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ACL897 not detecting headphone jack and not showing in pipewire sink
by GitHub issues - opened 11 Jun '23

11 Jun '23

alsa-project/alsa-ucm-conf issue #323 was opened from axet: Hello! I have Gigabyte Mother board with Realtek ACL897 chip. alsamixer shows "Headphone" output I can manually unmute it and fix volume. Even if motherboard does not have jack detection for front panel (back panel jack detection works) I would like to be able to select Headphones as output in Gnome Sound. I've told this is alsa-ucm-conf profile issue. But I have not found any related profiles in /usr/share/alsa for my device. 00:1f.3 Audio device [0403]: Intel Corporation Alder Lake-S HD Audio Controller [8086:7ad0] (rev 11 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/323 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[BUG] selftests: alsa: pcm-test: premature timeout severs the test
by Mirsad Goran Todorovac 11 Jun '23

11 Jun '23

Hi, On several platforms, and for some time, I've noticed alsa: pcm-test TIMEOUT. I have tried to increase the timeout in selftests/alsa/settings: timeout=300, so I think there is no more purpose to further increase it, so something appears generally stuck. The test runs up to "default.time4.1.8 section where it hangs for more than 200 seconds, possibly indefinitely. The output of the selftest is: # # default.time3.1.8.0.PLAYBACK - 44.1kHz stereo large periods # # default.time3.1.8.0.PLAYBACK hw_params.RW_INTERLEAVED.S16_LE.44100.2.22496.202464 sw_params.202464 # ok 61 default.time3.0.3.0.PLAYBACK # # default.time4.0.3.0.PLAYBACK - 48kHz stereo small periods # # default.time4.0.3.0.PLAYBACK hw_params.RW_INTERLEAVED.S16_LE.48000.2.512.4096 sw_params.4096 # ok 62 default.time4.0.3.0.PLAYBACK # # default.time5.0.3.0.PLAYBACK - 48kHz stereo large periods # # default.time5.0.3.0.PLAYBACK hw_params.RW_INTERLEAVED.S16_LE.48000.2.24000.192000 sw_params.192000 # ok 63 default.time5.0.3.0.PLAYBACK # # default.time6.0.3.0.PLAYBACK - 48kHz 6 channel large periods # # default.time6.0.3.0.PLAYBACK hw_params.RW_INTERLEAVED.S16_LE.48000.2.48000.576000 sw_params.576000 # ok 64 default.time6.0.3.0.PLAYBACK # # default.time7.0.3.0.PLAYBACK - 96kHz stereo large periods # # default.time7.0.3.0.PLAYBACK hw_params.RW_INTERLEAVED.S16_LE.96000.2.48000.192000 sw_params.192000 # not ok 65 default.time3.1.8.0.PLAYBACK # # time mismatch: expected 4000ms got 17005 # # default.time4.1.8.# not ok 2 selftests: alsa: pcm-test # TIMEOUT 300 seconds The platform is AMD Ryzen 9 assembled box with AsRock mainboard. Config and Lshw attached. CONTINUED: Just to test further, I have increased timeout even further, to 400 seconds. Only then the test passed, but with numerous errors, and this is Ryzen 9, so I guess it can only be worse on hardware like i3 or i5. But many subtests failed, so I am submitting the entire test log (due to mailing list constraints to 100K size of attachments, it is compressed). Best regards, Mirsad -------------- diff -u /dev/null tools/testing/selftests/alsa/settings --- /dev/null 2023-06-11 00:36:30.651447094 +0200 +++ tools/testing/selftests/alsa/settings 2023-06-11 00:37:32.067504069 +0200 @@ -0,0 +1 @@ +timeout=400

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[PATCH v5 3/4] ASoC: tas2781: Add tas2781 driver
by Shenghao Ding 10 Jun '23

10 Jun '23

Create tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v5: - fix the coding style on tasdev_dsp_cfg_blk_remove - fix the coding style on tasdev_dsp_prog_blk_remove - change enum channel to unsigned short - remove global addr handling in the code - remove Superfluous cast in tasdevice_fw_ready - capitialize the bin file name, excluding the file extention name - rename some variables in tasdev_dsp_prog_blk_remove to comply with coding style - rename some variables in tasdev_dsp_cfg_blk_remove to comply with coding style - unneeded ret init in tasdevice_digital_putvol - add force firmware load Kcontrol - correct the return value in tasdevice_set_profile_id --- sound/soc/codecs/Kconfig | 25 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/tas2781-comlib.c | 534 +++++++++++++++++++++ sound/soc/codecs/tas2781-i2c.c | 763 ++++++++++++++++++++++++++++++ 4 files changed, 1328 insertions(+) create mode 100644 sound/soc/codecs/tas2781-comlib.c create mode 100644 sound/soc/codecs/tas2781-i2c.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8020097d4e4c..0481ed52e1c3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -234,6 +234,9 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2764 imply SND_SOC_TAS2770 imply SND_SOC_TAS2780 + imply SND_SOC_TAS2781_COMLIB + imply SND_SOC_TAS2781_FMWLIB + imply SND_SOC_TAS2781 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1699,6 +1702,28 @@ config SND_SOC_TAS2780 Enable support for Texas Instruments TAS2780 high-efficiency digital input mono Class-D audio power amplifiers. +config SND_SOC_TAS2781_COMLIB + depends on I2C + select CRC8 + select REGMAP_I2C + tristate + +config SND_SOC_TAS2781_FMWLIB + tristate + default n + +config SND_SOC_TAS2781 + tristate "Texas Instruments TAS2781 speaker amplifier" + depends on I2C + select SND_SOC_TAS2781_COMLIB + select SND_SOC_TAS2781_FMWLIB + help + Enable support for Texas Instruments TAS2781 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + Note the TAS2781 driver implements a flexible and configurable + algo coefficient setting, for one, two or even multiple TAS2781 + chips. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5cdbae88e6e3..97c3881ed587 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -269,6 +269,9 @@ snd-soc-tas5805m-objs := tas5805m.o snd-soc-tas6424-objs := tas6424.o snd-soc-tda7419-objs := tda7419.o snd-soc-tas2770-objs := tas2770.o +snd-soc-tas2781-comlib-objs := tas2781-comlib.o +snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o +snd-soc-tas2781-objs := tas2781-i2c.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -633,6 +636,9 @@ obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS2562) += snd-soc-tas2562.o obj-$(CONFIG_SND_SOC_TAS2764) += snd-soc-tas2764.o obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o +obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o +obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o +obj-$(CONFIG_SND_SOC_TAS2781) += snd-soc-tas2781.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c new file mode 100644 index 000000000000..a88c6c28a394 --- /dev/null +++ b/sound/soc/codecs/tas2781-comlib.c @@ -0,0 +1,534 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// tas2781-lib.c -- TAS2781 Common functions for HDA and ASoC Audio drivers +// +// Copyright 2023 Texas Instruments, Inc. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> + +#include <linux/crc8.h> +#include <linux/firmware.h> +#include <linux/gpio/consumer.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_irq.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tas2781.h> + +#define TASDEVICE_CRC8_POLYNOMIAL 0x4d + +static const struct regmap_range_cfg tasdevice_ranges[] = { + { + .range_min = 0, + .range_max = 256 * 128, + .selector_reg = TASDEVICE_PAGE_SELECT, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 8, + .val_bits = 8, + .cache_type = REGCACHE_RBTREE, + .ranges = tasdevice_ranges, + .num_ranges = ARRAY_SIZE(tasdevice_ranges), + .max_register = 256 * 128, +}; + +static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, + unsigned short chn, int book) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + int ret = 0; + + if (chn < tas_priv->ndev) { + struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; + struct regmap *map = tas_priv->regmap; + + if (client->addr != tasdev->dev_addr) { + client->addr = tasdev->dev_addr; + if (tasdev->cur_book == book) { + ret = regmap_write(map, + TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", + __func__, ret); + goto out; + } + } + goto out; + } + + if (tasdev->cur_book != book) { + ret = regmap_write(map, TASDEVICE_BOOKCTL_REG, book); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", + __func__, ret); + goto out; + } + tasdev->cur_book = book; + } + } else { + ret = -EINVAL; + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + } + +out: + return ret; +} + +int tasdevice_dev_read(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned int *val) +{ + int ret = 0; + + if (chn < tas_priv->ndev) { + struct regmap *map = tas_priv->regmap; + + ret = tasdevice_change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_read(map, TASDEVICE_PGRG(reg), val); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else { + ret = -EINVAL; + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + } + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_read); + +int tasdevice_dev_write(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned int value) +{ + int ret = 0; + + if (chn < tas_priv->ndev) { + struct regmap *map = tas_priv->regmap; + + ret = tasdevice_change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_write(map, TASDEVICE_PGRG(reg), + value); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else { + ret = -EINVAL; + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + } + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_write); + +int tasdevice_dev_bulk_write( + struct tasdevice_priv *tas_priv, unsigned short chn, + unsigned int reg, unsigned char *data, + unsigned int len) +{ + int ret = 0; + + if (chn < tas_priv->ndev) { + struct regmap *map = tas_priv->regmap; + + ret = tasdevice_change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_bulk_write(map, TASDEVICE_PGRG(reg), + data, len); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else { + ret = -EINVAL; + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + } + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_bulk_write); + +int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned char *data, + unsigned int len) +{ + int ret = 0; + + if (chn < tas_priv->ndev) { + struct regmap *map = tas_priv->regmap; + + ret = tasdevice_change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_bulk_read(map, TASDEVICE_PGRG(reg), data, len); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_bulk_read); + +int tasdevice_dev_update_bits( + struct tasdevice_priv *tas_priv, unsigned short chn, + unsigned int reg, unsigned int mask, unsigned int value) +{ + int ret = 0; + + if (chn < tas_priv->ndev) { + struct regmap *map = tas_priv->regmap; + + ret = tasdevice_change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_update_bits(map, TASDEVICE_PGRG(reg), + mask, value); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else { + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + ret = -EINVAL; + } + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_update_bits); + +struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c) +{ + struct tasdevice_priv *tas_priv; + + tas_priv = devm_kzalloc(&i2c->dev, sizeof(*tas_priv), GFP_KERNEL); + if (!tas_priv) + return NULL; + tas_priv->dev = &i2c->dev; + tas_priv->client = (void *)i2c; + + return tas_priv; +} +EXPORT_SYMBOL_GPL(tasdevice_kzalloc); + +void tas2781_reset(struct tasdevice_priv *tas_dev) +{ + int ret, i; + + if (tas_dev->reset) { + gpiod_set_value_cansleep(tas_dev->reset, 0); + usleep_range(500, 1000); + gpiod_set_value_cansleep(tas_dev->reset, 1); + } else { + for (i = 0; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_write(tas_dev, i, + TAS2781_REG_SWRESET, + TAS2781_REG_SWRESET_RESET); + if (ret < 0) + dev_err(tas_dev->dev, + "dev %d swreset fail, %d\n", + i, ret); + } + } + usleep_range(1000, 1050); +} +EXPORT_SYMBOL_GPL(tas2781_reset); + +int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, + void (*cont)(const struct firmware *fw, void *context)) +{ + int ret = 0; + + /* Codec Lock Hold to ensure that codec_probe and firmware parsing and + * loading do not simultaneously execute. + */ + mutex_lock(&tas_priv->codec_lock); + + scnprintf(tas_priv->rca_binaryname, 64, "%sRCA%d.bin", + tas_priv->dev_name, tas_priv->ndev); + crc8_populate_msb(tas_priv->crc8_lkp_tbl, TASDEVICE_CRC8_POLYNOMIAL); + tas_priv->codec = codec; + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_priv->rca_binaryname, tas_priv->dev, GFP_KERNEL, tas_priv, + cont); + if (ret) + dev_err(tas_priv->dev, "request_firmware_nowait err:0x%08x\n", + ret); + + /* Codec Lock Release*/ + mutex_unlock(&tas_priv->codec_lock); + return ret; +} +EXPORT_SYMBOL_GPL(tascodec_init); + +int tasdevice_init(struct tasdevice_priv *tas_priv) +{ + int ret = 0; + int i; + + tas_priv->regmap = devm_regmap_init_i2c(tas_priv->client, + &tasdevice_regmap); + if (IS_ERR(tas_priv->regmap)) { + ret = PTR_ERR(tas_priv->regmap); + dev_err(tas_priv->dev, "Failed to allocate register map: %d\n", + ret); + goto out; + } + + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + for (i = 0; i < tas_priv->ndev; i++) { + tas_priv->tasdevice[i].cur_book = -1; + tas_priv->tasdevice[i].cur_prog = -1; + tas_priv->tasdevice[i].cur_conf = -1; + } + + dev_set_drvdata(tas_priv->dev, tas_priv); + + mutex_init(&tas_priv->codec_lock); + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_init); + +static void tasdev_dsp_prog_blk_remove(struct tasdevice_prog *prog) +{ + struct tasdevice_data *tas_dt; + struct tasdev_blk *blk; + unsigned int i; + + if (!prog) + return; + + tas_dt = &(prog->dev_data); + + if (!tas_dt->dev_blks) + return; + + for (i = 0; i < tas_dt->nr_blk; i++) { + blk = &(tas_dt->dev_blks[i]); + kfree(blk->data); + } + kfree(tas_dt->dev_blks); +} + +static void tasdev_dsp_prog_remove(struct tasdevice_prog *prog, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_prog_blk_remove(&prog[i]); + kfree(prog); +} + +static void tasdev_dsp_cfg_blk_remove(struct tasdevice_config *cfg) +{ + struct tasdevice_data *tas_dt; + struct tasdev_blk *blk; + unsigned int i; + + if (cfg) { + tas_dt = &(cfg->dev_data); + + if (!tas_dt->dev_blks) + return; + + for (i = 0; i < tas_dt->nr_blk; i++) { + blk = &(tas_dt->dev_blks[i]); + kfree(blk->data); + } + kfree(tas_dt->dev_blks); + } +} + +static void tasdev_dsp_cfg_remove(struct tasdevice_config *config, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_cfg_blk_remove(&config[i]); + kfree(config); +} + +void tasdevice_dsp_remove(void *context) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + struct tasdevice_fw *tas_fmw = tas_dev->fmw; + + if (!tas_dev->fmw) + return; + + if (tas_fmw->programs) + tasdev_dsp_prog_remove(tas_fmw->programs, + tas_fmw->nr_programs); + if (tas_fmw->configs) + tasdev_dsp_cfg_remove(tas_fmw->configs, + tas_fmw->nr_configurations); + kfree(tas_fmw); + tas_dev->fmw = NULL; +} +EXPORT_SYMBOL_GPL(tasdevice_dsp_remove); + +void tasdevice_remove(struct tasdevice_priv *tas_priv) +{ + if (gpio_is_valid(tas_priv->irq_info.irq_gpio)) + gpio_free(tas_priv->irq_info.irq_gpio); + kfree(tas_priv->acpi_subsystem_id); + mutex_destroy(&tas_priv->codec_lock); +} +EXPORT_SYMBOL_GPL(tasdevice_remove); + +static int tasdevice_clamp(int val, int max, unsigned int invert) +{ + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + return val; +} + +int tasdevice_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask; + int max = mc->max; + int err_cnt = 0; + int val, i, ret; + + mask = (1 << fls(max)) - 1; + mask <<= mc->shift; + val = tasdevice_clamp(ucontrol->value.integer.value[0], max, invert); + for (i = 0; i < tas_priv->ndev; i++) { + ret = tasdevice_dev_update_bits(tas_priv, i, + mc->reg, mask, (unsigned int)(val << mc->shift)); + if (!ret) + continue; + err_cnt++; + dev_err(tas_priv->dev, "set AMP vol error in dev %d\n", i); + } + + /* All the devices set error, return 0 */ + return (err_cnt == tas_priv->ndev) ? 0 : 1; +} +EXPORT_SYMBOL_GPL(tasdevice_amp_putvol); + +int tasdevice_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask = 0; + int max = mc->max; + int ret = 0; + int val; + + /* Read the primary device */ + ret = tasdevice_dev_read(tas_priv, 0, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + mask = (1 << fls(max)) - 1; + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + val = tasdevice_clamp(val, max, invert); + ucontrol->value.integer.value[0] = val; + +out: + return ret; + +} +EXPORT_SYMBOL_GPL(tasdevice_amp_getvol); + +int tasdevice_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int err_cnt = 0; + int ret; + int val, i; + + val = tasdevice_clamp(ucontrol->value.integer.value[0], max, invert); + + for (i = 0; i < tas_priv->ndev; i++) { + ret = tasdevice_dev_write(tas_priv, i, mc->reg, + (unsigned int)val); + if (!ret) + continue; + err_cnt++; + dev_err(tas_priv->dev, + "set digital vol err in dev %d\n", i); + } + + /* All the devices set error, return 0 */ + return (err_cnt == tas_priv->ndev) ? 0 : 1; + +} +EXPORT_SYMBOL_GPL(tasdevice_digital_putvol); + +int tasdevice_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int ret, val; + + /* Read the primary device as the whole */ + ret = tasdevice_dev_read(tas_priv, 0, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get digital vol error\n", + __func__); + goto out; + } + + val = tasdevice_clamp(val, max, invert); + ucontrol->value.integer.value[0] = val; + +out: + return ret; + +} +EXPORT_SYMBOL_GPL(tasdevice_digital_getvol); + +MODULE_DESCRIPTION("TAS2781 common library"); +MODULE_AUTHOR("Shenghao Ding, TI, <shenghao-ding(a)ti.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c new file mode 100644 index 000000000000..4c59429a42b7 --- /dev/null +++ b/sound/soc/codecs/tas2781-i2c.c @@ -0,0 +1,763 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#include <linux/crc8.h> +#include <linux/firmware.h> +#include <linux/gpio/consumer.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_irq.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tas2781.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +static const struct i2c_device_id tasdevice_id[] = { + { "tas2781", TAS2781 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, tasdevice_id); + +#ifdef CONFIG_OF +static const struct of_device_id tasdevice_of_match[] = { + { .compatible = "ti,tas2781" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tasdevice_of_match); +#endif + +/** + * tas2781_digital_getvol - get the volum control + * @kcontrol: control pointer + * @ucontrol: User data + * Customer Kcontrol for tas2781 is primarily for regmap booking, paging + * depends on internal regmap mechanism. + * tas2781 contains book and page two-level register map, especially + * book switching will set the register BXXP00R7F, after switching to the + * correct book, then leverage the mechanism for paging to access the + * register. + */ +static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_digital_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_digital_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_amp_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_amp_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = (int)tas_priv->force_fwload_status; + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + tas_priv->force_fwload_status ? "ON" : "OFF"); + + return 0; +} + +static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + bool change, val = (bool)ucontrol->value.integer.value[0]; + + if (tas_priv->force_fwload_status == val) + change = false; + else { + change = true; + tas_priv->force_fwload_status = val; + } + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + tas_priv->force_fwload_status ? "ON" : "OFF"); + + return change; +} + +static const struct snd_kcontrol_new tas2781_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), + SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + +static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int ret = 0; + + if (tas_priv->rcabin.profile_cfg_id != + ucontrol->value.integer.value[0]) { + tas_priv->rcabin.profile_cfg_id = + ucontrol->value.integer.value[0]; + ret = 1; + } + + return ret; +} + +static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)tas_fw->nr_programs; + + return 0; +} + +static int tasdevice_info_configurations( + struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = + snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)tas_fw->nr_configurations - 1; + + return 0; +} + +static int tasdevice_info_profile(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->rcabin.ncfgs - 1; + + return 0; +} + +static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + + return 0; +} + +static int tasdevice_create_control(struct tasdevice_priv *tas_priv) +{ + struct snd_kcontrol_new *prof_ctrls; + int nr_controls = 1; + int mix_index = 0; + int ret; + char *name; + + prof_ctrls = devm_kcalloc(tas_priv->dev, nr_controls, + sizeof(prof_ctrls[0]), GFP_KERNEL); + if (!prof_ctrls) { + ret = -ENOMEM; + goto out; + } + + /* Create a mixer item for selecting the active profile */ + name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + if (!name) { + ret = -ENOMEM; + goto out; + } + scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "Speaker Profile Id"); + prof_ctrls[mix_index].name = name; + prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + prof_ctrls[mix_index].info = tasdevice_info_profile; + prof_ctrls[mix_index].get = tasdevice_get_profile_id; + prof_ctrls[mix_index].put = tasdevice_set_profile_id; + mix_index++; + + ret = snd_soc_add_component_controls(tas_priv->codec, + prof_ctrls, nr_controls < mix_index ? nr_controls : mix_index); + +out: + return ret; +} + +static int tasdevice_program_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + + return 0; +} + +static int tasdevice_program_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + unsigned int nr_program = ucontrol->value.integer.value[0]; + int ret = 0; + + if (tas_priv->cur_prog != nr_program) { + tas_priv->cur_prog = nr_program; + ret = 1; + } + + return ret; +} + +static int tasdevice_configuration_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + + return 0; +} + +static int tasdevice_configuration_put( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + unsigned int nr_configuration = ucontrol->value.integer.value[0]; + int ret = 0; + + if (tas_priv->cur_conf != nr_configuration) { + tas_priv->cur_conf = nr_configuration; + ret = 1; + } + + return ret; +} + +static int tasdevice_dsp_create_ctrls( + struct tasdevice_priv *tas_priv) +{ + struct snd_kcontrol_new *dsp_ctrls; + char *prog_name, *conf_name; + int nr_controls = 2; + int mix_index = 0; + int ret; + + /* Alloc kcontrol via devm_kzalloc, which don't manually + * free the kcontrol + */ + dsp_ctrls = devm_kcalloc(tas_priv->dev, nr_controls, + sizeof(dsp_ctrls[0]), GFP_KERNEL); + if (!dsp_ctrls) { + ret = -ENOMEM; + goto out; + } + + /* Create a mixer item for selecting the active profile */ + prog_name = devm_kzalloc(tas_priv->dev, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); + conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + if (!prog_name || !conf_name) { + ret = -ENOMEM; + goto out; + } + + scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "Speaker Program Id"); + dsp_ctrls[mix_index].name = prog_name; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_programs; + dsp_ctrls[mix_index].get = tasdevice_program_get; + dsp_ctrls[mix_index].put = tasdevice_program_put; + mix_index++; + + scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "Speaker Config Id"); + dsp_ctrls[mix_index].name = conf_name; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_configurations; + dsp_ctrls[mix_index].get = tasdevice_configuration_get; + dsp_ctrls[mix_index].put = tasdevice_configuration_put; + mix_index++; + + ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, + nr_controls < mix_index ? nr_controls : mix_index); + +out: + return ret; +} + +static void tasdevice_fw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_priv = context; + int ret = 0; + int i; + + mutex_lock(&tas_priv->codec_lock); + + ret = tasdevice_rca_parser(tas_priv, fmw); + if (ret) + goto out; + tasdevice_create_control(tas_priv); + + tasdevice_dsp_remove(tas_priv); + tasdevice_calbin_remove(tas_priv); + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", + tas_priv->dev_name); + ret = tasdevice_dsp_parser(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dspfw load %s error\n", + tas_priv->coef_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + tasdevice_dsp_create_ctrls(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; + + /* If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + for (i = 0; i < tas_priv->ndev; i++) { + scnprintf(tas_priv->cal_binaryname[i], 64, "%s_cal_0x%02x.bin", + tas_priv->dev_name, tas_priv->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_priv, + tas_priv->cal_binaryname[i], i); + if (ret != 0) + dev_err(tas_priv->dev, + "%s: load %s error, default will effect\n", + __func__, tas_priv->cal_binaryname[i]); + } + + tasdevice_prmg_calibdata_load(tas_priv, 0); + tas_priv->cur_prog = 0; +out: + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + /*If DSP FW fail, kcontrol won't be created */ + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + } + mutex_unlock(&tas_priv->codec_lock); + if (fmw) + release_firmware(fmw); +} + +static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *codec = snd_soc_dapm_to_component(w->dapm); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int state = 0; + + /* Codec Lock Hold */ + mutex_lock(&tas_priv->codec_lock); + if (event == SND_SOC_DAPM_PRE_PMD) + state = 1; + tasdevice_tuning_switch(tas_priv, state); + /* Codec Lock Release*/ + mutex_unlock(&tas_priv->codec_lock); + + return 0; +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT_E("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, + 0, 0, tasdevice_dapm_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC") +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"SPK", NULL, "ASI"}, + {"OUT", NULL, "SPK"}, + {"ASI OUT", NULL, "DMIC"} +}; + +static int tasdevice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *codec = dai->component; + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int ret = 0; + + if (tas_priv->fw_state != TASDEVICE_DSP_FW_ALL_OK) { + dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + ret = -EINVAL; + } + + return ret; +} + +static int tasdevice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct tasdevice_priv *tas_priv = snd_soc_dai_get_drvdata(dai); + unsigned int slot_width; + unsigned int fsrate; + int bclk_rate; + int rc = 0; + + fsrate = params_rate(params); + switch (fsrate) { + case 48000: + case 44100: + break; + default: + dev_err(tas_priv->dev, "%s: incorrect sample rate = %u\n", + __func__, fsrate); + rc = -EINVAL; + goto out; + } + + slot_width = params_width(params); + switch (slot_width) { + case 16: + case 20: + case 24: + case 32: + break; + default: + dev_err(tas_priv->dev, "%s: incorrect slot width = %u\n", + __func__, slot_width); + rc = -EINVAL; + goto out; + } + + bclk_rate = snd_soc_params_to_bclk(params); + if (bclk_rate < 0) { + dev_err(tas_priv->dev, "%s: incorrect bclk rate = %d\n", + __func__, bclk_rate); + rc = bclk_rate; + goto out; + } + +out: + return rc; +} + +static int tasdevice_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct tasdevice_priv *tas_priv = snd_soc_dai_get_drvdata(codec_dai); + + tas_priv->sysclk = freq; + + return 0; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .startup = tasdevice_startup, + .hw_params = tasdevice_hw_params, + .set_sysclk = tasdevice_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2781_codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_codec_probe(struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + return tascodec_init(tas_priv, codec, tasdevice_fw_ready); +} + +static void tasdevice_deinit(void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + tasdevice_calbin_remove(tas_priv); + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; +} + +static void tasdevice_codec_remove( + struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + tasdevice_deinit(tas_priv); +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_codec_probe, + .remove = tasdevice_codec_remove, + .controls = tas2781_snd_controls, + .num_controls = ARRAY_SIZE(tas2781_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + unsigned int dev_addrs[TASDEVICE_MAX_CHANNELS]; + int rc, i, ndev = 0; + + if (tas_priv->isacpi) { + ndev = device_property_read_u32_array(&client->dev, + "ti,audio-slots", NULL, 0); + if (ndev <= 0) { + ndev = 1; + dev_addrs[0] = client->addr; + } else { + ndev = (ndev < ARRAY_SIZE(dev_addrs)) + ? ndev : ARRAY_SIZE(dev_addrs); + ndev = device_property_read_u32_array(&client->dev, + "ti,audio-slots", dev_addrs, ndev); + } + + tas_priv->irq_info.irq_gpio = + acpi_dev_gpio_irq_get(ACPI_COMPANION(&client->dev), 0); + } else { + struct device_node *np = tas_priv->dev->of_node; +#ifdef CONFIG_OF + const __be32 *reg, *reg_end; + int len, sw, aw; + + aw = of_n_addr_cells(np); + sw = of_n_size_cells(np); + if (sw == 0) { + reg = (const __be32 *)of_get_property(np, + "reg", &len); + reg_end = reg + len/sizeof(*reg); + ndev = 0; + do { + dev_addrs[ndev] = of_read_number(reg, aw); + reg += aw; + ndev++; + } while (reg < reg_end); + } else { + ndev = 1; + dev_addrs[0] = client->addr; + } +#else + ndev = 1; + dev_addrs[0] = client->addr; +#endif + tas_priv->irq_info.irq_gpio = of_irq_get(np, 0); + } + tas_priv->ndev = ndev; + for (i = 0; i < ndev; i++) + tas_priv->tasdevice[i].dev_addr = dev_addrs[i]; + + tas_priv->reset = devm_gpiod_get_optional(&client->dev, + "reset-gpios", GPIOD_OUT_HIGH); + if (IS_ERR(tas_priv->reset)) + dev_err(tas_priv->dev, "%s Can't get reset GPIO\n", + __func__); + + strcpy(tas_priv->dev_name, tasdevice_id[tas_priv->chip_id].name); + + if (gpio_is_valid(tas_priv->irq_info.irq_gpio)) { + rc = gpio_request(tas_priv->irq_info.irq_gpio, + "AUDEV-IRQ"); + if (!rc) { + gpio_direction_input( + tas_priv->irq_info.irq_gpio); + + tas_priv->irq_info.irq = + gpio_to_irq(tas_priv->irq_info.irq_gpio); + } else + dev_err(tas_priv->dev, "%s: GPIO %d request error\n", + __func__, tas_priv->irq_info.irq_gpio); + } else + dev_err(tas_priv->dev, + "Looking up irq-gpio property failed %d\n", + tas_priv->irq_info.irq_gpio); +} + +static int tasdevice_i2c_probe(struct i2c_client *i2c) +{ + const struct i2c_device_id *id = i2c_match_id(tasdevice_id, i2c); + const struct acpi_device_id *acpi_id; + struct tasdevice_priv *tas_priv; + int ret; + + tas_priv = tasdevice_kzalloc(i2c); + if (!tas_priv) + return -ENOMEM; + + if (ACPI_HANDLE(&i2c->dev)) { + acpi_id = acpi_match_device(i2c->dev.driver->acpi_match_table, + &i2c->dev); + if (!acpi_id) { + dev_err(&i2c->dev, "No driver data\n"); + ret = -EINVAL; + goto err; + } + tas_priv->chip_id = acpi_id->driver_data; + tas_priv->isacpi = true; + } else { + tas_priv->chip_id = id ? id->driver_data : 0; + tas_priv->isacpi = false; + } + + tasdevice_parse_dt(tas_priv); + + ret = tasdevice_init(tas_priv); + if (ret) + goto err; + + ret = devm_snd_soc_register_component(tas_priv->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_priv->dev, "%s: codec register error:0x%08x\n", + __func__, ret); + goto err; + } +err: + if (ret < 0) + tasdevice_remove(tas_priv); + return ret; +} + +static void tasdevice_i2c_remove(struct i2c_client *client) +{ + struct tasdevice_priv *tas_priv = i2c_get_clientdata(client); + + tasdevice_remove(tas_priv); +} + +#ifdef CONFIG_ACPI +static const struct acpi_device_id tasdevice_acpi_match[] = { + { "TAS2781", TAS2781 }, + {}, +}; + +MODULE_DEVICE_TABLE(acpi, tasdevice_acpi_match); +#endif + +static struct i2c_driver tasdevice_i2c_driver = { + .driver = { + .name = "tas2781-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tasdevice_of_match), +#ifdef CONFIG_ACPI + .acpi_match_table = ACPI_PTR(tasdevice_acpi_match), +#endif + }, + .probe = tasdevice_i2c_probe, + .remove = tasdevice_i2c_remove, + .id_table = tasdevice_id, +}; + +module_i2c_driver(tasdevice_i2c_driver); + +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_AUTHOR("Kevin Lu <kevin-lu(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2781 Driver"); +MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS(SND_SOC_TAS2781_FMWLIB); -- 2.34.1

