPLE_RATE2, 0x00000003 }, @@ -217,7 +217,7 @@ const struct reg_default cs42l43_reg_default[CS42L43_N_DEFAULTS] = { { CS42L43_CTRL_REG, 0x00000006 }, { CS42L43_FDIV_FRAC, 0x40000000 }, { CS42L43_CAL_RATIO, 0x00000080 }, - { CS42L43_SPI_CLK_CONFIG1, 0x00000000 }, + { CS42L43_SPI_CLK_CONFIG1, 0x00000001 }, { CS42L43_SPI_CONFIG1, 0x00000000 }, { CS42L43_SPI_CONFIG2, 0x00000000 }, { CS42L43_SPI_CONFIG3, 0x00000001 }, -- 2.30.2

3 2

[BUG] snd_fireface: frozen buffer with full system lockup
by edmund.raile 04 Mar '24

04 Mar '24

Hello, I want to report a bug with snd_fireface. issue audio suddenly freezes, repeating the last buffer to infinity other symptoms Most of the time, the GUI also freezes to a point that no interaction is possible. Not even a user console can be reached (ctrl + alt + F#). The only way to get the system back in this scenario is to use REISUB or power it down, I can not reach any console. Luckily, sometimes the kernel messages at the bottom are written to the logfile. occurrence This issue is triggered by more "stressful" circumstances, specifically while using the Windows VST Kontakt in linux-native reaper through yabridge (wine). The timeframe is 5-20 minutes. Reaper connects to the interface using ALSA, no other application is playing audio. My default sample rate is 96k, increasing buffer size to 512 samples decreases CPU load but does not prevent this crash. variables I tried to exclude OS Manjaro or fresh Artix install makes no difference host application the crash happens just as well with linux-native bitwig under similar circumstances wine variants wine, wine-staging, wine-tkg w esync and fsync patches The reason why I attribute both these issues to snd_firewire is that I get no freezes / instabilities at all under these circumstances: * in Linux, using an usb audio interface or the onboard audio instead. To me, this suggests Wine, Reaper and Kontakt are not likely the culprit. * in Windows, using RME's dirver. To me, this suggests FW643 and my platform have no issue working together and the FireFace works as it should. * in Linux, passing the FW643 card through to a windows VM with vfio-pci. Here I am just booting the same physical drive in QEMU, so using the same RME driver. To me, this suggests there is no strange interaction between linux and FW643. system Interface: RME FireFace 800 FireWire controller: LSI FW643 rev 8 Platform: MSI ms7d25 Z690 Kernel: 6.8.0-rc5-1-mainline #8 SMP PREEMPT_DYNAMIC OS: Artix Xfce video driver: mesa An related issue? During normal use, as of recent kernel versions, just playing back audio from any pulse audio application, I get a similar freeze but less severe: no other applications are blocked and it can be removed by just stoping playback, then after 5 seconds the audible freeze stops and I can start playback again. This happens less often, 5-60 minutes between freezes. I managed to capture this unicorn event yesterday in a relatively short file using 'sudo trace-cmd record -e snd_firewire_lib:amdtp_packet', maybe it can give you a hint as to what may be going on: https://pixeldrain.com/u/tgoSA1mk I tried to do the same for my real issue (the one with the full system freeze but I can not stop the recording and the output after rebooting are files split by cpu cores. I hope this is the right place to turn to for help. It has prevented me from using my interface in linux for audio production for a long time and I want to help fix it. If I can provide any form of further assistance or somehow collect any more data, please let me know. Kind regards, Edmund Raile. watchdog: Watchdog detected hard LOCKUP on cpu 13 Modules linked in: ntfs3 fuse 8021q garp mrp stp llc bnep bluetooth ecdh_generic rfkill vfat fat snd_sof_pci_intel_tgl snd_sof_intel_hda_common soundwire_intel snd_sof_intel_hda_mlink snd_hda_codec_hdmi soundwire_cadence snd_sof_intel_hda snd_sof_pci snd_sof_xtensa_dsp snd_sof intel_rapl_msr snd_sof_utils intel_rapl_common snd_soc_hdac_hda snd_hda_ext_core snd_soc_acpi_intel_match intel_uncore_frequency intel_uncore_frequency_common snd_soc_acpi intel_tcc_cooling soundwire_generic_allocation snd_hda_codec_realtek x86_pkg_temp_thermal soundwire_bus intel_powerclamp snd_hda_codec_generic snd_soc_core coretemp snd_compress kvm_intel ac97_bus snd_pcm_dmaengine snd_hda_intel snd_usb_audio snd_intel_dspcfg snd_fireface snd_intel_sdw_acpi kvm snd_usbmidi_lib snd_dice snd_hda_codec snd_firewire_lib snd_ump snd_hda_core snd_rawmidi irqbypass mc snd_hwdep snd_seq_device rapl iTCO_wdt snd_pcm intel_pmc_bxt intel_cstate mei_pxp mei_hdcp iTCO_vendor_support ee1004 snd_timer intel_uncore psmouse pcspkr crypto_user spi_nor snd i2c_i801 mei_me mousedev wacom joydev mtd igc soundcore i2c_smbus intel_pmc_core intel_vsec mei pmt_telemetry pmt_class mac_hid coreboot_table intel_scu_pltdrv ext4 crc32c_generic crc16 mbcache jbd2 hid_logitech ff_memless hid_logitech_hidpp hid_logitech_dj dm_crypt cbc uas usb_storage encrypted_keys trusted asn1_encoder tee dm_mod usbhid crct10dif_pclmul crc32_pclmul crc32c_intel polyval_clmulni polyval_generic gf128mul ghash_clmulni_intel sha512_ssse3 sha256_ssse3 sha1_ssse3 serio_raw aesni_intel atkbd nvme libps2 spi_intel_pci vivaldi_fmap firewire_ohci crypto_simd nvme_core firewire_core cryptd xhci_pci spi_intel nvme_auth xhci_pci_renesas i8042 crc_itu_t serio i915 i2c_algo_bit drm_buddy video wmi ttm intel_gtt drm_display_helper cec CPU: 13 PID: 1993 Comm: reaper/audio Not tainted 6.8.0-rc5-1-mainline #8 c2061649224b2c533bd915b824b700586979f2da Hardware name: Micro-Star International Co., Ltd. MS-7D25/PRO Z690-A DDR4(MS-7D25), BIOS Dasharo (coreboot+UEFI) v1.1.3 01/17/2024 RIP: 0010:tasklet_unlock_spin_wait+0xd/0x20 Code: c0 c3 cc cc cc cc 0f 1f 80 00 00 00 00 90 90 90 90 90 90 90 90 90 90 90 90 90 90 90 90 f3 0f 1e fa 0f 1f 44 00 00 eb 02 f3 90 <48> 8b 47 08 a8 02 75 f6 c3 cc cc cc cc 66 0f 1f 44 00 00 90 90 90 RSP: 0018:ffffb828c1cdbd08 EFLAGS: 00000002 RAX: 0000000000000002 RBX: ffff951841aba028 RCX: 0000000000000000 RDX: 0000000000000001 RSI: ffff951841a2aa70 RDI: ffff951841aba0a8 RBP: ffffb828c1cdbea0 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000075800 R14: ffff951846a9f600 R15: 0000000000000000 FS: 00007f7f394006c0(0000) GS:ffff952b9fb40000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007fc65c2a0000 CR3: 000000018516a000 CR4: 0000000000f50ef0 PKRU: 55555554 Call Trace: <NMI> ? watchdog_hardlockup_check+0xfc/0x1d0 ? __perf_event_overflow+0xe5/0x2a0 ? handle_pmi_common+0x16f/0x380 ? intel_pmu_handle_irq+0x104/0x480 ? perf_event_nmi_handler+0x2a/0x50 ? nmi_handle+0x5e/0x150 ? default_do_nmi+0x40/0x100 ? exc_nmi+0x139/0x1c0 ? end_repeat_nmi+0xf/0x60 ? tasklet_unlock_spin_wait+0xd/0x20 ? tasklet_unlock_spin_wait+0xd/0x20 ? tasklet_unlock_spin_wait+0xd/0x20 </NMI> <TASK> ohci_flush_iso_completions+0x1d/0x100 [firewire_ohci 83160820eab58a73fec0d1e78730dd51dd0ac98c] amdtp_domain_stream_pcm_pointer+0x41/0x50 [snd_firewire_lib 95578abd4cf50f7c2e87b0d8233b959be2154fc3] snd_pcm_update_hw_ptr0+0x6c/0x430 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] ? sched_clock_cpu+0xf/0x190 ? __smp_call_single_queue+0xad/0x120 snd_pcm_delay+0x33/0x140 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] snd_pcm_common_ioctl+0x922/0x12b0 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] ? wake_up_q+0x4e/0x90 ? futex_wake+0x159/0x190 snd_pcm_ioctl+0x2e/0x50 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] __x64_sys_ioctl+0x94/0xd0 do_syscall_64+0x86/0x170 ? do_syscall_64+0x96/0x170 ? do_syscall_64+0x96/0x170 ? do_syscall_64+0x96/0x170 ? do_syscall_64+0x96/0x170 ? __irq_exit_rcu+0x4b/0xc0 entry_SYSCALL_64_after_hwframe+0x6e/0x76 RIP: 0033:0x7f7f47cfd4ff Code: 00 48 89 44 24 18 31 c0 48 8d 44 24 60 c7 04 24 10 00 00 00 48 89 44 24 08 48 8d 44 24 20 48 89 44 24 10 b8 10 00 00 00 0f 05 <89> c2 3d 00 f0 ff ff 77 18 48 8b 44 24 18 64 48 2b 04 25 28 00 00 RSP: 002b:00007f7f393ffd50 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000003cfc060 RCX: 00007f7f47cfd4ff RDX: 00007f7f393ffdf0 RSI: 0000000080084121 RDI: 000000000000000a RBP: 0000000000000001 R08: 0000000000000001 R09: 0000000000000004 R10: 00007f7f393ffb20 R11: 0000000000000246 R12: 0000000000005000 R13: 0000000003272a80 R14: 0000000000000050 R15: 00000000032706c0 </TASK> watchdog: Watchdog detected hard LOCKUP on cpu 6 CPU: 6 PID: 0 Comm: swapper/6 Not tainted 6.8.0-rc5-1-mainline #8 c2061649224b2c533bd915b824b700586979f2da Hardware name: Micro-Star International Co., Ltd. MS-7D25/PRO Z690-A DDR4(MS-7D25), BIOS Dasharo (coreboot+UEFI) v1.1.3 01/17/2024 RIP: 0010:native_queued_spin_lock_slowpath+0x6e/0x2e0 Code: 77 7f f0 0f ba 2b 08 0f 92 c2 8b 03 0f b6 d2 c1 e2 08 30 e4 09 d0 3d ff 00 00 00 77 5b 85 c0 74 10 0f b6 03 84 c0 74 09 f3 90 <0f> b6 03 84 c0 75 f7 b8 01 00 00 00 66 89 03 65 48 ff 05 83 a6 23 RSP: 0018:ffffb828c01ece08 EFLAGS: 00000002 RAX: 0000000000000001 RBX: ffff951846a9f6f8 RCX: 0000000000000300 RDX: 0000000000000000 RSI: 0000000000000001 RDI: ffff951846a9f6f8 RBP: ffff95184684c438 R08: ffff951879f3f800 R09: ffff951841a94028 R10: ffff951879f3f800 R11: 0000000000000000 R12: 0000000000000031 R13: 0000000000000000 R14: ffff9518a562e000 R15: ffff9518c49e5800 FS: 0000000000000000(0000) GS:ffff952b9f980000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00000001010fdac8 CR3: 000000018516a000 CR4: 0000000000f50ef0 PKRU: 55555554 Call Trace: <NMI> ? watchdog_hardlockup_check+0xfc/0x1d0 ? __perf_event_overflow+0xe5/0x2a0 ? handle_pmi_common+0x16f/0x380 ? intel_pmu_handle_irq+0x104/0x480 ? perf_event_nmi_handler+0x2a/0x50 ? nmi_handle+0x5e/0x150 ? default_do_nmi+0x40/0x100 ? exc_nmi+0x139/0x1c0 ? end_repeat_nmi+0xf/0x60 ? native_queued_spin_lock_slowpath+0x6e/0x2e0 ? native_queued_spin_lock_slowpath+0x6e/0x2e0 ? native_queued_spin_lock_slowpath+0x6e/0x2e0 </NMI> <IRQ> _raw_spin_lock_irqsave+0x3d/0x50 snd_pcm_period_elapsed+0x18/0x40 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] process_rx_packets+0x245/0x5a0 [snd_firewire_lib 95578abd4cf50f7c2e87b0d8233b959be2154fc3] ? irqtime_account_irq+0x40/0xc0 irq_target_callback+0x16/0x20 [snd_firewire_lib 95578abd4cf50f7c2e87b0d8233b959be2154fc3] handle_it_packet+0x135/0x150 [firewire_ohci 83160820eab58a73fec0d1e78730dd51dd0ac98c] context_tasklet+0x140/0x160 [firewire_ohci 83160820eab58a73fec0d1e78730dd51dd0ac98c] tasklet_action_common.isra.0+0x146/0x240 __do_softirq+0xc9/0x2c8 __irq_exit_rcu+0xa3/0xc0 common_interrupt+0x86/0xa0 </IRQ> <TASK> asm_common_interrupt+0x26/0x40 RIP: 0010:cpuidle_enter_state+0xcc/0x440 Code: 2a 39 36 ff e8 05 f1 ff ff 8b 53 04 49 89 c5 0f 1f 44 00 00 31 ff e8 83 38 35 ff 45 84 ff 0f 85 56 02 00 00 fb 0f 1f 44 00 00 <45> 85 f6 0f 88 85 01 00 00 49 63 d6 48 8d 04 52 48 8d 04 82 49 8d RSP: 0018:ffffb828c00ffe90 EFLAGS: 00000246 RAX: ffff952b9f9b4800 RBX: ffff952b9f9be208 RCX: 000000000000001f RDX: 0000000000000006 RSI: 0000000022b8e38e RDI: 0000000000000000 RBP: 0000000000000001 R08: 0000000000000000 R09: 00000000000001c8 R10: 0000000000000018 R11: ffff952b9f9b31e4 R12: ffffffff83f49780 R13: 0000006b6114102b R14: 0000000000000001 R15: 0000000000000000 cpuidle_enter+0x2d/0x40 do_idle+0x1d8/0x230 cpu_startup_entry+0x2a/0x30 start_secondary+0x11e/0x140 secondary_startup_64_no_verify+0x184/0x18b </TASK> rcu: INFO: rcu_preempt detected stalls on CPUs/tasks: rcu: \x096-...0: (1 GPs behind) idle=d7ec/1/0x4000000000000002 softirq=52862/52863 fqs=5201 rcu: \x0913-...0: (1 GPs behind) idle=c9d4/1/0x4000000000000000 softirq=53654/53655 fqs=5201 rcu: \x09(detected by 2, t=18002 jiffies, g=76629, q=5124 ncpus=16) Sending NMI from CPU 2 to CPUs 6: NMI backtrace for cpu 6 CPU: 6 PID: 0 Comm: swapper/6 Not tainted 6.8.0-rc5-1-mainline #8 c2061649224b2c533bd915b824b700586979f2da Hardware name: Micro-Star International Co., Ltd. MS-7D25/PRO Z690-A DDR4(MS-7D25), BIOS Dasharo (coreboot+UEFI) v1.1.3 01/17/2024 RIP: 0010:native_queued_spin_lock_slowpath+0x6e/0x2e0 Code: 77 7f f0 0f ba 2b 08 0f 92 c2 8b 03 0f b6 d2 c1 e2 08 30 e4 09 d0 3d ff 00 00 00 77 5b 85 c0 74 10 0f b6 03 84 c0 74 09 f3 90 <0f> b6 03 84 c0 75 f7 b8 01 00 00 00 66 89 03 65 48 ff 05 83 a6 23 RSP: 0018:ffffb828c01ece08 EFLAGS: 00000002 RAX: 0000000000000001 RBX: ffff951846a9f6f8 RCX: 0000000000000300 RDX: 0000000000000000 RSI: 0000000000000001 RDI: ffff951846a9f6f8 RBP: ffff95184684c438 R08: ffff951879f3f800 R09: ffff951841a94028 R10: ffff951879f3f800 R11: 0000000000000000 R12: 0000000000000031 R13: 0000000000000000 R14: ffff9518a562e000 R15: ffff9518c49e5800 FS: 0000000000000000(0000) GS:ffff952b9f980000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00000001010fdac8 CR3: 000000018516a000 CR4: 0000000000f50ef0 PKRU: 55555554 Call Trace: <NMI> ? nmi_cpu_backtrace+0x99/0x110 ? nmi_cpu_backtrace_handler+0x11/0x20 ? nmi_handle+0x5e/0x150 ? default_do_nmi+0x40/0x100 ? exc_nmi+0x139/0x1c0 ? end_repeat_nmi+0xf/0x60 ? native_queued_spin_lock_slowpath+0x6e/0x2e0 ? native_queued_spin_lock_slowpath+0x6e/0x2e0 ? native_queued_spin_lock_slowpath+0x6e/0x2e0 </NMI> <IRQ> _raw_spin_lock_irqsave+0x3d/0x50 snd_pcm_period_elapsed+0x18/0x40 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] process_rx_packets+0x245/0x5a0 [snd_firewire_lib 95578abd4cf50f7c2e87b0d8233b959be2154fc3] ? irqtime_account_irq+0x40/0xc0 irq_target_callback+0x16/0x20 [snd_firewire_lib 95578abd4cf50f7c2e87b0d8233b959be2154fc3] handle_it_packet+0x135/0x150 [firewire_ohci 83160820eab58a73fec0d1e78730dd51dd0ac98c] context_tasklet+0x140/0x160 [firewire_ohci 83160820eab58a73fec0d1e78730dd51dd0ac98c] tasklet_action_common.isra.0+0x146/0x240 __do_softirq+0xc9/0x2c8 __irq_exit_rcu+0xa3/0xc0 common_interrupt+0x86/0xa0 </IRQ> <TASK> asm_common_interrupt+0x26/0x40 RIP: 0010:cpuidle_enter_state+0xcc/0x440 Code: 2a 39 36 ff e8 05 f1 ff ff 8b 53 04 49 89 c5 0f 1f 44 00 00 31 ff e8 83 38 35 ff 45 84 ff 0f 85 56 02 00 00 fb 0f 1f 44 00 00 <45> 85 f6 0f 88 85 01 00 00 49 63 d6 48 8d 04 52 48 8d 04 82 49 8d RSP: 0018:ffffb828c00ffe90 EFLAGS: 00000246 RAX: ffff952b9f9b4800 RBX: ffff952b9f9be208 RCX: 000000000000001f RDX: 0000000000000006 RSI: 0000000022b8e38e RDI: 0000000000000000 RBP: 0000000000000001 R08: 0000000000000000 R09: 00000000000001c8 R10: 0000000000000018 R11: ffff952b9f9b31e4 R12: ffffffff83f49780 R13: 0000006b6114102b R14: 0000000000000001 R15: 0000000000000000 cpuidle_enter+0x2d/0x40 do_idle+0x1d8/0x230 cpu_startup_entry+0x2a/0x30 start_secondary+0x11e/0x140 secondary_startup_64_no_verify+0x184/0x18b </TASK> Sending NMI from CPU 2 to CPUs 13: NMI backtrace for cpu 13 CPU: 13 PID: 1993 Comm: reaper/audio Not tainted 6.8.0-rc5-1-mainline #8 c2061649224b2c533bd915b824b700586979f2da Hardware name: Micro-Star International Co., Ltd. MS-7D25/PRO Z690-A DDR4(MS-7D25), BIOS Dasharo (coreboot+UEFI) v1.1.3 01/17/2024 RIP: 0010:tasklet_unlock_spin_wait+0xd/0x20 Code: c0 c3 cc cc cc cc 0f 1f 80 00 00 00 00 90 90 90 90 90 90 90 90 90 90 90 90 90 90 90 90 f3 0f 1e fa 0f 1f 44 00 00 eb 02 f3 90 <48> 8b 47 08 a8 02 75 f6 c3 cc cc cc cc 66 0f 1f 44 00 00 90 90 90 RSP: 0018:ffffb828c1cdbd08 EFLAGS: 00000002 RAX: 0000000000000002 RBX: ffff951841aba028 RCX: 0000000000000000 RDX: 0000000000000001 RSI: ffff951841a2aa70 RDI: ffff951841aba0a8 RBP: ffffb828c1cdbea0 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000075800 R14: ffff951846a9f600 R15: 0000000000000000 FS: 00007f7f394006c0(0000) GS:ffff952b9fb40000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007fc65c2a0000 CR3: 000000018516a000 CR4: 0000000000f50ef0 PKRU: 55555554 Call Trace: <NMI> ? nmi_cpu_backtrace+0x99/0x110 ? nmi_cpu_backtrace_handler+0x11/0x20 ? nmi_handle+0x5e/0x150 ? default_do_nmi+0x40/0x100 ? exc_nmi+0x139/0x1c0 ? end_repeat_nmi+0xf/0x60 ? tasklet_unlock_spin_wait+0xd/0x20 ? tasklet_unlock_spin_wait+0xd/0x20 ? tasklet_unlock_spin_wait+0xd/0x20 </NMI> <TASK> ohci_flush_iso_completions+0x1d/0x100 [firewire_ohci 83160820eab58a73fec0d1e78730dd51dd0ac98c] amdtp_domain_stream_pcm_pointer+0x41/0x50 [snd_firewire_lib 95578abd4cf50f7c2e87b0d8233b959be2154fc3] snd_pcm_update_hw_ptr0+0x6c/0x430 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] ? sched_clock_cpu+0xf/0x190 ? __smp_call_single_queue+0xad/0x120 snd_pcm_delay+0x33/0x140 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] snd_pcm_common_ioctl+0x922/0x12b0 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] ? wake_up_q+0x4e/0x90 ? futex_wake+0x159/0x190 snd_pcm_ioctl+0x2e/0x50 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] __x64_sys_ioctl+0x94/0xd0 do_syscall_64+0x86/0x170 ? do_syscall_64+0x96/0x170 ? do_syscall_64+0x96/0x170 ? do_syscall_64+0x96/0x170 ? do_syscall_64+0x96/0x170 ? __irq_exit_rcu+0x4b/0xc0 entry_SYSCALL_64_after_hwframe+0x6e/0x76 RIP: 0033:0x7f7f47cfd4ff Code: 00 48 89 44 24 18 31 c0 48 8d 44 24 60 c7 04 24 10 00 00 00 48 89 44 24 08 48 8d 44 24 20 48 89 44 24 10 b8 10 00 00 00 0f 05 <89> c2 3d 00 f0 ff ff 77 18 48 8b 44 24 18 64 48 2b 04 25 28 00 00 RSP: 002b:00007f7f393ffd50 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000003cfc060 RCX: 00007f7f47cfd4ff RDX: 00007f7f393ffdf0 RSI: 0000000080084121 RDI: 000000000000000a RBP: 0000000000000001 R08: 0000000000000001 R09: 0000000000000004 R10: 00007f7f393ffb20 R11: 0000000000000246 R12: 0000000000005000 R13: 0000000003272a80 R14: 0000000000000050 R15: 00000000032706c0 </TASK> INFO: task khugepaged:124 blocked for more than 122 seconds. Not tainted 6.8.0-rc5-1-mainline #8 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. task:khugepaged state:D stack:0 pid:124 tgid:124 ppid:2 flags:0x00004000 Call Trace: <TASK> __schedule+0x3e6/0x1520 ? place_entity+0x1b/0xf0 schedule+0x32/0xd0 schedule_timeout+0x151/0x160 wait_for_completion+0x8a/0x160 __flush_work.isra.0+0x173/0x280 ? __pfx_wq_barrier_func+0x10/0x10 __lru_add_drain_all+0x14c/0x200 khugepaged+0x72/0x950 ? __pfx_autoremove_wake_function+0x10/0x10 ? __pfx_khugepaged+0x10/0x10 kthread+0xe5/0x120 ? __pfx_kthread+0x10/0x10 ret_from_fork+0x31/0x50 ? __pfx_kthread+0x10/0x10 ret_from_fork_asm+0x1b/0x30 </TASK>

1 0

ASoC: rockchip_i2s_tdm calibration clocking problem
by Pavel Hofman 04 Mar '24

04 Mar '24

Hi, I am hitting a clock issue with rockchip_i2s_tdm.c + simple-audio-card (+ RK3308). At boot the mclk clk_i2s0_8ch_tx is (somehow) initialized to some (unimportant?) value 50176000 Hz. Note that this frequency is not multiple of either 48kHz or 44.1kHz. Method asoc_simple_parse_clk() reads this value and sets it to simple_dai->sysclk. Subsequently at asoc_simple_dai_init this "random" initial value is stored to i2s_tdm->mclk_tx_freq: 17.839330] rockchip_i2s_tdm_set_sysclk+0x50/0xbc [snd_soc_rockchip_i2s_tdm] [ 17.839367] snd_soc_dai_set_sysclk+0x38/0xb8 [snd_soc_core] [ 17.839596] asoc_simple_init_dai+0x94/0xc0 [snd_soc_simple_card_utils] [ 17.839640] asoc_simple_dai_init+0x130/0x230 [snd_soc_simple_card_utils] [ 17.839672] snd_soc_link_init+0x28/0x90 [snd_soc_core] [ 17.839843] snd_soc_bind_card+0x60c/0xbb4 [snd_soc_core] When starting playback, called by rockchip_i2s_tdm_hw_params(), rockchip_i2s_tdm_calibrate_mclk() correctly switches parent of mclk_parent to correct root pll clock mclk_root0/1 for the given samplerate and correctly configures mclk_parent frequency. https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rockchip_i… But right after that, the next line of rockchip_i2s_tdm_hw_params() calls rockchip_i2s_tdm_set_mclk() https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rockchip_i… This method calls clk_set_rate(i2s_tdm->mclk_tx, i2s_tdm->mclk_tx_freq), which resets the clock and its parental chain to the original incorrect value stored in i2s_tdm->mclk_tx_freq from the dai initialization. https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rockchip_i… As a result, no matter what sample rate is being played, the i2s mclk clock always ends up configured incorrectly. DTS I2S sets all clocks, therefore the clk calibration in rockchip_i2s_tdm.c should be (and is) used: i2s_8ch_0: i2s@ff300000 { compatible = "rockchip,rk3308-i2s-tdm"; reg = <0x0 0xff300000 0x0 0x1000>; interrupts = <GIC_SPI 48 IRQ_TYPE_LEVEL_HIGH>; clocks = <&cru SCLK_I2S0_8CH_TX>, <&cru SCLK_I2S0_8CH_RX>, <&cru HCLK_I2S0_8CH>, <&cru SCLK_I2S0_8CH_TX_SRC>, <&cru SCLK_I2S0_8CH_RX_SRC>, <&cru PLL_VPLL0>, <&cru PLL_VPLL1>; clock-names = "mclk_tx", "mclk_rx", "hclk", "mclk_tx_src", "mclk_rx_src", "mclk_root0", "mclk_root1"; ......... It seems to me that the calibration code should also rewrite the initially incorrect i2s_tdm->mclk_tx_freq and i2s_tdm->mclk_rx_freq with correct values corresponding to the momentary hw_params rate, or maybe rockchip_i2s_tdm_set_mclk() should not be called if rockchip_i2s_tdm_calibrate_mclk() is called a line above (i.e. putting the call to rockchip_i2s_tdm_set_mclk() into "else" branch). Thank you very much for help. With regards, Pavel.

2 1

[PATCH 5.10/5.15/6.1 0/1] soundwire: stream: use consistent pattern for freeing buffers
by Daniil Dulov 04 Mar '24

04 Mar '24

Svacer reports NULL-pointer dereference and double free issues in do_bank_switch() in case sdw_ml_sync_bank_switch() returns an error not on the first iteration of the list_for_each_entry() loop. These problems are present in 5.10, 5.15 and 6.1 stable releases. These problems have been fixed by the following upstream patch that can be cleanly applied to 5.10, 5.15 and 6.1 branches.

1 1

[PATCH] [alsa-lib][PATCH] conf: USB-Audio: Add Corsair HS60 Pro to the IEC958 blacklist
by David Senoner 04 Mar '24

04 Mar '24

This device is just an external USB soundcard with a female 3.5mm jack for a headset, no S/PDIF | IEC958 connector. Add it to the blacklist to prevent it being opened. Signed-off-by: David Senoner <seda18(a)rolmail.net> --- src/conf/cards/USB-Audio.conf | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 80631b2e..08346a1b 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -53,6 +53,7 @@ USB-Audio.pcm.iec958_device { "Blue Snowball" 999 "C-Media USB Headphone Set" 999 "Cmedia Audio" 999 + "Corsair HS60 PRO Surround USB S" 999 "DELL PROFESSIONAL SOUND BAR AE5" 999 "HP Digital Stereo Headset" 999 "GN 9330" 999 -- 2.43.2

3 2

[PATCH RESEND] conf: USB-Audio: Add more Scarlett devices to the IEC958 blacklist
by Geoffrey D. Bennett 04 Mar '24

04 Mar '24

The Scarlett Solo and 2i2 don't have S/PDIF outputs. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- src/conf/cards/USB-Audio.conf | 3 +++ 1 file changed, 3 insertions(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 80631b2e..05384a32 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -66,8 +66,11 @@ USB-Audio.pcm.iec958_device { "Plantronics USB Headset" 999 "Plantronics Wireless Audio" 999 "SB WoW Headset" 999 + "Scarlett 2i2 4th Gen" 999 "Scarlett 2i2 USB" 999 "Scarlett 2i4 USB" 999 + "Scarlett Solo 4th Gen" 999 + "Scarlett Solo USB" 999 "Sennheiser USB headset" 999 "SWTOR Gaming Headset by Razer" 999 "ThinkStation P620 Main" 999 -- 2.43.0

2 1

[PATCH v3 0/5] ALSA/ASoC: Conditionally skip i915 init and cleanups
by Cezary Rojewski 04 Mar '24

04 Mar '24

A small set of changes to improve initialization of the audio stack on HDAudio devices and pair of cleanups. As the first change is the most important one here, following is the technical background for it: Commit 78f613ba1efb ("drm/i915: finish removal of CNL") and its friends removed support for i915 for all CNL-based platforms. HDAudio library, however, still treats such platforms as valid candidates for i915 binding. Update query mechanism to reflect changes made in drm tree. At the same time, i915 support for LKF-based platforms has not been provided so remove them from valid binding candidates. The snd_soc_hda change is a follow up for the above and the cleanup patches do not bring any functional changes. Changes in v3: - snd_soc_hda_codec now returns -ENODEV on attach() if i915 is not present - denylist now const - added new patch for the avs-driver to address -ENODEV during probe_codec() - note: retained reviewed-by for patch 1/4 as changes are minimal Changes in v2: - list of problematic VGA devices is now declared locally, no more touching drm stuff Cezary Rojewski (5): ALSA: hda: Skip i915 initialization on CNL/LKF-based platforms ASoC: codecs: hda: Skip HDMI/DP registration if i915 is missing ASoC: Intel: avs: Ignore codecs with no suppoting driver ASoC: codecs: hda: Cleanup error messages ALSA: hda: Reuse for_each_pcm_streams() sound/hda/hdac_i915.c | 32 +++++++++++++++++++++++++++++--- sound/pci/hda/hda_codec.c | 2 +- sound/soc/codecs/hda.c | 15 ++++++++++----- sound/soc/intel/avs/core.c | 9 +++++---- 4 files changed, 45 insertions(+), 13 deletions(-) -- 2.25.1

3 9

[PATCH v1 0/3] ALSA: hda: cs35l41: Add support for a couple of Lenovo and Asus Laptops
by Stefan Binding 04 Mar '24

04 Mar '24

Add support for a couple of Lenovo Thinkbook 16P laptops using CS35L41 HDA. Also fix a _DSD BIOS error for the ASUS UM5302LA laptop. Stefan Binding (3): ALSA: hda: cs35l41: Support Lenovo Thinkbook 16P ALSA: hda/realtek: Add quirks for Lenovo Thinkbook 16P laptops ALSA: hda: cs35l41: Overwrite CS35L41 configuration for ASUS UM5302LA sound/pci/hda/cs35l41_hda_property.c | 6 ++++++ sound/pci/hda/patch_realtek.c | 2 ++ 2 files changed, 8 insertions(+) -- 2.34.1

2 4

Enable Headset Mic for Acer NB platform
by Kailang 04 Mar '24

04 Mar '24

Hi Takashi, This patch will enable headset Mic for Acer NB platform. Many Thanks. BR, Kailang

2 3

Scarlett Solo USB profiles missing or broken in 1.2.11
by GitHub issues - edited 03 Mar '24

03 Mar '24

alsa-project/alsa-ucm-conf issue #398 was edited from Slabity: Prior to version 1.2.10, my Scarlett Solo USB profiles looked like the following, where I could select profiles that explicitly enabled or disabled either the Inputs or Outputs: ![goodaudiooutput](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/b1bbc857-82b6-4fa5-b478-0c00c26b281c) As of 1.2.11, these profiles have disappeared and have left only a couple of profiles that do not provide the same functionality: ![badaudiooutputs](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/6c321638-3eb1-45ef-94a5-734c16477886) Other devices like my GPU's HDMI/DP Audio and motherboard's "Starship/Matisse HD Audio Controller" continue to expose the wider variety of profiles. It is only this device that seems to have the issue. I originally reported this on #346 as I thought it was related, but was told it might be a different issue. What makes me worried is the mention that this is "expected" now despite it breaking my audio setup. I can no longer select the "Digital Stereo (IEC958) Input" to exclusively enable the Input Device *without* enabling the Output Device. Here's some more information about the device: ``` $ lsusb | grep Focusrite Bus 007 Device 003: ID 1235:801c Focusrite-Novation Scarlett Solo USB $ alsaucm listcards 0: hw:2 Focusrite Scarlett Solo USB at usb-0000:10:00.3-3, high speed $ alsaucm -c hw:2 dump text Verb.HiFi { Comment Default Device.Line1 { Comment "Headphones / Line 1-2" Values { CaptureCTL "_ucm0001.hw:USB" PlaybackCTL "_ucm0001.hw:USB" PlaybackPCM "_ucm0001.hw:USB" PlaybackPriority 200 } } Device.Mic1 { Comment "Input 1 Mic" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 1 CapturePCM "_ucm0001.scarlett2i_mono_in:USB,0,0" CapturePriority 200 PlaybackCTL "_ucm0001.hw:USB" } } Device.Mic2 { Comment "Input 2 Inst/Line" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 1 CapturePCM "_ucm0001.scarlett2i_mono_in:USB,0,1" CapturePriority 100 PlaybackCTL "_ucm0001.hw:USB" } } } Verb.Direct { Comment "Direct Scarlett Solo USB" Device.Direct { Comment "Direct Scarlett Solo USB" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 2 CapturePCM "_ucm0001.hw:USB" CapturePriority 1000 PlaybackCTL "_ucm0001.hw:USB" PlaybackChannels 2 PlaybackPCM "_ucm0001.hw:USB" PlaybackPriority 1000 } } } ``` Some OS info it it helps: Linux distribution: NixOS on nixpkgs-unstable Kernel: 6.7.7 ALSA version: Advanced Linux Sound Architecture Driver Version k6.7.7 Pulseaudio version: 15.0.0 Pipewire version: 1.0.3 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/398 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Scarlett Solo USB profiles missing or broken in 1.2.11
by GitHub issues - edited 03 Mar '24

03 Mar '24

alsa-project/alsa-ucm-conf issue #398 was edited from Slabity: Prior to version 1.2.10, my Scarlett Solo USB profiles looked like the following, where I could select profiles that explicitly enabled or disabled either the Inputs or Outputs: ![goodaudiooutput](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/b1bbc857-82b6-4fa5-b478-0c00c26b281c) As of 1.2.11, these profiles have disappeared and have left only a couple of profiles that do not provide the same functionality: ![badaudiooutputs](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/6c321638-3eb1-45ef-94a5-734c16477886) Other devices like my GPU's HDMI/DP Audio and motherboard's "Starship/Matisse HD Audio Controller" continue to expose the wider variety of profiles. It is only this device that seems to have the issue. I originally reported this on #346 as I thought it was related, but was told it might be a different issue. What makes me worried is the mention that this is "expected" now despite it breaking my audio setup. I can no longer select the "Digital Stereo (IEC958) Input" to exclusively enable the Input Device *without* enabling the Output Device. Here's some more information about the device: ``` $ lsusb | grep Focusrite Bus 007 Device 003: ID 1235:801c Focusrite-Novation Scarlett Solo USB $ alsaucm listcards 0: hw:2 Focusrite Scarlett Solo USB at usb-0000:10:00.3-3, high speed $ alsaucm -c hw:2 dump text Verb.HiFi { Comment Default Device.Line1 { Comment "Headphones / Line 1-2" Values { CaptureCTL "_ucm0001.hw:USB" PlaybackCTL "_ucm0001.hw:USB" PlaybackPCM "_ucm0001.hw:USB" PlaybackPriority 200 } } Device.Mic1 { Comment "Input 1 Mic" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 1 CapturePCM "_ucm0001.scarlett2i_mono_in:USB,0,0" CapturePriority 200 PlaybackCTL "_ucm0001.hw:USB" } } Device.Mic2 { Comment "Input 2 Inst/Line" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 1 CapturePCM "_ucm0001.scarlett2i_mono_in:USB,0,1" CapturePriority 100 PlaybackCTL "_ucm0001.hw:USB" } } } Verb.Direct { Comment "Direct Scarlett Solo USB" Device.Direct { Comment "Direct Scarlett Solo USB" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 2 CapturePCM "_ucm0001.hw:USB" CapturePriority 1000 PlaybackCTL "_ucm0001.hw:USB" PlaybackChannels 2 PlaybackPCM "_ucm0001.hw:USB" PlaybackPriority 1000 } } } ``` Some OS info it it helps: Linux distribution: NixOS Kernel: 6.7.7 ALSA version: Advanced Linux Sound Architecture Driver Version k6.7.7 Pulseaudio version: 15.0.0 Pipewire version: 1.0.3 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/398 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Scarlett Solo USB profiles missing or broken in 1.2.11
by GitHub issues - edited 03 Mar '24

03 Mar '24

alsa-project/alsa-ucm-conf issue #398 was edited from Slabity: Prior to version 1.2.10, my Scarlett Solo USB profiles looked like the following, where I could select profiles that explicitly enabled or disabled either the Inputs or Outputs: ![goodaudiooutput](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/b1bbc857-82b6-4fa5-b478-0c00c26b281c) As of 1.2.11, these profiles have disappeared and have left only a couple of profiles that do not provide the same functionality: ![badaudiooutputs](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/6c321638-3eb1-45ef-94a5-734c16477886) Other devices like my GPU's HDMI/DP Audio and motherboard's "Starship/Matisse HD Audio Controller" continue to expose the wider variety of profiles. It is only this device that seems to have the issue. I originally reported this on #346 as I thought it was related, but was told it might be a different issue. What makes me worried is the mention that this is "expected" now despite it breaking my audio setup. I can no longer select the "Digital Stereo (IEC958) Input" to exclusively enable the Input Device *without* enabling the Output Device. Here's some more information about the device: ``` $ lsusb | grep Focusrite Bus 007 Device 003: ID 1235:801c Focusrite-Novation Scarlett Solo USB $ alsaucm listcards 0: hw:2 Focusrite Scarlett Solo USB at usb-0000:10:00.3-3, high speed $ alsaucm -c hw:2 dump text Verb.HiFi { Comment Default Device.Line1 { Comment "Headphones / Line 1-2" Values { CaptureCTL "_ucm0001.hw:USB" PlaybackCTL "_ucm0001.hw:USB" PlaybackPCM "_ucm0001.hw:USB" PlaybackPriority 200 } } Device.Mic1 { Comment "Input 1 Mic" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 1 CapturePCM "_ucm0001.scarlett2i_mono_in:USB,0,0" CapturePriority 200 PlaybackCTL "_ucm0001.hw:USB" } } Device.Mic2 { Comment "Input 2 Inst/Line" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 1 CapturePCM "_ucm0001.scarlett2i_mono_in:USB,0,1" CapturePriority 100 PlaybackCTL "_ucm0001.hw:USB" } } } Verb.Direct { Comment "Direct Scarlett Solo USB" Device.Direct { Comment "Direct Scarlett Solo USB" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 2 CapturePCM "_ucm0001.hw:USB" CapturePriority 1000 PlaybackCTL "_ucm0001.hw:USB" PlaybackChannels 2 PlaybackPCM "_ucm0001.hw:USB" PlaybackPriority 1000 } } } ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/398 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Scarlett Solo USB profiles missing or broken in 1.2.11
by GitHub issues - opened 03 Mar '24

03 Mar '24

alsa-project/alsa-ucm-conf issue #398 was opened from Slabity: Prior to version 1.2.10, my Scarlett Solo USB profiles looked like the following, where I could select profiles that explicitly enabled or disabled either the Inputs or Outputs: ![goodaudiooutput](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/b1bbc857-82b6-4fa5-b478-0c00c26b281c) As of 1.2.11, these profiles have disappeared and have left only a couple of profiles that do not provide the same functionality: ![badaudiooutputs](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/6c321638-3eb1-45ef-94a5-734c16477886) Other devices like my GPU's HDMI/DP Audio and motherboard's "Starship/Matisse HD Audio Controller" continue to expose the wider variety of profiles. It is only this device that seems to have the issue. I originally reported this on #346 as I thought it was related, but was told it might be a different issue. What makes me worried is the mention that this is "expected" now despite it breaking my audio setup. I can no longer select the "Digital Stereo (IEC958) Input" to exclusively enable the Input Device *without* enabling the Output Device. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/398 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] soundwire: Use snd_soc_substream_to_rtd() to obtain rtd
by Cezary Rojewski 03 Mar '24

03 Mar '24

Utilize the helper function instead of casting from ->private_data directly. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- drivers/soundwire/stream.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index f9c0adc0738d..4e9e7d2a942d 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1718,7 +1718,7 @@ EXPORT_SYMBOL(sdw_deprepare_stream); static int set_stream(struct snd_pcm_substream *substream, struct sdw_stream_runtime *sdw_stream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *dai; int ret = 0; int i; @@ -1771,7 +1771,7 @@ EXPORT_SYMBOL(sdw_alloc_stream); int sdw_startup_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; char *name; int ret; @@ -1815,7 +1815,7 @@ EXPORT_SYMBOL(sdw_startup_stream); void sdw_shutdown_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; struct snd_soc_dai *dai; -- 2.25.1

2 1

[PATCH] soundwire: constify the struct device_type usage
by Ricardo B. Marliere 03 Mar '24

03 Mar '24

Since commit aed65af1cc2f ("drivers: make device_type const"), the driver core can properly handle constant struct device_type. Move the sdw_master_type and sdw_slave_type variables to be constant structures as well, placing it into read-only memory which can not be modified at runtime. Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Ricardo B. Marliere <ricardo(a)marliere.net> --- drivers/soundwire/master.c | 2 +- drivers/soundwire/slave.c | 2 +- include/linux/soundwire/sdw_type.h | 4 ++-- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/master.c b/drivers/soundwire/master.c index 51abedbbaa66..b2c64512739d 100644 --- a/drivers/soundwire/master.c +++ b/drivers/soundwire/master.c @@ -112,7 +112,7 @@ static const struct dev_pm_ops master_dev_pm = { pm_generic_runtime_resume, NULL) }; -struct device_type sdw_master_type = { +const struct device_type sdw_master_type = { .name = "soundwire_master", .release = sdw_master_device_release, .pm = &master_dev_pm, diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 060c2982e26b..9963b92eb505 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -16,7 +16,7 @@ static void sdw_slave_release(struct device *dev) kfree(slave); } -struct device_type sdw_slave_type = { +const struct device_type sdw_slave_type = { .name = "sdw_slave", .release = sdw_slave_release, .uevent = sdw_slave_uevent, diff --git a/include/linux/soundwire/sdw_type.h b/include/linux/soundwire/sdw_type.h index b445f7200f06..693320b4f5c2 100644 --- a/include/linux/soundwire/sdw_type.h +++ b/include/linux/soundwire/sdw_type.h @@ -5,8 +5,8 @@ #define __SOUNDWIRE_TYPES_H extern const struct bus_type sdw_bus_type; -extern struct device_type sdw_slave_type; -extern struct device_type sdw_master_type; +extern const struct device_type sdw_slave_type; +extern const struct device_type sdw_master_type; static inline int is_sdw_slave(const struct device *dev) { --- base-commit: 81a7d0c4d059cb5c122110acbeec7bedfb91a741 change-id: 20240219-device_cleanup-soundwire-c037fffa2746 Best regards, -- Ricardo B. Marliere <ricardo(a)marliere.net>

2 1

[PATCH] soundwire: Use snd_soc_substream_to_rtd() to obtain rtd
by Amadeusz Sławiński 03 Mar '24

03 Mar '24

From: Cezary Rojewski <cezary.rojewski(a)intel.com> Utilize the helper function instead of casting from ->private_data directly. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- drivers/soundwire/stream.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 69719b335bcb..652aae4e221d 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1717,7 +1717,7 @@ EXPORT_SYMBOL(sdw_deprepare_stream); static int set_stream(struct snd_pcm_substream *substream, struct sdw_stream_runtime *sdw_stream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *dai; int ret = 0; int i; @@ -1770,7 +1770,7 @@ EXPORT_SYMBOL(sdw_alloc_stream); int sdw_startup_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; char *name; int ret; @@ -1814,7 +1814,7 @@ EXPORT_SYMBOL(sdw_startup_stream); void sdw_shutdown_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; struct snd_soc_dai *dai; -- 2.34.1

2 2

Internal mic does not work on IdeaPad 3 Chromebook
by GitHub issues - opened 03 Mar '24

03 Mar '24

alsa-project/alsa-ucm-conf issue #397 was opened from abhi-deshpande: Device - IdeaPad 3 Chromebook (Octopus Lick) OS - Fedora 39 Card - sof-glkda7219max I recently converted my Chromebook to Chultrabook and installed latest Fedora on it. It seems that most of the audio stuff works, but the internal DMIC doesn't work. I read some previous issues and commits and found that this is due to the fact that the DMIC needs channel split as CRAS does. This was being implemented in #154 I guess. I am currently on lastest alsa-lib and alsa-ucm-conf. What could be missing ? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/397 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH V3 00/13] soundwire/SOF: add SoundWire Interface support for AMD SOF stack
by Vijendar Mukunda 02 Mar '24

02 Mar '24

This patch series is to redesign existing platform device creation logic for SoundWire managers and Implement generic functions for SoundWire manager probe, start and exit sequence which are common for both Legacy (NO DSP enabled) and SOF stack, and add SoundWire Interface support for AMD SOF stack (ACP 6.3 based platform). Below patch series is reviewed in https://github.com/thesofproject/linux/pull/4699 Changes since V1: - add new commit to update license for amd soundwire manager driver files. - wrap the commit message with in 75 characters. - add amd tag in commit title for soundwire patches Changes since V2: - drop "drivers" tag from commit title for SoundWire patches. This patch series has dependencies. "drivers: soundwire: amd: refactor amd soundwire manager device node creation" patch has dependency on below patch. "ASoC/soundwire: implement generic api for scanning amd soundwire controller" Similarly SOF patches has dependency on Soundwire tree patches. This patch series has to be reviewed and merged in one go. Vijendar Mukunda (13): ASoC/soundwire: implement generic api for scanning amd soundwire controller soundwire: amd: update license soundwire: amd: refactor amd soundwire manager device node creation soundwire: amd: implement function to extract slave information soundwire: amd: refactor soundwire pads enable soundwire: amd: refactor register mask structure ASoC: SOF: amd: add code for invoking soundwire manager helper functions ASoC: SOF: amd: add interrupt handling for SoundWire manager devices ASoC: SOF: amd: Add Soundwire DAI configuration support for AMD platforms ASoC: SOF: amd: add machine select logic for soundwire based platforms ASoC: SOF: amd: update descriptor fields for acp6.3 based platform ASoC: SOF: amd: select soundwire dependency flag for acp6.3 based platform ASoC: SOF: amd: refactor acp driver pm ops drivers/soundwire/Makefile | 2 +- drivers/soundwire/amd_init.c | 235 +++++++++++++++++++++++++++++ drivers/soundwire/amd_init.h | 13 ++ drivers/soundwire/amd_manager.c | 45 ++---- drivers/soundwire/amd_manager.h | 14 +- include/linux/soundwire/sdw_amd.h | 81 ++++++++-- include/sound/sof/dai-amd.h | 7 + include/sound/sof/dai.h | 2 + include/uapi/sound/sof/tokens.h | 4 + sound/soc/amd/acp/Kconfig | 7 + sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/amd-sdw-acpi.c | 62 ++++++++ sound/soc/sof/amd/Kconfig | 18 +++ sound/soc/sof/amd/acp-common.c | 65 +++++++- sound/soc/sof/amd/acp-dsp-offset.h | 10 ++ sound/soc/sof/amd/acp.c | 202 ++++++++++++++++++++++++- sound/soc/sof/amd/acp.h | 26 +++- sound/soc/sof/amd/pci-acp63.c | 7 + sound/soc/sof/ipc3-pcm.c | 25 +++ sound/soc/sof/ipc3-topology.c | 40 +++++ sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/topology.c | 5 + 22 files changed, 802 insertions(+), 71 deletions(-) create mode 100644 drivers/soundwire/amd_init.c create mode 100644 drivers/soundwire/amd_init.h create mode 100644 sound/soc/amd/acp/amd-sdw-acpi.c -- 2.34.1

3 15

[PATCH v4 00/39] Add support for sam9x7 SoC family
by Varshini Rajendran 01 Mar '24

01 Mar '24

This patch series adds support for the new SoC family - sam9x7. - The device tree, configs and drivers are added - Clock driver for sam9x7 is added - Support for basic peripherals is added - Target board SAM9X75 Curiosity is added Changes in v4: -------------- - Addressed all the review comments in the patches - Picked up all Acked-by and Reviewed-by tags - Dropped applied patches from the series - Added pwm node and related dt binding documentation - Added support for exporting some clocks to DT - Dropped USB related patches and changes. See NOTE. - All the specific changes are captured in the corresponding patches NOTE: Owing to the discussion here https://lore.kernel.org/linux-devicetree/CAL_JsqJ9PrX6fj-EbffeJce09MXs=B7t+… the USB related changes are dropped from this series in order to enable us to work on the mentioned issues before adding new compatibles as said. The issues/warnings will be addressed in subsequent patches. After which the USB related support for sam9x7 SoCs will be added. Hope this works out fine. Changes in v3: -------------- - Fixed the DT documentation errors pointed out in v2. - Dropped Acked-by tag in tcb DT doc patch as it had to be adapted according to sam9x7 correctly. - Picked by the previously missed tags. - Dropped this patch "dt-bindings: usb: generic-ehci: Document clock-names property" as the warning was not found while validating DT-schema for at91-sam9x75_curiosity.dtb. - Dropped redundant words in the commit message. - Fixed the CHECK_DTBS warnings validated against at91-sam9x75_curiosity.dtb. - Renamed dt nodes according to naming convention. - Dropped unwanted status property in dts. - Removed nodes that are not in use from the board dts. - Removed spi DT doc patch from the series as it was already applied and a fix patch was applied subsequently. Added a patch to remove the compatible to adapt sam9x7. - Added sam9x7 compatibles in usb dt documentation. Changes in v2: -------------- - Added sam9x7 specific compatibles in DT with fallbacks - Documented all the newly added DT compatible strings - Added device tree for the target board sam9x75 curiosity and documented the same in the DT bindings documentation - Removed the dt nodes that are not supported at the moment - Removed the configs added by previous version that are not supported at the moment - Fixed all the corrections in the commit message - Changed all the instances of copyright year to 2023 - Added sam9x7 flag in PIT64B configuration - Moved macro definitions to header file - Added another divider in mck characteristics in the pmc driver - Fixed the memory leak in the pmc driver - Dropped patches that are no longer needed - Picked up Acked-by and Reviewed-by tags Varshini Rajendran (39): dt-bindings: net: cdns,macb: add sam9x7 ethernet interface dt-bindings: atmel-sysreg: add sam9x7 dt-bindings: crypto: add sam9x7 in Atmel AES dt-bindings: crypto: add sam9x7 in Atmel SHA dt-bindings: crypto: add sam9x7 in Atmel TDES dt-bindings: i2c: at91: Add sam9x7 compatible string dt-bindings: atmel-ssc: add microchip,sam9x7-ssc dt-bindings: atmel-nand: add microchip,sam9x7-pmecc dt-bindings: pinctrl: at91: add sam9x7 dt-bindings: rng: atmel,at91-trng: add sam9x7 TRNG dt-bindings: rtt: at91rm9260: add sam9x7 compatible dt-bindings: serial: atmel,at91-usart: add compatible for sam9x7. ASoC: dt-bindings: atmel-classd: add sam9x7 compatible dt-bindings: pwm: at91: Add sam9x7 compatible strings list dt-bindings: watchdog: sama5d4-wdt: add compatible for sam9x7-wdt spi: dt-bindings: atmel,at91rm9200-spi: remove 9x60 compatible from list ASoC: dt-bindings: microchip: add sam9x7 ARM: at91: pm: add support for sam9x7 SoC family ARM: at91: pm: add sam9x7 SoC init config ARM: at91: add support in SoC driver for new sam9x7 dt-bindings: clk: at91: add sam9x7 dt-bindings: clk: at91: add sam9x7 clock controller clk: at91: clk-sam9x60-pll: re-factor to support individual core freq outputs clk: at91: sam9x7: add support for HW PLL freq dividers clk: at91: sama7g5: move mux table macros to header file dt-bindings: clock: at91: Allow PLLs to be exported and referenced in DT clk: at91: sam9x7: add sam9x7 pmc driver dt-bindings: irqchip/atmel-aic5: Add support for sam9x7 aic irqchip/atmel-aic5: Add support to get nirqs from DT for sam9x60 & sam9x7 power: reset: at91-poweroff: lookup for proper pmc dt node for sam9x7 power: reset: at91-reset: add reset support for sam9x7 SoC power: reset: at91-reset: add sdhwc support for sam9x7 SoC dt-bindings: reset: atmel,at91sam9260-reset: add sam9x7 dt-bindings: power: reset: atmel,sama5d2-shdwc: add sam9x7 ARM: at91: Kconfig: add config flag for SAM9X7 SoC ARM: configs: at91: enable config flags for sam9x7 SoC family ARM: dts: at91: sam9x7: add device tree for SoC dt-bindings: arm: add sam9x75 curiosity board ARM: dts: at91: sam9x75_curiosity: add sam9x75 curiosity board .../devicetree/bindings/arm/atmel-at91.yaml | 6 + .../devicetree/bindings/arm/atmel-sysregs.txt | 7 +- .../bindings/clock/atmel,at91rm9200-pmc.yaml | 2 + .../bindings/clock/atmel,at91sam9x5-sckc.yaml | 4 +- .../crypto/atmel,at91sam9g46-aes.yaml | 6 +- .../crypto/atmel,at91sam9g46-sha.yaml | 6 +- .../crypto/atmel,at91sam9g46-tdes.yaml | 6 +- .../bindings/i2c/atmel,at91sam-i2c.yaml | 4 +- .../interrupt-controller/atmel,aic.txt | 2 +- .../devicetree/bindings/misc/atmel-ssc.txt | 1 + .../devicetree/bindings/mtd/atmel-nand.txt | 1 + .../devicetree/bindings/net/cdns,macb.yaml | 5 + .../bindings/pinctrl/atmel,at91-pinctrl.txt | 2 + .../power/reset/atmel,sama5d2-shdwc.yaml | 3 + .../bindings/pwm/atmel,at91sam-pwm.yaml | 3 + .../reset/atmel,at91sam9260-reset.yaml | 4 + .../bindings/rng/atmel,at91-trng.yaml | 4 + .../bindings/rtc/atmel,at91sam9260-rtt.yaml | 4 +- .../bindings/serial/atmel,at91-usart.yaml | 12 +- .../bindings/sound/atmel,sama5d2-classd.yaml | 7 +- .../sound/microchip,sama7g5-i2smcc.yaml | 11 +- .../bindings/spi/atmel,at91rm9200-spi.yaml | 1 - .../bindings/watchdog/atmel,sama5d4-wdt.yaml | 12 +- arch/arm/boot/dts/microchip/Makefile | 3 + .../dts/microchip/at91-sam9x75_curiosity.dts | 309 +++++ arch/arm/boot/dts/microchip/sam9x60.dtsi | 1 + arch/arm/boot/dts/microchip/sam9x7.dtsi | 1214 +++++++++++++++++ arch/arm/configs/at91_dt_defconfig | 1 + arch/arm/mach-at91/Kconfig | 23 +- arch/arm/mach-at91/Makefile | 1 + arch/arm/mach-at91/generic.h | 2 + arch/arm/mach-at91/pm.c | 35 + arch/arm/mach-at91/sam9x7.c | 34 + drivers/clk/at91/Makefile | 1 + drivers/clk/at91/clk-sam9x60-pll.c | 50 +- drivers/clk/at91/pmc.h | 18 + drivers/clk/at91/sam9x60.c | 7 + drivers/clk/at91/sam9x7.c | 946 +++++++++++++ drivers/clk/at91/sama7g5.c | 42 +- drivers/irqchip/irq-atmel-aic5.c | 12 +- drivers/power/reset/Kconfig | 4 +- drivers/power/reset/at91-sama5d2_shdwc.c | 1 + drivers/soc/atmel/soc.c | 23 + drivers/soc/atmel/soc.h | 9 + include/dt-bindings/clock/at91.h | 4 + 45 files changed, 2788 insertions(+), 65 deletions(-) create mode 100644 arch/arm/boot/dts/microchip/at91-sam9x75_curiosity.dts create mode 100644 arch/arm/boot/dts/microchip/sam9x7.dtsi create mode 100644 arch/arm/mach-at91/sam9x7.c create mode 100644 drivers/clk/at91/sam9x7.c -- 2.25.1

6 8

[PATCH v1] ALSA: hda: optimize the probe codec process
by songxiebing 01 Mar '24

01 Mar '24

From: songxiebing <songxiebing(a)kylinos.cn> In azx_probe_codecs function，when bus->codec_mask is becomes to 0(no codecs), execute azx_init_chip, bus->codec_mask will be initialized to a value again， this causes snd_hda_codec_new function to run, the process is as follows: -->snd_hda_codec_new -->snd_hda_codec_device_init -->snd_hdac_device_init---snd_hdac_read_parm(...AC_PAR_VENDOR_ID) 2s ---snd_hdac_read_parm(...AC_PAR_VENDOR_ID) 2s ---snd_hdac_read_parm(...AC_PAR_SUBSYSTEM_ID) 2s ---snd_hdac_read_parm(...AC_PAR_REV_ID) 2s ---snd_hdac_read_parm(...AC_PAR_NODE_COUNT) 2s when no codecs, read communication is error, each command will be polled for 2 second，a total of 10s, it is easy to some problem. like this: 2 [ 14.833404][ 6] [ T164] hda 0006:00: Codec #0 probe error; disabling it... 3 [ 14.844178][ 6] [ T164] hda 0006:00: codec_mask = 0x1 4 [ 14.880532][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0000 5 [ 15.891988][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0000 6 [ 16.978090][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0001 7 [ 18.140895][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0002 8 [ 19.135516][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0004 10 [ 19.900086][ 6] [ T164] hda 0006:00: no codecs initialized 11 [ 45.573398][ 2] [ C2] watchdog: BUG: soft lockup - CPU#2 stuck for 22s! [kworker/2:0:25] Here, when bus->codec_mask is 0, use a direct break to avoid execute snd_hda_codec_new function. Signed-off-by: songxiebing <songxiebing(a)kylinos.cn> --- sound/pci/hda/hda_controller.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index e5c53ad..e1ece0a 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1474,6 +1474,9 @@ int azx_probe_codecs(struct azx *chip, unsigned int max_slots) dev_warn(chip->card->dev, "Codec #%d probe error; disabling it...\n", c); bus->codec_mask &= ~(1 << c); + /* no codecs */ + if (bus->codec_mask == 0) + break; /* More badly, accessing to a non-existing * codec often screws up the controller chip, * and disturbs the further communications. -- 2.7.4

2 3

Headset Mic no show at resume back
by Kailang 01 Mar '24

01 Mar '24

Hi Takashi, Attach patch will fix headset Mic no show issue. BR, Kailang

2 1

[PATCH v2 0/2] ASoC: nau8325: Modify driver code and dtschema.
by Seven Lee 01 Mar '24

01 Mar '24

Remove the "Clock detection" switch control, and modify dtschema for the explanation of "nuvoton,dac-vref". Change: V1 -> V2 - Revise the driver description part for the C++ comment. - In the nau8325_clkdet_put function, modify the max variable to hard code. - Removed "Clock Detection" switch control. - modify the "ALC Enable" switch name. - Revise the dtschema for "nuvoton,dac-vref". Seven Lee (2): ASoC: dt-bindings: Added schema for "nuvoton,nau8325" ASoC: nau8325: new driver .../bindings/sound/nuvoton,nau8325.yaml | 82 ++ sound/soc/codecs/nau8325.c | 856 ++++++++++++++++++ sound/soc/codecs/nau8325.h | 391 ++++++++ 3 files changed, 1329 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml create mode 100755 sound/soc/codecs/nau8325.c create mode 100644 sound/soc/codecs/nau8325.h -- 2.25.1

4 7

[PATCH] ASoC: madera: Fix typo in madera_set_fll_clks shift value
by Stuart Henderson 29 Feb '24

29 Feb '24

Fix a typo in the shift value used in madera_set_fll_clks. Fixes: 3863857dd5ca3 ("ASoC: madera: Enable clocks for input pins when used for the FLL") Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> --- sound/soc/codecs/madera.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index b9f19fbd2911..b24d6472ad5f 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -3884,7 +3884,7 @@ static inline int madera_set_fll_clks(struct madera_fll *fll, int base, bool ena return madera_set_fll_clks_reg(fll, ena, base + MADERA_FLL_CONTROL_6_OFFS, MADERA_FLL1_REFCLK_SRC_MASK, - MADERA_FLL1_REFCLK_DIV_SHIFT); + MADERA_FLL1_REFCLK_SRC_SHIFT); } static inline int madera_set_fllao_clks(struct madera_fll *fll, int base, bool ena) -- 2.30.2

2 1

[linux-next:master] BUILD REGRESSION f303a3e2bcfba900efb5aee55236d17030e9f882
by kernel test robot 29 Feb '24

