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

PinePhone Pro support
by GitHub issues - opened 09 Dec '21

09 Dec '21

alsa-project/alsa-ucm-conf issue #125 was opened from tomfitzhenry: It'd be great to have PinePhone Pro support in alsa-ucm-conf. https://gitlab.com/pine64-org/pine64-alsa-ucm/-/issues/4 is another project that tracks this. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/125 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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PinePhone support
by GitHub issues - opened 09 Dec '21

09 Dec '21

alsa-project/alsa-ucm-conf issue #124 was opened from tomfitzhenry: It'd be great to have PinePhone support in alsa-ucm-conf. https://gitlab.com/pine64-org/pine64-alsa-ucm/-/issues/3 is the canonical source of PinePhone UCM. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/124 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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alsa-lib-1.2.6 regression, no audio
by GitHub issues - opened 09 Dec '21

09 Dec '21

alsa-project/alsa-lib issue #199 was opened from gkubaryk: after upgrading the following packages, there is no audio: =media-plugins/alsa-plugins-1.2.6 =media-sound/alsa-utils-1.2.6 =media-libs/alsa-ucm-conf-1.2.6 =media-libs/alsa-lib-1.2.6 I checked each of the four individually and the last one (alsa-lib, 1.2.5.1) is the one that restored audio. alsa-info.sh output, as run from 1.2.5.1 (but it downloaded the newer version of the script to /tmp) is at http://alsa-project.org/db/?f=f94a60ff547f9cce9e872acae67fdf35473fa754 let me know if I can be of further assistance; I need the audio working for the rest of the day but would be able to provide output from 1.2.6 tomorrow perhaps, if desired. Issue URL : https://github.com/alsa-project/alsa-lib/issues/199 Repository URL: https://github.com/alsa-project/alsa-lib

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Pacman cannot find alsa-lib=1.2.6
by GitHub issues - opened 09 Dec '21

09 Dec '21

alsa-project/alsa-lib issue #198 was opened from TonyDaraguon: Using Arco When I try to update lib32-alsa-lib, my package manager tells me that it couldn't find the dependencies (alsa-lib=1.2.6). What should I do? Issue URL : https://github.com/alsa-project/alsa-lib/issues/198 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v2] kselftest: alsa: Add simplistic test for ALSA mixer controls kselftest
by Mark Brown 08 Dec '21

08 Dec '21

Add a basic test for the mixer control interface. For every control on every sound card in the system it checks that it can read and write the default value where the control supports that and for writeable controls attempts to write all valid values, restoring the default values after each test to minimise disruption for users. There are quite a few areas for improvement - currently no coverage of the generation of notifications, several of the control types don't have any coverage for the values and we don't have any testing of error handling when we attempt to write out of range values - but this provides some basic coverage. This is added as a kselftest since unlike other ALSA test programs it does not require either physical setup of the device or interactive monitoring by users and kselftest is one of the test suites that is frequently run by people doing general automated testing so should increase coverage. It is written in terms of alsa-lib since tinyalsa is not generally packaged for distributions which makes things harder for general users interested in kselftest as a whole but it will be a barrier to people with Android. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- v2: Use pkg-config to get CFLAGS and LDLIBS for alsa-lib. MAINTAINERS | 7 + tools/testing/selftests/Makefile | 3 +- tools/testing/selftests/alsa/.gitignore | 1 + tools/testing/selftests/alsa/Makefile | 9 + tools/testing/selftests/alsa/mixer-test.c | 616 ++++++++++++++++++++++ 5 files changed, 635 insertions(+), 1 deletion(-) create mode 100644 tools/testing/selftests/alsa/.gitignore create mode 100644 tools/testing/selftests/alsa/Makefile create mode 100644 tools/testing/selftests/alsa/mixer-test.c diff --git a/MAINTAINERS b/MAINTAINERS index 79ef55bf2ca7..ba25b33e2f96 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -17946,6 +17946,7 @@ F: Documentation/sound/ F: include/sound/ F: include/uapi/sound/ F: sound/ +F: tools/testing/selftests/alsa SOUND - COMPRESSED AUDIO M: Vinod Koul <vkoul(a)kernel.org> @@ -17965,6 +17966,12 @@ F: include/sound/dmaengine_pcm.h F: sound/core/pcm_dmaengine.c F: sound/soc/soc-generic-dmaengine-pcm.c +SOUND - ALSA SELFTESTS +M: Mark Brown <broonie(a)kernel.org> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Supported +F: tools/testing/selftests/alsa + SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) M: Liam Girdwood <lgirdwood(a)gmail.com> M: Mark Brown <broonie(a)kernel.org> diff --git a/tools/testing/selftests/Makefile b/tools/testing/selftests/Makefile index c852eb40c4f7..d08fe4cfe811 100644 --- a/tools/testing/selftests/Makefile +++ b/tools/testing/selftests/Makefile @@ -1,5 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 -TARGETS = arm64 +TARGETS += alsa +TARGETS += arm64 TARGETS += bpf TARGETS += breakpoints TARGETS += capabilities diff --git a/tools/testing/selftests/alsa/.gitignore b/tools/testing/selftests/alsa/.gitignore new file mode 100644 index 000000000000..3bb7c41266a8 --- /dev/null +++ b/tools/testing/selftests/alsa/.gitignore @@ -0,0 +1 @@ +mixer-test diff --git a/tools/testing/selftests/alsa/Makefile b/tools/testing/selftests/alsa/Makefile new file mode 100644 index 000000000000..f64d9090426d --- /dev/null +++ b/tools/testing/selftests/alsa/Makefile @@ -0,0 +1,9 @@ +# SPDX-License-Identifier: GPL-2.0 +# + +CFLAGS += $(shell pkg-config --cflags alsa) +LDLIBS += $(shell pkg-config --libs alsa) + +TEST_GEN_PROGS := mixer-test + +include ../lib.mk diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c new file mode 100644 index 000000000000..6082efa0b426 --- /dev/null +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -0,0 +1,616 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// kselftest for the ALSA mixer API +// +// Original author: Mark Brown <broonie(a)kernel.org> +// Copyright (c) 2021 Arm Limited + +// This test will iterate over all cards detected in the system, exercising +// every mixer control it can find. This may conflict with other system +// software if there is audio activity so is best run on a system with a +// minimal active userspace. + +#include <stdio.h> +#include <stdlib.h> +#include <stdbool.h> +#include <string.h> +#include <getopt.h> +#include <stdarg.h> +#include <ctype.h> +#include <math.h> +#include <errno.h> +#include <assert.h> +#include <alsa/asoundlib.h> +#include <poll.h> +#include <stdint.h> + +#include "../kselftest.h" + +#define TESTS_PER_CONTROL 3 + +struct card_data { + snd_ctl_t *handle; + int card; + int num_ctls; + snd_ctl_elem_list_t *ctls; + struct card_data *next; +}; + +struct ctl_data { + const char *name; + snd_ctl_elem_id_t *id; + snd_ctl_elem_info_t *info; + snd_ctl_elem_value_t *def_val; + int elem; + struct card_data *card; + struct ctl_data *next; +}; + +int num_cards = 0; +int num_controls = 0; +struct card_data *card_list = NULL; +struct ctl_data *ctl_list = NULL; + +void find_controls(void) +{ + char name[32]; + int card, ctl, err; + struct card_data *card_data; + struct ctl_data *ctl_data; + + card = -1; + if (snd_card_next(&card) < 0 || card < 0) + return; + + while (card >= 0) { + sprintf(name, "hw:%d", card); + + card_data = malloc(sizeof(*card_data)); + if (!card_data) { + ksft_print_msg("Out of memory\n"); + ksft_exit_fail(); + } + + err = snd_ctl_open(&card_data->handle, name, 0); + if (err < 0) { + ksft_print_msg("Failed to get hctl for card %d: %s\n", + card, snd_strerror(err)); + goto next_card; + } + + /* Count controls */ + snd_ctl_elem_list_malloc(&card_data->ctls); + snd_ctl_elem_list(card_data->handle, card_data->ctls); + card_data->num_ctls = snd_ctl_elem_list_get_count(card_data->ctls); + + /* Enumerate control information */ + snd_ctl_elem_list_alloc_space(card_data->ctls, card_data->num_ctls); + snd_ctl_elem_list(card_data->handle, card_data->ctls); + + card_data->card = num_cards++; + card_data->next = card_list; + card_list = card_data; + + num_controls += card_data->num_ctls; + + for (ctl = 0; ctl < card_data->num_ctls; ctl++) { + ctl_data = malloc(sizeof(*ctl_data)); + if (!ctl_data) { + ksft_print_msg("Out of memory\n"); + ksft_exit_fail(); + } + + ctl_data->card = card_data; + ctl_data->elem = ctl; + ctl_data->name = snd_ctl_elem_list_get_name(card_data->ctls, + ctl); + + err = snd_ctl_elem_id_malloc(&ctl_data->id); + if (err < 0) { + ksft_print_msg("Out of memory\n"); + ksft_exit_fail(); + } + + err = snd_ctl_elem_info_malloc(&ctl_data->info); + if (err < 0) { + ksft_print_msg("Out of memory\n"); + ksft_exit_fail(); + } + + err = snd_ctl_elem_value_malloc(&ctl_data->def_val); + if (err < 0) { + ksft_print_msg("Out of memory\n"); + ksft_exit_fail(); + } + + snd_ctl_elem_list_get_id(card_data->ctls, ctl, + ctl_data->id); + snd_ctl_elem_info_set_id(ctl_data->info, ctl_data->id); + err = snd_ctl_elem_info(card_data->handle, + ctl_data->info); + if (err < 0) { + ksft_print_msg("%s getting info for %d\n", + snd_strerror(err), + ctl_data->name); + } + + snd_ctl_elem_value_set_id(ctl_data->def_val, + ctl_data->id); + + ctl_data->next = ctl_list; + ctl_list = ctl_data; + } + + next_card: + if (snd_card_next(&card) < 0) { + ksft_print_msg("snd_card_next"); + break; + } + } +} + +/* + * Check that we can read the default value and it is valid. Write + * tests use the read value to restore the default. + */ +void test_ctl_get_value(struct ctl_data *ctl) +{ + int err; + long int_val; + long long int64_val; + + /* If the control is turned off let's be polite */ + if (snd_ctl_elem_info_is_inactive(ctl->info)) { + ksft_print_msg("%s is inactive\n", ctl->name); + ksft_test_result_skip("get_value.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + /* Can't test reading on an unreadable control */ + if (!snd_ctl_elem_info_is_readable(ctl->info)) { + ksft_print_msg("%s is not readable\n", ctl->name); + ksft_test_result_skip("get_value.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + err = snd_ctl_elem_read(ctl->card->handle, ctl->def_val); + if (err < 0) { + ksft_print_msg("snd_ctl_elem_read() failed: %s\n", + snd_strerror(err)); + goto out; + } + + switch (snd_ctl_elem_info_get_type(ctl->info)) { + case SND_CTL_ELEM_TYPE_NONE: + ksft_print_msg("%s Invalid control type NONE\n", ctl->name); + err = -1; + break; + + case SND_CTL_ELEM_TYPE_BOOLEAN: + int_val = snd_ctl_elem_value_get_boolean(ctl->def_val, 0); + switch (int_val) { + case 0: + case 1: + break; + default: + ksft_print_msg("%s Invalid boolean value %ld\n", + ctl->name, int_val); + err = -1; + break; + } + break; + + case SND_CTL_ELEM_TYPE_INTEGER: + int_val = snd_ctl_elem_value_get_integer(ctl->def_val, 0); + + if (int_val < snd_ctl_elem_info_get_min(ctl->info)) { + ksft_print_msg("%s value %ld less than minimum %ld\n", + ctl->name, int_val, + snd_ctl_elem_info_get_min(ctl->info)); + err = -1; + } + + if (int_val > snd_ctl_elem_info_get_max(ctl->info)) { + ksft_print_msg("%s value %ld more than maximum %ld\n", + ctl->name, int_val, + snd_ctl_elem_info_get_max(ctl->info)); + err = -1; + } + + /* Only check step size if there is one and we're in bounds */ + if (err >= 0 && snd_ctl_elem_info_get_step(ctl->info) && + (int_val - snd_ctl_elem_info_get_min(ctl->info) % + snd_ctl_elem_info_get_step(ctl->info))) { + ksft_print_msg("%s value %ld invalid for step %ld minimum %ld\n", + ctl->name, int_val, + snd_ctl_elem_info_get_step(ctl->info), + snd_ctl_elem_info_get_min(ctl->info)); + err = -1; + } + break; + + case SND_CTL_ELEM_TYPE_INTEGER64: + int64_val = snd_ctl_elem_value_get_integer64(ctl->def_val, 0); + + if (int64_val < snd_ctl_elem_info_get_min64(ctl->info)) { + ksft_print_msg("%s value %lld less than minimum %lld\n", + ctl->name, int64_val, + snd_ctl_elem_info_get_min64(ctl->info)); + err = -1; + } + + if (int64_val > snd_ctl_elem_info_get_max64(ctl->info)) { + ksft_print_msg("%s value %lld more than maximum %lld\n", + ctl->name, int64_val, + snd_ctl_elem_info_get_max(ctl->info)); + err = -1; + } + + /* Only check step size if there is one and we're in bounds */ + if (err >= 0 && snd_ctl_elem_info_get_step64(ctl->info) && + (int64_val - snd_ctl_elem_info_get_min64(ctl->info)) % + snd_ctl_elem_info_get_step64(ctl->info)) { + ksft_print_msg("%s value %lld invalid for step %lld minimum %lld\n", + ctl->name, int64_val, + snd_ctl_elem_info_get_step64(ctl->info), + snd_ctl_elem_info_get_min64(ctl->info)); + err = -1; + } + break; + + default: + /* No tests for other types */ + ksft_test_result_skip("get_value.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + +out: + ksft_test_result(err >= 0, "get_value.%d.%d\n", + ctl->card->card, ctl->elem); +} + +bool show_mismatch(struct ctl_data *ctl, int index, + snd_ctl_elem_value_t *read_val, + snd_ctl_elem_value_t *expected_val) +{ + long long expected_int, read_int; + + /* + * We factor out the code to compare values representable as + * integers, ensure that check doesn't log otherwise. + */ + expected_int = 0; + read_int = 0; + + switch (snd_ctl_elem_info_get_type(ctl->info)) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + expected_int = snd_ctl_elem_value_get_boolean(expected_val, + index); + read_int = snd_ctl_elem_value_get_boolean(read_val, index); + break; + + case SND_CTL_ELEM_TYPE_INTEGER: + expected_int = snd_ctl_elem_value_get_integer(expected_val, + index); + read_int = snd_ctl_elem_value_get_integer(read_val, index); + break; + + case SND_CTL_ELEM_TYPE_INTEGER64: + expected_int = snd_ctl_elem_value_get_integer64(expected_val, + index); + read_int = snd_ctl_elem_value_get_integer64(read_val, + index); + break; + + case SND_CTL_ELEM_TYPE_ENUMERATED: + expected_int = snd_ctl_elem_value_get_enumerated(expected_val, + index); + read_int = snd_ctl_elem_value_get_enumerated(read_val, + index); + break; + + default: + break; + } + + if (expected_int != read_int) { + ksft_print_msg("%s.%d expected %lld but read %lld\n", + ctl->name, index, expected_int, read_int); + return true; + } else { + return false; + } +} + +/* + * Write a value then if possible verify that we get the expected + * result. An optional expected value can be provided if we expect + * the write to fail, for verifying that invalid writes don't corrupt + * anything. + */ +int write_and_verify(struct ctl_data *ctl, + snd_ctl_elem_value_t *write_val, + snd_ctl_elem_value_t *expected_val) +{ + int err, i; + bool error_expected, mismatch_shown; + snd_ctl_elem_value_t *read_val, *w_val; + snd_ctl_elem_value_alloca(&read_val); + snd_ctl_elem_value_alloca(&w_val); + + /* + * We need to copy the write value since writing can modify + * the value which causes surprises, and allocate an expected + * value if we expect to read back what we wrote. + */ + snd_ctl_elem_value_copy(w_val, write_val); + if (expected_val) { + error_expected = true; + } else { + error_expected = false; + snd_ctl_elem_value_alloca(&expected_val); + snd_ctl_elem_value_copy(expected_val, write_val); + } + + /* + * Do the write, if we have an expected value ignore the error + * and carry on to validate the expected value. + */ + err = snd_ctl_elem_write(ctl->card->handle, w_val); + if (err < 0 && !error_expected) { + ksft_print_msg("snd_ctl_elem_write() failed: %s\n", + snd_strerror(err)); + return err; + } + + /* Can we do the verification part? */ + if (!snd_ctl_elem_info_is_readable(ctl->info)) + return err; + + snd_ctl_elem_value_set_id(read_val, ctl->id); + + err = snd_ctl_elem_read(ctl->card->handle, read_val); + if (err < 0) { + ksft_print_msg("snd_ctl_elem_read() failed: %s\n", + snd_strerror(err)); + return err; + } + + /* + * Use the libray to compare values, if there's a mismatch + * carry on and try to provide a more useful diagnostic than + * just "mismatch". + */ + if (!snd_ctl_elem_value_compare(expected_val, read_val)) + return 0; + + mismatch_shown = false; + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) + if (show_mismatch(ctl, i, read_val, expected_val)) + mismatch_shown = true; + + if (!mismatch_shown) + ksft_print_msg("%s read and written values differ\n", + ctl->name); + + return -1; +} + +/* + * Make sure we can write the default value back to the control, this + * should validate that at least some write works. + */ +void test_ctl_write_default(struct ctl_data *ctl) +{ + int err; + + /* If the control is turned off let's be polite */ + if (snd_ctl_elem_info_is_inactive(ctl->info)) { + ksft_print_msg("%s is inactive\n", ctl->name); + ksft_test_result_skip("write_default.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + if (!snd_ctl_elem_info_is_writable(ctl->info)) { + ksft_print_msg("%s is not writeable\n", ctl->name); + ksft_test_result_skip("write_default.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + /* No idea what the default was for unreadable controls */ + if (!snd_ctl_elem_info_is_readable(ctl->info)) { + ksft_print_msg("%s couldn't read default\n", ctl->name); + ksft_test_result_skip("write_default.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + err = write_and_verify(ctl, ctl->def_val, NULL); + + ksft_test_result(err >= 0, "write_default.%d.%d\n", + ctl->card->card, ctl->elem); +} + +bool test_ctl_write_valid_boolean(struct ctl_data *ctl) +{ + int err, i, j; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + snd_ctl_elem_value_set_id(val, ctl->id); + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + for (j = 0; j < 2; j++) { + snd_ctl_elem_value_set_boolean(val, i, j); + err = write_and_verify(ctl, val, NULL); + if (err != 0) + fail = true; + } + } + + return !fail; +} + +bool test_ctl_write_valid_integer(struct ctl_data *ctl) +{ + int err; + int i; + long j, step; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + snd_ctl_elem_value_set_id(val, ctl->id); + + step = snd_ctl_elem_info_get_step(ctl->info); + if (!step) + step = 1; + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + for (j = snd_ctl_elem_info_get_min(ctl->info); + j <= snd_ctl_elem_info_get_max(ctl->info); j += step) { + + snd_ctl_elem_value_set_integer(val, i, j); + err = write_and_verify(ctl, val, NULL); + if (err != 0) + fail = true; + } + } + + + return !fail; +} + +bool test_ctl_write_valid_integer64(struct ctl_data *ctl) +{ + int err, i; + long long j, step; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + snd_ctl_elem_value_set_id(val, ctl->id); + + step = snd_ctl_elem_info_get_step64(ctl->info); + if (!step) + step = 1; + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + for (j = snd_ctl_elem_info_get_min64(ctl->info); + j <= snd_ctl_elem_info_get_max64(ctl->info); j += step) { + + snd_ctl_elem_value_set_integer64(val, i, j); + err = write_and_verify(ctl, val, NULL); + if (err != 0) + fail = true; + } + } + + + return !fail; +} + +bool test_ctl_write_valid_enumerated(struct ctl_data *ctl) +{ + int err, i, j; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + snd_ctl_elem_value_set_id(val, ctl->id); + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + for (j = 0; j < snd_ctl_elem_info_get_items(ctl->info); j++) { + snd_ctl_elem_value_set_enumerated(val, i, j); + err = write_and_verify(ctl, val, NULL); + if (err != 0) + fail = true; + } + } + + return !fail; +} + +void test_ctl_write_valid(struct ctl_data *ctl) +{ + bool pass; + int err; + + /* If the control is turned off let's be polite */ + if (snd_ctl_elem_info_is_inactive(ctl->info)) { + ksft_print_msg("%s is inactive\n", ctl->name); + ksft_test_result_skip("write_valid.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + if (!snd_ctl_elem_info_is_writable(ctl->info)) { + ksft_print_msg("%s is not writeable\n", ctl->name); + ksft_test_result_skip("write_valid.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + switch (snd_ctl_elem_info_get_type(ctl->info)) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + pass = test_ctl_write_valid_boolean(ctl); + break; + + case SND_CTL_ELEM_TYPE_INTEGER: + pass = test_ctl_write_valid_integer(ctl); + break; + + case SND_CTL_ELEM_TYPE_INTEGER64: + pass = test_ctl_write_valid_integer64(ctl); + break; + + case SND_CTL_ELEM_TYPE_ENUMERATED: + pass = test_ctl_write_valid_enumerated(ctl); + break; + + default: + /* No tests for this yet */ + ksft_test_result_skip("write_valid.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + /* Restore the default value to minimise disruption */ + err = write_and_verify(ctl, ctl->def_val, NULL); + if (err < 0) + pass = false; + + ksft_test_result(pass, "write_valid.%d.%d\n", + ctl->card->card, ctl->elem); +} + +int main(void) +{ + struct ctl_data *ctl; + + ksft_print_header(); + + find_controls(); + + ksft_set_plan(num_controls * TESTS_PER_CONTROL); + + for (ctl = ctl_list; ctl != NULL; ctl = ctl->next) { + /* + * Must test get_value() before we write anything, the + * test stores the default value for later cleanup. + */ + test_ctl_get_value(ctl); + test_ctl_write_default(ctl); + test_ctl_write_valid(ctl); + } + + ksft_exit_pass(); + + return 0; +} -- 2.30.2

6 18

[PATCH] kselftest: alsa: Use private alsa-lib configuration in mixer test
by Jaroslav Kysela 08 Dec '21

08 Dec '21

As mentined by Takashi Sakamoto, the system-wide alsa-lib configuration may override the standard device declarations. This patch use the private alsa-lib configuration to set the predictable environment. Cc: Mark Brown <broonie(a)kernel.org> Cc: Shuah Khan <shuah(a)kernel.org> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: linux-kselftest(a)vger.kernel.org Link: https://lore.kernel.org/alsa-devel/Ya7TAHdMe9i41bsC@workstation/ Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- tools/testing/selftests/alsa/mixer-test.c | 50 ++++++++++++++++++++++- 1 file changed, 49 insertions(+), 1 deletion(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index b87475fb7372..0f533707484c 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -46,22 +46,68 @@ struct ctl_data { struct ctl_data *next; }; +static const char *alsa_config = +"ctl.hw {\n" +" @args [ CARD ]\n" +" @args.CARD.type string\n" +" type hw\n" +" card $CARD\n" +"}\n" +; + int num_cards = 0; int num_controls = 0; struct card_data *card_list = NULL; struct ctl_data *ctl_list = NULL; +#if !defined(SND_LIB_VER) || SND_LIB_VERSION < SND_LIB_VER(1, 2, 6) +int snd_config_load_string(snd_config_t **config, const char *s, size_t size) +{ + snd_input_t *input; + snd_config_t *dst; + int err; + + assert(config && s); + if (size == 0) + size = strlen(s); + err = snd_input_buffer_open(&input, s, size); + if (err < 0) + return err; + err = snd_config_top(&dst); + if (err < 0) { + snd_input_close(input); + return err; + } + err = snd_config_load(dst, input); + snd_input_close(input); + if (err < 0) { + snd_config_delete(dst); + return err; + } + *config = dst; + return 0; +} +#endif + void find_controls(void) { char name[32]; int card, ctl, err; struct card_data *card_data; struct ctl_data *ctl_data; + snd_config_t *config; card = -1; if (snd_card_next(&card) < 0 || card < 0) return; + err = snd_config_load_string(&config, alsa_config, strlen(alsa_config)); + if (err < 0) { + ksft_print_msg("Unable to parse custom alsa-lib configuration: %s\n", + snd_strerror(err)); + ksft_exit_fail(); + } + while (card >= 0) { sprintf(name, "hw:%d", card); @@ -71,7 +117,7 @@ void find_controls(void) ksft_exit_fail(); } - err = snd_ctl_open(&card_data->handle, name, 0); + err = snd_ctl_open_lconf(&card_data->handle, name, 0, config); if (err < 0) { ksft_print_msg("Failed to get hctl for card %d: %s\n", card, snd_strerror(err)); @@ -147,6 +193,8 @@ void find_controls(void) break; } } + + snd_config_delete(config); } /* -- 2.31.1

3 5

vxloader: no VX-compatible cards found VX222v2
by GitHub issues - opened 08 Dec '21

08 Dec '21

alsa-project/alsa-tools issue #9 was opened from LGCW: I have an AMD AM4 motherboard with two legacy PCI slots. The bridge uses the ASMedia ASM1083. Running vxloader produces the following error. ``` vxloader vxloader: no VX-compatible cards found vxloader -D hw:0 vxloader: cannot open hwdep hw:0 ``` If it's anything like the 9632 the firmware has to be loaded in order to function? Best I can tell Debian 11 sees the card. ``` lsmod | grep vx snd_vx222 24576 0 snd_vx_lib 57344 1 snd_vx222 snd_pcm 147456 2 snd_hdsp,snd_vx_lib snd 110592 8 snd_seq_device,snd_hwdep,snd_hdsp,snd_timer,snd_vx_lib,snd_vx222,snd_pcm,snd_rawmidi ``` ``` cat /proc/asound/card0/vx-status Digigram VX222/v2 at 0xf100 & 0xf000, irq 25 Xilinx Firmware: Loaded Device Initialized: Yes DSP audio info: realtime linear8 linear16 linear24 Input Source: Analog Clock Mode: Internal Clock Source: Internal Frequency: 48000 Detected Frequency: 0 Detected UER type: Not Present Min/Max/Cur IBL: 252/2016/252 (granularity=126) ``` I noticed hdsploader picked it up as well. ``` hdsploader hdsploader - firmware loader for RME Hammerfall DSP cards Looking for HDSP + Multiface or Digiface cards : Card 0 : Digigram VX222/v2 at 0xf100 & 0xf000, irq 25 Card 1 : RME Hammerfall HDSP 9632 at 0xf7400000, irq 27 ``` Issue URL : https://github.com/alsa-project/alsa-tools/issues/9 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH v2 0/1] ASoC: rt5682s: Add dapm switch to mute/unmute HP playback output
by Ajit Kumar Pandey 08 Dec '21

08 Dec '21

changes since v1: - Fix typo in commit description Ajit Kumar Pandey (1): ASoC: rt5682s: Add dapm switch to mute/unmute HP playback output sound/soc/codecs/rt5682s.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) -- 2.25.1

2 2

[PATCH] ASoC: rt5682s: Add dapm switch to mute/unmute HP playback output
by Ajit Kumar Pandey 08 Dec '21

08 Dec '21

Add dapm switch in playback path to mute or unmute HP output data. We will set and reset MUTE_SFT bit in RT5682S_HP_CTRL_1 register based on switch value to mute or umute respective channel. Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> --- sound/soc/codecs/rt5682s.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index d49a4f68566d..21ab5f7df422 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -1573,6 +1573,14 @@ static const char * const rt5682s_adcdat_pin_select[] = { "ADCDAT1", "ADCDAT2", }; +/* Out Switch */ +static const struct snd_kcontrol_new hpol_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682S_HP_CTRL_1, + RT5682S_L_MUTE_SFT, 1, 1); +static const struct snd_kcontrol_new hpor_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682S_HP_CTRL_1, + RT5682S_R_MUTE_SFT, 1, 1); + static SOC_VALUE_ENUM_SINGLE_DECL(rt5682s_adcdat_pin_enum, RT5682S_GPIO_CTRL_1, RT5682S_GP4_PIN_SFT, RT5682S_GP4_PIN_MASK, rt5682s_adcdat_pin_select, rt5682s_adcdat_pin_values); @@ -1746,6 +1754,11 @@ static const struct snd_soc_dapm_widget rt5682s_dapm_widgets[] = { SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5682s_hp_amp_event, SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("HPOL Playback", SND_SOC_NOPM, 0, 0, + &hpol_switch), + SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0, + &hpor_switch), + /* CLK DET */ SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682S_CLK_DET, RT5682S_SYS_CLK_DET_SFT, 0, NULL, 0), @@ -1895,8 +1908,10 @@ static const struct snd_soc_dapm_route rt5682s_dapm_routes[] = { {"HP Amp", NULL, "CLKDET SYS"}, {"HP Amp", NULL, "SAR"}, - {"HPOL", NULL, "HP Amp"}, - {"HPOR", NULL, "HP Amp"}, + {"HPOL Playback", "Switch", "HP Amp"}, + {"HPOR Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HPOL Playback"}, + {"HPOR", NULL, "HPOR Playback"}, }; static int rt5682s_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, -- 2.25.1

1 0

[PATCH] ASoC: mediatek: mt8195: silence uninitialized variable warning
by Dan Carpenter 08 Dec '21

08 Dec '21

Smatch complains that we might hit the continue path on every iteration through the loop. sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c:831 mt8195_mt6359_rt1019_rt5682_card_late_probe() error: uninitialized symbol 'sof_comp'. Initialize "sof_comp" to NULL to silence this warning. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index b240610dcef0..11a185da0d96 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -813,7 +813,7 @@ static int mt8195_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, static int mt8195_mt6359_rt1019_rt5682_card_late_probe(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *runtime; - struct snd_soc_component *sof_comp; + struct snd_soc_component *sof_comp = NULL; int i; /* 1. find sof component */ -- 2.20.1

2 1

[PATCH] ASoC: AMD: acp-config: fix missing dependency on SND_SOC_ACPI
by Pierre-Louis Bossart 08 Dec '21

08 Dec '21

With a custom .config, the following error is thrown: ERROR: modpost: "snd_soc_acpi_codec_list" [sound/soc/amd/snd-acp-config.ko] undefined! Fix by adding a clear dependency on SND_SOC_ACPI Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/amd/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 8961b8fd23eb..bcfeb3fc2592 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_AMD_YC_MACH config SND_AMD_ACP_CONFIG tristate "AMD ACP configuration selection" + depends on SND_SOC_ACPI help This option adds an auto detection to determine which ACP driver modules to use -- 2.25.1

2 1

[PATCH 2/2] ASoC: rt5682s: add delay time to fix pop sound issue
by shumingf＠realtek.com 08 Dec '21

08 Dec '21

From: Shuming Fan <shumingf(a)realtek.com> There is a pop noise at the beginning of the capture data. This patch adds the delay time before stereo1 ADC unmute to fix the pop sound issue. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- include/sound/rt5682s.h | 1 + sound/soc/codecs/rt5682s.c | 34 +++++++++++++++++++++++++++++++--- 2 files changed, 32 insertions(+), 3 deletions(-) diff --git a/include/sound/rt5682s.h b/include/sound/rt5682s.h index accfbc2dcdd2..f18d91308b9a 100644 --- a/include/sound/rt5682s.h +++ b/include/sound/rt5682s.h @@ -40,6 +40,7 @@ struct rt5682s_platform_data { enum rt5682s_jd_src jd_src; unsigned int dmic_clk_rate; unsigned int dmic_delay; + unsigned int amic_delay; bool dmic_clk_driving_high; const char *dai_clk_names[RT5682S_DAI_NUM_CLKS]; diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index d49a4f68566d..efa1016831dd 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -1367,6 +1367,31 @@ static int rt5682s_hp_amp_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5682s_stereo1_adc_mixl_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + unsigned int delay = 0; + + if (rt5682s->pdata.amic_delay) + delay = rt5682s->pdata.amic_delay; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(delay); + snd_soc_component_update_bits(component, RT5682S_STO1_ADC_DIG_VOL, + RT5682S_L_MUTE, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_component_update_bits(component, RT5682S_STO1_ADC_DIG_VOL, + RT5682S_L_MUTE, RT5682S_L_MUTE); + break; + } + + return 0; +} + static int sar_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1680,9 +1705,10 @@ static const struct snd_soc_dapm_widget rt5682s_dapm_widgets[] = { /* ADC Mixer */ SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5682S_PWR_DIG_2, RT5682S_PWR_ADC_S1F_BIT, 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), - SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", RT5682S_STO1_ADC_DIG_VOL, - RT5682S_L_MUTE_SFT, 1, rt5682s_sto1_adc_l_mix, - ARRAY_SIZE(rt5682s_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER_E("Stereo1 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5682s_sto1_adc_l_mix, ARRAY_SIZE(rt5682s_sto1_adc_l_mix), + rt5682s_stereo1_adc_mixl_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682S_STO1_ADC_DIG_VOL, RT5682S_R_MUTE_SFT, 1, rt5682s_sto1_adc_r_mix, ARRAY_SIZE(rt5682s_sto1_adc_r_mix)), @@ -2885,6 +2911,8 @@ static int rt5682s_parse_dt(struct rt5682s_priv *rt5682s, struct device *dev) &rt5682s->pdata.dmic_clk_rate); device_property_read_u32(dev, "realtek,dmic-delay-ms", &rt5682s->pdata.dmic_delay); + device_property_read_u32(dev, "realtek,amic-delay-ms", + &rt5682s->pdata.amic_delay); rt5682s->pdata.ldo1_en = of_get_named_gpio(dev->of_node, "realtek,ldo1-en-gpios", 0); -- 2.34.1

2 1

[PATCH 1/2] ASoC: dt-bindings: rt5682s: add AMIC delay time property
by shumingf＠realtek.com 08 Dec '21

08 Dec '21

From: Shuming Fan <shumingf(a)realtek.com> Add the AMIC delay time to control how much delay time (ms) to unmute the stereo1 ADC. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml index 2b8b7b51fe55..d65c0ed5060c 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml @@ -61,6 +61,10 @@ properties: description: | Set the delay time (ms) for the requirement of the particular DMIC. + realtek,amic-delay-ms: + description: | + Set the delay time (ms) for the requirement of the particular platform or AMIC. + realtek,dmic-clk-driving-high: type: boolean description: | -- 2.34.1

2 1

[PATCH] ASoC: Intel: boards: add 'static' qualifiers for max98390 routes
by Pierre-Louis Bossart 08 Dec '21

08 Dec '21

Sparse warnings: sound/soc/intel/boards/sof_maxim_common.c:140:33: error: symbol 'max_98390_dapm_routes' was not declared. Should it be static? sound/soc/intel/boards/sof_maxim_common.c:156:33: error: symbol 'max_98390_tt_dapm_routes' was not declared. Should it be static? Fixes: f316c9d9ba8ea ('ASoC: Intel: boards: add max98390 2/4 speakers support') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/sof_maxim_common.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index 9171d9cd179e..112e89951da0 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -137,7 +137,7 @@ EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); /* * Maxim MAX98390 */ -const struct snd_soc_dapm_route max_98390_dapm_routes[] = { +static const struct snd_soc_dapm_route max_98390_dapm_routes[] = { /* speaker */ { "Left Spk", NULL, "Left BE_OUT" }, { "Right Spk", NULL, "Right BE_OUT" }, @@ -153,7 +153,7 @@ static const struct snd_soc_dapm_widget max_98390_tt_dapm_widgets[] = { SND_SOC_DAPM_SPK("TR Spk", NULL), }; -const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = { +static const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = { /* Tweeter speaker */ { "TL Spk", NULL, "Tweeter Left BE_OUT" }, { "TR Spk", NULL, "Tweeter Right BE_OUT" }, -- 2.25.1

2 1

[PATCH] ASoC: Intel: sof_rt5682: Move rt1015 speaker amp to common file
by Pierre-Louis Bossart 08 Dec '21

08 Dec '21

From: Yong Zhi <yong.zhi(a)intel.com> Move rt1015 driver code to common file to be consistent with rt1011 and rt1015p. No functional change. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/sof_realtek_common.c | 119 +++++++++++++++++++- sound/soc/intel/boards/sof_realtek_common.h | 7 ++ sound/soc/intel/boards/sof_rt5682.c | 107 +----------------- 3 files changed, 127 insertions(+), 106 deletions(-) diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index 2ec34f8df9e1..4cf131310ad3 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -12,12 +12,13 @@ #include <sound/soc-dapm.h> #include <uapi/sound/asound.h> #include "../../codecs/rt1011.h" +#include "../../codecs/rt1015.h" #include "sof_realtek_common.h" /* * Current only 2-amp configuration is supported for rt1011 */ -static const struct snd_soc_dapm_route rt1011_dapm_routes[] = { +static const struct snd_soc_dapm_route speaker_map_lr[] = { /* speaker */ { "Left Spk", NULL, "Left SPO" }, { "Right Spk", NULL, "Right SPO" }, @@ -117,8 +118,8 @@ static int rt1011_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card = rtd->card; int ret; - ret = snd_soc_dapm_add_routes(&card->dapm, rt1011_dapm_routes, - ARRAY_SIZE(rt1011_dapm_routes)); + ret = snd_soc_dapm_add_routes(&card->dapm, speaker_map_lr, + ARRAY_SIZE(speaker_map_lr)); if (ret) dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); return ret; @@ -241,3 +242,115 @@ void sof_rt1015p_codec_conf(struct snd_soc_card *card) card->codec_conf = rt1015p_codec_confs; card->num_configs = ARRAY_SIZE(rt1015p_codec_confs); } + +/* + * RT1015 audio amplifier + */ + +static int rt1015_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int i, fs = 64, ret; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, + params_rate(params) * fs, + params_rate(params) * 256); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT1015_SCLK_S_PLL, + params_rate(params) * 256, + SND_SOC_CLOCK_IN); + if (ret) + return ret; + } + + return 0; +} + +static int rt1015_hw_params_pll_and_tdm(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int i, fs = 100, ret; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, + params_rate(params) * fs, + params_rate(params) * 256); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT1015_SCLK_S_PLL, + params_rate(params) * 256, + SND_SOC_CLOCK_IN); + if (ret) + return ret; + } + /* rx slot 1 for RT1015_DEV0_NAME */ + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), + 0x0, 0x1, 4, 24); + if (ret) + return ret; + + /* rx slot 2 for RT1015_DEV1_NAME */ + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 1), + 0x0, 0x2, 4, 24); + if (ret) + return ret; + + return 0; +} + +static struct snd_soc_ops rt1015_ops = { + .hw_params = rt1015_hw_params, +}; + +static struct snd_soc_codec_conf rt1015_amp_conf[] = { + { + .dlc = COMP_CODEC_CONF(RT1015_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(RT1015_DEV1_NAME), + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link_component rt1015_components[] = { + { + .name = RT1015_DEV0_NAME, + .dai_name = RT1015_CODEC_DAI, + }, + { + .name = RT1015_DEV1_NAME, + .dai_name = RT1015_CODEC_DAI, + }, +}; + +static int speaker_codec_init_lr(struct snd_soc_pcm_runtime *rtd) +{ + return snd_soc_dapm_add_routes(&rtd->card->dapm, speaker_map_lr, + ARRAY_SIZE(speaker_map_lr)); +} + +void sof_rt1015_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = rt1015_amp_conf; + card->num_configs = ARRAY_SIZE(rt1015_amp_conf); +} + +void sof_rt1015_dai_link(struct snd_soc_dai_link *link, unsigned int fs) +{ + link->codecs = rt1015_components; + link->num_codecs = ARRAY_SIZE(rt1015_components); + link->init = speaker_codec_init_lr; + link->ops = &rt1015_ops; + + if (fs == 100) + rt1015_ops.hw_params = rt1015_hw_params_pll_and_tdm; +} diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h index cb0b49b2855c..228ac9c08430 100644 --- a/sound/soc/intel/boards/sof_realtek_common.h +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -28,4 +28,11 @@ void sof_rt1011_codec_conf(struct snd_soc_card *card); void sof_rt1015p_dai_link(struct snd_soc_dai_link *link); void sof_rt1015p_codec_conf(struct snd_soc_card *card); +#define RT1015_CODEC_DAI "rt1015-aif" +#define RT1015_DEV0_NAME "i2c-10EC1015:00" +#define RT1015_DEV1_NAME "i2c-10EC1015:01" + +void sof_rt1015_dai_link(struct snd_soc_dai_link *link, unsigned int fs); +void sof_rt1015_codec_conf(struct snd_soc_card *card); + #endif /* __SOF_REALTEK_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 6cadb5fb72e0..bd6d2e7dea53 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -20,7 +20,6 @@ #include <sound/rt5682.h> #include <sound/rt5682s.h> #include <sound/soc-acpi.h> -#include "../../codecs/rt1015.h" #include "../../codecs/rt5682.h" #include "../../codecs/rt5682s.h" #include "../../codecs/hdac_hdmi.h" @@ -400,67 +399,6 @@ static struct snd_soc_ops sof_rt5682_ops = { .hw_params = sof_rt5682_hw_params, }; -static int sof_rt1015_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai; - int i, fs, ret; - - if (!snd_soc_card_get_codec_dai(card, "rt1015-aif")) - return 0; - - if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_100FS) - fs = 100; - else - fs = 64; - - for_each_rtd_codec_dais(rtd, i, codec_dai) { - ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, - params_rate(params) * fs, - params_rate(params) * 256); - if (ret < 0) { - dev_err(card->dev, "failed to set pll\n"); - return ret; - } - /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, RT1015_SCLK_S_PLL, - params_rate(params) * 256, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "failed to set sysclk\n"); - return ret; - } - - if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_100FS) { - if (!strcmp(codec_dai->component->name, "i2c-10EC1015:00")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x0, 0x1, 4, 24); - if (ret < 0) { - dev_err(card->dev, "failed to set tdm slot\n"); - return ret; - } - } - - if (!strcmp(codec_dai->component->name, "i2c-10EC1015:01")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x0, 0x2, 4, 24); - if (ret < 0) { - dev_err(card->dev, "failed to set tdm slot\n"); - return ret; - } - } - } - } - - return 0; -} - -static struct snd_soc_ops sof_rt1015_ops = { - .hw_params = sof_rt1015_hw_params, -}; - static struct snd_soc_dai_link_component platform_component[] = { { /* name might be overridden during probe */ @@ -551,22 +489,11 @@ static const struct snd_soc_dapm_route sof_map[] = { { "IN1P", NULL, "Headset Mic" }, }; -static const struct snd_soc_dapm_route speaker_map_lr[] = { - { "Left Spk", NULL, "Left SPO" }, - { "Right Spk", NULL, "Right SPO" }, -}; - static const struct snd_soc_dapm_route dmic_map[] = { /* digital mics */ {"DMic", NULL, "SoC DMIC"}, }; -static int speaker_codec_init_lr(struct snd_soc_pcm_runtime *rtd) -{ - return snd_soc_dapm_add_routes(&rtd->card->dapm, speaker_map_lr, - ARRAY_SIZE(speaker_map_lr)); -} - static int dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -589,17 +516,6 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static struct snd_soc_codec_conf rt1015_amp_conf[] = { - { - .dlc = COMP_CODEC_CONF("i2c-10EC1015:00"), - .name_prefix = "Left", - }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1015:01"), - .name_prefix = "Right", - }, -}; - /* sof audio machine driver for rt5682 codec */ static struct snd_soc_card sof_audio_card_rt5682 = { .name = "rt5682", /* the sof- prefix is added by the core */ @@ -635,17 +551,6 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; -static struct snd_soc_dai_link_component rt1015_components[] = { - { - .name = "i2c-10EC1015:00", - .dai_name = "rt1015-aif", - }, - { - .name = "i2c-10EC1015:01", - .dai_name = "rt1015-aif", - }, -}; - static struct snd_soc_dai_link_component dummy_component[] = { { .name = "snd-soc-dummy", @@ -798,10 +703,8 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].id = id; if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) { - links[id].codecs = rt1015_components; - links[id].num_codecs = ARRAY_SIZE(rt1015_components); - links[id].init = speaker_codec_init_lr; - links[id].ops = &sof_rt1015_ops; + sof_rt1015_dai_link(&links[id], (sof_rt5682_quirk & + SOF_RT1015_SPEAKER_AMP_100FS) ? 100 : 64); } else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) { sof_rt1015p_dai_link(&links[id]); } else if (sof_rt5682_quirk & @@ -995,10 +898,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_audio_card_rt5682.dai_link = dai_links; - if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) { - sof_audio_card_rt5682.codec_conf = rt1015_amp_conf; - sof_audio_card_rt5682.num_configs = ARRAY_SIZE(rt1015_amp_conf); - } + if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) + sof_rt1015_codec_conf(&sof_audio_card_rt5682); INIT_LIST_HEAD(&ctx->hdmi_pcm_list); -- 2.25.1

2 1

[PATCH] sound: fsl: add missing put_device() call in imx_hdmi_probe()
by Qing Wang 08 Dec '21

08 Dec '21

From: Wang Qing <wangqing(a)vivo.com> of_find_device_by_node() takes a reference to the embedded struct device which needs to be dropped when error return. Signed-off-by: Wang Qing <wangqing(a)vivo.com> --- sound/soc/fsl/imx-hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index 929f69b..89fd8ad --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -125,6 +125,7 @@ static int imx_hdmi_probe(struct platform_device *pdev) data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); if (!data) { + put_device(&cpu_pdev->dev); ret = -ENOMEM; goto fail; } -- 2.7.4

2 1

Re: [Linux-stm32] [PATCH v2 1/4] ASoC: dt-bindings: stm32: i2s: add audio-graph-card port
by Alexandre TORGUE 08 Dec '21

08 Dec '21

Hi Ahmad On 12/7/21 2:59 PM, Ahmad Fatoum wrote: > Hello Alex, > > On 07.12.21 14:52, Alexandre TORGUE wrote: >> Hi Rob >> >> On 12/1/21 11:34 PM, Rob Herring wrote: >>> On Fri, Nov 26, 2021 at 11:25:27AM +0100, Olivier MOYSAN wrote: >>>> Hi Rob, >>>> >>>> On 11/25/21 10:26 PM, Rob Herring wrote: >>>>> On Thu, 25 Nov 2021 15:40:50 +0100, Olivier Moysan wrote: >>>>>> The STM2 I2S DAI can be connected via the audio-graph-card. >>>>>> Add port entry into the bindings. >>>>>> >>>>>> Signed-off-by: Olivier Moysan <olivier.moysan(a)foss.st.com> >>>>>> --- >>>>>> Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml | 5 +++++ >>>>>> 1 file changed, 5 insertions(+) >>>>>> >>>>> >>>>> Running 'make dtbs_check' with the schema in this patch gives the >>>>> following warnings. Consider if they are expected or the schema is >>>>> incorrect. These may not be new warnings. >>>>> >>>>> Note that it is not yet a requirement to have 0 warnings for dtbs_check. >>>>> This will change in the future. >>>>> >>>>> Full log is available here: https://patchwork.ozlabs.org/patch/1559750 >>>>> >>>>> >>>>> audio-controller@4000b000: 'port' does not match any of the regexes: '^port@[0-9]', 'pinctrl-[0-9]+' >>>>> arch/arm/boot/dts/stm32mp157a-dk1.dt.yaml >>>>> arch/arm/boot/dts/stm32mp157c-dk2.dt.yaml >>>>> >>>> >>>> This warning is not a new one. >>>> >>>> The i2s2 node in stm32mp15xx-dkx.dtsi would require the following binding: >>>> port: >>>> $ref: audio-graph-port.yaml# >>>> unevaluatedProperties: false >>>> >>>> However the spi binding requires to introduce a unit address: >>>> patternProperties: >>>> '^port@[0-9]': >>>> $ref: audio-graph-port.yaml# >>>> unevaluatedProperties: false >>>> >>>> The warning can be removed by re-ordering the bindings patches in the serie, >>>> as "additionalProperties: true" makes the check more tolerant on extra >>>> properties. >>> >>> That's never right. >>> >>>> The patch "ASoC: dt-bindings: stm32: i2s: add audio-graph-card port" can >>>> even be merely dropped. >>>> So, I suggest to resend the serie without audio-graph-card patch. >>> >>> Only if you aren't using audio-graph-card. >>> >>>> >>>> Does it sound too permissive to you ? >>> >>> I think perhaps you need to combine the schemas into 1. Or you need to >>> restructure your dtsi files such that you only add spi specific >>> properties when spi mode is enabled and only add i2s specific properties >>> when i2s mode is enabled. Or use the /delete-property/ directive. >> >> Initially the aim of this series was to fix a "make W=1" warnings seen on spi and i2s nodes (duplicate unit-address). Moving both nodes in a common node + using a different compatible depending on SPI or I2S usage sounded good) but it is not enough. In this series the common node is named as following: "spi2s2: spi@4000b000". It is fine for a spi usage but if we want to use this "common node" with I2S compatible and specific bindings, the node name remains spi@... and then specific spi checks are done. For this with this series applied we got this issue reported by spi-controller.yaml: >> >> spi@4000b000: port@0: 'compatible' is a required property >> >> So, if we use two separates nodes we got W=1 warning and if we use a common node we got yaml check issue. One possibility would be to use a common node with a new node name (for example i2spi@...) but I think it is not acceptable. >> >> How to progress ? > > Atmel Flexcom can be configured to be either UART, SPI or i2c. Functions > are child nodes of the flexcom node and the MFD driver matching against it, > just configure the operating mode and then calls of_platform_populate. > > Would something along these lines fit here as well? Yes it could but in my mind it was not a MFD as both feature cannot be used at the same time: it is either SPI or I2S and choice is done "statically" in device tree depending board usage. Lee, what it is your feeling about that ? Will you accept to add a MFD driver for this SPI/I2S peripheral whose prurpose is only to populate child node (either SPI or I2S) ? Cheers Alex > > Cheers, > Ahmad > >> >> Thanks >> Alex >> >> >>> Rob >>> >> >> _______________________________________________ >> Linux-stm32 mailing list >> Linux-stm32(a)st-md-mailman.stormreply.com >> https://st-md-mailman.stormreply.com/mailman/listinfo/linux-stm32 >> > >

2 2

[PATCH 1/1] ASoC: dt-bindings: audio-graph-port: enable both flag/phandle for bitclock/frame-master
by Kuninori Morimoto 08 Dec '21

08 Dec '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_daifmt_parse_clock_provider_raw() is handling both bitclock/frame-master, and is supporting both flag/phandle. Current DT is assuming it is flag style. This patch allows both case. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../bindings/sound/audio-graph-port.yaml | 23 +++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index 43e7f86e3b23..6c3645a0d36b 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -14,6 +14,24 @@ select: false allOf: - $ref: /schemas/graph.yaml#/$defs/port-base +definitions: + flag_or_phandle: + oneOf: + - type: boolean + const: true + - type: 'null' + - type: array + minItems: 1 + maxItems: 1 + items: + type: array + items: + phandle: true + type: integer + maximum: 0xffffffff + minItems: 1 + maxItems: 1 + properties: prefix: description: "device name prefix" @@ -42,10 +60,11 @@ patternProperties: $ref: /schemas/types.yaml#/definitions/flag frame-master: description: Indicates dai-link frame master. - $ref: /schemas/types.yaml#/definitions/phandle + $ref: "#/definitions/flag_or_phandle" bitclock-master: description: Indicates dai-link bit clock master - $ref: /schemas/types.yaml#/definitions/phandle + $ref: "#/definitions/flag_or_phandle" + dai-format: description: audio format. items: -- 2.25.1

1 1

topology: Add option to pass include path for conditional includes
by GitHub pull_request - opened 08 Dec '21

08 Dec '21

alsa-project/alsa-utils pull request #125 was opened from ranj063: The include path passed with -I option will override the relative include path based on the source file. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/125 Patch URL : https://github.com/alsa-project/alsa-utils/pull/125.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Feature request: ACPI NHLT generation for topology2
by GitHub issues - opened 07 Dec '21

07 Dec '21

alsa-project/alsa-utils issue #124 was opened from juimonen: I'd like to add ACPI NHLT generation into topology2 parsing. Why: In certain future sof firmware versions this is needed for DAI configuration Why to alsa-utils: adding it to kernel driver was internally rejected, also I can't link to topology2 parsing outside alsa-utils Good thing: if you ever wondered how acpi nhlt blobs are created and what they contain in intel platforms -> this code is doing it. Bad things: I can think many, but I'm trying to do this with least harm to every party (kernel driver, firmware) Example of the code can be found here: https://github.com/thesofproject/alsa-utils/pull/4 So mainly asking for comments/suggestions from @perexg @lrgirdwo @ranj063 FYI. Issue URL : https://github.com/alsa-project/alsa-utils/issues/124 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[RESEND, PATCH v6 1/2] dt-bindings: sound: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 07 Dec '21

07 Dec '21

DT bindings for tlv320adc3xxx driver, currently supporting Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 ++++++++++++++++++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 ++++ 2 files changed, 165 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..c4fed6335230 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw(a)axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + reset-gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,micbias1-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS1 pin + + ti,micbias2-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: tlv320adc3101@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/include/dt-bindings/sound/tlv320adc3xxx.h b/include/dt-bindings/sound/tlv320adc3xxx.h new file mode 100644 index 000000000000..3b3fa43fa961 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Devicetree bindings definitions for tlv320adc3xxx driver. + * + * Copyright (C) 2021 Axis Communications AB + */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +#define ADC3XXX_GPIO_DISABLED 0 /* I/O buffers powered down */ +#define ADC3XXX_GPIO_INPUT 1 /* Various non-GPIO inputs */ +#define ADC3XXX_GPIO_GPI 2 /* General purpose input */ +#define ADC3XXX_GPIO_GPO 3 /* General purpose output */ +#define ADC3XXX_GPIO_CLKOUT 4 /* Source set in reg. CLKOUT_MUX */ +#define ADC3XXX_GPIO_INT1 5 /* INT1 output */ +#define ADC3XXX_GPIO_INT2 6 /* INT2 output */ +/* value 7 is reserved */ +#define ADC3XXX_GPIO_SECONDARY_BCLK 8 /* Codec interface secondary BCLK */ +#define ADC3XXX_GPIO_SECONDARY_WCLK 9 /* Codec interface secondary WCLK */ +#define ADC3XXX_GPIO_ADC_MOD_CLK 10 /* Clock output for digital mics */ +/* values 11-15 reserved */ + +#define ADC3XXX_MICBIAS_OFF 0 /* Micbias pin powered off */ +#define ADC3XXX_MICBIAS_2_0V 1 /* Micbias pin set to 2.0V */ +#define ADC3XXX_MICBIAS_2_5V 2 /* Micbias pin set to 2.5V */ +#define ADC3XXX_MICBIAS_AVDD 3 /* Use AVDD voltage for micbias pin */ + +#endif /* __DT_TLV320ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 2

[PATCH] ALSA: seq: Set upper limit of processed events
by Takashi Iwai 07 Dec '21

07 Dec '21

Currently ALSA sequencer core tries to process the queued events as much as possible when they become dispatchable. If applications try to queue too massive events to be processed at the very same timing, the sequencer core would still try to process such all events, either in the interrupt context or via some notifier; in either away, it might be a cause of RCU stall or such problems. As a potential workaround for those problems, this patch adds the upper limit of the amount of events to be processed. The remaining events are processed in the next batch, so they won't be lost. For the time being, it's limited up to 1000 events per queue, which should be high enough for any normal usages. Reported-by: Zqiang <qiang.zhang1211(a)gmail.com> Reported-by: syzbot+bb950e68b400ab4f65f8(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20211102033222.3849-1-qiang.zhang1211@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_queue.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index d6c02dea976c..bc933104c3ee 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -235,12 +235,15 @@ struct snd_seq_queue *snd_seq_queue_find_name(char *name) /* -------------------------------------------------------- */ +#define MAX_CELL_PROCESSES_IN_QUEUE 1000 + void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) { unsigned long flags; struct snd_seq_event_cell *cell; snd_seq_tick_time_t cur_tick; snd_seq_real_time_t cur_time; + int processed = 0; if (q == NULL) return; @@ -263,6 +266,8 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + if (++processed >= MAX_CELL_PROCESSES_IN_QUEUE) + goto out; /* the rest processed at the next batch */ } /* Process time queue... */ @@ -272,14 +277,19 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + if (++processed >= MAX_CELL_PROCESSES_IN_QUEUE) + goto out; /* the rest processed at the next batch */ } + out: /* free lock */ spin_lock_irqsave(&q->check_lock, flags); if (q->check_again) { q->check_again = 0; - spin_unlock_irqrestore(&q->check_lock, flags); - goto __again; + if (processed < MAX_CELL_PROCESSES_IN_QUEUE) { + spin_unlock_irqrestore(&q->check_lock, flags); + goto __again; + } } q->check_blocked = 0; spin_unlock_irqrestore(&q->check_lock, flags); -- 2.31.1

1 0

[PATCH] ALSA: mixart: Add sanity check for timer notify streams
by Takashi Iwai 07 Dec '21

07 Dec '21

The miXart timer notification is a variable length, and if a hardware is screwed up, we may access over the actual data size. Let's add a sanity check and bail out if an invalid value is received. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/mixart/mixart_core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 853083dd4bad..a047ed0f84e9 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -443,6 +443,8 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) notify = (struct mixart_timer_notify *)mixart_msg_data; BUILD_BUG_ON(sizeof(notify) > sizeof(mixart_msg_data)); + if (snd_BUG_ON(notify->stream_count > ARRAY_SIZE(notify->streams))) + break; for(i=0; i<notify->stream_count; i++) { u32 buffer_id = notify->streams[i].buffer_id; -- 2.31.1

1 0

[PATCH v4 0/5] Add pin control support for lpass sc7280
by Srinivasa Rao Mandadapu 07 Dec '21

07 Dec '21

This patch series is to split lpass variant common pin control functions and SoC specific functions and to add lpass sc7280 pincontrol support. It also Adds dt-bindings for lpass sc7280 lpass lpi pincontrol. Changes Since V3: -- Update separate Kconfig fields for sm8250 and sc7280. -- Update module license and description. -- Move static variables to corresponding .c files from header file. Changes Since V2: -- Add new dt-bindings for sc7280 lpi driver. -- Make clock voting change as separate patch. -- Split existing pincontrol driver and make common functions as part of separate file. -- Rename lpass pincontrol lpi dt-bindings to sm8250 specific dt-bindings Changes Since V1: -- Make lpi pinctrl variant data structure as constant -- Add appropriate commit message -- Change signedoff by sequence. Srinivasa Rao Mandadapu (5): dt-bindings: pinctrl: qcom: Update lpass lpi file name to SoC specific dt-bindings: pinctrl: qcom: Add sc7280 lpass lpi pinctrl bindings pinctrl: qcom: Move chip specific functions to right files pinctrl: qcom: Update clock voting as optional pinctrl: qcom: Add SC7280 lpass pin configuration .../bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml | 130 ----------- .../pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 115 ++++++++++ .../pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml | 130 +++++++++++ drivers/pinctrl/qcom/Kconfig | 16 ++ drivers/pinctrl/qcom/Makefile | 2 + drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 252 +-------------------- drivers/pinctrl/qcom/pinctrl-lpass-lpi.h | 98 ++++++++ drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 169 ++++++++++++++ drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c | 166 ++++++++++++++ 9 files changed, 704 insertions(+), 374 deletions(-) delete mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml create mode 100644 drivers/pinctrl/qcom/pinctrl-lpass-lpi.h create mode 100644 drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c create mode 100644 drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 13

[PATCH] ASoC: soc-core: add the driver component name to the component struc
by Kory Maincent 07 Dec '21

07 Dec '21

The function fmt_single_name is using to sanitize namings, and it return the name set to component->name. This function is returning either the "device_name" or the "device_name.driver_name" for i2c devices. There is no use of the component driver name. If a non i2c driver register two components the function will return the same "device_name" for both components. This could cause unexpected issue, in my case it is a debugfs error which tries to create two directory with the same component name. I have fixed it by prefixing the component name with the driver component name. Signed-off-by: Kory Maincent <kory.maincent(a)bootlin.com> --- sound/soc/soc-core.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index dcf6be4c4aaa..21ff77b231b8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2342,10 +2342,10 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_card); * Simplify DAI link configuration by removing ".-1" from device names * and sanitizing names. */ -static char *fmt_single_name(struct device *dev, int *id) +static char *fmt_single_name(struct device *dev, const char *snd_drv_name, int *id) { const char *devname = dev_name(dev); - char *found, *name; + char *found, *name, *tmp; unsigned int id1, id2; if (devname == NULL) @@ -2380,6 +2380,14 @@ static char *fmt_single_name(struct device *dev, int *id) *id = 0; } + if (snd_drv_name != NULL) { + /* Add driver component name if present */ + tmp = devm_kasprintf(dev, GFP_KERNEL, "%s.%s", snd_drv_name, name); + devm_kfree(dev, name); + name = devm_kstrdup(dev, tmp, GFP_KERNEL); + devm_kfree(dev, tmp); + } + return name; } @@ -2444,7 +2452,7 @@ struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, */ if (legacy_dai_naming && (dai_drv->id == 0 || dai_drv->name == NULL)) { - dai->name = fmt_single_name(dev, &dai->id); + dai->name = fmt_single_name(dev, dai_drv->name, &dai->id); } else { dai->name = fmt_multiple_name(dev, dai_drv); if (dai_drv->id) @@ -2578,7 +2586,7 @@ int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->list); mutex_init(&component->io_mutex); - component->name = fmt_single_name(dev, &component->id); + component->name = fmt_single_name(dev, driver->name, &component->id); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; -- 2.25.1

2 4

[PATCH] sound: ppc: beep: fix clang -Wimplicit-fallthrough
by Anders Roxell 07 Dec '21

07 Dec '21

Clang warns: sound/ppc/beep.c:103:2: warning: unannotated fall-through between switch labels [-Wimplicit-fallthrough] case SND_TONE: break; ^ sound/ppc/beep.c:103:2: note: insert 'break;' to avoid fall-through case SND_TONE: break; ^ break; 1 warning generated. Clang is more pedantic than GCC, which does not warn when failing through to a case that is just break or return. Clang's version is more in line with the kernel's own stance in deprecated.rst. Add athe missing break to silence the warning. Reported-by: Naresh Kamboju <naresh.kamboju(a)linaro.org> Signed-off-by: Anders Roxell <anders.roxell(a)linaro.org> --- sound/ppc/beep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/ppc/beep.c b/sound/ppc/beep.c index 0f4bce1c0d4f..bf289783eafd 100644 --- a/sound/ppc/beep.c +++ b/sound/ppc/beep.c @@ -99,7 +99,7 @@ static int snd_pmac_beep_event(struct input_dev *dev, unsigned int type, return -1; switch (code) { - case SND_BELL: if (hz) hz = 1000; + case SND_BELL: if (hz) hz = 1000; break; case SND_TONE: break; default: return -1; } -- 2.33.0

2 1

Logitech Z10 USB speakers need a volume change before audio works
by Hans de Goede 07 Dec '21

07 Dec '21

Hi, I've a set of Logitech Z10 USB speakers, which act as a USB soundcard. They have this weird glitch where after turning off my PC (and their power-supply as well) and then turning things back on, they are silent until I change the PCM volume control for the speakers inside alsa-mixer. It seems like they need some "set-volume" command to be send over the USB bus to unmute them when initially powered-up / turned on. Is their some existing usb-audio quirk which I can try to work around this? Regards, Hans

2 8

Add headset Mic supported for Lenovo platform
by Kailang 07 Dec '21

07 Dec '21

Hi Takashi, Attach patch will enable headset Mic support on Lenovo platform. Many Thanks. BR, Kailang

2 1

[PATCH] ALSA: mixart: Reduce size of mixart_timer_notify
by Kees Cook 07 Dec '21

07 Dec '21

The mixart_timer_notify structure was larger than could be represented by the mixart_msg_data array storage. Adjust the size to as large as possible to fix the warning seen with -Warray-bounds builds: sound/pci/mixart/mixart_core.c: In function 'snd_mixart_threaded_irq': sound/pci/mixart/mixart_core.c:447:50: error: array subscript 'struct mixart_timer_notify[0]' is partly outside array bounds of 'u32[128]' {aka 'unsigned int[128]'} [-Werror=array-bounds] 447 | for(i=0; i<notify->stream_count; i++) { | ^~ sound/pci/mixart/mixart_core.c:328:12: note: while referencing 'mixart_msg_data' 328 | static u32 mixart_msg_data[MSG_DEFAULT_SIZE / 4]; | ^~~~~~~~~~~~~~~ Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/pci/mixart/mixart_core.c | 3 +-- sound/pci/mixart/mixart_core.h | 10 +++++++++- 2 files changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index fb8895af0363..853083dd4bad 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -23,8 +23,6 @@ #define MSG_DESCRIPTOR_SIZE 0x24 #define MSG_HEADER_SIZE (MSG_DESCRIPTOR_SIZE + 4) -#define MSG_DEFAULT_SIZE 512 - #define MSG_TYPE_MASK 0x00000003 /* mask for following types */ #define MSG_TYPE_NOTIFY 0 /* embedded -> driver (only notification, do not get_msg() !) */ #define MSG_TYPE_COMMAND 1 /* driver <-> embedded (a command has no answer) */ @@ -444,6 +442,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) struct mixart_timer_notify *notify; notify = (struct mixart_timer_notify *)mixart_msg_data; + BUILD_BUG_ON(sizeof(notify) > sizeof(mixart_msg_data)); for(i=0; i<notify->stream_count; i++) { u32 buffer_id = notify->streams[i].buffer_id; diff --git a/sound/pci/mixart/mixart_core.h b/sound/pci/mixart/mixart_core.h index fbf4731a276d..2f0e29ed5d63 100644 --- a/sound/pci/mixart/mixart_core.h +++ b/sound/pci/mixart/mixart_core.h @@ -49,6 +49,7 @@ enum mixart_message_id { MSG_CLOCK_SET_PROPERTIES = 0x200002, }; +#define MSG_DEFAULT_SIZE 512 struct mixart_msg { @@ -251,10 +252,17 @@ struct mixart_sample_pos u32 sample_pos_low_part; } __attribute__((packed)); +/* + * This structure is limited by the size of MSG_DEFAULT_SIZE. Instead of + * having MIXART_MAX_STREAM_PER_CARD * MIXART_MAX_CARDS many streams, + * this is capped to have a total size below MSG_DEFAULT_SIZE. + */ +#define MIXART_MAX_TIMER_NOTIFY_STREAMS \ + ((MSG_DEFAULT_SIZE - sizeof(u32)) / sizeof(struct mixart_sample_pos)) struct mixart_timer_notify { u32 stream_count; - struct mixart_sample_pos streams[MIXART_MAX_STREAM_PER_CARD * MIXART_MAX_CARDS]; + struct mixart_sample_pos streams[MIXART_MAX_TIMER_NOTIFY_STREAMS]; } __attribute__((packed)); -- 2.30.2

2 1

[GIT PULL] ASoC fixes for v5.16-rc4
by Mark Brown 07 Dec '21

07 Dec '21

The following changes since commit 86f74ba3fef56dd1cee19b7a15ae27fc0da5bb61: ASoC: SOF: hda: reset DAI widget before reconfiguring it (2021-11-24 12:57:11 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.16-rc4 for you to fetch changes up to 3fc27e9a1f619b50700f020e6cd270c1b74755f0: ASoC: codecs: wsa881x: fix return values from kcontrol put (2021-12-01 14:13:53 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.16 A relatively large collection of updates, the size is increased quite a bit by there being some repetitive changes for similar issues that occur multiple times with both notifying control value changes and runtime PM. The Rockchip update looks at first glance like a cleanup but fixes instantiation of the hardware on some systems. ---------------------------------------------------------------- Dan Carpenter (1): ASoC: amd: fix uninitialized variable in snd_acp6x_probe() David Heidelberg (1): ASoC: dt-bindings: wlf,wm8962: add missing interrupt property Hui Wang (1): ASoC: SOF: Intel: Retry codec probing if it fails Mark Brown (1): Suspend related fixes on Tegra Nicolas Frattaroli (1): ASoC: rockchip: i2s_tdm: Dup static DAI template Rob Clark (2): ASoC: rt5682: Fix crash due to out of scope stack vars ASoC: rt5682s: Fix crash due to out of scope stack vars Sameer Pujar (6): ASoC: tegra: Balance runtime PM count ASoC: tegra: Use normal system sleep for SFC ASoC: tegra: Use normal system sleep for MVC ASoC: tegra: Use normal system sleep for Mixer ASoC: tegra: Use normal system sleep for AMX ASoC: tegra: Use normal system sleep for ADX Srinivas Kandagatla (4): ASoC: qdsp6: q6routing: Fix return value from msm_routing_put_audio_mixer ASoC: codecs: wcd934x: handle channel mappping list correctly ASoC: codecs: wcd934x: return correct value from mixer put ASoC: codecs: wsa881x: fix return values from kcontrol put .../devicetree/bindings/sound/wlf,wm8962.yaml | 3 + sound/soc/amd/yc/pci-acp6x.c | 3 +- sound/soc/codecs/rt5682.c | 10 +- sound/soc/codecs/rt5682s.c | 10 +- sound/soc/codecs/wcd934x.c | 126 +++++++++++++++------ sound/soc/codecs/wsa881x.c | 16 ++- sound/soc/qcom/qdsp6/q6routing.c | 8 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 52 +++++---- sound/soc/sof/intel/hda-codec.c | 14 ++- sound/soc/tegra/tegra210_adx.c | 4 +- sound/soc/tegra/tegra210_amx.c | 4 +- sound/soc/tegra/tegra210_mixer.c | 4 +- sound/soc/tegra/tegra210_mvc.c | 8 +- sound/soc/tegra/tegra210_sfc.c | 4 +- 14 files changed, 179 insertions(+), 87 deletions(-)

2 1

Add support for some non-alphanumeric variable names in the math expr evaluator
by GitHub issues - edited 07 Dec '21

07 Dec '21

alsa-project/alsa-lib issue #197 was edited from ranj063: The math expression evaluator expects variable names to only be alpha-numeric. So if I have something like "[$DYNAMIC_PIPELINE + 2]", I get the error: ALSA lib confeval.c:263:(snd_config_evaluate_string) wrong expression '$[$DYNAMIC_PIPELINE + 2]' ALSA lib conf.c:5632:(snd_config_expand_custom) Expand error (walk): Invalid argument Failed to expand pre-processor definitions in input config Is it possible to add support for some non-alphanumeric characters in the variable names , esp '_'? I tried this change it seems to suffice: ``` diff --git a/src/confeval.c b/src/confeval.c index a971bf38..7330d67e 100644 --- a/src/confeval.c +++ b/src/confeval.c @@ -190,7 +190,7 @@ int _snd_eval_string(snd_config_t **dst, const char *s, } else { e = s + 1; while (*e) { - if (!isalnum(*e)) + if (!isalnum(*e) && (*e != '_')) break; e++; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/197 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Add support for some non-alphanumeric variable names in the math expr evaluator
by GitHub issues - edited 07 Dec '21

07 Dec '21

alsa-project/alsa-lib issue #197 was edited from ranj063: The math expression evaluator expects variable names to only be alpha-numeric. So if I have something like "[$DYNAMIC_PIPELINE + 2]", I get the error: ALSA lib confeval.c:263:(snd_config_evaluate_string) wrong expression '$[$DYNAMIC_PIPELINE + 77]' ALSA lib conf.c:5632:(snd_config_expand_custom) Expand error (walk): Invalid argument Failed to expand pre-processor definitions in input config Is it possible to add support for some non-alphanumeric characters in the variable names , esp '_'? I tried this change it seems to suffice: ``` diff --git a/src/confeval.c b/src/confeval.c index a971bf38..7330d67e 100644 --- a/src/confeval.c +++ b/src/confeval.c @@ -190,7 +190,7 @@ int _snd_eval_string(snd_config_t **dst, const char *s, } else { e = s + 1; while (*e) { - if (!isalnum(*e)) + if (!isalnum(*e) && (*e != '_')) break; e++; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/197 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Add support for some non-alphanumeric variable names in the math expr evaluator
by GitHub issues - opened 07 Dec '21

07 Dec '21

alsa-project/alsa-lib issue #197 was opened from ranj063: The math expression evaluator expects variable names to only be alpha-numeric. So if I have something like "[$DMIC_CH_COUNT + 2]", I get the error: Issue URL : https://github.com/alsa-project/alsa-lib/issues/197 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Issues with plughw working, however raw hw not on MAX98360A
by Dylanger Daly 07 Dec '21

07 Dec '21

Hi all, I'm currently having issues with my speakers on a Chromebook running the MAX98360A DAC. director@DRIFTWOOD:~$ aplay -D hw:sc7180rt5682max,1 --dump-hw-params /usr/share/sounds/alsa/Front_Center.wav XDG_RUNTIME_DIR (/run/user/0) is not owned by us (uid 1000), but by uid 0! (This could e.g. happen if you try to connect to a non-root PulseAudio as a root u) Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono HW Params of device "hw:sc7180rt5682max,1": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 64 CHANNELS: 4 RATE: 48000 PERIOD_TIME: 64000 PERIOD_SIZE: 3072 PERIOD_BYTES: 24576 PERIODS: 2 BUFFER_TIME: 128000 BUFFER_SIZE: 6144 BUFFER_BYTES: 49152 TICK_TIME: ALL -------------------- aplay: set_params:1358: Channels count non available director@DRIFTWOOD:~$ aplay -D plughw:sc7180rt5682max,1 --dump-hw-params /usr/share/sounds/alsa/Front_Center.wav XDG_RUNTIME_DIR (/run/user/0) is not owned by us (uid 1000), but by uid 0! (This could e.g. happen if you try to connect to a non-root PulseAudio as a root u) Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono HW Params of device "plughw:sc7180rt5682max,1": -------------------- ACCESS: MMAP_INTERLEAVED MMAP_NONINTERLEAVED MMAP_COMPLEX RW_INTERLEAVED RW_NONINTERLEAVED FORMAT: S8 U8 S16_LE S16_BE U16_LE U16_BE S24_LE S24_BE U24_LE U24_BE S32_LE S32_BE U32_LE U32_BE FLOAT_LE FLOAT_BE FLOAT64_LE FLOAT64_BE MU_LAW A_LAW IMA_AE SUBFORMAT: STD SAMPLE_BITS: [4 64] FRAME_BITS: [4 640000] CHANNELS: [1 10000] RATE: [4000 4294967295) PERIOD_TIME: 64000 PERIOD_SIZE: [256 274877907) PERIOD_BYTES: [128 4294967295) PERIODS: (0 2147484) BUFFER_TIME: [1 4294967295] BUFFER_SIZE: [512 549755813] BUFFER_BYTES: [256 4294967295] TICK_TIME: ALL -------------------- When I specify plughw, meaning "add software processing", I can hear perfect output, however if I attempt to use the raw hw handle, I get significantly less format support and the channel count isn't even known, and there is no sound output. I believe this is happening because the driver isn't being picked up, and plughw simply adds what it can to the device making it work. The end result is neither PipeWire or PluseAudio detect the speaker device, I assume this is because they're expecting to use the raw hw device handle Here's my ucm2 config: https://github.com/hexdump0815/imagebuilder/blob/d02a71ace7b93b8efbff842662… Any help would be appreciated, thank you