1 0

[PATCH v5 2/4] ASoC: tas2781: firmware lib
by Shenghao Ding 10 Jun '23

10 Jun '23

Create tas2781 firmware lib. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v5: - fixed issue | Reported-by: kernel test robot <lkp(a)intel.com> | Link: https://lore.kernel.org/oe-kbuild-all/202305022033.LiI7Ojm4-lkp@intel.com/ - Move back to codecs folder, this firmware lib is special for the audio cases - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202305221045.K3Bx6bts-lkp@intel.com/ - change enum channel to unsigned short - coding style correction - fixed active_device detection issue in different configs in tasdevice_add_config. - fixed the log output level from dev_info to dev_err at error branch in tas2781_load_calibration - correct the conf_no boundary in tasdevice_select_tuningprm_cfg - add force firmward load status in tasdevice_select_tuningprm_cfg - "Unneeded loading dsp conf" as the non-error --- sound/soc/codecs/tas2781-fmwlib.c | 2427 +++++++++++++++++++++++++++++ 1 file changed, 2427 insertions(+) create mode 100644 sound/soc/codecs/tas2781-fmwlib.c diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c new file mode 100644 index 000000000000..432b19ccec8c --- /dev/null +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -0,0 +1,2427 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// tasdevice-fmw.c -- TASDEVICE firmware support +// +// Copyright 2023 Texas Instruments, Inc. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> + +#include <linux/crc8.h> +#include <linux/firmware.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_irq.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/tas2781.h> + + +#define ERROR_PRAM_CRCCHK 0x0000000 +#define ERROR_YRAM_CRCCHK 0x0000001 +#define PPC_DRIVER_CRCCHK 0x00000200 + +#define TAS2781_SA_COEFF_SWAP_REG TASDEVICE_REG(0, 0x35, 0x2c) +#define TAS2781_YRAM_BOOK1 140 +#define TAS2781_YRAM1_PAGE 42 +#define TAS2781_YRAM1_START_REG 88 + +#define TAS2781_YRAM2_START_PAGE 43 +#define TAS2781_YRAM2_END_PAGE 49 +#define TAS2781_YRAM2_START_REG 8 +#define TAS2781_YRAM2_END_REG 127 + +#define TAS2781_YRAM3_PAGE 50 +#define TAS2781_YRAM3_START_REG 8 +#define TAS2781_YRAM3_END_REG 27 + +/*should not include B0_P53_R44-R47 */ +#define TAS2781_YRAM_BOOK2 0 +#define TAS2781_YRAM4_START_PAGE 50 +#define TAS2781_YRAM4_END_PAGE 60 + +#define TAS2781_YRAM5_PAGE 61 +#define TAS2781_YRAM5_START_REG TAS2781_YRAM3_START_REG +#define TAS2781_YRAM5_END_REG TAS2781_YRAM3_END_REG + +#define TASDEVICE_MAXPROGRAM_NUM_KERNEL 5 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS 64 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL 10 +#define MAIN_ALL_DEVICES_1X 0x01 +#define MAIN_DEVICE_A_1X 0x02 +#define MAIN_DEVICE_B_1X 0x03 +#define MAIN_DEVICE_C_1X 0x04 +#define MAIN_DEVICE_D_1X 0x05 +#define COEFF_DEVICE_A_1X 0x12 +#define COEFF_DEVICE_B_1X 0x13 +#define COEFF_DEVICE_C_1X 0x14 +#define COEFF_DEVICE_D_1X 0x15 +#define PRE_DEVICE_A_1X 0x22 +#define PRE_DEVICE_B_1X 0x23 +#define PRE_DEVICE_C_1X 0x24 +#define PRE_DEVICE_D_1X 0x25 +#define PRE_SOFTWARE_RESET_DEVICE_A 0x41 +#define PRE_SOFTWARE_RESET_DEVICE_B 0x42 +#define PRE_SOFTWARE_RESET_DEVICE_C 0x43 +#define PRE_SOFTWARE_RESET_DEVICE_D 0x44 +#define POST_SOFTWARE_RESET_DEVICE_A 0x45 +#define POST_SOFTWARE_RESET_DEVICE_B 0x46 +#define POST_SOFTWARE_RESET_DEVICE_C 0x47 +#define POST_SOFTWARE_RESET_DEVICE_D 0x48 + +struct tas_crc { + unsigned char offset; + unsigned char len; +}; + +static const char deviceNumber[TASDEVICE_DSP_TAS_MAX_DEVICE] = { + 1, 2, 1, 2, 1, 1, 0, 2, 4, 3, 1, 2, 3, 4 +}; + +static struct tasdevice_config_info *tasdevice_add_config( + struct tasdevice_priv *tas_priv, unsigned char *config_data, + unsigned int config_size, int *status) +{ + struct tasdevice_config_info *cfg_info; + struct tasdev_blk_data **bk_da; + unsigned int config_offset = 0; + unsigned int i; + + /* In most projects are many audio cases, such as music, handfree, + * receiver, games, audio-to-haptics, PMIC record, bypass mode, + * portrait, landscape, etc. Even in multiple audios, one or + * two of the chips will work for the special case, such as + * ultrasonic application. In order to support these variable-numbers + * of audio cases, flexible configs have been introduced in the + * dsp firmware. + */ + cfg_info = kzalloc(sizeof(struct tasdevice_config_info), GFP_KERNEL); + if (!cfg_info) { + *status = -ENOMEM; + goto out; + } + + if (tas_priv->rcabin.fw_hdr.binary_version_num >= 0x105) { + if (config_offset + 64 > (int)config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add conf: Out of boundary\n"); + goto out; + } + config_offset += 64; + } + + if (config_offset + 4 > (int)config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add config: Out of boundary\n"); + goto out; + } + + /* convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + cfg_info->nblocks = + be32_to_cpup((__be32 *)&config_data[config_offset]); + config_offset += 4; + + /* Several kinds of dsp/algorithm firmwares can run on tas2781, + * the number and size of blk are not fixed and different among + * these firmwares. + */ + bk_da = cfg_info->blk_data = kcalloc(cfg_info->nblocks, + sizeof(struct tasdev_blk_data *), GFP_KERNEL); + if (!bk_da) { + *status = -ENOMEM; + goto out; + } + cfg_info->real_nblocks = 0; + for (i = 0; i < cfg_info->nblocks; i++) { + if (config_offset + 12 > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d nblocks = %u!\n", + __func__, i, cfg_info->nblocks); + break; + } + bk_da[i] = kzalloc(sizeof(struct tasdev_blk_data), GFP_KERNEL); + if (!bk_da[i]) { + *status = -ENOMEM; + break; + } + + bk_da[i]->dev_idx = config_data[config_offset]; + config_offset++; + + bk_da[i]->block_type = config_data[config_offset]; + config_offset++; + + if (bk_da[i]->block_type == TASDEVICE_BIN_BLK_PRE_POWER_UP) { + if (bk_da[i]->dev_idx == 0) + cfg_info->active_dev = + (1 << tas_priv->ndev) - 1; + else + cfg_info->active_dev |= 1 << + (bk_da[i]->dev_idx - 1); + + } + bk_da[i]->yram_checksum = + be16_to_cpup((__be16 *)&config_data[config_offset]); + config_offset += 2; + bk_da[i]->block_size = + be32_to_cpup((__be32 *)&config_data[config_offset]); + config_offset += 4; + + bk_da[i]->n_subblks = + be32_to_cpup((__be32 *)&config_data[config_offset]); + + config_offset += 4; + + if (config_offset + bk_da[i]->block_size > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d blks = %u!\n", + __func__, i, cfg_info->nblocks); + break; + } + /* instead of kzalloc+memcpy */ + bk_da[i]->regdata = kmemdup(&config_data[config_offset], + bk_da[i]->block_size, GFP_KERNEL); + if (!bk_da[i]->regdata) { + *status = -ENOMEM; + goto out; + } + + config_offset += bk_da[i]->block_size; + cfg_info->real_nblocks += 1; + } + +out: + return cfg_info; +} + +int tasdevice_rca_parser(void *context, const struct firmware *fmw) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_config_info **cfg_info; + struct tasdevice_rca_hdr *fw_hdr; + struct tasdevice_rca *rca; + unsigned int total_config_sz = 0; + unsigned char *buf; + int offset = 0; + int ret = 0; + int i; + + rca = &(tas_priv->rcabin); + fw_hdr = &(rca->fw_hdr); + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "Failed to read %s\n", + tas_priv->rca_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + ret = -EINVAL; + goto out; + } + buf = (unsigned char *)fmw->data; + + fw_hdr->img_sz = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + if (fw_hdr->img_sz != fmw->size) { + dev_err(tas_priv->dev, + "File size not match, %d %u", (int)fmw->size, + fw_hdr->img_sz); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + ret = -EINVAL; + goto out; + } + + fw_hdr->checksum = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + fw_hdr->binary_version_num = be32_to_cpup((__be32 *)&buf[offset]); + if (fw_hdr->binary_version_num < 0x103) { + dev_err(tas_priv->dev, "File version 0x%04x is too low", + fw_hdr->binary_version_num); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + ret = -EINVAL; + goto out; + } + offset += 4; + fw_hdr->drv_fw_version = be32_to_cpup((__be32 *)&buf[offset]); + offset += 8; + fw_hdr->plat_type = buf[offset]; + offset += 1; + fw_hdr->dev_family = buf[offset]; + offset += 1; + fw_hdr->reserve = buf[offset]; + offset += 1; + fw_hdr->ndev = buf[offset]; + offset += 1; + if (fw_hdr->ndev != tas_priv->ndev) { + dev_err(tas_priv->dev, + "ndev(%u) in rcabin mismatch ndev(%u) in DTS\n", + fw_hdr->ndev, tas_priv->ndev); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + ret = -EINVAL; + goto out; + } + if (offset + TASDEVICE_DEVICE_SUM > fw_hdr->img_sz) { + dev_err(tas_priv->dev, "rca_ready: Out of boundary!\n"); + ret = -EINVAL; + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + + for (i = 0; i < TASDEVICE_DEVICE_SUM; i++, offset++) + fw_hdr->devs[i] = buf[offset]; + + fw_hdr->nconfig = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + + for (i = 0; i < TASDEVICE_CONFIG_SUM; i++) { + fw_hdr->config_size[i] = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + total_config_sz += fw_hdr->config_size[i]; + } + + if (fw_hdr->img_sz - total_config_sz != (unsigned int)offset) { + dev_err(tas_priv->dev, "Bin file error!\n"); + ret = -EINVAL; + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + + cfg_info = kcalloc(fw_hdr->nconfig, sizeof(*cfg_info), GFP_KERNEL); + if (!cfg_info) { + ret = -ENOMEM; + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + rca->cfg_info = cfg_info; + rca->ncfgs = 0; + for (i = 0; i < (int)fw_hdr->nconfig; i++) { + rca->ncfgs += 1; + cfg_info[i] = tasdevice_add_config(tas_priv, &buf[offset], + fw_hdr->config_size[i], &ret); + if (ret) { + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + offset += (int)fw_hdr->config_size[i]; + } +out: + return ret; +} +EXPORT_SYMBOL_NS_GPL(tasdevice_rca_parser, SND_SOC_TAS2781_FMWLIB); + +static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + + if (offset + 16 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + + /* convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + block->type = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + block->is_pchksum_present = data[offset]; + offset++; + + block->pchksum = data[offset]; + offset++; + + block->is_ychksum_present = data[offset]; + offset++; + + block->ychksum = data[offset]; + offset++; + + block->blk_size = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + block->nr_subblocks = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + if (offset + block->blk_size > fmw->size) { + dev_err(tas_fmw->dev, "%s: nSublocks error\n", __func__); + offset = -EINVAL; + goto out; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], block->blk_size, GFP_KERNEL); + if (!block->data) { + offset = -ENOMEM; + goto out; + } + offset += block->blk_size; + +out: + return offset; +} + +static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = fmw->data; + struct tasdev_blk *blk; + unsigned int i; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + img_data->nr_blk = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(struct tasdev_blk), GFP_KERNEL); + if (!img_data->dev_blks) { + offset = -ENOMEM; + goto out; + } + + for (i = 0; i < img_data->nr_blk; i++) { + blk = &(img_data->dev_blks[i]); + offset = fw_parse_block_data_kernel(tas_fmw, blk, fmw, offset); + if (offset < 0) { + offset = -EINVAL; + break; + } + } + +out: + return offset; +} + +static int fw_parse_program_data_kernel( + struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + struct tasdevice_prog *program; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &(tas_fmw->programs[i]); + if (offset + 72 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + offset = -EINVAL; + goto out; + } + /*skip 72 unused byts*/ + offset += 72; + + offset = fw_parse_data_kernel(tas_fmw, &(program->dev_data), + fmw, offset); + if (offset < 0) + goto out; + } + +out: + return offset; +} + +static int fw_parse_configuration_data_kernel( + struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + struct tasdevice_config *config; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &(tas_fmw->configs[i]); + if (offset + 80 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + offset = -EINVAL; + goto out; + } + memcpy(config->name, &data[offset], 64); + /*skip extra 16 bytes*/ + offset += 80; + + offset = fw_parse_data_kernel(tas_fmw, &(config->dev_data), + fmw, offset); + if (offset < 0) + goto out; + } + +out: + return offset; +} + +static int fw_parse_variable_header_kernel( + struct tasdevice_priv *tas_priv, const struct firmware *fmw, + int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + struct tasdevice_prog *program; + struct tasdevice_config *config; + const unsigned char *buf = fmw->data; + unsigned short max_confs; + unsigned int i; + + if (offset + 12 + 4 * TASDEVICE_MAXPROGRAM_NUM_KERNEL > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + fw_hdr->device_family = be16_to_cpup((__be16 *)&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s:not TAS device\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 2; + fw_hdr->device = be16_to_cpup((__be16 *)&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + offset = -EINVAL; + goto out; + } + offset += 2; + fw_hdr->ndev = deviceNumber[fw_hdr->device]; + + if (fw_hdr->ndev != tas_priv->ndev) { + dev_err(tas_priv->dev, + "%s: ndev(%u) in dspbin mismatch ndev(%u) in DTS\n", + __func__, fw_hdr->ndev, tas_priv->ndev); + offset = -EINVAL; + goto out; + } + + tas_fmw->nr_programs = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + + if (tas_fmw->nr_programs == 0 || tas_fmw->nr_programs > + TASDEVICE_MAXPROGRAM_NUM_KERNEL) { + dev_err(tas_priv->dev, "mnPrograms is invalid\n"); + offset = -EINVAL; + goto out; + } + + tas_fmw->programs = kcalloc(tas_fmw->nr_programs, + sizeof(struct tasdevice_prog), GFP_KERNEL); + if (!tas_fmw->programs) { + offset = -ENOMEM; + goto out; + } + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &(tas_fmw->programs[i]); + program->prog_size = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + } + + /* Skip the unused prog_size */ + offset += 4 * (TASDEVICE_MAXPROGRAM_NUM_KERNEL - tas_fmw->nr_programs); + + tas_fmw->nr_configurations = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + + /* The max number of config in firmware greater than 4 pieces of + * tas2781s is different from the one lower than 4 pieces of + * tas2781s. + */ + max_confs = (fw_hdr->ndev >= 4) ? + TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS : + TASDEVICE_MAXCONFIG_NUM_KERNEL; + if (tas_fmw->nr_configurations == 0 || + tas_fmw->nr_configurations > max_confs) { + dev_err(tas_priv->dev, "%s: Conf is invalid\n", __func__); + offset = -EINVAL; + goto out; + } + + if (offset + 4 * max_confs > fmw->size) { + dev_err(tas_priv->dev, "%s: mpConfigurations err\n", __func__); + offset = -EINVAL; + goto out; + } + + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(struct tasdevice_config), GFP_KERNEL); + if (!tas_fmw->configs) { + offset = -ENOMEM; + goto out; + } + + for (i = 0; i < tas_fmw->nr_programs; i++) { + config = &(tas_fmw->configs[i]); + config->cfg_size = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + } + + /* Skip the unused configs */ + offset += 4 * (max_confs - tas_fmw->nr_programs); + +out: + return offset; +} + +static int tasdevice_process_block(void *context, unsigned char *data, + unsigned char dev_idx, int sublocksize) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; + int subblk_offset, chn, chnend, rc; + unsigned char subblk_typ = data[1]; + int blktyp = dev_idx & 0xC0; + int idx = dev_idx & 0x3F; + bool is_err = false; + + if (idx) { + chn = idx - 1; + chnend = idx; + } else { + chn = 0; + chnend = tas_priv->ndev; + } + + for (; chn < chnend; chn++) { + if (tas_priv->tasdevice[chn].is_loading == false) + continue; + + is_err = false; + subblk_offset = 2; + switch (subblk_typ) { + case TASDEVICE_CMD_SING_W: { + int i; + unsigned short len = be16_to_cpup((__be16 *)&data[2]); + + subblk_offset += 2; + if (subblk_offset + 4 * len > sublocksize) { + dev_err(tas_priv->dev, + "process_block: Out of boundary\n"); + is_err = true; + break; + } + + for (i = 0; i < len; i++) { + rc = tasdevice_dev_write(tas_priv, chn, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + data[subblk_offset + 3]); + if (rc < 0) { + is_err = true; + dev_err(tas_priv->dev, + "process_block: single write error\n"); + } + subblk_offset += 4; + } + } + break; + case TASDEVICE_CMD_BURST: { + unsigned short len = be16_to_cpup((__be16 *)&data[2]); + + subblk_offset += 2; + if (subblk_offset + 4 + len > sublocksize) { + dev_err(tas_priv->dev, + "%s: BST Out of boundary\n", + __func__); + is_err = true; + break; + } + if (len % 4) { + dev_err(tas_priv->dev, + "%s:Bst-len(%u)not div by 4\n", + __func__, len); + break; + } + + rc = tasdevice_dev_bulk_write(tas_priv, chn, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + &(data[subblk_offset + 4]), len); + if (rc < 0) { + is_err = true; + dev_err(tas_priv->dev, + "%s: bulk_write error = %d\n", + __func__, rc); + } + subblk_offset += (len + 4); + } + break; + case TASDEVICE_CMD_DELAY: { + unsigned int sleep_time = 0; + + if (subblk_offset + 2 > sublocksize) { + dev_err(tas_priv->dev, + "%s: delay Out of boundary\n", + __func__); + is_err = true; + break; + } + sleep_time = be16_to_cpup((__be16 *)&data[2]) * 1000; + usleep_range(sleep_time, sleep_time + 50); + subblk_offset += 2; + } + break; + case TASDEVICE_CMD_FIELD_W: + if (subblk_offset + 6 > sublocksize) { + dev_err(tas_priv->dev, + "%s: bit write Out of boundary\n", + __func__); + is_err = true; + break; + } + rc = tasdevice_dev_update_bits(tas_priv, chn, + TASDEVICE_REG(data[subblk_offset + 2], + data[subblk_offset + 3], + data[subblk_offset + 4]), + data[subblk_offset + 1], + data[subblk_offset + 5]); + if (rc < 0) { + is_err = true; + dev_err(tas_priv->dev, + "%s: update_bits error = %d\n", + __func__, rc); + } + subblk_offset += 6; + break; + default: + break; + } + if (is_err == true && blktyp != 0) { + if (blktyp == 0x80) { + tas_priv->tasdevice[chn].cur_prog = -1; + tas_priv->tasdevice[chn].cur_conf = -1; + } else + tas_priv->tasdevice[chn].cur_conf = -1; + } + } + + return subblk_offset; +} + +void tasdevice_select_cfg_blk(void *pContext, int conf_no, + unsigned char block_type) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) pContext; + struct tasdevice_rca *rca = &(tas_priv->rcabin); + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdev_blk_data **blk_data; + int j, k, chn, chnend; + + if (conf_no >= rca->ncfgs || conf_no < 0 || !cfg_info) { + dev_err(tas_priv->dev, "conf_no should be not more than %u\n", + rca->ncfgs); + return; + } + blk_data = cfg_info[conf_no]->blk_data; + + for (j = 0; j < (int)cfg_info[conf_no]->real_nblocks; j++) { + unsigned int length = 0, rc = 0; + + if (block_type > 5 || block_type < 2) { + dev_err(tas_priv->dev, + "block_type should be in range from 2 to 5\n"); + break; + } + if (block_type != blk_data[j]->block_type) + continue; + + for (k = 0; k < (int)blk_data[j]->n_subblks; k++) { + if (blk_data[j]->dev_idx) { + chn = blk_data[j]->dev_idx - 1; + chnend = blk_data[j]->dev_idx; + } else { + chn = 0; + chnend = tas_priv->ndev; + } + for (; chn < chnend; chn++) + tas_priv->tasdevice[chn].is_loading = true; + + rc = tasdevice_process_block(tas_priv, + blk_data[j]->regdata + length, + blk_data[j]->dev_idx, + blk_data[j]->block_size - length); + length += rc; + if (blk_data[j]->block_size < length) { + dev_err(tas_priv->dev, + "%s: %u %u out of boundary\n", + __func__, length, + blk_data[j]->block_size); + break; + } + } + if (length != blk_data[j]->block_size) + dev_err(tas_priv->dev, "%s: %u %u size is not same\n", + __func__, length, blk_data[j]->block_size); + } +} +EXPORT_SYMBOL_NS_GPL(tasdevice_select_cfg_blk, SND_SOC_TAS2781_FMWLIB); + +static int tasdevice_load_block_kernel( + struct tasdevice_priv *tasdevice, struct tasdev_blk *block) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &(tasdevice->fmw->fw_hdr); + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &(fw_hdr->fixed_hdr); + const unsigned int blk_size = block->blk_size; + unsigned int i, length; + unsigned char *data = block->data; + unsigned char dev_idx = 0; + + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + switch (block->type) { + case MAIN_ALL_DEVICES_1X: + dev_idx = 0x80; + break; + case MAIN_DEVICE_A_1X: + dev_idx = 0x81; + break; + case COEFF_DEVICE_A_1X: + case PRE_DEVICE_A_1X: + case PRE_SOFTWARE_RESET_DEVICE_A: + case POST_SOFTWARE_RESET_DEVICE_A: + dev_idx = 0xC1; + break; + case MAIN_DEVICE_B_1X: + dev_idx = 0x82; + break; + case COEFF_DEVICE_B_1X: + case PRE_DEVICE_B_1X: + case PRE_SOFTWARE_RESET_DEVICE_B: + case POST_SOFTWARE_RESET_DEVICE_B: + dev_idx = 0xC2; + break; + case MAIN_DEVICE_C_1X: + dev_idx = 0x83; + break; + case COEFF_DEVICE_C_1X: + case PRE_DEVICE_C_1X: + case PRE_SOFTWARE_RESET_DEVICE_C: + case POST_SOFTWARE_RESET_DEVICE_C: + dev_idx = 0xC3; + break; + case MAIN_DEVICE_D_1X: + dev_idx = 0x84; + break; + case COEFF_DEVICE_D_1X: + case PRE_DEVICE_D_1X: + case PRE_SOFTWARE_RESET_DEVICE_D: + case POST_SOFTWARE_RESET_DEVICE_D: + dev_idx = 0xC4; + break; + default: + dev_info(tasdevice->dev, + "%s: load block: Other Type = 0x%02x\n", + __func__, block->type); + break; + } + } else if (fw_fixed_hdr->ppcver >= + PPC3_VERSION) { + switch (block->type) { + case MAIN_ALL_DEVICES_1X: + dev_idx = 0x80; + break; + case MAIN_DEVICE_A_1X: + dev_idx = 0x81; + break; + case COEFF_DEVICE_A_1X: + case PRE_DEVICE_A_1X: + dev_idx = 0xC1; + break; + case MAIN_DEVICE_B_1X: + dev_idx = 0x82; + break; + case COEFF_DEVICE_B_1X: + case PRE_DEVICE_B_1X: + dev_idx = 0xC2; + break; + case MAIN_DEVICE_C_1X: + dev_idx = 0x83; + break; + case COEFF_DEVICE_C_1X: + case PRE_DEVICE_C_1X: + dev_idx = 0xC3; + break; + case MAIN_DEVICE_D_1X: + dev_idx = 0x84; + break; + case COEFF_DEVICE_D_1X: + case PRE_DEVICE_D_1X: + dev_idx = 0xC4; + break; + default: + dev_info(tasdevice->dev, + "%s: load block: Other Type = 0x%02x\n", + __func__, block->type); + break; + } + } else { + switch (block->type) { + case MAIN_ALL_DEVICES: + dev_idx = 0|0x80; + break; + case MAIN_DEVICE_A: + dev_idx = 0x81; + break; + case COEFF_DEVICE_A: + case PRE_DEVICE_A: + dev_idx = 0xC1; + break; + case MAIN_DEVICE_B: + dev_idx = 0x82; + break; + case COEFF_DEVICE_B: + case PRE_DEVICE_B: + dev_idx = 0xC2; + break; + case MAIN_DEVICE_C: + dev_idx = 0x83; + break; + case COEFF_DEVICE_C: + case PRE_DEVICE_C: + dev_idx = 0xC3; + break; + case MAIN_DEVICE_D: + dev_idx = 0x84; + break; + case COEFF_DEVICE_D: + case PRE_DEVICE_D: + dev_idx = 0xC4; + break; + default: + dev_info(tasdevice->dev, + "%s: load block: Other Type = 0x%02x\n", + __func__, block->type); + break; + } + } + + for (i = 0, length = 0; i < block->nr_subblocks; i++) { + int rc = tasdevice_process_block(tasdevice, data + length, + dev_idx, blk_size - length); + if (rc < 0) { + dev_err(tasdevice->dev, + "%s: %u %u sublock write error\n", + __func__, length, blk_size); + break; + } + length += (unsigned int)rc; + if (blk_size < length) { + dev_err(tasdevice->dev, "%s: %u %u out of boundary\n", + __func__, length, blk_size); + break; + } + } + + return 0; +} + +static int fw_parse_variable_hdr(struct tasdevice_priv + *tas_priv, struct tasdevice_dspfw_hdr *fw_hdr, + const struct firmware *fmw, int offset) +{ + const unsigned char *buf = fmw->data; + int len = strlen((char *)&buf[offset]); + + len++; + + if (offset + len + 8 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + + offset += len; + + fw_hdr->device_family = be32_to_cpup((__be32 *)&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s: not TAS device\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 4; + + fw_hdr->device = be32_to_cpup((__be32 *)&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + offset = -EINVAL; + goto out; + } + offset += 4; + fw_hdr->ndev = deviceNumber[fw_hdr->device]; + +out: + return offset; +} + +static int fw_parse_variable_header_git(struct tasdevice_priv + *tas_priv, const struct firmware *fmw, int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + + offset = fw_parse_variable_hdr(tas_priv, fw_hdr, fmw, offset); + if (offset < 0) + goto out; + if (fw_hdr->ndev != tas_priv->ndev) { + dev_err(tas_priv->dev, + "%s: ndev(%u) in dspbin mismatch ndev(%u) in DTS\n", + __func__, fw_hdr->ndev, tas_priv->ndev); + offset = -EINVAL; + } + +out: + return offset; +} + +static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + int n; + + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Type error\n", __func__); + offset = -EINVAL; + goto out; + } + block->type = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + if (tas_fmw->fw_hdr.fixed_hdr.drv_ver >= PPC_DRIVER_CRCCHK) { + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "PChkSumPresent error\n"); + offset = -EINVAL; + goto out; + } + block->is_pchksum_present = data[offset]; + offset++; + + block->pchksum = data[offset]; + offset++; + + block->is_ychksum_present = data[offset]; + offset++; + + block->ychksum = data[offset]; + offset++; + } else { + block->is_pchksum_present = 0; + block->is_ychksum_present = 0; + } + + block->nr_cmds = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + n = block->nr_cmds * 4; + if (offset + n > fmw->size) { + dev_err(tas_fmw->dev, + "%s: File Size(%lu) error offset = %d n = %d\n", + __func__, (unsigned long)fmw->size, offset, n); + offset = -EINVAL; + goto out; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], n, GFP_KERNEL); + if (!block->data) { + offset = -ENOMEM; + goto out; + } + offset += n; + +out: + return offset; +} + +/* When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_data(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = (unsigned char *)fmw->data; + struct tasdev_blk *blk; + unsigned int i; + int n; + + if (offset + 64 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Name error\n", __func__); + offset = -EINVAL; + goto out; + } + memcpy(img_data->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n++; + if (offset + n + 2 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Description error\n", __func__); + offset = -EINVAL; + goto out; + } + offset += n; + img_data->nr_blk = be16_to_cpup((__be16 *)&data[offset]); + offset += 2; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(struct tasdev_blk), GFP_KERNEL); + if (!img_data->dev_blks) { + offset = -ENOMEM; + goto out; + } + for (i = 0; i < img_data->nr_blk; i++) { + blk = &(img_data->dev_blks[i]); + offset = fw_parse_block_data(tas_fmw, blk, fmw, offset); + if (offset < 0) { + offset = -EINVAL; + goto out; + } + } + +out: + return offset; +} + +/* When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_program_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + unsigned char *buf = (unsigned char *)fmw->data; + struct tasdevice_prog *program; + int i; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + tas_fmw->nr_programs = be16_to_cpup((__be16 *)&buf[offset]); + offset += 2; + + if (tas_fmw->nr_programs == 0) { + /*Not error in calibration Data file, return directly*/ + dev_info(tas_priv->dev, "%s: No Programs data, maybe calbin\n", + __func__); + goto out; + } + + tas_fmw->programs = + kcalloc(tas_fmw->nr_programs, sizeof(struct tasdevice_prog), + GFP_KERNEL); + if (!tas_fmw->programs) { + offset = -ENOMEM; + goto out; + } + for (i = 0; i < tas_fmw->nr_programs; i++) { + int n = 0; + + program = &(tas_fmw->programs[i]); + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 64; + + n = strlen((char *)&buf[offset]); + /* skip '\0' and 5 unused bytes */ + n += 6; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + offset = -EINVAL; + goto out; + } + + offset += n; + + offset = fw_parse_data(tas_fmw, &(program->dev_data), fmw, + offset); + if (offset < 0) + goto out; + } + +out: + return offset; +} + +/* When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_configuration_data( + struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + struct tasdevice_config *config; + unsigned int i; + int n; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + tas_fmw->nr_configurations = be16_to_cpup((__be16 *)&data[offset]); + offset += 2; + + if (tas_fmw->nr_configurations == 0) { + dev_err(tas_priv->dev, "%s: Conf is zero\n", __func__); + /*Not error for calibration Data file, return directly*/ + goto out; + } + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(struct tasdevice_config), GFP_KERNEL); + if (!tas_fmw->configs) { + offset = -ENOMEM; + goto out; + } + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &(tas_fmw->configs[i]); + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "File Size err\n"); + offset = -EINVAL; + goto out; + } + memcpy(config->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n += 15; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + offset = -EINVAL; + goto out; + } + + offset += n; + + offset = fw_parse_data(tas_fmw, &(config->dev_data), + fmw, offset); + if (offset < 0) + goto out; + } + +out: + return offset; +} + +static bool check_inpage_yram_rg(struct tas_crc *cd, + unsigned char reg, unsigned char len) +{ + bool in = false; + + + if (reg <= TAS2781_YRAM5_END_REG && + reg >= TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_END_REG) + cd->len = TAS2781_YRAM5_END_REG - reg + 1; + else + cd->len = len; + cd->offset = reg; + in = true; + } else if (reg < TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_START_REG) { + cd->offset = TAS2781_YRAM5_START_REG; + cd->len = len - TAS2781_YRAM5_START_REG + reg; + in = true; + } + } + + return in; +} + +static bool check_inpage_yram_bk1(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (page == TAS2781_YRAM1_PAGE) { + if (reg >= TAS2781_YRAM1_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg + len > TAS2781_YRAM1_START_REG) { + cd->offset = TAS2781_YRAM1_START_REG; + cd->len = len - TAS2781_YRAM1_START_REG + reg; + in = true; + } + } else if (page == TAS2781_YRAM3_PAGE) + in = check_inpage_yram_rg(cd, reg, len); + + return in; +} + +/* Return Code: + * true -- the registers are in the inpage yram + * false -- the registers are NOT in the inpage yram + */ +static bool check_inpage_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1) { + in = check_inpage_yram_bk1(cd, page, reg, len); + goto end; + } + if (book == TAS2781_YRAM_BOOK2 && page == TAS2781_YRAM5_PAGE) + in = check_inpage_yram_rg(cd, reg, len); + +end: + return in; +} + +static bool check_inblock_yram_bk(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if ((page >= TAS2781_YRAM4_START_PAGE && + page <= TAS2781_YRAM4_END_PAGE) || + (page >= TAS2781_YRAM2_START_PAGE && + page <= TAS2781_YRAM2_END_PAGE)) { + if (reg <= TAS2781_YRAM2_END_REG && + reg >= TAS2781_YRAM2_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg < TAS2781_YRAM2_START_REG) { + if (reg + len - 1 >= TAS2781_YRAM2_START_REG) { + cd->offset = TAS2781_YRAM2_START_REG; + cd->len = reg + len - TAS2781_YRAM2_START_REG; + in = true; + } + } + } + + return in; +} + +/* Return Code: + * true -- the registers are in the inblock yram + * false -- the registers are NOT in the inblock yram + */ +static bool check_inblock_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1 || book == TAS2781_YRAM_BOOK2) + in = check_inblock_yram_bk(cd, page, reg, len); + + return in; +} + +static bool check_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in; + + in = check_inpage_yram(cd, book, page, reg, len); + if (in) + goto end; + in = check_inblock_yram(cd, book, page, reg, len); + +end: + return in; +} + +static int tasdev_multibytes_chksum(struct tasdevice_priv *tasdevice, + unsigned short chn, unsigned char book, unsigned char page, + unsigned char reg, unsigned int len) +{ + struct tas_crc crc_data; + unsigned char crc_chksum = 0; + unsigned char nBuf1[128]; + int ret = 0; + int i; + bool in; + + if ((reg + len - 1) > 127) { + ret = -EINVAL; + dev_err(tasdevice->dev, "firmware error\n"); + goto end; + } + + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (reg == TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG)) + && (len == 4)) { + /*DSP swap command, pass */ + ret = 0; + goto end; + } + + in = check_yram(&crc_data, book, page, reg, len); + if (!in) + goto end; + + if (len == 1) { + dev_err(tasdevice->dev, "firmware error\n"); + ret = -EINVAL; + goto end; + } + + ret = tasdevice_dev_bulk_read(tasdevice, chn, + TASDEVICE_REG(book, page, crc_data.offset), + nBuf1, crc_data.len); + if (ret < 0) + goto end; + + for (i = 0; i < crc_data.len; i++) { + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (page == TASDEVICE_PAGE_ID( + TAS2781_SA_COEFF_SWAP_REG)) + && ((i + crc_data.offset) + >= TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG)) + && ((i + crc_data.offset) + <= (TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG) + + 4))) + /*DSP swap command, bypass */ + continue; + else + crc_chksum += crc8(tasdevice->crc8_lkp_tbl, &nBuf1[i], + 1, 0); + } + + ret = crc_chksum; + +end: + return ret; +} + +static int do_singlereg_checksum(struct tasdevice_priv *tasdevice, + unsigned short chl, unsigned char book, unsigned char page, + unsigned char reg, unsigned char val) +{ + struct tas_crc crc_data; + unsigned int nData1; + int ret = 0; + bool in; + + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (reg >= TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG)) + && (reg <= (TASDEVICE_PAGE_REG( + TAS2781_SA_COEFF_SWAP_REG) + 4))) { + /*DSP swap command, pass */ + ret = 0; + goto end; + } + + in = check_yram(&crc_data, book, page, reg, 1); + if (!in) + goto end; + ret = tasdevice_dev_read(tasdevice, chl, + TASDEVICE_REG(book, page, reg), &nData1); + if (ret < 0) + goto end; + + if (nData1 != val) { + dev_err(tasdevice->dev, + "B[0x%x]P[0x%x]R[0x%x] W[0x%x], R[0x%x]\n", + book, page, reg, val, nData1); + tasdevice->tasdevice[chl].err_code |= ERROR_YRAM_CRCCHK; + ret = -EAGAIN; + goto end; + } + + ret = crc8(tasdevice->crc8_lkp_tbl, &val, 1, 0); + +end: + return ret; +} + +static void set_err_prg_cfg(unsigned int type, struct tasdevice *dev) +{ + if ((type == MAIN_ALL_DEVICES) || (type == MAIN_DEVICE_A) + || (type == MAIN_DEVICE_B) || (type == MAIN_DEVICE_C) + || (type == MAIN_DEVICE_D)) + dev->cur_prog = -1; + else + dev->cur_conf = -1; +} + +static int tasdev_bytes_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, int chn, unsigned char book, + unsigned char page, unsigned char reg, unsigned int len, + unsigned char val, unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) + ret = tasdev_multibytes_chksum(tas_priv, chn, book, page, reg, + len); + else + ret = do_singlereg_checksum(tas_priv, chn, book, page, reg, + val); + + if (ret > 0) { + *crc_chksum += (unsigned char)ret; + goto end; + } + + if (ret != -EAGAIN) + goto end; + + block->nr_retry--; + if (block->nr_retry > 0) + goto end; + + set_err_prg_cfg(block->type, &tas_priv->tasdevice[chn]); + +end: + return ret; +} + +static int tasdev_multibytes_wr(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, int chn, unsigned char book, + unsigned char page, unsigned char reg, unsigned char *data, + unsigned int len, unsigned int *nr_cmds, + unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) { + ret = tasdevice_dev_bulk_write(tas_priv, chn, + TASDEVICE_REG(book, page, reg), data + 3, len); + if (ret < 0) + goto end; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, chn, + book, page, reg, len, 0, crc_chksum); + } else { + ret = tasdevice_dev_write(tas_priv, chn, + TASDEVICE_REG(book, page, reg), data[3]); + if (ret < 0) + goto end; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, chn, book, + page, reg, 1, data[3], crc_chksum); + } + + if (!block->is_ychksum_present || ret >= 0) { + *nr_cmds += 1; + if (len >= 2) + *nr_cmds += ((len - 2) / 4) + 1; + } + +end: + return ret; +} + +static int tasdev_block_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, int chn) +{ + unsigned int nr_value; + int ret; + + ret = tasdevice_dev_read(tas_priv, chn, TASDEVICE_I2CChecksum, + &nr_value); + if (ret < 0) { + dev_err(tas_priv->dev, "%s: Chn %d\n", __func__, chn); + set_err_prg_cfg(block->type, &tas_priv->tasdevice[chn]); + goto end; + } + + if ((nr_value & 0xff) != block->pchksum) { + dev_err(tas_priv->dev, "%s: Blk PChkSum Chn %d ", __func__, + chn); + dev_err(tas_priv->dev, "PChkSum = 0x%x, Reg = 0x%x\n", + block->pchksum, (nr_value & 0xff)); + tas_priv->tasdevice[chn].err_code |= ERROR_PRAM_CRCCHK; + ret = -EAGAIN; + block->nr_retry--; + + if (block->nr_retry <= 0) + set_err_prg_cfg(block->type, + &tas_priv->tasdevice[chn]); + } else + tas_priv->tasdevice[chn].err_code &= ~ERROR_PRAM_CRCCHK; + +end: + return ret; +} + +static int tasdev_load_blk(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, int chn) +{ + unsigned int sleep_time; + unsigned int len; + unsigned int nr_cmds; + unsigned char *data = block->data; + unsigned char crc_chksum = 0; + unsigned char offset; + unsigned char book; + unsigned char page; + unsigned char val; + int ret = 0; + + while (block->nr_retry > 0) { + if (block->is_pchksum_present) { + ret = tasdevice_dev_write(tas_priv, chn, + TASDEVICE_I2CChecksum, 0); + if (ret < 0) + break; + } + + if (block->is_ychksum_present) + crc_chksum = 0; + + nr_cmds = 0; + + while (nr_cmds < block->nr_cmds) { + data = block->data + nr_cmds * 4; + + book = data[0]; + page = data[1]; + offset = data[2]; + val = data[3]; + + nr_cmds++; + /*Single byte write*/ + if (offset <= 0x7F) { + ret = tasdevice_dev_write(tas_priv, chn, + TASDEVICE_REG(book, page, offset), + val); + if (ret < 0) + goto end; + if (block->is_ychksum_present) { + ret = tasdev_bytes_chksum(tas_priv, + block, chn, book, page, offset, + 1, val, &crc_chksum); + if (ret < 0) + break; + } + continue; + } + /*sleep command*/ + if (offset == 0x81) { + /*book -- data[0] page -- data[1]*/ + sleep_time = ((book << 8) + page)*1000; + usleep_range(sleep_time, sleep_time + 50); + continue; + } + /*Multiple bytes write*/ + if (offset == 0x85) { + data += 4; + len = (book << 8) + page; + book = data[0]; + page = data[1]; + offset = data[2]; + ret = tasdev_multibytes_wr(tas_priv, + block, chn, book, page, offset, data, + len, &nr_cmds, &crc_chksum); + if (ret < 0) + break; + } + } + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) /*err in current device, skip it*/ + break; + + if (block->is_pchksum_present) { + ret = tasdev_block_chksum(tas_priv, block, chn); + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) /*err in current device, skip it*/ + break; + } + + if (block->is_ychksum_present) { + /* TBD, open it when FW ready */ + dev_err(tas_priv->dev, + "Blk YChkSum: FW = 0x%x, YCRC = 0x%x\n", + block->ychksum, crc_chksum); + + tas_priv->tasdevice[chn].err_code &= + ~ERROR_YRAM_CRCCHK; + ret = 0; + } + /*skip current blk*/ + break; + } + +end: + return ret; +} + +static int tasdevice_load_block(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + int chnend = 0; + int ret = 0; + int chn = 0; + int rc = 0; + + switch (block->type) { + case MAIN_ALL_DEVICES: + chn = 0; + chnend = tas_priv->ndev; + break; + case MAIN_DEVICE_A: + case COEFF_DEVICE_A: + case PRE_DEVICE_A: + chn = 0; + chnend = 1; + break; + case MAIN_DEVICE_B: + case COEFF_DEVICE_B: + case PRE_DEVICE_B: + chn = 1; + chnend = 2; + break; + case MAIN_DEVICE_C: + case COEFF_DEVICE_C: + case PRE_DEVICE_C: + chn = 2; + chnend = 3; + break; + case MAIN_DEVICE_D: + case COEFF_DEVICE_D: + case PRE_DEVICE_D: + chn = 3; + chnend = 4; + break; + default: + dev_dbg(tas_priv->dev, "load blk: Other Type = 0x%02x\n", + block->type); + break; + } + + for (; chn < chnend; chn++) { + block->nr_retry = 6; + if (tas_priv->tasdevice[chn].is_loading == false) + continue; + ret = tasdev_load_blk(tas_priv, block, chn); + if (ret < 0) + dev_err(tas_priv->dev, "dev %d, Blk (%d) load error\n", + chn, block->type); + rc |= ret; + } + + return rc; +} + +static int dspfw_default_callback(struct tasdevice_priv *tas_priv, + unsigned int drv_ver, unsigned int ppcver) +{ + int rc = 0; + + if (drv_ver == 0x100) { + if (ppcver >= PPC3_VERSION) { + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + } else { + switch (ppcver) { + case 0x00: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + break; + default: + dev_err(tas_priv->dev, + "%s: PPCVer must be 0x0 or 0x%02x", + __func__, PPC3_VERSION); + dev_err(tas_priv->dev, " Current:0x%02x\n", + ppcver); + rc = -EINVAL; + break; + } + } + } else { + dev_err(tas_priv->dev, + "DrvVer must be 0x0, 0x230 or above 0x230 "); + dev_err(tas_priv->dev, "current is 0x%02x\n", drv_ver); + rc = -EINVAL; + } + + return rc; +} + +static int load_calib_data(struct tasdevice_priv *tas_priv, + struct tasdevice_data *dev_data) +{ + struct tasdev_blk *block; + unsigned int i; + int ret = 0; + + for (i = 0; i < dev_data->nr_blk; i++) { + block = &(dev_data->dev_blks[i]); + ret = tasdevice_load_block(tas_priv, block); + if (ret < 0) + break; + } + + return ret; +} + +static int fw_parse_header(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &(fw_hdr->fixed_hdr); + const unsigned char magic_number[] = { 0x35, 0x35, 0x35, 0x32 }; + const unsigned char *buf = (unsigned char *)fmw->data; + + if (offset + 92 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + if (memcmp(&buf[offset], magic_number, 4)) { + dev_err(tas_priv->dev, "%s: Magic num NOT match\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 4; + + /* Convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + fw_fixed_hdr->fwsize = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + if (fw_fixed_hdr->fwsize != fmw->size) { + dev_err(tas_priv->dev, "File size not match, %lu %u", + (unsigned long)fmw->size, fw_fixed_hdr->fwsize); + offset = -EINVAL; + goto out; + } + offset += 4; + fw_fixed_hdr->ppcver = be32_to_cpup((__be32 *)&buf[offset]); + offset += 8; + fw_fixed_hdr->drv_ver = be32_to_cpup((__be32 *)&buf[offset]); + offset += 72; + + out: + return offset; +} + +static int fw_parse_variable_hdr_cal(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + + offset = fw_parse_variable_hdr(tas_priv, fw_hdr, fmw, offset); + if (offset < 0) + goto out; + if (fw_hdr->ndev != 1) { + dev_err(tas_priv->dev, + "%s: calbin must be 1, but currently ndev(%u)\n", + __func__, fw_hdr->ndev); + offset = -EINVAL; + } + +out: + return offset; +} + +/* When calibrated data parsing error occurs, DSP can still work with default + * calibrated data, memory resource related to calibrated data will be + * released in the tasdevice_codec_remove. + */ +static int fw_parse_calibration_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_calibration *calibration; + unsigned char *data = (unsigned char *)fmw->data; + unsigned int i, n; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: Calibrations error\n", __func__); + offset = -EINVAL; + goto out; + } + tas_fmw->nr_calibrations = be16_to_cpup((__be16 *)&data[offset]); + offset += 2; + + if (tas_fmw->nr_calibrations != 1) { + dev_err(tas_priv->dev, + "%s: only support one calibraiton(%d)!\n", + __func__, tas_fmw->nr_calibrations); + goto out; + } + + tas_fmw->calibrations = kcalloc(tas_fmw->nr_calibrations, + sizeof(struct tasdevice_calibration), GFP_KERNEL); + if (!tas_fmw->calibrations) { + offset = -ENOMEM; + goto out; + } + for (i = 0; i < tas_fmw->nr_calibrations; i++) { + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "Calibrations error\n"); + offset = -EINVAL; + goto out; + } + calibration = &(tas_fmw->calibrations[i]); + offset += 64; + + n = strlen((char *)&data[offset]); + /* skip '\0' and 2 unused bytes */ + n += 3; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + offset = -EINVAL; + goto out; + } + offset += n; + + offset = fw_parse_data(tas_fmw, &(calibration->dev_data), fmw, + offset); + if (offset < 0) + goto out; + } + +out: + return offset; +} + +int tas2781_load_calibration(void *context, char *file_name, + unsigned short i) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; + struct tasdevice *tasdev = &(tas_priv->tasdevice[i]); + const struct firmware *fw_entry; + struct tasdevice_fw *tas_fmw; + struct firmware fmw; + int offset = 0; + int ret; + + ret = request_firmware(&fw_entry, file_name, tas_priv->dev); + if (ret) { + dev_err(tas_priv->dev, "%s: Request firmware %s failed\n", + __func__, file_name); + goto out; + } + + if (!fw_entry->size) { + dev_err(tas_priv->dev, "%s: file read error: size = %lu\n", + __func__, (unsigned long)fw_entry->size); + goto out; + } + fmw.size = fw_entry->size; + fmw.data = fw_entry->data; + + tas_fmw = tasdev->cali_data_fmw = kzalloc(sizeof(struct tasdevice_fw), + GFP_KERNEL); + if (!tasdev->cali_data_fmw) { + ret = -ENOMEM; + goto out; + } + tas_fmw->dev = tas_priv->dev; + offset = fw_parse_header(tas_priv, tas_fmw, &fmw, offset); + if (offset == -EINVAL) { + dev_err(tas_priv->dev, "fw_parse_header EXIT!\n"); + ret = offset; + goto out; + } + offset = fw_parse_variable_hdr_cal(tas_priv, tas_fmw, &fmw, offset); + if (offset == -EINVAL) { + dev_err(tas_priv->dev, + "%s: fw_parse_variable_header_cal EXIT!\n", __func__); + ret = offset; + goto out; + } + offset = fw_parse_program_data(tas_priv, tas_fmw, &fmw, offset); + if (offset < 0) { + dev_err(tas_priv->dev, "fw_parse_program_data EXIT!\n"); + ret = offset; + goto out; + } + offset = fw_parse_configuration_data(tas_priv, tas_fmw, &fmw, offset); + if (offset < 0) { + dev_err(tas_priv->dev, "fw_parse_configuration_data EXIT!\n"); + ret = offset; + goto out; + } + offset = fw_parse_calibration_data(tas_priv, tas_fmw, &fmw, offset); + if (offset < 0) { + dev_err(tas_priv->dev, "fw_parse_calibration_data EXIT!\n"); + ret = offset; + goto out; + } + +out: + if (fw_entry) + release_firmware(fw_entry); + + return ret; +} +EXPORT_SYMBOL_NS_GPL(tas2781_load_calibration, SND_SOC_TAS2781_FMWLIB); + +static int tasdevice_dspfw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr; + struct tasdevice_fw *tas_fmw; + int offset = 0; + int ret = 0; + + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "%s: Failed to read firmware %s\n", + __func__, tas_priv->coef_binaryname); + ret = -EINVAL; + goto out; + } + + tas_priv->fmw = kzalloc(sizeof(struct tasdevice_fw), GFP_KERNEL); + if (!tas_priv->fmw) { + ret = -ENOMEM; + goto out; + } + tas_fmw = tas_priv->fmw; + tas_fmw->dev = tas_priv->dev; + offset = fw_parse_header(tas_priv, tas_fmw, fmw, offset); + + if (offset == -EINVAL) { + ret = -EINVAL; + goto out; + } + fw_fixed_hdr = &(tas_fmw->fw_hdr.fixed_hdr); + /* Support different versions of firmware */ + switch (fw_fixed_hdr->drv_ver) { + case 0x301: + case 0x302: + case 0x502: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + break; + case 0x202: + case 0x400: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + break; + default: + ret = dspfw_default_callback(tas_priv, + fw_fixed_hdr->drv_ver, fw_fixed_hdr->ppcver); + if (ret) + goto out; + break; + } + + offset = tas_priv->fw_parse_variable_header(tas_priv, fmw, offset); + if (offset < 0) { + ret = offset; + goto out; + } + offset = tas_priv->fw_parse_program_data(tas_priv, tas_fmw, fmw, + offset); + if (offset < 0) { + ret = offset; + goto out; + } + offset = tas_priv->fw_parse_configuration_data(tas_priv, + tas_fmw, fmw, offset); + if (offset < 0) + ret = offset; + +out: + return ret; +} + +int tasdevice_dsp_parser(void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; + const struct firmware *fw_entry; + int ret; + + ret = request_firmware(&fw_entry, tas_priv->coef_binaryname, + tas_priv->dev); + if (ret) { + dev_err(tas_priv->dev, "%s: load %s error\n", __func__, + tas_priv->coef_binaryname); + goto out; + } + + ret = tasdevice_dspfw_ready(fw_entry, tas_priv); + release_firmware(fw_entry); + fw_entry = NULL; + +out: + return ret; +} +EXPORT_SYMBOL_NS_GPL(tasdevice_dsp_parser, SND_SOC_TAS2781_FMWLIB); + +static void tas2781_clear_calfirmware(struct tasdevice_fw *tas_fmw) +{ + struct tasdevice_calibration *calibration; + struct tasdev_blk *block; + struct tasdevice_data *im; + unsigned int blks; + int i; + + if (!tas_fmw->calibrations) + goto out; + + for (i = 0; i < tas_fmw->nr_calibrations; i++) { + calibration = &(tas_fmw->calibrations[i]); + if (!calibration) + continue; + + im = &(calibration->dev_data); + + if (!im->dev_blks) + continue; + + for (blks = 0; blks < im->nr_blk; blks++) { + block = &(im->dev_blks[blks]); + if (!block) + continue; + kfree(block->data); + } + kfree(im->dev_blks); + } + kfree(tas_fmw->calibrations); +out: + kfree(tas_fmw); +} + +void tasdevice_calbin_remove(void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice *tasdev; + int i; + + if (!tas_priv) + return; + + for (i = 0; i < tas_priv->ndev; i++) { + tasdev = &(tas_priv->tasdevice[i]); + if (!tasdev->cali_data_fmw) + continue; + tas2781_clear_calfirmware(tasdev->cali_data_fmw); + tasdev->cali_data_fmw = NULL; + } +} +EXPORT_SYMBOL_NS_GPL(tasdevice_calbin_remove, SND_SOC_TAS2781_FMWLIB); + +void tasdevice_config_info_remove(void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_rca *rca = &(tas_priv->rcabin); + struct tasdevice_config_info **ci = rca->cfg_info; + int i, j; + + if (!ci) + return; + for (i = 0; i < rca->ncfgs; i++) { + if (!ci[i]) + continue; + if (ci[i]->blk_data) { + for (j = 0; j < (int)ci[i]->real_nblocks; j++) { + if (!ci[i]->blk_data[j]) + continue; + kfree(ci[i]->blk_data[j]->regdata); + kfree(ci[i]->blk_data[j]); + } + kfree(ci[i]->blk_data); + } + kfree(ci[i]); + } + kfree(ci); +} +EXPORT_SYMBOL_NS_GPL(tasdevice_config_info_remove, SND_SOC_TAS2781_FMWLIB); + +static int tasdevice_load_data(struct tasdevice_priv *tas_priv, + struct tasdevice_data *dev_data) +{ + struct tasdev_blk *block; + unsigned int i; + int ret = 0; + + for (i = 0; i < dev_data->nr_blk; i++) { + block = &(dev_data->dev_blks[i]); + ret = tas_priv->tasdevice_load_block(tas_priv, block); + if (ret < 0) + break; + } + + return ret; +} + +int tasdevice_select_tuningprm_cfg(void *context, int prm_no, + int cfg_no, int rca_conf_no) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_rca *rca = &(tas_priv->rcabin); + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + struct tasdevice_config *conf; + int prog_status = 0; + int status, i; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + goto out; + } + + if (cfg_no >= tas_fmw->nr_configurations) { + dev_err(tas_priv->dev, + "%s: cfg(%d) is not in range of conf %u\n", + __func__, cfg_no, tas_fmw->nr_configurations); + goto out; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + goto out; + } + + if (rca_conf_no >= rca->ncfgs || rca_conf_no < 0 || + !cfg_info) { + dev_err(tas_priv->dev, + "conf_no:%d should be in range from 0 to %u\n", + rca_conf_no, rca->ncfgs-1); + goto out; + } + + conf = &(tas_fmw->configs[cfg_no]); + for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { + if (cfg_info[rca_conf_no]->active_dev & (1 << i)) { + if (tas_priv->tasdevice[i].cur_prog != prm_no + || tas_priv->force_fwload_status) { + tas_priv->tasdevice[i].cur_conf = -1; + tas_priv->tasdevice[i].is_loading = true; + prog_status++; + } + } else + tas_priv->tasdevice[i].is_loading = false; + tas_priv->tasdevice[i].is_loaderr = false; + } + + if (prog_status) { + program = &(tas_fmw->programs[prm_no]); + tasdevice_load_data(tas_priv, &(program->dev_data)); + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].is_loaderr == true) + continue; + else if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) { + struct tasdevice_fw *cal_fmw = + tas_priv->tasdevice[i].cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration + *cal = cal_fmw->calibrations; + + if (cal) + load_calib_data(tas_priv, + &(cal->dev_data)); + } + tas_priv->tasdevice[i].cur_prog = prm_no; + } + } + } + + for (i = 0, status = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].cur_conf != cfg_no + && (cfg_info[rca_conf_no]->active_dev & (1 << i)) + && (tas_priv->tasdevice[i].is_loaderr == false)) { + status++; + tas_priv->tasdevice[i].is_loading = true; + } else + tas_priv->tasdevice[i].is_loading = false; + } + + if (status) { + status = 0; + tasdevice_load_data(tas_priv, &(conf->dev_data)); + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].is_loaderr == true) { + status |= 1 << (i + 4); + continue; + } else if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) + tas_priv->tasdevice[i].cur_conf = cfg_no; + } + } else + dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", + __func__, cfg_no); + + status |= cfg_info[rca_conf_no]->active_dev; + +out: + return prog_status; +} +EXPORT_SYMBOL_NS_GPL(tasdevice_select_tuningprm_cfg, + SND_SOC_TAS2781_FMWLIB); + +int tasdevice_prmg_load(void *context, int prm_no) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + int prog_status = 0; + int i; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + goto out; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + goto out; + } + + for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].cur_prog != prm_no) { + tas_priv->tasdevice[i].cur_conf = -1; + tas_priv->tasdevice[i].is_loading = true; + prog_status++; + } + } + + if (prog_status) { + program = &(tas_fmw->programs[prm_no]); + tasdevice_load_data(tas_priv, &(program->dev_data)); + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].is_loaderr == true) + continue; + else if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) + tas_priv->tasdevice[i].cur_prog = prm_no; + } + } + +out: + return prog_status; +} +EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_load, SND_SOC_TAS2781_FMWLIB); + +int tasdevice_prmg_calibdata_load(void *context, int prm_no) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + int prog_status = 0; + int i; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + goto out; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + goto out; + } + + for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].cur_prog != prm_no) { + tas_priv->tasdevice[i].cur_conf = -1; + tas_priv->tasdevice[i].is_loading = true; + prog_status++; + } + tas_priv->tasdevice[i].is_loaderr = false; + } + + if (prog_status) { + program = &(tas_fmw->programs[prm_no]); + tasdevice_load_data(tas_priv, &(program->dev_data)); + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].is_loaderr == true) + continue; + else if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) { + struct tasdevice_fw *cal_fmw = + tas_priv->tasdevice[i].cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration *cal = + cal_fmw->calibrations; + + if (cal) + load_calib_data(tas_priv, + &(cal->dev_data)); + } + tas_priv->tasdevice[i].cur_prog = prm_no; + } + } + } + +out: + return prog_status; +} +EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_calibdata_load, + SND_SOC_TAS2781_FMWLIB); + +void tasdevice_tuning_switch(void *context, int state) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + int profile_cfg_id = tas_priv->rcabin.profile_cfg_id; + + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + return; + } + + if (state == 0) { + if (tas_priv->cur_prog < tas_fmw->nr_programs) { + /*dsp mode or tuning mode*/ + profile_cfg_id = tas_priv->rcabin.profile_cfg_id; + tasdevice_select_tuningprm_cfg(tas_priv, + tas_priv->cur_prog, tas_priv->cur_conf, + profile_cfg_id); + } + + tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + } else + tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN); +} +EXPORT_SYMBOL_NS_GPL(tasdevice_tuning_switch, + SND_SOC_TAS2781_FMWLIB); + +MODULE_DESCRIPTION("Texas Firmware Support"); +MODULE_AUTHOR("Shenghao Ding, TI, <shenghao-ding(a)ti.com>"); +MODULE_LICENSE("GPL"); -- 2.34.1