29 Feb '24

tree/branch: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master branch HEAD: f303a3e2bcfba900efb5aee55236d17030e9f882 Add linux-next specific files for 20240229 Error/Warning reports: https://lore.kernel.org/oe-kbuild-all/202402292319.eHMsyI8L-lkp@intel.com https://lore.kernel.org/oe-kbuild-all/202403010220.WelHZqIi-lkp@intel.com Error/Warning: (recently discovered and may have been fixed) (.text+0x721c): undefined reference to `__divdi3' arm-linux-gnueabi-ld: file.c:(.text+0x958): undefined reference to `__aeabi_ldivmod' arm-linux-gnueabi-ld: inode.c:(.text+0x48e): undefined reference to `__aeabi_ldivmod' data.o:(.text+0x33e0): undefined reference to `__moddi3' drivers/hid/amd-sfh-hid/amd_sfh_pcie.c:413:34: error: 'struct cpuinfo_loongarch' has no member named 'x86' drivers/leds/leds-gpio-register.c:23:25: warning: attribute declaration must precede definition [-Wignored-attributes] file.c:(.text+0x944): undefined reference to `__aeabi_ldivmod' file.o:(.text+0x820c): undefined reference to `__moddi3' file.o:(.text+0x97d0): undefined reference to `__divdi3' gc.o:(.text+0x4874): undefined reference to `__moddi3' include/linux/dpll.h:179:1: warning: control reaches end of non-void function [-Wreturn-type] inode.c:(.text+0x47a): undefined reference to `__aeabi_ldivmod' inode.o:(.text+0x1328): undefined reference to `__moddi3' microblaze-linux-ld: data.o:(.text+0x33fc): undefined reference to `__moddi3' microblaze-linux-ld: data.o:(.text+0x3418): undefined reference to `__divdi3' microblaze-linux-ld: file.o:(.text+0x8228): undefined reference to `__moddi3' microblaze-linux-ld: file.o:(.text+0x8244): undefined reference to `__divdi3' microblaze-linux-ld: gc.o:(.text+0x4890): undefined reference to `__moddi3' microblaze-linux-ld: gc.o:(.text+0x48ac): undefined reference to `__divdi3' microblaze-linux-ld: inode.o:(.text+0x1344): undefined reference to `__moddi3' microblaze-linux-ld: inode.o:(.text+0x1360): undefined reference to `__divdi3' microblaze-linux-ld: namei.o:(.text+0x1940): undefined reference to `__moddi3' microblaze-linux-ld: namei.o:(.text+0x195c): undefined reference to `__divdi3' microblaze-linux-ld: segment.o:(.text+0xda30): undefined reference to `__moddi3' microblaze-linux-ld: segment.o:(.text+0xda4c): undefined reference to `__divdi3' microblaze-linux-ld: xattr.o:(.text+0x1930): undefined reference to `__moddi3' microblaze-linux-ld: xattr.o:(.text+0x194c): undefined reference to `__divdi3' namei.o:(.text+0x1924): undefined reference to `__moddi3' segment.o:(.text+0xda14): undefined reference to `__moddi3' xattr.o:(.text+0x1914): undefined reference to `__moddi3' Error/Warning ids grouped by kconfigs: gcc_recent_errors |-- alpha-allyesconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- arc-allmodconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- arc-allyesconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- arm-allmodconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- arm-allyesconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- arm-randconfig-001-20240116 | |-- arm-linux-gnueabi-ld:file.c:(.text):undefined-reference-to-__aeabi_ldivmod | |-- arm-linux-gnueabi-ld:inode.c:(.text):undefined-reference-to-__aeabi_ldivmod | |-- file.c:(.text):undefined-reference-to-__aeabi_ldivmod | `-- inode.c:(.text):undefined-reference-to-__aeabi_ldivmod |-- arm-randconfig-r131-20240229 | |-- drivers-leds-flash-leds-ktd2692.c:sparse:sparse:symbol-ktd2692_timing-was-not-declared.-Should-it-be-static | `-- io_uring-io_uring.c:sparse:sparse:cast-to-restricted-io_req_flags_t |-- arm64-defconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- arm64-randconfig-002-20240229 | `-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg |-- csky-allmodconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- csky-allyesconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- csky-randconfig-002-20240229 | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- i386-allmodconfig | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-max_group_count-not-described-in-drm_dp_tunnel_mgr_create | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tracker-not-described-in-drm_dp_tunnel_put | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tunnel-not-described-in-drm_dp_tunnel_put | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:expecting-prototype-for-drm_dp_tunnel_atomic_get_allocated_bw().-Prototype-was-for-drm_dp_tunnel_get_allocated_bw()-instead | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- i386-allyesconfig | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-max_group_count-not-described-in-drm_dp_tunnel_mgr_create | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tracker-not-described-in-drm_dp_tunnel_put | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tunnel-not-described-in-drm_dp_tunnel_put | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:expecting-prototype-for-drm_dp_tunnel_atomic_get_allocated_bw().-Prototype-was-for-drm_dp_tunnel_get_allocated_bw()-instead | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- i386-buildonly-randconfig-003-20240229 | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- i386-buildonly-randconfig-005-20240229 | `-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg |-- i386-randconfig-012-20240229 | `-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg |-- i386-randconfig-061-20240229 | |-- drivers-leds-flash-leds-ktd2692.c:sparse:sparse:symbol-ktd2692_timing-was-not-declared.-Should-it-be-static | |-- drivers-video-backlight-ktd2801-backlight.c:sparse:sparse:symbol-ktd2801_timing-was-not-declared.-Should-it-be-static | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:cast-to-restricted-__le32 | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:cast-to-restricted-__le64 | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:restricted-__le32-degrades-to-integer | `-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:restricted-__le64-degrades-to-integer |-- i386-randconfig-062-20240229 | |-- drivers-video-backlight-ktd2801-backlight.c:sparse:sparse:symbol-ktd2801_timing-was-not-declared.-Should-it-be-static | `-- io_uring-io_uring.c:sparse:sparse:cast-to-restricted-io_req_flags_t |-- i386-randconfig-063-20240229 | |-- drivers-video-backlight-ktd2801-backlight.c:sparse:sparse:symbol-ktd2801_timing-was-not-declared.-Should-it-be-static | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:cast-to-restricted-__le32 | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:cast-to-restricted-__le64 | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:restricted-__le32-degrades-to-integer | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:restricted-__le64-degrades-to-integer | `-- io_uring-io_uring.c:sparse:sparse:cast-to-restricted-io_req_flags_t |-- i386-randconfig-141-20240228 | `-- include-linux-dpll.h:warning:control-reaches-end-of-non-void-function |-- loongarch-allmodconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- loongarch-allyesconfig | `-- drivers-hid-amd-sfh-hid-amd_sfh_pcie.c:error:struct-cpuinfo_loongarch-has-no-member-named-x86 |-- loongarch-defconfig | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- loongarch-randconfig-002-20240229 | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- m68k-allmodconfig | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- m68k-allyesconfig | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- microblaze-allmodconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- microblaze-allyesconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- microblaze-randconfig-r022-20220915 | |-- (.text):undefined-reference-to-__divdi3 | |-- data.o:(.text):undefined-reference-to-__moddi3 | |-- file.o:(.text):undefined-reference-to-__divdi3 | |-- file.o:(.text):undefined-reference-to-__moddi3 | |-- gc.o:(.text):undefined-reference-to-__moddi3 | |-- inode.o:(.text):undefined-reference-to-__moddi3 | |-- microblaze-linux-ld:data.o:(.text):undefined-reference-to-__divdi3 | |-- microblaze-linux-ld:data.o:(.text):undefined-reference-to-__moddi3 | |-- microblaze-linux-ld:file.o:(.text):undefined-reference-to-__divdi3 | |-- microblaze-linux-ld:file.o:(.text):undefined-reference-to-__moddi3 | |-- microblaze-linux-ld:gc.o:(.text):undefined-reference-to-__divdi3 | |-- microblaze-linux-ld:gc.o:(.text):undefined-reference-to-__moddi3 | |-- microblaze-linux-ld:inode.o:(.text):undefined-reference-to-__divdi3 | |-- microblaze-linux-ld:inode.o:(.text):undefined-reference-to-__moddi3 | |-- microblaze-linux-ld:namei.o:(.text):undefined-reference-to-__divdi3 | |-- microblaze-linux-ld:namei.o:(.text):undefined-reference-to-__moddi3 | |-- microblaze-linux-ld:segment.o:(.text):undefined-reference-to-__divdi3 | |-- microblaze-linux-ld:segment.o:(.text):undefined-reference-to-__moddi3 | |-- microblaze-linux-ld:xattr.o:(.text):undefined-reference-to-__divdi3 | |-- microblaze-linux-ld:xattr.o:(.text):undefined-reference-to-__moddi3 | |-- namei.o:(.text):undefined-reference-to-__moddi3 | |-- segment.o:(.text):undefined-reference-to-__moddi3 | `-- xattr.o:(.text):undefined-reference-to-__moddi3 |-- mips-allyesconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- nios2-allmodconfig | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- nios2-allyesconfig | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- nios2-randconfig-001-20240229 | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- openrisc-allyesconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- openrisc-randconfig-r122-20240229 | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- drivers-leds-flash-leds-ktd2692.c:sparse:sparse:symbol-ktd2692_timing-was-not-declared.-Should-it-be-static |-- parisc-allmodconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- parisc-allyesconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- parisc-defconfig | `-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg |-- parisc64-defconfig | `-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg |-- powerpc-allmodconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- powerpc-randconfig-r121-20240229 | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | |-- drivers-leds-flash-leds-ktd2692.c:sparse:sparse:symbol-ktd2692_timing-was-not-declared.-Should-it-be-static | |-- kernel-power-energy_model.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-struct-em_perf_state-table-got-struct-em_perf_state-noderef-__rcu | |-- kernel-power-energy_model.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-struct-kref-kref-got-struct-kref-noderef-__rcu | |-- kernel-power-energy_model.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-void-const-objp-got-struct-em_perf_table-noderef-__rcu-assigned-em_table | `-- kernel-power-energy_model.c:sparse:sparse:incorrect-type-in-assignment-(different-address-spaces)-expected-struct-em_perf_state-new_ps-got-struct-em_perf_state-noderef-__rcu |-- powerpc64-randconfig-001-20240229 | `-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg |-- s390-allyesconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- s390-randconfig-002-20240229 | `-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg |-- sh-allmodconfig | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- sh-allyesconfig | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- sparc-allmodconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- sparc-randconfig-002-20240229 | `-- (.head.text):relocation-truncated-to-fit:R_SPARC_WDISP22-against-init.text |-- sparc64-allmodconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- sparc64-allyesconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- um-allyesconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- x86_64-randconfig-011-20240229 | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- x86_64-randconfig-013-20240229 | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- x86_64-randconfig-014-20240229 | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- x86_64-randconfig-016-20240229 | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-max_group_count-not-described-in-drm_dp_tunnel_mgr_create | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tracker-not-described-in-drm_dp_tunnel_put | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tunnel-not-described-in-drm_dp_tunnel_put | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:expecting-prototype-for-drm_dp_tunnel_atomic_get_allocated_bw().-Prototype-was-for-drm_dp_tunnel_get_allocated_bw()-instead | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- x86_64-randconfig-075-20240229 | `-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg |-- x86_64-randconfig-121-20240229 | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-leds-flash-leds-ktd2692.c:sparse:sparse:symbol-ktd2692_timing-was-not-declared.-Should-it-be-static | |-- drivers-video-backlight-ktd2801-backlight.c:sparse:sparse:symbol-ktd2801_timing-was-not-declared.-Should-it-be-static | |-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead | |-- kernel-power-energy_model.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-struct-em_perf_state-table-got-struct-em_perf_state-noderef-__rcu | |-- kernel-power-energy_model.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-struct-kref-kref-got-struct-kref-noderef-__rcu | |-- kernel-power-energy_model.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-void-const-objp-got-struct-em_perf_table-noderef-__rcu-assigned-em_table | `-- kernel-power-energy_model.c:sparse:sparse:incorrect-type-in-assignment-(different-address-spaces)-expected-struct-em_perf_state-new_ps-got-struct-em_perf_state-noderef-__rcu |-- x86_64-randconfig-161-20240229 | |-- drivers-acpi-property.c-acpi_data_add_buffer_props()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code | |-- fs-proc-proc_sysctl.c-proc_sys_call_handler()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code | |-- io_uring-filetable.c-io_alloc_file_tables()-error:kvcalloc()-does-not-make-sense-for-no-sleep-code | |-- kernel-resource.c-walk_system_ram_res_rev()-error:kvcalloc()-does-not-make-sense-for-no-sleep-code | |-- lib-stackdepot.c-stack_depot_init()-error:kvcalloc()-does-not-make-sense-for-no-sleep-code | |-- net-ethtool-common.c-ethtool_get_max_rxnfc_channel()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code | |-- net-ipv4-fib_semantics.c-fib_create_info()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code | `-- net-ipv4-tcp_metrics.c-tcp_metrics_hash_alloc()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code |-- x86_64-randconfig-r113-20240229 | |-- drivers-leds-flash-leds-ktd2692.c:sparse:sparse:symbol-ktd2692_timing-was-not-declared.-Should-it-be-static | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:cast-to-restricted-__le32 | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:cast-to-restricted-__le64 | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:restricted-__le32-degrades-to-integer | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:restricted-__le64-degrades-to-integer | `-- io_uring-io_uring.c:sparse:sparse:cast-to-restricted-io_req_flags_t |-- xtensa-allyesconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-gpu-drm-xe-xe_ggtt.c:error:implicit-declaration-of-function-writeq | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead `-- xtensa-randconfig-002-20240229 `-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg clang_recent_errors |-- arm-defconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- arm64-allmodconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- hexagon-allmodconfig | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- hexagon-allyesconfig | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- i386-randconfig-003-20240229 | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-max_group_count-not-described-in-drm_dp_tunnel_mgr_create | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tracker-not-described-in-drm_dp_tunnel_put | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tunnel-not-described-in-drm_dp_tunnel_put | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:expecting-prototype-for-drm_dp_tunnel_atomic_get_allocated_bw().-Prototype-was-for-drm_dp_tunnel_get_allocated_bw()-instead | `-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg |-- i386-randconfig-013-20240229 | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- i386-randconfig-015-20240229 | `-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg |-- i386-randconfig-141-20240229 | |-- drivers-acpi-property.c-acpi_data_add_buffer_props()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code | |-- drivers-gpu-drm-drm_property.c-drm_property_create_blob()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code | |-- drivers-media-common-videobuf2-videobuf2-dma-sg.c-vb2_dma_sg_alloc()-error:kvcalloc()-does-not-make-sense-for-no-sleep-code | |-- drivers-media-v4l2-core-v4l2-event.c-v4l2_event_subscribe()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code | |-- drivers-media-v4l2-core-v4l2-subdev.c-__v4l2_subdev_state_alloc()-error:kvcalloc()-does-not-make-sense-for-no-sleep-code | |-- drivers-media-v4l2-core-v4l2-subdev.c-v4l2_subdev_init_stream_configs()-error:kvcalloc()-does-not-make-sense-for-no-sleep-code | |-- drivers-scsi-sd_zbc.c-sd_zbc_revalidate_zones()-error:kvcalloc()-does-not-make-sense-for-no-sleep-code | |-- fs-btrfs-free-space-tree.c-alloc_bitmap()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code | |-- fs-btrfs-raid56.c-btrfs_alloc_stripe_hash_table()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code | |-- fs-btrfs-scrub.c-scrub_setup_ctx()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code | |-- fs-proc-proc_sysctl.c-proc_sys_call_handler()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code | |-- io_uring-filetable.c-io_alloc_file_tables()-error:kvcalloc()-does-not-make-sense-for-no-sleep-code | |-- kernel-resource.c-walk_system_ram_res_rev()-error:kvcalloc()-does-not-make-sense-for-no-sleep-code | |-- lib-stackdepot.c-stack_depot_init()-error:kvcalloc()-does-not-make-sense-for-no-sleep-code | |-- net-ethtool-common.c-ethtool_get_max_rxnfc_channel()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code | |-- net-ipv4-fib_semantics.c-fib_create_info()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code | |-- net-ipv4-tcp_metrics.c-tcp_metrics_hash_alloc()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code | |-- sound-core-info.c-snd_info_text_entry_write()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code | |-- sound-core-memalloc.c-snd_dma_sg_fallback_alloc()-error:kvcalloc()-does-not-make-sense-for-no-sleep-code | `-- sound-soc-intel-atom-sst-sst.c-intel_sst_suspend()-error:kvzalloc()-does-not-make-sense-for-no-sleep-code |-- i386-randconfig-r123-20240229 | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-leds-flash-leds-ktd2692.c:sparse:sparse:symbol-ktd2692_timing-was-not-declared.-Should-it-be-static | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:cast-to-restricted-__le32 | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:cast-to-restricted-__le64 | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:restricted-__le32-degrades-to-integer | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:restricted-__le64-degrades-to-integer | `-- io_uring-io_uring.c:sparse:sparse:cast-to-restricted-io_req_flags_t |-- mips-bcm47xx_defconfig | `-- drivers-leds-leds-gpio-register.c:warning:attribute-declaration-must-precede-definition |-- powerpc-allyesconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- powerpc-randconfig-003-20240229 | `-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg |-- powerpc64-randconfig-002-20240229 | `-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg |-- riscv-allmodconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- riscv-allyesconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- riscv-defconfig | `-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg |-- s390-allmodconfig | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- x86_64-allmodconfig | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-max_group_count-not-described-in-drm_dp_tunnel_mgr_create | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tracker-not-described-in-drm_dp_tunnel_put | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tunnel-not-described-in-drm_dp_tunnel_put | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:expecting-prototype-for-drm_dp_tunnel_atomic_get_allocated_bw().-Prototype-was-for-drm_dp_tunnel_get_allocated_bw()-instead | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- x86_64-allyesconfig | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-max_group_count-not-described-in-drm_dp_tunnel_mgr_create | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tracker-not-described-in-drm_dp_tunnel_put | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tunnel-not-described-in-drm_dp_tunnel_put | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:expecting-prototype-for-drm_dp_tunnel_atomic_get_allocated_bw().-Prototype-was-for-drm_dp_tunnel_get_allocated_bw()-instead | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- x86_64-randconfig-001-20240229 | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-max_group_count-not-described-in-drm_dp_tunnel_mgr_create | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tracker-not-described-in-drm_dp_tunnel_put | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tunnel-not-described-in-drm_dp_tunnel_put | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:expecting-prototype-for-drm_dp_tunnel_atomic_get_allocated_bw().-Prototype-was-for-drm_dp_tunnel_get_allocated_bw()-instead | `-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg |-- x86_64-randconfig-003-20240229 | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-max_group_count-not-described-in-drm_dp_tunnel_mgr_create | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tracker-not-described-in-drm_dp_tunnel_put | |-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:Function-parameter-or-struct-member-tunnel-not-described-in-drm_dp_tunnel_put | `-- drivers-gpu-drm-display-drm_dp_tunnel.c:warning:expecting-prototype-for-drm_dp_tunnel_atomic_get_allocated_bw().-Prototype-was-for-drm_dp_tunnel_get_allocated_bw()-instead |-- x86_64-randconfig-004-20240229 | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- x86_64-randconfig-005-20240229 | `-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead |-- x86_64-randconfig-122-20240229 | |-- drivers-gpu-drm-nouveau-nvkm-subdev-gsp-r535.c:warning:Function-parameter-or-struct-member-gsp-not-described-in-nvkm_gsp_radix3_sg | |-- drivers-video-backlight-ktd2801-backlight.c:sparse:sparse:symbol-ktd2801_timing-was-not-declared.-Should-it-be-static | |-- fs-ubifs-journal.c:warning:expecting-prototype-for-wake_up_reservation().-Prototype-was-for-add_or_start_queue()-instead | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:cast-to-restricted-__le32 | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:cast-to-restricted-__le64 | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:restricted-__le32-degrades-to-integer | |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:restricted-__le64-degrades-to-integer | `-- io_uring-io_uring.c:sparse:sparse:cast-to-restricted-io_req_flags_t `-- x86_64-randconfig-123-20240229 |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:cast-to-restricted-__le32 |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:cast-to-restricted-__le64 |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:restricted-__le32-degrades-to-integer |-- include-trace-..-..-drivers-bus-mhi-host-trace.h:sparse:sparse:restricted-__le64-degrades-to-integer `-- io_uring-io_uring.c:sparse:sparse:cast-to-restricted-io_req_flags_t elapsed time: 730m configs tested: 179 configs skipped: 3 tested configs: alpha allnoconfig gcc alpha allyesconfig gcc alpha defconfig gcc arc allmodconfig gcc arc allnoconfig gcc arc allyesconfig gcc arc defconfig gcc arc randconfig-001-20240229 gcc arc randconfig-002-20240229 gcc arc vdk_hs38_smp_defconfig gcc arm allmodconfig gcc arm allnoconfig clang arm allyesconfig gcc arm defconfig clang arm gemini_defconfig clang arm integrator_defconfig clang arm jornada720_defconfig clang arm mv78xx0_defconfig clang arm pxa3xx_defconfig clang arm randconfig-001-20240229 gcc arm randconfig-002-20240229 gcc arm randconfig-003-20240229 clang arm randconfig-004-20240229 gcc arm spitz_defconfig gcc arm64 allmodconfig clang arm64 allnoconfig gcc arm64 defconfig gcc arm64 randconfig-001-20240229 clang arm64 randconfig-002-20240229 gcc arm64 randconfig-003-20240229 clang arm64 randconfig-004-20240229 clang csky allmodconfig gcc csky allnoconfig gcc csky allyesconfig gcc csky defconfig gcc csky randconfig-001-20240229 gcc csky randconfig-002-20240229 gcc hexagon allmodconfig clang hexagon allnoconfig clang hexagon allyesconfig clang hexagon defconfig clang hexagon randconfig-001-20240229 clang hexagon randconfig-002-20240229 clang i386 allmodconfig gcc i386 allnoconfig gcc i386 allyesconfig gcc i386 buildonly-randconfig-001-20240229 clang i386 buildonly-randconfig-002-20240229 gcc i386 buildonly-randconfig-003-20240229 gcc i386 buildonly-randconfig-004-20240229 clang i386 buildonly-randconfig-005-20240229 gcc i386 buildonly-randconfig-006-20240229 gcc i386 defconfig clang i386 randconfig-001-20240229 clang i386 randconfig-002-20240229 gcc i386 randconfig-003-20240229 clang i386 randconfig-004-20240229 gcc i386 randconfig-005-20240229 gcc i386 randconfig-006-20240229 gcc i386 randconfig-011-20240229 gcc i386 randconfig-012-20240229 gcc i386 randconfig-013-20240229 clang i386 randconfig-014-20240229 clang i386 randconfig-015-20240229 clang i386 randconfig-016-20240229 clang loongarch allmodconfig gcc loongarch allnoconfig gcc loongarch defconfig gcc loongarch randconfig-001-20240229 gcc loongarch randconfig-002-20240229 gcc m68k allmodconfig gcc m68k allnoconfig gcc m68k allyesconfig gcc m68k defconfig gcc m68k sun3_defconfig gcc microblaze allmodconfig gcc microblaze allnoconfig gcc microblaze allyesconfig gcc microblaze defconfig gcc mips allnoconfig gcc mips allyesconfig gcc mips jazz_defconfig clang mips rm200_defconfig gcc mips vocore2_defconfig clang nios2 allmodconfig gcc nios2 allnoconfig gcc nios2 allyesconfig gcc nios2 defconfig gcc nios2 randconfig-001-20240229 gcc nios2 randconfig-002-20240229 gcc openrisc allnoconfig gcc openrisc allyesconfig gcc openrisc defconfig gcc parisc allmodconfig gcc parisc allnoconfig gcc parisc allyesconfig gcc parisc defconfig gcc parisc randconfig-001-20240229 gcc parisc randconfig-002-20240229 gcc parisc64 defconfig gcc powerpc allmodconfig gcc powerpc allnoconfig gcc powerpc allyesconfig clang powerpc ep8248e_defconfig gcc powerpc ksi8560_defconfig gcc powerpc randconfig-001-20240229 clang powerpc randconfig-002-20240229 gcc powerpc randconfig-003-20240229 clang powerpc xes_mpc85xx_defconfig gcc powerpc64 randconfig-001-20240229 gcc powerpc64 randconfig-002-20240229 clang powerpc64 randconfig-003-20240229 clang riscv allmodconfig clang riscv allnoconfig gcc riscv allyesconfig clang riscv defconfig clang riscv randconfig-001-20240229 clang riscv randconfig-002-20240229 clang s390 allmodconfig clang s390 allnoconfig clang s390 allyesconfig gcc s390 defconfig clang s390 randconfig-001-20240229 clang s390 randconfig-002-20240229 gcc sh alldefconfig gcc sh allmodconfig gcc sh allnoconfig gcc sh allyesconfig gcc sh defconfig gcc sh randconfig-001-20240229 gcc sh randconfig-002-20240229 gcc sh sdk7780_defconfig gcc sparc allmodconfig gcc sparc allnoconfig gcc sparc defconfig gcc sparc64 allmodconfig gcc sparc64 allyesconfig gcc sparc64 defconfig gcc sparc64 randconfig-001-20240229 gcc sparc64 randconfig-002-20240229 gcc um allmodconfig clang um allnoconfig clang um allyesconfig gcc um defconfig clang um i386_defconfig gcc um randconfig-001-20240229 gcc um randconfig-002-20240229 gcc um x86_64_defconfig clang x86_64 allnoconfig clang x86_64 allyesconfig clang x86_64 buildonly-randconfig-001-20240229 gcc x86_64 buildonly-randconfig-002-20240229 clang x86_64 buildonly-randconfig-003-20240229 gcc x86_64 buildonly-randconfig-004-20240229 gcc x86_64 buildonly-randconfig-005-20240229 gcc x86_64 buildonly-randconfig-006-20240229 gcc x86_64 defconfig gcc x86_64 randconfig-001-20240229 clang x86_64 randconfig-002-20240229 clang x86_64 randconfig-003-20240229 clang x86_64 randconfig-004-20240229 clang x86_64 randconfig-005-20240229 clang x86_64 randconfig-006-20240229 clang x86_64 randconfig-011-20240229 gcc x86_64 randconfig-012-20240229 gcc x86_64 randconfig-013-20240229 gcc x86_64 randconfig-014-20240229 gcc x86_64 randconfig-015-20240229 gcc x86_64 randconfig-016-20240229 gcc x86_64 randconfig-071-20240229 clang x86_64 randconfig-072-20240229 clang x86_64 randconfig-073-20240229 clang x86_64 randconfig-074-20240229 clang x86_64 randconfig-075-20240229 gcc x86_64 randconfig-076-20240229 clang x86_64 rhel-8.3-rust clang xtensa allnoconfig gcc xtensa randconfig-001-20240229 gcc xtensa randconfig-002-20240229 gcc -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH] mfd: cs42l43: Fix wrong register defaults
by Maciej Strozek 29 Feb '24

29 Feb '24

A few regs have unnecessary values in defaults, change them to match the datasheet Fixes: ace6d1448138 ("mfd: cs42l43: Add support for cs42l43 core driver") Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- drivers/mfd/cs42l43.c | 68 +++++++++++++++++++++---------------------- 1 file changed, 34 insertions(+), 34 deletions(-) diff --git a/drivers/mfd/cs42l43.c b/drivers/mfd/cs42l43.c index 7b6d07cbe6fc..73c88ee6a866 100644 --- a/drivers/mfd/cs42l43.c +++ b/drivers/mfd/cs42l43.c @@ -131,38 +131,38 @@ const struct reg_default cs42l43_reg_default[CS42L43_N_DEFAULTS] = { { CS42L43_ASP_TX_CH4_CTRL, 0x00170091 }, { CS42L43_ASP_TX_CH5_CTRL, 0x001700C1 }, { CS42L43_ASP_TX_CH6_CTRL, 0x001700F1 }, - { CS42L43_ASPTX1_INPUT, 0x00800000 }, - { CS42L43_ASPTX2_INPUT, 0x00800000 }, - { CS42L43_ASPTX3_INPUT, 0x00800000 }, - { CS42L43_ASPTX4_INPUT, 0x00800000 }, - { CS42L43_ASPTX5_INPUT, 0x00800000 }, - { CS42L43_ASPTX6_INPUT, 0x00800000 }, - { CS42L43_SWIRE_DP1_CH1_INPUT, 0x00800000 }, - { CS42L43_SWIRE_DP1_CH2_INPUT, 0x00800000 }, - { CS42L43_SWIRE_DP1_CH3_INPUT, 0x00800000 }, - { CS42L43_SWIRE_DP1_CH4_INPUT, 0x00800000 }, - { CS42L43_SWIRE_DP2_CH1_INPUT, 0x00800000 }, - { CS42L43_SWIRE_DP2_CH2_INPUT, 0x00800000 }, - { CS42L43_SWIRE_DP3_CH1_INPUT, 0x00800000 }, - { CS42L43_SWIRE_DP3_CH2_INPUT, 0x00800000 }, - { CS42L43_SWIRE_DP4_CH1_INPUT, 0x00800000 }, - { CS42L43_SWIRE_DP4_CH2_INPUT, 0x00800000 }, - { CS42L43_ASRC_INT1_INPUT1, 0x00800000 }, - { CS42L43_ASRC_INT2_INPUT1, 0x00800000 }, - { CS42L43_ASRC_INT3_INPUT1, 0x00800000 }, - { CS42L43_ASRC_INT4_INPUT1, 0x00800000 }, - { CS42L43_ASRC_DEC1_INPUT1, 0x00800000 }, - { CS42L43_ASRC_DEC2_INPUT1, 0x00800000 }, - { CS42L43_ASRC_DEC3_INPUT1, 0x00800000 }, - { CS42L43_ASRC_DEC4_INPUT1, 0x00800000 }, - { CS42L43_ISRC1INT1_INPUT1, 0x00800000 }, - { CS42L43_ISRC1INT2_INPUT1, 0x00800000 }, - { CS42L43_ISRC1DEC1_INPUT1, 0x00800000 }, - { CS42L43_ISRC1DEC2_INPUT1, 0x00800000 }, - { CS42L43_ISRC2INT1_INPUT1, 0x00800000 }, - { CS42L43_ISRC2INT2_INPUT1, 0x00800000 }, - { CS42L43_ISRC2DEC1_INPUT1, 0x00800000 }, - { CS42L43_ISRC2DEC2_INPUT1, 0x00800000 }, + { CS42L43_ASPTX1_INPUT, 0x00000000 }, + { CS42L43_ASPTX2_INPUT, 0x00000000 }, + { CS42L43_ASPTX3_INPUT, 0x00000000 }, + { CS42L43_ASPTX4_INPUT, 0x00000000 }, + { CS42L43_ASPTX5_INPUT, 0x00000000 }, + { CS42L43_ASPTX6_INPUT, 0x00000000 }, + { CS42L43_SWIRE_DP1_CH1_INPUT, 0x00000000 }, + { CS42L43_SWIRE_DP1_CH2_INPUT, 0x00000000 }, + { CS42L43_SWIRE_DP1_CH3_INPUT, 0x00000000 }, + { CS42L43_SWIRE_DP1_CH4_INPUT, 0x00000000 }, + { CS42L43_SWIRE_DP2_CH1_INPUT, 0x00000000 }, + { CS42L43_SWIRE_DP2_CH2_INPUT, 0x00000000 }, + { CS42L43_SWIRE_DP3_CH1_INPUT, 0x00000000 }, + { CS42L43_SWIRE_DP3_CH2_INPUT, 0x00000000 }, + { CS42L43_SWIRE_DP4_CH1_INPUT, 0x00000000 }, + { CS42L43_SWIRE_DP4_CH2_INPUT, 0x00000000 }, + { CS42L43_ASRC_INT1_INPUT1, 0x00000000 }, + { CS42L43_ASRC_INT2_INPUT1, 0x00000000 }, + { CS42L43_ASRC_INT3_INPUT1, 0x00000000 }, + { CS42L43_ASRC_INT4_INPUT1, 0x00000000 }, + { CS42L43_ASRC_DEC1_INPUT1, 0x00000000 }, + { CS42L43_ASRC_DEC2_INPUT1, 0x00000000 }, + { CS42L43_ASRC_DEC3_INPUT1, 0x00000000 }, + { CS42L43_ASRC_DEC4_INPUT1, 0x00000000 }, + { CS42L43_ISRC1INT1_INPUT1, 0x00000000 }, + { CS42L43_ISRC1INT2_INPUT1, 0x00000000 }, + { CS42L43_ISRC1DEC1_INPUT1, 0x00000000 }, + { CS42L43_ISRC1DEC2_INPUT1, 0x00000000 }, + { CS42L43_ISRC2INT1_INPUT1, 0x00000000 }, + { CS42L43_ISRC2INT2_INPUT1, 0x00000000 }, + { CS42L43_ISRC2DEC1_INPUT1, 0x00000000 }, + { CS42L43_ISRC2DEC2_INPUT1, 0x00000000 }, { CS42L43_EQ1MIX_INPUT1, 0x00800000 }, { CS42L43_EQ1MIX_INPUT2, 0x00800000 }, { CS42L43_EQ1MIX_INPUT3, 0x00800000 }, @@ -171,8 +171,8 @@ const struct reg_default cs42l43_reg_default[CS42L43_N_DEFAULTS] = { { CS42L43_EQ2MIX_INPUT2, 0x00800000 }, { CS42L43_EQ2MIX_INPUT3, 0x00800000 }, { CS42L43_EQ2MIX_INPUT4, 0x00800000 }, - { CS42L43_SPDIF1_INPUT1, 0x00800000 }, - { CS42L43_SPDIF2_INPUT1, 0x00800000 }, + { CS42L43_SPDIF1_INPUT1, 0x00000000 }, + { CS42L43_SPDIF2_INPUT1, 0x00000000 }, { CS42L43_AMP1MIX_INPUT1, 0x00800000 }, { CS42L43_AMP1MIX_INPUT2, 0x00800000 }, { CS42L43_AMP1MIX_INPUT3, 0x00800000 }, -- 2.30.2

2 1

Re: [PATCH 2/4] reset: add GPIO-based reset controller
by Tim Harvey 29 Feb '24

29 Feb '24

> On 1/9/24 04:41, Krzysztof Kozlowski wrote: > > I think a separate pseudo-device is necessary a generic solution. So I > guess I will revive my patchset. > Sean and Krzysztof, I see a lot of value in a generic reset-gpio driver that you have both tried to get accepted in the past. I support boards that have a number of SPI and I2C devices that often have GPIO resets wired to them that are pulled in hardware to the in-reset state and find no support in the various drivers for reset handling. I've often wondered why a generic gpio reset wasn't supported in the SPI/I2C cores like it is for some other subsystems. The approach of a gpios-reset solution makes sense to me. Will you be submitting a follow-on series to your previous attempts? Best Regards, Tim

2 1

[PATCH AUTOSEL 6.6 04/21] ASoC: cs42l43: Handle error from devm_pm_runtime_enable
by Sasha Levin 29 Feb '24

29 Feb '24

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit d1722057477a3786b8c0d60c28fc281f6ecf1cc3 ] As devm_pm_runtime_enable can fail due to memory allocations, it is best to handle the error. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20240206113850.719888-1-ckeepax@opensource.cirrus… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l43.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 5643c666d7d04..8015f4b7a5b32 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2175,7 +2175,10 @@ static int cs42l43_codec_probe(struct platform_device *pdev) pm_runtime_use_autosuspend(priv->dev); pm_runtime_set_active(priv->dev); pm_runtime_get_noresume(priv->dev); - devm_pm_runtime_enable(priv->dev); + + ret = devm_pm_runtime_enable(priv->dev); + if (ret) + goto err_pm; for (i = 0; i < ARRAY_SIZE(cs42l43_irqs); i++) { ret = cs42l43_request_irq(priv, dom, cs42l43_irqs[i].name, -- 2.43.0

1 0

[PATCH AUTOSEL 6.7 24/26] ALSA: hda/realtek: cs35l41: Add internal speaker support for ASUS UM3402 with missing DSD
by Sasha Levin 29 Feb '24

29 Feb '24

From: Jean-Loïc Charroud <lagiraudiere+linux(a)free.fr> [ Upstream commit 706c1fa1ab09f11a131fc4d699ce4c0224b1cb2d ] Add the values for the missing DSD properties to the cs35l41 config table. Signed-off-by: Jean-Loïc Charroud <lagiraudiere+linux(a)free.fr> Link: https://lore.kernel.org/r/1435594585.650325975.1707867511062.JavaMail.zimbr… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 35277ce890a46..ffdff336678cc 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -89,6 +89,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431D1F", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431DA2", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "10431E02", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "10431E12", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "10431EE2", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "10431F12", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, @@ -423,6 +424,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431D1F", generic_dsd_config }, { "CSC3551", "10431DA2", generic_dsd_config }, { "CSC3551", "10431E02", generic_dsd_config }, + { "CSC3551", "10431E12", generic_dsd_config }, { "CSC3551", "10431EE2", generic_dsd_config }, { "CSC3551", "10431F12", generic_dsd_config }, { "CSC3551", "10431F1F", generic_dsd_config }, -- 2.43.0

1 0

[PATCH AUTOSEL 6.7 05/26] ASoC: cs42l43: Handle error from devm_pm_runtime_enable
by Sasha Levin 29 Feb '24

29 Feb '24

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit d1722057477a3786b8c0d60c28fc281f6ecf1cc3 ] As devm_pm_runtime_enable can fail due to memory allocations, it is best to handle the error. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20240206113850.719888-1-ckeepax@opensource.cirrus… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l43.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index d62c9f26c6325..5009cf64124ed 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2175,7 +2175,10 @@ static int cs42l43_codec_probe(struct platform_device *pdev) pm_runtime_use_autosuspend(priv->dev); pm_runtime_set_active(priv->dev); pm_runtime_get_noresume(priv->dev); - devm_pm_runtime_enable(priv->dev); + + ret = devm_pm_runtime_enable(priv->dev); + if (ret) + goto err_pm; for (i = 0; i < ARRAY_SIZE(cs42l43_irqs); i++) { ret = cs42l43_request_irq(priv, dom, cs42l43_irqs[i].name, -- 2.43.0

1 0

[GIT PULL] ASoC fixes for v6.8-rc5
by Mark Brown 29 Feb '24

29 Feb '24

The following changes since commit 0db0c1770834f39e11a2902e20e1f11a482f4465: ASoC: cs35l56: Workaround for ACPI with broken spk-id-gpios property (2024-02-13 19:37:25 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.8-rc5 for you to fetch changes up to ed00a6945dc32462c2d3744a3518d2316da66fcc: ASoC: amd: yc: Fix non-functional mic on Lenovo 21J2 (2024-02-28 12:24:59 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.8 A few small fixes, some driver specific and one slightly larger one from Richard which adds a new core helper and updates a small clutch of drivers to deal with the fact that they were using a helper which requires that the lock for the list of controls without holding that lock. We also have some quirks for new AMD based Lenovo systems. ---------------------------------------------------------------- Colin Ian King (1): ASoC: qcom: Fix uninitialized pointer dmactl Jiawei Wang (2): ASoC: amd: yc: add new YC platform variant (0x63) support ASoC: amd: yc: Fix non-functional mic on Lenovo 21J2 Johnny Hsieh (1): ASoC: amd: yc: Add Lenovo ThinkBook 21J0 into DMI quirk table Richard Fitzgerald (2): ASoC: cs35l56: Must clear HALO_STATE before issuing SYSTEM_RESET ASoC: soc-card: Fix missing locking in snd_soc_card_get_kcontrol() include/sound/soc-card.h | 2 ++ sound/soc/amd/yc/acp6x-mach.c | 14 ++++++++++++++ sound/soc/amd/yc/pci-acp6x.c | 1 + sound/soc/codecs/cs35l45.c | 2 +- sound/soc/codecs/cs35l56-shared.c | 1 + sound/soc/codecs/cs35l56.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 12 +++++++++++- sound/soc/qcom/lpass-cdc-dma.c | 2 +- sound/soc/soc-card.c | 24 ++++++++++++++++++++++-- 9 files changed, 54 insertions(+), 6 deletions(-)

2 1

alsa cant initialize on raspberry pi 5
by GitHub issues - opened 29 Feb '24

29 Feb '24

alsa-project/alsa-lib issue #386 was opened from sy-project: I can use alsa command ``` # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sndrpihifiberry [snd_rpi_hifiberry_dacplusadcpro], device 0: HiFiBerry DAC+ADC Pro HiFi multicodec-0 [HiFiBerry DAC+ADC Pro HiFi multicodec-0] Subdevices: 1/1 Subdevice #0: subdevice #0 # arecord -l **** List of CAPTURE Hardware Devices **** card 0: sndrpihifiberry [snd_rpi_hifiberry_dacplusadcpro], device 0: HiFiBerry DAC+ADC Pro HiFi multicodec-0 [HiFiBerry DAC+ADC Pro HiFi multicodec-0] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` and I can aplay to use wav file. ``` # aplay -D hw:0,0 /root/C_Code/audiotest/test1.wav Playing WAVE '/root/C_Code/audiotest/test1.wav' : Signed 32 bit Little Endian, Rate 48000 Hz, Stereo ``` but I cant initialize on c code. my code is here. ```c printf("alsa start\n"); int err; snd_pcm_t *capture_handle, *playback_handle; snd_pcm_hw_params_t *hw_params; unsigned char *buffer; int sample_rate = SAMPLE_RATE; int dir; snd_pcm_uframes_t frames = 32; snd_pcm_uframes_t bufferSize, periodSize; double x_prev = 0.0; double y_prev = 0.0; double b0 = 1.0; double b1 = -0.8; double a1 = 0.0; printf("alsa open capture handle\n"); err = snd_pcm_open(&capture_handle, "default", SND_PCM_STREAM_CAPTURE, 0); if(err < 0) { fprintf(stderr, "Capture open error: %s\n", snd_strerror(err)); exit(1); } printf("alsa open playback handle\n"); err = snd_pcm_open(&playback_handle, "default", SND_PCM_STREAM_PLAYBACK, 0); if(err < 0) { fprintf(stderr, "Playback open error: %s\n", snd_strerror(err)); exit(1); } printf("alsa pcm hw param alloc\n"); snd_pcm_hw_params_alloca(&hw_params); printf("alsa capture_handle set start\n"); err = snd_pcm_hw_params_any(capture_handle, hw_params); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_any cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } err = snd_pcm_hw_params_set_access(capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_set_access cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } err = snd_pcm_hw_params_set_format(capture_handle, hw_params, SND_PCM_FORMAT_S16_LE); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_set_format cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } err = snd_pcm_hw_params_set_channels(capture_handle, hw_params, NUM_CHANNELS); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_set_channels cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } err = snd_pcm_hw_params_set_rate_near(capture_handle, hw_params, &sample_rate, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_set_rate_near cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } err = snd_pcm_hw_params_set_buffer_size_near(capture_handle, hw_params, &bufferSize); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_set_buffer_size_near cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } err = snd_pcm_hw_params_set_period_size_near(capture_handle, hw_params, &periodSize, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_set_period_size_near cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } err = snd_pcm_hw_params(capture_handle, hw_params); if (err < 0) { fprintf(stderr, "alsa cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } snd_pcm_hw_params_get_buffer_size(hw_params, &bufferSize); snd_pcm_hw_params_get_period_size(hw_params, &periodSize, &dir); printf("Buffer size: %lu frames\n", bufferSize); printf("Period size: %lu frames\n", periodSize); printf("alsa playback_handle set start\n"); snd_pcm_hw_params_any(playback_handle, hw_params); snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(playback_handle, hw_params, NUM_CHANNELS); snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &sample_rate, 0); snd_pcm_hw_params(playback_handle, hw_params); printf("alsa playback_handle set success\n"); printf("alsa buf malloc\n"); buffer = (unsigned char*)malloc(BUFFER_SIZE*NUM_CHANNELS*2); while(1) { printf("main loop start\n"); err = snd_pcm_readi(capture_handle, buffer, BUFFER_SIZE); if(err<0) { fprintf(stderr, "Capture read error[%d]: %s\n", err, snd_strerror(err)); break; } for(int i = 0; i < BUFFER_SIZE * NUM_CHANNELS * 2; i += 2) { short sample = ((short)buffer[i + 1] << 8) | buffer[i]; double x = (double)sample / 32768.0; double y = b0 * x + b1 * x_prev + a1 * y_prev; x_prev = x; y_prev = y; short filter_sample = (short)(y * 32768.0); buffer[i] = (unsigned char)(filter_sample & 0xFF); buffer[i+1] = (unsigned char)((filter_sample >> 8) & 0xFF); } err = snd_pcm_writei(playback_handle, buffer, BUFFER_SIZE); if(err < 0) { fprintf(stderr, "Playback write error: %s\n", snd_strerror(err)); break; } } free(buffer); snd_pcm_close(capture_handle); snd_pcm_close(playback_handle); exit(0); ``` This code said ``` alsa start alsa open capture handle alsa open playback handle alsa pcm hw param alloc alsa capture_handle set start snd_pcm_hw_params_set_buffer_size_near cant set hw params. [Invalid argument] ``` what should I do? Issue URL : https://github.com/alsa-project/alsa-lib/issues/386 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: dice: add stream format parameters for Weiss devices
by Michele Perrone 29 Feb '24

29 Feb '24

Hard-coded stream format parameters are added for Weiss Engineering FireWire devices. When the device vendor and model match, the parameters are copied into the stream format cache. This allows for setting all supported sampling rates up to 192kHz, and consequently adjusting the number of available I/O channels. Signed-off-by: Rolf Anderegg <rolf.anderegg(a)weiss.ch> Signed-off-by: Michele Perrone <michele.perrone(a)weiss.ch> --- sound/firewire/dice/Makefile | 2 +- sound/firewire/dice/dice-weiss.c | 120 +++++++++++++++++++++++++++++++ sound/firewire/dice/dice.c | 72 +++++++++++++++++++ 3 files changed, 193 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/dice/dice-weiss.c diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index a5f3fbf28b8c..bac8712f9014 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -2,5 +2,5 @@ snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \ dice-alesis.o dice-extension.o dice-mytek.o dice-presonus.o \ - dice-harman.o dice-focusrite.o + dice-harman.o dice-focusrite.o dice-weiss.o obj-$(CONFIG_SND_DICE) += snd-dice.o diff --git a/sound/firewire/dice/dice-weiss.c b/sound/firewire/dice/dice-weiss.c new file mode 100644 index 000000000000..ad22bfafd324 --- /dev/null +++ b/sound/firewire/dice/dice-weiss.c @@ -0,0 +1,120 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * dice-weiss.c - a part of driver for DICE based devices + * + * Copyright (c) 2023 Rolf Anderegg and Michele Perrone + */ + +#include "dice.h" + +struct dice_weiss_spec { + unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + bool has_midi; +}; + +/* Weiss DAC202: 192kHz 2-channel DAC */ +static const struct dice_weiss_spec dac202 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss MAN301: 192kHz 2-channel music archive network player */ +static const struct dice_weiss_spec man301 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss INT202: 192kHz unidirectional 2-channel digital Firewire interface */ +static const struct dice_weiss_spec int202 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{0, 0, 0}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss INT203: 192kHz bidirectional 2-channel digital Firewire interface */ +static const struct dice_weiss_spec int203 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss ADC2: 192kHz A/D converter with microphone preamps and line inputs */ +static const struct dice_weiss_spec adc2 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss DAC2/Minerva: 192kHz 2-channel DAC */ +static const struct dice_weiss_spec dac2_minerva = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss Vesta: 192kHz 2-channel Firewire to AES/EBU interface */ +static const struct dice_weiss_spec vesta = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss AFI1: 192kHz 24-channel Firewire to ADAT or AES/EBU interface */ +static const struct dice_weiss_spec afi1 = { + .tx_pcm_chs = {{24, 16, 8}, {0, 0, 0} }, + .rx_pcm_chs = {{24, 16, 8}, {0, 0, 0} }, + .has_midi = false +}; + +int snd_dice_detect_weiss_formats(struct snd_dice *dice) +{ + static const struct { + u32 model_id; + const struct dice_weiss_spec *spec; + } *entry, entries[] = { + {0x000007, &dac202}, + {0x000008, &dac202}, /* Maya edition: same audio I/O as DAC202 */ + {0x000006, &int202}, + {0x00000a, &int203}, + {0x00000b, &man301}, + {0x000001, &adc2}, + {0x000003, &dac2_minerva}, + {0x000002, &vesta}, + {0x000004, &afi1}, + }; + struct fw_csr_iterator it; + int key, val, model_id; + int i; + + model_id = 0; + fw_csr_iterator_init(&it, dice->unit->directory); + while (fw_csr_iterator_next(&it, &key, &val)) { + if (key == CSR_MODEL) { + model_id = val; + break; + } + } + + for (i = 0; i < ARRAY_SIZE(entries); ++i) { + entry = entries + i; + if (entry->model_id == model_id) + break; + } + if (i == ARRAY_SIZE(entries)) + return -ENODEV; + + memcpy(dice->tx_pcm_chs, entry->spec->tx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + memcpy(dice->rx_pcm_chs, entry->spec->rx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + + if (entry->spec->has_midi) { + dice->tx_midi_ports[0] = 1; + dice->rx_midi_ports[0] = 1; + } + + return 0; +} diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 6c93e6e4982c..4bcd026a0b3f 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -396,6 +396,78 @@ static const struct ieee1394_device_id dice_id_table[] = { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, }, + /* Weiss DAC202: 192kHz 2-channel DAC */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000007, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss DAC202: 192kHz 2-channel DAC (Maya edition) */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000008, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss MAN301: 192kHz 2-channel music archive network player */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x00000b, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss INT202: 192kHz unidirectional 2-channel digital Firewire interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000006, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss INT203: 192kHz bidirectional 2-channel digital Firewire interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x00000a, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss ADC2: 192kHz A/D converter with microphone preamps and line inputs */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000001, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss DAC2/Minerva: 192kHz 2-channel DAC */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000003, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss Vesta: 192kHz 2-channel Firewire to AES/EBU interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000002, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss AFI1: 192kHz 24-channel Firewire to ADAT or AES/EBU interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000004, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, dice_id_table); base-commit: 9b4469410cf9a0fcbccc92c480fd42f7c815a745 -- 2.41.0

3 26

[PATCH] Convert the Imagination Technologies SPDIF Input Controllerto DT schema.
by Javier García 28 Feb '24

28 Feb '24

Convert the Imagination Technologies SPDIF Input Controllerto DT schema. Signed-off-by: Javier García <javier.garcia.ta(a)udima.es> --- .../bindings/sound/img,spdif-in.txt | 41 ---------- .../bindings/sound/img,spdif-in.yaml | 80 +++++++++++++++++++ 2 files changed, 80 insertions(+), 41 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/img,spdif-in.txt create mode 100644 Documentation/devicetree/bindings/sound/img,spdif-in.yaml diff --git a/Documentation/devicetree/bindings/sound/img,spdif-in.txt b/Documentation/devicetree/bindings/sound/img,spdif-in.txt deleted file mode 100644 index f7ea8c87bf34..000000000000 --- a/Documentation/devicetree/bindings/sound/img,spdif-in.txt +++ /dev/null @@ -1,41 +0,0 @@ -Imagination Technologies SPDIF Input Controller - -Required Properties: - - - compatible : Compatible list, must contain "img,spdif-in" - - - #sound-dai-cells : Must be equal to 0 - - - reg : Offset and length of the register set for the device - - - dmas: Contains an entry for each entry in dma-names. - - - dma-names: Must include the following entry: - "rx" - - - clocks : Contains an entry for each entry in clock-names - - - clock-names : Includes the following entries: - "sys" The system clock - -Optional Properties: - - - resets: Should contain a phandle to the spdif in reset signal, if any - - - reset-names: Should contain the reset signal name "rst", if a - reset phandle is given - - - interrupts : Contains the spdif in interrupt, if present - -Example: - -spdif_in: spdif-in@18100e00 { - compatible = "img,spdif-in"; - reg = <0x18100E00 0x100>; - interrupts = <GIC_SHARED 20 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&mdc 15 0xffffffff 0>; - dma-names = "rx"; - clocks = <&cr_periph SYS_CLK_SPDIF_IN>; - clock-names = "sys"; - #sound-dai-cells = <0>; -}; diff --git a/Documentation/devicetree/bindings/sound/img,spdif-in.yaml b/Documentation/devicetree/bindings/sound/img,spdif-in.yaml new file mode 100644 index 000000000000..d201293d63c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/img,spdif-in.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/img,spdif-in.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Imagination Technologies SPDIF Input Controller + +maintainers: + - Rob Herring <robh+dt(a)kernel.org> + - Krzysztof Kozlowski <krzysztof.kozlowski+dt(a)linaro.org> + +properties: + compatible: + enum: + - img,spdif-in + + reg: + description: + Offset and length of the register set for the device. + maxItems: 1 + + interrupts: + maxItems: 1 + + dmas: + maxItems: 1 + + dma-names: + items: + - const: rx + + clocks: + items: + - description: The system clock + + clock-names: + items: + - const: sys + + '#sound-dai-cells': + const: 0 + + resets: + items: + - description: Should contain a phandle to the spdif in reset signal, if any + + reset-names: + items: + - const: rst + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/pistachio-clk.h> + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/interrupt-controller/mips-gic.h> + #include <dt-bindings/reset/pistachio-resets.h> + spdif_in: spdif-in@18100e00 { + compatible = "img,spdif-in"; + reg = <0x18100E00 0x100>; + interrupts = <GIC_SHARED 20 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 15 0xffffffff 0>; + dma-names = "rx"; + clocks = <&cr_periph SYS_CLK_SPDIF_IN>; + clock-names = "sys"; + + #sound-dai-cells = <0>; + }; -- 2.43.0

4 9

[PATCH v2] ASoC: meson: axg-fifo: use FIELD helpers
by Jerome Brunet 27 Feb '24

27 Feb '24

Use FIELD_GET() and FIELD_PREP() helpers instead of doing it manually. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- Changes since v1: [1] * Add missing include <linux/bitfield.h> causing build error with x86 compile test. [1]: http://lore.kernel.org/alsa-devel/20240223175116.2005407-7-jbrunet@baylibre… sound/soc/meson/axg-fifo.c | 25 +++++++++++++------------ sound/soc/meson/axg-fifo.h | 12 +++++------- sound/soc/meson/axg-frddr.c | 5 +++-- sound/soc/meson/axg-toddr.c | 22 ++++++++++------------ 4 files changed, 31 insertions(+), 33 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 65541fdb0038..bebee0ca8e38 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -3,6 +3,7 @@ // Copyright (c) 2018 BayLibre, SAS. // Author: Jerome Brunet <jbrunet(a)baylibre.com> +#include <linux/bitfield.h> #include <linux/clk.h> #include <linux/of_irq.h> #include <linux/of_platform.h> @@ -145,8 +146,8 @@ int axg_fifo_pcm_hw_params(struct snd_soc_component *component, /* Enable irq if necessary */ irq_en = runtime->no_period_wakeup ? 0 : FIFO_INT_COUNT_REPEAT; regmap_update_bits(fifo->map, FIFO_CTRL0, - CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), - CTRL0_INT_EN(irq_en)); + CTRL0_INT_EN, + FIELD_PREP(CTRL0_INT_EN, irq_en)); return 0; } @@ -176,9 +177,9 @@ int axg_fifo_pcm_hw_free(struct snd_soc_component *component, { struct axg_fifo *fifo = axg_fifo_data(ss); - /* Disable the block count irq */ + /* Disable irqs */ regmap_update_bits(fifo->map, FIFO_CTRL0, - CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), 0); + CTRL0_INT_EN, 0); return 0; } @@ -187,13 +188,13 @@ EXPORT_SYMBOL_GPL(axg_fifo_pcm_hw_free); static void axg_fifo_ack_irq(struct axg_fifo *fifo, u8 mask) { regmap_update_bits(fifo->map, FIFO_CTRL1, - CTRL1_INT_CLR(FIFO_INT_MASK), - CTRL1_INT_CLR(mask)); + CTRL1_INT_CLR, + FIELD_PREP(CTRL1_INT_CLR, mask)); /* Clear must also be cleared */ regmap_update_bits(fifo->map, FIFO_CTRL1, - CTRL1_INT_CLR(FIFO_INT_MASK), - 0); + CTRL1_INT_CLR, + FIELD_PREP(CTRL1_INT_CLR, 0)); } static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) @@ -204,7 +205,7 @@ static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) regmap_read(fifo->map, FIFO_STATUS1, &status); - status = STATUS1_INT_STS(status) & FIFO_INT_MASK; + status = FIELD_GET(STATUS1_INT_STS, status); if (status & FIFO_INT_COUNT_REPEAT) snd_pcm_period_elapsed(ss); else @@ -254,15 +255,15 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, /* Setup status2 so it reports the memory pointer */ regmap_update_bits(fifo->map, FIFO_CTRL1, - CTRL1_STATUS2_SEL_MASK, - CTRL1_STATUS2_SEL(STATUS2_SEL_DDR_READ)); + CTRL1_STATUS2_SEL, + FIELD_PREP(CTRL1_STATUS2_SEL, STATUS2_SEL_DDR_READ)); /* Make sure the dma is initially disabled */ __dma_enable(fifo, false); /* Disable irqs until params are ready */ regmap_update_bits(fifo->map, FIFO_CTRL0, - CTRL0_INT_EN(FIFO_INT_MASK), 0); + CTRL0_INT_EN, 0); /* Clear any pending interrupt */ axg_fifo_ack_irq(fifo, FIFO_INT_MASK); diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h index a14c31eb06d8..4c48c0a08481 100644 --- a/sound/soc/meson/axg-fifo.h +++ b/sound/soc/meson/axg-fifo.h @@ -40,21 +40,19 @@ struct snd_soc_pcm_runtime; #define FIFO_CTRL0 0x00 #define CTRL0_DMA_EN BIT(31) -#define CTRL0_INT_EN(x) ((x) << 16) +#define CTRL0_INT_EN GENMASK(23, 16) #define CTRL0_SEL_MASK GENMASK(2, 0) #define CTRL0_SEL_SHIFT 0 #define FIFO_CTRL1 0x04 -#define CTRL1_INT_CLR(x) ((x) << 0) -#define CTRL1_STATUS2_SEL_MASK GENMASK(11, 8) -#define CTRL1_STATUS2_SEL(x) ((x) << 8) +#define CTRL1_INT_CLR GENMASK(7, 0) +#define CTRL1_STATUS2_SEL GENMASK(11, 8) #define STATUS2_SEL_DDR_READ 0 -#define CTRL1_FRDDR_DEPTH_MASK GENMASK(31, 24) -#define CTRL1_FRDDR_DEPTH(x) ((x) << 24) +#define CTRL1_FRDDR_DEPTH GENMASK(31, 24) #define FIFO_START_ADDR 0x08 #define FIFO_FINISH_ADDR 0x0c #define FIFO_INT_ADDR 0x10 #define FIFO_STATUS1 0x14 -#define STATUS1_INT_STS(x) ((x) << 0) +#define STATUS1_INT_STS GENMASK(7, 0) #define FIFO_STATUS2 0x18 #define FIFO_INIT_ADDR 0x24 #define FIFO_CTRL2 0x28 diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c index 98140f449eb3..e97d43ae7fd2 100644 --- a/sound/soc/meson/axg-frddr.c +++ b/sound/soc/meson/axg-frddr.c @@ -7,6 +7,7 @@ * This driver implements the frontend playback DAI of AXG and G12A based SoCs */ +#include <linux/bitfield.h> #include <linux/clk.h> #include <linux/regmap.h> #include <linux/module.h> @@ -59,8 +60,8 @@ static int axg_frddr_dai_hw_params(struct snd_pcm_substream *substream, /* Trim the FIFO depth if the period is small to improve latency */ depth = min(period, fifo->depth); val = (depth / AXG_FIFO_BURST) - 1; - regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH_MASK, - CTRL1_FRDDR_DEPTH(val)); + regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH, + FIELD_PREP(CTRL1_FRDDR_DEPTH, val)); return 0; } diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c index 32ee45cce7f8..e03a6e21c1c6 100644 --- a/sound/soc/meson/axg-toddr.c +++ b/sound/soc/meson/axg-toddr.c @@ -5,6 +5,7 @@ /* This driver implements the frontend capture DAI of AXG based SoCs */ +#include <linux/bitfield.h> #include <linux/clk.h> #include <linux/regmap.h> #include <linux/module.h> @@ -19,12 +20,9 @@ #define CTRL0_TODDR_EXT_SIGNED BIT(29) #define CTRL0_TODDR_PP_MODE BIT(28) #define CTRL0_TODDR_SYNC_CH BIT(27) -#define CTRL0_TODDR_TYPE_MASK GENMASK(15, 13) -#define CTRL0_TODDR_TYPE(x) ((x) << 13) -#define CTRL0_TODDR_MSB_POS_MASK GENMASK(12, 8) -#define CTRL0_TODDR_MSB_POS(x) ((x) << 8) -#define CTRL0_TODDR_LSB_POS_MASK GENMASK(7, 3) -#define CTRL0_TODDR_LSB_POS(x) ((x) << 3) +#define CTRL0_TODDR_TYPE GENMASK(15, 13) +#define CTRL0_TODDR_MSB_POS GENMASK(12, 8) +#define CTRL0_TODDR_LSB_POS GENMASK(7, 3) #define CTRL1_TODDR_FORCE_FINISH BIT(25) #define CTRL1_SEL_SHIFT 28 @@ -76,12 +74,12 @@ static int axg_toddr_dai_hw_params(struct snd_pcm_substream *substream, width = params_width(params); regmap_update_bits(fifo->map, FIFO_CTRL0, - CTRL0_TODDR_TYPE_MASK | - CTRL0_TODDR_MSB_POS_MASK | - CTRL0_TODDR_LSB_POS_MASK, - CTRL0_TODDR_TYPE(type) | - CTRL0_TODDR_MSB_POS(TODDR_MSB_POS) | - CTRL0_TODDR_LSB_POS(TODDR_MSB_POS - (width - 1))); + CTRL0_TODDR_TYPE | + CTRL0_TODDR_MSB_POS | + CTRL0_TODDR_LSB_POS, + FIELD_PREP(CTRL0_TODDR_TYPE, type) | + FIELD_PREP(CTRL0_TODDR_MSB_POS, TODDR_MSB_POS) | + FIELD_PREP(CTRL0_TODDR_LSB_POS, TODDR_MSB_POS - (width - 1))); return 0; } -- 2.43.0

2 1

Audio dsp recovery using remoteproc
by noman pouigt 27 Feb '24

27 Feb '24

mailbox to dsp_1 is currently modeling platform pcm driver. mailbox to dsp_2 is also doing the same. Platform driver callbacks cause IPC to be sent to dsp's. Lifecycle of two dsp's are managed by separate remoteproc drivers. Single sound card is exposed. Separate watchdog interrupts from the corresponding dsp's are connected to remoteproc to manage crashing of the individual dsp's. How can I restart both the dsp when either of them crashes using the kernel device model? Remoteproc driver currently only restarts the crashed dsp instead of restarting both the dsp. It is needed to bring up the hardware in a consistent state as both the dsp's are connected to a common hardware. I thought of making a virtual parent remoteproc device and then managing individual dsp as a subdevice of that parent device but remoteproc device node is associated with the individual elf file i.e. it can manage only a single dsp. How can I model remoteproc drivers using linux device model so that when either of them crashes both the dsp's get reloaded by the remoteproc framework. MailBox ---- DSP_1 --- | | Linux ------ common_hw -> speaker/mic | | MailBox ---- DSP_2 ---

2 1

[PATCH 00/13] irqchip: Convert to platform remove callback returning void
by Uwe Kleine-König 27 Feb '24

27 Feb '24

this series converts all drivers below drivers/irqchip to use .remove_new(). See commit 5c5a7680e67b ("platform: Provide a remove callback that returns no value") for an extended explanation and the eventual goal. The TL;DR; is to make it harder for driver authors to leak resources. The drivers touched here are all fine though and don't return early in .remove(). So all conversions in this series are trivial. Best regards Uwe Uwe Kleine-König (13): irqchip/imgpdc: Convert to platform remove callback returning void irqchip/imx-intmux: Convert to platform remove callback returning void irqchip/imx-irqsteer: Convert to platform remove callback returning void irqchip/keystone: Convert to platform remove callback returning void irqchip/ls-scfg-msi: Convert to platform remove callback returning void irqchip/madera: Convert to platform remove callback returning void irqchip/mvebu-pic: Convert to platform remove callback returning void irqchip/pruss-intc: Convert to platform remove callback returning void irqchip/renesas-intc-irqpin: Convert to platform remove callback returning void irqchip/renesas-irqc: Convert to platform remove callback returning void irqchip/renesas-rza1: Convert to platform remove callback returning void irqchip/stm32-exti: Convert to platform remove callback returning void irqchip/ts4800: Convert to platform remove callback returning void drivers/irqchip/irq-imgpdc.c | 5 ++--- drivers/irqchip/irq-imx-intmux.c | 6 ++---- drivers/irqchip/irq-imx-irqsteer.c | 6 ++---- drivers/irqchip/irq-keystone.c | 5 ++--- drivers/irqchip/irq-ls-scfg-msi.c | 6 ++---- drivers/irqchip/irq-madera.c | 6 ++---- drivers/irqchip/irq-mvebu-pic.c | 6 ++---- drivers/irqchip/irq-pruss-intc.c | 6 ++---- drivers/irqchip/irq-renesas-intc-irqpin.c | 5 ++--- drivers/irqchip/irq-renesas-irqc.c | 5 ++--- drivers/irqchip/irq-renesas-rza1.c | 5 ++--- drivers/irqchip/irq-stm32-exti.c | 5 ++--- drivers/irqchip/irq-ts4800.c | 6 ++---- 13 files changed, 26 insertions(+), 46 deletions(-) base-commit: 39676dfe52331dba909c617f213fdb21015c8d10 -- 2.42.0

3 4

[PATCH] ASoC: cs35l56: Prevent bad sign extension in cs35l56_read_silicon_uid()
by Richard Fitzgerald 27 Feb '24

27 Feb '24

Cast u8 values to u32 when using them to build a 32-bit unsigned value that is then stored in a u64. This avoids the possibility of a bad sign extension where the u8 is implicitly extended to an int, thus changing it from an unsigned to a signed value. Whether this is a real problem is debatable, but it does no harm to ensure that the u8 are cast to a suitable type for shifting. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: e1830f66f6c6 ("ASoC: cs35l56: Add helper functions for amp calibration") --- sound/soc/codecs/cs35l56-shared.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 20b6dbd3fbab..f3670bf85a95 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -659,9 +659,10 @@ static int cs35l56_read_silicon_uid(struct cs35l56_base *cs35l56_base, u64 *uid) return ret; } - unique_id = pte.lot[2] | (pte.lot[1] << 8) | (pte.lot[0] << 16); + unique_id = (u32)pte.lot[2] | ((u32)pte.lot[1] << 8) | ((u32)pte.lot[0] << 16); unique_id <<= 32; - unique_id |= pte.x | (pte.y << 8) | (pte.wafer_id << 16) | (pte.dvs << 24); + unique_id |= (u32)pte.x | ((u32)pte.y << 8) | ((u32)pte.wafer_id << 16) | + ((u32)pte.dvs << 24); dev_dbg(cs35l56_base->dev, "UniqueID = %#llx\n", unique_id); -- 2.39.2

2 1

ALC285 reduce HP pop noise
by Kailang 27 Feb '24

27 Feb '24

Hi Takashi, Lenovo Thinkpad platform with ALC285 which had pop noise from Headphone port on reboot state. Attach patch will reduce it. BR, Kailang

2 1

[PATCH 0/9] ALSA: cs35l56: Apply calibration from EFI
by Richard Fitzgerald 27 Feb '24

27 Feb '24

Factory calibration of the speakers stores the calibration information into an EFI variable. This set of patches adds support for applying speaker calibration data from that EFI variable. Patch #1 and #2 are bugfixes that are prerequisites. The HDA patches #7 and #8 depend on the ASoC patches. Patch #9 is a KUnit test for the new cs-amp-lib library. Richard Fitzgerald (8): ASoC: wm_adsp: Fix missing mutex_lock in wm_adsp_write_ctl() ASoC: wm_adsp: Add wm_adsp_start() and wm_adsp_stop() ASoC: cs-amp-lib: Add helpers for factory calibration data ASoC: cs35l56: Add helper functions for amp calibration ASoC: cs35l56: Apply amp calibration from EFI data ALSA: hda: hda_cs_dsp_ctl: Add helper function to write calibration ALSA: hda: cs35l56: Apply amp calibration from EFI data ASoC: cs-amp-lib: Add KUnit test for calibration helpers Simon Trimmer (1): ALSA: hda: hda_cs_dsp_ctl: Only call notify when a control has been added to a card include/sound/cs-amp-lib.h | 66 +++ include/sound/cs35l56.h | 10 + sound/pci/hda/Kconfig | 2 + sound/pci/hda/cs35l56_hda.c | 36 +- sound/pci/hda/hda_cs_dsp_ctl.c | 52 ++- sound/pci/hda/hda_cs_dsp_ctl.h | 4 + sound/soc/codecs/Kconfig | 18 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs-amp-lib-test.c | 709 +++++++++++++++++++++++++++++ sound/soc/codecs/cs-amp-lib.c | 270 +++++++++++ sound/soc/codecs/cs35l56-sdw.c | 20 + sound/soc/codecs/cs35l56-shared.c | 83 ++++ sound/soc/codecs/cs35l56.c | 43 +- sound/soc/codecs/wm_adsp.c | 30 +- sound/soc/codecs/wm_adsp.h | 2 + 15 files changed, 1328 insertions(+), 21 deletions(-) create mode 100644 include/sound/cs-amp-lib.h create mode 100644 sound/soc/codecs/cs-amp-lib-test.c create mode 100644 sound/soc/codecs/cs-amp-lib.c -- 2.30.2