1 0

Support for 8086:ae20 Intel Smart Sound Technology on Fujitsu Lifebook
by Norbert Preining 07 Dec '21

07 Dec '21

Dear all, (please Cc) today I realized that the built-in microphone (next to the webcam) of my Fujitsu Lifebook U-series is not detected. (I usully use a headset, but due to travelling was forced to use the internal). I tried hard to get this device working, but neither with my self-compiled kernel (5.15.6) nor the Debian kernel distribution kernel 5.15 the microphone seems to be detected. I booted into Windows and tried to find out what the device might be. My **guess** is what is mentioned as Intel Smart Sound Technology with Vendor ID 8086 Device ID ae20 I git grepped the kernel sources for this, and snd-hda-intel contains quite some mentioning about Vendor ID 8086, but ae20 is not appearing. Now the interesting thing is that booting into linux and using lspci and lsusb, I cannot see a single device with the above specifications: lspci: root@sakefilet /usr/src/git/linux/Documentation # lspci 00:00.0 Host bridge: Intel Corporation Comet Lake-U v1 4c Host Bridge/DRAM Controller (rev 0c) 00:02.0 VGA compatible controller: Intel Corporation CometLake-U GT2 [UHD Graphics] (rev 02) 00:04.0 Signal processing controller: Intel Corporation Xeon E3-1200 v5/E3-1500 v5/6th Gen Core Processor Thermal Subsystem (rev 0c) 00:12.0 Signal processing controller: Intel Corporation Comet Lake Thermal Subsytem 00:14.0 USB controller: Intel Corporation Comet Lake PCH-LP USB 3.1 xHCI Host Controller 00:14.2 RAM memory: Intel Corporation Comet Lake PCH-LP Shared SRAM 00:14.3 Network controller: Intel Corporation Comet Lake PCH-LP CNVi WiFi 00:15.0 Serial bus controller [0c80]: Intel Corporation Serial IO I2C Host Controller 00:15.1 Serial bus controller [0c80]: Intel Corporation Comet Lake Serial IO I2C Host Controller 00:15.3 Serial bus controller [0c80]: Intel Corporation Device 02eb 00:16.0 Communication controller: Intel Corporation Comet Lake Management Engine Interface 00:16.3 Serial controller: Intel Corporation Device 02e3 00:1c.0 PCI bridge: Intel Corporation Comet Lake PCI Express Root Port #1 (rev f0) 00:1c.4 PCI bridge: Intel Corporation Comet Lake PCI Express Root Port #5 (rev f0) 00:1d.0 PCI bridge: Intel Corporation Comet Lake PCI Express Root Port #9 (rev f0) 00:1d.4 PCI bridge: Intel Corporation Comet Lake PCI Express Root Port #13 (rev f0) 00:1e.0 Communication controller: Intel Corporation Device 02a8 00:1e.1 Communication controller: Intel Corporation Device 02a9 00:1f.0 ISA bridge: Intel Corporation Comet Lake PCH-LP LPC Premium Controller/eSPI Controller 00:1f.3 Multimedia audio controller: Intel Corporation Comet Lake PCH-LP cAVS 00:1f.4 SMBus: Intel Corporation Comet Lake PCH-LP SMBus Host Controller 00:1f.5 Serial bus controller [0c80]: Intel Corporation Comet Lake SPI (flash) Controller 00:1f.6 Ethernet controller: Intel Corporation Ethernet Connection (10) I219-LM 06:00.0 PCI bridge: Intel Corporation JHL7540 Thunderbolt 3 Bridge [Titan Ridge 4C 2018] (rev 06) 07:00.0 PCI bridge: Intel Corporation JHL7540 Thunderbolt 3 Bridge [Titan Ridge 4C 2018] (rev 06) 07:01.0 PCI bridge: Intel Corporation JHL7540 Thunderbolt 3 Bridge [Titan Ridge 4C 2018] (rev 06) 07:02.0 PCI bridge: Intel Corporation JHL7540 Thunderbolt 3 Bridge [Titan Ridge 4C 2018] (rev 06) 07:04.0 PCI bridge: Intel Corporation JHL7540 Thunderbolt 3 Bridge [Titan Ridge 4C 2018] (rev 06) 08:00.0 System peripheral: Intel Corporation JHL7540 Thunderbolt 3 NHI [Titan Ridge 4C 2018] (rev 06) 3d:00.0 USB controller: Intel Corporation JHL7540 Thunderbolt 3 USB Controller [Titan Ridge 4C 2018] (rev 06) 72:00.0 Non-Volatile memory controller: Samsung Electronics Co Ltd NVMe SSD Controller SM981/PM981/PM983 77:00.0 SD Host controller: O2 Micro, Inc. SD/MMC Card Reader Controller (rev 01) lsusb: Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 001 Device 004: ID 04f2:b6e1 Chicony Electronics Co., Ltd FJ Camera Bus 001 Device 003: ID 298d:2033 Next Biometrics NB-2033-U Bus 001 Device 002: ID 058f:9540 Alcor Micro Corp. AU9540 Smartcard Reader Bus 001 Device 005: ID 8087:0026 Intel Corp. AX201 Bluetooth Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub I attach the output of alsa-info for completeness. The kernel version is 5.15.6 with some patches from xannmod, but I could reproduce the same behaviour with standard Debian kernel 5.15. Am I missing some config option, or is the vendorid/devid not supported by now? Thanks for every suggestion, (Please CC!) Al the best Norbert -- PREINING Norbert https://www.preining.info Fujitsu Research + IFMGA Guide + TU Wien + TeX Live + Debian Dev GPG: 0x860CDC13 fp: F7D8 A928 26E3 16A1 9FA0 ACF0 6CAC A448 860C DC13

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pre-processor only works with relative path based on input file with the complete path
by GitHub issues - opened 06 Dec '21

06 Dec '21

alsa-project/alsa-utils issue #123 was opened from ranj063: For ex: `alsatplg -P sof-hda-generic.conf -o test.conf` fails with: Unable to open included conf file sof-hda-generic.conf/include/components/eqiir/highpass_40hz_0db_48khz.conf Unable to process include file Failed to process includes Failed to process conditional includes in input config whereas: `alsatplg -P ~/work/sof/sof/tools/topology/topology2/sof-hda-generic.conf -o test.conf` works OK. Would it be possible to make the first command work with the use of the ALSA_CONFIG_DIR environment variable? Issue URL : https://github.com/alsa-project/alsa-utils/issues/123 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH AUTOSEL 5.10 08/15] ALSA: hda/hdmi: fix HDA codec entry table order for ADL-P
by Sasha Levin 06 Dec '21

06 Dec '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 289047db1143c42c81820352f195a393ff639a52 ] Keep the HDA_CODEC_ENTRY entries sorted by the codec VID. ADL-P is the only misplaced Intel HDMI codec. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211130124732.696896-2-kai.vehmanen@linux.intel.… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7b91615bcac32..fe725f0f09312 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4362,11 +4362,11 @@ HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), -HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), +HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- 2.33.0

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[PATCH AUTOSEL 5.10 07/15] ALSA: hda: Add Intel DG2 PCI ID and HDMI codec vid
by Sasha Levin 06 Dec '21

06 Dec '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit d85ffff5302b1509efc482e8877c253b0a668b33 ] Add HD Audio PCI ID and HDMI codec vendor ID for Intel DG2. Reviewed-by: Uma Shankar <uma.shankar(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211130124732.696896-1-kai.vehmanen@linux.intel.… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 12 +++++++++++- sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 64115a796af06..3cc936f2cbf8d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -369,7 +369,10 @@ enum { ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c) || \ - ((pci)->device == 0x490d)) + ((pci)->device == 0x490d) || \ + ((pci)->device == 0x4f90) || \ + ((pci)->device == 0x4f91) || \ + ((pci)->device == 0x4f92)) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) @@ -2540,6 +2543,13 @@ static const struct pci_device_id azx_ids[] = { /* DG1 */ { PCI_DEVICE(0x8086, 0x490d), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* DG2 */ + { PCI_DEVICE(0x8086, 0x4f90), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4f91), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4f92), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Alderlake-S */ { PCI_DEVICE(0x8086, 0x7ad0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c65144715af78..7b91615bcac32 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4364,6 +4364,7 @@ HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), -- 2.33.0

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[PATCH AUTOSEL 5.15 12/24] ALSA: hda/hdmi: fix HDA codec entry table order for ADL-P
by Sasha Levin 06 Dec '21

06 Dec '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 289047db1143c42c81820352f195a393ff639a52 ] Keep the HDA_CODEC_ENTRY entries sorted by the codec VID. ADL-P is the only misplaced Intel HDMI codec. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211130124732.696896-2-kai.vehmanen@linux.intel.… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 98633d2684deb..415701bd10ac8 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4380,11 +4380,11 @@ HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), -HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), +HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- 2.33.0

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[PATCH AUTOSEL 5.15 11/24] ALSA: hda: Add Intel DG2 PCI ID and HDMI codec vid
by Sasha Levin 06 Dec '21

06 Dec '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit d85ffff5302b1509efc482e8877c253b0a668b33 ] Add HD Audio PCI ID and HDMI codec vendor ID for Intel DG2. Reviewed-by: Uma Shankar <uma.shankar(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211130124732.696896-1-kai.vehmanen@linux.intel.… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 12 +++++++++++- sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 90e9263ac0bd7..21fec82489bd7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -335,7 +335,10 @@ enum { ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c) || \ - ((pci)->device == 0x490d)) + ((pci)->device == 0x490d) || \ + ((pci)->device == 0x4f90) || \ + ((pci)->device == 0x4f91) || \ + ((pci)->device == 0x4f92)) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) @@ -2472,6 +2475,13 @@ static const struct pci_device_id azx_ids[] = { /* DG1 */ { PCI_DEVICE(0x8086, 0x490d), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* DG2 */ + { PCI_DEVICE(0x8086, 0x4f90), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4f91), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4f92), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Alderlake-S */ { PCI_DEVICE(0x8086, 0x7ad0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 65d2c55399195..98633d2684deb 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4382,6 +4382,7 @@ HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), -- 2.33.0

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GLK board with ESSX8336 sound. Would like to get sound working on this system
by Theodore Kilgore 06 Dec '21

06 Dec '21

Hello, I assume that someone is already working on this but I do not know who. I have some experience with coding and kernel stuff (see drivers/media/usb/gspca). Not experienced with doing sound support. Theodore Kilgore

3 2

[PATCH RFC v1 0/2] ASoC: meson: aiu: two fixes
by Martin Blumenstingl 06 Dec '21

06 Dec '21

Hello Jerome, in this series I am proposing two fixes for the "aiu" driver, used on Amlogic Meson8, Meson8b, Meson8m2, GXBB, GXL and GXM SoCs. The first patch is the result of me trying to understand the way how we get the DMA buffer and address for the audio data. I'm not an expert in terms of DMA. From what I understand we need to inform DMA core about the limitations of the hardware. In case of AIU it's DMA address registers only take 32 bits, so DMA core should be aware of this. The second patch is what I could come up with to fix the infamous I2S buffer underrun issue, also called the "machine gun noise" (MGN) bug. After a lot of testing, debugging and comparing vendor code with the upstream "aiu" driver I have come up with this fix. I have written down my thoughts in the description of that patch. To be clear: these are my thoughts, unfortunately I have no way of proving this other than asking other people to test this patch (off-list I have already received positive feedback along with confirmation that both 2-ch and 6-ch audio are still working fine. Even with Kodi's menu - which is an easy way to reproduce the MGN bug - sound output is fine with this patch). Please let me know what you think about these patches and especially the patch descriptions. Martin Blumenstingl (2): ASoC: meson: aiu: fifo: Add missing dma_coerce_mask_and_coherent() ASoC: meson: aiu: Move AIU_I2S_MISC hold setting to aiu-fifo-i2s sound/soc/meson/aiu-encoder-i2s.c | 33 ------------------------------- sound/soc/meson/aiu-fifo-i2s.c | 12 +++++++++++ sound/soc/meson/aiu-fifo.c | 6 ++++++ 3 files changed, 18 insertions(+), 33 deletions(-) -- 2.34.1

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[PATCH 1/1] ASoC: fsl-asoc-card: Add missing Kconfig option for tlv320aic31xx
by Ariel D'Alessandro 06 Dec '21

06 Dec '21

This is a follow up of patchsets: [RFC patch 0/5] Support BCLK input clock in tlv320aic31xx [PATCH 0/4] fsl-asoc-card: Add optional dt property for setting mclk-id Patch "ASoC: fsl-asoc-card: Support fsl,imx-audio-tlv320aic31xx codec" in "[RFC patch 0/5] Support BCLK input clock in tlv320aic31xx" missed a Kconfig option. Sending incremental patch fix. Regards, Ariel Ariel D'Alessandro (1): ASoC: fsl-asoc-card: Add missing Kconfig option for tlv320aic31xx sound/soc/fsl/Kconfig | 1 + 1 file changed, 1 insertion(+) -- 2.30.2

2 3

[PATCH] sound/soc: remove useless bool conversion to bool variable
by Bernard Zhao 06 Dec '21

06 Dec '21

This patch remove useless bool conversion to bool variable Signed-off-by: Bernard Zhao <bernard(a)vivo.com> --- sound/soc/codecs/cs35l35.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 7a5588f1df01..961a3e07e70f 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -1311,7 +1311,7 @@ static int cs35l35_handle_of_data(struct i2c_client *i2c_client, pdata->gain_zc = of_property_read_bool(np, "cirrus,amp-gain-zc"); classh = of_get_child_by_name(np, "cirrus,classh-internal-algo"); - classh_config->classh_algo_enable = classh ? true : false; + classh_config->classh_algo_enable = (classh != NULL); if (classh_config->classh_algo_enable) { classh_config->classh_bst_override = -- 2.33.1

2 2

[PATCH] ASoC: amd: acp6x-pdm-dma: Constify static snd_soc_dai_ops
by Rikard Falkeborn 06 Dec '21

06 Dec '21

The only usage of acp6x_pdm_dai_ops is to assign its address to the ops field in the snd_soc_dai_driver struct, which is a pointer to const snd_soc_dai_ops. Make it const to allow the compiler to put it in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/amd/yc/acp6x-pdm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/yc/acp6x-pdm-dma.c b/sound/soc/amd/yc/acp6x-pdm-dma.c index e604f4ea524f..7e66393e4153 100644 --- a/sound/soc/amd/yc/acp6x-pdm-dma.c +++ b/sound/soc/amd/yc/acp6x-pdm-dma.c @@ -318,7 +318,7 @@ static int acp6x_pdm_dai_trigger(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_dai_ops acp6x_pdm_dai_ops = { +static const struct snd_soc_dai_ops acp6x_pdm_dai_ops = { .trigger = acp6x_pdm_dai_trigger, }; -- 2.34.1

2 1

[PATCH] ASoC: SOF: Intel: fix build issue related to CODEC_PROBE_ENTRIES
by Kai Vehmanen 06 Dec '21

06 Dec '21

Fix following error: sound/soc/sof/intel/hda-codec.c:132:35: error: use of undeclared identifier 'CODEC_PROBE_RETRIES' Found with config: i386-randconfig-r033-20211202 (https://download.01.org/0day-ci/archive/20211203/202112031943.Twg19fWT-lkp@…) Fixes: 046aede2f847 ("ASoC: SOF: Intel: Retry codec probing if it fails") Reported-by: kernel test robot <lkp(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/intel/hda-codec.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 13cd96e6724a..2f3f4a733d9e 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -20,9 +20,10 @@ #include "../../codecs/hdac_hda.h" #endif /* CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC */ +#define CODEC_PROBE_RETRIES 3 + #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) #define IDISP_VID_INTEL 0x80860000 -#define CODEC_PROBE_RETRIES 3 /* load the legacy HDA codec driver */ static int request_codec_module(struct hda_codec *codec) base-commit: 3fc27e9a1f619b50700f020e6cd270c1b74755f0 -- 2.34.1

2 1

[PATCH v1 1/1] ASoC: zl38060: Setup parent device and get rid of unnecessary of_node assignment
by Andy Shevchenko 06 Dec '21

06 Dec '21

Some of the drivers do not set parent device. This may lead to obstacles during debugging or understanding the device relations from the Linux point of view. Assign parent device for GPIO chips created by these drivers. While at it, let GPIO library to assign of_node from the parent device. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/codecs/zl38060.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/zl38060.c b/sound/soc/codecs/zl38060.c index d20ec1571010..6cae0fb08093 100644 --- a/sound/soc/codecs/zl38060.c +++ b/sound/soc/codecs/zl38060.c @@ -589,9 +589,7 @@ static int zl38_spi_probe(struct spi_device *spi) sizeof(template_chip), GFP_KERNEL); if (!priv->gpio_chip) return -ENOMEM; -#ifdef CONFIG_OF_GPIO - priv->gpio_chip->of_node = dev->of_node; -#endif + priv->gpio_chip->parent = dev; err = devm_gpiochip_add_data(dev, priv->gpio_chip, priv->regmap); if (err) return err; -- 2.33.0

2 1

[PATCH v1 1/2] ASoC: ti: davinci-mcasp: Get rid of duplicate of_node assignment
by Andy Shevchenko 06 Dec '21

06 Dec '21

GPIO library does copy the of_node from the parent device of the GPIO chip, there is no need to repeat this in the individual drivers. Remove assignment here. For the details one may look into the of_gpio_dev_init() implementation. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/ti/davinci-mcasp.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 56a19eeec5c7..81c1ccec5904 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2230,9 +2230,6 @@ static int davinci_mcasp_init_gpiochip(struct davinci_mcasp *mcasp) mcasp->gpio_chip = davinci_mcasp_template_chip; mcasp->gpio_chip.label = dev_name(mcasp->dev); mcasp->gpio_chip.parent = mcasp->dev; -#ifdef CONFIG_OF_GPIO - mcasp->gpio_chip.of_node = mcasp->dev->of_node; -#endif return devm_gpiochip_add_data(mcasp->dev, &mcasp->gpio_chip, mcasp); } -- 2.33.0

3 5

[PATCH 0/2] ASoC: mediatek: support memory-region assignment
by Trevor Wu 06 Dec '21

06 Dec '21

This series of patches adds support for memory-region assignment, so the access region of DMA engine could be restricted. Patches are based on broonie tree "for-next" branch. Trevor Wu (2): ASoC: mediatek: mt8195: support reserved memory assignment dt-bindings: mediatek: mt8195: add memory-region property .../devicetree/bindings/sound/mt8195-afe-pcm.yaml | 8 ++++++++ sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 7 +++++++ 2 files changed, 15 insertions(+) -- 2.18.0

2 3

[PATCH v2 0/4] ASoC: mediatek: Update MT8195 machine driver
by Trevor Wu 06 Dec '21

06 Dec '21

This series of patches adds support for RT5682s headset codec in mt8195 machine drivers, and SOF support on card mt8195-mt6359-rt1019-rt5682 is also included. Patches are based on broonie tree "for-next" branch. Changes since v1: - remove patch3 and patch4 in v1 - add SOF support on card mt8195-mt6359-rt1012-rt5682 - add new propertes to dt-bindings for mt8195-mt6359-rt1019-rt5682 Trevor Wu (4): ASoC: mediatek: mt8195: add headset codec rt5682s support dt-bindings: mediatek: mt8195: add model property ASoC: mediatek: mt8195: add sof support on mt8195-mt6359-rt1019-rt5682 dt-bindings: mediatek: mt8195: add adsp and dai-link property .../sound/mt8195-mt6359-rt1011-rt5682.yaml | 4 + .../sound/mt8195-mt6359-rt1019-rt5682.yaml | 14 + sound/soc/mediatek/Kconfig | 2 + .../mt8195/mt8195-mt6359-rt1011-rt5682.c | 29 +- .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 347 +++++++++++++++++- 5 files changed, 370 insertions(+), 26 deletions(-) -- 2.18.0

2 5

[PATCH 1/2] ASoC: SOF: Intel: pci-tgl: add ADL-N support
by Kai Vehmanen 06 Dec '21

06 Dec '21

Add PCI DID for Intel AlderLake-N. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/pci-tgl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index f2ea34df9741..302068fd0b81 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -114,6 +114,8 @@ static const struct pci_device_id sof_pci_ids[] = { .driver_data = (unsigned long)&adl_desc}, { PCI_DEVICE(0x8086, 0x51cc), /* ADL-M */ .driver_data = (unsigned long)&adl_desc}, + { PCI_DEVICE(0x8086, 0x54c8), /* ADL-N */ + .driver_data = (unsigned long)&adl_desc}, { 0, } }; MODULE_DEVICE_TABLE(pci, sof_pci_ids); base-commit: 0695ad92fe1a0bb7697eb92c6a145a73c5ab0e24 -- 2.34.1

2 2

GIT Project URLS
by GitHub issues - edited 06 Dec '21

06 Dec '21

alsa-project/alsa-lib issue #196 was edited from h1z1: Hi Not really sure where to report this, are the URLS listed on alsa-project.org accurate? They include GIT and HTTP, not HTTPS and are not correct regardless. ex (from https://alsa-project.org/wiki/GIT_Server) ``` Web interface http://git.alsa-project.org/?p=alsa-driver.git ``` Results in what I"m assuming is a 404 ``` $ git clone https://git.alsa-project.org/?p=alsa-driver.git alsa-driver.git Cloning into 'alsa-driver.git'... fatal: repository 'https://git.alsa-project.org/?p=alsa-driver.git/' not found ``` Worth noting gitweb is reachable. Should those URLs be accessible via HTTPS? Issue URL : https://github.com/alsa-project/alsa-lib/issues/196 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

GIT Project URLS
by GitHub issues - opened 06 Dec '21

06 Dec '21

alsa-project/alsa-lib issue #196 was opened from h1z1: Hi Not really sure where to report this, are the URLS listed on alsa-project.org accurate? They include GIT and HTTP, not HTTPS and are not correct regardless. ex ``` Web interface http://git.alsa-project.org/?p=alsa-driver.git ``` Results in what I"m assuming is a 404 ``` $ git clone https://git.alsa-project.org/?p=alsa-driver.git alsa-driver.git Cloning into 'alsa-driver.git'... fatal: repository 'https://git.alsa-project.org/?p=alsa-driver.git/' not found ``` Should those URLs be accessilble via HTTPS? Issue URL : https://github.com/alsa-project/alsa-lib/issues/196 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSA 1.2.6 release
by Jaroslav Kysela 06 Dec '21

06 Dec '21

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-lib alsa-utils alsa-plugins alsa-ucm-conf pyalsa Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.5.1_v1.2.6 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

[PATCH] kselftest: alsa: Add simplistic test for ALSA mixer controls kselftest
by Mark Brown 06 Dec '21

06 Dec '21

Add a basic test for the mixer control interface. For every control on every sound card in the system it checks that it can read and write the default value where the control supports that and for writeable controls attempts to write all valid values, restoring the default values after each test to minimise disruption for users. There are quite a few areas for improvement - currently no coverage of the generation of notifications, several of the control types don't have any coverage for the values and we don't have any testing of error handling when we attempt to write out of range values - but this provides some basic coverage. This is added as a kselftest since unlike other ALSA test programs it does not require either physical setup of the device or interactive monitoring by users and kselftest is one of the test suites that is frequently run by people doing general automated testing so should increase coverage. It is written in terms of alsa-lib since tinyalsa is not generally packaged for distributions which makes things harder for general users interested in kselftest as a whole but it will be a barrier to people with Android. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- MAINTAINERS | 7 + tools/testing/selftests/Makefile | 3 +- tools/testing/selftests/alsa/.gitignore | 1 + tools/testing/selftests/alsa/Makefile | 8 + tools/testing/selftests/alsa/mixer-test.c | 616 ++++++++++++++++++++++ 5 files changed, 634 insertions(+), 1 deletion(-) create mode 100644 tools/testing/selftests/alsa/.gitignore create mode 100644 tools/testing/selftests/alsa/Makefile create mode 100644 tools/testing/selftests/alsa/mixer-test.c diff --git a/MAINTAINERS b/MAINTAINERS index 5b1e79f8e3d8..92deb9a83b8c 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -17934,6 +17934,7 @@ F: Documentation/sound/ F: include/sound/ F: include/uapi/sound/ F: sound/ +F: tools/testing/selftests/alsa SOUND - COMPRESSED AUDIO M: Vinod Koul <vkoul(a)kernel.org> @@ -17953,6 +17954,12 @@ F: include/sound/dmaengine_pcm.h F: sound/core/pcm_dmaengine.c F: sound/soc/soc-generic-dmaengine-pcm.c +SOUND - ALSA SELFTESTS +M: Mark Brown <broonie(a)kernel.org> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Supported +F: tools/testing/selftests/alsa + SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) M: Liam Girdwood <lgirdwood(a)gmail.com> M: Mark Brown <broonie(a)kernel.org> diff --git a/tools/testing/selftests/Makefile b/tools/testing/selftests/Makefile index c852eb40c4f7..d08fe4cfe811 100644 --- a/tools/testing/selftests/Makefile +++ b/tools/testing/selftests/Makefile @@ -1,5 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 -TARGETS = arm64 +TARGETS += alsa +TARGETS += arm64 TARGETS += bpf TARGETS += breakpoints TARGETS += capabilities diff --git a/tools/testing/selftests/alsa/.gitignore b/tools/testing/selftests/alsa/.gitignore new file mode 100644 index 000000000000..3bb7c41266a8 --- /dev/null +++ b/tools/testing/selftests/alsa/.gitignore @@ -0,0 +1 @@ +mixer-test diff --git a/tools/testing/selftests/alsa/Makefile b/tools/testing/selftests/alsa/Makefile new file mode 100644 index 000000000000..e62f227ce02c --- /dev/null +++ b/tools/testing/selftests/alsa/Makefile @@ -0,0 +1,8 @@ +# SPDX-License-Identifier: GPL-2.0 +# + +LDLIBS += -lasound + +TEST_GEN_PROGS := mixer-test + +include ../lib.mk diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c new file mode 100644 index 000000000000..6082efa0b426 --- /dev/null +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -0,0 +1,616 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// kselftest for the ALSA mixer API +// +// Original author: Mark Brown <broonie(a)kernel.org> +// Copyright (c) 2021 Arm Limited + +// This test will iterate over all cards detected in the system, exercising +// every mixer control it can find. This may conflict with other system +// software if there is audio activity so is best run on a system with a +// minimal active userspace. + +#include <stdio.h> +#include <stdlib.h> +#include <stdbool.h> +#include <string.h> +#include <getopt.h> +#include <stdarg.h> +#include <ctype.h> +#include <math.h> +#include <errno.h> +#include <assert.h> +#include <alsa/asoundlib.h> +#include <poll.h> +#include <stdint.h> + +#include "../kselftest.h" + +#define TESTS_PER_CONTROL 3 + +struct card_data { + snd_ctl_t *handle; + int card; + int num_ctls; + snd_ctl_elem_list_t *ctls; + struct card_data *next; +}; + +struct ctl_data { + const char *name; + snd_ctl_elem_id_t *id; + snd_ctl_elem_info_t *info; + snd_ctl_elem_value_t *def_val; + int elem; + struct card_data *card; + struct ctl_data *next; +}; + +int num_cards = 0; +int num_controls = 0; +struct card_data *card_list = NULL; +struct ctl_data *ctl_list = NULL; + +void find_controls(void) +{ + char name[32]; + int card, ctl, err; + struct card_data *card_data; + struct ctl_data *ctl_data; + + card = -1; + if (snd_card_next(&card) < 0 || card < 0) + return; + + while (card >= 0) { + sprintf(name, "hw:%d", card); + + card_data = malloc(sizeof(*card_data)); + if (!card_data) { + ksft_print_msg("Out of memory\n"); + ksft_exit_fail(); + } + + err = snd_ctl_open(&card_data->handle, name, 0); + if (err < 0) { + ksft_print_msg("Failed to get hctl for card %d: %s\n", + card, snd_strerror(err)); + goto next_card; + } + + /* Count controls */ + snd_ctl_elem_list_malloc(&card_data->ctls); + snd_ctl_elem_list(card_data->handle, card_data->ctls); + card_data->num_ctls = snd_ctl_elem_list_get_count(card_data->ctls); + + /* Enumerate control information */ + snd_ctl_elem_list_alloc_space(card_data->ctls, card_data->num_ctls); + snd_ctl_elem_list(card_data->handle, card_data->ctls); + + card_data->card = num_cards++; + card_data->next = card_list; + card_list = card_data; + + num_controls += card_data->num_ctls; + + for (ctl = 0; ctl < card_data->num_ctls; ctl++) { + ctl_data = malloc(sizeof(*ctl_data)); + if (!ctl_data) { + ksft_print_msg("Out of memory\n"); + ksft_exit_fail(); + } + + ctl_data->card = card_data; + ctl_data->elem = ctl; + ctl_data->name = snd_ctl_elem_list_get_name(card_data->ctls, + ctl); + + err = snd_ctl_elem_id_malloc(&ctl_data->id); + if (err < 0) { + ksft_print_msg("Out of memory\n"); + ksft_exit_fail(); + } + + err = snd_ctl_elem_info_malloc(&ctl_data->info); + if (err < 0) { + ksft_print_msg("Out of memory\n"); + ksft_exit_fail(); + } + + err = snd_ctl_elem_value_malloc(&ctl_data->def_val); + if (err < 0) { + ksft_print_msg("Out of memory\n"); + ksft_exit_fail(); + } + + snd_ctl_elem_list_get_id(card_data->ctls, ctl, + ctl_data->id); + snd_ctl_elem_info_set_id(ctl_data->info, ctl_data->id); + err = snd_ctl_elem_info(card_data->handle, + ctl_data->info); + if (err < 0) { + ksft_print_msg("%s getting info for %d\n", + snd_strerror(err), + ctl_data->name); + } + + snd_ctl_elem_value_set_id(ctl_data->def_val, + ctl_data->id); + + ctl_data->next = ctl_list; + ctl_list = ctl_data; + } + + next_card: + if (snd_card_next(&card) < 0) { + ksft_print_msg("snd_card_next"); + break; + } + } +} + +/* + * Check that we can read the default value and it is valid. Write + * tests use the read value to restore the default. + */ +void test_ctl_get_value(struct ctl_data *ctl) +{ + int err; + long int_val; + long long int64_val; + + /* If the control is turned off let's be polite */ + if (snd_ctl_elem_info_is_inactive(ctl->info)) { + ksft_print_msg("%s is inactive\n", ctl->name); + ksft_test_result_skip("get_value.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + /* Can't test reading on an unreadable control */ + if (!snd_ctl_elem_info_is_readable(ctl->info)) { + ksft_print_msg("%s is not readable\n", ctl->name); + ksft_test_result_skip("get_value.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + err = snd_ctl_elem_read(ctl->card->handle, ctl->def_val); + if (err < 0) { + ksft_print_msg("snd_ctl_elem_read() failed: %s\n", + snd_strerror(err)); + goto out; + } + + switch (snd_ctl_elem_info_get_type(ctl->info)) { + case SND_CTL_ELEM_TYPE_NONE: + ksft_print_msg("%s Invalid control type NONE\n", ctl->name); + err = -1; + break; + + case SND_CTL_ELEM_TYPE_BOOLEAN: + int_val = snd_ctl_elem_value_get_boolean(ctl->def_val, 0); + switch (int_val) { + case 0: + case 1: + break; + default: + ksft_print_msg("%s Invalid boolean value %ld\n", + ctl->name, int_val); + err = -1; + break; + } + break; + + case SND_CTL_ELEM_TYPE_INTEGER: + int_val = snd_ctl_elem_value_get_integer(ctl->def_val, 0); + + if (int_val < snd_ctl_elem_info_get_min(ctl->info)) { + ksft_print_msg("%s value %ld less than minimum %ld\n", + ctl->name, int_val, + snd_ctl_elem_info_get_min(ctl->info)); + err = -1; + } + + if (int_val > snd_ctl_elem_info_get_max(ctl->info)) { + ksft_print_msg("%s value %ld more than maximum %ld\n", + ctl->name, int_val, + snd_ctl_elem_info_get_max(ctl->info)); + err = -1; + } + + /* Only check step size if there is one and we're in bounds */ + if (err >= 0 && snd_ctl_elem_info_get_step(ctl->info) && + (int_val - snd_ctl_elem_info_get_min(ctl->info) % + snd_ctl_elem_info_get_step(ctl->info))) { + ksft_print_msg("%s value %ld invalid for step %ld minimum %ld\n", + ctl->name, int_val, + snd_ctl_elem_info_get_step(ctl->info), + snd_ctl_elem_info_get_min(ctl->info)); + err = -1; + } + break; + + case SND_CTL_ELEM_TYPE_INTEGER64: + int64_val = snd_ctl_elem_value_get_integer64(ctl->def_val, 0); + + if (int64_val < snd_ctl_elem_info_get_min64(ctl->info)) { + ksft_print_msg("%s value %lld less than minimum %lld\n", + ctl->name, int64_val, + snd_ctl_elem_info_get_min64(ctl->info)); + err = -1; + } + + if (int64_val > snd_ctl_elem_info_get_max64(ctl->info)) { + ksft_print_msg("%s value %lld more than maximum %lld\n", + ctl->name, int64_val, + snd_ctl_elem_info_get_max(ctl->info)); + err = -1; + } + + /* Only check step size if there is one and we're in bounds */ + if (err >= 0 && snd_ctl_elem_info_get_step64(ctl->info) && + (int64_val - snd_ctl_elem_info_get_min64(ctl->info)) % + snd_ctl_elem_info_get_step64(ctl->info)) { + ksft_print_msg("%s value %lld invalid for step %lld minimum %lld\n", + ctl->name, int64_val, + snd_ctl_elem_info_get_step64(ctl->info), + snd_ctl_elem_info_get_min64(ctl->info)); + err = -1; + } + break; + + default: + /* No tests for other types */ + ksft_test_result_skip("get_value.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + +out: + ksft_test_result(err >= 0, "get_value.%d.%d\n", + ctl->card->card, ctl->elem); +} + +bool show_mismatch(struct ctl_data *ctl, int index, + snd_ctl_elem_value_t *read_val, + snd_ctl_elem_value_t *expected_val) +{ + long long expected_int, read_int; + + /* + * We factor out the code to compare values representable as + * integers, ensure that check doesn't log otherwise. + */ + expected_int = 0; + read_int = 0; + + switch (snd_ctl_elem_info_get_type(ctl->info)) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + expected_int = snd_ctl_elem_value_get_boolean(expected_val, + index); + read_int = snd_ctl_elem_value_get_boolean(read_val, index); + break; + + case SND_CTL_ELEM_TYPE_INTEGER: + expected_int = snd_ctl_elem_value_get_integer(expected_val, + index); + read_int = snd_ctl_elem_value_get_integer(read_val, index); + break; + + case SND_CTL_ELEM_TYPE_INTEGER64: + expected_int = snd_ctl_elem_value_get_integer64(expected_val, + index); + read_int = snd_ctl_elem_value_get_integer64(read_val, + index); + break; + + case SND_CTL_ELEM_TYPE_ENUMERATED: + expected_int = snd_ctl_elem_value_get_enumerated(expected_val, + index); + read_int = snd_ctl_elem_value_get_enumerated(read_val, + index); + break; + + default: + break; + } + + if (expected_int != read_int) { + ksft_print_msg("%s.%d expected %lld but read %lld\n", + ctl->name, index, expected_int, read_int); + return true; + } else { + return false; + } +} + +/* + * Write a value then if possible verify that we get the expected + * result. An optional expected value can be provided if we expect + * the write to fail, for verifying that invalid writes don't corrupt + * anything. + */ +int write_and_verify(struct ctl_data *ctl, + snd_ctl_elem_value_t *write_val, + snd_ctl_elem_value_t *expected_val) +{ + int err, i; + bool error_expected, mismatch_shown; + snd_ctl_elem_value_t *read_val, *w_val; + snd_ctl_elem_value_alloca(&read_val); + snd_ctl_elem_value_alloca(&w_val); + + /* + * We need to copy the write value since writing can modify + * the value which causes surprises, and allocate an expected + * value if we expect to read back what we wrote. + */ + snd_ctl_elem_value_copy(w_val, write_val); + if (expected_val) { + error_expected = true; + } else { + error_expected = false; + snd_ctl_elem_value_alloca(&expected_val); + snd_ctl_elem_value_copy(expected_val, write_val); + } + + /* + * Do the write, if we have an expected value ignore the error + * and carry on to validate the expected value. + */ + err = snd_ctl_elem_write(ctl->card->handle, w_val); + if (err < 0 && !error_expected) { + ksft_print_msg("snd_ctl_elem_write() failed: %s\n", + snd_strerror(err)); + return err; + } + + /* Can we do the verification part? */ + if (!snd_ctl_elem_info_is_readable(ctl->info)) + return err; + + snd_ctl_elem_value_set_id(read_val, ctl->id); + + err = snd_ctl_elem_read(ctl->card->handle, read_val); + if (err < 0) { + ksft_print_msg("snd_ctl_elem_read() failed: %s\n", + snd_strerror(err)); + return err; + } + + /* + * Use the libray to compare values, if there's a mismatch + * carry on and try to provide a more useful diagnostic than + * just "mismatch". + */ + if (!snd_ctl_elem_value_compare(expected_val, read_val)) + return 0; + + mismatch_shown = false; + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) + if (show_mismatch(ctl, i, read_val, expected_val)) + mismatch_shown = true; + + if (!mismatch_shown) + ksft_print_msg("%s read and written values differ\n", + ctl->name); + + return -1; +} + +/* + * Make sure we can write the default value back to the control, this + * should validate that at least some write works. + */ +void test_ctl_write_default(struct ctl_data *ctl) +{ + int err; + + /* If the control is turned off let's be polite */ + if (snd_ctl_elem_info_is_inactive(ctl->info)) { + ksft_print_msg("%s is inactive\n", ctl->name); + ksft_test_result_skip("write_default.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + if (!snd_ctl_elem_info_is_writable(ctl->info)) { + ksft_print_msg("%s is not writeable\n", ctl->name); + ksft_test_result_skip("write_default.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + /* No idea what the default was for unreadable controls */ + if (!snd_ctl_elem_info_is_readable(ctl->info)) { + ksft_print_msg("%s couldn't read default\n", ctl->name); + ksft_test_result_skip("write_default.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + err = write_and_verify(ctl, ctl->def_val, NULL); + + ksft_test_result(err >= 0, "write_default.%d.%d\n", + ctl->card->card, ctl->elem); +} + +bool test_ctl_write_valid_boolean(struct ctl_data *ctl) +{ + int err, i, j; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + snd_ctl_elem_value_set_id(val, ctl->id); + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + for (j = 0; j < 2; j++) { + snd_ctl_elem_value_set_boolean(val, i, j); + err = write_and_verify(ctl, val, NULL); + if (err != 0) + fail = true; + } + } + + return !fail; +} + +bool test_ctl_write_valid_integer(struct ctl_data *ctl) +{ + int err; + int i; + long j, step; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + snd_ctl_elem_value_set_id(val, ctl->id); + + step = snd_ctl_elem_info_get_step(ctl->info); + if (!step) + step = 1; + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + for (j = snd_ctl_elem_info_get_min(ctl->info); + j <= snd_ctl_elem_info_get_max(ctl->info); j += step) { + + snd_ctl_elem_value_set_integer(val, i, j); + err = write_and_verify(ctl, val, NULL); + if (err != 0) + fail = true; + } + } + + + return !fail; +} + +bool test_ctl_write_valid_integer64(struct ctl_data *ctl) +{ + int err, i; + long long j, step; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + snd_ctl_elem_value_set_id(val, ctl->id); + + step = snd_ctl_elem_info_get_step64(ctl->info); + if (!step) + step = 1; + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + for (j = snd_ctl_elem_info_get_min64(ctl->info); + j <= snd_ctl_elem_info_get_max64(ctl->info); j += step) { + + snd_ctl_elem_value_set_integer64(val, i, j); + err = write_and_verify(ctl, val, NULL); + if (err != 0) + fail = true; + } + } + + + return !fail; +} + +bool test_ctl_write_valid_enumerated(struct ctl_data *ctl) +{ + int err, i, j; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + snd_ctl_elem_value_set_id(val, ctl->id); + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + for (j = 0; j < snd_ctl_elem_info_get_items(ctl->info); j++) { + snd_ctl_elem_value_set_enumerated(val, i, j); + err = write_and_verify(ctl, val, NULL); + if (err != 0) + fail = true; + } + } + + return !fail; +} + +void test_ctl_write_valid(struct ctl_data *ctl) +{ + bool pass; + int err; + + /* If the control is turned off let's be polite */ + if (snd_ctl_elem_info_is_inactive(ctl->info)) { + ksft_print_msg("%s is inactive\n", ctl->name); + ksft_test_result_skip("write_valid.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + if (!snd_ctl_elem_info_is_writable(ctl->info)) { + ksft_print_msg("%s is not writeable\n", ctl->name); + ksft_test_result_skip("write_valid.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + switch (snd_ctl_elem_info_get_type(ctl->info)) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + pass = test_ctl_write_valid_boolean(ctl); + break; + + case SND_CTL_ELEM_TYPE_INTEGER: + pass = test_ctl_write_valid_integer(ctl); + break; + + case SND_CTL_ELEM_TYPE_INTEGER64: + pass = test_ctl_write_valid_integer64(ctl); + break; + + case SND_CTL_ELEM_TYPE_ENUMERATED: + pass = test_ctl_write_valid_enumerated(ctl); + break; + + default: + /* No tests for this yet */ + ksft_test_result_skip("write_valid.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + /* Restore the default value to minimise disruption */ + err = write_and_verify(ctl, ctl->def_val, NULL); + if (err < 0) + pass = false; + + ksft_test_result(pass, "write_valid.%d.%d\n", + ctl->card->card, ctl->elem); +} + +int main(void) +{ + struct ctl_data *ctl; + + ksft_print_header(); + + find_controls(); + + ksft_set_plan(num_controls * TESTS_PER_CONTROL); + + for (ctl = ctl_list; ctl != NULL; ctl = ctl->next) { + /* + * Must test get_value() before we write anything, the + * test stores the default value for later cleanup. + */ + test_ctl_get_value(ctl); + test_ctl_write_default(ctl); + test_ctl_write_valid(ctl); + } + + ksft_exit_pass(); + + return 0; +} -- 2.30.2

2 2

[PATCH v9 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 06 Dec '21

06 Dec '21

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V8: -- Fix errors in sc7280 lpass cpu dt-bindings. -- Move to quicinc domain email id's. Changes Since V7: -- Fix indentation errors. -- Bisect patches to avoid interdependency. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: SC7280: Update config for building codec dma drivers ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add regmap config support for codec dma driver ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 75 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 275 +++++++++ sound/soc/qcom/lpass-cpu.c | 244 +++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 617 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 441 +++++++++++++++ sound/soc/qcom/lpass.h | 162 ++++++ 9 files changed, 1869 insertions(+), 87 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- 2.7.4

1 10

[PATCH] sound/core: remove useless NULL check before kfree
by Bernard Zhao 06 Dec '21

06 Dec '21

Tis patch try to remove useless NULL check before kfree Signed-off-by: Bernard Zhao <bernard(a)vivo.com> --- sound/core/info_oss.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/core/info_oss.c b/sound/core/info_oss.c index 1ba887c7954e..ebc714b2f46b 100644 --- a/sound/core/info_oss.c +++ b/sound/core/info_oss.c @@ -32,10 +32,8 @@ int snd_oss_info_register(int dev, int num, char *string) mutex_lock(&strings); if (string == NULL) { x = snd_sndstat_strings[num][dev]; - if (x) { - kfree(x); - x = NULL; - } + kfree(x); + x = NULL; } else { x = kstrdup(string, GFP_KERNEL); if (x == NULL) { -- 2.33.1

2 1

[PATCH v2] ALSA: usb-audio: Reorder snd_djm_devices[] entries
by Geraldo Nascimento 06 Dec '21

06 Dec '21

Olivia Mackintosh has posted to alsa-devel reporting that there's a potential bug that could break mixer quirks for Pioneer devices introduced by 6d27788160362a7ee6c0d317636fe4b1ddbe59a7 "ALSA: usb-audio: Add support for the Pioneer DJM 750MK2 Mixer/Soundcard". This happened because the DJM 750 MK2 was added last to the Pioneer DJM device table index and defined as 0x4 but was added to snd_djm_devices[] just after the DJM 750 (MK1) entry instead of last, after the DJM 900 NXS2. This escaped review. To prevent that from ever happening again, Takashi Iwai suggested to use C99 array designators in snd_djm_devices[] instead of simply reordering the entries. Reported-by: Olivia Mackintosh <livvy(a)base.nu> Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- V1 -> V2: added C99 array designators to prevent the order of entries ever being an issue again. --- Hi Takashi, First, thanks for the suggestion regarding C99 array designators. This will serve us well. As I said before, I don't own any DJM mixers to test this, and although it's a simple change, I'm always for testing. Thanks, Geraldo Nascimento --- diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index d489c1de3bae..823b6b8de942 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3016,11 +3016,11 @@ static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = { static const struct snd_djm_device snd_djm_devices[] = { - SND_DJM_DEVICE(250mk2), - SND_DJM_DEVICE(750), - SND_DJM_DEVICE(750mk2), - SND_DJM_DEVICE(850), - SND_DJM_DEVICE(900nxs2) + [SND_DJM_250MK2_IDX] = SND_DJM_DEVICE(250mk2), + [SND_DJM_750_IDX] = SND_DJM_DEVICE(750), + [SND_DJM_850_IDX] = SND_DJM_DEVICE(850), + [SND_DJM_900NXS2_IDX] = SND_DJM_DEVICE(900nxs2), + [SND_DJM_750MK2_IDX] = SND_DJM_DEVICE(750mk2), };

2 1

[PATCH] sound: Fix some typo
by Christophe JAILLET 06 Dec '21

06 Dec '21

Some comments and include guards are not consistent with the name of the file where they can be found. This is likely some typo or cut'n'paste issues. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/pci/ac97/ac97_pcm.c | 2 +- sound/pci/hda/hda_generic.h | 2 +- sound/soc/codecs/sta350.h | 2 +- sound/soc/codecs/tlv320aic26.h | 6 +++--- sound/usb/usx2y/usbusx2y.c | 2 +- 5 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c index 491de1a623cb..5fee8e89790f 100644 --- a/sound/pci/ac97/ac97_pcm.c +++ b/sound/pci/ac97/ac97_pcm.c @@ -231,7 +231,7 @@ static int set_spdif_rate(struct snd_ac97 *ac97, unsigned short rate) * If the codec doesn't support VAR, the rate must be 48000 (except * for SPDIF). * - * The valid registers are AC97_PMC_MIC_ADC_RATE, + * The valid registers are AC97_PCM_MIC_ADC_RATE, * AC97_PCM_FRONT_DAC_RATE, AC97_PCM_LR_ADC_RATE. * AC97_PCM_SURR_DAC_RATE and AC97_PCM_LFE_DAC_RATE are accepted * if the codec supports them. diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index c43bd0f0338e..8e1bc8ea74fc 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -183,7 +183,7 @@ struct hda_gen_spec { struct automic_entry am_entry[MAX_AUTO_MIC_PINS]; /* for pin sensing */ - /* current status; set in hda_geneic.c */ + /* current status; set in hda_generic.c */ unsigned int hp_jack_present:1; unsigned int line_jack_present:1; unsigned int speaker_muted:1; /* current status of speaker mute */ diff --git a/sound/soc/codecs/sta350.h b/sound/soc/codecs/sta350.h index f16900e00afa..80bf56093d94 100644 --- a/sound/soc/codecs/sta350.h +++ b/sound/soc/codecs/sta350.h @@ -14,7 +14,7 @@ #ifndef _ASOC_STA_350_H #define _ASOC_STA_350_H -/* STA50 register addresses */ +/* STA350 register addresses */ #define STA350_REGISTER_COUNT 0x4D #define STA350_COEF_COUNT 62 diff --git a/sound/soc/codecs/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h index 1f2879b7a080..c86569883e0c 100644 --- a/sound/soc/codecs/tlv320aic26.h +++ b/sound/soc/codecs/tlv320aic26.h @@ -6,8 +6,8 @@ * Copyright (C) 2008 Secret Lab Technologies Ltd. */ -#ifndef _TLV320AIC16_H_ -#define _TLV320AIC16_H_ +#ifndef _TLV320AIC26_H_ +#define _TLV320AIC26_H_ /* AIC26 Registers */ #define AIC26_PAGE_ADDR(page, offset) ((page << 11) | offset << 5) @@ -88,4 +88,4 @@ enum aic26_wlen { AIC26_WLEN_32 = 3 << 10, }; -#endif /* _TLV320AIC16_H_ */ +#endif /* _TLV320AIC26_H_ */ diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 099bee662af6..52f4e6652407 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-or-later /* - * usbusy2y.c - ALSA USB US-428 Driver + * usbusx2y.c - ALSA USB US-428 Driver * 2005-04-14 Karsten Wiese Version 0.8.7.2: -- 2.30.2

2 1

[PATCH] ASoC: SOF: Intel: move CODEC_PROBE_RETRIES definition
by Arnd Bergmann 06 Dec '21

06 Dec '21

From: Arnd Bergmann <arnd(a)arndb.de> The macro is defined in an #ifdef but used outside: sound/soc/sof/intel/hda-codec.c: In function 'hda_codec_probe': sound/soc/sof/intel/hda-codec.c:132:42: error: 'CODEC_PROBE_RETRIES' undeclared (first use in this function) 132 | } while (resp == -1 && retry++ < CODEC_PROBE_RETRIES); | ^~~~~~~~~~~~~~~~~~~ Move it to a place where it can be seen unconditionally. Fixes: 046aede2f847 ("ASoC: SOF: Intel: Retry codec probing if it fails") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/sof/intel/hda-codec.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 13cd96e6724a..4324178b3ca6 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -20,9 +20,10 @@ #include "../../codecs/hdac_hda.h" #endif /* CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC */ +#define CODEC_PROBE_RETRIES 3 + #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) #define IDISP_VID_INTEL 0x80860000 -#define CODEC_PROBE_RETRIES 3 /* load the legacy HDA codec driver */ static int request_codec_module(struct hda_codec *codec) -- 2.29.2

2 1

[PATCH] ALSA: usb-audio: Reorder snd_djm_devices entries
by Geraldo Nascimento 04 Dec '21

04 Dec '21

Olivia Mackintosh has posted to alsa-devel reporting that there's a potential bug that could break mixer quirks for Pioneer devices introduced by 6d27788160362a7ee6c0d317636fe4b1ddbe59a7 "ALSA: usb-audio: Add support for the Pioneer DJM 750MK2 Mixer/Soundcard". This happened because the DJM 750 MK2 was added last to the Pioneer DJM device table index and defined as 0x4 but was added to snd_djm_devices[] just after the DJM 750 (MK1) entry instead of last, after the DJM 900 NXS2. This escaped review. Let's reorder the entries in snd_djm_devices[] so they match the Pioneer DJM device table index #define's. Reported-by: Olivia Mackintosh <livvy(a)base.nu> Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index d489c1de3bae..d4cd880b5c6c 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3018,9 +3018,9 @@ static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = { static const struct snd_djm_device snd_djm_devices[] = { SND_DJM_DEVICE(250mk2), SND_DJM_DEVICE(750), - SND_DJM_DEVICE(750mk2), SND_DJM_DEVICE(850), - SND_DJM_DEVICE(900nxs2) + SND_DJM_DEVICE(900nxs2), + SND_DJM_DEVICE(750mk2) };

2 2

[PATCH v8 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 04 Dec '21

04 Dec '21

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V7: -- Fix indentation errors. -- Bisect patches to avoid interdependency. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: SC7280: Update config for building codec dma drivers ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add regmap config support for codec dma driver ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 70 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 275 +++++++++ sound/soc/qcom/lpass-cpu.c | 244 +++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 617 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 441 +++++++++++++++ sound/soc/qcom/lpass.h | 162 ++++++ 9 files changed, 1864 insertions(+), 87 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

5 15

[PATCH v8 0/3] Machine driver to support LPASS SC7280 sound card registration
by Srinivasa Rao Mandadapu 04 Dec '21

04 Dec '21

This patch set is to add support for SC7280 sound card registration and to add dt-bindings documentation file. Srinivasa Rao Mandadapu (3): ASoC: google: dt-bindings: Add sc7280-herobrine machine bindings ASoC: qcom: Add macro for lpass DAI id's max limit ASoC: qcom: SC7280: Add machine driver Changes Since V7: -- Create separate patch for dai id's macro. -- Remove unused dapm widget structure. Changes Since V6: -- Remove redundant headers. -- Move max ports macro to lpass.h header. -- Arrange structure alignment. -- Fix indentation errors. -- Update module license. Changes Since V5: -- Add required properties to dt-bindings Changes Since V4: -- Add COMPILE_TEST flag in sc7280 configuration. -- Remove redundant startup and shutdown callbacks of snd_soc_ops. -- Fix typo errors. Changes Since V3: -- Change audio jack playpause key value. Changes Since V2: -- updated required field in bindings -- updated Return type check with proper enum in sc7280.c -- Updated Header name and Typos in sc7280.c Changes Since V1: -- Indentation changes and typo. .../bindings/sound/google,sc7280-herobrine.yaml | 171 +++++++++++++ sound/soc/qcom/Kconfig | 14 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass.h | 1 + sound/soc/qcom/sc7280.c | 284 +++++++++++++++++++++ 5 files changed, 472 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml create mode 100644 sound/soc/qcom/sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 4

[bug report] ASoC: Intel: mrfld - create separate module for pci part
by Dan Carpenter 04 Dec '21

04 Dec '21

Hello Vinod Koul, The patch f533a035e4da: "ASoC: Intel: mrfld - create separate module for pci part" from Nov 4, 2014, leads to the following Smatch static checker warning: sound/soc/intel/atom/sst/sst_pci.c:102 sst_platform_get_resources() warn: resource freed on success: 'ctx->pci' sound/soc/intel/atom/sst/sst_pci.c 25 static int sst_platform_get_resources(struct intel_sst_drv *ctx) 26 { 27 int ddr_base, ret = 0; 28 struct pci_dev *pci = ctx->pci; 29 30 ret = pci_request_regions(pci, SST_DRV_NAME); 31 if (ret) 32 return ret; 33 34 /* map registers */ 35 /* DDR base */ 36 if (ctx->dev_id == SST_MRFLD_PCI_ID) { 37 ctx->ddr_base = pci_resource_start(pci, 0); 38 /* check that the relocated IMR base matches with FW Binary */ 39 ddr_base = relocate_imr_addr_mrfld(ctx->ddr_base); 40 if (!ctx->pdata->lib_info) { 41 dev_err(ctx->dev, "lib_info pointer NULL\n"); 42 ret = -EINVAL; 43 goto do_release_regions; 44 } 45 if (ddr_base != ctx->pdata->lib_info->mod_base) { 46 dev_err(ctx->dev, 47 "FW LSP DDR BASE does not match with IFWI\n"); 48 ret = -EINVAL; 49 goto do_release_regions; 50 } 51 ctx->ddr_end = pci_resource_end(pci, 0); 52 53 ctx->ddr = pcim_iomap(pci, 0, 54 pci_resource_len(pci, 0)); 55 if (!ctx->ddr) { 56 ret = -EINVAL; 57 goto do_release_regions; 58 } 59 dev_dbg(ctx->dev, "sst: DDR Ptr %p\n", ctx->ddr); 60 } else { 61 ctx->ddr = NULL; 62 } 63 /* SHIM */ 64 ctx->shim_phy_add = pci_resource_start(pci, 1); 65 ctx->shim = pcim_iomap(pci, 1, pci_resource_len(pci, 1)); 66 if (!ctx->shim) { 67 ret = -EINVAL; 68 goto do_release_regions; 69 } 70 dev_dbg(ctx->dev, "SST Shim Ptr %p\n", ctx->shim); 71 72 /* Shared SRAM */ 73 ctx->mailbox_add = pci_resource_start(pci, 2); 74 ctx->mailbox = pcim_iomap(pci, 2, pci_resource_len(pci, 2)); 75 if (!ctx->mailbox) { 76 ret = -EINVAL; 77 goto do_release_regions; 78 } 79 dev_dbg(ctx->dev, "SRAM Ptr %p\n", ctx->mailbox); 80 81 /* IRAM */ 82 ctx->iram_end = pci_resource_end(pci, 3); 83 ctx->iram_base = pci_resource_start(pci, 3); 84 ctx->iram = pcim_iomap(pci, 3, pci_resource_len(pci, 3)); 85 if (!ctx->iram) { 86 ret = -EINVAL; 87 goto do_release_regions; 88 } 89 dev_dbg(ctx->dev, "IRAM Ptr %p\n", ctx->iram); 90 91 /* DRAM */ 92 ctx->dram_end = pci_resource_end(pci, 4); 93 ctx->dram_base = pci_resource_start(pci, 4); 94 ctx->dram = pcim_iomap(pci, 4, pci_resource_len(pci, 4)); 95 if (!ctx->dram) { 96 ret = -EINVAL; 97 goto do_release_regions; 98 } 99 dev_dbg(ctx->dev, "DRAM Ptr %p\n", ctx->dram); 100 do_release_regions: 101 pci_release_regions(pci); --> 102 return ret; 103 } Surely there should be a "return 0;" before the do_release_regions: label? How does this code work? regards, dan carpenter

6 8

[RFC patch 0/5] Support BCLK input clock in tlv320aic31xx
by Ariel D'Alessandro 03 Dec '21

03 Dec '21

The tlv320aic31xx codec allows using BCLK as the input clock for PLL, deriving all the frequencies through a set of divisors. In this case, codec sysclk is determined by the hwparams sample rate/format. So its frequency must be updated from the codec itself when these are changed. This patchset modifies the tlv320aic31xx driver to update its sysclk if BCLK is used as the input clock. This allows to be used by the generic fsl-asoc-card, without having to add a specific driver. Ariel D'Alessandro (5): ASoC: tlv320aic31xx: Fix typo in BCLK clock name ASoC: tlv320aic31xx: Add support for pll_r coefficient ASoC: tlv320aic31xx: Add divs for bclk as clk_in ASoC: tlv320aic31xx: Handle BCLK set as PLL input configuration ASoC: fsl-asoc-card: Support fsl,imx-audio-tlv320aic31xx codec sound/soc/codecs/tlv320aic31xx.c | 105 ++++++++++++++++++++----------- sound/soc/codecs/tlv320aic31xx.h | 2 +- sound/soc/fsl/fsl-asoc-card.c | 12 ++++ 3 files changed, 83 insertions(+), 36 deletions(-) -- 2.30.2

3 13

[PATCH v7 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 03 Dec '21

03 Dec '21

From: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add regmap config support for codec dma driver ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio ASoC: qcom: SC7280: Update config for building codec dma drivers .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 70 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 275 +++++++++ sound/soc/qcom/lpass-cpu.c | 244 +++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 617 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 441 +++++++++++++++ sound/soc/qcom/lpass.h | 162 ++++++ 9 files changed, 1864 insertions(+), 87 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 18

[PATCH v7 0/2] Machine driver to support LPASS SC7280 sound card registration
by Srinivasa Rao Mandadapu 03 Dec '21

03 Dec '21

This patch set is to add support for SC7280 sound card registration and to add dt-bindings documentation file. Srinivasa Rao Mandadapu (2): ASoC: google: dt-bindings: Add sc7280-herobrine machine bindings ASoC: qcom: SC7280: Add machine driver Changes Since V6: -- Remove redundant headers. -- Move max ports macro to lpass.h header. -- Arrange structure alignment. -- Fix indentation errors. -- Update module license. Changes Since V5: -- Add required properties to dt-bindings Changes Since V4: -- Add COMPILE_TEST flag in sc7280 configuration. -- Remove redundant startup and shutdown callbacks of snd_soc_ops. -- Fix typo errors. Changes Since V3: -- Change audio jack playpause key value. Changes Since V2: -- updated required field in bindings -- updated Return type check with proper enum in sc7280.c -- Updated Header name and Typos in sc7280.c Changes Since V1: -- Indentation changes and typo. .../bindings/sound/google,sc7280-herobrine.yaml | 171 ++++++++++++ sound/soc/qcom/Kconfig | 14 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass.h | 1 + sound/soc/qcom/sc7280.c | 290 +++++++++++++++++++++ 5 files changed, 478 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml create mode 100644 sound/soc/qcom/sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 3

Re: [linux-next:master 3956/5128] sound/soc/sof/intel/hda-codec.c:132:35: error: use of undeclared identifier 'CODEC_PROBE_RETRIES'
by Pierre-Louis Bossart 03 Dec '21

03 Dec '21

[adding alsa-devel] On 12/3/21 5:25 AM, kernel test robot wrote: > tree: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master > head: 7afeac307a9561e3a93682c1e7eb22f918aa1187 > commit: 046aede2f847676f93a2ea4f48b77909c51dba40 [3956/5128] ASoC: SOF: Intel: Retry codec probing if it fails > config: i386-randconfig-r033-20211202 (https://download.01.org/0day-ci/archive/20211203/202112031943.Twg19fWT-lkp@…) > compiler: clang version 14.0.0 (https://github.com/llvm/llvm-project 4b553297ef3ee4dc2119d5429adf3072e90fac38) > reproduce (this is a W=1 build): > wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross > chmod +x ~/bin/make.cross > # https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git/commit/… > git remote add linux-next https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git > git fetch --no-tags linux-next master > git checkout 046aede2f847676f93a2ea4f48b77909c51dba40 > # save the config file to linux build tree > mkdir build_dir > COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang make.cross W=1 O=build_dir ARCH=i386 SHELL=/bin/bash > > If you fix the issue, kindly add following tag as appropriate > Reported-by: kernel test robot <lkp(a)intel.com> > > All errors (new ones prefixed by >>): > >>> sound/soc/sof/intel/hda-codec.c:132:35: error: use of undeclared identifier 'CODEC_PROBE_RETRIES' > } while (resp == -1 && retry++ < CODEC_PROBE_RETRIES); > ^ > 1 error generated. > > > vim +/CODEC_PROBE_RETRIES +132 sound/soc/sof/intel/hda-codec.c > > 110 > 111 /* probe individual codec */ > 112 static int hda_codec_probe(struct snd_sof_dev *sdev, int address, > 113 bool hda_codec_use_common_hdmi) > 114 { > 115 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) > 116 struct hdac_hda_priv *hda_priv; > 117 struct hda_codec *codec; > 118 int type = HDA_DEV_LEGACY; > 119 #endif > 120 struct hda_bus *hbus = sof_to_hbus(sdev); > 121 struct hdac_device *hdev; > 122 u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) | > 123 (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; > 124 u32 resp = -1; > 125 int ret, retry = 0; > 126 > 127 do { > 128 mutex_lock(&hbus->core.cmd_mutex); > 129 snd_hdac_bus_send_cmd(&hbus->core, hda_cmd); > 130 snd_hdac_bus_get_response(&hbus->core, address, &resp); > 131 mutex_unlock(&hbus->core.cmd_mutex); > > 132 } while (resp == -1 && retry++ < CODEC_PROBE_RETRIES); Indeed, something's not right here. CODEC_PROBE_RETRIES is defined conditionally #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) #define IDISP_VID_INTEL 0x80860000 #define CODEC_PROBE_RETRIES 3 but it's used unconditionally. We could define this constant unconditionally as a quick fix, but this compilation problem might expose a larger problem. Kai, I wonder if this is code from lines 120 to 139 that we didn't update when we moved to support HDMI with the generic HDaudio parts? I don't see why we could even try to send a command on the bus is there's no audio codec support? hda_codec_use_common_hdmi should be the default assumption now, I don't think we support the old solution, do we? > 133 > 134 if (resp == -1) > 135 return -EIO; > 136 dev_dbg(sdev->dev, "HDA codec #%d probed OK: response: %x\n", > 137 address, resp); > 138 > > --- > 0-DAY CI Kernel Test Service, Intel Corporation > https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org >

3 3

[PATCH v2] ALSA: hda/realtek: Fix quirk for TongFang PHxTxX1
by Werner Sembach 03 Dec '21

03 Dec '21

This fixes the SND_PCI_QUIRK(...) of the TongFang PHxTxX1 barebone. This fixes the issue of sound not working after s3 suspend. When waking up from s3 suspend the Coef 0x10 is set to 0x0220 instead of 0x0020. Setting the value manually makes the sound work again. This patch does this automatically. While being on it, I also fixed the comment formatting of the quirk and shortened variable and function names. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Fixes: dd6dd6e3c791 ("ALSA: hda/realtek: Add quirk for TongFang PHxTxX1") Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/patch_realtek.c | 40 +++++++++++++++++++---------------- 1 file changed, 22 insertions(+), 18 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9ce7457533c96..d361a1260d5a1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6503,22 +6503,26 @@ static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, /* for alc285_fixup_ideapad_s740_coef() */ #include "ideapad_s740_helper.c" -static void alc256_fixup_tongfang_reset_persistent_settings(struct hda_codec *codec, - const struct hda_fixup *fix, - int action) +static const struct coef_fw alc256_fixup_set_coef_defaults_coefs[] = { + WRITE_COEF(0x10, 0x0020), WRITE_COEF(0x24, 0x0000), + WRITE_COEF(0x26, 0x0000), WRITE_COEF(0x29, 0x3000), + WRITE_COEF(0x37, 0xfe05), WRITE_COEF(0x45, 0x5089), + {} +}; + +static void alc256_fixup_set_coef_defaults(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) { /* - * A certain other OS sets these coeffs to different values. On at least one TongFang - * barebone these settings might survive even a cold reboot. So to restore a clean slate the - * values are explicitly reset to default here. Without this, the external microphone is - * always in a plugged-in state, while the internal microphone is always in an unplugged - * state, breaking the ability to use the internal microphone. - */ - alc_write_coef_idx(codec, 0x24, 0x0000); - alc_write_coef_idx(codec, 0x26, 0x0000); - alc_write_coef_idx(codec, 0x29, 0x3000); - alc_write_coef_idx(codec, 0x37, 0xfe05); - alc_write_coef_idx(codec, 0x45, 0x5089); + * A certain other OS sets these coeffs to different values. On at least + * one TongFang barebone these settings might survive even a cold + * reboot. So to restore a clean slate the values are explicitly reset + * to default here. Without this, the external microphone is always in a + * plugged-in state, while the internal microphone is always in an + * unplugged state, breaking the ability to use the internal microphone. + */ + alc_process_coef_fw(codec, alc256_fixup_set_coef_defaults_coefs); } static const struct coef_fw alc233_fixup_no_audio_jack_coefs[] = { @@ -6759,7 +6763,7 @@ enum { ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, ALC287_FIXUP_13S_GEN2_SPEAKERS, - ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS, + ALC256_FIXUP_SET_COEF_DEFAULTS, ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, ALC233_FIXUP_NO_AUDIO_JACK, }; @@ -8465,9 +8469,9 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE, }, - [ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS] = { + [ALC256_FIXUP_SET_COEF_DEFAULTS] = { .type = HDA_FIXUP_FUNC, - .v.func = alc256_fixup_tongfang_reset_persistent_settings, + .v.func = alc256_fixup_set_coef_defaults, }, [ALC245_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, @@ -8929,7 +8933,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), - SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS), + SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_SET_COEF_DEFAULTS), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), -- 2.25.1

2 1

[PATCH v6 0/2] Machine driver to support LPASS SC7280 sound card registration
by Srinivasa Rao Mandadapu 03 Dec '21

03 Dec '21

This patch set is to add support for SC7280 sound card registration and to add dt-bindings documentation file. Srinivasa Rao Mandadapu (2): ASoC: google: dt-bindings: Add sc7280-herobrine machine bindings ASoC: qcom: SC7280: Add machine driver Changes Since V5: -- Add required properties to dt-bindings Changes Since V4: -- Add COMPILE_TEST flag in sc7280 configuration. -- Remove redundant startup and shutdown callbacks of snd_soc_ops. -- Fix typo errors. Changes Since V3: -- Change audio jack playpause key value. Changes Since V2: -- updated required field in bindings -- updated Return type check with proper enum in sc7280.c -- Updated Header name and Typos in sc7280.c Changes Since V1: -- Indentation changes and typo. .../bindings/sound/google,sc7280-herobrine.yaml | 171 ++++++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7280.c | 299 +++++++++++++++++++++ 4 files changed, 484 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml create mode 100644 sound/soc/qcom/sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 5

Practical questions about mainlining a downstream codec driver
by Vincent Knecht 03 Dec '21

03 Dec '21

Hi ! I'm looking at submitting a patch to add support for (i2c) ak4375 codec (a headphones amplifier). It works in close-to-mainline kernel forks used in postmarketOS for msm8916-alcatel-idol347 and msm8939-alcatel-idol3 phones (the latter is not in mainline yet). Currently the driver is mostly a cleanup and update of the downstream one to use modern framework (eg. regmap, snd_soc_component_* functions, etc.) The question is : how much should it be trimmed down ? For example : IC supports PLL and MCLK modes, and both testing devices use PLL mode. - should the driver only support the mode that I could test ? - should the UCM control(s) related to untested mode/features also be trimmed down ? or can they be kept, with an error/warning that selected value is unsupported ? Another question : While both devices are quite close, downstream drivers have little differences. For example, the idol347 one has an additional initialization register sequence. What are the options ? - allow setting this optional sequence from devicetree ? with register/values in driver and a boolean devicetree property to enable it ? or with an optional devicetree array property - put it in driver, and associate it with a specific additional compatible ? - ... ? Bonus question : Sadly, there's no detailed datasheet for ak4375 at https://www.akm.com/eu/en/products/audio/audio-dac/ and only a brief one is floating on the Internets, eg. https://datasheetspdf.com/pdf-file/1400317/AKM/AK4375A/1 I sent multiple inquiries through akm.com, even mailed directly to persons with akm.com addresses I could find in alsa mailing list so as to obtain a complete datasheet, with no success. Can anyone help with this ? Work on this driver is there : https://github.com/msm8916-mainline/linux/pull/203 Thank you

1 0

init procedure is executed during a store and restore process when the final snd_ctl_elem is not readable
by GitHub issues - edited 03 Dec '21

03 Dec '21

alsa-project/alsa-utils issue #122 was edited from yixingwang0517: When i execute the following commands sequencitally, alsactl store -f 1.txt alsactl restore -f 1.txt my playing volume was initialized. After a debugging process, I found the problem: During the "store" process, a control element is saved to the configuration file only when this element is readable.While during the "restore" process, after all the control element from the configuration file is processed, an "additional control element check" which causes a "init procedure" if the result is true is processed. ![image](https://user-images.githubusercontent.com/19835921/144555692-6b3cee21-73fc-4ade-a1f4-cc8f99679a9d.png) Thus, if total control element count is 5 with the 5th element is not readable, then there will be only 4 control element stored in the configuration file, ranges from "control.1" to "control.4". Then the "additional control checking" process produces "true" and the variable "err" is evaluated as "-EAGAIN". Issue URL : https://github.com/alsa-project/alsa-utils/issues/122 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

init procedure is executed during a store and restore process when the final snd_ctl_elem is not readable
by GitHub issues - opened 03 Dec '21

03 Dec '21

alsa-project/alsa-utils issue #122 was opened from yixingwang0517: When i execute the following commands sequencitally, alsa store -f 1.txt alsa restore -f 1.txt my playing volume was initialized. After a debugging process, I found the problem: During the "store" process, a control element is saved to the configuration file only when this element is readable.While during the "restore" process, after all the control element from the configuration file is processed, an "additional control element check" which causes a "init procedure" if the result is true is processed. ![image](https://user-images.githubusercontent.com/19835921/144555692-6b3cee21-73fc-4ade-a1f4-cc8f99679a9d.png) Thus, if total control element count is 5 with the 5th element is not readable, then there will be only 4 control element stored in the configuration file, ranges from "control.1" to "control.4". Then the "additional control checking" process produces "true" and the variable "err" is evaluated as "-EAGAIN". Issue URL : https://github.com/alsa-project/alsa-utils/issues/122 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] Sound: Add support for the Pioneer DJM 750MK2 Mixer/Soundcard
by William Overton 02 Dec '21

02 Dec '21

The kernel already has support for very similar Pioneer djm products and this work is based on that. Added device to quirks-table.h and added control info to mixer_quirks.c. Tested on my hardware and all working. Signed-off-by: William Overton <willovertonuk(a)gmail.com> --- sound/usb/mixer_quirks.c | 34 +++++++++++++++++++++++ sound/usb/quirks-table.h | 58 ++++++++++++++++++++++++++++++++++++++++ 2 files changed, 92 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 46082dc57be0..d489c1de3bae 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2795,6 +2795,7 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_750_IDX 0x1 #define SND_DJM_850_IDX 0x2 #define SND_DJM_900NXS2_IDX 0x3 +#define SND_DJM_750MK2_IDX 0x4 #define SND_DJM_CTL(_name, suffix, _default_value, _windex) { \ @@ -2984,10 +2985,40 @@ static const struct snd_djm_ctl snd_djm_ctls_900nxs2[] = { SND_DJM_CTL("Ch5 Input", 900nxs2_cap5, 3, SND_DJM_WINDEX_CAP) }; +// DJM-750MK2 +static const u16 snd_djm_opts_750mk2_cap1[] = { + 0x0100, 0x0102, 0x0103, 0x0106, 0x0107, 0x0108, 0x0109, 0x010a }; +static const u16 snd_djm_opts_750mk2_cap2[] = { + 0x0200, 0x0202, 0x0203, 0x0206, 0x0207, 0x0208, 0x0209, 0x020a }; +static const u16 snd_djm_opts_750mk2_cap3[] = { + 0x0300, 0x0302, 0x0303, 0x0306, 0x0307, 0x0308, 0x0309, 0x030a }; +static const u16 snd_djm_opts_750mk2_cap4[] = { + 0x0400, 0x0402, 0x0403, 0x0406, 0x0407, 0x0408, 0x0409, 0x040a }; +static const u16 snd_djm_opts_750mk2_cap5[] = { + 0x0507, 0x0508, 0x0509, 0x050a, 0x0511, 0x0512, 0x0513, 0x0514 }; + +static const u16 snd_djm_opts_750mk2_pb1[] = { 0x0100, 0x0101, 0x0104 }; +static const u16 snd_djm_opts_750mk2_pb2[] = { 0x0200, 0x0201, 0x0204 }; +static const u16 snd_djm_opts_750mk2_pb3[] = { 0x0300, 0x0301, 0x0304 }; + + +static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = { + SND_DJM_CTL("Capture Level", cap_level, 0, SND_DJM_WINDEX_CAPLVL), + SND_DJM_CTL("Ch1 Input", 750mk2_cap1, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch2 Input", 750mk2_cap2, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch3 Input", 750mk2_cap3, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch4 Input", 750mk2_cap4, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch5 Input", 750mk2_cap5, 3, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch1 Output", 750mk2_pb1, 0, SND_DJM_WINDEX_PB), + SND_DJM_CTL("Ch2 Output", 750mk2_pb2, 1, SND_DJM_WINDEX_PB), + SND_DJM_CTL("Ch3 Output", 750mk2_pb3, 2, SND_DJM_WINDEX_PB) +}; + static const struct snd_djm_device snd_djm_devices[] = { SND_DJM_DEVICE(250mk2), SND_DJM_DEVICE(750), + SND_DJM_DEVICE(750mk2), SND_DJM_DEVICE(850), SND_DJM_DEVICE(900nxs2) }; @@ -3235,6 +3266,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x08e4, 0x017f): /* Pioneer DJ DJM-750 */ err = snd_djm_controls_create(mixer, SND_DJM_750_IDX); break; + case USB_ID(0x2b73, 0x001b): /* Pioneer DJ DJM-750MK2 */ + err = snd_djm_controls_create(mixer, SND_DJM_750MK2_IDX); + break; case USB_ID(0x08e4, 0x0163): /* Pioneer DJ DJM-850 */ err = snd_djm_controls_create(mixer, SND_DJM_850_IDX); break; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e03043f7dad3..bc0116273e94 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3850,6 +3850,64 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Pioneer DJ DJM-750MK2 + * 10 channels playback & 12 channels capture @ 48kHz S24LE + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x001b), + .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 = 10, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .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_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = -1 + } + } + } +}, { /* * Pioneer DJ DJM-850 -- 2.30.2