1 0

[PATCH v5 1/4] ASoC: tas2781: Add Header file for tas2781 driver
by Shenghao Ding 10 Jun '23

10 Jun '23

Create Header file for tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v5: - remove /* __TAS2781_LIB_H__ */ at the end of the file - remove #include <linux/kernel.h> in the tas2781.h - remove enum channel in tas2781-dsp.h - Add TASDEVICE_MAX_CHANNELS in tas2781-dsp.h - remove global addr related definition in tas2781.h - reorder the member in tasdevice_priv - fix the firmware version compatibility issue --- include/sound/tas2781-dsp.h | 183 ++++++++++++++++++++++++++++++++++++ include/sound/tas2781-tlv.h | 21 +++++ include/sound/tas2781.h | 164 ++++++++++++++++++++++++++++++++ 3 files changed, 368 insertions(+) create mode 100644 include/sound/tas2781-dsp.h create mode 100644 include/sound/tas2781-tlv.h create mode 100644 include/sound/tas2781.h diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h new file mode 100644 index 000000000000..bd1b72bf47a5 --- /dev/null +++ b/include/sound/tas2781-dsp.h @@ -0,0 +1,183 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TASDEVICE_DSP_H__ +#define __TASDEVICE_DSP_H__ + +#define MAIN_ALL_DEVICES 0x0d +#define MAIN_DEVICE_A 0x01 +#define MAIN_DEVICE_B 0x08 +#define MAIN_DEVICE_C 0x10 +#define MAIN_DEVICE_D 0x14 +#define COEFF_DEVICE_A 0x03 +#define COEFF_DEVICE_B 0x0a +#define COEFF_DEVICE_C 0x11 +#define COEFF_DEVICE_D 0x15 +#define PRE_DEVICE_A 0x04 +#define PRE_DEVICE_B 0x0b +#define PRE_DEVICE_C 0x12 +#define PRE_DEVICE_D 0x16 + +#define PPC3_VERSION 0x4100 +#define PPC3_VERSION_TAS2781 0x14600 +#define TASDEVICE_DEVICE_SUM 8 +#define TASDEVICE_CONFIG_SUM 64 + +#define TASDEVICE_MAX_CHANNELS 8 + +enum tasdevice_dsp_dev_idx { + TASDEVICE_DSP_TAS_2555 = 0, + TASDEVICE_DSP_TAS_2555_STEREO, + TASDEVICE_DSP_TAS_2557_MONO, + TASDEVICE_DSP_TAS_2557_DUAL_MONO, + TASDEVICE_DSP_TAS_2559, + TASDEVICE_DSP_TAS_2563, + TASDEVICE_DSP_TAS_2563_DUAL_MONO = 7, + TASDEVICE_DSP_TAS_2563_QUAD, + TASDEVICE_DSP_TAS_2563_21, + TASDEVICE_DSP_TAS_2781, + TASDEVICE_DSP_TAS_2781_DUAL_MONO, + TASDEVICE_DSP_TAS_2781_21, + TASDEVICE_DSP_TAS_2781_QUAD, + TASDEVICE_DSP_TAS_MAX_DEVICE +}; + +struct tasdevice_fw_fixed_hdr { + unsigned int fwsize; + unsigned int ppcver; + unsigned int drv_ver; +}; + +struct tasdevice_dspfw_hdr { + struct tasdevice_fw_fixed_hdr fixed_hdr; + unsigned short device_family; + unsigned short device; + unsigned char ndev; +}; + +struct tasdev_blk { + int nr_retry; + unsigned int type; + unsigned char is_pchksum_present; + unsigned char pchksum; + unsigned char is_ychksum_present; + unsigned char ychksum; + unsigned int nr_cmds; + unsigned int blk_size; + unsigned int nr_subblocks; + unsigned char *data; +}; + +struct tasdevice_data { + char name[64]; + unsigned int nr_blk; + struct tasdev_blk *dev_blks; +}; + +struct tasdevice_prog { + unsigned int prog_size; + struct tasdevice_data dev_data; +}; + +struct tasdevice_config { + unsigned int cfg_size; + char name[64]; + struct tasdevice_data dev_data; +}; + +struct tasdevice_calibration { + struct tasdevice_data dev_data; +}; + +struct tasdevice_fw { + struct tasdevice_dspfw_hdr fw_hdr; + unsigned short nr_programs; + struct tasdevice_prog *programs; + unsigned short nr_configurations; + struct tasdevice_config *configs; + unsigned short nr_calibrations; + struct tasdevice_calibration *calibrations; + struct device *dev; +}; + +enum tasdevice_dsp_fw_state { + TASDEVICE_DSP_FW_NONE = 0, + TASDEVICE_DSP_FW_PENDING, + TASDEVICE_DSP_FW_FAIL, + TASDEVICE_DSP_FW_ALL_OK, +}; + +enum tasdevice_bin_blk_type { + TASDEVICE_BIN_BLK_COEFF = 1, + TASDEVICE_BIN_BLK_POST_POWER_UP, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN, + TASDEVICE_BIN_BLK_PRE_POWER_UP, + TASDEVICE_BIN_BLK_POST_SHUTDOWN +}; + +struct tasdevice_rca_hdr { + unsigned int img_sz; + unsigned int checksum; + unsigned int binary_version_num; + unsigned int drv_fw_version; + unsigned char plat_type; + unsigned char dev_family; + unsigned char reserve; + unsigned char ndev; + unsigned char devs[TASDEVICE_DEVICE_SUM]; + unsigned int nconfig; + unsigned int config_size[TASDEVICE_CONFIG_SUM]; +}; + +struct tasdev_blk_data { + unsigned char dev_idx; + unsigned char block_type; + unsigned short yram_checksum; + unsigned int block_size; + unsigned int n_subblks; + unsigned char *regdata; +}; + +struct tasdevice_config_info { + unsigned int nblocks; + unsigned int real_nblocks; + unsigned char active_dev; + struct tasdev_blk_data **blk_data; +}; + +struct tasdevice_rca { + struct tasdevice_rca_hdr fw_hdr; + int ncfgs; + struct tasdevice_config_info **cfg_info; + int profile_cfg_id; +}; + +void tasdevice_select_cfg_blk(void *context, int conf_no, + unsigned char block_type); +void tasdevice_config_info_remove(void *context); +void tasdevice_dsp_remove(void *context); +int tasdevice_dsp_parser(void *context); +int tasdevice_rca_parser(void *context, const struct firmware *fmw); +void tasdevice_dsp_remove(void *context); +void tasdevice_calbin_remove(void *context); +int tasdevice_select_tuningprm_cfg(void *context, int prm, + int cfg_no, int rca_conf_no); +int tasdevice_prmg_load(void *context, int prm_no); +int tasdevice_prmg_calibdata_load(void *context, int prm_no); +void tasdevice_tuning_switch(void *context, int state); +int tas2781_load_calibration(void *context, char *file_name, + unsigned short i); + +#endif diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h new file mode 100644 index 000000000000..4038dd421150 --- /dev/null +++ b/include/sound/tas2781-tlv.h @@ -0,0 +1,21 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// + +#ifndef __TAS2781_TLV_H__ +#define __TAS2781_TLV_H__ + +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); +static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); + +#endif diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h new file mode 100644 index 000000000000..9de5c57c148c --- /dev/null +++ b/include/sound/tas2781.h @@ -0,0 +1,164 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TAS2781_H__ +#define __TAS2781_H__ + +#include "tas2781-dsp.h" + +/* version number */ +#define TAS2781_DRV_VER 1 +#define SMARTAMP_MODULE_NAME "tas2781" +#define TAS2781_GLOBAL_ADDR 0x40 +#define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_88200) + +#define TASDEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/*PAGE Control Register (available in page0 of each book) */ +#define TASDEVICE_PAGE_SELECT 0x00 +#define TASDEVICE_BOOKCTL_PAGE 0x00 +#define TASDEVICE_BOOKCTL_REG 127 +#define TASDEVICE_BOOK_ID(reg) (reg / (256 * 128)) +#define TASDEVICE_PAGE_ID(reg) ((reg % (256 * 128)) / 128) +#define TASDEVICE_PAGE_REG(reg) ((reg % (256 * 128)) % 128) +#define TASDEVICE_PGRG(reg) (reg % (256 * 128)) +#define TASDEVICE_REG(book, page, reg) (((book * 256 * 128) + \ + (page * 128)) + reg) + +/*Software Reset */ +#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) +#define TAS2781_REG_SWRESET_RESET BIT(0) + +/*I2C Checksum */ +#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) + +/* Volume control */ +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) + +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + +enum audio_device { + TAS2781 = 0, +}; + +enum device_catlog_id { + LENOVO = 0, + OTHERS +}; + +struct tasdevice { + struct tasdevice_fw *cali_data_fmw; + unsigned int dev_addr; + unsigned int err_code; + unsigned char cur_book; + short cur_prog; + short cur_conf; + bool is_loading; + bool is_loaderr; +}; + +struct tasdevice_irqinfo { + int irq_gpio; + int irq; +}; + +struct calidata { + unsigned char *data; + unsigned long total_sz; +}; + +struct tasdevice_priv { + struct tasdevice tasdevice[TASDEVICE_MAX_CHANNELS]; + struct tasdevice_irqinfo irq_info; + struct tasdevice_rca rcabin; + struct calidata cali_data; + struct tasdevice_fw *fmw; + struct gpio_desc *reset; + struct mutex codec_lock; + struct regmap *regmap; + struct device *dev; + struct tm tm; + + enum device_catlog_id catlog_id; + const char *acpi_subsystem_id; + unsigned char cal_binaryname[TASDEVICE_MAX_CHANNELS][64]; + unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; + unsigned char coef_binaryname[64]; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + unsigned char ndev; + unsigned int magic_num; + unsigned int chip_id; + unsigned int sysclk; + + int cur_prog; + int cur_conf; + int fw_state; + int index; + void *client; + void *codec; + bool force_fwload_status; + bool playback_started; + bool isacpi; + int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, int offset); + int (*fw_parse_program_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block); +}; + +void tas2781_reset(struct tasdevice_priv *tas_dev); +int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, + void (*cont)(const struct firmware *fw, void *context)); +struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c); +int tasdevice_init(struct tasdevice_priv *tas_priv); +void tasdevice_remove(struct tasdevice_priv *tas_priv); +int tasdevice_dev_read(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned int *value); +int tasdevice_dev_write(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned int value); +int tasdevice_dev_bulk_write( + struct tasdevice_priv *tas_priv, unsigned short chn, + unsigned int reg, unsigned char *p_data, unsigned int n_length); +int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned char *p_data, + unsigned int n_length); +int tasdevice_dev_update_bits( + struct tasdevice_priv *tasdevice, unsigned short chn, + unsigned int reg, unsigned int mask, unsigned int value); +int tasdevice_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); + +#endif /* __TAS2781_H__ */ -- 2.34.1

1 0

[PATCH] ALSA: hda/realtek: Add a quirk for Compaq N14JP6
by Edson Juliano Drosdeck 10 Jun '23

10 Jun '23

Add a quirk for Compaq N14JP6 to fixup ALC897 headset MIC no sound. Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a5d55a7063d3..308ec7034cc9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11740,6 +11740,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1b0a, 0x01b8, "ACER Veriton", ALC662_FIXUP_ACER_VERITON), SND_PCI_QUIRK(0x1b35, 0x1234, "CZC ET26", ALC662_FIXUP_CZC_ET26), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), + SND_PCI_QUIRK(0x1c6c, 0x1239, "Compaq N14JP6-V2", ALC897_FIXUP_HP_HSMIC_VERB), #if 0 /* Below is a quirk table taken from the old code. -- 2.30.2

2 1

[PATCH 1/2] ASoC: dt-bindings: max98388: add amplifier driver
by “Ryan 10 Jun '23

10 Jun '23

From: Ryan Lee <ryans.lee(a)analog.com> Add dt-bindings information for Analog Devices MAX98388 I2S Amplifier Signed-off-by: Ryan Lee <ryans.lee(a)analog.com> --- .../bindings/sound/adi,max98388.yaml | 79 +++++++++++++++++++ 1 file changed, 79 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,max98388.yaml diff --git a/Documentation/devicetree/bindings/sound/adi,max98388.yaml b/Documentation/devicetree/bindings/sound/adi,max98388.yaml new file mode 100644 index 000000000000..fc0ac8d8c3ae --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,max98388.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,max98388.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices MAX98388 Speaker Amplifier + +maintainers: + - Ryan Lee <ryans.lee(a)analog.com> + +description: + The MAX98388 is a mono Class-D speaker amplifier with I/V feedback. + The device provides a PCM interface for audio data and a standard + I2C interface for control data communication. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - adi,max98388 + reg: + maxItems: 1 + description: I2C address of the device. + + '#sound-dai-cells': + const: 0 + + adi,vmon-slot-no: + description: slot number of the voltage feedback monitor + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 15 + default: 0 + + adi,imon-slot-no: + description: slot number of the current feedback monitor + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 15 + default: 1 + + adi,interleave-mode: + description: + For cases where a single combined channel for the I/V feedback data + is not sufficient, the device can also be configured to share + a single data output channel on alternating frames. + In this configuration, the current and voltage data will be frame + interleaved on a single output channel. + type: boolean + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + - "#sound-dai-cells" + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98388: amplifier@39 { + compatible = "adi,max98388"; + reg = <0x39>; + #sound-dai-cells = <0>; + adi,vmon-slot-no = <0>; + adi,imon-slot-no = <1>; + adi,interleave-mode; + reset-gpios = <&gpio 4 GPIO_ACTIVE_LOW>; + }; + }; -- 2.34.1

4 4

[PATCH] Revert "ASoC: mediatek: mt8192-mt6359: Remove " Jack" from Headphone pin name"
by Nícolas F. R. A. Prado 09 Jun '23

09 Jun '23

This reverts commit cbbc0ec6dea09c815f1d1ef0abaf3f2ec89ff11f. That commit removed the " Jack" suffix with the reasoning that it is automatically added to the name of the kcontrol created, which is true, but this name is also used to look for the DAPM widget that will be toggled when the jack status is updated. Since the widget is still called "Headphone Jack" the jack can't link to the widget and the following error is shown: mt8192_mt6359 sound: ASoC: DAPM unknown pin Headphone It is not possible to also rename the headphone DAPM widget because its name is used by a switch kcontrol, "Headphone Jack Switch", both to link to the headphone widget and to assemble its name. This switch's name is referenced in the upstream UCM file, so renaming it would break userspace. Since the original commit didn't bring any benefit, besides sparing a few CPU cycles, simply revert it. Signed-off-by: Nícolas F. R. A. Prado <nfraprado(a)collabora.com> --- sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index 4e0d5bf12b47..5e163e23a207 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -46,7 +46,7 @@ struct mt8192_mt6359_priv { /* Headset jack detection DAPM pins */ static struct snd_soc_jack_pin mt8192_jack_pins[] = { { - .pin = "Headphone", + .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE, }, { -- 2.41.0

3 6

[PATCH] ASoC: audio-graph-card2-custom-sample: add missing CPU:Codec = 1:N sample
by Kuninori Morimoto 09 Jun '23

09 Jun '23

It has CPU:Codec = 1:1 and N:N samples, but missing 1:N settings. This patch adds it. One note here is that because of registering timing, probing and CPU/Codec numbering are mismatching. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../audio-graph-card2-custom-sample.dtsi | 26 +++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi index 994db61a26b3..f5e7d669da45 100644 --- a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi +++ b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi @@ -42,6 +42,15 @@ / { * [Normal] * cpu0 <-@-----------------> codec0 * + * [Semi-Multi] + * + * CPU:Codec = 1:N + * + * +-+ + * cpu7 <-@------->| |-> codec12 + * | |-> codec13 + * +-+ + * * [Multi-CPU/Codec] * +-+ +-+ * cpu1 <--| |<-@--------->| |-> codec1 @@ -128,6 +137,9 @@ audio-graph-card2-custom-sample { */ &cpu0 + /* [Semi-Multi] */ + &sm0 + /* * [Multi-CPU/Codec]: cpu side only * cpu1/cpu2/codec1/codec2 @@ -194,6 +206,13 @@ ports@4 { port@1 { mc2c10_ep: endpoint { remote-endpoint = <&codec10_ep>; }; }; port@2 { mc2c11_ep: endpoint { remote-endpoint = <&codec11_ep>; }; }; }; + + /* [Semi-Multi] */ + ports@5 { + port@0 { smcodec0_ep: endpoint { remote-endpoint = <&cpu7_ep>; }; }; + port@1 { smcodec1_ep: endpoint { remote-endpoint = <&codec12_ep>; }; }; + port@2 { smcodec2_ep: endpoint { remote-endpoint = <&codec13_ep>; }; }; + }; }; dpcm { @@ -261,6 +280,9 @@ ports { /* [DPCM-Multi]::FE */ port@5 { cpu5_ep: endpoint { remote-endpoint = <&fe10_ep>; }; }; port@6 { cpu6_ep: endpoint { remote-endpoint = <&fe11_ep>; }; }; + + /* [Semi-Multi] */ + sm0: port@7 { cpu7_ep: endpoint { remote-endpoint = <&smcodec0_ep>; }; }; }; }; @@ -311,6 +333,10 @@ port@3 { port@9 { codec9_ep: endpoint { remote-endpoint = <&mc2c01_ep>; }; }; port@10 { codec10_ep: endpoint { remote-endpoint = <&mc2c10_ep>; }; }; port@11 { codec11_ep: endpoint { remote-endpoint = <&mc2c11_ep>; }; }; + + /* [Semi-Multi] */ + port@12 { codec12_ep: endpoint { remote-endpoint = <&smcodec1_ep>; }; }; + port@13 { codec13_ep: endpoint { remote-endpoint = <&smcodec2_ep>; }; }; }; }; }; -- 2.25.1

2 1

[PATCH 1/2] ASoC: SOF: Refactor rx function for fuzzing
by cujomalainey＠chromium.org 09 Jun '23

09 Jun '23

From: Curtis Malainey <cujomalainey(a)chromium.org> Refactor the function so reading the data is done outside the work function so fuzzing can pass data directly into the work callbacks. Also expose the inner function outside the module so we can call it from the injector. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc3-priv.h | 2 ++ sound/soc/sof/ipc3.c | 63 ++++++++++++++++++++++++--------------- 2 files changed, 41 insertions(+), 24 deletions(-) diff --git a/sound/soc/sof/ipc3-priv.h b/sound/soc/sof/ipc3-priv.h index f5044202f3c5a..0bbca418e67e6 100644 --- a/sound/soc/sof/ipc3-priv.h +++ b/sound/soc/sof/ipc3-priv.h @@ -28,6 +28,8 @@ int sof_ipc3_validate_fw_version(struct snd_sof_dev *sdev); /* dtrace position update */ int ipc3_dtrace_posn_update(struct snd_sof_dev *sdev, struct sof_ipc_dma_trace_posn *posn); +/* RX handler backend */ +void sof_ipc3_do_rx_work(struct snd_sof_dev *sdev, struct sof_ipc_cmd_hdr *hdr, void *msg_buf); /* dtrace platform callback wrappers */ static inline int sof_dtrace_host_init(struct snd_sof_dev *sdev, diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index c677677420939..ec1ac0fb2d9f8 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -954,31 +954,21 @@ static void ipc3_trace_message(struct snd_sof_dev *sdev, void *msg_buf) } } -/* DSP firmware has sent host a message */ -static void sof_ipc3_rx_msg(struct snd_sof_dev *sdev) +void sof_ipc3_do_rx_work(struct snd_sof_dev *sdev, struct sof_ipc_cmd_hdr *hdr, void *msg_buf) { ipc3_rx_callback rx_callback = NULL; - struct sof_ipc_cmd_hdr hdr; - void *msg_buf; u32 cmd; int err; - /* read back header */ - err = snd_sof_ipc_msg_data(sdev, NULL, &hdr, sizeof(hdr)); - if (err < 0) { - dev_warn(sdev->dev, "failed to read IPC header: %d\n", err); - return; - } + ipc3_log_header(sdev->dev, "ipc rx", hdr->cmd); - if (hdr.size < sizeof(hdr) || hdr.size > SOF_IPC_MSG_MAX_SIZE) { + if (hdr->size < sizeof(hdr) || hdr->size > SOF_IPC_MSG_MAX_SIZE) { dev_err(sdev->dev, "The received message size is invalid: %u\n", - hdr.size); + hdr->size); return; } - ipc3_log_header(sdev->dev, "ipc rx", hdr.cmd); - - cmd = hdr.cmd & SOF_GLB_TYPE_MASK; + cmd = hdr->cmd & SOF_GLB_TYPE_MASK; /* check message type */ switch (cmd) { @@ -1016,6 +1006,36 @@ static void sof_ipc3_rx_msg(struct snd_sof_dev *sdev) break; } + /* Call local handler for the message */ + if (rx_callback) + rx_callback(sdev, msg_buf); + + /* Notify registered clients */ + sof_client_ipc_rx_dispatcher(sdev, msg_buf); + + ipc3_log_header(sdev->dev, "ipc rx done", hdr->cmd); +} +EXPORT_SYMBOL(sof_ipc3_do_rx_work); + +/* DSP firmware has sent host a message */ +static void sof_ipc3_rx_msg(struct snd_sof_dev *sdev) +{ + struct sof_ipc_cmd_hdr hdr; + void *msg_buf; + int err; + + /* read back header */ + err = snd_sof_ipc_msg_data(sdev, NULL, &hdr, sizeof(hdr)); + if (err < 0) { + dev_warn(sdev->dev, "failed to read IPC header: %d\n", err); + return; + } + + if (hdr.size < sizeof(hdr)) { + dev_err(sdev->dev, "The received message size is invalid\n"); + return; + } + /* read the full message */ msg_buf = kmalloc(hdr.size, GFP_KERNEL); if (!msg_buf) @@ -1024,18 +1044,13 @@ static void sof_ipc3_rx_msg(struct snd_sof_dev *sdev) err = snd_sof_ipc_msg_data(sdev, NULL, msg_buf, hdr.size); if (err < 0) { dev_err(sdev->dev, "%s: Failed to read message: %d\n", __func__, err); - } else { - /* Call local handler for the message */ - if (rx_callback) - rx_callback(sdev, msg_buf); - - /* Notify registered clients */ - sof_client_ipc_rx_dispatcher(sdev, msg_buf); + kfree(msg_buf); + return; } - kfree(msg_buf); + sof_ipc3_do_rx_work(sdev, &hdr, msg_buf); - ipc3_log_header(sdev->dev, "ipc rx done", hdr.cmd); + kfree(msg_buf); } static int sof_ipc3_set_core_state(struct snd_sof_dev *sdev, int core_idx, bool on) -- 2.41.0.162.gfafddb0af9-goog