4 13

[PATCH v9] ASoc: tas2783: Add tas2783 codec driver
by Shenghao Ding 27 Feb '24

27 Feb '24

The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp speaker protection algorithm. The integrated speaker voltage and current sense provides for real-time monitoring of lodspeakers. The ASoC component provides the majority of the functionality of the device, all the audio functions. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v9: - simplify tasdevice_set_sdw_stream - fixed some Linux coding style - fixed the spelling mistakes - Select left/right channel based on unique id - a longer description has been added - remove unused crc8 in KCONFIG - Invert of RE here means reciprocal of RE - correct indentation - correct the wrong pow calibrated register - correct comment on RESET register - fixed usually lines don't end with an open parenthesis - fixed tasdevice_sdw_shutdown - use defines for function, entity, etc. for readable - remove unnecessary codec_lock in component_probe() - remove struct sdw_stream_data - remove unused macro definitions - use BIT(2) and BIT(1) instead of 0x2 and 0x1 - implement pm_runtime - reg_start is u8, reg_start + i only adds i byte, it should be reg_start + i * 4, because the calibration values - fixed total_sz in tas2783_calibration - add a define for TASDEVICE_REG(0, 0, 0x0a) - rework macro style in header file - Add select CRC32 - rewrite the mute and the return value handling - Add comments on bin file - Add link_id into the fw name to distinguish the same sdw id on different sdw id. - remove unnecessary mutex_lock - use set_active() in tasdevice_io_init and move all other parts in the probe. - move pm_runtime_get_noresume to the probe - fix the return value of putvol function - add put_noidle in tasdevice_sdw_remove - remove unused header file(of.h) - correct the style of the multi-line comment - Set the readable registers in tas2783_readable_register for debugfs - Use new PM macros - Use default put and get inline SOC_SINGLE_RANGE_TLV and SOC_SINGLE_SX_TLV - Use respective %pt to log the calibrated time - Remove unnecessary struct tm - Add trailing comma tasdevice_dapm_widgets and tasdevice_audio_map - Rewrite the return value handling in tasdevice_sdw_probe - use dev_num instead of id.unique_id - rework the log - Add code to check the size of calibrated data in UEFI - Add description of Layout of calibrated Data in UEFI - switch back link_id and unique_id to identify the devcie - unique_id offset as the device index of the calibrated data --- sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 851 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 106 ++++ 4 files changed, 975 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 59f9742e9ff4..6ba3fd0c0811 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -247,6 +247,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1847,6 +1848,21 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + depends on EFI + select REGMAP + select REGMAP_SOUNDWIRE + select CRC32 + select CRC32_SARWATE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + TAS2783 supports the MIPI SoundWire v1.2 compatible interface + for audio and control data. At the end of calibrated data appends + the crc data with crc32_sarwate for error detection. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f53baa2b9565..6afceb126ae3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -284,6 +284,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -664,6 +665,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..42a6f70dd108 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,851 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Kevin Lu <kevin-lu(a)ti.com> +// Shenghao Ding <shenghao-ding(a)ti.com> +// + +#include <asm/unaligned.h> +#include <linux/acpi.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/pcm_params.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include <sound/tas2781-tlv.h> +#include "tas2783.h" + +static const unsigned int tas2783_cali_reg[] = { + TAS2783_CALIBRATION_RE, /* Resistance */ + TAS2783_CALIBRATION_RE_LOW, /* Low limitation of RE */ + TAS2783_CALIBRATION_INV_RE, /* Reciprocal of RE */ + TAS2783_CALIBRATION_POW, /* RMS Power */ + TAS2783_CALIBRATION_TLIMIT /* Temperature limitation */ +}; + +static const struct reg_default tas2783_reg_defaults[] = { + /* Default values for ROM mode. Activated. */ + { 0x8002, 0x1a }, /* AMP inactive. */ + { 0x8097, 0xc8 }, + { 0x80b5, 0x74 }, + { 0x8099, 0x20 }, + { 0xfe8d, 0x0d }, + { 0xfebe, 0x4a }, + { 0x8230, 0x00 }, + { 0x8231, 0x00 }, + { 0x8232, 0x00 }, + { 0x8233, 0x01 }, + { 0x8418, 0x00 }, /* Set volume to 0 dB. */ + { 0x8419, 0x00 }, + { 0x841a, 0x00 }, + { 0x841b, 0x00 }, + { 0x8428, 0x40 }, /* Unmute channel */ + { 0x8429, 0x00 }, + { 0x842a, 0x00 }, + { 0x842b, 0x00 }, + { 0x8548, 0x00 }, /* Set volume to 0 dB. */ + { 0x8549, 0x00 }, + { 0x854a, 0x00 }, + { 0x854b, 0x00 }, + { 0x8558, 0x40 }, /* Unmute channel */ + { 0x8559, 0x00 }, + { 0x855a, 0x00 }, + { 0x855b, 0x00 }, + { 0x800a, 0x3a }, /* Enable both channel */ + { 0x800e, 0x44 }, + { 0x800f, 0x40 }, + { 0x805c, 0x99 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_VOLUME, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0x807F: /* Page 0 */ + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + /* + * Only software reset register was volatiled. + * Bit is self clearing. + * 0b = Don't reset + * 1b = reset + */ + case 0x8001: + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x44ffffff, + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, dvc_tlv), + SOC_SINGLE_SX_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, amp_vol_tlv), +}; + +static void tas2783_apply_calib(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + struct regmap *map = tas_dev->regmap; + u8 *cali_start; + u16 offset; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + offset = tas_dev->sdw_peripheral->id.unique_id - + TAS2783_UNIQUE_ID_BASE; + if (offset >= TAS2783_MAX_DEV_NUM) { + dev_err(tas_dev->dev, "%s: offset(%u) is out of range.\n", + __func__, offset); + return; + } + + /* + * The area saving tas2783 calibrated data is specified by its + * unique_id offset. cali_start is the first address of current + * tas2783's calibrated data. + */ + cali_start = (u8 *)(cali_data + offset * sizeof(tas2783_cali_reg)); + for (int i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + ret = regmap_bulk_write(map, tas2783_cali_reg[i], + &cali_start[4 * i], 4); + if (ret) { + dev_err(tas_dev->dev, "Cali failed %x:%d\n", + tas2783_cali_reg[i], ret); + break; + } + } +} + +/* + * Load the calibration data, including speaker impedance, f0, etc. + * Calibration is done by the manufacturer in the factory. The calibration + * data are used by the algorithm for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * The DSP will work with default data values if calibrated data are + * missing or are invalid. + * Layout of calibrated Data in UEFI: + * Calibrated Data of Dev 0(unique_id offset 0) (20 bytes) + * Calibrated Data of Dev 1(unique_id offset 1) (20 bytes) + * Calibrated Data of Dev 2(unique_id offset 2) (20 bytes) + * Calibrated Data of Dev 3(unique_id offset 3) (20 bytes) + * Calibrated Data of Dev 4(unique_id offset 4) (20 bytes) + * Calibrated Data of Dev 5(unique_id offset 5) (20 bytes) + * Calibrated Data of Dev 6(unique_id offset 6) (20 bytes) + * Calibrated Data of Dev 7(unique_id offset 7) (20 bytes) + * TimeStamp of Calibration (4 bytes) + * CRC (4 bytes) + * Reserved (88 bytes) + */ +static void tas2783_calibration(struct tasdevice_priv *tas_dev) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct calibration_data cali_data; + unsigned int *tmp_val; + unsigned int crc; + efi_status_t status; + + cali_data.total_sz = 0; + + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, NULL); + if (status == EFI_BUFFER_TOO_SMALL + && cali_data.total_sz < TAS2783_MAX_CALIDATA_SIZE) { + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, + cali_data.data); + dev_dbg(tas_dev->dev, "%s: cali get %lx bytes result:%ld\n", + __func__, cali_data.total_sz, status); + } + if (status != 0) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return; + } + + tmp_val = (unsigned int *)cali_data.data; + + /* 8 devs * 20 bytes calibrated data/dev + 4 bytes Timestamp */ + crc = crc32(~0, cali_data.data, 164) ^ ~0; + + if (crc == tmp_val[41]) { + /* Date and time of when calibration was done. */ + tas2783_apply_calib(tas_dev, tmp_val); + dev_dbg(tas_dev->dev, "%ptTs", &tmp_val[40]); + } else { + dev_dbg(tas_dev->dev, "CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[41]); + } +} + +static void tasdevice_dspfw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct tas2783_firmware_node *p; + struct regmap *map = tas_dev->regmap; + unsigned char *buf = NULL; + int offset = 0, img_sz; + int ret; + + if (!fmw || !fmw->data) { + dev_warn(tas_dev->dev, + "%s: failed to read %s: work in bypass-dsp mode.\n", + __func__, tas_dev->dspfw_binaryname); + return; + } + buf = (unsigned char *)fmw->data; + + img_sz = get_unaligned_le32(&buf[offset]); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_warn(tas_dev->dev, "%s: size not matching, %d %u.", + __func__, (int)fmw->size, img_sz); + return; + } + + while (offset < img_sz) { + p = (struct tas2783_firmware_node *)(buf + offset); + if (p->length > 1) { + ret = regmap_bulk_write(map, p->download_addr, + buf + offset + sizeof(unsigned int) * 5, p->length); + } else { + ret = regmap_write(map, p->download_addr, + *(buf + offset + sizeof(unsigned int) * 5)); + } + + if (ret != 0) { + dev_dbg(tas_dev->dev, + "%s: load FW fail: %d, work in bypass.\n", + __func__, ret); + return; + } + offset += sizeof(unsigned int) * 5 + p->length; + } + + tas2783_calibration(tas_dev); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC"), +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"}, +}; + +static int tasdevice_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + int ret; + + dev_dbg(dai->dev, "%s: dai_name %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(tas_priv->sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) { + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); + return ret; + } + + dev_dbg(dai->dev, "%s: format: %i rate: %i\n", __func__, + params_format(params), params_rate(params)); + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tas_priv->sdw_peripheral, sdw_stream); + + return 0; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct regmap *map = tas_dev->regmap; + int ret; + + dev_dbg(tas_dev->dev, "%s: %d.\n", __func__, mute); + + if (mute) { + if (direction == SNDRV_PCM_STREAM_CAPTURE) { + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_INACTIVE); + if (ret) + dev_err(tas_dev->dev, + "%s: Disable AEF failed.\n", __func__); + } else { + /* FU23 mute (0x40400108) */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 1); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 mute failed.\n", __func__); + goto out; + } + /* + * Both playback and echo data will be shutdown in + * playback stream. + */ + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK | + TAS2783_REG_AEF_MASK, + TAS2783_REG_PWR_MODE_ACTIVE | + TAS2783_REG_PWR_MODE_SW_PWD); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR&AEF shutdown failed.\n", + __func__); + goto out; + } + tas_dev->pstream = false; + } + } else { + /* FU23 Unmute, 0x40400108. */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 Unmute failed.\n", __func__); + goto out; + } + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK, + TAS2783_REG_PWR_MODE_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR Unmute failed.\n", __func__); + goto out; + } + tas_dev->pstream = true; + } else { + /* Capture stream is the echo ref data for voice. + * Without playback, it can't be active. + */ + if (tas_dev->pstream == true) { + ret = regmap_update_bits(map, + TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: AEF enable failed.\n", + __func__); + goto out; + } + } else { + dev_err(tas_dev->dev, + "%s: No playback, no AEF!", __func__); + ret = -EINVAL; + } + } + } +out: + if (ret) + dev_err(tas_dev->dev, "Mute or unmute %d failed %d.\n", + mute, ret); + + return ret; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_comp_probe(struct snd_soc_component *comp) +{ + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(comp); + acpi_handle handle = ACPI_HANDLE(tas_dev->dev); + const struct firmware *fw_entry = NULL; + const char *sub = NULL; + int ret; + + if (handle) { + sub = acpi_get_subsystem_id(handle); + if (IS_ERR(sub)) + sub = NULL; + } + + tas_dev->component = comp; + + /* + * Each tas2783 in the system has its own dspfw. + */ + if (comp->name_prefix) { + /* + * name_prefix.bin stores the dsp firmware including speaker + * protection algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied into + * firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, "%s-tas2783.bin", + comp->name_prefix); + } else { + /* Compatible with the previous naming rule */ + if (sub) { + /* + * subsys_id-link_id[0,1,...,N]-unique_id[1,...,4].bin + * stores the dsp firmware including speaker protection + * algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied + * into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "%s-%d-%d.bin", sub, + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } else { + /* + * tas2783-link_id[0,1,...,N]-unique_id[1,...,4].bin + * stores the dsp firmware including speaker protection + * algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied + * into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "tas2783-%d-%d.bin", + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } + } + + ret = request_firmware(&fw_entry, tas_dev->dspfw_binaryname, + tas_dev->dev); + if (ret) { + dev_err(tas_dev->dev, + "%s: request_firmware %x open status: %d.\n", __func__, + tas_dev->sdw_peripheral->id.unique_id, ret); + goto out; + } + + tasdevice_dspfw_ready(fw_entry, tas_dev); + +out: + if (fw_entry) + release_firmware(fw_entry); + return 0; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_comp_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + return ret; + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + pm_runtime_enable(tas_dev->dev); + + return 0; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + struct sdw_dpn_prop *dpn; + unsigned long addr; + int nval, i, j; + u32 bit; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = BIT(2); /* BITMAP: 00000100 */ + prop->sink_ports = BIT(1); /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static int tasdevice_io_init(struct device *dev, + struct sdw_slave *slave) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(tas_priv->regmap, false); + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + * Update count of parent 'active' children. + */ + pm_runtime_set_active(&slave->dev); + } + + /* sw reset */ + ret = regmap_write(tas_priv->regmap, TAS2873_REG_SWRESET, + TAS2873_REG_SWRESET_RESET); + if (ret) { + dev_err(tas_priv->dev, "Reset failed.\n"); + goto out; + } + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, false); + regcache_mark_dirty(tas_priv->regmap); + } + + tas_priv->first_hw_init = true; + tas_priv->hw_init = true; + +out: + return ret; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tas_priv->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tas_priv->hw_init = false; + /* + * Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_priv->hw_init || tas_priv->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return -ENOMEM; + + tas_dev->dev = dev; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_peripheral = peripheral; + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) + return dev_err_probe(dev, PTR_ERR(tas_dev->regmap), + "Failed devm_regmap_init_sdw."); + + return tasdevice_init(tas_dev); +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(&peripheral->dev); + + if (tas_dev->first_hw_init) + pm_runtime_disable(tas_dev->dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {} +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static int __maybe_unused tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (!tas_priv->hw_init) + return 0; + + regcache_cache_only(tas_priv->regmap, true); + + return 0; +} + +#define TAS2783_PROBE_TIMEOUT_MS 5000 + +static int __maybe_unused tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long time; + + if (!tas_priv->first_hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT_MS)); + if (!time) { + dev_err(&slave->dev, "Init not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(tas_priv->regmap, false); + regcache_sync(tas_priv->regmap); + + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume) + RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume, NULL) +}; + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = &tas2783_sdca_pm, + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..631776dcec4c --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,106 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2023 - 2024 Texas Instruments Incorporated + * https://www.ti.com + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + * Shenghao Ding <shenghao-ding(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +/* Unique id start */ +#define TAS2783_UNIQUE_ID_BASE 0x08 +#define TAS2783_MAX_DEV_NUM 8 + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 + \ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0x00, 0x01) +#define TAS2873_REG_SWRESET_MASK BIT(0) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +#define TAS2873_REG_PWR_CTRL TASDEVICE_REG(0x0, 0x00, 0x02) +#define TAS2783_REG_AEF_MASK GENMASK(4, 3) +#define TAS2783_REG_AEF_ACTIVE 0 +#define TAS2783_REG_AEF_INACTIVE GENMASK(4, 3) +#define TAS2783_REG_PWR_MODE_MASK GENMASK(2, 0) +#define TAS2783_REG_PWR_MODE_ACTIVE 0 +#define TAS2783_REG_PWR_MODE_SW_PWD BIT(1) + +#define TAS2783_REG_TDM_RX_CFG TASDEVICE_REG(0x0, 0x00, 0x0a) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1a) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x13, 0x70) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +/* TAS2783 SDCA Control - function number */ +#define TAS2783_FUNC_TYPE_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_FU_VOLUME 0x02 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 +#define TAS2783_DSPFW_FILENAME_LEN 64 + +#define TAS2783_MAX_CALIDATA_SIZE 256 + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct snd_soc_component *component; + struct sdw_slave *sdw_peripheral; + enum sdw_slave_status status; + struct sdw_bus_params params; + struct regmap *regmap; + struct device *dev; + unsigned char dspfw_binaryname[TAS2783_DSPFW_FILENAME_LEN]; + unsigned char dev_name[32]; + unsigned int chip_id; + bool pstream; + bool hw_init; + bool first_hw_init; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

4 5

[PATCH 0/6] ASoC: meson: axg fixes and clean-up
by Jerome Brunet 26 Feb '24

26 Feb '24

This are various fixes and clean up gathered while working on Amlogic audio support. These help better handle higher and unusual clock configuration for TDM, SPDIF or PDM. Jerome Brunet (6): ASoC: meson: axg-tdm-interface: fix mclk setup without mclk-fs ASoC: meson: axg-tdm-interface: add frame rate constraint ASoC: meson: axg-tdm-interface: update error format error traces ASoC: meson: axg-spdifin: use max width for rate detection ASoC: meson: axg-fifo: take continuous rates ASoC: meson: axg-fifo: use FIELD helpers sound/soc/meson/axg-fifo.c | 24 +++++++++++----------- sound/soc/meson/axg-fifo.h | 14 +++++-------- sound/soc/meson/axg-frddr.c | 12 +++++++---- sound/soc/meson/axg-spdifin.c | 6 +++--- sound/soc/meson/axg-tdm-interface.c | 31 +++++++++++++++++++---------- sound/soc/meson/axg-toddr.c | 29 ++++++++++++++------------- 6 files changed, 64 insertions(+), 52 deletions(-) -- 2.43.0

2 10

[PATCH 0/6] ASoC: codecs: Simplify mclk initialization
by Cezary Rojewski 26 Feb '24

26 Feb '24

The patchset may not cover all codecs found in the codecs/ directory - noticed a possible improvement and grepped for similar pattern across C files found in the directory. Those addressed here seem pretty straightforward. Most of clk_xxx() functions do check if provided clk-pointer is non-NULL. These do not check if the pointer is an error-pointer. Providing such to a clk_xxx() results in a panic. By utilizing _optional() variant of devm_clk_get() the driver code is both simplified and more robust. There is no need to remember about IS_ERR(clk) checks each time mclk is accessed. Cezary Rojewski (6): ASoC: codecs: da7213: Simplify mclk initialization ASoC: codecs: nau8825: Simplify mclk initialization ASoC: codecs: rt5514: Simplify mclk initialization ASoC: codecs: rt5616: Simplify mclk initialization ASoC: codecs: rt5640: Simplify mclk initialization ASoC: codecs: rt5660: Simplify mclk initialization sound/soc/codecs/da7213.c | 12 ++++-------- sound/soc/codecs/nau8825.c | 12 ++++-------- sound/soc/codecs/rt5514.c | 9 +++------ sound/soc/codecs/rt5616.c | 9 +++------ sound/soc/codecs/rt5640.c | 9 +++------ sound/soc/codecs/rt5660.c | 9 +++------ 6 files changed, 20 insertions(+), 40 deletions(-) -- 2.25.1

2 7

[PATCH v4 0/2] ASoC: codecs: tx-macro: correct TX SMIC MUXn widgets on SM8350+
by Krzysztof Kozlowski 26 Feb '24

26 Feb '24

Hi, Changelog in individual patches. v2: https://lore.kernel.org/alsa-devel/20240125153110.410295-1-krzysztof.kozlow… v1: https://lore.kernel.org/all/20230717140138.201745-1-krzysztof.kozlowski@lin… Best regards, Krzysztof Krzysztof Kozlowski (2): ASoC: codecs: tx-macro: split widgets per different LPASS versions ASoC: codecs: tx-macro: correct TX SMIC MUXn widgets on SM8350+ sound/soc/codecs/lpass-macro-common.h | 7 + sound/soc/codecs/lpass-tx-macro.c | 679 +++++++++++++++++++------- 2 files changed, 514 insertions(+), 172 deletions(-) -- 2.34.1

2 4

[PATCH v3 0/2] ASoC: codecs: tx-macro: correct TX SMIC MUXn widgets on SM8350+
by Krzysztof Kozlowski 26 Feb '24

26 Feb '24

Hi, Changelog in individual patches. v2: https://lore.kernel.org/alsa-devel/20240125153110.410295-1-krzysztof.kozlow… v1: https://lore.kernel.org/all/20230717140138.201745-1-krzysztof.kozlowski@lin… Best regards, Krzysztof Krzysztof Kozlowski (2): ASoC: codecs: tx-macro: split widgets per different LPASS versions ASoC: codecs: tx-macro: correct TX SMIC MUXn widgets on SM8350+ sound/soc/codecs/lpass-macro-common.h | 7 + sound/soc/codecs/lpass-tx-macro.c | 671 +++++++++++++++++++------- 2 files changed, 506 insertions(+), 172 deletions(-) -- 2.34.1

2 4

[PATCH v2 0/4] ALSA/ASoC: Conditionally skip i915 init and cleanups
by Cezary Rojewski 26 Feb '24

26 Feb '24

A small set of changes to improve initialization of the audio stack on HDAudio devices and pair of cleanups. As the first change is the most important one here, following is the technical background for it: Commit 78f613ba1efb ("drm/i915: finish removal of CNL") and its friends removed support for i915 for all CNL-based platforms. HDAudio library, however, still treats such platforms as valid candidates for i915 binding. Update query mechanism to reflect changes made in drm tree. At the same time, i915 support for LKF-based platforms has not been provided so remove them from valid binding candidates. The snd_soc_hda change is a follow up for the above and the cleanup patches do not bring any functional changes. Changes in v2: - list of problematic VGA devices is now declared locally, no more touching drm stuff Cezary Rojewski (4): ALSA: hda: Skip i915 initialization on CNL/LKF-based platforms ASoC: codecs: hda: Skip HDMI/DP registration if i915 is missing ASoC: codecs: hda: Cleanup error messages ALSA: hda: Reuse for_each_pcm_streams() sound/hda/hdac_i915.c | 32 +++++++++++++++++++++++++++++--- sound/pci/hda/hda_codec.c | 2 +- sound/soc/codecs/hda.c | 15 ++++++++++----- 3 files changed, 40 insertions(+), 9 deletions(-) -- 2.25.1

4 12

[tiwai-sound:test/guard 50/70] sound/core/rawmidi.c:1012:6: error: cannot jump from this asm goto statement to one of its possible targets
by kernel test robot 25 Feb '24

25 Feb '24

Hi Takashi, FYI, the error/warning was bisected to this commit, please ignore it if it's irrelevant. tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/guard head: 0ba3f403c3b7fdd2683aac6285903f92ba81c32d commit: 369c285f8405ec663691db39ec59a01c70f80ded [50/70] ALSA: rawmidi: Use guard() for locking config: powerpc-randconfig-r081-20240225 (https://download.01.org/0day-ci/archive/20240225/202402251935.CGUH0Id9-lkp@…) compiler: clang version 15.0.7 (https://github.com/llvm/llvm-project.git 8dfdcc7b7bf66834a761bd8de445840ef68e4d1a) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240225/202402251935.CGUH0Id9-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202402251935.CGUH0Id9-lkp@intel.com/ All errors (new ones prefixed by >>): >> sound/core/rawmidi.c:1012:6: error: cannot jump from this asm goto statement to one of its possible targets if (put_user(device, argp)) ^ arch/powerpc/include/asm/uaccess.h:66:5: note: expanded from macro 'put_user' __put_user(x, _pu_addr) : -EFAULT; \ ^ arch/powerpc/include/asm/uaccess.h:48:3: note: expanded from macro '__put_user' __put_user_size_goto(__pu_val, __pu_addr, __pu_size, __pu_failed); \ ^ arch/powerpc/include/asm/uaccess.h:116:10: note: expanded from macro '__put_user_size_goto' case 1: __put_user_asm_goto(x, __pus_addr, label, "stb"); break; \ ^ arch/powerpc/include/asm/uaccess.h:86:2: note: expanded from macro '__put_user_asm_goto' asm_volatile_goto( \ ^ include/linux/compiler_types.h:366:33: note: expanded from macro 'asm_volatile_goto' #define asm_volatile_goto(x...) asm goto(x) ^ sound/core/rawmidi.c:996:6: note: possible target of asm goto statement if (get_user(device, argp)) ^ arch/powerpc/include/asm/uaccess.h:295:5: note: expanded from macro 'get_user' __get_user(x, _gu_addr) : \ ^ arch/powerpc/include/asm/uaccess.h:283:2: note: expanded from macro '__get_user' __get_user_size_allowed(__gu_val, __gu_addr, __gu_size, __gu_err); \ ^ arch/powerpc/include/asm/uaccess.h:201:16: note: expanded from macro '__get_user_size_allowed' break; \ ^ sound/core/rawmidi.c:1000:2: note: jump exits scope of variable with __attribute__((cleanup)) guard(mutex)(&register_mutex); ^ include/linux/cleanup.h:164:15: note: expanded from macro 'guard' CLASS(_name, __UNIQUE_ID(guard)) ^ include/linux/compiler.h:180:29: note: expanded from macro '__UNIQUE_ID' #define __UNIQUE_ID(prefix) __PASTE(__PASTE(__UNIQUE_ID_, prefix), __COUNTER__) ^ include/linux/compiler_types.h:84:22: note: expanded from macro '__PASTE' #define __PASTE(a,b) ___PASTE(a,b) ^ include/linux/compiler_types.h:83:23: note: expanded from macro '___PASTE' #define ___PASTE(a,b) a##b ^ <scratch space>:5:1: note: expanded from here __UNIQUE_ID_guard309 ^ sound/core/rawmidi.c:996:6: error: cannot jump from this asm goto statement to one of its possible targets if (get_user(device, argp)) ^ arch/powerpc/include/asm/uaccess.h:295:5: note: expanded from macro 'get_user' __get_user(x, _gu_addr) : \ ^ arch/powerpc/include/asm/uaccess.h:283:2: note: expanded from macro '__get_user' __get_user_size_allowed(__gu_val, __gu_addr, __gu_size, __gu_err); \ ^ arch/powerpc/include/asm/uaccess.h:199:3: note: expanded from macro '__get_user_size_allowed' __get_user_size_goto(x, ptr, size, __gus_failed); \ ^ arch/powerpc/include/asm/uaccess.h:187:10: note: expanded from macro '__get_user_size_goto' case 1: __get_user_asm_goto(x, (u8 __user *)ptr, label, "lbz"); break; \ ^ arch/powerpc/include/asm/uaccess.h:158:2: note: expanded from macro '__get_user_asm_goto' asm_volatile_goto( \ ^ include/linux/compiler_types.h:366:33: note: expanded from macro 'asm_volatile_goto' #define asm_volatile_goto(x...) asm goto(x) ^ sound/core/rawmidi.c:1012:6: note: possible target of asm goto statement if (put_user(device, argp)) ^ arch/powerpc/include/asm/uaccess.h:66:5: note: expanded from macro 'put_user' __put_user(x, _pu_addr) : -EFAULT; \ ^ arch/powerpc/include/asm/uaccess.h:52:9: note: expanded from macro '__put_user' \ ^ sound/core/rawmidi.c:1000:2: note: jump bypasses initialization of variable with __attribute__((cleanup)) guard(mutex)(&register_mutex); ^ include/linux/cleanup.h:164:15: note: expanded from macro 'guard' CLASS(_name, __UNIQUE_ID(guard)) ^ include/linux/compiler.h:180:29: note: expanded from macro '__UNIQUE_ID' #define __UNIQUE_ID(prefix) __PASTE(__PASTE(__UNIQUE_ID_, prefix), __COUNTER__) ^ include/linux/compiler_types.h:84:22: note: expanded from macro '__PASTE' #define __PASTE(a,b) ___PASTE(a,b) ^ include/linux/compiler_types.h:83:23: note: expanded from macro '___PASTE' #define ___PASTE(a,b) a##b ^ <scratch space>:5:1: note: expanded from here __UNIQUE_ID_guard309 ^ 2 errors generated. Kconfig warnings: (for reference only) WARNING: unmet direct dependencies detected for SND_SOC_MPC5200_AC97 Depends on [n]: SOUND [=m] && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_MPC52xx [=y] && PPC_BESTCOMM [=n] Selected by [m]: - SND_MPC52xx_SOC_EFIKA [=m] && SOUND [=m] && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_EFIKA [=y] WARNING: unmet direct dependencies detected for ADB_CUDA Depends on [n]: MACINTOSH_DRIVERS [=n] && (ADB [=n] || PPC_PMAC [=y]) && !PPC_PMAC64 [=n] Selected by [y]: - PPC_PMAC [=y] && PPC_BOOK3S [=y] && CPU_BIG_ENDIAN [=y] && POWER_RESET [=y] && PPC32 [=y] vim +1012 sound/core/rawmidi.c ^1da177e4c3f41 Linus Torvalds 2005-04-16 986 127ae6f6dad2ed Takashi Iwai 2023-05-23 987 /* ioctl to find the next device; either legacy or UMP depending on @find_ump */ 127ae6f6dad2ed Takashi Iwai 2023-05-23 988 static int snd_rawmidi_next_device(struct snd_card *card, int __user *argp, 127ae6f6dad2ed Takashi Iwai 2023-05-23 989 bool find_ump) ^1da177e4c3f41 Linus Torvalds 2005-04-16 990 ^1da177e4c3f41 Linus Torvalds 2005-04-16 991 { 127ae6f6dad2ed Takashi Iwai 2023-05-23 992 struct snd_rawmidi *rmidi; ^1da177e4c3f41 Linus Torvalds 2005-04-16 993 int device; 127ae6f6dad2ed Takashi Iwai 2023-05-23 994 bool is_ump; ^1da177e4c3f41 Linus Torvalds 2005-04-16 995 127ae6f6dad2ed Takashi Iwai 2023-05-23 996 if (get_user(device, argp)) ^1da177e4c3f41 Linus Torvalds 2005-04-16 997 return -EFAULT; a7a13d0676335a Dan Carpenter 2010-09-09 998 if (device >= SNDRV_RAWMIDI_DEVICES) /* next device is -1 */ a7a13d0676335a Dan Carpenter 2010-09-09 999 device = SNDRV_RAWMIDI_DEVICES - 1; 369c285f8405ec Takashi Iwai 2024-02-21 1000 guard(mutex)(&register_mutex); ^1da177e4c3f41 Linus Torvalds 2005-04-16 1001 device = device < 0 ? 0 : device + 1; 127ae6f6dad2ed Takashi Iwai 2023-05-23 1002 for (; device < SNDRV_RAWMIDI_DEVICES; device++) { 127ae6f6dad2ed Takashi Iwai 2023-05-23 1003 rmidi = snd_rawmidi_search(card, device); 127ae6f6dad2ed Takashi Iwai 2023-05-23 1004 if (!rmidi) 127ae6f6dad2ed Takashi Iwai 2023-05-23 1005 continue; 127ae6f6dad2ed Takashi Iwai 2023-05-23 1006 is_ump = rawmidi_is_ump(rmidi); 127ae6f6dad2ed Takashi Iwai 2023-05-23 1007 if (find_ump == is_ump) ^1da177e4c3f41 Linus Torvalds 2005-04-16 1008 break; ^1da177e4c3f41 Linus Torvalds 2005-04-16 1009 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 1010 if (device == SNDRV_RAWMIDI_DEVICES) ^1da177e4c3f41 Linus Torvalds 2005-04-16 1011 device = -1; 127ae6f6dad2ed Takashi Iwai 2023-05-23 @1012 if (put_user(device, argp)) ^1da177e4c3f41 Linus Torvalds 2005-04-16 1013 return -EFAULT; ^1da177e4c3f41 Linus Torvalds 2005-04-16 1014 return 0; ^1da177e4c3f41 Linus Torvalds 2005-04-16 1015 } 127ae6f6dad2ed Takashi Iwai 2023-05-23 1016 :::::: The code at line 1012 was first introduced by commit :::::: 127ae6f6dad2edb2201e27b7e6fa72994b537fad ALSA: rawmidi: Skip UMP devices at SNDRV_CTL_IOCTL_RAWMIDI_NEXT_DEVICE :::::: TO: Takashi Iwai <tiwai(a)suse.de> :::::: CC: Takashi Iwai <tiwai(a)suse.de> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

Attempt to Fix ALC285 on Linux for HP Envy x360 Laptops
by Athaariq Ardhiansyah 25 Feb '24

25 Feb '24

Dear Mr. Yang and ALSA contributors, Currently, I'm trying to fix ALC285 for HP Envy x360 13-ar0107au convertible laptop on Linux v6.7.6 kernel. It has several problems including B&O soundbar speakers won't work and volume too low even though it's 100%. For now, the B&O soundbar is working. However, I'm still struggling to fix the low volume problem. The draft patch is attached to this mail. If you don't mind, please let me know the information about the correct coefficients, so I can help you fix the problem and test it on my device. Thank you for your attention. Sincerely, Eric

1 0

[tiwai-sound:test/guard 46/70] sound/core/hwdep.c:291:8: error: cannot jump from this asm goto statement to one of its possible targets
by kernel test robot 25 Feb '24

25 Feb '24

Hi Takashi, FYI, the error/warning was bisected to this commit, please ignore it if it's irrelevant. tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/guard head: 08e50f26c5431175dc7207664c292f3ecc134121 commit: 9c86a0c433a5a99e2d5cc4713609efed801632a1 [46/70] ALSA: hwdep: Use guard() for locking config: powerpc-randconfig-r081-20240225 (https://download.01.org/0day-ci/archive/20240225/202402251039.08XAcTR2-lkp@…) compiler: clang version 15.0.7 (https://github.com/llvm/llvm-project.git 8dfdcc7b7bf66834a761bd8de445840ef68e4d1a) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240225/202402251039.08XAcTR2-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202402251039.08XAcTR2-lkp@intel.com/ All errors (new ones prefixed by >>): >> sound/core/hwdep.c:291:8: error: cannot jump from this asm goto statement to one of its possible targets if (put_user(device, (int __user *)arg)) ^ arch/powerpc/include/asm/uaccess.h:66:5: note: expanded from macro 'put_user' __put_user(x, _pu_addr) : -EFAULT; \ ^ arch/powerpc/include/asm/uaccess.h:48:3: note: expanded from macro '__put_user' __put_user_size_goto(__pu_val, __pu_addr, __pu_size, __pu_failed); \ ^ arch/powerpc/include/asm/uaccess.h:116:10: note: expanded from macro '__put_user_size_goto' case 1: __put_user_asm_goto(x, __pus_addr, label, "stb"); break; \ ^ arch/powerpc/include/asm/uaccess.h:86:2: note: expanded from macro '__put_user_asm_goto' asm_volatile_goto( \ ^ include/linux/compiler_types.h:366:33: note: expanded from macro 'asm_volatile_goto' #define asm_volatile_goto(x...) asm goto(x) ^ sound/core/hwdep.c:273:8: note: possible target of asm goto statement if (get_user(device, (int __user *)arg)) ^ arch/powerpc/include/asm/uaccess.h:295:5: note: expanded from macro 'get_user' __get_user(x, _gu_addr) : \ ^ arch/powerpc/include/asm/uaccess.h:283:2: note: expanded from macro '__get_user' __get_user_size_allowed(__gu_val, __gu_addr, __gu_size, __gu_err); \ ^ arch/powerpc/include/asm/uaccess.h:201:16: note: expanded from macro '__get_user_size_allowed' break; \ ^ sound/core/hwdep.c:276:4: note: jump exits scope of variable with __attribute__((cleanup)) guard(mutex)(&register_mutex); ^ include/linux/cleanup.h:164:15: note: expanded from macro 'guard' CLASS(_name, __UNIQUE_ID(guard)) ^ include/linux/compiler.h:180:29: note: expanded from macro '__UNIQUE_ID' #define __UNIQUE_ID(prefix) __PASTE(__PASTE(__UNIQUE_ID_, prefix), __COUNTER__) ^ include/linux/compiler_types.h:84:22: note: expanded from macro '__PASTE' #define __PASTE(a,b) ___PASTE(a,b) ^ include/linux/compiler_types.h:83:23: note: expanded from macro '___PASTE' #define ___PASTE(a,b) a##b ^ <scratch space>:163:1: note: expanded from here __UNIQUE_ID_guard260 ^ sound/core/hwdep.c:273:8: error: cannot jump from this asm goto statement to one of its possible targets if (get_user(device, (int __user *)arg)) ^ arch/powerpc/include/asm/uaccess.h:295:5: note: expanded from macro 'get_user' __get_user(x, _gu_addr) : \ ^ arch/powerpc/include/asm/uaccess.h:283:2: note: expanded from macro '__get_user' __get_user_size_allowed(__gu_val, __gu_addr, __gu_size, __gu_err); \ ^ arch/powerpc/include/asm/uaccess.h:199:3: note: expanded from macro '__get_user_size_allowed' __get_user_size_goto(x, ptr, size, __gus_failed); \ ^ arch/powerpc/include/asm/uaccess.h:187:10: note: expanded from macro '__get_user_size_goto' case 1: __get_user_asm_goto(x, (u8 __user *)ptr, label, "lbz"); break; \ ^ arch/powerpc/include/asm/uaccess.h:158:2: note: expanded from macro '__get_user_asm_goto' asm_volatile_goto( \ ^ include/linux/compiler_types.h:366:33: note: expanded from macro 'asm_volatile_goto' #define asm_volatile_goto(x...) asm goto(x) ^ sound/core/hwdep.c:291:8: note: possible target of asm goto statement if (put_user(device, (int __user *)arg)) ^ arch/powerpc/include/asm/uaccess.h:66:5: note: expanded from macro 'put_user' __put_user(x, _pu_addr) : -EFAULT; \ ^ arch/powerpc/include/asm/uaccess.h:52:9: note: expanded from macro '__put_user' \ ^ sound/core/hwdep.c:276:4: note: jump bypasses initialization of variable with __attribute__((cleanup)) guard(mutex)(&register_mutex); ^ include/linux/cleanup.h:164:15: note: expanded from macro 'guard' CLASS(_name, __UNIQUE_ID(guard)) ^ include/linux/compiler.h:180:29: note: expanded from macro '__UNIQUE_ID' #define __UNIQUE_ID(prefix) __PASTE(__PASTE(__UNIQUE_ID_, prefix), __COUNTER__) ^ include/linux/compiler_types.h:84:22: note: expanded from macro '__PASTE' #define __PASTE(a,b) ___PASTE(a,b) ^ include/linux/compiler_types.h:83:23: note: expanded from macro '___PASTE' #define ___PASTE(a,b) a##b ^ <scratch space>:163:1: note: expanded from here __UNIQUE_ID_guard260 ^ 2 errors generated. Kconfig warnings: (for reference only) WARNING: unmet direct dependencies detected for SND_SOC_MPC5200_AC97 Depends on [n]: SOUND [=m] && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_MPC52xx [=y] && PPC_BESTCOMM [=n] Selected by [m]: - SND_MPC52xx_SOC_EFIKA [=m] && SOUND [=m] && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_EFIKA [=y] WARNING: unmet direct dependencies detected for ADB_CUDA Depends on [n]: MACINTOSH_DRIVERS [=n] && (ADB [=n] || PPC_PMAC [=y]) && !PPC_PMAC64 [=n] Selected by [y]: - PPC_PMAC [=y] && PPC_BOOK3S [=y] && CPU_BIG_ENDIAN [=y] && POWER_RESET [=y] && PPC32 [=y] vim +291 sound/core/hwdep.c ^1da177e4c3f41 Linus Torvalds 2005-04-16 263 d9a98de218ce18 Takashi Iwai 2005-11-17 264 static int snd_hwdep_control_ioctl(struct snd_card *card, d9a98de218ce18 Takashi Iwai 2005-11-17 265 struct snd_ctl_file * control, ^1da177e4c3f41 Linus Torvalds 2005-04-16 266 unsigned int cmd, unsigned long arg) ^1da177e4c3f41 Linus Torvalds 2005-04-16 267 { ^1da177e4c3f41 Linus Torvalds 2005-04-16 268 switch (cmd) { ^1da177e4c3f41 Linus Torvalds 2005-04-16 269 case SNDRV_CTL_IOCTL_HWDEP_NEXT_DEVICE: ^1da177e4c3f41 Linus Torvalds 2005-04-16 270 { ^1da177e4c3f41 Linus Torvalds 2005-04-16 271 int device; ^1da177e4c3f41 Linus Torvalds 2005-04-16 272 ^1da177e4c3f41 Linus Torvalds 2005-04-16 273 if (get_user(device, (int __user *)arg)) ^1da177e4c3f41 Linus Torvalds 2005-04-16 274 return -EFAULT; f7b2bb8549e352 Dan Carpenter 2011-10-28 275 9c86a0c433a5a9 Takashi Iwai 2024-02-21 276 guard(mutex)(&register_mutex); f7b2bb8549e352 Dan Carpenter 2011-10-28 277 if (device < 0) f7b2bb8549e352 Dan Carpenter 2011-10-28 278 device = 0; f7b2bb8549e352 Dan Carpenter 2011-10-28 279 else if (device < SNDRV_MINOR_HWDEPS) f7b2bb8549e352 Dan Carpenter 2011-10-28 280 device++; f7b2bb8549e352 Dan Carpenter 2011-10-28 281 else f7b2bb8549e352 Dan Carpenter 2011-10-28 282 device = SNDRV_MINOR_HWDEPS; f7b2bb8549e352 Dan Carpenter 2011-10-28 283 ^1da177e4c3f41 Linus Torvalds 2005-04-16 284 while (device < SNDRV_MINOR_HWDEPS) { f87135f56cb266 Clemens Ladisch 2005-11-20 285 if (snd_hwdep_search(card, device)) ^1da177e4c3f41 Linus Torvalds 2005-04-16 286 break; ^1da177e4c3f41 Linus Torvalds 2005-04-16 287 device++; ^1da177e4c3f41 Linus Torvalds 2005-04-16 288 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 289 if (device >= SNDRV_MINOR_HWDEPS) ^1da177e4c3f41 Linus Torvalds 2005-04-16 290 device = -1; ^1da177e4c3f41 Linus Torvalds 2005-04-16 @291 if (put_user(device, (int __user *)arg)) ^1da177e4c3f41 Linus Torvalds 2005-04-16 292 return -EFAULT; ^1da177e4c3f41 Linus Torvalds 2005-04-16 293 return 0; ^1da177e4c3f41 Linus Torvalds 2005-04-16 294 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 295 case SNDRV_CTL_IOCTL_HWDEP_INFO: ^1da177e4c3f41 Linus Torvalds 2005-04-16 296 { d9a98de218ce18 Takashi Iwai 2005-11-17 297 struct snd_hwdep_info __user *info = (struct snd_hwdep_info __user *)arg; 9c86a0c433a5a9 Takashi Iwai 2024-02-21 298 int device; d9a98de218ce18 Takashi Iwai 2005-11-17 299 struct snd_hwdep *hwdep; ^1da177e4c3f41 Linus Torvalds 2005-04-16 300 ^1da177e4c3f41 Linus Torvalds 2005-04-16 301 if (get_user(device, &info->device)) ^1da177e4c3f41 Linus Torvalds 2005-04-16 302 return -EFAULT; 9c86a0c433a5a9 Takashi Iwai 2024-02-21 303 guard(mutex)(&register_mutex); f87135f56cb266 Clemens Ladisch 2005-11-20 304 hwdep = snd_hwdep_search(card, device); 9c86a0c433a5a9 Takashi Iwai 2024-02-21 305 if (!hwdep) 9c86a0c433a5a9 Takashi Iwai 2024-02-21 306 return -ENXIO; 9c86a0c433a5a9 Takashi Iwai 2024-02-21 307 return snd_hwdep_info(hwdep, info); ^1da177e4c3f41 Linus Torvalds 2005-04-16 308 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 309 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 310 return -ENOIOCTLCMD; ^1da177e4c3f41 Linus Torvalds 2005-04-16 311 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 312 :::::: The code at line 291 was first introduced by commit :::::: 1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 Linux-2.6.12-rc2 :::::: TO: Linus Torvalds <torvalds(a)ppc970.osdl.org> :::::: CC: Linus Torvalds <torvalds(a)ppc970.osdl.org> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

pulseaudio: rewind does not seem to work properly
by GitHub issues - opened 24 Feb '24

24 Feb '24

alsa-project/alsa-plugins issue #56 was opened from z-s-e: In my application, after I use `snd_pcm_rewind()` strange things happen, such as `snd_pcm_status_get_avail()` reporting a value larger than `period_count * frames_per_period` which my app depends on to never happen. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/56 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

Re: PCI SSID and Codec SSID collision?
by Willian Wang 24 Feb '24

24 Feb '24

Hi, I'm curious about the reasons for the fixup table to be the same for PCI and Codec SSIDs and if the PCI SSID is usually more important for it to be checked before the Codec. Anyway, would a simple conditional statement to match these exact PCI and Codec SSID to set the fixup manually be enough? There seems to already exist a manual fix for TX300 and ROG Strix G17 there (not the same kind of fix and with a FIXME comment, though). I have this exact laptop model and would like to try to contribute to the code if it's not too complicated.

3 4

[PATCH v14 00/53] Introduce QC USB SND audio offloading support
by Wesley Cheng 24 Feb '24

24 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 v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) 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 | 109 + 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 | 72 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- 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, 6228 insertions(+), 177 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

3 71

[PATCH v2 0/6] ALSA: cs35l56: Apply calibration from EFI
by Richard Fitzgerald 24 Feb '24

24 Feb '24

Factory calibration of the speakers stores the calibration information into an EFI variable. This set of patches adds support for applying speaker calibration data from that EFI variable. The HDA patch (#5) depends on the ASoC patches #2 and #3 Patch #6 is a KUnit test for the new cs-amp-lib library. Richard Fitzgerald (6): ASoC: wm_adsp: Add wm_adsp_start() and wm_adsp_stop() ASoC: cs-amp-lib: Add helpers for factory calibration data ASoC: cs35l56: Add helper functions for amp calibration ASoC: cs35l56: Apply amp calibration from EFI data ALSA: hda: cs35l56: Apply amp calibration from EFI data ASoC: cs-amp-lib: Add KUnit test for calibration helpers include/sound/cs-amp-lib.h | 66 +++ include/sound/cs35l56.h | 10 + sound/pci/hda/Kconfig | 2 + sound/pci/hda/cs35l56_hda.c | 39 +- sound/soc/codecs/Kconfig | 17 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs-amp-lib-test.c | 715 +++++++++++++++++++++++++++++ sound/soc/codecs/cs-amp-lib.c | 277 +++++++++++ sound/soc/codecs/cs35l56-sdw.c | 20 + sound/soc/codecs/cs35l56-shared.c | 83 ++++ sound/soc/codecs/cs35l56.c | 44 +- sound/soc/codecs/wm_adsp.c | 27 +- sound/soc/codecs/wm_adsp.h | 2 + 13 files changed, 1287 insertions(+), 19 deletions(-) create mode 100644 include/sound/cs-amp-lib.h create mode 100644 sound/soc/codecs/cs-amp-lib-test.c create mode 100644 sound/soc/codecs/cs-amp-lib.c -- 2.30.2

2 10

[PATCH v12 00/15] Add audio support in v4l2 framework
by Shengjiu Wang 23 Feb '24

23 Feb '24

Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. It is a specific use case, there is no reference in current kernel. v4l2 memory to memory is the closed implementation, v4l2 current support video, image, radio, tuner, touch devices, so it is not complicated to add support for this specific audio case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the first 3 patches is for refining the code to make it can be shared by the "memory->asrc->memory" driver. The main change is in the v4l2 side, A /dev/vl4-audioX will be created, user applications only use the ioctl of v4l2 framework. Other change is to add memory to memory support for two kinds of i.MX ASRC module. changes in v12 - minor changes according to comments - drop min_buffers_needed = 1 and V4L2_CTRL_FLAG_UPDATE flag - drop bus_info changes in v11 - add add-fixed-point-test-controls in vivid. - add v4l2_ctrl_fp_compose() helper function for min and max changes in v10 - remove FIXED_POINT type - change code base on media: v4l2-ctrls: add support for fraction_bits - fix issue reported by kernel test robot - remove module_alias changes in v9: - add MEDIA_ENT_F_PROC_AUDIO_RESAMPLER. - add MEDIA_INTF_T_V4L_AUDIO - add media controller support - refine the vim2m-audio to support 8k<->16k conversion. changes in v8: - refine V4L2_CAP_AUDIO_M2M to be 0x00000008 - update doc for FIXED_POINT - address comments for imx-asrc changes in v7: - add acked-by from Mark - separate commit for fixed point, m2m audio class, audio rate controls - use INTEGER_MENU for rate, FIXED_POINT for rate offset - remove used fmts - address other comments for Hans changes in v6: - use m2m_prepare/m2m_unprepare/m2m_start/m2m_stop to replace m2m_start_part_one/m2m_stop_part_one, m2m_start_part_two/m2m_stop_part_two. - change V4L2_CTRL_TYPE_ASRC_RATE to V4L2_CTRL_TYPE_FIXED_POINT - fix warning by kernel test rebot - remove some unused format V4L2_AUDIO_FMT_XX - Get SNDRV_PCM_FORMAT from V4L2_AUDIO_FMT in driver. - rename audm2m to viaudm2m. changes in v5: - remove V4L2_AUDIO_FMT_LPCM - define audio pixel format like V4L2_AUDIO_FMT_S8... - remove rate and format in struct v4l2_audio_format. - Add V4L2_CID_ASRC_SOURCE_RATE and V4L2_CID_ASRC_DEST_RATE controls - updata document accordingly. changes in v4: - update document style - separate V4L2_AUDIO_FMT_LPCM and V4L2_CAP_AUDIO_M2M in separate commit changes in v3: - Modify documents for adding audio m2m support - Add audio virtual m2m driver - Defined V4L2_AUDIO_FMT_LPCM format type for audio. - Defined V4L2_CAP_AUDIO_M2M capability type for audio m2m case. - with modification in v4l-utils, pass v4l2-compliance test. changes in v2: - decouple the implementation in v4l2 and ALSA - implement the memory to memory driver as a platfrom driver and move it to driver/media - move fsl_asrc_common.h to include/sound folder Shengjiu Wang (15): ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc: move fsl_asrc_common.h to include/sound ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device media: uapi: Add V4L2_CAP_AUDIO_M2M capability flag media: v4l2: Add audio capture and output support media: uapi: Define audio sample format fourcc type media: uapi: Add V4L2_CTRL_CLASS_M2M_AUDIO media: uapi: Add audio rate controls support media: uapi: Declare interface types for Audio media: uapi: Add an entity type for audio resampler media: vivid: add fixed point test controls media: imx-asrc: Add memory to memory driver media: vim2m-audio: add virtual driver for audio memory to memory .../media/mediactl/media-types.rst | 11 + .../userspace-api/media/v4l/buffer.rst | 6 + .../userspace-api/media/v4l/common.rst | 1 + .../media/v4l/dev-audio-mem2mem.rst | 71 + .../userspace-api/media/v4l/devices.rst | 1 + .../media/v4l/ext-ctrls-audio-m2m.rst | 41 + .../userspace-api/media/v4l/pixfmt-audio.rst | 87 ++ .../userspace-api/media/v4l/pixfmt.rst | 1 + .../media/v4l/vidioc-enum-fmt.rst | 2 + .../media/v4l/vidioc-g-ext-ctrls.rst | 4 + .../userspace-api/media/v4l/vidioc-g-fmt.rst | 4 + .../media/v4l/vidioc-querycap.rst | 3 + .../media/videodev2.h.rst.exceptions | 3 + .../media/common/videobuf2/videobuf2-v4l2.c | 4 + drivers/media/platform/nxp/Kconfig | 13 + drivers/media/platform/nxp/Makefile | 1 + drivers/media/platform/nxp/imx-asrc.c | 1256 +++++++++++++++++ drivers/media/test-drivers/Kconfig | 10 + drivers/media/test-drivers/Makefile | 1 + drivers/media/test-drivers/vim2m-audio.c | 793 +++++++++++ drivers/media/test-drivers/vivid/vivid-core.h | 2 + .../media/test-drivers/vivid/vivid-ctrls.c | 26 + drivers/media/v4l2-core/v4l2-compat-ioctl32.c | 9 + drivers/media/v4l2-core/v4l2-ctrls-defs.c | 10 + drivers/media/v4l2-core/v4l2-dev.c | 21 + drivers/media/v4l2-core/v4l2-ioctl.c | 66 + drivers/media/v4l2-core/v4l2-mem2mem.c | 13 +- include/media/v4l2-ctrls.h | 6 + include/media/v4l2-dev.h | 2 + include/media/v4l2-ioctl.h | 34 + .../fsl => include/sound}/fsl_asrc_common.h | 60 + include/uapi/linux/media.h | 2 + include/uapi/linux/v4l2-controls.h | 9 + include/uapi/linux/videodev2.h | 41 + sound/soc/fsl/fsl_asrc.c | 144 ++ sound/soc/fsl/fsl_asrc.h | 4 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 233 +++ sound/soc/fsl/fsl_easrc.h | 6 +- 39 files changed, 2996 insertions(+), 7 deletions(-) create mode 100644 Documentation/userspace-api/media/v4l/dev-audio-mem2mem.rst create mode 100644 Documentation/userspace-api/media/v4l/ext-ctrls-audio-m2m.rst create mode 100644 Documentation/userspace-api/media/v4l/pixfmt-audio.rst create mode 100644 drivers/media/platform/nxp/imx-asrc.c create mode 100644 drivers/media/test-drivers/vim2m-audio.c rename {sound/soc/fsl => include/sound}/fsl_asrc_common.h (60%) -- 2.34.1

5 31

PCI SSID and Codec SSID collision?
by Gergo Koteles 23 Feb '24

23 Feb '24

Hi, The Lenovo Yoga Slim Pro 9 14IRP8 has tas2781 amplifiers, but it can only pickup its fixup with the hda_model=17aa:38be module parameter. The 17aa:38be is the SSID of its Codec. The PCI SSID is 17aa:3802. The patch_realtek.c has these fixup: SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", ALC287_FIXUP_TAS2781_I2C), And the snd_hda_pick_fixup() picks the former fixup because of the same PCI SSID, but that is a Codec SSID? At least it worked without that line. https://bugzilla.kernel.org/show_bug.cgi?id=208555#c839 It would be nice if it can be work without the module parameter. What is the best way to add this strangeness to patch_realtek.c? Thanks, Gergo

2 1

[PATCH] ALSA: hda/realtek: tas2781: enable subwoofer volume control
by Gergo Koteles 23 Feb '24

23 Feb '24

The volume of subwoofer channels is always at maximum with the ALC269_FIXUP_THINKPAD_ACPI chain. Use ALC285_FIXUP_THINKPAD_HEADSET_JACK to align it to the master volume. Link: https://bugzilla.kernel.org/show_bug.cgi?id=208555#c827 Fixes: 3babae915f4c ("ALSA: hda/tas2781: Add tas2781 HDA driver") Cc: <stable(a)vger.kernel.org> Signed-off-by: Gergo Koteles <soyer(a)irl.hu> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0ec1312bffd5..24a26959070f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9585,7 +9585,7 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = tas2781_fixup_i2c, .chained = true, - .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK, }, [ALC287_FIXUP_YOGA7_14ARB7_I2C] = { .type = HDA_FIXUP_FUNC, base-commit: b401b621758e46812da61fa58a67c3fd8d91de0d -- 2.43.2

2 1

[PATCH 0/4] ALSA/ASoC: Conditionally skip i915 init and cleanups
by Cezary Rojewski 23 Feb '24

23 Feb '24

A small set of changes to improve initialization of the audio stack on HDAudio devices and pair of cleanups. While I did touch i915_pciids.h header which is part of drm, I'd like the changes to go through Mark's tree entirely - being cohesive with the rest of the code by adding LKF IDs where they belong rather than hiding them within the sound tree. As the first change is the most important one here, following is the technical background for it: Commit 78f613ba1efb ("drm/i915: finish removal of CNL") and its friends removed support for i915 for all CNL-based platforms. HDAudio library, however, still treats such platforms as valid candidates for i915 binding. Update query mechanism to reflect changes made in drm tree. At the same time, i915 support for LKF-based platforms has not been provided so remove them from valid binding candidates. The snd_soc_hda change is a follow up for the above and the cleanup patches do not bring any functional changes. Cezary Rojewski (4): ALSA: hda: Skip i915 initialization on CNL/LKF-based platforms ASoC: codecs: hda: Skip HDMI/DP registration if i915 is missing ASoC: codecs: hda: Cleanup error messages ALSA: hda: Reuse for_each_pcm_streams() include/drm/i915_pciids.h | 4 ++++ sound/hda/hdac_i915.c | 18 +++++++++++++++--- sound/pci/hda/hda_codec.c | 2 +- sound/soc/codecs/hda.c | 15 ++++++++++----- 4 files changed, 30 insertions(+), 9 deletions(-) -- 2.25.1

4 9

[PATCH v17 00/51] Introduce QC USB SND audio offloading support
by Wesley Cheng 23 Feb '24

23 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 v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) 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 (11): 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: Introduce SND kcontrols to track USB offloading state 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: Add USB offloading capable kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ALSA: usb-audio: qcom: Use card and PCM index from QMI request 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 | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 39 +- 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 | 182 ++ sound/soc/Kconfig | 9 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 10 + sound/soc/qcom/Makefile | 2 + 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 | 408 ++++ sound/soc/qcom/sm8250.c | 15 +- sound/soc/qcom/usb_offload_utils.c | 51 + sound/soc/qcom/usb_offload_utils.h | 26 + sound/soc/soc-usb.c | 660 ++++++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 2 +- sound/usb/card.c | 109 + sound/usb/card.h | 26 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 6 + sound/usb/qcom/mixer_usb_offload.c | 65 + sound/usb/qcom/mixer_usb_offload.h | 17 + sound/usb/qcom/qc_audio_offload.c | 1914 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 50 files changed, 6496 insertions(+), 167 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/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/mixer_usb_offload.c create mode 100644 sound/usb/qcom/mixer_usb_offload.h 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

5 61

[PATCH usb next] ASoC: Revert "ASoC: dt-bindings: Update example for enabling USB offload on SM8250"
by Krzysztof Kozlowski 23 Feb '24

23 Feb '24

This reverts commit a9c83252bff616cf3a38d55b7c6a6ad63667f2dd from USB tree, because it depends on other DT bindings changes which were not applied. This commit alone causes dt_binding_check failures: Documentation/devicetree/bindings/sound/qcom,sm8250.example.dts:97.44-45 syntax error Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Greg, Please take this revert. Original commit should go via Mark's ASoC. --- .../devicetree/bindings/sound/qcom,sm8250.yaml | 15 --------------- 1 file changed, 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index 49e4f5bbe9dd..2ab6871e89e5 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -227,21 +227,6 @@ examples: sound-dai = <&vamacro 0>; }; }; - - usb-dai-link { - link-name = "USB Playback"; - cpu { - sound-dai = <&q6afedai USB_RX>; - }; - - codec { - sound-dai = <&usbdai USB_RX>; - }; - - platform { - sound-dai = <&q6routing>; - }; - }; }; - | -- 2.34.1

2 1

Skip missing ladspa directories
by GitHub pull_request - opened 23 Feb '24

23 Feb '24

alsa-project/alsa-lib pull request #385 was opened from m42a: If there is a nonexistent directory in the user's ladspa path, skip it instead of immediately failing the search. Request URL : https://github.com/alsa-project/alsa-lib/pull/385 Patch URL : https://github.com/alsa-project/alsa-lib/pull/385.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[tiwai-sound:test/guard 44/62] sound/core/oss/mixer_oss.c:819:3: error: cannot jump from this goto statement to its label
by kernel test robot 23 Feb '24

23 Feb '24

Hi Takashi, FYI, the error/warning was bisected to this commit, please ignore it if it's irrelevant. tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/guard head: e4545d528ae658c35d02593567613c06f1386ffa commit: b398454ce784b8a931d9fe80a0c1fb9dec644d29 [44/62] ALSA: mixer_oss: Use guard() for locking config: hexagon-allmodconfig (https://download.01.org/0day-ci/archive/20240223/202402231259.Tx6bs1or-lkp@…) compiler: clang version 19.0.0git (https://github.com/llvm/llvm-project edd4aee4dd9b5b98b2576a6f783e4086173d902a) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240223/202402231259.Tx6bs1or-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202402231259.Tx6bs1or-lkp@intel.com/ All errors (new ones prefixed by >>): >> sound/core/oss/mixer_oss.c:819:3: error: cannot jump from this goto statement to its label 819 | goto error; | ^ sound/core/oss/mixer_oss.c:821:2: note: jump bypasses initialization of variable with __attribute__((cleanup)) 821 | guard(rwsem_read)(&card->controls_rwsem); | ^ include/linux/cleanup.h:164:15: note: expanded from macro 'guard' 164 | CLASS(_name, __UNIQUE_ID(guard)) | ^ include/linux/compiler.h:180:29: note: expanded from macro '__UNIQUE_ID' 180 | #define __UNIQUE_ID(prefix) __PASTE(__PASTE(__UNIQUE_ID_, prefix), __COUNTER__) | ^ include/linux/compiler_types.h:84:22: note: expanded from macro '__PASTE' 84 | #define __PASTE(a,b) ___PASTE(a,b) | ^ include/linux/compiler_types.h:83:23: note: expanded from macro '___PASTE' 83 | #define ___PASTE(a,b) a##b | ^ <scratch space>:36:1: note: expanded from here 36 | __UNIQUE_ID_guard239 | ^ sound/core/oss/mixer_oss.c:870:3: error: cannot jump from this goto statement to its label 870 | goto error; | ^ sound/core/oss/mixer_oss.c:872:2: note: jump bypasses initialization of variable with __attribute__((cleanup)) 872 | guard(rwsem_read)(&card->controls_rwsem); | ^ include/linux/cleanup.h:164:15: note: expanded from macro 'guard' 164 | CLASS(_name, __UNIQUE_ID(guard)) | ^ include/linux/compiler.h:180:29: note: expanded from macro '__UNIQUE_ID' 180 | #define __UNIQUE_ID(prefix) __PASTE(__PASTE(__UNIQUE_ID_, prefix), __COUNTER__) | ^ include/linux/compiler_types.h:84:22: note: expanded from macro '__PASTE' 84 | #define __PASTE(a,b) ___PASTE(a,b) | ^ include/linux/compiler_types.h:83:23: note: expanded from macro '___PASTE' 83 | #define ___PASTE(a,b) a##b | ^ <scratch space>:45:1: note: expanded from here 45 | __UNIQUE_ID_guard240 | ^ 2 errors generated. vim +819 sound/core/oss/mixer_oss.c 803 804 static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned int *active_index) 805 { 806 struct snd_card *card = fmixer->card; 807 struct snd_mixer_oss *mixer = fmixer->mixer; 808 struct snd_kcontrol *kctl; 809 struct snd_mixer_oss_slot *pslot; 810 struct slot *slot; 811 struct snd_ctl_elem_info *uinfo; 812 struct snd_ctl_elem_value *uctl; 813 int err, idx; 814 815 uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); 816 uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); 817 if (uinfo == NULL || uctl == NULL) { 818 err = -ENOMEM; > 819 goto error; 820 } 821 guard(rwsem_read)(&card->controls_rwsem); 822 kctl = snd_mixer_oss_test_id(mixer, "Capture Source", 0); 823 if (! kctl) { 824 err = -ENOENT; 825 goto error; 826 } 827 err = kctl->info(kctl, uinfo); 828 if (err < 0) 829 goto error; 830 err = kctl->get(kctl, uctl); 831 if (err < 0) 832 goto error; 833 for (idx = 0; idx < 32; idx++) { 834 if (!(mixer->mask_recsrc & (1 << idx))) 835 continue; 836 pslot = &mixer->slots[idx]; 837 slot = pslot->private_data; 838 if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE) 839 continue; 840 if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE)) 841 continue; 842 if (slot->capture_item == uctl->value.enumerated.item[0]) { 843 *active_index = idx; 844 break; 845 } 846 } 847 err = 0; 848 error: 849 kfree(uctl); 850 kfree(uinfo); 851 return err; 852 } 853 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH] ASoC: soc-core: tidyup strcmp() param on snd_soc_is_matching_dai()
by Kuninori Morimoto 22 Feb '24

22 Feb '24

snd_soc_is_matching_dai() checks DAI name, which is paired function with snd_soc_dai_name_get(). It checks dlc->dai_name and dai->name (A) or dai->driver_name (B) or dai->component->name (C) static int snd_soc_is_matching_dai(...) { ... if (strcmp(dlc->dai_name, dai->name) == 0) ~~~~~~~~~~~~~ ^^^^^^^^^(A) if (... strcmp(dai->driver->name, dlc->dai_name) == 0) (B)^^^^^^^^^^^^^^^^ ~~~~~~~~~~~~~ if (... strcmp(dlc->dai_name, dai->component->name) == 0) ~~~~~~~~~~~~~ ^^^^^^^^^^^^^^^^^^(C) ... } But (B) part order is different with (A) and (C) (= ^^^^ and ~~~~). This is not a big deal, but confusable to read. Fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b11b2ca5d939..507cd3015ff4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -287,7 +287,7 @@ static int snd_soc_is_matching_dai(const struct snd_soc_dai_link_component *dlc, return 1; if (dai->driver->name && - strcmp(dai->driver->name, dlc->dai_name) == 0) + strcmp(dlc->dai_name, dai->driver->name) == 0) return 1; if (dai->component->name && -- 2.25.1

2 1

[PATCH 1/2] ASoC: amd: ps: update license
by Vijendar Mukunda 22 Feb '24

22 Feb '24

To align with AMD SoundWire manager driver license, update license as GPL-2.0-only for Pink Sardine ACP PCI driver and corresponding child drivers. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/Makefile | 2 +- sound/soc/amd/ps/pci-ps.c | 2 +- sound/soc/amd/ps/ps-mach.c | 2 +- sound/soc/amd/ps/ps-pdm-dma.c | 2 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/ps/Makefile b/sound/soc/amd/ps/Makefile index f2a5eaf2fa4d..b3c254886fd9 100644 --- a/sound/soc/amd/ps/Makefile +++ b/sound/soc/amd/ps/Makefile @@ -1,4 +1,4 @@ -# SPDX-License-Identifier: GPL-2.0+ +# SPDX-License-Identifier: GPL-2.0-only # Pink Sardine platform Support snd-pci-ps-objs := pci-ps.o snd-ps-pdm-dma-objs := ps-pdm-dma.o diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 205bca95aa06..c72d666d51bd 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0+ +// SPDX-License-Identifier: GPL-2.0-only /* * AMD Pink Sardine ACP PCI Driver * diff --git a/sound/soc/amd/ps/ps-mach.c b/sound/soc/amd/ps/ps-mach.c index 3ffbe4fdafdf..e675b8f569eb 100644 --- a/sound/soc/amd/ps/ps-mach.c +++ b/sound/soc/amd/ps/ps-mach.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0+ +// SPDX-License-Identifier: GPL-2.0-only /* * Machine driver for AMD Pink Sardine platform using DMIC * diff --git a/sound/soc/amd/ps/ps-pdm-dma.c b/sound/soc/amd/ps/ps-pdm-dma.c index d48f7c5af289..7bbacbab1095 100644 --- a/sound/soc/amd/ps/ps-pdm-dma.c +++ b/sound/soc/amd/ps/ps-pdm-dma.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0+ +// SPDX-License-Identifier: GPL-2.0-only /* * AMD ALSA SoC Pink Sardine PDM Driver * diff --git a/sound/soc/amd/ps/ps-sdw-dma.c b/sound/soc/amd/ps/ps-sdw-dma.c index 9b59063798f2..66b800962f8c 100644 --- a/sound/soc/amd/ps/ps-sdw-dma.c +++ b/sound/soc/amd/ps/ps-sdw-dma.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0+ +// SPDX-License-Identifier: GPL-2.0-only /* * AMD ALSA SoC Pink Sardine SoundWire DMA Driver * -- 2.34.1

2 2

[PATCH v6 0/6] reset: gpio: ASoC: shared GPIO resets
by Krzysztof Kozlowski 22 Feb '24

22 Feb '24

Hi, Dependencies / Merging ====================== 1. Depends on !GPIOLIB stub: https://lore.kernel.org/all/20240125081601.118051-3-krzysztof.kozlowski@lin… 2. Patch #2 (cpufreq: do not open-code of_phandle_args_equal()) and patch #4 (reset: Instantiate reset GPIO controller for shared reset-gpios) depend on OF change (patch #1). Changes in v6 ============= 1. reset/core.c: Add check for number of GPIO cells==2 (Neil). 2. Add Rb/Ack tags. Changes in v5 ============= 1. Minor comments from Philipp: missing cleanup.h, correct indentation of pr_err(), shorten gpio_device_find_by_fwnode() line. 2. Acks/Rbs. Changes in v4 ============= 1. New patches: of: add of_phandle_args_equal() helper cpufreq: do not open-code of_phandle_args_equal() 2. reset-gpio.c: - Drop unneeded comment (Bartosz), add Rb tag. - Do not assign of_node. 3. reset/core.c: - Implement most of Bartosz feedback (I responded to one which I did not implement) and comments from Philipp. - Expect either rcdev->of_args or rcdev->of_node. - Drop __reset_gpios_args_match() and use common helper (Philipp). - Move declarations of automatic-cleanup variables in __reset_add_reset_gpio_lookup() to place of use (Bartosz). - Separate gpio_device_get_label() and kstrdup() (Philipp). - Correct doc for __reset_add_reset_gpio_device(), rewrite few comments. - Drop unneeded "r" variable in __reset_find_rcdev() (Philipp). - Drop of_phandle_args initialization in __of_reset_control_get (Philipp). - Check if CONFIG_RESET_GPIO is enabled before trying to look up reset-gpios. 4. Drop Chris' patch: "i2c: muxes: pca954x: Allow sharing reset GPIO", because discussion is on going. Changes in v3 ============= 1. reset-gpio.c: - Add reset_gpio_of_xlate (Philipp). - reset_gpio_of_args_put->reset_gpio_of_node_put (Philipp). - Expect via platdata of_phandle_args. - Do not call device_set_node() to attach itself to reset consumer (the final device). This was questionable idea in the first place. Bartosz suggested to use GPIO_LOOKUP to solve this. 2. reset/core.c, implement Philipp's feedback. That was a lot: - Commit msg fixes. - Add new platform_device earlier, when reset core found "reset-gpios" but not "resets". - Do not overwrite of_phandle_args. - Expect matching .of_reset_n_cells. - Pass of_phandle_args as platdata to reset-gpio. - Rename reset_gpio_device->reset_gpio_lookup and others. Fix few comments and code cleanup pointed on review. - From Bartosz: Use GPIO_LOOKUP and a lot of cleanup.h in __reset_add_reset_gpio_lookup(). 3. Include here Chris' patch: "i2c: muxes: pca954x: Allow sharing reset GPIO". Changes in v2 ============= 1. wsa884x.c: add missing return in wsa884x_get_reset(), correct comment. 2. qcom,wsa8840.yaml: fix oneOf syntax. 3. reset-gpio.c: - Fix smatch warning on platdata evaluation. - Parse GPIO args and store them in rc.of_args. 4. reset/core.c: - Revise approach based on Bartosz comments: parse the reset-gpios phandle with arguments, do not use deprecated API and do not rely on gpio_desc pointer. - Create a list of instantiated platform devices to avoid any duplicates. - After creating reset-gpio platform device, try to get new reset controller or return EPROBE_DEFER. - Drop the "cookie" member and add new "of_args" to "struct reset_controller_dev". Description =========== We have at least few cases where hardware engineers decided to use one powerdown/shutdown/reset GPIO line for multiple devices: 1. WSA884x (this and previous patch): https://lore.kernel.org/all/b7aeda24-d638-45b7-8e30-80d287f498f8@sirena.org… 2. https://lore.kernel.org/all/20231027033104.1348921-1-chris.packham@alliedte… 3. https://lore.kernel.org/lkml/20191030120440.3699-1-peter.ujfalusi@ti.com/ 4. https://lore.kernel.org/all/20211018234923.1769028-1-sean.anderson@seco.com/ 5. https://social.treehouse.systems/@marcan/111268780311634160 I try to solve my case, hopefuly Chris' (2), partially Sean's (4) and maybe Hectors (5), using Rob's suggestion: https://lore.kernel.org/all/YXi5CUCEi7YmNxXM@robh.at.kernel.org/ Best regards, Krzysztof Cc: Chris Packham <chris.packham(a)alliedtelesis.co.nz> Cc: Bartosz Golaszewski <brgl(a)bgdev.pl> Cc: Sean Anderson <sean.anderson(a)seco.com> Krzysztof Kozlowski (6): of: Add of_phandle_args_equal() helper cpufreq: do not open-code of_phandle_args_equal() reset: gpio: Add GPIO-based reset controller reset: Instantiate reset GPIO controller for shared reset-gpios ASoC: dt-bindings: qcom,wsa8840: Add reset-gpios for shared line ASoC: codecs: wsa884x: Allow sharing reset GPIO .../bindings/sound/qcom,wsa8840.yaml | 11 +- MAINTAINERS | 5 + drivers/reset/Kconfig | 9 + drivers/reset/Makefile | 1 + drivers/reset/core.c | 224 +++++++++++++++++- drivers/reset/reset-gpio.c | 119 ++++++++++ include/linux/cpufreq.h | 3 +- include/linux/of.h | 16 ++ include/linux/reset-controller.h | 4 + sound/soc/codecs/wsa884x.c | 53 ++++- 10 files changed, 419 insertions(+), 26 deletions(-) create mode 100644 drivers/reset/reset-gpio.c -- 2.34.1

5 25

[PATCH AUTOSEL 6.7 07/44] ASoC: codecs: wcd934x: drop unneeded regulator include
by Sasha Levin 22 Feb '24

22 Feb '24

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 35314e39dabcfb256832654ad0e856a9fba744bd ] Driver does not use any regulator code, so drop redundant include of regulator/consumer.h header. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Link: https://msgid.link/r/20240117151208.1219755-3-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wcd934x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 1b6e376f3833..6813268e6a19 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -13,7 +13,6 @@ #include <linux/of.h> #include <linux/platform_device.h> #include <linux/regmap.h> -#include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/slimbus.h> #include <sound/pcm_params.h> -- 2.43.0