4 3

[PATCH] ALSA: hda/realtek: Fix quirk for TongFang PHxTxX1
by Werner Sembach 02 Dec '21

02 Dec '21

This fixes the SND_PCI_QUIRK(...) of the TongFang PHxTxX1 barebone. This fixes the issue of sound not working after s3 suspend. When waking up from s3 suspend the Coef 0x10 is set to 0x0220 instead of 0x0020. Setting the value manually makes the sound work again. This patch does this automatically. While being on it, I also fixed the comment formatting of the quirk. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/patch_realtek.c | 25 ++++++++++++++----------- 1 file changed, 14 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9ce7457533c96..d3a16843c7afd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6503,22 +6503,25 @@ static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, /* for alc285_fixup_ideapad_s740_coef() */ #include "ideapad_s740_helper.c" +static const struct coef_fw alc256_fixup_tongfang_reset_persistent_settings_coefs[] = { + WRITE_COEF(0x10, 0x0020), WRITE_COEF(0x24, 0x0000), WRITE_COEF(0x26, 0x0000), + WRITE_COEF(0x29, 0x3000), WRITE_COEF(0x37, 0xfe05), WRITE_COEF(0x45, 0x5089), + {} +}; + static void alc256_fixup_tongfang_reset_persistent_settings(struct hda_codec *codec, const struct hda_fixup *fix, int action) { /* - * A certain other OS sets these coeffs to different values. On at least one TongFang - * barebone these settings might survive even a cold reboot. So to restore a clean slate the - * values are explicitly reset to default here. Without this, the external microphone is - * always in a plugged-in state, while the internal microphone is always in an unplugged - * state, breaking the ability to use the internal microphone. - */ - alc_write_coef_idx(codec, 0x24, 0x0000); - alc_write_coef_idx(codec, 0x26, 0x0000); - alc_write_coef_idx(codec, 0x29, 0x3000); - alc_write_coef_idx(codec, 0x37, 0xfe05); - alc_write_coef_idx(codec, 0x45, 0x5089); + * A certain other OS sets these coeffs to different values. On at least + * one TongFang barebone these settings might survive even a cold + * reboot. So to restore a clean slate the values are explicitly reset + * to default here. Without this, the external microphone is always in a + * plugged-in state, while the internal microphone is always in an + * unplugged state, breaking the ability to use the internal microphone. + */ + alc_process_coef_fw(codec, alc256_fixup_tongfang_reset_persistent_settings_coefs); } static const struct coef_fw alc233_fixup_no_audio_jack_coefs[] = { -- 2.25.1

2 6

[PATCH v3 00/20] Support HDMI audio on NVIDIA Tegra20
by Dmitry Osipenko 02 Dec '21

02 Dec '21

This series revives Tegra20 S/PDIF driver which was upstreamed long time ago, but never was used. It also turns Tegra DRM HDMI driver into HDMI audio CODEC provider. Finally, HDMI audio is enabled in device-trees. For now the audio is enable only for Acer A500 tablet and Toshiba AC100 netbook because they're already supported by upstream, later on ASUS TF101 tablet will join them. I based S/PDIF patches on Arnd's Bergmann patch from a separate series [1] that removes obsolete slave_id. This eases merging of the patches by removing the merge conflict. This is a note for Mark Brown. I also based this series on top of power management series [2]. I.e. [2] should be applied first, otherwise "Add S/PDIF node to Tegra20 device-tree" patch should have merge conflict. This is a note for Thierry. [1] https://patchwork.ozlabs.org/project/linux-tegra/list/?series=273312 [2] https://patchwork.ozlabs.org/project/linux-tegra/list/?series=274534 Changelog: v3: - Renamed S/PDIF device-tree clocks as was suggested by Rob Herring. - Added r-bs and acks that were given by Rob Herring to v2. v2: - Corrected I2S yaml problem that was reported by the DT bot for v1 by removing the non-existent required clock-names property. - Removed assigned-clocks property from S/PDIF yaml since this property is now inherited from the clocks property. - Reordered the "tegra20: spdif: Set FIFO trigger level" patch, making it the first sound/soc patch in the series, like it was suggested by Mark Brown in the comment to v1. Also reworded commit message of this patch to *not* make it looks like it should be backported to stable kernels. Arnd Bergmann (1): ASoC: tegra20-spdif: stop setting slave_id Dmitry Osipenko (19): ASoC: dt-bindings: Add binding for Tegra20 S/PDIF ASoC: dt-bindings: tegra20-i2s: Convert to schema ASoC: dt-bindings: tegra20-i2s: Document new nvidia,fixed-parent-rate property dt-bindings: host1x: Document optional HDMI sound-dai-cells ASoC: tegra20: spdif: Set FIFO trigger level ASoC: tegra20: spdif: Support device-tree ASoC: tegra20: spdif: Improve driver's code ASoC: tegra20: spdif: Use more resource-managed helpers ASoC: tegra20: spdif: Reset hardware ASoC: tegra20: spdif: Support system suspend ASoC: tegra20: spdif: Filter out unsupported rates ASoC: tegra20: i2s: Filter out unsupported rates drm/tegra: hdmi: Unwind tegra_hdmi_init() errors drm/tegra: hdmi: Register audio CODEC on Tegra20 ARM: tegra_defconfig: Enable S/PDIF driver ARM: tegra: Add S/PDIF node to Tegra20 device-tree ARM: tegra: Add HDMI audio graph to Tegra20 device-tree ARM: tegra: acer-a500: Enable S/PDIF and HDMI audio ARM: tegra: paz00: Enable S/PDIF and HDMI audio .../display/tegra/nvidia,tegra20-host1x.txt | 1 + .../bindings/sound/nvidia,tegra20-i2s.txt | 30 --- .../bindings/sound/nvidia,tegra20-i2s.yaml | 77 +++++++ .../bindings/sound/nvidia,tegra20-spdif.yaml | 85 ++++++++ .../boot/dts/tegra20-acer-a500-picasso.dts | 8 + arch/arm/boot/dts/tegra20-paz00.dts | 8 + arch/arm/boot/dts/tegra20.dtsi | 40 +++- arch/arm/configs/tegra_defconfig | 1 + drivers/gpu/drm/tegra/Kconfig | 3 + drivers/gpu/drm/tegra/hdmi.c | 168 +++++++++++++-- sound/soc/tegra/tegra20_i2s.c | 49 +++++ sound/soc/tegra/tegra20_spdif.c | 197 ++++++++++++------ sound/soc/tegra/tegra20_spdif.h | 1 + sound/soc/tegra/tegra_pcm.c | 6 + sound/soc/tegra/tegra_pcm.h | 1 + 15 files changed, 572 insertions(+), 103 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml -- 2.33.1

1 20

[PATCH] ALSA: ctl: Fix copy of updated id with element read/write
by Alan Young 02 Dec '21

02 Dec '21

When control_compat.c:copy_ctl_value_to_user() is used, by ctl_elem_read_user() & ctl_elem_write_user(), it must also copy back the snd_ctl_elem_id value that may have been updated (filled in) by the call to snd_ctl_elem_read/snd_ctl_elem_write(). This matches the functionality provided by snd_ctl_elem_read_user() and snd_ctl_elem_write_user(), via snd_ctl_build_ioff(). Without this, and without making additional calls to snd_ctl_info() which are unnecessary when using the non-compat calls, a userspace application will not know the numid value for the element and consequently will not be able to use the poll/read interface on the control file to determine which elements have updates. Signed-off-by: Alan Young <consult.awy(a)gmail.com> --- sound/core/control_compat.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 470dabc60aa0..edff063e088d 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -264,6 +264,7 @@ static int copy_ctl_value_to_user(void __user *userdata, struct snd_ctl_elem_value *data, int type, int count) { + struct snd_ctl_elem_value32 __user *data32 = userdata; int i, size; if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || @@ -280,6 +281,8 @@ static int copy_ctl_value_to_user(void __user *userdata, if (copy_to_user(valuep, data->value.bytes.data, size)) return -EFAULT; } + if (copy_to_user(&data32->id, &data->id, sizeof(data32->id))) + return -EFAULT; return 0; } -- 2.25.1

2 1

[PATCH] ALSA: ctl: Fix copy of updated id with element read/write
by Alan Young 02 Dec '21

02 Dec '21

When control_compat.c:copy_ctl_value_to_user() is used, by ctl_elem_read_user() & ctl_elem_write_user(), it must also copy back the snd_ctl_elem_id value that may have been updated (filled in) by the call to snd_ctl_elem_read/snd_ctl_elem_write(). This matches the functionality provided by snd_ctl_elem_read_user() and snd_ctl_elem_write_user(), via snd_ctl_build_ioff(). Without this, and without making additional calls to snd_ctl_info() which are unnecessary when using the non-compat calls, a userspace application will not know the numid value for the element and consequently will not be able to use the poll/read interface on the control file to determine which elements have updates. Signed-off-by: Alan Young [1]<consult.awy(a)gmail.com> --- sound/core/control_compat.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 470dabc60aa0..edff063e088d 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -264,6 +264,7 @@ static int copy_ctl_value_to_user(void __user *userdata, struct snd_ctl_elem_value *data, int type, int count) { + struct snd_ctl_elem_value32 __user *data32 = userdata; int i, size; if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || @@ -280,6 +281,8 @@ static int copy_ctl_value_to_user(void __user *userdata, if (copy_to_user(valuep, data->value.bytes.data, size)) return -EFAULT; } + if (copy_to_user(&data32->id, &data->id, sizeof(data32->id))) + return -EFAULT; return 0; } -- 2.25.1 References 1. mailto:consult.awy@gmail.com

1 0

[PATCH] ASoC: rt5640: Fix the wrong state of the JD in the HDA header
by Oder Chiou 02 Dec '21

02 Dec '21

The patch fixes the wrong state of the JD with 1M pull up resistor in the HDA header. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 08b37878cb00..f3659b14c74e 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1973,7 +1973,7 @@ static int rt5640_set_bias_level(struct snd_soc_component *component, RT5640_PWR_FV1 | RT5640_PWR_FV2, RT5640_PWR_FV1 | RT5640_PWR_FV2); snd_soc_component_update_bits(component, RT5640_DUMMY1, - 0x0301, 0x0301); + 0x1, 0x1); snd_soc_component_update_bits(component, RT5640_MICBIAS, 0x0030, 0x0030); } @@ -2533,7 +2533,7 @@ static void rt5640_enable_hda_jack_detect( snd_soc_component_update_bits(component, RT5640_GPIO_CTRL3, RT5640_GP1_PF_MASK, RT5640_GP1_PF_OUT); - snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x700, 0x300); + snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x400, 0x0); rt5640->jack = jack; @@ -2651,13 +2651,16 @@ static int rt5640_probe(struct snd_soc_component *component) if (device_property_read_u32(component->dev, "realtek,jack-detect-source", &val) == 0) { - if (val <= RT5640_JD_SRC_GPIO4) + if (val <= RT5640_JD_SRC_GPIO4) { rt5640->jd_src = val << RT5640_JD_SFT; - else if (val == RT5640_JD_SRC_HDA_HEADER) + } else if (val == RT5640_JD_SRC_HDA_HEADER) { rt5640->jd_src = RT5640_JD_SRC_HDA_HEADER; - else + snd_soc_component_update_bits(component, RT5640_DUMMY1, + 0x0300, 0x0); + } else { dev_warn(component->dev, "Warning: Invalid jack-detect-source value: %d, leaving jack-detect disabled\n", val); + } } if (!device_property_read_bool(component->dev, "realtek,jack-detect-not-inverted")) -- 2.34.1

2 2

[PATCH v2 1/2] ASoC: amd: Fix dependency for SPI master
by Lucas Tanure 02 Dec '21

02 Dec '21

Set SPI_MASTER as dependency as is using CS35L41 SPI driver Fixes: 96792fdd77cd1 ("ASoC: amd: enable vangogh platform machine driver build") Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Reported-by: kernel test robot <lkp(a)intel.com> --- v2 -> Fix patch description sound/soc/amd/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 092966ff5ea7..8961b8fd23eb 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -68,7 +68,7 @@ config SND_SOC_AMD_VANGOGH_MACH tristate "AMD Vangogh support for NAU8821 CS35L41" select SND_SOC_NAU8821 select SND_SOC_CS35L41_SPI - depends on SND_SOC_AMD_ACP5x && I2C + depends on SND_SOC_AMD_ACP5x && I2C && SPI_MASTER help This option enables machine driver for Vangogh platform using NAU8821 and CS35L41 codecs. -- 2.34.1

2 2

[PATCH 1/2] ASoC: amd: Fix dependency for SPI master
by Lucas Tanure 02 Dec '21

02 Dec '21

Select SPI_MASTER as is using CS35L41 SPI driver Fixes: 96792fdd77cd1 ("ASoC: amd: enable vangogh platform machine driver build") Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Reported-by: kernel test robot <lkp(a)intel.com> --- sound/soc/amd/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 092966ff5ea7..8961b8fd23eb 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -68,7 +68,7 @@ config SND_SOC_AMD_VANGOGH_MACH tristate "AMD Vangogh support for NAU8821 CS35L41" select SND_SOC_NAU8821 select SND_SOC_CS35L41_SPI - depends on SND_SOC_AMD_ACP5x && I2C + depends on SND_SOC_AMD_ACP5x && I2C && SPI_MASTER help This option enables machine driver for Vangogh platform using NAU8821 and CS35L41 codecs. -- 2.34.1

2 3

[PATCH] ASoC: cs42l42: Implement system suspend
by Richard Fitzgerald 02 Dec '21

02 Dec '21

Add system suspend functions to handle clean power-down on suspend and restoring state on resume. The jack state could change during suspend. Plug->unplug and unplug->plug are straightforward because this looks no different from any other plug state change. However, the jack could be unplugged and a different type of jack plugged, and on resume the plug state would not have changed. Some code changes are needed to the jack handling so that on resume a plugged state will be re-evaluated instead of filtered out. In this case there would not have been a previous unplug event to clear the reported jack state so the full state of all jack type and button flags must be reported. Note that during system suspend any jack plug/unplug and button events will not be reported or generate a system wakeup. If the plug state or headset type has changed it will be reported after resume. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 212 ++++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/cs42l42.h | 8 +- 2 files changed, 209 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 43d98bdb5b5b..5abac0dc8e8b 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -47,7 +47,6 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_I2C_STRETCH, 0x03 }, { CS42L42_I2C_TIMEOUT, 0xB7 }, { CS42L42_PWR_CTL1, 0xFF }, - { CS42L42_PWR_CTL2, 0x84 }, { CS42L42_PWR_CTL3, 0x20 }, { CS42L42_RSENSE_CTL1, 0x40 }, { CS42L42_RSENSE_CTL2, 0x00 }, @@ -331,6 +330,11 @@ static bool cs42l42_volatile_register(struct device *dev, unsigned int reg) case CS42L42_DEVID_CD: case CS42L42_DEVID_E: case CS42L42_MCLK_STATUS: + /* + * PWR_CTL2 must be volatile so it can be used as a canary bit to + * detect a reset during system suspend. + */ + case CS42L42_PWR_CTL2: case CS42L42_OSC_SWITCH_STATUS: case CS42L42_TRSENSE_STATUS: case CS42L42_HS_DET_STATUS: @@ -550,7 +554,7 @@ static int cs42l42_set_jack(struct snd_soc_component *component, struct snd_soc_ struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); /* Prevent race with interrupt handler */ - mutex_lock(&cs42l42->jack_detect_mutex); + mutex_lock(&cs42l42->irq_lock); cs42l42->jack = jk; if (jk) { @@ -566,7 +570,7 @@ static int cs42l42_set_jack(struct snd_soc_component *component, struct snd_soc_ break; } } - mutex_unlock(&cs42l42->jack_detect_mutex); + mutex_unlock(&cs42l42->irq_lock); return 0; } @@ -1613,6 +1617,12 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) unsigned int i; int report = 0; + mutex_lock(&cs42l42->irq_lock); + + if (cs42l42->suspended) { + mutex_unlock(&cs42l42->irq_lock); + return IRQ_NONE; + } /* Read sticky registers to clear interurpt */ for (i = 0; i < ARRAY_SIZE(stickies); i++) { @@ -1635,9 +1645,12 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) CS42L42_M_DETECT_FT_MASK | CS42L42_M_HSBIAS_HIZ_MASK); - mutex_lock(&cs42l42->jack_detect_mutex); - - /* Check auto-detect status */ + /* + * Check auto-detect status. During system suspend the jack could be + * unplugged and a different type plugged. On resume there will not + * have been a previous unplug event to clear the reported flags, so + * here the full state of all flags must be reported. + */ if ((~masks[5]) & irq_params_table[5].mask) { if (stickies[5] & CS42L42_HSDET_AUTO_DONE_MASK) { cs42l42_process_hs_type_detect(cs42l42); @@ -1645,11 +1658,15 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) case CS42L42_PLUG_CTIA: case CS42L42_PLUG_OMTP: snd_soc_jack_report(cs42l42->jack, SND_JACK_HEADSET, - SND_JACK_HEADSET); + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); break; case CS42L42_PLUG_HEADPHONE: snd_soc_jack_report(cs42l42->jack, SND_JACK_HEADPHONE, - SND_JACK_HEADPHONE); + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); break; default: break; @@ -1705,7 +1722,7 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) } } - mutex_unlock(&cs42l42->jack_detect_mutex); + mutex_unlock(&cs42l42->irq_lock); return IRQ_HANDLED; } @@ -2053,8 +2070,9 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, return -ENOMEM; cs42l42->dev = &i2c_client->dev; + cs42l42->irq = i2c_client->irq; i2c_set_clientdata(i2c_client, cs42l42); - mutex_init(&cs42l42->jack_detect_mutex); + mutex_init(&cs42l42->irq_lock); cs42l42->regmap = devm_regmap_init_i2c(i2c_client, &cs42l42_regmap); if (IS_ERR(cs42l42->regmap)) { @@ -2210,6 +2228,179 @@ static int cs42l42_i2c_remove(struct i2c_client *i2c_client) return 0; } +/* Suspend sequence from datasheet */ +static const struct reg_sequence __maybe_unused cs42l42_suspend_seq[] = { + REG_SEQ0(CS42L42_MIC_DET_CTL1, 0x9f), + REG_SEQ0(CS42L42_ADC_OVFL_INT_MASK, 0x01), + REG_SEQ0(CS42L42_MIXER_INT_MASK, 0x0F), + REG_SEQ0(CS42L42_SRC_INT_MASK, 0x0F), + REG_SEQ0(CS42L42_ASP_RX_INT_MASK, 0x1F), + REG_SEQ0(CS42L42_ASP_TX_INT_MASK, 0x0F), + REG_SEQ0(CS42L42_CODEC_INT_MASK, 0x03), + REG_SEQ0(CS42L42_SRCPL_INT_MASK, 0x7F), + REG_SEQ0(CS42L42_VPMON_INT_MASK, 0x01), + REG_SEQ0(CS42L42_PLL_LOCK_INT_MASK, 0x01), + REG_SEQ0(CS42L42_TSRS_PLUG_INT_MASK, 0x0F), + REG_SEQ0(CS42L42_WAKE_CTL, 0xE1), + REG_SEQ0(CS42L42_DET_INT1_MASK, 0xE0), + REG_SEQ0(CS42L42_DET_INT2_MASK, 0xFF), + REG_SEQ0(CS42L42_MIXER_CHA_VOL, 0x3f), + REG_SEQ0(CS42L42_MIXER_ADC_VOL, 0x3f), + REG_SEQ0(CS42L42_MIXER_CHB_VOL, 0x3f), + REG_SEQ0(CS42L42_HP_CTL, 0x0f), + REG_SEQ0(CS42L42_ASP_RX_DAI0_EN, 0x00), + REG_SEQ0(CS42L42_ASP_CLK_CFG, 0x00), + REG_SEQ0(CS42L42_HSDET_CTL2, 0x00), + REG_SEQ0(CS42L42_PWR_CTL1, 0xfe), + REG_SEQ0(CS42L42_PWR_CTL2, 0x8c), + REG_SEQ0(CS42L42_DAC_CTL2, 0x02), + REG_SEQ0(CS42L42_HS_CLAMP_DISABLE, 0x00), + REG_SEQ0(CS42L42_MISC_DET_CTL, 0x03), + REG_SEQ0(CS42L42_TIPSENSE_CTL, 0x02), + REG_SEQ0(CS42L42_HSBIAS_SC_AUTOCTL, 0x03), + REG_SEQ0(CS42L42_PWR_CTL1, 0xff), +}; + +static const struct reg_sequence __maybe_unused cs42l42_suspend_post_seq[] = { + REG_SEQ0(CS42L42_PWR_CTL2, 0x9c), +}; + +static int __maybe_unused cs42l42_suspend(struct device *dev) +{ + struct cs42l42_private *cs42l42 = dev_get_drvdata(dev); + unsigned int reg; + int ret; + + if (cs42l42->irq >= 0) + disable_irq(cs42l42->irq); + + /* + * Wait for threaded irq handler to be idle and stop it processing + * future interrupts. This ensures a safe disable if the interrupt + * is shared. + */ + mutex_lock(&cs42l42->irq_lock); + cs42l42->suspended = true; + + /* Clear any previous PDN_DONE */ + regmap_read(cs42l42->regmap, CS42L42_CODEC_STATUS, &reg); + + /* Preserve cached values so they can be restored. */ + regmap_multi_reg_write_bypassed(cs42l42->regmap, + cs42l42_suspend_seq, + ARRAY_SIZE(cs42l42_suspend_seq)); + + /* The cached address page register value is now stale */ + regcache_drop_region(cs42l42->regmap, CS42L42_PAGE_REGISTER, CS42L42_PAGE_REGISTER); + + /* Wait for power-down complete */ + msleep(CS42L42_PDN_DONE_TIME_MS); + ret = regmap_read_poll_timeout(cs42l42->regmap, CS42L42_CODEC_STATUS, reg, + (reg & CS42L42_PDN_DONE_MASK), + CS42L42_PDN_DONE_POLL_US, + CS42L42_PDN_DONE_TIMEOUT_US); + if (ret) + dev_warn(dev, "Failed to get PDN_DONE: %d\n", ret); + + regmap_multi_reg_write_bypassed(cs42l42->regmap, + cs42l42_suspend_post_seq, + ARRAY_SIZE(cs42l42_suspend_post_seq)); + msleep(CS42L42_FILT_DISCHARGE_TIME_MS); + + /* The cached address page register value is now stale */ + regcache_drop_region(cs42l42->regmap, CS42L42_PAGE_REGISTER, CS42L42_PAGE_REGISTER); + + regcache_cache_only(cs42l42->regmap, true); + + gpiod_set_value_cansleep(cs42l42->reset_gpio, 0); + regulator_bulk_disable(ARRAY_SIZE(cs42l42->supplies), cs42l42->supplies); + + mutex_unlock(&cs42l42->irq_lock); + + dev_dbg(dev, "System suspended\n"); + + return 0; +} + +static int __maybe_unused cs42l42_resume(struct device *dev) +{ + struct cs42l42_private *cs42l42 = dev_get_drvdata(dev); + unsigned int val; + int ret; + + mutex_lock(&cs42l42->irq_lock); + + /* + * If jack was unplugged and re-plugged during suspend it could have + * changed type but the tip-sense state hasn't changed. Force a + * plugged state to be re-evaluated if it is still plugged. + */ + if (cs42l42->plug_state != CS42L42_TS_UNPLUG) + cs42l42->plug_state = CS42L42_TS_TRANS; + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42l42->supplies), cs42l42->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); + + regcache_cache_only(cs42l42->regmap, false); + + if (cs42l42->reset_gpio) { + /* Registers have reset to defaults */ + regcache_mark_dirty(cs42l42->regmap); + } else { + /* + * Only call regcache_mark_dirty() if cs42l42 reset, so + * regcache_sync() writes all non-default cached values. + * If it didn't reset we don't want to filter out syncing + * dirty cache entries that have default value. + * DISCHARGE_FILT==1 after suspend. If the cs42l42 reset + * it will now be 0. + */ + ret = regmap_read(cs42l42->regmap, CS42L42_PWR_CTL2, &val); + if (ret) { + dev_err(dev, "failed to read PWR_CTL2: %d\n", ret); + goto err; + } + + if ((val & CS42L42_DISCHARGE_FILT_MASK) == 0) { + dev_dbg(dev, "Has reset in suspend\n"); + regcache_mark_dirty(cs42l42->regmap); + } else { + dev_dbg(dev, "Did not reset in suspend\n"); + } + } + + /* Sync LATCH_TO_VP first so the VP domain registers sync correctly */ + regcache_sync_region(cs42l42->regmap, CS42L42_MIC_DET_CTL1, CS42L42_MIC_DET_CTL1); + regcache_sync(cs42l42->regmap); + + cs42l42->suspended = false; + mutex_unlock(&cs42l42->irq_lock); + + if (cs42l42->irq >= 0) + enable_irq(cs42l42->irq); + + dev_dbg(dev, "System resumed\n"); + + return 0; +err: + gpiod_set_value_cansleep(cs42l42->reset_gpio, 0); + regulator_bulk_disable(ARRAY_SIZE(cs42l42->supplies), cs42l42->supplies); + + mutex_unlock(&cs42l42->irq_lock); + + return ret; +} + +static const struct dev_pm_ops cs42l42_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(cs42l42_suspend, cs42l42_resume) +}; + #ifdef CONFIG_OF static const struct of_device_id cs42l42_of_match[] = { { .compatible = "cirrus,cs42l42", }, @@ -2236,6 +2427,7 @@ MODULE_DEVICE_TABLE(i2c, cs42l42_id); static struct i2c_driver cs42l42_i2c_driver = { .driver = { .name = "cs42l42", + .pm = &cs42l42_pm_ops, .of_match_table = of_match_ptr(cs42l42_of_match), .acpi_match_table = ACPI_PTR(cs42l42_acpi_match), }, diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 9fff183dce8e..b497ca1618c2 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -826,6 +826,10 @@ #define CS42L42_PLL_LOCK_POLL_US 250 #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 static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = { "VA", @@ -842,7 +846,8 @@ struct cs42l42_private { struct gpio_desc *reset_gpio; struct completion pdn_done; struct snd_soc_jack *jack; - struct mutex jack_detect_mutex; + int irq; + struct mutex irq_lock; int pll_config; int bclk; u32 sclk; @@ -860,6 +865,7 @@ struct cs42l42_private { u8 hs_bias_sense_en; u8 stream_use; bool hp_adc_up_pending; + bool suspended; }; #endif /* __CS42L42_H__ */ -- 2.11.0

3 5

[PATCH v10 0/3] This patches provide ADSP IPC support for MT8195
by allen-kh.cheng 02 Dec '21

02 Dec '21

From: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> Mediatek ADSP IPC is used to send notification or short message between processors with dsp.· It will place the message to the share buffer and will access the ADSP mailbox registers to kick dsp. Two mailboxes used to send notification or short message between processors with dsp changes since v9: - rename adsp_mbox_chan_ops to mtk_adsp_mbox_chan_ops changes since v8: - remove struct adsp_mbox_ch_info - add get_mtk_adsp_mbox_priv - use mtk_adsp_mbox_priv va_mboxreg address in adsp mbox driver - add struct mtk_adsp_mbox_cfg for DSP mbox register offset - remove adsp mbox register offset hard code define - remove mtk-adsp-ipc.h reference in adsp mbox driver changes since v7: - add mtk prefix for adsp ipc functions - rename adsp_mbox_ch_info to mtk_adsp_mbox_ch_info - remove incorrect reviewers in commit message changes since v6: - dt-bindings: change example dtsi node. - rename config MTK_ADSP_IPC_MBOX to MTK_ADSP_MBOX - remove unused variable - add reviewers changes since v5: - remove some redundant code changes since v4: - use switch ... case in adsp_ipc_recv - add error handling path for chan_name pointer - refine some code to be concise changes since v3: - reorder MTK_ADSP_IPC_MBOX config - remove some redundant code - remove lock in mtk-adsp-mailbox changes since v2: - separate adsp_mailbox into two instances changes since v1: - fix dt_binding_check error Allen-KH Cheng (3): dt-bindings: mediatek: add adsp-mbox document firmware: mediatek: add adsp ipc protocol interface mailbox: mediatek: add support for adsp mailbox controller .../bindings/mailbox/mtk,adsp-mbox.yaml | 52 ++++++ drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 9 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 135 ++++++++++++++ drivers/mailbox/Kconfig | 7 + drivers/mailbox/Makefile | 2 + drivers/mailbox/mtk-adsp-mailbox.c | 176 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 65 +++++++ 10 files changed, 450 insertions(+) create mode 100644 Documentation/devicetree/bindings/mailbox/mtk,adsp-mbox.yaml create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 drivers/mailbox/mtk-adsp-mailbox.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

1 3

[PATCH v6 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 02 Dec '21

02 Dec '21

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add regmap config support for codec dma driver ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio ASoC: qcom: SC7280: Update config for building codec dma drivers .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 69 ++- sound/soc/qcom/Kconfig | 13 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 275 ++++++++++ sound/soc/qcom/lpass-cpu.c | 244 ++++++++- sound/soc/qcom/lpass-lpaif-reg.h | 103 +++- sound/soc/qcom/lpass-platform.c | 578 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 434 ++++++++++++++++ sound/soc/qcom/lpass.h | 151 ++++++ 9 files changed, 1784 insertions(+), 87 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 22

alsautil: init procedure is executed during a store and restore process when the final snd_ctl_elem is not readable
by 易先生 02 Dec '21

02 Dec '21

hi, team When i execute the following commands sequencitally, alsa store -f 1.txt alsa restore -f 1.txt my playing volume was initialized. After a debugging process, I found the problem: During the "store" process, a control element is saved to the configuration file only when this element is readable.While during the "restore" process, after all the control element from the configuration file is processed, an "additional control element check" which causes a "init procedure" if the result is true is processed. Thus, if total control element count is 5 with the 5th element is not readable, then there will be only 4 control element stored in the configuration file, ranges from "control.1" to "control.4". Then the "additional control checking" process produces "true" and the variable "err" is evaluated as "-EAGAIN".

1 0

[PATCH] ALSA: usb-audio: Drop CONFIG_PM
by Takashi Iwai 02 Dec '21

02 Dec '21

Practically seen, CONFIG_PM is almost mandatory. Let's drop the ugly ifdef lines and simplify the code. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 7 ------- sound/usb/mixer.c | 4 ---- sound/usb/mixer.h | 2 -- sound/usb/mixer_quirks.c | 2 -- sound/usb/mixer_quirks.h | 2 -- sound/usb/power.h | 10 ---------- 6 files changed, 27 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 1764b9302d46..376962291c4d 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -987,8 +987,6 @@ void snd_usb_unlock_shutdown(struct snd_usb_audio *chip) wake_up(&chip->shutdown_wait); } -#ifdef CONFIG_PM - int snd_usb_autoresume(struct snd_usb_audio *chip) { int i, err; @@ -1100,11 +1098,6 @@ static int usb_audio_resume(struct usb_interface *intf) atomic_dec(&chip->active); /* allow autopm after this point */ return err; } -#else -#define usb_audio_suspend NULL -#define usb_audio_resume NULL -#define usb_audio_resume NULL -#endif /* CONFIG_PM */ static const struct usb_device_id usb_audio_ids [] = { #include "quirks-table.h" diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 5b9fd07ce2a2..e8f3f8d622ec 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3629,7 +3629,6 @@ void snd_usb_mixer_disconnect(struct usb_mixer_interface *mixer) mixer->disconnected = true; } -#ifdef CONFIG_PM /* stop any bus activity of a mixer */ static void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer) { @@ -3711,7 +3710,6 @@ int snd_usb_mixer_resume(struct usb_mixer_interface *mixer) return snd_usb_mixer_activate(mixer); } -#endif void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, struct usb_mixer_interface *mixer, @@ -3720,7 +3718,5 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, list->mixer = mixer; list->id = unitid; list->dump = snd_usb_mixer_dump_cval; -#ifdef CONFIG_PM list->resume = restore_mixer_value; -#endif } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 98ea24d91d80..d43895c1ae5c 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -118,10 +118,8 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv); -#ifdef CONFIG_PM int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer); int snd_usb_mixer_resume(struct usb_mixer_interface *mixer); -#endif int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int index, int value); diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index db194ad168d0..1f9863725c7c 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3280,7 +3280,6 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) return err; } -#ifdef CONFIG_PM void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer) { switch (mixer->chip->usb_id) { @@ -3289,7 +3288,6 @@ void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer) break; } } -#endif void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, int unitid) diff --git a/sound/usb/mixer_quirks.h b/sound/usb/mixer_quirks.h index 52be26db558f..4ba01ba3fe8b 100644 --- a/sound/usb/mixer_quirks.h +++ b/sound/usb/mixer_quirks.h @@ -14,9 +14,7 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, struct usb_mixer_elem_info *cval, int unitid, struct snd_kcontrol *kctl); -#ifdef CONFIG_PM void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer); -#endif #endif /* SND_USB_MIXER_QUIRKS_H */ diff --git a/sound/usb/power.h b/sound/usb/power.h index 6004231a7c75..396e3e51440a 100644 --- a/sound/usb/power.h +++ b/sound/usb/power.h @@ -21,17 +21,7 @@ struct snd_usb_power_domain * snd_usb_find_power_domain(struct usb_host_interface *ctrl_iface, unsigned char id); -#ifdef CONFIG_PM int snd_usb_autoresume(struct snd_usb_audio *chip); void snd_usb_autosuspend(struct snd_usb_audio *chip); -#else -static inline int snd_usb_autoresume(struct snd_usb_audio *chip) -{ - return 0; -} -static inline void snd_usb_autosuspend(struct snd_usb_audio *chip) -{ -} -#endif #endif /* __USBAUDIO_POWER_H */ -- 2.31.1

1 0

[PATCH] ALSA: usb-audio: Drop superfluous '0' in Presonus Studio 1810c's ID
by Takashi Iwai 02 Dec '21

02 Dec '21

The vendor ID of Presonus Studio 1810c had a superfluous '0' in its USB ID. Drop it. Fixes: 8dc5efe3d17c ("ALSA: usb-audio: Add support for Presonus Studio 1810c") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/format.c | 2 +- sound/usb/mixer_quirks.c | 2 +- sound/usb/quirks.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index f5e676a51b30..405dc0bf6678 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -375,7 +375,7 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, for (rate = min; rate <= max; rate += res) { /* Filter out invalid rates on Presonus Studio 1810c */ - if (chip->usb_id == USB_ID(0x0194f, 0x010c) && + if (chip->usb_id == USB_ID(0x194f, 0x010c) && !s1810c_valid_sample_rate(fp, rate)) goto skip_rate; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index d489c1de3bae..db194ad168d0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3254,7 +3254,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_rme_controls_create(mixer); break; - case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */ + case USB_ID(0x194f, 0x010c): /* Presonus Studio 1810c */ err = snd_sc1810_init_mixer(mixer); break; case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 64e1c20311ed..ab9f3da49941 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1290,7 +1290,7 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, if (chip->usb_id == USB_ID(0x0763, 0x2012)) return fasttrackpro_skip_setting_quirk(chip, iface, altno); /* presonus studio 1810c: skip altsets incompatible with device_setup */ - if (chip->usb_id == USB_ID(0x0194f, 0x010c)) + if (chip->usb_id == USB_ID(0x194f, 0x010c)) return s1810c_skip_setting_quirk(chip, iface, altno); -- 2.31.1

1 0

[PATCH] ALSA: hda/hdmi: Consider ELD is invalid when no SAD is present
by Kai-Heng Feng 02 Dec '21

02 Dec '21

There's a system that reports a bogus HDMI audio interface: $ cat eld#2.0 monitor_present 1 eld_valid 1 monitor_name connection_type DisplayPort eld_version [0x2] CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0xe430 product_id 0x690 port_id 0x0 support_hdcp 0 support_ai 0 audio_sync_delay 0 speakers [0xffff] FL/FR LFE FC RL/RR RC FLC/FRC RLC/RRC FLW/FRW FLH/FRH TC FCH sad_count 0 Since playing audio is not possible without SAD, also consider ELD is invalid for this case. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 415701bd10ac8..e7c2f3167f311 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1535,7 +1535,7 @@ static void update_eld(struct hda_codec *codec, } } - if (!eld->eld_valid || eld->eld_size <= 0) { + if (!eld->eld_valid || eld->eld_size <= 0 || eld->info.sad_count <= 0) { eld->eld_valid = false; eld->eld_size = 0; } -- 2.32.0

2 1

[PATCH v9 0/3] This patches provide ADSP IPC support for MT8195
by allen-kh.cheng 02 Dec '21

02 Dec '21

From: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> Mediatek ADSP IPC is used to send notification or short message between processors with dsp.· It will place the message to the share buffer and will access the ADSP mailbox registers to kick dsp. Two mailboxes used to send notification or short message between processors with dsp changes since v8: - remove struct adsp_mbox_ch_info - add get_mtk_adsp_mbox_priv - use mtk_adsp_mbox_priv va_mboxreg address in adsp mbox driver - add struct mtk_adsp_mbox_cfg for DSP mbox register offset - remove adsp mbox register offset hard code define - remove mtk-adsp-ipc.h reference in adsp mbox driver changes since v7: - add mtk prefix for adsp ipc functions - rename adsp_mbox_ch_info to mtk_adsp_mbox_ch_info - rename adsp_mbox_chan_ops to mtk_adsp_mbox_chan_ops - remove incorrect reviewers in commit message changes since v6: - dt-bindings: change example dtsi node. - rename config MTK_ADSP_IPC_MBOX to MTK_ADSP_MBOX - remove unused variable - add reviewers changes since v5: - remove some redundant code changes since v4: - use switch ... case in adsp_ipc_recv - add error handling path for chan_name pointer - refine some code to be concise changes since v3: - reorder MTK_ADSP_IPC_MBOX config - remove some redundant code - remove lock in mtk-adsp-mailbox changes since v2: - separate adsp_mailbox into two instances changes since v1: - fix dt_binding_check error Allen-KH Cheng (3): dt-bindings: mediatek: add adsp-mbox document firmware: mediatek: add adsp ipc protocol interface mailbox: mediatek: add support for adsp mailbox controller .../bindings/mailbox/mtk,adsp-mbox.yaml | 52 ++++++ drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 9 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 135 ++++++++++++++ drivers/mailbox/Kconfig | 7 + drivers/mailbox/Makefile | 2 + drivers/mailbox/mtk-adsp-mailbox.c | 176 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 65 +++++++ 10 files changed, 450 insertions(+) create mode 100644 Documentation/devicetree/bindings/mailbox/mtk,adsp-mbox.yaml create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 drivers/mailbox/mtk-adsp-mailbox.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

2 5

[PATCH v2 00/20] Support HDMI audio on NVIDIA Tegra20
by Dmitry Osipenko 02 Dec '21

02 Dec '21

This series revives Tegra20 S/PDIF driver which was upstreamed long time ago, but never was used. It also turns Tegra DRM HDMI driver into HDMI audio CODEC provider. Finally, HDMI audio is enabled in device-trees. For now the audio is enable only for Acer A500 tablet and Toshiba AC100 netbook because they're already supported by upstream, later on ASUS TF101 tablet will join them. I based S/PDIF patches on Arnd's Bergmann patch from a separate series [1] that removes obsolete slave_id. This eases merging of the patches by removing the merge conflict. [1] https://patchwork.ozlabs.org/project/linux-tegra/list/?submitter=80402 Changelog: v1: - Corrected I2S yaml problem that was reported by the DT bot for v1 by removing the non-existent required clock-names property. - Removed assigned-clocks property from S/PDIF yaml since this property is now inherited from the clocks property. - Reordered the "tegra20: spdif: Set FIFO trigger level" patch, making it the first sound/soc patch in the series, like it was suggested by Mark Brown in the comment to v1. Also reworded commit message of this patch to *not* make it looks like it should be backported to stable kernels. Arnd Bergmann (1): ASoC: tegra20-spdif: stop setting slave_id Dmitry Osipenko (19): ASoC: dt-bindings: Add binding for Tegra20 S/PDIF ASoC: dt-bindings: tegra20-i2s: Convert to schema ASoC: dt-bindings: tegra20-i2s: Document new nvidia,fixed-parent-rate property dt-bindings: host1x: Document optional HDMI sound-dai-cells ASoC: tegra20: spdif: Set FIFO trigger level ASoC: tegra20: spdif: Support device-tree ASoC: tegra20: spdif: Improve driver's code ASoC: tegra20: spdif: Use more resource-managed helpers ASoC: tegra20: spdif: Reset hardware ASoC: tegra20: spdif: Support system suspend ASoC: tegra20: spdif: Filter out unsupported rates ASoC: tegra20: i2s: Filter out unsupported rates drm/tegra: hdmi: Unwind tegra_hdmi_init() errors drm/tegra: hdmi: Register audio CODEC on Tegra20 ARM: tegra_defconfig: Enable S/PDIF driver ARM: tegra: Add S/PDIF node to Tegra20 device-tree ARM: tegra: Add HDMI audio graph to Tegra20 device-tree ARM: tegra: acer-a500: Enable S/PDIF and HDMI audio ARM: tegra: paz00: Enable S/PDIF and HDMI audio .../display/tegra/nvidia,tegra20-host1x.txt | 1 + .../bindings/sound/nvidia,tegra20-i2s.txt | 30 --- .../bindings/sound/nvidia,tegra20-i2s.yaml | 77 +++++++ .../bindings/sound/nvidia,tegra20-spdif.yaml | 85 ++++++++ .../boot/dts/tegra20-acer-a500-picasso.dts | 8 + arch/arm/boot/dts/tegra20-paz00.dts | 8 + arch/arm/boot/dts/tegra20.dtsi | 40 +++- arch/arm/configs/tegra_defconfig | 1 + drivers/gpu/drm/tegra/Kconfig | 3 + drivers/gpu/drm/tegra/hdmi.c | 168 +++++++++++++-- sound/soc/tegra/tegra20_i2s.c | 49 +++++ sound/soc/tegra/tegra20_spdif.c | 195 +++++++++++++----- sound/soc/tegra/tegra20_spdif.h | 1 + sound/soc/tegra/tegra_pcm.c | 6 + sound/soc/tegra/tegra_pcm.h | 1 + 15 files changed, 571 insertions(+), 102 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml -- 2.33.1

2 24

[PATCH v2 0/4] ARM: dts: stm32: merge spi and i2s nodes
by Olivier Moysan 02 Dec '21

02 Dec '21

When a STM32 SPI instance offers I2S feature, two nodes are defined in STM32 SoC device tree to support both SPI and I2S. Merge SPI node and I2S nodes into a single node, to avoid hardware description duplication and compilation warnings. Changes in v2: - Squash some patches in the serie to avoid dtbs check warnings on intermediate compilation steps. Olivier Moysan (4): ASoC: dt-bindings: stm32: i2s: add audio-graph-card port ASoC: dt-bindings: stm32: i2s: allow additional properties. ASoC: dt-bindings: stm32: i2s: update example ARM: dts: stm32: merge spi and i2s nodes .../bindings/sound/st,stm32-i2s.yaml | 9 ++++- arch/arm/boot/dts/stm32mp151.dtsi | 39 ++----------------- arch/arm/boot/dts/stm32mp157c-ev1.dts | 2 +- .../arm/boot/dts/stm32mp15xx-dhcom-drc02.dtsi | 2 +- .../boot/dts/stm32mp15xx-dhcor-avenger96.dtsi | 2 +- arch/arm/boot/dts/stm32mp15xx-dkx.dtsi | 7 +++- 6 files changed, 18 insertions(+), 43 deletions(-) -- 2.17.1

2 8

[PATCH v2] ASoC: tegra: Add master volume/mute control support
by Sameer Pujar 01 Dec '21

01 Dec '21

The MVC module has a per channel control bit, based on which it decides to apply channel specific volume/mute settings. When per channel control bit is enabled (which is the default HW configuration), all MVC channel volume/mute can be independently controlled. If the control is disabled, channel-0 volume/mute setting is applied by HW to all remaining channels. Thus add support to leverage this HW feature by exposing master controls for volume/mute. With this, now there are per channel and master volume/mute controls. Users need to just use controls which are suitable for their applications. The per channel control enable/disable is mananged in driver and hidden from users, so that they need to just worry about respective volume/mute controls. Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- changes in v2: * Kcontrol put() related comments, received during v1, are addressed in another series [0]. * Thus v2 is rebased on top of recent changes. * In doing so, this patch also addresses optimization comment received on [0] which suggested usage of regmap_update_bits_check() in the MVC driver. [0] https://lkml.org/lkml/2021/11/18/930 sound/soc/tegra/tegra210_mvc.c | 209 ++++++++++++++++++++++++++++++++--------- sound/soc/tegra/tegra210_mvc.h | 5 + 2 files changed, 169 insertions(+), 45 deletions(-) diff --git a/sound/soc/tegra/tegra210_mvc.c b/sound/soc/tegra/tegra210_mvc.c index 93cbeb6..2d79138 100644 --- a/sound/soc/tegra/tegra210_mvc.c +++ b/sound/soc/tegra/tegra210_mvc.c @@ -108,67 +108,152 @@ static void tegra210_mvc_conv_vol(struct tegra210_mvc *mvc, u8 chan, s32 val) } } -static int tegra210_mvc_get_mute(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static u32 tegra210_mvc_get_ctrl_reg(struct snd_kcontrol *kcontrol) { struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_mvc *mvc = snd_soc_component_get_drvdata(cmpnt); - u8 mute_mask; u32 val; pm_runtime_get_sync(cmpnt->dev); regmap_read(mvc->regmap, TEGRA210_MVC_CTRL, &val); pm_runtime_put(cmpnt->dev); - mute_mask = (val >> TEGRA210_MVC_MUTE_SHIFT) & - TEGRA210_MUTE_MASK_EN; + return val; +} + +static int tegra210_mvc_get_mute(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 val = tegra210_mvc_get_ctrl_reg(kcontrol); + u8 mute_mask = TEGRA210_GET_MUTE_VAL(val); - ucontrol->value.integer.value[0] = mute_mask; + /* + * If per channel control is enabled, then return + * exact mute/unmute setting of all channels. + * + * Else report setting based on CH0 bit to reflect + * the correct HW state. + */ + if (val & TEGRA210_MVC_PER_CHAN_CTRL_EN) { + ucontrol->value.integer.value[0] = mute_mask; + } else { + if (mute_mask & TEGRA210_MVC_CH0_MUTE_EN) + ucontrol->value.integer.value[0] = + TEGRA210_MUTE_MASK_EN; + else + ucontrol->value.integer.value[0] = 0; + } return 0; } -static int tegra210_mvc_put_mute(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int tegra210_mvc_get_master_mute(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 val = tegra210_mvc_get_ctrl_reg(kcontrol); + u8 mute_mask = TEGRA210_GET_MUTE_VAL(val); + + /* + * If per channel control is disabled, then return + * master mute/unmute setting based on CH0 bit. + * + * Else report settings based on state of all + * channels. + */ + if (!(val & TEGRA210_MVC_PER_CHAN_CTRL_EN)) { + ucontrol->value.integer.value[0] = + mute_mask & TEGRA210_MVC_CH0_MUTE_EN; + } else { + if (mute_mask == TEGRA210_MUTE_MASK_EN) + ucontrol->value.integer.value[0] = + TEGRA210_MVC_CH0_MUTE_EN; + else + ucontrol->value.integer.value[0] = 0; + } + + return 0; +} + +static int tegra210_mvc_volume_switch_timeout(struct snd_soc_component *cmpnt) { - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_mvc *mvc = snd_soc_component_get_drvdata(cmpnt); - unsigned int value; - u8 new_mask, old_mask; + u32 value; int err; - pm_runtime_get_sync(cmpnt->dev); - - /* Check if VOLUME_SWITCH is triggered */ err = regmap_read_poll_timeout(mvc->regmap, TEGRA210_MVC_SWITCH, value, !(value & TEGRA210_MVC_VOLUME_SWITCH_MASK), 10, 10000); if (err < 0) - goto end; + dev_err(cmpnt->dev, + "Volume switch trigger is still active, err = %d\n", + err); - regmap_read(mvc->regmap, TEGRA210_MVC_CTRL, &value); + return err; +} - old_mask = (value >> TEGRA210_MVC_MUTE_SHIFT) & TEGRA210_MUTE_MASK_EN; - new_mask = ucontrol->value.integer.value[0]; +static int tegra210_mvc_update_mute(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, + bool per_chan_ctrl) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct tegra210_mvc *mvc = snd_soc_component_get_drvdata(cmpnt); + u32 mute_val = ucontrol->value.integer.value[0]; + u32 per_ch_ctrl_val; + bool change = false; + int err; - if (new_mask == old_mask) { - err = 0; + pm_runtime_get_sync(cmpnt->dev); + + err = tegra210_mvc_volume_switch_timeout(cmpnt); + if (err < 0) goto end; + + if (per_chan_ctrl) { + per_ch_ctrl_val = TEGRA210_MVC_PER_CHAN_CTRL_EN; + } else { + per_ch_ctrl_val = 0; + + if (mute_val) + mute_val = TEGRA210_MUTE_MASK_EN; } - err = regmap_update_bits(mvc->regmap, mc->reg, + regmap_update_bits_check(mvc->regmap, TEGRA210_MVC_CTRL, TEGRA210_MVC_MUTE_MASK, - new_mask << TEGRA210_MVC_MUTE_SHIFT); - if (err < 0) - goto end; + mute_val << TEGRA210_MVC_MUTE_SHIFT, + &change); - err = 1; + if (change) { + regmap_update_bits(mvc->regmap, TEGRA210_MVC_CTRL, + TEGRA210_MVC_PER_CHAN_CTRL_EN_MASK, + per_ch_ctrl_val); + + regmap_update_bits(mvc->regmap, TEGRA210_MVC_SWITCH, + TEGRA210_MVC_VOLUME_SWITCH_MASK, + TEGRA210_MVC_VOLUME_SWITCH_TRIGGER); + } end: pm_runtime_put(cmpnt->dev); - return err; + + if (err < 0) + return err; + + if (change) + return 1; + + return 0; +} + +static int tegra210_mvc_put_mute(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return tegra210_mvc_update_mute(kcontrol, ucontrol, true); +} + +static int tegra210_mvc_put_master_mute(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return tegra210_mvc_update_mute(kcontrol, ucontrol, false); } static int tegra210_mvc_get_vol(struct snd_kcontrol *kcontrol, @@ -178,7 +263,7 @@ static int tegra210_mvc_get_vol(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_mvc *mvc = snd_soc_component_get_drvdata(cmpnt); - u8 chan = (mc->reg - TEGRA210_MVC_TARGET_VOL) / REG_SIZE; + u8 chan = TEGRA210_MVC_GET_CHAN(mc->reg, TEGRA210_MVC_TARGET_VOL); s32 val = mvc->volume[chan]; if (mvc->curve_type == CURVE_POLY) { @@ -193,44 +278,55 @@ static int tegra210_mvc_get_vol(struct snd_kcontrol *kcontrol, return 0; } -static int tegra210_mvc_put_vol(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int tegra210_mvc_get_master_vol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return tegra210_mvc_get_vol(kcontrol, ucontrol); +} + +static int tegra210_mvc_update_vol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, + bool per_ch_enable) { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_mvc *mvc = snd_soc_component_get_drvdata(cmpnt); - unsigned int reg = mc->reg; - unsigned int value; - u8 chan; - int err, old_volume; + u8 chan = TEGRA210_MVC_GET_CHAN(mc->reg, TEGRA210_MVC_TARGET_VOL); + int old_volume = mvc->volume[chan]; + int err, i; pm_runtime_get_sync(cmpnt->dev); - /* Check if VOLUME_SWITCH is triggered */ - err = regmap_read_poll_timeout(mvc->regmap, TEGRA210_MVC_SWITCH, - value, !(value & TEGRA210_MVC_VOLUME_SWITCH_MASK), - 10, 10000); + err = tegra210_mvc_volume_switch_timeout(cmpnt); if (err < 0) goto end; - chan = (reg - TEGRA210_MVC_TARGET_VOL) / REG_SIZE; - old_volume = mvc->volume[chan]; - - tegra210_mvc_conv_vol(mvc, chan, - ucontrol->value.integer.value[0]); + tegra210_mvc_conv_vol(mvc, chan, ucontrol->value.integer.value[0]); if (mvc->volume[chan] == old_volume) { err = 0; goto end; } + if (per_ch_enable) { + regmap_update_bits(mvc->regmap, TEGRA210_MVC_CTRL, + TEGRA210_MVC_PER_CHAN_CTRL_EN_MASK, + TEGRA210_MVC_PER_CHAN_CTRL_EN); + } else { + regmap_update_bits(mvc->regmap, TEGRA210_MVC_CTRL, + TEGRA210_MVC_PER_CHAN_CTRL_EN_MASK, 0); + + for (i = 1; i < TEGRA210_MVC_MAX_CHAN_COUNT; i++) + mvc->volume[i] = mvc->volume[chan]; + } + /* Configure init volume same as target volume */ regmap_write(mvc->regmap, TEGRA210_MVC_REG_OFFSET(TEGRA210_MVC_INIT_VOL, chan), mvc->volume[chan]); - regmap_write(mvc->regmap, reg, mvc->volume[chan]); + regmap_write(mvc->regmap, mc->reg, mvc->volume[chan]); regmap_update_bits(mvc->regmap, TEGRA210_MVC_SWITCH, TEGRA210_MVC_VOLUME_SWITCH_MASK, @@ -240,9 +336,22 @@ static int tegra210_mvc_put_vol(struct snd_kcontrol *kcontrol, end: pm_runtime_put(cmpnt->dev); + return err; } +static int tegra210_mvc_put_vol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return tegra210_mvc_update_vol(kcontrol, ucontrol, true); +} + +static int tegra210_mvc_put_master_vol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return tegra210_mvc_update_vol(kcontrol, ucontrol, false); +} + static void tegra210_mvc_reset_vol_settings(struct tegra210_mvc *mvc, struct device *dev) { @@ -438,6 +547,16 @@ static const struct snd_kcontrol_new tegra210_mvc_vol_ctrl[] = { TEGRA210_MVC_CTRL, 0, TEGRA210_MUTE_MASK_EN, 0, tegra210_mvc_get_mute, tegra210_mvc_put_mute), + /* Master volume */ + SOC_SINGLE_EXT("Volume", TEGRA210_MVC_TARGET_VOL, 0, 16000, 0, + tegra210_mvc_get_master_vol, + tegra210_mvc_put_master_vol), + + /* Master mute */ + SOC_SINGLE_EXT("Mute", TEGRA210_MVC_CTRL, 0, 1, 0, + tegra210_mvc_get_master_mute, + tegra210_mvc_put_master_mute), + SOC_ENUM_EXT("Curve Type", tegra210_mvc_curve_type_ctrl, tegra210_mvc_get_curve_type, tegra210_mvc_put_curve_type), }; diff --git a/sound/soc/tegra/tegra210_mvc.h b/sound/soc/tegra/tegra210_mvc.h index def29c4..d775335 100644 --- a/sound/soc/tegra/tegra210_mvc.h +++ b/sound/soc/tegra/tegra210_mvc.h @@ -59,6 +59,7 @@ #define TEGRA210_MUTE_MASK_EN 0xff #define TEGRA210_MVC_MUTE_MASK (TEGRA210_MUTE_MASK_EN << TEGRA210_MVC_MUTE_SHIFT) #define TEGRA210_MVC_MUTE_EN (TEGRA210_MUTE_MASK_EN << TEGRA210_MVC_MUTE_SHIFT) +#define TEGRA210_MVC_CH0_MUTE_EN 1 #define TEGRA210_MVC_PER_CHAN_CTRL_EN_SHIFT 30 #define TEGRA210_MVC_PER_CHAN_CTRL_EN_MASK (1 << TEGRA210_MVC_PER_CHAN_CTRL_EN_SHIFT) @@ -92,6 +93,10 @@ #define TEGRA210_MVC_MAX_CHAN_COUNT 8 #define TEGRA210_MVC_REG_OFFSET(reg, i) (reg + (REG_SIZE * i)) +#define TEGRA210_MVC_GET_CHAN(reg, base) (((reg) - (base)) / REG_SIZE) + +#define TEGRA210_GET_MUTE_VAL(val) (((val) >> TEGRA210_MVC_MUTE_SHIFT) & TEGRA210_MUTE_MASK_EN) + #define NUM_GAIN_POLY_COEFFS 9 enum { -- 2.7.4

2 1

[PATCH] firmware: cs_dsp: Move lockdep asserts to avoid potential null pointer
by Charles Keepax 01 Dec '21

01 Dec '21

Move the lockdep asserts until after the ctl pointer has been checked for NULL, to avoid potentially NULL pointer dereferences. Fixes: fb2f364fb5b9 ("firmware: cs_dsp: Add lockdep asserts to interface functions") Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- drivers/firmware/cirrus/cs_dsp.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index 3814cbba0a544..5af8171d6cedf 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -759,11 +759,11 @@ int cs_dsp_coeff_write_ctrl(struct cs_dsp_coeff_ctl *ctl, { int ret = 0; - lockdep_assert_held(&ctl->dsp->pwr_lock); - if (!ctl) return -ENOENT; + lockdep_assert_held(&ctl->dsp->pwr_lock); + if (len + off * sizeof(u32) > ctl->len) return -EINVAL; @@ -827,11 +827,11 @@ int cs_dsp_coeff_read_ctrl(struct cs_dsp_coeff_ctl *ctl, { int ret = 0; - lockdep_assert_held(&ctl->dsp->pwr_lock); - if (!ctl) return -ENOENT; + lockdep_assert_held(&ctl->dsp->pwr_lock); + if (len + off * sizeof(u32) > ctl->len) return -EINVAL; -- 2.11.0

2 1

[PATCH] ASoC: mediatek: mt8195: make several arrays static const
by Colin Ian King 01 Dec '21

01 Dec '21

Don't populate various arrays on the stack but instead make them static const. Also makes the object code smaller by a few hundred bytes. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-clk.c b/sound/soc/mediatek/mt8195/mt8195-afe-clk.c index 8420b2c71332..c2543f4cffb7 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-clk.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-clk.c @@ -326,7 +326,7 @@ int mt8195_afe_enable_reg_rw_clk(struct mtk_base_afe *afe) { struct mt8195_afe_private *afe_priv = afe->platform_priv; int i; - unsigned int clk_array[] = { + static const unsigned int clk_array[] = { MT8195_CLK_SCP_ADSP_AUDIODSP, /* bus clock for infra */ MT8195_CLK_TOP_AUDIO_H_SEL, /* clock for ADSP bus */ MT8195_CLK_TOP_AUDIO_LOCAL_BUS_SEL, /* bus clock for DRAM access */ @@ -347,7 +347,7 @@ int mt8195_afe_disable_reg_rw_clk(struct mtk_base_afe *afe) { struct mt8195_afe_private *afe_priv = afe->platform_priv; int i; - unsigned int clk_array[] = { + static const unsigned int clk_array[] = { MT8195_CLK_AUD_A1SYS, MT8195_CLK_AUD_A1SYS_HP, MT8195_CLK_AUD_AFE, @@ -380,11 +380,11 @@ static int mt8195_afe_enable_timing_sys(struct mtk_base_afe *afe) { struct mt8195_afe_private *afe_priv = afe->platform_priv; int i; - unsigned int clk_array[] = { + static const unsigned int clk_array[] = { MT8195_CLK_AUD_A1SYS, MT8195_CLK_AUD_A2SYS, }; - unsigned int cg_array[] = { + static const unsigned int cg_array[] = { MT8195_TOP_CG_A1SYS_TIMING, MT8195_TOP_CG_A2SYS_TIMING, MT8195_TOP_CG_26M_TIMING, @@ -403,11 +403,11 @@ static int mt8195_afe_disable_timing_sys(struct mtk_base_afe *afe) { struct mt8195_afe_private *afe_priv = afe->platform_priv; int i; - unsigned int clk_array[] = { + static const unsigned int clk_array[] = { MT8195_CLK_AUD_A2SYS, MT8195_CLK_AUD_A1SYS, }; - unsigned int cg_array[] = { + static const unsigned int cg_array[] = { MT8195_TOP_CG_26M_TIMING, MT8195_TOP_CG_A2SYS_TIMING, MT8195_TOP_CG_A1SYS_TIMING, -- 2.33.1

2 1

[PATCH v4] ASoC: Intel: boards: add max98390 2/4 speakers support
by Mac Chiang 01 Dec '21

01 Dec '21

support 2 hw boards. 1. SSP2 connects max98390, 2 speakers. 2. SSP1 connects max98390, 2/4 speakers. 2 or 4 speakers playback add echo reference capture add bt offload support add DMI_OEM_STRING for board variants add ALC5682I-VS support Signed-off-by: Mark Hsieh <mark_hsieh(a)wistron.corp-partner.google.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Kieth Tzeng <keith.tzeng(a)quantatw.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- v3 -> v4: - use comp_ids to scan acpi support ALC5682I-VD or ALC5682I-VS - remove redundant drv_name "adl_max98390_rt5682s" from soc-acpi-intel-adl-match.c and device table in sof_rt5682.c --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_maxim_common.c | 180 ++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 16 ++ sound/soc/intel/boards/sof_rt5682.c | 72 +++++++ .../intel/common/soc-acpi-intel-adl-match.c | 13 ++ sound/soc/sof/sof-pci-dev.c | 9 + 6 files changed, 291 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 849445fcc05d..34ccefcc30c7 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -467,6 +467,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) select SND_SOC_MAX98373_I2C + select SND_SOC_MAX98390 select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT1015P diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index e66dfe666915..9171d9cd179e 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -5,6 +5,7 @@ #include <linux/string.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include <sound/soc-dai.h> #include <sound/soc-dapm.h> #include <uapi/sound/asound.h> @@ -133,6 +134,185 @@ void max_98373_set_codec_conf(struct snd_soc_card *card) } EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); +/* + * Maxim MAX98390 + */ +const struct snd_soc_dapm_route max_98390_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, +}; + +static const struct snd_kcontrol_new max_98390_tt_kcontrols[] = { + SOC_DAPM_PIN_SWITCH("TL Spk"), + SOC_DAPM_PIN_SWITCH("TR Spk"), +}; + +static const struct snd_soc_dapm_widget max_98390_tt_dapm_widgets[] = { + SND_SOC_DAPM_SPK("TL Spk", NULL), + SND_SOC_DAPM_SPK("TR Spk", NULL), +}; + +const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = { + /* Tweeter speaker */ + { "TL Spk", NULL, "Tweeter Left BE_OUT" }, + { "TR Spk", NULL, "Tweeter Right BE_OUT" }, +}; + +static struct snd_soc_codec_conf max_98390_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +static struct snd_soc_codec_conf max_98390_4spk_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV2_NAME), + .name_prefix = "Tweeter Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV3_NAME), + .name_prefix = "Tweeter Left", + }, +}; + +struct snd_soc_dai_link_component max_98390_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +struct snd_soc_dai_link_component max_98390_4spk_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV2_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV3_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_4spk_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static int max_98390_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int i; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + if (i >= ARRAY_SIZE(max_98390_4spk_components)) { + dev_err(codec_dai->dev, "invalid codec index %d\n", i); + return -ENODEV; + } + + if (!strcmp(codec_dai->component->name, MAX_98390_DEV0_NAME)) { + /* DEV0 tdm slot configuration Right */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x01, 3, 4, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV1_NAME)) { + /* DEV1 tdm slot configuration Left */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x02, 3, 4, 32); + } + + if (!strcmp(codec_dai->component->name, MAX_98390_DEV2_NAME)) { + /* DEVi2 tdm slot configuration Tweeter Right */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x04, 3, 4, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV3_NAME)) { + /* DEV3 tdm slot configuration Tweeter Left */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x08, 3, 4, 32); + } + } + return 0; +} + +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + /* add regular speakers dapm route */ + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_dapm_routes, + ARRAY_SIZE(max_98390_dapm_routes)); + if (ret) { + dev_err(rtd->dev, "unable to add Left/Right Speaker dapm, ret %d\n", ret); + return ret; + } + + /* add widgets/controls/dapm for tweeter speakers */ + if (acpi_dev_present("MX98390", "3", -1)) { + ret = snd_soc_dapm_new_controls(&card->dapm, max_98390_tt_dapm_widgets, + ARRAY_SIZE(max_98390_tt_dapm_widgets)); + + if (ret) { + dev_err(rtd->dev, "unable to add tweeter dapm controls, ret %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_add_card_controls(card, max_98390_tt_kcontrols, + ARRAY_SIZE(max_98390_tt_kcontrols)); + if (ret) { + dev_err(rtd->dev, "unable to add tweeter card controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_tt_dapm_routes, + ARRAY_SIZE(max_98390_tt_dapm_routes)); + if (ret) + dev_err(rtd->dev, + "unable to add Tweeter Left/Right Speaker dapm, ret %d\n", ret); + } + return ret; +} +EXPORT_SYMBOL_NS(max_98390_spk_codec_init, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +const struct snd_soc_ops max_98390_ops = { + .hw_params = max_98390_hw_params, +}; +EXPORT_SYMBOL_NS(max_98390_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch) +{ + if (ch == ARRAY_SIZE(max_98390_4spk_codec_conf)) { + card->codec_conf = max_98390_4spk_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_4spk_codec_conf); + } else { + card->codec_conf = max_98390_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_codec_conf); + } +} +EXPORT_SYMBOL_NS(max_98390_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); + /* * Maxim MAX98357A/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 3ff5e8fec4de..7a8c53049e4d 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -24,6 +24,22 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); void max_98373_set_codec_conf(struct snd_soc_card *card); int max_98373_trigger(struct snd_pcm_substream *substream, int cmd); +/* + * Maxim MAX98390 + */ +#define MAX_98390_CODEC_DAI "max98390-aif1" +#define MAX_98390_DEV0_NAME "i2c-MX98390:00" +#define MAX_98390_DEV1_NAME "i2c-MX98390:01" +#define MAX_98390_DEV2_NAME "i2c-MX98390:02" +#define MAX_98390_DEV3_NAME "i2c-MX98390:03" + +extern struct snd_soc_dai_link_component max_98390_components[2]; +extern struct snd_soc_dai_link_component max_98390_4spk_components[4]; +extern const struct snd_soc_ops max_98390_ops; + +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch); +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd); + /* * Maxim MAX98357A/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index c41f386b4138..6cadb5fb72e0 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -59,6 +59,9 @@ (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) #define SOF_SSP_BT_OFFLOAD_PRESENT BIT(22) #define SOF_RT5682S_HEADPHONE_CODEC_PRESENT BIT(23) +#define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) +#define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) + /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -179,6 +182,36 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S_4SPK"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_MAX98390_TWEETER_SPEAKER_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, {} }; @@ -486,6 +519,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) if (err < 0) return err; } + return hdac_hdmi_jack_port_init(component, &card->dapm); } @@ -784,6 +818,20 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) { sof_rt1011_dai_link(&links[id]); + } else if (sof_rt5682_quirk & + SOF_MAX98390_SPEAKER_AMP_PRESENT) { + if (sof_rt5682_quirk & + SOF_MAX98390_TWEETER_SPEAKER_PRESENT) { + links[id].codecs = max_98390_4spk_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_4spk_components); + } else { + links[id].codecs = max_98390_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_components); + } + links[id].init = max_98390_spk_codec_init; + links[id].ops = &max_98390_ops; + links[id].dpcm_capture = 1; + } else { max_98357a_dai_link(&links[id]); } @@ -868,6 +916,10 @@ static int sof_audio_probe(struct platform_device *pdev) if (acpi_dev_present("RTL5682", NULL, -1)) sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + /* Detect the headset codec variant to support machines in DMI quirk */ + if (acpi_dev_present("RTL5682", NULL, -1)) + sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; dmic_be_num = 0; @@ -924,6 +976,14 @@ static int sof_audio_probe(struct platform_device *pdev) sof_rt1011_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) sof_rt1015p_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_MAX98390_SPEAKER_AMP_PRESENT) { + if (sof_rt5682_quirk & SOF_MAX98390_TWEETER_SPEAKER_PRESENT) + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_4spk_components)); + else + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_components)); + } if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) sof_audio_card_rt5682.num_links++; @@ -1050,6 +1110,17 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "adl_max98390_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, { .name = "adl_mx98360_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | @@ -1080,6 +1151,7 @@ MODULE_DESCRIPTION("SOF Audio Machine driver"); MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index fde310e5724b..f32bcb2b2e09 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -379,6 +379,11 @@ static const struct snd_soc_acpi_codecs adl_rt1019p_amp = { .codecs = {"RTL1019"} }; +static const struct snd_soc_acpi_codecs adl_max98390_amp = { + .num_codecs = 1, + .codecs = {"MX98390"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .comp_ids = &adl_rt5682_rt5682s_hp, @@ -434,6 +439,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-nau8825.tplg", }, + { + .comp_ids = &adl_rt5682_rt5682s_hp, + .drv_name = "adl_max98390_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index bc9e70765678..be90a5f79766 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -59,6 +59,15 @@ static const struct dmi_system_id sof_tplg_table[] = { }, .driver_data = "sof-adl-rt5682-ssp0-max98373-ssp2.tplg", }, + { + .callback = sof_tplg_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), + }, + .driver_data = "sof-adl-max98390-ssp2-rt5682-ssp0.tplg", + }, + {} }; -- 2.20.1

3 2

[PATCH] ASoC: mediatek: mt8195: Constify static snd_soc_ops
by Rikard Falkeborn 01 Dec '21

01 Dec '21

These are only assigned to the ops field in the snd_soc_dai_link which is a pointer to const struct snd_soc_ops. Make them const to allow the compiler to put them in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c | 2 +- sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c index e103102d7ef6..9e6b54e19c23 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c @@ -421,7 +421,7 @@ static int mt8195_dptx_hw_params(struct snd_pcm_substream *substream, SND_SOC_CLOCK_OUT); } -static struct snd_soc_ops mt8195_dptx_ops = { +static const struct snd_soc_ops mt8195_dptx_ops = { .hw_params = mt8195_dptx_hw_params, }; diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index 95abaadcd842..e22e5fd40984 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -391,7 +391,7 @@ static int mt8195_dptx_hw_params(struct snd_pcm_substream *substream, SND_SOC_CLOCK_OUT); } -static struct snd_soc_ops mt8195_dptx_ops = { +static const struct snd_soc_ops mt8195_dptx_ops = { .hw_params = mt8195_dptx_hw_params, }; -- 2.34.1

2 1

[PATCH] ASoC: Intel: hda_dsp_common: don't multiline PCM topology warning
by Chris Down 01 Dec '21

01 Dec '21

On my T14s Gen2 I saw the following: [ 16.057258] skl_hda_dsp_generic skl_hda_dsp_generic: hda_dsp_hdmi_build_controls: no PCM in topology for HDMI converter 3 [ 16.057261] skl_hda_dsp_generic skl_hda_dsp_generic: hda_dsp_hdmi_build_controls: no PCM in topology for HDMI converter 4 [ 16.057263] skl_hda_dsp_generic skl_hda_dsp_generic: hda_dsp_hdmi_build_controls: no PCM in topology for HDMI converter 5 [...and so on.] It looks like the double newline is a mistake, so remove one. Signed-off-by: Chris Down <chris(a)chrisdown.name> --- sound/soc/intel/boards/hda_dsp_common.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/hda_dsp_common.c b/sound/soc/intel/boards/hda_dsp_common.c index efdc4bc4bb1f..5c31ddc0884a 100644 --- a/sound/soc/intel/boards/hda_dsp_common.c +++ b/sound/soc/intel/boards/hda_dsp_common.c @@ -68,7 +68,7 @@ int hda_dsp_hdmi_build_controls(struct snd_soc_card *card, hpcm->pcm = NULL; hpcm->device = SNDRV_PCM_INVALID_DEVICE; dev_warn(card->dev, - "%s: no PCM in topology for HDMI converter %d\n\n", + "%s: no PCM in topology for HDMI converter %d\n", __func__, i); } i++; -- 2.34.1

3 2

[PATCH] ASoC: intel: boards: bytcht*: Constify static snd_soc_ops
by Rikard Falkeborn 01 Dec '21

01 Dec '21

These are only assigned to the ops fields in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make them const to allow the compiler to put them in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_nocodec.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index 0a736308052a..ffd497a5b5a5 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -147,7 +147,7 @@ static int byt_cht_cx2072x_aif1_startup(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_RATE, 48000); } -static struct snd_soc_ops byt_cht_cx2072x_aif1_ops = { +static const struct snd_soc_ops byt_cht_cx2072x_aif1_ops = { .startup = byt_cht_cx2072x_aif1_startup, }; diff --git a/sound/soc/intel/boards/bytcht_nocodec.c b/sound/soc/intel/boards/bytcht_nocodec.c index 67b3c4e97864..115c2bcaabd4 100644 --- a/sound/soc/intel/boards/bytcht_nocodec.c +++ b/sound/soc/intel/boards/bytcht_nocodec.c @@ -93,7 +93,7 @@ static int aif1_startup(struct snd_pcm_substream *substream) &constraints_48000); } -static struct snd_soc_ops aif1_ops = { +static const struct snd_soc_ops aif1_ops = { .startup = aif1_startup, }; -- 2.34.1