2 2

ASUS RoG Strix Z790-F: No SPDIF Sound output
by GitHub issues - opened 09 Jun '23

09 Jun '23

alsa-project/alsa-lib issue #325 was opened from Nightwing0815: Hey guys, what Can I do to get my optical toslink output ( pcm signal undecoded ) working? Here are an inxi report of my system: ``` System: Host: Ultron Kernel: 6.1.0-1013-oem x86_64 bits: 64 compiler: N/A parameters: BOOT_IMAGE=/vmlinuz-6.1.0-1013-oem root=UUID=aa125f57-ee1e-4fd4-93fc-a77a180ec451 ro quiet splash Desktop: Cinnamon 5.6.8 tk: GTK 3.24.33 wm: muffin vt: 7 dm: LightDM 1.30.0 Distro: Linux Mint 21.1 Vera base: Ubuntu 22.04 jammy Machine: Type: Desktop System: ASUS product: N/A v: N/A serial: <superuser required> Mobo: ASUSTeK model: ROG STRIX Z790-F GAMING WIFI v: Rev 1.xx serial: <superuser required> UEFI: American Megatrends v: 0904 date: 03/29/2023 Battery: Message: No system battery data found. Is one present? Memory: RAM: total: 62.57 GiB used: 2.13 GiB (3.4%) RAM Report: unknown-error: Unknown dmidecode error. Unable to generate data. PCI Slots: Unknown-error: Unknown dmidecode error. Unable to generate data. CPU: Info: model: 13th Gen Intel Core i7-13700K bits: 64 type: MST AMCP arch: N/A family: 6 model-id: 0xB7 (183) stepping: 1 microcode: 0x113 Topology: cpus: 1x cores: 16 mt: 8 tpc: 2 st: 8 threads: 24 smt: enabled cache: L1: 1.4 MiB desc: d-8x32 KiB, 8x48 KiB; i-8x32 KiB, 8x64 KiB L2: 24 MiB desc: 8x2 MiB, 2x4 MiB L3: 30 MiB desc: 1x30 MiB Speed (MHz): avg: 3100 high: 3400 min/max: 800/5300:5400:4200 scaling: driver: intel_pstate governor: powersave cores: 1: 1100 2: 3400 3: 3400 4: 3400 5: 3400 6: 3400 7: 3400 8: 3400 9: 3400 10: 3400 11: 3400 12: 3400 13: 3400 14: 3400 15: 1100 16: 3400 17: 800 18: 3400 19: 3400 20: 3400 21: 3400 22: 3400 23: 3400 24: 3400 bogomips: 164044 Graphics: Device-1: NVIDIA vendor: CardExpert driver: nvidia v: 530.41.03 alternate: nvidiafb,nouveau,nvidia_drm ports: active: none off: DP-1,HDMI-A-1 empty: DP-2,DP-3 bus-ID: 0000:01:00.0 chip-ID: 10de:2782 class-ID: 0300 Display: x11 server: X.Org v: 1.21.1.4 driver: X: loaded: nvidia gpu: nvidia display-ID: :0 screens: 1 Screen-1: 0 s-res: 1920x2160 s-dpi: 46 s-size: 1060x1167mm (41.7x45.9") s-diag: 1577mm (62.1") Monitor-1: DP-1 pos: primary,bottom res: 1920x1080 hz: 60 dpi: 102 size: 477x268mm (18.8x10.6") diag: 547mm (21.5") Monitor-2: HDMI-0 pos: primary,top res: 1920x1080 hz: 60 dpi: 38 size: 1280x720mm (50.4x28.3") diag: 1469mm (57.8") OpenGL: renderer: NVIDIA GeForce RTX 4070 Ti/PCIe/SSE2 v: 4.6.0 NVIDIA 530.41.03 direct render: Yes Audio: Device-1: Intel vendor: ASUSTeK driver: snd_hda_intel v: kernel alternate: snd_sof_pci_intel_tgl bus-ID: 0000:00:1f.3 chip-ID: 8086:7a50 class-ID: 0403 Device-2: NVIDIA vendor: CardExpert driver: snd_hda_intel v: kernel bus-ID: 0000:01:00.1 chip-ID: 10de:22bc class-ID: 0403 Device-3: ASUSTek USB Audio type: USB driver: hid-generic,snd-usb-audio,usbhid bus-ID: 1-2:2 chip-ID: 0b05:1a52 class-ID: 0300 Sound Server-1: ALSA v: k6.1.0-1013-oem running: yes Sound Server-2: PulseAudio v: 15.99.1 running: no Sound Server-3: PipeWire v: 0.3.48 running: yes Network: Device-1: Intel vendor: ASUSTeK driver: igc v: kernel port: N/A bus-ID: 0000:05:00.0 chip-ID: 8086:125c class-ID: 0200 IF: eno2 state: up speed: 1000 Mbps duplex: full mac: c8:7f:54:6a:7e:55 IP v4: 192.168.178.10/24 type: noprefixroute scope: global broadcast: 192.168.178.255 IP v6: 2a02:8070:8a80:77c0:aac9:dade:e3e2:8611/64 type: temporary dynamic scope: global IP v6: 2a02:8070:8a80:77c0:64ef:c8d5:1d2b:3a7f/64 type: dynamic mngtmpaddr noprefixroute scope: global IP v6: fe80::6d94:62d1:90b5:608b/64 type: noprefixroute scope: link IF-ID-1: virbr0 state: down mac: 52:54:00:e9:bf:38 IP v4: 192.168.100.1/24 scope: global broadcast: 192.168.100.255 WAN IP: 46.5.221.128 Bluetooth: Message: No bluetooth data found. Logical: Message: No logical block device data found. RAID: Hardware-1: Intel Device driver: vmd v: 0.6 port: N/A bus-ID: 0000:00:0e.0 chip-ID: 8086:a77f rev: class-ID: 0104 Unmounted: Message: No unmounted partitions found. USB: Hub-1: 1-0:1 info: Hi-speed hub with single TT ports: 16 rev: 2.0 speed: 480 Mb/s chip-ID: 1d6b:0002 class-ID: 0900 Device-1: 1-2:2 info: ASUSTek USB Audio type: Audio,HID driver: hid-generic,snd-usb-audio,usbhid interfaces: 8 rev: 2.0 speed: 480 Mb/s power: 100mA chip-ID: 0b05:1a52 class-ID: 0300 Device-2: 1-4:3 info: ASUSTek AURA LED Controller type: HID driver: hid-generic,usbhid interfaces: 2 rev: 2.0 speed: 12 Mb/s power: 16mA chip-ID: 0b05:19af class-ID: 0300 serial: 9876543210 Hub-2: 1-10:4 info: ASMedia ASM1074 High-Speed hub ports: 4 rev: 2.1 speed: 480 Mb/s power: 100mA chip-ID: 174c:2074 class-ID: 0900 Device-1: 1-10.3:6 info: Fujitsu Siemens s KB955 USB Keyboard type: Keyboard,HID driver: hid-generic,usbhid interfaces: 2 rev: 1.1 speed: 1.5 Mb/s power: 100mA chip-ID: 0bf8:102e class-ID: 0300 Device-2: 1-10.4:8 info: Logitech G5 Laser Mouse type: Mouse,HID driver: hid-generic,usbhid interfaces: 2 rev: 2.0 speed: 12 Mb/s power: 98mA chip-ID: 046d:c049 class-ID: 0300 Hub-3: 1-11:5 info: Genesys Logic Hub ports: 4 rev: 2.0 speed: 480 Mb/s power: 100mA chip-ID: 05e3:0608 class-ID: 0900 Hub-4: 1-13:7 info: Genesys Logic Hub ports: 4 rev: 2.0 speed: 480 Mb/s power: 100mA chip-ID: 05e3:0608 class-ID: 0900 Hub-5: 2-0:1 info: Super-speed hub ports: 8 rev: 3.1 speed: 20 Gb/s chip-ID: 1d6b:0003 class-ID: 0900 Hub-6: 2-8:2 info: ASMedia ASM1074 SuperSpeed hub ports: 4 rev: 3.0 speed: 5 Gb/s power: 8mA chip-ID: 174c:3074 class-ID: 0900 Sensors: System Temperatures: cpu: 32.5 C mobo: 30.0 C gpu: nvidia temp: 36 C Fan Speeds (RPM): fan-1: 0 fan-2: 808 fan-3: 0 fan-4: 0 fan-5: 447 fan-6: 453 fan-7: 2288 gpu: nvidia fan: 0% ``` Thanks in advance, ToM Issue URL : https://github.com/alsa-project/alsa-lib/issues/325 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ucm2: MediaTek: mt8395-evk: Add alsa-ucm support
by GitHub pull_request - opened 09 Jun '23

09 Jun '23

alsa-project/alsa-ucm-conf pull request #322 was opened from zoran055: Add alsa-ucm support for the MediaTek mt8395-evk platform. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/322 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/322.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: MediaTek: mt8390-evk: Add alsa-ucm support
by GitHub pull_request - opened 09 Jun '23

09 Jun '23

alsa-project/alsa-ucm-conf pull request #321 was opened from zoran055: Add alsa-ucm support for the MediaTek mt8390-evk platform. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/321 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/321.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 0/2] Fix error check and cleanup for JH7110 TDM
by Walker Chen 09 Jun '23

09 Jun '23

Some minor issues were found during addtional testing and static analysis. The first patch fix the error check for the return value of devm_reset_control_array_get_exclusive(). The second patch drop some unused macros. Fixes: fd4762b6b5cf ("ASoC: starfive: Add JH7110 TDM driver") Changes since v1: - Fix an error check in jh7110_tdm_clk_reset_get(). - Return to use *_BIT to indicate the shift and remove some unused macros. --- v1: https://lore.kernel.org/all/20230607081439.1517-1-walker.chen@starfivetech.… Walker Chen (2): ASoC: starfive: Fix an error check in jh7110_tdm_clk_reset_get() ASoC: starfive: Remove some unused macros sound/soc/starfive/jh7110_tdm.c | 15 +++------------ 1 file changed, 3 insertions(+), 12 deletions(-) -- 2.17.1

3 5

[PATCH v1] ASoC: starfive: Cleanup and fix error check for JH7110 TDM
by Walker Chen 09 Jun '23

09 Jun '23

Some minor issues were found during addtional testing and static analysis. The patch fixed these minor issues. 1.Use BIT() macro to indicate configuration for TDM registers. 2.Fix the check for devm_reset_control_array_get_exclusive return value. The devm_reset_control_array_get_exclusive() function may return NULL if it's an optional request. If optional is intended then NULL should not be treated as an error case, but as a special kind of success case. So here the IS_ERR() is used to check better. Fixes: fd4762b6b5cf ("ASoC: starfive: Add JH7110 TDM driver") Signed-off-by: Walker Chen <walker.chen(a)starfivetech.com> --- Fix the following issue: https://lore.kernel.org/all/ZH7t6Nc+NTcGpq%2F3@moroto/ --- sound/soc/starfive/jh7110_tdm.c | 200 +++++++++++--------------------- 1 file changed, 68 insertions(+), 132 deletions(-) diff --git a/sound/soc/starfive/jh7110_tdm.c b/sound/soc/starfive/jh7110_tdm.c index 973b910d2d3e..139ff091672e 100644 --- a/sound/soc/starfive/jh7110_tdm.c +++ b/sound/soc/starfive/jh7110_tdm.c @@ -24,63 +24,33 @@ #include <sound/soc.h> #include <sound/soc-dai.h> +/* Register offsets for JH7110 TDM device */ #define TDM_PCMGBCR 0x00 - #define PCMGBCR_MASK 0x1e - #define PCMGBCR_ENABLE BIT(0) - #define PCMGBCR_TRITXEN BIT(4) - #define CLKPOL_BIT 5 - #define TRITXEN_BIT 4 - #define ELM_BIT 3 - #define SYNCM_BIT 2 - #define MS_BIT 1 #define TDM_PCMTXCR 0x04 - #define PCMTXCR_TXEN BIT(0) - #define IFL_BIT 11 - #define WL_BIT 8 - #define SSCALE_BIT 4 - #define SL_BIT 2 - #define LRJ_BIT 1 #define TDM_PCMRXCR 0x08 - #define PCMRXCR_RXEN BIT(0) - #define PCMRXCR_RXSL_MASK 0xc - #define PCMRXCR_RXSL_16BIT 0x4 - #define PCMRXCR_RXSL_32BIT 0x8 - #define PCMRXCR_SCALE_MASK 0xf0 - #define PCMRXCR_SCALE_1CH 0x10 #define TDM_PCMDIV 0x0c -#define JH7110_TDM_FIFO 0x170c0000 -#define JH7110_TDM_FIFO_DEPTH 32 +/* Bit definition for TDM_PCMGBCR */ +#define TDM_PCMGBCR_SYNCM_LONG BIT(2) +#define TDM_PCMGBCR_MS_SLAVE BIT(1) +#define TDM_PCMGBCR_EN BIT(0) -enum TDM_MASTER_SLAVE_MODE { - TDM_AS_MASTER = 0, - TDM_AS_SLAVE, -}; +/* Bit definition for TDM_PCMTXCR */ +#define TDM_PCMTXCR_LEFT_J BIT(1) +#define TDM_PCMTXCR_TXEN BIT(0) -enum TDM_CLKPOL { - /* tx raising and rx falling */ - TDM_TX_RASING_RX_FALLING = 0, - /* tx falling and rx raising */ - TDM_TX_FALLING_RX_RASING, -}; +/* Bit definition for TDM_PCMRXCR */ +#define TDM_PCMRXCR_LEFT_J BIT(1) +#define TDM_PCMRXCR_RXEN BIT(0) -enum TDM_ELM { - /* only work while SYNCM=0 */ - TDM_ELM_LATE = 0, - TDM_ELM_EARLY, -}; - -enum TDM_SYNCM { - /* short frame sync */ - TDM_SYNCM_SHORT = 0, - /* long frame sync */ - TDM_SYNCM_LONG, -}; +#define JH7110_TDM_FIFO 0x170c0000 +#define JH7110_TDM_FIFO_DEPTH 32 -enum TDM_IFL { - /* FIFO to send or received : half-1/2, Quarter-1/4 */ - TDM_FIFO_HALF = 0, - TDM_FIFO_QUARTER, +enum TDM_SL { + /* send or received slot length */ + TDM_8BIT_SLOT_LEN = 0, + TDM_16BIT_SLOT_LEN, + TDM_32BIT_SLOT_LEN, }; enum TDM_WL { @@ -92,43 +62,24 @@ enum TDM_WL { TDM_32BIT_WORD_LEN, }; -enum TDM_SL { - /* send or received slot length */ - TDM_8BIT_SLOT_LEN = 0, - TDM_16BIT_SLOT_LEN, - TDM_32BIT_SLOT_LEN, -}; - -enum TDM_LRJ { - /* left-justify or right-justify */ - TDM_RIGHT_JUSTIFY = 0, - TDM_LEFT_JUSTIFT, -}; - -struct tdm_chan_cfg { - enum TDM_IFL ifl; - enum TDM_WL wl; - unsigned char sscale; - enum TDM_SL sl; - enum TDM_LRJ lrj; - unsigned char enable; -}; - struct jh7110_tdm_dev { void __iomem *tdm_base; struct device *dev; struct clk_bulk_data clks[6]; struct reset_control *resets; - enum TDM_CLKPOL clkpolity; - enum TDM_ELM elm; - enum TDM_SYNCM syncm; - enum TDM_MASTER_SLAVE_MODE ms_mode; + /* related to PCMTXCR */ + u16 txwl; + u16 txsscale; + u16 txsl; - struct tdm_chan_cfg tx; - struct tdm_chan_cfg rx; + /* related to PCMRXCR */ + u16 rxwl; + u16 rxsscale; + u16 rxsl; u16 syncdiv; + u16 syncm; u32 samplerate; u32 pcmclk; @@ -166,13 +117,13 @@ static void jh7110_tdm_start(struct jh7110_tdm_dev *tdm, u32 data; data = jh7110_tdm_readl(tdm, TDM_PCMGBCR); - jh7110_tdm_writel(tdm, TDM_PCMGBCR, data | PCMGBCR_ENABLE); + jh7110_tdm_writel(tdm, TDM_PCMGBCR, data | TDM_PCMGBCR_EN); /* restore context */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - jh7110_tdm_writel(tdm, TDM_PCMTXCR, tdm->saved_pcmtxcr | PCMTXCR_TXEN); + jh7110_tdm_writel(tdm, TDM_PCMTXCR, tdm->saved_pcmtxcr | TDM_PCMTXCR_TXEN); else - jh7110_tdm_writel(tdm, TDM_PCMRXCR, tdm->saved_pcmrxcr | PCMRXCR_RXEN); + jh7110_tdm_writel(tdm, TDM_PCMRXCR, tdm->saved_pcmrxcr | TDM_PCMRXCR_RXEN); } static void jh7110_tdm_stop(struct jh7110_tdm_dev *tdm, @@ -182,11 +133,11 @@ static void jh7110_tdm_stop(struct jh7110_tdm_dev *tdm, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { val = jh7110_tdm_readl(tdm, TDM_PCMTXCR); - val &= ~PCMTXCR_TXEN; + val &= ~TDM_PCMTXCR_TXEN; jh7110_tdm_writel(tdm, TDM_PCMTXCR, val); } else { val = jh7110_tdm_readl(tdm, TDM_PCMRXCR); - val &= ~PCMRXCR_RXEN; + val &= ~TDM_PCMRXCR_RXEN; jh7110_tdm_writel(tdm, TDM_PCMRXCR, val); } } @@ -195,15 +146,15 @@ static int jh7110_tdm_syncdiv(struct jh7110_tdm_dev *tdm) { u32 sl, sscale, syncdiv; - if (tdm->rx.sl >= tdm->tx.sl) - sl = tdm->rx.sl; + if (tdm->rxsl >= tdm->txsl) + sl = tdm->rxsl; else - sl = tdm->tx.sl; + sl = tdm->txsl; - if (tdm->rx.sscale >= tdm->tx.sscale) - sscale = tdm->rx.sscale; + if (tdm->rxsscale >= tdm->txsscale) + sscale = tdm->rxsscale; else - sscale = tdm->tx.sscale; + sscale = tdm->txsscale; syncdiv = tdm->pcmclk / tdm->samplerate - 1; @@ -212,10 +163,10 @@ static int jh7110_tdm_syncdiv(struct jh7110_tdm_dev *tdm) return -EINVAL; } - if (tdm->syncm == TDM_SYNCM_LONG && - (tdm->rx.sscale <= 1 || tdm->tx.sscale <= 1) && + if (tdm->syncm == TDM_PCMGBCR_SYNCM_LONG && + (tdm->rxsscale <= 1 || tdm->txsscale <= 1) && ((syncdiv + 1) <= sl)) { - dev_err(tdm->dev, "Wrong syncdiv! It must be (syncdiv+1) > max[tx.sl, rx.sl]\n"); + dev_err(tdm->dev, "Wrong syncdiv! It must be (syncdiv+1) > max[txsl, rxsl]\n"); return -EINVAL; } @@ -233,17 +184,15 @@ static int jh7110_tdm_config(struct jh7110_tdm_dev *tdm, if (ret) return ret; - datarx = (tdm->rx.ifl << IFL_BIT) | - (tdm->rx.wl << WL_BIT) | - (tdm->rx.sscale << SSCALE_BIT) | - (tdm->rx.sl << SL_BIT) | - (tdm->rx.lrj << LRJ_BIT); + datarx = (tdm->rxwl << 8) | + (tdm->rxsscale << 4) | + (tdm->rxsl << 2) | + TDM_PCMRXCR_LEFT_J; - datatx = (tdm->tx.ifl << IFL_BIT) | - (tdm->tx.wl << WL_BIT) | - (tdm->tx.sscale << SSCALE_BIT) | - (tdm->tx.sl << SL_BIT) | - (tdm->tx.lrj << LRJ_BIT); + datatx = (tdm->txwl << 8) | + (tdm->txsscale << 4) | + (tdm->txsl << 2) | + TDM_PCMTXCR_LEFT_J; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) jh7110_tdm_writel(tdm, TDM_PCMTXCR, datatx); @@ -346,7 +295,8 @@ static int jh7110_tdm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct jh7110_tdm_dev *tdm = snd_soc_dai_get_drvdata(dai); - int chan_wl, chan_sl, chan_nr; + int chan_nr; + unsigned short chan_wl, chan_sl; unsigned int data_width; unsigned int dma_bus_width; struct snd_dmaengine_dai_dma_data *dma_data = NULL; @@ -389,15 +339,15 @@ static int jh7110_tdm_hw_params(struct snd_pcm_substream *substream, } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - tdm->tx.wl = chan_wl; - tdm->tx.sl = chan_sl; - tdm->tx.sscale = chan_nr; + tdm->txwl = chan_wl; + tdm->txsl = chan_sl; + tdm->txsscale = chan_nr; tdm->play_dma_data.addr_width = dma_bus_width; dma_data = &tdm->play_dma_data; } else { - tdm->rx.wl = chan_wl; - tdm->rx.sl = chan_sl; - tdm->rx.sscale = chan_nr; + tdm->rxwl = chan_wl; + tdm->rxsl = chan_sl; + tdm->rxsscale = chan_nr; tdm->capture_dma_data.addr_width = dma_bus_width; dma_data = &tdm->capture_dma_data; } @@ -444,15 +394,17 @@ static int jh7110_tdm_set_dai_fmt(struct snd_soc_dai *cpu_dai, struct jh7110_tdm_dev *tdm = snd_soc_dai_get_drvdata(cpu_dai); unsigned int gbcr; + gbcr = tdm->syncm; + /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { case SND_SOC_DAIFMT_BP_FP: /* cpu is master */ - tdm->ms_mode = TDM_AS_MASTER; + gbcr &= ~TDM_PCMGBCR_MS_SLAVE; break; case SND_SOC_DAIFMT_BC_FC: /* codec is master */ - tdm->ms_mode = TDM_AS_SLAVE; + gbcr |= TDM_PCMGBCR_MS_SLAVE; break; case SND_SOC_DAIFMT_BC_FP: case SND_SOC_DAIFMT_BP_FC: @@ -462,10 +414,6 @@ static int jh7110_tdm_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - gbcr = (tdm->clkpolity << CLKPOL_BIT) | - (tdm->elm << ELM_BIT) | - (tdm->syncm << SYNCM_BIT) | - (tdm->ms_mode << MS_BIT); jh7110_tdm_writel(tdm, TDM_PCMGBCR, gbcr); return 0; @@ -537,18 +485,7 @@ static const struct snd_dmaengine_pcm_config jh7110_dmaengine_pcm_config = { static void jh7110_tdm_init_params(struct jh7110_tdm_dev *tdm) { - tdm->clkpolity = TDM_TX_RASING_RX_FALLING; - tdm->elm = TDM_ELM_LATE; - tdm->syncm = TDM_SYNCM_SHORT; - - tdm->rx.ifl = TDM_FIFO_HALF; - tdm->tx.ifl = TDM_FIFO_HALF; - tdm->rx.wl = TDM_16BIT_WORD_LEN; - tdm->tx.wl = TDM_16BIT_WORD_LEN; - tdm->rx.sscale = 2; - tdm->tx.sscale = 2; - tdm->rx.lrj = TDM_LEFT_JUSTIFT; - tdm->tx.lrj = TDM_LEFT_JUSTIFT; + tdm->syncm = 0; tdm->play_dma_data.addr = JH7110_TDM_FIFO; tdm->play_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; @@ -580,10 +517,9 @@ static int jh7110_tdm_clk_reset_get(struct platform_device *pdev, } tdm->resets = devm_reset_control_array_get_exclusive(&pdev->dev); - if (IS_ERR_OR_NULL(tdm->resets)) { - ret = PTR_ERR(tdm->resets); - dev_err(&pdev->dev, "Failed to get tdm resets"); - return ret; + if (IS_ERR(tdm->resets)) { + dev_err(&pdev->dev, "Failed to get tdm resets\n"); + return PTR_ERR(tdm->resets); } return 0; @@ -649,12 +585,12 @@ static int jh7110_tdm_dev_remove(struct platform_device *pdev) return 0; } -static const struct of_device_id jh7110_tdm_of_match[] = { +static const struct of_device_id jh7110_tdm_match[] = { { .compatible = "starfive,jh7110-tdm", }, {} }; -MODULE_DEVICE_TABLE(of, jh7110_tdm_of_match); +MODULE_DEVICE_TABLE(of, jh7110_tdm_match); static const struct dev_pm_ops jh7110_tdm_pm_ops = { RUNTIME_PM_OPS(jh7110_tdm_runtime_suspend, @@ -666,7 +602,7 @@ static const struct dev_pm_ops jh7110_tdm_pm_ops = { static struct platform_driver jh7110_tdm_driver = { .driver = { .name = "jh7110-tdm", - .of_match_table = jh7110_tdm_of_match, + .of_match_table = jh7110_tdm_match, .pm = pm_ptr(&jh7110_tdm_pm_ops), }, .probe = jh7110_tdm_probe, -- 2.17.1

3 7

[PATCH v4 00/68] clk: Make determine_rate mandatory for muxes
by Maxime Ripard 09 Jun '23