2 2

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

21 Feb '24

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

2 1

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

21 Feb '24

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

2 11

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

21 Feb '24

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

1 1

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

21 Feb '24

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

2 2

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

21 Feb '24

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

3 2

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

21 Feb '24

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

2 1

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

21 Feb '24

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

2 1

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

21 Feb '24

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

2 5

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

20 Feb '24

The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp speaker protection algorithm. The integrated speaker voltage and current sense provides for real-time monitoring of lodspeakers. The ASoC component provides the majority of the functionality of the device, all the audio functions. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v8: - simplify tasdevice_set_sdw_stream - fixed some Linux coding style - fixed the spelling mistakes - Select left/right channel based on unique id - a longer description has been added - remove unused crc8 in KCONFIG - Invert of RE here means reciprocal of RE - correct indentation - correct the wrong pow calibrated register - correct comment on RESET register - fixed usually lines don't end with an open parenthesis - fixed tasdevice_sdw_shutdown - use defines for function, entity, etc. for readable - remove unnecessary codec_lock in component_probe() - remove struct sdw_stream_data - remove unused macro definitions - use BIT(2) and BIT(1) instead of 0x2 and 0x1 - implement pm_runtime - reg_start is u8, reg_start + i only adds i byte, it should be reg_start + i * 4, because the calibration values - fixed total_sz in tas2783_calibration - add a define for TASDEVICE_REG(0, 0, 0x0a) - rework macro style in header file - Add select CRC32 - rewrite the mute and the return value handling - Add comments on bin file - Add link_id into the fw name to distinguish the same sdw id on different sdw id. - remove unnecessary mutex_lock - use set_active() in tasdevice_io_init and move all other parts in the probe. - move pm_runtime_get_noresume to the probe - fix the return value of putvol function - add put_noidle in tasdevice_sdw_remove - remove unused header file(of.h) - correct the style of the multi-line comment - Set the readable registers in tas2783_readable_register for debugfs - Use new PM macros - Use default put and get inline SOC_SINGLE_RANGE_TLV and SOC_SINGLE_SX_TLV - Use respective %pt to log the calibrated time - Remove unnecessary struct tm - Add trailing comma tasdevice_dapm_widgets and tasdevice_audio_map - Rewrite the return value handling in tasdevice_sdw_probe - use dev_num instead of id.unique_id - rework the log --- sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 844 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 103 ++++ 4 files changed, 965 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 59f9742e9ff4..6ba3fd0c0811 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -247,6 +247,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1847,6 +1848,21 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + depends on EFI + select REGMAP + select REGMAP_SOUNDWIRE + select CRC32 + select CRC32_SARWATE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + TAS2783 supports the MIPI SoundWire v1.2 compatible interface + for audio and control data. At the end of calibrated data appends + the crc data with crc32_sarwate for error detection. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f53baa2b9565..6afceb126ae3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -284,6 +284,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -664,6 +665,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..ebb7102828c5 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,844 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Kevin Lu <kevin-lu(a)ti.com> +// Shenghao Ding <shenghao-ding(a)ti.com> +// + +#include <asm/unaligned.h> +#include <linux/acpi.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/pcm_params.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include <sound/tas2781-tlv.h> +#include "tas2783.h" + +static const unsigned int tas2783_cali_reg[] = { + TAS2783_CALIBRATION_RE, /* Resistance */ + TAS2783_CALIBRATION_RE_LOW, /* Low limitation of RE */ + TAS2783_CALIBRATION_INV_RE, /* Reciprocal of RE */ + TAS2783_CALIBRATION_POW, /* RMS Power */ + TAS2783_CALIBRATION_TLIMIT /* Temperature limitation */ +}; + +static const struct reg_default tas2783_reg_defaults[] = { + /* Default values for ROM mode. Activated. */ + { 0x8002, 0x1a }, /* AMP inactive. */ + { 0x8097, 0xc8 }, + { 0x80b5, 0x74 }, + { 0x8099, 0x20 }, + { 0xfe8d, 0x0d }, + { 0xfebe, 0x4a }, + { 0x8230, 0x00 }, + { 0x8231, 0x00 }, + { 0x8232, 0x00 }, + { 0x8233, 0x01 }, + { 0x8418, 0x00 }, /* Set volume to 0 dB. */ + { 0x8419, 0x00 }, + { 0x841a, 0x00 }, + { 0x841b, 0x00 }, + { 0x8428, 0x40 }, /* Unmute channel */ + { 0x8429, 0x00 }, + { 0x842a, 0x00 }, + { 0x842b, 0x00 }, + { 0x8548, 0x00 }, /* Set volume to 0 dB. */ + { 0x8549, 0x00 }, + { 0x854a, 0x00 }, + { 0x854b, 0x00 }, + { 0x8558, 0x40 }, /* Unmute channel */ + { 0x8559, 0x00 }, + { 0x855a, 0x00 }, + { 0x855b, 0x00 }, + { 0x800a, 0x3a }, /* Enable both channel */ + { 0x800e, 0x44 }, + { 0x800f, 0x40 }, + { 0x805c, 0x99 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_VOLUME, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0x807F: /* Page 0 */ + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + /* + * Only software reset register was volatiled. + * Bit is self clearing. + * 0b = Don't reset + * 1b = reset + */ + case 0x8001: + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x44ffffff, + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, dvc_tlv), + SOC_SINGLE_SX_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, amp_vol_tlv), +}; + +static void tas2783_apply_calib(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + struct regmap *map = tas_dev->regmap; + u8 *cali_start; + u16 dev_num; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + dev_num = clamp(tas_dev->sdw_peripheral->dev_num, 1, 4) - 1; + /* + * The area saving tas2783 calibrated data is specified by its + * dev_num. cali_start is the first address of current tas2783's + * calibrated data. + */ + cali_start = (u8 *)(cali_data + dev_num * sizeof(tas2783_cali_reg)); + for (int i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + ret = regmap_bulk_write(map, tas2783_cali_reg[i], + &cali_start[4 * i], 4); + if (ret) { + dev_err(tas_dev->dev, "Cali failed %x:%d\n", + tas2783_cali_reg[i], ret); + break; + } + } +} + +/* + * Load the calibration data, including speaker impedance, f0, etc. + * Calibration is done by the manufacturer in the factory. The calibration + * data are used by the algorithm for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * The DSP will work with default data values if calibrated data are + * missing or are invalid. + * Layout of calibrated Data in UEFI: + * Calibrated Data of Dev 0 (20 bytes) + * Calibrated Data of Dev 1 (20 bytes) + * Calibrated Data of Dev 2 (20 bytes) + * Calibrated Data of Dev 3 (20 bytes) + * CRC (4 bytes) + */ +static void tas2783_calibration(struct tasdevice_priv *tas_dev) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + unsigned int attr = 0, crc; + unsigned int *tmp_val; + efi_status_t status; + + tas_dev->cali_data.total_sz = 0; + + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_dev->cali_data.total_sz, tas_dev->cali_data.data); + if (status == EFI_BUFFER_TOO_SMALL) { + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_dev->cali_data.total_sz, + tas_dev->cali_data.data); + dev_dbg(tas_dev->dev, "cali get %lx bytes result:%ld\n", + tas_dev->cali_data.total_sz, status); + } + if (status != 0) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return; + } + + tmp_val = (unsigned int *)tas_dev->cali_data.data; + + crc = crc32(~0, tas_dev->cali_data.data, 84) ^ ~0; + + if (crc == tmp_val[21]) { + /* Date and time of when calibration was done. */ + tas2783_apply_calib(tas_dev, tmp_val); + dev_dbg(tas_dev->dev, "%ptTs", &tmp_val[20]); + } else { + dev_dbg(tas_dev->dev, "CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[21]); + tas_dev->cali_data.total_sz = 0; + } +} + +static void tasdevice_dspfw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct tas2783_firmware_node *p; + struct regmap *map = tas_dev->regmap; + unsigned char *buf = NULL; + int offset = 0, img_sz; + int ret; + + if (!fmw || !fmw->data) { + dev_warn(tas_dev->dev, + "%s: failed to read %s: work in bypass-dsp mode.\n", + __func__, tas_dev->dspfw_binaryname); + return; + } + buf = (unsigned char *)fmw->data; + + img_sz = get_unaligned_le32(&buf[offset]); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_warn(tas_dev->dev, "%s: size not matching, %d %u.", + __func__, (int)fmw->size, img_sz); + return; + } + + while (offset < img_sz) { + p = (struct tas2783_firmware_node *)(buf + offset); + if (p->length > 1) { + ret = regmap_bulk_write(map, p->download_addr, + buf + offset + sizeof(unsigned int) * 5, p->length); + } else { + ret = regmap_write(map, p->download_addr, + *(buf + offset + sizeof(unsigned int) * 5)); + } + + if (ret != 0) { + dev_dbg(tas_dev->dev, + "%s: load FW fail: %d, work in bypass.\n", + __func__, ret); + return; + } + offset += sizeof(unsigned int) * 5 + p->length; + } + + tas2783_calibration(tas_dev); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC"), +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"}, +}; + +static int tasdevice_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + int ret; + + dev_dbg(dai->dev, "%s: dai_name %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(tas_priv->sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) { + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); + return ret; + } + + dev_dbg(dai->dev, "%s: format: %i rate: %i\n", __func__, + params_format(params), params_rate(params)); + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tas_priv->sdw_peripheral, sdw_stream); + + return 0; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct regmap *map = tas_dev->regmap; + int ret; + + dev_dbg(tas_dev->dev, "%s: %d.\n", __func__, mute); + + if (mute) { + if (direction == SNDRV_PCM_STREAM_CAPTURE) { + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_INACTIVE); + if (ret) + dev_err(tas_dev->dev, + "%s: Disable AEF failed.\n", __func__); + } else { + /* FU23 mute (0x40400108) */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 1); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 mute failed.\n", __func__); + goto out; + } + /* + * Both playback and echo data will be shutdown in + * playback stream. + */ + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK | + TAS2783_REG_AEF_MASK, + TAS2783_REG_PWR_MODE_ACTIVE | + TAS2783_REG_PWR_MODE_SW_PWD); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR&AEF shutdown failed.\n", + __func__); + goto out; + } + tas_dev->pstream = false; + } + } else { + /* FU23 Unmute, 0x40400108. */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 Unmute failed.\n", __func__); + goto out; + } + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK, + TAS2783_REG_PWR_MODE_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR Unmute failed.\n", __func__); + goto out; + } + tas_dev->pstream = true; + } else { + /* Capture stream is the echo ref data for voice. + * Without playback, it can't be active. + */ + if (tas_dev->pstream == true) { + ret = regmap_update_bits(map, + TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: AEF enable failed.\n", + __func__); + goto out; + } + } else { + dev_err(tas_dev->dev, + "%s: No playback, no AEF!", __func__); + ret = -EINVAL; + } + } + } +out: + if (ret) + dev_err(tas_dev->dev, "Mute or unmute %d failed %d.\n", + mute, ret); + + return ret; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_comp_probe(struct snd_soc_component *comp) +{ + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(comp); + acpi_handle handle = ACPI_HANDLE(tas_dev->dev); + const struct firmware *fw_entry = NULL; + const char *sub = NULL; + int ret, value_sdw; + + if (handle) { + sub = acpi_get_subsystem_id(handle); + if (IS_ERR(sub)) + sub = NULL; + } + + tas_dev->component = comp; + + /* + * Each tas2783 in the system has its own dspfw. + */ + if (comp->name_prefix) { + /* + * name_prefix.bin stores the dsp firmware including speaker + * protection algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied into + * firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, "%s-tas2783.bin", + comp->name_prefix); + } else { + /* Compatible with the previous naming rule */ + if (sub) { + /* + * subsystem_id-link_id[0,1,...,N]-dev_num[1,...,4].bin stores + * the dsp firmware including speaker protection algorithm, + * audio acoustic algorithm, speaker characters and algorithm + * params, it must be copied into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "%s-%d-%d.bin", sub, + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->dev_num); + } else { + /* + * tas2783-link_id[0,1,...,N]-dev_num[1,...,4].bin stores + * the dsp firmware including speaker protection algorithm, + * audio acoustic algorithm, speaker characters and algorithm + * params, it must be copied into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "tas2783-%d-%d.bin", + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->dev_num); + } + } + + ret = request_firmware(&fw_entry, tas_dev->dspfw_binaryname, + tas_dev->dev); + if (ret) { + dev_err(tas_dev->dev, + "%s: request_firmware %x open status: %d.\n", __func__, + tas_dev->sdw_peripheral->id.unique_id, ret); + goto out; + } + + tasdevice_dspfw_ready(fw_entry, tas_dev); + + /* Select left/right channel based on device number. */ + value_sdw = 0x1a; + value_sdw |= (tas_dev->sdw_peripheral->dev_num & BIT(0)) << 4; + dev_dbg(tas_dev->dev, "%s: dev_num = %u", __func__, + tas_dev->sdw_peripheral->dev_num); + regmap_write(tas_dev->regmap, TAS2783_REG_TDM_RX_CFG, value_sdw); + if (ret != 0) + dev_warn(tas_dev->dev, "%s: L/R setting failed: %d.\n", + __func__, ret); + +out: + if (fw_entry) + release_firmware(fw_entry); + return 0; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_comp_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + return ret; + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + pm_runtime_enable(tas_dev->dev); + + return 0; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + struct sdw_dpn_prop *dpn; + unsigned long addr; + int nval, i, j; + u32 bit; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = BIT(2); /* BITMAP: 00000100 */ + prop->sink_ports = BIT(1); /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static int tasdevice_io_init(struct device *dev, + struct sdw_slave *slave) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(tas_priv->regmap, false); + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + * Update count of parent 'active' children + */ + pm_runtime_set_active(&slave->dev); + } + + /* sw reset */ + ret = regmap_write(tas_priv->regmap, TAS2873_REG_SWRESET, + TAS2873_REG_SWRESET_RESET); + if (ret) { + dev_err(tas_priv->dev, "Reset failed.\n"); + goto out; + } + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, false); + regcache_mark_dirty(tas_priv->regmap); + } + + tas_priv->first_hw_init = true; + tas_priv->hw_init = true; + +out: + return ret; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tas_priv->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tas_priv->hw_init = false; + /* + * Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_priv->hw_init || tas_priv->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return -ENOMEM; + + tas_dev->dev = dev; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_peripheral = peripheral; + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) + return dev_err_probe(dev, PTR_ERR(tas_dev->regmap), + "Failed devm_regmap_init_sdw."); + + return tasdevice_init(tas_dev); +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(&peripheral->dev); + + if (tas_dev->first_hw_init) + pm_runtime_disable(tas_dev->dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {} +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static int __maybe_unused tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (!tas_priv->hw_init) + return 0; + + regcache_cache_only(tas_priv->regmap, true); + + return 0; +} + +#define TAS2783_PROBE_TIMEOUT_MS 5000 + +static int __maybe_unused tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long time; + + if (!tas_priv->first_hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT_MS)); + if (!time) { + dev_err(&slave->dev, "Init not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(tas_priv->regmap, false); + regcache_sync(tas_priv->regmap); + + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume) + RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume, NULL) +}; + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = &tas2783_sdca_pm, + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..41bdce1964ff --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,103 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2023 - 2024 Texas Instruments Incorporated + * https://www.ti.com + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + * Shenghao Ding <shenghao-ding(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 + \ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0x00, 0x01) +#define TAS2873_REG_SWRESET_MASK BIT(0) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +#define TAS2873_REG_PWR_CTRL TASDEVICE_REG(0x0, 0x00, 0x02) +#define TAS2783_REG_AEF_MASK GENMASK(4, 3) +#define TAS2783_REG_AEF_ACTIVE 0 +#define TAS2783_REG_AEF_INACTIVE GENMASK(4, 3) +#define TAS2783_REG_PWR_MODE_MASK GENMASK(2, 0) +#define TAS2783_REG_PWR_MODE_ACTIVE 0 +#define TAS2783_REG_PWR_MODE_SW_PWD BIT(1) + +#define TAS2783_REG_TDM_RX_CFG TASDEVICE_REG(0x0, 0x00, 0x0a) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1a) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x13, 0x70) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +/* TAS2783 SDCA Control - function number */ +#define TAS2783_FUNC_TYPE_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_FU_VOLUME 0x02 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 +#define TAS2783_DSPFW_FILENAME_LEN 64 + +#define TAS2783_MAX_CALIDATA_SIZE 252 + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct snd_soc_component *component; + struct calibration_data cali_data; + struct sdw_slave *sdw_peripheral; + enum sdw_slave_status status; + struct sdw_bus_params params; + struct regmap *regmap; + struct device *dev; + unsigned char dspfw_binaryname[TAS2783_DSPFW_FILENAME_LEN]; + unsigned char dev_name[32]; + unsigned int chip_id; + bool pstream; + bool hw_init; + bool first_hw_init; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

4 3

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

20 Feb '24

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

4 4

(no subject)
by Takashi Iwai 19 Feb '24

19 Feb '24

Linux regressions mailing list <regressions(a)lists.linux.dev> Subject: Re: [REGRESSION] Acp5x probing regression introduced between kernel 6.7.2 -> 6.7.4 In-Reply-To: <7a0cd63f-8a83-4dc5-8763-63dcdae8d68a(a)leemhuis.info> References: <CAD_nV8BG0t7US=+C28kQOR==712MPfZ9m-fuKksgoZCgrEByCw(a)mail.gmail.com> <7a0cd63f-8a83-4dc5-8763-63dcdae8d68a(a)leemhuis.info> User-Agent: Wanderlust/2.15.9 (Almost Unreal) Emacs/27.2 Mule/6.0 On Sun, 11 Feb 2024 18:19:25 +0100, Linux regression tracking (Thorsten Leemhuis) wrote: > > [CCing a few people] > > On 11.02.24 15:34, Ted Chang wrote: > > > > I noticed 6.7.4 has introduced a regression for the steam deck. The LCD > > steam deck can no longer probe the acp5x audio chipset anymore. This > > regression does not affect the 6.8.x series. I did not test kernel > > 6.7.3 because Opensuse tumbleweed skipped the update on my machine. > > Thx for your report. FWIW, problems like this can be caused by all > sorts of changes, but obviously those in the area of audio support > are most likely to cause this. There are just a few in the > v6.7.2..v6.7.4 range[1]. Among them a commit that is related to > acp5x, that's why I CCed its author as well (Venkata Prasad Potturu). > > Maybe one of the new recipients will have an idea. If not, you most > likely will have to bisect this and check if mainline is affected > as well.[2] > > Ciao, Thorsten > > [1] > $ git log --oneline v6.7.2..v6.7.4 sound/ > f3570675bf09af ASoC: codecs: wsa883x: fix PA volume control > 2f8e9b77ca2fea ASoC: codecs: lpass-wsa-macro: fix compander volume hack > 5b465d6384e4eb ASoC: codecs: wcd938x: fix headphones volume controls > 1673211a38012e ASoC: qcom: sc8280xp: limit speaker volumes > 242b5bffa23a9c ASoC: codecs: rtq9128: Fix TDM enable and DAI format control flow > 2c272ff9859601 ASoC: codecs: rtq9128: Fix PM_RUNTIME usage > 4a28302b2c681e ALSA: hda/conexant: Fix headset auto detect fail in cx8070 and SN6140 > e37a96941fdd53 ALSA: hda: intel-dspcfg: add filters for ARL-S and ARL > ffa3eea886c6fe ALSA: hda: Intel: add HDA_ARL PCI ID support > 4b6986b170f2f2 ASoC: amd: Add new dmi entries for acp5x platform This one is the only relevant change, I suppose. The machine matches with 'Valve Jupiter'. Interestingly, the system seems working with 6.8-rc3, so some piece might be missing. Or simply reverting this patch should fix. Takashi > e38ad4ace20b4d ALSA: hda: Refer to correct stream index at loops > a434c75e0671f9 soundwire: fix initializing sysfs for same devices on different buses > > [2] I'm working on a guide that describes what's needed: > https://www.leemhuis.info/files/misc/How%20to%20bisect%20a%20Linux%20kernel… > > > Steps to reproduce the problem > > 1. Obtain a steam deck > > 2. Install kernel 6.7.4 > > 3. Boot the device and you will see dummy output in gnome shell > > > > Observed kernel logs. > > > > [ 8.755614] cs35l41 spi-VLV1776:00: supply VA not found, using dummy regulator > > [ 8.760506] cs35l41 spi-VLV1776:00: supply VP not found, using dummy regulator > > [ 8.777148] cs35l41 spi-VLV1776:00: Cirrus Logic CS35L41 (35a40), Revision: B2 > > [ 8.777471] cs35l41 spi-VLV1776:01: supply VA not found, using dummy regulator > > [ 8.777532] cs35l41 spi-VLV1776:01: supply VP not found, using dummy regulator > > [ 8.777709] cs35l41 spi-VLV1776:01: Reset line busy, assuming shared reset > > [ 8.788465] cs35l41 spi-VLV1776:01: Cirrus Logic CS35L41 (35a40), Revision: B2 > > [ 8.877280] snd_hda_intel 0000:04:00.1: enabling device (0000 -> 0002) > > [ 8.877595] snd_hda_intel 0000:04:00.1: Handle vga_switcheroo audio client > > [ 8.889913] snd_acp_pci 0000:04:00.5: enabling device (0000 -> 0002) > > [ 8.890063] snd_acp_pci 0000:04:00.5: Unsupported device revision:0x50 > > [ 8.890129] snd_acp_pci: probe of 0000:04:00.5 failed with error -22 > > [ 8.906136] snd_hda_intel 0000:04:00.1: bound 0000:04:00.0 (ops amdgpu_dm_audio_component_bind_ops [amdgpu] > > > > > > No kernel module in use shown. > > > > 04:00.5 Multimedia controller [0480]: Advanced Micro Devices, Inc. [AMD] > > ACP/ACP3X/ACP6x Audio Coprocessor [1022:15e2] (rev 50) > > Subsystem: Valve Software Device [1e44:1776] > > Flags: fast devsel, IRQ 70, IOMMU group 4 > > Memory at 80380000 (32-bit, non-prefetchable) [size=256K] > > Capabilities: <access denied> > > Kernel modules: snd_pci_acp3x, snd_rn_pci_acp3x, snd_pci_acp5x, > > snd_pci_acp6x, snd_acp_pci, snd_rpl_pci_acp6x, snd_pci_ps, > > snd_sof_amd_renoir, snd_sof_amd_rembrandt, snd_sof_amd_vangogh, > > snd_sof_amd_acp63 > > > > > > Information for package kernel-default: > > --------------------------------------- > > Repository : openSUSE-Tumbleweed-Oss > > Name : kernel-default > > Version : 6.7.4-1.1 > > Arch : x86_64 > > Vendor : openSUSE > > Installed Size : 240.3 MiB > > Installed : Yes > > Status : up-to-date > > Source package : kernel-default-6.7.4-1.1.nosrc > > Upstream URL : https://www.kernel.org/ <https://www.kernel.org/> > > Summary : The Standard Kernel > > Description : > > The standard kernel for both uniprocessor and multiprocessor systems. > > > > > > Source Timestamp: 2024-02-06 05:32:37 +0000 > > GIT Revision: 01735a3e65287585dd830a6a3d33d909a4f9ae7f > > GIT Branch: stable > > > > Handle 0x0000, DMI type 0, 26 bytes > > BIOS Information > > Vendor: Valve > > Version: F7A0120 > > Release Date: 12/01/2023 > > Address: 0xE0000 > > Runtime Size: 128 kB > > BIOS Revision: 1.20 > > Firmware Revision: 1.16 > > > > #regzbot introduced: v6.7.2..v6.7.4 > >

3 5

[PATCH] ALSA: firewire-lib: fix to check cycle continuity
by Takashi Sakamoto 19 Feb '24

19 Feb '24

The local helper function to compare the given pair of cycle count evaluates them. If the left value is less than the right value, the function returns negative value. If the safe cycle is less than the current cycle, it is the case of cycle lost. However, it is not currently handled properly. This commit fixes the bug. Cc: <stable(a)vger.kernel.org> Fixes: 705794c53b00 ("ALSA: firewire-lib: check cycle continuity") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index a13c0b408aad..7be17bca257f 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -951,7 +951,7 @@ static int generate_tx_packet_descs(struct amdtp_stream *s, struct pkt_desc *des // to the reason. unsigned int safe_cycle = increment_ohci_cycle_count(next_cycle, IR_JUMBO_PAYLOAD_MAX_SKIP_CYCLES); - lost = (compare_ohci_cycle_count(safe_cycle, cycle) > 0); + lost = (compare_ohci_cycle_count(safe_cycle, cycle) < 0); } if (lost) { dev_err(&s->unit->device, "Detect discontinuity of cycle: %d %d\n", -- 2.40.1

2 1

[PATCH 0/4] ALSA: oxfw: add support for Miglia Harmony Audio
by Takashi Sakamoto 19 Feb '24

19 Feb '24

Hi, There used to be a company called Miglia Technologies Ltd. in the UK, and the company was selling a product called Harmony Audio in 2004. Upon obtaining the product by chance recently and examining its internals, it was revealed that it utilizes Oxford Semiconductor OXFW970 for its communication function in IEEE 1394 bus. This patchset adds support for the model. Takashi Sakamoto (4): ALSA: oxfw: use const qualifier for immutable argument ALSA: oxfw: support the case that AV/C Stream Format Information command is not available ALSA: firewire-lib: handle quirk to calculate payload quadlets as data block counter ALSA: oxfw: add support for Miglia Harmony Audio sound/firewire/amdtp-stream.c | 12 ++-- sound/firewire/amdtp-stream.h | 4 ++ sound/firewire/oxfw/oxfw-stream.c | 100 +++++++++++++++++++++++------- sound/firewire/oxfw/oxfw.c | 10 ++- sound/firewire/oxfw/oxfw.h | 7 ++- 5 files changed, 103 insertions(+), 30 deletions(-) -- 2.40.1

2 5

[PATCH] clk: constify the of_phandle_args argument of of_clk_provider
by Krzysztof Kozlowski 19 Feb '24

19 Feb '24

None of the implementations of the get() and get_hw() callbacks of "struct of_clk_provider" modify the contents of received of_phandle_args pointer. They treat it as read-only variable used to find the clock to return. Make obvious that implementations are not supposed to modify the of_phandle_args, by making it a pointer to const. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/clk/at91/pmc.c | 3 +- drivers/clk/at91/pmc.h | 3 +- drivers/clk/baikal-t1/clk-ccu-div.c | 2 +- drivers/clk/baikal-t1/clk-ccu-pll.c | 2 +- drivers/clk/bcm/clk-kona-setup.c | 2 +- drivers/clk/clk-axm5516.c | 2 +- drivers/clk/clk-cdce706.c | 2 +- drivers/clk/clk-cdce925.c | 2 +- drivers/clk/clk-k210.c | 2 +- drivers/clk/clk-lochnagar.c | 2 +- drivers/clk/clk-max77686.c | 2 +- drivers/clk/clk-max9485.c | 2 +- drivers/clk/clk-qoriq.c | 3 +- drivers/clk/clk-renesas-pcie.c | 2 +- drivers/clk/clk-rk808.c | 2 +- drivers/clk/clk-scpi.c | 2 +- drivers/clk/clk-si521xx.c | 2 +- drivers/clk/clk-si5341.c | 2 +- drivers/clk/clk-si5351.c | 4 +-- drivers/clk/clk-sparx5.c | 3 +- drivers/clk/clk-stm32f4.c | 2 +- drivers/clk/clk-versaclock3.c | 2 +- drivers/clk/clk-versaclock5.c | 2 +- drivers/clk/clk-versaclock7.c | 2 +- drivers/clk/clk.c | 30 ++++++++++--------- drivers/clk/hisilicon/clk-hi3660-stub.c | 2 +- drivers/clk/imx/clk-imx8qxp-lpcg.c | 2 +- drivers/clk/imx/clk-scu.c | 2 +- drivers/clk/imx/clk-scu.h | 2 +- drivers/clk/keystone/sci-clk.c | 3 +- drivers/clk/meson/meson-clkc-utils.c | 3 +- drivers/clk/meson/meson-clkc-utils.h | 3 +- drivers/clk/mvebu/common.c | 2 +- drivers/clk/mvebu/cp110-system-controller.c | 2 +- drivers/clk/mvebu/kirkwood.c | 2 +- drivers/clk/nxp/clk-lpc18xx-ccu.c | 2 +- drivers/clk/qcom/clk-rpm.c | 2 +- drivers/clk/qcom/clk-rpmh.c | 2 +- drivers/clk/qcom/clk-smd-rpm.c | 2 +- drivers/clk/qcom/clk-spmi-pmic-div.c | 2 +- drivers/clk/qcom/common.c | 2 +- drivers/clk/qcom/krait-cc.c | 2 +- drivers/clk/renesas/r9a06g032-clocks.c | 2 +- drivers/clk/renesas/renesas-cpg-mssr.c | 2 +- drivers/clk/renesas/rzg2l-cpg.c | 2 +- .../clk/starfive/clk-starfive-jh7100-audio.c | 3 +- drivers/clk/starfive/clk-starfive-jh7100.c | 3 +- .../clk/starfive/clk-starfive-jh7110-aon.c | 3 +- .../clk/starfive/clk-starfive-jh7110-isp.c | 3 +- .../clk/starfive/clk-starfive-jh7110-pll.c | 3 +- .../clk/starfive/clk-starfive-jh7110-stg.c | 3 +- .../clk/starfive/clk-starfive-jh7110-sys.c | 3 +- .../clk/starfive/clk-starfive-jh7110-vout.c | 3 +- drivers/clk/tegra/clk-bpmp.c | 2 +- drivers/clk/tegra/clk-tegra124.c | 2 +- drivers/clk/tegra/clk-tegra20.c | 2 +- drivers/clk/tegra/clk-tegra30.c | 2 +- drivers/clk/ti/clkctrl.c | 2 +- drivers/clk/ux500/u8500_of_clk.c | 4 +-- drivers/clk/versatile/clk-sp810.c | 2 +- drivers/media/platform/ti/omap3isp/isp.c | 3 +- drivers/phy/qualcomm/phy-qcom-qmp-combo.c | 6 ++-- include/linux/clk-provider.h | 28 ++++++++--------- include/linux/clk.h | 4 +-- sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 2 +- 65 files changed, 115 insertions(+), 95 deletions(-) diff --git a/drivers/clk/at91/pmc.c b/drivers/clk/at91/pmc.c index 5aa9c1f1c886..64ec624ffc15 100644 --- a/drivers/clk/at91/pmc.c +++ b/drivers/clk/at91/pmc.c @@ -44,7 +44,8 @@ int of_at91_get_clk_range(struct device_node *np, const char *propname, } EXPORT_SYMBOL_GPL(of_at91_get_clk_range); -struct clk_hw *of_clk_hw_pmc_get(struct of_phandle_args *clkspec, void *data) +struct clk_hw *of_clk_hw_pmc_get(const struct of_phandle_args *clkspec, + void *data) { unsigned int type = clkspec->args[0]; unsigned int idx = clkspec->args[1]; diff --git a/drivers/clk/at91/pmc.h b/drivers/clk/at91/pmc.h index 0f52e80bcd49..09ae26d0cece 100644 --- a/drivers/clk/at91/pmc.h +++ b/drivers/clk/at91/pmc.h @@ -126,7 +126,8 @@ struct pmc_data *pmc_data_allocate(unsigned int ncore, unsigned int nsystem, int of_at91_get_clk_range(struct device_node *np, const char *propname, struct clk_range *range); -struct clk_hw *of_clk_hw_pmc_get(struct of_phandle_args *clkspec, void *data); +struct clk_hw *of_clk_hw_pmc_get(const struct of_phandle_args *clkspec, + void *data); struct clk_hw * __init at91_clk_register_audio_pll_frac(struct regmap *regmap, const char *name, diff --git a/drivers/clk/baikal-t1/clk-ccu-div.c b/drivers/clk/baikal-t1/clk-ccu-div.c index 84555a00f950..97ea3d747a78 100644 --- a/drivers/clk/baikal-t1/clk-ccu-div.c +++ b/drivers/clk/baikal-t1/clk-ccu-div.c @@ -322,7 +322,7 @@ static int ccu_div_find_sys_regs(struct ccu_div_data *data) return 0; } -static struct clk_hw *ccu_div_of_clk_hw_get(struct of_phandle_args *clkspec, +static struct clk_hw *ccu_div_of_clk_hw_get(const struct of_phandle_args *clkspec, void *priv) { struct ccu_div_data *data = priv; diff --git a/drivers/clk/baikal-t1/clk-ccu-pll.c b/drivers/clk/baikal-t1/clk-ccu-pll.c index fce02ce77347..0e71d5967794 100644 --- a/drivers/clk/baikal-t1/clk-ccu-pll.c +++ b/drivers/clk/baikal-t1/clk-ccu-pll.c @@ -126,7 +126,7 @@ static int ccu_pll_find_sys_regs(struct ccu_pll_data *data) return 0; } -static struct clk_hw *ccu_pll_of_clk_hw_get(struct of_phandle_args *clkspec, +static struct clk_hw *ccu_pll_of_clk_hw_get(const struct of_phandle_args *clkspec, void *priv) { struct ccu_pll_data *data = priv; diff --git a/drivers/clk/bcm/clk-kona-setup.c b/drivers/clk/bcm/clk-kona-setup.c index 338558f6fbae..1149bcd7617f 100644 --- a/drivers/clk/bcm/clk-kona-setup.c +++ b/drivers/clk/bcm/clk-kona-setup.c @@ -773,7 +773,7 @@ static bool ccu_data_valid(struct ccu_data *ccu) } static struct clk_hw * -of_clk_kona_onecell_get(struct of_phandle_args *clkspec, void *data) +of_clk_kona_onecell_get(const struct of_phandle_args *clkspec, void *data) { struct ccu_data *ccu = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/clk-axm5516.c b/drivers/clk/clk-axm5516.c index 4a3462ee8f3e..458f44f4c08e 100644 --- a/drivers/clk/clk-axm5516.c +++ b/drivers/clk/clk-axm5516.c @@ -512,7 +512,7 @@ static struct axxia_clk *axmclk_clocks[] = { }; static struct clk_hw * -of_clk_axmclk_get(struct of_phandle_args *clkspec, void *unused) +of_clk_axmclk_get(const struct of_phandle_args *clkspec, void *unused) { unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/clk-cdce706.c b/drivers/clk/clk-cdce706.c index dd3d42d9ad86..1db9fc5d2af7 100644 --- a/drivers/clk/clk-cdce706.c +++ b/drivers/clk/clk-cdce706.c @@ -618,7 +618,7 @@ static int cdce706_register_clkouts(struct cdce706_dev_data *cdce) } static struct clk_hw * -of_clk_cdce_get(struct of_phandle_args *clkspec, void *data) +of_clk_cdce_get(const struct of_phandle_args *clkspec, void *data) { struct cdce706_dev_data *cdce = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/clk-cdce925.c b/drivers/clk/clk-cdce925.c index b0122093c6ff..48135ee578c5 100644 --- a/drivers/clk/clk-cdce925.c +++ b/drivers/clk/clk-cdce925.c @@ -576,7 +576,7 @@ static int cdce925_regmap_i2c_read(void *context, } static struct clk_hw * -of_clk_cdce925_get(struct of_phandle_args *clkspec, void *_data) +of_clk_cdce925_get(const struct of_phandle_args *clkspec, void *_data) { struct clk_cdce925_chip *data = _data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/clk-k210.c b/drivers/clk/clk-k210.c index 7b7329a907ab..2333ed3ed7a2 100644 --- a/drivers/clk/clk-k210.c +++ b/drivers/clk/clk-k210.c @@ -875,7 +875,7 @@ static inline void __init k210_register_clk_child(struct device_node *np, k210_register_clk(np, ksc, id, &parent_data, 1, 0); } -static struct clk_hw *k210_clk_hw_onecell_get(struct of_phandle_args *clkspec, +static struct clk_hw *k210_clk_hw_onecell_get(const struct of_phandle_args *clkspec, void *data) { struct k210_sysclk *ksc = data; diff --git a/drivers/clk/clk-lochnagar.c b/drivers/clk/clk-lochnagar.c index 5561a2c66b69..5b773787a225 100644 --- a/drivers/clk/clk-lochnagar.c +++ b/drivers/clk/clk-lochnagar.c @@ -215,7 +215,7 @@ static const struct clk_ops lochnagar_clk_ops = { }; static struct clk_hw * -lochnagar_of_clk_hw_get(struct of_phandle_args *clkspec, void *data) +lochnagar_of_clk_hw_get(const struct of_phandle_args *clkspec, void *data) { struct lochnagar_clk_priv *priv = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/clk-max77686.c b/drivers/clk/clk-max77686.c index 3727d5472450..3ce2453c116c 100644 --- a/drivers/clk/clk-max77686.c +++ b/drivers/clk/clk-max77686.c @@ -145,7 +145,7 @@ static const struct clk_ops max77686_clk_ops = { }; static struct clk_hw * -of_clk_max77686_get(struct of_phandle_args *clkspec, void *data) +of_clk_max77686_get(const struct of_phandle_args *clkspec, void *data) { struct max77686_clk_driver_data *drv_data = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/clk-max9485.c b/drivers/clk/clk-max9485.c index be9020b6c789..050246008514 100644 --- a/drivers/clk/clk-max9485.c +++ b/drivers/clk/clk-max9485.c @@ -246,7 +246,7 @@ static const struct max9485_clk max9485_clks[MAX9485_NUM_CLKS] = { }; static struct clk_hw * -max9485_of_clk_get(struct of_phandle_args *clkspec, void *data) +max9485_of_clk_get(const struct of_phandle_args *clkspec, void *data) { struct max9485_driver_data *drvdata = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/clk-qoriq.c b/drivers/clk/clk-qoriq.c index 4dcde305944c..95baeca8c031 100644 --- a/drivers/clk/clk-qoriq.c +++ b/drivers/clk/clk-qoriq.c @@ -1384,7 +1384,8 @@ static void __init core_pll_init(struct device_node *np) } } -static struct clk *clockgen_clk_get(struct of_phandle_args *clkspec, void *data) +static struct clk *clockgen_clk_get(const struct of_phandle_args *clkspec, + void *data) { struct clockgen *cg = data; struct clk *clk; diff --git a/drivers/clk/clk-renesas-pcie.c b/drivers/clk/clk-renesas-pcie.c index 53e21ac302e6..0fad48ff0f95 100644 --- a/drivers/clk/clk-renesas-pcie.c +++ b/drivers/clk/clk-renesas-pcie.c @@ -270,7 +270,7 @@ static void rs9_update_config(struct rs9_driver_data *rs9) } static struct clk_hw * -rs9_of_clk_get(struct of_phandle_args *clkspec, void *data) +rs9_of_clk_get(const struct of_phandle_args *clkspec, void *data) { struct rs9_driver_data *rs9 = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/clk-rk808.c b/drivers/clk/clk-rk808.c index f7412b137e5e..05e5047e7ce8 100644 --- a/drivers/clk/clk-rk808.c +++ b/drivers/clk/clk-rk808.c @@ -72,7 +72,7 @@ static const struct clk_ops rk808_clkout2_ops = { }; static struct clk_hw * -of_clk_rk808_get(struct of_phandle_args *clkspec, void *data) +of_clk_rk808_get(const struct of_phandle_args *clkspec, void *data) { struct rk808_clkout *rk808_clkout = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/clk-scpi.c b/drivers/clk/clk-scpi.c index 108b697bd317..8125e6ac7916 100644 --- a/drivers/clk/clk-scpi.c +++ b/drivers/clk/clk-scpi.c @@ -172,7 +172,7 @@ struct scpi_clk_data { }; static struct clk_hw * -scpi_of_clk_src_get(struct of_phandle_args *clkspec, void *data) +scpi_of_clk_src_get(const struct of_phandle_args *clkspec, void *data) { struct scpi_clk *sclk; struct scpi_clk_data *clk_data = data; diff --git a/drivers/clk/clk-si521xx.c b/drivers/clk/clk-si521xx.c index 4f7b74f889f1..5912bc5ad375 100644 --- a/drivers/clk/clk-si521xx.c +++ b/drivers/clk/clk-si521xx.c @@ -269,7 +269,7 @@ static void si521xx_diff_idx_to_reg_bit(const u16 chip_info, const int idx, } static struct clk_hw * -si521xx_of_clk_get(struct of_phandle_args *clkspec, void *data) +si521xx_of_clk_get(const struct of_phandle_args *clkspec, void *data) { struct si521xx *si = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/clk-si5341.c b/drivers/clk/clk-si5341.c index 6e8dd7387cfd..4acf1daabcdf 100644 --- a/drivers/clk/clk-si5341.c +++ b/drivers/clk/clk-si5341.c @@ -957,7 +957,7 @@ static int si5341_is_programmed_already(struct clk_si5341 *data) } static struct clk_hw * -of_clk_si5341_get(struct of_phandle_args *clkspec, void *_data) +of_clk_si5341_get(const struct of_phandle_args *clkspec, void *_data) { struct clk_si5341 *data = _data; unsigned int idx = clkspec->args[1]; diff --git a/drivers/clk/clk-si5351.c b/drivers/clk/clk-si5351.c index 4ce83c5265b8..3b325167c072 100644 --- a/drivers/clk/clk-si5351.c +++ b/drivers/clk/clk-si5351.c @@ -1390,7 +1390,7 @@ static int si5351_dt_parse(struct i2c_client *client, } static struct clk_hw * -si53351_of_clk_get(struct of_phandle_args *clkspec, void *data) +si53351_of_clk_get(const struct of_phandle_args *clkspec, void *data) { struct si5351_driver_data *drvdata = data; unsigned int idx = clkspec->args[0]; @@ -1409,7 +1409,7 @@ static int si5351_dt_parse(struct i2c_client *client, enum si5351_variant varian } static struct clk_hw * -si53351_of_clk_get(struct of_phandle_args *clkspec, void *data) +si53351_of_clk_get(const struct of_phandle_args *clkspec, void *data) { return NULL; } diff --git a/drivers/clk/clk-sparx5.c b/drivers/clk/clk-sparx5.c index 0fad0c1a0186..b352cdc289b6 100644 --- a/drivers/clk/clk-sparx5.c +++ b/drivers/clk/clk-sparx5.c @@ -229,7 +229,8 @@ static const struct clk_ops s5_pll_ops = { .recalc_rate = s5_pll_recalc_rate, }; -static struct clk_hw *s5_clk_hw_get(struct of_phandle_args *clkspec, void *data) +static struct clk_hw *s5_clk_hw_get(const struct of_phandle_args *clkspec, + void *data) { struct s5_clk_data *s5_clk = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/clk-stm32f4.c b/drivers/clk/clk-stm32f4.c index 07c13ebe327d..daa21cc8851b 100644 --- a/drivers/clk/clk-stm32f4.c +++ b/drivers/clk/clk-stm32f4.c @@ -872,7 +872,7 @@ static int stm32f4_rcc_lookup_clk_idx(u8 primary, u8 secondary) } static struct clk_hw * -stm32f4_rcc_lookup_clk(struct of_phandle_args *clkspec, void *data) +stm32f4_rcc_lookup_clk(const struct of_phandle_args *clkspec, void *data) { int i = stm32f4_rcc_lookup_clk_idx(clkspec->args[0], clkspec->args[1]); diff --git a/drivers/clk/clk-versaclock3.c b/drivers/clk/clk-versaclock3.c index 76d7ea1964c3..257160509310 100644 --- a/drivers/clk/clk-versaclock3.c +++ b/drivers/clk/clk-versaclock3.c @@ -964,7 +964,7 @@ static struct vc3_hw_data clk_mux[] = { } }; -static struct clk_hw *vc3_of_clk_get(struct of_phandle_args *clkspec, +static struct clk_hw *vc3_of_clk_get(const struct of_phandle_args *clkspec, void *data) { unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/clk-versaclock5.c b/drivers/clk/clk-versaclock5.c index 6d31cd54d7cf..9a25e96955bd 100644 --- a/drivers/clk/clk-versaclock5.c +++ b/drivers/clk/clk-versaclock5.c @@ -731,7 +731,7 @@ static const struct clk_ops vc5_clk_out_ops = { .get_parent = vc5_clk_out_get_parent, }; -static struct clk_hw *vc5_of_clk_get(struct of_phandle_args *clkspec, +static struct clk_hw *vc5_of_clk_get(const struct of_phandle_args *clkspec, void *data) { struct vc5_driver_data *vc5 = data; diff --git a/drivers/clk/clk-versaclock7.c b/drivers/clk/clk-versaclock7.c index f323263e32c3..8345a946c1ec 100644 --- a/drivers/clk/clk-versaclock7.c +++ b/drivers/clk/clk-versaclock7.c @@ -174,7 +174,7 @@ struct vc7_bank_src_map { } src; }; -static struct clk_hw *vc7_of_clk_get(struct of_phandle_args *clkspec, +static struct clk_hw *vc7_of_clk_get(const struct of_phandle_args *clkspec, void *data) { struct vc7_driver_data *vc7 = data; diff --git a/drivers/clk/clk.c b/drivers/clk/clk.c index 2253c154a824..cfb1dfeeb329 100644 --- a/drivers/clk/clk.c +++ b/drivers/clk/clk.c @@ -342,7 +342,7 @@ static struct clk_core *clk_core_lookup(const char *name) static int of_parse_clkspec(const struct device_node *np, int index, const char *name, struct of_phandle_args *out_args); static struct clk_hw * -of_clk_get_hw_from_clkspec(struct of_phandle_args *clkspec); +of_clk_get_hw_from_clkspec(const struct of_phandle_args *clkspec); #else static inline int of_parse_clkspec(const struct device_node *np, int index, const char *name, @@ -351,7 +351,7 @@ static inline int of_parse_clkspec(const struct device_node *np, int index, return -ENOENT; } static inline struct clk_hw * -of_clk_get_hw_from_clkspec(struct of_phandle_args *clkspec) +of_clk_get_hw_from_clkspec(const struct of_phandle_args *clkspec) { return ERR_PTR(-ENOENT); } @@ -4818,8 +4818,8 @@ struct of_clk_provider { struct list_head link; struct device_node *node; - struct clk *(*get)(struct of_phandle_args *clkspec, void *data); - struct clk_hw *(*get_hw)(struct of_phandle_args *clkspec, void *data); + struct clk *(*get)(const struct of_phandle_args *clkspec, void *data); + struct clk_hw *(*get_hw)(const struct of_phandle_args *clkspec, void *data); void *data; }; @@ -4830,20 +4830,22 @@ static const struct of_device_id __clk_of_table_sentinel static LIST_HEAD(of_clk_providers); static DEFINE_MUTEX(of_clk_mutex); -struct clk *of_clk_src_simple_get(struct of_phandle_args *clkspec, +struct clk *of_clk_src_simple_get(const struct of_phandle_args *clkspec, void *data) { return data; } EXPORT_SYMBOL_GPL(of_clk_src_simple_get); -struct clk_hw *of_clk_hw_simple_get(struct of_phandle_args *clkspec, void *data) +struct clk_hw *of_clk_hw_simple_get(const struct of_phandle_args *clkspec, + void *data) { return data; } EXPORT_SYMBOL_GPL(of_clk_hw_simple_get); -struct clk *of_clk_src_onecell_get(struct of_phandle_args *clkspec, void *data) +struct clk *of_clk_src_onecell_get(const struct of_phandle_args *clkspec, + void *data) { struct clk_onecell_data *clk_data = data; unsigned int idx = clkspec->args[0]; @@ -4858,7 +4860,7 @@ struct clk *of_clk_src_onecell_get(struct of_phandle_args *clkspec, void *data) EXPORT_SYMBOL_GPL(of_clk_src_onecell_get); struct clk_hw * -of_clk_hw_onecell_get(struct of_phandle_args *clkspec, void *data) +of_clk_hw_onecell_get(const struct of_phandle_args *clkspec, void *data) { struct clk_hw_onecell_data *hw_data = data; unsigned int idx = clkspec->args[0]; @@ -4881,7 +4883,7 @@ EXPORT_SYMBOL_GPL(of_clk_hw_onecell_get); * This function is *deprecated*. Use of_clk_add_hw_provider() instead. */ int of_clk_add_provider(struct device_node *np, - struct clk *(*clk_src_get)(struct of_phandle_args *clkspec, + struct clk *(*clk_src_get)(const struct of_phandle_args *clkspec, void *data), void *data) { @@ -4923,7 +4925,7 @@ EXPORT_SYMBOL_GPL(of_clk_add_provider); * @data: context pointer for @get callback. */ int of_clk_add_hw_provider(struct device_node *np, - struct clk_hw *(*get)(struct of_phandle_args *clkspec, + struct clk_hw *(*get)(const struct of_phandle_args *clkspec, void *data), void *data) { @@ -4997,7 +4999,7 @@ static struct device_node *get_clk_provider_node(struct device *dev) * Return: 0 on success or an errno on failure. */ int devm_of_clk_add_hw_provider(struct device *dev, - struct clk_hw *(*get)(struct of_phandle_args *clkspec, + struct clk_hw *(*get)(const struct of_phandle_args *clkspec, void *data), void *data) { @@ -5123,7 +5125,7 @@ static int of_parse_clkspec(const struct device_node *np, int index, static struct clk_hw * __of_clk_get_hw_from_provider(struct of_clk_provider *provider, - struct of_phandle_args *clkspec) + const struct of_phandle_args *clkspec) { struct clk *clk; @@ -5137,7 +5139,7 @@ __of_clk_get_hw_from_provider(struct of_clk_provider *provider, } static struct clk_hw * -of_clk_get_hw_from_clkspec(struct of_phandle_args *clkspec) +of_clk_get_hw_from_clkspec(const struct of_phandle_args *clkspec) { struct of_clk_provider *provider; struct clk_hw *hw = ERR_PTR(-EPROBE_DEFER); @@ -5166,7 +5168,7 @@ of_clk_get_hw_from_clkspec(struct of_phandle_args *clkspec) * providers, an input is a clock specifier data structure as returned * from the of_parse_phandle_with_args() function call. */ -struct clk *of_clk_get_from_provider(struct of_phandle_args *clkspec) +struct clk *of_clk_get_from_provider(const struct of_phandle_args *clkspec) { struct clk_hw *hw = of_clk_get_hw_from_clkspec(clkspec); diff --git a/drivers/clk/hisilicon/clk-hi3660-stub.c b/drivers/clk/hisilicon/clk-hi3660-stub.c index 3a653d54bee0..1dd7fb7cfa5b 100644 --- a/drivers/clk/hisilicon/clk-hi3660-stub.c +++ b/drivers/clk/hisilicon/clk-hi3660-stub.c @@ -108,7 +108,7 @@ static struct hi3660_stub_clk hi3660_stub_clks[HI3660_CLK_STUB_NUM] = { DEFINE_CLK_STUB(HI3660_CLK_STUB_DDR, 0x00040309, "clk-ddrc") }; -static struct clk_hw *hi3660_stub_clk_hw_get(struct of_phandle_args *clkspec, +static struct clk_hw *hi3660_stub_clk_hw_get(const struct of_phandle_args *clkspec, void *data) { unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/imx/clk-imx8qxp-lpcg.c b/drivers/clk/imx/clk-imx8qxp-lpcg.c index d0ccaa040225..feb2e4a61b6c 100644 --- a/drivers/clk/imx/clk-imx8qxp-lpcg.c +++ b/drivers/clk/imx/clk-imx8qxp-lpcg.c @@ -159,7 +159,7 @@ static const struct imx8qxp_ss_lpcg imx8qxp_ss_lsio = { #define IMX_LPCG_MAX_CLKS 8 -static struct clk_hw *imx_lpcg_of_clk_src_get(struct of_phandle_args *clkspec, +static struct clk_hw *imx_lpcg_of_clk_src_get(const struct of_phandle_args *clkspec, void *data) { struct clk_hw_onecell_data *hw_data = data; diff --git a/drivers/clk/imx/clk-scu.c b/drivers/clk/imx/clk-scu.c index e48a904c0013..d357dc337741 100644 --- a/drivers/clk/imx/clk-scu.c +++ b/drivers/clk/imx/clk-scu.c @@ -517,7 +517,7 @@ struct clk_hw *__imx_clk_scu(struct device *dev, const char *name, return hw; } -struct clk_hw *imx_scu_of_clk_src_get(struct of_phandle_args *clkspec, +struct clk_hw *imx_scu_of_clk_src_get(const struct of_phandle_args *clkspec, void *data) { unsigned int rsrc = clkspec->args[0]; diff --git a/drivers/clk/imx/clk-scu.h b/drivers/clk/imx/clk-scu.h index af7b697f51ca..38db6a1905d0 100644 --- a/drivers/clk/imx/clk-scu.h +++ b/drivers/clk/imx/clk-scu.h @@ -27,7 +27,7 @@ extern const struct imx_clk_scu_rsrc_table imx_clk_scu_rsrc_imx8qm; int imx_clk_scu_init(struct device_node *np, const struct imx_clk_scu_rsrc_table *data); -struct clk_hw *imx_scu_of_clk_src_get(struct of_phandle_args *clkspec, +struct clk_hw *imx_scu_of_clk_src_get(const struct of_phandle_args *clkspec, void *data); struct clk_hw *imx_clk_scu_alloc_dev(const char *name, const char * const *parents, diff --git a/drivers/clk/keystone/sci-clk.c b/drivers/clk/keystone/sci-clk.c index 35fe197dd303..ba27fb22f973 100644 --- a/drivers/clk/keystone/sci-clk.c +++ b/drivers/clk/keystone/sci-clk.c @@ -375,7 +375,8 @@ static int _cmp_sci_clk(const void *a, const void *b) * when a corresponding of_clk_get call is executed. Returns a pointer * to the TI SCI hw clock struct, or ERR_PTR value in failure. */ -static struct clk_hw *sci_clk_get(struct of_phandle_args *clkspec, void *data) +static struct clk_hw *sci_clk_get(const struct of_phandle_args *clkspec, + void *data) { struct sci_clk_provider *provider = data; struct sci_clk **clk; diff --git a/drivers/clk/meson/meson-clkc-utils.c b/drivers/clk/meson/meson-clkc-utils.c index 7370644e8092..904b482bf47b 100644 --- a/drivers/clk/meson/meson-clkc-utils.c +++ b/drivers/clk/meson/meson-clkc-utils.c @@ -8,7 +8,8 @@ #include <linux/module.h> #include "meson-clkc-utils.h" -struct clk_hw *meson_clk_hw_get(struct of_phandle_args *clkspec, void *clk_hw_data) +struct clk_hw *meson_clk_hw_get(const struct of_phandle_args *clkspec, + void *clk_hw_data) { const struct meson_clk_hw_data *data = clk_hw_data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/meson/meson-clkc-utils.h b/drivers/clk/meson/meson-clkc-utils.h index fe6f40728949..3889ead76a07 100644 --- a/drivers/clk/meson/meson-clkc-utils.h +++ b/drivers/clk/meson/meson-clkc-utils.h @@ -14,6 +14,7 @@ struct meson_clk_hw_data { unsigned int num; }; -struct clk_hw *meson_clk_hw_get(struct of_phandle_args *clkspec, void *clk_hw_data); +struct clk_hw *meson_clk_hw_get(const struct of_phandle_args *clkspec, + void *clk_hw_data); #endif diff --git a/drivers/clk/mvebu/common.c b/drivers/clk/mvebu/common.c index 785dbede4835..19cd62152175 100644 --- a/drivers/clk/mvebu/common.c +++ b/drivers/clk/mvebu/common.c @@ -199,7 +199,7 @@ struct clk_gating_ctrl { static struct clk_gating_ctrl *ctrl; static struct clk *clk_gating_get_src( - struct of_phandle_args *clkspec, void *data) + const struct of_phandle_args *clkspec, void *data) { int n; diff --git a/drivers/clk/mvebu/cp110-system-controller.c b/drivers/clk/mvebu/cp110-system-controller.c index 03c59bf22106..8f6e2dd3e044 100644 --- a/drivers/clk/mvebu/cp110-system-controller.c +++ b/drivers/clk/mvebu/cp110-system-controller.c @@ -192,7 +192,7 @@ static void cp110_unregister_gate(struct clk_hw *hw) kfree(to_cp110_gate_clk(hw)); } -static struct clk_hw *cp110_of_clk_get(struct of_phandle_args *clkspec, +static struct clk_hw *cp110_of_clk_get(const struct of_phandle_args *clkspec, void *data) { struct clk_hw_onecell_data *clk_data = data; diff --git a/drivers/clk/mvebu/kirkwood.c b/drivers/clk/mvebu/kirkwood.c index 8bc893df4736..b2b9a7a7a8fb 100644 --- a/drivers/clk/mvebu/kirkwood.c +++ b/drivers/clk/mvebu/kirkwood.c @@ -269,7 +269,7 @@ static const struct clk_muxing_soc_desc kirkwood_mux_desc[] __initconst = { }; static struct clk *clk_muxing_get_src( - struct of_phandle_args *clkspec, void *data) + const struct of_phandle_args *clkspec, void *data) { struct clk_muxing_ctrl *ctrl = (struct clk_muxing_ctrl *)data; int n; diff --git a/drivers/clk/nxp/clk-lpc18xx-ccu.c b/drivers/clk/nxp/clk-lpc18xx-ccu.c index ddb28b38f549..3d8299b7e790 100644 --- a/drivers/clk/nxp/clk-lpc18xx-ccu.c +++ b/drivers/clk/nxp/clk-lpc18xx-ccu.c @@ -109,7 +109,7 @@ static struct lpc18xx_clk_branch clk_branches[] = { {"base_sdio_clk", "sdio", CLK_SDIO, 0}, }; -static struct clk *lpc18xx_ccu_branch_clk_get(struct of_phandle_args *clkspec, +static struct clk *lpc18xx_ccu_branch_clk_get(const struct of_phandle_args *clkspec, void *data) { struct lpc18xx_branch_clk_data *clk_data = data; diff --git a/drivers/clk/qcom/clk-rpm.c b/drivers/clk/qcom/clk-rpm.c index 745026ef4d9c..b7e048237bc0 100644 --- a/drivers/clk/qcom/clk-rpm.c +++ b/drivers/clk/qcom/clk-rpm.c @@ -514,7 +514,7 @@ static const struct of_device_id rpm_clk_match_table[] = { }; MODULE_DEVICE_TABLE(of, rpm_clk_match_table); -static struct clk_hw *qcom_rpm_clk_hw_get(struct of_phandle_args *clkspec, +static struct clk_hw *qcom_rpm_clk_hw_get(const struct of_phandle_args *clkspec, void *data) { struct rpm_cc *rcc = data; diff --git a/drivers/clk/qcom/clk-rpmh.c b/drivers/clk/qcom/clk-rpmh.c index bb82abeed88f..8c2c6aa3a793 100644 --- a/drivers/clk/qcom/clk-rpmh.c +++ b/drivers/clk/qcom/clk-rpmh.c @@ -793,7 +793,7 @@ static const struct clk_rpmh_desc clk_rpmh_x1e80100 = { .num_clks = ARRAY_SIZE(x1e80100_rpmh_clocks), }; -static struct clk_hw *of_clk_rpmh_hw_get(struct of_phandle_args *clkspec, +static struct clk_hw *of_clk_rpmh_hw_get(const struct of_phandle_args *clkspec, void *data) { struct clk_rpmh_desc *rpmh = data; diff --git a/drivers/clk/qcom/clk-smd-rpm.c b/drivers/clk/qcom/clk-smd-rpm.c index 8602c02047d0..5d7e0cd2d450 100644 --- a/drivers/clk/qcom/clk-smd-rpm.c +++ b/drivers/clk/qcom/clk-smd-rpm.c @@ -1232,7 +1232,7 @@ static const struct of_device_id rpm_smd_clk_match_table[] = { }; MODULE_DEVICE_TABLE(of, rpm_smd_clk_match_table); -static struct clk_hw *qcom_smdrpm_clk_hw_get(struct of_phandle_args *clkspec, +static struct clk_hw *qcom_smdrpm_clk_hw_get(const struct of_phandle_args *clkspec, void *data) { const struct rpm_smd_clk_desc *desc = data; diff --git a/drivers/clk/qcom/clk-spmi-pmic-div.c b/drivers/clk/qcom/clk-spmi-pmic-div.c index f394031eb0e5..3f65aead519d 100644 --- a/drivers/clk/qcom/clk-spmi-pmic-div.c +++ b/drivers/clk/qcom/clk-spmi-pmic-div.c @@ -181,7 +181,7 @@ struct spmi_pmic_div_clk_cc { }; static struct clk_hw * -spmi_pmic_div_clk_hw_get(struct of_phandle_args *clkspec, void *data) +spmi_pmic_div_clk_hw_get(const struct of_phandle_args *clkspec, void *data) { struct spmi_pmic_div_clk_cc *cc = data; int idx = clkspec->args[0] - 1; /* Start at 1 instead of 0 */ diff --git a/drivers/clk/qcom/common.c b/drivers/clk/qcom/common.c index 75f09e6e057e..0ae6f8ea607a 100644 --- a/drivers/clk/qcom/common.c +++ b/drivers/clk/qcom/common.c @@ -220,7 +220,7 @@ static void qcom_cc_drop_protected(struct device *dev, struct qcom_cc *cc) } } -static struct clk_hw *qcom_cc_clk_hw_get(struct of_phandle_args *clkspec, +static struct clk_hw *qcom_cc_clk_hw_get(const struct of_phandle_args *clkspec, void *data) { struct qcom_cc *cc = data; diff --git a/drivers/clk/qcom/krait-cc.c b/drivers/clk/qcom/krait-cc.c index ae325f4e1047..181f4096a506 100644 --- a/drivers/clk/qcom/krait-cc.c +++ b/drivers/clk/qcom/krait-cc.c @@ -324,7 +324,7 @@ static struct clk_hw *krait_add_clks(struct device *dev, int id, bool unique_aux return pri_mux; } -static struct clk *krait_of_get(struct of_phandle_args *clkspec, void *data) +static struct clk *krait_of_get(const struct of_phandle_args *clkspec, void *data) { unsigned int idx = clkspec->args[0]; struct clk **clks = data; diff --git a/drivers/clk/renesas/r9a06g032-clocks.c b/drivers/clk/renesas/r9a06g032-clocks.c index c1348e2d450c..f8ea88663fca 100644 --- a/drivers/clk/renesas/r9a06g032-clocks.c +++ b/drivers/clk/renesas/r9a06g032-clocks.c @@ -733,7 +733,7 @@ struct r9a06g032_clk_gate { #define to_r9a06g032_gate(_hw) container_of(_hw, struct r9a06g032_clk_gate, hw) -static int create_add_module_clock(struct of_phandle_args *clkspec, +static int create_add_module_clock(const struct of_phandle_args *clkspec, struct device *dev) { struct clk *clk; diff --git a/drivers/clk/renesas/renesas-cpg-mssr.c b/drivers/clk/renesas/renesas-cpg-mssr.c index 1b421b809796..c172444ee766 100644 --- a/drivers/clk/renesas/renesas-cpg-mssr.c +++ b/drivers/clk/renesas/renesas-cpg-mssr.c @@ -273,7 +273,7 @@ static const struct clk_ops cpg_mstp_clock_ops = { }; static -struct clk *cpg_mssr_clk_src_twocell_get(struct of_phandle_args *clkspec, +struct clk *cpg_mssr_clk_src_twocell_get(const struct of_phandle_args *clkspec, void *data) { unsigned int clkidx = clkspec->args[1]; diff --git a/drivers/clk/renesas/rzg2l-cpg.c b/drivers/clk/renesas/rzg2l-cpg.c index 3d2daa4ba2a4..7464a298b2e5 100644 --- a/drivers/clk/renesas/rzg2l-cpg.c +++ b/drivers/clk/renesas/rzg2l-cpg.c @@ -1052,7 +1052,7 @@ rzg2l_cpg_pll_clk_register(const struct cpg_core_clk *core, } static struct clk -*rzg2l_cpg_clk_src_twocell_get(struct of_phandle_args *clkspec, +*rzg2l_cpg_clk_src_twocell_get(const struct of_phandle_args *clkspec, void *data) { unsigned int clkidx = clkspec->args[1]; diff --git a/drivers/clk/starfive/clk-starfive-jh7100-audio.c b/drivers/clk/starfive/clk-starfive-jh7100-audio.c index 1fcf4e62f347..79feba75ea43 100644 --- a/drivers/clk/starfive/clk-starfive-jh7100-audio.c +++ b/drivers/clk/starfive/clk-starfive-jh7100-audio.c @@ -84,7 +84,8 @@ static const struct jh71x0_clk_data jh7100_audclk_data[] = { JH7100_AUDCLK_AUDIO_12288), }; -static struct clk_hw *jh7100_audclk_get(struct of_phandle_args *clkspec, void *data) +static struct clk_hw *jh7100_audclk_get(const struct of_phandle_args *clkspec, + void *data) { struct jh71x0_clk_priv *priv = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/starfive/clk-starfive-jh7100.c b/drivers/clk/starfive/clk-starfive-jh7100.c index 03f6f26a15d8..b7e43f4847d2 100644 --- a/drivers/clk/starfive/clk-starfive-jh7100.c +++ b/drivers/clk/starfive/clk-starfive-jh7100.c @@ -267,7 +267,8 @@ static const struct jh71x0_clk_data jh7100_clk_data[] __initconst = { JH71X0_GATE(JH7100_CLK_SYSERR_APB, "syserr_apb", 0, JH7100_CLK_APB2_BUS), }; -static struct clk_hw *jh7100_clk_get(struct of_phandle_args *clkspec, void *data) +static struct clk_hw *jh7100_clk_get(const struct of_phandle_args *clkspec, + void *data) { struct jh71x0_clk_priv *priv = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/starfive/clk-starfive-jh7110-aon.c b/drivers/clk/starfive/clk-starfive-jh7110-aon.c index 418efdad719b..d05da683a544 100644 --- a/drivers/clk/starfive/clk-starfive-jh7110-aon.c +++ b/drivers/clk/starfive/clk-starfive-jh7110-aon.c @@ -54,7 +54,8 @@ static const struct jh71x0_clk_data jh7110_aonclk_data[] = { JH71X0_GATE(JH7110_AONCLK_RTC_CAL, "rtc_cal", 0, JH7110_AONCLK_OSC), }; -static struct clk_hw *jh7110_aonclk_get(struct of_phandle_args *clkspec, void *data) +static struct clk_hw *jh7110_aonclk_get(const struct of_phandle_args *clkspec, + void *data) { struct jh71x0_clk_priv *priv = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/starfive/clk-starfive-jh7110-isp.c b/drivers/clk/starfive/clk-starfive-jh7110-isp.c index 929b8788279e..518c4a8d4200 100644 --- a/drivers/clk/starfive/clk-starfive-jh7110-isp.c +++ b/drivers/clk/starfive/clk-starfive-jh7110-isp.c @@ -75,7 +75,8 @@ static inline int jh7110_isp_top_rst_init(struct jh71x0_clk_priv *priv) return reset_control_deassert(top_rsts); } -static struct clk_hw *jh7110_ispclk_get(struct of_phandle_args *clkspec, void *data) +static struct clk_hw *jh7110_ispclk_get(const struct of_phandle_args *clkspec, + void *data) { struct jh71x0_clk_priv *priv = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/starfive/clk-starfive-jh7110-pll.c b/drivers/clk/starfive/clk-starfive-jh7110-pll.c index 3598390e8fd0..3c53459c554b 100644 --- a/drivers/clk/starfive/clk-starfive-jh7110-pll.c +++ b/drivers/clk/starfive/clk-starfive-jh7110-pll.c @@ -442,7 +442,8 @@ static const struct clk_ops jh7110_pll_ops = { .debug_init = jh7110_pll_debug_init, }; -static struct clk_hw *jh7110_pll_get(struct of_phandle_args *clkspec, void *data) +static struct clk_hw *jh7110_pll_get(const struct of_phandle_args *clkspec, + void *data) { struct jh7110_pll_priv *priv = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/starfive/clk-starfive-jh7110-stg.c b/drivers/clk/starfive/clk-starfive-jh7110-stg.c index dafcb7190592..0fba5de830e9 100644 --- a/drivers/clk/starfive/clk-starfive-jh7110-stg.c +++ b/drivers/clk/starfive/clk-starfive-jh7110-stg.c @@ -75,7 +75,8 @@ static const struct jh71x0_clk_data jh7110_stgclk_data[] = { JH71X0_GATE(JH7110_STGCLK_DMA1P_AHB, "dma1p_ahb", 0, JH7110_STGCLK_STG_AXIAHB), }; -static struct clk_hw *jh7110_stgclk_get(struct of_phandle_args *clkspec, void *data) +static struct clk_hw *jh7110_stgclk_get(const struct of_phandle_args *clkspec, + void *data) { struct jh71x0_clk_priv *priv = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/starfive/clk-starfive-jh7110-sys.c b/drivers/clk/starfive/clk-starfive-jh7110-sys.c index 8f5e5abfa178..6ea25b959091 100644 --- a/drivers/clk/starfive/clk-starfive-jh7110-sys.c +++ b/drivers/clk/starfive/clk-starfive-jh7110-sys.c @@ -323,7 +323,8 @@ static const struct jh71x0_clk_data jh7110_sysclk_data[] __initconst = { JH7110_SYSCLK_OSC), }; -static struct clk_hw *jh7110_sysclk_get(struct of_phandle_args *clkspec, void *data) +static struct clk_hw *jh7110_sysclk_get(const struct of_phandle_args *clkspec, + void *data) { struct jh71x0_clk_priv *priv = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/starfive/clk-starfive-jh7110-vout.c b/drivers/clk/starfive/clk-starfive-jh7110-vout.c index 10cc1ec43925..7f69a4d75126 100644 --- a/drivers/clk/starfive/clk-starfive-jh7110-vout.c +++ b/drivers/clk/starfive/clk-starfive-jh7110-vout.c @@ -80,7 +80,8 @@ static int jh7110_vout_top_rst_init(struct jh71x0_clk_priv *priv) return reset_control_deassert(top_rst); } -static struct clk_hw *jh7110_voutclk_get(struct of_phandle_args *clkspec, void *data) +static struct clk_hw *jh7110_voutclk_get(const struct of_phandle_args *clkspec, + void *data) { struct jh71x0_clk_priv *priv = data; unsigned int idx = clkspec->args[0]; diff --git a/drivers/clk/tegra/clk-bpmp.c b/drivers/clk/tegra/clk-bpmp.c index 7bfba0afd778..9c53ab9b8a00 100644 --- a/drivers/clk/tegra/clk-bpmp.c +++ b/drivers/clk/tegra/clk-bpmp.c @@ -654,7 +654,7 @@ static void tegra_bpmp_unregister_clocks(struct tegra_bpmp *bpmp) clk_hw_unregister(&bpmp->clocks[i]->hw); } -static struct clk_hw *tegra_bpmp_clk_of_xlate(struct of_phandle_args *clkspec, +static struct clk_hw *tegra_bpmp_clk_of_xlate(const struct of_phandle_args *clkspec, void *data) { unsigned int id = clkspec->args[0], i; diff --git a/drivers/clk/tegra/clk-tegra124.c b/drivers/clk/tegra/clk-tegra124.c index 6c46592d794e..87f1e5bc5c0e 100644 --- a/drivers/clk/tegra/clk-tegra124.c +++ b/drivers/clk/tegra/clk-tegra124.c @@ -1502,7 +1502,7 @@ static void __init tegra124_132_clock_init_pre(struct device_node *np) writel(plld_base, clk_base + PLLD_BASE); } -static struct clk *tegra124_clk_src_onecell_get(struct of_phandle_args *clkspec, +static struct clk *tegra124_clk_src_onecell_get(const struct of_phandle_args *clkspec, void *data) { struct clk_hw *hw; diff --git a/drivers/clk/tegra/clk-tegra20.c b/drivers/clk/tegra/clk-tegra20.c index 2c58ce25af75..8a7e17d3379a 100644 --- a/drivers/clk/tegra/clk-tegra20.c +++ b/drivers/clk/tegra/clk-tegra20.c @@ -1069,7 +1069,7 @@ static const struct of_device_id pmc_match[] __initconst = { static bool tegra20_car_initialized; -static struct clk *tegra20_clk_src_onecell_get(struct of_phandle_args *clkspec, +static struct clk *tegra20_clk_src_onecell_get(const struct of_phandle_args *clkspec, void *data) { struct clk_hw *parent_hw; diff --git a/drivers/clk/tegra/clk-tegra30.c b/drivers/clk/tegra/clk-tegra30.c index 82a8cb9545eb..4317c68cdeb4 100644 --- a/drivers/clk/tegra/clk-tegra30.c +++ b/drivers/clk/tegra/clk-tegra30.c @@ -1273,7 +1273,7 @@ static struct tegra_audio_clk_info tegra30_audio_plls[] = { static bool tegra30_car_initialized; -static struct clk *tegra30_clk_src_onecell_get(struct of_phandle_args *clkspec, +static struct clk *tegra30_clk_src_onecell_get(const struct of_phandle_args *clkspec, void *data) { struct clk_hw *hw; diff --git a/drivers/clk/ti/clkctrl.c b/drivers/clk/ti/clkctrl.c index 607e34d8e289..31df051c116c 100644 --- a/drivers/clk/ti/clkctrl.c +++ b/drivers/clk/ti/clkctrl.c @@ -221,7 +221,7 @@ static const struct clk_ops omap4_clkctrl_clk_ops = { .init = omap2_init_clk_clkdm, }; -static struct clk_hw *_ti_omap4_clkctrl_xlate(struct of_phandle_args *clkspec, +static struct clk_hw *_ti_omap4_clkctrl_xlate(const struct of_phandle_args *clkspec, void *data) { struct omap_clkctrl_provider *provider = data; diff --git a/drivers/clk/ux500/u8500_of_clk.c b/drivers/clk/ux500/u8500_of_clk.c index 8e2f6c65db2a..aba91ca5ca07 100644 --- a/drivers/clk/ux500/u8500_of_clk.c +++ b/drivers/clk/ux500/u8500_of_clk.c @@ -26,7 +26,7 @@ static struct clk_hw *clkout_clk[2]; #define PRCC_KCLK_STORE(clk, base, bit) \ prcc_kclk[(base * PRCC_PERIPHS_PER_CLUSTER) + bit] = clk -static struct clk *ux500_twocell_get(struct of_phandle_args *clkspec, +static struct clk *ux500_twocell_get(const struct of_phandle_args *clkspec, void *data) { struct clk **clk_data = data; @@ -71,7 +71,7 @@ static const char * const u8500_clkout_parents[] = { "clk009", }; -static struct clk_hw *ux500_clkout_get(struct of_phandle_args *clkspec, +static struct clk_hw *ux500_clkout_get(const struct of_phandle_args *clkspec, void *data) { u32 id, source, divider; diff --git a/drivers/clk/versatile/clk-sp810.c b/drivers/clk/versatile/clk-sp810.c index 45adac1b4630..5fd21c13f546 100644 --- a/drivers/clk/versatile/clk-sp810.c +++ b/drivers/clk/versatile/clk-sp810.c @@ -68,7 +68,7 @@ static const struct clk_ops clk_sp810_timerclken_ops = { .set_parent = clk_sp810_timerclken_set_parent, }; -static struct clk *clk_sp810_timerclken_of_get(struct of_phandle_args *clkspec, +static struct clk *clk_sp810_timerclken_of_get(const struct of_phandle_args *clkspec, void *data) { struct clk_sp810 *sp810 = data; diff --git a/drivers/media/platform/ti/omap3isp/isp.c b/drivers/media/platform/ti/omap3isp/isp.c index 1cda23244c7b..e2c78f9ab797 100644 --- a/drivers/media/platform/ti/omap3isp/isp.c +++ b/drivers/media/platform/ti/omap3isp/isp.c @@ -281,7 +281,8 @@ static const struct clk_ops isp_xclk_ops = { static const char *isp_xclk_parent_name = "cam_mclk"; -static struct clk *isp_xclk_src_get(struct of_phandle_args *clkspec, void *data) +static struct clk *isp_xclk_src_get(const struct of_phandle_args *clkspec, + void *data) { unsigned int idx = clkspec->args[0]; struct isp_device *isp = data; diff --git a/drivers/phy/qualcomm/phy-qcom-qmp-combo.c b/drivers/phy/qualcomm/phy-qcom-qmp-combo.c index b6908a03da58..46a77df32dff 100644 --- a/drivers/phy/qualcomm/phy-qcom-qmp-combo.c +++ b/drivers/phy/qualcomm/phy-qcom-qmp-combo.c @@ -3136,7 +3136,8 @@ static const struct clk_ops qmp_dp_link_clk_ops = { .recalc_rate = qmp_dp_link_clk_recalc_rate, }; -static struct clk_hw *qmp_dp_clks_hw_get(struct of_phandle_args *clkspec, void *data) +static struct clk_hw *qmp_dp_clks_hw_get(const struct of_phandle_args *clkspec, + void *data) { struct qmp_combo *qmp = data; unsigned int idx = clkspec->args[0]; @@ -3177,7 +3178,8 @@ static int phy_dp_clks_register(struct qmp_combo *qmp, struct device_node *np) return 0; } -static struct clk_hw *qmp_combo_clk_hw_get(struct of_phandle_args *clkspec, void *data) +static struct clk_hw *qmp_combo_clk_hw_get(const struct of_phandle_args *clkspec, + void *data) { struct qmp_combo *qmp = data; diff --git a/include/linux/clk-provider.h b/include/linux/clk-provider.h index 1293c38ddb7f..d58313368924 100644 --- a/include/linux/clk-provider.h +++ b/include/linux/clk-provider.h @@ -1532,25 +1532,25 @@ struct clk_hw_onecell_data { #ifdef CONFIG_OF int of_clk_add_provider(struct device_node *np, - struct clk *(*clk_src_get)(struct of_phandle_args *args, + struct clk *(*clk_src_get)(const struct of_phandle_args *args, void *data), void *data); int of_clk_add_hw_provider(struct device_node *np, - struct clk_hw *(*get)(struct of_phandle_args *clkspec, + struct clk_hw *(*get)(const struct of_phandle_args *clkspec, void *data), void *data); int devm_of_clk_add_hw_provider(struct device *dev, - struct clk_hw *(*get)(struct of_phandle_args *clkspec, + struct clk_hw *(*get)(const struct of_phandle_args *clkspec, void *data), void *data); void of_clk_del_provider(struct device_node *np); -struct clk *of_clk_src_simple_get(struct of_phandle_args *clkspec, +struct clk *of_clk_src_simple_get(const struct of_phandle_args *clkspec, void *data); -struct clk_hw *of_clk_hw_simple_get(struct of_phandle_args *clkspec, +struct clk_hw *of_clk_hw_simple_get(const struct of_phandle_args *clkspec, void *data); -struct clk *of_clk_src_onecell_get(struct of_phandle_args *clkspec, void *data); -struct clk_hw *of_clk_hw_onecell_get(struct of_phandle_args *clkspec, +struct clk *of_clk_src_onecell_get(const struct of_phandle_args *clkspec, void *data); +struct clk_hw *of_clk_hw_onecell_get(const struct of_phandle_args *clkspec, void *data); int of_clk_parent_fill(struct device_node *np, const char **parents, unsigned int size); @@ -1560,21 +1560,21 @@ int of_clk_detect_critical(struct device_node *np, int index, #else /* !CONFIG_OF */ static inline int of_clk_add_provider(struct device_node *np, - struct clk *(*clk_src_get)(struct of_phandle_args *args, + struct clk *(*clk_src_get)(const struct of_phandle_args *args, void *data), void *data) { return 0; } static inline int of_clk_add_hw_provider(struct device_node *np, - struct clk_hw *(*get)(struct of_phandle_args *clkspec, + struct clk_hw *(*get)(const struct of_phandle_args *clkspec, void *data), void *data) { return 0; } static inline int devm_of_clk_add_hw_provider(struct device *dev, - struct clk_hw *(*get)(struct of_phandle_args *clkspec, + struct clk_hw *(*get)(const struct of_phandle_args *clkspec, void *data), void *data) { @@ -1583,22 +1583,22 @@ static inline int devm_of_clk_add_hw_provider(struct device *dev, static inline void of_clk_del_provider(struct device_node *np) {} static inline struct clk *of_clk_src_simple_get( - struct of_phandle_args *clkspec, void *data) + const struct of_phandle_args *clkspec, void *data) { return ERR_PTR(-ENOENT); } static inline struct clk_hw * -of_clk_hw_simple_get(struct of_phandle_args *clkspec, void *data) +of_clk_hw_simple_get(const struct of_phandle_args *clkspec, void *data) { return ERR_PTR(-ENOENT); } static inline struct clk *of_clk_src_onecell_get( - struct of_phandle_args *clkspec, void *data) + const struct of_phandle_args *clkspec, void *data) { return ERR_PTR(-ENOENT); } static inline struct clk_hw * -of_clk_hw_onecell_get(struct of_phandle_args *clkspec, void *data) +of_clk_hw_onecell_get(const struct of_phandle_args *clkspec, void *data) { return ERR_PTR(-ENOENT); } diff --git a/include/linux/clk.h b/include/linux/clk.h index 06f1b292f8a0..8e3c44ff89da 100644 --- a/include/linux/clk.h +++ b/include/linux/clk.h @@ -1141,7 +1141,7 @@ static inline struct clk *clk_get_optional(struct device *dev, const char *id) #if defined(CONFIG_OF) && defined(CONFIG_COMMON_CLK) struct clk *of_clk_get(struct device_node *np, int index); struct clk *of_clk_get_by_name(struct device_node *np, const char *name); -struct clk *of_clk_get_from_provider(struct of_phandle_args *clkspec); +struct clk *of_clk_get_from_provider(const struct of_phandle_args *clkspec); #else static inline struct clk *of_clk_get(struct device_node *np, int index) { @@ -1152,7 +1152,7 @@ static inline struct clk *of_clk_get_by_name(struct device_node *np, { return ERR_PTR(-ENOENT); } -static inline struct clk *of_clk_get_from_provider(struct of_phandle_args *clkspec) +static inline struct clk *of_clk_get_from_provider(const struct of_phandle_args *clkspec) { return ERR_PTR(-ENOENT); } diff --git a/sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c b/sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c index e758411603be..b846707baa89 100644 --- a/sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c +++ b/sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c @@ -106,7 +106,7 @@ static const struct clk_ops clk_vote_q6dsp_ops = { }; -static struct clk_hw *q6dsp_of_clk_hw_get(struct of_phandle_args *clkspec, +static struct clk_hw *q6dsp_of_clk_hw_get(const struct of_phandle_args *clkspec, void *data) { struct q6dsp_cc *cc = data; -- 2.34.1