3 2

[PATCH v2] ASoC: qdsp6: q6routing: Fix return value from msm_routing_put_audio_mixer
by Srinivas Kandagatla 01 Dec '21

01 Dec '21

msm_routing_put_audio_mixer() can return incorrect value in various scenarios. scenario 1: amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 1 amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 0 return value is 0 instead of 1 eventhough value was changed scenario 2: amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 1 amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 1 return value is 1 instead of 0 eventhough the value was not changed scenario 3: amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 0 return value is 1 instead of 0 eventhough the value was not changed Fix this by adding checks, so that change notifications are sent correctly. Fixes: e3a33673e845 ("ASoC: qdsp6: q6routing: Add q6routing driver") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6routing.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index cd74681e811e..928fd23e2c27 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -498,14 +498,16 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, struct session_data *session = &data->sessions[session_id]; if (ucontrol->value.integer.value[0]) { + if (session->port_id == be_id) + return 0; + session->port_id = be_id; snd_soc_dapm_mixer_update_power(dapm, kcontrol, 1, update); } else { - if (session->port_id == be_id) { - session->port_id = -1; + if (session->port_id == -1 || session->port_id != be_id) return 0; - } + session->port_id = -1; snd_soc_dapm_mixer_update_power(dapm, kcontrol, 0, update); } -- 2.21.0

2 1

[PATCH 0/4] ASoC: codecs: Qualcomm codecs fix kcontrol put return values
by Srinivas Kandagatla 01 Dec '21

01 Dec '21

Hi Mark, Some recent testing found few issues with wcd934x and wsa881x codec drivers that are not handling kcontrol put correctly. This patchset fixes those instances. Along with this there is also a bug fix for the way channel list was updated for wcd934x dais. --srini Srinivas Kandagatla (4): ASoC: codecs: wcd934x: handle channel mappping list correctly ASoC: codecs: wcd934x: remove redundant ret variable ASoC: codecs: wcd934x: return correct value from mixer put ASoC: codecs: wsa881x: fix return values from kcontrol put sound/soc/codecs/wcd934x.c | 132 ++++++++++++++++++++++++++----------- sound/soc/codecs/wsa881x.c | 16 +++-- 2 files changed, 107 insertions(+), 41 deletions(-) -- 2.21.0

2 8

[PATCH] ASoC: amd: fix uninitialized variable in snd_acp6x_probe()
by Dan Carpenter 01 Dec '21

01 Dec '21

The "index" is potentially used without being initialized on the error path. Fixes: fc329c1de498 ("ASoC: amd: add platform devices for acp6x pdm driver and dmic driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/amd/yc/pci-acp6x.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/yc/pci-acp6x.c b/sound/soc/amd/yc/pci-acp6x.c index 957eeb6fb8e3..7e9a9a9d8ddd 100644 --- a/sound/soc/amd/yc/pci-acp6x.c +++ b/sound/soc/amd/yc/pci-acp6x.c @@ -146,10 +146,11 @@ static int snd_acp6x_probe(struct pci_dev *pci, { struct acp6x_dev_data *adata; struct platform_device_info pdevinfo[ACP6x_DEVS]; - int ret, index; + int index = 0; int val = 0x00; u32 addr; unsigned int irqflags; + int ret; irqflags = IRQF_SHARED; /* Yellow Carp device check */ -- 2.20.1

2 1

[PATCH] ASoC: SOF: Intel: Retry codec probing if it fails
by Kai Vehmanen 01 Dec '21

01 Dec '21

From: Hui Wang <hui.wang(a)canonical.com> On the latest Lenovo Thinkstation laptops, we often experience the speaker failure after rebooting, check the dmesg, we could see: sof-audio-pci-intel-tgl 0000:00:1f.3: codec #0 probe error, ret: -5 The analogue codec on the machine is ALC287, then we designed a testcase to reboot and check the codec probing result repeatedly, we found the analogue codec probing always failed at least once within several minutes to several hours (roughly 1 reboot per min). This issue happens on all laptops of this Thinkstation model, but with legacy HDA driver, we couldn't reproduce this issue on those laptops. And so far, this issue is not reproduced on machines which don't belong to this model. We tried to make the hda_dsp_ctrl_init_chip() same as hda_intel_init_chip() which is the controller init routine in the legacy HDA driver, but it didn't help. We found when issue happens, the resp is -1, and if we let driver re-run send_cmd() and get_response(), it will get the correct response 10ec0287, then driver continues the rest work, finally boot to the desktop and all audio function work well. Here adding codec probing retries to 3 times, it could fix the issue on this Thinkstation model, and it doesn't bring impact to other machines. Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/intel/hda-codec.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) Notes: - patch for-5.16 on top of for-5.16 8a724d5f60 diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 6744318de612..13cd96e6724a 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -22,6 +22,7 @@ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) #define IDISP_VID_INTEL 0x80860000 +#define CODEC_PROBE_RETRIES 3 /* load the legacy HDA codec driver */ static int request_codec_module(struct hda_codec *codec) @@ -121,12 +122,15 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; u32 resp = -1; - int ret; + int ret, retry = 0; + + do { + mutex_lock(&hbus->core.cmd_mutex); + snd_hdac_bus_send_cmd(&hbus->core, hda_cmd); + snd_hdac_bus_get_response(&hbus->core, address, &resp); + mutex_unlock(&hbus->core.cmd_mutex); + } while (resp == -1 && retry++ < CODEC_PROBE_RETRIES); - mutex_lock(&hbus->core.cmd_mutex); - snd_hdac_bus_send_cmd(&hbus->core, hda_cmd); - snd_hdac_bus_get_response(&hbus->core, address, &resp); - mutex_unlock(&hbus->core.cmd_mutex); if (resp == -1) return -EIO; dev_dbg(sdev->dev, "HDA codec #%d probed OK: response: %x\n", base-commit: 8a724d5f60904084e09de2685ff9e77bc8a40815 -- 2.34.1

2 1

[PATCH v3] ASoC: rockchip: i2s_tdm: Dup static DAI template
by Nicolas Frattaroli 01 Dec '21

01 Dec '21

Previously, the DAI template was used directly, which lead to fun bugs such as "why is my channels_max changing?" when one instantiated more than one i2s_tdm IP block in a device tree. This change makes it so that we instead duplicate the template struct, and then use that. Fixes: 081068fd6414 ("ASoC: rockchip: add support for i2s-tdm controller") Signed-off-by: Nicolas Frattaroli <frattaroli.nicolas(a)gmail.com> --- Changes in v3: - constify the i2s_tdm_dai struct - remove remaining direct i2s_tdm_dai uses - move symmetric_rate setting to _init_dai - store pointer to dai struct in rk_i2s_tdm_dev struct Changes in v2: - remove accidental whitespace changes sound/soc/rockchip/rockchip_i2s_tdm.c | 52 ++++++++++++++++----------- 1 file changed, 31 insertions(+), 21 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index 17b9b287853a..5f9cb5c4c7f0 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -95,6 +95,7 @@ struct rk_i2s_tdm_dev { spinlock_t lock; /* xfer lock */ bool has_playback; bool has_capture; + struct snd_soc_dai_driver *dai; }; static int to_ch_num(unsigned int val) @@ -1310,19 +1311,14 @@ static const struct of_device_id rockchip_i2s_tdm_match[] = { {}, }; -static struct snd_soc_dai_driver i2s_tdm_dai = { +static const struct snd_soc_dai_driver i2s_tdm_dai = { .probe = rockchip_i2s_tdm_dai_probe, - .playback = { - .stream_name = "Playback", - }, - .capture = { - .stream_name = "Capture", - }, .ops = &rockchip_i2s_tdm_dai_ops, }; -static void rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) +static int rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) { + struct snd_soc_dai_driver *dai; struct property *dma_names; const char *dma_name; u64 formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | @@ -1337,19 +1333,33 @@ static void rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) i2s_tdm->has_capture = true; } + dai = devm_kmemdup(i2s_tdm->dev, &i2s_tdm_dai, + sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + if (i2s_tdm->has_playback) { - i2s_tdm_dai.playback.channels_min = 2; - i2s_tdm_dai.playback.channels_max = 8; - i2s_tdm_dai.playback.rates = SNDRV_PCM_RATE_8000_192000; - i2s_tdm_dai.playback.formats = formats; + dai->playback.stream_name = "Playback"; + dai->playback.channels_min = 2; + dai->playback.channels_max = 8; + dai->playback.rates = SNDRV_PCM_RATE_8000_192000; + dai->playback.formats = formats; } if (i2s_tdm->has_capture) { - i2s_tdm_dai.capture.channels_min = 2; - i2s_tdm_dai.capture.channels_max = 8; - i2s_tdm_dai.capture.rates = SNDRV_PCM_RATE_8000_192000; - i2s_tdm_dai.capture.formats = formats; + dai->capture.stream_name = "Capture"; + dai->capture.channels_min = 2; + dai->capture.channels_max = 8; + dai->capture.rates = SNDRV_PCM_RATE_8000_192000; + dai->capture.formats = formats; } + + if (i2s_tdm->clk_trcm != TRCM_TXRX) + dai->symmetric_rate = 1; + + i2s_tdm->dai = dai; + + return 0; } static int rockchip_i2s_tdm_path_check(struct rk_i2s_tdm_dev *i2s_tdm, @@ -1541,8 +1551,6 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev) spin_lock_init(&i2s_tdm->lock); i2s_tdm->soc_data = (struct rk_i2s_soc_data *)of_id->data; - rockchip_i2s_tdm_init_dai(i2s_tdm); - i2s_tdm->frame_width = 64; i2s_tdm->clk_trcm = TRCM_TXRX; @@ -1555,8 +1563,10 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev) } i2s_tdm->clk_trcm = TRCM_RX; } - if (i2s_tdm->clk_trcm != TRCM_TXRX) - i2s_tdm_dai.symmetric_rate = 1; + + ret = rockchip_i2s_tdm_init_dai(i2s_tdm); + if (ret) + return ret; i2s_tdm->grf = syscon_regmap_lookup_by_phandle(node, "rockchip,grf"); if (IS_ERR(i2s_tdm->grf)) @@ -1678,7 +1688,7 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(&pdev->dev, &rockchip_i2s_tdm_component, - &i2s_tdm_dai, 1); + i2s_tdm->dai, 1); if (ret) { dev_err(&pdev->dev, "Could not register DAI\n"); -- 2.34.0

2 1

[PATCH] ASoC: rockchip: i2s_tdm: Dup static DAI template
by Nicolas Frattaroli 01 Dec '21

01 Dec '21

Previously, the DAI template was used directly, which lead to fun bugs such as "why is my channels_max changing?" when one instantiated more than one i2s_tdm IP block in a device tree. This change makes it so that we instead duplicate the template struct, and then use that. Fixes: 081068fd6414 ("ASoC: rockchip: add support for i2s-tdm controller") Signed-off-by: Nicolas Frattaroli <frattaroli.nicolas(a)gmail.com> --- sound/soc/rockchip/rockchip_i2s_tdm.c | 42 +++++++++++++++------------ 1 file changed, 24 insertions(+), 18 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index 17b9b287853a..e9ddbbf4563e 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -1312,22 +1312,17 @@ static const struct of_device_id rockchip_i2s_tdm_match[] = { static struct snd_soc_dai_driver i2s_tdm_dai = { .probe = rockchip_i2s_tdm_dai_probe, - .playback = { - .stream_name = "Playback", - }, - .capture = { - .stream_name = "Capture", - }, .ops = &rockchip_i2s_tdm_dai_ops, }; -static void rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) +static int rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) { + struct snd_soc_dai_driver *dai; struct property *dma_names; const char *dma_name; u64 formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE); + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE); struct device_node *node = i2s_tdm->dev->of_node; of_property_for_each_string(node, "dma-names", dma_names, dma_name) { @@ -1337,19 +1332,28 @@ static void rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) i2s_tdm->has_capture = true; } + dai = devm_kmemdup(i2s_tdm->dev, &i2s_tdm_dai, + sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + if (i2s_tdm->has_playback) { - i2s_tdm_dai.playback.channels_min = 2; - i2s_tdm_dai.playback.channels_max = 8; - i2s_tdm_dai.playback.rates = SNDRV_PCM_RATE_8000_192000; - i2s_tdm_dai.playback.formats = formats; + dai->playback.stream_name = "Playback"; + dai->playback.channels_min = 2; + dai->playback.channels_max = 8; + dai->playback.rates = SNDRV_PCM_RATE_8000_192000; + dai->playback.formats = formats; } if (i2s_tdm->has_capture) { - i2s_tdm_dai.capture.channels_min = 2; - i2s_tdm_dai.capture.channels_max = 8; - i2s_tdm_dai.capture.rates = SNDRV_PCM_RATE_8000_192000; - i2s_tdm_dai.capture.formats = formats; + dai->capture.stream_name = "Capture"; + dai->capture.channels_min = 2; + dai->capture.channels_max = 8; + dai->capture.rates = SNDRV_PCM_RATE_8000_192000; + dai->capture.formats = formats; } + + return 0; } static int rockchip_i2s_tdm_path_check(struct rk_i2s_tdm_dev *i2s_tdm, @@ -1541,7 +1545,9 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev) spin_lock_init(&i2s_tdm->lock); i2s_tdm->soc_data = (struct rk_i2s_soc_data *)of_id->data; - rockchip_i2s_tdm_init_dai(i2s_tdm); + ret = rockchip_i2s_tdm_init_dai(i2s_tdm); + if (ret) + return ret; i2s_tdm->frame_width = 64; -- 2.34.0

2 1

What is "DMAR: PTE Read access is not set"?
by Mark Hills 01 Dec '21

01 Dec '21

Recently, seeing from the kernel when my audio device (Layla3G, PCI) is open: [ 2333.598177] DMAR: [DMA Read NO_PASID] Request device [0a:00.0] fault addr 0xfffe0000 [fault reason 0x06] PTE Read access is not set [ 2333.768836] DMAR: DRHD: handling fault status reg 3 [ 2338.376854] dmar_fault: 80 callbacks suppressed I'm not really sure where to begin here; does it mean that snd-echo3g should be making some up-front declaration of how it uses some DMA area? Seems to have appeared in the 5.14.x series (currently 5.14.16) but also I did a BIOS update in that time for another issue. 5.15.6 (latest stable) is also still affected. Audio plays fine; but my logs are full of these excess messages. -- Mark

1 0

[PATCH v3 0/5] Add pin control support for lpass sc7280
by Srinivasa Rao Mandadapu 01 Dec '21

01 Dec '21

This patch series is to split lpass variant common pin control functions and SoC specific functions and to add lpass sc7280 pincontrol support. It also Adds dt-bindings for lpass sc7280 lpass lpi pincontrol. Changes Since V2: -- Add new dt-bindings for sc7280 lpi driver. -- Make clock voting change as separate patch. -- Split existing pincontrol driver and make common functions as part of separate file. -- Rename lpass pincontrol lpi dt-bindings to sm8250 specific dt-bindings Changes Since V1: -- Make lpi pinctrl variant data structure as constant -- Add appropriate commit message -- Change signedoff by sequence. Srinivasa Rao Mandadapu (5): dt-bindings: pinctrl: qcom: Update lpass lpi file name to SoC specific dt-bindings: pinctrl: qcom: Add sc7280 lpass lpi pinctrl bindings pinctrl: qcom: Move chip specific functions to right files pinctrl: qcom: Update clock voting as optional pinctrl: qcom: Add SC7280 lpass pin configuration .../bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml | 130 ----------- .../pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 115 ++++++++++ .../pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml | 130 +++++++++++ drivers/pinctrl/qcom/Makefile | 2 + drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 253 +-------------------- drivers/pinctrl/qcom/pinctrl-lpass-lpi.h | 140 ++++++++++++ drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 127 +++++++++++ drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c | 125 ++++++++++ 8 files changed, 646 insertions(+), 376 deletions(-) delete mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml create mode 100644 drivers/pinctrl/qcom/pinctrl-lpass-lpi.h create mode 100644 drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c create mode 100644 drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH v6 1/2] dt-bindings: sound: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 01 Dec '21

01 Dec '21

DT bindings for tlv320adc3xxx driver, currently supporting Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 ++++++++++++++++++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 ++++ 2 files changed, 165 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..c4fed6335230 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw(a)axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + reset-gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,micbias1-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS1 pin + + ti,micbias2-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: tlv320adc3101@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/include/dt-bindings/sound/tlv320adc3xxx.h b/include/dt-bindings/sound/tlv320adc3xxx.h new file mode 100644 index 000000000000..3b3fa43fa961 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Devicetree bindings definitions for tlv320adc3xxx driver. + * + * Copyright (C) 2021 Axis Communications AB + */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +#define ADC3XXX_GPIO_DISABLED 0 /* I/O buffers powered down */ +#define ADC3XXX_GPIO_INPUT 1 /* Various non-GPIO inputs */ +#define ADC3XXX_GPIO_GPI 2 /* General purpose input */ +#define ADC3XXX_GPIO_GPO 3 /* General purpose output */ +#define ADC3XXX_GPIO_CLKOUT 4 /* Source set in reg. CLKOUT_MUX */ +#define ADC3XXX_GPIO_INT1 5 /* INT1 output */ +#define ADC3XXX_GPIO_INT2 6 /* INT2 output */ +/* value 7 is reserved */ +#define ADC3XXX_GPIO_SECONDARY_BCLK 8 /* Codec interface secondary BCLK */ +#define ADC3XXX_GPIO_SECONDARY_WCLK 9 /* Codec interface secondary WCLK */ +#define ADC3XXX_GPIO_ADC_MOD_CLK 10 /* Clock output for digital mics */ +/* values 11-15 reserved */ + +#define ADC3XXX_MICBIAS_OFF 0 /* Micbias pin powered off */ +#define ADC3XXX_MICBIAS_2_0V 1 /* Micbias pin set to 2.0V */ +#define ADC3XXX_MICBIAS_2_5V 2 /* Micbias pin set to 2.5V */ +#define ADC3XXX_MICBIAS_AVDD 3 /* Use AVDD voltage for micbias pin */ + +#endif /* __DT_TLV320ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 2

[PATCH v6 0/2] Codec driver for TLV320ADC3001 and 3101
by Ricard Wanderlof 01 Dec '21

01 Dec '21

I've submitted this a couple of times, but the kernel test robot has come back with various remarks, so I've reworked it a bit more hopefully with positive results this time. This is a codec driver for the TLV320ADC3001 and TLV320ADC3101 2 channel audio ADC chips from Texas Instruments. Based on an old driver for these chips that I was referred to by Texas Instruments, but which apparently was never upstreamed, I've taken patches from various incarnations, ported them and augmented them with various new features in order to make a working driver. There is currently no support for the on-chip "miniDSP". Although the TLV320ADC3001 and TLV320ADC3101 have similarities with other TI codec chips for which there are existing drivers, upon closer inspection I decided the differences were too great to warrant attempting to modify an existing driver, especially without access to all supported chips and resources for testing them all. There are certainly things that could be improved, among others in the area of PLL register value calculations, but it has been tested on the Axis ARTPEC-8 platform and has been found to work satisfactorily at least with a TLV320ADC3101 operating as an I2S slave at 48 kHz with 32 bit bit depth, 1 or 2 channels, both with the on-chip PLL enabled and disabled. I have not tested this on a TLV320ADC3001 chip; in the original driver the only differences were related to the now-removed initialization code, so I've just retained the compatible strings for both chips. Limitations: - GPIO pins on chip can only be set up as outputs. - Using BCLK as an input to clock chain is supported by the hw but not by driver. Changes v5->v6: - Added missing "depends on I2C" in the Kconfig (reported by lkp(a)intel.com). - Fold include file into source as it is not needed elsewhere. - Fix SPDX-License-Identifier which was using a deprecated license. - Clean up authorship information in file header, removing history. - Remove useless GPIOLIB ifdefs. - Use devm_gpiod_get for reset gpio rather than devm_gpio_request etc. - Handle EPROBE_DEFER properly for reset gpio request. - Use dev_err_probe rather than explicit EPROBE_DEFER checking. - Properly check value from DT parsing for mic bias and codec gpio pins. No changes to the devicetree bindings. Changes v4->v5: Yet another fix based on kernel test robot (lkp(a)intel.com), this time a legitimate issue: the return value from of_get_named_gpio_flags() was crammed into an unsigned int so any potential negative error codes were lost. No changes to the devicetree bindings. Changes v3->v4: More fixes reported by kernel test robot (lkp(a)intel.com). Don't know why these weren't reported the first time. Changes v2->v3: Fix a couple of issues reported by the kernel test robot (lkp(a)intel.com). Changes v1->v2: Large portions of the driver have been reworked from v1 based on remarks from Mark. More specifically: - Remove an unneeded #ifdef and EXPORT_SYMBOL_GPL. - Add ADC3XXX_ prefix to all macro names to avoid namespace contention. - Clean up certain macro names with mixed lower case - upper case names. - Do away with the D0..D7 macros representing bit values. - Fix general formatting: spaces before and after {, }, replace multi spaces with tabs, maximum 2 blank lines anywhere, and other minor issues. - Page register does not need to be volatile. - Rather than use 0 / -6dB enum for input attenuator, use standard TLV. - The dither offset control contains reserved values so use VALUE_ENUM instead of straight ENUM. - Replace vector of SOC_ENUMs with individual variables. - Fix ALSA control names: - Use Volume at the end of all gain controls, and remove "Gain" where applicable. - Use "Switch" at the end of all switch controls, either adding it or changing "switch" into "Switch" as applicable. - Switches and Volumes related to capture should be set up as such, so add "Capture" to widget names where appropriate. This is mostly important for the PGA Capture and Switch widgets, not just for consistency, but also because amixer simple Switch controls expect different names (cap/nocap instead of on/off) for control values for capture. - Make widget names have capital first letters for all consitutent words. - When resuming, the registers will not have changed, so no need to call snd_soc_component_cache_sync(). - Move reset gpio init to I2C probe - Remove needless _priv at end of private struct name. - Make hex constants lower case - _CBM_CFM/_CBS_CFS -> _CBP_CFP/_CBC_CFC - Instead of using device specific ALSA controls to control the two GPIO pins, use GPIOLIB. GPIO pin functions still can be set up in the devicetree as the pins can be used for other functions which are not GPIO related. - Limitation: The GPIO pins can only be used as outputs, even though the hardware alternatively supports using them as inputs. - Handle GPIO configs in a vector rather than separately. - Use more appropriate module_i2c_driver instead of generic module_init/module_exit. - Remove useless initialization data. - As the ADC mute function had no control, add it. - Clean up GPIO initialization which was done both in the I2C probe and in the component probe. - Remove probe, remove, suspend and resume callbacks, which either do not do anything at all anymore, or uselessly just do set_bias_level. - The PLL should not be configured in the devicetree, so move the corresponding definitions to tlv320adc3xxx.h where they can be picked up by a machine driver if needed. - Default to PLL AUTO mode. - Use "reset-gpios" instead of too-generic "gpios" for dt property for configuring codec reset pin. - Use function for printing PLL mode rather than macro. - dev_info is too verbose for most of the printouts, change them to dev_dbg. - Add support for configuring micbias voltage in DT. - Remove ALSA control for mic bias. - Move PLL and NADC, MADC divisor enable to DAPM, including 10 ms post PLL start up delay via POST_PMU event. - ADC is powered on using DAPM, so needless to do it in set_bias_level, thus remove it. - Let DAPM handle BCLK divisor enable. With this, we can remove set_bias_level as it doesn't do anything anymore. - Fix incorrect argument order to a snd_soc_component_update_bits call. - Although the chip itself can operate using BCLK as the clock source, either directly to the divider chain or via the PLL, this is not yet supported in the driver, so require the clocks property for the MCLK for now. - Move PLL rate structure out of .h file where it does not belong. /Ricard Ricard Wanderlof (2): dt-bindings: sound: tlv320adc3xxx: New codec driver ASoC: codec: tlv320adc3xxx: New codec driver .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 ++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1311 +++++++++++++++++ 5 files changed, 1486 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h create mode 100644 sound/soc/codecs/tlv320adc3xxx.c -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 2

[PATCH v6 2/2] ASoC: codec: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 01 Dec '21

01 Dec '21

New codec driver for Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1311 ++++++++++++++++++++++++++++++ 3 files changed, 1321 insertions(+) create mode 100644 sound/soc/codecs/tlv320adc3xxx.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b4f70e27342c..d638113c5988 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -220,6 +220,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TDA7419 imply SND_SOC_TFA9879 imply SND_SOC_TFA989X + imply SND_SOC_TLV320ADC3XXX imply SND_SOC_TLV320ADCX140 imply SND_SOC_TLV320AIC23_I2C imply SND_SOC_TLV320AIC23_SPI @@ -1494,6 +1495,13 @@ config SND_SOC_TFA989X Note that the driver currently bypasses the built-in "CoolFlux DSP" and does not support (hardware) volume control. +config SND_SOC_TLV320ADC3XXX + tristate "Texas Instruments TLV320ADC3001/3101 audio ADC" + depends on I2C + help + Enable support for Texas Instruments TLV320ADC3001 and TLV320ADC3101 + ADCs. + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 485eee75502b..24bc6b34ba2f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -239,6 +239,7 @@ snd-soc-tda7419-objs := tda7419.o snd-soc-tas2770-objs := tas2770.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o +snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o @@ -574,6 +575,7 @@ obj-$(CONFIG_SND_SOC_TDA7419) += snd-soc-tda7419.o obj-$(CONFIG_SND_SOC_TAS2770) += snd-soc-tas2770.o obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TFA989X) += snd-soc-tfa989x.o +obj-$(CONFIG_SND_SOC_TLV320ADC3XXX) += snd-soc-tlv320adc3xxx.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c new file mode 100644 index 000000000000..fc4b75e0d944 --- /dev/null +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -0,0 +1,1311 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Based on sound/soc/codecs/tlv320aic3x.c by Vladimir Barinov +// +// Copyright (C) 2010 Mistral Solutions Pvt Ltd. +// Author: Shahina Shaik <shahina.s(a)mistralsolutions.com> +// +// Copyright (C) 2014-2018, Ambarella, Inc. +// Author: Dongge wu <dgwu(a)ambarella.com> +// +// Copyright (C) 2021 Axis Communications AB +// Author: Ricard Wanderlof <ricardw(a)axis.com> +// + +#include <dt-bindings/sound/tlv320adc3xxx.h> +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio/driver.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/cdev.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> + +/* + * General definitions defining exported functionality. + */ + +#define ADC3XXX_MICBIAS_PINS 2 + +/* Number of GPIO pins exposed via the gpiolib interface */ +#define ADC3XXX_GPIOS_MAX 2 + +#define ADC3XXX_RATES SNDRV_PCM_RATE_8000_96000 +#define ADC3XXX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* + * PLL modes, to be used for clk_id for set_sysclk callback. + * + * The default behavior (AUTO) is to take the first matching entry in the clock + * table, which is intended to be the PLL based one if there is more than one. + * + * Setting the clock source using simple-card (clocks or + * system-clock-frequency property) sets clk_id = 0 = ADC3XXX_PLL_AUTO. + */ +#define ADC3XXX_PLL_AUTO 0 /* Use first available mode */ +#define ADC3XXX_PLL_ENABLE 1 /* Use PLL for clock generation */ +#define ADC3XXX_PLL_BYPASS 2 /* Don't use PLL for clock generation */ + +/* Register definitions. */ + +#define ADC3XXX_PAGE_SIZE 128 +#define ADC3XXX_REG(page, reg) ((page * ADC3XXX_PAGE_SIZE) + reg) + +/* + * Page 0 registers. + */ + +#define ADC3XXX_PAGE_SELECT ADC3XXX_REG(0, 0) +#define ADC3XXX_RESET ADC3XXX_REG(0, 1) + +/* 2-3 Reserved */ + +#define ADC3XXX_CLKGEN_MUX ADC3XXX_REG(0, 4) +#define ADC3XXX_PLL_PROG_PR ADC3XXX_REG(0, 5) +#define ADC3XXX_PLL_PROG_J ADC3XXX_REG(0, 6) +#define ADC3XXX_PLL_PROG_D_MSB ADC3XXX_REG(0, 7) +#define ADC3XXX_PLL_PROG_D_LSB ADC3XXX_REG(0, 8) + +/* 9-17 Reserved */ + +#define ADC3XXX_ADC_NADC ADC3XXX_REG(0, 18) +#define ADC3XXX_ADC_MADC ADC3XXX_REG(0, 19) +#define ADC3XXX_ADC_AOSR ADC3XXX_REG(0, 20) +#define ADC3XXX_ADC_IADC ADC3XXX_REG(0, 21) + +/* 23-24 Reserved */ + +#define ADC3XXX_CLKOUT_MUX ADC3XXX_REG(0, 25) +#define ADC3XXX_CLKOUT_M_DIV ADC3XXX_REG(0, 26) +#define ADC3XXX_INTERFACE_CTRL_1 ADC3XXX_REG(0, 27) +#define ADC3XXX_CH_OFFSET_1 ADC3XXX_REG(0, 28) +#define ADC3XXX_INTERFACE_CTRL_2 ADC3XXX_REG(0, 29) +#define ADC3XXX_BCLK_N_DIV ADC3XXX_REG(0, 30) +#define ADC3XXX_INTERFACE_CTRL_3 ADC3XXX_REG(0, 31) +#define ADC3XXX_INTERFACE_CTRL_4 ADC3XXX_REG(0, 32) +#define ADC3XXX_INTERFACE_CTRL_5 ADC3XXX_REG(0, 33) +#define ADC3XXX_I2S_SYNC ADC3XXX_REG(0, 34) +/* 35 Reserved */ +#define ADC3XXX_ADC_FLAG ADC3XXX_REG(0, 36) +#define ADC3XXX_CH_OFFSET_2 ADC3XXX_REG(0, 37) +#define ADC3XXX_I2S_TDM_CTRL ADC3XXX_REG(0, 38) +/* 39-41 Reserved */ +#define ADC3XXX_INTR_FLAG_1 ADC3XXX_REG(0, 42) +#define ADC3XXX_INTR_FLAG_2 ADC3XXX_REG(0, 43) +/* 44 Reserved */ +#define ADC3XXX_INTR_FLAG_ADC1 ADC3XXX_REG(0, 45) +/* 46 Reserved */ +#define ADC3XXX_INTR_FLAG_ADC2 ADC3XXX_REG(0, 47) +#define ADC3XXX_INT1_CTRL ADC3XXX_REG(0, 48) +#define ADC3XXX_INT2_CTRL ADC3XXX_REG(0, 49) +/* 50 Reserved */ +#define ADC3XXX_GPIO2_CTRL ADC3XXX_REG(0, 51) +#define ADC3XXX_GPIO1_CTRL ADC3XXX_REG(0, 52) +#define ADC3XXX_DOUT_CTRL ADC3XXX_REG(0, 53) +/* 54-56 Reserved */ +#define ADC3XXX_SYNC_CTRL_1 ADC3XXX_REG(0, 57) +#define ADC3XXX_SYNC_CTRL_2 ADC3XXX_REG(0, 58) +#define ADC3XXX_CIC_GAIN_CTRL ADC3XXX_REG(0, 59) +/* 60 Reserved */ +#define ADC3XXX_PRB_SELECT ADC3XXX_REG(0, 61) +#define ADC3XXX_INST_MODE_CTRL ADC3XXX_REG(0, 62) +/* 63-79 Reserved */ +#define ADC3XXX_MIC_POLARITY_CTRL ADC3XXX_REG(0, 80) +#define ADC3XXX_ADC_DIGITAL ADC3XXX_REG(0, 81) +#define ADC3XXX_ADC_FGA ADC3XXX_REG(0, 82) +#define ADC3XXX_LADC_VOL ADC3XXX_REG(0, 83) +#define ADC3XXX_RADC_VOL ADC3XXX_REG(0, 84) +#define ADC3XXX_ADC_PHASE_COMP ADC3XXX_REG(0, 85) +#define ADC3XXX_LEFT_CHN_AGC_1 ADC3XXX_REG(0, 86) +#define ADC3XXX_LEFT_CHN_AGC_2 ADC3XXX_REG(0, 87) +#define ADC3XXX_LEFT_CHN_AGC_3 ADC3XXX_REG(0, 88) +#define ADC3XXX_LEFT_CHN_AGC_4 ADC3XXX_REG(0, 89) +#define ADC3XXX_LEFT_CHN_AGC_5 ADC3XXX_REG(0, 90) +#define ADC3XXX_LEFT_CHN_AGC_6 ADC3XXX_REG(0, 91) +#define ADC3XXX_LEFT_CHN_AGC_7 ADC3XXX_REG(0, 92) +#define ADC3XXX_LEFT_AGC_GAIN ADC3XXX_REG(0, 93) +#define ADC3XXX_RIGHT_CHN_AGC_1 ADC3XXX_REG(0, 94) +#define ADC3XXX_RIGHT_CHN_AGC_2 ADC3XXX_REG(0, 95) +#define ADC3XXX_RIGHT_CHN_AGC_3 ADC3XXX_REG(0, 96) +#define ADC3XXX_RIGHT_CHN_AGC_4 ADC3XXX_REG(0, 97) +#define ADC3XXX_RIGHT_CHN_AGC_5 ADC3XXX_REG(0, 98) +#define ADC3XXX_RIGHT_CHN_AGC_6 ADC3XXX_REG(0, 99) +#define ADC3XXX_RIGHT_CHN_AGC_7 ADC3XXX_REG(0, 100) +#define ADC3XXX_RIGHT_AGC_GAIN ADC3XXX_REG(0, 101) +/* 102-127 Reserved */ + +/* + * Page 1 registers. + */ + +/* 1-25 Reserved */ +#define ADC3XXX_DITHER_CTRL ADC3XXX_REG(1, 26) +/* 27-50 Reserved */ +#define ADC3XXX_MICBIAS_CTRL ADC3XXX_REG(1, 51) +#define ADC3XXX_LEFT_PGA_SEL_1 ADC3XXX_REG(1, 52) +/* 53 Reserved */ +#define ADC3XXX_LEFT_PGA_SEL_2 ADC3XXX_REG(1, 54) +#define ADC3XXX_RIGHT_PGA_SEL_1 ADC3XXX_REG(1, 55) +#define ADC3XXX_RIGHT_PGA_SEL_2 ADC3XXX_REG(1, 57) +#define ADC3XXX_LEFT_APGA_CTRL ADC3XXX_REG(1, 59) +#define ADC3XXX_RIGHT_APGA_CTRL ADC3XXX_REG(1, 60) +#define ADC3XXX_LOW_CURRENT_MODES ADC3XXX_REG(1, 61) +#define ADC3XXX_ANALOG_PGA_FLAGS ADC3XXX_REG(1, 62) +/* 63-127 Reserved */ + +/* + * Register bits. + */ + +/* PLL Enable bits */ +#define ADC3XXX_ENABLE_PLL_SHIFT 7 +#define ADC3XXX_ENABLE_PLL (1 << ADC3XXX_ENABLE_PLL_SHIFT) +#define ADC3XXX_ENABLE_NADC_SHIFT 7 +#define ADC3XXX_ENABLE_NADC (1 << ADC3XXX_ENABLE_NADC_SHIFT) +#define ADC3XXX_ENABLE_MADC_SHIFT 7 +#define ADC3XXX_ENABLE_MADC (1 << ADC3XXX_ENABLE_MADC_SHIFT) +#define ADC3XXX_ENABLE_BCLK_SHIFT 7 +#define ADC3XXX_ENABLE_BCLK (1 << ADC3XXX_ENABLE_BCLK_SHIFT) + +/* Power bits */ +#define ADC3XXX_LADC_PWR_ON 0x80 +#define ADC3XXX_RADC_PWR_ON 0x40 + +#define ADC3XXX_SOFT_RESET 0x01 +#define ADC3XXX_BCLK_MASTER 0x08 +#define ADC3XXX_WCLK_MASTER 0x04 + +/* Interface register masks */ +#define ADC3XXX_FORMAT_MASK 0xc0 +#define ADC3XXX_FORMAT_SHIFT 6 +#define ADC3XXX_WLENGTH_MASK 0x30 +#define ADC3XXX_WLENGTH_SHIFT 4 +#define ADC3XXX_CLKDIR_MASK 0x0c +#define ADC3XXX_CLKDIR_SHIFT 2 + +/* Interface register bit patterns */ +#define ADC3XXX_FORMAT_I2S (0 << ADC3XXX_FORMAT_SHIFT) +#define ADC3XXX_FORMAT_DSP (1 << ADC3XXX_FORMAT_SHIFT) +#define ADC3XXX_FORMAT_RJF (2 << ADC3XXX_FORMAT_SHIFT) +#define ADC3XXX_FORMAT_LJF (3 << ADC3XXX_FORMAT_SHIFT) + +#define ADC3XXX_IFACE_16BITS (0 << ADC3XXX_WLENGTH_SHIFT) +#define ADC3XXX_IFACE_20BITS (1 << ADC3XXX_WLENGTH_SHIFT) +#define ADC3XXX_IFACE_24BITS (2 << ADC3XXX_WLENGTH_SHIFT) +#define ADC3XXX_IFACE_32BITS (3 << ADC3XXX_WLENGTH_SHIFT) + +/* PLL P/R bit offsets */ +#define ADC3XXX_PLLP_SHIFT 4 +#define ADC3XXX_PLLR_SHIFT 0 +#define ADC3XXX_PLL_PR_MASK 0x7f +#define ADC3XXX_PLLJ_MASK 0x3f +#define ADC3XXX_PLLD_MSB_MASK 0x3f +#define ADC3XXX_PLLD_LSB_MASK 0xff +#define ADC3XXX_NADC_MASK 0x7f +#define ADC3XXX_MADC_MASK 0x7f +#define ADC3XXX_AOSR_MASK 0xff +#define ADC3XXX_IADC_MASK 0xff +#define ADC3XXX_BDIV_MASK 0x7f + +/* PLL_CLKIN bits */ +#define ADC3XXX_PLL_CLKIN_SHIFT 2 +#define ADC3XXX_PLL_CLKIN_MCLK 0x0 +#define ADC3XXX_PLL_CLKIN_BCLK 0x1 +#define ADC3XXX_PLL_CLKIN_ZERO 0x3 + +/* CODEC_CLKIN bits */ +#define ADC3XXX_CODEC_CLKIN_SHIFT 0 +#define ADC3XXX_CODEC_CLKIN_MCLK 0x0 +#define ADC3XXX_CODEC_CLKIN_BCLK 0x1 +#define ADC3XXX_CODEC_CLKIN_PLL_CLK 0x3 + +#define ADC3XXX_USE_PLL ((ADC3XXX_PLL_CLKIN_MCLK << ADC3XXX_PLL_CLKIN_SHIFT) | \ + (ADC3XXX_CODEC_CLKIN_PLL_CLK << ADC3XXX_CODEC_CLKIN_SHIFT)) +#define ADC3XXX_NO_PLL ((ADC3XXX_PLL_CLKIN_ZERO << ADC3XXX_PLL_CLKIN_SHIFT) | \ + (ADC3XXX_CODEC_CLKIN_MCLK << ADC3XXX_CODEC_CLKIN_SHIFT)) + +/* Analog PGA control bits */ +#define ADC3XXX_LPGA_MUTE 0x80 +#define ADC3XXX_RPGA_MUTE 0x80 + +#define ADC3XXX_LPGA_GAIN_MASK 0x7f +#define ADC3XXX_RPGA_GAIN_MASK 0x7f + +/* ADC current modes */ +#define ADC3XXX_ADC_LOW_CURR_MODE 0x01 + +/* Left ADC Input selection bits */ +#define ADC3XXX_LCH_SEL1_SHIFT 0 +#define ADC3XXX_LCH_SEL2_SHIFT 2 +#define ADC3XXX_LCH_SEL3_SHIFT 4 +#define ADC3XXX_LCH_SEL4_SHIFT 6 + +#define ADC3XXX_LCH_SEL1X_SHIFT 0 +#define ADC3XXX_LCH_SEL2X_SHIFT 2 +#define ADC3XXX_LCH_SEL3X_SHIFT 4 +#define ADC3XXX_LCH_COMMON_MODE 0x40 +#define ADC3XXX_BYPASS_LPGA 0x80 + +/* Right ADC Input selection bits */ +#define ADC3XXX_RCH_SEL1_SHIFT 0 +#define ADC3XXX_RCH_SEL2_SHIFT 2 +#define ADC3XXX_RCH_SEL3_SHIFT 4 +#define ADC3XXX_RCH_SEL4_SHIFT 6 + +#define ADC3XXX_RCH_SEL1X_SHIFT 0 +#define ADC3XXX_RCH_SEL2X_SHIFT 2 +#define ADC3XXX_RCH_SEL3X_SHIFT 4 +#define ADC3XXX_RCH_COMMON_MODE 0x40 +#define ADC3XXX_BYPASS_RPGA 0x80 + +/* MICBIAS control bits */ +#define ADC3XXX_MICBIAS_MASK 0x2 +#define ADC3XXX_MICBIAS1_SHIFT 5 +#define ADC3XXX_MICBIAS2_SHIFT 3 + +#define ADC3XXX_ADC_MAX_VOLUME 64 +#define ADC3XXX_ADC_POS_VOL 24 + +/* GPIO control bits (GPIO1_CTRL and GPIO2_CTRL) */ +#define ADC3XXX_GPIO_CTRL_CFG_MASK 0x3c +#define ADC3XXX_GPIO_CTRL_CFG_SHIFT 2 +#define ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_MASK 0x01 +#define ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_SHIFT 0 +#define ADC3XXX_GPIO_CTRL_INPUT_VALUE_MASK 0x02 +#define ADC3XXX_GPIO_CTRL_INPUT_VALUE_SHIFT 1 + +enum adc3xxx_type { + ADC3001 = 0, + ADC3101 +}; + +struct adc3xxx { + struct device *dev; + enum adc3xxx_type type; + struct clk *mclk; + struct regmap *regmap; + struct gpio_desc *rst_pin; + unsigned int pll_mode; + unsigned int sysclk; + unsigned int gpio_cfg[ADC3XXX_GPIOS_MAX]; /* value+1 (0 => not set) */ + unsigned int micbias_vg[ADC3XXX_MICBIAS_PINS]; + int master; + u8 page_no; + int use_pll; + struct gpio_chip gpio_chip; +}; + +static const unsigned int adc3xxx_gpio_ctrl_reg[ADC3XXX_GPIOS_MAX] = { + ADC3XXX_GPIO1_CTRL, + ADC3XXX_GPIO2_CTRL +}; + +static const unsigned int adc3xxx_micbias_shift[ADC3XXX_MICBIAS_PINS] = { + ADC3XXX_MICBIAS1_SHIFT, + ADC3XXX_MICBIAS2_SHIFT +}; + +static const struct reg_default adc3xxx_defaults[] = { + /* Page 0 */ + { 0, 0x00 }, { 1, 0x00 }, { 2, 0x00 }, { 3, 0x00 }, + { 4, 0x00 }, { 5, 0x11 }, { 6, 0x04 }, { 7, 0x00 }, + { 8, 0x00 }, { 9, 0x00 }, { 10, 0x00 }, { 11, 0x00 }, + { 12, 0x00 }, { 13, 0x00 }, { 14, 0x00 }, { 15, 0x00 }, + { 16, 0x00 }, { 17, 0x00 }, { 18, 0x01 }, { 19, 0x01 }, + { 20, 0x80 }, { 21, 0x80 }, { 22, 0x04 }, { 23, 0x00 }, + { 24, 0x00 }, { 25, 0x00 }, { 26, 0x01 }, { 27, 0x00 }, + { 28, 0x00 }, { 29, 0x02 }, { 30, 0x01 }, { 31, 0x00 }, + { 32, 0x00 }, { 33, 0x10 }, { 34, 0x00 }, { 35, 0x00 }, + { 36, 0x00 }, { 37, 0x00 }, { 38, 0x02 }, { 39, 0x00 }, + { 40, 0x00 }, { 41, 0x00 }, { 42, 0x00 }, { 43, 0x00 }, + { 44, 0x00 }, { 45, 0x00 }, { 46, 0x00 }, { 47, 0x00 }, + { 48, 0x00 }, { 49, 0x00 }, { 50, 0x00 }, { 51, 0x00 }, + { 52, 0x00 }, { 53, 0x12 }, { 54, 0x00 }, { 55, 0x00 }, + { 56, 0x00 }, { 57, 0x00 }, { 58, 0x00 }, { 59, 0x44 }, + { 60, 0x00 }, { 61, 0x01 }, { 62, 0x00 }, { 63, 0x00 }, + { 64, 0x00 }, { 65, 0x00 }, { 66, 0x00 }, { 67, 0x00 }, + { 68, 0x00 }, { 69, 0x00 }, { 70, 0x00 }, { 71, 0x00 }, + { 72, 0x00 }, { 73, 0x00 }, { 74, 0x00 }, { 75, 0x00 }, + { 76, 0x00 }, { 77, 0x00 }, { 78, 0x00 }, { 79, 0x00 }, + { 80, 0x00 }, { 81, 0x00 }, { 82, 0x88 }, { 83, 0x00 }, + { 84, 0x00 }, { 85, 0x00 }, { 86, 0x00 }, { 87, 0x00 }, + { 88, 0x7f }, { 89, 0x00 }, { 90, 0x00 }, { 91, 0x00 }, + { 92, 0x00 }, { 93, 0x00 }, { 94, 0x00 }, { 95, 0x00 }, + { 96, 0x7f }, { 97, 0x00 }, { 98, 0x00 }, { 99, 0x00 }, + { 100, 0x00 }, { 101, 0x00 }, { 102, 0x00 }, { 103, 0x00 }, + { 104, 0x00 }, { 105, 0x00 }, { 106, 0x00 }, { 107, 0x00 }, + { 108, 0x00 }, { 109, 0x00 }, { 110, 0x00 }, { 111, 0x00 }, + { 112, 0x00 }, { 113, 0x00 }, { 114, 0x00 }, { 115, 0x00 }, + { 116, 0x00 }, { 117, 0x00 }, { 118, 0x00 }, { 119, 0x00 }, + { 120, 0x00 }, { 121, 0x00 }, { 122, 0x00 }, { 123, 0x00 }, + { 124, 0x00 }, { 125, 0x00 }, { 126, 0x00 }, { 127, 0x00 }, + + /* Page 1 */ + { 128, 0x00 }, { 129, 0x00 }, { 130, 0x00 }, { 131, 0x00 }, + { 132, 0x00 }, { 133, 0x00 }, { 134, 0x00 }, { 135, 0x00 }, + { 136, 0x00 }, { 137, 0x00 }, { 138, 0x00 }, { 139, 0x00 }, + { 140, 0x00 }, { 141, 0x00 }, { 142, 0x00 }, { 143, 0x00 }, + { 144, 0x00 }, { 145, 0x00 }, { 146, 0x00 }, { 147, 0x00 }, + { 148, 0x00 }, { 149, 0x00 }, { 150, 0x00 }, { 151, 0x00 }, + { 152, 0x00 }, { 153, 0x00 }, { 154, 0x00 }, { 155, 0x00 }, + { 156, 0x00 }, { 157, 0x00 }, { 158, 0x00 }, { 159, 0x00 }, + { 160, 0x00 }, { 161, 0x00 }, { 162, 0x00 }, { 163, 0x00 }, + { 164, 0x00 }, { 165, 0x00 }, { 166, 0x00 }, { 167, 0x00 }, + { 168, 0x00 }, { 169, 0x00 }, { 170, 0x00 }, { 171, 0x00 }, + { 172, 0x00 }, { 173, 0x00 }, { 174, 0x00 }, { 175, 0x00 }, + { 176, 0x00 }, { 177, 0x00 }, { 178, 0x00 }, { 179, 0x00 }, + { 180, 0xff }, { 181, 0x00 }, { 182, 0x3f }, { 183, 0xff }, + { 184, 0x00 }, { 185, 0x3f }, { 186, 0x00 }, { 187, 0x80 }, + { 188, 0x80 }, { 189, 0x00 }, { 190, 0x00 }, { 191, 0x00 }, +}; + +static bool adc3xxx_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADC3XXX_RESET: + return true; + default: + return false; + } +} + +static const struct regmap_range_cfg adc3xxx_ranges[] = { + { + .range_min = 0, + .range_max = 2 * ADC3XXX_PAGE_SIZE, + .selector_reg = ADC3XXX_PAGE_SELECT, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = ADC3XXX_PAGE_SIZE, + } +}; + +static const struct regmap_config adc3xxx_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = adc3xxx_defaults, + .num_reg_defaults = ARRAY_SIZE(adc3xxx_defaults), + + .volatile_reg = adc3xxx_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + + .ranges = adc3xxx_ranges, + .num_ranges = ARRAY_SIZE(adc3xxx_ranges), + .max_register = 2 * ADC3XXX_PAGE_SIZE, +}; + +struct adc3xxx_rate_divs { + u32 mclk; + u32 rate; + u8 pll_p; + u8 pll_r; + u8 pll_j; + u16 pll_d; + u8 nadc; + u8 madc; + u8 aosr; +}; + +/* + * PLL and Clock settings. + * If p member is 0, PLL is not used. + * The order of the entries in this table have the PLL entries before + * the non-PLL entries, so that the PLL modes are preferred unless + * the PLL mode setting says otherwise. + */ +static const struct adc3xxx_rate_divs adc3xxx_divs[] = { + /* mclk, rate, p, r, j, d, nadc, madc, aosr */ + /* 8k rate */ + { 12000000, 8000, 1, 1, 7, 1680, 42, 2, 128 }, + { 12288000, 8000, 1, 1, 7, 0000, 42, 2, 128 }, + /* 11.025k rate */ + { 12000000, 11025, 1, 1, 6, 8208, 29, 2, 128 }, + /* 16k rate */ + { 12000000, 16000, 1, 1, 7, 1680, 21, 2, 128 }, + { 12288000, 16000, 1, 1, 7, 0000, 21, 2, 128 }, + /* 22.05k rate */ + { 12000000, 22050, 1, 1, 7, 560, 15, 2, 128 }, + /* 32k rate */ + { 12000000, 32000, 1, 1, 8, 1920, 12, 2, 128 }, + { 12288000, 32000, 1, 1, 8, 0000, 12, 2, 128 }, + /* 44.1k rate */ + { 12000000, 44100, 1, 1, 7, 5264, 8, 2, 128 }, + /* 48k rate */ + { 12000000, 48000, 1, 1, 7, 1680, 7, 2, 128 }, + { 12288000, 48000, 1, 1, 7, 0000, 7, 2, 128 }, + { 24576000, 48000, 1, 1, 3, 5000, 7, 2, 128 }, /* With PLL */ + { 24576000, 48000, 0, 0, 0, 0000, 2, 2, 128 }, /* Without PLL */ + /* 88.2k rate */ + { 12000000, 88200, 1, 1, 7, 5264, 4, 4, 64 }, + /* 96k rate */ + { 12000000, 96000, 1, 1, 8, 1920, 4, 4, 64 }, +}; + +static int adc3xxx_get_divs(struct device *dev, int mclk, int rate, int pll_mode) +{ + int i; + + dev_dbg(dev, "mclk = %d, rate = %d, clock mode %u\n", + mclk, rate, pll_mode); + for (i = 0; i < ARRAY_SIZE(adc3xxx_divs); i++) { + const struct adc3xxx_rate_divs *mode = &adc3xxx_divs[i]; + + /* Skip this entry if it doesn't fulfill the intended clock + * mode requirement. We consider anything besides the two + * modes below to be the same as ADC3XXX_PLL_AUTO. + */ + if ((pll_mode == ADC3XXX_PLL_BYPASS && mode->pll_p) || + (pll_mode == ADC3XXX_PLL_ENABLE && !mode->pll_p)) + continue; + + if (mode->rate == rate && mode->mclk == mclk) + return i; + } + + dev_info(dev, "Master clock rate %d and sample rate %d is not supported\n", + mclk, rate); + return -EINVAL; +} + +static int adc3xxx_pll_delay(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* 10msec delay needed after PLL power-up to allow + * PLL and dividers to stabilize (datasheet p13). + */ + usleep_range(10000, 20000); + + return 0; +} + +static const char * const adc_softstepping_text[] = { "1 step", "2 step", "off" }; +static SOC_ENUM_SINGLE_DECL(adc_softstepping_enum, ADC3XXX_ADC_DIGITAL, 0, + adc_softstepping_text); + +static const char * const multiplier_text[] = { "1", "2", "4", "8", "16", "32", "64", "128" }; +static SOC_ENUM_SINGLE_DECL(left_agc_attack_mult_enum, + ADC3XXX_LEFT_CHN_AGC_4, 0, multiplier_text); +static SOC_ENUM_SINGLE_DECL(right_agc_attack_mult_enum, + ADC3XXX_RIGHT_CHN_AGC_4, 0, multiplier_text); +static SOC_ENUM_SINGLE_DECL(left_agc_decay_mult_enum, + ADC3XXX_LEFT_CHN_AGC_5, 0, multiplier_text); +static SOC_ENUM_SINGLE_DECL(right_agc_decay_mult_enum, + ADC3XXX_RIGHT_CHN_AGC_5, 0, multiplier_text); + +static const char * const dither_dc_offset_text[] = { + "0mV", "15mV", "30mV", "45mV", "60mV", "75mV", "90mV", "105mV", + "-15mV", "-30mV", "-45mV", "-60mV", "-75mV", "-90mV", "-105mV" +}; +static const unsigned int dither_dc_offset_values[] = { + 0, 1, 2, 3, 4, 5, 6, 7, 9, 10, 11, 12, 13, 14, 15 +}; +static SOC_VALUE_ENUM_DOUBLE_DECL(dither_dc_offset_enum, + ADC3XXX_DITHER_CTRL, + 4, 0, 0xf, dither_dc_offset_text, + dither_dc_offset_values); + +static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_fine_tlv, -40, 10, 0); +/* AGC target: 8 values: -5.5, -8, -10, -12, -14, -17, -20, -24 dB */ +/* It would be nice to declare these in the order above, but empirically + * TLV_DB_SCALE_ITEM doesn't take lightly to the increment (second) parameter + * being negative, despite there being examples to the contrary in other + * drivers. So declare these in the order from lowest to highest, and + * set the invert flag in the SOC_DOUBLE_R_TLV declaration instead. + */ +static const DECLARE_TLV_DB_RANGE(agc_target_tlv, + 0, 0, TLV_DB_SCALE_ITEM(-2400, 0, 0), + 1, 3, TLV_DB_SCALE_ITEM(-2000, 300, 0), + 4, 6, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 7, 7, TLV_DB_SCALE_ITEM(-550, 0, 0)); +/* Since the 'disabled' value (mute) is at the highest value in the dB + * range (i.e. just before -32 dB) rather than the lowest, we need to resort + * to using a TLV_DB_RANGE in order to get the mute value in the right place. + */ +static const DECLARE_TLV_DB_RANGE(agc_thresh_tlv, + 0, 30, TLV_DB_SCALE_ITEM(-9000, 200, 0), + 31, 31, TLV_DB_SCALE_ITEM(0, 0, 1)); /* disabled = mute */ +/* AGC hysteresis: 4 values: 1, 2, 4 dB, disabled (= mute) */ +static const DECLARE_TLV_DB_RANGE(agc_hysteresis_tlv, + 0, 1, TLV_DB_SCALE_ITEM(100, 100, 0), + 2, 2, TLV_DB_SCALE_ITEM(400, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(0, 0, 1)); /* disabled = mute */ +static const DECLARE_TLV_DB_SCALE(agc_max_tlv, 0, 50, 0); +/* Input attenuation: -6 dB or 0 dB */ +static const DECLARE_TLV_DB_SCALE(input_attenuation_tlv, -600, 600, 0); + +static const struct snd_kcontrol_new adc3xxx_snd_controls[] = { + SOC_DOUBLE_R_TLV("PGA Capture Volume", ADC3XXX_LEFT_APGA_CTRL, + ADC3XXX_RIGHT_APGA_CTRL, 0, 80, 0, pga_tlv), + SOC_DOUBLE("PGA Capture Switch", ADC3XXX_ADC_FGA, 7, 3, 1, 1), + SOC_DOUBLE_R("AGC Capture Switch", ADC3XXX_LEFT_CHN_AGC_1, + ADC3XXX_RIGHT_CHN_AGC_1, 7, 1, 0), + SOC_DOUBLE_R_TLV("AGC Target Level Capture Volume", ADC3XXX_LEFT_CHN_AGC_1, + ADC3XXX_RIGHT_CHN_AGC_2, 4, 0x07, 1, agc_target_tlv), + SOC_DOUBLE_R_TLV("AGC Noise Threshold Capture Volume", ADC3XXX_LEFT_CHN_AGC_2, + ADC3XXX_RIGHT_CHN_AGC_2, 1, 0x1f, 1, agc_thresh_tlv), + SOC_DOUBLE_R_TLV("AGC Hysteresis Capture Volume", ADC3XXX_LEFT_CHN_AGC_2, + ADC3XXX_RIGHT_CHN_AGC_2, 6, 3, 0, agc_hysteresis_tlv), + SOC_DOUBLE_R("AGC Clip Stepping Capture Switch", ADC3XXX_LEFT_CHN_AGC_2, + ADC3XXX_RIGHT_CHN_AGC_2, 0, 1, 0), + /* + * Oddly enough, the data sheet says the default value + * for the left/right AGC maximum gain register field + * (ADC3XXX_LEFT/RIGHT_CHN_AGC_3 bits 0..6) is 0x7f = 127 + * (verified empirically) even though this value (indeed, above + * 0x50) is specified as 'Reserved. Do not use.' in the accompanying + * table in the data sheet. + */ + SOC_DOUBLE_R_TLV("AGC Maximum Capture Volume", ADC3XXX_LEFT_CHN_AGC_3, + ADC3XXX_RIGHT_CHN_AGC_3, 0, 0x50, 0, agc_max_tlv), + SOC_DOUBLE_R("AGC Attack Time", ADC3XXX_LEFT_CHN_AGC_4, + ADC3XXX_RIGHT_CHN_AGC_4, 3, 0x1f, 0), + /* Would like to have the multipliers as LR pairs, but there is + * no SOC_ENUM_foo which accepts two values in separate registers. + */ + SOC_ENUM("AGC Left Attack Time Multiplier", left_agc_attack_mult_enum), + SOC_ENUM("AGC Right Attack Time Multiplier", right_agc_attack_mult_enum), + SOC_DOUBLE_R("AGC Decay Time", ADC3XXX_LEFT_CHN_AGC_5, + ADC3XXX_RIGHT_CHN_AGC_5, 3, 0x1f, 0), + SOC_ENUM("AGC Left Decay Time Multiplier", left_agc_decay_mult_enum), + SOC_ENUM("AGC Right Decay Time Multiplier", right_agc_decay_mult_enum), + SOC_DOUBLE_R("AGC Noise Debounce", ADC3XXX_LEFT_CHN_AGC_6, + ADC3XXX_RIGHT_CHN_AGC_6, 0, 0x1f, 0), + SOC_DOUBLE_R("AGC Signal Debounce", ADC3XXX_LEFT_CHN_AGC_7, + ADC3XXX_RIGHT_CHN_AGC_7, 0, 0x0f, 0), + /* Read only register */ + SOC_DOUBLE_R_S_TLV("AGC Applied Capture Volume", ADC3XXX_LEFT_AGC_GAIN, + ADC3XXX_RIGHT_AGC_GAIN, 0, -24, 40, 6, 0, adc_tlv), + /* ADC soft stepping */ + SOC_ENUM("ADC Soft Stepping", adc_softstepping_enum), + /* Left/Right Input attenuation */ + SOC_SINGLE_TLV("Left Input IN_1L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input IN_2L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 2, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input IN_3L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input IN_1R Capture Volume", + ADC3XXX_LEFT_PGA_SEL_2, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input DIF_2L_3L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 6, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input DIF_1L_1R Capture Volume", + ADC3XXX_LEFT_PGA_SEL_2, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input DIF_2R_3R Capture Volume", + ADC3XXX_LEFT_PGA_SEL_2, 2, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_1R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_2R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 2, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_3R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_1L Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_2, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input DIF_2R_3R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 6, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input DIF_1L_1R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_2, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input DIF_2L_3L Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_2, 2, 1, 1, input_attenuation_tlv), + SOC_DOUBLE_R_S_TLV("ADC Volume Control Capture Volume", ADC3XXX_LADC_VOL, + ADC3XXX_RADC_VOL, 0, -24, 40, 6, 0, adc_tlv), + /* Empirically, the following doesn't work the way it's supposed + * to. Values 0, -0.1, -0.2 and -0.3 dB result in the same level, and + * -0.4 dB drops about 0.12 dB on a specific chip. + */ + SOC_DOUBLE_TLV("ADC Fine Volume Control Capture Volume", ADC3XXX_ADC_FGA, + 4, 0, 4, 1, adc_fine_tlv), + SOC_SINGLE("Left ADC Unselected CM Bias Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 6, 1, 0), + SOC_SINGLE("Right ADC Unselected CM Bias Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 6, 1, 0), + SOC_ENUM("Dither Control DC Offset", dither_dc_offset_enum), +}; + +/* Left input selection, Single Ended inputs and Differential inputs */ +static const struct snd_kcontrol_new left_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN_1L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 1, 0x1, 1), + SOC_DAPM_SINGLE("IN_2L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_3L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2L_3L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 7, 0x1, 1), + SOC_DAPM_SINGLE("DIF_1L_1R Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2R_3R Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_1R Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 1, 0x1, 1), +}; + +/* Right input selection, Single Ended inputs and Differential inputs */ +static const struct snd_kcontrol_new right_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN_1R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 1, 0x1, 1), + SOC_DAPM_SINGLE("IN_2R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_3R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2R_3R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 7, 0x1, 1), + SOC_DAPM_SINGLE("DIF_1L_1R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2L_3L Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_1L Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 1, 0x1, 1), +}; + +/* Left Digital Mic input for left ADC */ +static const struct snd_kcontrol_new left_input_dmic_controls[] = { + SOC_DAPM_SINGLE("Left ADC Capture Switch", + ADC3XXX_ADC_DIGITAL, 3, 0x1, 0), +}; + +/* Right Digital Mic input for Right ADC */ +static const struct snd_kcontrol_new right_input_dmic_controls[] = { + SOC_DAPM_SINGLE("Right ADC Capture Switch", + ADC3XXX_ADC_DIGITAL, 2, 0x1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget adc3xxx_dapm_widgets[] = { + + /* Left Input Selection */ + SND_SOC_DAPM_MIXER("Left Input", SND_SOC_NOPM, 0, 0, + &left_input_mixer_controls[0], + ARRAY_SIZE(left_input_mixer_controls)), + /* Right Input Selection */ + SND_SOC_DAPM_MIXER("Right Input", SND_SOC_NOPM, 0, 0, + &right_input_mixer_controls[0], + ARRAY_SIZE(right_input_mixer_controls)), + /* PGA selection */ + SND_SOC_DAPM_PGA("Left PGA", ADC3XXX_LEFT_APGA_CTRL, 7, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right PGA", ADC3XXX_RIGHT_APGA_CTRL, 7, 1, NULL, 0), + + /* Digital Microphone Input Control for Left/Right ADC */ + SND_SOC_DAPM_MIXER("Left DMic Input", SND_SOC_NOPM, 0, 0, + &left_input_dmic_controls[0], + ARRAY_SIZE(left_input_dmic_controls)), + SND_SOC_DAPM_MIXER("Right DMic Input", SND_SOC_NOPM, 0, 0, + &right_input_dmic_controls[0], + ARRAY_SIZE(right_input_dmic_controls)), + + /* Left/Right ADC */ + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", ADC3XXX_ADC_DIGITAL, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", ADC3XXX_ADC_DIGITAL, 6, 0), + + /* Inputs */ + SND_SOC_DAPM_INPUT("IN_1L"), + SND_SOC_DAPM_INPUT("IN_1R"), + SND_SOC_DAPM_INPUT("IN_2L"), + SND_SOC_DAPM_INPUT("IN_2R"), + SND_SOC_DAPM_INPUT("IN_3L"), + SND_SOC_DAPM_INPUT("IN_3R"), + SND_SOC_DAPM_INPUT("DIFL_1L_1R"), + SND_SOC_DAPM_INPUT("DIFL_2L_3L"), + SND_SOC_DAPM_INPUT("DIFL_2R_3R"), + SND_SOC_DAPM_INPUT("DIFR_1L_1R"), + SND_SOC_DAPM_INPUT("DIFR_2L_3L"), + SND_SOC_DAPM_INPUT("DIFR_2R_3R"), + SND_SOC_DAPM_INPUT("DMic_L"), + SND_SOC_DAPM_INPUT("DMic_R"), + + /* Digital audio interface output */ + SND_SOC_DAPM_AIF_OUT("AIF_OUT", "Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Clocks */ + SND_SOC_DAPM_SUPPLY("PLL_CLK", ADC3XXX_PLL_PROG_PR, ADC3XXX_ENABLE_PLL_SHIFT, + 0, adc3xxx_pll_delay, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_SUPPLY("ADC_CLK", ADC3XXX_ADC_NADC, ADC3XXX_ENABLE_NADC_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_MOD_CLK", ADC3XXX_ADC_MADC, ADC3XXX_ENABLE_MADC_SHIFT, + 0, NULL, 0), + + /* This refers to the generated BCLK in master mode. */ + SND_SOC_DAPM_SUPPLY("BCLK", ADC3XXX_BCLK_N_DIV, ADC3XXX_ENABLE_BCLK_SHIFT, + 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route adc3xxx_intercon[] = { + /* Left input selection from switches */ + { "Left Input", "IN_1L Capture Switch", "IN_1L" }, + { "Left Input", "IN_2L Capture Switch", "IN_2L" }, + { "Left Input", "IN_3L Capture Switch", "IN_3L" }, + { "Left Input", "DIF_2L_3L Capture Switch", "DIFL_2L_3L" }, + { "Left Input", "DIF_1L_1R Capture Switch", "DIFL_1L_1R" }, + { "Left Input", "DIF_2R_3R Capture Switch", "DIFL_2R_3R" }, + { "Left Input", "IN_1R Capture Switch", "IN_1R" }, + + /* Left input selection to left PGA */ + { "Left PGA", NULL, "Left Input" }, + + /* Left PGA to left ADC */ + { "Left ADC", NULL, "Left PGA" }, + + /* Right input selection from switches */ + { "Right Input", "IN_1R Capture Switch", "IN_1R" }, + { "Right Input", "IN_2R Capture Switch", "IN_2R" }, + { "Right Input", "IN_3R Capture Switch", "IN_3R" }, + { "Right Input", "DIF_2R_3R Capture Switch", "DIFR_2R_3R" }, + { "Right Input", "DIF_1L_1R Capture Switch", "DIFR_1L_1R" }, + { "Right Input", "DIF_2L_3L Capture Switch", "DIFR_2L_3L" }, + { "Right Input", "IN_1L Capture Switch", "IN_1L" }, + + /* Right input selection to right PGA */ + { "Right PGA", NULL, "Right Input" }, + + /* Right PGA to right ADC */ + { "Right ADC", NULL, "Right PGA" }, + + /* Left DMic Input selection from switch */ + { "Left DMic Input", "Left ADC Capture Switch", "DMic_L" }, + + /* Left DMic to left ADC */ + { "Left ADC", NULL, "Left DMic Input" }, + + /* Right DMic Input selection from switch */ + { "Right DMic Input", "Right ADC Capture Switch", "DMic_R" }, + + /* Right DMic to right ADC */ + { "Right ADC", NULL, "Right DMic Input" }, + + /* ADC to AIF output */ + { "AIF_OUT", NULL, "Left ADC" }, + { "AIF_OUT", NULL, "Right ADC" }, + + /* Clocking */ + { "ADC_MOD_CLK", NULL, "ADC_CLK" }, + { "Left ADC", NULL, "ADC_MOD_CLK" }, + { "Right ADC", NULL, "ADC_MOD_CLK" }, + + { "BCLK", NULL, "ADC_CLK" }, +}; + +static const struct snd_soc_dapm_route adc3xxx_pll_intercon[] = { + { "ADC_CLK", NULL, "PLL_CLK" }, +}; + +static const struct snd_soc_dapm_route adc3xxx_bclk_out_intercon[] = { + { "AIF_OUT", NULL, "BCLK" } +}; + +static int adc3xxx_gpio_request(struct gpio_chip *chip, unsigned int offset) +{ + struct adc3xxx *adc3xxx = gpiochip_get_data(chip); + + if (offset >= ADC3XXX_GPIOS_MAX) + return -EINVAL; + + /* GPIO1 is offset 0, GPIO2 is offset 1 */ + /* We check here that the GPIO pins are either not configured in the + * DT, or that they purposely are set as outputs. + * (Input mode not yet implemented). + */ + if (adc3xxx->gpio_cfg[offset] != 0 && + adc3xxx->gpio_cfg[offset] != ADC3XXX_GPIO_GPO + 1) + return -EINVAL; + + return 0; +} + +static int adc3xxx_gpio_direction_out(struct gpio_chip *chip, + unsigned int offset, int value) +{ + struct adc3xxx *adc3xxx = gpiochip_get_data(chip); + + /* Set GPIO output function. */ + return regmap_update_bits(adc3xxx->regmap, + adc3xxx_gpio_ctrl_reg[offset], + ADC3XXX_GPIO_CTRL_CFG_MASK | + ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_MASK, + ADC3XXX_GPIO_GPO << ADC3XXX_GPIO_CTRL_CFG_SHIFT | + !!value << ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_SHIFT); +} + +/* With only GPIO outputs configured, we never get the .direction_out call, + * so we set the output mode and output value in the same call. Hence + * .set in practice does the same thing as .direction_out . + */ +static void adc3xxx_gpio_set(struct gpio_chip *chip, unsigned int offset, + int value) +{ + (void) adc3xxx_gpio_direction_out(chip, offset, value); +} + +/* Even though we only support GPIO output for now, some GPIO clients + * want to read the current pin state using the .get callback. + */ +static int adc3xxx_gpio_get(struct gpio_chip *chip, unsigned int offset) +{ + struct adc3xxx *adc3xxx = gpiochip_get_data(chip); + unsigned int regval; + int ret; + + /* We only allow output pins, so just read the value set in the output + * pin register field. + */ + ret = regmap_read(adc3xxx->regmap, adc3xxx_gpio_ctrl_reg[offset], &regval); + if (ret) + return ret; + return !!(regval & ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_MASK); +} + +static const struct gpio_chip adc3xxx_gpio_chip = { + .label = "adc3xxx", + .owner = THIS_MODULE, + .request = adc3xxx_gpio_request, + .direction_output = adc3xxx_gpio_direction_out, + .set = adc3xxx_gpio_set, + .get = adc3xxx_gpio_get, + .can_sleep = 1, +}; + +static void adc3xxx_free_gpio(struct adc3xxx *adc3xxx) +{ + gpiochip_remove(&adc3xxx->gpio_chip); +} + +static void adc3xxx_init_gpio(struct adc3xxx *adc3xxx) +{ + int gpio, micbias; + int ret; + + adc3xxx->gpio_chip = adc3xxx_gpio_chip; + adc3xxx->gpio_chip.ngpio = ADC3XXX_GPIOS_MAX; + adc3xxx->gpio_chip.parent = adc3xxx->dev; + adc3xxx->gpio_chip.base = -1; + + ret = gpiochip_add_data(&adc3xxx->gpio_chip, adc3xxx); + if (ret) + dev_err(adc3xxx->dev, "Failed to add gpios: %d\n", ret); + + /* Set up potential GPIO configuration from the devicetree. + * This allows us to set up things which are not software + * controllable GPIOs, such as PDM microphone I/O, + */ + for (gpio = 0; gpio < ADC3XXX_GPIOS_MAX; gpio++) { + unsigned int cfg = adc3xxx->gpio_cfg[gpio]; + + if (cfg) { + cfg--; /* actual value to use is stored +1 */ + regmap_update_bits(adc3xxx->regmap, + adc3xxx_gpio_ctrl_reg[gpio], + ADC3XXX_GPIO_CTRL_CFG_MASK, + cfg << ADC3XXX_GPIO_CTRL_CFG_SHIFT); + } + } + + /* Set up micbias voltage */ + for (micbias = 0; micbias < ADC3XXX_MICBIAS_PINS; micbias++) { + unsigned int vg = adc3xxx->micbias_vg[micbias]; + + regmap_update_bits(adc3xxx->regmap, + ADC3XXX_MICBIAS_CTRL, + ADC3XXX_MICBIAS_MASK << adc3xxx_micbias_shift[micbias], + vg << adc3xxx_micbias_shift[micbias]); + } +} + +static int adc3xxx_parse_dt_gpio(struct adc3xxx *adc3xxx, + const char *propname, unsigned int *cfg) +{ + struct device *dev = adc3xxx->dev; + struct device_node *np = dev->of_node; + unsigned int val; + + if (!of_property_read_u32(np, propname, &val)) { + if (val & ~15 || val == 7 || val >= 11) { + dev_err(dev, "Invalid property value for '%s'\n", propname); + return -EINVAL; + } + if (val == ADC3XXX_GPIO_GPI) + dev_warn(dev, "GPIO Input read not yet implemented\n"); + *cfg = val + 1; /* 0 => not set up, all others shifted +1 */ + } + return 0; +} + +static int adc3xxx_parse_dt_micbias(struct adc3xxx *adc3xxx, + const char *propname, unsigned int *vg) +{ + struct device *dev = adc3xxx->dev; + struct device_node *np = dev->of_node; + unsigned int val; + + if (!of_property_read_u32(np, propname, &val)) { + if (val >= ADC3XXX_MICBIAS_AVDD) { + dev_err(dev, "Invalid property value for '%s'\n", propname); + return -EINVAL; + } + *vg = val; + } + return 0; +} + +static int adc3xxx_parse_pll_mode(uint32_t val, unsigned int *pll_mode) +{ + if (val != ADC3XXX_PLL_ENABLE && val != ADC3XXX_PLL_BYPASS && + val != ADC3XXX_PLL_AUTO) + return -EINVAL; + + *pll_mode = val; + + return 0; +} + +static void adc3xxx_setup_pll(struct snd_soc_component *component, + int div_entry) +{ + int i = div_entry; + + /* P & R values */ + snd_soc_component_write(component, ADC3XXX_PLL_PROG_PR, + (adc3xxx_divs[i].pll_p << ADC3XXX_PLLP_SHIFT) | + (adc3xxx_divs[i].pll_r << ADC3XXX_PLLR_SHIFT)); + /* J value */ + snd_soc_component_write(component, ADC3XXX_PLL_PROG_J, + adc3xxx_divs[i].pll_j & ADC3XXX_PLLJ_MASK); + /* D value */ + snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_LSB, + adc3xxx_divs[i].pll_d & ADC3XXX_PLLD_LSB_MASK); + snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_MSB, + (adc3xxx_divs[i].pll_d >> 8) & ADC3XXX_PLLD_MSB_MASK); +} + +static int adc3xxx_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 snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(dai->component); + struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); + int i, width = 16; + u8 iface_len, bdiv; + + i = adc3xxx_get_divs(component->dev, adc3xxx->sysclk, + params_rate(params), adc3xxx->pll_mode); + + if (i < 0) + return i; + + /* select data word length */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iface_len = ADC3XXX_IFACE_16BITS; + width = 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_len = ADC3XXX_IFACE_20BITS; + width = 20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_len = ADC3XXX_IFACE_24BITS; + width = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_len = ADC3XXX_IFACE_32BITS; + width = 32; + break; + default: + dev_err(component->dev, "Unsupported serial data format\n"); + return -EINVAL; + } + snd_soc_component_update_bits(component, ADC3XXX_INTERFACE_CTRL_1, + ADC3XXX_WLENGTH_MASK, iface_len); + if (adc3xxx_divs[i].pll_p) { /* If PLL used for this mode */ + adc3xxx_setup_pll(component, i); + snd_soc_component_write(component, ADC3XXX_CLKGEN_MUX, ADC3XXX_USE_PLL); + if (!adc3xxx->use_pll) { + snd_soc_dapm_add_routes(dapm, adc3xxx_pll_intercon, + ARRAY_SIZE(adc3xxx_pll_intercon)); + adc3xxx->use_pll = 1; + } + } else { + snd_soc_component_write(component, ADC3XXX_CLKGEN_MUX, ADC3XXX_NO_PLL); + if (adc3xxx->use_pll) { + snd_soc_dapm_del_routes(dapm, adc3xxx_pll_intercon, + ARRAY_SIZE(adc3xxx_pll_intercon)); + adc3xxx->use_pll = 0; + } + } + + /* NADC */ + snd_soc_component_update_bits(component, ADC3XXX_ADC_NADC, + ADC3XXX_NADC_MASK, adc3xxx_divs[i].nadc); + /* MADC */ + snd_soc_component_update_bits(component, ADC3XXX_ADC_MADC, + ADC3XXX_MADC_MASK, adc3xxx_divs[i].madc); + /* AOSR */ + snd_soc_component_update_bits(component, ADC3XXX_ADC_AOSR, + ADC3XXX_AOSR_MASK, adc3xxx_divs[i].aosr); + /* BDIV N Value */ + /* BCLK is (by default) set up to be derived from ADC_CLK */ + bdiv = (adc3xxx_divs[i].aosr * adc3xxx_divs[i].madc) / (2 * width); + snd_soc_component_update_bits(component, ADC3XXX_BCLK_N_DIV, + ADC3XXX_BDIV_MASK, bdiv); + + return 0; +} + +static const char *adc3xxx_pll_mode_text(int pll_mode) +{ + switch (pll_mode) { + case ADC3XXX_PLL_AUTO: + return "PLL auto"; + case ADC3XXX_PLL_ENABLE: + return "PLL enable"; + case ADC3XXX_PLL_BYPASS: + return "PLL bypass"; + default: + break; + } + + return "PLL unknown"; +} + +static int adc3xxx_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); + int ret; + + ret = adc3xxx_parse_pll_mode(clk_id, &adc3xxx->pll_mode); + if (ret < 0) + return ret; + + adc3xxx->sysclk = freq; + dev_dbg(component->dev, "Set sysclk to %u Hz, %s\n", + freq, adc3xxx_pll_mode_text(adc3xxx->pll_mode)); + return 0; +} + +static int adc3xxx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); + u8 clkdir = 0, format = 0; + int master = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + master = 1; + clkdir = ADC3XXX_BCLK_MASTER | ADC3XXX_WCLK_MASTER; + break; + case SND_SOC_DAIFMT_CBC_CFC: + master = 0; + break; + default: + dev_err(component->dev, "Invalid DAI clock setup\n"); + return -EINVAL; + } + + /* + * match both interface format and signal polarities since they + * are fixed + */ + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF: + format = ADC3XXX_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF: + format = ADC3XXX_FORMAT_DSP; + break; + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF: + format = ADC3XXX_FORMAT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF: + format = ADC3XXX_FORMAT_RJF; + break; + case SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF: + format = ADC3XXX_FORMAT_LJF; + break; + default: + dev_err(component->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + /* Add/del route enabling BCLK output as applicable */ + if (master && !adc3xxx->master) + snd_soc_dapm_add_routes(dapm, adc3xxx_bclk_out_intercon, + ARRAY_SIZE(adc3xxx_bclk_out_intercon)); + else if (!master && adc3xxx->master) + snd_soc_dapm_del_routes(dapm, adc3xxx_bclk_out_intercon, + ARRAY_SIZE(adc3xxx_bclk_out_intercon)); + adc3xxx->master = master; + + /* set clock direction and format */ + return snd_soc_component_update_bits(component, + ADC3XXX_INTERFACE_CTRL_1, + ADC3XXX_CLKDIR_MASK | ADC3XXX_FORMAT_MASK, + clkdir | format); +} + +static const struct snd_soc_dai_ops adc3xxx_dai_ops = { + .hw_params = adc3xxx_hw_params, + .set_sysclk = adc3xxx_set_dai_sysclk, + .set_fmt = adc3xxx_set_dai_fmt, +}; + +static struct snd_soc_dai_driver adc3xxx_dai = { + .name = "tlv320adc3xxx-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ADC3XXX_RATES, + .formats = ADC3XXX_FORMATS, + }, + .ops = &adc3xxx_dai_ops, +}; + +static const struct snd_soc_component_driver soc_component_dev_adc3xxx = { + .controls = adc3xxx_snd_controls, + .num_controls = ARRAY_SIZE(adc3xxx_snd_controls), + .dapm_widgets = adc3xxx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adc3xxx_dapm_widgets), + .dapm_routes = adc3xxx_intercon, + .num_dapm_routes = ARRAY_SIZE(adc3xxx_intercon), +}; + +static int adc3xxx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct adc3xxx *adc3xxx = NULL; + int ret; + + adc3xxx = devm_kzalloc(dev, sizeof(struct adc3xxx), GFP_KERNEL); + if (!adc3xxx) + return -ENOMEM; + adc3xxx->dev = dev; + + adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); + if (IS_ERR(adc3xxx->rst_pin)) { + return dev_err_probe(dev, PTR_ERR(adc3xxx->rst_pin), + "Failed to request rst_pin\n"); + } + + adc3xxx->mclk = devm_clk_get(dev, NULL); + if (IS_ERR(adc3xxx->mclk)) { + /* + * The chip itself supports running off the BCLK either + * directly or via the PLL, but the driver does not (yet), so + * having a specified mclk is required. Otherwise, we could + * use the lack of a clocks property to indicate when BCLK is + * intended as the clock source. + */ + return dev_err_probe(dev, PTR_ERR(adc3xxx->mclk), + "Failed to acquire MCLK\n"); + } else if (adc3xxx->mclk) { + ret = clk_prepare_enable(adc3xxx->mclk); + if (ret < 0) + return ret; + dev_dbg(dev, "Enabled MCLK, freq %lu Hz\n", clk_get_rate(adc3xxx->mclk)); + } + + ret = adc3xxx_parse_dt_gpio(adc3xxx, "ti,dmdin-gpio1", &adc3xxx->gpio_cfg[0]); + if (ret < 0) + return ret; + ret = adc3xxx_parse_dt_gpio(adc3xxx, "ti,dmclk-gpio2", &adc3xxx->gpio_cfg[1]); + if (ret < 0) + return ret; + ret = adc3xxx_parse_dt_micbias(adc3xxx, "ti,micbias1-vg", &adc3xxx->micbias_vg[0]); + if (ret < 0) + return ret; + ret = adc3xxx_parse_dt_micbias(adc3xxx, "ti,micbias2-vg", &adc3xxx->micbias_vg[1]); + if (ret < 0) + return ret; + + adc3xxx->regmap = devm_regmap_init_i2c(i2c, &adc3xxx_regmap); + if (IS_ERR(adc3xxx->regmap)) { + ret = PTR_ERR(adc3xxx->regmap); + return ret; + } + + i2c_set_clientdata(i2c, adc3xxx); + + adc3xxx->type = id->driver_data; + + /* Reset codec chip */ + gpiod_set_value_cansleep(adc3xxx->rst_pin, 1); + usleep_range(2000, 100000); /* Requirement: > 10 ns (datasheet p13) */ + gpiod_set_value_cansleep(adc3xxx->rst_pin, 0); + + /* Potentially set up pins used as GPIOs */ + adc3xxx_init_gpio(adc3xxx); + + ret = snd_soc_register_component(dev, + &soc_component_dev_adc3xxx, &adc3xxx_dai, 1); + if (ret < 0) + dev_err(dev, "Failed to register codec: %d\n", ret); + + return ret; +} + +static int __exit adc3xxx_i2c_remove(struct i2c_client *client) +{ + struct adc3xxx *adc3xxx = i2c_get_clientdata(client); + + if (adc3xxx->mclk) + clk_disable_unprepare(adc3xxx->mclk); + adc3xxx_free_gpio(adc3xxx); + snd_soc_unregister_component(&client->dev); + return 0; +} + +static const struct of_device_id tlv320adc3xxx_of_match[] = { + { .compatible = "ti,tlv320adc3001", }, + { .compatible = "ti,tlv320adc3101", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320adc3xxx_of_match); + +static const struct i2c_device_id adc3xxx_i2c_id[] = { + { "tlv320adc3001", ADC3001 }, + { "tlv320adc3101", ADC3101 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, adc3xxx_i2c_id); + +static struct i2c_driver adc3xxx_i2c_driver = { + .driver = { + .name = "tlv320adc3xxx-codec", + .of_match_table = tlv320adc3xxx_of_match, + }, + .probe = adc3xxx_i2c_probe, + .remove = adc3xxx_i2c_remove, + .id_table = adc3xxx_i2c_id, +}; + +module_i2c_driver(adc3xxx_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320ADC3xxx codec driver"); +MODULE_AUTHOR("shahina.s(a)mistralsolutions.com"); +MODULE_LICENSE("GPL v2"); -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[PATCH] ASoC: rt1011: revert 'I2S Reference' to SOC_ENUM_EXT
by Jack Yu 01 Dec '21