09 Jun '23

Hi, This is a follow-up to a previous series that was printing a warning when a mux has a set_parent implementation but is missing determine_rate(). The rationale is that set_parent() is very likely to be useful when changing the rate, but it's determine_rate() that takes the parenting decision. If we're missing it, then the current parent is always going to be used, and thus set_parent() will not be used. The only exception being a direct call to clk_set_parent(), but those are fairly rare compared to clk_set_rate(). Stephen then asked to promote the warning to an error, and to fix up all the muxes that are in that situation first. So here it is :) It was build-tested on x86, arm and arm64. Affected drivers have been tracked down by the following coccinelle script: virtual report @ clk_ops @ identifier ops; position p; @@ struct clk_ops ops@p = { ... }; @ has_set_parent @ identifier clk_ops.ops; identifier set_parent_f; @@ struct clk_ops ops = { .set_parent = set_parent_f, }; @ has_determine_rate @ identifier clk_ops.ops; identifier determine_rate_f; @@ struct clk_ops ops = { .determine_rate = determine_rate_f, }; @ script:python depends on report && has_set_parent && !has_determine_rate @ ops << clk_ops.ops; set_parent_f << has_set_parent.set_parent_f; p << clk_ops.p; @@ coccilib.report.print_report(p[0], "ERROR: %s has set_parent (%s)" % (ops, set_parent_f)) Berlin is the only user still matching after this series has been applied, but it's because it uses a composite clock which throws the script off. The driver has been converted and shouldn't be a problem. Let me know what you think, Maxime Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- Changes in v4: - Switch from __clk_mux_determine_rate to a new helper - Introduced unit tests for that new helper - Fix kunit regression - Reworded most of the commit logs - Link to v3: https://lore.kernel.org/r/20221018-clk-range-checks-fixes-v3-0-9a1358472d52… Changes in v3: - Rebased on top of next-20230404 - Link to v2: https://lore.kernel.org/r/20221018-clk-range-checks-fixes-v2-0-f6736dec138e… Changes in v2: - Drop all the patches already applied - Promote the clk registration warning to an error - Make all muxes use determine_rate - Link to v1: https://lore.kernel.org/r/20221018-clk-range-checks-fixes-v1-0-f3ef80518140… --- Maxime Ripard (66): clk: Export clk_hw_forward_rate_request() clk: test: Fix type sign of rounded rate variables clk: lan966x: Remove unused round_rate hook clk: nodrv: Add a determine_rate hook clk: test: Add a determine_rate hook clk: actions: composite: Add a determine_rate hook for pass clk clk: at91: main: Add a determine_rate hook clk: at91: sckc: Add a determine_rate hook clk: berlin: div: Add a determine_rate hook clk: cdce706: Add a determine_rate hook clk: k210: pll: Add a determine_rate hook clk: k210: aclk: Add a determine_rate hook clk: k210: mux: Add a determine_rate hook clk: lmk04832: clkout: Add a determine_rate hook clk: lochnagar: Add a determine_rate hook clk: qoriq: Add a determine_rate hook clk: si5341: Add a determine_rate hook clk: stm32f4: mux: Add a determine_rate hook clk: vc5: mux: Add a determine_rate hook clk: vc5: clkout: Add a determine_rate hook clk: wm831x: clkout: Add a determine_rate hook clk: davinci: da8xx-cfgchip: Add a determine_rate hook clk: davinci: da8xx-cfgchip: Add a determine_rate hook clk: imx: busy: Add a determine_rate hook clk: imx: fixup-mux: Add a determine_rate hook clk: imx: scu: Add a determine_rate hook clk: mediatek: cpumux: Add a determine_rate hook clk: pxa: Add a determine_rate hook clk: renesas: r9a06g032: Add a determine_rate hook clk: socfpga: gate: Add a determine_rate hook clk: stm32: core: Add a determine_rate hook clk: tegra: bpmp: Add a determine_rate hook clk: tegra: super: Add a determine_rate hook clk: tegra: periph: Add a determine_rate hook clk: ux500: prcmu: Add a determine_rate hook clk: ux500: sysctrl: Add a determine_rate hook clk: versatile: sp810: Add a determine_rate hook drm/tegra: sor: Add a determine_rate hook phy: cadence: sierra: Add a determine_rate hook phy: cadence: torrent: Add a determine_rate hook phy: ti: am654-serdes: Add a determine_rate hook phy: ti: j721e-wiz: Add a determine_rate hook rtc: sun6i: Add a determine_rate hook ASoC: tlv320aic32x4: Add a determine_rate hook clk: actions: composite: div: Switch to determine_rate clk: actions: composite: fact: Switch to determine_rate clk: at91: smd: Switch to determine_rate clk: axi-clkgen: Switch to determine_rate clk: cdce706: divider: Switch to determine_rate clk: cdce706: clkout: Switch to determine_rate clk: si5341: Switch to determine_rate clk: si5351: pll: Switch to determine_rate clk: si5351: msynth: Switch to determine_rate clk: si5351: clkout: Switch to determine_rate clk: da8xx: clk48: Switch to determine_rate clk: imx: scu: Switch to determine_rate clk: ingenic: cgu: Switch to determine_rate clk: ingenic: tcu: Switch to determine_rate clk: sprd: composite: Switch to determine_rate clk: st: flexgen: Switch to determine_rate clk: stm32: composite: Switch to determine_rate clk: tegra: periph: Switch to determine_rate clk: tegra: super: Switch to determine_rate ASoC: tlv320aic32x4: pll: Switch to determine_rate ASoC: tlv320aic32x4: div: Switch to determine_rate clk: Forbid to register a mux without determine_rate Stephen Boyd (2): clk: Move no reparent case into a separate function clk: Introduce clk_hw_determine_rate_no_reparent() drivers/clk/actions/owl-composite.c | 35 ++++-- drivers/clk/at91/clk-main.c | 1 + drivers/clk/at91/clk-smd.c | 29 +++-- drivers/clk/at91/sckc.c | 1 + drivers/clk/berlin/berlin2-div.c | 1 + drivers/clk/clk-axi-clkgen.c | 14 ++- drivers/clk/clk-cdce706.c | 30 ++--- drivers/clk/clk-k210.c | 3 + drivers/clk/clk-lan966x.c | 17 --- drivers/clk/clk-lmk04832.c | 1 + drivers/clk/clk-lochnagar.c | 1 + drivers/clk/clk-qoriq.c | 1 + drivers/clk/clk-si5341.c | 19 ++-- drivers/clk/clk-si5351.c | 67 ++++++----- drivers/clk/clk-stm32f4.c | 1 + drivers/clk/clk-versaclock5.c | 2 + drivers/clk/clk-wm831x.c | 1 + drivers/clk/clk.c | 108 ++++++++++++------ drivers/clk/clk_test.c | 180 +++++++++++++++++++++++++++++- drivers/clk/davinci/da8xx-cfgchip.c | 12 +- drivers/clk/imx/clk-busy.c | 1 + drivers/clk/imx/clk-fixup-mux.c | 1 + drivers/clk/imx/clk-scu.c | 20 +++- drivers/clk/ingenic/cgu.c | 15 +-- drivers/clk/ingenic/tcu.c | 19 ++-- drivers/clk/mediatek/clk-cpumux.c | 1 + drivers/clk/pxa/clk-pxa.c | 1 + drivers/clk/renesas/r9a06g032-clocks.c | 1 + drivers/clk/socfpga/clk-gate.c | 1 + drivers/clk/sprd/composite.c | 16 ++- drivers/clk/st/clk-flexgen.c | 15 +-- drivers/clk/stm32/clk-stm32-core.c | 33 ++++-- drivers/clk/tegra/clk-bpmp.c | 1 + drivers/clk/tegra/clk-periph.c | 17 ++- drivers/clk/tegra/clk-super.c | 16 ++- drivers/clk/ux500/clk-prcmu.c | 1 + drivers/clk/ux500/clk-sysctrl.c | 1 + drivers/clk/versatile/clk-sp810.c | 1 + drivers/gpu/drm/tegra/sor.c | 1 + drivers/phy/cadence/phy-cadence-sierra.c | 1 + drivers/phy/cadence/phy-cadence-torrent.c | 1 + drivers/phy/ti/phy-am654-serdes.c | 1 + drivers/phy/ti/phy-j721e-wiz.c | 1 + drivers/rtc/rtc-sun6i.c | 1 + include/linux/clk-provider.h | 2 + sound/soc/codecs/tlv320aic32x4-clk.c | 33 +++--- 46 files changed, 527 insertions(+), 199 deletions(-) --- base-commit: 145e5cddfe8b4bf607510b2dcf630d95f4db420f change-id: 20221018-clk-range-checks-fixes-2039f3523240 Best regards, -- Maxime Ripard <maxime(a)cerno.tech>

2 4

[PATCH] ASoC: stac9766: fix build errors with REGMAP_AC97
by Randy Dunlap 09 Jun '23

09 Jun '23

Select REGMAP_AC97 to fix these build errors: ERROR: modpost: "regmap_ac97_default_volatile" [sound/soc/codecs/snd-soc-stac9766.ko] undefined! ERROR: modpost: "__regmap_init_ac97" [sound/soc/codecs/snd-soc-stac9766.ko] undefined! Fixes: 6bbf787bb70c ("ASoC: stac9766: Convert to regmap") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff -- a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1693,6 +1693,7 @@ config SND_SOC_STA529 config SND_SOC_STAC9766 tristate depends on SND_SOC_AC97_BUS + select REGMAP_AC97 config SND_SOC_STI_SAS tristate "codec Audio support for STI SAS codec"

1 0

[PATCH] ASoC: max98088: clean up some inconsistent indenting
by Jiapeng Chong 08 Jun '23

08 Jun '23

No functional modification involved. sound/soc/codecs/max98088.c:316 m98088_eq_band() warn: inconsistent indenting. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Closes: https://bugzilla.openanolis.cn/show_bug.cgi?id=5461 Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/max98088.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index fb136e2b849b..8b56ee550c09 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -310,24 +310,24 @@ static const struct regmap_config max98088_regmap = { static void m98088_eq_band(struct snd_soc_component *component, unsigned int dai, unsigned int band, u16 *coefs) { - unsigned int eq_reg; - unsigned int i; + unsigned int eq_reg; + unsigned int i; if (WARN_ON(band > 4) || WARN_ON(dai > 1)) return; - /* Load the base register address */ - eq_reg = dai ? M98088_REG_84_DAI2_EQ_BASE : M98088_REG_52_DAI1_EQ_BASE; + /* Load the base register address */ + eq_reg = dai ? M98088_REG_84_DAI2_EQ_BASE : M98088_REG_52_DAI1_EQ_BASE; - /* Add the band address offset, note adjustment for word address */ - eq_reg += band * (M98088_COEFS_PER_BAND << 1); + /* Add the band address offset, note adjustment for word address */ + eq_reg += band * (M98088_COEFS_PER_BAND << 1); - /* Step through the registers and coefs */ - for (i = 0; i < M98088_COEFS_PER_BAND; i++) { - snd_soc_component_write(component, eq_reg++, M98088_BYTE1(coefs[i])); - snd_soc_component_write(component, eq_reg++, M98088_BYTE0(coefs[i])); - } + /* Step through the registers and coefs */ + for (i = 0; i < M98088_COEFS_PER_BAND; i++) { + snd_soc_component_write(component, eq_reg++, M98088_BYTE1(coefs[i])); + snd_soc_component_write(component, eq_reg++, M98088_BYTE0(coefs[i])); + } } /* -- 2.20.1.7.g153144c

2 1

[PATCH] ASoC: tegra: Simplify code around clk_get_rate() handling
by Christophe JAILLET 08 Jun '23

08 Jun '23

clk_get_rate() returns an unsigned long, so there is no point in storing it in a long, and test for negative values. So, turn 'parent_rate' into an unsigned long, simplify the sanity check, the error message and the return value, in case of error (i.e. 0). Doing so also turns 'i' and 'valid_rates' into unsigned long, but it is fine and harmless. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/tegra/tegra20_i2s.c | 9 ++++----- sound/soc/tegra/tegra20_spdif.c | 9 ++++----- 2 files changed, 8 insertions(+), 10 deletions(-) diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index e1a0f50969c1..d38b58305c6b 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -273,13 +273,12 @@ static int tegra20_i2s_filter_rates(struct snd_pcm_hw_params *params, struct snd_soc_dai *dai = rule->private; struct tegra20_i2s *i2s = dev_get_drvdata(dai->dev); struct clk *parent = clk_get_parent(i2s->clk_i2s); - long i, parent_rate, valid_rates = 0; + unsigned long i, parent_rate, valid_rates = 0; parent_rate = clk_get_rate(parent); - if (parent_rate <= 0) { - dev_err(dai->dev, "Can't get parent clock rate: %ld\n", - parent_rate); - return parent_rate ?: -EINVAL; + if (!parent_rate) { + dev_err(dai->dev, "Can't get parent clock rate\n"); + return -EINVAL; } for (i = 0; i < ARRAY_SIZE(tegra20_i2s_rates); i++) { diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 86bef54dfdf2..d034803695a0 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -187,13 +187,12 @@ static int tegra20_spdif_filter_rates(struct snd_pcm_hw_params *params, struct tegra20_spdif *spdif = dev_get_drvdata(dai->dev); struct clk *parent = clk_get_parent(spdif->clk_spdif_out); static const unsigned int rates[] = { 32000, 44100, 48000 }; - long i, parent_rate, valid_rates = 0; + unsigned long i, parent_rate, valid_rates = 0; parent_rate = clk_get_rate(parent); - if (parent_rate <= 0) { - dev_err(dai->dev, "Can't get parent clock rate: %ld\n", - parent_rate); - return parent_rate ?: -EINVAL; + if (!parent_rate) { + dev_err(dai->dev, "Can't get parent clock rate\n"); + return -EINVAL; } for (i = 0; i < ARRAY_SIZE(rates); i++) { -- 2.34.1

2 1

[PATCH v2 0/3] ASoC: stm32: fix dtbs_check warnings
by Olivier Moysan 08 Jun '23

08 Jun '23

Fix dtbs_check warnings in STM32MP15 DK boards Devices Trees for STM32 I2S and Cirrus CS42L51 codec. - Add OF graph port property in I2S and CS42L51 DT bindings. Fixes warnings: audio-controller@4000b000: Unevaluated properties are not allowed ('port' was unexpected) cs42l51@4a: Unevaluated properties are not allowed ('port' was unexpected) - Correct OF graph DAI audio format property for STM32MP15x Dkx I2S node Changes in v2: - Add port example in i2s and cs42l51 binding Olivier Moysan (3): ASoC: dt-bindings: stm32: document audio of graph port for i2s ASoC: dt-bindings: document audio of graph port for cs42l51 ARM: dts: stm32: fix i2s endpoint format property for stm32mp15xx-dkx .../devicetree/bindings/sound/cirrus,cs42l51.yaml | 11 +++++++++++ .../devicetree/bindings/sound/st,stm32-i2s.yaml | 11 +++++++++++ arch/arm/boot/dts/stm32mp15xx-dkx.dtsi | 2 +- 3 files changed, 23 insertions(+), 1 deletion(-) -- 2.25.1

4 6

[PATCH] soundwire: debugfs: Add missing SCP registers
by Bard Liao 08 Jun '23

08 Jun '23

From: Uday M Bhat <uday.m.bhat(a)intel.com> SCP registers needs to be updated to accommodate additional register entries as per the Soundwire 1.2 specification. Signed-off-by: Uday M Bhat <uday.m.bhat(a)intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/debugfs.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/debugfs.c b/drivers/soundwire/debugfs.c index c3a1a359ee5c..d1553cb77187 100644 --- a/drivers/soundwire/debugfs.c +++ b/drivers/soundwire/debugfs.c @@ -86,10 +86,17 @@ static int sdw_slave_reg_show(struct seq_file *s_file, void *data) /* SCP registers */ ret += scnprintf(buf + ret, RD_BUF - ret, "\nSCP\n"); - for (i = SDW_SCP_INT1; i <= SDW_SCP_BANKDELAY; i++) + for (i = SDW_SCP_INT1; i <= SDW_SCP_BUS_CLOCK_BASE; i++) ret += sdw_sprintf(slave, buf, ret, i); for (i = SDW_SCP_DEVID_0; i <= SDW_SCP_DEVID_5; i++) ret += sdw_sprintf(slave, buf, ret, i); + for (i = SDW_SCP_FRAMECTRL_B0; i <= SDW_SCP_BUSCLOCK_SCALE_B0; i++) + ret += sdw_sprintf(slave, buf, ret, i); + for (i = SDW_SCP_FRAMECTRL_B1; i <= SDW_SCP_BUSCLOCK_SCALE_B1; i++) + ret += sdw_sprintf(slave, buf, ret, i); + for (i = SDW_SCP_PHY_OUT_CTRL_0; i <= SDW_SCP_PHY_OUT_CTRL_7; i++) + ret += sdw_sprintf(slave, buf, ret, i); + /* * SCP Bank 0/1 registers are read-only and cannot be -- 2.25.1

2 1

[PATCH v2 1/5] soundwire: stream: Add missing clear of alloc_slave_rt
by Charles Keepax 08 Jun '23

08 Jun '23

The current path that skips allocating the slave runtime does not clear the alloc_slave_rt flag, this is clearly incorrect. Add the missing clear, so the runtime won't be erroneously cleaned up. Fixes: f3016b891c8c ("soundwire: stream: sdw_stream_add_ functions can be called multiple times") Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Changes since v1: - Added missing fixes tag Thanks, Charles drivers/soundwire/stream.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index c2191c07442b0..379228f221869 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -2021,8 +2021,10 @@ int sdw_stream_add_slave(struct sdw_slave *slave, skip_alloc_master_rt: s_rt = sdw_slave_rt_find(slave, stream); - if (s_rt) + if (s_rt) { + alloc_slave_rt = false; goto skip_alloc_slave_rt; + } s_rt = sdw_slave_rt_alloc(slave, m_rt); if (!s_rt) { -- 2.30.2

3 8

[PATCH 0/5] ASoC: mt8188-mt6359: Cleanups
by AngeloGioacchino Del Regno 08 Jun '23

08 Jun '23

This series performs some cleanups to the mt8188-mt6359 driver, including usage of bitfield macros, adding definitions of register fields and some others for readability and consistency. AngeloGioacchino Del Regno (4): ASoC: mediatek: mt8188-mt6359: Compress of_device_id entries ASoC: mediatek: mt8188-mt6359: Cleanup return 0 disguised as return ret ASoC: mediatek: mt8188-mt6359: Clean up log levels ASoC: mediatek: mt8188-mt6359: Use bitfield macros for registers Dan Carpenter (1): ASoC: mediatek: mt8188-mt6359: clean up a return in codec_init sound/soc/mediatek/mt8188/mt8188-mt6359.c | 87 ++++++++++++----------- 1 file changed, 45 insertions(+), 42 deletions(-) -- 2.40.1

5 20

[PATCH 1/2] ASoC: dt-bindings: qcom,wsa8840: Add WSA884x family of speakers
by Krzysztof Kozlowski 08 Jun '23

08 Jun '23

Add binding for WSA8840/WSA8845/WSA8845H smart speaker amplifiers used in Qualcomm QRD8550 board with SM8550 SoC. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../bindings/sound/qcom,wsa8840.yaml | 66 +++++++++++++++++++ 1 file changed, 66 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml new file mode 100644 index 000000000000..a999f787aa4d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wsa8840.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WSA8840/WSA8845/WSA8845H smart speaker amplifier + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + +description: + WSA884X is a family of Qualcomm Aqstic smart speaker amplifiers using + SoundWire digital audio interface. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: sdw10217020400 + + reg: + maxItems: 1 + + powerdown-gpios: + description: Powerdown/Shutdown line to use (pin SD_N) + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + vdd-1p8-supply: true + vdd-io-supply: true + +required: + - compatible + - reg + - powerdown-gpios + - '#sound-dai-cells' + - vdd-1p8-supply + - vdd-io-supply + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + soundwire-controller { + #address-cells = <2>; + #size-cells = <0>; + + speaker@0,1 { + compatible = "sdw10217020400"; + reg = <0 1>; + pinctrl-names = "default"; + pinctrl-0 = <&spkr_2_sd_n_active>; + powerdown-gpios = <&lpass_tlmm 18 GPIO_ACTIVE_LOW>; + #sound-dai-cells = <0>; + sound-name-prefix = "SpkrRight"; + vdd-1p8-supply = <&vreg_l15b_1p8>; + vdd-io-supply = <&vreg_l3g_1p2>; + }; + }; -- 2.34.1

1 2

[PATCH v2] SoC: mediatek: mt8188-mt6359: clean up a return in codec_init
by Dan Carpenter 08 Jun '23

08 Jun '23

This code triggers a Smatch static checker warning and does sort of look like an error path. sound/soc/mediatek/mt8188/mt8188-mt6359.c:597 mt8188_max98390_codec_init() warn: missing error code? 'ret' However, returning 0 is intentional. Make that explicit. Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- v2: At first I thought this was an error path but it is a not. sound/soc/mediatek/mt8188/mt8188-mt6359.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index bc4b74970a46..5ad8d2a5bbc4 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -594,7 +594,7 @@ static int mt8188_max98390_codec_init(struct snd_soc_pcm_runtime *rtd) } if (rtd->dai_link->num_codecs <= 2) - return ret; + return 0; /* add widgets/controls/dapm for rear speakers */ ret = snd_soc_dapm_new_controls(&card->dapm, mt8188_rear_spk_widgets, -- 2.39.2

3 3

[PATCH V3 0/9] ASoC: amd: ps: add SoundWire support
by Vijendar Mukunda 08 Jun '23

08 Jun '23

This patch series add support for - Platform device creation for SoundWire Manager instances and PDM controller. - SoundWire DMA driver. - Interrupt handling for SoundWire manager related interrupts, SoundWire DMA interrupts and ACP error interrupts. - ACP PCI driver PM ops modification with respect to SoundWire Power modes. Changes since v1: - update "soundwire" as "SoundWire" in code. - add comments for platform device mask and platform device count - remove unused variables in acp pci driver private data structure - refactor dma enable register structures in SoundWire DMA driver - add TODO comments in IRQ handler - update IRQ mask values using bit macros - Fix build error reported in Makefile - rename "sdw_dma_stream_instance" structure name as "acp_sdw_dma_stream" changes since v2: - add comments in irq handler. - remove workqueue for SoundWire DMA interrupts and use thread implementation for DMA interrupt handling. - add error checks in sdw_amd_scan_controller() - remove passing "acp_lock" as platform resource for SoundWire DMA driver and PDM driver. - retrieve "acp_lock" reference from parent driver directly and use the reference in SoundWire DMA driver. - add handling for acp pci driver probe even when no ACP PDM or SoundWire manager platform devices created. - Fix indentation for acp_sdw_dma_count structure variables. - Use macro instead of hard coded values for FIFO offset and PTE offset. - Change pm_runtime enable sequence in SoundWire DMA driver probe function. - Refactor system level resume callback in SoundWire DMA driver. Vijendar Mukunda (9): ASoC: amd: ps: create platform devices based on acp config ASoC: amd: ps: handle SoundWire interrupts in acp pci driver ASoC: amd: ps: add SoundWire dma driver ASoC: amd: ps: add SoundWire dma driver dma ops ASoC: amd: ps: add support for SoundWire DMA interrupts ASoC: amd: ps: add pm ops support for SoundWire dma driver ASoC: amd: ps: enable SoundWire dma driver build ASoC: amd: update comments in Kconfig file ASoC: amd: ps: Add SoundWire specific checks in pci driver in pm ops. sound/soc/amd/Kconfig | 3 +- sound/soc/amd/ps/Makefile | 2 + sound/soc/amd/ps/acp63.h | 180 ++++++++++- sound/soc/amd/ps/pci-ps.c | 400 +++++++++++++++++++++-- sound/soc/amd/ps/ps-sdw-dma.c | 579 ++++++++++++++++++++++++++++++++++ 5 files changed, 1138 insertions(+), 26 deletions(-) create mode 100644 sound/soc/amd/ps/ps-sdw-dma.c -- 2.34.1

4 23

[PATCH v3 1/3] docs: sound: add 'pcmtest' driver documentation
by Ivan Orlov 08 Jun '23

08 Jun '23

Add documentation for the new Virtual PCM Test Driver. It covers all possible usage cases: errors and delay injections, random and pattern-based data generation, playback and ioctl redefinition functionalities testing. We have a lot of different virtual media drivers, which can be used for testing of the userspace applications and media subsystem middle layer. However, all of them are aimed at testing the video functionality and simulating the video devices. For audio devices we have only snd-dummy module, which is good in simulating the correct behavior of an ALSA device. I decided to write a tool, which would help to test the userspace ALSA programs (and the PCM middle layer as well) under unusual circumstances to figure out how they would behave. So I came up with this Virtual PCM Test Driver. This new Virtual PCM Test Driver has several features which can be useful during the userspace ALSA applications testing/fuzzing, or testing/fuzzing of the PCM middle layer. Not all of them can be implemented using the existing virtual drivers (like dummy or loopback). Here is what can this driver do: - Simulate both capture and playback processes - Check the playback stream for containing the looped pattern - Generate random or pattern-based capture data - Inject delays into the playback and capturing processes - Inject errors during the PCM callbacks Also, this driver can check the playback stream for containing the predefined pattern, which is used in the corresponding selftest to check the PCM middle layer data transferring functionality. Additionally, this driver redefines the default RESET ioctl, and the selftest covers this PCM API functionality as well. The driver supports both interleaved and non-interleaved access modes, and have separate pattern buffers for each channel. The driver supports up to 4 channels and up to 8 substreams. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- V1 -> V2: - Rename the driver from from 'valsa' to 'pcmtest'. - Implement support for interleaved and non-interleaved access modes - Add support for 8 substreams and 4 channels - Extend supported formats - Extend and rewrite in C the selftest for the driver V2 -> V3: - Add separate pattern buffers for each channel - Speed up the capture data generation when using interleaved access mode - Extend the corresponding selftest to cover the multiple channels capturing and playback functionalities when using interleaved access mode. - Fix documentation issues Documentation/sound/cards/index.rst | 1 + Documentation/sound/cards/pcmtest.rst | 120 ++++++++++++++++++++++++++ 2 files changed, 121 insertions(+) create mode 100644 Documentation/sound/cards/pcmtest.rst diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst index c016f8c3b88b..49c1f2f688f8 100644 --- a/Documentation/sound/cards/index.rst +++ b/Documentation/sound/cards/index.rst @@ -17,3 +17,4 @@ Card-Specific Information hdspm serial-u16550 img-spdif-in + pcmtest diff --git a/Documentation/sound/cards/pcmtest.rst b/Documentation/sound/cards/pcmtest.rst new file mode 100644 index 000000000000..e163522f3205 --- /dev/null +++ b/Documentation/sound/cards/pcmtest.rst @@ -0,0 +1,120 @@ +.. SPDX-License-Identifier: GPL-2.0 + +The Virtual PCM Test Driver +=========================== + +The Virtual PCM Test Driver emulates a generic PCM device, and can be used for +testing/fuzzing of the userspace ALSA applications, as well as for testing/fuzzing of +the PCM middle layer. Additionally, it can be used for simulating hard to reproduce +problems with PCM devices. + +What can this driver do? +~~~~~~~~~~~~~~~~~~~~~~~~ + +At this moment the driver can do the following things: + * Simulate both capture and playback processes + * Generate random or pattern-based capturing data + * Inject delays into the playback and capturing processes + * Inject errors during the PCM callbacks + +It supports up to 8 substreams and 4 channels. Also it supports both interleaved and +non-interleaved access modes. + +Also, this driver can check the playback stream for containing the predefined pattern, +which is used in the corresponding selftest (alsa/pcmtest-test.sh) to check the PCM middle +layer data transferring functionality. Additionally, this driver redefines the default +RESET ioctl, and the selftest covers this PCM API functionality as well. + +Configuration +------------- + +The driver has several parameters besides the common ALSA module parameters: + + * fill_mode (bool) - Buffer fill mode (see below) + * inject_delay (int) + * inject_hwpars_err (bool) + * inject_prepare_err (bool) + * inject_trigger_err (bool) + + +Capture Data Generation +----------------------- + +The driver has two modes of data generation: the first (0 in the fill_mode parameter) +means random data generation, the second (1 in the fill_mode) - pattern-based +data generation. Let's look at the second mode. + +First of all, you may want to specify the pattern for data generation. You can do it +by writing the pattern to the debugfs file. There are pattern buffer debugfs entries +for each channel, as well as entries which contain the pattern buffer length. + + * /sys/kernel/debug/pcmtest/fill_pattern[0-3] + * /sys/kernel/debug/pcmtest/fill_pattern[0-3]_len + +To set the pattern for the channel 0 you can execute the following command: + +.. code-block:: bash + + echo -n mycoolpattern > /sys/kernel/debug/pcmtest/fill_pattern0 + +Then, after every capture action performed on the 'pcmtest' device the buffer for the +channel 0 will contain 'mycoolpatternmycoolpatternmycoolpatternmy...'. + +The pattern itself can be up to 4096 bytes long. + +Delay injection +--------------- + +The driver has 'inject_delay' parameter, which has very self-descriptive name and +can be used for time delay/speedup simulations. The parameter has integer type, and +it means the delay added between module's internal timer ticks. + +If the 'inject_delay' value is positive, the buffer will be filled slower, if it is +negative - faster. You can try it yourself by starting a recording in any +audiorecording application (like Audacity) and selecting the 'pcmtest' device as a +source. + +This parameter can be also used for generating a huge amount of sound data in a very +short period of time (with the negative 'inject_delay' value). + +Errors injection +---------------- + +This module can be used for injecting errors into the PCM communication process. This +action can help you to figure out how the userspace ALSA program behaves under unusual +circumstances. + +For example, you can make all 'hw_params' PCM callback calls return EBUSY error by +writing '1' to the 'inject_hwpars_err' module parameter: + +.. code-block:: bash + + echo 1 > /sys/module/snd_pcmtest/parameters/inject_hwpars_err + +Errors can be injected into the following PCM callbacks: + + * hw_params (EBUSY) + * prepare (EINVAL) + * trigger (EINVAL) + +Playback test +------------- + +This driver can be also used for the playback functionality testing - every time you +write the playback data to the 'pcmtest' PCM device and close it, the driver checks the +buffer for containing the looped pattern (which is specified in the fill_pattern +debugfs file for each channel). If the playback buffer content represents the looped +pattern, 'pc_test' debugfs entry is set into '1'. Otherwise, the driver sets it to '0'. + +ioctl redefinition test +----------------------- + +The driver redefines the 'reset' ioctl, which is default for all PCM devices. To test +this functionality, we can trigger the reset ioctl and check the 'ioctl_test' debugfs +entry: + +.. code-block:: bash + + cat /sys/kernel/debug/pcmtest/ioctl_test + +If the ioctl is triggered successfully, this file will contain '1', and '0' otherwise. -- 2.34.1

3 6

[PATCH 0/4] ASoC: add new trigger ordering method
by Kuninori Morimoto 08 Jun '23

08 Jun '23

Hi Mark This patch-set adds new "trigger" starting/stopping method. Kuninori Morimoto (4): ASoC: add new trigger ordering method ASoC: amd: use use new trigger ordering method ASoC: atmel: use use new trigger ordering method ASoC: remove old trigger ordering method include/sound/soc-component.h | 11 ++- include/sound/soc.h | 23 ++++-- sound/soc/amd/acp-da7219-max98357a.c | 20 ++--- sound/soc/amd/acp-es8336.c | 2 +- sound/soc/atmel/mchp-pdmc.c | 2 +- sound/soc/soc-pcm.c | 105 ++++++++++++++------------- 6 files changed, 93 insertions(+), 70 deletions(-) -- 2.25.1