7 10

[PATCH AUTOSEL 5.10 04/16] ASoC: codecs: wcd934x: drop unneeded regulator include
by Sasha Levin 18 Feb '24

18 Feb '24

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 35314e39dabcfb256832654ad0e856a9fba744bd ] Driver does not use any regulator code, so drop redundant include of regulator/consumer.h header. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Link: https://msgid.link/r/20240117151208.1219755-3-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wcd934x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 104751ac6cd1..dbed3a646524 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -13,7 +13,6 @@ #include <linux/of.h> #include <linux/platform_device.h> #include <linux/regmap.h> -#include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/slimbus.h> #include <sound/pcm_params.h> -- 2.43.0

2 1

[PATCH] conf: USB-Audio: Add more Scarlett devices to the IEC958 blacklist
by Geoffrey D. Bennett 18 Feb '24

18 Feb '24

The Scarlett Solo and 2i2 don't have S/PDIF outputs. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- src/conf/cards/USB-Audio.conf | 3 +++ 1 file changed, 3 insertions(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 80631b2e..05384a32 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -66,8 +66,11 @@ USB-Audio.pcm.iec958_device { "Plantronics USB Headset" 999 "Plantronics Wireless Audio" 999 "SB WoW Headset" 999 + "Scarlett 2i2 4th Gen" 999 "Scarlett 2i2 USB" 999 "Scarlett 2i4 USB" 999 + "Scarlett Solo 4th Gen" 999 + "Scarlett Solo USB" 999 "Sennheiser USB headset" 999 "SWTOR Gaming Headset by Razer" 999 "ThinkStation P620 Main" 999 -- 2.43.0

1 0

[PATCH v16 00/50] Introduce QC USB SND audio offloading support
by Wesley Cheng 17 Feb '24

17 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 v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) 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 (11): 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 (39): 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: Introduce SND kcontrols to track USB offloading state 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: Add USB offloading capable kcontrol ALSA: usb-audio: qcom: Use card and PCM index from QMI request 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 | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 39 +- 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 | 182 ++ sound/soc/Kconfig | 9 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 10 + sound/soc/qcom/Makefile | 2 + 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 | 408 ++++ sound/soc/qcom/sm8250.c | 15 +- sound/soc/qcom/usb_offload_utils.c | 51 + sound/soc/qcom/usb_offload_utils.h | 26 + sound/soc/soc-usb.c | 660 ++++++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 2 +- sound/usb/card.c | 109 + sound/usb/card.h | 15 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 6 + sound/usb/qcom/mixer_usb_offload.c | 65 + sound/usb/qcom/mixer_usb_offload.h | 17 + sound/usb/qcom/qc_audio_offload.c | 1914 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 50 files changed, 6485 insertions(+), 167 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/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/mixer_usb_offload.c create mode 100644 sound/usb/qcom/mixer_usb_offload.h 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

1 51

[v4 PATCH] ALSA: virtio: Fix "Coverity: virtsnd_kctl_tlv_op(): Uninitialized variables" warning.
by Aiswarya Cyriac 16 Feb '24

16 Feb '24

This commit fixes the following warning when building virtio_snd driver. " *** CID 1583619: Uninitialized variables (UNINIT) sound/virtio/virtio_kctl.c:294 in virtsnd_kctl_tlv_op() 288 289 break; 290 } 291 292 kfree(tlv); 293 vvv CID 1583619: Uninitialized variables (UNINIT) vvv Using uninitialized value "rc". 294 return rc; 295 } 296 297 /** 298 * virtsnd_kctl_get_enum_items() - Query items for the ENUMERATED element type. 299 * @snd: VirtIO sound device. " This warning is caused by the absence of the "default" branch in the switch-block, and is a false positive because the kernel calls virtsnd_kctl_tlv_op() only with values for op_flag processed in this block. Also, this commit unifies the cleanup path for all possible control paths in the callback function. Signed-off-by: Anton Yakovlev <anton.yakovlev(a)opensynergy.com> Signed-off-by: Aiswarya Cyriac <aiswarya.cyriac(a)opensynergy.com> Reported-by: coverity-bot <keescook+coverity-bot(a)chromium.org> Addresses-Coverity-ID: 1583619 ("Uninitialized variables") Fixes: d6568e3de42d ("ALSA: virtio: add support for audio controls") --- sound/virtio/virtio_kctl.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/virtio/virtio_kctl.c b/sound/virtio/virtio_kctl.c index 0c6ac74aca1e..7aa79c05b464 100644 --- a/sound/virtio/virtio_kctl.c +++ b/sound/virtio/virtio_kctl.c @@ -253,8 +253,8 @@ static int virtsnd_kctl_tlv_op(struct snd_kcontrol *kcontrol, int op_flag, tlv = kzalloc(size, GFP_KERNEL); if (!tlv) { - virtsnd_ctl_msg_unref(msg); - return -ENOMEM; + rc = -ENOMEM; + goto on_msg_unref; } sg_init_one(&sg, tlv, size); @@ -281,14 +281,25 @@ static int virtsnd_kctl_tlv_op(struct snd_kcontrol *kcontrol, int op_flag, hdr->hdr.code = cpu_to_le32(VIRTIO_SND_R_CTL_TLV_COMMAND); - if (copy_from_user(tlv, utlv, size)) + if (copy_from_user(tlv, utlv, size)) { rc = -EFAULT; - else + goto on_msg_unref; + } else { rc = virtsnd_ctl_msg_send(snd, msg, &sg, NULL, false); + } break; + default: + rc = -EINVAL; + /* We never get here - we listed all values for op_flag */ + WARN_ON(1); + goto on_msg_unref; } + kfree(tlv); + return rc; +on_msg_unref: + virtsnd_ctl_msg_unref(msg); kfree(tlv); return rc; -- 2.43.2

3 4

[PATCH] soundwire: bus_type: make sdw_bus_type const
by Ricardo B. Marliere 16 Feb '24

16 Feb '24

Since commit d492cc2573a0 ("driver core: device.h: make struct bus_type a const *"), the driver core can properly handle constant struct bus_type, move the sdw_bus_type variable to be a constant structure as well, placing it into read-only memory which can not be modified at runtime. Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Suggested-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Ricardo B. Marliere <ricardo(a)marliere.net> --- drivers/soundwire/bus_type.c | 2 +- include/linux/soundwire/sdw_type.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/bus_type.c b/drivers/soundwire/bus_type.c index 9fa93bb923d7..fd65b2360fc1 100644 --- a/drivers/soundwire/bus_type.c +++ b/drivers/soundwire/bus_type.c @@ -72,7 +72,7 @@ int sdw_slave_uevent(const struct device *dev, struct kobj_uevent_env *env) return 0; } -struct bus_type sdw_bus_type = { +const struct bus_type sdw_bus_type = { .name = "soundwire", .match = sdw_bus_match, }; diff --git a/include/linux/soundwire/sdw_type.h b/include/linux/soundwire/sdw_type.h index d8c27f1e5559..b445f7200f06 100644 --- a/include/linux/soundwire/sdw_type.h +++ b/include/linux/soundwire/sdw_type.h @@ -4,7 +4,7 @@ #ifndef __SOUNDWIRE_TYPES_H #define __SOUNDWIRE_TYPES_H -extern struct bus_type sdw_bus_type; +extern const struct bus_type sdw_bus_type; extern struct device_type sdw_slave_type; extern struct device_type sdw_master_type; --- base-commit: 9282cfa2eb080e3bbb95f488af35618b614cdf76 change-id: 20240213-bus_cleanup-soundwire-8b56c6862ef1 Best regards, -- Ricardo B. Marliere <ricardo(a)marliere.net>

3 2

[PATCH] ALSA: usb-audio: More relaxed check of MIDI jack names
by Takashi Iwai 15 Feb '24

15 Feb '24

The USB audio driver tries to retrieve MIDI jack name strings that can be used for rawmidi substream names and sequencer port names, but its checking is too strict: often the firmware provides the jack info for unexpected directions, and then we miss the info although it's present. In this patch, the code to extract the jack info is changed to allow both in and out directions in a single loop. That is, the former two functions to obtain the descriptor pointers for jack in and out are changed to a single function that returns iJack of the corresponding jack ID, no matter which direction is used. It's a code simplification at the same time as well as the fix. Fixes: eb596e0fd13c ("ALSA: usb-audio: generate midi streaming substream names from jack names") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/midi.c | 77 +++++++++++++++++------------------------------- 1 file changed, 27 insertions(+), 50 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 6b0993258e03..c1f2e5a03de9 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1742,50 +1742,44 @@ static void snd_usbmidi_get_port_info(struct snd_rawmidi *rmidi, int number, } } -static struct usb_midi_in_jack_descriptor *find_usb_in_jack_descriptor( - struct usb_host_interface *hostif, uint8_t jack_id) +/* return iJack for the corresponding jackID */ +static int find_usb_ijack(struct usb_host_interface *hostif, uint8_t jack_id) { unsigned char *extra = hostif->extra; int extralen = hostif->extralen; + struct usb_descriptor_header *h; + struct usb_midi_out_jack_descriptor *outjd; + struct usb_midi_in_jack_descriptor *injd; + size_t sz; while (extralen > 4) { - struct usb_midi_in_jack_descriptor *injd = - (struct usb_midi_in_jack_descriptor *)extra; + h = (struct usb_descriptor_header *)extra; + if (h->bDescriptorType != USB_DT_CS_INTERFACE) + goto next; - if (injd->bLength >= sizeof(*injd) && - injd->bDescriptorType == USB_DT_CS_INTERFACE && - injd->bDescriptorSubtype == UAC_MIDI_IN_JACK && - injd->bJackID == jack_id) - return injd; - if (!extra[0]) - break; - extralen -= extra[0]; - extra += extra[0]; - } - return NULL; -} - -static struct usb_midi_out_jack_descriptor *find_usb_out_jack_descriptor( - struct usb_host_interface *hostif, uint8_t jack_id) -{ - unsigned char *extra = hostif->extra; - int extralen = hostif->extralen; - - while (extralen > 4) { - struct usb_midi_out_jack_descriptor *outjd = - (struct usb_midi_out_jack_descriptor *)extra; - - if (outjd->bLength >= sizeof(*outjd) && - outjd->bDescriptorType == USB_DT_CS_INTERFACE && + outjd = (struct usb_midi_out_jack_descriptor *)h; + if (h->bLength >= sizeof(*outjd) && outjd->bDescriptorSubtype == UAC_MIDI_OUT_JACK && - outjd->bJackID == jack_id) - return outjd; + outjd->bJackID == jack_id) { + sz = USB_DT_MIDI_OUT_SIZE(outjd->bNrInputPins); + if (outjd->bLength < sz) + goto next; + return *(extra + sz - 1); + } + + injd = (struct usb_midi_in_jack_descriptor *)h; + if (injd->bLength >= sizeof(*injd) && + injd->bDescriptorSubtype == UAC_MIDI_IN_JACK && + injd->bJackID == jack_id) + return injd->iJack; + +next: if (!extra[0]) break; extralen -= extra[0]; extra += extra[0]; } - return NULL; + return 0; } static void snd_usbmidi_init_substream(struct snd_usb_midi *umidi, @@ -1796,13 +1790,10 @@ static void snd_usbmidi_init_substream(struct snd_usb_midi *umidi, const char *name_format; struct usb_interface *intf; struct usb_host_interface *hostif; - struct usb_midi_in_jack_descriptor *injd; - struct usb_midi_out_jack_descriptor *outjd; uint8_t jack_name_buf[32]; uint8_t *default_jack_name = "MIDI"; uint8_t *jack_name = default_jack_name; uint8_t iJack; - size_t sz; int res; struct snd_rawmidi_substream *substream = @@ -1816,21 +1807,7 @@ static void snd_usbmidi_init_substream(struct snd_usb_midi *umidi, intf = umidi->iface; if (intf && jack_id >= 0) { hostif = intf->cur_altsetting; - iJack = 0; - if (stream != SNDRV_RAWMIDI_STREAM_OUTPUT) { - /* in jacks connect to outs */ - outjd = find_usb_out_jack_descriptor(hostif, jack_id); - if (outjd) { - sz = USB_DT_MIDI_OUT_SIZE(outjd->bNrInputPins); - if (outjd->bLength >= sz) - iJack = *(((uint8_t *) outjd) + sz - sizeof(uint8_t)); - } - } else { - /* and out jacks connect to ins */ - injd = find_usb_in_jack_descriptor(hostif, jack_id); - if (injd) - iJack = injd->iJack; - } + iJack = find_usb_ijack(hostif, jack_id); if (iJack != 0) { res = usb_string(umidi->dev, iJack, jack_name_buf, ARRAY_SIZE(jack_name_buf)); -- 2.35.3

1 0

[ASOC][tlv320aic3110] Error resetting registers
by Jesús Jiménez Sánchez 15 Feb '24

15 Feb '24

Hello, I am having an issue with the tlv320aic3110 codec. I get these two errors during boot up in my embedded device: [ 10.079068] tlv320aic31xx-codec 0-0018: Unable to sync registers 0x1d-0x1d. -6 [ 10.161668] tlv320aic31xx-codec 0-0018: Failed to restore cache: -6 According to the datasheet, 0x1d can be "Table 7-69. Page 0 / Register 29 (0x1D): Codec Interface Control 2" or "Table 7-136. Page 1 / Register 1 (0x01) Through Page 1 / Register 29 (0x1D): Reserved". After about a minute, there are also these errors: [ 66.858820] tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -16 [ 66.943057] tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -16 [ 67.092200] tlv320aic31xx-codec 0-0018: ASoC: POST_PMU: HPL Driver event failed: -16 [ 67.383700] tlv320aic31xx-codec 0-0018: ASoC: error at soc_component_read_no_lock on tlv320aic31xx-codec.0-0018: -16 [ 67.482291] tlv320aic31xx-codec 0-0018: ASoC: error at soc_component_read_no_lock on tlv320aic31xx-codec.0-0018: -16 [ 67.545386] tlv320aic31xx-codec 0-0018: ASoC: POST_PMU: SPL ClassD event failed: -16 [ 67.823211] tlv320aic31xx-codec 0-0018: ASoC: error at soc_component_read_no_lock on tlv320aic31xx-codec.0-0018: -16 [ 67.895536] tlv320aic31xx-codec 0-0018: ASoC: error at soc_component_read_no_lock on tlv320aic31xx-codec.0-0018: -16 And finally, after suspend/resume, I get these last errors: [ 5981.210062] tlv320aic31xx-codec 0-0018: ASoC: POST_PMD: HPL Driver event failed: -16 [ 5981.210103] tlv320aic31xx-codec 0-0018: ASoC: POST_PMD: SPL ClassD event failed: -16 [ 5981.210170] v1v8_ao: Underflow of regulator enable count [ 5981.210184] Failed to disable DVDD: -EINVAL [ 5981.223267] tlv320aic31xx-codec 0-0018: Unable to sync registers 0x1d-0x1d. -6 [ 5981.223303] tlv320aic31xx-codec 0-0018: Failed to restore cache: -6 [ 5981.233551] tlv320aic31xx-codec 0-0018: aic31xx_wait_bits: Failed! 0x25 was 0xc1855ddb expected 0x20 (-16, 0x20, 0 us) [ 5981.233591] tlv320aic31xx-codec 0-0018: ASoC: POST_PMU: HPL Driver event failed: -1 [ 5981.233669] tlv320aic31xx-codec 0-0018: ASoC: POST_PMU: SPL ClassD event failed: -16 Here are all the relevant parts of our device tree: / { sound0: sound@0 { compatible = "simple-audio-card"; simple-audio-card,name = "Sound"; simple-audio-card,widgets = "Microphone", "Microphone Jack", "Headphone", "Headphone Jack", "Speaker", "Speaker"; simple-audio-card,routing = "MIC1LP", "Microphone Jack", "MIC1RP", "Microphone Jack", "MIC1LP", "MICBIAS", "MIC1RP", "MICBIAS", "Headphone Jack", "HPL", "Headphone Jack", "HPR", "Speaker", "SPL", "Speaker", "SPR"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&sound0_master>; simple-audio-card,frame-master = <&sound0_master>; simple-audio-card,bitclock-inversion; simple-audio-card,convert-rate = <48000>; sound0_master: simple-audio-card,cpu { sound-dai = <&i2s1>; system-clock-frequency = <12000000>; }; simple-audio-card,codec { sound-dai = <&codec>; system-clock-frequency = <12000000>; }; }; }; &pinctrl { codec_pins_a: codec_pins_a { pins1 { pinmux = <STM32_PINMUX('H', 2, GPIO)>; /* AUDIO_PWR_EN1 */ }; pins2 { pinmux = <STM32_PINMUX('C', 0, AF6)>, /* CODEC_I2S1_SDO */ <STM32_PINMUX('A', 6, AF5)>, /* CODEC_I2S1_SDI */ <STM32_PINMUX('B', 1, AF5)>, /* CODEC_I2S1_BITCLK / I2S1_CK */ <STM32_PINMUX('B', 0, AF5)>, /* I2S1_MCK */ <STM32_PINMUX('A', 4, AF5)>; /* CODEC_I2S1_WORDCLK / I2S1_WS */ }; }; codec_sleep_pins_a: codec_sleep_pins_a { pins { pinmux = <STM32_PINMUX('H', 2, ANALOG)>, /* AUDIO_PWR_EN1 */ <STM32_PINMUX('C', 0, ANALOG)>, /* CODEC_I2S1_SDO */ <STM32_PINMUX('A', 6, ANALOG)>, /* CODEC_I2S1_SDI */ <STM32_PINMUX('B', 1, ANALOG)>, /* CODEC_I2S1_BITCLK / I2S1_CK */ <STM32_PINMUX('B', 0, ANALOG)>, /* I2S1_MCK */ <STM32_PINMUX('A', 4, ANALOG)>; /* CODEC_I2S1_WORDCLK / I2S1_WS */ }; }; }; &i2s1 { status = "okay"; pinctrl-names = "default", "sleep"; pinctrl-0 = <&codec_pins_a>; pinctrl-1 = <&codec_sleep_pins_a>; /* Sleep pins */ clock-names = "pclk", "i2sclk", "x8k", "x11k"; clocks = <&rcc SPI1>, <&rcc SPI1_K>, <&scmi_clk CK_SCMI_PLL3_Q>, <&scmi_clk CK_SCMI_PLL3_R>; }; &i2c1 { pinctrl-names = "default", "sleep"; pinctrl-0 = <&i2c1_pins_a>; pinctrl-1 = <&i2c1_sleep_pins_a>; i2c-scl-rising-time-ns = <96>; i2c-scl-falling-time-ns = <3>; clock-frequency = <100000>; status = "okay"; /* spare dmas for other usage */ /delete-property/dmas; /delete-property/dma-names; codec: codec@18 { compatible = "ti,tlv320aic3110"; reg = <0x18>; #sound-dai-cells = <0>; status = "okay"; reset-gpios = <&gpiof 6 (GPIO_ACTIVE_LOW | GPIO_OPEN_DRAIN)>; ai31xx-micbias-vg = <MICBIAS_2_0V>; /* Regulators */ HPVDD-supply = <&scmi_v3v3_sw>; /* 3V3_CODEC */ SPRVDD-supply = <&v5v0_ao>; /* 5V0_CODEC */ SPLVDD-supply = <&v5v0_ao>; /* 5V0_CODEC */ AVDD-supply = <&scmi_v3v3_sw>; /* 3V3_CODEC */ IOVDD-supply = <&scmi_v3v3_sw>; /* 3V3_CODEC */ DVDD-supply = <&scmi_v1v8_periph>; /* 1V8_CODEC */ }; }; I have been trying to find a solution for a few days but I haven't been able to find anything important. Is there anything I could do to maybe get closer to the root cause of these errors? Has anyone seen something like this before? Let me know if you need any other information from me that might help. Thanks in advance!

1 0

[GIT PULL] ASoC fixes for v6.8-rc4
by Mark Brown 15 Feb '24

15 Feb '24

The following changes since commit e81fdba0208666b65bafeaba814874b4d6e5edde: ALSA: Various fixes for Cirrus Logic CS35L56 support (2024-02-01 17:45:32 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.8-rc4 for you to fetch changes up to 0db0c1770834f39e11a2902e20e1f11a482f4465: ASoC: cs35l56: Workaround for ACPI with broken spk-id-gpios property (2024-02-13 19:37:25 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.8 A relatively large set of fixes and quirk additions here but they're all driver specific, people seem to be back into the swing of things after the holidays. This is all driver specific and much of it fairly minor. ---------------------------------------------------------------- Alexey Khoroshilov (1): ASoC: rt5645: Fix deadlock in rt5645_jack_detect_work() Amadeusz Sławiński (1): ASoC: Intel: avs: Fix dynamic port assignment when TDM is set Arnd Bergmann (1): ASoC: q6dsp: fix event handler prototype Attila Tőkés (1): ASoC: amd: yc: Fix non-functional mic on Lenovo 82UU Cezary Rojewski (1): ASoC: Intel: avs: Fix pci_probe() error path Charles Keepax (2): ASoC: cs42l43: Handle error from devm_pm_runtime_enable ASoC: cs42l43: Add system suspend ops to disable IRQ Cristian Ciocaltea (1): ASoC: SOF: amd: Fix locking in ACP IRQ handler Curtis Malainey (1): ASoC: SOF: IPC3: fix message bounds on ipc ops Dan Carpenter (1): ASoC: cs35l56: fix reversed if statement in cs35l56_dspwait_asp1tx_put() Gergo Koteles (1): ASoC: tas2781: add module parameter to tascodec_init() Hans de Goede (4): ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards ASoC: Intel: cht_bsw_rt5645: Cleanup codec_name handling ASoC: rt5645: Make LattePanda board DMI match more precise ASoC: rt5645: Add DMI quirk for inverted jack-detect on MeeGoPad T8 Jeffrey Hugo (1): ASoC: dt-bindings: google,sc7280-herobrine: Drop bouncing @codeaurora Mario Limonciello (1): ASoC: amd: yc: Add DMI quirk for Lenovo Ideapad Pro 5 16ARP8 Mark Brown (2): ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT ASoC: SOF: Intel: pci-tgl/lnl: Change default paths Peter Ujfalusi (3): ASoC: SOF: Intel: pci-tgl: Change the default paths and firmware names ASoC: SOF: Intel: pci-lnl: Change the topology path to intel/sof-ipc4-tplg ASoC: SOF: ipc4-pcm: Workaround for crashed firmware on system suspend Ranjani Sridharan (1): ASoC: SOF: ipc3-topology: Fix pipeline tear down logic Richard Fitzgerald (3): ASoC: cs35l56: Fix deadlock in ASP1 mixer register initialization ASoC: cs35l56: Remove default from IRQ1_CFG register ASoC: cs35l56: Workaround for ACPI with broken spk-id-gpios property .../bindings/sound/google,sc7280-herobrine.yaml | 1 - include/sound/tas2781.h | 1 + sound/pci/hda/tas2781_hda_i2c.c | 2 +- sound/soc/amd/yc/acp6x-mach.c | 14 ++ sound/soc/codecs/cs35l56-shared.c | 1 - sound/soc/codecs/cs35l56.c | 246 ++++++++++++++------- sound/soc/codecs/cs35l56.h | 2 +- sound/soc/codecs/cs42l43.c | 48 +++- sound/soc/codecs/rt5645.c | 26 +++ sound/soc/codecs/tas2781-comlib.c | 3 +- sound/soc/codecs/tas2781-i2c.c | 2 +- sound/soc/intel/avs/core.c | 3 + sound/soc/intel/avs/topology.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 3 +- sound/soc/intel/boards/bytcht_da7213.c | 3 +- sound/soc/intel/boards/bytcht_es8316.c | 3 +- sound/soc/intel/boards/bytcr_rt5640.c | 3 +- sound/soc/intel/boards/bytcr_rt5651.c | 3 +- sound/soc/intel/boards/bytcr_wm5102.c | 3 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 7 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 3 +- sound/soc/qcom/qdsp6/q6apm-dai.c | 8 +- sound/soc/sof/amd/acp-ipc.c | 2 + sound/soc/sof/amd/acp.c | 17 +- sound/soc/sof/intel/pci-lnl.c | 2 +- sound/soc/sof/intel/pci-tgl.c | 64 +++--- sound/soc/sof/ipc3-topology.c | 55 +++-- sound/soc/sof/ipc3.c | 2 +- sound/soc/sof/ipc4-pcm.c | 13 +- 29 files changed, 378 insertions(+), 164 deletions(-)

2 1

[PATCH 0/2] ALSA: struct bus_type cleanup
by Ricardo B. Marliere 15 Feb '24

15 Feb '24

This series is part of an effort to cleanup the users of the driver core, as can be seen in many recent patches authored by Greg across the tree (e.g. [1]). --- [1]: https://lore.kernel.org/lkml/?q=f%3Agregkh%40linuxfoundation.org+s%3A%22mak… To: Johannes Berg <johannes(a)sipsolutions.net> To: Jaroslav Kysela <perex(a)perex.cz> To: Takashi Iwai <tiwai(a)suse.com> Cc: <linuxppc-dev(a)lists.ozlabs.org> Cc: <alsa-devel(a)alsa-project.org> Cc: <linux-sound(a)vger.kernel.org> Cc: <linux-kernel(a)vger.kernel.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Ricardo B. Marliere <ricardo(a)marliere.net> --- Ricardo B. Marliere (2): ALSA: aoa: make soundbus_bus_type const ALSA: seq: make snd_seq_bus_type const sound/aoa/soundbus/core.c | 2 +- sound/core/seq_device.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) --- base-commit: d7bf73809849463f76de42aad62c850305dd6c5d change-id: 20240214-bus_cleanup-alsa-1d05ffc6507b Best regards, -- Ricardo B. Marliere <ricardo(a)marliere.net>

3 4

Codec stops working after sleep and resume
by Jesús Jiménez Sánchez 15 Feb '24

15 Feb '24

I am using the tlv320aic31xx codec in my embedded device and I've noticed that, after forcing the unit into sleep and then, resume, I get a bunch of these errors: ``` tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: aic31xx_wait_bits: Failed! 0x25 was 0xcc239ca8 expected 0x0 (-108, 0x20, 0 us) tlv320aic31xx-codec 0-0018: ASoC: POST_PMD: HPL Driver event failed: -1 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: POST_PMD: SPL ClassD event failed: -108 tlv320aic31xx-codec 0-0018: ASoC: POST_PMD: SPR ClassD event failed: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 tlv320aic31xx-codec 0-0018: ASoC: error at snd_soc_component_update_bits on tlv320aic31xx-codec.0-0018: -108 ``` When I look at he output of `dmesg`, I found this: ``` ------------[ cut here ]------------ WARNING: CPU: 0 PID: 1506 at drivers/i2c/i2c-core.h:54 __i2c_transfer+0xac/0xb4 i2c i2c-0: Transfer while suspended Modules linked in: rfcomm stm32_adc stm32_timer_trigger stm32_lptimer_trigger sha256_generic libsha256 sha256_arm snd_soc_simple_card snd_soc_simple_card_utils cfg80211 stm32_crc32 hci_uart btqca stm32_cryp libdes crypto_engine snd_soc_stm32_i2s snd_soc_tlv320aic31xx bluetooth stm32_adc_core snd_soc_core atmel_mxt_ts snd_pcm_dmaengine snd_pcm ecdh_generic snd_timer snd ecc sgm4151x_charger soundcore libaes [last unloaded: wlan] CPU: 0 PID: 1506 Comm: sh Tainted: G O 5.15.67 #1 Hardware name: STM32 (Device Tree Support) [<c010e8cc>] (unwind_backtrace) from [<c010c120>] (show_stack+0x10/0x14) [<c010c120>] (show_stack) from [<c011dc1c>] (__warn+0xec/0x148) [<c011dc1c>] (__warn) from [<c0bed42c>] (warn_slowpath_fmt+0x78/0xac) [<c0bed42c>] (warn_slowpath_fmt) from [<c092ecdc>] (__i2c_transfer+0xac/0xb4) [<c092ecdc>] (__i2c_transfer) from [<c092ed80>] (i2c_transfer+0x9c/0x138) [<c092ed80>] (i2c_transfer) from [<c06f5a40>] (regmap_i2c_read+0x5c/0x98) [<c06f5a40>] (regmap_i2c_read) from [<c06f0e2c>] (_regmap_raw_read+0x10c/0x2a8) [<c06f0e2c>] (_regmap_raw_read) from [<c06f1008>] (_regmap_bus_read+0x40/0x6c) [<c06f1008>] (_regmap_bus_read) from [<c06ef9c4>] (_regmap_read+0x60/0x14c) [<c06ef9c4>] (_regmap_read) from [<c06effdc>] (_regmap_update_bits+0xb4/0xf4) [<c06effdc>] (_regmap_update_bits) from [<c06f00c0>] (_regmap_select_page+0xa4/0x108) [<c06f00c0>] (_regmap_select_page) from [<c06f07a8>] (_regmap_raw_write_impl+0x684/0x968) [<c06f07a8>] (_regmap_raw_write_impl) from [<c06f0b04>] (_regmap_bus_raw_write+0x78/0xa0) [<c06f0b04>] (_regmap_bus_raw_write) from [<c06efe40>] (_regmap_write+0x4c/0x134) [<c06efe40>] (_regmap_write) from [<c06f0010>] (_regmap_update_bits+0xe8/0xf4) [<c06f0010>] (_regmap_update_bits) from [<c06f1820>] (regmap_update_bits_base+0x50/0x74) [<c06f1820>] (regmap_update_bits_base) from [<bf0d0980>] (snd_soc_component_update_bits+0x78/0x124 [snd_soc_core]) [<bf0d0980>] (snd_soc_component_update_bits [snd_soc_core]) from [<bf0daa20>] (dapm_seq_run_coalesced+0x14c/0x1a4 [snd_soc_core]) [<bf0daa20>] (dapm_seq_run_coalesced [snd_soc_core]) from [<bf0dab68>] (dapm_seq_run+0xf0/0x350 [snd_soc_core]) [<bf0dab68>] (dapm_seq_run [snd_soc_core]) from [<bf0db170>] (dapm_power_widgets+0x3a8/0x530 [snd_soc_core]) [<bf0db170>] (dapm_power_widgets [snd_soc_core]) from [<bf0cc214>] (snd_soc_dapm_sync+0x48/0x54 [snd_soc_core]) [<bf0cc214>] (snd_soc_dapm_sync [snd_soc_core]) from [<bf0c652c>] (snd_soc_suspend+0xf0/0x230 [snd_soc_core]) [<bf0c652c>] (snd_soc_suspend [snd_soc_core]) from [<c06df188>] (dpm_run_callback+0x48/0x134) [<c06df188>] (dpm_run_callback) from [<c06df58c>] (__device_suspend+0x11c/0x4c4) [<c06df58c>] (__device_suspend) from [<c06e1c10>] (dpm_suspend+0x148/0x2cc) [<c06e1c10>] (dpm_suspend) from [<c06e20e8>] (dpm_suspend_start+0x64/0x6c) [<c06e20e8>] (dpm_suspend_start) from [<c0159f00>] (suspend_devices_and_enter+0x150/0x2b4) [<c0159f00>] (suspend_devices_and_enter) from [<c0bede38>] (enter_state+0x204/0x250) [<c0bede38>] (enter_state) from [<c015a0a4>] (pm_suspend+0x40/0x98) [<c015a0a4>] (pm_suspend) from [<c0158ae4>] (state_store+0x68/0xc8) [<c0158ae4>] (state_store) from [<c0330a68>] (kernfs_fop_write_iter+0x10c/0x1cc) [<c0330a68>] (kernfs_fop_write_iter) from [<c028ea40>] (vfs_write+0x298/0x45c) [<c028ea40>] (vfs_write) from [<c028ed4c>] (ksys_write+0x60/0xe4) [<c028ed4c>] (ksys_write) from [<c0100060>] (ret_fast_syscall+0x0/0x48) Exception stack(0xcffcbfa8 to 0xcffcbff0) bfa0: 00000004 005ac7d8 00000001 005ac7d8 00000004 00000000 bfc0: 00000004 005ac7d8 00000001 00000004 b6f01680 bee896d8 005a8448 005a8590 bfe0: 005a6dfc bee89690 004fcda4 b6dcb4f8 ---[ end trace 75457fd792755cae ]--- ``` I've been trying for a few days, but I can't find the source of the problem or a way to solve it. Is there anything I might be missing? I assume the problem is that it's trying to write to the registers after the suspend has happened, but I haven't been able to prove that either, and I wouldn't know how to fix that. Could that be what's happening here? Thanks.

1 1

Fwd: No sound on Lenovo P1 Gen 6 (00:1f.3 Multimedia audio controller: Intel Corporation Device 51ca (rev 01))
by Jeff Chua 15 Feb '24

15 Feb '24

[Reposted to ALSA Deveopment] I've just about everything but still couldn't get souns on the Lenovo P1 Gen 6. I've even tried SOF modules in the kernel but not working either. I'm just using standard alsa without other sound utils. Alsa has served me fine on IBM/Lenovo for many years. Latest on Lenovo X1 Gen 10. Sound works fine on the headphone plugged in to the headphone jack. Attached is my kernel .config ... # lspci 00:1f.3 Multimedia audio controller: Intel Corporation Device 51ca (rev 01) Subsystem: Lenovo Device 2317 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 Interrupt: pin A routed to IRQ 130 Region 0: Memory at 6452b30000 (64-bit, non-prefetchable) [size=16K] Region 4: Memory at 6452a00000 (64-bit, non-prefetchable) [size=1M] Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0-,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst+ PME-Enable- DSel=0 DScale=0 PME- Capabilities: [80] Vendor Specific Information: Len=14 <?> Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee3c000 Data: 0020 Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel # lsmod ... snd_ctl_led 20480 0 snd_hda_codec_realtek 147456 1 snd_hda_codec_generic 73728 1 snd_hda_codec_realtek snd_hda_codec_hdmi 61440 1 snd_hda_intel 36864 0 snd_intel_dspcfg 12288 1 snd_hda_intel snd_hda_codec 114688 4 snd_hda_codec_generic,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec_realtek snd_hwdep 12288 1 snd_hda_codec snd_hda_core 69632 5 snd_hda_codec_generic,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec,snd_hda_codec_realtek # dmesg ... snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC287: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 snd_hda_codec_realtek hdaudioC0D0: inputs: snd_hda_codec_realtek hdaudioC0D0: Mic=0x19 snd_hda_codec_realtek hdaudioC0D0: cs35l41_generic_fixup snd_hda_codec_realtek hdaudioC0D0: cs35l41_generic_fixup snd_hda_codec_realtek hdaudioC0D0: cs35l41_generic_fixup # uname -a Linux p1 6.8.0-rc4 #10 SMP PREEMPT_DYNAMIC Thu Feb 15 09:09:25 SGT 2024 x86_64 GNU/Linux I've the latest linux, alsa-lib, alsa-utils, alsa-plugins inatall ... # linux-kernel commit 8d3dea210042f54b952b481838c1e7dfc4ec751d (HEAD -> master, origin/master, origin/HEAD) Merge: 6004b044f771 8fa507083388 Author: Linus Torvalds <torvalds(a)linux-foundation.org> Date: Wed Feb 14 16:06:31 2024 -0800 And all the latest alsa-lib, alsa-utils, alsa-plugins install from git. # alsa-lib commit d8ce72f2561f23293ad0d98d30060c4a80a74c36 (HEAD -> master, origin/master, origin/HEAD) Author: Jaroslav Kysela <perex(a)perex.cz> Date: Fri Feb 9 11:21:04 2024 +0100 # alsa-utils commit 7cae8880234149b42d99abd603b387f91a8c7013 (HEAD -> master, tag: v1.2.11, origin/master, origin/HEAD) Author: Jaroslav Kysela <perex(a)perex.cz> Date: Mon Jan 29 13:57:30 2024 +0100 Release v1.2.11 # alsa-plugins commit 873a2966d22cd2956dc3c844fdc5d5573178502a (HEAD -> master, origin/master, origin/HEAD) Author: Takashi Iwai <tiwai(a)suse.de> Date: Sat Oct 7 09:20:41 2023 +0200 # cat /proc/cmdline BOOT_IMAGE=/linux/bzc1 root=/dev/nvme0n1p2 ro resume=/dev/nvme0n1p12 nosplash master=nvme0n1p fstype=f2fs reboot=acpi loop.max_part=63 modules=\"i915\" drm_kms_helper.poll=0 nomodeset vga=755 i915.modeset=1 i915.enable_fbc=1 i915.enable_guc=3 i915.enable_gvt=1 i915.enable_psr=0 i915.fastboot=1 iwlwifi.enable_ini=0 split_lock_detect=off thinkpad_acpi.fan_control=1 thinkpad_acpi.volume_control=1 x11 sd_mod.ssd=1 psmouse.synaptics_intertouch=1 Any help is very much appreciated! My best., Jeff

1 0

[v3 PATCH] ALSA: virtio: Fix "Coverity: virtsnd_kctl_tlv_op(): Uninitialized variables" warning.
by Aiswarya Cyriac 15 Feb '24

15 Feb '24

This commit fixes the following warning when building virtio_snd driver. " *** CID 1583619: Uninitialized variables (UNINIT) sound/virtio/virtio_kctl.c:294 in virtsnd_kctl_tlv_op() 288 289 break; 290 } 291 292 kfree(tlv); 293 vvv CID 1583619: Uninitialized variables (UNINIT) vvv Using uninitialized value "rc". 294 return rc; 295 } 296 297 /** 298 * virtsnd_kctl_get_enum_items() - Query items for the ENUMERATED element type. 299 * @snd: VirtIO sound device. " This warning is caused by the absence of the "default" branch in the switch-block, and is a false positive because the kernel calls virtsnd_kctl_tlv_op() only with values for op_flag processed in this block. Also, this commit unifies the cleanup path for all possible control paths in the callback function. Signed-off-by: Anton Yakovlev <anton.yakovlev(a)opensynergy.com> Signed-off-by: Aiswarya Cyriac <aiswarya.cyriac(a)opensynergy.com> Reported-by: coverity-bot <keescook+coverity-bot(a)chromium.org> Addresses-Coverity-ID: 1583619 ("Uninitialized variables") Fixes: d6568e3de42d ("ALSA: virtio: add support for audio controls") --- sound/virtio/virtio_kctl.c | 25 +++++++++++++++++++++---- 1 file changed, 21 insertions(+), 4 deletions(-) diff --git a/sound/virtio/virtio_kctl.c b/sound/virtio/virtio_kctl.c index 0c6ac74aca1e..40606eb381af 100644 --- a/sound/virtio/virtio_kctl.c +++ b/sound/virtio/virtio_kctl.c @@ -253,8 +253,8 @@ static int virtsnd_kctl_tlv_op(struct snd_kcontrol *kcontrol, int op_flag, tlv = kzalloc(size, GFP_KERNEL); if (!tlv) { - virtsnd_ctl_msg_unref(msg); - return -ENOMEM; + rc = -ENOMEM; + goto on_cleanup; } sg_init_one(&sg, tlv, size); @@ -266,6 +266,11 @@ static int virtsnd_kctl_tlv_op(struct snd_kcontrol *kcontrol, int op_flag, case SNDRV_CTL_TLV_OP_READ: hdr->hdr.code = cpu_to_le32(VIRTIO_SND_R_CTL_TLV_READ); + /* Since virtsnd_ctl_msg_send() drops the reference, we increase + * the counter to be consistent with the on_cleanup path. + */ + virtsnd_ctl_msg_ref(msg); + rc = virtsnd_ctl_msg_send(snd, msg, NULL, &sg, false); if (!rc) { if (copy_to_user(utlv, tlv, size)) @@ -281,14 +286,26 @@ static int virtsnd_kctl_tlv_op(struct snd_kcontrol *kcontrol, int op_flag, hdr->hdr.code = cpu_to_le32(VIRTIO_SND_R_CTL_TLV_COMMAND); - if (copy_from_user(tlv, utlv, size)) + if (copy_from_user(tlv, utlv, size)) { rc = -EFAULT; - else + } else { + /* Same as the comment above */ + virtsnd_ctl_msg_ref(msg); + rc = virtsnd_ctl_msg_send(snd, msg, &sg, NULL, false); + } + + break; + default: + rc = -EINVAL; + WARN_ON(1); break; } +on_cleanup: + virtsnd_ctl_msg_unref(msg); + kfree(tlv); return rc; -- 2.43.0

2 2

[PATCH v15 00/50] Introduce QC USB SND audio offloading support
by Wesley Cheng 15 Feb '24

15 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 v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) 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 (11): 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 (39): 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: Add USB offloading capable kcontrol ALSA: usb-audio: qcom: Use card and PCM index from QMI request 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 | 39 +- 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 | 171 ++ sound/soc/Kconfig | 9 + sound/soc/Makefile | 4 + 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 | 537 +++++ sound/usb/Kconfig | 19 + sound/usb/Makefile | 3 +- sound/usb/card.c | 109 + sound/usb/card.h | 15 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 65 + sound/usb/mixer_usb_offload.h | 11 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1913 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 49 files changed, 6289 insertions(+), 169 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

2 60

[PATCH] soc: qcom: apr: make aprbus const
by Ricardo B. Marliere 14 Feb '24

14 Feb '24

Since commit d492cc2573a0 ("driver core: device.h: make struct bus_type a const *"), the driver core can properly handle constant struct bus_type, move the aprbus variable to be a constant structure as well, placing it into read-only memory which can not be modified at runtime. Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Suggested-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Ricardo B. Marliere <ricardo(a)marliere.net> --- drivers/soc/qcom/apr.c | 2 +- include/linux/soc/qcom/apr.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soc/qcom/apr.c b/drivers/soc/qcom/apr.c index 1f8b315576a4..50749e870efa 100644 --- a/drivers/soc/qcom/apr.c +++ b/drivers/soc/qcom/apr.c @@ -399,7 +399,7 @@ static int apr_uevent(const struct device *dev, struct kobj_uevent_env *env) return add_uevent_var(env, "MODALIAS=apr:%s", adev->name); } -struct bus_type aprbus = { +const struct bus_type aprbus = { .name = "aprbus", .match = apr_device_match, .probe = apr_device_probe, diff --git a/include/linux/soc/qcom/apr.h b/include/linux/soc/qcom/apr.h index be98aebcb3e1..7161a3183eda 100644 --- a/include/linux/soc/qcom/apr.h +++ b/include/linux/soc/qcom/apr.h @@ -9,7 +9,7 @@ #include <dt-bindings/soc/qcom,apr.h> #include <dt-bindings/soc/qcom,gpr.h> -extern struct bus_type aprbus; +extern const struct bus_type aprbus; #define APR_HDR_LEN(hdr_len) ((hdr_len)/4) --- base-commit: f7b0172f074a435a34bc0902b6cdbf1a3e935e27 change-id: 20240213-bus_cleanup-apr-305f89f9058c Best regards, -- Ricardo B. Marliere <ricardo(a)marliere.net>

3 2

Re: [REGRESSION] Acp5x probing regression introduced between kernel 6.7.2 -> 6.7.4
by Ted Chang 14 Feb '24

14 Feb '24

Hi everyone, Takashi Iwai suggested that I test OBS home:tiwai:bsc1219789 repo. His comment suggest he reverted commit c3ab23a10771bbe06300e5374efa809789c65455 ASoC: amd: Add new dmi entries for acp5x platform. My audio seems to work again [ 7.420289] acpi_cpufreq: overriding BIOS provided _PSD data [ 7.439331] snd_pci_acp5x 0000:04:00.5: enabling device (0000 -> 0002) [ 7.574796] snd_hda_intel 0000:04:00.1: enabling device (0000 -> 0002) [ 7.574980] snd_hda_intel 0000:04:00.1: Handle vga_switcheroo audio client [ 7.577788] kvm_amd: TSC scaling supported [ 7.577794] kvm_amd: Nested Virtualization enabled [ 7.577796] kvm_amd: Nested Paging enabled [ 7.577799] kvm_amd: SEV enabled (ASIDs 1 - 14) [ 7.577802] kvm_amd: SEV-ES disabled (ASIDs 0 - 0) [ 7.577824] kvm_amd: Virtual VMLOAD VMSAVE supported [ 7.577825] kvm_amd: Virtual GIF supported [ 7.577827] kvm_amd: LBR virtualization supported [ 7.589223] MCE: In-kernel MCE decoding enabled. [ 7.590386] snd_hda_intel 0000:04:00.1: bound 0000:04:00.0 (ops amdgpu_dm_audio_component_bind_ops [amdgpu]) [ 7.591577] cs35l41 spi-VLV1776:00: supply VA not found, using dummy regulator [ 7.591644] cs35l41 spi-VLV1776:00: supply VP not found, using dummy regulator [ 7.595880] input: HD-Audio Generic HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input17 [ 7.596790] input: HD-Audio Generic HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input18 https://build.opensuse.org/project/show/home:tiwai:bsc1219789 Now I'm building a test kernel with the revert of the suspected backport patch. It's being built in OBS home:tiwai:bsc1219789 repo. Please give it a try later once after the build finishes (takes an hour or so). So the culprit was the stable commit 4b6986b170f2f23e390bbd2d50784caa9cb67093, which is the upstream commit c3ab23a10771bbe06300e5374efa809789c65455 ASoC: amd: Add new dmi entries for acp5x platform On the other note, I am not the recipient of your reply email Ciao, Thorsten. I had to grab the emails from the MAINTAINER list. On Sun, Feb 11, 2024 at 6:34 AM Ted Chang <tedchang2010(a)gmail.com> wrote: > Hi kernel maintainers, > > I noticed 6.7.4 has introduced a regression for the steam deck. The LCD > steam deck can no longer probe the acp5x audio chipset anymore. This > regression does not affect the 6.8.x series. I did not test kernel 6.7.3 > because Opensuse tumbleweed skipped the update on my machine. > > Steps to reproduce the problem > 1. Obtain a steam deck > 2. Install kernel 6.7.4 > 3. Boot the device and you will see dummy output in gnome shell > > Observed kernel logs. > > [ 8.755614] cs35l41 spi-VLV1776:00: supply VA not found, using dummy regulator > [ 8.760506] cs35l41 spi-VLV1776:00: supply VP not found, using dummy regulator > [ 8.777148] cs35l41 spi-VLV1776:00: Cirrus Logic CS35L41 (35a40), Revision: B2 > [ 8.777471] cs35l41 spi-VLV1776:01: supply VA not found, using dummy regulator > [ 8.777532] cs35l41 spi-VLV1776:01: supply VP not found, using dummy regulator > [ 8.777709] cs35l41 spi-VLV1776:01: Reset line busy, assuming shared reset > [ 8.788465] cs35l41 spi-VLV1776:01: Cirrus Logic CS35L41 (35a40), Revision: B2 > [ 8.877280] snd_hda_intel 0000:04:00.1: enabling device (0000 -> 0002) > [ 8.877595] snd_hda_intel 0000:04:00.1: Handle vga_switcheroo audio client > [ 8.889913] snd_acp_pci 0000:04:00.5: enabling device (0000 -> 0002) > [ 8.890063] snd_acp_pci 0000:04:00.5: Unsupported device revision:0x50 > [ 8.890129] snd_acp_pci: probe of 0000:04:00.5 failed with error -22 > [ 8.906136] snd_hda_intel 0000:04:00.1: bound 0000:04:00.0 (ops amdgpu_dm_audio_component_bind_ops [amdgpu] > > > No kernel module in use shown. > > 04:00.5 Multimedia controller [0480]: Advanced Micro Devices, Inc. [AMD] > ACP/ACP3X/ACP6x Audio Coprocessor [1022:15e2] (rev 50) > Subsystem: Valve Software Device [1e44:1776] > Flags: fast devsel, IRQ 70, IOMMU group 4 > Memory at 80380000 (32-bit, non-prefetchable) [size=256K] > Capabilities: <access denied> > Kernel modules: snd_pci_acp3x, snd_rn_pci_acp3x, snd_pci_acp5x, > snd_pci_acp6x, snd_acp_pci, snd_rpl_pci_acp6x, snd_pci_ps, > snd_sof_amd_renoir, snd_sof_amd_rembrandt, snd_sof_amd_vangogh, > snd_sof_amd_acp63 > > > Information for package kernel-default: > --------------------------------------- > Repository : openSUSE-Tumbleweed-Oss > Name : kernel-default > Version : 6.7.4-1.1 > Arch : x86_64 > Vendor : openSUSE > Installed Size : 240.3 MiB > Installed : Yes > Status : up-to-date > Source package : kernel-default-6.7.4-1.1.nosrc > Upstream URL : https://www.kernel.org/ > Summary : The Standard Kernel > Description : > The standard kernel for both uniprocessor and multiprocessor systems. > > > Source Timestamp: 2024-02-06 05:32:37 +0000 > GIT Revision: 01735a3e65287585dd830a6a3d33d909a4f9ae7f > GIT Branch: stable > > Handle 0x0000, DMI type 0, 26 bytes > BIOS Information > Vendor: Valve > Version: F7A0120 > Release Date: 12/01/2023 > Address: 0xE0000 > Runtime Size: 128 kB > BIOS Revision: 1.20 > Firmware Revision: 1.16 > > #regzbot introduced: v6.7.2..v6.7.4 > >

4 3

[PATCH v2 0/2] ASoC: meson: aiu: fix function pointer type mismatch
by Jerome Brunet 14 Feb '24

14 Feb '24

This patchset fixes 2 -Wcast-function-type-strict warning in amlogic audio drivers with clang 16. Changes since v1: [0] * use devm_clk_get_enabled() instead of adding a dedicated helper in each driver. * Split the patch, 1 per fixed commit to make it easier for stable. [0]: https://lore.kernel.org/lkml/20240213101220.459641-1-arnd@kernel.org/ Jerome Brunet (2): ASoC: meson: aiu: fix function pointer type mismatch ASoC: meson: t9015: fix function pointer type mismatch sound/soc/meson/aiu.c | 19 ++++--------------- sound/soc/meson/aiu.h | 1 - sound/soc/meson/t9015.c | 20 ++++---------------- 3 files changed, 8 insertions(+), 32 deletions(-) -- 2.43.0

3 5

[PATCH] ALSA: virtio: Fix "Coverity: virtsnd_kctl_tlv_op(): Uninitialized variables" warning.
by Aiswarya Cyriac 14 Feb '24

14 Feb '24

Fix the following warning when building virtio_snd driver. " *** CID 1583619: Uninitialized variables (UNINIT) sound/virtio/virtio_kctl.c:294 in virtsnd_kctl_tlv_op() 288 289 break; 290 } 291 292 kfree(tlv); 293 vvv CID 1583619: Uninitialized variables (UNINIT) vvv Using uninitialized value "rc". 294 return rc; 295 } 296 297 /** 298 * virtsnd_kctl_get_enum_items() - Query items for the ENUMERATED element type. 299 * @snd: VirtIO sound device. " Signed-off-by: Anton Yakovlev <anton.yakovlev(a)opensynergy.com> Signed-off-by: Aiswarya Cyriac <aiswarya.cyriac(a)opensynergy.com> Reported-by: coverity-bot <keescook+coverity-bot(a)chromium.org> Addresses-Coverity-ID: 1583619 ("Uninitialized variables") Fixes: d6568e3de42d ("ALSA: virtio: add support for audio controls") --- sound/virtio/virtio_kctl.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/virtio/virtio_kctl.c b/sound/virtio/virtio_kctl.c index 0c6ac74aca1e..d7a160c5db03 100644 --- a/sound/virtio/virtio_kctl.c +++ b/sound/virtio/virtio_kctl.c @@ -286,6 +286,11 @@ static int virtsnd_kctl_tlv_op(struct snd_kcontrol *kcontrol, int op_flag, else rc = virtsnd_ctl_msg_send(snd, msg, &sg, NULL, false); + break; + default: + virtsnd_ctl_msg_unref(msg); + rc = -EINVAL; + break; } -- 2.43.0

3 9

[v2 PATCH] ALSA: virtio: Fix "Coverity: virtsnd_kctl_tlv_op(): Uninitialized variables" warning.
by Aiswarya Cyriac 14 Feb '24

14 Feb '24

This commit fixes the following warning when building virtio_snd driver. " *** CID 1583619: Uninitialized variables (UNINIT) sound/virtio/virtio_kctl.c:294 in virtsnd_kctl_tlv_op() 288 289 break; 290 } 291 292 kfree(tlv); 293 vvv CID 1583619: Uninitialized variables (UNINIT) vvv Using uninitialized value "rc". 294 return rc; 295 } 296 297 /** 298 * virtsnd_kctl_get_enum_items() - Query items for the ENUMERATED element type. 299 * @snd: VirtIO sound device. " This warning is caused by the absence of the "default" branch in the switch-block, and is a false positive because the kernel calls virtsnd_kctl_tlv_op() only with values for op_flag processed in this block. Also, this commit unifies the cleanup path for all possible control paths in the callback function. Signed-off-by: Anton Yakovlev <anton.yakovlev(a)opensynergy.com> Signed-off-by: Aiswarya Cyriac <aiswarya.cyriac(a)opensynergy.com> Reported-by: coverity-bot <keescook+coverity-bot(a)chromium.org> Addresses-Coverity-ID: 1583619 ("Uninitialized variables") Fixes: d6568e3de42d ("ALSA: virtio: add support for audio controls") --- sound/virtio/virtio_kctl.c | 24 ++++++++++++++++++++---- 1 file changed, 20 insertions(+), 4 deletions(-) diff --git a/sound/virtio/virtio_kctl.c b/sound/virtio/virtio_kctl.c index 0c6ac74aca1e..89e29a496dbe 100644 --- a/sound/virtio/virtio_kctl.c +++ b/sound/virtio/virtio_kctl.c @@ -253,8 +253,8 @@ static int virtsnd_kctl_tlv_op(struct snd_kcontrol *kcontrol, int op_flag, tlv = kzalloc(size, GFP_KERNEL); if (!tlv) { - virtsnd_ctl_msg_unref(msg); - return -ENOMEM; + rc = -ENOMEM; + goto on_cleanup; } sg_init_one(&sg, tlv, size); @@ -266,6 +266,11 @@ static int virtsnd_kctl_tlv_op(struct snd_kcontrol *kcontrol, int op_flag, case SNDRV_CTL_TLV_OP_READ: hdr->hdr.code = cpu_to_le32(VIRTIO_SND_R_CTL_TLV_READ); + /* Since virtsnd_ctl_msg_send() drops the reference, we increase + * the counter to be consistent with the on_cleanup path. + */ + virtsnd_ctl_msg_ref(msg); + rc = virtsnd_ctl_msg_send(snd, msg, NULL, &sg, false); if (!rc) { if (copy_to_user(utlv, tlv, size)) @@ -281,14 +286,25 @@ static int virtsnd_kctl_tlv_op(struct snd_kcontrol *kcontrol, int op_flag, hdr->hdr.code = cpu_to_le32(VIRTIO_SND_R_CTL_TLV_COMMAND); - if (copy_from_user(tlv, utlv, size)) + if (copy_from_user(tlv, utlv, size)) { rc = -EFAULT; - else + } else { + /* Same as the comment above */ + virtsnd_ctl_msg_ref(msg); + rc = virtsnd_ctl_msg_send(snd, msg, &sg, NULL, false); + } + + break; + default: + rc = -EINVAL; break; } +on_cleanup: + virtsnd_ctl_msg_unref(msg); + kfree(tlv); return rc; -- 2.43.0

1 0

[PATCH 0/3] ASoC: codecs: qcom: tx-macro: minor cleanups
by Krzysztof Kozlowski 14 Feb '24

14 Feb '24

Hi, Few minor cleanups. The series will conflict with my series: https://lore.kernel.org/alsa-devel/20240129143534.109196-1-krzysztof.kozlow… Depending what will come in first, I will rebase the other patchset. Best regards, Krzysztof Krzysztof Kozlowski (3): ASoC: codecs: tx-macro: Drop unimplemented DMIC clock divider ASoC: codecs: tx-macro: Mark AMIC control registers as volatile ASoC: codecs: tx-macro: Simplify setting AMIC control sound/soc/codecs/lpass-tx-macro.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) -- 2.34.1

2 4

[PATCH v2] ASoC: codecs: va-macro: add npl clk
by neil.armstrong＠linaro.org 14 Feb '24

14 Feb '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> New versions of VA Macro has soundwire integrated, so handle the soundwire npl clock correctly in the codec driver. Introduce has_npl_clk and handle the sm8550 case separately because it has soundwire integrated but doesn't have an npl clock. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Signed-off-by: Neil Armstrong <neil.armstrong(a)linaro.org> --- Changes since v2: - Dropped already applied/upstreamed patch 1 - Rebased on next-20240202 - Switch to has_npl_clk and add sm8550 compat using Krzysztof's changes - Link to v2: https://lore.kernel.org/all/20221118071849.25506-1-srinivas.kandagatla@lina… Changes since v1: - updated bindings as per Kryz comments. - added error handling during npl and mclk enable. - Link to v1: https://lore.kernel.org/all/20221115105541.16322-1-srinivas.kandagatla@lina… --- sound/soc/codecs/lpass-va-macro.c | 57 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index b71ef03c4aef..6eceeff10bf6 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -201,10 +201,12 @@ struct va_macro { unsigned long active_ch_cnt[VA_MACRO_MAX_DAIS]; u16 dmic_clk_div; bool has_swr_master; + bool has_npl_clk; int dec_mode[VA_MACRO_NUM_DECIMATORS]; struct regmap *regmap; struct clk *mclk; + struct clk *npl; struct clk *macro; struct clk *dcodec; struct clk *fsgen; @@ -225,14 +227,22 @@ struct va_macro { struct va_macro_data { bool has_swr_master; + bool has_npl_clk; }; static const struct va_macro_data sm8250_va_data = { .has_swr_master = false, + .has_npl_clk = false, }; static const struct va_macro_data sm8450_va_data = { .has_swr_master = true, + .has_npl_clk = true, +}; + +static const struct va_macro_data sm8550_va_data = { + .has_swr_master = true, + .has_npl_clk = false, }; static bool va_is_volatile_register(struct device *dev, unsigned int reg) @@ -1332,6 +1342,12 @@ static int fsgen_gate_enable(struct clk_hw *hw) struct regmap *regmap = va->regmap; int ret; + if (va->has_swr_master) { + ret = clk_prepare_enable(va->mclk); + if (ret) + return ret; + } + ret = va_macro_mclk_enable(va, true); if (va->has_swr_master) regmap_update_bits(regmap, CDC_VA_CLK_RST_CTRL_SWR_CONTROL, @@ -1350,6 +1366,8 @@ static void fsgen_gate_disable(struct clk_hw *hw) CDC_VA_SWR_CLK_EN_MASK, 0x0); va_macro_mclk_enable(va, false); + if (va->has_swr_master) + clk_disable_unprepare(va->mclk); } static int fsgen_gate_is_enabled(struct clk_hw *hw) @@ -1378,6 +1396,9 @@ static int va_macro_register_fsgen_output(struct va_macro *va) struct clk_init_data init; int ret; + if (va->has_npl_clk) + parent = va->npl; + parent_clk_name = __clk_get_name(parent); of_property_read_string(np, "clock-output-names", &clk_name); @@ -1500,10 +1521,21 @@ static int va_macro_probe(struct platform_device *pdev) data = of_device_get_match_data(dev); va->has_swr_master = data->has_swr_master; + va->has_npl_clk = data->has_npl_clk; /* mclk rate */ clk_set_rate(va->mclk, 2 * VA_MACRO_MCLK_FREQ); + if (va->has_npl_clk) { + va->npl = devm_clk_get(dev, "npl"); + if (IS_ERR(va->npl)) { + ret = PTR_ERR(va->npl); + goto err; + } + + clk_set_rate(va->npl, 2 * VA_MACRO_MCLK_FREQ); + } + ret = clk_prepare_enable(va->macro); if (ret) goto err; @@ -1516,6 +1548,12 @@ static int va_macro_probe(struct platform_device *pdev) if (ret) goto err_mclk; + if (va->has_npl_clk) { + ret = clk_prepare_enable(va->npl); + if (ret) + goto err_npl; + } + if (va->has_swr_master) { /* Set default CLK div to 1 */ regmap_update_bits(va->regmap, CDC_VA_TOP_CSR_SWR_MIC_CTL0, @@ -1564,6 +1602,9 @@ static int va_macro_probe(struct platform_device *pdev) return 0; err_clkout: + if (va->has_npl_clk) + clk_disable_unprepare(va->npl); +err_npl: clk_disable_unprepare(va->mclk); err_mclk: clk_disable_unprepare(va->dcodec); @@ -1579,6 +1620,9 @@ static void va_macro_remove(struct platform_device *pdev) { struct va_macro *va = dev_get_drvdata(&pdev->dev); + if (va->has_npl_clk) + clk_disable_unprepare(va->npl); + clk_disable_unprepare(va->mclk); clk_disable_unprepare(va->dcodec); clk_disable_unprepare(va->macro); @@ -1593,6 +1637,9 @@ static int __maybe_unused va_macro_runtime_suspend(struct device *dev) regcache_cache_only(va->regmap, true); regcache_mark_dirty(va->regmap); + if (va->has_npl_clk) + clk_disable_unprepare(va->npl); + clk_disable_unprepare(va->mclk); return 0; @@ -1609,6 +1656,15 @@ static int __maybe_unused va_macro_runtime_resume(struct device *dev) return ret; } + if (va->has_npl_clk) { + ret = clk_prepare_enable(va->npl); + if (ret) { + clk_disable_unprepare(va->mclk); + dev_err(va->dev, "unable to prepare npl\n"); + return ret; + } + } + regcache_cache_only(va->regmap, false); regcache_sync(va->regmap); @@ -1624,6 +1680,7 @@ static const struct of_device_id va_macro_dt_match[] = { { .compatible = "qcom,sc7280-lpass-va-macro", .data = &sm8250_va_data }, { .compatible = "qcom,sm8250-lpass-va-macro", .data = &sm8250_va_data }, { .compatible = "qcom,sm8450-lpass-va-macro", .data = &sm8450_va_data }, + { .compatible = "qcom,sm8550-lpass-va-macro", .data = &sm8550_va_data }, { .compatible = "qcom,sc8280xp-lpass-va-macro", .data = &sm8450_va_data }, {} }; --- base-commit: 076d56d74f17e625b3d63cf4743b3d7d02180379 change-id: 20240203-topic-sm8x50-upstream-va-macro-npl-41fb6697232d Best regards, -- Neil Armstrong <neil.armstrong(a)linaro.org>