01 Dec '21

Revert 'I2S Reference' to SOC_ENUM_EXT because the settings are specific for some platforms, the default setting for 'I2S Reference' does nothing, only some SoC platform need to configure it. Previous 'I2S Reference' in SOC_ENUM format only toggles one bit of RT1011_TDM1_SET_1 register, which isn't enough for specific platform. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1011.c | 55 ++++++++++++++++++++++++++++++++++----- sound/soc/codecs/rt1011.h | 7 +++++ 2 files changed, 56 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 297af7ff824c..b62301a6281f 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1311,13 +1311,54 @@ static int rt1011_r0_load_info(struct snd_kcontrol *kcontrol, .put = rt1011_r0_load_mode_put \ } -static const char * const rt1011_i2s_ref_texts[] = { - "Left Channel", "Right Channel" +static const char * const rt1011_i2s_ref[] = { + "None", "Left Channel", "Right Channel" }; -static SOC_ENUM_SINGLE_DECL(rt1011_i2s_ref_enum, - RT1011_TDM1_SET_1, 7, - rt1011_i2s_ref_texts); +static SOC_ENUM_SINGLE_DECL(rt1011_i2s_ref_enum, 0, 0, + rt1011_i2s_ref); + +static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct rt1011_priv *rt1011 = + snd_soc_component_get_drvdata(component); + + rt1011->i2s_ref = ucontrol->value.enumerated.item[0]; + switch (rt1011->i2s_ref) { + case RT1011_I2S_REF_LEFT_CH: + regmap_write(rt1011->regmap, RT1011_TDM_TOTAL_SET, 0x0240); + regmap_write(rt1011->regmap, RT1011_TDM1_SET_2, 0x8); + regmap_write(rt1011->regmap, RT1011_TDM1_SET_1, 0x1022); + regmap_write(rt1011->regmap, RT1011_ADCDAT_OUT_SOURCE, 0x4); + break; + case RT1011_I2S_REF_RIGHT_CH: + regmap_write(rt1011->regmap, RT1011_TDM_TOTAL_SET, 0x0240); + regmap_write(rt1011->regmap, RT1011_TDM1_SET_2, 0x8); + regmap_write(rt1011->regmap, RT1011_TDM1_SET_1, 0x10a2); + regmap_write(rt1011->regmap, RT1011_ADCDAT_OUT_SOURCE, 0x4); + break; + default: + dev_info(component->dev, "I2S Reference: Do nothing\n"); + } + + return 0; +} + +static int rt1011_i2s_ref_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct rt1011_priv *rt1011 = + snd_soc_component_get_drvdata(component); + + ucontrol->value.enumerated.item[0] = rt1011->i2s_ref; + + return 0; +} static const struct snd_kcontrol_new rt1011_snd_controls[] = { /* I2S Data In Selection */ @@ -1358,7 +1399,8 @@ static const struct snd_kcontrol_new rt1011_snd_controls[] = { SOC_SINGLE("R0 Temperature", RT1011_STP_INITIAL_RESISTANCE_TEMP, 2, 255, 0), /* I2S Reference */ - SOC_ENUM("I2S Reference", rt1011_i2s_ref_enum), + SOC_ENUM_EXT("I2S Reference", rt1011_i2s_ref_enum, + rt1011_i2s_ref_get, rt1011_i2s_ref_put), }; static int rt1011_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, @@ -2017,6 +2059,7 @@ static int rt1011_probe(struct snd_soc_component *component) schedule_work(&rt1011->cali_work); + rt1011->i2s_ref = 0; rt1011->bq_drc_params = devm_kcalloc(component->dev, RT1011_ADVMODE_NUM, sizeof(struct rt1011_bq_drc_params *), GFP_KERNEL); diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 68fadc15fa8c..4d6e7492d99c 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -654,6 +654,12 @@ enum { RT1011_AIFS }; +enum { + RT1011_I2S_REF_NONE, + RT1011_I2S_REF_LEFT_CH, + RT1011_I2S_REF_RIGHT_CH, +}; + /* BiQual & DRC related settings */ #define RT1011_BQ_DRC_NUM 128 struct rt1011_bq_drc_params { @@ -692,6 +698,7 @@ struct rt1011_priv { unsigned int r0_reg, cali_done; unsigned int r0_calib, temperature_calib; int recv_spk_mode; + int i2s_ref; }; #endif /* end of _RT1011_H_ */ -- 2.33.0

3 5

ALC1220 Broken Front Panel
by GitHub issues - opened 01 Dec '21

01 Dec '21

alsa-project/alsa-ucm-conf issue #123 was opened from vs49688: Since upgrading from v1.2.4 -> v1.2.5.1, I can no longer get any input/output from my front-panel. I've bisected the issue to 5947daef124c84fae511233248196576dfbfab7c. Reverting fixes it, but this may break other cards. The hardware is an "AX370-Gaming K7", with a ALC1220 chip. ##### alsa-info Before revert [alsainfo_before.txt](https://github.com/alsa-project/alsa-ucm-conf/files/76… ##### alsa-info After revert [alsainfo_after.txt](https://github.com/alsa-project/alsa-ucm-conf/files/763… ##### Relevant dmesg output: ``` [ 11.455796] snd_hda_intel 0000:0b:00.3: enabling device (0000 -> 0002) [ 11.532977] snd_hda_codec_realtek hdaudioC1D0: autoconfig for ALC1220: line_outs=3 (0x14/0x15/0x16/0x0/0x0) type:line [ 11.532980] snd_hda_codec_realtek hdaudioC1D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 11.532981] snd_hda_codec_realtek hdaudioC1D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 11.532982] snd_hda_codec_realtek hdaudioC1D0: mono: mono_out=0x0 [ 11.532983] snd_hda_codec_realtek hdaudioC1D0: dig-out=0x1e/0x0 [ 11.532984] snd_hda_codec_realtek hdaudioC1D0: inputs: [ 11.532985] snd_hda_codec_realtek hdaudioC1D0: Rear Mic=0x18 [ 11.532986] snd_hda_codec_realtek hdaudioC1D0: Line=0x1a [ 11.543432] snd_hda_codec_realtek hdaudioC1D1: autoconfig for ALC1220: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:hp [ 11.543434] snd_hda_codec_realtek hdaudioC1D1: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 11.543435] snd_hda_codec_realtek hdaudioC1D1: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 11.543436] snd_hda_codec_realtek hdaudioC1D1: mono: mono_out=0x0 [ 11.543436] snd_hda_codec_realtek hdaudioC1D1: inputs: [ 11.543437] snd_hda_codec_realtek hdaudioC1D1: Front Mic=0x1a [ 11.548516] input: HD-Audio Generic Rear Mic as /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/sound/card1/input19 [ 11.548554] input: HD-Audio Generic Line as /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/sound/card1/input20 [ 11.548583] input: HD-Audio Generic Line Out Front as /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/sound/card1/input21 [ 11.548613] input: HD-Audio Generic Line Out Surround as /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/sound/card1/input22 [ 11.548641] input: HD-Audio Generic Line Out CLFE as /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/sound/card1/input23 [ 11.548669] input: HD-Audio Generic Front Mic as /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/sound/card1/input24 [ 11.548695] input: HD-Audio Generic Front Headphone as /devices/pci0000:00/0000:00:08.1/0000:0b:00.3/sound/card1/input25 ``` ##### pavucontrol Before revert (doesn't work): ![ssbad](https://user-images.githubusercontent.com/4423262/144226384-0d338bef-9cda-439c-9bba-cc0cb63551eb.png) ##### pavucontrol After revert (works): Appears the same as it did on v1.2.4: ![ssgood](https://user-images.githubusercontent.com/4423262/144226410-14e3cd07-b179-4ce3-a224-17e7a35ae1dd.png) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/123 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 00/21] ASoC: SOF: Platform updates for AMD and Mediatek
by Daniel Baluta 01 Dec '21

01 Dec '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries adds Renoir ACP HW support and Mediatek MT8195 platform. Notes: - I first send the meditatek support in a previous emails but it failed to apply because there are some dependencies on AMD patches related to DAI ids. Ajit Kumar Pandey (12): ASoC: SOF: amd: Add Renoir ACP HW support ASoC: SOF: amd: Add helper callbacks for ACP's DMA configuration ASoC: SOF: amd: Add fw loader and renoir dsp ops to load firmware ASoC: SOF: amd: Add IPC support for ACP IP block ASoC: SOF: amd: Add dai driver dsp ops callback for Renoir ASoC: SOF: amd: Add PCM stream callback for Renoir dai's ASoC: amd: Add module to determine ACP configuration ASoC: SOF: amd: Add machine driver dsp ops for Renoir platform ASoC: SOF: amd: Add Renoir PCI driver interface ASoC: amd: acp-config: Remove legacy acpi based machine struct ASoC: SOF: topology: Add support for AMD ACP DAIs ASoC: SOF: amd: Add support for SOF firmware authentication V sujith kumar Reddy (1): ASoC: SOF: amd: Add trace logger support YC Hung (8): ASoC: SOF: mediatek: Add mt8195 hardware support ASoC: SOF: tokens: add token for Mediatek AFE ASoC: SOF: topology: Add support for Mediatek AFE DAI ASoC: SOF: mediatek: Add fw loader and mt8195 dsp ops to load firmware ASoC: SOF: Add mt8195 device descriptor ASoC: SOF: mediatek: Add dai driver dsp ops callback for mt8195 ASoC: SOF: mediatek: Add mt8195 dsp clock support ASoC: SOF: mediatek: Add DSP system PM callback for mt8195 include/sound/sof/dai-amd.h | 21 + include/sound/sof/dai-mediatek.h | 23 + include/sound/sof/dai.h | 10 + include/uapi/sound/sof/tokens.h | 5 + sound/soc/amd/Kconfig | 6 + sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-config.c | 71 +++ sound/soc/amd/mach-config.h | 28 ++ sound/soc/sof/Kconfig | 4 +- sound/soc/sof/Makefile | 2 + sound/soc/sof/amd/Kconfig | 33 ++ sound/soc/sof/amd/Makefile | 11 + sound/soc/sof/amd/acp-dsp-offset.h | 78 +++ sound/soc/sof/amd/acp-ipc.c | 187 ++++++++ sound/soc/sof/amd/acp-loader.c | 199 ++++++++ sound/soc/sof/amd/acp-pcm.c | 82 ++++ sound/soc/sof/amd/acp-stream.c | 181 +++++++ sound/soc/sof/amd/acp-trace.c | 84 ++++ sound/soc/sof/amd/acp.c | 446 ++++++++++++++++++ sound/soc/sof/amd/acp.h | 226 +++++++++ sound/soc/sof/amd/pci-rn.c | 165 +++++++ sound/soc/sof/amd/renoir.c | 185 ++++++++ sound/soc/sof/mediatek/Kconfig | 33 ++ sound/soc/sof/mediatek/Makefile | 2 + sound/soc/sof/mediatek/adsp_helper.h | 49 ++ sound/soc/sof/mediatek/mediatek-ops.h | 8 + sound/soc/sof/mediatek/mt8195/Makefile | 3 + sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 158 +++++++ sound/soc/sof/mediatek/mt8195/mt8195-clk.h | 28 ++ sound/soc/sof/mediatek/mt8195/mt8195-loader.c | 56 +++ sound/soc/sof/mediatek/mt8195/mt8195.c | 437 +++++++++++++++++ sound/soc/sof/mediatek/mt8195/mt8195.h | 158 +++++++ sound/soc/sof/pcm.c | 48 ++ sound/soc/sof/sof-of-dev.c | 13 + sound/soc/sof/topology.c | 168 +++++++ 35 files changed, 3209 insertions(+), 1 deletion(-) create mode 100644 include/sound/sof/dai-amd.h create mode 100644 include/sound/sof/dai-mediatek.h create mode 100644 sound/soc/amd/acp-config.c create mode 100644 sound/soc/amd/mach-config.h create mode 100644 sound/soc/sof/amd/Kconfig create mode 100644 sound/soc/sof/amd/Makefile create mode 100644 sound/soc/sof/amd/acp-dsp-offset.h create mode 100644 sound/soc/sof/amd/acp-ipc.c create mode 100644 sound/soc/sof/amd/acp-loader.c create mode 100644 sound/soc/sof/amd/acp-pcm.c create mode 100644 sound/soc/sof/amd/acp-stream.c create mode 100644 sound/soc/sof/amd/acp-trace.c create mode 100644 sound/soc/sof/amd/acp.c create mode 100644 sound/soc/sof/amd/acp.h create mode 100644 sound/soc/sof/amd/pci-rn.c create mode 100644 sound/soc/sof/amd/renoir.c create mode 100644 sound/soc/sof/mediatek/Kconfig create mode 100644 sound/soc/sof/mediatek/Makefile create mode 100644 sound/soc/sof/mediatek/adsp_helper.h create mode 100644 sound/soc/sof/mediatek/mediatek-ops.h create mode 100644 sound/soc/sof/mediatek/mt8195/Makefile create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.c create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.h create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-loader.c create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195.c create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195.h -- 2.27.0

7 32

[PATCH -next] ALSA: oss: fix compile error when OSS_DEBUG is enabled
by Bixuan Cui 01 Dec '21

01 Dec '21

Fix compile error when OSS_DEBUG is enabled: sound/core/oss/pcm_oss.c: In function 'snd_pcm_oss_set_trigger': sound/core/oss/pcm_oss.c:2055:10: error: 'substream' undeclared (first use in this function); did you mean 'csubstream'? pcm_dbg(substream->pcm, "pcm_oss: trigger = 0x%x\n", trigger); ^ Fixes: 61efcee8608c ("ALSA: oss: Use standard printk helpers") Signed-off-by: Bixuan Cui <cuibixuan(a)linux.alibaba.com> --- sound/core/oss/pcm_oss.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 82a8187..bb37665 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2052,7 +2052,7 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr int err, cmd; #ifdef OSS_DEBUG - pcm_dbg(substream->pcm, "pcm_oss: trigger = 0x%x\n", trigger); + pr_debug("pcm_oss: trigger = 0x%x\n", trigger); #endif psubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; -- 1.8.3.1

2 1

[PATCH -next] ALSA: Fix oversized kvmalloc() calls
by Bixuan Cui 01 Dec '21

01 Dec '21

The commit 7661809d493b ("mm: don't allow oversized kvmalloc() calls") limits the max allocatable memory via kvzalloc() to MAX_INT. Reported-by: syzbot+bb348e9f9a954d42746f(a)syzkaller.appspotmail.com Signed-off-by: Bixuan Cui <cuibixuan(a)linux.alibaba.com> --- sound/core/oss/pcm_plugin.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 061ba06..61fccb5 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -68,6 +68,10 @@ static int snd_pcm_plugin_alloc(struct snd_pcm_plugin *plugin, snd_pcm_uframes_t size /= 8; if (plugin->buf_frames < frames) { kvfree(plugin->buf); + + if (size > INT_MAX) + return -ENOMEM; + plugin->buf = kvzalloc(size, GFP_KERNEL); plugin->buf_frames = frames; } -- 1.8.3.1

2 5

[PATCH 0/3] ALSA: pcm: oss: various fixes
by Takashi Iwai 01 Dec '21

01 Dec '21

Hi, here is a patch set to cover several corner-cases that are found in PCM OSS layer. Mostly for addressing the missing error handlings, but also limit the period size to a reasonable value. Takashi === Takashi Iwai (3): ALSA: pcm: oss: Fix negative period/buffer sizes ALSA: pcm: oss: Limit the period size to 16MB ALSA: pcm: oss: Handle missing errors in snd_pcm_oss_change_params*() sound/core/oss/pcm_oss.c | 37 +++++++++++++++++++++++++------------ 1 file changed, 25 insertions(+), 12 deletions(-) -- 2.31.1

1 3

[PATCH v5 0/6] Multiple headphone codec driver support
by Brent Lu 30 Nov '21

30 Nov '21

Support multiple headphone drivers in same machine driver. In this case, both rt5682 and rt5682s are supported and enumerated by different ACPI HID "10EC5682" and "RTL5682". V2 Changes: - remove useless 'NULL', 'false' in if-condition - can use 'comp_ids' field alone to enumerate driver - add comma to the end of entry in structure initialization - keep the table of byt/cht/cml/icl untouched V3 Changes: - upstreamd from SOF github, PR#3200 - use new compatiable IDs to shrink the enumerate table of BYT and CHT - add 'const' to snd_soc_acpi_codecs structures V4 Changes: - add signoff to patch 4~6 V5 Changes: - none, just rebase for patch 3 conflict Brent Lu (3): ASoC: soc-acpi: add comp_ids field for machine driver matching ASoC: Intel: sof_rt5682: detect codec variant in probe function ASoC: Intel: sof_rt5682: use comp_ids to enumerate rt5682s Pierre-Louis Bossart (3): ASoC: Intel: soc-acpi-byt: shrink tables using compatible IDs ASoC: Intel: soc-acpi-cht: shrink tables using compatible IDs ASoC: Intel: soc-acpi: use const for all uses of snd_soc_acpi_codecs include/sound/soc-acpi.h | 3 + sound/soc/intel/boards/sof_rt5682.c | 34 ++------- .../intel/common/soc-acpi-intel-adl-match.c | 11 ++- .../intel/common/soc-acpi-intel-bxt-match.c | 2 +- .../intel/common/soc-acpi-intel-byt-match.c | 68 +++++++----------- .../intel/common/soc-acpi-intel-cht-match.c | 69 +++++++------------ .../intel/common/soc-acpi-intel-cml-match.c | 8 +-- .../intel/common/soc-acpi-intel-glk-match.c | 2 +- .../intel/common/soc-acpi-intel-jsl-match.c | 43 ++++-------- .../intel/common/soc-acpi-intel-kbl-match.c | 12 ++-- .../intel/common/soc-acpi-intel-skl-match.c | 2 +- .../intel/common/soc-acpi-intel-tgl-match.c | 11 ++- sound/soc/soc-acpi.c | 24 ++++++- 13 files changed, 119 insertions(+), 170 deletions(-) -- 2.25.1

4 9

[PATCH] ASoC: qdsp6: q6routing: Fix return value from msm_routing_put_audio_mixer
by Srinivas Kandagatla 30 Nov '21

30 Nov '21

msm_routing_put_audio_mixer() can return incorrect value in various scenarios. Fix this, so that change notifications are sent correctly. Fixes: e3a33673e845 ("ASoC: qdsp6: q6routing: Add q6routing driver") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6routing.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index cd74681e811e..928fd23e2c27 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -498,14 +498,16 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, struct session_data *session = &data->sessions[session_id]; if (ucontrol->value.integer.value[0]) { + if (session->port_id == be_id) + return 0; + session->port_id = be_id; snd_soc_dapm_mixer_update_power(dapm, kcontrol, 1, update); } else { - if (session->port_id == be_id) { - session->port_id = -1; + if (session->port_id == -1 || session->port_id != be_id) return 0; - } + session->port_id = -1; snd_soc_dapm_mixer_update_power(dapm, kcontrol, 0, update); } -- 2.21.0

2 2

[PATCH v5 0/3] ALSA: hda: New NHLT functions and cleanup
by Cezary Rojewski 30 Nov '21

30 Nov '21

Changes add two crucial functions: endpoint presence-check and retrieval of endpoint's BLOB (hardware configuration) to NHLT API. Outside of that, "is DMIC in NHLT?" check is being re-visited. No need to check for channel count or anything DMIC-configuration related, just straight up verify link_type presence. Changes in v5: - no code changes - actually removed the tags mentioned in v3 Changes in v4: - patch renaming nhlt_acpi_table struct to acpi_nhlt_table has been dropped due to naming conflicts with other parts of the kernel - revoked previously appended tags from Mark and Pierre as every patch required an update due to above Changes in v3: - no code changes - appended Mark's Acked-by tag for patch 4/4 - appended Pierre's Reviewed-by tag for all patches Changes in v2: - patch "ALSA hda: Drop device-argument in NHLT functions" has been dropped - updated newly added declarations in intel-nhlt.h so warning: "no-previous-prototype-for-function" and error: "use-of-undeclared-identifier" are no longer observed when CONFIG_SND_INTEL_NHLT is not enabled - added Mark's tag to the last patch of the series Amadeusz Sławiński (3): ALSA: hda: Fill gaps in NHLT endpoint-interface ALSA: hda: Simplify DMIC-in-NHLT check ASoC: Intel: Skylake: Use NHLT API to search for blob include/sound/intel-nhlt.h | 37 +++++++-- sound/hda/intel-dsp-config.c | 2 +- sound/hda/intel-nhlt.c | 102 +++++++++++++++++++++++++ sound/soc/intel/skylake/skl-nhlt.c | 102 ------------------------- sound/soc/intel/skylake/skl-pcm.c | 3 + sound/soc/intel/skylake/skl-topology.c | 29 ++++--- sound/soc/intel/skylake/skl-topology.h | 1 + sound/soc/intel/skylake/skl.h | 4 - 8 files changed, 153 insertions(+), 127 deletions(-) -- 2.25.1

3 5

[PATCH AUTOSEL 5.15 01/68] ASoC: mediatek: mt8173-rt5650: Rename Speaker control to Ext Spk
by Sasha Levin 30 Nov '21

30 Nov '21

From: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> [ Upstream commit 0a8facac0d1e38dc8b86ade6d3f0d8b33dae7c58 ] Some RT5645 and RT5650 powered platforms are using "Ext Spk" instead of "Speaker", and this is also reflected in alsa-lib configurations for the generic RT5645 usecase manager configs. Rename the "Speaker" control to "Ext Spk" in order to be able to make the userspace reuse/inherit the same configurations also for this machine, along with the others. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Link: https://lore.kernel.org/r/20211105152013.75252-1-angelogioacchino.delregno@… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index c28ebf891cb05..2cbf679f5c74b 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -30,15 +30,15 @@ static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { }; static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { - {"Speaker", NULL, "SPOL"}, - {"Speaker", NULL, "SPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, {"DMIC L1", NULL, "Int Mic"}, {"DMIC R1", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, @@ -48,7 +48,7 @@ static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { }; static const struct snd_kcontrol_new mt8173_rt5650_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), -- 2.33.0

2 16

[PATCH AUTOSEL 4.9 01/12] ASoC: mediatek: mt8173-rt5650: Rename Speaker control to Ext Spk
by Sasha Levin 30 Nov '21

30 Nov '21

From: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> [ Upstream commit 0a8facac0d1e38dc8b86ade6d3f0d8b33dae7c58 ] Some RT5645 and RT5650 powered platforms are using "Ext Spk" instead of "Speaker", and this is also reflected in alsa-lib configurations for the generic RT5645 usecase manager configs. Rename the "Speaker" control to "Ext Spk" in order to be able to make the userspace reuse/inherit the same configurations also for this machine, along with the others. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Link: https://lore.kernel.org/r/20211105152013.75252-1-angelogioacchino.delregno@… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index ba65f4157a7e0..eeb7f4c66d47a 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -38,15 +38,15 @@ static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { }; static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { - {"Speaker", NULL, "SPOL"}, - {"Speaker", NULL, "SPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, {"DMIC L1", NULL, "Int Mic"}, {"DMIC R1", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, @@ -58,7 +58,7 @@ static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { }; static const struct snd_kcontrol_new mt8173_rt5650_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), -- 2.33.0

1 0

[PATCH AUTOSEL 4.14 01/14] ASoC: mediatek: mt8173-rt5650: Rename Speaker control to Ext Spk
by Sasha Levin 30 Nov '21

30 Nov '21

From: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> [ Upstream commit 0a8facac0d1e38dc8b86ade6d3f0d8b33dae7c58 ] Some RT5645 and RT5650 powered platforms are using "Ext Spk" instead of "Speaker", and this is also reflected in alsa-lib configurations for the generic RT5645 usecase manager configs. Rename the "Speaker" control to "Ext Spk" in order to be able to make the userspace reuse/inherit the same configurations also for this machine, along with the others. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Link: https://lore.kernel.org/r/20211105152013.75252-1-angelogioacchino.delregno@… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index e69c141d8ed4c..60a6a83aa772c 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -38,15 +38,15 @@ static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { }; static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { - {"Speaker", NULL, "SPOL"}, - {"Speaker", NULL, "SPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, {"DMIC L1", NULL, "Int Mic"}, {"DMIC R1", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, @@ -58,7 +58,7 @@ static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { }; static const struct snd_kcontrol_new mt8173_rt5650_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), -- 2.33.0

1 0

[PATCH AUTOSEL 4.19 01/17] ASoC: mediatek: mt8173-rt5650: Rename Speaker control to Ext Spk
by Sasha Levin 30 Nov '21

30 Nov '21

From: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> [ Upstream commit 0a8facac0d1e38dc8b86ade6d3f0d8b33dae7c58 ] Some RT5645 and RT5650 powered platforms are using "Ext Spk" instead of "Speaker", and this is also reflected in alsa-lib configurations for the generic RT5645 usecase manager configs. Rename the "Speaker" control to "Ext Spk" in order to be able to make the userspace reuse/inherit the same configurations also for this machine, along with the others. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Link: https://lore.kernel.org/r/20211105152013.75252-1-angelogioacchino.delregno@… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index 7a89b4aad182f..6cae0e4d33710 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -30,15 +30,15 @@ static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { }; static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { - {"Speaker", NULL, "SPOL"}, - {"Speaker", NULL, "SPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, {"DMIC L1", NULL, "Int Mic"}, {"DMIC R1", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, @@ -48,7 +48,7 @@ static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { }; static const struct snd_kcontrol_new mt8173_rt5650_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), -- 2.33.0

1 2

[PATCH AUTOSEL 5.4 01/25] ASoC: mediatek: mt8173-rt5650: Rename Speaker control to Ext Spk
by Sasha Levin 30 Nov '21

30 Nov '21

From: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> [ Upstream commit 0a8facac0d1e38dc8b86ade6d3f0d8b33dae7c58 ] Some RT5645 and RT5650 powered platforms are using "Ext Spk" instead of "Speaker", and this is also reflected in alsa-lib configurations for the generic RT5645 usecase manager configs. Rename the "Speaker" control to "Ext Spk" in order to be able to make the userspace reuse/inherit the same configurations also for this machine, along with the others. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Link: https://lore.kernel.org/r/20211105152013.75252-1-angelogioacchino.delregno@… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index ef6f236752867..de3e3a55de4bf 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -30,15 +30,15 @@ static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { }; static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { - {"Speaker", NULL, "SPOL"}, - {"Speaker", NULL, "SPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, {"DMIC L1", NULL, "Int Mic"}, {"DMIC R1", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, @@ -48,7 +48,7 @@ static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { }; static const struct snd_kcontrol_new mt8173_rt5650_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), -- 2.33.0

1 2

[PATCH AUTOSEL 5.10 01/43] ASoC: mediatek: mt8173-rt5650: Rename Speaker control to Ext Spk
by Sasha Levin 30 Nov '21

30 Nov '21

From: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> [ Upstream commit 0a8facac0d1e38dc8b86ade6d3f0d8b33dae7c58 ] Some RT5645 and RT5650 powered platforms are using "Ext Spk" instead of "Speaker", and this is also reflected in alsa-lib configurations for the generic RT5645 usecase manager configs. Rename the "Speaker" control to "Ext Spk" in order to be able to make the userspace reuse/inherit the same configurations also for this machine, along with the others. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Link: https://lore.kernel.org/r/20211105152013.75252-1-angelogioacchino.delregno@… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index c28ebf891cb05..2cbf679f5c74b 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -30,15 +30,15 @@ static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { }; static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { - {"Speaker", NULL, "SPOL"}, - {"Speaker", NULL, "SPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, {"DMIC L1", NULL, "Int Mic"}, {"DMIC R1", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, @@ -48,7 +48,7 @@ static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { }; static const struct snd_kcontrol_new mt8173_rt5650_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), -- 2.33.0

1 10

Dual simulatenous output on single card not recognized
by Eldred HABERT 30 Nov '21

30 Nov '21

Hello, I bought a laptop with integrated sound output (either to speakers or headphones), and a HDMI output. Attached is the output of `aplay -l`. Under both PipeWire and PulseAudio, I only get a single output/sink (by default to the built-in output), but I can get simultaneous output to both: - under PulseAudio, by adding a `load-module module-alsa-sink device=hw:0,7` line to the config; - under PipeWire, by adding a profile to the card to enable both outputs, which correctly provides two outputs that work fine concurrently. I was directed to this mailing list after the following discussion: > It's a kernel ALSA bug if SOF actually must contain enough data for the driver to figure this out (or at least pass that data on to userspace to deal with it) > if [we are] right that hardware itself does not contain the required data, then it can only be fixed in userspace via a card database > which is one thing to do, another would be maybe a heuristic rule that attempts to guess if such cards with analog and digital sub-devices might be independent Please tell me if more details are needed—it would be nice if hardware configurations like mine could be fully supported out of the box ^^ Cheers, ~ ISSOtm

2 3

Channel distortion when converting S16_LE -> S32_LE ('plug' plugin)
by GitHub issues - opened 30 Nov '21

30 Nov '21

alsa-project/alsa-lib issue #195 was opened from dh1dm: When using a Terretec EWX2496 (ICE1712) a sample format conversion is required in some cases as S32_LE is the only supported format. This can be done by using the 'plug' plugin (e.g. 'plughw:0,0'). However, the left audio channel is distorted in a varying intensity, it sounds like a resolution loss in the lower bits. Applications with native S32_LE support (e.g. Audacity) do not show this effect. I'd suppose that the responsible code is in 'alsa-lib/src/pcm/pcm_plug.c', but up to now I could not track down this issue. Issue URL : https://github.com/alsa-project/alsa-lib/issues/195 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: meson: use dev_err_probe
by Jerome Brunet 30 Nov '21

30 Nov '21

Use dev_err_probe() helper function to handle probe deferral. It removes the open coded test for -EPROBE_DEFER but more importantly, it sets the deferral reason in debugfs which is great for debugging. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/aiu.c | 17 +++++++---------- sound/soc/meson/axg-fifo.c | 9 +++------ sound/soc/meson/axg-pdm.c | 9 +++------ sound/soc/meson/axg-spdifin.c | 6 ++---- sound/soc/meson/axg-spdifout.c | 6 ++---- sound/soc/meson/axg-tdm-formatter.c | 18 ++++++------------ sound/soc/meson/axg-tdm-interface.c | 9 +++------ sound/soc/meson/meson-card-utils.c | 4 ++-- sound/soc/meson/t9015.c | 8 ++++---- 9 files changed, 32 insertions(+), 54 deletions(-) diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index ba15d5762b0b..37036adf14ce 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -219,31 +219,29 @@ static int aiu_clk_get(struct device *dev) aiu->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(aiu->pclk)) { - if (PTR_ERR(aiu->pclk) != -EPROBE_DEFER) - dev_err(dev, "Can't get the aiu pclk\n"); + dev_err_probe(dev, PTR_ERR(aiu->pclk), + "Can't get the aiu pclk\n"); return PTR_ERR(aiu->pclk); } aiu->spdif_mclk = devm_clk_get(dev, "spdif_mclk"); if (IS_ERR(aiu->spdif_mclk)) { - if (PTR_ERR(aiu->spdif_mclk) != -EPROBE_DEFER) - dev_err(dev, "Can't get the aiu spdif master clock\n"); + dev_err_probe(dev, PTR_ERR(aiu->spdif_mclk), + "Can't get the aiu spdif master clock\n"); return PTR_ERR(aiu->spdif_mclk); } ret = aiu_clk_bulk_get(dev, aiu_i2s_ids, ARRAY_SIZE(aiu_i2s_ids), &aiu->i2s); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "Can't get the i2s clocks\n"); + dev_err_probe(dev, ret, "Can't get the i2s clocks\n"); return ret; } ret = aiu_clk_bulk_get(dev, aiu_spdif_ids, ARRAY_SIZE(aiu_spdif_ids), &aiu->spdif); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "Can't get the spdif clocks\n"); + dev_err_probe(dev, ret, "Can't get the spdif clocks\n"); return ret; } @@ -282,8 +280,7 @@ static int aiu_probe(struct platform_device *pdev) ret = device_reset(dev); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "Failed to reset device\n"); + dev_err_probe(dev, ret, "Failed to reset device\n"); return ret; } diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index b9af2d513e09..8d3184aeba60 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -352,17 +352,14 @@ int axg_fifo_probe(struct platform_device *pdev) fifo->pclk = devm_clk_get(dev, NULL); if (IS_ERR(fifo->pclk)) { - if (PTR_ERR(fifo->pclk) != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %ld\n", - PTR_ERR(fifo->pclk)); + dev_err_probe(dev, PTR_ERR(fifo->pclk), "failed to get pclk\n"); return PTR_ERR(fifo->pclk); } fifo->arb = devm_reset_control_get_exclusive(dev, NULL); if (IS_ERR(fifo->arb)) { - if (PTR_ERR(fifo->arb) != -EPROBE_DEFER) - dev_err(dev, "failed to get arb reset: %ld\n", - PTR_ERR(fifo->arb)); + dev_err_probe(dev, PTR_ERR(fifo->arb), + "failed to get arb reset\n"); return PTR_ERR(fifo->arb); } diff --git a/sound/soc/meson/axg-pdm.c b/sound/soc/meson/axg-pdm.c index bfd37d49a73e..9537fbcf56e0 100644 --- a/sound/soc/meson/axg-pdm.c +++ b/sound/soc/meson/axg-pdm.c @@ -613,24 +613,21 @@ static int axg_pdm_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { ret = PTR_ERR(priv->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk\n"); return ret; } priv->dclk = devm_clk_get(dev, "dclk"); if (IS_ERR(priv->dclk)) { ret = PTR_ERR(priv->dclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get dclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get dclk\n"); return ret; } priv->sysclk = devm_clk_get(dev, "sysclk"); if (IS_ERR(priv->sysclk)) { ret = PTR_ERR(priv->sysclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get dclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get dclk\n"); return ret; } diff --git a/sound/soc/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c index d0d09f945b48..e10bc9352667 100644 --- a/sound/soc/meson/axg-spdifin.c +++ b/sound/soc/meson/axg-spdifin.c @@ -481,16 +481,14 @@ static int axg_spdifin_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { ret = PTR_ERR(priv->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk\n"); return ret; } priv->refclk = devm_clk_get(dev, "refclk"); if (IS_ERR(priv->refclk)) { ret = PTR_ERR(priv->refclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get mclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get mclk\n"); return ret; } diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c index e769a5ee6e27..179654d76c8f 100644 --- a/sound/soc/meson/axg-spdifout.c +++ b/sound/soc/meson/axg-spdifout.c @@ -424,16 +424,14 @@ static int axg_spdifout_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { ret = PTR_ERR(priv->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk\n"); return ret; } priv->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(priv->mclk)) { ret = PTR_ERR(priv->mclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get mclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get mclk\n"); return ret; } diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index cab7fa2851aa..b8bbeda87a72 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -284,8 +284,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(formatter->pclk)) { ret = PTR_ERR(formatter->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk\n"); return ret; } @@ -293,8 +292,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->sclk = devm_clk_get(dev, "sclk"); if (IS_ERR(formatter->sclk)) { ret = PTR_ERR(formatter->sclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get sclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get sclk\n"); return ret; } @@ -302,8 +300,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->lrclk = devm_clk_get(dev, "lrclk"); if (IS_ERR(formatter->lrclk)) { ret = PTR_ERR(formatter->lrclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get lrclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get lrclk\n"); return ret; } @@ -311,8 +308,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->sclk_sel = devm_clk_get(dev, "sclk_sel"); if (IS_ERR(formatter->sclk_sel)) { ret = PTR_ERR(formatter->sclk_sel); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get sclk_sel: %d\n", ret); + dev_err_probe(dev, ret, "failed to get sclk_sel\n"); return ret; } @@ -320,8 +316,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->lrclk_sel = devm_clk_get(dev, "lrclk_sel"); if (IS_ERR(formatter->lrclk_sel)) { ret = PTR_ERR(formatter->lrclk_sel); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get lrclk_sel: %d\n", ret); + dev_err_probe(dev, ret, "failed to get lrclk_sel\n"); return ret; } @@ -329,8 +324,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->reset = devm_reset_control_get_optional_exclusive(dev, NULL); if (IS_ERR(formatter->reset)) { ret = PTR_ERR(formatter->reset); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get reset: %d\n", ret); + dev_err_probe(dev, ret, "failed to get reset\n"); return ret; } diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index db077773af7a..8f8b126de158 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -535,8 +535,7 @@ static int axg_tdm_iface_probe(struct platform_device *pdev) iface->sclk = devm_clk_get(dev, "sclk"); if (IS_ERR(iface->sclk)) { ret = PTR_ERR(iface->sclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get sclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get sclk\n"); return ret; } @@ -544,8 +543,7 @@ static int axg_tdm_iface_probe(struct platform_device *pdev) iface->lrclk = devm_clk_get(dev, "lrclk"); if (IS_ERR(iface->lrclk)) { ret = PTR_ERR(iface->lrclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get lrclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get lrclk\n"); return ret; } @@ -561,8 +559,7 @@ static int axg_tdm_iface_probe(struct platform_device *pdev) if (ret == -ENOENT) { iface->mclk = NULL; } else { - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get mclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get mclk\n"); return ret; } } diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index 29b0174f4b5c..cc6e16e31f9c 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -86,8 +86,8 @@ int meson_card_parse_dai(struct snd_soc_card *card, ret = of_parse_phandle_with_args(node, "sound-dai", "#sound-dai-cells", 0, &args); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(card->dev, "can't parse dai %d\n", ret); + dev_err_probe(card->dev, ret, + "can't parse sound-dai at %pOFn\n", node); return ret; } *dai_of_node = args.np; diff --git a/sound/soc/meson/t9015.c b/sound/soc/meson/t9015.c index 4c1349dd1e06..7a0a807f494c 100644 --- a/sound/soc/meson/t9015.c +++ b/sound/soc/meson/t9015.c @@ -259,15 +259,15 @@ static int t9015_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { - if (PTR_ERR(priv->pclk) != -EPROBE_DEFER) - dev_err(dev, "failed to get core clock\n"); + dev_err_probe(dev, PTR_ERR(priv->pclk), + "failed to get core clock\n"); return PTR_ERR(priv->pclk); } priv->avdd = devm_regulator_get(dev, "AVDD"); if (IS_ERR(priv->avdd)) { - if (PTR_ERR(priv->avdd) != -EPROBE_DEFER) - dev_err(dev, "failed to AVDD\n"); + dev_err_probe(dev, PTR_ERR(priv->avdd), + "failed to get AVDD\n"); return PTR_ERR(priv->avdd); } -- 2.34.0

2 2

[PATCH v8 0/3] This patches provide ADSP IPC support for MT8195
by allen-kh.cheng 30 Nov '21

30 Nov '21

From: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> Mediatek ADSP IPC is used to send notification or short message between processors with dsp.· It will place the message to the share buffer and will access the ADSP mailbox registers to kick dsp. Two mailboxes used to send notification or short message between processors with dsp changes since v7: - add mtk prefix for adsp ipc functions - rename adsp_mbox_ch_info to mtk_adsp_mbox_ch_info - rename adsp_mbox_chan_ops to mtk_adsp_mbox_chan_ops - remove incorrect reviewers in commit message changes since v6: - dt-bindings: change example dtsi node. - rename config MTK_ADSP_IPC_MBOX to MTK_ADSP_MBOX - remove unused variable - add reviewers changes since v5: - remove some redundant code changes since v4: - use switch ... case in adsp_ipc_recv - add error handling path for chan_name pointer - refine some code to be concise changes since v3: - reorder MTK_ADSP_IPC_MBOX config - remove some redundant code - remove lock in mtk-adsp-mailbox changes since v2: - separate adsp_mailbox into two instances changes since v1: - fix dt_binding_check error Allen-KH Cheng (3): dt-bindings: mediatek: add adsp-mbox document firmware: mediatek: add adsp ipc protocol interface mailbox: mediatek: add support for adsp mailbox controller .../bindings/mailbox/mtk,adsp-mbox.yaml | 52 ++++++ drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 9 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 138 ++++++++++++++ drivers/mailbox/Kconfig | 7 + drivers/mailbox/Makefile | 2 + drivers/mailbox/mtk-adsp-mailbox.c | 175 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 70 +++++++ 10 files changed, 457 insertions(+) create mode 100644 Documentation/devicetree/bindings/mailbox/mtk,adsp-mbox.yaml create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 drivers/mailbox/mtk-adsp-mailbox.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

2 4

[PATCH v3] ASoC: dt-bindings: davinci-mcasp: convert McASP bindings to yaml schema
by Jayesh Choudhary 29 Nov '21

29 Nov '21

Convert the bindings for McASP controllers for TI SOCs from txt to YAML schema. Adds additional properties 'clocks', 'clock-names', 'power-domains', '#sound-dai-cells', 'num-serializer' and 'port' which were not there in the txt file. Adds 'dmas' and 'dma-names' in the example which were not there in the txt file. Changes 'interrupts' and 'interrupt-names' from optional to required properties. Signed-off-by: Jayesh Choudhary <j-choudhary(a)ti.com> --- Changelog: v3: - removes maxItems from 'clock-names' v2: - changes the commit message - modifies the properties 'clocks', 'clock-names', 'dma-names', 'dmas', 'interrupts' and 'interrupt-names' according to the arm SOCs - adds 'port' and 'num-serializer' as node properties .../bindings/sound/davinci-mcasp-audio.txt | 86 --------- .../bindings/sound/davinci-mcasp-audio.yaml | 178 ++++++++++++++++++ 2 files changed, 178 insertions(+), 86 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt deleted file mode 100644 index bd863bd69501..000000000000 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ /dev/null @@ -1,86 +0,0 @@ -Texas Instruments McASP controller - -Required properties: -- compatible : - "ti,dm646x-mcasp-audio" : for DM646x platforms - "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms - "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx) - "ti,dra7-mcasp-audio" : for DRA7xx platforms - "ti,omap4-mcasp-audio" : for OMAP4 - -- reg : Should contain reg specifiers for the entries in the reg-names property. -- reg-names : Should contain: - * "mpu" for the main registers (required). For compatibility with - existing software, it is recommended this is the first entry. - * "dat" for separate data port register access (optional). -- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF, - IEC60958-1, and AES-3 formats. -- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted - or received over one serializer. -- serial-dir : A list of serializer configuration. Each entry is a number - indication for serializer pin direction. - (0 - INACTIVE, 1 - TX, 2 - RX) -- dmas: two element list of DMA controller phandles and DMA request line - ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. The dma - identifiers must be "rx" and "tx". - -Optional properties: - -- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0 -- tx-num-evt : FIFO levels. -- rx-num-evt : FIFO levels. -- dismod : Specify the drive on TX pin during inactive slots - 0 : 3-state - 2 : logic low - 3 : logic high - Defaults to 'logic low' when the property is not present -- sram-size-playback : size of sram to be allocated during playback -- sram-size-capture : size of sram to be allocated during capture -- interrupts : Interrupt numbers for McASP -- interrupt-names : Known interrupt names are "tx" and "rx" -- pinctrl-0: Should specify pin control group used for this controller. -- pinctrl-names: Should contain only one value - "default", for more details - please refer to pinctrl-bindings.txt -- fck_parent : Should contain a valid clock name which will be used as parent - for the McASP fck -- auxclk-fs-ratio: When McASP is bus master indicates the ratio between AUCLK - and FS rate if applicable: - AUCLK rate = auxclk-fs-ratio * FS rate - -Optional GPIO support: -If any McASP pin need to be used as GPIO then the McASP node must have: -... - gpio-controller - #gpio-cells = <2>; -... - -When requesting a GPIO, the first parameter is the PIN index in McASP_P* -registers. -For example to request the AXR2 pin of mcasp8: -function-gpios = <&mcasp8 2 0>; - -Or to request the ACLKR pin of mcasp8: -function-gpios = <&mcasp8 29 0>; - -For generic gpio information, please refer to bindings/gpio/gpio.txt - -Example: - -mcasp0: mcasp0@1d00000 { - compatible = "ti,da830-mcasp-audio"; - reg = <0x100000 0x3000>; - reg-names "mpu"; - interrupts = <82>, <83>; - interrupt-names = "tx", "rx"; - op-mode = <0>; /* MCASP_IIS_MODE */ - tdm-slots = <2>; - serial-dir = < - 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ - 0 0 0 0 - 0 0 0 1 - 2 0 0 0 >; - tx-num-evt = <1>; - rx-num-evt = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml new file mode 100644 index 000000000000..c4d3f56470bf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -0,0 +1,178 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/davinci-mcasp-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: McASP Controller for TI SoCs + +maintainers: + - Jayesh Choudhary <j-choudhary(a)ti.com> + +properties: + compatible: + enum: + - ti,dm646x-mcasp-audio + - ti,da830-mcasp-audio + - ti,am33xx-mcasp-audio + - ti,dra7-mcasp-audio + - ti,omap4-mcasp-audio + + reg: + minItems: 1 + items: + - description: main registers + - description: data port register + + reg-names: + minItems: 1 + items: + - const: mpu + - const: dat + + op-mode: + description: I2S - 0 or DIT - 1 mode + enum: + - 0 + - 1 + + tdm-slots: + maxItems: 1 + + serial-dir: + description: + A list of serializer configuration + Entry is indication for serializer pin direction + 0 - Inactive, 1 - TX, 2 - RX + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 16 + items: + minimum: 0 + maximum: 2 + default: 0 + + dmas: + minItems: 1 + items: + - description: transmission DMA channel + - description: reception DMA channel + + dma-names: + minItems: 1 + items: + - const: tx + - const: rx + + ti,hwmods: + $ref: /schemas/types.yaml#/definitions/string + description: Name of hwmod associated with McASP + maxItems: 1 + deprecated: true + + tx-num-evt: + maxItems: 1 + + rx-num-evt: + maxItems: 1 + + dismod: + enum: + - 0 + - 2 + - 3 + default: 2 + + sram-size-playback: + maxItems: 1 + + sram-size-capture: + maxItems: 1 + + interrupts: + minItems: 1 + items: + - description: TX FIFO interrupt + - description: RX FIFO interrupt + + interrupt-names: + oneOf: + - minItems: 1 + items: + - const: tx + - const: rx + - const: common + + fck_parent: + description: parent clock for McASP fck + maxItems: 1 + + auxclk-fs-ratio: + description: ratio of AUCLK and FS if applicable + maxItems: 1 + + gpio-controller: true + + "#gpio-cells": + const: 2 + + function-gpios: + maxItems: 1 + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + minItems: 1 + items: + - const: fck + - const: ahclkx + - const: ahclkr + + power-domains: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + num-serializer: + maxItems: 1 + + port: + type: object + +required: + - compatible + - reg + - reg-names + - dmas + - dma-names + - interrupts + - interrupt-names + - serial-dir + - op-mode + - tdm-slots + +additionalProperties: false + +examples: + - | + mcasp0: mcasp0@1d00000 { + compatible = "ti,da830-mcasp-audio"; + reg = <0x100000 0x3000>; + reg-names = "mpu"; + interrupts = <82>, <83>; + interrupt-names = "tx", "rx"; + op-mode = <0>; /* MCASP_IIS_MODE */ + tdm-slots = <2>; + dmas = <&main_udmap 0xc400>, <&main_udmap 0x4400>; + dma-names = "tx", "rx"; + serial-dir = < + 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ + 0 0 0 0 + 0 0 0 1 + 2 0 0 0 >; + tx-num-evt = <1>; + rx-num-evt = <1>; + }; -- 2.17.1

3 5

[PATCH 0/5] ASoC: soc-pcm: tidyup soc_pcm_pointer()'s delay update method
by Kuninori Morimoto 29 Nov '21

29 Nov '21

Hi Mark Current soc_pcm_pointer() is checking runtime->delay, but it might be updated silently by component's .point callback. It is strange and difficult to find/know the issue. This patch adds .delay callback for component, and solve the issue. Kuninori Morimoto (5): ASoC: soc-dai: update snd_soc_dai_delay() to snd_soc_pcm_dai_delay() ASoC: soc-component: add snd_soc_pcm_component_delay() ASoC: amd: acp-pcm-dma: add .delay support ASoC: intel: sst-mfld-platform-pcm: add .delay support ASoC: soc-pcm: tidyup soc_pcm_pointer()'s delay update method include/sound/soc-component.h | 4 ++ include/sound/soc-dai.h | 4 +- sound/soc/amd/acp-pcm-dma.c | 15 +++++++- sound/soc/amd/acp.h | 1 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 14 ++++++- sound/soc/soc-component.c | 28 ++++++++++++++ sound/soc/soc-dai.c | 40 ++++++++++++++------ sound/soc/soc-pcm.c | 29 +++----------- 8 files changed, 95 insertions(+), 40 deletions(-) -- 2.25.1

3 12

[PATCH v2] ASoC: sunxi: sun4i-spdif: Implement IEC958 control
by Jernej Skrabec 29 Nov '21

29 Nov '21

SPDIF core is capable of sending custom status. Implement IEC958 control handling. Signed-off-by: Jernej Skrabec <jernej.skrabec(a)gmail.com> --- Changes from v1: - return 1 if there was a change in sun4i_spdif_set_status() - added locking for reading and writing status sound/soc/sunxi/sun4i-spdif.c | 115 ++++++++++++++++++++++++++++++++++ 1 file changed, 115 insertions(+) diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index a10949bf0ca1..17090f43150e 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -21,6 +21,8 @@ #include <linux/platform_device.h> #include <linux/pm_runtime.h> #include <linux/reset.h> +#include <linux/spinlock.h> +#include <sound/asoundef.h> #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -186,6 +188,7 @@ struct sun4i_spdif_dev { struct regmap *regmap; struct snd_dmaengine_dai_dma_data dma_params_tx; const struct sun4i_spdif_quirks *quirks; + spinlock_t lock; }; static void sun4i_spdif_configure(struct sun4i_spdif_dev *host) @@ -385,11 +388,122 @@ static int sun4i_spdif_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int sun4i_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + + return 0; +} + +static int sun4i_spdif_get_status_mask(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 *status = ucontrol->value.iec958.status; + + status[0] = 0xff; + status[1] = 0xff; + status[2] = 0xff; + status[3] = 0xff; + status[4] = 0xff; + status[5] = 0x03; + + return 0; +} + +static int sun4i_spdif_get_status(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + u8 *status = ucontrol->value.iec958.status; + unsigned long flags; + unsigned int reg; + + spin_lock_irqsave(&host->lock, flags); + + regmap_read(host->regmap, SUN4I_SPDIF_TXCHSTA0, &reg); + + status[0] = reg & 0xff; + status[1] = (reg >> 8) & 0xff; + status[2] = (reg >> 16) & 0xff; + status[3] = (reg >> 24) & 0xff; + + regmap_read(host->regmap, SUN4I_SPDIF_TXCHSTA1, &reg); + + status[4] = reg & 0xff; + status[5] = (reg >> 8) & 0x3; + + spin_unlock_irqrestore(&host->lock, flags); + + return 0; +} + +static int sun4i_spdif_set_status(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + u8 *status = ucontrol->value.iec958.status; + unsigned long flags; + unsigned int reg; + bool chg0, chg1; + + spin_lock_irqsave(&host->lock, flags); + + reg = (u32)status[3] << 24; + reg |= (u32)status[2] << 16; + reg |= (u32)status[1] << 8; + reg |= (u32)status[0]; + + regmap_update_bits_check(host->regmap, SUN4I_SPDIF_TXCHSTA0, + GENMASK(31,0), reg, &chg0); + + reg = (u32)status[5] << 8; + reg |= (u32)status[4]; + + regmap_update_bits_check(host->regmap, SUN4I_SPDIF_TXCHSTA1, + GENMASK(9,0), reg, &chg1); + + reg = SUN4I_SPDIF_TXCFG_CHSTMODE; + if (status[0] & IEC958_AES0_NONAUDIO) + reg |= SUN4I_SPDIF_TXCFG_NONAUDIO; + + regmap_update_bits(host->regmap, SUN4I_SPDIF_TXCFG, + SUN4I_SPDIF_TXCFG_CHSTMODE | + SUN4I_SPDIF_TXCFG_NONAUDIO, reg); + + spin_unlock_irqrestore(&host->lock, flags); + + return chg0 || chg1; +} + +static struct snd_kcontrol_new sun4i_spdif_controls[] = { + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, MASK), + .info = sun4i_spdif_info, + .get = sun4i_spdif_get_status_mask + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = sun4i_spdif_info, + .get = sun4i_spdif_get_status, + .put = sun4i_spdif_set_status + } +}; + static int sun4i_spdif_soc_dai_probe(struct snd_soc_dai *dai) { struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(dai); snd_soc_dai_init_dma_data(dai, &host->dma_params_tx, NULL); + snd_soc_add_dai_controls(dai, sun4i_spdif_controls, + ARRAY_SIZE(sun4i_spdif_controls)); + return 0; } @@ -512,6 +626,7 @@ static int sun4i_spdif_probe(struct platform_device *pdev) return -ENOMEM; host->pdev = pdev; + spin_lock_init(&host->lock); /* Initialize this copy of the CPU DAI driver structure */ memcpy(&host->cpu_dai_drv, &sun4i_spdif_dai, sizeof(sun4i_spdif_dai)); -- 2.34.0

2 1

[PATCH] ASoC: sun8i-codec: Add AIF, ADC, and DAC volume controls
by Samuel Holland 29 Nov '21

29 Nov '21

This allows changing the volume of each digital input/output independently, and provides the only "master volume" for the DAC. (The ADC also has a gain control on the analog side.) While the hardware supports digital gain up to +72dB, the controls here are limited to +24dB maximum, as any gain above that level makes volume sliders difficult to use, and is extremely likely to cause clipping. Signed-off-by: Samuel Holland <samuel(a)sholland.org> --- sound/soc/sunxi/sun8i-codec.c | 56 +++++++++++++++++++++++++++++++++++ 1 file changed, 56 insertions(+) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 518bfb724a5b..0bea2162f68d 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -21,6 +21,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/tlv.h> #define SUN8I_SYSCLK_CTL 0x00c #define SUN8I_SYSCLK_CTL_AIF1CLK_ENA 11 @@ -72,6 +73,12 @@ #define SUN8I_AIF1_MXR_SRC_AD0R_MXR_SRC_AIF2DACR 10 #define SUN8I_AIF1_MXR_SRC_AD0R_MXR_SRC_ADCR 9 #define SUN8I_AIF1_MXR_SRC_AD0R_MXR_SRC_AIF2DACL 8 +#define SUN8I_AIF1_VOL_CTRL1 0x050 +#define SUN8I_AIF1_VOL_CTRL1_AD0L_VOL 8 +#define SUN8I_AIF1_VOL_CTRL1_AD0R_VOL 0 +#define SUN8I_AIF1_VOL_CTRL3 0x058 +#define SUN8I_AIF1_VOL_CTRL3_DA0L_VOL 8 +#define SUN8I_AIF1_VOL_CTRL3_DA0R_VOL 0 #define SUN8I_AIF2_ADCDAT_CTRL 0x084 #define SUN8I_AIF2_ADCDAT_CTRL_AIF2_ADCL_ENA 15 #define SUN8I_AIF2_ADCDAT_CTRL_AIF2_ADCR_ENA 14 @@ -91,6 +98,12 @@ #define SUN8I_AIF2_MXR_SRC_ADCR_MXR_SRC_AIF1DA1R 10 #define SUN8I_AIF2_MXR_SRC_ADCR_MXR_SRC_AIF2DACL 9 #define SUN8I_AIF2_MXR_SRC_ADCR_MXR_SRC_ADCR 8 +#define SUN8I_AIF2_VOL_CTRL1 0x090 +#define SUN8I_AIF2_VOL_CTRL1_ADCL_VOL 8 +#define SUN8I_AIF2_VOL_CTRL1_ADCR_VOL 0 +#define SUN8I_AIF2_VOL_CTRL2 0x098 +#define SUN8I_AIF2_VOL_CTRL2_DACL_VOL 8 +#define SUN8I_AIF2_VOL_CTRL2_DACR_VOL 0 #define SUN8I_AIF3_CLK_CTRL_AIF3_CLK_SRC_AIF1 (0x0 << 0) #define SUN8I_AIF3_CLK_CTRL_AIF3_CLK_SRC_AIF2 (0x1 << 0) #define SUN8I_AIF3_CLK_CTRL_AIF3_CLK_SRC_AIF1CLK (0x2 << 0) @@ -102,8 +115,14 @@ #define SUN8I_ADC_DIG_CTRL_ENAD 15 #define SUN8I_ADC_DIG_CTRL_ADOUT_DTS 2 #define SUN8I_ADC_DIG_CTRL_ADOUT_DLY 1 +#define SUN8I_ADC_VOL_CTRL 0x104 +#define SUN8I_ADC_VOL_CTRL_ADCL_VOL 8 +#define SUN8I_ADC_VOL_CTRL_ADCR_VOL 0 #define SUN8I_DAC_DIG_CTRL 0x120 #define SUN8I_DAC_DIG_CTRL_ENDA 15 +#define SUN8I_DAC_VOL_CTRL 0x124 +#define SUN8I_DAC_VOL_CTRL_DACL_VOL 8 +#define SUN8I_DAC_VOL_CTRL_DACR_VOL 0 #define SUN8I_DAC_MXR_SRC 0x130 #define SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_AIF1DA0L 15 #define SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_AIF1DA1L 14 @@ -696,6 +715,41 @@ static struct snd_soc_dai_driver sun8i_codec_dais[] = { }, }; +static const DECLARE_TLV_DB_SCALE(sun8i_codec_vol_scale, -12000, 75, 1); + +static const struct snd_kcontrol_new sun8i_codec_controls[] = { + SOC_DOUBLE_TLV("AIF1 AD0 Capture Volume", + SUN8I_AIF1_VOL_CTRL1, + SUN8I_AIF1_VOL_CTRL1_AD0L_VOL, + SUN8I_AIF1_VOL_CTRL1_AD0R_VOL, + 0xc0, 0, sun8i_codec_vol_scale), + SOC_DOUBLE_TLV("AIF1 DA0 Playback Volume", + SUN8I_AIF1_VOL_CTRL3, + SUN8I_AIF1_VOL_CTRL3_DA0L_VOL, + SUN8I_AIF1_VOL_CTRL3_DA0R_VOL, + 0xc0, 0, sun8i_codec_vol_scale), + SOC_DOUBLE_TLV("AIF2 ADC Capture Volume", + SUN8I_AIF2_VOL_CTRL1, + SUN8I_AIF2_VOL_CTRL1_ADCL_VOL, + SUN8I_AIF2_VOL_CTRL1_ADCR_VOL, + 0xc0, 0, sun8i_codec_vol_scale), + SOC_DOUBLE_TLV("AIF2 DAC Playback Volume", + SUN8I_AIF2_VOL_CTRL2, + SUN8I_AIF2_VOL_CTRL2_DACL_VOL, + SUN8I_AIF2_VOL_CTRL2_DACR_VOL, + 0xc0, 0, sun8i_codec_vol_scale), + SOC_DOUBLE_TLV("ADC Capture Volume", + SUN8I_ADC_VOL_CTRL, + SUN8I_ADC_VOL_CTRL_ADCL_VOL, + SUN8I_ADC_VOL_CTRL_ADCR_VOL, + 0xc0, 0, sun8i_codec_vol_scale), + SOC_DOUBLE_TLV("DAC Playback Volume", + SUN8I_DAC_VOL_CTRL, + SUN8I_DAC_VOL_CTRL_DACL_VOL, + SUN8I_DAC_VOL_CTRL_DACR_VOL, + 0xc0, 0, sun8i_codec_vol_scale), +}; + static int sun8i_codec_aif_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1215,6 +1269,8 @@ static int sun8i_codec_component_probe(struct snd_soc_component *component) } static const struct snd_soc_component_driver sun8i_soc_component = { + .controls = sun8i_codec_controls, + .num_controls = ARRAY_SIZE(sun8i_codec_controls), .dapm_widgets = sun8i_codec_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(sun8i_codec_dapm_widgets), .dapm_routes = sun8i_codec_dapm_routes, -- 2.32.0

2 1

[PATCH] ASoC: qcom: Add support for ALC5682I-VS codec
by lvzhaoxiong 29 Nov '21

29 Nov '21

Qcom machine driver adds rt5682s support in this patch. Card name can be specified from dts by model property, and driver makes use of the name to distinguish which headset codec is on the board. Signed-off-by: lvzhaoxiong <lvzhaoxiong(a)huaqin.corp-partner.google.com> --- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/sc7180.c | 24 +++++++++++++++++++----- 2 files changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index b2173847dc47..cf3e151bb635 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -169,6 +169,7 @@ config SND_SOC_SC7180 select SND_SOC_LPASS_SC7180 select SND_SOC_MAX98357A select SND_SOC_RT5682_I2C + select SND_SOC_RT5682S select SND_SOC_ADAU7002 help To add support for audio on Qualcomm Technologies Inc. diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index 768566bb57a5..2fff764a00a7 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -17,6 +17,7 @@ #include <uapi/linux/input-event-codes.h> #include "../codecs/rt5682.h" +#include "../codecs/rt5682s.h" #include "common.h" #include "lpass.h" @@ -128,7 +129,21 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream) struct sc7180_snd_data *data = snd_soc_card_get_drvdata(card); struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - int ret; + int pll_id, pll_source, pll_in, pll_out, clk_id, ret; + + if (!(strcmp(card->name, "sc7180-rt5682-max98357a-1mic"))) { + pll_source = RT5682_PLL1_S_MCLK; + pll_id = 0; + clk_id = RT5682_SCLK_S_PLL1; + pll_out = RT5682_PLL1_FREQ; + pll_in = DEFAULT_MCLK_RATE; + } else if (!(strcmp(card->name, "sc7180-rt5682s-max98357a-1mic"))) { + pll_source = RT5682S_PLL_S_MCLK; + pll_id = RT5682S_PLL2; + clk_id = RT5682S_SCLK_S_PLL2; + pll_out = RT5682_PLL1_FREQ; + pll_in = DEFAULT_MCLK_RATE; + } switch (cpu_dai->id) { case MI2S_PRIMARY: @@ -145,16 +160,15 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream) SND_SOC_DAIFMT_I2S); /* Configure PLL1 for codec */ - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, - DEFAULT_MCLK_RATE, RT5682_PLL1_FREQ); + ret = snd_soc_dai_set_pll(codec_dai, pll_id, pll_source, + pll_in, pll_out); if (ret) { dev_err(rtd->dev, "can't set codec pll: %d\n", ret); return ret; } /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, - RT5682_PLL1_FREQ, + ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, pll_out, SND_SOC_CLOCK_IN); if (ret) dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", -- 2.17.1

2 1

ALSA: hda: Make proper use of timecounter
by Thomas Gleixner 29 Nov '21

29 Nov '21

HDA uses a timecounter to read a hardware clock running at 24 MHz. The conversion factor is set with a mult value of 125 and a shift value of 0, which is not converting the hardware clock to nanoseconds, it is converting to 1/3 nanoseconds because the conversion factor from 24Mhz to nanoseconds is 125/3. The usage sites divide the "nanoseconds" value returned by timecounter_read() by 3 to get a real nanoseconds value. There is a lengthy comment in azx_timecounter_init() explaining this choice. That comment makes blatantly wrong assumptions about how timecounters work and what can overflow. The comment says: * Applying the 1/3 factor as part of the multiplication * requires at least 20 bits for a decent precision, however * overflows occur after about 4 hours or less, not a option. timecounters operate on time deltas between two readouts of a clock and use the mult/shift pair to calculate a precise nanoseconds value: delta_nsec = (delta_clock * mult) >> shift; The fractional part is also taken into account and preserved to prevent accumulated rounding errors. For details see cyclecounter_cyc2ns(). The mult/shift pair has to be chosen so that the multiplication of the maximum expected delta value does not result in a 64bit overflow. As the counter wraps around on 32bit, the maximum observable delta between two reads is (1 << 32) - 1 which is about 178.9 seconds. That in turn means the maximum multiplication factor which fits into an u32 will not cause a 64bit overflow ever because it's guaranteed that: ((1 << 32) - 1) ^ 2 < (1 << 64) The resulting correct multiplication factor is 2796202667 and the shift value is 26, i.e. 26 bit precision. The overflow of the multiplication would happen exactly at a clock readout delta of 6597069765 which is way after the wrap around of the hardware clock at around 274.8 seconds which is off from the claimed 4 hours by more than an order of magnitude. If the counter ever wraps around the last read value then the calculation is off by the number of wrap arounds times 178.9 seconds because the overflow cannot be observed. Use clocks_calc_mult_shift(), which calculates the most accurate mult/shift pair based on the given clock frequency, and remove the bogus comment along with the divisions at the readout sites. Fixes: 5d890f591d15 ("ALSA: hda: support for wallclock timestamps") Signed-off-by: Thomas Gleixner <tglx(a)linutronix.de> --- sound/hda/hdac_stream.c | 14 ++++---------- sound/pci/hda/hda_controller.c | 1 - sound/soc/intel/skylake/skl-pcm.c | 1 - 3 files changed, 4 insertions(+), 12 deletions(-) --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -534,17 +534,11 @@ static void azx_timecounter_init(struct cc->mask = CLOCKSOURCE_MASK(32); /* - * Converting from 24 MHz to ns means applying a 125/3 factor. - * To avoid any saturation issues in intermediate operations, - * the 125 factor is applied first. The division is applied - * last after reading the timecounter value. - * Applying the 1/3 factor as part of the multiplication - * requires at least 20 bits for a decent precision, however - * overflows occur after about 4 hours or less, not a option. + * Calculate the optimal mult/shift values. The counter wraps + * around after ~178.9 seconds. */ - - cc->mult = 125; /* saturation after 195 years */ - cc->shift = 0; + clocks_calc_mult_shift(&cc->mult, &cc->shift, 24000000, + NSEC_PER_SEC, 178); nsec = 0; /* audio time is elapsed time since trigger */ timecounter_init(tc, cc, nsec); --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -504,7 +504,6 @@ static int azx_get_time_info(struct snd_ snd_pcm_gettime(substream->runtime, system_ts); nsec = timecounter_read(&azx_dev->core.tc); - nsec = div_u64(nsec, 3); /* can be optimized */ if (audio_tstamp_config->report_delay) nsec = azx_adjust_codec_delay(substream, nsec); --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1251,7 +1251,6 @@ static int skl_platform_soc_get_time_inf snd_pcm_gettime(substream->runtime, system_ts); nsec = timecounter_read(&hstr->tc); - nsec = div_u64(nsec, 3); /* can be optimized */ if (audio_tstamp_config->report_delay) nsec = skl_adjust_codec_delay(substream, nsec);