1 5

[PATCH] ASoC: codecs: wsa883x: use existing define instead of raw value
by Krzysztof Kozlowski 07 Jun '23

07 Jun '23

Use existing define for WSA883X_GLOBAL_PA_ENABLE instead of hard-coded value, just like in other places in this driver. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/wsa883x.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index c609cb63dae6..5c1cfceb2956 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -1334,7 +1334,8 @@ static int wsa883x_digital_mute(struct snd_soc_dai *dai, int mute, int stream) WSA883X_DRE_GAIN_EN_MASK, WSA883X_DRE_GAIN_FROM_CSR); snd_soc_component_write_field(component, WSA883X_PA_FSM_CTL, - WSA883X_GLOBAL_PA_EN_MASK, 1); + WSA883X_GLOBAL_PA_EN_MASK, + WSA883X_GLOBAL_PA_ENABLE); } -- 2.34.1

2 1

[PATCH 0/3] ASoC: stm32: fix dtbs_check warnings
by Olivier Moysan 07 Jun '23

07 Jun '23

Fix dtbs_check warnings in STM32MP15 DK boards Devices Trees for STM32 I2S and Cirrus CS42L51 codec. - Add OF graph port property in I2S and CS42L51 DT bindings. Fixes warnings: audio-controller@4000b000: Unevaluated properties are not allowed ('port' was unexpected) cs42l51@4a: Unevaluated properties are not allowed ('port' was unexpected) - Correct OF graph DAI audio format property for STM32MP15x Dkx I2S node Olivier Moysan (3): ASoC: dt-bindings: stm32: document audio of graph port for i2s ASoC: dt-bindings: document audio of graph port for cs42l51 ARM: dts: stm32: fix i2s endpoint format property for stm32mp15xx-dkx Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml | 4 ++++ Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml | 4 ++++ arch/arm/boot/dts/stm32mp15xx-dkx.dtsi | 2 +- 3 files changed, 9 insertions(+), 1 deletion(-) -- 2.25.1

4 6

[PATCH] ASoC: mediatek: mt8188-mt6359: add i2c dependency
by Arnd Bergmann 07 Jun '23

07 Jun '23

From: Arnd Bergmann <arnd(a)arndb.de> The newly added driver is missing this dependency, causing a possible build failure: WARNING: unmet direct dependencies detected for SND_SOC_MAX98390 WARNING: unmet direct dependencies detected for SND_SOC_NAU8825 Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=m] Selected by [y]: - SND_SOC_MT8188_MT6359 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_MT8188 [=y] && MTK_PMIC_WRAP [=y] aarch64-linux-ld: sound/soc/codecs/max98390.o: in function `max98390_i2c_probe': max98390.c:(.text+0x514): undefined reference to `__devm_regmap_init_i2c' Fixes: 9f08dcbddeb30 ("ASoC: mediatek: mt8188-mt6359: support new board with nau88255") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/mediatek/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 4ea012342b522..90db67e0ce4fb 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -223,6 +223,7 @@ config SND_SOC_MT8188 config SND_SOC_MT8188_MT6359 tristate "ASoC Audio driver for MT8188 with MT6359 and I2S codecs" depends on SND_SOC_MT8188 && MTK_PMIC_WRAP + depends on I2C select SND_SOC_MT6359 select SND_SOC_HDMI_CODEC select SND_SOC_DMIC -- 2.39.2

2 1

[PATCH 0/2] ASoC: Intel: HDA: two cleanups
by Pierre-Louis Bossart 07 Jun '23

07 Jun '23

One removal of a non-existent Kconfig initialization and one correction to enable SoundWire link when no HDaudio external codec is detected. This will generate more information and help resolve cases where the ACPI information is incorrect. Pierre-Louis Bossart (2): ASoC: SOF: Intel: fix SoundWire/HDaudio mutual exclusion ASoC: SOF: Intel: hda-pcm: remove kernel parameter init sound/soc/sof/intel/hda-pcm.c | 2 +- sound/soc/sof/intel/hda.c | 12 +++++++++++- 2 files changed, 12 insertions(+), 2 deletions(-) -- 2.37.2

2 3

[PATCH v2 0/9] Add support for IIO devices in ASoC
by Herve Codina 07 Jun '23

07 Jun '23

Several weeks ago, I sent a series [1] for adding a potentiometer as an auxiliary device in ASoC. The feedback was that the potentiometer should be directly handled in IIO (as other potentiometers) and something more generic should be present in ASoC in order to have a binding to import some IIO devices into sound cards. The series related to the IIO potentiometer device is already under review [2]. This series introduces audio-iio-aux. Its goal is to offer the binding between IIO and ASoC. It exposes attached IIO devices as ASoC auxiliary devices and allows to control them through mixer controls. On my system, the IIO device is a potentiometer and it is present in an amplifier design present in the audio path. Compare to the previous iteration https://lore.kernel.org/linux-kernel/20230421124122.324820-1-herve.codina@b… This v2 series mainly: - updates the binding using a simple-card subnode and handles this new subnode in the simple-card driver. - Improves existing IIO code and documentation. - Renames simple-iio-aux to audio-iio-aux and fixes the driver itself. Best regards, Hervé [1] https://lore.kernel.org/linux-kernel/20230203111422.142479-1-herve.codina@b… [2] https://lore.kernel.org/linux-kernel/20230421085245.302169-1-herve.codina@b… Changes v1 -> v2 - Patch 1 Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the /schemas/iio/iio-consumer.yaml reference Remove the unneeded '|' after description - Patch 2 (new in v2) Introduce the simple-audio-card additional-devs subnode - Patch 3 (new in v2) Check err before switch() in iio_channel_read_max() - Patch 4 (new in v2) Fix raw reads and raw writes documentation - Patch 5 (patch 2 in v1) Check err before switch() in iio_channel_read_min() Fix documentation - Patch 6 (path 3 in v1) No changes - Patch 7 (patch 4 in v1) Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the mask usage from audio_iio_aux_{get,put}_volsw helpers Use directly PTR_ERR() in dev_err_probe() parameter Remove the '!!' construction Remove of_match_ptr() - Patch 8 (new in v2) Add a missing of_node_put() in the simple-card driver - Patch 9 (new in v2) Handle additional-devs in the simple-card driver Herve Codina (9): ASoC: dt-bindings: Add audio-iio-aux ASoC: dt-bindings: simple-card: Add additional-devs subnode iio: inkern: Check error explicitly in iio_channel_read_max() iio: consumer.h: Fix raw values documentation notes iio: inkern: Add a helper to query an available minimum raw value ASoC: soc-dapm.h: Add a helper to build a DAPM widget dynamically ASoC: codecs: Add support for the generic IIO auxiliary devices ASoC: simple-card: Add missing of_node_put() in case of error ASoC: simple-card: Handle additional devices .../bindings/sound/audio-iio-aux.yaml | 64 ++++ .../bindings/sound/simple-card.yaml | 53 +++ drivers/iio/inkern.c | 75 ++++- include/linux/iio/consumer.h | 37 ++- include/sound/soc-dapm.h | 12 +- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/audio-iio-aux.c | 302 ++++++++++++++++++ sound/soc/generic/simple-card.c | 53 ++- 9 files changed, 596 insertions(+), 14 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-iio-aux.yaml create mode 100644 sound/soc/codecs/audio-iio-aux.c -- 2.40.1

7 37

[PATCH v2 00/26] ASoC/soundwire: add support for ACE2.x
by Bard Liao 07 Jun '23

07 Jun '23

This series uses the abstraction added in past kernel cycles to provide support for the ACE2.x integration. The existing SHIM and Cadence registers are now split in 3 (SHIM, IP, SHIM vendor-specific), with some parts also moved to the HDaudio Extended Multi link structures. Nothing fundamentally different except for the register map. This series only provides the basic mechanisms to expose SoundWire-based DAIs. The PCI parts and DSP management will be contributed later, and the DAI ops are now empty as well. The change is mainly on SoundWire. It would be better to go through SoundWire tree. v2: - Some cleanup for free, trigger, hw_params callbacks before introducing ACE2.x callbacks. Pierre-Louis Bossart (26): ASoC: SOF: Intel: shim: add enum for ACE 2.0 IP used in LunarLake soundwire: intel: add ACE2.x SHIM definitions soundwire: intel_ace2x: add empty new ops for LunarLake soundwire/ASOC: Intel: update offsets for LunarLake soundwire: intel/cadence: set ip_offset at run-time ASoC/soundwire: intel: pass hdac_bus pointer for link management soundwire: intel: add eml_lock in the interface for new platforms ASoC: SOF: Intel: hda: retrieve SoundWire eml_lock and pass pointer soundwire: intel_init: use eml_lock parameter soundwire: intel_ace2x: add debugfs support soundwire: intel_ace2x: add link power-up/down helpers soundwire: intel_ace2x: set SYNCPRD before powering-up soundwire: intel_ace2x: configure link PHY soundwire: intel_ace2x: add DAI registration soundwire: intel_ace2x: add sync_arm/sync_go helpers soundwire: intel_ace2x: use common helpers for bus start/stop soundwire: intel_ace2x: enable wake support soundwire: intel_ace2x: add check_cmdsync_unlocked helper soundwire: bus: add new manager callback to deal with peripheral enumeration soundwire: intel_ace2x: add new_peripheral_assigned callback soundwire: intel_ace2x: add pre/post bank switch callbacks ASoC: SOF/soundwire: re-add substream in params_stream structure soundwire: intel: remove .trigger callback implementation soundwire: intel: use substream for .trigger callback soundwire: intel: remove .free callback implementation soundwire: intel: use substream for .free callback drivers/soundwire/Makefile | 3 +- drivers/soundwire/bus.c | 3 + drivers/soundwire/cadence_master.h | 2 + drivers/soundwire/intel.c | 54 +--- drivers/soundwire/intel.h | 16 + drivers/soundwire/intel_ace2x.c | 393 ++++++++++++++++++++++++ drivers/soundwire/intel_ace2x_debugfs.c | 147 +++++++++ drivers/soundwire/intel_auxdevice.c | 17 + drivers/soundwire/intel_init.c | 21 +- include/linux/soundwire/sdw.h | 3 +- include/linux/soundwire/sdw_intel.h | 94 +++++- sound/soc/sof/intel/hda.c | 33 +- sound/soc/sof/intel/shim.h | 1 + 13 files changed, 723 insertions(+), 64 deletions(-) create mode 100644 drivers/soundwire/intel_ace2x.c create mode 100644 drivers/soundwire/intel_ace2x_debugfs.c -- 2.25.1

3 30

[PATCH v4 1/3] docs: sound: add 'pcmtest' driver documentation
by Ivan Orlov 07 Jun '23

07 Jun '23

Add documentation for the new Virtual PCM Test Driver. It covers all possible usage cases: errors and delay injections, random and pattern-based data generation, playback and ioctl redefinition functionalities testing. We have a lot of different virtual media drivers, which can be used for testing of the userspace applications and media subsystem middle layer. However, all of them are aimed at testing the video functionality and simulating the video devices. For audio devices we have only snd-dummy module, which is good in simulating the correct behavior of an ALSA device. I decided to write a tool, which would help to test the userspace ALSA programs (and the PCM middle layer as well) under unusual circumstances to figure out how they would behave. So I came up with this Virtual PCM Test Driver. This new Virtual PCM Test Driver has several features which can be useful during the userspace ALSA applications testing/fuzzing, or testing/fuzzing of the PCM middle layer. Not all of them can be implemented using the existing virtual drivers (like dummy or loopback). Here is what can this driver do: - Simulate both capture and playback processes - Check the playback stream for containing the looped pattern - Generate random or pattern-based capture data - Inject delays into the playback and capturing processes - Inject errors during the PCM callbacks Also, this driver can check the playback stream for containing the predefined pattern, which is used in the corresponding selftest to check the PCM middle layer data transferring functionality. Additionally, this driver redefines the default RESET ioctl, and the selftest covers this PCM API functionality as well. The driver supports both interleaved and non-interleaved access modes, and have separate pattern buffers for each channel. The driver supports up to 4 channels and up to 8 substreams. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- V1 -> V2: - Rename the driver from from 'valsa' to 'pcmtest'. - Implement support for interleaved and non-interleaved access modes - Add support for 8 substreams and 4 channels - Extend supported formats - Extend and rewrite in C the selftest for the driver V2 -> V3: - Add separate pattern buffers for each channel - Speed up the capture data generation when using interleaved access mode - Extend the corresponding selftest to cover the multiple channels capturing and playback functionalities when using interleaved access mode. - Fix documentation issues V3 -> V4: - Fix issue in the selftest: there was a typo in the fscanf argument. Documentation/sound/cards/index.rst | 1 + Documentation/sound/cards/pcmtest.rst | 120 ++++++++++++++++++++++++++ 2 files changed, 121 insertions(+) create mode 100644 Documentation/sound/cards/pcmtest.rst diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst index c016f8c3b88b..49c1f2f688f8 100644 --- a/Documentation/sound/cards/index.rst +++ b/Documentation/sound/cards/index.rst @@ -17,3 +17,4 @@ Card-Specific Information hdspm serial-u16550 img-spdif-in + pcmtest diff --git a/Documentation/sound/cards/pcmtest.rst b/Documentation/sound/cards/pcmtest.rst new file mode 100644 index 000000000000..e163522f3205 --- /dev/null +++ b/Documentation/sound/cards/pcmtest.rst @@ -0,0 +1,120 @@ +.. SPDX-License-Identifier: GPL-2.0 + +The Virtual PCM Test Driver +=========================== + +The Virtual PCM Test Driver emulates a generic PCM device, and can be used for +testing/fuzzing of the userspace ALSA applications, as well as for testing/fuzzing of +the PCM middle layer. Additionally, it can be used for simulating hard to reproduce +problems with PCM devices. + +What can this driver do? +~~~~~~~~~~~~~~~~~~~~~~~~ + +At this moment the driver can do the following things: + * Simulate both capture and playback processes + * Generate random or pattern-based capturing data + * Inject delays into the playback and capturing processes + * Inject errors during the PCM callbacks + +It supports up to 8 substreams and 4 channels. Also it supports both interleaved and +non-interleaved access modes. + +Also, this driver can check the playback stream for containing the predefined pattern, +which is used in the corresponding selftest (alsa/pcmtest-test.sh) to check the PCM middle +layer data transferring functionality. Additionally, this driver redefines the default +RESET ioctl, and the selftest covers this PCM API functionality as well. + +Configuration +------------- + +The driver has several parameters besides the common ALSA module parameters: + + * fill_mode (bool) - Buffer fill mode (see below) + * inject_delay (int) + * inject_hwpars_err (bool) + * inject_prepare_err (bool) + * inject_trigger_err (bool) + + +Capture Data Generation +----------------------- + +The driver has two modes of data generation: the first (0 in the fill_mode parameter) +means random data generation, the second (1 in the fill_mode) - pattern-based +data generation. Let's look at the second mode. + +First of all, you may want to specify the pattern for data generation. You can do it +by writing the pattern to the debugfs file. There are pattern buffer debugfs entries +for each channel, as well as entries which contain the pattern buffer length. + + * /sys/kernel/debug/pcmtest/fill_pattern[0-3] + * /sys/kernel/debug/pcmtest/fill_pattern[0-3]_len + +To set the pattern for the channel 0 you can execute the following command: + +.. code-block:: bash + + echo -n mycoolpattern > /sys/kernel/debug/pcmtest/fill_pattern0 + +Then, after every capture action performed on the 'pcmtest' device the buffer for the +channel 0 will contain 'mycoolpatternmycoolpatternmycoolpatternmy...'. + +The pattern itself can be up to 4096 bytes long. + +Delay injection +--------------- + +The driver has 'inject_delay' parameter, which has very self-descriptive name and +can be used for time delay/speedup simulations. The parameter has integer type, and +it means the delay added between module's internal timer ticks. + +If the 'inject_delay' value is positive, the buffer will be filled slower, if it is +negative - faster. You can try it yourself by starting a recording in any +audiorecording application (like Audacity) and selecting the 'pcmtest' device as a +source. + +This parameter can be also used for generating a huge amount of sound data in a very +short period of time (with the negative 'inject_delay' value). + +Errors injection +---------------- + +This module can be used for injecting errors into the PCM communication process. This +action can help you to figure out how the userspace ALSA program behaves under unusual +circumstances. + +For example, you can make all 'hw_params' PCM callback calls return EBUSY error by +writing '1' to the 'inject_hwpars_err' module parameter: + +.. code-block:: bash + + echo 1 > /sys/module/snd_pcmtest/parameters/inject_hwpars_err + +Errors can be injected into the following PCM callbacks: + + * hw_params (EBUSY) + * prepare (EINVAL) + * trigger (EINVAL) + +Playback test +------------- + +This driver can be also used for the playback functionality testing - every time you +write the playback data to the 'pcmtest' PCM device and close it, the driver checks the +buffer for containing the looped pattern (which is specified in the fill_pattern +debugfs file for each channel). If the playback buffer content represents the looped +pattern, 'pc_test' debugfs entry is set into '1'. Otherwise, the driver sets it to '0'. + +ioctl redefinition test +----------------------- + +The driver redefines the 'reset' ioctl, which is default for all PCM devices. To test +this functionality, we can trigger the reset ioctl and check the 'ioctl_test' debugfs +entry: + +.. code-block:: bash + + cat /sys/kernel/debug/pcmtest/ioctl_test + +If the ioctl is triggered successfully, this file will contain '1', and '0' otherwise. -- 2.34.1

2 2

[PATCH] ASoC: soc.h: remove snd_soc_compr_ops :: trigger
by Kuninori Morimoto 07 Jun '23

07 Jun '23

ASoC framework is not using trigger call-back for snd_soc_compr_ops. This patch remove it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ed79b27b0392..10e4ea0664af 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -633,7 +633,6 @@ struct snd_soc_compr_ops { int (*startup)(struct snd_compr_stream *); void (*shutdown)(struct snd_compr_stream *); int (*set_params)(struct snd_compr_stream *); - int (*trigger)(struct snd_compr_stream *); }; struct snd_soc_component* -- 2.25.1

2 1

[PATCH] ASoC: add snd_soc_get_stream_cpu()
by Kuninori Morimoto 07 Jun '23

07 Jun '23

We are using get_stream_cpu() to get CPU stream which cares Codec2Codec. But it is static function for now, and we want to use it from other files. This patch makes it global function. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 34 ++++++++++++++++++++++++++++++++++ sound/soc/soc-dapm.c | 35 +---------------------------------- sound/soc/soc-pcm.c | 6 ++---- 4 files changed, 38 insertions(+), 38 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 533e553a343f..ed79b27b0392 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1292,6 +1292,7 @@ unsigned int snd_soc_daifmt_parse_clock_provider_raw(struct device_node *np, snd_soc_daifmt_clock_provider_from_bitmap( \ snd_soc_daifmt_parse_clock_provider_as_bitmap(np, prefix)) +int snd_soc_get_stream_cpu(struct snd_soc_dai_link *dai_link, int stream); int snd_soc_get_dai_id(struct device_node *ep); int snd_soc_get_dai_name(const struct of_phandle_args *args, const char **dai_name); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b48efc3a08d2..e8308926bd98 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3196,6 +3196,40 @@ unsigned int snd_soc_daifmt_parse_clock_provider_raw(struct device_node *np, } EXPORT_SYMBOL_GPL(snd_soc_daifmt_parse_clock_provider_raw); +int snd_soc_get_stream_cpu(struct snd_soc_dai_link *dai_link, int stream) +{ + /* + * [Normal] + * + * Playback + * CPU : SNDRV_PCM_STREAM_PLAYBACK + * Codec: SNDRV_PCM_STREAM_PLAYBACK + * + * Capture + * CPU : SNDRV_PCM_STREAM_CAPTURE + * Codec: SNDRV_PCM_STREAM_CAPTURE + */ + if (!dai_link->c2c_params) + return stream; + + /* + * [Codec2Codec] + * + * Playback + * CPU : SNDRV_PCM_STREAM_CAPTURE + * Codec: SNDRV_PCM_STREAM_PLAYBACK + * + * Capture + * CPU : SNDRV_PCM_STREAM_PLAYBACK + * Codec: SNDRV_PCM_STREAM_CAPTURE + */ + if (stream == SNDRV_PCM_STREAM_CAPTURE) + return SNDRV_PCM_STREAM_PLAYBACK; + + return SNDRV_PCM_STREAM_CAPTURE; +} +EXPORT_SYMBOL_GPL(snd_soc_get_stream_cpu); + int snd_soc_get_dai_id(struct device_node *ep) { struct snd_soc_component *component; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f2f04ce693a1..09a197ad6c2c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4322,39 +4322,6 @@ static void dapm_connect_dai_routes(struct snd_soc_dapm_context *dapm, snd_soc_dapm_add_path(dapm, src, sink, NULL, NULL); } -static int get_stream_cpu(struct snd_soc_dai_link *dai_link, int stream) -{ - /* - * [Normal] - * - * Playback - * CPU : SNDRV_PCM_STREAM_PLAYBACK - * Codec: SNDRV_PCM_STREAM_PLAYBACK - * - * Playback - * CPU : SNDRV_PCM_STREAM_CAPTURE - * Codec: SNDRV_PCM_STREAM_CAPTURE - */ - if (!dai_link->c2c_params) - return stream; - - /* - * [Codec2Codec] - * - * Playback - * CPU : SNDRV_PCM_STREAM_CAPTURE - * Codec: SNDRV_PCM_STREAM_PLAYBACK - * - * Capture - * CPU : SNDRV_PCM_STREAM_PLAYBACK - * Codec: SNDRV_PCM_STREAM_CAPTURE - */ - if (stream == SNDRV_PCM_STREAM_CAPTURE) - return SNDRV_PCM_STREAM_PLAYBACK; - - return SNDRV_PCM_STREAM_CAPTURE; -} - static void dapm_connect_dai_pair(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *codec_dai, @@ -4372,7 +4339,7 @@ static void dapm_connect_dai_pair(struct snd_soc_card *card, for_each_pcm_streams(stream) { int stream_cpu, stream_codec; - stream_cpu = get_stream_cpu(dai_link, stream); + stream_cpu = snd_soc_get_stream_cpu(dai_link, stream); stream_codec = stream; /* connect BE DAI playback if widgets are valid */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2abeb6f6a4a8..f16402ee5a19 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2781,10 +2781,8 @@ static int soc_get_playback_capture(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *codec_dai; /* Adapt stream for codec2codec links */ - int cpu_capture = dai_link->c2c_params ? - SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; - int cpu_playback = dai_link->c2c_params ? - SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + int cpu_capture = snd_soc_get_stream_cpu(dai_link, SNDRV_PCM_STREAM_CAPTURE); + int cpu_playback = snd_soc_get_stream_cpu(dai_link, SNDRV_PCM_STREAM_PLAYBACK); for_each_rtd_codec_dais(rtd, i, codec_dai) { if (dai_link->num_cpus == 1) { -- 2.25.1

2 1

[PATCH] ASoC: mediatek: mt8188-mt6359: Fix error code in codec_init
by Dan Carpenter 07 Jun '23

07 Jun '23

Return -EINVAL on this error path instead of returning success. Fixes: 9f08dcbddeb3 ("ASoC: mediatek: mt8188-mt6359: support new board with nau88255") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- This is based on static analysis and not tested. sound/soc/mediatek/mt8188/mt8188-mt6359.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index bc4b74970a46..ae02bbcd5d4e 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -594,7 +594,7 @@ static int mt8188_max98390_codec_init(struct snd_soc_pcm_runtime *rtd) } if (rtd->dai_link->num_codecs <= 2) - return ret; + return -EINVAL; /* add widgets/controls/dapm for rear speakers */ ret = snd_soc_dapm_new_controls(&card->dapm, mt8188_rear_spk_widgets, -- 2.39.2

2 2

[PATCH RESEND v3 0/4] Add Loongson HD Audio support
by Yanteng Si 07 Jun '23

07 Jun '23

v3: * Adding some comments. * Seting polling_mode flag. v2: * Handling with two new flags added to struct hdac_bus. v1: * The Loongson HDA controller is compatible with High Definition Audio Specification Revision 1.0a. See <https://loongson.github.io/LoongArch-Documentation/Loongson- 7A1000-usermanual-EN.html#hda-controller> Yanteng Si (4): ALSA: hda: Add Loongson LS7A HD-Audio support ALSA: hda: Using polling mode for loongson controller by default ALSA: hda: Workaround for SDnCTL register on loongson ALSA: hda/intel: Workaround for WALLCLK register for loongson controller include/linux/pci_ids.h | 3 +++ include/sound/hdaudio.h | 2 ++ sound/hda/hdac_controller.c | 5 ++++- sound/hda/hdac_device.c | 1 + sound/hda/hdac_stream.c | 6 +++++- sound/pci/hda/hda_intel.c | 20 ++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 1 + 7 files changed, 36 insertions(+), 2 deletions(-) -- 2.31.4

3 6

[PATCH v3 0/4] Add Loongson HD Audio support
by Yanteng Si 07 Jun '23

07 Jun '23

v3: * Adding some comments. * Seting polling_mode flag. v2: * Handling with two new flags added to struct hdac_bus. v1: * The Loongson HDA controller is compatible with High Definition Audio Specification Revision 1.0a. See <https://loongson.github.io/LoongArch-Documentation/Loongson- 7A1000-usermanual-EN.html#hda-controller> Yanteng Si (4): ALSA: hda: Add Loongson LS7A HD-Audio support ALSA: hda: Using polling mode for loongson controller by default ALSA: hda: Workaround for SDnCTL register on loongson ALSA: hda/intel: Workaround for WALLCLK register for loongson controller include/linux/pci_ids.h | 3 +++ include/sound/hdaudio.h | 3 +++ sound/hda/hdac_controller.c | 5 ++++- sound/hda/hdac_device.c | 1 + sound/hda/hdac_stream.c | 6 +++++- sound/pci/hda/hda_intel.c | 20 ++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 1 + 7 files changed, 37 insertions(+), 2 deletions(-) -- 2.31.4

1 5

[PATCH v2 0/4] Add Loongson HD Audio support
by Yanteng Si 07 Jun '23

07 Jun '23

v2: * Handling with two new flags added to struct hdac_bus. v1: * The Loongson HDA controller is compatible with High Definition Audio Specification Revision 1.0a. See <https://loongson.github.io/LoongArch-Documentation/Loongson- 7A1000-usermanual-EN.html#hda-controller> Yanteng Si (4): ALSA: hda: Add Loongson LS7A HD-Audio support ALSA: hda: Using polling mode for loongson controller by default ALSA: hda: Workaround for SDnCTL register on loongson controller ALSA: hda/intel: Workaround for WALLCLK register for loongson controller include/linux/pci_ids.h | 3 +++ include/sound/hdaudio.h | 2 ++ sound/hda/hdac_controller.c | 5 ++++- sound/hda/hdac_device.c | 1 + sound/hda/hdac_stream.c | 6 +++++- sound/pci/hda/hda_intel.c | 15 +++++++++++++++ sound/pci/hda/patch_hdmi.c | 1 + 7 files changed, 31 insertions(+), 2 deletions(-) -- 2.31.4

2 7

[PATCH v4 2/3] ALSA: Implement the new Virtual PCM Test Driver
by ivan.orlov0322＠gmail.com 06 Jun '23

06 Jun '23

The content of this message was lost. It was probably cross-posted to multiple lists and previously handled on another list.