2 1

[PATCH] ASoC: cs35l56: Workaround for ACPI with broken spk-id-gpios property
by Richard Fitzgerald 14 Feb '24

14 Feb '24

The ACPI in some SoundWire laptops has a spk-id-gpios property but it points to the wrong Device node. This patch adds a workaround to try to get the GPIO directly from the correct Device node. If the attempt to get the GPIOs from the property fails, the workaround looks for the SDCA node "AF01", which is where the GpioIo resource is defined. If this exists, a spk-id-gpios mapping is added to that node and then the GPIO is got from that node using the property. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56.c | 93 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 93 insertions(+) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 98d3957c66e7..2c1313e34cce 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -5,6 +5,7 @@ // Copyright (C) 2023 Cirrus Logic, Inc. and // Cirrus Logic International Semiconductor Ltd. +#include <linux/acpi.h> #include <linux/completion.h> #include <linux/debugfs.h> #include <linux/delay.h> @@ -15,6 +16,7 @@ #include <linux/module.h> #include <linux/pm.h> #include <linux/pm_runtime.h> +#include <linux/property.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> @@ -1260,6 +1262,94 @@ static int cs35l56_get_firmware_uid(struct cs35l56_private *cs35l56) return 0; } +/* + * Some SoundWire laptops have a spk-id-gpios property but it points to + * the wrong ACPI Device node so can't be used to get the GPIO. Try to + * find the SDCA node containing the GpioIo resource and add a GPIO + * mapping to it. + */ +static const struct acpi_gpio_params cs35l56_af01_first_gpio = { 0, 0, false }; +static const struct acpi_gpio_mapping cs35l56_af01_spkid_gpios_mapping[] = { + { "spk-id-gpios", &cs35l56_af01_first_gpio, 1 }, + { } +}; + +static void cs35l56_acpi_dev_release_driver_gpios(void *adev) +{ + acpi_dev_remove_driver_gpios(adev); +} + +static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l56) +{ + struct fwnode_handle *af01_fwnode; + const union acpi_object *obj; + struct gpio_desc *desc; + int ret; + + /* Find the SDCA node containing the GpioIo */ + af01_fwnode = device_get_named_child_node(cs35l56->base.dev, "AF01"); + if (!af01_fwnode) { + dev_dbg(cs35l56->base.dev, "No AF01 node\n"); + return -ENOENT; + } + + ret = acpi_dev_get_property(ACPI_COMPANION(cs35l56->base.dev), + "spk-id-gpios", ACPI_TYPE_PACKAGE, &obj); + if (ret) { + dev_dbg(cs35l56->base.dev, "Could not get spk-id-gpios package: %d\n", ret); + return -ENOENT; + } + + /* The broken properties we can handle are a 4-element package (one GPIO) */ + if (obj->package.count != 4) { + dev_warn(cs35l56->base.dev, "Unexpected spk-id element count %d\n", + obj->package.count); + return -ENOENT; + } + + /* Add a GPIO mapping if it doesn't already have one */ + if (!fwnode_property_present(af01_fwnode, "spk-id-gpios")) { + struct acpi_device *adev = to_acpi_device_node(af01_fwnode); + + /* + * Can't use devm_acpi_dev_add_driver_gpios() because the + * mapping isn't being added to the node pointed to by + * ACPI_COMPANION(). + */ + ret = acpi_dev_add_driver_gpios(adev, cs35l56_af01_spkid_gpios_mapping); + if (ret) { + return dev_err_probe(cs35l56->base.dev, ret, + "Failed to add gpio mapping to AF01\n"); + } + + ret = devm_add_action_or_reset(cs35l56->base.dev, + cs35l56_acpi_dev_release_driver_gpios, + adev); + if (ret) + return ret; + + dev_dbg(cs35l56->base.dev, "Added spk-id-gpios mapping to AF01\n"); + } + + desc = fwnode_gpiod_get_index(af01_fwnode, "spk-id", 0, GPIOD_IN, NULL); + if (IS_ERR(desc)) { + ret = PTR_ERR(desc); + return dev_err_probe(cs35l56->base.dev, ret, "Get GPIO from AF01 failed\n"); + } + + ret = gpiod_get_value_cansleep(desc); + gpiod_put(desc); + + if (ret < 0) { + dev_err_probe(cs35l56->base.dev, ret, "Error reading spk-id GPIO\n"); + return ret; + } + + dev_info(cs35l56->base.dev, "Got spk-id from AF01\n"); + + return ret; +} + int cs35l56_common_probe(struct cs35l56_private *cs35l56) { int ret; @@ -1304,6 +1394,9 @@ int cs35l56_common_probe(struct cs35l56_private *cs35l56) } ret = cs35l56_get_speaker_id(&cs35l56->base); + if (ACPI_COMPANION(cs35l56->base.dev) && cs35l56->sdw_peripheral && (ret == -ENOENT)) + ret = cs35l56_try_get_broken_sdca_spkid_gpio(cs35l56); + if ((ret < 0) && (ret != -ENOENT)) goto err; -- 2.30.2

2 1

[PATCH] ASoC: q6dsp: fix event handler prototype
by Arnd Bergmann 13 Feb '24

13 Feb '24

From: Arnd Bergmann <arnd(a)arndb.de> clang-16 points out a mismatch in function types that was hidden by a typecast: sound/soc/qcom/qdsp6/q6apm-dai.c:355:38: error: cast from 'void (*)(uint32_t, uint32_t, uint32_t *, void *)' (aka 'void (*)(unsigned int, unsigned int, unsigned int *, void *)') to 'q6apm_cb' (aka 'void (*)(unsigned int, unsigned int, void *, void *)') converts to incompatible function type [-Werror,-Wcast-function-type-strict] 355 | prtd->graph = q6apm_graph_open(dev, (q6apm_cb)event_handler, prtd, graph_id); sound/soc/qcom/qdsp6/q6apm-dai.c:499:38: error: cast from 'void (*)(uint32_t, uint32_t, uint32_t *, void *)' (aka 'void (*)(unsigned int, unsigned int, unsigned int *, void *)') to 'q6apm_cb' (aka 'void (*)(unsigned int, unsigned int, void *, void *)') converts to incompatible function type [-Werror,-Wcast-function-type-strict] 499 | prtd->graph = q6apm_graph_open(dev, (q6apm_cb)event_handler_compr, prtd, graph_id); The only difference here is the 'payload' argument, which is not even used in this function, so just fix its type and remove the cast. Fixes: 88b60bf047fd ("ASoC: q6dsp: q6apm-dai: Add open/free compress DAI callbacks") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/qcom/qdsp6/q6apm-dai.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 052e40cb38fe..00bbd291be5c 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -123,7 +123,7 @@ static struct snd_pcm_hardware q6apm_dai_hardware_playback = { .fifo_size = 0, }; -static void event_handler(uint32_t opcode, uint32_t token, uint32_t *payload, void *priv) +static void event_handler(uint32_t opcode, uint32_t token, void *payload, void *priv) { struct q6apm_dai_rtd *prtd = priv; struct snd_pcm_substream *substream = prtd->substream; @@ -157,7 +157,7 @@ static void event_handler(uint32_t opcode, uint32_t token, uint32_t *payload, vo } static void event_handler_compr(uint32_t opcode, uint32_t token, - uint32_t *payload, void *priv) + void *payload, void *priv) { struct q6apm_dai_rtd *prtd = priv; struct snd_compr_stream *substream = prtd->cstream; @@ -352,7 +352,7 @@ static int q6apm_dai_open(struct snd_soc_component *component, spin_lock_init(&prtd->lock); prtd->substream = substream; - prtd->graph = q6apm_graph_open(dev, (q6apm_cb)event_handler, prtd, graph_id); + prtd->graph = q6apm_graph_open(dev, event_handler, prtd, graph_id); if (IS_ERR(prtd->graph)) { dev_err(dev, "%s: Could not allocate memory\n", __func__); ret = PTR_ERR(prtd->graph); @@ -496,7 +496,7 @@ static int q6apm_dai_compr_open(struct snd_soc_component *component, return -ENOMEM; prtd->cstream = stream; - prtd->graph = q6apm_graph_open(dev, (q6apm_cb)event_handler_compr, prtd, graph_id); + prtd->graph = q6apm_graph_open(dev, event_handler_compr, prtd, graph_id); if (IS_ERR(prtd->graph)) { ret = PTR_ERR(prtd->graph); kfree(prtd); -- 2.39.2

2 1

[PATCH 6.1.y 0/7] ASoC: codecs: es8326: fix support
by kovalev＠altlinux.org 13 Feb '24

13 Feb '24

These patches were backported from v6.6 upstream and are intended for 6.1.y stable kernel. Patches have been successfully tested on the latest 6.1.75 kernel. [PATCH 6.1.y 1/7] ASoC: codecs: es8326: Convert to i2c's .probe_new() [PATCH 6.1.y 2/7] ASoC: codecs: ES8326: Add es8326_mute function [PATCH 6.1.y 3/7] ASoC: codecs: ES8326: Change Hp_detect register names [PATCH 6.1.y 4/7] ASoC: codecs: ES8326: Change Volatile Reg function [PATCH 6.1.y 5/7] ASoC: codecs: ES8326: Fix power-up sequence [PATCH 6.1.y 6/7] ASOC: codecs: ES8326: Add calibration support for [PATCH 6.1.y 7/7] ASoC: codecs: ES8326: Update jact detection function

3 10

[PATCH] Add Lenovo Legion 7i gen7 sound quirk
by T. Kudela 13 Feb '24

13 Feb '24

Add sound support for the Legion 7i gen7 laptop (16IAX7). Signed-off-by: T. Kudela <ramzes005(a)gmail.com> --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ sound/pci/hda/patch_realtek.c | 1 + 2 files changed, 3 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index d74cf11eef1e..8a6b484b2184 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -95,6 +95,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431F12", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, { "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "17AA38B4", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "17AA38B5", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "17AA38B6", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, @@ -431,6 +432,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431F12", generic_dsd_config }, { "CSC3551", "10431F1F", generic_dsd_config }, { "CSC3551", "10431F62", generic_dsd_config }, + { "CSC3551", "17AA386F", generic_dsd_config }, { "CSC3551", "17AA38B4", generic_dsd_config }, { "CSC3551", "17AA38B5", generic_dsd_config }, { "CSC3551", "17AA38B6", generic_dsd_config }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6994c4c5073c..0029f61d4693 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10260,6 +10260,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x386f, "Legion 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C), SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C), -- 2.34.1

5 7

[PATCH] ASoC: dt-bindings: qcom,sm8250: Allow up to 8 codec DAIs
by Krzysztof Kozlowski 13 Feb '24

13 Feb '24

Sound card on Qualcomm X1E80100 CRD board has eight DAIs in one DAI link (for WSA speakers). Boards with older SoCs could technically have similar setup, even if it was not observed on mainlined devices. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- DTS with X1E80100 sound card using this: https://lore.kernel.org/linux-devicetree/20240212172335.124845-5-krzysztof.… --- Documentation/devicetree/bindings/sound/qcom,sm8250.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index 6f419747273e..2ab6871e89e5 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -107,7 +107,7 @@ patternProperties: properties: sound-dai: minItems: 1 - maxItems: 4 + maxItems: 8 required: - link-name -- 2.34.1

2 2

[PATCH] ASoC: meson: add helpers for clk_disable_unprepare
by Arnd Bergmann 13 Feb '24

13 Feb '24

From: Arnd Bergmann <arnd(a)arndb.de> Casting between incompatible function types causes a warning with clang-16 because it breaks control flow integrity (KCFI) rules: sound/soc/meson/aiu.c:243:12: error: cast from 'void (*)(struct clk *)' to 'void (*)(void *)' converts to incompatible function type [-Werror,-Wcast-function-type-strict] 243 | (void(*)(void *))clk_disable_unprepare, | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ include/linux/device.h:406:34: note: expanded from macro 'devm_add_action_or_reset' 406 | __devm_add_action_or_reset(dev, action, data, #action) | ^~~~~~ sound/soc/meson/t9015.c:274:4: error: cast from 'void (*)(struct clk *)' to 'void (*)(void *)' converts to incompatible function type [-Werror,-Wcast-function-type-strict] 274 | (void(*)(void *))clk_disable_unprepare, | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ include/linux/device.h:406:34: note: expanded from macro 'devm_add_action_or_reset' 406 | __devm_add_action_or_reset(dev, action, data, #action) | ^~~~~~ These two meson drivers cast clk_disable_unprepare() into a different type in order to have it automatically called from the driver relase. Add trivial helpers to do the same using correct types. Fixes: 33901f5b9b16 ("ASoC: meson: add t9015 internal DAC driver") Fixes: 6ae9ca9ce986 ("ASoC: meson: aiu: add i2s and spdif support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/meson/aiu.c | 9 ++++++--- sound/soc/meson/t9015.c | 9 ++++++--- 2 files changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index 7109b81cc3d0..211f360df284 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -209,6 +209,11 @@ static const char * const aiu_spdif_ids[] = { [MCLK] = "spdif_mclk_sel" }; +static void aiu_clk_disable(void *clk) +{ + clk_disable_unprepare(clk); +} + static int aiu_clk_get(struct device *dev) { struct aiu *aiu = dev_get_drvdata(dev); @@ -239,9 +244,7 @@ static int aiu_clk_get(struct device *dev) return ret; } - ret = devm_add_action_or_reset(dev, - (void(*)(void *))clk_disable_unprepare, - aiu->pclk); + ret = devm_add_action_or_reset(dev, aiu_clk_disable, aiu->pclk); if (ret) dev_err(dev, "failed to add reset action on pclk"); diff --git a/sound/soc/meson/t9015.c b/sound/soc/meson/t9015.c index 9c6b4dac6893..e0f9a603a5e9 100644 --- a/sound/soc/meson/t9015.c +++ b/sound/soc/meson/t9015.c @@ -243,6 +243,11 @@ static const struct regmap_config t9015_regmap_config = { .max_register = POWER_CFG, }; +static void t9015_clk_disable(void *clk) +{ + clk_disable_unprepare(clk); +} + static int t9015_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; @@ -270,9 +275,7 @@ static int t9015_probe(struct platform_device *pdev) return ret; } - ret = devm_add_action_or_reset(dev, - (void(*)(void *))clk_disable_unprepare, - priv->pclk); + ret = devm_add_action_or_reset(dev, t9015_clk_disable, priv->pclk); if (ret) return ret; -- 2.39.2

4 5

[PATCH] USB: Always select config with the highest supported UAC version
by Alexander Tsoy 13 Feb '24

13 Feb '24

Config with the highest supported UAC version is always preferable because it usually provides more features. Main motivation for this change is to improve support for several UAC2 devices which have UAC1 config as the first one for compatibility reasons. UAC2 mode provides a wider range of supported sampling rates on these devices. Also when UAC4 support is implemented, it will need just one additional case line without changing the logic. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- drivers/usb/core/generic.c | 39 +++++++++++++++++++++++++------------- 1 file changed, 26 insertions(+), 13 deletions(-) diff --git a/drivers/usb/core/generic.c b/drivers/usb/core/generic.c index b134bff5c3fe..8aeb180e1cf9 100644 --- a/drivers/usb/core/generic.c +++ b/drivers/usb/core/generic.c @@ -48,9 +48,16 @@ static bool is_audio(struct usb_interface_descriptor *desc) return desc->bInterfaceClass == USB_CLASS_AUDIO; } -static bool is_uac3_config(struct usb_interface_descriptor *desc) +static bool is_supported_uac(struct usb_interface_descriptor *desc) { - return desc->bInterfaceProtocol == UAC_VERSION_3; + switch(desc->bInterfaceProtocol) { + case UAC_VERSION_1: + case UAC_VERSION_2: + case UAC_VERSION_3: + return true; + default: + return false; + } } int usb_choose_configuration(struct usb_device *udev) @@ -135,22 +142,28 @@ int usb_choose_configuration(struct usb_device *udev) } /* - * Select first configuration as default for audio so that - * devices that don't comply with UAC3 protocol are supported. - * But, still iterate through other configurations and - * select UAC3 compliant config if present. + * Iterate through audio configurations and select the first of + * the highest supported UAC versions. Select the first config + * if no supported UAC configs are found. */ if (desc && is_audio(desc)) { - /* Always prefer the first found UAC3 config */ - if (is_uac3_config(desc)) { - best = c; - break; - } + struct usb_interface_descriptor *best_desc = NULL; + + if (best) + best_desc = &best->intf_cache[0]->altsetting->desc; - /* If there is no UAC3 config, prefer the first config */ - else if (i == 0) + if (i == 0) best = c; + /* Assume that bInterfaceProtocol value is always + * growing when UAC versions are incremented, so that + * the direct comparison is possible. */ + else if (is_supported_uac(desc) && best_desc && + (!is_supported_uac(best_desc) || + (desc->bInterfaceProtocol > + best_desc->bInterfaceProtocol))) + best = c; + /* Unconditional continue, because the rest of the code * in the loop is irrelevant for audio devices, and * because it can reassign best, which for audio devices -- 2.43.0

3 4

Coverity: virtsnd_kctl_tlv_op(): Uninitialized variables
by coverity-bot 12 Feb '24

12 Feb '24

Hello! This is an experimental semi-automated report about issues detected by Coverity from a scan of next-20240212 as part of the linux-next scan project: https://scan.coverity.com/projects/linux-next-weekly-scan You're getting this email because you were associated with the identified lines of code (noted below) that were touched by commits: Fri Feb 9 14:01:15 2024 +0100 d6568e3de42d ("ALSA: virtio: add support for audio controls") Coverity reported the following: *** CID 1583619: Uninitialized variables (UNINIT) sound/virtio/virtio_kctl.c:294 in virtsnd_kctl_tlv_op() 288 289 break; 290 } 291 292 kfree(tlv); 293 vvv CID 1583619: Uninitialized variables (UNINIT) vvv Using uninitialized value "rc". 294 return rc; 295 } 296 297 /** 298 * virtsnd_kctl_get_enum_items() - Query items for the ENUMERATED element type. 299 * @snd: VirtIO sound device. If this is a false positive, please let us know so we can mark it as such, or teach the Coverity rules to be smarter. If not, please make sure fixes get into linux-next. :) For patches fixing this, please include these lines (but double-check the "Fixes" first): Reported-by: coverity-bot <keescook+coverity-bot(a)chromium.org> Addresses-Coverity-ID: 1583619 ("Uninitialized variables") Fixes: d6568e3de42d ("ALSA: virtio: add support for audio controls") Thanks for your attention! (Human comment: looks like there's no "default" case in the switch statement.) -- Coverity-bot

1 0

[PATCH v2 00/33] spi: get rid of some legacy macros
by Uwe Kleine-König 12 Feb '24

12 Feb '24

Hello, this is v2 of this patch set. Changes since (implicit) v1, sent with Message-Id: cover.1705348269.git.u.kleine-koenig(a)pengutronix.de: - Rebase to v6.8-rc1 - Fix a build failure on sh - Added the tags received in (implicit) v1. The slave-mt27xx driver needs some more work. The patch presented here is enough however to get rid of the defines handled in patch 32. Cleaning that up is out-of-scope for this series, so I'll delay that until later. Note that Jonathan Cameron has already applied patch 3 to his tree, it didn't appear in a public tree though yet. I still included it here to make the kernel build bots happy. Best regards Uwe Uwe Kleine-König (33): fpga: ice40-spi: Follow renaming of SPI "master" to "controller" ieee802154: ca8210: Follow renaming of SPI "master" to "controller" iio: adc: ad_sigma_delta: Follow renaming of SPI "master" to "controller" Input: pxspad - follow renaming of SPI "master" to "controller" Input: synaptics-rmi4 - follow renaming of SPI "master" to "controller" media: mgb4: Follow renaming of SPI "master" to "controller" media: netup_unidvb: Follow renaming of SPI "master" to "controller" media: usb/msi2500: Follow renaming of SPI "master" to "controller" media: v4l2-subdev: Follow renaming of SPI "master" to "controller" misc: gehc-achc: Follow renaming of SPI "master" to "controller" mmc: mmc_spi: Follow renaming of SPI "master" to "controller" mtd: dataflash: Follow renaming of SPI "master" to "controller" mtd: rawnand: fsl_elbc: Let .probe retry if local bus is missing net: ks8851: Follow renaming of SPI "master" to "controller" net: vertexcom: mse102x: Follow renaming of SPI "master" to "controller" platform/chrome: cros_ec_spi: Follow renaming of SPI "master" to "controller" spi: bitbang: Follow renaming of SPI "master" to "controller" spi: cadence-quadspi: Don't emit error message on allocation error spi: cadence-quadspi: Follow renaming of SPI "master" to "controller" spi: cavium: Follow renaming of SPI "master" to "controller" spi: geni-qcom: Follow renaming of SPI "master" to "controller" spi: loopback-test: Follow renaming of SPI "master" to "controller" spi: slave-mt27xx: Follow renaming of SPI "master" to "controller" spi: spidev: Follow renaming of SPI "master" to "controller" staging: fbtft: Follow renaming of SPI "master" to "controller" staging: greybus: spi: Follow renaming of SPI "master" to "controller" tpm_tis_spi: Follow renaming of SPI "master" to "controller" usb: gadget: max3420_udc: Follow renaming of SPI "master" to "controller" video: fbdev: mmp: Follow renaming of SPI "master" to "controller" wifi: libertas: Follow renaming of SPI "master" to "controller" spi: fsl-lib: Follow renaming of SPI "master" to "controller" spi: Drop compat layer from renaming "master" to "controller" Documentation: spi: Update documentation for renaming "master" to "controller" .../driver-api/driver-model/devres.rst | 2 +- Documentation/spi/spi-summary.rst | 74 +++++++++---------- drivers/char/tpm/tpm_tis_spi_main.c | 4 +- drivers/fpga/ice40-spi.c | 4 +- drivers/iio/adc/ad_sigma_delta.c | 14 ++-- drivers/input/joystick/psxpad-spi.c | 4 +- drivers/input/rmi4/rmi_spi.c | 2 +- drivers/media/pci/mgb4/mgb4_core.c | 14 ++-- .../media/pci/netup_unidvb/netup_unidvb_spi.c | 48 ++++++------ drivers/media/usb/msi2500/msi2500.c | 38 +++++----- drivers/media/v4l2-core/v4l2-spi.c | 4 +- drivers/misc/gehc-achc.c | 8 +- drivers/mmc/host/mmc_spi.c | 6 +- drivers/mtd/devices/mtd_dataflash.c | 2 +- drivers/mtd/nand/raw/fsl_elbc_nand.c | 3 +- drivers/net/ethernet/micrel/ks8851_spi.c | 4 +- drivers/net/ethernet/vertexcom/mse102x.c | 2 +- drivers/net/ieee802154/ca8210.c | 2 +- .../net/wireless/marvell/libertas/if_spi.c | 2 +- drivers/platform/chrome/cros_ec_spi.c | 8 +- drivers/spi/spi-ath79.c | 4 +- drivers/spi/spi-bitbang.c | 64 ++++++++-------- drivers/spi/spi-butterfly.c | 6 +- drivers/spi/spi-cadence-quadspi.c | 7 +- drivers/spi/spi-cavium.c | 6 +- drivers/spi/spi-cavium.h | 2 +- drivers/spi/spi-davinci.c | 6 +- drivers/spi/spi-fsl-lib.c | 14 ++-- drivers/spi/spi-geni-qcom.c | 2 +- drivers/spi/spi-gpio.c | 2 +- drivers/spi/spi-lm70llp.c | 6 +- drivers/spi/spi-loopback-test.c | 4 +- drivers/spi/spi-oc-tiny.c | 6 +- drivers/spi/spi-omap-uwire.c | 4 +- drivers/spi/spi-sh-sci.c | 10 +-- drivers/spi/spi-slave-mt27xx.c | 2 +- drivers/spi/spi-xilinx.c | 4 +- drivers/spi/spi-xtensa-xtfpga.c | 2 +- drivers/spi/spi.c | 2 +- drivers/spi/spidev.c | 2 +- drivers/staging/fbtft/fbtft-core.c | 4 +- drivers/staging/greybus/spilib.c | 66 ++++++++--------- drivers/usb/gadget/udc/max3420_udc.c | 2 +- drivers/video/fbdev/mmp/hw/mmp_spi.c | 26 +++---- include/linux/spi/spi.h | 20 +---- include/linux/spi/spi_bitbang.h | 2 +- include/media/v4l2-common.h | 6 +- 47 files changed, 254 insertions(+), 272 deletions(-) base-commit: 6613476e225e090cc9aad49be7fa504e290dd33d -- 2.43.0

6 10

[PATCH 1/2] ASoC: rt5645: Make LattePanda board DMI match more precise
by Hans de Goede 12 Feb '24

12 Feb '24

The DMI strings used for the LattePanda board DMI quirks are very generic. Using the dmidecode database from https://linux-hardware.org/ shows that the chosen DMI strings also match the following 2 laptops which also have a rt5645 codec: Insignia NS-P11W7100 https://linux-hardware.org/?computer=E092FFF8BA04 Insignia NS-P10W8100 https://linux-hardware.org/?computer=AFB6C0BF7934 All 4 hw revisions of the LattePanda board have "S70CR" in their BIOS version DMI strings: DF-BI-7-S70CR100-* DF-BI-7-S70CR110-* DF-BI-7-S70CR200-* LP-BS-7-S70CR700-* See e.g. https://linux-hardware.org/?computer=D98250A817C0 Add a partial (non exact) DMI match on this string to make the LattePanda board DMI match more precise to avoid false-positive matches. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/rt5645.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 5150d6ee3748..b3865e3c3259 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3837,6 +3837,16 @@ static const struct dmi_system_id dmi_platform_data[] = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), DMI_EXACT_MATCH(DMI_BOARD_NAME, "Cherry Trail CR"), DMI_EXACT_MATCH(DMI_BOARD_VERSION, "Default string"), + /* + * Above strings are too generic, LattePanda BIOS versions for + * all 4 hw revisions are: + * DF-BI-7-S70CR100-* + * DF-BI-7-S70CR110-* + * DF-BI-7-S70CR200-* + * LP-BS-7-S70CR700-* + * Do a partial match for S70CR to avoid false positive matches. + */ + DMI_MATCH(DMI_BIOS_VERSION, "S70CR"), }, .driver_data = (void *)&lattepanda_board_platform_data, }, -- 2.43.0

3 3

[PATCH] soundwire: fix double free of pointer
by Daniil Dulov 12 Feb '24

12 Feb '24

If sdw_ml_sync_bank_switch() returns error not on the first iteration, it leads to freeing prevously freed memory. So, set the pointer to NULL after each successful bank switch. Signed-off-by: Daniil Dulov <d.dulov(a)aladdin.ru> --- drivers/soundwire/stream.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 304ff2ee7d75..d650e6f0f8e7 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -833,6 +833,7 @@ static int do_bank_switch(struct sdw_stream_runtime *stream) "multi link bank switch failed: %d\n", ret); goto error; } + bus->defer_msg.msg = NULL; if (multi_link) mutex_unlock(&bus->msg_lock); -- 2.25.1

2 1

ALSA-LIB: correct IEC958 preambles for PCM multichannel?
by Pavel Hofman 12 Feb '24

12 Feb '24

Hi, I wonder if the pcm_iec958.c correctly sets preambles for multichannel PCM/LPCM. It looks as if coded only for stereo. IIUC the X/Y preambles should alternate for even/odd channels. This check https://github.com/alsa-project/alsa-lib/blob/master/src/pcm/pcm_iec958.c#L… : if (channel) data |= iec->preamble[PREAMBLE_Y]; /* odd sub frame, 'Y' */ else if (! iec->counter) data |= iec->preamble[PREAMBLE_Z]; /* Block start, 'Z' */ else data |= iec->preamble[PREAMBLE_X]; /* even sub frame, 'X' */ IMO the condition should be "if (channel % 2)", because for multichannel only the first even channel ch0 gets Y, and the remaining even channels ch2, ch4, ... get X. This is confirmed by analyzing the generated stream captured with the file plugin. Also I am not sure if counting the subframes for Z preamble in https://github.com/alsa-project/alsa-lib/blob/master/src/pcm/pcm_iec958.c#L… respects the multichannel. I do not know what "parallel IEC958 streams" mean, how the Z preamble should be placed for PCM multichannel. A) Either it can count all channel pairs sequentially up to 192, i.e. preamble Z after 192 pair of (ch0, ch1), (ch2, ch3),... If so, IMO the incrementation in https://github.com/alsa-project/alsa-lib/blob/master/src/pcm/pcm_iec958.c#L… should increment by channels/2, not by 1 (which fits for channels=2) B) Or it can count each channel pair separately, i.e. 4 preables Z (for 8ch) in a row after 192 pairs of (ch0+ch1), 192 pairs of (ch2+ch3), etc. If so, IMO the counter should be reset after looping for all channel frames also for PCM, not only for nonaudio/single_stream https://github.com/alsa-project/alsa-lib/blob/master/src/pcm/pcm_iec958.c#L… The reason I am looking at this code are the problems with channel swaps at xrun handling of RPi HDMI https://forums.raspberrypi.com/viewtopic.php?p=2187582#p2187540. I suspect the iec958 plugin does not generate correct multichannel IEC958 stream which confuses the HDMI receiver at the stream recovery. Thanks a lot, Pavel.

1 1

[PATCH] sound: drivers: fix typo 'reguest' to 'request'
by Yinchuan Guo 12 Feb '24

12 Feb '24

This patch fixes a widespread spelling mistake of the word "request" (incorrectly spelled as "reguest") across multiple files. Signed-off-by: Yinchuan Guo <guoych37(a)mail2.sysu.edu.cn> --- drivers/mfd/da9063-irq.c | 2 +- drivers/scsi/lpfc/lpfc_sli.c | 2 +- drivers/tty/serial/max310x.c | 2 +- drivers/tty/serial/sccnxp.c | 2 +- sound/soc/codecs/rt274.c | 2 +- sound/soc/codecs/rt286.c | 2 +- sound/soc/codecs/rt298.c | 2 +- sound/soc/codecs/rt5514-spi.c | 2 +- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5665.c | 2 +- sound/soc/codecs/rt5668.c | 2 +- sound/soc/codecs/rt5682-i2c.c | 2 +- 16 files changed, 16 insertions(+), 16 deletions(-) diff --git a/drivers/mfd/da9063-irq.c b/drivers/mfd/da9063-irq.c index e2bbedf58e68..698d8ea50f32 100644 --- a/drivers/mfd/da9063-irq.c +++ b/drivers/mfd/da9063-irq.c @@ -188,7 +188,7 @@ int da9063_irq_init(struct da9063 *da9063) IRQF_TRIGGER_LOW | IRQF_ONESHOT | IRQF_SHARED, da9063->irq_base, irq_chip, &da9063->regmap_irq); if (ret) { - dev_err(da9063->dev, "Failed to reguest IRQ %d: %d\n", + dev_err(da9063->dev, "Failed to request IRQ %d: %d\n", da9063->chip_irq, ret); return ret; } diff --git a/drivers/scsi/lpfc/lpfc_sli.c b/drivers/scsi/lpfc/lpfc_sli.c index fc3682f15f50..a4fd45bb4ea5 100644 --- a/drivers/scsi/lpfc/lpfc_sli.c +++ b/drivers/scsi/lpfc/lpfc_sli.c @@ -9724,7 +9724,7 @@ lpfc_sli4_iocb2wqe(struct lpfc_hba *phba, struct lpfc_iocbq *iocbq, } wqe->els_req.payload_len = xmit_len; - /* Els_reguest64 has a TMO */ + /* Els_request64 has a TMO */ bf_set(wqe_tmo, &wqe->els_req.wqe_com, iocbq->iocb.ulpTimeout); /* Need a VF for word 4 set the vf bit*/ diff --git a/drivers/tty/serial/max310x.c b/drivers/tty/serial/max310x.c index 3cbc757d7be7..6e259b292c41 100644 --- a/drivers/tty/serial/max310x.c +++ b/drivers/tty/serial/max310x.c @@ -1402,7 +1402,7 @@ static int max310x_probe(struct device *dev, const struct max310x_devtype *devty if (!ret) return 0; - dev_err(dev, "Unable to reguest IRQ %i\n", irq); + dev_err(dev, "Unable to request IRQ %i\n", irq); out_uart: for (i = 0; i < devtype->nr; i++) { diff --git a/drivers/tty/serial/sccnxp.c b/drivers/tty/serial/sccnxp.c index 10cc16a71f26..8968d214da2f 100644 --- a/drivers/tty/serial/sccnxp.c +++ b/drivers/tty/serial/sccnxp.c @@ -1016,7 +1016,7 @@ static int sccnxp_probe(struct platform_device *pdev) if (!ret) return 0; - dev_err(&pdev->dev, "Unable to reguest IRQ %i\n", s->irq); + dev_err(&pdev->dev, "Unable to request IRQ %i\n", s->irq); } else { timer_setup(&s->timer, sccnxp_timer, 0); mod_timer(&s->timer, jiffies + diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 0d3773c576f8..b385b859373b 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1196,7 +1196,7 @@ static int rt274_i2c_probe(struct i2c_client *i2c, IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt274", rt274); if (ret != 0) { dev_err(&i2c->dev, - "Failed to reguest IRQ: %d\n", ret); + "Failed to request IRQ: %d\n", ret); return ret; } } diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 802f4851c3df..061211f0fd65 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1244,7 +1244,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); if (ret != 0) { dev_err(&i2c->dev, - "Failed to reguest IRQ: %d\n", ret); + "Failed to request IRQ: %d\n", ret); return ret; } } diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index c592c40a7ab3..ac81edfd2320 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1286,7 +1286,7 @@ static int rt298_i2c_probe(struct i2c_client *i2c, IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt298", rt298); if (ret != 0) { dev_err(&i2c->dev, - "Failed to reguest IRQ: %d\n", ret); + "Failed to request IRQ: %d\n", ret); return ret; } } diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 1a25a3787935..98e12f8113bc 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -280,7 +280,7 @@ static int rt5514_spi_pcm_probe(struct snd_soc_component *component) rt5514_dsp); if (ret) dev_err(&rt5514_spi->dev, - "%s Failed to reguest IRQ: %d\n", __func__, + "%s Failed to request IRQ: %d\n", __func__, ret); else device_init_wakeup(rt5514_dsp->dev, true); diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 9523f4b5c800..5b7ae70be1be 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2843,7 +2843,7 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, /* Gets re-enabled by rt5640_set_jack() */ disable_irq(rt5640->irq); } else { - dev_warn(&i2c->dev, "Failed to reguest IRQ %d: %d\n", + dev_warn(&i2c->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret); rt5640->irq = -ENXIO; } diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9408ee63cb26..cc868a9742bc 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4127,7 +4127,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5645", rt5645); if (ret) { - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); goto err_enable; } } diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index fc0c83b73f09..e603886627e4 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -2266,7 +2266,7 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, /* Gets re-enabled by rt5651_set_jack() */ disable_irq(rt5651->irq); } else { - dev_warn(&i2c->dev, "Failed to reguest IRQ %d: %d\n", + dev_warn(&i2c->dev, "Failed to request IRQ %d: %d\n", rt5651->irq, ret); rt5651->irq = -ENXIO; } diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 4a50b169fe03..acfbdcf2fbd9 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -4299,7 +4299,7 @@ static int rt5659_i2c_probe(struct i2c_client *i2c, rt5659_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5659", rt5659); if (ret) - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); /* Enable IRQ output for GPIO1 pin any way */ regmap_update_bits(rt5659->regmap, RT5659_GPIO_CTRL_1, diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index be9fc58ff681..8f7db0703ea2 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3676,7 +3676,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5663", rt5663); if (ret) { - dev_err(&i2c->dev, "%s Failed to reguest IRQ: %d\n", + dev_err(&i2c->dev, "%s Failed to request IRQ: %d\n", __func__, ret); goto err_enable; } diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index e59323fd5bf2..d82ead0e5caf 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4930,7 +4930,7 @@ static int rt5665_i2c_probe(struct i2c_client *i2c, rt5665_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5665", rt5665); if (ret) - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); } diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 6ab1a8bc3735..f1abfbc62a78 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -2581,7 +2581,7 @@ static int rt5668_i2c_probe(struct i2c_client *i2c, rt5668_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5668", rt5668); if (ret) - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); } diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 8ea9f1d9fec0..d03307585806 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -261,7 +261,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5682", rt5682); if (ret) - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); } return devm_snd_soc_register_component(&i2c->dev, -- 2.34.1

4 5

MAG Z790 TOMAHAWK WIFI DDR4 no Sound Front Audio Realtek 4080
by GitHub issues - opened 12 Feb '24

12 Feb '24

alsa-project/alsa-ucm-conf issue #395 was opened from Acer6920: Greetings, back audio is working the sound card is recognized as Generic USB Audio 3 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-0000:00:14.0-11, high speed alsa information: http://alsa-project.org/db/?f=655312de1f9b11ddc33de5fa41c6cad87bcbab9d if i missed something excuse it please i use zorin os/linux since yesterday and need some advice if you need more information. thanks a lot. ace Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/395 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] Subject: [PATCH] sound: drivers: fix typo 'reguest' to 'request'
by Yinchuan Guo 12 Feb '24

12 Feb '24

This patch fixes a widespread spelling mistake of the word "request" (incorrectly spelled as "reguest") across multiple files. Signed-off-by: Yinchuan Guo <guoych37(a)mail2.sysu.edu.cn> --- drivers/mfd/da9063-irq.c | 2 +- drivers/scsi/lpfc/lpfc_sli.c | 2 +- drivers/tty/serial/max310x.c | 2 +- drivers/tty/serial/sccnxp.c | 2 +- sound/soc/codecs/rt274.c | 2 +- sound/soc/codecs/rt286.c | 2 +- sound/soc/codecs/rt298.c | 2 +- sound/soc/codecs/rt5514-spi.c | 2 +- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5665.c | 2 +- sound/soc/codecs/rt5668.c | 2 +- sound/soc/codecs/rt5682-i2c.c | 2 +- 16 files changed, 16 insertions(+), 16 deletions(-) diff --git a/drivers/mfd/da9063-irq.c b/drivers/mfd/da9063-irq.c index e2bbedf58e68..698d8ea50f32 100644 --- a/drivers/mfd/da9063-irq.c +++ b/drivers/mfd/da9063-irq.c @@ -188,7 +188,7 @@ int da9063_irq_init(struct da9063 *da9063) IRQF_TRIGGER_LOW | IRQF_ONESHOT | IRQF_SHARED, da9063->irq_base, irq_chip, &da9063->regmap_irq); if (ret) { - dev_err(da9063->dev, "Failed to reguest IRQ %d: %d\n", + dev_err(da9063->dev, "Failed to request IRQ %d: %d\n", da9063->chip_irq, ret); return ret; } diff --git a/drivers/scsi/lpfc/lpfc_sli.c b/drivers/scsi/lpfc/lpfc_sli.c index fc3682f15f50..a4fd45bb4ea5 100644 --- a/drivers/scsi/lpfc/lpfc_sli.c +++ b/drivers/scsi/lpfc/lpfc_sli.c @@ -9724,7 +9724,7 @@ lpfc_sli4_iocb2wqe(struct lpfc_hba *phba, struct lpfc_iocbq *iocbq, } wqe->els_req.payload_len = xmit_len; - /* Els_reguest64 has a TMO */ + /* Els_request64 has a TMO */ bf_set(wqe_tmo, &wqe->els_req.wqe_com, iocbq->iocb.ulpTimeout); /* Need a VF for word 4 set the vf bit*/ diff --git a/drivers/tty/serial/max310x.c b/drivers/tty/serial/max310x.c index 3cbc757d7be7..6e259b292c41 100644 --- a/drivers/tty/serial/max310x.c +++ b/drivers/tty/serial/max310x.c @@ -1402,7 +1402,7 @@ static int max310x_probe(struct device *dev, const struct max310x_devtype *devty if (!ret) return 0; - dev_err(dev, "Unable to reguest IRQ %i\n", irq); + dev_err(dev, "Unable to request IRQ %i\n", irq); out_uart: for (i = 0; i < devtype->nr; i++) { diff --git a/drivers/tty/serial/sccnxp.c b/drivers/tty/serial/sccnxp.c index 10cc16a71f26..8968d214da2f 100644 --- a/drivers/tty/serial/sccnxp.c +++ b/drivers/tty/serial/sccnxp.c @@ -1016,7 +1016,7 @@ static int sccnxp_probe(struct platform_device *pdev) if (!ret) return 0; - dev_err(&pdev->dev, "Unable to reguest IRQ %i\n", s->irq); + dev_err(&pdev->dev, "Unable to request IRQ %i\n", s->irq); } else { timer_setup(&s->timer, sccnxp_timer, 0); mod_timer(&s->timer, jiffies + diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 0d3773c576f8..b385b859373b 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1196,7 +1196,7 @@ static int rt274_i2c_probe(struct i2c_client *i2c, IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt274", rt274); if (ret != 0) { dev_err(&i2c->dev, - "Failed to reguest IRQ: %d\n", ret); + "Failed to request IRQ: %d\n", ret); return ret; } } diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 802f4851c3df..061211f0fd65 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1244,7 +1244,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); if (ret != 0) { dev_err(&i2c->dev, - "Failed to reguest IRQ: %d\n", ret); + "Failed to request IRQ: %d\n", ret); return ret; } } diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index c592c40a7ab3..ac81edfd2320 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1286,7 +1286,7 @@ static int rt298_i2c_probe(struct i2c_client *i2c, IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt298", rt298); if (ret != 0) { dev_err(&i2c->dev, - "Failed to reguest IRQ: %d\n", ret); + "Failed to request IRQ: %d\n", ret); return ret; } } diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 1a25a3787935..98e12f8113bc 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -280,7 +280,7 @@ static int rt5514_spi_pcm_probe(struct snd_soc_component *component) rt5514_dsp); if (ret) dev_err(&rt5514_spi->dev, - "%s Failed to reguest IRQ: %d\n", __func__, + "%s Failed to request IRQ: %d\n", __func__, ret); else device_init_wakeup(rt5514_dsp->dev, true); diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 9523f4b5c800..5b7ae70be1be 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2843,7 +2843,7 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, /* Gets re-enabled by rt5640_set_jack() */ disable_irq(rt5640->irq); } else { - dev_warn(&i2c->dev, "Failed to reguest IRQ %d: %d\n", + dev_warn(&i2c->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret); rt5640->irq = -ENXIO; } diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9408ee63cb26..cc868a9742bc 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4127,7 +4127,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5645", rt5645); if (ret) { - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); goto err_enable; } } diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index fc0c83b73f09..e603886627e4 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -2266,7 +2266,7 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, /* Gets re-enabled by rt5651_set_jack() */ disable_irq(rt5651->irq); } else { - dev_warn(&i2c->dev, "Failed to reguest IRQ %d: %d\n", + dev_warn(&i2c->dev, "Failed to request IRQ %d: %d\n", rt5651->irq, ret); rt5651->irq = -ENXIO; } diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 4a50b169fe03..acfbdcf2fbd9 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -4299,7 +4299,7 @@ static int rt5659_i2c_probe(struct i2c_client *i2c, rt5659_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5659", rt5659); if (ret) - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); /* Enable IRQ output for GPIO1 pin any way */ regmap_update_bits(rt5659->regmap, RT5659_GPIO_CTRL_1, diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index be9fc58ff681..8f7db0703ea2 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3676,7 +3676,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5663", rt5663); if (ret) { - dev_err(&i2c->dev, "%s Failed to reguest IRQ: %d\n", + dev_err(&i2c->dev, "%s Failed to request IRQ: %d\n", __func__, ret); goto err_enable; } diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index e59323fd5bf2..d82ead0e5caf 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4930,7 +4930,7 @@ static int rt5665_i2c_probe(struct i2c_client *i2c, rt5665_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5665", rt5665); if (ret) - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); } diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 6ab1a8bc3735..f1abfbc62a78 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -2581,7 +2581,7 @@ static int rt5668_i2c_probe(struct i2c_client *i2c, rt5668_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5668", rt5668); if (ret) - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); } diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 8ea9f1d9fec0..d03307585806 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -261,7 +261,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5682", rt5682); if (ret) - dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); } return devm_snd_soc_register_component(&i2c->dev, -- 2.34.1

3 2

Asus ROG Strix Z790-A Gaming WIFI II audio codec detected as generic USB audio, headphones and SPDIF not working
by GitHub issues - opened 12 Feb '24

12 Feb '24

alsa-project/alsa-ucm-conf issue #394 was opened from danielton1: Hi, I have an Asus ROG STRIX Z790-A GAMING WIFI II motherboard with a Realtek ALC4080 audio codec. Sound is working through the speakers via the line out jack, and it does detect when I plug headphones in, but everything I'm finding is that the ALSA UCM is failing to detect the device as a Realtek ALC4080 and instead is just using a generic USB profile, probably due to the USB ID being unknown. Audio always comes out the line out jack no matter what is plugged in to the front panel or SPDIF or how pipewire/ALSA are configured. This is what lsusb gives as the ID: Bus 001 Device 002: ID 0b05:1af1 ASUSTek Computer, Inc. USB Audio This is the output of alsa-info.sh: http://alsa-project.org/db/?f=3dc5eb049583da76a18466a6cf16d144dae95f53 I've tried modifying the files in /usr/share/alsa/ucm2/USB-Audio several different ways per several forum threads I came across online and ended up with no sound at all, so I'm not sure what's going on or how to fix this. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/394 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 6.8 regression fix 0/2] ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards
by Hans de Goede 12 Feb '24

12 Feb '24

Hi All, While testing 6.8 on a Bay Trail device with a ALC5640 codec I noticed a regression in 6.8 which causes a NULL pointer deref in probe(). All BYT/CHT Intel machine drivers are affected. Patch 1/2 of this series fixes all of them. Patch 2/2 adds some small cleanups to cht_bsw_rt5645.c for issues which I noticed while working on 1/2. Regards, Hans Hans de Goede (2): ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards ASoC: Intel: cht_bsw_rt5645: Cleanup codec_name handling sound/soc/intel/boards/bytcht_cx2072x.c | 3 ++- sound/soc/intel/boards/bytcht_da7213.c | 3 ++- sound/soc/intel/boards/bytcht_es8316.c | 3 ++- sound/soc/intel/boards/bytcr_rt5640.c | 3 ++- sound/soc/intel/boards/bytcr_rt5651.c | 3 ++- sound/soc/intel/boards/bytcr_wm5102.c | 3 ++- sound/soc/intel/boards/cht_bsw_rt5645.c | 7 +++---- sound/soc/intel/boards/cht_bsw_rt5672.c | 3 ++- 8 files changed, 17 insertions(+), 11 deletions(-) -- 2.43.0

2 3

Re: [REGRESSION] Acp5x probing regression introduced between kernel 6.7.2 -> 6.7.4
by Linux regression tracking (Thorsten Leemhuis) 11 Feb '24

11 Feb '24

[CCing a few people] On 11.02.24 15:34, Ted Chang wrote: > > I noticed 6.7.4 has introduced a regression for the steam deck. The LCD > steam deck can no longer probe the acp5x audio chipset anymore. This > regression does not affect the 6.8.x series. I did not test kernel > 6.7.3 because Opensuse tumbleweed skipped the update on my machine. Thx for your report. FWIW, problems like this can be caused by all sorts of changes, but obviously those in the area of audio support are most likely to cause this. There are just a few in the v6.7.2..v6.7.4 range[1]. Among them a commit that is related to acp5x, that's why I CCed its author as well (Venkata Prasad Potturu). Maybe one of the new recipients will have an idea. If not, you most likely will have to bisect this and check if mainline is affected as well.[2] Ciao, Thorsten [1] $ git log --oneline v6.7.2..v6.7.4 sound/ f3570675bf09af ASoC: codecs: wsa883x: fix PA volume control 2f8e9b77ca2fea ASoC: codecs: lpass-wsa-macro: fix compander volume hack 5b465d6384e4eb ASoC: codecs: wcd938x: fix headphones volume controls 1673211a38012e ASoC: qcom: sc8280xp: limit speaker volumes 242b5bffa23a9c ASoC: codecs: rtq9128: Fix TDM enable and DAI format control flow 2c272ff9859601 ASoC: codecs: rtq9128: Fix PM_RUNTIME usage 4a28302b2c681e ALSA: hda/conexant: Fix headset auto detect fail in cx8070 and SN6140 e37a96941fdd53 ALSA: hda: intel-dspcfg: add filters for ARL-S and ARL ffa3eea886c6fe ALSA: hda: Intel: add HDA_ARL PCI ID support 4b6986b170f2f2 ASoC: amd: Add new dmi entries for acp5x platform e38ad4ace20b4d ALSA: hda: Refer to correct stream index at loops a434c75e0671f9 soundwire: fix initializing sysfs for same devices on different buses [2] I'm working on a guide that describes what's needed: https://www.leemhuis.info/files/misc/How%20to%20bisect%20a%20Linux%20kernel… > Steps to reproduce the problem > 1. Obtain a steam deck > 2. Install kernel 6.7.4 > 3. Boot the device and you will see dummy output in gnome shell > > Observed kernel logs. > > [ 8.755614] cs35l41 spi-VLV1776:00: supply VA not found, using dummy regulator > [ 8.760506] cs35l41 spi-VLV1776:00: supply VP not found, using dummy regulator > [ 8.777148] cs35l41 spi-VLV1776:00: Cirrus Logic CS35L41 (35a40), Revision: B2 > [ 8.777471] cs35l41 spi-VLV1776:01: supply VA not found, using dummy regulator > [ 8.777532] cs35l41 spi-VLV1776:01: supply VP not found, using dummy regulator > [ 8.777709] cs35l41 spi-VLV1776:01: Reset line busy, assuming shared reset > [ 8.788465] cs35l41 spi-VLV1776:01: Cirrus Logic CS35L41 (35a40), Revision: B2 > [ 8.877280] snd_hda_intel 0000:04:00.1: enabling device (0000 -> 0002) > [ 8.877595] snd_hda_intel 0000:04:00.1: Handle vga_switcheroo audio client > [ 8.889913] snd_acp_pci 0000:04:00.5: enabling device (0000 -> 0002) > [ 8.890063] snd_acp_pci 0000:04:00.5: Unsupported device revision:0x50 > [ 8.890129] snd_acp_pci: probe of 0000:04:00.5 failed with error -22 > [ 8.906136] snd_hda_intel 0000:04:00.1: bound 0000:04:00.0 (ops amdgpu_dm_audio_component_bind_ops [amdgpu] > > > No kernel module in use shown. > > 04:00.5 Multimedia controller [0480]: Advanced Micro Devices, Inc. [AMD] > ACP/ACP3X/ACP6x Audio Coprocessor [1022:15e2] (rev 50) > Subsystem: Valve Software Device [1e44:1776] > Flags: fast devsel, IRQ 70, IOMMU group 4 > Memory at 80380000 (32-bit, non-prefetchable) [size=256K] > Capabilities: <access denied> > Kernel modules: snd_pci_acp3x, snd_rn_pci_acp3x, snd_pci_acp5x, > snd_pci_acp6x, snd_acp_pci, snd_rpl_pci_acp6x, snd_pci_ps, > snd_sof_amd_renoir, snd_sof_amd_rembrandt, snd_sof_amd_vangogh, > snd_sof_amd_acp63 > > > Information for package kernel-default: > --------------------------------------- > Repository : openSUSE-Tumbleweed-Oss > Name : kernel-default > Version : 6.7.4-1.1 > Arch : x86_64 > Vendor : openSUSE > Installed Size : 240.3 MiB > Installed : Yes > Status : up-to-date > Source package : kernel-default-6.7.4-1.1.nosrc > Upstream URL : https://www.kernel.org/ <https://www.kernel.org/> > Summary : The Standard Kernel > Description : > The standard kernel for both uniprocessor and multiprocessor systems. > > > Source Timestamp: 2024-02-06 05:32:37 +0000 > GIT Revision: 01735a3e65287585dd830a6a3d33d909a4f9ae7f > GIT Branch: stable > > Handle 0x0000, DMI type 0, 26 bytes > BIOS Information > Vendor: Valve > Version: F7A0120 > Release Date: 12/01/2023 > Address: 0xE0000 > Runtime Size: 128 kB > BIOS Revision: 1.20 > Firmware Revision: 1.16 > > #regzbot introduced: v6.7.2..v6.7.4 >

1 0

[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

2 1

[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

2 1

[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

2 1

[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

2 4

[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

4 73

[PATCH] ASoC: tas2781: remove unused acpi_subysystem_id
by Gergo Koteles 08 Feb '24

08 Feb '24

The acpi_subysystem_id is only written and freed, not read, so unnecessary. Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver") CC: stable(a)vger.kernel.org Signed-off-by: Gergo Koteles <soyer(a)irl.hu> --- include/sound/tas2781.h | 1 - sound/pci/hda/tas2781_hda_i2c.c | 10 ---------- sound/soc/codecs/tas2781-comlib.c | 1 - 3 files changed, 12 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..02c85084aca5 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -129,18 +129,8 @@ 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: 610010737f74482a61896596a0116876ecf9e65c -- 2.43.0

3 7

[PATCH AUTOSEL 5.15 05/23] ASoC: codecs: wcd934x: drop unneeded regulator include
by Sasha Levin 07 Feb '24

07 Feb '24

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 35314e39dabcfb256832654ad0e856a9fba744bd ] Driver does not use any regulator code, so drop redundant include of regulator/consumer.h header. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Link: https://msgid.link/r/20240117151208.1219755-3-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wcd934x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 94ffd2ba29ae..a12bc454e085 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -13,7 +13,6 @@ #include <linux/of.h> #include <linux/platform_device.h> #include <linux/regmap.h> -#include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/slimbus.h> #include <sound/pcm_params.h> -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 06/29] ASoC: codecs: wcd934x: drop unneeded regulator include
by Sasha Levin 07 Feb '24

07 Feb '24

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 35314e39dabcfb256832654ad0e856a9fba744bd ] Driver does not use any regulator code, so drop redundant include of regulator/consumer.h header. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Link: https://msgid.link/r/20240117151208.1219755-3-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wcd934x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 0b5999c819db..86d0907a5f97 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -13,7 +13,6 @@ #include <linux/of.h> #include <linux/platform_device.h> #include <linux/regmap.h> -#include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/slimbus.h> #include <sound/pcm_params.h> -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 07/38] ASoC: codecs: wcd934x: drop unneeded regulator include
by Sasha Levin 07 Feb '24

07 Feb '24

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 35314e39dabcfb256832654ad0e856a9fba744bd ] Driver does not use any regulator code, so drop redundant include of regulator/consumer.h header. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Link: https://msgid.link/r/20240117151208.1219755-3-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wcd934x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 1b6e376f3833..6813268e6a19 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -13,7 +13,6 @@ #include <linux/of.h> #include <linux/platform_device.h> #include <linux/regmap.h> -#include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/slimbus.h> #include <sound/pcm_params.h> -- 2.43.0

1 0

[PATCH AUTOSEL 6.7 18/44] ALSA: hda: cs35l41: Support ASUS Zenbook UM3402YAR
by Sasha Levin 07 Feb '24

07 Feb '24

From: Chhayly Leang <clw.leang(a)gmail.com> [ Upstream commit be220d2e5544ff094142d263db5cf94d034b5e39 ] Adds sound support for ASUS Zenbook UM3402YAR with missing DSD Signed-off-by: Chhayly Leang <clw.leang(a)gmail.com> Link: https://lore.kernel.org/r/20240126080912.87422-1-clw.leang@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 59504852adc6..d74cf11eef1e 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -76,6 +76,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431533", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431573", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431663", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, + { "10431683", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "104316A3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "104316D3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "104316F3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, @@ -411,6 +412,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431533", generic_dsd_config }, { "CSC3551", "10431573", generic_dsd_config }, { "CSC3551", "10431663", generic_dsd_config }, + { "CSC3551", "10431683", generic_dsd_config }, { "CSC3551", "104316A3", generic_dsd_config }, { "CSC3551", "104316D3", generic_dsd_config }, { "CSC3551", "104316F3", generic_dsd_config }, -- 2.43.0

1 0

[PATCH AUTOSEL 6.7 17/44] ALSA: hda: cs35l41: Support additional ASUS Zenbook UX3402VA
by Sasha Levin 07 Feb '24

07 Feb '24

From: Kenzo Gomez <kenzo.sgomez(a)gmail.com> [ Upstream commit c16dfab33f99fc3ff43d48253bc2784ccb84c1de ] Add new model entry into configuration table. Signed-off-by: Kenzo Gomez <kenzo.sgomez(a)gmail.com> Link: https://lore.kernel.org/r/20240127164621.26431-1-kenzo.sgomez@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 35277ce890a4..59504852adc6 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -76,6 +76,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431533", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431573", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431663", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, + { "104316A3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "104316D3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "104316F3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "104317F3", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, @@ -410,6 +411,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431533", generic_dsd_config }, { "CSC3551", "10431573", generic_dsd_config }, { "CSC3551", "10431663", generic_dsd_config }, + { "CSC3551", "104316A3", generic_dsd_config }, { "CSC3551", "104316D3", generic_dsd_config }, { "CSC3551", "104316F3", generic_dsd_config }, { "CSC3551", "104317F3", generic_dsd_config }, -- 2.43.0

1 0

[PATCH 0/3] ASoC: Intel: avs: Add support for sending initial module config
by Amadeusz Sławiński 07 Feb '24

07 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 | 163 +++++++++++++++++++++++++- sound/soc/intel/avs/topology.h | 13 ++ 4 files changed, 216 insertions(+), 2 deletions(-) -- 2.34.1

2 4

[PATCH] ASoC: Intel: avs: Fix dynamic port assignment when TDM is set
by Amadeusz Sławiński 07 Feb '24

07 Feb '24

In case TDM is set in topology on SSP0, parser will overwrite vindex value, because it only checks if port is set. Fix this by checking whole field value. Fixes: e6d50e474e45 ("ASoC: Intel: avs: Improve topology parsing of dynamic strings") Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/intel/avs/topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index 778236d3fd280..48b3c67c91032 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -857,7 +857,7 @@ assign_copier_gtw_instance(struct snd_soc_component *comp, struct avs_tplg_modcf } /* If topology sets value don't overwrite it */ - if (cfg->copier.vindex.i2s.instance) + if (cfg->copier.vindex.val) return; mach = dev_get_platdata(comp->card->dev); -- 2.34.1

2 1

[PATCH] ASoC: Intel: avs: Expose FW version with sysfs
by Amadeusz Sławiński 07 Feb '24

07 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> --- .../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..abfdecd85f204 --- /dev/null +++ b/Documentation/ABI/testing/sysfs-bus-pci-devices-avs @@ -0,0 +1,8 @@ +What: /sys/devices/pci0000:00/<dev>/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

2 1

[PATCH] ASoC: dt-bindings: atmel,asoc-wm8904: Convert to json-schema
by Dharma Balasubiramani 07 Feb '24

07 Feb '24

Convert atmel,asoc-wm8904 devicetree binding to json-schema. Signed-off-by: Dharma Balasubiramani <dharma.b(a)microchip.com> --- .../bindings/sound/atmel,asoc-wm8904.yaml | 84 +++++++++++++++++++ .../bindings/sound/atmel-wm8904.txt | 55 ------------ 2 files changed, 84 insertions(+), 55 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel,asoc-wm8904.yaml delete mode 100644 Documentation/devicetree/bindings/sound/atmel-wm8904.txt diff --git a/Documentation/devicetree/bindings/sound/atmel,asoc-wm8904.yaml b/Documentation/devicetree/bindings/sound/atmel,asoc-wm8904.yaml new file mode 100644 index 000000000000..89a67f8e33b6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,asoc-wm8904.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,asoc-wm8904.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel wm8904 audio codec complex + +maintainers: + - Dharma Balasubiramani <dharma.b(a)microchip.com> + +description: + The ASoC audio complex configuration for Atmel with WM8904 audio codec. + +properties: + compatible: + const: atmel,asoc-wm8904 + + atmel,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + atmel,ssc-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the SSC controller. + + atmel,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the WM8731 audio codec. + + atmel,audio-routing: + description: + A list of the connections between audio components. Each entry is a pair + of strings, the first being the connection's sink, the second being the + connection's source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + items: + enum: + # Board Connectors + - Headphone Jack + - Line In Jack + - Mic + # WM8904 CODEC Pins + - IN1L + - IN1R + - IN2L + - IN2R + - IN3L + - IN3R + - HPOUTL + - HPOUTR + - LINEOUTL + - LINEOUTR + - MICBIAS + +required: + - compatible + - atmel,model + - atmel,audio-routing + - atmel,ssc-controller + - atmel,audio-codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "atmel,asoc-wm8904"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pck0_as_mck>; + + atmel,model = "wm8904 @ AT91SAM9N12EK"; + + atmel,audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "IN2L", "Line In Jack", + "IN2R", "Line In Jack", + "Mic", "MICBIAS", + "IN1L", "Mic"; + + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8904>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel-wm8904.txt b/Documentation/devicetree/bindings/sound/atmel-wm8904.txt deleted file mode 100644 index 8bbe50c884b6..000000000000 --- a/Documentation/devicetree/bindings/sound/atmel-wm8904.txt +++ /dev/null @@ -1,55 +0,0 @@ -Atmel ASoC driver with wm8904 audio codec complex - -Required properties: - - compatible: "atmel,asoc-wm8904" - - atmel,model: The user-visible name of this sound complex. - - atmel,audio-routing: A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the WM8904's pins, and the jacks on the board: - - WM8904 pins: - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * MICBIAS - - Board connectors: - - * Headphone Jack - * Line In Jack - * Mic - - - atmel,ssc-controller: The phandle of the SSC controller - - atmel,audio-codec: The phandle of the WM8904 audio codec - -Optional properties: - - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt - -Example: -sound { - compatible = "atmel,asoc-wm8904"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_pck0_as_mck>; - - atmel,model = "wm8904 @ AT91SAM9N12EK"; - - atmel,audio-routing = - "Headphone Jack", "HPOUTL", - "Headphone Jack", "HPOUTR", - "IN2L", "Line In Jack", - "IN2R", "Line In Jack", - "Mic", "MICBIAS", - "IN1L", "Mic"; - - atmel,ssc-controller = <&ssc0>; - atmel,audio-codec = <&wm8904>; -}; -- 2.25.1