3 3

[PATCH] ALSA: hda/cs8409: Set PMSG_ON earlier inside cs8409 driver
by Vitaly Rodionov 29 Nov '21

29 Nov '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> For cs8409, it is required to run Jack Detect on resume. Jack Detect on cs8409+cs42l42 requires an interrupt from cs42l42 to be sent to cs8409 which is propogated to the driver via an unsolicited event. However, the hda_codec drops unsolicited events if the power_state is not set to PMSG_ON. Which is set at the end of the resume call. This means there is a race condition between setting power_state to PMSG_ON and receiving the interrupt. To solve this, we can add an API to set the power_state earlier and call that before we start Jack Detect. This does not cause issues, since we know inside our driver that we are already initialized, and ready to handle the unsolicited events. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/pci/hda/hda_local.h | 8 ++++++++ sound/pci/hda/patch_cs8409.c | 5 +++++ 2 files changed, 13 insertions(+) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 4662a47add7e..96f78daf90f0 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -439,6 +439,14 @@ int snd_hda_codec_set_pin_target(struct hda_codec *codec, hda_nid_t nid, #define for_each_hda_codec_node(nid, codec) \ for ((nid) = (codec)->core.start_nid; (nid) < (codec)->core.end_nid; (nid)++) +/* Set the codec power_state flag to indicate to allow unsol event handling; + * see hda_codec_unsol_event() in hda_bind.c. Calling this might confuse the + * state tracking, so use with care. + */ +static inline void snd_hda_codec_allow_unsol_events(struct hda_codec *codec) { + codec->core.dev.power.power_state = PMSG_ON; +} + /* * get widget capabilities */ diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 31ff11ab868e..039b9f2f8e94 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -750,6 +750,11 @@ static void cs42l42_resume(struct sub_codec *cs42l42) if (cs42l42->full_scale_vol) cs8409_i2c_write(cs42l42, 0x2001, 0x01); + /* we have to explicitly allow unsol event handling even during the + * resume phase so that the jack event is processed properly + */ + snd_hda_codec_allow_unsol_events(cs42l42->codec); + cs42l42_enable_jack_detect(cs42l42); } -- 2.25.1

2 1

[PATCH v7 0/3] This patches provide ADSP IPC support for MT8195
by allen-kh.cheng 29 Nov '21

29 Nov '21

From: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> Mediatek ADSP IPC is used to send notification or short message between processors with dsp.· It will place the message to the share buffer and will access the ADSP mailbox registers to kick dsp. Two mailboxes used to send notification or short message between processors with dsp changes since v6: - dt-bindings: change example dtsi node. - change config MTK_ADSP_IPC_MBOX to MTK_ADSP_MBOX - remove unused variable - add reviewers changes since v5: - remove some redundant code changes since v4: - use switch ... case in adsp_ipc_recv - add error handling path for chan_name pointer - refine some code to be concise changes since v3: - reorder MTK_ADSP_IPC_MBOX config - remove some redundant code - remove lock in mtk-adsp-mailbox changes since v2: - separate adsp_mailbox into two instances changes since v1: - fix dt_binding_check error Allen-KH Cheng (3): dt-bindings: mediatek: add adsp-mbox document firmware: mediatek: add adsp ipc protocol interface mailbox: mediatek: add support for adsp mailbox controller .../bindings/mailbox/mtk,adsp-mbox.yaml | 52 ++++++ drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 9 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 138 ++++++++++++++ drivers/mailbox/Kconfig | 7 + drivers/mailbox/Makefile | 2 + drivers/mailbox/mtk-adsp-mailbox.c | 175 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 70 +++++++ 10 files changed, 457 insertions(+) create mode 100644 Documentation/devicetree/bindings/mailbox/mtk,adsp-mbox.yaml create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 drivers/mailbox/mtk-adsp-mailbox.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

2 6

Shouldn't there be an error..
by GitHub issues - edited 28 Nov '21

28 Nov '21

alsa-project/alsa-utils issue #121 was edited from auwsom: ..when I type `aplay test.wav` and there is "no soundcards found" when I type `aplay -l`? Instead of nothing. How would I know to check if there is a soundcard, versus some other component of was causing no sound to be heard. Thanks Issue URL : https://github.com/alsa-project/alsa-utils/issues/121 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Shouldn't there be an error..
by GitHub issues - opened 28 Nov '21

28 Nov '21

alsa-project/alsa-utils issue #121 was opened from auwsom: ..when I type `aplay test.wav` and there is "no soundcards found" when I type `aplay -l`? Issue URL : https://github.com/alsa-project/alsa-utils/issues/121 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Is it a regression? 5.15.5: No sound with speaker-test. 5.12.13: it works.
by u34＠net9.ga 27 Nov '21

27 Nov '21

The following looks to me a regression in the linux sound driver. alsa-info.sh output is included at the bottom of this message. There are 2 outputs. An output with a 5.12.13 kernel. The other output is with a 5.15.5. They were obtained with the same PC. With 5.12.13, the speaker-test program works as expected. There is no sound when the speaker-test program is ran with 5.15.5. 5.15.5 has options snd-hda-intel id=PCH index=0 options snd-hda-intel id=HDMI index=1 , which 5.12.13 does not have. I have tried various combinations of the knobs of alsamixer. The dmesg section for the 5.15.5 kernal is 2 lines shorter then for the former kernel. In addition, the total number of lines in the output of the 5.15.5 kernel is greater, compared to the 5.12.13 output. I don't know if it is of any significance. alsa-info output with 5.12.13: ============================== upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.5.0 !!################################ !!Script ran on: Fri Nov 26 05:51:28 UTC 2021 !!Linux Distribution !!------------------ Arch Linux \r (\l) Arch Linux release DISTRIB_ID=Arch DISTRIB_DESCRIPTION="Arch Linux" NAME="ArchBang Linux" ID=arch PRETTY_NAME="ArchBang Linux" !!DMI Information !!--------------- Manufacturer: Product Name: Product Version: Firmware Version: FLB8510H.86A.0131.2017.0516.1705 System SKU: To be filled by O.E.M. Board Vendor: Intel Corporation Board Name: DB85FL !!ACPI Device Status Information !!--------------- /sys/bus/acpi/devices/INT3F0D:00/status 15 /sys/bus/acpi/devices/PNP0103:00/status 15 /sys/bus/acpi/devices/PNP0C01:00/status 15 /sys/bus/acpi/devices/PNP0C04:00/status 31 /sys/bus/acpi/devices/PNP0C0C:00/status 11 /sys/bus/acpi/devices/PNP0C0F:00/status 9 /sys/bus/acpi/devices/PNP0C0F:01/status 9 /sys/bus/acpi/devices/PNP0C0F:02/status 9 /sys/bus/acpi/devices/PNP0C0F:03/status 9 /sys/bus/acpi/devices/PNP0C0F:04/status 9 /sys/bus/acpi/devices/PNP0C0F:05/status 9 /sys/bus/acpi/devices/PNP0C0F:06/status 9 /sys/bus/acpi/devices/PNP0C0F:07/status 9 /sys/bus/acpi/devices/device:29/status 15 /sys/bus/acpi/devices/device:2a/status 15 /sys/bus/acpi/devices/device:2b/status 15 /sys/bus/acpi/devices/device:2c/status 15 /sys/bus/acpi/devices/device:2d/status 15 /sys/bus/acpi/devices/device:2e/status 15 /sys/bus/acpi/devices/device:2f/status 15 /sys/bus/acpi/devices/device:30/status 15 /sys/bus/acpi/devices/device:31/status 15 /sys/bus/acpi/devices/device:32/status 15 /sys/bus/acpi/devices/device:33/status 15 /sys/bus/acpi/devices/device:34/status 15 /sys/bus/acpi/devices/device:35/status 15 /sys/bus/acpi/devices/device:36/status 15 /sys/bus/acpi/devices/device:37/status 15 /sys/bus/acpi/devices/device:38/status 15 /sys/bus/acpi/devices/device:39/status 15 /sys/bus/acpi/devices/device:3a/status 15 /sys/bus/acpi/devices/device:3b/status 15 /sys/bus/acpi/devices/device:3c/status 15 /sys/bus/acpi/devices/device:3d/status 15 /sys/bus/acpi/devices/device:4e/status 15 !!Kernel Information !!------------------ Kernel release: 5.12.13-arch1-1 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k5.12.13-arch1-1 Library version: 1.2.5.1 Utilities version: 1.2.5.1 !!Loaded ALSA modules !!------------------- snd_hda_intel (card 0) snd_hda_intel (card 1) !!Sound Servers on this system !!---------------------------- Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xf7c34000 irq 29 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf7c30000 irq 30 !!PCI Soundcards installed in the system !!-------------------------------------- 00:03.0 Audio device [0403]: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller [8086:0c0c] (rev 06) Subsystem: Intel Corporation Device [8086:2052] 00:1b.0 Audio device [0403]: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller [8086:8c20] (rev 04) DeviceName: CODEC2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N dmic_detect : Y enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) pm_blacklist : Y position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 1 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : -1 snoop : -1 !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N dmic_detect : Y enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) pm_blacklist : Y position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 1 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : -1 snoop : -1 !!Sysfs card info !!--------------- !!Card: /sys/class/sound/card0 Driver: /sys/bus/pci/drivers/snd_hda_intel Tree: /sys/class/sound/card0 |-- controlC0 | |-- dev | |-- device -> ../../card0 | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- device -> ../../../0000:00:03.0 |-- hwC0D0 | |-- afg | |-- chip_name | |-- clear | |-- dev | |-- device -> ../../card0 | |-- driver_pin_configs | |-- hints | |-- init_pin_configs | |-- init_verbs | |-- mfg | |-- modelname | |-- power | |-- power_off_acct | |-- power_on_acct | |-- reconfig | |-- revision_id | |-- subsystem -> ../../../../../../class/sound | |-- subsystem_id | |-- uevent | |-- user_pin_configs | |-- vendor_id | `-- vendor_name |-- id |-- input5 | |-- capabilities | |-- device -> ../../card0 | |-- event5 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input6 | |-- capabilities | |-- device -> ../../card0 | |-- event6 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input7 | |-- capabilities | |-- device -> ../../card0 | |-- event7 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- number |-- pcmC0D3p | |-- dev | |-- device -> ../../card0 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC0D7p | |-- dev | |-- device -> ../../card0 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC0D8p | |-- dev | |-- device -> ../../card0 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- power | |-- autosuspend_delay_ms | |-- control | |-- runtime_active_time | |-- runtime_status | `-- runtime_suspended_time |-- subsystem -> ../../../../../class/sound `-- uevent !!Card: /sys/class/sound/card1 Driver: /sys/bus/pci/drivers/snd_hda_intel Tree: /sys/class/sound/card1 |-- controlC1 | |-- dev | |-- device -> ../../card1 | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- device -> ../../../0000:00:1b.0 |-- hwC1D2 | |-- afg | |-- chip_name | |-- clear | |-- dev | |-- device -> ../../card1 | |-- driver_pin_configs | |-- hints | |-- init_pin_configs | |-- init_verbs | |-- mfg | |-- modelname | |-- power | |-- power_off_acct | |-- power_on_acct | |-- reconfig | |-- revision_id | |-- subsystem -> ../../../../../../class/sound | |-- subsystem_id | |-- uevent | |-- user_pin_configs | |-- vendor_id | `-- vendor_name |-- id |-- input10 | |-- capabilities | |-- device -> ../../card1 | |-- event10 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input8 | |-- capabilities | |-- device -> ../../card1 | |-- event8 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input9 | |-- capabilities | |-- device -> ../../card1 | |-- event9 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- number |-- pcmC1D0c | |-- dev | |-- device -> ../../card1 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC1D0p | |-- dev | |-- device -> ../../card1 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC1D1p | |-- dev | |-- device -> ../../card1 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC1D2c | |-- dev | |-- device -> ../../card1 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- power | |-- autosuspend_delay_ms | |-- control | |-- runtime_active_time | |-- runtime_status | `-- runtime_suspended_time |-- subsystem -> ../../../../../class/sound `-- uevent !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Intel Haswell HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862807 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560010: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560020: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560030: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono Codec: Realtek ALC662 rev1 Address: 2 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0662 Subsystem Id: 0x80862052 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Headphone+LO Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC662 rev1 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Center Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="ALC662 rev1 Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC662 rev1 Analog", type="Audio", device=0 Amp-In caps: ofs=0x09, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x11 0x11] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC662 rev1 Alt Analog", type="Audio", device=2 Amp-In caps: ofs=0x09, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 9 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x04 0x0b Node 0x0f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=04, enabled=1 Connection: 1 0x0c Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00010034: IN OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x4017c040: [N/A] Speaker at Ext N/A Conn = Analog, Color = UNKNOWN DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00000034: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Center Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Switch", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Control: name="Rear Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00001734: IN OUT Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19030: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 1 0x0e Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Front Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x02a19140: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0e Node 0x1a [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00000034: IN OUT Detect Pin Default 0x0181303f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xf Pin-ctls: 0x20: IN Unsolicited: tag=03, enabled=1 Connection: 1 0x0d Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x02214150: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x5, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0e Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x40b4c601: [N/A] Telephony at Ext N/A Conn = RCA, Color = UNKNOWN DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x90460120: [Fixed] SPDIF Out at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=12 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x26 [Vendor Defined Widget] wcaps 0xf00000: Mono --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 6 Nov 26 05:37 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 12 Nov 26 05:37 /dev/snd/controlC1 crw-rw----+ 1 root audio 116, 5 Nov 26 05:37 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 11 Nov 26 05:37 /dev/snd/hwC1D2 crw-rw----+ 1 root audio 116, 2 Nov 26 05:37 /dev/snd/pcmC0D3p crw-rw----+ 1 root audio 116, 3 Nov 26 05:37 /dev/snd/pcmC0D7p crw-rw----+ 1 root audio 116, 4 Nov 26 05:37 /dev/snd/pcmC0D8p crw-rw----+ 1 root audio 116, 8 Nov 26 05:37 /dev/snd/pcmC1D0c crw-rw----+ 1 root audio 116, 7 Nov 26 05:37 /dev/snd/pcmC1D0p crw-rw----+ 1 root audio 116, 9 Nov 26 05:37 /dev/snd/pcmC1D1p crw-rw----+ 1 root audio 116, 10 Nov 26 05:37 /dev/snd/pcmC1D2c crw-rw----+ 1 root audio 116, 1 Nov 26 05:37 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Nov 26 05:37 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 80 Nov 26 05:37 . drwxr-xr-x 3 root root 320 Nov 26 05:37 .. lrwxrwxrwx 1 root root 12 Nov 26 05:37 pci-0000:00:03.0 -> ../controlC0 lrwxrwxrwx 1 root root 12 Nov 26 05:37 pci-0000:00:1b.0 -> ../controlC1 !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 1: ALC662 rev1 Digital [ALC662 rev1 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 1: PCH [HDA Intel PCH], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 2: ALC662 rev1 Alt Analog [ALC662 rev1 Alt Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card HDMI Card sysdefault:0 'HDMI'/'HDA Intel HDMI at 0xf7c34000 irq 29' Mixer name : 'Intel Haswell HDMI' Components : 'HDA:80862807,80860101,00100000' Controls : 21 Simple ctrls : 3 Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',1 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',2 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] !!-------Mixer controls for card PCH Card sysdefault:1 'PCH'/'HDA Intel PCH at 0xf7c30000 irq 30' Mixer name : 'Realtek ALC662 rev1' Components : 'HDA:10ec0662,80862052,00100101' Controls : 44 Simple ctrls : 21 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 44 [69%] [-20.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'Headphone+LO',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] Front Right: Playback 64 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] [on] Front Right: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 17 [55%] [12.00dB] [on] Front Right: Capture 17 [55%] [12.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-13.50dB] [off] Front Right: Capture 0 [0%] [-13.50dB] [off] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '4ch' '6ch' Item0: '2ch' Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Loopback Mixing',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' Simple mixer control 'Rear Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Rear Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] !!Alsactl output !!-------------- --startcollapse-- state.HDMI { control.1 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.2 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.3 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.4 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.5 { iface MIXER name 'IEC958 Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.6 { iface PCM device 3 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.7 { iface CARD name 'HDMI/DP,pcm=7 Jack' value false comment { access read type BOOLEAN count 1 } } control.8 { iface MIXER name 'IEC958 Playback Con Mask' index 1 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.9 { iface MIXER name 'IEC958 Playback Pro Mask' index 1 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.10 { iface MIXER name 'IEC958 Playback Default' index 1 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.11 { iface MIXER name 'IEC958 Playback Switch' index 1 value true comment { access 'read write' type BOOLEAN count 1 } } control.12 { iface PCM device 7 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.13 { iface CARD name 'HDMI/DP,pcm=8 Jack' value false comment { access read type BOOLEAN count 1 } } control.14 { iface MIXER name 'IEC958 Playback Con Mask' index 2 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.15 { iface MIXER name 'IEC958 Playback Pro Mask' index 2 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.16 { iface MIXER name 'IEC958 Playback Default' index 2 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.17 { iface MIXER name 'IEC958 Playback Switch' index 2 value true comment { access 'read write' type BOOLEAN count 1 } } control.18 { iface PCM device 8 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.19 { iface PCM device 3 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.20 { iface PCM device 7 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.21 { iface PCM device 8 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } } state.PCH { control.1 { iface MIXER name 'Channel Mode' value '2ch' comment { access 'read write' type ENUMERATED count 1 item.0 '2ch' item.1 '4ch' item.2 '6ch' } } control.2 { iface MIXER name 'Headphone+LO Playback Volume' value.0 64 value.1 64 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.3 { iface MIXER name 'Front Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.4 { iface MIXER name 'Surround Playback Volume' value.0 64 value.1 64 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.5 { iface MIXER name 'Surround Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.6 { iface MIXER name 'Center Playback Volume' value 64 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 } } control.7 { iface MIXER name 'LFE Playback Volume' value 64 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 } } control.8 { iface MIXER name 'Center Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.9 { iface MIXER name 'LFE Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.10 { iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.11 { iface MIXER name 'Loopback Mixing' value Disabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.12 { iface MIXER name 'Front Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.13 { iface MIXER name 'Front Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.14 { iface MIXER name 'Rear Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.15 { iface MIXER name 'Rear Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.16 { iface MIXER name 'Line Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.17 { iface MIXER name 'Line Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.18 { iface MIXER name 'Input Source' value 'Front Mic' comment { access 'read write' type ENUMERATED count 1 item.0 'Front Mic' item.1 'Rear Mic' item.2 Line } } control.19 { iface MIXER name 'Input Source' index 1 value 'Front Mic' comment { access 'read write' type ENUMERATED count 1 item.0 'Front Mic' item.1 'Rear Mic' item.2 Line } } control.20 { iface MIXER name 'Capture Volume' value.0 17 value.1 17 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -1350 dbmax 3300 dbvalue.0 1200 dbvalue.1 1200 } } control.21 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.22 { iface MIXER name 'Capture Volume' index 1 value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -1350 dbmax 3300 dbvalue.0 -1350 dbvalue.1 -1350 } } control.23 { iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.24 { iface MIXER name 'Front Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.25 { iface MIXER name 'Rear Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.26 { iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.27 { iface MIXER name 'Beep Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.28 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.29 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.30 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.31 { iface MIXER name 'IEC958 Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.32 { iface MIXER name 'IEC958 Default PCM Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.33 { iface MIXER name 'Master Playback Volume' value 44 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -2000 } } control.34 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.35 { iface CARD name 'Front Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.36 { iface CARD name 'Rear Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.37 { iface CARD name 'Line Jack' value false comment { access read type BOOLEAN count 1 } } control.38 { iface CARD name 'Line Out Jack' value true comment { access read type BOOLEAN count 1 } } control.39 { iface CARD name 'Front Headphone Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.40 { iface CARD name 'SPDIF Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.41 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.42 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.43 { iface PCM device 1 name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.44 { iface PCM device 2 name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } } --endcollapse-- !!All Loaded Modules !!------------------ aesni_intel agpgart at24 bpf_preload cdc_ether cdrom cec coretemp crc32_pclmul crc32c_intel crct10dif_pclmul cryptd crypto_simd dm_mod drm drm_kms_helper e1000e fat fb_sys_fops fuse ghash_clmulni_intel i2c_algo_bit i2c_i801 i2c_smbus i915 iTCO_vendor_support iTCO_wdt intel_cstate intel_gtt intel_pmc_bxt intel_powerclamp intel_rapl_common intel_rapl_msr intel_uncore ip_tables irqbypass isofs kvm kvm_intel ledtrig_audio loop lpc_ich mac_hid mei mei_hdcp mei_me mei_wdt mii mousedev overlay rapl rfkill snd snd_hda_codec snd_hda_codec_generic snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_core snd_hda_intel snd_hwdep snd_intel_dspcfg snd_intel_sdw_acpi snd_pcm snd_timer soundcore squashfs sr_mod syscopyarea sysfillrect sysimgblt uas usb_storage usbhid usbnet vfat video x86_pkg_temp_thermal x_tables xhci_pci xhci_pci_renesas !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x05 0x58560010 0x06 0x58560020 0x07 0x18560030 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: /sys/class/sound/hwC1D2/init_pin_configs: 0x14 0x01014010 0x15 0x4017c040 0x16 0x411111f0 0x18 0x01a19030 0x19 0x02a19140 0x1a 0x0181303f 0x1b 0x02214150 0x1c 0x411111f0 0x1d 0x40b4c601 0x1e 0x90460120 /sys/class/sound/hwC1D2/driver_pin_configs: /sys/class/sound/hwC1D2/user_pin_configs: /sys/class/sound/hwC1D2/init_verbs: /sys/class/sound/hwC1D2/hints: !!ALSA/HDA dmesg !!-------------- [ 0.089835] ACPI: Added _OSI(Linux-Dell-Video) [ 0.089835] ACPI: Added _OSI(Linux-Lenovo-NV-HDMI-Audio) [ 0.089835] ACPI: Added _OSI(Linux-HPI-Hybrid-Graphics) -- [ 13.947834] iTCO_vendor_support: vendor-support=0 [ 14.140544] snd_hda_intel 0000:00:03.0: enabling device (0000 -> 0002) [ 14.140729] snd_hda_intel 0000:00:03.0: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) [ 14.141149] snd_hda_intel 0000:00:1b.0: enabling device (0000 -> 0002) [ 14.186225] iTCO_wdt iTCO_wdt.1.auto: Found a Lynx Point TCO device (Version=2, TCOBASE=0x1860) [ 14.186295] iTCO_wdt iTCO_wdt.1.auto: initialized. heartbeat=30 sec (nowayout=0) [ 14.354484] input: HDA Intel HDMI HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:03.0/sound/card0/input5 [ 14.354528] input: HDA Intel HDMI HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:03.0/sound/card0/input6 [ 14.354556] input: HDA Intel HDMI HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:03.0/sound/card0/input7 [ 14.584436] snd_hda_codec_realtek hdaudioC1D2: autoconfig for ALC662 rev1: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:line [ 14.584441] snd_hda_codec_realtek hdaudioC1D2: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 14.584443] snd_hda_codec_realtek hdaudioC1D2: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) [ 14.584444] snd_hda_codec_realtek hdaudioC1D2: mono: mono_out=0x0 [ 14.584445] snd_hda_codec_realtek hdaudioC1D2: dig-out=0x1e/0x0 [ 14.584446] snd_hda_codec_realtek hdaudioC1D2: inputs: [ 14.584447] snd_hda_codec_realtek hdaudioC1D2: Front Mic=0x19 [ 14.584448] snd_hda_codec_realtek hdaudioC1D2: Rear Mic=0x18 [ 14.584449] snd_hda_codec_realtek hdaudioC1D2: Line=0x1a [ 14.596570] input: HDA Intel PCH Rear Mic as /devices/pci0000:00/0000:00:1b.0/sound/card1/input8 [ 14.596620] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1b.0/sound/card1/input9 [ 14.596661] input: HDA Intel PCH Line Out as /devices/pci0000:00/0000:00:1b.0/sound/card1/input10 [ 14.936847] intel_rapl_common: Found RAPL domain package alsa-info output with 5.15.5: ============================= upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.5.0 !!################################ !!Script ran on: Fri Nov 26 23:39:13 UTC 2021 !!Linux Distribution !!------------------ Arch Linux \r (\l) NAME="Arch Linux" PRETTY_NAME="Arch Linux" ID=arch HOME_URL="https://archlinux.org/" DOCUMENTATION_URL="https://wiki.archlinux.org/" SUPPORT_URL="https://bbs.archlinux.org/" BUG_REPORT_URL="https://bugs.archlinux.org/" LOGO=archlinux-logo !!DMI Information !!--------------- Manufacturer: Product Name: Product Version: Firmware Version: FLB8510H.86A.0131.2017.0516.1705 System SKU: To be filled by O.E.M. Board Vendor: Intel Corporation Board Name: DB85FL !!ACPI Device Status Information !!--------------- /sys/bus/acpi/devices/INT3F0D:00/status 15 /sys/bus/acpi/devices/PNP0103:00/status 15 /sys/bus/acpi/devices/PNP0C01:00/status 15 /sys/bus/acpi/devices/PNP0C04:00/status 31 /sys/bus/acpi/devices/PNP0C0C:00/status 11 /sys/bus/acpi/devices/PNP0C0F:00/status 9 /sys/bus/acpi/devices/PNP0C0F:01/status 9 /sys/bus/acpi/devices/PNP0C0F:02/status 9 /sys/bus/acpi/devices/PNP0C0F:03/status 9 /sys/bus/acpi/devices/PNP0C0F:04/status 9 /sys/bus/acpi/devices/PNP0C0F:05/status 9 /sys/bus/acpi/devices/PNP0C0F:06/status 9 /sys/bus/acpi/devices/PNP0C0F:07/status 9 /sys/bus/acpi/devices/device:29/status 15 /sys/bus/acpi/devices/device:2a/status 15 /sys/bus/acpi/devices/device:2b/status 15 /sys/bus/acpi/devices/device:2c/status 15 /sys/bus/acpi/devices/device:2d/status 15 /sys/bus/acpi/devices/device:2e/status 15 /sys/bus/acpi/devices/device:2f/status 15 /sys/bus/acpi/devices/device:30/status 15 /sys/bus/acpi/devices/device:31/status 15 /sys/bus/acpi/devices/device:32/status 15 /sys/bus/acpi/devices/device:33/status 15 /sys/bus/acpi/devices/device:34/status 15 /sys/bus/acpi/devices/device:35/status 15 /sys/bus/acpi/devices/device:36/status 15 /sys/bus/acpi/devices/device:37/status 15 /sys/bus/acpi/devices/device:38/status 15 /sys/bus/acpi/devices/device:39/status 15 /sys/bus/acpi/devices/device:3a/status 15 /sys/bus/acpi/devices/device:3b/status 15 /sys/bus/acpi/devices/device:3c/status 15 /sys/bus/acpi/devices/device:3d/status 15 /sys/bus/acpi/devices/device:4e/status 15 !!Kernel Information !!------------------ Kernel release: 5.15.5-arch1-1 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k5.15.5-arch1-1 Library version: 1.2.5.1 Utilities version: 1.2.5.1 !!Loaded ALSA modules !!------------------- snd_hda_intel (card 0) snd_hda_intel (card 1) !!Sound Servers on this system !!---------------------------- Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf7c30000 irq 30 1 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xf7c34000 irq 29 !!PCI Soundcards installed in the system !!-------------------------------------- 00:03.0 Audio device [0403]: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller [8086:0c0c] (rev 06) Subsystem: Intel Corporation Device [8086:2052] 00:1b.0 Audio device [0403]: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller [8086:8c20] (rev 04) DeviceName: CODEC2 !!Modprobe options (Sound related) !!-------------------------------- snd_hda_intel: id=PCH index=0 snd_hda_intel: id=HDMI index=1 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N dmic_detect : Y enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : HDMI,(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) pm_blacklist : Y position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 1 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : -1 snoop : -1 !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N dmic_detect : Y enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : HDMI,(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) pm_blacklist : Y position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 1 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : -1 snoop : -1 !!Sysfs card info !!--------------- !!Card: /sys/class/sound/card0 Driver: /sys/bus/pci/drivers/snd_hda_intel Tree: /sys/class/sound/card0 |-- controlC0 | |-- dev | |-- device -> ../../card0 | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- device -> ../../../0000:00:1b.0 |-- hwC0D2 | |-- afg | |-- chip_name | |-- clear | |-- dev | |-- device -> ../../card0 | |-- driver_pin_configs | |-- hints | |-- init_pin_configs | |-- init_verbs | |-- mfg | |-- modelname | |-- power | |-- power_off_acct | |-- power_on_acct | |-- reconfig | |-- revision_id | |-- subsystem -> ../../../../../../class/sound | |-- subsystem_id | |-- uevent | |-- user_pin_configs | |-- vendor_id | `-- vendor_name |-- id |-- input10 | |-- capabilities | |-- device -> ../../card0 | |-- event10 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input11 | |-- capabilities | |-- device -> ../../card0 | |-- event11 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input9 | |-- capabilities | |-- device -> ../../card0 | |-- event9 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- number |-- pcmC0D0c | |-- dev | |-- device -> ../../card0 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC0D0p | |-- dev | |-- device -> ../../card0 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC0D1p | |-- dev | |-- device -> ../../card0 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC0D2c | |-- dev | |-- device -> ../../card0 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- power | |-- autosuspend_delay_ms | |-- control | |-- runtime_active_time | |-- runtime_status | `-- runtime_suspended_time |-- subsystem -> ../../../../../class/sound `-- uevent !!Card: /sys/class/sound/card1 Driver: /sys/bus/pci/drivers/snd_hda_intel Tree: /sys/class/sound/card1 |-- controlC1 | |-- dev | |-- device -> ../../card1 | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- device -> ../../../0000:00:03.0 |-- hwC1D0 | |-- afg | |-- chip_name | |-- clear | |-- dev | |-- device -> ../../card1 | |-- driver_pin_configs | |-- hints | |-- init_pin_configs | |-- init_verbs | |-- mfg | |-- modelname | |-- power | |-- power_off_acct | |-- power_on_acct | |-- reconfig | |-- revision_id | |-- subsystem -> ../../../../../../class/sound | |-- subsystem_id | |-- uevent | |-- user_pin_configs | |-- vendor_id | `-- vendor_name |-- id |-- input6 | |-- capabilities | |-- device -> ../../card1 | |-- event6 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input7 | |-- capabilities | |-- device -> ../../card1 | |-- event7 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- input8 | |-- capabilities | |-- device -> ../../card1 | |-- event8 | |-- id | |-- inhibited | |-- modalias | |-- name | |-- phys | |-- power | |-- properties | |-- subsystem -> ../../../../../../class/input | |-- uevent | `-- uniq |-- number |-- pcmC1D3p | |-- dev | |-- device -> ../../card1 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC1D7p | |-- dev | |-- device -> ../../card1 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- pcmC1D8p | |-- dev | |-- device -> ../../card1 | |-- pcm_class | |-- power | |-- subsystem -> ../../../../../../class/sound | `-- uevent |-- power | |-- autosuspend_delay_ms | |-- control | |-- runtime_active_time | |-- runtime_status | `-- runtime_suspended_time |-- subsystem -> ../../../../../class/sound `-- uevent !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC662 rev1 Address: 2 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0662 Subsystem Id: 0x80862052 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Headphone+LO Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC662 rev1 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=5, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x1c 0x1c] Converter: stream=5, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Center Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x1c 0x1c] Converter: stream=5, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="ALC662 rev1 Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC662 rev1 Analog", type="Audio", device=0 Amp-In caps: ofs=0x09, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC662 rev1 Alt Analog", type="Audio", device=2 Amp-In caps: ofs=0x09, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 9 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x04 0x0b Node 0x0f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=04, enabled=1 Connection: 1 0x0c Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00010034: IN OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x4017c040: [N/A] Speaker at Ext N/A Conn = Analog, Color = UNKNOWN DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00000034: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Center Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Switch", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Control: name="Rear Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00001734: IN OUT Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19030: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 1 0x0e Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Front Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x02a19140: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0e Node 0x1a [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00000034: IN OUT Detect Pin Default 0x0181303f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0xf Pin-ctls: 0x20: IN Unsolicited: tag=03, enabled=1 Connection: 1 0x0d Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x02214150: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x5, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0e Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x40b4c601: [N/A] Telephony at Ext N/A Conn = RCA, Color = UNKNOWN DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x90460120: [Fixed] SPDIF Out at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=12 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x26 [Vendor Defined Widget] wcaps 0xf00000: Mono Codec: Intel Haswell HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862807 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560010: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560020: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560030: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 12 Nov 26 23:37 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 6 Nov 26 23:37 /dev/snd/controlC1 crw-rw----+ 1 root audio 116, 11 Nov 26 23:37 /dev/snd/hwC0D2 crw-rw----+ 1 root audio 116, 5 Nov 26 23:37 /dev/snd/hwC1D0 crw-rw----+ 1 root audio 116, 8 Nov 26 23:37 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 7 Nov 26 23:38 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 9 Nov 26 23:37 /dev/snd/pcmC0D1p crw-rw----+ 1 root audio 116, 10 Nov 26 23:37 /dev/snd/pcmC0D2c crw-rw----+ 1 root audio 116, 2 Nov 26 23:37 /dev/snd/pcmC1D3p crw-rw----+ 1 root audio 116, 3 Nov 26 23:37 /dev/snd/pcmC1D7p crw-rw----+ 1 root audio 116, 4 Nov 26 23:37 /dev/snd/pcmC1D8p crw-rw----+ 1 root audio 116, 1 Nov 26 23:37 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Nov 26 23:37 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 80 Nov 26 23:37 . drwxr-xr-x 3 root root 320 Nov 26 23:37 .. lrwxrwxrwx 1 root root 12 Nov 26 23:37 pci-0000:00:03.0 -> ../controlC1 lrwxrwxrwx 1 root root 12 Nov 26 23:37 pci-0000:00:1b.0 -> ../controlC0 !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC662 rev1 Digital [ALC662 rev1 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 2: ALC662 rev1 Alt Analog [ALC662 rev1 Alt Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card PCH Card sysdefault:0 'PCH'/'HDA Intel PCH at 0xf7c30000 irq 30' Mixer name : 'Realtek ALC662 rev1' Components : 'HDA:10ec0662,80862052,00100101' Controls : 45 Simple ctrls : 22 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 46 [72%] [-18.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [off] Front Right: Playback [off] Simple mixer control 'Headphone+LO',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 0 [0%] [-64.00dB] Front Right: Playback 0 [0%] [-64.00dB] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 173 [68%] [-16.40dB] Front Right: Playback 173 [68%] [-16.40dB] Simple mixer control 'Front',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [off] Front Right: Playback [off] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 46 [72%] [-18.00dB] [off] Front Right: Playback 46 [72%] [-18.00dB] [off] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 46 [72%] [-18.00dB] [off] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 46 [72%] [-18.00dB] [off] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 20 [65%] [-4.50dB] [on] Front Right: Playback 20 [65%] [-4.50dB] [on] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-13.50dB] [on] Front Right: Capture 0 [0%] [-13.50dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-13.50dB] [off] Front Right: Capture 0 [0%] [-13.50dB] [off] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '4ch' '6ch' Item0: '2ch' Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Loopback Mixing',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' Simple mixer control 'Rear Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Rear Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] !!-------Mixer controls for card HDMI Card sysdefault:1 'HDMI'/'HDA Intel HDMI at 0xf7c34000 irq 29' Mixer name : 'Intel Haswell HDMI' Components : 'HDA:80862807,80860101,00100000' Controls : 21 Simple ctrls : 3 Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958',1 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958',2 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] !!Alsactl output !!-------------- --startcollapse-- state.PCH { control.1 { iface MIXER name 'Channel Mode' value '2ch' comment { access 'read write' type ENUMERATED count 1 item.0 '2ch' item.1 '4ch' item.2 '6ch' } } control.2 { iface MIXER name 'Headphone+LO Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -6400 dbvalue.1 -6400 } } control.3 { iface MIXER name 'Front Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.4 { iface MIXER name 'Surround Playback Volume' value.0 46 value.1 46 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -1800 dbvalue.1 -1800 } } control.5 { iface MIXER name 'Surround Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.6 { iface MIXER name 'Center Playback Volume' value 46 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -1800 } } control.7 { iface MIXER name 'LFE Playback Volume' value 46 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -1800 } } control.8 { iface MIXER name 'Center Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.9 { iface MIXER name 'LFE Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.10 { iface MIXER name 'Headphone Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.11 { iface MIXER name 'Loopback Mixing' value Disabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.12 { iface MIXER name 'Front Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.13 { iface MIXER name 'Front Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.14 { iface MIXER name 'Rear Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.15 { iface MIXER name 'Rear Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.16 { iface MIXER name 'Line Playback Volume' value.0 20 value.1 20 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -450 dbvalue.1 -450 } } control.17 { iface MIXER name 'Line Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.18 { iface MIXER name 'Input Source' value 'Front Mic' comment { access 'read write' type ENUMERATED count 1 item.0 'Front Mic' item.1 'Rear Mic' item.2 Line } } control.19 { iface MIXER name 'Input Source' index 1 value 'Front Mic' comment { access 'read write' type ENUMERATED count 1 item.0 'Front Mic' item.1 'Rear Mic' item.2 Line } } control.20 { iface MIXER name 'Capture Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -1350 dbmax 3300 dbvalue.0 -1350 dbvalue.1 -1350 } } control.21 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.22 { iface MIXER name 'Capture Volume' index 1 value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -1350 dbmax 3300 dbvalue.0 -1350 dbvalue.1 -1350 } } control.23 { iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.24 { iface MIXER name 'Front Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.25 { iface MIXER name 'Rear Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.26 { iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.27 { iface MIXER name 'Beep Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.28 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.29 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.30 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.31 { iface MIXER name 'IEC958 Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.32 { iface MIXER name 'IEC958 Default PCM Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.33 { iface MIXER name 'Master Playback Volume' value 46 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -1800 } } control.34 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.35 { iface CARD name 'Front Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.36 { iface CARD name 'Rear Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.37 { iface CARD name 'Line Jack' value false comment { access read type BOOLEAN count 1 } } control.38 { iface CARD name 'Line Out Jack' value true comment { access read type BOOLEAN count 1 } } control.39 { iface CARD name 'Front Headphone Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.40 { iface CARD name 'SPDIF Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.41 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.42 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.43 { iface PCM device 1 name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.44 { iface PCM device 2 name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.45 { iface MIXER name 'PCM Playback Volume' value.0 173 value.1 173 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 -1640 dbvalue.1 -1640 } } } state.HDMI { control.1 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.2 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.3 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.4 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.5 { iface MIXER name 'IEC958 Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.6 { iface PCM device 3 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.7 { iface CARD name 'HDMI/DP,pcm=7 Jack' value false comment { access read type BOOLEAN count 1 } } control.8 { iface MIXER name 'IEC958 Playback Con Mask' index 1 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.9 { iface MIXER name 'IEC958 Playback Pro Mask' index 1 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.10 { iface MIXER name 'IEC958 Playback Default' index 1 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.11 { iface MIXER name 'IEC958 Playback Switch' index 1 value false comment { access 'read write' type BOOLEAN count 1 } } control.12 { iface PCM device 7 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.13 { iface CARD name 'HDMI/DP,pcm=8 Jack' value false comment { access read type BOOLEAN count 1 } } control.14 { iface MIXER name 'IEC958 Playback Con Mask' index 2 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.15 { iface MIXER name 'IEC958 Playback Pro Mask' index 2 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.16 { iface MIXER name 'IEC958 Playback Default' index 2 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.17 { iface MIXER name 'IEC958 Playback Switch' index 2 value false comment { access 'read write' type BOOLEAN count 1 } } control.18 { iface PCM device 8 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.19 { iface PCM device 3 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.20 { iface PCM device 7 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.21 { iface PCM device 8 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } } --endcollapse-- !!All Loaded Modules !!------------------ aesni_intel at24 cdc_ether cdrom coretemp crc16 crc32_pclmul crc32c_generic crc32c_intel crct10dif_pclmul cryptd crypto_simd dm_mod e1000e ext4 fat fuse ghash_clmulni_intel i2c_i801 i2c_smbus i915 iTCO_vendor_support iTCO_wdt intel_cstate intel_gtt intel_pmc_bxt intel_powerclamp intel_rapl_common intel_rapl_msr intel_spi intel_spi_platform intel_uncore ip_tables irqbypass jbd2 kvm kvm_intel ledtrig_audio libcrc32c lpc_ich mac_hid mbcache mei mei_hdcp mei_me mei_wdt mii mousedev mtd nct6683 nf_conntrack nf_defrag_ipv4 nf_defrag_ipv6 nf_reject_ipv4 nf_reject_ipv6 nf_tables nfnetlink nft_counter nft_ct nft_limit nft_reject nft_reject_inet nls_iso8859_1 pcspkr rapl sg snd snd_hda_codec snd_hda_codec_generic snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_core snd_hda_intel snd_hwdep snd_intel_dspcfg snd_intel_sdw_acpi snd_pcm snd_timer soundcore spi_nor sr_mod ttm uas usb_storage usbhid usbnet vfat video x86_pkg_temp_thermal x_tables xhci_pci xhci_pci_renesas !!Sysfs Files !!----------- /sys/class/sound/hwC0D2/init_pin_configs: 0x14 0x01014010 0x15 0x4017c040 0x16 0x411111f0 0x18 0x01a19030 0x19 0x02a19140 0x1a 0x0181303f 0x1b 0x02214150 0x1c 0x411111f0 0x1d 0x40b4c601 0x1e 0x90460120 /sys/class/sound/hwC0D2/driver_pin_configs: /sys/class/sound/hwC0D2/user_pin_configs: /sys/class/sound/hwC0D2/init_verbs: /sys/class/sound/hwC0D2/hints: /sys/class/sound/hwC1D0/init_pin_configs: 0x05 0x58560010 0x06 0x58560020 0x07 0x18560030 /sys/class/sound/hwC1D0/driver_pin_configs: /sys/class/sound/hwC1D0/user_pin_configs: /sys/class/sound/hwC1D0/init_verbs: /sys/class/sound/hwC1D0/hints: !!ALSA/HDA dmesg !!-------------- [ 0.099082] ACPI: Added _OSI(Linux-Dell-Video) [ 0.099082] ACPI: Added _OSI(Linux-Lenovo-NV-HDMI-Audio) [ 0.099082] ACPI: Added _OSI(Linux-HPI-Hybrid-Graphics) -- [ 12.324700] AES CTR mode by8 optimization enabled [ 12.404051] snd_hda_intel 0000:00:03.0: enabling device (0000 -> 0002) [ 12.404225] snd_hda_intel 0000:00:03.0: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) [ 12.404685] snd_hda_intel 0000:00:1b.0: enabling device (0000 -> 0002) [ 12.535527] input: HDA Intel HDMI HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:03.0/sound/card1/input6 [ 12.535599] input: HDA Intel HDMI HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:03.0/sound/card1/input7 [ 12.535660] input: HDA Intel HDMI HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:03.0/sound/card1/input8 [ 12.877211] snd_hda_codec_realtek hdaudioC0D2: autoconfig for ALC662 rev1: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:line [ 12.877227] snd_hda_codec_realtek hdaudioC0D2: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 12.877233] snd_hda_codec_realtek hdaudioC0D2: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) [ 12.877239] snd_hda_codec_realtek hdaudioC0D2: mono: mono_out=0x0 [ 12.877243] snd_hda_codec_realtek hdaudioC0D2: dig-out=0x1e/0x0 [ 12.877247] snd_hda_codec_realtek hdaudioC0D2: inputs: [ 12.877251] snd_hda_codec_realtek hdaudioC0D2: Front Mic=0x19 [ 12.877256] snd_hda_codec_realtek hdaudioC0D2: Rear Mic=0x18 [ 12.877260] snd_hda_codec_realtek hdaudioC0D2: Line=0x1a [ 12.893002] input: HDA Intel PCH Rear Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 [ 12.893066] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 [ 12.893118] input: HDA Intel PCH Line Out as /devices/pci0000:00/0000:00:1b.0/sound/card0/input11 [ 13.779298] intel_rapl_common: Found RAPL domain package -- u34

1 0

[PATCH] ASoC: codecs/jz4770: Add missing gain control after DAC/ADC mixer
by Paul Cercueil 27 Nov '21

27 Nov '21

The capture and playback paths both have a configurable gain after their respective mixer, which can be set from -31 dB to 0 dB in 32 steps. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> --- sound/soc/codecs/jz4770.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/jz4770.c b/sound/soc/codecs/jz4770.c index 6b60120f59a6..1d0c467ab57b 100644 --- a/sound/soc/codecs/jz4770.c +++ b/sound/soc/codecs/jz4770.c @@ -307,6 +307,7 @@ static const DECLARE_TLV_DB_MINMAX_MUTE(dac_tlv, -3100, 0); static const DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0); static const DECLARE_TLV_DB_MINMAX(out_tlv, -2500, 600); static const DECLARE_TLV_DB_SCALE(linein_tlv, -2500, 100, 0); +static const DECLARE_TLV_DB_MINMAX(mixer_tlv, -3100, 0); /* Unconditional controls. */ static const struct snd_kcontrol_new jz4770_codec_snd_controls[] = { @@ -319,6 +320,14 @@ static const struct snd_kcontrol_new jz4770_codec_snd_controls[] = { SOC_DOUBLE_R_TLV("Line In Bypass Playback Volume", JZ4770_CODEC_REG_GCR_LIBYL, JZ4770_CODEC_REG_GCR_LIBYR, REG_GCR_GAIN_OFFSET, REG_GCR_GAIN_MAX, 1, linein_tlv), + + SOC_SINGLE_TLV("Mixer Capture Volume", + JZ4770_CODEC_REG_GCR_MIXADC, + REG_GCR_GAIN_OFFSET, REG_GCR_GAIN_MAX, 1, mixer_tlv), + + SOC_SINGLE_TLV("Mixer Playback Volume", + JZ4770_CODEC_REG_GCR_MIXDAC, + REG_GCR_GAIN_OFFSET, REG_GCR_GAIN_MAX, 1, mixer_tlv), }; static const struct snd_kcontrol_new jz4770_codec_pcm_playback_controls[] = { -- 2.33.0

2 1

[PATCH] ASoC: imx-hdmi: add put_device() after of_find_device_by_node()
by cgel.zte＠gmail.com 27 Nov '21

27 Nov '21

From: Ye Guojin <ye.guojin(a)zte.com.cn> This was found by coccicheck: ./sound/soc/fsl/imx-hdmi.c,209,1-7,ERROR missing put_device; call of_find_device_by_node on line 119, but without a corresponding object release within this function. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Ye Guojin <ye.guojin(a)zte.com.cn> --- sound/soc/fsl/imx-hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index f10359a28800..929f69b758af 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -145,6 +145,8 @@ static int imx_hdmi_probe(struct platform_device *pdev) data->dai.capture_only = false; data->dai.init = imx_hdmi_init; + put_device(&cpu_pdev->dev); + if (of_node_name_eq(cpu_np, "sai")) { data->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1; data->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; -- 2.25.1

2 1

[PATCH] ASoC: cs35l41: Fix link problem
by Lucas Tanure 27 Nov '21

27 Nov '21

Can't link I2C and SPI to the same binary, better to move CS35L41 to 3 modules approach. And instead of exposing cs35l41_reg, volatile_reg, readable_reg and precious_reg arrays, move cs35l41_regmap_i2c/spi to new module and expose it. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> --- sound/soc/codecs/Kconfig | 5 ++++ sound/soc/codecs/Makefile | 6 +++-- sound/soc/codecs/cs35l41-i2c.c | 15 ----------- sound/soc/codecs/cs35l41-spi.c | 16 ------------ sound/soc/codecs/cs35l41-tables.c | 41 ++++++++++++++++++++++++++++--- sound/soc/codecs/cs35l41.c | 2 ++ sound/soc/codecs/cs35l41.h | 7 ++---- 7 files changed, 50 insertions(+), 42 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3fe62df32238..6ca1edf0d066 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -613,14 +613,19 @@ config SND_SOC_CS35L36 tristate "Cirrus Logic CS35L36 CODEC" depends on I2C +config SND_SOC_CS35L41 + tristate + config SND_SOC_CS35L41_SPI tristate "Cirrus Logic CS35L41 CODEC (SPI)" depends on SPI_MASTER + select SND_SOC_CS35L41 select REGMAP_SPI config SND_SOC_CS35L41_I2C tristate "Cirrus Logic CS35L41 CODEC (I2C)" depends on I2C + select SND_SOC_CS35L41 select REGMAP_I2C config SND_SOC_CS42L42 diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9acfbcbfc46d..485eee75502b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -54,8 +54,9 @@ snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o snd-soc-cs35l35-objs := cs35l35.o snd-soc-cs35l36-objs := cs35l36.o -snd-soc-cs35l41-spi-objs := cs35l41-spi.o cs35l41.o cs35l41-tables.o -snd-soc-cs35l41-i2c-objs := cs35l41-i2c.o cs35l41.o cs35l41-tables.o +snd-soc-cs35l41-objs := cs35l41.o cs35l41-tables.o +snd-soc-cs35l41-spi-objs := cs35l41-spi.o +snd-soc-cs35l41-i2c-objs := cs35l41-i2c.o snd-soc-cs42l42-objs := cs42l42.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o @@ -391,6 +392,7 @@ obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o obj-$(CONFIG_SND_SOC_CS35L35) += snd-soc-cs35l35.o obj-$(CONFIG_SND_SOC_CS35L36) += snd-soc-cs35l36.o +obj-$(CONFIG_SND_SOC_CS35L41) += snd-soc-cs35l41.o obj-$(CONFIG_SND_SOC_CS35L41_SPI) += snd-soc-cs35l41-spi.o obj-$(CONFIG_SND_SOC_CS35L41_I2C) += snd-soc-cs35l41-i2c.o obj-$(CONFIG_SND_SOC_CS42L42) += snd-soc-cs42l42.o diff --git a/sound/soc/codecs/cs35l41-i2c.c b/sound/soc/codecs/cs35l41-i2c.c index d5fa8d2c4a70..c9b604af6b71 100644 --- a/sound/soc/codecs/cs35l41-i2c.c +++ b/sound/soc/codecs/cs35l41-i2c.c @@ -20,21 +20,6 @@ #include <sound/cs35l41.h> #include "cs35l41.h" -static struct regmap_config cs35l41_regmap_i2c = { - .reg_bits = 32, - .val_bits = 32, - .reg_stride = CS35L41_REGSTRIDE, - .reg_format_endian = REGMAP_ENDIAN_BIG, - .val_format_endian = REGMAP_ENDIAN_BIG, - .max_register = CS35L41_LASTREG, - .reg_defaults = cs35l41_reg, - .num_reg_defaults = ARRAY_SIZE(cs35l41_reg), - .volatile_reg = cs35l41_volatile_reg, - .readable_reg = cs35l41_readable_reg, - .precious_reg = cs35l41_precious_reg, - .cache_type = REGCACHE_RBTREE, -}; - static const struct i2c_device_id cs35l41_id_i2c[] = { { "cs35l40", 0 }, { "cs35l41", 0 }, diff --git a/sound/soc/codecs/cs35l41-spi.c b/sound/soc/codecs/cs35l41-spi.c index 3fa99741779a..c202d9df70ee 100644 --- a/sound/soc/codecs/cs35l41-spi.c +++ b/sound/soc/codecs/cs35l41-spi.c @@ -18,22 +18,6 @@ #include <sound/cs35l41.h> #include "cs35l41.h" -static struct regmap_config cs35l41_regmap_spi = { - .reg_bits = 32, - .val_bits = 32, - .pad_bits = 16, - .reg_stride = CS35L41_REGSTRIDE, - .reg_format_endian = REGMAP_ENDIAN_BIG, - .val_format_endian = REGMAP_ENDIAN_BIG, - .max_register = CS35L41_LASTREG, - .reg_defaults = cs35l41_reg, - .num_reg_defaults = ARRAY_SIZE(cs35l41_reg), - .volatile_reg = cs35l41_volatile_reg, - .readable_reg = cs35l41_readable_reg, - .precious_reg = cs35l41_precious_reg, - .cache_type = REGCACHE_RBTREE, -}; - static const struct spi_device_id cs35l41_id_spi[] = { { "cs35l40", 0 }, { "cs35l41", 0 }, diff --git a/sound/soc/codecs/cs35l41-tables.c b/sound/soc/codecs/cs35l41-tables.c index 9d1a7d7dd24d..3eb18b17a7b0 100644 --- a/sound/soc/codecs/cs35l41-tables.c +++ b/sound/soc/codecs/cs35l41-tables.c @@ -8,7 +8,7 @@ #include "cs35l41.h" -const struct reg_default cs35l41_reg[CS35L41_MAX_CACHE_REG] = { +static const struct reg_default cs35l41_reg[] = { { CS35L41_PWR_CTRL1, 0x00000000 }, { CS35L41_PWR_CTRL3, 0x01000010 }, { CS35L41_GPIO_PAD_CONTROL, 0x00000000 }, @@ -47,7 +47,7 @@ const struct reg_default cs35l41_reg[CS35L41_MAX_CACHE_REG] = { { CS35L41_MIXER_NGATE_CH2_CFG, 0x00000303 }, }; -bool cs35l41_readable_reg(struct device *dev, unsigned int reg) +static bool cs35l41_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { case CS35L41_DEVID: @@ -331,7 +331,7 @@ bool cs35l41_readable_reg(struct device *dev, unsigned int reg) } } -bool cs35l41_precious_reg(struct device *dev, unsigned int reg) +static bool cs35l41_precious_reg(struct device *dev, unsigned int reg) { switch (reg) { case CS35L41_OTP_MEM0 ... CS35L41_OTP_MEM31: @@ -344,7 +344,7 @@ bool cs35l41_precious_reg(struct device *dev, unsigned int reg) } } -bool cs35l41_volatile_reg(struct device *dev, unsigned int reg) +static bool cs35l41_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case CS35L41_DEVID: @@ -688,3 +688,36 @@ const struct cs35l41_otp_map_element_t cs35l41_otp_map_map[CS35L41_NUM_OTP_MAPS] .word_offset = 2, }, }; + +struct regmap_config cs35l41_regmap_i2c = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = CS35L41_REGSTRIDE, + .reg_format_endian = REGMAP_ENDIAN_BIG, + .val_format_endian = REGMAP_ENDIAN_BIG, + .max_register = CS35L41_LASTREG, + .reg_defaults = cs35l41_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l41_reg), + .volatile_reg = cs35l41_volatile_reg, + .readable_reg = cs35l41_readable_reg, + .precious_reg = cs35l41_precious_reg, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(cs35l41_regmap_i2c); + +struct regmap_config cs35l41_regmap_spi = { + .reg_bits = 32, + .val_bits = 32, + .pad_bits = 16, + .reg_stride = CS35L41_REGSTRIDE, + .reg_format_endian = REGMAP_ENDIAN_BIG, + .val_format_endian = REGMAP_ENDIAN_BIG, + .max_register = CS35L41_LASTREG, + .reg_defaults = cs35l41_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l41_reg), + .volatile_reg = cs35l41_volatile_reg, + .readable_reg = cs35l41_readable_reg, + .precious_reg = cs35l41_precious_reg, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(cs35l41_regmap_spi); diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index e8e997efaa8b..60332eae1162 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -1724,6 +1724,7 @@ int cs35l41_probe(struct cs35l41_private *cs35l41, return ret; } +EXPORT_SYMBOL_GPL(cs35l41_probe); void cs35l41_remove(struct cs35l41_private *cs35l41) { @@ -1732,6 +1733,7 @@ void cs35l41_remove(struct cs35l41_private *cs35l41) regulator_bulk_disable(CS35L41_NUM_SUPPLIES, cs35l41->supplies); gpiod_set_value_cansleep(cs35l41->reset_gpio, 0); } +EXPORT_SYMBOL_GPL(cs35l41_remove); MODULE_DESCRIPTION("ASoC CS35L41 driver"); MODULE_AUTHOR("David Rhodes, Cirrus Logic Inc, <david.rhodes(a)cirrus.com>"); diff --git a/sound/soc/codecs/cs35l41.h b/sound/soc/codecs/cs35l41.h index a23eabff074f..c7c45f19754b 100644 --- a/sound/soc/codecs/cs35l41.h +++ b/sound/soc/codecs/cs35l41.h @@ -538,7 +538,6 @@ #define CS35L41_OTP_TRIM_35 0x0000400C #define CS35L41_OTP_TRIM_36 0x00002030 -#define CS35L41_MAX_CACHE_REG 36 #define CS35L41_OTP_SIZE_WORDS 32 #define CS35L41_NUM_OTP_ELEM 100 #define CS35L41_NUM_OTP_MAPS 5 @@ -734,9 +733,8 @@ #define CS35L41_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) #define CS35L41_TX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) -bool cs35l41_readable_reg(struct device *dev, unsigned int reg); -bool cs35l41_precious_reg(struct device *dev, unsigned int reg); -bool cs35l41_volatile_reg(struct device *dev, unsigned int reg); +extern struct regmap_config cs35l41_regmap_i2c; +extern struct regmap_config cs35l41_regmap_spi; struct cs35l41_otp_packed_element_t { u32 reg; @@ -752,7 +750,6 @@ struct cs35l41_otp_map_element_t { u32 word_offset; }; -extern const struct reg_default cs35l41_reg[CS35L41_MAX_CACHE_REG]; extern const struct cs35l41_otp_map_element_t cs35l41_otp_map_map[CS35L41_NUM_OTP_MAPS]; -- 2.34.0

3 2

[PATCH] ASoC: SOF: mediatek: Use %pR/%pa to print resources/physical addresses
by Geert Uytterhoeven 27 Nov '21

27 Nov '21

On 32-bit with CONFIG_ARCH_DMA_ADDR_T_64BIT=n: sound/soc/sof/mediatek/mt8195/mt8195.c: In function ‘platform_parse_resource’: sound/soc/sof/mediatek/mt8195/mt8195.c:51:15: error: format ‘%llx’ expects argument of type ‘long long unsigned int’, but argument 4 has type ‘unsigned int’ [-Werror=format=] 51 | dev_dbg(dev, "DMA pbase=0x%llx, size=0x%llx\n", | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/sof/mediatek/mt8195/mt8195.c: In function ‘adsp_memory_remap_init’: sound/soc/sof/mediatek/mt8195/mt8195.c:167:15: error: format ‘%llx’ expects argument of type ‘long long unsigned int’, but argument 4 has type ‘phys_addr_t’ {aka ‘unsigned int’} [-Werror=format=] 167 | dev_dbg(dev, "adsp->pa_dram %llx, offset %#x\n", adsp->pa_dram, offset); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/sof/mediatek/mt8195/mt8195.c: In function ‘adsp_shared_base_ioremap’: sound/soc/sof/mediatek/mt8195/mt8195.c:196:15: error: format ‘%llx’ expects argument of type ‘long long unsigned int’, but argument 5 has type ‘phys_addr_t’ {aka ‘unsigned int’} [-Werror=format=] 196 | dev_dbg(dev, "shared-dram vbase=%p, phy addr :%llx, size=%#x\n", | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Fix the first cases by printing the full resource using %pR. Fix the other cases by printing the physical addresses using %pa. Reported-by: noreply(a)ellerman.id.au Fixes: 32d7e03d26fd9318 ("ASoC: SOF: mediatek: Add mt8195 hardware support") Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- sound/soc/sof/mediatek/mt8195/mt8195.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 40e5a25875a6a93c..b1580b6ac8e19e61 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -48,8 +48,7 @@ static int platform_parse_resource(struct platform_device *pdev, void *data) return ret; } - dev_dbg(dev, "DMA pbase=0x%llx, size=0x%llx\n", - (phys_addr_t)res.start, resource_size(&res)); + dev_dbg(dev, "DMA %pR\n", &res); ret = of_reserved_mem_device_init(dev); if (ret) { @@ -164,7 +163,7 @@ static int adsp_memory_remap_init(struct device *dev, struct mtk_adsp_chip_info offset = adsp->pa_dram - DRAM_PHYS_BASE_FROM_DSP_VIEW; adsp->dram_offset = offset; offset >>= DRAM_REMAP_SHIFT; - dev_dbg(dev, "adsp->pa_dram %llx, offset %#x\n", adsp->pa_dram, offset); + dev_dbg(dev, "adsp->pa_dram %pa, offset %#x\n", &adsp->pa_dram, offset); writel(offset, vaddr_emi_map); if (offset != readl(vaddr_emi_map)) { dev_err(dev, "write emi map fail : %#x\n", readl(vaddr_emi_map)); @@ -193,8 +192,8 @@ static int adsp_shared_base_ioremap(struct platform_device *pdev, void *data) return -ENOMEM; } } - dev_dbg(dev, "shared-dram vbase=%p, phy addr :%llx, size=%#x\n", - adsp->shared_dram, adsp->pa_shared_dram, shared_size); + dev_dbg(dev, "shared-dram vbase=%p, phy addr :%pa, size=%#x\n", + adsp->shared_dram, &adsp->pa_shared_dram, shared_size); return 0; } -- 2.25.1

2 1

[PATCH 00/10] ASoC: SOF: Fixes for Intel HD-Audio DMA stopping
by Kai Vehmanen 27 Nov '21

27 Nov '21

Implement an updated programming sequence to handle DMA stop for Intel HD-Audio DMA. The new flow is only used if the firmware is sufficiently new to support the feature. SOF1.9.2 is the first release with the updated flow. The kernel changes are backwards compatible with old firmware releases. Likewise new firmware releases will work with old kernel. Series reviewed originally at: https://github.com/thesofproject/linux/pull/3167 Buglink: https://github.com/thesofproject/sof/issues/4779 Ranjani Sridharan (10): ASoC: SOF: Intel: hda: clear stream before freeing the DAI widget ASoC: SOF: Intel: hda: Add a helper function for stream reset ASoC: SOF: Intel: hda: reset stream before coupling host and link DMA's ASoC: SOF: pcm: invoke platform hw_free for STOP/SUSPEND triggers ASoC: SOF: call platform hw_free for paused streams during suspend ASoC: SOF: Add a helper for freeing PCM stream ASoC: SOF: pcm: move the check for prepared flag ASoC: SOF: align the hw_free sequence with stop ASoC: SOF: IPC: dai: Expand DAI_CONFIG IPC flags ASoC: SOF: Intel: hda: send DAI_CONFIG IPC during pause include/sound/sof/dai.h | 25 +++++++--- sound/soc/sof/intel/hda-dai.c | 53 +++++++++++++++++---- sound/soc/sof/intel/hda-stream.c | 81 +++++++++++++++++++------------- sound/soc/sof/intel/hda.c | 26 ++++++---- sound/soc/sof/intel/hda.h | 6 ++- sound/soc/sof/pcm.c | 36 +++++++------- sound/soc/sof/sof-audio.c | 35 ++++++++++---- sound/soc/sof/sof-audio.h | 2 + 8 files changed, 180 insertions(+), 84 deletions(-) base-commit: cd2f33e93d578e9e1c47ef8981ec69298da9cb38 -- 2.33.0

2 11

[PATCH 1/2] ASoC: rt5682: Fix crash due to out of scope stack vars
by Rob Clark 27 Nov '21

27 Nov '21

From: Rob Clark <robdclark(a)chromium.org> Move the declaration of temporary arrays to somewhere that won't go out of scope before the devm_clk_hw_register() call, lest we be at the whim of the compiler for whether those stack variables get overwritten. Fixes a crash seen with gcc version 11.2.1 20210728 (Red Hat 11.2.1-1) Fixes: edbd24ea1e5c ("ASoC: rt5682: Drop usage of __clk_get_name()") Signed-off-by: Rob Clark <robdclark(a)chromium.org> --- sound/soc/codecs/rt5682.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 78b4cb5fb6c8..0d59ec2a5460 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2844,6 +2844,8 @@ int rt5682_register_dai_clks(struct rt5682_priv *rt5682) for (i = 0; i < RT5682_DAI_NUM_CLKS; ++i) { struct clk_init_data init = { }; + struct clk_parent_data parent_data; + const struct clk_hw *parent; dai_clk_hw = &rt5682->dai_clks_hw[i]; @@ -2851,17 +2853,17 @@ int rt5682_register_dai_clks(struct rt5682_priv *rt5682) case RT5682_DAI_WCLK_IDX: /* Make MCLK the parent of WCLK */ if (rt5682->mclk) { - init.parent_data = &(struct clk_parent_data){ + parent_data = (struct clk_parent_data){ .fw_name = "mclk", }; + init.parent_data = &parent_data; init.num_parents = 1; } break; case RT5682_DAI_BCLK_IDX: /* Make WCLK the parent of BCLK */ - init.parent_hws = &(const struct clk_hw *){ - &rt5682->dai_clks_hw[RT5682_DAI_WCLK_IDX] - }; + parent = &rt5682->dai_clks_hw[RT5682_DAI_WCLK_IDX]; + init.parent_hws = &parent; init.num_parents = 1; break; default: -- 2.33.1

3 4

[PATCH 0/6] Suspend related fixes on Tegra
by Sameer Pujar 27 Nov '21

27 Nov '21

This series addresses following problems: * The runtime PM is not balanced in MVC driver, whenever mute or volume mixer controls are set. * Some of the AHUB devices (SFC, MVC, Mixer, AMX and ADX) use late system sleep. Suspend failure is seen on Jetson TX2 platform. Sameer Pujar (6): ASoC: tegra: Balance runtime PM count ASoC: tegra: Use normal system sleep for SFC ASoC: tegra: Use normal system sleep for MVC ASoC: tegra: Use normal system sleep for Mixer ASoC: tegra: Use normal system sleep for AMX ASoC: tegra: Use normal system sleep for ADX sound/soc/tegra/tegra210_adx.c | 4 ++-- sound/soc/tegra/tegra210_amx.c | 4 ++-- sound/soc/tegra/tegra210_mixer.c | 4 ++-- sound/soc/tegra/tegra210_mvc.c | 8 ++++---- sound/soc/tegra/tegra210_sfc.c | 4 ++-- 5 files changed, 12 insertions(+), 12 deletions(-) -- 2.7.4

2 7

[bug report] ASoC: amd: add vangogh machine driver
by Dan Carpenter 26 Nov '21

26 Nov '21

Hello Vijendar Mukunda, The patch 34a0094b9ff7: "ASoC: amd: add vangogh machine driver" from Oct 14, 2021, leads to the following Smatch static checker warning: sound/soc/amd/vangogh/acp5x-mach.c:190 acp5x_cs35l41_hw_params() error: uninitialized symbol 'ret'. sound/soc/amd/vangogh/acp5x-mach.c 158 static int acp5x_cs35l41_hw_params(struct snd_pcm_substream *substream, 159 struct snd_pcm_hw_params *params) 160 { 161 struct snd_soc_pcm_runtime *rtd = substream->private_data; 162 struct snd_soc_card *card = rtd->card; 163 struct snd_soc_dai *codec_dai; 164 int ret, i; 165 unsigned int num_codecs = rtd->num_codecs; 166 unsigned int bclk_val; 167 168 for (i = 0; i < num_codecs; i++) { 169 codec_dai = asoc_rtd_to_codec(rtd, i); 170 if ((strcmp(codec_dai->name, "spi-VLV1776:00") == 0) || 171 (strcmp(codec_dai->name, "spi-VLV1776:01") == 0)) { How positive are we that we're always going to find one of these codecs? Smatch is worried we might not find them. 172 switch (params_rate(params)) { 173 case 48000: 174 bclk_val = 1536000; 175 break; 176 default: 177 dev_err(card->dev, "Invalid Samplerate:0x%x\n", 178 params_rate(params)); 179 return -EINVAL; 180 } 181 ret = snd_soc_component_set_sysclk(codec_dai->component, 182 0, 0, bclk_val, SND_SOC_CLOCK_IN); 183 if (ret < 0) { 184 dev_err(card->dev, "failed to set sysclk for CS35l41 dai\n"); 185 return ret; 186 } 187 } 188 } 189 --> 190 return ret; ^^^^^^^^^^ Also it's a bit more readable to "return 0;" if we know this is a success path. 191 } regards, dan carpenter

2 1

[PATCH v4 0/3] ALSA: hda: New NHLT functions and cleanup
by Cezary Rojewski 26 Nov '21

26 Nov '21

Changes add two crucial functions: endpoint presence-check and retrieval of endpoint's BLOB (hardware configuration) to NHLT API. Outside of that, "is DMIC in NHLT?" check is being re-visited. No need to check for channel count or anything DMIC-configuration related, just straight up verify link_type presence. Changes in v4: - patch renaming nhlt_acpi_table struct to acpi_nhlt_table has been dropped due to naming conflicts with other parts of the kernel - revoked previously appended tags from Mark and Pierre as every patch required an update due to above Changes in v3: - no code changes - appended Mark's Acked-by tag for patch 4/4 - appended Pierre's Reviewed-by tag for all patches Changes in v2: - patch "ALSA hda: Drop device-argument in NHLT functions" has been dropped - updated newly added declarations in intel-nhlt.h so warning: "no-previous-prototype-for-function" and error: "use-of-undeclared-identifier" are no longer observed when CONFIG_SND_INTEL_NHLT is not enabled - added Mark's tag to the last patch of the series Amadeusz Sławiński (3): ALSA: hda: Fill gaps in NHLT endpoint-interface ALSA: hda: Simplify DMIC-in-NHLT check ASoC: Intel: Skylake: Use NHLT API to search for blob include/sound/intel-nhlt.h | 37 +++++++-- sound/hda/intel-dsp-config.c | 2 +- sound/hda/intel-nhlt.c | 102 +++++++++++++++++++++++++ sound/soc/intel/skylake/skl-nhlt.c | 102 ------------------------- sound/soc/intel/skylake/skl-pcm.c | 3 + sound/soc/intel/skylake/skl-topology.c | 29 ++++--- sound/soc/intel/skylake/skl-topology.h | 1 + sound/soc/intel/skylake/skl.h | 4 - 8 files changed, 153 insertions(+), 127 deletions(-) -- 2.25.1

1 4

[PATCH v3 0/4] ALSA: hda: New NHLT functions and cleanup
by Cezary Rojewski 26 Nov '21

26 Nov '21

Changes add two crucial functions: endpoint presence-check and retrieval of endpoint's BLOB (hardware configuration) to NHLT API. Few cleanups accompany the above: Work is done to align NHLT-struct naming with other, commonly used ACPI-structs. While cleaning up, don't forget about "is DMIC in NHLT?" check. No need to check for channel count or anything DMIC-configuration related, just straight up verify link_type presence. Changes in v3: - no code changes - appended Mark's Acked-by tag for patch 4/4 - appended Pierre's Reviewed-by tag for all patches Changes in v2: - patch "ALSA hda: Drop device-argument in NHLT functions" has been dropped - updated newly added declarations in intel-nhlt.h so warning: "no-previous-prototype-for-function" and error: "use-of-undeclared-identifier" are no longer observed when CONFIG_SND_INTEL_NHLT is not enabled - added Mark's tag to the last patch of the series Amadeusz Sławiński (4): ALSA: hda: Follow ACPI convention in NHLT struct naming ALSA: hda: Fill gaps in NHLT endpoint-interface ALSA: hda: Simplify DMIC-in-NHLT check ASoC: Intel: Skylake: Use NHLT API to search for blob include/sound/intel-nhlt.h | 53 ++++++++---- sound/hda/intel-dsp-config.c | 4 +- sound/hda/intel-nhlt.c | 110 ++++++++++++++++++++++++- sound/soc/intel/skylake/skl-nhlt.c | 108 +----------------------- sound/soc/intel/skylake/skl-pcm.c | 3 + sound/soc/intel/skylake/skl-topology.c | 29 ++++--- sound/soc/intel/skylake/skl-topology.h | 1 + sound/soc/intel/skylake/skl.h | 6 +- sound/soc/sof/intel/hda.c | 2 +- 9 files changed, 171 insertions(+), 145 deletions(-) -- 2.25.1

3 8

[PATCH v2 0/8] ASoC: SOF: Add support for Mediatek MT8195
by Daniel Baluta 26 Nov '21

26 Nov '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This adds sound open firmware driver support for MT8915 platform. Changes since v1: - removed AMD patches since they were already applied - fixed compilation error reported by Mark, generated by the fact that mt8195_run was declared too early in the series. YC Hung (8): ASoC: SOF: mediatek: Add mt8195 hardware support ASoC: SOF: tokens: add token for Mediatek AFE ASoC: SOF: topology: Add support for Mediatek AFE DAI ASoC: SOF: mediatek: Add fw loader and mt8195 dsp ops to load firmware ASoC: SOF: Add mt8195 device descriptor ASoC: SOF: mediatek: Add dai driver dsp ops callback for mt8195 ASoC: SOF: mediatek: Add mt8195 dsp clock support ASoC: SOF: mediatek: Add DSP system PM callback for mt8195 include/sound/sof/dai-mediatek.h | 23 + include/sound/sof/dai.h | 3 + include/uapi/sound/sof/tokens.h | 5 + sound/soc/sof/Kconfig | 3 +- sound/soc/sof/Makefile | 1 + sound/soc/sof/mediatek/Kconfig | 33 ++ sound/soc/sof/mediatek/Makefile | 2 + sound/soc/sof/mediatek/adsp_helper.h | 49 ++ sound/soc/sof/mediatek/mediatek-ops.h | 8 + sound/soc/sof/mediatek/mt8195/Makefile | 3 + sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 158 +++++++ sound/soc/sof/mediatek/mt8195/mt8195-clk.h | 28 ++ sound/soc/sof/mediatek/mt8195/mt8195-loader.c | 56 +++ sound/soc/sof/mediatek/mt8195/mt8195.c | 437 ++++++++++++++++++ sound/soc/sof/mediatek/mt8195/mt8195.h | 158 +++++++ sound/soc/sof/pcm.c | 12 + sound/soc/sof/sof-of-dev.c | 13 + sound/soc/sof/topology.c | 59 +++ 18 files changed, 1050 insertions(+), 1 deletion(-) create mode 100644 include/sound/sof/dai-mediatek.h create mode 100644 sound/soc/sof/mediatek/Kconfig create mode 100644 sound/soc/sof/mediatek/Makefile create mode 100644 sound/soc/sof/mediatek/adsp_helper.h create mode 100644 sound/soc/sof/mediatek/mediatek-ops.h create mode 100644 sound/soc/sof/mediatek/mt8195/Makefile create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.c create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.h create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-loader.c create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195.c create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195.h -- 2.27.0

3 17

[PATCH v4 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 26 Nov '21

26 Nov '21

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. This patch set depends on: -- https://patchwork.kernel.org/project/alsa-devel/list/?series=570161 -- https://patchwork.kernel.org/project/alsa-devel/list/?series=572615 -- https://patchwork.kernel.org/project/alsa-devel/list/?series=559677 Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add regmap config support for codec dma driver ASoC: dt-bindings: Add SC7280 sound card bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio ASoC: qcom: SC7280: Update config for building codec dma drivers .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 69 ++- sound/soc/qcom/Kconfig | 13 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/common.c | 39 ++ sound/soc/qcom/common.h | 1 + sound/soc/qcom/lpass-cdc-dma.c | 195 ++++++++ sound/soc/qcom/lpass-cpu.c | 245 +++++++++- sound/soc/qcom/lpass-lpaif-reg.h | 103 ++++- sound/soc/qcom/lpass-platform.c | 513 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 417 +++++++++++++++++ sound/soc/qcom/lpass.h | 150 ++++++ 11 files changed, 1670 insertions(+), 79 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 15

[PATCH v5 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 26 Nov '21

26 Nov '21

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. This patch set depends on: -- https://patchwork.kernel.org/project/alsa-devel/list/?series=570161 -- https://patchwork.kernel.org/project/alsa-devel/list/?series=572615 -- https://patchwork.kernel.org/project/alsa-devel/list/?series=559677 Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add regmap config support for codec dma driver ASoC: dt-bindings: Add SC7280 sound card bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio ASoC: qcom: SC7280: Update config for building codec dma drivers .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 69 ++- sound/soc/qcom/Kconfig | 13 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/common.c | 39 ++ sound/soc/qcom/common.h | 1 + sound/soc/qcom/lpass-cdc-dma.c | 195 ++++++++ sound/soc/qcom/lpass-cpu.c | 245 +++++++++- sound/soc/qcom/lpass-lpaif-reg.h | 103 ++++- sound/soc/qcom/lpass-platform.c | 513 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 416 +++++++++++++++++ sound/soc/qcom/lpass.h | 150 ++++++ 11 files changed, 1669 insertions(+), 79 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 9

[PATCH v6 0/3] This patches provide ADSP IPC support for MT8195
by allen-kh.cheng 26 Nov '21

26 Nov '21

Mediatek ADSP IPC is used to send notification or short message between processors with dsp.· It will place the message to the share buffer and will access the ADSP mailbox registers to kick dsp. Two mailboxes used to send notification or short message between processors with dsp changes since v5: - remove some redundant code changes since v4: - use switch ... case in adsp_ipc_recv - add error handling path for chan_name pointer - refine some code to be concise changes since v3: - reorder MTK_ADSP_IPC_MBOX config - remove some redundant code - remove lock in mtk-adsp-mailbox changes since v2: - separate adsp_mailbox into two instances changes since v1: - fix dt_binding_check error Allen-KH Cheng (3): dt-bindings: mediatek: add adsp-mbox document firmware: mediatek: add adsp ipc protocol interface mailbox: mediatek: add support for adsp mailbox controller .../bindings/mailbox/mtk,adsp-mbox.yaml | 52 +++++ drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 9 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 138 ++++++++++++++ drivers/mailbox/Kconfig | 7 + drivers/mailbox/Makefile | 2 + drivers/mailbox/mtk-adsp-mailbox.c | 178 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 70 +++++++ 10 files changed, 460 insertions(+) create mode 100644 Documentation/devicetree/bindings/mailbox/mtk,adsp-mbox.yaml create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 drivers/mailbox/mtk-adsp-mailbox.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

3 7

[PATCH v5 05/10] ASoC: qcom: Add helper function to get dma control and lpaif handle
by Srinivasa Rao Mandadapu 26 Nov '21

26 Nov '21

Add support function to get dma control and lpaif handle to avoid repeated code in platform driver Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> Co-developed-by: Venkata Prasad Potturu <potturu(a)codeaurora.org> Signed-off-by: Venkata Prasad Potturu <potturu(a)codeaurora.org> --- sound/soc/qcom/lpass-platform.c | 90 ++++++++++++++++++++--------------------- 1 file changed, 43 insertions(+), 47 deletions(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index a44162c..59c0884 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -177,6 +177,44 @@ static int lpass_platform_pcmops_close(struct snd_soc_component *component, return 0; } +static void __get_lpaif_handle(struct snd_pcm_substream *substream, + struct snd_soc_component *component, + struct lpaif_dmactl **dmactl, int *id, struct regmap **map) +{ + struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); + struct lpass_data *drvdata = snd_soc_component_get_drvdata(component); + struct snd_pcm_runtime *rt = substream->runtime; + struct lpass_pcm_data *pcm_data = rt->private_data; + struct lpass_variant *v = drvdata->variant; + int dir = substream->stream; + unsigned int dai_id = cpu_dai->driver->id; + struct lpaif_dmactl *l_dmactl; + struct regmap *l_map; + int l_id; + + if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + l_id = pcm_data->dma_ch; + if (dai_id == LPASS_DP_RX) { + l_dmactl = drvdata->hdmi_rd_dmactl; + l_map = drvdata->hdmiif_map; + } else { + l_dmactl = drvdata->rd_dmactl; + l_map = drvdata->lpaif_map; + } + } else { + l_dmactl = drvdata->wr_dmactl; + l_id = pcm_data->dma_ch - v->wrdma_channel_start; + l_map = drvdata->lpaif_map; + } + if (dmactl) + *dmactl = l_dmactl; + if (id) + *id = l_id; + if (map) + *map = l_map; +} + static int lpass_platform_pcmops_hw_params(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -191,22 +229,12 @@ static int lpass_platform_pcmops_hw_params(struct snd_soc_component *component, unsigned int channels = params_channels(params); unsigned int regval; struct lpaif_dmactl *dmactl; - int id, dir = substream->stream; + int id; int bitwidth; int ret, dma_port = pcm_data->i2s_port + v->dmactl_audif_start; unsigned int dai_id = cpu_dai->driver->id; - if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - id = pcm_data->dma_ch; - if (dai_id == LPASS_DP_RX) - dmactl = drvdata->hdmi_rd_dmactl; - else - dmactl = drvdata->rd_dmactl; - - } else { - dmactl = drvdata->wr_dmactl; - id = pcm_data->dma_ch - v->wrdma_channel_start; - } + __get_lpaif_handle(substream, component, &dmactl, &id, NULL); bitwidth = snd_pcm_format_width(format); if (bitwidth < 0) { @@ -379,24 +407,9 @@ static int lpass_platform_pcmops_prepare(struct snd_soc_component *component, int ret, id, ch, dir = substream->stream; unsigned int dai_id = cpu_dai->driver->id; - ch = pcm_data->dma_ch; - if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - if (dai_id == LPASS_DP_RX) { - dmactl = drvdata->hdmi_rd_dmactl; - map = drvdata->hdmiif_map; - } else { - dmactl = drvdata->rd_dmactl; - map = drvdata->lpaif_map; - } - - id = pcm_data->dma_ch; - } else { - dmactl = drvdata->wr_dmactl; - id = pcm_data->dma_ch - v->wrdma_channel_start; - map = drvdata->lpaif_map; - } + __get_lpaif_handle(substream, component, &dmactl, &id, &map); ret = regmap_write(map, LPAIF_DMABASE_REG(v, ch, dir, dai_id), runtime->dma_addr); if (ret) { @@ -444,26 +457,12 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, struct lpaif_dmactl *dmactl; struct regmap *map; int ret, ch, id; - int dir = substream->stream; unsigned int reg_irqclr = 0, val_irqclr = 0; unsigned int reg_irqen = 0, val_irqen = 0, val_mask = 0; unsigned int dai_id = cpu_dai->driver->id; ch = pcm_data->dma_ch; - if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - id = pcm_data->dma_ch; - if (dai_id == LPASS_DP_RX) { - dmactl = drvdata->hdmi_rd_dmactl; - map = drvdata->hdmiif_map; - } else { - dmactl = drvdata->rd_dmactl; - map = drvdata->lpaif_map; - } - } else { - dmactl = drvdata->wr_dmactl; - id = pcm_data->dma_ch - v->wrdma_channel_start; - map = drvdata->lpaif_map; - } + __get_lpaif_handle(substream, component, &dmactl, &id, &map); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -597,10 +596,7 @@ static snd_pcm_uframes_t lpass_platform_pcmops_pointer( struct regmap *map; unsigned int dai_id = cpu_dai->driver->id; - if (dai_id == LPASS_DP_RX) - map = drvdata->hdmiif_map; - else - map = drvdata->lpaif_map; + __get_lpaif_handle(substream, component, NULL, NULL, &map); ch = pcm_data->dma_ch; -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 11

[PATCH v5 0/3] This patches provide ADSP IPC support for MT8195
by allen-kh.cheng 26 Nov '21

26 Nov '21

Mediatek ADSP IPC is used to send notification or short message between processors with dsp.· It will place the message to the share buffer and will access the ADSP mailbox registers to kick dsp. Two mailboxes used to send notification or short message between processors with dsp changes since v4: - use switch ... case in adsp_ipc_recv - add error handling path for chan_name pointer - refine some code to be concise changes since v3: - reorder MTK_ADSP_IPC_MBOX config - remove some redundant code - remove lock in mtk-adsp-mailbox changes since v2: - separate adsp_mailbox into two instances changes since v1: - fix dt_binding_check error Allen-KH Cheng (3): dt-bindings: mediatek: add adsp-mbox document firmware: mediatek: add adsp ipc protocol interface mailbox: mediatek: add support for adsp mailbox controller .../bindings/mailbox/mtk,adsp-mbox.yaml | 52 +++++ drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 9 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 138 ++++++++++++++ drivers/mailbox/Kconfig | 7 + drivers/mailbox/Makefile | 2 + drivers/mailbox/mtk-adsp-mailbox.c | 178 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 70 +++++++ 10 files changed, 460 insertions(+) create mode 100644 Documentation/devicetree/bindings/mailbox/mtk,adsp-mbox.yaml create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 drivers/mailbox/mtk-adsp-mailbox.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

2 5

[PATCH v4 0/3] This patches provide ADSP IPC support for MT8195
by allen-kh.cheng 26 Nov '21

26 Nov '21

Mediatek ADSP IPC is used to send notification or short message between processors with dsp.· It will place the message to the share buffer and will access the ADSP mailbox registers to kick dsp. Two mailboxes used to send notification or short message between processors with dsp changes since v3: - reorder MTK_ADSP_IPC_MBOX config - remove some redundant code - remove lock in mtk-adsp-mailbox changes since v2: - seperate adsp_mailbox into two instances changes since v1: - fix dt_binding_check error Allen-KH Cheng (3): dt-bindings: mediatek: add adsp-mbox document firmware: mediatek: add adsp ipc protocol interface mailbox: mediatek: add support for adsp mailbox controller .../bindings/mailbox/mtk,adsp-mbox.yaml | 52 +++++ drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 9 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 132 +++++++++++++ drivers/mailbox/Kconfig | 7 + drivers/mailbox/Makefile | 2 + drivers/mailbox/mtk-adsp-mailbox.c | 184 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 70 +++++++ 10 files changed, 460 insertions(+) create mode 100644 Documentation/devicetree/bindings/mailbox/mtk,adsp-mbox.yaml create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 drivers/mailbox/mtk-adsp-mailbox.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

2 5

[PATCH v1 00/20] Support HDMI audio on NVIDIA Tegra20
by Dmitry Osipenko 26 Nov '21

26 Nov '21

This series revives Tegra20 S/PDIF driver which was upstreamed long time ago, but never was used. It also turns Tegra DRM HDMI driver into HDMI audio CODEC provider. Finally, HDMI audio is enabled in device-trees. For now the audio is enable only for Acer A500 tablet and Toshiba AC100 netbook because they're already supported by upstream, later on ASUS TF101 tablet will join them. I based S/PDIF patches on Arnd's Bergmann patch from a separate series [1] that removes obsolete slave_id. This eases merging of the patches by removing the merge conflict. [1] https://patchwork.ozlabs.org/project/linux-tegra/list/?submitter=80402 Arnd Bergmann (1): ASoC: tegra20-spdif: stop setting slave_id Dmitry Osipenko (19): ASoC: dt-bindings: Add binding for Tegra20 S/PDIF ASoC: dt-bindings: tegra20-i2s: Convert to schema ASoC: dt-bindings: tegra20-i2s: Document new nvidia,fixed-parent-rate property dt-bindings: host1x: Document optional HDMI sound-dai-cells ASoC: tegra20: spdif: Support device-tree ASoC: tegra20: spdif: Set FIFO trigger level ASoC: tegra20: spdif: Improve driver's code ASoC: tegra20: spdif: Use more resource-managed helpers ASoC: tegra20: spdif: Reset hardware ASoC: tegra20: spdif: Support system suspend ASoC: tegra20: spdif: Filter out unsupported rates ASoC: tegra20: i2s: Filter out unsupported rates drm/tegra: hdmi: Unwind tegra_hdmi_init() errors drm/tegra: hdmi: Register audio CODEC on Tegra20 ARM: tegra_defconfig: Enable S/PDIF driver ARM: tegra: Add S/PDIF node to Tegra20 device-tree ARM: tegra: Add HDMI audio graph to Tegra20 device-tree ARM: tegra: acer-a500: Enable S/PDIF and HDMI audio ARM: tegra: paz00: Enable S/PDIF and HDMI audio .../display/tegra/nvidia,tegra20-host1x.txt | 1 + .../bindings/sound/nvidia,tegra20-i2s.txt | 30 --- .../bindings/sound/nvidia,tegra20-i2s.yaml | 78 +++++++ .../bindings/sound/nvidia,tegra20-spdif.yaml | 88 ++++++++ .../boot/dts/tegra20-acer-a500-picasso.dts | 8 + arch/arm/boot/dts/tegra20-paz00.dts | 8 + arch/arm/boot/dts/tegra20.dtsi | 40 +++- arch/arm/configs/tegra_defconfig | 1 + drivers/gpu/drm/tegra/Kconfig | 3 + drivers/gpu/drm/tegra/hdmi.c | 168 +++++++++++++-- sound/soc/tegra/tegra20_i2s.c | 49 +++++ sound/soc/tegra/tegra20_spdif.c | 195 +++++++++++++----- sound/soc/tegra/tegra20_spdif.h | 1 + sound/soc/tegra/tegra_pcm.c | 6 + sound/soc/tegra/tegra_pcm.h | 1 + 15 files changed, 575 insertions(+), 102 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-i2s.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-spdif.yaml -- 2.33.1

3 29

[PATCH v2] ASoC: dt-bindings: davinci-mcasp: convert McASP bindings to yaml schema
by Jayesh Choudhary 25 Nov '21

25 Nov '21

Convert the bindings for McASP controllers for TI SOCs from txt to YAML schema. Adds additional properties 'clocks', 'clock-names', 'power-domains' '#sound-dai-cells', 'num-serializer' and 'port' which were not there in the txt file. Adds 'dmas' and 'dma-names' in the example which were not there in the txt file. Changes 'interrupts' and 'interrupt-names' from optional to required properties. Signed-off-by: Jayesh Choudhary <j-choudhary(a)ti.com> --- Changelog: v2: - changes the commit message - modifies the properties 'clocks', 'clock-names', 'dma-names', 'dmas', 'interrupts' and 'interrupt-names' according to the arm SOCs - adds 'port' and 'num-serializer' as node properties. .../bindings/sound/davinci-mcasp-audio.txt | 86 --------- .../bindings/sound/davinci-mcasp-audio.yaml | 179 ++++++++++++++++++ 2 files changed, 179 insertions(+), 86 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt deleted file mode 100644 index bd863bd69501..000000000000 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ /dev/null @@ -1,86 +0,0 @@ -Texas Instruments McASP controller - -Required properties: -- compatible : - "ti,dm646x-mcasp-audio" : for DM646x platforms - "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms - "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx) - "ti,dra7-mcasp-audio" : for DRA7xx platforms - "ti,omap4-mcasp-audio" : for OMAP4 - -- reg : Should contain reg specifiers for the entries in the reg-names property. -- reg-names : Should contain: - * "mpu" for the main registers (required). For compatibility with - existing software, it is recommended this is the first entry. - * "dat" for separate data port register access (optional). -- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF, - IEC60958-1, and AES-3 formats. -- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted - or received over one serializer. -- serial-dir : A list of serializer configuration. Each entry is a number - indication for serializer pin direction. - (0 - INACTIVE, 1 - TX, 2 - RX) -- dmas: two element list of DMA controller phandles and DMA request line - ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. The dma - identifiers must be "rx" and "tx". - -Optional properties: - -- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0 -- tx-num-evt : FIFO levels. -- rx-num-evt : FIFO levels. -- dismod : Specify the drive on TX pin during inactive slots - 0 : 3-state - 2 : logic low - 3 : logic high - Defaults to 'logic low' when the property is not present -- sram-size-playback : size of sram to be allocated during playback -- sram-size-capture : size of sram to be allocated during capture -- interrupts : Interrupt numbers for McASP -- interrupt-names : Known interrupt names are "tx" and "rx" -- pinctrl-0: Should specify pin control group used for this controller. -- pinctrl-names: Should contain only one value - "default", for more details - please refer to pinctrl-bindings.txt -- fck_parent : Should contain a valid clock name which will be used as parent - for the McASP fck -- auxclk-fs-ratio: When McASP is bus master indicates the ratio between AUCLK - and FS rate if applicable: - AUCLK rate = auxclk-fs-ratio * FS rate - -Optional GPIO support: -If any McASP pin need to be used as GPIO then the McASP node must have: -... - gpio-controller - #gpio-cells = <2>; -... - -When requesting a GPIO, the first parameter is the PIN index in McASP_P* -registers. -For example to request the AXR2 pin of mcasp8: -function-gpios = <&mcasp8 2 0>; - -Or to request the ACLKR pin of mcasp8: -function-gpios = <&mcasp8 29 0>; - -For generic gpio information, please refer to bindings/gpio/gpio.txt - -Example: - -mcasp0: mcasp0@1d00000 { - compatible = "ti,da830-mcasp-audio"; - reg = <0x100000 0x3000>; - reg-names "mpu"; - interrupts = <82>, <83>; - interrupt-names = "tx", "rx"; - op-mode = <0>; /* MCASP_IIS_MODE */ - tdm-slots = <2>; - serial-dir = < - 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ - 0 0 0 0 - 0 0 0 1 - 2 0 0 0 >; - tx-num-evt = <1>; - rx-num-evt = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml new file mode 100644 index 000000000000..2f8b2aca8c34 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -0,0 +1,179 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/davinci-mcasp-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: McASP Controller for TI SoCs + +maintainers: + - Jayesh Choudhary <j-choudhary(a)ti.com> + +properties: + compatible: + enum: + - ti,dm646x-mcasp-audio + - ti,da830-mcasp-audio + - ti,am33xx-mcasp-audio + - ti,dra7-mcasp-audio + - ti,omap4-mcasp-audio + + reg: + minItems: 1 + items: + - description: main registers + - description: data port register + + reg-names: + minItems: 1 + items: + - const: mpu + - const: dat + + op-mode: + description: I2S - 0 or DIT - 1 mode + enum: + - 0 + - 1 + + tdm-slots: + maxItems: 1 + + serial-dir: + description: + A list of serializer configuration + Entry is indication for serializer pin direction + 0 - Inactive, 1 - TX, 2 - RX + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 16 + items: + minimum: 0 + maximum: 2 + default: 0 + + dmas: + minItems: 1 + items: + - description: transmission DMA channel + - description: reception DMA channel + + dma-names: + minItems: 1 + items: + - const: tx + - const: rx + + ti,hwmods: + $ref: /schemas/types.yaml#/definitions/string + description: Name of hwmod associated with McASP + maxItems: 1 + deprecated: true + + tx-num-evt: + maxItems: 1 + + rx-num-evt: + maxItems: 1 + + dismod: + enum: + - 0 + - 2 + - 3 + default: 2 + + sram-size-playback: + maxItems: 1 + + sram-size-capture: + maxItems: 1 + + interrupts: + minItems: 1 + items: + - description: TX FIFO interrupt + - description: RX FIFO interrupt + + interrupt-names: + oneOf: + - minItems: 1 + items: + - const: tx + - const: rx + - const: common + + fck_parent: + description: parent clock for McASP fck + maxItems: 1 + + auxclk-fs-ratio: + description: ratio of AUCLK and FS if applicable + maxItems: 1 + + gpio-controller: true + + "#gpio-cells": + const: 2 + + function-gpios: + maxItems: 1 + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + minItems: 1 + maxItems: 3 + items: + - const: fck + - const: ahclkx + - const: ahclkr + + power-domains: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + num-serializer: + maxItems: 1 + + port: + type: object + +required: + - compatible + - reg + - reg-names + - dmas + - dma-names + - interrupts + - interrupt-names + - serial-dir + - op-mode + - tdm-slots + +additionalProperties: false + +examples: + - | + mcasp0: mcasp0@1d00000 { + compatible = "ti,da830-mcasp-audio"; + reg = <0x100000 0x3000>; + reg-names = "mpu"; + interrupts = <82>, <83>; + interrupt-names = "tx", "rx"; + op-mode = <0>; /* MCASP_IIS_MODE */ + tdm-slots = <2>; + dmas = <&main_udmap 0xc400>, <&main_udmap 0x4400>; + dma-names = "tx", "rx"; + serial-dir = < + 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ + 0 0 0 0 + 0 0 0 1 + 2 0 0 0 >; + tx-num-evt = <1>; + rx-num-evt = <1>; + }; -- 2.17.1

2 2

[PATCH 0/5] xen: cleanup detection of non-essential pv devices
by Juergen Gross 25 Nov '21

25 Nov '21

Today the non-essential pv devices are hard coded in the xenbus driver and this list is lacking multiple entries. This series reworks the detection logic of non-essential devices by adding a flag for that purpose to struct xenbus_driver. Juergen Gross (5): xen: add "not_essential" flag to struct xenbus_driver xen: flag xen_drm_front to be not essential for system boot xen: flag hvc_xen to be not essential for system boot xen: flag pvcalls-front to be not essential for system boot xen: flag xen_snd_front to be not essential for system boot drivers/gpu/drm/xen/xen_drm_front.c | 1 + drivers/input/misc/xen-kbdfront.c | 1 + drivers/tty/hvc/hvc_xen.c | 1 + drivers/video/fbdev/xen-fbfront.c | 1 + drivers/xen/pvcalls-front.c | 1 + drivers/xen/xenbus/xenbus_probe_frontend.c | 14 +++----------- include/xen/xenbus.h | 1 + sound/xen/xen_snd_front.c | 1 + 8 files changed, 10 insertions(+), 11 deletions(-) -- 2.26.2

5 8

[GIT PULL] ASoC fixes for v5.16-rc3
by Mark Brown 25 Nov '21

25 Nov '21

The following changes since commit 424fe7edbed18d47f7b97f7e1322a6f8969b77ae: ASoC: stm32: i2s: fix 32 bits channel length without mclk (2021-11-17 13:04:38 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.16-rc3 for you to fetch changes up to 86f74ba3fef56dd1cee19b7a15ae27fc0da5bb61: ASoC: SOF: hda: reset DAI widget before reconfiguring it (2021-11-24 12:57:11 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.16 There's a large but repetitive set of fixes here for issues with the Tegra kcontrols not correctly reporting changes to userspace, a fix for some issues with matching on older x86 platforms introduced during the merge window together with a set of smaller fixes and one new system quirk. ---------------------------------------------------------------- Hans de Goede (1): ASoC: soc-acpi: Set mach->id field on comp_ids matches Lucas Tanure (1): ASoC: cs35l41: Set the max SPI speed for the whole device Nicolas Frattaroli (1): ASoC: rk817: Add module alias for rk817-codec Pierre-Louis Bossart (1): ASoC: Intel: soc-acpi: add entry for ESSX8336 on CML Ranjani Sridharan (1): ASoC: SOF: hda: reset DAI widget before reconfiguring it Sameer Pujar (16): ASoC: tegra: Fix wrong value type in ADMAIF ASoC: tegra: Fix wrong value type in I2S ASoC: tegra: Fix wrong value type in DMIC ASoC: tegra: Fix wrong value type in DSPK ASoC: tegra: Fix wrong value type in SFC ASoC: tegra: Fix wrong value type in MVC ASoC: tegra: Fix kcontrol put callback in ADMAIF ASoC: tegra: Fix kcontrol put callback in I2S ASoC: tegra: Fix kcontrol put callback in DMIC ASoC: tegra: Fix kcontrol put callback in DSPK ASoC: tegra: Fix kcontrol put callback in AHUB ASoC: tegra: Fix kcontrol put callback in MVC ASoC: tegra: Fix kcontrol put callback in SFC ASoC: tegra: Fix kcontrol put callback in AMX ASoC: tegra: Fix kcontrol put callback in ADX ASoC: tegra: Fix kcontrol put callback in Mixer include/sound/soc-acpi.h | 2 +- sound/soc/codecs/cs35l41-spi.c | 32 +-- sound/soc/codecs/cs35l41.c | 7 - sound/soc/codecs/cs35l41.h | 4 +- sound/soc/codecs/rk817_codec.c | 1 + sound/soc/intel/common/soc-acpi-intel-cml-match.c | 6 + sound/soc/soc-acpi.c | 4 +- sound/soc/sof/intel/hda.c | 7 + sound/soc/tegra/tegra186_dspk.c | 181 ++++++++++--- sound/soc/tegra/tegra210_admaif.c | 140 +++++++--- sound/soc/tegra/tegra210_adx.c | 3 + sound/soc/tegra/tegra210_ahub.c | 11 +- sound/soc/tegra/tegra210_amx.c | 3 + sound/soc/tegra/tegra210_dmic.c | 184 +++++++++++--- sound/soc/tegra/tegra210_i2s.c | 296 +++++++++++++++++----- sound/soc/tegra/tegra210_mixer.c | 26 +- sound/soc/tegra/tegra210_mvc.c | 30 ++- sound/soc/tegra/tegra210_sfc.c | 123 ++++++--- 18 files changed, 802 insertions(+), 258 deletions(-)

2 1

[PATCH 0/2] Kconfig symbol clean-up for sound
by Lukas Bulwahn 25 Nov '21

25 Nov '21

Dear sound maintainers, The script ./scripts/checkkconfigsymbols.py warns on invalid references to Kconfig symbols (often, minor typos, name confusions or outdated references). This is a patch series addressing the issues reported by ./scripts/checkkconfigsymbols.py in the ./sound/ directory for Kconfig files, quickly filtered down with: ./scripts/checkkconfigsymbols.py | grep "sound.*Kconfig" -B 1 This patch series addresses: SND_SOC_UNIPHIER_AIO_DMA Referencing files: sound/soc/uniphier/Kconfig SND_SOC_WCD937X Referencing files: sound/soc/codecs/Kconfig This patch series does not address: CLK_FIXED_FCH Referencing files: sound/soc/amd/Kconfig, sound/soc/amd/acp/Kconfig Rationale: config definition has been submitted, but is still pending for inclusion through another tree. SND_SOC_AC97_BUS_NEW Referencing files: sound/soc/pxa/Kconfig Rationale: waiting for response; the kernel developer has been informed here: https://lore.kernel.org/all/CAKXUXMzqgyNGEnxAMQqZiXJYSK-X8uB-nxHWwPg41H6yS0… Both patches in this series can be discussed and applied individually. They are sent in one patch series, as they orginate from the investigation guided by the same tool and hence share similar topics and resolutions. Please pick this series of minor clean-up patches. It applies cleanly on next-20211124. Best regards, Lukas Lukas Bulwahn (2): ASoC: uniphier: drop selecting non-existing SND_SOC_UNIPHIER_AIO_DMA ASoC: codecs: wcd938x: add SND_SOC_WCD938_SDW to codec list instead sound/soc/codecs/Kconfig | 3 +-- sound/soc/uniphier/Kconfig | 2 -- 2 files changed, 1 insertion(+), 4 deletions(-) -- 2.26.2

2 3

[PATCH -next] ASoC: SOF: mediatek: Add missing of_node_put() in platform_parse_resource()
by Yang Yingliang 25 Nov '21

25 Nov '21

The node pointer is returned by of_parse_phandle() with refcount incremented in platform_parse_resource(). Calling of_node_put() to aovid the refcount leak. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 40e5a25875a6..55d9812870a4 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -43,6 +43,7 @@ static int platform_parse_resource(struct platform_device *pdev, void *data) } ret = of_address_to_resource(mem_region, 0, &res); + of_node_put(mem_region); if (ret) { dev_err(dev, "of_address_to_resource dma failed\n"); return ret; @@ -64,6 +65,7 @@ static int platform_parse_resource(struct platform_device *pdev, void *data) } ret = of_address_to_resource(mem_region, 0, &res); + of_node_put(mem_region); if (ret) { dev_err(dev, "of_address_to_resource sysmem failed\n"); return ret; -- 2.25.1

2 1

[PATCH 1/2] ASoC: rt5640: Add the binding include file for the HDA header support
by Oder Chiou 25 Nov '21

25 Nov '21

The patch adds the binding include file for the HDA header support. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/dt-bindings/sound/rt5640.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/dt-bindings/sound/rt5640.h b/include/dt-bindings/sound/rt5640.h index 154c9b4414f2..655f6946388a 100644 --- a/include/dt-bindings/sound/rt5640.h +++ b/include/dt-bindings/sound/rt5640.h @@ -16,6 +16,7 @@ #define RT5640_JD_SRC_GPIO2 4 #define RT5640_JD_SRC_GPIO3 5 #define RT5640_JD_SRC_GPIO4 6 +#define RT5640_JD_SRC_HDA_HEADER 7 #define RT5640_OVCD_SF_0P5 0 #define RT5640_OVCD_SF_0P75 1 -- 2.33.1

2 2

[PATCH] ASoC: mediatek: remove unnecessary CONFIG_PM
by Jiaxin Yu 25 Nov '21

25 Nov '21

The unnecessary conditional inclusion caused the following warning. Such as: >> sound/soc/mediatek/mt8192/mt8192-afe-pcm.c:2368:32: warning: unused >> variable 'mt8192_afe_pm_ops' [-Wunused-const-variable] static const struct dev_pm_ops mt8192_afe_pm_ops = { Because runtime_pm already handles the case without CONFIG_PM, we can remove CONFIG_PM condition. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> Reported-by: kernel test robot <lkp(a)intel.com> --- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 -- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 -- sound/soc/mediatek/mt8173/mt8173-max98090.c | 2 -- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 -- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 -- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 -- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 -- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 2 -- 8 files changed, 16 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index bc3d0466472b..0f178de92a0f 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -1474,9 +1474,7 @@ static struct platform_driver mt2701_afe_pcm_driver = { .driver = { .name = "mt2701-audio", .of_match_table = mt2701_afe_pcm_dt_match, -#ifdef CONFIG_PM .pm = &mt2701_afe_pm_ops, -#endif }, .probe = mt2701_afe_pcm_dev_probe, .remove = mt2701_afe_pcm_dev_remove, diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 3d68e4726ea2..fb4abec9aa5f 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -901,9 +901,7 @@ static struct platform_driver mt6797_afe_pcm_driver = { .driver = { .name = "mt6797-audio", .of_match_table = mt6797_afe_pcm_dt_match, -#ifdef CONFIG_PM .pm = &mt6797_afe_pm_ops, -#endif }, .probe = mt6797_afe_pcm_dev_probe, .remove = mt6797_afe_pcm_dev_remove, diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c index fc94314bfc02..2408c9d3d9b3 100644 --- a/sound/soc/mediatek/mt8173/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c @@ -193,9 +193,7 @@ static struct platform_driver mt8173_max98090_driver = { .driver = { .name = "mt8173-max98090", .of_match_table = mt8173_max98090_dt_match, -#ifdef CONFIG_PM .pm = &snd_soc_pm_ops, -#endif }, .probe = mt8173_max98090_dev_probe, }; diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 0f28dc2217c0..e6e824f3d24a 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -231,9 +231,7 @@ static struct platform_driver mt8173_rt5650_rt5514_driver = { .driver = { .name = "mtk-rt5650-rt5514", .of_match_table = mt8173_rt5650_rt5514_dt_match, -#ifdef CONFIG_PM .pm = &snd_soc_pm_ops, -#endif }, .probe = mt8173_rt5650_rt5514_dev_probe, }; diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index 077c6ee06780..ba6fe3d90bfc 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -298,9 +298,7 @@ static struct platform_driver mt8173_rt5650_rt5676_driver = { .driver = { .name = "mtk-rt5650-rt5676", .of_match_table = mt8173_rt5650_rt5676_dt_match, -#ifdef CONFIG_PM .pm = &snd_soc_pm_ops, -#endif }, .probe = mt8173_rt5650_rt5676_dev_probe, }; diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index 2cbf679f5c74..9b933cce0b20 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -336,9 +336,7 @@ static struct platform_driver mt8173_rt5650_driver = { .driver = { .name = "mtk-rt5650", .of_match_table = mt8173_rt5650_dt_match, -#ifdef CONFIG_PM .pm = &snd_soc_pm_ops, -#endif }, .probe = mt8173_rt5650_dev_probe, }; diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index 14e77df06b01..86c8a523fe9e 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -1279,9 +1279,7 @@ static struct platform_driver mt8183_afe_pcm_driver = { .driver = { .name = "mt8183-audio", .of_match_table = mt8183_afe_pcm_dt_match, -#ifdef CONFIG_PM .pm = &mt8183_afe_pm_ops, -#endif }, .probe = mt8183_afe_pcm_dev_probe, .remove = mt8183_afe_pcm_dev_remove, diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c index 31c280339c50..e1e4ca931551 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c @@ -2381,9 +2381,7 @@ static struct platform_driver mt8192_afe_pcm_driver = { .driver = { .name = "mt8192-audio", .of_match_table = mt8192_afe_pcm_dt_match, -#ifdef CONFIG_PM .pm = &mt8192_afe_pm_ops, -#endif }, .probe = mt8192_afe_pcm_dev_probe, .remove = mt8192_afe_pcm_dev_remove, -- 2.25.1

3 2

[PATCH] ASoC: dt-bindings: wlf, wm8962: add missing interrupt property
by David Heidelberg 25 Nov '21

25 Nov '21

Both, hardware and drivers does support interrupts. Fix warnings as: arch/arm/boot/dts/tegra30-microsoft-surface-rt-efi.dt.yaml: audio-codec@1a: 'interrupt-parent', 'interrupts' do not match any of the regexes: 'pinctrl-[0-9]+' From schema: /home/runner/work/linux/linux/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml Fixes: cd51b942f344 ("ASoC: dt-bindings: wlf,wm8962: Convert to json-schema") Signed-off-by: David Heidelberg <david(a)ixit.cz> --- Documentation/devicetree/bindings/sound/wlf,wm8962.yaml | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml index 0e6249d7c133..5e172e9462b9 100644 --- a/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml +++ b/Documentation/devicetree/bindings/sound/wlf,wm8962.yaml @@ -19,6 +19,9 @@ properties: clocks: maxItems: 1 + interrupts: + maxItems: 1 + "#sound-dai-cells": const: 0 -- 2.33.0

4 3

[PATCH] ASoC: dt-bindings: davinci-mcasp: convert McASP bindings to yaml schema
by Jayesh Choudhary 25 Nov '21

25 Nov '21

Convert the bindings for McASP controllers for TI SOCs from txt to YAML schema. Adds additional properties 'clocks', 'clock-names', 'power-domains' and '#sound-dai-cells' which were not there in txt file. Adds 'dmas' and 'dma-names' in the example which were not there in txt file. Changes 'interrupts' and 'interrupt-names' from optional to required properties. Signed-off-by: Jayesh Choudhary <j-choudhary(a)ti.com> --- .../bindings/sound/davinci-mcasp-audio.txt | 86 ---------- .../bindings/sound/davinci-mcasp-audio.yaml | 161 ++++++++++++++++++ 2 files changed, 161 insertions(+), 86 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt deleted file mode 100644 index bd863bd69501..000000000000 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ /dev/null @@ -1,86 +0,0 @@ -Texas Instruments McASP controller - -Required properties: -- compatible : - "ti,dm646x-mcasp-audio" : for DM646x platforms - "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms - "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx) - "ti,dra7-mcasp-audio" : for DRA7xx platforms - "ti,omap4-mcasp-audio" : for OMAP4 - -- reg : Should contain reg specifiers for the entries in the reg-names property. -- reg-names : Should contain: - * "mpu" for the main registers (required). For compatibility with - existing software, it is recommended this is the first entry. - * "dat" for separate data port register access (optional). -- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF, - IEC60958-1, and AES-3 formats. -- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted - or received over one serializer. -- serial-dir : A list of serializer configuration. Each entry is a number - indication for serializer pin direction. - (0 - INACTIVE, 1 - TX, 2 - RX) -- dmas: two element list of DMA controller phandles and DMA request line - ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. The dma - identifiers must be "rx" and "tx". - -Optional properties: - -- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0 -- tx-num-evt : FIFO levels. -- rx-num-evt : FIFO levels. -- dismod : Specify the drive on TX pin during inactive slots - 0 : 3-state - 2 : logic low - 3 : logic high - Defaults to 'logic low' when the property is not present -- sram-size-playback : size of sram to be allocated during playback -- sram-size-capture : size of sram to be allocated during capture -- interrupts : Interrupt numbers for McASP -- interrupt-names : Known interrupt names are "tx" and "rx" -- pinctrl-0: Should specify pin control group used for this controller. -- pinctrl-names: Should contain only one value - "default", for more details - please refer to pinctrl-bindings.txt -- fck_parent : Should contain a valid clock name which will be used as parent - for the McASP fck -- auxclk-fs-ratio: When McASP is bus master indicates the ratio between AUCLK - and FS rate if applicable: - AUCLK rate = auxclk-fs-ratio * FS rate - -Optional GPIO support: -If any McASP pin need to be used as GPIO then the McASP node must have: -... - gpio-controller - #gpio-cells = <2>; -... - -When requesting a GPIO, the first parameter is the PIN index in McASP_P* -registers. -For example to request the AXR2 pin of mcasp8: -function-gpios = <&mcasp8 2 0>; - -Or to request the ACLKR pin of mcasp8: -function-gpios = <&mcasp8 29 0>; - -For generic gpio information, please refer to bindings/gpio/gpio.txt - -Example: - -mcasp0: mcasp0@1d00000 { - compatible = "ti,da830-mcasp-audio"; - reg = <0x100000 0x3000>; - reg-names "mpu"; - interrupts = <82>, <83>; - interrupt-names = "tx", "rx"; - op-mode = <0>; /* MCASP_IIS_MODE */ - tdm-slots = <2>; - serial-dir = < - 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ - 0 0 0 0 - 0 0 0 1 - 2 0 0 0 >; - tx-num-evt = <1>; - rx-num-evt = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml new file mode 100644 index 000000000000..ce1bd02decd3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -0,0 +1,161 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/davinci-mcasp-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: McASP Controller for TI SoCs + +maintainers: + - Jayesh Choudhary <j-choudhary(a)ti.com> + +properties: + compatible: + enum: + - ti,dm646x-mcasp-audio + - ti,da830-mcasp-audio + - ti,am33xx-mcasp-audio + - ti,dra7-mcasp-audio + - ti,omap4-mcasp-audio + + reg: + minItems: 1 + items: + - description: main registers + - description: data port register + + reg-names: + minItems: 1 + items: + - const: mpu + - const: dat + + op-mode: + description: I2S - 0 or DIT - 1 mode + enum: + - 0 + - 1 + + tdm-slots: + maxItems: 1 + + serial-dir: + description: + A list of serializer configuration + Entry is indication for serializer pin direction + 0 - Inactive, 1 - TX, 2 - RX + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 16 + items: + minimum: 0 + maximum: 2 + default: 0 + + dmas: + items: + - description: transmission DMA channel + - description: reception DMA channel + + dma-names: + items: + - const: tx + - const: rx + + ti,hwmods: + $ref: /schemas/types.yaml#/definitions/string + description: Name of hwmod associated with McASP + maxItems: 1 + deprecated: true + + tx-num-evt: + maxItems: 1 + + rx-num-evt: + maxItems: 1 + + dismod: + enum: + - 0 + - 2 + - 3 + default: 2 + + sram-size-playback: + maxItems: 1 + + sram-size-capture: + maxItems: 1 + + interrupts: + items: + - description: TX FIFO interrupt + - description: RX FIFO interrupt + + interrupt-names: + items: + - const: tx + - const: rx + + fck_parent: + description: parent clock for McASP fck + maxItems: 1 + + auxclk-fs-ratio: + description: ratio of AUCLK and FS if applicable + maxItems: 1 + + gpio-controller: true + + "#gpio-cells": + const: 2 + + function-gpios: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: fck + + power-domains: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - reg-names + - dmas + - dma-names + - interrupts + - interrupt-names + - serial-dir + - op-mode + - tdm-slots + +additionalProperties: false + +examples: + - | + mcasp0: mcasp0@1d00000 { + compatible = "ti,da830-mcasp-audio"; + reg = <0x100000 0x3000>; + reg-names = "mpu"; + interrupts = <82>, <83>; + interrupt-names = "tx", "rx"; + op-mode = <0>; /* MCASP_IIS_MODE */ + tdm-slots = <2>; + dmas = <&main_udmap 0xc400>, <&main_udmap 0x4400>; + dma-names = "tx", "rx"; + serial-dir = < + 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ + 0 0 0 0 + 0 0 0 1 + 2 0 0 0 >; + tx-num-evt = <1>; + rx-num-evt = <1>; + }; -- 2.17.1

3 3

sound: X86: Xilinx : PCIe : ALSA PCM sample driver code
by Muni Sekhar 25 Nov '21

25 Nov '21

Hi All, I am using a Linux X86 machine where Xilinx FPGA sound card connected to X86 machine's PCIe slot. The sound card supports PCM playback and capture capabilities. I need to write an ALSA driver for my sound card. Since I am using X86 machine and it don't have the devicetree, can someone please suggest the what driver architecture I should use for this type of sound card, also please point me to the ALSA driver reference code for this type of sound card. -- Thanks, Sekhar

1 0

[PATCH 00/11] dmaengine: kill off dma_slave_config->slave_id
by Arnd Bergmann 25 Nov '21

25 Nov '21

From: Arnd Bergmann <arnd(a)arndb.de> I recently came across some new uses of the 'slave_id' field that I had (almost) removed a few years ago. There are no legitimate uses of this field in the kernel, only a few stale references and two drivers that abuse the field as a side-channel between the dmaengine driver and its client. Let's change the xilinx and qualcomm drivers to use the documented side-channel (peripheral_data) instead, and remove the remnants of it to prevent new users from coming in. As the last patch in the series depends on all the others, it would be nice have everything merged into the dmaengine tree for v5.17. Arnd Arnd Bergmann (11): ASoC: dai_dma: remove slave_id field spi: pic32: stop setting dma_config->slave_id mmc: bcm2835: stop setting chan_config->slave_id dmaengine: shdma: remove legacy slave_id parsing dmaengine: pxa/mmp: stop referencing config->slave_id dmaengine: sprd: stop referencing config->slave_id dmaengine: qcom-adm: stop abusing slave_id config dmaengine: xilinx_dpdma: stop using slave_id field dmaengine: tegra20-apb: stop checking config->slave_id staging: ralink-gdma: stop using slave_id config dmaengine: remove slave_id config field drivers/dma/mmp_pdma.c | 6 --- drivers/dma/pxa_dma.c | 7 --- drivers/dma/qcom/qcom_adm.c | 56 ++++++++++++++++++++--- drivers/dma/sh/shdma-base.c | 8 ---- drivers/dma/sprd-dma.c | 3 -- drivers/dma/tegra20-apb-dma.c | 6 --- drivers/dma/xilinx/xilinx_dpdma.c | 12 +++-- drivers/gpu/drm/xlnx/zynqmp_disp.c | 9 +++- drivers/mmc/host/bcm2835.c | 2 - drivers/mtd/nand/raw/qcom_nandc.c | 14 +++++- drivers/spi/spi-pic32.c | 2 - drivers/staging/ralink-gdma/ralink-gdma.c | 12 ++--- drivers/tty/serial/msm_serial.c | 15 +++++- include/linux/dma/qcom_adm.h | 12 +++++ include/linux/dma/xilinx_dpdma.h | 11 +++++ include/linux/dmaengine.h | 4 -- include/sound/dmaengine_pcm.h | 2 - sound/core/pcm_dmaengine.c | 5 +- sound/soc/tegra/tegra20_spdif.c | 1 - 19 files changed, 119 insertions(+), 68 deletions(-) create mode 100644 include/linux/dma/qcom_adm.h create mode 100644 include/linux/dma/xilinx_dpdma.h -- 2.30.2 Cc: Andy Gross <agross(a)kernel.org> Cc: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Baolin Wang <baolin.wang7(a)gmail.com> Cc: Bjorn Andersson <bjorn.andersson(a)linaro.org> Cc: Chunyan Zhang <zhang.lyra(a)gmail.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Hyun Kwon <hyun.kwon(a)xilinx.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Jon Hunter <jonathanh(a)nvidia.com> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> Cc: Laxman Dewangan <ldewangan(a)nvidia.com> Cc: Manivannan Sadhasivam <mani(a)kernel.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Michal Simek <michal.simek(a)xilinx.com> Cc: Nicolas Saenz Julienne <nsaenz(a)kernel.org> Cc: Orson Zhai <orsonzhai(a)gmail.com> Cc: Robert Jarzmik <robert.jarzmik(a)free.fr> Cc: Scott Branden <sbranden(a)broadcom.com> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Thierry Reding <thierry.reding(a)gmail.com> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: bcm-kernel-feedback-list(a)broadcom.com Cc: dmaengine(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-arm-msm(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: linux-mmc(a)vger.kernel.org Cc: linux-mtd(a)lists.infradead.org Cc: linux-rpi-kernel(a)lists.infradead.org Cc: linux-serial(a)vger.kernel.org Cc: linux-spi(a)vger.kernel.org Cc: linux-staging(a)lists.linux.dev Cc: linux-tegra(a)vger.kernel.org

11 35

[PATCH v3 0/3] This patches provide ADSP IPC support for MT8195.
by allen-kh.cheng 25 Nov '21

25 Nov '21

Mediatek ADSP IPC is used to send notification or short message between processors with dsp. It will place the message to the share buffer and will access the ADSP mailbox registers to kick dsp. Two mailboxes used to send notification or short message between processors with dsp Add changes history from series changes since v2: - seperate adsp_mailbox into two instances changes since v1: - fix dt_binding_check error Allen-KH Cheng (3): dt-bindings: mediatek: add adsp-mbox document mailbox: mediatek: add support for adsp mailbox controller firmware: mediatek: add adsp ipc protocol interface .../bindings/mailbox/mtk,adsp-mbox.yaml | 53 +++++ drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 10 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 130 ++++++++++++ drivers/mailbox/Kconfig | 7 + drivers/mailbox/Makefile | 2 + drivers/mailbox/mtk-adsp-mailbox.c | 187 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 72 +++++++ 10 files changed, 465 insertions(+) create mode 100644 Documentation/devicetree/bindings/mailbox/mtk,adsp-mbox.yaml create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 drivers/mailbox/mtk-adsp-mailbox.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

2 10

[PATCH] ALSA: hda/cs8409: Schedule delayed work for jack detect on resume
by Vitaly Rodionov 25 Nov '21

25 Nov '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> CS42L42 runs jack detect on resume, however this requires unsol events, and unsol events are ignored whilst the power state is not set to ON. The power state is set to ON only after the resume finishes. Schedule a delayed work timer to run jack detect after the resume call finishes. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/pci/hda/patch_cs8409.c | 79 +++++++++++++++++++++++++++++------- sound/pci/hda/patch_cs8409.h | 1 + 2 files changed, 65 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 31ff11ab868e..88213e95f0b3 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -634,6 +634,30 @@ static void cs42l42_run_jack_detect(struct sub_codec *cs42l42) cs8409_i2c_write(cs42l42, 0x1120, 0xc0); } +static void cs42l42_run_jack_detect_all(struct hda_codec *codec) +{ + struct cs8409_spec *spec = codec->spec; + struct sub_codec *cs42l42; + int i; + + for (i = 0; i < spec->num_scodecs; i++) { + cs42l42 = spec->scodecs[i]; + cs42l42_enable_jack_detect(cs42l42); + if (!cs42l42->hp_jack_in) + cs42l42_run_jack_detect(cs42l42); + } +} + +/* + * cs42l42_jack_detect_worker - Worker that retries jack detect + */ +static void cs42l42_jack_detect_worker(struct work_struct *work) +{ + struct cs8409_spec *spec = container_of(work, struct cs8409_spec, jack_detect_work.work); + + cs42l42_run_jack_detect_all(spec->codec); +} + static int cs42l42_handle_tip_sense(struct sub_codec *cs42l42, unsigned int reg_ts_status) { int status_changed = 0; @@ -749,8 +773,6 @@ static void cs42l42_resume(struct sub_codec *cs42l42) if (cs42l42->full_scale_vol) cs8409_i2c_write(cs42l42, 0x2001, 0x01); - - cs42l42_enable_jack_detect(cs42l42); } #ifdef CONFIG_PM @@ -800,6 +822,7 @@ static void cs8409_free(struct hda_codec *codec) /* Cancel i2c clock disable timer, and disable clock if left enabled */ cancel_delayed_work_sync(&spec->i2c_clk_work); + cancel_delayed_work_sync(&spec->jack_detect_work); cs8409_disable_i2c_clock(codec); snd_hda_gen_free(codec); @@ -863,6 +886,7 @@ static int cs8409_cs42l42_suspend(struct hda_codec *codec) /* Cancel i2c clock disable timer, and disable clock if left enabled */ cancel_delayed_work_sync(&spec->i2c_clk_work); + cancel_delayed_work_sync(&spec->jack_detect_work); cs8409_disable_i2c_clock(codec); snd_hda_shutup_pins(codec); @@ -970,6 +994,8 @@ void cs8409_cs42l42_fixups(struct hda_codec *codec, const struct hda_fixup *fix, spec->scodecs[CS8409_CODEC0]->codec = codec; codec->patch_ops = cs8409_cs42l42_patch_ops; + INIT_DELAYED_WORK(&spec->jack_detect_work, cs42l42_jack_detect_worker); + spec->gen.suppress_auto_mute = 1; spec->gen.no_primary_hp = 1; spec->gen.suppress_vmaster = 1; @@ -1029,9 +1055,16 @@ void cs8409_cs42l42_fixups(struct hda_codec *codec, const struct hda_fixup *fix, case HDA_FIXUP_ACT_INIT: cs8409_cs42l42_hw_init(codec); spec->init_done = 1; - if (spec->init_done && spec->build_ctrl_done - && !spec->scodecs[CS8409_CODEC0]->hp_jack_in) - cs42l42_run_jack_detect(spec->scodecs[CS8409_CODEC0]); + if (spec->init_done && spec->build_ctrl_done) { + /* No point in running jack detect until we have fully resumed */ + if (codec->core.dev.power.power_state.event != PM_EVENT_ON) { + codec_warn(codec, "Not ready to detect jack, deferring...\n"); + schedule_delayed_work(&spec->jack_detect_work, msecs_to_jiffies(25)); + return; + } else { + cs42l42_run_jack_detect_all(codec); + } + } break; case HDA_FIXUP_ACT_BUILD: spec->build_ctrl_done = 1; @@ -1040,9 +1073,16 @@ void cs8409_cs42l42_fixups(struct hda_codec *codec, const struct hda_fixup *fix, * been already plugged in. * Run immediately after init. */ - if (spec->init_done && spec->build_ctrl_done - && !spec->scodecs[CS8409_CODEC0]->hp_jack_in) - cs42l42_run_jack_detect(spec->scodecs[CS8409_CODEC0]); + if (spec->init_done && spec->build_ctrl_done) { + /* No point in running jack detect until we have fully resumed */ + if (codec->core.dev.power.power_state.event != PM_EVENT_ON) { + codec_warn(codec, "Not ready to detect jack, deferring...\n"); + schedule_delayed_work(&spec->jack_detect_work, msecs_to_jiffies(25)); + return; + } else { + cs42l42_run_jack_detect_all(codec); + } + } break; default: break; @@ -1178,7 +1218,6 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac { struct cs8409_spec *spec = codec->spec; struct snd_kcontrol_new *kctrl; - int i; switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: @@ -1193,6 +1232,8 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac spec->scodecs[CS8409_CODEC1]->codec = codec; spec->num_scodecs = 2; + INIT_DELAYED_WORK(&spec->jack_detect_work, cs42l42_jack_detect_worker); + codec->patch_ops = cs8409_dolphin_patch_ops; /* GPIO 1,5 out, 0,4 in */ @@ -1237,9 +1278,13 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac dolphin_hw_init(codec); spec->init_done = 1; if (spec->init_done && spec->build_ctrl_done) { - for (i = 0; i < spec->num_scodecs; i++) { - if (!spec->scodecs[i]->hp_jack_in) - cs42l42_run_jack_detect(spec->scodecs[i]); + /* No point in running jack detect until we have fully resumed */ + if (codec->core.dev.power.power_state.event != PM_EVENT_ON) { + codec_warn(codec, "Not ready to detect jack, deferring...\n"); + schedule_delayed_work(&spec->jack_detect_work, msecs_to_jiffies(25)); + return; + } else { + cs42l42_run_jack_detect_all(codec); } } break; @@ -1251,9 +1296,13 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac * Run immediately after init. */ if (spec->init_done && spec->build_ctrl_done) { - for (i = 0; i < spec->num_scodecs; i++) { - if (!spec->scodecs[i]->hp_jack_in) - cs42l42_run_jack_detect(spec->scodecs[i]); + /* No point in running jack detect until we have fully resumed */ + if (codec->core.dev.power.power_state.event != PM_EVENT_ON) { + codec_warn(codec, "Not ready to detect jack, deferring...\n"); + schedule_delayed_work(&spec->jack_detect_work, msecs_to_jiffies(25)); + return; + } else { + cs42l42_run_jack_detect_all(codec); } } break; diff --git a/sound/pci/hda/patch_cs8409.h b/sound/pci/hda/patch_cs8409.h index ade2b838590c..632d3ec8322d 100644 --- a/sound/pci/hda/patch_cs8409.h +++ b/sound/pci/hda/patch_cs8409.h @@ -330,6 +330,7 @@ struct cs8409_spec { unsigned int i2c_clck_enabled; unsigned int dev_addr; struct delayed_work i2c_clk_work; + struct delayed_work jack_detect_work; unsigned int playback_started:1; unsigned int capture_started:1; -- 2.25.1

2 1

[PATCH] dt-bindings: sound: nvidia, tegra-audio: Convert multiple txt bindings to yaml
by David Heidelberg 25 Nov '21

25 Nov '21

Convert Tegra audio complex with the * ALC5632 * MAX98090 * RT5640 * RT5677 * SGTL5000 * TrimSlice * WM8753 * WM8903 * WM9712 codec to the YAML format. Signed-off-by: David Heidelberg <david(a)ixit.cz> --- .../sound/nvidia,tegra-audio-alc5632.txt | 48 ----- .../sound/nvidia,tegra-audio-max98090.txt | 53 ----- .../sound/nvidia,tegra-audio-rt5640.txt | 52 ----- .../sound/nvidia,tegra-audio-rt5677.txt | 67 ------ .../sound/nvidia,tegra-audio-sgtl5000.txt | 42 ---- .../sound/nvidia,tegra-audio-trimslice.txt | 21 -- .../sound/nvidia,tegra-audio-wm8753.txt | 40 ---- .../sound/nvidia,tegra-audio-wm8903.txt | 62 ------ .../sound/nvidia,tegra-audio-wm9712.txt | 60 ------ .../bindings/sound/nvidia,tegra-audio.yaml | 190 ++++++++++++++++++ 10 files changed, 190 insertions(+), 445 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt deleted file mode 100644 index 57f40f93453e..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt +++ /dev/null @@ -1,48 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-alc5632" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,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 ALC5632's pins as documented in the binding for the device - and: - - * Headset Stereophone - * Int Spk - * Headset Mic - * Digital Mic - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller -- nvidia,audio-codec : The phandle of the ALC5632 audio codec - -Example: - -sound { - compatible = "nvidia,tegra-audio-alc5632-paz00", - "nvidia,tegra-audio-alc5632"; - - nvidia,model = "Compal PAZ00"; - - nvidia,audio-routing = - "Int Spk", "SPK_OUTP", - "Int Spk", "SPK_OUTN", - "Headset Mic","MICBIAS1", - "MIC1_N", "Headset Mic", - "MIC1_P", "Headset Mic", - "Headset Stereophone", "HP_OUT_R", - "Headset Stereophone", "HP_OUT_L"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&alc5632>; - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt deleted file mode 100644 index c3495beba358..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt +++ /dev/null @@ -1,53 +0,0 @@ -NVIDIA Tegra audio complex, with MAX98090 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-max98090" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,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 MAX98090's pins (as documented in its binding), and the jacks - on the board: - - * Headphones - * Speakers - * Mic Jack - * Int Mic - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the MAX98090 audio codec. - -Optional properties: -- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in -- nvidia,mic-det-gpios : The GPIO that detect microphones are plugged in - -Example: - -sound { - compatible = "nvidia,tegra-audio-max98090-venice2", - "nvidia,tegra-audio-max98090"; - nvidia,model = "NVIDIA Tegra Venice2"; - - nvidia,audio-routing = - "Headphones", "HPR", - "Headphones", "HPL", - "Speakers", "SPKR", - "Speakers", "SPKL", - "Mic Jack", "MICBIAS", - "IN34", "Mic Jack"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&acodec>; - - clocks = <&tegra_car TEGRA124_CLK_PLL_A>, - <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, - <&tegra_car TEGRA124_CLK_EXTERN1>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt deleted file mode 100644 index 7788808dcd0b..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt +++ /dev/null @@ -1,52 +0,0 @@ -NVIDIA Tegra audio complex, with RT5640 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-rt5640" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,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 RT5640's pins (as documented in its binding), and the jacks - on the board: - - * Headphones - * Speakers - * Mic Jack - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the RT5640 audio codec. This binding - assumes that AIF1 on the CODEC is connected to Tegra. - -Optional properties: -- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in - -Example: - -sound { - compatible = "nvidia,tegra-audio-rt5640-dalmore", - "nvidia,tegra-audio-rt5640"; - nvidia,model = "NVIDIA Tegra Dalmore"; - - nvidia,audio-routing = - "Headphones", "HPOR", - "Headphones", "HPOL", - "Speakers", "SPORP", - "Speakers", "SPORN", - "Speakers", "SPOLP", - "Speakers", "SPOLN"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&rt5640>; - - nvidia,hp-det-gpios = <&gpio 143 0>; /* GPIO PR7 */ - - clocks = <&tegra_car 216>, <&tegra_car 217>, <&tegra_car 120>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt deleted file mode 100644 index a4589cda214e..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt +++ /dev/null @@ -1,67 +0,0 @@ -NVIDIA Tegra audio complex, with RT5677 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-rt5677" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,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 RT5677's pins (as documented in its binding), and the jacks - on the board: - - * Headphone - * Speaker - * Headset Mic - * Internal Mic 1 - * Internal Mic 2 - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the RT5677 audio codec. This binding - assumes that AIF1 on the CODEC is connected to Tegra. - -Optional properties: -- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in -- nvidia,hp-en-gpios : The GPIO that enables headphone amplifier -- nvidia,mic-present-gpios: The GPIO that mic jack is plugged in -- nvidia,dmic-clk-en-gpios : The GPIO that gates DMIC clock signal - -Example: - -sound { - compatible = "nvidia,tegra-audio-rt5677-ryu", - "nvidia,tegra-audio-rt5677"; - nvidia,model = "NVIDIA Tegra Ryu"; - - nvidia,audio-routing = - "Headphone", "LOUT2", - "Headphone", "LOUT1", - "Headset Mic", "MICBIAS1", - "IN1P", "Headset Mic", - "IN1N", "Headset Mic", - "DMIC L1", "Internal Mic 1", - "DMIC R1", "Internal Mic 1", - "DMIC L2", "Internal Mic 2", - "DMIC R2", "Internal Mic 2", - "Speaker", "PDM1L", - "Speaker", "PDM1R"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&rt5677>; - - nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>; - nvidia,mic-present-gpios = <&gpio TEGRA_GPIO(O, 5) GPIO_ACTIVE_LOW>; - nvidia,hp-en-gpios = <&rt5677 1 GPIO_ACTIVE_HIGH>; - nvidia,dmic-clk-en-gpios = <&rt5677 2 GPIO_ACTIVE_HIGH>; - - clocks = <&tegra_car TEGRA124_CLK_PLL_A>, - <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, - <&tegra_car TEGRA124_CLK_EXTERN1>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt deleted file mode 100644 index 5da7da4ea07a..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt +++ /dev/null @@ -1,42 +0,0 @@ -NVIDIA Tegra audio complex, with SGTL5000 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-sgtl5000" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,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 SGTL5000's pins (as documented in its binding), and the jacks - on the board: - - * Headphone Jack - * Line In Jack - * Mic Jack - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the SGTL5000 audio codec. - -Example: - -sound { - compatible = "toradex,tegra-audio-sgtl5000-apalis_t30", - "nvidia,tegra-audio-sgtl5000"; - nvidia,model = "Toradex Apalis T30"; - nvidia,audio-routing = - "Headphone Jack", "HP_OUT", - "LINE_IN", "Line In Jack", - "MIC_IN", "Mic Jack"; - nvidia,i2s-controller = <&tegra_i2s2>; - nvidia,audio-codec = <&sgtl5000>; - clocks = <&tegra_car TEGRA30_CLK_PLL_A>, - <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, - <&tegra_car TEGRA30_CLK_EXTERN1>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt deleted file mode 100644 index ef1fe7358279..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt +++ /dev/null @@ -1,21 +0,0 @@ -NVIDIA Tegra audio complex for TrimSlice - -Required properties: -- compatible : "nvidia,tegra-audio-trimslice" -- clocks : Must contain an entry for each entry in clock-names. -- clock-names : Must include the following entries: - "pll_a" (The Tegra clock of that name), - "pll_a_out0" (The Tegra clock of that name), - "mclk" (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller -- nvidia,audio-codec : The phandle of the WM8903 audio codec - -Example: - -sound { - compatible = "nvidia,tegra-audio-trimslice"; - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&codec>; - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt deleted file mode 100644 index 96f6a57dd6b4..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt +++ /dev/null @@ -1,40 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-wm8753" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,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 WM8753's pins as documented in the binding for the WM8753, - and the jacks on the board: - - * Headphone Jack - * Mic Jack - -- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller -- nvidia,audio-codec : The phandle of the WM8753 audio codec -Example: - -sound { - compatible = "nvidia,tegra-audio-wm8753-whistler", - "nvidia,tegra-audio-wm8753" - nvidia,model = "tegra-wm8753-harmony"; - - nvidia,audio-routing = - "Headphone Jack", "LOUT1", - "Headphone Jack", "ROUT1"; - - nvidia,i2s-controller = <&i2s1>; - nvidia,audio-codec = <&wm8753>; - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; - diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt deleted file mode 100644 index bbd581a8c5bc..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt +++ /dev/null @@ -1,62 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-wm8903" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,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 WM8903's pins (documented in the WM8903 binding document), - and the jacks on the board: - - * Headphone Jack - * Int Spk - * Mic Jack - * Int Mic - -- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller -- nvidia,audio-codec : The phandle of the WM8903 audio codec - -Optional properties: -- nvidia,spkr-en-gpios : The GPIO that enables the speakers -- nvidia,hp-mute-gpios : The GPIO that mutes the headphones -- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in -- nvidia,int-mic-en-gpios : The GPIO that enables the internal microphone -- nvidia,ext-mic-en-gpios : The GPIO that enables the external microphone -- nvidia,headset : The Mic Jack represents state of the headset microphone pin - -Example: - -sound { - compatible = "nvidia,tegra-audio-wm8903-harmony", - "nvidia,tegra-audio-wm8903" - nvidia,model = "tegra-wm8903-harmony"; - - nvidia,audio-routing = - "Headphone Jack", "HPOUTR", - "Headphone Jack", "HPOUTL", - "Int Spk", "ROP", - "Int Spk", "RON", - "Int Spk", "LOP", - "Int Spk", "LON", - "Mic Jack", "MICBIAS", - "IN1L", "Mic Jack"; - - nvidia,i2s-controller = <&i2s1>; - nvidia,audio-codec = <&wm8903>; - - nvidia,spkr-en-gpios = <&codec 2 0>; - nvidia,hp-det-gpios = <&gpio 178 0>; /* gpio PW2 */ - nvidia,int-mic-en-gpios = <&gpio 184 0>; /*gpio PX0 */ - nvidia,ext-mic-en-gpios = <&gpio 185 0>; /* gpio PX1 */ - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; - diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt deleted file mode 100644 index 436f6cd9d07c..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt +++ /dev/null @@ -1,60 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-wm9712" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,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 WM9712's pins, and the jacks on the board: - - WM9712 pins: - - * MONOOUT - * HPOUTL - * HPOUTR - * LOUT2 - * ROUT2 - * OUT3 - * LINEINL - * LINEINR - * PHONE - * PCBEEP - * MIC1 - * MIC2 - * Mic Bias - - Board connectors: - - * Headphone - * LineIn - * Mic - -- nvidia,ac97-controller : The phandle of the Tegra AC97 controller - - -Example: - -sound { - compatible = "nvidia,tegra-audio-wm9712-colibri_t20", - "nvidia,tegra-audio-wm9712"; - nvidia,model = "Toradex Colibri T20"; - - nvidia,audio-routing = - "Headphone", "HPOUTL", - "Headphone", "HPOUTR", - "LineIn", "LINEINL", - "LineIn", "LINEINR", - "Mic", "MIC1"; - - nvidia,ac97-controller = <&ac97>; - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio.yaml new file mode 100644 index 000000000000..6b8876bfa40f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio.yaml @@ -0,0 +1,190 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/nvidia,tegra-audio.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: NVIDIA Tegra audio complex + +maintainers: + - Jon Hunter <jonathanh(a)nvidia.com> + - Thierry Reding <thierry.reding(a)gmail.com> + +properties: + compatible: + oneOf: + - items: + - pattern: '^[a-z0-9]+,tegra-audio-alc5632(-[a-z0-9])+' + - const: nvidia,tegra-audio-alc5632 + - items: + - enum: + - nvidia,tegra-audio-max98090-nyan-big + - nvidia,tegra-audio-max98090-nyan-blaze + - const: nvidia,tegra-audio-max98090-nyan + - const: nvidia,tegra-audio-max98090 + - items: + - pattern: '^[a-z0-9]+,tegra-audio-max98090(-[a-z0-9])+' + - const: nvidia,tegra-audio-max98090 + - items: + - pattern: '^[a-z0-9]+,tegra-audio-rt56(39|40)(-[a-z0-9])+' + - const: nvidia,tegra-audio-rt5640 + - items: + - pattern: '^[a-z0-9]+,tegra-audio-rt5677(-[a-z0-9])+' + - const: nvidia,tegra-audio-rt5677 + - items: + - enum: + - toradex,tegra-audio-sgtl5000-apalis_t30 + - toradex,tegra-audio-sgtl5000-colibri_t30 + - toradex,tegra-audio-sgtl5000-apalis_tk1 + - const: nvidia,tegra-audio-sgtl5000 + - const: nvidia,tegra-audio-trimslice + - items: + - pattern: '^[a-z0-9]+,tegra-audio-wm8753(-[a-z0-9])+' + - const: nvidia,tegra-audio-wm8753 + - items: + - pattern: '^[a-z0-9]+,tegra-audio-(plutux|wm8903(-[a-z0-9])+)' + - const: nvidia,tegra-audio-wm8903 + - items: + - pattern: '^[a-z0-9]+,tegra-audio-wm9712(-[a-z0-9])+' + - const: nvidia,tegra-audio-wm9712 + + clocks: + items: + - description: PLL A clock + - description: PLL A OUT0 clock + - description: The Tegra cdev1/extern1 clock, which feeds the card's mclk + + clock-names: + items: + - const: pll_a + - const: pll_a_out0 + - const: mclk + + assigned-clocks: true + + assigned-clock-parents: true + + assigned-clock-rates: true + + nvidia,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + 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. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board: + ALC5632: + * Headset Stereophone + * Int Spk + * Headset Mic + * Digital Mic + MAXX98090: + * Headphones + * SPeakers + * Mic Jack + * Int Mic + RT5640: + * Headphones + * Speakers + * Mic Jack + RT5677: + * Headphone + * Speaker + * Headset Mic + * Internal Mic 1 + * Internal Mic 2 + SGTL5000: + * Headphone Jack + * Line In Jack + * Mic Jack + WM8753: + * Headphone Jack + * Mic Jack + WM8903: + * Headphone Jack + * Int Spk + * Mic Jack + * Int Mic + WM9712: + * Headphone + * LineIn + * Mic + + nvidia,ac97-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the AC97 controller + + nvidia,i2s-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the Tegra I2S controller + + nvidia,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of audio codec + + nvidia,spkr-en-gpios: + maxItems: 1 + description: The GPIO that enables the speakers + + nvidia,hp-mute-gpios: + maxItems: 1 + description: The GPIO that mutes the headphones + + nvidia,hp-det-gpios: + maxItems: 1 + description: The GPIO that detect headphones are plugged in + + nvidia,mic-det-gpios: + maxItems: 1 + description: The GPIO that detect microphone is plugged in + + nvidia,int-mic-en-gpios: + maxItems: 1 + description: The GPIO that enables the internal microphone + + nvidia,ext-mic-en-gpios: + maxItems: 1 + description: The GPIO that enables the external microphone + + nvidia,headset: + type: boolean + description: The Mic Jack represents state of the headset microphone pin + +additionalProperties: false + +required: + - compatible + - clocks + - clock-names + +oneOf: + - required: + - nvidia,audio-codec + - nvidia,i2s-controller + - required: + - nvidia,ac97-controller + +examples: + - | + #include <dt-bindings/clock/tegra30-car.h> + + sound { + compatible = "toradex,tegra-audio-sgtl5000-apalis_t30", + "nvidia,tegra-audio-sgtl5000"; + nvidia,model = "Toradex Apalis T30 SGTL5000"; + nvidia,audio-routing = + "Headphone Jack", "HP_OUT", + "LINE_IN", "Line In Jack", + "MIC_IN", "Mic Jack"; + nvidia,i2s-controller = <&tegra_i2s2>; + nvidia,audio-codec = <&codec>; + clocks = <&tegra_car TEGRA30_CLK_PLL_A>, + <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA30_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; -- 2.33.0

3 5

[PATCH v2] ASoC: rockchip: i2s_tdm: Dup static DAI template
by Nicolas Frattaroli 25 Nov '21

25 Nov '21

Previously, the DAI template was used directly, which lead to fun bugs such as "why is my channels_max changing?" when one instantiated more than one i2s_tdm IP block in a device tree. This change makes it so that we instead duplicate the template struct, and then use that. Fixes: 081068fd6414 ("ASoC: rockchip: add support for i2s-tdm controller") Signed-off-by: Nicolas Frattaroli <frattaroli.nicolas(a)gmail.com> --- sound/soc/rockchip/rockchip_i2s_tdm.c | 38 ++++++++++++++++----------- 1 file changed, 22 insertions(+), 16 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index 17b9b287853a..4328c10ea830 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -1312,17 +1312,12 @@ static const struct of_device_id rockchip_i2s_tdm_match[] = { static struct snd_soc_dai_driver i2s_tdm_dai = { .probe = rockchip_i2s_tdm_dai_probe, - .playback = { - .stream_name = "Playback", - }, - .capture = { - .stream_name = "Capture", - }, .ops = &rockchip_i2s_tdm_dai_ops, }; -static void rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) +static int rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) { + struct snd_soc_dai_driver *dai; struct property *dma_names; const char *dma_name; u64 formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | @@ -1337,19 +1332,28 @@ static void rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) i2s_tdm->has_capture = true; } + dai = devm_kmemdup(i2s_tdm->dev, &i2s_tdm_dai, + sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + if (i2s_tdm->has_playback) { - i2s_tdm_dai.playback.channels_min = 2; - i2s_tdm_dai.playback.channels_max = 8; - i2s_tdm_dai.playback.rates = SNDRV_PCM_RATE_8000_192000; - i2s_tdm_dai.playback.formats = formats; + dai->playback.stream_name = "Playback"; + dai->playback.channels_min = 2; + dai->playback.channels_max = 8; + dai->playback.rates = SNDRV_PCM_RATE_8000_192000; + dai->playback.formats = formats; } if (i2s_tdm->has_capture) { - i2s_tdm_dai.capture.channels_min = 2; - i2s_tdm_dai.capture.channels_max = 8; - i2s_tdm_dai.capture.rates = SNDRV_PCM_RATE_8000_192000; - i2s_tdm_dai.capture.formats = formats; + dai->capture.stream_name = "Capture"; + dai->capture.channels_min = 2; + dai->capture.channels_max = 8; + dai->capture.rates = SNDRV_PCM_RATE_8000_192000; + dai->capture.formats = formats; } + + return 0; } static int rockchip_i2s_tdm_path_check(struct rk_i2s_tdm_dev *i2s_tdm, @@ -1541,7 +1545,9 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev) spin_lock_init(&i2s_tdm->lock); i2s_tdm->soc_data = (struct rk_i2s_soc_data *)of_id->data; - rockchip_i2s_tdm_init_dai(i2s_tdm); + ret = rockchip_i2s_tdm_init_dai(i2s_tdm); + if (ret) + return ret; i2s_tdm->frame_width = 64; -- 2.34.0

1 1

[PATCH v2 1/3] ASoC: SOF: sof-audio: setup sched widgets during pipeline complete step
by Kai Vehmanen 24 Nov '21