3 2

[PATCH v7] ASoc: tas2783: Add tas2783 codec driver
by Shenghao Ding 07 Feb '24

07 Feb '24

The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp speaker protection algorithm. The integrated speaker voltage and current sense provides for real-time monitoring of lodspeakers. The ASoC component provides the majority of the functionality of the device, all the audio functions. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v7: - simplify tasdevice_set_sdw_stream - fixed some Linux coding style - fixed the spelling mistakes - Select left/right channel based on unique id - a longer description has been added - remove unused crc8 in KCONFIG - Invert of RE here means reciprocal of RE - correct indentation - correct the wrong pow calibrated register - correct comment on RESET register - fixed usually lines don't end with an open parenthesis - fixed tasdevice_sdw_shutdown - use defines for function, entity, etc. for readable - remove unnecessary codec_lock in component_probe() - remove struct sdw_stream_data - remove unused macro definitions - use BIT(2) and BIT(1) instead of 0x2 and 0x1 - implement pm_runtime - reg_start is u8, reg_start + i only adds i byte, it should be reg_start + i * 4, because the calibration values - fixed total_sz in tas2783_calibration - add a define for TASDEVICE_REG(0, 0, 0x0a) - rework macro style in header file - Add select CRC32 - rewrite the mute and the return value handling - Add comments on bin file - Add link_id into the fw name to distinguish the same sdw id on different sdw id. - remove unnecessary mutex_lock - use set_active() in tasdevice_io_init and move all other parts in the probe. - move pm_runtime_get_noresume to the probe - fix the return value of putvol function - add put_noidle in tasdevice_sdw_remove --- sound/soc/codecs/Kconfig | 14 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 937 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 108 ++++ 4 files changed, 1061 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 59f9742e9ff4..f7b35e614801 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -247,6 +247,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1847,6 +1848,19 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + depends on EFI + select REGMAP + select REGMAP_SOUNDWIRE + select CRC32_SARWATE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + TAS2783 supports the MIPI SoundWire v1.2 compatible interface + for audio and control data. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f53baa2b9565..6afceb126ae3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -284,6 +284,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -664,6 +665,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..69e1f5cbfa0f --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,937 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Kevin Lu <kevin-lu(a)ti.com> +// Shenghao Ding <shenghao-ding(a)ti.com> +// + +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/pcm_params.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2783.h" + +static const unsigned int tas2783_cali_reg[] = { + TAS2783_CALIBRATION_RE, /* Resistance */ + TAS2783_CALIBRATION_RE_LOW, /* Low limitation of RE */ + TAS2783_CALIBRATION_INV_RE, /* Reciprocal of RE */ + TAS2783_CALIBRATION_POW, /* RMS Power */ + TAS2783_CALIBRATION_TLIMIT /* Temperature limitation */ +}; + +static const struct reg_default tas2783_reg_defaults[] = { + /* Default values for ROM mode. Activated. */ + { 0x8002, 0x1a }, /* AMP inactive. */ + { 0x8097, 0xc8 }, + { 0x80b5, 0x74 }, + { 0x8099, 0x20 }, + { 0xfe8d, 0x0d }, + { 0xfebe, 0x4a }, + { 0x8230, 0x00 }, + { 0x8231, 0x00 }, + { 0x8232, 0x00 }, + { 0x8233, 0x01 }, + { 0x8418, 0x00 }, /* Set volume to 0 dB. */ + { 0x8419, 0x00 }, + { 0x841a, 0x00 }, + { 0x841b, 0x00 }, + { 0x8428, 0x40 }, /* Unmute channel */ + { 0x8429, 0x00 }, + { 0x842a, 0x00 }, + { 0x842b, 0x00 }, + { 0x8548, 0x00 }, /* Set volume to 0 dB. */ + { 0x8549, 0x00 }, + { 0x854a, 0x00 }, + { 0x854b, 0x00 }, + { 0x8558, 0x40 }, /* Unmute channel */ + { 0x8559, 0x00 }, + { 0x855a, 0x00 }, + { 0x855b, 0x00 }, + { 0x800a, 0x3a }, /* Enable both channel */ + { 0x800e, 0x44 }, + { 0x800f, 0x40 }, + { 0x805c, 0x99 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_VOLUME, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0xc000: /* Page 0 ~ 127. */ + case 0xfe80 ... 0xfeff: /* Page 253. */ + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8001: + /* Only reset register was volatiled. + * Software reset. Bit is self clearing. + * 0b = Don't reset + * 1b = reset + */ + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x44ffffff, + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static int tasdevice_clamp(int val, int max, unsigned int invert) +{ + /* Keep in valid area, out of range value don't care. */ + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + return val; +} + +static int tas2783_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + int val = 0, ret; + + if (!ucontrol) { + dev_err(tas_dev->dev, "%s, wrong parameter.\n", __func__); + return -EINVAL; + } + /* Read current volume from the device. */ + ret = regmap_read(map, mc->reg, &val); + if (ret < 0) { + dev_err(tas_dev->dev, "%s, get digital vol error %x.\n", + __func__, ret); + return ret; + } + ucontrol->value.integer.value[0] = + tasdevice_clamp(val, mc->max, mc->invert); + + return ret; +} + +static int tas2783_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + int val, ret; + + if (!ucontrol) { + dev_err(tas_dev->dev, "%s, wrong parameter.\n", __func__); + return 0; + } + val = tasdevice_clamp(ucontrol->value.integer.value[0], + mc->max, mc->invert); + + ret = regmap_write(map, mc->reg, val); + if (ret < 0) { + dev_err(tas_dev->dev, "%s, Put vol %d into %x %x.\n", + __func__, val, mc->reg, ret); + return 0; + } + + return 1; +} + +static int tas2783_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + unsigned char mask = 0; + int ret, val = 0; + + if (!ucontrol) { + dev_err(tas_dev->dev, "%s, wrong parameter.\n", __func__); + return -EINVAL; + } + /* Read current volume from the device. */ + ret = regmap_read(map, mc->reg, &val); + if (ret < 0) { + dev_err(tas_dev->dev, "%s get AMP vol from %x with %d.\n", + __func__, mc->reg, ret); + return ret; + } + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + ucontrol->value.integer.value[0] = tasdevice_clamp(val, mc->max, + mc->invert); + + return ret; +} + +static int tas2783_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + unsigned char mask; + int val, ret; + + if (!ucontrol) { + dev_err(tas_dev->dev, "%s, wrong parameter.\n", __func__); + return 0; + } + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = tasdevice_clamp(ucontrol->value.integer.value[0], mc->max, + mc->invert); + ret = regmap_update_bits(map, mc->reg, mask, val << mc->shift); + if (ret < 0) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + return 0; + } + + return 1; +} + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, 0, tas2783_amp_getvol, + tas2783_amp_putvol, amp_vol_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, tas2783_digital_getvol, + tas2783_digital_putvol, dvc_tlv), +}; + +static void tas2783_apply_calib(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + struct regmap *map = tas_dev->regmap; + u8 *reg_start; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s, slaver doesn't exist.\n", + __func__); + return; + } + if ((tas_dev->sdw_peripheral->id.unique_id < TAS2783_ID_MIN) || + (tas_dev->sdw_peripheral->id.unique_id > TAS2783_ID_MAX)) { + dev_err(tas_dev->dev, "%s, error unique_id.\n", + __func__); + return; + } + + reg_start = (u8 *)(cali_data + (tas_dev->sdw_peripheral->id.unique_id + - TAS2783_ID_MIN) * sizeof(tas2783_cali_reg)); + for (int i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + ret = regmap_bulk_write(map, tas2783_cali_reg[i], + &reg_start[4 * i], 4); + if (ret) { + dev_err(tas_dev->dev, "Cali failed %x:%d\n", + tas2783_cali_reg[i], ret); + break; + } + } +} + +/* Load the calibration data, including speaker impedance, f0, etc. + * Calibration is done by the manufacturer in the factory. The calibration + * data are used by the algorithm for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * The DSP will work with default data values if calibrated data are + * missing or are invalid. + */ +static int tas2783_calibration(struct tasdevice_priv *tas_dev) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct tm *tm = &tas_dev->tm; + unsigned int attr = 0, crc; + unsigned int *tmp_val; + efi_status_t status; + + tas_dev->cali_data.total_sz = 0; + + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_dev->cali_data.total_sz, tas_dev->cali_data.data); + if (status == EFI_BUFFER_TOO_SMALL) { + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_dev->cali_data.total_sz, + tas_dev->cali_data.data); + dev_dbg(tas_dev->dev, "cali get %lx bytes result:%ld\n", + tas_dev->cali_data.total_sz, status); + } + if (status != 0) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return 0; + } + + tmp_val = (unsigned int *)tas_dev->cali_data.data; + + crc = crc32(~0, tas_dev->cali_data.data, 84) ^ ~0; + + if (crc == tmp_val[21]) { + /* Date and time of calibration was done. */ + time64_to_tm(tmp_val[20], 0, tm); + dev_dbg(tas_dev->dev, "%4ld-%2d-%2d, %2d:%2d:%2d\n", + tm->tm_year, tm->tm_mon, tm->tm_mday, + tm->tm_hour, tm->tm_min, tm->tm_sec); + tas2783_apply_calib(tas_dev, tmp_val); + } else { + dev_dbg(tas_dev->dev, "CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[21]); + tas_dev->cali_data.total_sz = 0; + } + + return 0; +} + +static void tasdevice_rca_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct tas2783_firmware_node *p; + struct regmap *map = tas_dev->regmap; + unsigned char *buf = NULL; + int offset = 0, img_sz; + int ret, value_sdw; + + if (!fmw || !fmw->data) { + /* No firmware binary, devices will work in ROM mode. */ + dev_err(tas_dev->dev, + "Failed to read %s, no side-effect on driver\n", + tas_dev->rca_binaryname); + ret = -EINVAL; + goto out; + } + buf = (unsigned char *)fmw->data; + + img_sz = le32_to_cpup((__le32 *)&buf[offset]); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_err(tas_dev->dev, "Size not matching, %d %u", + (int)fmw->size, img_sz); + ret = -EINVAL; + goto out; + } + + while (offset < img_sz) { + p = (struct tas2783_firmware_node *)(buf + offset); + if (p->length > 1) { + ret = regmap_bulk_write(map, p->download_addr, + buf + offset + sizeof(unsigned int)*5, p->length); + } else + ret = regmap_write(map, p->download_addr, + *(buf + offset + sizeof(unsigned int) * 5)); + + if (ret != 0) { + dev_dbg(tas_dev->dev, "Load FW fail: %d.\n", ret); + goto out; + } + offset += sizeof(unsigned int)*5 + p->length; + } + /* Select left/right channel based on unique id. */ + value_sdw = 0x1a; + value_sdw |= ((tas_dev->sdw_peripheral->dev_num & 1) << 4); + dev_dbg(tas_dev->dev, "%s dev_num = %u", __func__, + tas_dev->sdw_peripheral->dev_num); + regmap_write(map, TAS2783_REG_TDM_RX_CFG, value_sdw); + + tas2783_calibration(tas_dev); + +out: + if (fmw) + release_firmware(fmw); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, + 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC") +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"} +}; + +static int tasdevice_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + int ret; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(tas_priv->sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) { + dev_err(dai->dev, "Unable to configure port\n"); + return ret; + } + + dev_dbg(dai->dev, "%s format: %i rate: %i\n", __func__, + params_format(params), params_rate(params)); + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tas_priv->sdw_peripheral, sdw_stream); + + return 0; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct regmap *map = tas_dev->regmap; + int ret; + + dev_dbg(tas_dev->dev, "Mute or unmute %d.\n", mute); + + if (mute) { + if (direction == SNDRV_PCM_STREAM_CAPTURE) { + ret = regmap_update_bits(map, + TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_INACTIVE); + if (ret) + dev_err(tas_dev->dev, "Disable AEF failed.\n"); + } else { + /* FU23 mute (0x40400108) */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 1); + if (ret) { + dev_err(tas_dev->dev, "FU23 mute failed.\n"); + goto out; + } + /* Both playback and echo data will be shutdown in + * playback stream. + */ + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK | + TAS2783_REG_AEF_MASK, + TAS2783_REG_PWR_MODE_ACTIVE | + TAS2783_REG_PWR_MODE_SW_PWD); + if (ret) { + dev_err(tas_dev->dev, + "PWR&AEF shutdown failed.\n"); + goto out; + } + tas_dev->pstream = false; + } + } else { + /* FU23 Unmute, 0x40400108. */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0); + if (ret) { + dev_err(tas_dev->dev, "FU23 Unmute failed.\n"); + goto out; + } + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK, + TAS2783_REG_PWR_MODE_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, "PWR Unmute failed.\n"); + goto out; + } + tas_dev->pstream = true; + } else { + /* Capture stream is the echo ref data for voice. + * Without playback, it can't be active. + */ + if (tas_dev->pstream == true) { + ret = regmap_update_bits(map, + TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, "AEF failed.\n"); + goto out; + } + } else { + dev_err(tas_dev->dev, + "No playback, no AEF!"); + ret = -EINVAL; + } + } + } +out: + if (ret) + dev_err(tas_dev->dev, "Mute or unmute %d failed %d.\n", + mute, ret); + + return ret; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_component_probe(struct snd_soc_component *component) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + + tas_dev->component = component; + + dev_dbg(tas_dev->dev, "%s was called.\n", __func__); + + return 0; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_component_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + goto out; + } + + /* tas2783-link_id[0,1,...,N]-unique_id[8,9,...,f].bin stores the dsp + * firmware including speaker protection algorithm, audio acoustic + * algorithm, speaker characters and algorithm params, it must be + * copied into firmware folder. Each tas2783 has its own bin file. + */ + scnprintf(tas_dev->rca_binaryname, 64, "tas2783-%d-%x.bin", + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_dev->rca_binaryname, tas_dev->dev, GFP_KERNEL, + tas_dev, tasdevice_rca_ready); + if (ret) { + dev_err(tas_dev->dev, + "%s: request_firmware %x open status: %d.\n", + __func__, tas_dev->sdw_peripheral->id.unique_id, ret); + goto out; + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + pm_runtime_get_noresume(tas_dev->dev); + pm_runtime_enable(tas_dev->dev); + +out: + return ret; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = BIT(2); /* BITMAP: 00000100 */ + prop->sink_ports = BIT(1); /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static int tasdevice_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(tas_priv->regmap, false); + + if (tas_priv->first_hw_init) + regcache_cache_bypass(tas_priv->regmap, true); + else + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* sw reset */ + ret = regmap_write(tas_priv->regmap, TAS2873_REG_SWRESET, + TAS2873_REG_SWRESET_RESET); + if (ret) { + dev_err(tas_priv->dev, "Reset failed.\n"); + goto out; + } + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, false); + regcache_mark_dirty(tas_priv->regmap); + } + + tas_priv->first_hw_init = true; + tas_priv->hw_init = true; + +out: + return ret; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tas_priv->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tas_priv->hw_init = false; + + /* Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_priv->hw_init || tas_priv->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + int ret; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return -ENOMEM; + + tas_dev->dev = dev; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_peripheral = peripheral; + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, + &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) { + ret = PTR_ERR(tas_dev->regmap); + dev_err(dev, "Failed %d of devm_regmap_init_sdw.", ret); + } else + ret = tasdevice_init(tas_dev); + + return ret; +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(&peripheral->dev); + + if (tas_dev->first_hw_init) + pm_runtime_disable(tas_dev->dev); + + pm_runtime_put_noidle(tas_dev->dev); + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {}, +}; + +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static int __maybe_unused tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (!tas_priv->hw_init) + return 0; + + regcache_cache_only(tas_priv->regmap, true); + + return 0; +} + +#define TAS2783_PROBE_TIMEOUT 5000 + +static int __maybe_unused tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long time; + + if (!tas_priv->first_hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Init not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(tas_priv->regmap, false); + regcache_sync(tas_priv->regmap); + + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SET_SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume) + SET_RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume, NULL) +}; + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = &tas2783_sdca_pm, + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..55b0cb09d65a --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,108 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2023 - 2024 Texas Instruments Incorporated + * https://www.ti.com + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + * Shenghao Ding <shenghao-ding(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 + \ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0x00, 0x01) +#define TAS2873_REG_SWRESET_MASK BIT(0) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +#define TAS2873_REG_PWR_CTRL TASDEVICE_REG(0x0, 0x00, 0x02) +#define TAS2783_REG_AEF_MASK GENMASK(4, 3) +#define TAS2783_REG_AEF_ACTIVE 0 +#define TAS2783_REG_AEF_INACTIVE GENMASK(4, 3) +#define TAS2783_REG_PWR_MODE_MASK GENMASK(2, 0) +#define TAS2783_REG_PWR_MODE_ACTIVE 0 +#define TAS2783_REG_PWR_MODE_SW_PWD BIT(1) + +#define TAS2783_REG_TDM_RX_CFG TASDEVICE_REG(0x0, 0x00, 0x0a) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1a) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x13, 0x70) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +/* Unique id start */ +#define TAS2783_ID_MIN 0x08 +/* Unique id end */ +#define TAS2783_ID_MAX 0x0f + +/* TAS2783 SDCA Control - function number */ +#define TAS2783_FUNC_TYPE_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_FU_VOLUME 0x02 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 + +#define TAS2783_MAX_CALIDATA_SIZE 252 + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct snd_soc_component *component; + struct calibration_data cali_data; + struct sdw_slave *sdw_peripheral; + enum sdw_slave_status status; + struct sdw_bus_params params; + struct regmap *regmap; + struct device *dev; + struct tm tm; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + unsigned int chip_id; + bool pstream; + bool hw_init; + bool first_hw_init; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

4 4

[PATCH] ASoC: cs35l56: Fix deadlock in ASP1 mixer register initialization
by Richard Fitzgerald 07 Feb '24

07 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") --- sound/soc/codecs/cs35l56.c | 158 ++++++++++++++++++------------------- sound/soc/codecs/cs35l56.h | 2 +- 2 files changed, 76 insertions(+), 84 deletions(-) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index ebed5ab1245b..7ec70cca1ac4 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -68,6 +68,63 @@ 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; + + /* + * 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 +135,11 @@ 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); + if (!cs35l56->asp1_mixer_widgets_initialized) { + 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 +165,11 @@ 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); + if (!cs35l56->asp1_mixer_widgets_initialized) { + 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 +185,6 @@ 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 +933,6 @@ 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 +978,13 @@ 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 +995,6 @@ 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 +1065,8 @@ 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 +1217,6 @@ 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

2 1

[PATCH 1/2] ASoC: cs42l43: Handle error from devm_pm_runtime_enable
by Charles Keepax 07 Feb '24

07 Feb '24

As devm_pm_runtime_enable can fail due to memory allocations, it is best to handle the error. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l43.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 256767ef4c03..f1e119277622 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2283,7 +2283,10 @@ static int cs42l43_codec_probe(struct platform_device *pdev) pm_runtime_use_autosuspend(priv->dev); pm_runtime_set_active(priv->dev); pm_runtime_get_noresume(priv->dev); - devm_pm_runtime_enable(priv->dev); + + ret = devm_pm_runtime_enable(priv->dev); + if (ret) + goto err_pm; for (i = 0; i < ARRAY_SIZE(cs42l43_irqs); i++) { ret = cs42l43_request_irq(priv, dom, cs42l43_irqs[i].name, -- 2.30.2

2 2

[PATCH][next] soundwire: intel_auxdevice: remove redundant assignment to variable link_flags
by Colin Ian King 07 Feb '24

07 Feb '24

The variable link_flags is being initialized with a value that is never read, it is being re-assigned later on. The initialization is redundant and can be removed. Cleans up clang scan build warning: drivers/soundwire/intel_auxdevice.c:624:2: warning: Value stored to 'link_flags' is never read [deadcode.DeadStores] Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- drivers/soundwire/intel_auxdevice.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/drivers/soundwire/intel_auxdevice.c b/drivers/soundwire/intel_auxdevice.c index 93698532deac..95125cc2fc59 100644 --- a/drivers/soundwire/intel_auxdevice.c +++ b/drivers/soundwire/intel_auxdevice.c @@ -621,8 +621,6 @@ static int __maybe_unused intel_resume(struct device *dev) return 0; } - link_flags = md_flags >> (bus->link_id * 8); - if (pm_runtime_suspended(dev)) { dev_dbg(dev, "pm_runtime status was suspended, forcing active\n"); -- 2.39.2

3 2

[linux][PATCH] ASoC: dt-bindings: atmel,sam9x5-wm8731: Convert to json-schema
by Dharma Balasubiramani 06 Feb '24

06 Feb '24

Convert atmel sam9x5-wm8731-audio devicetree binding to json-schema. Signed-off-by: Dharma Balasubiramani <dharma.b(a)microchip.com> --- .../sound/atmel,sam9x5-wm8731-audio.yaml | 76 +++++++++++++++++++ .../sound/atmel-sam9x5-wm8731-audio.txt | 35 --------- 2 files changed, 76 insertions(+), 35 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel,sam9x5-wm8731-audio.yaml delete mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt diff --git a/Documentation/devicetree/bindings/sound/atmel,sam9x5-wm8731-audio.yaml b/Documentation/devicetree/bindings/sound/atmel,sam9x5-wm8731-audio.yaml new file mode 100644 index 000000000000..33717b728f63 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,sam9x5-wm8731-audio.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,sam9x5-wm8731-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel at91sam9x5ek wm8731 audio complex + +maintainers: + - Dharma Balasubiramani <dharma.b(a)microchip.com> + +description: + The audio complex configuration for Atmel at91sam9x5ek with WM8731 audio codec. + +properties: + compatible: + const: atmel,sam9x5-wm8731-audio + + atmel,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + atmel,ssc-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the SSC controller. + + atmel,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the WM8731 audio codec. + + atmel,audio-routing: + description: + A list of the connections between audio components. Each entry is a pair + of strings, the first being the connection's sink, the second being the + connection's source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + items: + enum: + # Board Connectors + - Headphone Jack + - Line In Jack + + # CODEC Pins + - LOUT + - ROUT + - LHPOUT + - RHPOUT + - LLINEIN + - RLINEIN + - MICIN + +required: + - compatible + - atmel,model + - atmel,ssc-controller + - atmel,audio-codec + - atmel,audio-routing + +additionalProperties: false + +examples: + - | + sound { + compatible = "atmel,sam9x5-wm8731-audio"; + + atmel,model = "wm8731 @ AT91SAM9X5EK"; + + atmel,audio-routing = + "Headphone Jack", "RHPOUT", + "Headphone Jack", "LHPOUT", + "LLINEIN", "Line In Jack", + "RLINEIN", "Line In Jack"; + + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8731>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt b/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt deleted file mode 100644 index 8facbce53db8..000000000000 --- a/Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt +++ /dev/null @@ -1,35 +0,0 @@ -* Atmel at91sam9x5ek wm8731 audio complex - -Required properties: - - compatible: "atmel,sam9x5-wm8731-audio" - - atmel,model: The user-visible name of this sound complex. - - atmel,ssc-controller: The phandle of the SSC controller - - atmel,audio-codec: The phandle of the WM8731 audio codec - - atmel,audio-routing: A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. - -Available audio endpoints for the audio-routing table: - -Board connectors: - * Headphone Jack - * Line In Jack - -wm8731 pins: -cf Documentation/devicetree/bindings/sound/wlf,wm8731.yaml - -Example: -sound { - compatible = "atmel,sam9x5-wm8731-audio"; - - atmel,model = "wm8731 @ AT91SAM9X5EK"; - - atmel,audio-routing = - "Headphone Jack", "RHPOUT", - "Headphone Jack", "LHPOUT", - "LLINEIN", "Line In Jack", - "RLINEIN", "Line In Jack"; - - atmel,ssc-controller = <&ssc0>; - atmel,audio-codec = <&wm8731>; -}; -- 2.25.1

3 2

[PATCH] ASoC: amd: yc: Add DMI quirk for Lenovo Ideapad Pro 5 16ARP8
by Mario Limonciello 06 Feb '24

06 Feb '24

The laptop requires a quirk ID to enable its internal microphone. Add it to the DMI quirk table. Reported-by: Stanislav Petrov <stanislav.i.petrov(a)gmail.com> Closes: https://bugzilla.kernel.org/show_bug.cgi?id=216925 Cc: stable(a)vger.kernel.org Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 23d44a50d815..80ad60d485ea 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -248,6 +248,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "82YM"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "83AS"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.34.1

2 1

[PATCH 1/3] ASoC: cs42l43: Add clear of stashed pointer on component remove
by Charles Keepax 06 Feb '24

06 Feb '24

If the component is removed the stashed component pointer in the CODECs private struct should also be cleared to prevent use of a stale pointer. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l43.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 2c402086924d..9e1deb3242cb 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2111,10 +2111,18 @@ static int cs42l43_component_probe(struct snd_soc_component *component) return 0; } +static void cs42l43_component_remove(struct snd_soc_component *component) +{ + struct cs42l43_codec *priv = snd_soc_component_get_drvdata(component); + + priv->component = NULL; +} + static const struct snd_soc_component_driver cs42l43_component_drv = { .name = "cs42l43-codec", .probe = cs42l43_component_probe, + .remove = cs42l43_component_remove, .set_sysclk = cs42l43_set_sysclk, .set_jack = cs42l43_set_jack, -- 2.30.2

2 3

[PATCH] ASoC: tas2781: add module parameter to tascodec_init()
by Gergo Koteles 05 Feb '24

05 Feb '24

The tascodec_init() of the snd-soc-tas2781-comlib module is called from snd-soc-tas2781-i2c and snd-hda-scodec-tas2781-i2c modules. It calls request_firmware_nowait() with parameter THIS_MODULE and a cont/callback from the latter modules. The latter modules can be removed while their callbacks are running, resulting in a general protection failure. Add module parameter to tascodec_init() so request_firmware_nowait() can be called with the module of the callback. Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver") CC: stable(a)vger.kernel.org Signed-off-by: Gergo Koteles <soyer(a)irl.hu> --- include/sound/tas2781.h | 1 + sound/pci/hda/tas2781_hda_i2c.c | 2 +- sound/soc/codecs/tas2781-comlib.c | 3 ++- sound/soc/codecs/tas2781-i2c.c | 2 +- 4 files changed, 5 insertions(+), 3 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index b00d65417c31..9aff384941de 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -142,6 +142,7 @@ struct tasdevice_priv { void tas2781_reset(struct tasdevice_priv *tas_dev); int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, + struct module *module, void (*cont)(const struct firmware *fw, void *context)); struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c); int tasdevice_init(struct tasdevice_priv *tas_priv); diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 2dd809de62e5..1bfb00102a77 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -710,7 +710,7 @@ static int tas2781_hda_bind(struct device *dev, struct device *master, strscpy(comps->name, dev_name(dev), sizeof(comps->name)); - ret = tascodec_init(tas_hda->priv, codec, tasdev_fw_ready); + ret = tascodec_init(tas_hda->priv, codec, THIS_MODULE, tasdev_fw_ready); if (!ret) comps->playback_hook = tas2781_hda_playback_hook; diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index b7e56ceb1acf..5d0e5348b361 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -267,6 +267,7 @@ void tas2781_reset(struct tasdevice_priv *tas_dev) EXPORT_SYMBOL_GPL(tas2781_reset); int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, + struct module *module, void (*cont)(const struct firmware *fw, void *context)) { int ret = 0; @@ -280,7 +281,7 @@ int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, tas_priv->dev_name, tas_priv->ndev); crc8_populate_msb(tas_priv->crc8_lkp_tbl, TASDEVICE_CRC8_POLYNOMIAL); tas_priv->codec = codec; - ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + ret = request_firmware_nowait(module, FW_ACTION_UEVENT, tas_priv->rca_binaryname, tas_priv->dev, GFP_KERNEL, tas_priv, cont); if (ret) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 32913bd1a623..b5abff230e43 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -566,7 +566,7 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) { struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); - return tascodec_init(tas_priv, codec, tasdevice_fw_ready); + return tascodec_init(tas_priv, codec, THIS_MODULE, tasdevice_fw_ready); } static void tasdevice_deinit(void *context) base-commit: d4ea2bd1bb502c54380cc44a4130660494679bb8 -- 2.43.0

2 1

[PATCH] ASoC: Intel: avs: Fix pci_probe() error path
by Cezary Rojewski 05 Feb '24

05 Feb '24

Recent changes modified operation-order in the probe() function without updating its error path accordingly. If snd_hdac_i915_init() exists with status EPROBE_DEFER the error path must cleanup allocated IRQs before leaving the scope. Fixes: 2dddc514b6e4 ("ASoC: Intel: avs: Move snd_hdac_i915_init to before probe_work.") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/avs/core.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 59c3793f65df..db78eb2f0108 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -477,6 +477,9 @@ static int avs_pci_probe(struct pci_dev *pci, const struct pci_device_id *id) return 0; err_i915_init: + pci_free_irq(pci, 0, adev); + pci_free_irq(pci, 0, bus); + pci_free_irq_vectors(pci); pci_clear_master(pci); pci_set_drvdata(pci, NULL); err_acquire_irq: -- 2.25.1

2 1

[PATCH] ASoC: cs35l56: fix reversed if statement in cs35l56_dspwait_asp1tx_put()
by Dan Carpenter 05 Feb '24

05 Feb '24

It looks like the "!" character was added accidentally. The regmap_update_bits_check() function is normally going to succeed. This means the rest of the function is unreachable and we don't handle the situation where "changed" is true correctly. Fixes: 07f7d6e7a124 ("ASoC: cs35l56: Fix for initializing ASP1 mixer registers") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- From static analysis and review, not tested. --- sound/soc/codecs/cs35l56.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index c23e29da4cfb..ebed5ab1245b 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -115,7 +115,7 @@ static int cs35l56_dspwait_asp1tx_put(struct snd_kcontrol *kcontrol, ret = regmap_update_bits_check(cs35l56->base.regmap, addr, CS35L56_ASP_TXn_SRC_MASK, val, &changed); - if (!ret) + if (ret) return ret; if (changed) -- 2.43.0

3 2

[PATCH v2 00/13] of: property: add port base loop
by Kuninori Morimoto 05 Feb '24

05 Feb '24

Hi Rob This is v2 of port base loop patch-set We have endpoint base functions - of_graph_get_next_endpoint() - of_graph_get_endpoint_count() - for_each_endpoint_of_node() But to handling "port" base things, it is not useful. We want to have "port" base functions, too. This patch-set adds it. Because current existing drivers couldn't use "port" base functions, it were implemented in a different way. This patch-set doesn't try to full-replace to avoid unknown bug, try easy / quick replace only for now, but easy to know how "port" base functions are needed. Because I can't test the driver which I can't use, non-ASoC drivers needs Tested-by, Acked-by. v1 -> v2 - tidyup function explain - add missing header on each files Kuninori Morimoto (13): of: property: add port base loop of: property: use of_graph_get_next_port() on of_graph_get_next_endpoint() of: property: add of_graph_get_next_endpoint_raw() drm: omapdrm: use of_graph_get_next_endpoint_raw() media: xilinx-tpg: use of_graph_get_next_endpoint_raw() ASoC: audio-graph-card: use of_graph_get_next_endpoint_raw() ASoC: audio-graph-card2: use of_graph_get_next_port() ASoC: audio-graph-card2: use of_graph_get_next_endpoint_raw() ASoC: test-component: use for_each_port_of_node() fbdev: omapfb: use of_graph_get_remote_port() fbdev: omapfb: use of_graph_get_next_port() fbdev: omapfb: use of_graph_get_next_endpoint_raw() fbdev: omapfb: use of_graph_get_next_endpoint() drivers/gpu/drm/omapdrm/dss/dpi.c | 2 +- drivers/gpu/drm/omapdrm/dss/sdi.c | 2 +- drivers/media/platform/xilinx/xilinx-tpg.c | 2 +- drivers/of/property.c | 92 +++++++++++++--- drivers/video/fbdev/omap2/omapfb/dss/dpi.c | 3 +- drivers/video/fbdev/omap2/omapfb/dss/dsi.c | 3 +- drivers/video/fbdev/omap2/omapfb/dss/dss-of.c | 101 +----------------- drivers/video/fbdev/omap2/omapfb/dss/dss.c | 9 +- drivers/video/fbdev/omap2/omapfb/dss/hdmi4.c | 3 +- drivers/video/fbdev/omap2/omapfb/dss/hdmi5.c | 3 +- drivers/video/fbdev/omap2/omapfb/dss/sdi.c | 3 +- drivers/video/fbdev/omap2/omapfb/dss/venc.c | 3 +- include/linux/of_graph.h | 23 ++++ include/video/omapfb_dss.h | 11 -- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/audio-graph-card2.c | 31 ++---- sound/soc/generic/test-component.c | 2 +- 17 files changed, 133 insertions(+), 162 deletions(-) -- 2.25.1

6 37

[PATCH] ALSA: hda/realtek: add IDs for Dell dual spk platform
by shumingf＠realtek.com 05 Feb '24

05 Feb '24

From: Shuming Fan <shumingf(a)realtek.com> This patch adds another two IDs for the Dell dual speaker platform. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dde21bebed17..497c3e829b0f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9581,7 +9581,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x0beb, "Dell XPS 15 9530 (2023)", ALC289_FIXUP_DELL_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0c03, "Dell Precision 5340", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0c0b, "Dell Oasis 14 RPL-P", ALC289_FIXUP_RTK_AMP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0c0d, "Dell Oasis", ALC289_FIXUP_RTK_AMP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0c0e, "Dell Oasis 16", ALC289_FIXUP_RTK_AMP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0c19, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS), SND_PCI_QUIRK(0x1028, 0x0c1a, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS), SND_PCI_QUIRK(0x1028, 0x0c1b, "Dell Precision 3440", ALC236_FIXUP_DELL_DUAL_CODECS), -- 2.34.1

2 1

[PATCH v2] ALSA: hda/realtek: Remove two HP Laptops using CS35L41
by Stefan Binding 04 Feb '24

04 Feb '24

The SKUs, and associated SSIDs, are no longer going to include the CS35L41. They may come back, but will need a different quirk. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/patch_realtek.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 348505edf063..3d7e7b32fcf4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9855,8 +9855,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8cdd, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8cde, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8d01, "HP EliteBook G12", ALC287_FIXUP_CS35L41_I2C_4), - SND_PCI_QUIRK(0x103c, 0x8d08, "HP EliteBook 1045 G12", ALC287_FIXUP_CS35L41_I2C_4), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.34.1

2 1

[PATCH v8 00/13] Introduce STM32 Firewall framework
by Gatien Chevallier 03 Feb '24

03 Feb '24

Introduce STM32 Firewall framework for STM32MP1x and STM32MP2x platforms. STM32MP1x(ETZPC) and STM32MP2x(RIFSC) Firewall controllers register to the framework to offer firewall services such as access granting. This series of patches is a new approach on the previous STM32 system bus, history is available here: https://lore.kernel.org/lkml/20230127164040.1047583/ The need for such framework arises from the fact that there are now multiple hardware firewalls implemented across multiple products. Drivers are shared between different products, using the same code. When it comes to firewalls, the purpose mostly stays the same: Protect hardware resources. But the implementation differs, and there are multiple types of firewalls: peripheral, memory, ... Some hardware firewall controllers such as the RIFSC implemented on STM32MP2x platforms may require to take ownership of a resource before being able to use it, hence the requirement for firewall services to take/release the ownership of such resources. On the other hand, hardware firewall configurations are becoming more and more complex. These mecanisms prevent platform crashes or other firewall-related incoveniences by denying access to some resources. The stm32 firewall framework offers an API that is defined in firewall controllers drivers to best fit the specificity of each firewall. For every peripherals protected by either the ETZPC or the RIFSC, the firewall framework checks the firewall controlelr registers to see if the peripheral's access is granted to the Linux kernel. If not, the peripheral is configured as secure, the node is marked populated, so that the driver is not probed for that device. The firewall framework relies on the access-controller device tree binding. It is used by peripherals to reference a domain access controller. In this case a firewall controller. The bus uses the ID referenced by the access-controller property to know where to look in the firewall to get the security configuration for the peripheral. This allows a device tree description rather than a hardcoded peripheral table in the bus driver. The STM32 ETZPC device is responsible for filtering accesses based on security level, or co-processor isolation for any resource connected to it. The RIFSC is responsible for filtering accesses based on Compartment ID / security level / privilege level for any resource connected to it. STM32MP13/15/25 SoC device tree files are updated in this series to implement this mecanism. Changes in V8: - Add missing "simple-bus" compatible in STM32MP13/25 SoC device tree files for the ETZPC/RIFSC nodes - Add missing dependency on OF for OF_DYNAMIC that is selected by STM32_FIREWALL Changes in V7: - Separate indentation changes from access-controllers changes in the device tree file commits - Select OF_DYNAMIC when STM32_FIREWALL is set in order to use of_detach_node() in the firewall framework - Handle previously non-present RNG and HASH nodes in the STM32MP13 device tree file Changes in V6: - Rename access-controller to access-controllers - Remove access-controller-provider - Update device trees and other bindings accordingly - Rework ETZPC/RIFSC bindings to define what access-controllers cells contain inside #access-controller-cells - Some other minor fixes Changes in V5: - Integrate and rework the "feature-domains" binding patch in this patchset. The binding is renamed to "access-controller" - Rename every feature-domain* reference to access-control* ones - Correct loop bug and missing select STM32_FIREWALL in 32-bit platform Kconfig Changes in V4: - Fix typo in commit message and YAML check errors in "dt-bindings: Document common device controller bindings" Note: This patch should be ignored as stated in the cover letter. I've done this to avoid errors on this series of patch - Correct code syntax/style issues reported by Simon Horman - Added Jonathan's tag for IIO on the treewide patch Changes in V3: Change incorrect ordering for bindings commits leading to an error while running "make DT_CHECKER_FLAGS=-m dt_binding_check" Changes in V2: generic: - Add fw_devlink dependency for "feature-domains" property. bindings: - Corrected YAMLS errors highlighted by Rob's robot - Firewall controllers YAMLs no longer define the maxItems for the "feature-domains" property - Renamed st,stm32-rifsc.yaml to st,stm32mp25-rifsc.yaml - Fix examples in YAML files - Change feature-domains maxItems to 2 in firewall consumer files as there should not be more than 2 entries for now - Declare "feature-domain-names" as an optional property for firewall controllers child nodes. - Add missing "feature-domains" property declaration in bosch,m_can.yaml and st,stm32-cryp.yaml files firewall framework: - Support multiple entries for "feature-domains" property - Better handle the device-tree parsing using phandle+args APIs - Remove "resource firewall" type - Add a field for the name of the firewall entry - Fix licenses RIFSC: - Add controller name - Driver is now a module_platform_driver - Fix license ETZPC: - Add controller name - Driver is now a module_platform_driver - Fix license Device trees: - Fix rifsc node name - Move the "ranges" property under the "feature-domains" one Gatien Chevallier (12): dt-bindings: treewide: add access-controllers description dt-bindings: bus: document RIFSC dt-bindings: bus: document ETZPC firewall: introduce stm32_firewall framework of: property: fw_devlink: Add support for "access-controller" bus: rifsc: introduce RIFSC firewall controller driver arm64: dts: st: add RIFSC as an access controller for STM32MP25x boards bus: etzpc: introduce ETZPC firewall controller driver ARM: dts: stm32: add ETZPC as a system bus for STM32MP15x boards ARM: dts: stm32: put ETZPC as an access controller for STM32MP15x boards ARM: dts: stm32: add ETZPC as a system bus for STM32MP13x boards ARM: dts: stm32: put ETZPC as an access controller for STM32MP13x boards Oleksii Moisieiev (1): dt-bindings: document generic access controllers .../access-controllers.yaml | 84 + .../bindings/bus/st,stm32-etzpc.yaml | 87 + .../bindings/bus/st,stm32mp25-rifsc.yaml | 96 + .../bindings/crypto/st,stm32-cryp.yaml | 4 + .../bindings/crypto/st,stm32-hash.yaml | 4 + .../devicetree/bindings/dma/st,stm32-dma.yaml | 4 + .../bindings/dma/st,stm32-dmamux.yaml | 4 + .../devicetree/bindings/i2c/st,stm32-i2c.yaml | 4 + .../bindings/iio/adc/st,stm32-adc.yaml | 4 + .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 4 + .../bindings/iio/dac/st,stm32-dac.yaml | 4 + .../bindings/media/cec/st,stm32-cec.yaml | 4 + .../bindings/media/st,stm32-dcmi.yaml | 4 + .../memory-controllers/st,stm32-fmc2-ebi.yaml | 4 + .../bindings/mfd/st,stm32-lptimer.yaml | 4 + .../bindings/mfd/st,stm32-timers.yaml | 4 + .../devicetree/bindings/mmc/arm,pl18x.yaml | 4 + .../bindings/net/can/bosch,m_can.yaml | 4 + .../devicetree/bindings/net/stm32-dwmac.yaml | 4 + .../bindings/phy/phy-stm32-usbphyc.yaml | 4 + .../bindings/regulator/st,stm32-vrefbuf.yaml | 4 + .../devicetree/bindings/rng/st,stm32-rng.yaml | 4 + .../bindings/serial/st,stm32-uart.yaml | 4 + .../bindings/sound/st,stm32-i2s.yaml | 4 + .../bindings/sound/st,stm32-sai.yaml | 4 + .../bindings/sound/st,stm32-spdifrx.yaml | 4 + .../bindings/spi/st,stm32-qspi.yaml | 4 + .../devicetree/bindings/spi/st,stm32-spi.yaml | 4 + .../devicetree/bindings/usb/dwc2.yaml | 4 + MAINTAINERS | 7 + arch/arm/boot/dts/st/stm32mp131.dtsi | 1063 ++++--- arch/arm/boot/dts/st/stm32mp133.dtsi | 51 +- arch/arm/boot/dts/st/stm32mp13xc.dtsi | 19 +- arch/arm/boot/dts/st/stm32mp13xf.dtsi | 19 +- arch/arm/boot/dts/st/stm32mp151.dtsi | 2756 +++++++++-------- arch/arm/boot/dts/st/stm32mp153.dtsi | 52 +- arch/arm/boot/dts/st/stm32mp15xc.dtsi | 19 +- arch/arm/mach-stm32/Kconfig | 1 + arch/arm64/Kconfig.platforms | 1 + arch/arm64/boot/dts/st/stm32mp251.dtsi | 7 +- drivers/bus/Kconfig | 10 + drivers/bus/Makefile | 1 + drivers/bus/stm32_etzpc.c | 141 + drivers/bus/stm32_firewall.c | 294 ++ drivers/bus/stm32_firewall.h | 83 + drivers/bus/stm32_rifsc.c | 252 ++ drivers/of/property.c | 2 + include/linux/bus/stm32_firewall_device.h | 141 + 48 files changed, 3352 insertions(+), 1938 deletions(-) create mode 100644 Documentation/devicetree/bindings/access-controllers/access-controllers.yaml create mode 100644 Documentation/devicetree/bindings/bus/st,stm32-etzpc.yaml create mode 100644 Documentation/devicetree/bindings/bus/st,stm32mp25-rifsc.yaml create mode 100644 drivers/bus/stm32_etzpc.c create mode 100644 drivers/bus/stm32_firewall.c create mode 100644 drivers/bus/stm32_firewall.h create mode 100644 drivers/bus/stm32_rifsc.c create mode 100644 include/linux/bus/stm32_firewall_device.h -- 2.25.1

3 19

[PATCH v5 0/4] ASoC: qcom: volume fixes and codec cleanups
by Johan Hovold 03 Feb '24

03 Feb '24

To reduce the risk of speaker damage the PA gain needs to be limited on machines like the Lenovo Thinkpad X13s until we have active speaker protection in place. Limit the gain to the current default setting provided by the UCM configuration which most user have so far been using (due to a bug in the configuration files which prevented hardware volume control [1]). The wsa883x PA volume control also turned out to be broken, which meant that the default setting used by UCM configuration is actually the lowest level (-3 dB). With the codec driver fixed, hardware volume control also works as expected. Note that the new wsa884x driver most likely suffers from a similar bug, I'll send a fix for that once I've got that confirmed. Included is also a related fix for the LPASS WSA macro driver, which was changing the digital gain setting behind the back of user space and which can result in excessive (or too low) digital gain. There are further Qualcomm codec drivers that similarly appear to manipulate various gain settings, but on closer inspection it turns out that they only write back the current settings. Tests reveal that these writes are indeed needed for any prior updates to take effect (at least for the WSA and RX macros). Johan [1] https://github.com/alsa-project/alsa-ucm-conf/pull/382 Changes in v5 - rebase on v6.8-rc1 Changes in v4 - keep the full PA volume control range and only fix the exported dB values - use a combined -3 dB machine limit as limiting just the PA volume confuses PulseAudio - drop the PA gain initialisation which is no longer needed Changes in v3 - fix the wsa883x PA volume control and update the machine limits accordingly Changes in v2 - keep the volume register write on power-on in lpass-wsa-macro - drop the other patches removing volume register writes on DAPM events - only drop the constant-zero gain offsets in wcd9335 Johan Hovold (4): ASoC: codecs: wsa883x: fix PA volume control ASoC: qcom: sc8280xp: limit speaker volumes ASoC: codecs: lpass-wsa-macro: fix compander volume hack ASoC: codecs: wcd9335: drop unused gain hack remnant sound/soc/codecs/lpass-wsa-macro.c | 7 ------- sound/soc/codecs/wcd9335.c | 4 ---- sound/soc/codecs/wsa883x.c | 6 +++++- sound/soc/qcom/sc8280xp.c | 12 +++++++----- 4 files changed, 12 insertions(+), 17 deletions(-) -- 2.43.0

4 9

[PATCH AUTOSEL 6.6 16/21] spi: cs42l43: Handle error from devm_pm_runtime_enable
by Sasha Levin 02 Feb '24

02 Feb '24

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit f9f4b0c6425eb9ffd9bf62b8b8143e786b6ba695 ] As it devm_pm_runtime_enable can fail due to memory allocations, it is best to handle the error. Suggested-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://msgid.link/r/20240124174101.2270249-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/spi/spi-cs42l43.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/drivers/spi/spi-cs42l43.c b/drivers/spi/spi-cs42l43.c index d239fc5a49cc..c1556b652909 100644 --- a/drivers/spi/spi-cs42l43.c +++ b/drivers/spi/spi-cs42l43.c @@ -244,7 +244,10 @@ static int cs42l43_spi_probe(struct platform_device *pdev) priv->ctlr->use_gpio_descriptors = true; priv->ctlr->auto_runtime_pm = true; - devm_pm_runtime_enable(priv->dev); + ret = devm_pm_runtime_enable(priv->dev); + if (ret) + return ret; + pm_runtime_idle(priv->dev); regmap_write(priv->regmap, CS42L43_TRAN_CONFIG6, CS42L43_FIFO_SIZE - 1); -- 2.43.0

1 0

[PATCH AUTOSEL 6.7 17/23] spi: cs42l43: Handle error from devm_pm_runtime_enable
by Sasha Levin 02 Feb '24

02 Feb '24

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit f9f4b0c6425eb9ffd9bf62b8b8143e786b6ba695 ] As it devm_pm_runtime_enable can fail due to memory allocations, it is best to handle the error. Suggested-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://msgid.link/r/20240124174101.2270249-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/spi/spi-cs42l43.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/drivers/spi/spi-cs42l43.c b/drivers/spi/spi-cs42l43.c index d239fc5a49cc..c1556b652909 100644 --- a/drivers/spi/spi-cs42l43.c +++ b/drivers/spi/spi-cs42l43.c @@ -244,7 +244,10 @@ static int cs42l43_spi_probe(struct platform_device *pdev) priv->ctlr->use_gpio_descriptors = true; priv->ctlr->auto_runtime_pm = true; - devm_pm_runtime_enable(priv->dev); + ret = devm_pm_runtime_enable(priv->dev); + if (ret) + return ret; + pm_runtime_idle(priv->dev); regmap_write(priv->regmap, CS42L43_TRAN_CONFIG6, CS42L43_FIFO_SIZE - 1); -- 2.43.0

1 0

[PATCH v1] ALSA: hda/realtek: Remove two HP Laptops using CS35L41
by Stefan Binding 02 Feb '24

02 Feb '24

These two HP laptops are changing configurations. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/patch_realtek.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 348505edf063..3d7e7b32fcf4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9855,8 +9855,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8cdd, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8cde, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8d01, "HP EliteBook G12", ALC287_FIXUP_CS35L41_I2C_4), - SND_PCI_QUIRK(0x103c, 0x8d08, "HP EliteBook 1045 G12", ALC287_FIXUP_CS35L41_I2C_4), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.34.1

2 3

[PATCH 1/2] ASoC: dt-bindings: Added schema for "nuvoton,nau8325"
by Seven Lee 02 Feb '24

02 Feb '24

Added a DT schema for describing nau8325 audio amplifiers. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../bindings/sound/nuvoton,nau8325.yaml | 82 +++++++++++++++++++ 1 file changed, 82 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml new file mode 100644 index 000000000000..9105985357aa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml @@ -0,0 +1,82 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8325.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8325 audio Amplifier + +maintainers: + - Seven Lee <WTLI(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nuvoton,nau8325 + + reg: + maxItems: 1 + + nuvoton,vref-impedance: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + VREF impedance selection. + enum: + - 0 # Open + - 1 # 25kOhm + - 2 # 125kOhm + - 3 # 2.5kOhm + default: 2 + + nuvoton,dac-vref: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + DAC Reference Voltage Setting. + enum: + - 0 # VDDA + - 1 # VDDA*1.5/1.8V + - 2 # VDDA*1.6/1.8V + - 3 # VDDA*1.7/1.8V + default: 2 + + nuvoton,alc-enable: + description: + Enable digital automatic level control (ALC) function. + type: boolean + + nuvoton,clock-detection-disable: + description: + When clock detection is enabled, it will detect whether MCLK + and FS are within the range. MCLK range is from 2.048MHz to 24.576MHz. + FS range is from 8kHz to 96kHz. + type: boolean + + nuvoton,clock-det-data: + description: + Request clock detection to require 2048 non-zero samples before enabling + the audio paths. If set then non-zero samples is required, otherwise it + doesn't matter. + type: boolean + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@21 { + compatible = "nuvoton,nau8325"; + reg = <0x21>; + nuvoton,vref-impedance = <2>; + nuvoton,dac-vref = <2>; + nuvoton,alc-enable; + nuvoton,clock-det-data; + }; + }; -- 2.25.1

4 11

[GIT PULL] ASoC fixes for v6.8-rc2-2
by Mark Brown 01 Feb '24

01 Feb '24

The following changes since commit 41bccc98fb7931d63d03f326a746ac4d429c1dd3: Linux 6.8-rc2 (2024-01-28 17:01:12 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.8-rc2-2 for you to fetch changes up to e81fdba0208666b65bafeaba814874b4d6e5edde: ALSA: Various fixes for Cirrus Logic CS35L56 support (2024-02-01 17:45:32 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.8 This pull request adds Richard Fitzgerald's series with extensive fixes for the CS35L56, he said: These patches fix various things that were undocumented, unknown or uncertain when the original driver code was written. And also a few things that were just bugs. ---------------------------------------------------------------- Chen-Yu Tsai (3): ASoC: sunxi: sun4i-spdif: Add support for Allwinner H616 ASoC: sun4i-spdif: Fix requirements for H6 ASoC: sun4i-spdif: Add Allwinner H616 compatible Fei Shao (1): ASoC: codecs: ES8326: Remove executable bit Johan Hovold (5): ASoC: codecs: wsa883x: fix PA volume control ASoC: codecs: lpass-wsa-macro: fix compander volume hack ASoC: codecs: wcd9335: drop unused gain hack remnant ASoC: codecs: wcd938x: fix headphones volume controls ASoC: qcom: sc8280xp: limit speaker volumes Krzysztof Kozlowski (4): ASoC: codecs: wcd938x: handle deferred probe ASoC: codecs: wcd938x: skip printing deferred probe failuers ASoC: codecs: wcd934x: drop unneeded regulator include ASoC: allow up to eight CPU/codec DAIs Marian Postevca (1): ASoC: amd: acp: Fix support for a Huawei Matebook laptop Mark Brown (4): ASoC: codecs: fix ES8326 performance and pop noise ASoC: qcom: volume fixes and codec cleanups arm64: sun50i-h616: Add DMA and SPDIF controllers ALSA: Various fixes for Cirrus Logic CS35L56 support Richard Fitzgerald (18): ASoC: wm_adsp: Fix firmware file search order ASoC: wm_adsp: Don't overwrite fwf_name with the default ASoC: cs35l56: cs35l56_component_remove() must clear cs35l56->component ASoC: cs35l56: cs35l56_component_remove() must clean up wm_adsp ASoC: cs35l56: Don't add the same register patch multiple times ASoC: cs35l56: Remove buggy checks from cs35l56_is_fw_reload_needed() ASoC: cs35l56: Fix to ensure ASP1 registers match cache ASoC: cs35l56: Fix default SDW TX mixer registers ALSA: hda: cs35l56: Initialize all ASP1 registers ASoC: cs35l56: Fix for initializing ASP1 mixer registers ASoC: cs35l56: Fix misuse of wm_adsp 'part' string for silicon revision ASoC: cs35l56: Firmware file must match the version of preloaded firmware ASoC: cs35l56: Load tunings for the correct speaker models ASoC: cs35l56: Allow more time for firmware to boot ALSA: hda: cs35l56: Fix order of searching for firmware files ALSA: hda: cs35l56: Fix filename string field layout ALSA: hda: cs35l56: Firmware file must match the version of preloaded firmware ALSA: hda: cs35l56: Remove unused test stub function Techno Mooney (1): ASoC: amd: yc: Add DMI quirk for MSI Bravo 15 C7VF Venkata Prasad Potturu (3): ASoC: amd: acp: Enable rt5682s clocks in acp slave mode ASoC: amd: acp: Update platform name for different boards ASoC: amd: acp: Add check for cpu dai link initialization Zhu Ning (4): ASoC: codecs: ES8326: improving crosstalk performance ASoC: codecs: ES8326: Improving the THD+N performance ASoC: codecs: ES8326: Minimize the pop noise on headphone ASoC: codecs: ES8326: fix the capture noise issue .../bindings/sound/allwinner,sun4i-a10-spdif.yaml | 5 +- include/sound/cs35l56.h | 7 +- sound/pci/hda/cs35l56_hda.c | 136 ++++++---- sound/soc/amd/acp/acp-mach-common.c | 16 +- sound/soc/amd/acp/acp-sof-mach.c | 4 + sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 8 + sound/soc/amd/yc/acp6x-mach.c | 7 + sound/soc/codecs/cs35l56-shared.c | 140 ++++++++-- sound/soc/codecs/cs35l56.c | 301 ++++++++++++++++++--- sound/soc/codecs/cs35l56.h | 2 + sound/soc/codecs/es8326.c | 186 +++++++++---- sound/soc/codecs/es8326.h | 3 +- sound/soc/codecs/lpass-wsa-macro.c | 7 - sound/soc/codecs/wcd9335.c | 4 - sound/soc/codecs/wcd934x.c | 1 - sound/soc/codecs/wcd938x.c | 8 +- sound/soc/codecs/wm_adsp.c | 73 +++-- sound/soc/codecs/wsa883x.c | 6 +- sound/soc/qcom/sc8280xp.c | 12 +- sound/soc/soc-core.c | 5 +- sound/soc/sunxi/sun4i-spdif.c | 5 + 21 files changed, 696 insertions(+), 240 deletions(-) mode change 100755 => 100644 sound/soc/codecs/es8326.c

2 2

[PATCH 00/18] ALSA: Various fixes for Cirrus Logic CS35L56 support
by Richard Fitzgerald 01 Feb '24

01 Feb '24

This chain of patches fixes various things that were undocumented, unknown or uncertain when the original driver code was written. And also a few things that were just bugs. The HDA patches have dependencies on the ASoC patches, except for the final patch that removes a bogus test stub function. Richard Fitzgerald (18): ASoC: wm_adsp: Fix firmware file search order ASoC: wm_adsp: Don't overwrite fwf_name with the default ASoC: cs35l56: cs35l56_component_remove() must clear cs35l56->component ASoC: cs35l56: cs35l56_component_remove() must clean up wm_adsp ASoC: cs35l56: Don't add the same register patch multiple times ASoC: cs35l56: Remove buggy checks from cs35l56_is_fw_reload_needed() ASoC: cs35l56: Fix to ensure ASP1 registers match cache ASoC: cs35l56: Fix default SDW TX mixer registers ALSA: hda: cs35l56: Initialize all ASP1 registers ASoC: cs35l56: Fix for initializing ASP1 mixer registers ASoC: cs35l56: Fix misuse of wm_adsp 'part' string for silicon revision ASoC: cs35l56: Firmware file must match the version of preloaded firmware ASoC: cs35l56: Load tunings for the correct speaker models ASoC: cs35l56: Allow more time for firmware to boot ALSA: hda: cs35l56: Fix order of searching for firmware files ALSA: hda: cs35l56: Fix filename string field layout ALSA: hda: cs35l56: Firmware file must match the version of preloaded firmware ALSA: hda: cs35l56: Remove unused test stub function include/sound/cs35l56.h | 7 +- sound/pci/hda/cs35l56_hda.c | 138 ++++++++------ sound/soc/codecs/cs35l56-shared.c | 140 ++++++++++++-- sound/soc/codecs/cs35l56.c | 307 +++++++++++++++++++++++++----- sound/soc/codecs/cs35l56.h | 2 + sound/soc/codecs/wm_adsp.c | 73 +++---- 6 files changed, 498 insertions(+), 169 deletions(-) -- 2.39.2

4 24

[PATCH] ALSA: usb-audio: Ignore clock selector errors for single connection
by Alexander Tsoy 01 Feb '24

01 Feb '24

For devices with multiple clock sources connected to a selector, we need to check what a clock selector control request has returned. This is needed to ensure that a requested clock source is indeed selected and for autoclock feature to work. For devices with single clock source connected, if we get an error there is nothing else we can do about it. We can't skip clock selector setup as it is required by some devices. So lets just ignore error in this case. This should fix various buggy Mackie devices: [ 649.109785] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32) [ 649.111946] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32) [ 649.113822] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32) There is also interesting info from the Windows documentation [1] (this is probably why manufacturers dont't even test this feature): "The USB Audio 2.0 driver doesn't support clock selection. The driver uses the Clock Source Entity, which is selected by default and never issues a Clock Selector Control SET CUR request." Link: https://learn.microsoft.com/en-us/windows-hardware/drivers/audio/usb-2-0-au… [1] Link: https://bugzilla.kernel.org/show_bug.cgi?id=217314 Link: https://bugzilla.kernel.org/show_bug.cgi?id=218175 Link: https://bugzilla.kernel.org/show_bug.cgi?id=218342 Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/clock.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index a8204c6d6fac..60fcb872a80b 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -347,8 +347,16 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, !writeable) return ret; err = uac_clock_selector_set_val(chip, entity_id, cur); - if (err < 0) + if (err < 0) { + if (pins == 1) { + usb_audio_dbg(chip, + "%s(): selector returned an error, " + "assuming a firmware bug, id %d, ret %d\n", + __func__, clock_id, err); + return ret; + } return err; + } } if (!validate || ret > 0 || !chip->autoclock) -- 2.43.0

2 3

[PATCH] ALSA: usb-audio: retry clock validate when it takes some times
by vincenttew 01 Feb '24

01 Feb '24

Behringer UMC202HD and UMC404HD take some times to validate the clock source. To prevent more DAC have the same delay which return the false in the first time, remove the USB id check to have a chance to retry the clock validate. Signed-off-by: vincenttew <vincenttew(a)google.com> --- sound/usb/clock.c | 43 ++++++++++++++++++------------------------- 1 file changed, 18 insertions(+), 25 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b118cf97607f..16d7a456dd10 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -181,34 +181,27 @@ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, return true; } - /* - * MOTU MicroBook IIc - * Sample rate changes takes more than 2 seconds for this device. Clock - * validity request returns false during that period. - */ - if (chip->usb_id == USB_ID(0x07fd, 0x0004)) { - count = 0; + count = 0; - while ((!ret) && (count < 50)) { - int err; + while ((!ret) && (count < 50)) { + int err; - msleep(100); - - err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_CLOCK_VALID << 8, - snd_usb_ctrl_intf(chip) | (source_id << 8), - &data, sizeof(data)); - if (err < 0) { - dev_warn(&dev->dev, - "%s(): cannot get clock validity for id %d\n", - __func__, source_id); - return false; - } - - ret = !!data; - count++; + msleep(100); + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_CLOCK_VALID << 8, + snd_usb_ctrl_intf(chip) | (source_id << 8), + &data, sizeof(data)); + if (err < 0) { + dev_warn(&dev->dev, + "%s(): cannot get clock validity for id %d\n", + __func__, source_id); + return false; } + + ret = !!data; + count++; } return ret; -- 2.43.0.429.g432eaa2c6b-goog

3 2

[PATCH] ALSA: usb-audio: Add quirk for PioneerDJ DDJ-1000
by rp＠nrp.de 01 Feb '24

01 Feb '24

Problems: - PioneerDJ DDJ-1000 audio channels are not recognized. - Jog dial displays are showing permanently "NO AUDIO DRIVER" The attached patch fixes this. The jog displays and all output and input channels are working now. Signed-off-by: Ralf Pachali <rp(a)nrp.de> --- sound/usb/quirks-table.h +++ sound/usb/quirks-table.h @@ -3942,6 +3942,64 @@ } }, +{ + /* + * PIONEER DJ DDJ-1000 + * PCM is 6 channels out, 12 channels in @ 44.1 fixed + * The Feedback for the output is the input + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0020), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 6, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 12, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = -1 + } + } + } +}, + /* * MacroSilicon MS2100/MS2106 based AV capture cards * --- sound/usb/quirks.c +++ sound/usb/quirks.c @@ -1390,6 +1390,16 @@ return 0; } +static int snd_usb_pioneer_ddj1000_boot_quirk(struct usb_device *dev) +{ + /* disable "no audio driver" message on jog wheel displays */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0, + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0, 0x8003, NULL, 0); + + return 0; +} + /* * Setup quirks */ @@ -1630,6 +1640,8 @@ switch (id) { case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ return snd_usb_motu_m_series_boot_quirk(dev); + case USB_ID(0x2b73, 0x0020): /* Pioneer DDJ-1000 */ + return snd_usb_pioneer_ddj1000_boot_quirk(dev); } return 0;

2 1

[GIT PULL] ASoC fixes for v6.8-rc2
by Mark Brown 01 Feb '24

01 Feb '24

The following changes since commit 6613476e225e090cc9aad49be7fa504e290dd33d: Linux 6.8-rc1 (2024-01-21 14:11:32 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.8-rc2 for you to fetch changes up to 5513c5d0fb3d509cdd0a11afc18441c57eb7c94c: ASoC: amd: acp: Fix support for a Huawei Matebook laptop (2024-01-29 20:16:36 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.8 Quite a lot of fixes that came in since the merge window, a large portion for for Qualcomm and ES8326. The 8 DAI support for Qualcomm is just raising a constant to allow for devies that otherwise only need DTs, and there's a few other device ID updates for sunxi (Allwinner) and AMD platforms. ---------------------------------------------------------------- Chen-Yu Tsai (3): ASoC: sunxi: sun4i-spdif: Add support for Allwinner H616 ASoC: sun4i-spdif: Fix requirements for H6 ASoC: sun4i-spdif: Add Allwinner H616 compatible Fei Shao (1): ASoC: codecs: ES8326: Remove executable bit Johan Hovold (5): ASoC: codecs: wsa883x: fix PA volume control ASoC: codecs: lpass-wsa-macro: fix compander volume hack ASoC: codecs: wcd9335: drop unused gain hack remnant ASoC: codecs: wcd938x: fix headphones volume controls ASoC: qcom: sc8280xp: limit speaker volumes Krzysztof Kozlowski (4): ASoC: codecs: wcd938x: handle deferred probe ASoC: codecs: wcd938x: skip printing deferred probe failuers ASoC: codecs: wcd934x: drop unneeded regulator include ASoC: allow up to eight CPU/codec DAIs Marian Postevca (1): ASoC: amd: acp: Fix support for a Huawei Matebook laptop Mark Brown (3): ASoC: codecs: fix ES8326 performance and pop noise ASoC: qcom: volume fixes and codec cleanups arm64: sun50i-h616: Add DMA and SPDIF controllers Techno Mooney (1): ASoC: amd: yc: Add DMI quirk for MSI Bravo 15 C7VF Venkata Prasad Potturu (3): ASoC: amd: acp: Enable rt5682s clocks in acp slave mode ASoC: amd: acp: Update platform name for different boards ASoC: amd: acp: Add check for cpu dai link initialization Zhu Ning (4): ASoC: codecs: ES8326: improving crosstalk performance ASoC: codecs: ES8326: Improving the THD+N performance ASoC: codecs: ES8326: Minimize the pop noise on headphone ASoC: codecs: ES8326: fix the capture noise issue .../bindings/sound/allwinner,sun4i-a10-spdif.yaml | 5 +- sound/soc/amd/acp/acp-mach-common.c | 16 +- sound/soc/amd/acp/acp-sof-mach.c | 4 + sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 8 + sound/soc/amd/yc/acp6x-mach.c | 7 + sound/soc/codecs/es8326.c | 186 +++++++++++++++------ sound/soc/codecs/es8326.h | 3 +- sound/soc/codecs/lpass-wsa-macro.c | 7 - sound/soc/codecs/wcd9335.c | 4 - sound/soc/codecs/wcd934x.c | 1 - sound/soc/codecs/wcd938x.c | 8 +- sound/soc/codecs/wsa883x.c | 6 +- sound/soc/qcom/sc8280xp.c | 12 +- sound/soc/soc-core.c | 5 +- sound/soc/sunxi/sun4i-spdif.c | 5 + 15 files changed, 202 insertions(+), 75 deletions(-) mode change 100755 => 100644 sound/soc/codecs/es8326.c

2 1

[GIT PULL] ASoC fixes for v6.8-rc2
by Mark Brown 01 Feb '24

01 Feb '24

The following changes since commit 6613476e225e090cc9aad49be7fa504e290dd33d: Linux 6.8-rc1 (2024-01-21 14:11:32 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.8-rc2 for you to fetch changes up to eeab239d6a2418fc5d2cd7ea76187085a97acde0: ASoC: wcd934x: fix an incorrect use of kstrndup() (2024-01-29 23:08:56 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.8 Quite a lot of fixes that came in since the merge window, a large portion for for Qualcomm and ES8326. The 8 DAI support for Qualcomm is just raising a constant to allow for devies that otherwise only need DTs, and there's a few other device ID updates for sunxi (Allwinner) and AMD platforms. ---------------------------------------------------------------- Chen-Yu Tsai (3): ASoC: sunxi: sun4i-spdif: Add support for Allwinner H616 ASoC: sun4i-spdif: Fix requirements for H6 ASoC: sun4i-spdif: Add Allwinner H616 compatible Fei Shao (1): ASoC: codecs: ES8326: Remove executable bit Fullway Wang (1): ASoC: wcd934x: fix an incorrect use of kstrndup() Johan Hovold (5): ASoC: codecs: wsa883x: fix PA volume control ASoC: codecs: lpass-wsa-macro: fix compander volume hack ASoC: codecs: wcd9335: drop unused gain hack remnant ASoC: codecs: wcd938x: fix headphones volume controls ASoC: qcom: sc8280xp: limit speaker volumes Krzysztof Kozlowski (4): ASoC: codecs: wcd938x: handle deferred probe ASoC: codecs: wcd938x: skip printing deferred probe failuers ASoC: codecs: wcd934x: drop unneeded regulator include ASoC: allow up to eight CPU/codec DAIs Marian Postevca (1): ASoC: amd: acp: Fix support for a Huawei Matebook laptop Mark Brown (3): ASoC: codecs: fix ES8326 performance and pop noise ASoC: qcom: volume fixes and codec cleanups arm64: sun50i-h616: Add DMA and SPDIF controllers Techno Mooney (1): ASoC: amd: yc: Add DMI quirk for MSI Bravo 15 C7VF Venkata Prasad Potturu (3): ASoC: amd: acp: Enable rt5682s clocks in acp slave mode ASoC: amd: acp: Update platform name for different boards ASoC: amd: acp: Add check for cpu dai link initialization Zhu Ning (4): ASoC: codecs: ES8326: improving crosstalk performance ASoC: codecs: ES8326: Improving the THD+N performance ASoC: codecs: ES8326: Minimize the pop noise on headphone ASoC: codecs: ES8326: fix the capture noise issue .../bindings/sound/allwinner,sun4i-a10-spdif.yaml | 5 +- sound/soc/amd/acp/acp-mach-common.c | 16 +- sound/soc/amd/acp/acp-sof-mach.c | 4 + sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 8 + sound/soc/amd/yc/acp6x-mach.c | 7 + sound/soc/codecs/es8326.c | 186 +++++++++++++++------ sound/soc/codecs/es8326.h | 3 +- sound/soc/codecs/lpass-wsa-macro.c | 7 - sound/soc/codecs/wcd9335.c | 4 - sound/soc/codecs/wcd934x.c | 3 +- sound/soc/codecs/wcd938x.c | 8 +- sound/soc/codecs/wsa883x.c | 6 +- sound/soc/qcom/sc8280xp.c | 12 +- sound/soc/soc-core.c | 5 +- sound/soc/sunxi/sun4i-spdif.c | 5 + 15 files changed, 203 insertions(+), 76 deletions(-) mode change 100755 => 100644 sound/soc/codecs/es8326.c

1 1

Add UCM2 Configs for Qualcomm sm8550 HDK platform
by GitHub pull_request - opened 01 Feb '24

01 Feb '24

alsa-project/alsa-ucm-conf pull request #393 was opened from superna9999: This adds the UCM2 Configs for Qualcomm SM8550 HDK platform. Tested with PulseAudio 16.1. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/393 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/393.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v12 00/41] Introduce QC USB SND audio offloading support
by Wesley Cheng 01 Feb '24

01 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 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 (3): xhci: add support to allocate several interrupters xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (38): 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 dt-bindings: usb: xhci: Add num-hc-interrupters definition dt-bindings: usb: dwc3: Limit num-hc-interrupters definition 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 + .../devicetree/bindings/usb/snps,dwc3.yaml | 4 + .../devicetree/bindings/usb/usb-xhci.yaml | 6 + 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-debugfs.c | 2 +- drivers/usb/host/xhci-mem.c | 113 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 48 +- drivers/usb/host/xhci-sideband.c | 374 ++++ drivers/usb/host/xhci.c | 115 +- drivers/usb/host/xhci.h | 19 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-sideband.h | 67 + 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 | 81 + 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 | 1909 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 52 files changed, 6163 insertions(+), 82 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

2 49

[PATCH v7 0/6] Add cs42l43 PC focused SoundWire CODEC
by Charles Keepax 31 Jan '24

31 Jan '24

This patch chain adds support for the Cirrus Logic cs42l43 PC focused SoundWire CODEC. The chain is currently based of Lee's for-mfd-next branch. This series is mostly just a resend keeping pace with the kernel under it, except for a minor fixup in the ASoC stuff. Thanks, Charles Charles Keepax (4): dt-bindings: mfd: cirrus,cs42l43: Add initial DT binding mfd: cs42l43: Add support for cs42l43 core driver pinctrl: cs42l43: Add support for the cs42l43 ASoC: cs42l43: Add support for the cs42l43 Lucas Tanure (2): soundwire: bus: Allow SoundWire peripherals to register IRQ handlers spi: cs42l43: Add SPI controller support .../bindings/sound/cirrus,cs42l43.yaml | 313 +++ MAINTAINERS | 4 + drivers/mfd/Kconfig | 23 + drivers/mfd/Makefile | 3 + drivers/mfd/cs42l43-i2c.c | 98 + drivers/mfd/cs42l43-sdw.c | 239 ++ drivers/mfd/cs42l43.c | 1188 +++++++++ drivers/mfd/cs42l43.h | 28 + drivers/pinctrl/cirrus/Kconfig | 11 + drivers/pinctrl/cirrus/Makefile | 2 + drivers/pinctrl/cirrus/pinctrl-cs42l43.c | 609 +++++ drivers/soundwire/bus.c | 32 + drivers/soundwire/bus_type.c | 12 + drivers/spi/Kconfig | 7 + drivers/spi/Makefile | 1 + drivers/spi/spi-cs42l43.c | 284 ++ include/linux/mfd/cs42l43-regs.h | 1184 +++++++++ include/linux/mfd/cs42l43.h | 102 + include/linux/soundwire/sdw.h | 9 + include/sound/cs42l43.h | 17 + sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42l43-jack.c | 946 +++++++ sound/soc/codecs/cs42l43-sdw.c | 74 + sound/soc/codecs/cs42l43.c | 2278 +++++++++++++++++ sound/soc/codecs/cs42l43.h | 131 + 26 files changed, 7615 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml create mode 100644 drivers/mfd/cs42l43-i2c.c create mode 100644 drivers/mfd/cs42l43-sdw.c create mode 100644 drivers/mfd/cs42l43.c create mode 100644 drivers/mfd/cs42l43.h create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs42l43.c create mode 100644 drivers/spi/spi-cs42l43.c create mode 100644 include/linux/mfd/cs42l43-regs.h create mode 100644 include/linux/mfd/cs42l43.h create mode 100644 include/sound/cs42l43.h create mode 100644 sound/soc/codecs/cs42l43-jack.c create mode 100644 sound/soc/codecs/cs42l43-sdw.c create mode 100644 sound/soc/codecs/cs42l43.c create mode 100644 sound/soc/codecs/cs42l43.h -- 2.30.2

6 32

[v3 0/3] ASoC: mediatek:mt8186: fix both the speaker and hdmi
by Jiaxin Yu 31 Jan '24

31 Jan '24

The speaker and hdmi of mt8186 platform are shared the same port of I2S, when connect the external display, use build-in speakers to play audio, they both playback at the same time. So we want to manage the playback device through DAPM events. Jiaxin Yu (3): ASoC: hdmi-codec: Add event handler for hdmi TX ASoC: mediatek: mt8186: correct the HDMI widgets drm/bridge: it6505: Add audio support drivers/gpu/drm/bridge/ite-it6505.c | 81 +++++++++++++++++-- include/sound/hdmi-codec.h | 6 ++ sound/soc/codecs/hdmi-codec.c | 32 +++++++- .../mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 5 files changed, 113 insertions(+), 10 deletions(-) -- 2.25.1

5 13

ALSA-LIB: xrun recovery in iec958 plugin for 8ch HDMI?
by Pavel Hofman 31 Jan '24

31 Jan '24

Hi, Users of RPi's hdmi:vc4-hdmi report random channel swapping after an xrun. Tested (among other clients) on speaker-test, details https://forums.raspberrypi.com/viewtopic.php?p=2187582#p2187540 Experienced RPi developer has diagnosed the driver - HDMI device and DMA are correctly reset at xrun recovery (i.e. snd_pcm_stop(XRUN) in the driver -> snd_pcm_prepare in the client). Therefore an option is that alsa supplies misaligned 8ch samples after the xrun recovery. Users checked the older HDMI driver bcm2835_audio (bcm2835-pcm.c) which accepts standard S16/24_LE formats and does not require the iec958 plugin and xruns did not cause any channel swaps. Hence aiming at the iec958 plugin. It seems to keep state (e.g. iec->counter) which perhaps may not be reset at snd_pcm_prepare. Thanks a lot, Pavel.