24 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Older firmware prior to ABI 3.19 has a dependency where the scheduler widgets need to be setup last. Moving the call to sof_widget_setup() before the pipeline_complete() call also helps remove the need for the 'reverse' direction when walking through the widget list - this was only working because of the topology macros but the topology does not require any order. Fixes: 5fcdbb2d45df ("ASoC: SOF: Add support for dynamic pipelines") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/sof-audio.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 0f2566f7c094..f4e142ec0fbd 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -637,16 +637,25 @@ const struct sof_ipc_pipe_new *snd_sof_pipeline_find(struct snd_sof_dev *sdev, int sof_set_up_pipelines(struct snd_sof_dev *sdev, bool verify) { + struct sof_ipc_fw_version *v = &sdev->fw_ready.version; struct snd_sof_widget *swidget; struct snd_sof_route *sroute; int ret; /* restore pipeline components */ - list_for_each_entry_reverse(swidget, &sdev->widget_list, list) { + list_for_each_entry(swidget, &sdev->widget_list, list) { /* only set up the widgets belonging to static pipelines */ if (!verify && swidget->dynamic_pipeline_widget) continue; + /* + * For older firmware, skip scheduler widgets in this loop, + * sof_widget_setup() will be called in the 'complete pipeline' loop + */ + if (v->abi_version < SOF_ABI_VER(3, 19, 0) && + swidget->id == snd_soc_dapm_scheduler) + continue; + /* update DAI config. The IPC will be sent in sof_widget_setup() */ if (WIDGET_IS_DAI(swidget->id)) { struct snd_sof_dai *dai = swidget->private; @@ -694,6 +703,12 @@ int sof_set_up_pipelines(struct snd_sof_dev *sdev, bool verify) if (!verify && swidget->dynamic_pipeline_widget) continue; + if (v->abi_version < SOF_ABI_VER(3, 19, 0)) { + ret = sof_widget_setup(sdev, swidget); + if (ret < 0) + return ret; + } + swidget->complete = snd_sof_complete_pipeline(sdev, swidget); break; @@ -722,7 +737,7 @@ int sof_tear_down_pipelines(struct snd_sof_dev *sdev, bool verify) * sroute->setup because during suspend all streams are suspended and during topology * loading the sound card unavailable to open PCMs. */ - list_for_each_entry_reverse(swidget, &sdev->widget_list, list) { + list_for_each_entry(swidget, &sdev->widget_list, list) { if (swidget->dynamic_pipeline_widget) continue; base-commit: 65c16dd2942f4476a3f96a2625b1475c6137c09a -- 2.33.0

2 6

[PATCH] ASoC: SOF: handle paused streams during system suspend
by Kai Vehmanen 24 Nov '21

24 Nov '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> During system suspend, paused streams do not get suspended. Therefore, we need to explicitly free these PCMs in the DSP and free the associated DAPM widgets so that they can be set up again during resume. Fixes: 5fcdbb2d45df ("ASoC: SOF: Add support for dynamic pipelines") Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/pcm.c | 5 ++- sound/soc/sof/sof-audio.c | 74 +++++++++++++++++++++++++++++++++++++-- sound/soc/sof/sof-audio.h | 2 ++ 3 files changed, 76 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 31dd79b794f1..0ceb1a9cbf73 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -100,9 +100,8 @@ void snd_sof_pcm_period_elapsed(struct snd_pcm_substream *substream) } EXPORT_SYMBOL(snd_sof_pcm_period_elapsed); -static int sof_pcm_dsp_pcm_free(struct snd_pcm_substream *substream, - struct snd_sof_dev *sdev, - struct snd_sof_pcm *spcm) +int sof_pcm_dsp_pcm_free(struct snd_pcm_substream *substream, struct snd_sof_dev *sdev, + struct snd_sof_pcm *spcm) { struct sof_ipc_stream stream; struct sof_ipc_reply reply; diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 0f2566f7c094..038cdb4a9aa2 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -129,6 +129,14 @@ int sof_widget_free(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget) case snd_soc_dapm_buffer: ipc_free.hdr.cmd |= SOF_IPC_TPLG_BUFFER_FREE; break; + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + { + struct snd_sof_dai *dai = swidget->private; + + dai->configured = false; + fallthrough; + } default: ipc_free.hdr.cmd |= SOF_IPC_TPLG_COMP_FREE; break; @@ -705,6 +713,55 @@ int sof_set_up_pipelines(struct snd_sof_dev *sdev, bool verify) return 0; } +/* + * Free the PCM, its associated widgets and set the prepared flag to false for all PCMs that + * did not get suspended(ex: paused streams) so the widgets can be set up again during resume. + */ +static int sof_tear_down_left_over_pipelines(struct snd_sof_dev *sdev) +{ + struct snd_sof_widget *swidget; + struct snd_sof_pcm *spcm; + int dir, ret; + + /* + * free all PCMs and their associated DAPM widgets if their connected DAPM widget + * list is not NULL. This should only be true for paused streams at this point. + * This is equivalent to the handling of FE DAI suspend trigger for running streams. + */ + list_for_each_entry(spcm, &sdev->pcm_list, list) + for_each_pcm_streams(dir) { + struct snd_pcm_substream *substream = spcm->stream[dir].substream; + + if (!substream || !substream->runtime) + continue; + + if (spcm->stream[dir].list) { + ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm); + if (ret < 0) + return ret; + + ret = sof_widget_list_free(sdev, spcm, dir); + if (ret < 0) { + dev_err(sdev->dev, "failed to free widgets during suspend\n"); + return ret; + } + } + } + + /* + * free any left over DAI widgets. This is equivalent to the handling of suspend trigger + * for the BE DAI for running streams. + */ + list_for_each_entry(swidget, &sdev->widget_list, list) + if (WIDGET_IS_DAI(swidget->id) && swidget->use_count == 1) { + ret = sof_widget_free(sdev, swidget); + if (ret < 0) + return ret; + } + + return 0; +} + /* * For older firmware, this function doesn't free widgets for static pipelines during suspend. * It only resets use_count for all widgets. @@ -719,8 +776,8 @@ int sof_tear_down_pipelines(struct snd_sof_dev *sdev, bool verify) /* * This function is called during suspend and for one-time topology verification during * first boot. In both cases, there is no need to protect swidget->use_count and - * sroute->setup because during suspend all streams are suspended and during topology - * loading the sound card unavailable to open PCMs. + * sroute->setup because during suspend all running streams are suspended and during + * topology loading the sound card unavailable to open PCMs. */ list_for_each_entry_reverse(swidget, &sdev->widget_list, list) { if (swidget->dynamic_pipeline_widget) @@ -739,6 +796,19 @@ int sof_tear_down_pipelines(struct snd_sof_dev *sdev, bool verify) return ret; } + /* + * Tear down all pipelines associated with PCMs that did not get suspended + * and unset the prepare flag so that they can be set up again during resume. + * Skip this step for older firmware. + */ + if (!verify && v->abi_version >= SOF_ABI_VER(3, 19, 0)) { + ret = sof_tear_down_left_over_pipelines(sdev); + if (ret < 0) { + dev_err(sdev->dev, "failed to tear down paused pipelines\n"); + return ret; + } + } + list_for_each_entry(sroute, &sdev->route_list, list) sroute->setup = false; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 389d56ac3aba..1c4f59d34717 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -265,4 +265,6 @@ int sof_widget_free(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget); /* PCM */ int sof_widget_list_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm, int dir); int sof_widget_list_free(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm, int dir); +int sof_pcm_dsp_pcm_free(struct snd_pcm_substream *substream, struct snd_sof_dev *sdev, + struct snd_sof_pcm *spcm); #endif base-commit: 65c16dd2942f4476a3f96a2625b1475c6137c09a -- 2.33.0

2 1

[PATCH 1/2] ASoC: SOF: sof-audio: setup sched widgets during pipeline complete step
by Kai Vehmanen 24 Nov '21

24 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Older firmware prior to ABI 3.19 has a dependency where the scheduler widgets need to be setup last. Moving the call to sof_widget_setup() before the pipeline_complete() call also helps remove the need for the 'reverse' direction when walking through the widget list - this was only working because of the topology macros but the topology does not require any order. Fixes: 5fcdbb2d45df ("ASoC: SOF: Add support for dynamic pipelines") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/sof-audio.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 0f2566f7c094..f4e142ec0fbd 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -637,16 +637,25 @@ const struct sof_ipc_pipe_new *snd_sof_pipeline_find(struct snd_sof_dev *sdev, int sof_set_up_pipelines(struct snd_sof_dev *sdev, bool verify) { + struct sof_ipc_fw_version *v = &sdev->fw_ready.version; struct snd_sof_widget *swidget; struct snd_sof_route *sroute; int ret; /* restore pipeline components */ - list_for_each_entry_reverse(swidget, &sdev->widget_list, list) { + list_for_each_entry(swidget, &sdev->widget_list, list) { /* only set up the widgets belonging to static pipelines */ if (!verify && swidget->dynamic_pipeline_widget) continue; + /* + * For older firmware, skip scheduler widgets in this loop, + * sof_widget_setup() will be called in the 'complete pipeline' loop + */ + if (v->abi_version < SOF_ABI_VER(3, 19, 0) && + swidget->id == snd_soc_dapm_scheduler) + continue; + /* update DAI config. The IPC will be sent in sof_widget_setup() */ if (WIDGET_IS_DAI(swidget->id)) { struct snd_sof_dai *dai = swidget->private; @@ -694,6 +703,12 @@ int sof_set_up_pipelines(struct snd_sof_dev *sdev, bool verify) if (!verify && swidget->dynamic_pipeline_widget) continue; + if (v->abi_version < SOF_ABI_VER(3, 19, 0)) { + ret = sof_widget_setup(sdev, swidget); + if (ret < 0) + return ret; + } + swidget->complete = snd_sof_complete_pipeline(sdev, swidget); break; @@ -722,7 +737,7 @@ int sof_tear_down_pipelines(struct snd_sof_dev *sdev, bool verify) * sroute->setup because during suspend all streams are suspended and during topology * loading the sound card unavailable to open PCMs. */ - list_for_each_entry_reverse(swidget, &sdev->widget_list, list) { + list_for_each_entry(swidget, &sdev->widget_list, list) { if (swidget->dynamic_pipeline_widget) continue; base-commit: 65c16dd2942f4476a3f96a2625b1475c6137c09a -- 2.33.0

2 2

[PATCH v4 0/4] ASoC: SOF: Intel: power optimizations with HDaudio SPIB register
by Pierre-Louis Bossart 24 Nov '21

24 Nov '21

This patchset was initially provided in a larger series that was split in two [1]. This part only provides support for the SPIB register support, added on Intel platforms since Skylake (2015). The use of the SPIB register helps reduce power consumption - though to a smaller degree than DMI_L1. This hardware capability is however incompatible with userspace-initiated rewinds typically used by PulseAudio. In the past (2015..2017) Intel suggested an API extension to let applications disable rewinds. At the time the feedback was that such a capability was too Intel-specific and SPIB remained unused except for loading DSP code. We now see devices with smaller batteries being released, and it's time to revisit Linux support for SPIB to extend battery life. In this update the rewinds are disabled via an opt-in kernel parameter. In the previous reviews, there was consensus that a Kconfig option was too complicated for distributions to set, and we are missing a TBD API to expose such capabilities to user-space. The debate on whether or not to use rewinds, and the impact of disabling rewinds, will likely be closed when Intel releases the 'deep-buffer' support, currently under development [2][3]. With this solution, rewinds will not be needed, ever. When an application deals with content that is not latency-sensitive (e.g. music playback), it will be able to reduce power consumption by selecting a different PCM device with increased buffering capabilities. Low-latency streams will be handled by the 'regular' path. In other words, the impossible compromise between power and latency will be handled with different PCM devices/profiles for the same endpoint, and we can push the design of capability negotiation to a later time when all the building blocks (firmware topology, kernel, userspace) are ready - we still have firmware xruns, DPCM race conditions to solve, and a need to describe these alternate PCM devices with UCM using 'modifiers'. [1] https://lore.kernel.org/alsa-devel/20210610205326.1176400-1-pierre-louis.bo… [2] https://github.com/thesofproject/linux/pull/3146 [3] https://github.com/thesofproject/sof/pull/4611 Changes since v3: Added Takashi's Reviewed-by tags on patches 1 and 2 Used SYNC_APPLPTR instead of EXPLICIT_SYNC as suggested by Takashi Added a missing .ack callback for IceLake Changes since v2: Added check on appl_ptr (Takashi) Moved rewind detection to deal with SYNC_PTR (Takashi) Pierre-Louis Bossart (2): ALSA: pcm: unconditionally check if appl_ptr is in 0..boundary range ALSA: pcm: introduce INFO_NO_REWINDS flag Ranjani Sridharan (2): ASoC: SOF: pcm: add .ack callback support ASoC: SOF: Intel: add .ack support for HDaudio platforms include/uapi/sound/asound.h | 2 +- sound/core/pcm_lib.c | 17 +++++++++++++ sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 1 + sound/soc/sof/intel/hda-pcm.c | 41 ++++++++++++++++++++++++++++++-- sound/soc/sof/intel/hda-stream.c | 2 ++ sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/icl.c | 1 + sound/soc/sof/intel/tgl.c | 1 + sound/soc/sof/ops.h | 10 ++++++++ sound/soc/sof/pcm.c | 9 +++++++ sound/soc/sof/sof-priv.h | 3 +++ 12 files changed, 86 insertions(+), 3 deletions(-) -- 2.25.1

2 5

[PATCH] ASoC: SOF: hda: reset DAI widget before reconfiguring it
by Kai Vehmanen 24 Nov '21

24 Nov '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> It is not unusual for ALSA/ASoC hw_params callbacks to be invoked multiple times. Reset and free the DAI widget before reconfiguring it to keep the DAI widget use_count balanced. Fixes: 0acb48dd31e3 ("ASoC: SOF: Intel: hda: make sure DAI widget is set up before IPC") Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/intel/hda.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 1e1e9659ea86..54dd18dce8d0 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -58,6 +58,13 @@ int hda_ctrl_dai_widget_setup(struct snd_soc_dapm_widget *w) return -EINVAL; } + /* DAI already configured, reset it before reconfiguring it */ + if (sof_dai->configured) { + ret = hda_ctrl_dai_widget_free(w); + if (ret < 0) + return ret; + } + config = &sof_dai->dai_config[sof_dai->current_config]; /* base-commit: 65c16dd2942f4476a3f96a2625b1475c6137c09a -- 2.33.0

2 1

[PATCH] ASoC: Intel: soc-acpi: add entry for ESSX8336 on CML
by Pierre-Louis Bossart 24 Nov '21

24 Nov '21

We have configurations for this codec on APL, GLK, JSL and TGL, somehow the information that some designs rely on CometLake was not shared. BugLink: https://github.com/thesofproject/linux/issues/3248 Fixes: 790049fb6623 ("ASoC: Intel: soc-acpi: apl/glk/tgl: add entry for devices based on ES8336 codec") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index b4eb0c97edf1..4eebc79d4b48 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -81,6 +81,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-da7219-max98390.tplg", }, + { + .id = "ESSX8336", + .drv_name = "sof-essx8336", + .sof_fw_filename = "sof-cml.ri", + .sof_tplg_filename = "sof-cml-es8336.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines); -- 2.25.1

2 1

[PATCH 4.19 271/323] ALSA: ISA: not for M68K
by Greg Kroah-Hartman 24 Nov '21

24 Nov '21

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 3c05f1477e62ea5a0a8797ba6a545b1dc751fb31 ] On m68k, compiling drivers under SND_ISA causes build errors: ../sound/core/isadma.c: In function 'snd_dma_program': ../sound/core/isadma.c:33:17: error: implicit declaration of function 'claim_dma_lock' [-Werror=implicit-function-declaration] 33 | flags = claim_dma_lock(); | ^~~~~~~~~~~~~~ ../sound/core/isadma.c:41:9: error: implicit declaration of function 'release_dma_lock' [-Werror=implicit-function-declaration] 41 | release_dma_lock(flags); | ^~~~~~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_playback_prepare': ../sound/isa/sb/sb16_main.c:253:72: error: 'DMA_AUTOINIT' undeclared (first use in this function) 253 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); | ^~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c:253:72: note: each undeclared identifier is reported only once for each function it appears in ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_capture_prepare': ../sound/isa/sb/sb16_main.c:322:71: error: 'DMA_AUTOINIT' undeclared (first use in this function) 322 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); | ^~~~~~~~~~~~ and more... Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-m68k(a)lists.linux-m68k.org Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Link: https://lore.kernel.org/r/20211016062602.3588-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/Makefile | 2 ++ sound/isa/Kconfig | 2 +- sound/pci/Kconfig | 1 + 3 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/Makefile b/sound/core/Makefile index ee4a4a6b99ba7..d123587c0fd8f 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -9,7 +9,9 @@ ifneq ($(CONFIG_SND_PROC_FS),) snd-y += info.o snd-$(CONFIG_SND_OSSEMUL) += info_oss.o endif +ifneq ($(CONFIG_M68K),y) snd-$(CONFIG_ISA_DMA_API) += isadma.o +endif snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index d7db1eeebc844..f8f3433925bb4 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -21,7 +21,7 @@ config SND_SB16_DSP menuconfig SND_ISA bool "ISA sound devices" depends on ISA || COMPILE_TEST - depends on ISA_DMA_API + depends on ISA_DMA_API && !M68K default y help Support for sound devices connected via the ISA bus. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 4105d9f653d90..bbaf46dc3f804 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -278,6 +278,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" depends on ISA_DMA_API && (X86_32 || COMPILE_TEST) + depends on !M68K select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. -- 2.33.0

1 0

[PATCH 5.4 021/100] ALSA: ISA: not for M68K
by Greg Kroah-Hartman 24 Nov '21

24 Nov '21

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 3c05f1477e62ea5a0a8797ba6a545b1dc751fb31 ] On m68k, compiling drivers under SND_ISA causes build errors: ../sound/core/isadma.c: In function 'snd_dma_program': ../sound/core/isadma.c:33:17: error: implicit declaration of function 'claim_dma_lock' [-Werror=implicit-function-declaration] 33 | flags = claim_dma_lock(); | ^~~~~~~~~~~~~~ ../sound/core/isadma.c:41:9: error: implicit declaration of function 'release_dma_lock' [-Werror=implicit-function-declaration] 41 | release_dma_lock(flags); | ^~~~~~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_playback_prepare': ../sound/isa/sb/sb16_main.c:253:72: error: 'DMA_AUTOINIT' undeclared (first use in this function) 253 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); | ^~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c:253:72: note: each undeclared identifier is reported only once for each function it appears in ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_capture_prepare': ../sound/isa/sb/sb16_main.c:322:71: error: 'DMA_AUTOINIT' undeclared (first use in this function) 322 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); | ^~~~~~~~~~~~ and more... Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-m68k(a)lists.linux-m68k.org Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Link: https://lore.kernel.org/r/20211016062602.3588-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/Makefile | 2 ++ sound/isa/Kconfig | 2 +- sound/pci/Kconfig | 1 + 3 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/Makefile b/sound/core/Makefile index ee4a4a6b99ba7..d123587c0fd8f 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -9,7 +9,9 @@ ifneq ($(CONFIG_SND_PROC_FS),) snd-y += info.o snd-$(CONFIG_SND_OSSEMUL) += info_oss.o endif +ifneq ($(CONFIG_M68K),y) snd-$(CONFIG_ISA_DMA_API) += isadma.o +endif snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b690ed937cbe8..df2e45c8814e9 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -22,7 +22,7 @@ config SND_SB16_DSP menuconfig SND_ISA bool "ISA sound devices" depends on ISA || COMPILE_TEST - depends on ISA_DMA_API + depends on ISA_DMA_API && !M68K default y help Support for sound devices connected via the ISA bus. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 7630f808d087c..6edde2f145025 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -279,6 +279,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" depends on ISA_DMA_API && (X86_32 || COMPILE_TEST) + depends on !M68K select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. -- 2.33.0

1 0

[PATCH 5.10 035/154] ALSA: ISA: not for M68K
by Greg Kroah-Hartman 24 Nov '21

24 Nov '21

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 3c05f1477e62ea5a0a8797ba6a545b1dc751fb31 ] On m68k, compiling drivers under SND_ISA causes build errors: ../sound/core/isadma.c: In function 'snd_dma_program': ../sound/core/isadma.c:33:17: error: implicit declaration of function 'claim_dma_lock' [-Werror=implicit-function-declaration] 33 | flags = claim_dma_lock(); | ^~~~~~~~~~~~~~ ../sound/core/isadma.c:41:9: error: implicit declaration of function 'release_dma_lock' [-Werror=implicit-function-declaration] 41 | release_dma_lock(flags); | ^~~~~~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_playback_prepare': ../sound/isa/sb/sb16_main.c:253:72: error: 'DMA_AUTOINIT' undeclared (first use in this function) 253 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); | ^~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c:253:72: note: each undeclared identifier is reported only once for each function it appears in ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_capture_prepare': ../sound/isa/sb/sb16_main.c:322:71: error: 'DMA_AUTOINIT' undeclared (first use in this function) 322 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); | ^~~~~~~~~~~~ and more... Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-m68k(a)lists.linux-m68k.org Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Link: https://lore.kernel.org/r/20211016062602.3588-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/Makefile | 2 ++ sound/isa/Kconfig | 2 +- sound/pci/Kconfig | 1 + 3 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/Makefile b/sound/core/Makefile index ee4a4a6b99ba7..d123587c0fd8f 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -9,7 +9,9 @@ ifneq ($(CONFIG_SND_PROC_FS),) snd-y += info.o snd-$(CONFIG_SND_OSSEMUL) += info_oss.o endif +ifneq ($(CONFIG_M68K),y) snd-$(CONFIG_ISA_DMA_API) += isadma.o +endif snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 6ffa48dd59830..570b88e0b2018 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -22,7 +22,7 @@ config SND_SB16_DSP menuconfig SND_ISA bool "ISA sound devices" depends on ISA || COMPILE_TEST - depends on ISA_DMA_API + depends on ISA_DMA_API && !M68K default y help Support for sound devices connected via the ISA bus. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 93bc9bef7641f..41ce125971777 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -279,6 +279,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" depends on ISA_DMA_API && (X86_32 || COMPILE_TEST) + depends on !M68K select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. -- 2.33.0

1 0

[PATCH 5.15 057/279] ALSA: ISA: not for M68K
by Greg Kroah-Hartman 24 Nov '21

24 Nov '21

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 3c05f1477e62ea5a0a8797ba6a545b1dc751fb31 ] On m68k, compiling drivers under SND_ISA causes build errors: ../sound/core/isadma.c: In function 'snd_dma_program': ../sound/core/isadma.c:33:17: error: implicit declaration of function 'claim_dma_lock' [-Werror=implicit-function-declaration] 33 | flags = claim_dma_lock(); | ^~~~~~~~~~~~~~ ../sound/core/isadma.c:41:9: error: implicit declaration of function 'release_dma_lock' [-Werror=implicit-function-declaration] 41 | release_dma_lock(flags); | ^~~~~~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_playback_prepare': ../sound/isa/sb/sb16_main.c:253:72: error: 'DMA_AUTOINIT' undeclared (first use in this function) 253 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); | ^~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c:253:72: note: each undeclared identifier is reported only once for each function it appears in ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_capture_prepare': ../sound/isa/sb/sb16_main.c:322:71: error: 'DMA_AUTOINIT' undeclared (first use in this function) 322 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); | ^~~~~~~~~~~~ and more... Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-m68k(a)lists.linux-m68k.org Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Link: https://lore.kernel.org/r/20211016062602.3588-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/Makefile | 2 ++ sound/isa/Kconfig | 2 +- sound/pci/Kconfig | 1 + 3 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/Makefile b/sound/core/Makefile index d774792850f31..79e1407cd0de7 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -9,7 +9,9 @@ ifneq ($(CONFIG_SND_PROC_FS),) snd-y += info.o snd-$(CONFIG_SND_OSSEMUL) += info_oss.o endif +ifneq ($(CONFIG_M68K),y) snd-$(CONFIG_ISA_DMA_API) += isadma.o +endif snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 6ffa48dd59830..570b88e0b2018 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -22,7 +22,7 @@ config SND_SB16_DSP menuconfig SND_ISA bool "ISA sound devices" depends on ISA || COMPILE_TEST - depends on ISA_DMA_API + depends on ISA_DMA_API && !M68K default y help Support for sound devices connected via the ISA bus. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 93bc9bef7641f..41ce125971777 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -279,6 +279,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" depends on ISA_DMA_API && (X86_32 || COMPILE_TEST) + depends on !M68K select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. -- 2.33.0

1 0

Creating UCM2 profiles for devices to split multichannel streams
by GitHub issues - opened 24 Nov '21

24 Nov '21

alsa-project/alsa-ucm-conf issue #122 was opened from brndd: Hey. Apologies for this question-issue, but I wasn't sure where to ask and figured it was better to do it here in public (so that others may benefit from the discussion) than ask by email or something. I randomly stumbled upon #121 which is about creating a UCM2 profile for the TC-Helicon GoXLR device to split some multichannel streams to better match the physical layout of the device. I had no idea this was possible with UCM2 (in fact I looked at UCM2 for this a few months ago and got the impression that it was *not* possible), and would be interested in creating a similar profile for my device, a Behringer UMC204HD. This device has a similar problem of reporting a stereo input stream when the physical layout is two mono inputs, and a 4-channel output stream when the physical layout is two stereo outputs. Thus far I've been using PulseAudio/PipeWire remappings to get around the issue, but a UCM2 profile would be a far superior solution, if indeed it is possible. So, my questions are: - Is being able to split streams like this a new feature of ALSA/UCM2? How do I know if my system (Fedora 34) supports this feature? - Is there documentation somewhere to get me started with writing a config for my device? I have no idea what half the stuff in https://github.com/alsa-project/alsa-ucm-conf/commit/97beee6ded9e3dddfb5658… means so I'm expecting to fall into some pits during the process ;) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/122 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 00/17] Non-const bitfield helper conversions
by Geert Uytterhoeven 24 Nov '21

24 Nov '21

Hi all, <linux/bitfield.h> contains various helpers for accessing bitfields, as typically used in hardware registers for memory-mapped I/O blocks. These helpers ensure type safety, and deduce automatically shift values from mask values, avoiding mistakes due to inconsistent shifts and masks, and leading to a reduction in source code size. I have already submitted a few conversions to the FIELD_{GET,PREP}() helpers that were fixes for real bugs: - [PATCH] mips: cm: Convert to bitfield API to fix out-of-bounds access https://lore.kernel.org/r/0471c545117c5fa05bd9c73005cda9b74608a61e.16355013… - [PATCH] drm/armada: Fix off-by-one error in armada_overlay_get_property() https://lore.kernel.org/r/5818c8b04834e6a9525441bc181580a230354b69.16355012… Plus several patches for normal conversions: - [PATCH] ARM: ptrace: Use bitfield helpers https://lore.kernel.org/r/a1445d3abb45cfc95cb1b03180fd53caf122035b.16375932… - [PATCH] MIPS: CPC: Use bitfield helpers https://lore.kernel.org/r/35f0f17e3d987afaa9cd09cdcb8131d42a53c3e1.16375932… - [PATCH] MIPS: CPS: Use bitfield helpers https://lore.kernel.org/r/8bd8b1b9a3787e594285addcf2057754540d0a5f.16375932… - [PATCH] crypto: sa2ul - Use bitfield helpers https://lore.kernel.org/r/ca89d204ef2e40193479db2742eadf0d9cf3c0ff.16375932… - [PATCH] dmaengine: stm32-mdma: Use bitfield helpers https://lore.kernel.org/r/36ceab242a594233dc7dc6f1dddb4ac32d1e846f.16375932… - [PATCH] intel_th: Use bitfield helpers https://lore.kernel.org/r/b1e4f027aa88acfbdfaa771b0920bd1d977828ba.16375932… - [PATCH] Input: palmas-pwrbutton - use bitfield helpers https://lore.kernel.org/r/f8831b88346b36fc6e01e0910d0db6c94287d2b4.16375932… - [PATCH] irqchip/mips-gic: Use bitfield helpers https://lore.kernel.org/r/74f9d126961a90d3e311b92a54870eaac5b3ae57.16375932… - [PATCH] mfd: mc13xxx: Use bitfield helpers https://lore.kernel.org/r/afa46868cf8c1666e9cbbbec42767ca2294b024d.16375932… - [PATCH] regulator: lp873x: Use bitfield helpers https://lore.kernel.org/r/44d60384b640c8586b4ca7edbc9287a34ce21c5b.16375932… - [PATCH] regulator: lp87565: Use bitfield helpers https://lore.kernel.org/r/941c2dfd5b5b124b8950bcce42db4c343dfe9821.16375932… The existing FIELD_{GET,PREP}() macros are limited to compile-time constants. However, it is very common to prepare or extract bitfield elements where the bitfield mask is not a compile-time constant. To avoid this limitation, the AT91 clock driver already has its own field_{prep,get}() macros. This patch series makes them available for general use, and converts several drivers to the existing FIELD_{GET,PREP}() and the new field_{get,prep}() helpers. I can take the first two patches through the reneas-clk tree for v5.17, but probably it is best for the remaining patches to be postponed to v5.18. Thanks for your comments! Geert Uytterhoeven (17): bitfield: Add non-constant field_{prep,get}() helpers clk: renesas: Use bitfield helpers [RFC] soc: renesas: Use bitfield helpers [RFC] ARM: OMAP2+: Use bitfield helpers [RFC] bus: omap_l3_noc: Use bitfield helpers [RFC] clk: ti: Use bitfield helpers [RFC] iio: st_sensors: Use bitfield helpers [RFC] iio: humidity: hts221: Use bitfield helpers [RFC] iio: imu: st_lsm6dsx: Use bitfield helpers [RFC] media: ti-vpe: cal: Use bitfield helpers [RFC] mmc: sdhci-of-aspeed: Use bitfield helpers [RFC] pinctrl: aspeed: Use bitfield helpers [RFC] pinctl: ti: iodelay: Use bitfield helpers [RFC] regulator: ti-abb: Use bitfield helpers [RFC] thermal/ti-soc-thermal: Use bitfield helpers [RFC] ALSA: ice1724: Use bitfield helpers [RFC] rtw89: Use bitfield helpers arch/arm/mach-omap2/clkt2xxx_dpllcore.c | 5 +- arch/arm/mach-omap2/cm2xxx.c | 11 ++- arch/arm/mach-omap2/cm2xxx_3xxx.h | 9 +-- arch/arm/mach-omap2/cm33xx.c | 9 +-- arch/arm/mach-omap2/cm3xxx.c | 7 +- arch/arm/mach-omap2/cminst44xx.c | 9 +-- arch/arm/mach-omap2/powerdomains3xxx_data.c | 3 +- arch/arm/mach-omap2/prm.h | 2 - arch/arm/mach-omap2/prm2xxx.c | 4 +- arch/arm/mach-omap2/prm2xxx_3xxx.c | 7 +- arch/arm/mach-omap2/prm2xxx_3xxx.h | 9 +-- arch/arm/mach-omap2/prm33xx.c | 53 +++++------- arch/arm/mach-omap2/prm3xxx.c | 3 +- arch/arm/mach-omap2/prm44xx.c | 53 ++++-------- arch/arm/mach-omap2/vc.c | 12 +-- arch/arm/mach-omap2/vp.c | 11 +-- drivers/bus/omap_l3_noc.c | 4 +- drivers/clk/at91/clk-peripheral.c | 1 + drivers/clk/at91/pmc.h | 3 - drivers/clk/renesas/clk-div6.c | 6 +- drivers/clk/renesas/r8a779a0-cpg-mssr.c | 9 +-- drivers/clk/renesas/rcar-gen3-cpg.c | 15 ++-- drivers/clk/ti/apll.c | 25 +++--- drivers/clk/ti/dpll3xxx.c | 81 ++++++++----------- .../iio/common/st_sensors/st_sensors_core.c | 5 +- drivers/iio/humidity/hts221_core.c | 8 +- drivers/iio/imu/st_lsm6dsx/st_lsm6dsx.h | 1 - .../iio/imu/st_lsm6dsx/st_lsm6dsx_buffer.c | 7 +- drivers/iio/imu/st_lsm6dsx/st_lsm6dsx_core.c | 45 +++++------ drivers/iio/imu/st_lsm6dsx/st_lsm6dsx_shub.c | 11 +-- drivers/media/platform/ti-vpe/cal.h | 4 +- drivers/mmc/host/sdhci-of-aspeed.c | 5 +- drivers/net/wireless/realtek/rtw89/core.h | 38 ++------- drivers/pinctrl/aspeed/pinctrl-aspeed-g4.c | 3 +- drivers/pinctrl/aspeed/pinctrl-aspeed-g5.c | 3 +- drivers/pinctrl/aspeed/pinctrl-aspeed-g6.c | 3 +- drivers/pinctrl/aspeed/pinctrl-aspeed.c | 5 +- drivers/pinctrl/aspeed/pinmux-aspeed.c | 6 +- drivers/pinctrl/ti/pinctrl-ti-iodelay.c | 35 +++----- drivers/regulator/ti-abb-regulator.c | 7 +- drivers/soc/renesas/renesas-soc.c | 4 +- drivers/thermal/ti-soc-thermal/ti-bandgap.c | 11 ++- include/linux/bitfield.h | 30 +++++++ sound/pci/ice1712/wm8766.c | 14 ++-- sound/pci/ice1712/wm8776.c | 14 ++-- 45 files changed, 263 insertions(+), 347 deletions(-) -- 2.25.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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[PATCH v3 00/11] Add support for Legion 7 16ACHg6 laptop
by Lucas Tanure 24 Nov '21

24 Nov '21

Add support for laptops that have CS35L41 connected to an HDA codec by I2S and direct I2C/SPI connection to the CPU Hardware: - The 16ACHg6 laptop has two CS35L41 amplifiers, connected to Realtek ALC287 by an I2S bus and by and direct I2C to the CPU. - The ALC287 codec is connected to the CPU by an HDA bus. - The CS35L41 has a DSP which will require firmware to be loaded. Architecture: - Creation of a library of shared functions for CS35L41 ASoC and HDA - HDA codec driver (RealTek) and CS35L41 HDA driver are combined by using component binding, where it uses device names to find the components and bind to the master - The HDA CS35L41 driver applies pre-defined registers sequences for each action in playback for HDA Sound card Changes from V2: - Not an RFC - Create a new HDA driver for CS35L41 instead of using the ASoC one - Use component binding and device names to find the CS35L41 driver - Create a shared library for ASoC and HDA CS35L41 drivers Lucas Tanure (11): ASoC: cs35l41: Set the max SPI speed for the whole device ASoC: cs35l41: Convert tables to shared source code ASoC: cs35l41: Move regmap config struct to shared code ASoC: cs35l41: Create function for init array of Supplies ASoC: cs35l41: Move cs35l41_otp_unpack to shared code ASoC: cs35l41: Move power initializations to reg_sequence ASoC: cs35l41: Create shared function for errata patches ASoC: cs35l41: Create shared function for setting channels ASoC: cs35l41: Create shared function for boost configuration hda: cs35l41: Add support for CS35L41 in HDA systems ACPI / scan: Create platform device for CLSA0100 ACPI nodes MAINTAINERS | 2 + drivers/acpi/scan.c | 1 + drivers/platform/x86/i2c-multi-instantiate.c | 7 + include/sound/cs35l41.h | 741 ++++++++++++++++++ sound/pci/hda/Kconfig | 10 + sound/pci/hda/Makefile | 28 +- sound/pci/hda/cs35l41_hda.c | 522 ++++++++++++ sound/pci/hda/cs35l41_hda.h | 69 ++ sound/pci/hda/cs35l41_hda_i2c.c | 61 ++ sound/pci/hda/hda_component.h | 20 + sound/pci/hda/patch_realtek.c | 103 +++ sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/cs35l41-i2c.c | 16 - .../{cs35l41-tables.c => cs35l41-lib.c} | 400 +++++++++- sound/soc/codecs/cs35l41-spi.c | 49 +- sound/soc/codecs/cs35l41.c | 390 +-------- sound/soc/codecs/cs35l41.h | 739 ----------------- 17 files changed, 1964 insertions(+), 1198 deletions(-) create mode 100644 sound/pci/hda/cs35l41_hda.c create mode 100644 sound/pci/hda/cs35l41_hda.h create mode 100644 sound/pci/hda/cs35l41_hda_i2c.c create mode 100644 sound/pci/hda/hda_component.h rename sound/soc/codecs/{cs35l41-tables.c => cs35l41-lib.c} (68%) -- 2.34.0

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[PATCH] ALSA: led: Use restricted type for iface assignment
by Takashi Iwai 23 Nov '21

23 Nov '21

Fix a sparse warning that complains about the inconsistent type assignment for iface, which is a restricted type of snd_ctl_elem_iface_t. Fixes: a135dfb5de15 ("ALSA: led control - add sysfs kcontrol LED marking layer") Reported-by: kernel test robot <lkp(a)intel.com> Link: https://lore.kernel.org/r/202111201028.xduVYgH5-lkp@intel.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control_led.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index a95332b2b90b..207828f30983 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -509,7 +509,7 @@ static char *parse_string(char *s, char *val, size_t val_size) return s; } -static char *parse_iface(char *s, unsigned int *val) +static char *parse_iface(char *s, snd_ctl_elem_iface_t *val) { if (!strncasecmp(s, "card", 4)) *val = SNDRV_CTL_ELEM_IFACE_CARD; -- 2.31.1

2 1

[PATCH] ALSA: intel-dsp-config: add quirk for CML devices based on ES8336 codec
by Pierre-Louis Bossart 23 Nov '21

23 Nov '21

We've added quirks for ESS8336 but missed CML, add quirks for both LP and H versions. BugLink: https://github.com/thesofproject/linux/issues/3248 Fixes: 9d36ceab9415 ("ALSA: intel-dsp-config: add quirk for APL/GLK/TGL devices based on ES8336 codec") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index b9ac9e9e45a4..3b1402a1f4a9 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -252,6 +252,11 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x02c8, }, + { + .flags = FLAG_SOF, + .device = 0x02c8, + .codec_hid = "ESSX8336", + }, /* Cometlake-H */ { .flags = FLAG_SOF, @@ -276,6 +281,11 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x06c8, }, + { + .flags = FLAG_SOF, + .device = 0x06c8, + .codec_hid = "ESSX8336", + }, #endif /* Icelake */ -- 2.25.1

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[PATCH v2 00/11] dmaengine: kill off dma_slave_config->slave_id
by Arnd Bergmann 23 Nov '21

23 Nov '21

From: Arnd Bergmann <arnd(a)arndb.de> I recently came across some new uses of the 'slave_id' field that I had (almost) removed a few years ago. There are no legitimate uses of this field in the kernel, only a few stale references and two drivers that abuse the field as a side-channel between the dmaengine driver and its client. Let's change the xilinx and qualcomm drivers to use the documented side-channel (peripheral_data) instead, and remove the remnants of it to prevent new users from coming in. I think I got all the necessary Acks on v1 already, so if there are no further concerns, please merge this through the dmaengine tree for v5.17, or pull from git://git.kernel.org/pub/scm/linux/kernel/git/arnd/asm-generic.git tags/dma-slave-id-removal Arnd Arnd Bergmann (13): ASoC: tegra20-spdif: stop setting slave_id ASoC: dai_dma: remove slave_id field spi: pic32: stop setting dma_config->slave_id mmc: bcm2835: stop setting chan_config->slave_id dmaengine: shdma: remove legacy slave_id parsing dmaengine: pxa/mmp: stop referencing config->slave_id dmaengine: sprd: stop referencing config->slave_id dmaengine: qcom-adm: stop abusing slave_id config dmaengine: xilinx_dpdma: stop using slave_id field dmaengine: tegra20-apb: stop checking config->slave_id dmaengine: remove slave_id config field --- Changes in v2: - drop obsolete drivers/staging/patch - split tegra change into a separate patch - improve some patch descriptions. drivers/dma/mmp_pdma.c | 6 --- drivers/dma/pxa_dma.c | 7 ---- drivers/dma/qcom/qcom_adm.c | 56 ++++++++++++++++++++++---- drivers/dma/sh/shdma-base.c | 8 ---- drivers/dma/sprd-dma.c | 3 -- drivers/dma/tegra20-apb-dma.c | 6 --- drivers/dma/xilinx/xilinx_dpdma.c | 17 +++++--- drivers/gpu/drm/xlnx/zynqmp_disp.c | 9 ++++- drivers/mmc/host/bcm2835.c | 2 - drivers/mtd/nand/raw/qcom_nandc.c | 14 ++++++- drivers/spi/spi-pic32.c | 2 - drivers/tty/serial/msm_serial.c | 15 ++++++- include/linux/dma/qcom_adm.h | 12 ++++++ include/linux/dma/xilinx_dpdma.h | 11 +++++ include/linux/dmaengine.h | 4 -- include/linux/platform_data/asoc-s3c.h | 6 --- include/sound/dmaengine_pcm.h | 4 -- sound/core/pcm_dmaengine.c | 26 +----------- sound/soc/tegra/tegra20_spdif.c | 1 - 27 files changed, 134 insertions(+), 146 deletions(-) create mode 100644 include/linux/dma/qcom_adm.h create mode 100644 include/linux/dma/xilinx_dpdma.h -- 2.29.2 Cc: Andy Gross <agross(a)kernel.org> Cc: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Baolin Wang <baolin.wang7(a)gmail.com> Cc: Bjorn Andersson <bjorn.andersson(a)linaro.org> Cc: Chunyan Zhang <zhang.lyra(a)gmail.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Hyun Kwon <hyun.kwon(a)xilinx.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Jon Hunter <jonathanh(a)nvidia.com> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> Cc: Laxman Dewangan <ldewangan(a)nvidia.com> Cc: Manivannan Sadhasivam <mani(a)kernel.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Michal Simek <michal.simek(a)xilinx.com> Cc: Nicolas Saenz Julienne <nsaenz(a)kernel.org> Cc: Orson Zhai <orsonzhai(a)gmail.com> Cc: Robert Jarzmik <robert.jarzmik(a)free.fr> Cc: Scott Branden <sbranden(a)broadcom.com> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Thierry Reding <thierry.reding(a)gmail.com> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: bcm-kernel-feedback-list(a)broadcom.com Cc: dmaengine(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-arm-msm(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: linux-mmc(a)vger.kernel.org Cc: linux-mtd(a)lists.infradead.org Cc: linux-rpi-kernel(a)lists.infradead.org Cc: linux-serial(a)vger.kernel.org Cc: linux-spi(a)vger.kernel.org Cc: linux-staging(a)lists.linux.dev Cc: linux-tegra(a)vger.kernel.org

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[PATCH] soundwire: qcom: remove redundant version number read
by Srinivas Kandagatla 23 Nov '21

23 Nov '21

Controller version is already available in struct qcom_swrm_ctrl, Just make use of it instead of reading this again. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index a317bea2d42d..46995bb382eb 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1156,11 +1156,7 @@ static int qcom_swrm_get_port_config(struct qcom_swrm_ctrl *ctrl) ret = of_property_read_u8_array(np, "qcom,ports-block-pack-mode", bp_mode, nports); if (ret) { - u32 version; - - ctrl->reg_read(ctrl, SWRM_COMP_HW_VERSION, &version); - - if (version <= 0x01030000) + if (ctrl->version <= 0x01030000) memset(bp_mode, SWR_INVALID_PARAM, QCOM_SDW_MAX_PORTS); else return ret; -- 2.21.0

2 1

ASoC: mt8192: remove unnecessary CONFIG_PM
by Jiaxin Yu 23 Nov '21

23 Nov '21

[PATCH] ASoC: mt8192: remove unnecessary CONFIG_PM

2 3

Re: [PATCH] ALSA: hda/via - fix occasionally no sound with VT1802
by Takashi Iwai 23 Nov '21

23 Nov '21

On Tue, 23 Nov 2021 09:07:14 +0100, tianyucode wrote: > > From: tianyu <tianyu(a)kylinos.cn> > > Hi guys, I'm a new guy for audio driver. > > I have encountered a problem. When using the VT1802 codec, > there will be no sound occasionally. After checking the initialization process, > I found that the cause of the problem is that the power state does not have > hda_set_power_state(codec, AC_PWRST_D0). > > 1. The power state is set to AC_PWRST_D0 during the first initialization： > > azx_probe_continue > azx_probe_codecs > snd_hda_codec_new > snd_hda_codec_device_new > hda_set_power_state(codec, AC_PWRST_D0); > > 2. Then, the power state is set to AC_PWRST_D3 in VT1802, but not set in realtek (ALC269). > > azx_probe_continue > azx_codec_configure > … > patch_vt2002 > via_parse_auto_config > snd_hda_gen_parse_auto_config > add_all_pin_power_ctls > add_pin_power_ctls > set_path_power > > static void add_all_pin_power_ctls(struct hda_codec *codec, bool on) > { > /**/ > if (!codec->power_save_node) > return; > > add_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins, on); > /**/ > } > > The power_save_node is set to 0 in patch_alc269 and it returns here, > but it is set to 1 in VT1802. > > patch_vt2002P > via_new_spec > codec->power_save_node = 1; > via_parse_auto_config > snd_hda_gen_parse_auto_config > codec->power_save_node = 0;//it set to 0 here,but add_all_pin_power_ctls has been done > > patch_alc269 > codec->power_save_node = 0; > > 3. Next there is a suspend and resume process, the resume process will also set the > power state to AC_PWRST_D0, but I think there is a problem with this process. > > (1)suspend: > > azx_probe_continue > snd_card_register > .. > snd_hda_codec_dev_register > snd_hda_codec_register > snd_hda_power_down > pm_runtime_put_autosuspend > __pm_runtime_suspend(dev, RPM_GET_PUT | RPM_ASYNC | RPM_AUTO); > rpm_suspend > > static int rpm_suspend(struct device *dev, int rpmflags) > { > /**/ > if (rpmflags & RPM_ASYNC) { > dev->power.request = (rpmflags & RPM_AUTO) ? > RPM_REQ_AUTOSUSPEND : RPM_REQ_SUSPEND; > if (!dev->power.request_pending) { > dev->power.request_pending = true; > queue_work(pm_wq, &dev->power.work); > } > goto out; > } > > __update_runtime_status(dev, RPM_SUSPENDING); > > /**/ > retval = rpm_callback(callback, dev); > __update_runtime_status(dev, RPM_SUSPENDED); > /**/ > } > > The first call to rpm_suspend has the RPM_ASYNC flag, so queue_work is called, > and the work queue is used to enter rpm_suspend > > pm_runtime_init > INIT_WORK(&dev->power.work, pm_runtime_work); > > (2)resume: > > azx_probe_continue > set_default_power_save > snd_hda_set_power_save(&chip->bus, val * 1000); > codec_set_power_save(c, delay); > > int val = power_save; > static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;//It is 0 in my .config > > static void codec_set_power_save(struct hda_codec *codec, int delay) > { > struct device *dev = hda_codec_dev(codec); > if (delay == 0 && codec->auto_runtime_pm) > delay = 3000; > > if (delay > 0) { > pm_runtime_set_autosuspend_delay(dev, delay); > pm_runtime_use_autosuspend(dev); > pm_runtime_allow(dev); > if (!pm_runtime_suspended(dev)) > pm_runtime_mark_last_busy(dev); > } else { > pm_runtime_dont_use_autosuspend(dev); > pm_runtime_forbid(dev); > } > } > > pm_runtime_forbid > rpm_resume(dev, 0); > > static int rpm_resume(struct device *dev, int rpmflags) > { > /**/ > if (dev->power.runtime_status == RPM_ACTIVE) { > retval = 1; > goto out; > } > > /**/ > _update_runtime_status(dev, RPM_RESUMING); > retval = rpm_callback(callback, dev); > /**/ > __update_runtime_status(dev, RPM_ACTIVE); > /**/ > } > > The callback functions of suspend and resume are as follows, which set the power state: > > hda_call_codec_suspend > state = hda_set_power_state(codec, AC_PWRST_D3); > > hda_call_codec_resume > hda_set_power_state(codec, AC_PWRST_D0); > > You can see that the resume function relies on dev->power.runtime_status, > and the status is set in rpm_suspend. The operation of rpm_suspend depends > on the scheduling of the work queue. I added print debugging on my machine, > and occasionally there will be pm_runtime_work in Run after rpm_resume. > At this time, the suspend and resume processes are not performed correctly. > In VT1802, the power_state is still D3, and the machine has no sound. > > 4. I searched the Internet and did not find the relevant modification, > but found the commit 317d9313925c (ALSA: hda/realtek-Set default power save node to 0). > > Does VIA need to be modified like this? > > Since I am not familiar with ALSA and runtime pm, so come here to consult. For VIA codecs, there is an additional fixup table to turn off the power_save_node. Add an entry with the PCI SSID of your device to vt2002p_fixups[] table with VIA_FIXUP_POWER_SAVE. thanks, Takashi

2 1

[PATCH] ALSA: hda/via - fix occasionally no sound with VT1802
by tianyucode 23 Nov '21

23 Nov '21

From: tianyu <tianyu(a)kylinos.cn> Hi guys, I'm a new guy for audio driver. I have encountered a problem. When using the VT1802 codec, there will be no sound occasionally. After checking the initialization process, I found that the cause of the problem is that the power state does not have hda_set_power_state(codec, AC_PWRST_D0). 1. The power state is set to AC_PWRST_D0 during the first initialization： azx_probe_continue azx_probe_codecs snd_hda_codec_new snd_hda_codec_device_new hda_set_power_state(codec, AC_PWRST_D0); 2. Then, the power state is set to AC_PWRST_D3 in VT1802, but not set in realtek (ALC269). azx_probe_continue azx_codec_configure … patch_vt2002 via_parse_auto_config snd_hda_gen_parse_auto_config add_all_pin_power_ctls add_pin_power_ctls set_path_power static void add_all_pin_power_ctls(struct hda_codec *codec, bool on) { /**/ if (!codec->power_save_node) return; add_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins, on); /**/ } The power_save_node is set to 0 in patch_alc269 and it returns here, but it is set to 1 in VT1802. patch_vt2002P via_new_spec codec->power_save_node = 1; via_parse_auto_config snd_hda_gen_parse_auto_config codec->power_save_node = 0;//it set to 0 here,but add_all_pin_power_ctls has been done patch_alc269 codec->power_save_node = 0; 3. Next there is a suspend and resume process, the resume process will also set the power state to AC_PWRST_D0, but I think there is a problem with this process. (1)suspend: azx_probe_continue snd_card_register .. snd_hda_codec_dev_register snd_hda_codec_register snd_hda_power_down pm_runtime_put_autosuspend __pm_runtime_suspend(dev, RPM_GET_PUT | RPM_ASYNC | RPM_AUTO); rpm_suspend static int rpm_suspend(struct device *dev, int rpmflags) { /**/ if (rpmflags & RPM_ASYNC) { dev->power.request = (rpmflags & RPM_AUTO) ? RPM_REQ_AUTOSUSPEND : RPM_REQ_SUSPEND; if (!dev->power.request_pending) { dev->power.request_pending = true; queue_work(pm_wq, &dev->power.work); } goto out; } __update_runtime_status(dev, RPM_SUSPENDING); /**/ retval = rpm_callback(callback, dev); __update_runtime_status(dev, RPM_SUSPENDED); /**/ } The first call to rpm_suspend has the RPM_ASYNC flag, so queue_work is called, and the work queue is used to enter rpm_suspend pm_runtime_init INIT_WORK(&dev->power.work, pm_runtime_work); (2)resume: azx_probe_continue set_default_power_save snd_hda_set_power_save(&chip->bus, val * 1000); codec_set_power_save(c, delay); int val = power_save; static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;//It is 0 in my .config static void codec_set_power_save(struct hda_codec *codec, int delay) { struct device *dev = hda_codec_dev(codec); if (delay == 0 && codec->auto_runtime_pm) delay = 3000; if (delay > 0) { pm_runtime_set_autosuspend_delay(dev, delay); pm_runtime_use_autosuspend(dev); pm_runtime_allow(dev); if (!pm_runtime_suspended(dev)) pm_runtime_mark_last_busy(dev); } else { pm_runtime_dont_use_autosuspend(dev); pm_runtime_forbid(dev); } } pm_runtime_forbid rpm_resume(dev, 0); static int rpm_resume(struct device *dev, int rpmflags) { /**/ if (dev->power.runtime_status == RPM_ACTIVE) { retval = 1; goto out; } /**/ _update_runtime_status(dev, RPM_RESUMING); retval = rpm_callback(callback, dev); /**/ __update_runtime_status(dev, RPM_ACTIVE); /**/ } The callback functions of suspend and resume are as follows, which set the power state: hda_call_codec_suspend state = hda_set_power_state(codec, AC_PWRST_D3); hda_call_codec_resume hda_set_power_state(codec, AC_PWRST_D0); You can see that the resume function relies on dev->power.runtime_status, and the status is set in rpm_suspend. The operation of rpm_suspend depends on the scheduling of the work queue. I added print debugging on my machine, and occasionally there will be pm_runtime_work in Run after rpm_resume. At this time, the suspend and resume processes are not performed correctly. In VT1802, the power_state is still D3, and the machine has no sound. 4. I searched the Internet and did not find the relevant modification, but found the commit 317d9313925c (ALSA: hda/realtek-Set default power save node to 0). Does VIA need to be modified like this? Since I am not familiar with ALSA and runtime pm, so come here to consult. --- sound/pci/hda/patch_via.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index a29d7e989fe6..d01ea582cf25 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1073,8 +1073,10 @@ static int patch_vt2002P(struct hda_codec *codec) snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - if (spec->codec_type == VT1802) + if (spec->codec_type == VT1802) { + codec->power_save_node = 0; err = snd_hda_add_verbs(codec, vt1802_init_verbs); + } else err = snd_hda_add_verbs(codec, vt2002P_init_verbs); if (err < 0) -- 2.25.1

1 0

[RESEND,v3] ASoC: Intel: boards: add max98390 2/4 speakers support
by Mac Chiang 23 Nov '21

23 Nov '21

support 2 hw boards. 1. SSP2 connects max98390, 2 speakers. 2. SSP1 connects max98390, 2/4 speakers. 2 or 4 speakers playback add echo reference capture add bt offload support add DMI_OEM_STRING for board variants add ALC5682I-VS support Signed-off-by: Mark Hsieh <mark_hsieh(a)wistron.corp-partner.google.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Kieth Tzeng <keith.tzeng(a)quantatw.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- changes in resend v3: - use .comp_ids to scan acpi support ALC5682I-VD or ALC5682I-VS --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_maxim_common.c | 180 ++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 16 ++ sound/soc/intel/boards/sof_rt5682.c | 84 ++++++++ .../intel/common/soc-acpi-intel-adl-match.c | 21 ++ sound/soc/sof/sof-pci-dev.c | 9 + 6 files changed, 311 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 849445fcc05d..34ccefcc30c7 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -467,6 +467,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) select SND_SOC_MAX98373_I2C + select SND_SOC_MAX98390 select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT1015P diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index e66dfe666915..9171d9cd179e 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -5,6 +5,7 @@ #include <linux/string.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include <sound/soc-dai.h> #include <sound/soc-dapm.h> #include <uapi/sound/asound.h> @@ -133,6 +134,185 @@ void max_98373_set_codec_conf(struct snd_soc_card *card) } EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); +/* + * Maxim MAX98390 + */ +const struct snd_soc_dapm_route max_98390_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, +}; + +static const struct snd_kcontrol_new max_98390_tt_kcontrols[] = { + SOC_DAPM_PIN_SWITCH("TL Spk"), + SOC_DAPM_PIN_SWITCH("TR Spk"), +}; + +static const struct snd_soc_dapm_widget max_98390_tt_dapm_widgets[] = { + SND_SOC_DAPM_SPK("TL Spk", NULL), + SND_SOC_DAPM_SPK("TR Spk", NULL), +}; + +const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = { + /* Tweeter speaker */ + { "TL Spk", NULL, "Tweeter Left BE_OUT" }, + { "TR Spk", NULL, "Tweeter Right BE_OUT" }, +}; + +static struct snd_soc_codec_conf max_98390_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +static struct snd_soc_codec_conf max_98390_4spk_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV2_NAME), + .name_prefix = "Tweeter Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV3_NAME), + .name_prefix = "Tweeter Left", + }, +}; + +struct snd_soc_dai_link_component max_98390_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +struct snd_soc_dai_link_component max_98390_4spk_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV2_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV3_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_4spk_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static int max_98390_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int i; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + if (i >= ARRAY_SIZE(max_98390_4spk_components)) { + dev_err(codec_dai->dev, "invalid codec index %d\n", i); + return -ENODEV; + } + + if (!strcmp(codec_dai->component->name, MAX_98390_DEV0_NAME)) { + /* DEV0 tdm slot configuration Right */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x01, 3, 4, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV1_NAME)) { + /* DEV1 tdm slot configuration Left */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x02, 3, 4, 32); + } + + if (!strcmp(codec_dai->component->name, MAX_98390_DEV2_NAME)) { + /* DEVi2 tdm slot configuration Tweeter Right */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x04, 3, 4, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV3_NAME)) { + /* DEV3 tdm slot configuration Tweeter Left */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x08, 3, 4, 32); + } + } + return 0; +} + +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + /* add regular speakers dapm route */ + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_dapm_routes, + ARRAY_SIZE(max_98390_dapm_routes)); + if (ret) { + dev_err(rtd->dev, "unable to add Left/Right Speaker dapm, ret %d\n", ret); + return ret; + } + + /* add widgets/controls/dapm for tweeter speakers */ + if (acpi_dev_present("MX98390", "3", -1)) { + ret = snd_soc_dapm_new_controls(&card->dapm, max_98390_tt_dapm_widgets, + ARRAY_SIZE(max_98390_tt_dapm_widgets)); + + if (ret) { + dev_err(rtd->dev, "unable to add tweeter dapm controls, ret %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_add_card_controls(card, max_98390_tt_kcontrols, + ARRAY_SIZE(max_98390_tt_kcontrols)); + if (ret) { + dev_err(rtd->dev, "unable to add tweeter card controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_tt_dapm_routes, + ARRAY_SIZE(max_98390_tt_dapm_routes)); + if (ret) + dev_err(rtd->dev, + "unable to add Tweeter Left/Right Speaker dapm, ret %d\n", ret); + } + return ret; +} +EXPORT_SYMBOL_NS(max_98390_spk_codec_init, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +const struct snd_soc_ops max_98390_ops = { + .hw_params = max_98390_hw_params, +}; +EXPORT_SYMBOL_NS(max_98390_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch) +{ + if (ch == ARRAY_SIZE(max_98390_4spk_codec_conf)) { + card->codec_conf = max_98390_4spk_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_4spk_codec_conf); + } else { + card->codec_conf = max_98390_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_codec_conf); + } +} +EXPORT_SYMBOL_NS(max_98390_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); + /* * Maxim MAX98357A/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 3ff5e8fec4de..7a8c53049e4d 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -24,6 +24,22 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); void max_98373_set_codec_conf(struct snd_soc_card *card); int max_98373_trigger(struct snd_pcm_substream *substream, int cmd); +/* + * Maxim MAX98390 + */ +#define MAX_98390_CODEC_DAI "max98390-aif1" +#define MAX_98390_DEV0_NAME "i2c-MX98390:00" +#define MAX_98390_DEV1_NAME "i2c-MX98390:01" +#define MAX_98390_DEV2_NAME "i2c-MX98390:02" +#define MAX_98390_DEV3_NAME "i2c-MX98390:03" + +extern struct snd_soc_dai_link_component max_98390_components[2]; +extern struct snd_soc_dai_link_component max_98390_4spk_components[4]; +extern const struct snd_soc_ops max_98390_ops; + +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch); +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd); + /* * Maxim MAX98357A/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index c41f386b4138..3869b38274b9 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -59,6 +59,9 @@ (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) #define SOF_SSP_BT_OFFLOAD_PRESENT BIT(22) #define SOF_RT5682S_HEADPHONE_CODEC_PRESENT BIT(23) +#define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) +#define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) + /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -179,6 +182,36 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S_4SPK"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_MAX98390_TWEETER_SPEAKER_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, {} }; @@ -486,6 +519,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) if (err < 0) return err; } + return hdac_hdmi_jack_port_init(component, &card->dapm); } @@ -784,6 +818,20 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) { sof_rt1011_dai_link(&links[id]); + } else if (sof_rt5682_quirk & + SOF_MAX98390_SPEAKER_AMP_PRESENT) { + if (sof_rt5682_quirk & + SOF_MAX98390_TWEETER_SPEAKER_PRESENT) { + links[id].codecs = max_98390_4spk_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_4spk_components); + } else { + links[id].codecs = max_98390_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_components); + } + links[id].init = max_98390_spk_codec_init; + links[id].ops = &max_98390_ops; + links[id].dpcm_capture = 1; + } else { max_98357a_dai_link(&links[id]); } @@ -868,6 +916,10 @@ static int sof_audio_probe(struct platform_device *pdev) if (acpi_dev_present("RTL5682", NULL, -1)) sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + /* Detect the headset codec variant to support machines in DMI quirk */ + if (acpi_dev_present("RTL5682", NULL, -1)) + sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; dmic_be_num = 0; @@ -924,6 +976,14 @@ static int sof_audio_probe(struct platform_device *pdev) sof_rt1011_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) sof_rt1015p_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_MAX98390_SPEAKER_AMP_PRESENT) { + if (sof_rt5682_quirk & SOF_MAX98390_TWEETER_SPEAKER_PRESENT) + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_4spk_components)); + else + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_components)); + } if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) sof_audio_card_rt5682.num_links++; @@ -1050,6 +1110,29 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "adl_max98390_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, + { + .name = "adl_mx98390_rt5682s", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_RT5682S_HEADPHONE_CODEC_PRESENT | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, { .name = "adl_mx98360_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | @@ -1080,6 +1163,7 @@ MODULE_DESCRIPTION("SOF Audio Machine driver"); MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index fde310e5724b..0d04166b9d64 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -379,6 +379,11 @@ static const struct snd_soc_acpi_codecs adl_rt1019p_amp = { .codecs = {"RTL1019"} }; +static const struct snd_soc_acpi_codecs adl_max98390_amp = { + .num_codecs = 1, + .codecs = {"MX98390"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .comp_ids = &adl_rt5682_rt5682s_hp, @@ -434,6 +439,22 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-nau8825.tplg", }, + { + .comp_ids = &adl_rt5682_rt5682s_hp, + .drv_name = "adl_max98390_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, + { + .comp_ids = &adl_rt5682_rt5682s_hp, + .drv_name = "adl_mx98390_rt5682s", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index bc9e70765678..be90a5f79766 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -59,6 +59,15 @@ static const struct dmi_system_id sof_tplg_table[] = { }, .driver_data = "sof-adl-rt5682-ssp0-max98373-ssp2.tplg", }, + { + .callback = sof_tplg_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), + }, + .driver_data = "sof-adl-max98390-ssp2-rt5682-ssp0.tplg", + }, + {} }; -- 2.20.1

3 2

[PATCH] ASoC: qcom: Add support for ALC5682I-VS codec
by lvzhaoxiong 23 Nov '21

23 Nov '21

Qcom machine driver adds rt5682s support in this patch. Card name can be specified from dts by model property, and driver makes use of the name to distinguish which headset codec is on the board. Signed-off-by: lvzhaoxiong <lvzhaoxiong(a)huaqin.corp-partner.google.com> --- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/sc7180.c | 24 +++++++++++++++++++----- 2 files changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index b2173847dc47..cf3e151bb635 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -169,6 +169,7 @@ config SND_SOC_SC7180 select SND_SOC_LPASS_SC7180 select SND_SOC_MAX98357A select SND_SOC_RT5682_I2C + select SND_SOC_RT5682S select SND_SOC_ADAU7002 help To add support for audio on Qualcomm Technologies Inc. diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index 768566bb57a5..2fff764a00a7 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -17,6 +17,7 @@ #include <uapi/linux/input-event-codes.h> #include "../codecs/rt5682.h" +#include "../codecs/rt5682s.h" #include "common.h" #include "lpass.h" @@ -128,7 +129,21 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream) struct sc7180_snd_data *data = snd_soc_card_get_drvdata(card); struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - int ret; + int pll_id, pll_source, pll_in, pll_out, clk_id, ret; + + if (!(strcmp(card->name, "sc7180-rt5682-max98357a-1mic"))) { + pll_source = RT5682_PLL1_S_MCLK; + pll_id = 0; + clk_id = RT5682_SCLK_S_PLL1; + pll_out = RT5682_PLL1_FREQ; + pll_in = DEFAULT_MCLK_RATE; + } else if (!(strcmp(card->name, "sc7180-rt5682s-max98357a-1mic"))) { + pll_source = RT5682S_PLL_S_MCLK; + pll_id = RT5682S_PLL2; + clk_id = RT5682S_SCLK_S_PLL2; + pll_out = RT5682_PLL1_FREQ; + pll_in = DEFAULT_MCLK_RATE; + } switch (cpu_dai->id) { case MI2S_PRIMARY: @@ -145,16 +160,15 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream) SND_SOC_DAIFMT_I2S); /* Configure PLL1 for codec */ - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, - DEFAULT_MCLK_RATE, RT5682_PLL1_FREQ); + ret = snd_soc_dai_set_pll(codec_dai, pll_id, pll_source, + pll_in, pll_out); if (ret) { dev_err(rtd->dev, "can't set codec pll: %d\n", ret); return ret; } /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, - RT5682_PLL1_FREQ, + ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, pll_out, SND_SOC_CLOCK_IN); if (ret) dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", -- 2.17.1

1 0

[PATCH] ASoC: rk817: Add module alias for rk817-codec
by Nicolas Frattaroli 23 Nov '21

23 Nov '21

Without a module alias, autoloading the driver does not occurr when it is built as a module. By adding a module alias, the driver now probes fine automatically and therefore analog audio output works as it should. Fixes: 0d6a04da9b25 ("ASoC: Add Rockchip rk817 audio CODEC support") Signed-off-by: Nicolas Frattaroli <frattaroli.nicolas(a)gmail.com> --- sound/soc/codecs/rk817_codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 943d7d933e81..03f24edfe4f6 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -539,3 +539,4 @@ module_platform_driver(rk817_codec_driver); MODULE_DESCRIPTION("ASoC RK817 codec driver"); MODULE_AUTHOR("binyuan <kevan.lan(a)rock-chips.com>"); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:rk817-codec"); -- 2.34.0

3 2

[PATCH] ASoC: SOF: sof-pci-dev: use community key on all Up boards
by Pierre-Louis Bossart 23 Nov '21

23 Nov '21

There are already 3 versions of the Up boards with support for the SOF community key (ApolloLake, WhiskyLake, TigerLake). Rather than continue to add quirks for each version, let's add a wildcard. For WHL and TGL, the authentication supports both the SOF community key and the firmware signed with the Intel production key. Given two choices, the community key is the preferred option to allow developers to sign their own firmware. The firmware signed with production key can still be selected if needed with a kernel module option (snd-sof-pci.fw_path="intel/sof") Tested-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/sof-pci-dev.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index bc9e70765678..b4bc4f887b43 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -64,17 +64,9 @@ static const struct dmi_system_id sof_tplg_table[] = { static const struct dmi_system_id community_key_platforms[] = { { - .ident = "Up Squared", + .ident = "Up boards", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), - DMI_MATCH(DMI_BOARD_NAME, "UP-APL01"), - } - }, - { - .ident = "Up Extreme", - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), - DMI_MATCH(DMI_BOARD_NAME, "UP-WHL01"), } }, { -- 2.25.1

2 1

[PATCH 00/10] ASoC: SOF: enable multicore with dynamic pipelines
by Kai Vehmanen 23 Nov '21

23 Nov '21

Hi, initial support for SOF dynamic pipelines was added in commit 5fcdbb2d45df ("ASoC: SOF: Add support for dynamic pipelines"). When a pipeline is marked dynamic in the SOF DSP firmware topology definition (the tplg file kernel loads from filesystem), it means the pipeline resources are not allocated when DSP is booted (at driver probe, or at runtime resume), but rather delayed until the pipeline is actually used. Until now, it has not been possible to use multiple DSP cores in a topology that uses dynamic pipeline capability. This patchset addresses this gap and allows to freely mix dynamic pipelines and multicore usages in topology definitions. The series was originally reviewed in https://github.com/thesofproject/linux/pull/3019 Ranjani Sridharan (10): ASoC: SOF: Intel: hda: expose get_chip_info() ASoC: SOF: Introduce num_cores and ref count per core ASoC: SOF: Add ops for core_get and core_put ASoC: SOF: Intel: TGL: set core_get/put ops ASoC: SOF: Intel: CNL/ICL/APL: set core_get/core_put ops ASoC: SOF: topology: remove sof_load_pipeline_ipc() ASoC: SOF: free widgets in sof_tear_down_pipelines() for static pipelines ASoC: SOF: hda: don't use the core op for power up/power down ASoC: SOF: add support for dynamic pipelines with multi-core ASoC: SOF: Intel: hda: free DAI widget during stop and suspend sound/soc/sof/imx/imx8.c | 1 + sound/soc/sof/imx/imx8m.c | 1 + sound/soc/sof/intel/apl.c | 5 +- sound/soc/sof/intel/bdw.c | 9 +++ sound/soc/sof/intel/byt.c | 9 +++ sound/soc/sof/intel/cnl.c | 5 +- sound/soc/sof/intel/hda-dai.c | 23 +------ sound/soc/sof/intel/hda-dsp.c | 52 +++++++++++++++- sound/soc/sof/intel/hda-loader.c | 24 +++++--- sound/soc/sof/intel/hda.c | 17 ++---- sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/icl.c | 5 +- sound/soc/sof/intel/pci-tng.c | 9 +++ sound/soc/sof/intel/shim.h | 7 +++ sound/soc/sof/intel/tgl.c | 46 +++++++++++++- sound/soc/sof/ops.h | 71 ++++++++++++++++------ sound/soc/sof/sof-audio.c | 83 +++++++++++++++++++------ sound/soc/sof/sof-audio.h | 6 -- sound/soc/sof/sof-priv.h | 21 +++++-- sound/soc/sof/topology.c | 100 ------------------------------- 20 files changed, 294 insertions(+), 201 deletions(-) base-commit: 626a3dfbdb5d11f92e709680135abf272057ef59 -- 2.33.0

2 11

[PATCH] ASoC: cs42l42: Report initial jack state
by Richard Fitzgerald 23 Nov '21

23 Nov '21

When a jack handler is registered in cs42l42_set_jack() the initial state should be reported if an attached headphone/headset has already been detected. The jack detect sequence takes around 1 second: typically long enough for the machine driver to probe and register the jack handler in time to receive the first report from the interrupt handler. So it is possible on some systems that the correct initial state was seen simply because of lucky timing. Modular builds were more likely to miss the reporting of the initial state. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 4ca239f33737 ("ASoC: cs42l42: Always enable TS_PLUG and TS_UNPLUG interrupts") --- sound/soc/codecs/cs42l42.c | 22 ++++++++++++++++++++++ sound/soc/codecs/cs42l42.h | 2 ++ 2 files changed, 24 insertions(+) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 0c4303547fd8..43d98bdb5b5b 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -549,8 +549,25 @@ static int cs42l42_set_jack(struct snd_soc_component *component, struct snd_soc_ { struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); + /* Prevent race with interrupt handler */ + mutex_lock(&cs42l42->jack_detect_mutex); cs42l42->jack = jk; + if (jk) { + switch (cs42l42->hs_type) { + case CS42L42_PLUG_CTIA: + case CS42L42_PLUG_OMTP: + snd_soc_jack_report(jk, SND_JACK_HEADSET, SND_JACK_HEADSET); + break; + case CS42L42_PLUG_HEADPHONE: + snd_soc_jack_report(jk, SND_JACK_HEADPHONE, SND_JACK_HEADPHONE); + break; + default: + break; + } + } + mutex_unlock(&cs42l42->jack_detect_mutex); + return 0; } @@ -1618,6 +1635,8 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) CS42L42_M_DETECT_FT_MASK | CS42L42_M_HSBIAS_HIZ_MASK); + mutex_lock(&cs42l42->jack_detect_mutex); + /* Check auto-detect status */ if ((~masks[5]) & irq_params_table[5].mask) { if (stickies[5] & CS42L42_HSDET_AUTO_DONE_MASK) { @@ -1686,6 +1705,8 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) } } + mutex_unlock(&cs42l42->jack_detect_mutex); + return IRQ_HANDLED; } @@ -2033,6 +2054,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, cs42l42->dev = &i2c_client->dev; i2c_set_clientdata(i2c_client, cs42l42); + mutex_init(&cs42l42->jack_detect_mutex); cs42l42->regmap = devm_regmap_init_i2c(i2c_client, &cs42l42_regmap); if (IS_ERR(cs42l42->regmap)) { diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 75ade987d0db..9fff183dce8e 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -12,6 +12,7 @@ #ifndef __CS42L42_H__ #define __CS42L42_H__ +#include <linux/mutex.h> #include <sound/jack.h> #define CS42L42_PAGE_REGISTER 0x00 /* Page Select Register */ @@ -841,6 +842,7 @@ struct cs42l42_private { struct gpio_desc *reset_gpio; struct completion pdn_done; struct snd_soc_jack *jack; + struct mutex jack_detect_mutex; int pll_config; int bclk; u32 sclk; -- 2.11.0

2 1

[PATCH v2 0/3] ASoC: stm32: add pm runtime support
by Olivier Moysan 23 Nov '21

23 Nov '21

Enable support of pm runtime on STM32 SPDIFRX, I2S and DFSDM drivers to allow power state monitoring. Changes in v2: - Move pm runtime enabling before component registration Olivier Moysan (3): ASoC: stm32: i2s: add pm_runtime support ASoC: stm32: dfsdm: add pm_runtime support for audio ASoC: stm32: spdifrx: add pm_runtime support sound/soc/stm/stm32_adfsdm.c | 5 ++++- sound/soc/stm/stm32_i2s.c | 4 ++++ sound/soc/stm/stm32_spdifrx.c | 4 ++++ 3 files changed, 12 insertions(+), 1 deletion(-) -- 2.17.1

2 4

[PATCH v2 0/5] ASoC: SOF: Add PM support for i.MX8/i.MX8X/i.MX8M
by Daniel Baluta 23 Nov '21

23 Nov '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patch series adds support for System PM and Runtime PM with SOF for i.MX8 platforms. First patch adds common code to manage clocks, patch 2/3 adds the actual PM support, patch 4/5 implement the DSP start / reset callbacks for i.mx8m. Changes since v1: * rebased on top of Mark's for-next branch Daniel Baluta (5): ASoC: SOF: imx: Add code to manage DSP related clocks ASoC: SOF: imx8: Add runtime PM / System PM support ASoC: SOF: imx8m: Add runtime PM / System PM support ASoC: SOF: imx8m: Implement DSP start ASoC: SOF: imx8m: Implement reset callback sound/soc/sof/imx/imx-common.c | 24 +++++ sound/soc/sof/imx/imx-common.h | 11 ++ sound/soc/sof/imx/imx8.c | 137 +++++++++++++++++++++++ sound/soc/sof/imx/imx8m.c | 191 ++++++++++++++++++++++++++++++++- 4 files changed, 362 insertions(+), 1 deletion(-) -- 2.27.0

2 6