1 1

ASoC: rockchip_i2s_tdm calibration clocking problem
by Pavel Hofman 31 Jan '24

31 Jan '24

Hi, I am hitting a clock issue with rockchip_i2s_tdm.c + simple-audio-card (+ RK3308). At boot the mclk clk_i2s0_8ch_tx is (somehow) initialized to some (unimportant?) value 50176000 Hz. Note that this frequency is not multiple of either 48kHz or 44.1kHz. Method asoc_simple_parse_clk() reads this value and sets it to simple_dai->sysclk. Subsequently at asoc_simple_dai_init this "random" initial value is stored to i2s_tdm->mclk_tx_freq: 17.839330] rockchip_i2s_tdm_set_sysclk+0x50/0xbc [snd_soc_rockchip_i2s_tdm] [ 17.839367] snd_soc_dai_set_sysclk+0x38/0xb8 [snd_soc_core] [ 17.839596] asoc_simple_init_dai+0x94/0xc0 [snd_soc_simple_card_utils] [ 17.839640] asoc_simple_dai_init+0x130/0x230 [snd_soc_simple_card_utils] [ 17.839672] snd_soc_link_init+0x28/0x90 [snd_soc_core] [ 17.839843] snd_soc_bind_card+0x60c/0xbb4 [snd_soc_core] When starting playback, called by rockchip_i2s_tdm_hw_params(), rockchip_i2s_tdm_calibrate_mclk() correctly switches parent of mclk_parent to correct root pll clock mclk_root0/1 for the given samplerate and correctly configures mclk_parent frequency. https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rockchip_i… But right after that, the next line of rockchip_i2s_tdm_hw_params() calls rockchip_i2s_tdm_set_mclk() https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rockchip_i… This method calls clk_set_rate(i2s_tdm->mclk_tx, i2s_tdm->mclk_tx_freq), which resets the clock and its parental chain to the original incorrect value stored in i2s_tdm->mclk_tx_freq from the dai initialization. https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rockchip_i… As a result, no matter what sample rate is being played, the i2s mclk clock always ends up configured incorrectly. DTS I2S sets all clocks, therefore the clk calibration in rockchip_i2s_tdm.c should be (and is) used: i2s_8ch_0: i2s@ff300000 { compatible = "rockchip,rk3308-i2s-tdm"; reg = <0x0 0xff300000 0x0 0x1000>; interrupts = <GIC_SPI 48 IRQ_TYPE_LEVEL_HIGH>; clocks = <&cru SCLK_I2S0_8CH_TX>, <&cru SCLK_I2S0_8CH_RX>, <&cru HCLK_I2S0_8CH>, <&cru SCLK_I2S0_8CH_TX_SRC>, <&cru SCLK_I2S0_8CH_RX_SRC>, <&cru PLL_VPLL0>, <&cru PLL_VPLL1>; clock-names = "mclk_tx", "mclk_rx", "hclk", "mclk_tx_src", "mclk_rx_src", "mclk_root0", "mclk_root1"; ......... It seems to me that the calibration code should also rewrite the initially incorrect i2s_tdm->mclk_tx_freq and i2s_tdm->mclk_rx_freq with correct values corresponding to the momentary hw_params rate, or maybe rockchip_i2s_tdm_set_mclk() should not be called if rockchip_i2s_tdm_calibrate_mclk() is called a line above (i.e. putting the call to rockchip_i2s_tdm_set_mclk() into "else" branch). Thank you very much for help. With regards, Pavel.

1 0

[PATCH 1/2] dt-bindings: pinctrl: Unify "input-debounce" schema
by Rob Herring 31 Jan '24

31 Jan '24

nuvoton,npcm845-pinctrl defines the common "input-debounce" property as an array rather than an scalar. Update the common definition to expand it to an uint32-array, and update all the users of the property with array constraints. Signed-off-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml | 3 ++- .../devicetree/bindings/pinctrl/nuvoton,npcm845-pinctrl.yaml | 1 - .../devicetree/bindings/pinctrl/nuvoton,wpcm450-pinctrl.yaml | 3 ++- Documentation/devicetree/bindings/pinctrl/pincfg-node.yaml | 2 +- 4 files changed, 5 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml b/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml index bb61a30321a1..482acda88e73 100644 --- a/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml +++ b/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml @@ -93,7 +93,8 @@ properties: input-schmitt-disable: true - input-debounce: true + input-debounce: + maxItems: 1 output-low: true diff --git a/Documentation/devicetree/bindings/pinctrl/nuvoton,npcm845-pinctrl.yaml b/Documentation/devicetree/bindings/pinctrl/nuvoton,npcm845-pinctrl.yaml index 3e8472898800..20cf0102aa63 100644 --- a/Documentation/devicetree/bindings/pinctrl/nuvoton,npcm845-pinctrl.yaml +++ b/Documentation/devicetree/bindings/pinctrl/nuvoton,npcm845-pinctrl.yaml @@ -152,7 +152,6 @@ patternProperties: description: Debouncing periods in microseconds, one period per interrupt bank found in the controller - $ref: /schemas/types.yaml#/definitions/uint32-array minItems: 1 maxItems: 4 diff --git a/Documentation/devicetree/bindings/pinctrl/nuvoton,wpcm450-pinctrl.yaml b/Documentation/devicetree/bindings/pinctrl/nuvoton,wpcm450-pinctrl.yaml index 7b7f840ffc4c..08442c880f07 100644 --- a/Documentation/devicetree/bindings/pinctrl/nuvoton,wpcm450-pinctrl.yaml +++ b/Documentation/devicetree/bindings/pinctrl/nuvoton,wpcm450-pinctrl.yaml @@ -103,7 +103,8 @@ patternProperties: items: pattern: "^gpio1?[0-9]{1,2}$" - input-debounce: true + input-debounce: + maxItems: 1 additionalProperties: false diff --git a/Documentation/devicetree/bindings/pinctrl/pincfg-node.yaml b/Documentation/devicetree/bindings/pinctrl/pincfg-node.yaml index be81ed22a036..d0af21a564b4 100644 --- a/Documentation/devicetree/bindings/pinctrl/pincfg-node.yaml +++ b/Documentation/devicetree/bindings/pinctrl/pincfg-node.yaml @@ -97,7 +97,7 @@ properties: description: disable schmitt-trigger mode input-debounce: - $ref: /schemas/types.yaml#/definitions/uint32 + $ref: /schemas/types.yaml#/definitions/uint32-array description: Takes the debounce time in usec as argument or 0 to disable debouncing -- 2.43.0

4 5

[PATCH v3] ASoC: Intel: mtl-match: Add cs42l43_l0 cs35l56_l23 for MTL
by Maciej Strozek 30 Jan '24

30 Jan '24

The layout is configured as: - Link0: CS42L43 Jack and mics (2ch) - Link2: 2x CS35L56 Speaker (amps 3 and 4, right) - Link3: 2x CS35L56 Speaker (amps 1 and 2, left) Corresponding SOF topology: https://github.com/thesofproject/sof/pull/8773 Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- v2 -> v3: added layout description and link to required SOF topology v1 -> v2: swapped AMP3 and AMP4 prefixes .../intel/common/soc-acpi-intel-mtl-match.c | 57 +++++++++++++++++++ 1 file changed, 57 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index feb12c6c85d1..23eaf47b3f24 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -377,6 +377,37 @@ static const struct snd_soc_acpi_adr_device cs35l56_2_adr[] = { } }; +static const struct snd_soc_acpi_adr_device cs35l56_2_r_adr[] = { + { + .adr = 0x00023201FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP3" + }, + { + .adr = 0x00023301FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_3_endpoint, + .name_prefix = "AMP4" + } + +}; + +static const struct snd_soc_acpi_adr_device cs35l56_3_l_adr[] = { + { + .adr = 0x00033001fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00033101fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_2_endpoint, + .name_prefix = "AMP2" + } +}; + static const struct snd_soc_acpi_link_adr rt5682_link2_max98373_link0[] = { /* Expected order: jack -> amp */ { @@ -554,6 +585,26 @@ static const struct snd_soc_acpi_link_adr mtl_cs42l43_cs35l56[] = { {} }; +static const struct snd_soc_acpi_link_adr cs42l43_link0_cs35l56_link2_link3[] = { + /* Expected order: jack -> amp */ + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43_0_adr), + .adr_d = cs42l43_0_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(cs35l56_2_r_adr), + .adr_d = cs35l56_2_r_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(cs35l56_3_l_adr), + .adr_d = cs35l56_3_l_adr, + }, + {} +}; + /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { /* mockup tests need to be first */ @@ -599,6 +650,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-mtl-rt1318-l12-rt714-l0.tplg" }, + { + .link_mask = BIT(0) | BIT(2) | BIT(3), + .links = cs42l43_link0_cs35l56_link2_link3, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-cs42l43-l0-cs35l56-l23.tplg", + }, { .link_mask = GENMASK(2, 0), .links = mtl_cs42l43_cs35l56, -- 2.34.1

4 3

topology: correct version script path
by GitHub pull_request - opened 30 Jan '24

30 Jan '24

alsa-project/alsa-lib pull request #383 was opened from jpalus: contrary to libasound, version script for libatopology is a regular source file. while it's often the case that $(builddir) and $(srcdir) point to the same directory, they don't always have to. therefore path needs to point explicitly to $(srcdir) for Versions script in topology Fixes: GH-382 Fixes: dc7da761f3a2 ("topology: separate Versions linker script") Request URL : https://github.com/alsa-project/alsa-lib/pull/383 Patch URL : https://github.com/alsa-project/alsa-lib/pull/383.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Linking failure when using separate builddir
by GitHub issues - opened 30 Jan '24

30 Jan '24

alsa-project/alsa-lib issue #382 was opened from jpalus: If `$(builddir)` is different than `$(srcdir)` linking of `libatopology` fails: ``` $ mkdir builddir $ cd builddir $ ../configure ... $ make ... /usr/bin/ld: cannot open linker script file Versions: No such file or directory ``` `alsa-lib 1.2.11` Issue URL : https://github.com/alsa-project/alsa-lib/issues/382 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: hda: cs35l41: Support additional ASUS Zenbook UX3402VA
by Kenzo Gomez 30 Jan '24

30 Jan '24

Add new model entry into configuration table. Signed-off-by: Kenzo Gomez <kenzo.sgomez(a)gmail.com> --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index c1afb721b4c6..adfb31a6bf1e 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -67,6 +67,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431D1F", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431DA2", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "10431E02", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "104316A3", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "10431EE2", I2C, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "10431F12", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431F1F", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, @@ -371,6 +372,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431D1F", generic_dsd_config }, { "CSC3551", "10431DA2", generic_dsd_config }, { "CSC3551", "10431E02", generic_dsd_config }, + { "CSC3551", "104316A3", generic_dsd_config }, { "CSC3551", "10431EE2", generic_dsd_config }, { "CSC3551", "10431F12", generic_dsd_config }, { "CSC3551", "10431F1F", generic_dsd_config }, -- 2.43.0

3 7

[PATCH v2 1/7] ASoC: cs42l43: Tidy up header includes
by Charles Keepax 30 Jan '24

30 Jan '24

Use more forward declarations, move header guards to cover other includes, and rely less on including headers through other headers. Suggested-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Changes since v1: - Shuffle forward declarations around Thanks, Charles sound/soc/codecs/cs42l43-jack.c | 5 +++++ sound/soc/codecs/cs42l43-sdw.c | 1 + sound/soc/codecs/cs42l43.c | 8 ++++++++ sound/soc/codecs/cs42l43.h | 21 ++++++++++++--------- 4 files changed, 26 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c index 24a598f2ed9a3..1d8d7bf0a6b0d 100644 --- a/sound/soc/codecs/cs42l43-jack.c +++ b/sound/soc/codecs/cs42l43-jack.c @@ -6,19 +6,24 @@ // Cirrus Logic International Semiconductor Ltd. #include <linux/build_bug.h> +#include <linux/completion.h> #include <linux/delay.h> #include <linux/errno.h> #include <linux/irq.h> #include <linux/jiffies.h> #include <linux/mfd/cs42l43.h> #include <linux/mfd/cs42l43-regs.h> +#include <linux/mutex.h> #include <linux/pm_runtime.h> #include <linux/property.h> +#include <linux/regmap.h> +#include <linux/workqueue.h> #include <sound/control.h> #include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc-component.h> +#include <sound/soc-jack.h> #include <sound/soc.h> #include "cs42l43.h" diff --git a/sound/soc/codecs/cs42l43-sdw.c b/sound/soc/codecs/cs42l43-sdw.c index 388f95853b699..60c00c05da055 100644 --- a/sound/soc/codecs/cs42l43-sdw.c +++ b/sound/soc/codecs/cs42l43-sdw.c @@ -9,6 +9,7 @@ #include <linux/mfd/cs42l43.h> #include <linux/mfd/cs42l43-regs.h> #include <linux/module.h> +#include <linux/soundwire/sdw.h> #include <sound/pcm.h> #include <sound/sdw.h> #include <sound/soc-component.h> diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 6a64681767de8..f2332f90f8337 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -6,17 +6,25 @@ // Cirrus Logic International Semiconductor Ltd. #include <linux/bitops.h> +#include <linux/clk.h> +#include <linux/device.h> #include <linux/err.h> #include <linux/errno.h> #include <linux/gcd.h> #include <linux/irq.h> +#include <linux/irqdomain.h> #include <linux/jiffies.h> #include <linux/mfd/cs42l43.h> #include <linux/mfd/cs42l43-regs.h> +#include <linux/mod_devicetable.h> #include <linux/module.h> +#include <linux/platform_device.h> #include <linux/pm_runtime.h> +#include <linux/regmap.h> #include <linux/string.h> +#include <linux/workqueue.h> #include <sound/control.h> +#include <sound/cs42l43.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc-component.h> diff --git a/sound/soc/codecs/cs42l43.h b/sound/soc/codecs/cs42l43.h index 125e36861d5d5..9924c13e1eb53 100644 --- a/sound/soc/codecs/cs42l43.h +++ b/sound/soc/codecs/cs42l43.h @@ -6,19 +6,14 @@ * Cirrus Logic International Semiconductor Ltd. */ -#include <linux/clk.h> +#ifndef CS42L43_ASOC_INT_H +#define CS42L43_ASOC_INT_H + #include <linux/completion.h> -#include <linux/device.h> #include <linux/mutex.h> -#include <linux/regmap.h> -#include <linux/soundwire/sdw.h> #include <linux/types.h> -#include <sound/cs42l43.h> +#include <linux/workqueue.h> #include <sound/pcm.h> -#include <sound/soc-jack.h> - -#ifndef CS42L43_ASOC_INT_H -#define CS42L43_ASOC_INT_H #define CS42L43_INTERNAL_SYSCLK 24576000 #define CS42L43_DEFAULT_SLOTS 0x3F @@ -37,6 +32,14 @@ #define CS42L43_N_BUTTONS 6 +struct clk; +struct device; + +struct snd_soc_component; +struct snd_soc_jack; + +struct cs42l43; + struct cs42l43_codec { struct device *dev; struct cs42l43 *core; -- 2.30.2

3 12

[PATCH] ASoC: amd: acp: Fix support for a Huawei Matebook laptop
by Marian Postevca 30 Jan '24

30 Jan '24

Previous commit that added support for Huawei MateBook D16 2021 with Ryzen 4600H (HVY-WXX9 M1010) was incomplete. To activate support for this laptop, the DMI table in acp3x-es83xx machine driver must also be updated. Fixes: b5338b1b901e ("ASoC: amd: acp: Add support for a new Huawei Matebook laptop") Signed-off-by: Marian Postevca <posteuca(a)mutex.one> --- sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c index f85b85ea4be9..2b0aa270a3e9 100644 --- a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c +++ b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c @@ -354,6 +354,14 @@ static const struct dmi_system_id acp3x_es83xx_dmi_table[] = { }, .driver_data = (void *)(ES83XX_ENABLE_DMIC|ES83XX_48_MHZ_MCLK), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "HVY-WXX9"), + DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "M1010"), + }, + .driver_data = (void *)(ES83XX_ENABLE_DMIC), + }, { .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"), -- 2.43.0

2 1

[PATCH] ALSA: usb-audio: Check presence of valid altsetting control
by Alexander Tsoy 29 Jan '24

29 Jan '24

Many devices with a single alternate setting do not have a Valid Alternate Setting Control and validation performed by validate_sample_rate_table_v2v3() doesn't work on them and is not really needed. So check the presense of control before sending altsetting validation requests. MOTU Microbook IIc is suffering the most without this check. It takes up to 40 seconds to bootup due to how slow it switches sampling rates: [ 2659.164824] usb 3-2: New USB device found, idVendor=07fd, idProduct=0004, bcdDevice= 0.60 [ 2659.164827] usb 3-2: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [ 2659.164829] usb 3-2: Product: MicroBook IIc [ 2659.164830] usb 3-2: Manufacturer: MOTU [ 2659.166204] usb 3-2: Found last interface = 3 [ 2679.322298] usb 3-2: No valid sample rate available for 1:1, assuming a firmware bug [ 2679.322306] usb 3-2: 1:1: add audio endpoint 0x3 [ 2679.322321] usb 3-2: Creating new data endpoint #3 [ 2679.322552] usb 3-2: 1:1 Set sample rate 96000, clock 1 [ 2684.362250] usb 3-2: 2:1: cannot get freq (v2/v3): err -110 [ 2694.444700] usb 3-2: No valid sample rate available for 2:1, assuming a firmware bug [ 2694.444707] usb 3-2: 2:1: add audio endpoint 0x84 [ 2694.444721] usb 3-2: Creating new data endpoint #84 [ 2699.482103] usb 3-2: 2:1 Set sample rate 96000, clock 1 Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/format.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index ab5fed9f55b6..3b45d0ee7693 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -470,9 +470,11 @@ static int validate_sample_rate_table_v2v3(struct snd_usb_audio *chip, int clock) { struct usb_device *dev = chip->dev; + struct usb_host_interface *alts; unsigned int *table; unsigned int nr_rates; int i, err; + u32 bmControls; /* performing the rate verification may lead to unexpected USB bus * behavior afterwards by some unknown reason. Do this only for the @@ -481,6 +483,24 @@ static int validate_sample_rate_table_v2v3(struct snd_usb_audio *chip, if (!(chip->quirk_flags & QUIRK_FLAG_VALIDATE_RATES)) return 0; /* don't perform the validation as default */ + alts = snd_usb_get_host_interface(chip, fp->iface, fp->altsetting); + if (!alts) + return 0; + + if (fp->protocol == UAC_VERSION_3) { + struct uac3_as_header_descriptor *as = snd_usb_find_csint_desc( + alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); + bmControls = le32_to_cpu(as->bmControls); + } else { + struct uac2_as_header_descriptor *as = snd_usb_find_csint_desc( + alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); + bmControls = as->bmControls; + } + + if (!uac_v2v3_control_is_readable(bmControls, + UAC2_AS_VAL_ALT_SETTINGS)) + return 0; + table = kcalloc(fp->nr_rates, sizeof(*table), GFP_KERNEL); if (!table) return -ENOMEM; -- 2.43.0

2 1

pcm: document interaction of drain silence and sw silence parameter
by GitHub pull_request - opened 29 Jan '24

29 Jan '24

alsa-project/alsa-lib pull request #381 was opened from z-s-e: I'm not sure this is the best way to describe the interaction, but I think this needs to be mentioned in some form at least. Request URL : https://github.com/alsa-project/alsa-lib/pull/381 Patch URL : https://github.com/alsa-project/alsa-lib/pull/381.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

USB-Audio: ALC4080: Add support for MSI MPG B650 Carbon Wifi (0db0:70d3)
by GitHub pull_request - edited 29 Jan '24

29 Jan '24

alsa-project/alsa-ucm-conf pull request #392 was edited from lngr: This PR adds support for the MSI MPG B650 Carbon Wifi board. This makes the front headphone and microphone ports working (tested on my local hardware). The microphone sound volume is a bit low compared to windows, [as mentioned previously](https://github.com/alsa-project/alsa-ucm-conf/pull/143/files#r8…, but in general it works. Fixes #359 ``` $ sudo lsusb | grep Audio Bus 001 Device 003: ID 0db0:70d3 Micro Star International USB Audio $ alsaucm listcards 0: hw:4 Generic USB Audio at usb-0000:0f:00.0-6, high speed ``` Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/392 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/392.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: ALC4080: Add support for MSI MPG B650 Carbon Wifi (0db0:70d3)
by GitHub pull_request - opened 29 Jan '24

29 Jan '24

alsa-project/alsa-ucm-conf pull request #392 was opened from lngr: This PR adds support for the MSI MPG B650 Carbon Wifi board. This makes the front headphone and microphone ports working (tested on my local hardware). The microphone sound volume is a bit low compared to windows, [as mentioned previously](https://github.com/alsa-project/alsa-ucm-conf/pull/143/files#r8…, but in general it works. Fixes #359 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/392 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/392.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v5 0/6] reset: gpio: ASoC: shared GPIO resets
by Krzysztof Kozlowski 29 Jan '24

29 Jan '24

Hi, Patch #2 (cpufreq: do not open-code of_phandle_args_equal()) and patch #4 (reset: Instantiate reset GPIO controller for shared reset-gpios) depend on OF change (patch #1). Changes in v5 ============= 1. Minor comments from Philipp: missing cleanup.h, correct indentation of pr_err(), shorten gpio_device_find_by_fwnode() line. 2. Acks/Rbs. Changes in v4 ============= 1. New patches: of: add of_phandle_args_equal() helper cpufreq: do not open-code of_phandle_args_equal() 2. reset-gpio.c: - Drop unneeded comment (Bartosz), add Rb tag. - Do not assign of_node. 3. reset/core.c: - Implement most of Bartosz feedback (I responded to one which I did not implement) and comments from Philipp. - Expect either rcdev->of_args or rcdev->of_node. - Drop __reset_gpios_args_match() and use common helper (Philipp). - Move declarations of automatic-cleanup variables in __reset_add_reset_gpio_lookup() to place of use (Bartosz). - Separate gpio_device_get_label() and kstrdup() (Philipp). - Correct doc for __reset_add_reset_gpio_device(), rewrite few comments. - Drop unneeded "r" variable in __reset_find_rcdev() (Philipp). - Drop of_phandle_args initialization in __of_reset_control_get (Philipp). - Check if CONFIG_RESET_GPIO is enabled before trying to look up reset-gpios. 4. Drop Chris' patch: "i2c: muxes: pca954x: Allow sharing reset GPIO", because discussion is on going. Changes in v3 ============= 1. reset-gpio.c: - Add reset_gpio_of_xlate (Philipp). - reset_gpio_of_args_put->reset_gpio_of_node_put (Philipp). - Expect via platdata of_phandle_args. - Do not call device_set_node() to attach itself to reset consumer (the final device). This was questionable idea in the first place. Bartosz suggested to use GPIO_LOOKUP to solve this. 2. reset/core.c, implement Philipp's feedback. That was a lot: - Commit msg fixes. - Add new platform_device earlier, when reset core found "reset-gpios" but not "resets". - Do not overwrite of_phandle_args. - Expect matching .of_reset_n_cells. - Pass of_phandle_args as platdata to reset-gpio. - Rename reset_gpio_device->reset_gpio_lookup and others. Fix few comments and code cleanup pointed on review. - From Bartosz: Use GPIO_LOOKUP and a lot of cleanup.h in __reset_add_reset_gpio_lookup(). 3. Include here Chris' patch: "i2c: muxes: pca954x: Allow sharing reset GPIO". Changes in v2 ============= 1. wsa884x.c: add missing return in wsa884x_get_reset(), correct comment. 2. qcom,wsa8840.yaml: fix oneOf syntax. 3. reset-gpio.c: - Fix smatch warning on platdata evaluation. - Parse GPIO args and store them in rc.of_args. 4. reset/core.c: - Revise approach based on Bartosz comments: parse the reset-gpios phandle with arguments, do not use deprecated API and do not rely on gpio_desc pointer. - Create a list of instantiated platform devices to avoid any duplicates. - After creating reset-gpio platform device, try to get new reset controller or return EPROBE_DEFER. - Drop the "cookie" member and add new "of_args" to "struct reset_controller_dev". Description =========== We have at least few cases where hardware engineers decided to use one powerdown/shutdown/reset GPIO line for multiple devices: 1. WSA884x (this and previous patch): https://lore.kernel.org/all/b7aeda24-d638-45b7-8e30-80d287f498f8@sirena.org… 2. https://lore.kernel.org/all/20231027033104.1348921-1-chris.packham@alliedte… 3. https://lore.kernel.org/lkml/20191030120440.3699-1-peter.ujfalusi@ti.com/ 4. https://lore.kernel.org/all/20211018234923.1769028-1-sean.anderson@seco.com/ 5. https://social.treehouse.systems/@marcan/111268780311634160 I try to solve my case, hopefuly Chris' (2), partially Sean's (4) and maybe Hectors (5), using Rob's suggestion: https://lore.kernel.org/all/YXi5CUCEi7YmNxXM@robh.at.kernel.org/ Best regards, Krzysztof Cc: Chris Packham <chris.packham(a)alliedtelesis.co.nz> Cc: Bartosz Golaszewski <brgl(a)bgdev.pl> Cc: Sean Anderson <sean.anderson(a)seco.com> Krzysztof Kozlowski (6): of: Add of_phandle_args_equal() helper cpufreq: do not open-code of_phandle_args_equal() reset: gpio: Add GPIO-based reset controller reset: Instantiate reset GPIO controller for shared reset-gpios ASoC: dt-bindings: qcom,wsa8840: Add reset-gpios for shared line ASoC: codecs: wsa884x: Allow sharing reset GPIO .../bindings/sound/qcom,wsa8840.yaml | 11 +- MAINTAINERS | 5 + drivers/reset/Kconfig | 9 + drivers/reset/Makefile | 1 + drivers/reset/core.c | 215 ++++++++++++++++-- drivers/reset/reset-gpio.c | 119 ++++++++++ include/linux/cpufreq.h | 3 +- include/linux/of.h | 16 ++ include/linux/reset-controller.h | 4 + sound/soc/codecs/wsa884x.c | 53 ++++- 10 files changed, 410 insertions(+), 26 deletions(-) create mode 100644 drivers/reset/reset-gpio.c -- 2.34.1

5 14

[PATCH] Revert "ALSA: usb-audio: Skip setting clock selector for single connections"
by Alexander Tsoy 29 Jan '24

29 Jan '24

This reverts commit 67794f882adca00d043899ac248bc002751da9f6. We need to explicitly set up the clock selector to workaround a problem with the Behringer mixers. This was originally done in d2e8f641257d ("ALSA: usb-audio: Explicitly set up the clock selector") The problem with MOTU M Series mentioned in commit message was fixed in a different way by checking control capabilities of clock selectors. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/clock.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 7b259641adb5..a8204c6d6fac 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -344,7 +344,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, if (ret > 0) { /* Skip setting clock selector again for some devices */ if (chip->quirk_flags & QUIRK_FLAG_SKIP_CLOCK_SELECTOR || - pins == 1 || !writeable) + !writeable) return ret; err = uac_clock_selector_set_val(chip, entity_id, cur); if (err < 0) -- 2.43.0

2 1

[PATCH] ALSA: hda/realtek: Enable Mute LED on HP Laptop 14-fq0xxx
by Luka Guzenko 29 Jan '24

29 Jan '24

This HP Laptop uses ALC236 codec with COEF 0x07 controlling the mute LED. Enable existing quirk for this device. Signed-off-by: Luka Guzenko <l.guzenko(a)web.de> --- 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..0d2fbb8b24b3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9852,6 +9852,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8786, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8787, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8788, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x87b7, "HP Laptop 14-fq0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), -- 2.43.0

2 1

[PATCH v2] ALSA: firewire-motu: add support for MOTU 896 mk3 FireWire and Hybrid
by Takashi Sakamoto 29 Jan '24

29 Jan '24

Mark of the Unicorn released 896 mk3 FireWire in 2008 as part of the third generation of its FireWire series. In 2011, 896 mk3 hybrid was released to support USB protocol. It supports sampling transfer frequency up to 192.0 kHz. The packet format differs depending on both of current sampling transfer frequency and the type of signal in optical interfaces. The model supports transmission of PCM frames as well as MIDI messages. The 896 mk3 FireWire consists of below ICs: * Texas Instruments TSB41AB2 * Xilinx Spartan-3A FPGA, XC3S500E * Texas Instruments TMS320C6722 * Microchip (Atmel) AT91SAM SAM7S256 It supports sampling transfer frequency up to 192.0 kHz. The packet format differs depending on both of current sampling transfer frequency and the type of signal in two pairs of optical interfaces. The model supports transmission of PCM frames, while has no port for MIDi messages. The model supports command mechanism to configure internal DSP. Hardware meter information is available in the first 2 chunks of each data block of tx packet. This commit adds support for it. The 896 mk3 FireWire is just tested, but the 896 mk3 Hybrid is not yet. $ config-rom-pretty-printer < motu-896mk3fw.img ROM header and bus information block ----------------------------------------------------------------- 1024 04100ce1 bus_info_length 4, crc_length 16, crc 3297 1028 31333934 bus_name "1394" 1032 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 1036 0001f200 company_id 0001f2 | 1040 00093add device_id 0000604893 | EUI-64 0547556791237341 root directory ----------------------------------------------------------------- 1044 0004ef04 directory_length 4, crc 61188 1048 030001f2 vendor 1052 0c0083c0 node capabilities: per IEEE 1394 1056 d1000002 --> unit directory at 1064 1060 8d000005 --> eui-64 leaf at 1080 unit directory at 1064 ----------------------------------------------------------------- 1064 0003998d directory_length 3, crc 39309 1068 120001f2 specifier id 1072 13000017 version 1076 17101800 model eui-64 leaf at 1080 ----------------------------------------------------------------- 1080 0002cc82 leaf_length 2, crc 52354 1084 0001f200 company_id 0001f2 | 1088 00093add device_id 0000604893 | EUI-64 0547556791237341 $ config-rom-pretty-printer < motu-896mk3hybrid.img ROM header and bus information block ----------------------------------------------------------------- 1024 04103cbe bus_info_length 4, crc_length 16, crc 15550 1028 31333934 bus_name "1394" 1032 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 1036 0001f200 company_id 0001f2 | 1040 000ae601 device_id 0000714241 | EUI-64 0547556791346689 root directory ----------------------------------------------------------------- 1044 0004ef04 directory_length 4, crc 61188 1048 030001f2 vendor 1052 0c0083c0 node capabilities: per IEEE 1394 1056 d1000002 --> unit directory at 1064 1060 8d000005 --> eui-64 leaf at 1080 unit directory at 1064 ----------------------------------------------------------------- 1064 000394ac directory_length 3, crc 38060 1068 120001f2 specifier id 1072 13000037 version 1076 17102800 model eui-64 leaf at 1080 ----------------------------------------------------------------- 1080 0002cf69 leaf_length 2, crc 53097 1084 0001f200 company_id 0001f2 | 1088 000ae601 device_id 0000714241 | EUI-64 0547556791346689 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 2 ++ sound/firewire/motu/motu-protocol-v3.c | 9 +++++++++ sound/firewire/motu/motu.c | 2 ++ sound/firewire/motu/motu.h | 1 + 4 files changed, 14 insertions(+) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 22b6c779682a..5973c25c2add 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -175,6 +175,8 @@ config SND_FIREWIRE_MOTU * 8pre * 828mk3 (FireWire only) * 828mk3 (Hybrid) + * 896mk3 (FireWire only) + * 896mk3 (Hybrid) * Ultralite mk3 (FireWire only) * Ultralite mk3 (Hybrid) * Traveler mk3 diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index 8a0426920a76..7254fdfe046a 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -261,6 +261,7 @@ int snd_motu_protocol_v3_cache_packet_formats(struct snd_motu *motu) if (motu->spec == &snd_motu_spec_828mk3_fw || motu->spec == &snd_motu_spec_828mk3_hybrid || + motu->spec == &snd_motu_spec_896mk3 || motu->spec == &snd_motu_spec_traveler_mk3 || motu->spec == &snd_motu_spec_track16) return detect_packet_formats_with_opt_ifaces(motu, data); @@ -288,6 +289,14 @@ const struct snd_motu_spec snd_motu_spec_828mk3_hybrid = { .rx_fixed_pcm_chunks = {14, 14, 14}, // Additional 4 dummy chunks at higher rate. }; +const struct snd_motu_spec snd_motu_spec_896mk3 = { + .name = "896mk3", + .protocol_version = SND_MOTU_PROTOCOL_V3, + .flags = SND_MOTU_SPEC_COMMAND_DSP, + .tx_fixed_pcm_chunks = {18, 14, 10}, + .rx_fixed_pcm_chunks = {18, 14, 10}, +}; + const struct snd_motu_spec snd_motu_spec_traveler_mk3 = { .name = "TravelerMk3", .protocol_version = SND_MOTU_PROTOCOL_V3, diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index d73599eb7d5a..d14ab5dd5bea 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -168,10 +168,12 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x00000d, &snd_motu_spec_ultralite), SND_MOTU_DEV_ENTRY(0x00000f, &snd_motu_spec_8pre), SND_MOTU_DEV_ENTRY(0x000015, &snd_motu_spec_828mk3_fw), // FireWire only. + SND_MOTU_DEV_ENTRY(0x000017, &snd_motu_spec_896mk3), // FireWire only. SND_MOTU_DEV_ENTRY(0x000019, &snd_motu_spec_ultralite_mk3), // FireWire only. SND_MOTU_DEV_ENTRY(0x00001b, &snd_motu_spec_traveler_mk3), SND_MOTU_DEV_ENTRY(0x000030, &snd_motu_spec_ultralite_mk3), // Hybrid. SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3_hybrid), // Hybrid. + SND_MOTU_DEV_ENTRY(0x000037, &snd_motu_spec_896mk3), // Hybrid. SND_MOTU_DEV_ENTRY(0x000033, &snd_motu_spec_audio_express), SND_MOTU_DEV_ENTRY(0x000039, &snd_motu_spec_track16), SND_MOTU_DEV_ENTRY(0x000045, &snd_motu_spec_4pre), diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 3b1dc98a7be0..c66be0a89ccf 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -138,6 +138,7 @@ extern const struct snd_motu_spec snd_motu_spec_8pre; extern const struct snd_motu_spec snd_motu_spec_828mk3_fw; extern const struct snd_motu_spec snd_motu_spec_828mk3_hybrid; +extern const struct snd_motu_spec snd_motu_spec_896mk3; extern const struct snd_motu_spec snd_motu_spec_traveler_mk3; extern const struct snd_motu_spec snd_motu_spec_ultralite_mk3; extern const struct snd_motu_spec snd_motu_spec_audio_express; -- 2.40.1

2 1

[PATCH] ALSA: firewire-motu: add support for MOTU 896 mk3 FireWire and Hybrid
by Takashi Sakamoto 29 Jan '24

29 Jan '24

Mark of the Unicorn released 896 mk3 FireWire in 2008 as part of the third generation of its FireWire series. In 2011, 896 mk3 hybrid was released to support USB protocol. It supports sampling transfer frequency up to 192.0 kHz. The packet format differs depending on both of current sampling transfer frequency and the type of signal in optical interfaces. The model supports transmission of PCM frames as well as MIDI messages. The 896 mk3 FireWire consists of below ICs: * Texas Instruments TSB41AB2 * Microchip (SMSC) USB3300 * Xilinx Spartan-3A FPGA, XC3S500E * Texas Instruments TMS320C6722 * Microchip (Atmel) AT91SAM SAM7S256 It supports sampling transfer frequency up to 192.0 kHz. The packet format differs depending on both of current sampling transfer frequency and the type of signal in two pairs of optical interfaces. The model supports transmission of PCM frames, while has no port for MIDi messages. The model supports command mechanism to configure internal DSP. Hardware meter information is available in the first 2 chunks of each data block of tx packet. This commit adds support for it. The 896 mk3 FireWire is just tested, but the 896 mk3 Hybrid is not yet. $ config-rom-pretty-printer < motu-896mk3fw.img ROM header and bus information block ----------------------------------------------------------------- 1024 04100ce1 bus_info_length 4, crc_length 16, crc 3297 1028 31333934 bus_name "1394" 1032 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 1036 0001f200 company_id 0001f2 | 1040 00093add device_id 0000604893 | EUI-64 0547556791237341 root directory ----------------------------------------------------------------- 1044 0004ef04 directory_length 4, crc 61188 1048 030001f2 vendor 1052 0c0083c0 node capabilities: per IEEE 1394 1056 d1000002 --> unit directory at 1064 1060 8d000005 --> eui-64 leaf at 1080 unit directory at 1064 ----------------------------------------------------------------- 1064 0003998d directory_length 3, crc 39309 1068 120001f2 specifier id 1072 13000017 version 1076 17101800 model eui-64 leaf at 1080 ----------------------------------------------------------------- 1080 0002cc82 leaf_length 2, crc 52354 1084 0001f200 company_id 0001f2 | 1088 00093add device_id 0000604893 | EUI-64 0547556791237341 $ config-rom-pretty-printer < motu-896mk3hybrid.img ROM header and bus information block ----------------------------------------------------------------- 1024 04103cbe bus_info_length 4, crc_length 16, crc 15550 1028 31333934 bus_name "1394" 1032 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 1036 0001f200 company_id 0001f2 | 1040 000ae601 device_id 0000714241 | EUI-64 0547556791346689 root directory ----------------------------------------------------------------- 1044 0004ef04 directory_length 4, crc 61188 1048 030001f2 vendor 1052 0c0083c0 node capabilities: per IEEE 1394 1056 d1000002 --> unit directory at 1064 1060 8d000005 --> eui-64 leaf at 1080 unit directory at 1064 ----------------------------------------------------------------- 1064 000394ac directory_length 3, crc 38060 1068 120001f2 specifier id 1072 13000037 version 1076 17102800 model eui-64 leaf at 1080 ----------------------------------------------------------------- 1080 0002cf69 leaf_length 2, crc 53097 1084 0001f200 company_id 0001f2 | 1088 000ae601 device_id 0000714241 | EUI-64 0547556791346689 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-protocol-v3.c | 9 +++++++++ sound/firewire/motu/motu.c | 2 ++ sound/firewire/motu/motu.h | 1 + 3 files changed, 12 insertions(+) diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index 8a0426920a76..7254fdfe046a 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -261,6 +261,7 @@ int snd_motu_protocol_v3_cache_packet_formats(struct snd_motu *motu) if (motu->spec == &snd_motu_spec_828mk3_fw || motu->spec == &snd_motu_spec_828mk3_hybrid || + motu->spec == &snd_motu_spec_896mk3 || motu->spec == &snd_motu_spec_traveler_mk3 || motu->spec == &snd_motu_spec_track16) return detect_packet_formats_with_opt_ifaces(motu, data); @@ -288,6 +289,14 @@ const struct snd_motu_spec snd_motu_spec_828mk3_hybrid = { .rx_fixed_pcm_chunks = {14, 14, 14}, // Additional 4 dummy chunks at higher rate. }; +const struct snd_motu_spec snd_motu_spec_896mk3 = { + .name = "896mk3", + .protocol_version = SND_MOTU_PROTOCOL_V3, + .flags = SND_MOTU_SPEC_COMMAND_DSP, + .tx_fixed_pcm_chunks = {18, 14, 10}, + .rx_fixed_pcm_chunks = {18, 14, 10}, +}; + const struct snd_motu_spec snd_motu_spec_traveler_mk3 = { .name = "TravelerMk3", .protocol_version = SND_MOTU_PROTOCOL_V3, diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index d73599eb7d5a..d14ab5dd5bea 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -168,10 +168,12 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x00000d, &snd_motu_spec_ultralite), SND_MOTU_DEV_ENTRY(0x00000f, &snd_motu_spec_8pre), SND_MOTU_DEV_ENTRY(0x000015, &snd_motu_spec_828mk3_fw), // FireWire only. + SND_MOTU_DEV_ENTRY(0x000017, &snd_motu_spec_896mk3), // FireWire only. SND_MOTU_DEV_ENTRY(0x000019, &snd_motu_spec_ultralite_mk3), // FireWire only. SND_MOTU_DEV_ENTRY(0x00001b, &snd_motu_spec_traveler_mk3), SND_MOTU_DEV_ENTRY(0x000030, &snd_motu_spec_ultralite_mk3), // Hybrid. SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3_hybrid), // Hybrid. + SND_MOTU_DEV_ENTRY(0x000037, &snd_motu_spec_896mk3), // Hybrid. SND_MOTU_DEV_ENTRY(0x000033, &snd_motu_spec_audio_express), SND_MOTU_DEV_ENTRY(0x000039, &snd_motu_spec_track16), SND_MOTU_DEV_ENTRY(0x000045, &snd_motu_spec_4pre), diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 3b1dc98a7be0..c66be0a89ccf 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -138,6 +138,7 @@ extern const struct snd_motu_spec snd_motu_spec_8pre; extern const struct snd_motu_spec snd_motu_spec_828mk3_fw; extern const struct snd_motu_spec snd_motu_spec_828mk3_hybrid; +extern const struct snd_motu_spec snd_motu_spec_896mk3; extern const struct snd_motu_spec snd_motu_spec_traveler_mk3; extern const struct snd_motu_spec snd_motu_spec_ultralite_mk3; extern const struct snd_motu_spec snd_motu_spec_audio_express; -- 2.40.1

1 1

[PATCH] ALSA: usb-audio: Skip setting clock selector for single connections
by Alexander Tsoy 28 Jan '24

28 Jan '24

Since commit 086b957cc17f5 ("ALSA: usb-audio: Skip the clock selector inquiry for single connections") we are already skipping clock selector inquiry if only one clock source is connected, but we are still sending a set request. Lets skip that too. This should fix errors when setting a sample rate on devices that don't have any controls present within the clock selector. An example of such device is the new revision of MOTU M Series (07fd:000b): AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 11 (CLOCK_SELECTOR) bClockID 1 bNrInPins 1 baCSourceID(0) 2 bmControls 0x00 iClockSelector 0 Perhaps we also should check if clock selectors are readable and writeable like we already do for clock sources, but this is out of scope of this patch. Link: https://bugzilla.kernel.org/show_bug.cgi?id=217601 Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/clock.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 33db334e6556..94e4aaeafe58 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -325,7 +325,8 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, visited, validate); if (ret > 0) { /* Skip setting clock selector again for some devices */ - if (chip->quirk_flags & QUIRK_FLAG_SKIP_CLOCK_SELECTOR) + if (chip->quirk_flags & QUIRK_FLAG_SKIP_CLOCK_SELECTOR || + pins == 1) return ret; err = uac_clock_selector_set_val(chip, entity_id, cur); if (err < 0) -- 2.43.0

2 4

[PATCH v2 0/4] Support HP Models without _DSD
by Stefan Binding 28 Jan '24

28 Jan '24

Add Quirks and driver properties for Dell models. Driver properties are required since these models do not have _DSD. Some models require special handing for Speaker ID and cannot use the configuration table to add properties. Also fix an issue for Channel Index property, when set through the configuration table, to use the same method as when loading _DSD properties. This is needed for laptops with 4 amps where the channels do not alternate. Changes from v1: - Rebase on latest for-next Stefan Binding (4): ALSA: hda: cs35l41: Set Channel Index correctly when system is missing _DSD ALSA: hda: cs35l41: Support additional HP Envy Models ALSA: hda: cs35l41: Support HP models without _DSD using dual Speaker ID ALSA: hda/realtek: Add quirks for various HP ENVY models sound/pci/hda/cs35l41_hda_property.c | 90 +++++++++++++++++++++++----- sound/pci/hda/patch_realtek.c | 40 +++++++++++++ 2 files changed, 114 insertions(+), 16 deletions(-) -- 2.34.1

2 5

Tascam US-428 crash
by GitHub issues - opened 28 Jan '24

28 Jan '24

alsa-project/alsa-tools issue #23 was opened from mgsx-dev: Hi, i'm not sure that's the right place to post the issue. (sorry in advance) I have a Tascam US-428 Hardware and tried to make it works on Ubuntu 22.04. I followed setup instruction for US-224 and adapted it for the US 428 : https://alsa.opensrc.org/Tascam_US-224 First i had to install firmware manually since hotplug package seems to be no longer available : * `lsusb` shows my hardware correctly : `Bus 001 Device 004: ID 1604:8000 Tascam US-428 Audio/Midi Controller (without fw)` * I downloaded firmwares from https://github.com/alsa-project/alsa-firmware * Loaded it to the Tascam : `sudo fxload -s alsa-firmware/usx2yloader/tascam_loader.ihx -I alsa-firmware/usx2yloader/us428fw.ihx -D /dev/bus/usb/001/004` * I manually copied `alsa-firmware/usx2yloader/` files (us428.conf, us428.prepad, and us428.rbt) to `/usr/lib/firmware/usx2yloader/` folder to make `usx2yloader` working. * Running `lsusb` confirmed it works. It shows now : `Bus 001 Device 005: ID 1604:8001 Tascam US-428 Audio/Midi Controller` * Also, the USB led on my Tascam turned up green. Then I tried to launch us428control to test midi : * `us428control -v 2` properly logs midi event when i move some controls on the Tascam. So far so good. Then i tried to test audio playback. ALSA properly list my device : ``` aplay -l ... card 1: USX2Y [TASCAM US-X2Y], device 0: US-X2Y Audio [US-X2Y Audio #0] Subdevices: 1/1 Subdevice #0: subdevice #0 aplay -L ... hw:CARD=USX2Y,DEV=0 TASCAM US-X2Y, US-X2Y Audio #0 Direct hardware device without any conversions plughw:CARD=USX2Y,DEV=0 TASCAM US-X2Y, US-X2Y Audio #0 Hardware device with all software conversions sysdefault:CARD=USX2Y TASCAM US-X2Y, US-X2Y Audio #0 Default Audio Device dmix:CARD=USX2Y,DEV=0 TASCAM US-X2Y, US-X2Y Audio #0 Direct sample mixing device usbstream:CARD=USX2Y TASCAM US-X2Y USB Stream Output ``` Then i tried to playback a wav file : `aplay -D dmix:CARD=USX2Y,DEV=0 test.wav` but when the song finish, or when i interrupt it, my whole system freeze (desktop GUI and mouse) and i had to shutdown my laptop manually. Note that i tried with other device name but it freeze as well, i also launched Audacity which freeze at startup as well. I'm a bit lost and i can't find useful information on the internet about it. Any help appreciated. Thanks for reading. Issue URL : https://github.com/alsa-project/alsa-tools/issues/23 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[tiwai-sound:topic/pm-ops 11/29] sound/arm/aaci.c:742:22: warning: unused variable 'aaci'
by kernel test robot 28 Jan '24

28 Jan '24

Hi Takashi, FYI, the error/warning was bisected to this commit, please ignore it if it's irrelevant. tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/pm-ops head: 2fe6ffd7c79f740b73533d44de3ae0e542674857 commit: 8c975a919c0e6139ddf3692fa44992364587070c [11/29] ALSA: aaci: Replace with DEFINE_SIMPLE_DEV_PM_OPS() config: arm-randconfig-004-20240127 (https://download.01.org/0day-ci/archive/20240128/202401281542.hlZQZx6m-lkp@…) compiler: arm-linux-gnueabi-gcc (GCC) 13.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240128/202401281542.hlZQZx6m-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202401281542.hlZQZx6m-lkp@intel.com/ All warnings (new ones prefixed by >>): sound/arm/aaci.c: In function 'aaci_do_suspend': >> sound/arm/aaci.c:742:22: warning: unused variable 'aaci' [-Wunused-variable] 742 | struct aaci *aaci = card->private_data; | ^~~~ vim +/aaci +742 sound/arm/aaci.c cb5a6ffc5c09bc Russell King 2005-05-12 736 cb5a6ffc5c09bc Russell King 2005-05-12 737 /* cb5a6ffc5c09bc Russell King 2005-05-12 738 * Power Management. cb5a6ffc5c09bc Russell King 2005-05-12 739 */ b13a714923fe4d Ulf Hansson 2013-12-03 740 static int aaci_do_suspend(struct snd_card *card) cb5a6ffc5c09bc Russell King 2005-05-12 741 { cb5a6ffc5c09bc Russell King 2005-05-12 @742 struct aaci *aaci = card->private_data; 792a6c51875c9d Takashi Iwai 2005-11-17 743 snd_power_change_state(card, SNDRV_CTL_POWER_D3cold); cb5a6ffc5c09bc Russell King 2005-05-12 744 return 0; cb5a6ffc5c09bc Russell King 2005-05-12 745 } cb5a6ffc5c09bc Russell King 2005-05-12 746 :::::: The code at line 742 was first introduced by commit :::::: cb5a6ffc5c09bc354af69407dae710dcddcced37 [ALSA] ARM AACI primecell driver :::::: TO: Russell King <rmk(a)arm.linux.org.uk> :::::: CC: Jaroslav Kysela <perex(a)suse.cz> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:topic/pm-ops 5/29] include/sound/sb.h:294:20: error: cannot combine with previous 'void' declaration specifier
by kernel test robot 28 Jan '24

28 Jan '24

Hi Takashi, FYI, the error/warning was bisected to this commit, please ignore it if it's irrelevant. tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/pm-ops head: 2fe6ffd7c79f740b73533d44de3ae0e542674857 commit: 9757c2b93cfed03065a14a6cd817474891eb55fe [5/29] ALSA: als4000: Replace with DEFINE_SIMPLE_DEV_PM_OPS() config: i386-randconfig-141-20240128 (https://download.01.org/0day-ci/archive/20240128/202401281302.usiU86hP-lkp@…) compiler: clang version 17.0.6 (https://github.com/llvm/llvm-project 6009708b4367171ccdbf4b5905cb6a803753fe18) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240128/202401281302.usiU86hP-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202401281302.usiU86hP-lkp@intel.com/ All errors (new ones prefixed by >>): In file included from sound/isa/als100.c:24: >> include/sound/sb.h:294:20: error: cannot combine with previous 'void' declaration specifier 294 | static inline void void snd_sbmixer_suspend(struct snd_sb *chip) {} | ^ include/sound/sb.h:295:20: error: cannot combine with previous 'void' declaration specifier 295 | static inline void void snd_sbmixer_resume(struct snd_sb *chip) {} | ^ 2 errors generated. vim +/void +294 include/sound/sb.h 274 275 /* sb_common.c */ 276 int snd_sbdsp_command(struct snd_sb *chip, unsigned char val); 277 int snd_sbdsp_get_byte(struct snd_sb *chip); 278 int snd_sbdsp_reset(struct snd_sb *chip); 279 int snd_sbdsp_create(struct snd_card *card, 280 unsigned long port, 281 int irq, 282 irq_handler_t irq_handler, 283 int dma8, int dma16, 284 unsigned short hardware, 285 struct snd_sb **r_chip); 286 /* sb_mixer.c */ 287 void snd_sbmixer_write(struct snd_sb *chip, unsigned char reg, unsigned char data); 288 unsigned char snd_sbmixer_read(struct snd_sb *chip, unsigned char reg); 289 int snd_sbmixer_new(struct snd_sb *chip); 290 #ifdef CONFIG_PM 291 void snd_sbmixer_suspend(struct snd_sb *chip); 292 void snd_sbmixer_resume(struct snd_sb *chip); 293 #else > 294 static inline void void snd_sbmixer_suspend(struct snd_sb *chip) {} 295 static inline void void snd_sbmixer_resume(struct snd_sb *chip) {} 296 #endif 297 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:topic/pm-ops 5/29] include/sound/sb.h:294:20: error: two or more data types in declaration specifiers
by kernel test robot 28 Jan '24

28 Jan '24

Hi Takashi, FYI, the error/warning was bisected to this commit, please ignore it if it's irrelevant. tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/pm-ops head: 2fe6ffd7c79f740b73533d44de3ae0e542674857 commit: 9757c2b93cfed03065a14a6cd817474891eb55fe [5/29] ALSA: als4000: Replace with DEFINE_SIMPLE_DEV_PM_OPS() config: i386-randconfig-052-20240127 (https://download.01.org/0day-ci/archive/20240128/202401281001.74RRLOzP-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240128/202401281001.74RRLOzP-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202401281001.74RRLOzP-lkp@intel.com/ All errors (new ones prefixed by >>): In file included from sound/isa/sb/sb_common.c:17: >> include/sound/sb.h:294:20: error: two or more data types in declaration specifiers 294 | static inline void void snd_sbmixer_suspend(struct snd_sb *chip) {} | ^~~~ include/sound/sb.h:295:20: error: two or more data types in declaration specifiers 295 | static inline void void snd_sbmixer_resume(struct snd_sb *chip) {} | ^~~~ vim +294 include/sound/sb.h 274 275 /* sb_common.c */ 276 int snd_sbdsp_command(struct snd_sb *chip, unsigned char val); 277 int snd_sbdsp_get_byte(struct snd_sb *chip); 278 int snd_sbdsp_reset(struct snd_sb *chip); 279 int snd_sbdsp_create(struct snd_card *card, 280 unsigned long port, 281 int irq, 282 irq_handler_t irq_handler, 283 int dma8, int dma16, 284 unsigned short hardware, 285 struct snd_sb **r_chip); 286 /* sb_mixer.c */ 287 void snd_sbmixer_write(struct snd_sb *chip, unsigned char reg, unsigned char data); 288 unsigned char snd_sbmixer_read(struct snd_sb *chip, unsigned char reg); 289 int snd_sbmixer_new(struct snd_sb *chip); 290 #ifdef CONFIG_PM 291 void snd_sbmixer_suspend(struct snd_sb *chip); 292 void snd_sbmixer_resume(struct snd_sb *chip); 293 #else > 294 static inline void void snd_sbmixer_suspend(struct snd_sb *chip) {} 295 static inline void void snd_sbmixer_resume(struct snd_sb *chip) {} 296 #endif 297 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:topic/pm-ops 5/29] sound/isa/cmi8330.c:55:12: warning: 'index' defined but not used
by kernel test robot 28 Jan '24

28 Jan '24

Hi Takashi, FYI, the error/warning was bisected to this commit, please ignore it if it's irrelevant. tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/pm-ops head: 2fe6ffd7c79f740b73533d44de3ae0e542674857 commit: 9757c2b93cfed03065a14a6cd817474891eb55fe [5/29] ALSA: als4000: Replace with DEFINE_SIMPLE_DEV_PM_OPS() config: powerpc-randconfig-002-20240127 (https://download.01.org/0day-ci/archive/20240128/202401280946.VcZLwAbo-lkp@…) compiler: powerpc-linux-gcc (GCC) 13.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240128/202401280946.VcZLwAbo-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202401280946.VcZLwAbo-lkp@intel.com/ All warnings (new ones prefixed by >>): In file included from sound/isa/cmi8330.c:41: include/sound/sb.h:294:20: error: two or more data types in declaration specifiers 294 | static inline void void snd_sbmixer_suspend(struct snd_sb *chip) {} | ^~~~ include/sound/sb.h:295:20: error: two or more data types in declaration specifiers 295 | static inline void void snd_sbmixer_resume(struct snd_sb *chip) {} | ^~~~ In file included from include/linux/init.h:5, from sound/isa/cmi8330.c:32: include/linux/build_bug.h:16:51: error: bit-field '<anonymous>' width not an integer constant 16 | #define BUILD_BUG_ON_ZERO(e) ((int)(sizeof(struct { int:(-!!(e)); }))) | ^ include/linux/compiler.h:228:33: note: in expansion of macro 'BUILD_BUG_ON_ZERO' 228 | #define __must_be_array(a) BUILD_BUG_ON_ZERO(__same_type((a), &(a)[0])) | ^~~~~~~~~~~~~~~~~ include/linux/array_size.h:11:59: note: in expansion of macro '__must_be_array' 11 | #define ARRAY_SIZE(arr) (sizeof(arr) / sizeof((arr)[0]) + __must_be_array(arr)) | ^~~~~~~~~~~~~~~ include/linux/moduleparam.h:521:20: note: in expansion of macro 'ARRAY_SIZE' 521 | = { .max = ARRAY_SIZE(array), .num = nump, \ | ^~~~~~~~~~ include/linux/moduleparam.h:505:9: note: in expansion of macro 'module_param_array_named' 505 | module_param_array_named(name, name, type, nump, perm) | ^~~~~~~~~~~~~~~~~~~~~~~~ sound/isa/cmi8330.c:72:1: note: in expansion of macro 'module_param_array' 72 | module_param_array(index, int, NULL, 0444); | ^~~~~~~~~~~~~~~~~~ >> sound/isa/cmi8330.c:55:12: warning: 'index' defined but not used [-Wunused-variable] 55 | static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; | ^~~~~ -- In file included from sound/pci/cs5530.c:35: include/sound/sb.h:294:20: error: two or more data types in declaration specifiers 294 | static inline void void snd_sbmixer_suspend(struct snd_sb *chip) {} | ^~~~ include/sound/sb.h:295:20: error: two or more data types in declaration specifiers 295 | static inline void void snd_sbmixer_resume(struct snd_sb *chip) {} | ^~~~ In file included from include/linux/bits.h:21, from include/linux/bitops.h:6, from include/linux/log2.h:12, from include/asm-generic/div64.h:55, from ./arch/powerpc/include/generated/asm/div64.h:1, from include/linux/math.h:6, from include/linux/delay.h:22, from sound/pci/cs5530.c:30: include/linux/build_bug.h:16:51: error: bit-field '<anonymous>' width not an integer constant 16 | #define BUILD_BUG_ON_ZERO(e) ((int)(sizeof(struct { int:(-!!(e)); }))) | ^ include/linux/compiler.h:228:33: note: in expansion of macro 'BUILD_BUG_ON_ZERO' 228 | #define __must_be_array(a) BUILD_BUG_ON_ZERO(__same_type((a), &(a)[0])) | ^~~~~~~~~~~~~~~~~ include/linux/array_size.h:11:59: note: in expansion of macro '__must_be_array' 11 | #define ARRAY_SIZE(arr) (sizeof(arr) / sizeof((arr)[0]) + __must_be_array(arr)) | ^~~~~~~~~~~~~~~ include/linux/moduleparam.h:521:20: note: in expansion of macro 'ARRAY_SIZE' 521 | = { .max = ARRAY_SIZE(array), .num = nump, \ | ^~~~~~~~~~ include/linux/moduleparam.h:505:9: note: in expansion of macro 'module_param_array_named' 505 | module_param_array_named(name, name, type, nump, perm) | ^~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/cs5530.c:46:1: note: in expansion of macro 'module_param_array' 46 | module_param_array(index, int, NULL, 0444); | ^~~~~~~~~~~~~~~~~~ >> sound/pci/cs5530.c:42:12: warning: 'index' defined but not used [-Wunused-variable] 42 | static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; | ^~~~~ -- In file included from sound/isa/sb/sb16.c:14, from sound/isa/sb/sbawe.c:2: include/sound/sb.h:294:20: error: two or more data types in declaration specifiers 294 | static inline void void snd_sbmixer_suspend(struct snd_sb *chip) {} | ^~~~ include/sound/sb.h:295:20: error: two or more data types in declaration specifiers 295 | static inline void void snd_sbmixer_resume(struct snd_sb *chip) {} | ^~~~ In file included from include/linux/bits.h:21, from include/linux/ratelimit_types.h:5, from include/linux/ratelimit.h:5, from include/linux/dev_printk.h:16, from include/linux/device.h:15, from arch/powerpc/include/asm/io.h:22, from arch/powerpc/include/asm/dma.h:22, from sound/isa/sb/sb16.c:7: include/linux/build_bug.h:16:51: error: bit-field '<anonymous>' width not an integer constant 16 | #define BUILD_BUG_ON_ZERO(e) ((int)(sizeof(struct { int:(-!!(e)); }))) | ^ include/linux/compiler.h:228:33: note: in expansion of macro 'BUILD_BUG_ON_ZERO' 228 | #define __must_be_array(a) BUILD_BUG_ON_ZERO(__same_type((a), &(a)[0])) | ^~~~~~~~~~~~~~~~~ include/linux/array_size.h:11:59: note: in expansion of macro '__must_be_array' 11 | #define ARRAY_SIZE(arr) (sizeof(arr) / sizeof((arr)[0]) + __must_be_array(arr)) | ^~~~~~~~~~~~~~~ include/linux/moduleparam.h:521:20: note: in expansion of macro 'ARRAY_SIZE' 521 | = { .max = ARRAY_SIZE(array), .num = nump, \ | ^~~~~~~~~~ include/linux/moduleparam.h:505:9: note: in expansion of macro 'module_param_array_named' 505 | module_param_array_named(name, name, type, nump, perm) | ^~~~~~~~~~~~~~~~~~~~~~~~ sound/isa/sb/sb16.c:69:1: note: in expansion of macro 'module_param_array' 69 | module_param_array(index, int, NULL, 0444); | ^~~~~~~~~~~~~~~~~~ >> sound/isa/sb/sb16.c:46:12: warning: 'index' defined but not used [-Wunused-variable] 46 | static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ | ^~~~~ -- In file included from sound/isa/sb/jazz16.c:23: include/sound/sb.h:294:20: error: two or more data types in declaration specifiers 294 | static inline void void snd_sbmixer_suspend(struct snd_sb *chip) {} | ^~~~ include/sound/sb.h:295:20: error: two or more data types in declaration specifiers 295 | static inline void void snd_sbmixer_resume(struct snd_sb *chip) {} | ^~~~ In file included from include/linux/init.h:5, from sound/isa/sb/jazz16.c:14: include/linux/build_bug.h:16:51: error: bit-field '<anonymous>' width not an integer constant 16 | #define BUILD_BUG_ON_ZERO(e) ((int)(sizeof(struct { int:(-!!(e)); }))) | ^ include/linux/compiler.h:228:33: note: in expansion of macro 'BUILD_BUG_ON_ZERO' 228 | #define __must_be_array(a) BUILD_BUG_ON_ZERO(__same_type((a), &(a)[0])) | ^~~~~~~~~~~~~~~~~ include/linux/array_size.h:11:59: note: in expansion of macro '__must_be_array' 11 | #define ARRAY_SIZE(arr) (sizeof(arr) / sizeof((arr)[0]) + __must_be_array(arr)) | ^~~~~~~~~~~~~~~ include/linux/moduleparam.h:521:20: note: in expansion of macro 'ARRAY_SIZE' 521 | = { .max = ARRAY_SIZE(array), .num = nump, \ | ^~~~~~~~~~ include/linux/moduleparam.h:505:9: note: in expansion of macro 'module_param_array_named' 505 | module_param_array_named(name, name, type, nump, perm) | ^~~~~~~~~~~~~~~~~~~~~~~~ sound/isa/sb/jazz16.c:44:1: note: in expansion of macro 'module_param_array' 44 | module_param_array(index, int, NULL, 0444); | ^~~~~~~~~~~~~~~~~~ >> sound/isa/sb/jazz16.c:34:12: warning: 'index' defined but not used [-Wunused-variable] 34 | static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ | ^~~~~ vim +/index +55 sound/isa/cmi8330.c ^1da177e4c3f41 Linus Torvalds 2005-04-16 54 ^1da177e4c3f41 Linus Torvalds 2005-04-16 @55 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; ^1da177e4c3f41 Linus Torvalds 2005-04-16 56 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; a67ff6a54095e2 Rusty Russell 2011-12-15 57 static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; ^1da177e4c3f41 Linus Torvalds 2005-04-16 58 #ifdef CONFIG_PNP a67ff6a54095e2 Rusty Russell 2011-12-15 59 static bool isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; ^1da177e4c3f41 Linus Torvalds 2005-04-16 60 #endif ^1da177e4c3f41 Linus Torvalds 2005-04-16 61 static long sbport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; ^1da177e4c3f41 Linus Torvalds 2005-04-16 62 static int sbirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; ^1da177e4c3f41 Linus Torvalds 2005-04-16 63 static int sbdma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; ^1da177e4c3f41 Linus Torvalds 2005-04-16 64 static int sbdma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; ^1da177e4c3f41 Linus Torvalds 2005-04-16 65 static long wssport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; ^1da177e4c3f41 Linus Torvalds 2005-04-16 66 static int wssirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; ^1da177e4c3f41 Linus Torvalds 2005-04-16 67 static int wssdma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; 48972cc5101dee Krzysztof Helt 2009-01-21 68 static long fmport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; c97dff84e0d9a4 Krzysztof Helt 2009-01-29 69 static long mpuport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; c97dff84e0d9a4 Krzysztof Helt 2009-01-29 70 static int mpuirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; ^1da177e4c3f41 Linus Torvalds 2005-04-16 71 ^1da177e4c3f41 Linus Torvalds 2005-04-16 @72 module_param_array(index, int, NULL, 0444); 69eb88825a7a56 Ondrej Zary 2009-07-04 73 MODULE_PARM_DESC(index, "Index value for CMI8330/CMI8329 soundcard."); ^1da177e4c3f41 Linus Torvalds 2005-04-16 74 module_param_array(id, charp, NULL, 0444); 69eb88825a7a56 Ondrej Zary 2009-07-04 75 MODULE_PARM_DESC(id, "ID string for CMI8330/CMI8329 soundcard."); ^1da177e4c3f41 Linus Torvalds 2005-04-16 76 module_param_array(enable, bool, NULL, 0444); 69eb88825a7a56 Ondrej Zary 2009-07-04 77 MODULE_PARM_DESC(enable, "Enable CMI8330/CMI8329 soundcard."); ^1da177e4c3f41 Linus Torvalds 2005-04-16 78 #ifdef CONFIG_PNP ^1da177e4c3f41 Linus Torvalds 2005-04-16 79 module_param_array(isapnp, bool, NULL, 0444); ^1da177e4c3f41 Linus Torvalds 2005-04-16 80 MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); ^1da177e4c3f41 Linus Torvalds 2005-04-16 81 #endif ^1da177e4c3f41 Linus Torvalds 2005-04-16 82 :::::: The code at line 55 was first introduced by commit :::::: 1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 Linux-2.6.12-rc2 :::::: TO: Linus Torvalds <torvalds(a)ppc970.osdl.org> :::::: CC: Linus Torvalds <torvalds(a)ppc970.osdl.org> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

ucm2: acp3x-es83xx: introduce UCM support for acp3x-es83xx sound card
by GitHub pull_request - opened 27 Jan '24

27 Jan '24

alsa-project/alsa-ucm-conf pull request #391 was opened from codepayne: Based on already existing UCM support in ucm2/Intel/sof-essx8336 This commit adds support for a line of Huawei laptops that use the AMD ACP controller and a es8336 codec. The kernel machine driver is already upstream. The proposed settings are based on the intel es83xx support already present, but with the difference that this line of laptops do not use an analog microphone that is connected to the codec. [alsa-info.zip](https://github.com/alsa-project/alsa-ucm-conf/files/14073662… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/391 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/391.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 00/13] of: property: add port base loop
by Kuninori Morimoto 27 Jan '24

27 Jan '24

Hi Rob We have endpoint base functions - of_graph_get_next_endpoint() - of_graph_get_endpoint_count() - for_each_endpoint_of_node() But to handling "port" base things, it is not useful. We want to have "port" base functions, too. This patch-set adds it. Because current existing drivers couldn't use "port" base functions, it were implemented in a different way. This patch-set doesn't try to full-replace to avoid unknown bug, try easy / quick replace only for now, but easy to know how "port" base functions are needed. Because I can't test the driver which I can't use, non-ASoC drivers needs Tested-by, Acked-by. Kuninori Morimoto (13): of: property: add port base loop of: property: use of_graph_get_next_port() on of_graph_get_next_endpoint() of: property: add of_graph_get_next_endpoint_raw() drm: omapdrm: use of_graph_get_next_endpoint_raw() media: xilinx-tpg: use of_graph_get_next_endpoint_raw() ASoC: audio-graph-card.c: use of_graph_get_next_endpoint_raw() ASoC: audio-graph-card2: use of_graph_get_next_port() ASoC: audio-graph-card2.c: use of_graph_get_next_endpoint_raw() ASoC: test-component: use for_each_port_of_node() fbdev: omapfb: use of_graph_get_remote_port() fbdev: omapfb: use of_graph_get_next_port() fbdev: omapfb: use of_graph_get_next_endpoint_raw() fbdev: omapfb: use of_graph_get_next_endpoint() drivers/gpu/drm/omapdrm/dss/dpi.c | 2 +- drivers/gpu/drm/omapdrm/dss/sdi.c | 2 +- drivers/media/platform/xilinx/xilinx-tpg.c | 2 +- drivers/of/property.c | 92 +++++++++++++--- drivers/video/fbdev/omap2/omapfb/dss/dpi.c | 2 +- drivers/video/fbdev/omap2/omapfb/dss/dsi.c | 2 +- drivers/video/fbdev/omap2/omapfb/dss/dss-of.c | 101 +----------------- drivers/video/fbdev/omap2/omapfb/dss/dss.c | 8 +- drivers/video/fbdev/omap2/omapfb/dss/hdmi4.c | 2 +- drivers/video/fbdev/omap2/omapfb/dss/hdmi5.c | 2 +- drivers/video/fbdev/omap2/omapfb/dss/sdi.c | 2 +- drivers/video/fbdev/omap2/omapfb/dss/venc.c | 2 +- include/linux/of_graph.h | 23 ++++ include/video/omapfb_dss.h | 11 -- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/audio-graph-card2.c | 31 ++---- sound/soc/generic/test-component.c | 2 +- 17 files changed, 126 insertions(+), 162 deletions(-) -- 2.25.1

2 15

[PATCH] ASoC: dt-bindings: samsung,tm2: Correct "audio-codec" constraints
by Rob Herring 26 Jan '24

26 Jan '24

The "audio-codec" constraints define how many entries(2), but not the size of each entry. Each entry is a single phandle. Define the size with an inner 'items' list. Signed-off-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/sound/samsung,tm2.yaml | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2.yaml b/Documentation/devicetree/bindings/sound/samsung,tm2.yaml index 760592599143..cbc7ba37362a 100644 --- a/Documentation/devicetree/bindings/sound/samsung,tm2.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,tm2.yaml @@ -25,8 +25,11 @@ properties: description: Phandles to the codecs. $ref: /schemas/types.yaml#/definitions/phandle-array items: - - description: Phandle to the WM5110 audio codec. - - description: Phandle to the HDMI transmitter node. + - items: + - description: Phandle to the WM5110 audio codec. + - items: + - description: Phandle to the HDMI transmitter node. + samsung,audio-routing: description: | -- 2.43.0

3 2