1 0

[PATCH v2 0/3] ES8316 audio codec fixes on Rock5B
by Cristian Ciocaltea 06 Jun '23

06 Jun '23

This patch series handles a few issues related to the ES8316 audio codec, discovered while doing some testing on the Rock 5B board. Changes in v2: - Preserved original dB gain range in PATCH 1 - Rewrote PATCH 2 conditional statement, per Mark's review - Rebased series onto next-20230530 - v1: https://lore.kernel.org/all/20230524074156.147387-1-cristian.ciocaltea@coll… Cristian Ciocaltea (3): ASoC: es8316: Increment max value for ALC Capture Target Volume control ASoC: es8316: Do not set rate constraints for unsupported MCLKs arm64: dts: rockchip: Assign ES8316 MCLK rate on rk3588-rock-5b .../boot/dts/rockchip/rk3588-rock-5b.dts | 2 ++ sound/soc/codecs/es8316.c | 23 +++++++++++-------- 2 files changed, 16 insertions(+), 9 deletions(-) -- 2.40.1

3 5

[PATCH v4 5/6] ALSA: hda/tas2781: Add tas2781 HDA driver
by Shenghao Ding 06 Jun '23

06 Jun '23

Create tas2781 HDA driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v4: - Add tiwai(a)suse.de into Cc list - remove unused ret in tas2781_hda_playback - in all *__put function, return 0, if the value is unchanged - remove superfluous - rewrite the subid judgement - dev_info to dev_dbg Changes to be committed: modified: sound/pci/hda/Kconfig modified: sound/pci/hda/Makefile new file: sound/pci/hda/tas2781_hda_i2c.c --- sound/pci/hda/Kconfig | 15 + sound/pci/hda/Makefile | 2 + sound/pci/hda/tas2781_hda_i2c.c | 834 ++++++++++++++++++++++++++++++++ 3 files changed, 851 insertions(+) create mode 100644 sound/pci/hda/tas2781_hda_i2c.c diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 886255a03e8b..e66257277492 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -130,6 +130,21 @@ config SND_HDA_SCODEC_CS35L41_SPI comment "Set to Y if you want auto-loading the side codec driver" depends on SND_HDA=y && SND_HDA_SCODEC_CS35L41_SPI=m +config SND_HDA_SCODEC_TAS2781_I2C + tristate "Build TAS2781 HD-audio side codec support for I2C Bus" + depends on I2C + depends on ACPI + depends on SND_SOC + select SND_SOC_TAS2781_COMLIB + select SND_SOC_TAS2781_FMWLIB + select CRC32_SARWATE + help + Say Y or M here to include TAS2781 I2C HD-audio side codec support + in snd-hda-intel driver, such as ALC287. + +comment "Set to Y if you want auto-loading the side codec driver" + depends on SND_HDA=y && SND_HDA_SCODEC_TAS2781_I2C=m + config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 00d306104484..1c76609690d6 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -32,6 +32,7 @@ snd-hda-scodec-cs35l41-objs := cs35l41_hda.o snd-hda-scodec-cs35l41-i2c-objs := cs35l41_hda_i2c.o snd-hda-scodec-cs35l41-spi-objs := cs35l41_hda_spi.o snd-hda-cs-dsp-ctls-objs := hda_cs_dsp_ctl.o +snd-hda-scodec-tas2781-i2c-objs := tas2781_hda_i2c.o # common driver obj-$(CONFIG_SND_HDA) := snd-hda-codec.o @@ -56,6 +57,7 @@ obj-$(CONFIG_SND_HDA_SCODEC_CS35L41) += snd-hda-scodec-cs35l41.o obj-$(CONFIG_SND_HDA_SCODEC_CS35L41_I2C) += snd-hda-scodec-cs35l41-i2c.o obj-$(CONFIG_SND_HDA_SCODEC_CS35L41_SPI) += snd-hda-scodec-cs35l41-spi.o obj-$(CONFIG_SND_HDA_CS_DSP_CONTROLS) += snd-hda-cs-dsp-ctls.o +obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_I2C) += snd-hda-scodec-tas2781-i2c.o # this must be the last entry after codec drivers; # otherwise the codec patches won't be hooked before the PCI probe diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c new file mode 100644 index 000000000000..e124575d3737 --- /dev/null +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -0,0 +1,834 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2781 HDA I2C driver +// +// Copyright 2023 Texas Instruments, Inc. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> + +#include <linux/acpi.h> +#include <linux/crc8.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/firmware.h> +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <sound/hda_codec.h> +#include <sound/soc.h> +#include <sound/tas2781.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "hda_local.h" +#include "hda_auto_parser.h" +#include "hda_component.h" +#include "hda_jack.h" +#include "hda_generic.h" + +#define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 + +#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .invert = xinvert} } + +enum calib_data { + R0_VAL = 0, + INV_R0, + R0LOW, + POWER, + TLIM, + CALIB_MAX +}; + +static int tas2781_acpi_get_i2c_resource(struct acpi_resource + *ares, void *data) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)data; + struct acpi_resource_i2c_serialbus *sb; + + if (i2c_acpi_get_i2c_resource(ares, &sb)) { + if (sb->slave_address != TAS2781_GLOBAL_ADDR) { + tas_priv->tasdevice[tas_priv->ndev].dev_addr = + (unsigned int)sb->slave_address; + tas_priv->ndev++; + } else + tas_priv->glb_addr.dev_addr = TAS2781_GLOBAL_ADDR; + + } + + return 1; +} + +static int tas2781_hda_read_acpi(struct tasdevice_priv *tas_priv, + const char *hid) +{ + struct acpi_device *adev; + struct device *physdev; + LIST_HEAD(resources); + const char *sub; + int ret; + + adev = acpi_dev_get_first_match_dev(hid, NULL, -1); + if (!adev) { + dev_err(tas_priv->dev, + "Failed to find an ACPI device for %s\n", hid); + return -ENODEV; + } + strcpy(tas_priv->dev_name, hid); + physdev = get_device(acpi_get_first_physical_node(adev)); + acpi_dev_put(adev); + + sub = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); + if (IS_ERR(sub)) + sub = NULL; + dev_info(tas_priv->dev, "subid = %s\n", sub); + + tas_priv->acpi_subsystem_id = sub; + + ret = acpi_dev_get_resources(adev, &resources, + tas2781_acpi_get_i2c_resource, tas_priv); + if (ret < 0) + goto err; + acpi_dev_free_resource_list(&resources); + + put_device(physdev); + dev_info(tas_priv->dev, "global addr: 0x%02x devcnt = %d\n", + tas_priv->glb_addr.dev_addr, tas_priv->ndev); + + return 0; + +err: + dev_err(tas_priv->dev, "Failed acpi ret: %d\n", ret); + put_device(physdev); + + return ret; +} + +static void tas2781_hda_playback_hook(struct device *dev, int action) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + dev_dbg(tas_priv->dev, "%s: action = %d\n", __func__, action); + switch (action) { + case HDA_GEN_PCM_ACT_OPEN: + pm_runtime_get_sync(dev); + mutex_lock(&tas_priv->codec_lock); + tas_priv->cur_conf = 0; + tas_priv->rcabin.profile_cfg_id = 1; + tasdevice_tuning_switch(tas_priv, 0); + mutex_unlock(&tas_priv->codec_lock); + break; + case HDA_GEN_PCM_ACT_CLOSE: + mutex_lock(&tas_priv->codec_lock); + tasdevice_tuning_switch(tas_priv, 1); + mutex_unlock(&tas_priv->codec_lock); + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + break; + default: + dev_warn(tas_priv->dev, "Playback action not supported: %d\n", + action); + break; + } +} + +static int tasdevice_info_profile(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->rcabin.ncfgs - 1; + + return 0; +} + +static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + + return 0; +} + +static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int ret = 0; + + if (tas_priv->rcabin.profile_cfg_id != + ucontrol->value.integer.value[0]) { + tas_priv->rcabin.profile_cfg_id = + ucontrol->value.integer.value[0]; + ret = 0; + } + + return ret; +} + +static int tasdevice_create_control(struct tasdevice_priv *tas_priv) +{ + char prof_ctrl_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + struct hda_codec *codec = tas_priv->codec; + struct snd_kcontrol_new prof_ctrl = { + .name = prof_ctrl_name, + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_profile, + .get = tasdevice_get_profile_id, + .put = tasdevice_set_profile_id, + }; + int ret; + + /* Create a mixer item for selecting the active profile */ + scnprintf(prof_ctrl_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "spk-profile-id"); + ret = snd_ctl_add(codec->card, snd_ctl_new1(&prof_ctrl, tas_priv)); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + prof_ctrl.name, ret); + goto out; + } + + dev_dbg(tas_priv->dev, "Added Control %s\n", prof_ctrl.name); + +out: + return ret; +} + +static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_fw->nr_programs; + + return 0; +} + +static int tasdevice_info_configurations( + struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_fw->nr_configurations - 1; + + return 0; +} + +static int tasdevice_program_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + + return 0; +} + +static int tasdevice_program_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + unsigned int nr_program = ucontrol->value.integer.value[0]; + int ret = 0; + + if (tas_priv->cur_prog != nr_program) { + tas_priv->cur_prog = nr_program; + ret = 1; + } + + return ret; +} + +static int tasdevice_config_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + + return 0; +} + +static int tasdevice_config_put( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + unsigned int nr_configuration = ucontrol->value.integer.value[0]; + int ret = 0; + + if (tas_priv->cur_conf != nr_configuration) { + tas_priv->cur_conf = nr_configuration; + ret = 1; + } + + return ret; +} + +/* + * tas2781_digital_getvol - get the volum control + * @kcontrol: control pointer + * @ucontrol: User data + * Customer Kcontrol for tas2781 is primarily for regmap booking, paging + * depends on internal regmap mechanism. + * tas2781 contains book and page two-level register map, especially + * book switching will set the register BXXP00R7F, after switching to the + * correct book, then leverage the mechanism for paging to access the + * register. + */ +static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_digital_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_amp_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_digital_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_amp_putvol(tas_priv, ucontrol, mc); +} + +static const struct snd_kcontrol_new tas2781_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Digital Volume", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), +}; + +static int tasdevice_dsp_create_ctrls(struct tasdevice_priv + *tas_priv) +{ + char prog_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char conf_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + struct hda_codec *codec = tas_priv->codec; + struct snd_kcontrol_new prog_ctl = { + .name = prog_name, + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_programs, + .get = tasdevice_program_get, + .put = tasdevice_program_put, + }; + struct snd_kcontrol_new conf_ctl = { + .name = conf_name, + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_configurations, + .get = tasdevice_config_get, + .put = tasdevice_config_put, + }; + int ret; + + scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "spk-prog-id"); + scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "spk-conf-id"); + + ret = snd_ctl_add(codec->card, snd_ctl_new1(&prog_ctl, tas_priv)); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + prog_ctl.name, ret); + goto out; + } + + dev_dbg(tas_priv->dev, "Added Control %s\n", prog_ctl.name); + + ret = snd_ctl_add(codec->card, snd_ctl_new1(&conf_ctl, tas_priv)); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + conf_ctl.name, ret); + goto out; + } + + dev_dbg(tas_priv->dev, "Added Control %s\n", conf_ctl.name); +out: + return ret; +} + +static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) +{ + static const unsigned char page_array[CALIB_MAX] = { + 0x17, 0x18, 0x18, 0x0d, 0x18 + }; + static const unsigned char rgno_array[CALIB_MAX] = { + 0x74, 0x0c, 0x14, 0x3c, 0x7c + }; + unsigned char *data; + int i, j, rc; + + for (i = 0; i < tas_priv->ndev; i++) { + data = tas_priv->cali_data.data + + i * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + for (j = 0; j < CALIB_MAX; j++) { + rc = tasdevice_dev_bulk_write(tas_priv, i, + TASDEVICE_REG(0, page_array[j], rgno_array[j]), + &(data[4 * j]), 4); + if (rc < 0) + dev_err(tas_priv->dev, + "chn %d calib %d bulk_wr err = %d\n", + i, j, rc); + } + } +} + +static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) +{ + efi_guid_t efi_guid = EFI_GUID(0x02f9af02, 0x7734, 0x4233, 0xb4, 0x3d, + 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct tm *tm = &tas_priv->tm; + unsigned int attr, crc; + unsigned int *tmp_val; + efi_status_t status; + int ret = 0; + + //Lenovo devices + if (tas_priv->catlog_id == LENOVO) + efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, 0x09, + 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + + tas_priv->cali_data.total_sz = 0; + /* Get real size of UEFI variable */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, tas_priv->cali_data.data); + if (status == EFI_BUFFER_TOO_SMALL) { + ret = -ENODEV; + /* Allocate data buffer of data_size bytes */ + tas_priv->cali_data.data = devm_kzalloc(tas_priv->dev, + tas_priv->cali_data.total_sz, GFP_KERNEL); + if (!tas_priv->cali_data.data) + return -ENOMEM; + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, + tas_priv->cali_data.data); + if (status != EFI_SUCCESS) { + ret = -EINVAL; + goto out; + } + } + + tmp_val = (unsigned int *)tas_priv->cali_data.data; + + crc = crc32(~0, tas_priv->cali_data.data, 84) ^ ~0; + dev_dbg(tas_priv->dev, "cali crc 0x%08x PK tmp_val 0x%08x\n", + crc, tmp_val[21]); + + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + dev_dbg(tas_priv->dev, "%4ld-%2d-%2d, %2d:%2d:%2d\n", + tm->tm_year, tm->tm_mon, tm->tm_mday, + tm->tm_hour, tm->tm_min, tm->tm_sec); + tas2781_apply_calib(tas_priv); + } +out: + return ret; +} + +static void tasdevice_fw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; + struct hda_codec *codec = tas_priv->codec; + int i, ret = 0; + + pm_runtime_get_sync(tas_priv->dev); + mutex_lock(&tas_priv->codec_lock); + + ret = tasdevice_rca_parser(tas_priv, fmw); + if (ret) + goto out; + tasdevice_create_control(tas_priv); + + for (i = 0; i < ARRAY_SIZE(tas2781_snd_controls); i++) { + ret = snd_ctl_add(codec->card, + snd_ctl_new1(&tas2781_snd_controls[i], tas_priv)); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_snd_controls[i].name, ret); + goto out; + } + } + + tasdevice_dsp_remove(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%04X.bin", + codec->core.subsystem_id & 0xffff); + ret = tasdevice_dsp_parser(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dspfw load %s error\n", + tas_priv->coef_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + tasdevice_dsp_create_ctrls(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; + tasdevice_prmg_load(tas_priv, 0); + tas_priv->cur_prog = 0; + + /* If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + tas2781_save_calibration(tas_priv); + +out: + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + /*If DSP FW fail, kcontrol won't be created */ + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + } + mutex_unlock(&tas_priv->codec_lock); + if (fmw) + release_firmware(fmw); + pm_runtime_mark_last_busy(tas_priv->dev); + pm_runtime_put_autosuspend(tas_priv->dev); +} + +static int tas2781_hda_bind(struct device *dev, struct device *master, + void *master_data) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct hda_component *comps = master_data; + struct hda_codec *codec; + unsigned int subid; + int ret; + + if (!comps || tas_priv->index < 0 || + tas_priv->index >= HDA_MAX_COMPONENTS) + return -EINVAL; + + comps = &comps[tas_priv->index]; + if (comps->dev) + return -EBUSY; + + codec = comps->codec; + subid = codec->core.subsystem_id & 0xFFFF; + + //Lenovo devices + if ((subid == 0x387d) || (subid == 0x387e) || (subid == 0x3881) + || (subid == 0x3884) || (subid == 0x3886) || (subid == 0x38a7) + || (subid == 0x38a8) || (subid == 0x38ba) || (subid == 0x38bb) + || (subid == 0x38be) || (subid == 0x38bf) || (subid == 0x38c3) + || (subid == 0x38cb) || (subid == 0x38cd)) + tas_priv->catlog_id = LENOVO; + else + tas_priv->catlog_id = OTHERS; + + pm_runtime_get_sync(dev); + + comps->dev = dev; + + strscpy(comps->name, dev_name(dev), sizeof(comps->name)); + + ret = tascodec_init(tas_priv, codec, tasdevice_fw_ready); + if (ret) + return ret; + + comps->playback_hook = tas2781_hda_playback_hook; + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + + return 0; +} + +static void tas2781_hda_unbind(struct device *dev, + struct device *master, void *master_data) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct hda_component *comps = master_data; + + if (comps[tas_priv->index].dev == dev) + memset(&comps[tas_priv->index], 0, sizeof(*comps)); + + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; +} + +static const struct component_ops tas2781_hda_comp_ops = { + .bind = tas2781_hda_bind, + .unbind = tas2781_hda_unbind, +}; + +static void tas2781_hda_remove(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + pm_runtime_get_sync(tas_priv->dev); + pm_runtime_disable(tas_priv->dev); + + + component_del(tas_priv->dev, &tas2781_hda_comp_ops); + + pm_runtime_put_noidle(tas_priv->dev); + + tasdevice_remove(tas_priv); +} + +static int tas2781_hda_i2c_probe(struct i2c_client *clt) +{ + struct tasdevice_priv *tas_priv; + const char *device_name; + int ret; + + if (strstr(dev_name(&clt->dev), "TIAS2781")) + device_name = "TIAS2781"; + else + return -ENODEV; + + tas_priv = tasdevice_kzalloc(clt); + if (!tas_priv) + return -ENOMEM; + + tas_priv->irq_info.irq = clt->irq; + ret = tas2781_hda_read_acpi(tas_priv, device_name); + if (ret) + return dev_err_probe(tas_priv->dev, ret, + "Platform not supported\n"); + + ret = tasdevice_init(tas_priv); + if (ret) + goto err; + + pm_runtime_set_autosuspend_delay(tas_priv->dev, 3000); + pm_runtime_use_autosuspend(tas_priv->dev); + pm_runtime_mark_last_busy(tas_priv->dev); + pm_runtime_set_active(tas_priv->dev); + pm_runtime_get_noresume(tas_priv->dev); + pm_runtime_enable(tas_priv->dev); + + pm_runtime_put_autosuspend(tas_priv->dev); + + ret = component_add(tas_priv->dev, &tas2781_hda_comp_ops); + if (ret) { + dev_err(tas_priv->dev, "Register component failed: %d\n", ret); + pm_runtime_disable(tas_priv->dev); + } + +err: + if (ret) + tas2781_hda_remove(&clt->dev); + return ret; +} + +static void tas2781_hda_i2c_remove(struct i2c_client *clt) +{ + tas2781_hda_remove(&clt->dev); +} + +static int tas2781_runtime_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int i; + + dev_info(tas_priv->dev, "Runtime Suspend\n"); + + mutex_lock(&tas_priv->codec_lock); + + if (tas_priv->playback_started) { + tasdevice_tuning_switch(tas_priv, 1); + tas_priv->playback_started = false; + } + + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + for (i = 0; i < tas_priv->ndev; i++) { + tas_priv->tasdevice[i].cur_book = -1; + tas_priv->tasdevice[i].cur_prog = -1; + tas_priv->tasdevice[i].cur_conf = -1; + } + + regcache_cache_only(tas_priv->regmap, true); + regcache_mark_dirty(tas_priv->regmap); + + mutex_unlock(&tas_priv->codec_lock); + + return 0; +} + +static int tas2781_runtime_resume(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long calib_data_sz = + tas_priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + int ret; + + dev_info(tas_priv->dev, "Runtime Resume\n"); + + mutex_lock(&tas_priv->codec_lock); + + regcache_cache_only(tas_priv->regmap, false); + ret = regcache_sync(tas_priv->regmap); + if (ret) { + dev_err(tas_priv->dev, + "Failed to restore register cache: %d\n", ret); + goto out; + } + + tasdevice_prmg_load(tas_priv, 0); + tas_priv->cur_prog = 0; + + /* If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + if (tas_priv->cali_data.total_sz > calib_data_sz) + tas2781_apply_calib(tas_priv); + +out: + mutex_unlock(&tas_priv->codec_lock); + + return ret; +} + +static int tas2781_system_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + dev_dbg(tas_priv->dev, "System Suspend\n"); + + ret = pm_runtime_force_suspend(dev); + if (ret) + return ret; + + /* Shutdown chip before system suspend */ + regcache_cache_only(tas_priv->regmap, false); + tasdevice_tuning_switch(tas_priv, 1); + regcache_cache_only(tas_priv->regmap, true); + regcache_mark_dirty(tas_priv->regmap); + /* + * Reset GPIO may be shared, so cannot reset here. + * However beyond this point, amps may be powered down. + */ + return 0; +} + +static int tas2781_system_resume(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long calib_data_sz = + tas_priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + int i, ret; + + dev_dbg(tas_priv->dev, "System Resume\n"); + + ret = pm_runtime_force_resume(dev); + if (ret) + return ret; + + mutex_lock(&tas_priv->codec_lock); + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + for (i = 0; i < tas_priv->ndev; i++) { + tas_priv->tasdevice[i].cur_book = -1; + tas_priv->tasdevice[i].cur_prog = -1; + tas_priv->tasdevice[i].cur_conf = -1; + } + tas2781_reset(tas_priv); + tasdevice_prmg_load(tas_priv, 0); + tas_priv->cur_prog = 0; + + /* If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + if (tas_priv->cali_data.total_sz > calib_data_sz) + tas2781_apply_calib(tas_priv); + mutex_unlock(&tas_priv->codec_lock); + + return 0; +} + +static const struct dev_pm_ops tas2781_hda_pm_ops = { + RUNTIME_PM_OPS(tas2781_runtime_suspend, tas2781_runtime_resume, + NULL) + SYSTEM_SLEEP_PM_OPS(tas2781_system_suspend, tas2781_system_resume) +}; + +static const struct i2c_device_id tas2781_hda_i2c_id[] = { + { "tas2781-hda", 0 }, + {} +}; + +static const struct acpi_device_id tas2781_acpi_hda_match[] = { + {"TIAS2781", 0 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); + +static struct i2c_driver tas2781_hda_i2c_driver = { + .driver = { + .name = "tas2781-hda", + .acpi_match_table = tas2781_acpi_hda_match, + .pm = &tas2781_hda_pm_ops, + }, + .id_table = tas2781_hda_i2c_id, + .probe_new = tas2781_hda_i2c_probe, + .remove = tas2781_hda_i2c_remove, +}; +module_i2c_driver(tas2781_hda_i2c_driver); + +MODULE_DESCRIPTION("TAS2781 HDA Driver"); +MODULE_AUTHOR("Shenghao Ding, TI, <shenghao-ding(a)ti.com>"); +MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS(SND_SOC_TAS2781_FMWLIB); -- 2.34.1

4 4

[PATCH] ALSA: hda/realtek: Enable 4 amplifiers instead of 2 on a HP platform
by Chris Chiu 06 Jun '23

06 Jun '23

In the commit 7bb62340951a ("ALSA: hda/realtek: fix speaker, mute/micmute LEDs not work on a HP platform"), speakers and LEDs are fixed but only 2 CS35L41 amplifiers on SPI bus connected to Realtek codec are enabled. Need the ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED to get all amplifiers working. Signed-off-by: Chris Chiu <chris.chiu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f10790ace5c1..0e76938f56df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9500,7 +9500,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b8a, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8b, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8d, "HP", ALC236_FIXUP_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8b8f, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8b8f, "HP", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b97, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), -- 2.25.1

2 1

[PATCH 00/28] ASoC: Intel: machine driver updates for 6.5
by Pierre-Louis Bossart 06 Jun '23

06 Jun '23

An unusually large set of patches to deal with new SoundWire-based platforms. The bulk of the patches addresses support for MTL using RT712, RT713, MAX98363, CS42L42 jack codec and amplifiers. The sof_sdw machine driver had to be updated to remove limitations on number of codecs per links, dai types, dai naming, etc. We also moved parts of the Realtek and Maxim support in common code to avoid duplication. Community users also reported two Dell SKUs which were not supported in the mainline due to hardware permutations. Balamurugan C (3): ASoC: Intel: ADL: Enable HDMI-In capture feature support for non-I2S codec boards. ASoC: Intel: ADL: Moving amp only boards into end of the table. ASoC: Intel: Sof_ssp_amp: Correcting author name. Bard Liao (14): ASoC: Intel: sof_sdw: add missing exit callback ASoC: Intel: sof_sdw: add dai info ASoC: Intel: sof_sdw: use predefine dailink id ASoC: Intel: sof_sdw: add codec_info pointer ASoC: Intel: sdw_sof: append dai_type and remove codec_type ASoC: Intel: sof_sdw: add multi dailink support for a codec ASoC: Intel: sof_sdw_rt_sdca_jack_common: test SOF_JACK_JDSRC in _exit ASoC: Intel: sof_sdw: rename SOF_RT711_JDSRC to SOF_JACK_JDSRC ASoC: Intel: sof_sdw: make rt711_sdca be generic ASoC: Intel: sof_sdw: add rt712 support ASoC: Intel: soc-acpi-intel-tgl-match: add rt712 ID ASoC: Intel: soc-acpi-intel-mtl-match: add rt712 ID ASoC: Intel: sof_sdw: add rt713 support ASoC: Intel: sof_sdw: increase sdw pin index for each sdw link Pierre-Louis Bossart (4): ASoC: Intel: soc-acpi: add table for RPL Dell SKU 0BDA ASoC: Intel: sof_sdw: add quick for Dell SKU 0BDA ASoC: Intel: soc-acpi: add tables for Dell SKU 0B34 ASoC: Intel: sof-sdw: add Dell SKU 0B34 Terry Cheong (1): ASoC: Intel: Add rpl_rt1019_rt5682 driver Uday M Bhat (6): ASoC: Intel: sof_sdw: Add helper function for cs42l42 codec ASoC: Intel: sof_sdw: Rename sof_sdw_max98373.c file to sof_sdw_maxim.c ASoC: Intel: sof_sdw: Modify maxim helper functions and structure names ASoC: Intel: sof_sdw: Add support for MAX98363 codec ASoC: Intel: sof_sdw: Add support for Rex soundwire ASoC: Intel: soc-acpi: add Rex CS42l42 and MAX98363 SoundWire entries sound/soc/intel/boards/Kconfig | 4 + sound/soc/intel/boards/Makefile | 10 +- sound/soc/intel/boards/sof_rt5682.c | 11 + sound/soc/intel/boards/sof_sdw.c | 542 +++++++++++++----- sound/soc/intel/boards/sof_sdw_common.h | 86 ++- sound/soc/intel/boards/sof_sdw_cs42l42.c | 131 +++++ .../{sof_sdw_max98373.c => sof_sdw_maxim.c} | 58 +- sound/soc/intel/boards/sof_sdw_rt711.c | 4 +- sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 102 ++++ ...1_sdca.c => sof_sdw_rt_sdca_jack_common.c} | 73 ++- sound/soc/intel/boards/sof_ssp_amp.c | 11 +- .../intel/common/soc-acpi-intel-adl-match.c | 46 +- .../intel/common/soc-acpi-intel-mtl-match.c | 98 ++++ .../intel/common/soc-acpi-intel-rpl-match.c | 42 ++ .../intel/common/soc-acpi-intel-tgl-match.c | 53 ++ 15 files changed, 1045 insertions(+), 226 deletions(-) create mode 100644 sound/soc/intel/boards/sof_sdw_cs42l42.c rename sound/soc/intel/boards/{sof_sdw_max98373.c => sof_sdw_maxim.c} (66%) create mode 100644 sound/soc/intel/boards/sof_sdw_rt712_sdca.c rename sound/soc/intel/boards/{sof_sdw_rt711_sdca.c => sof_sdw_rt_sdca_jack_common.c} (61%) -- 2.37.2

2 29

[PATCH 0/4] ASoC: support dai-links with symmetric clock roles
by Alvin Šipraga 06 Jun '23

06 Jun '23

From: Alvin Šipraga <alsi(a)bang-olufsen.dk> Currently the ASoC core always assumes that one end of a dai-link is a clock provider and the other a consumer. This series adds support for configuring dai-links where both ends are actually clock consumers. Alvin Šipraga (4): ASoC: dt-bindings: document new symmetric-clock-role flag ASoC: core: add support for dai-links with symmetric clock roles ASoC: audio-graph-card2: parse symmetric-clock-roles property ASoC: simple-card: parse symmetric-clock-roles property .../devicetree/bindings/sound/simple-card.yaml | 11 +++++++++++ include/sound/soc.h | 3 +++ sound/soc/generic/audio-graph-card2.c | 7 ++++++- sound/soc/generic/simple-card.c | 4 ++++ sound/soc/soc-core.c | 4 +++- 5 files changed, 27 insertions(+), 2 deletions(-) -- 2.40.0

5 15

[GIT PULL] ASoC fixes for v6.4-rc6
by Mark Brown 06 Jun '23

06 Jun '23

The following changes since commit 011a8719d6105dcb48077ea7a6a88ac019d4aa50: ASoC: dwc: move DMA init to snd_soc_dai_driver probe() (2023-05-23 21:39:30 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.4-rc6 for you to fetch changes up to 32cf0046a652116d6a216d575f3049a9ff9dd80d: ASoC: fsl_sai: Enable BCI bit if SAI works on synchronous mode with BYP asserted (2023-06-02 12:54:07 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.4 A lot of routine driver specific fixes here, nothing in the core though there are a couple of fixes for the generic cards. There's also a few new quirks for x86 platforms. ---------------------------------------------------------------- Chancel Liu (1): ASoC: fsl_sai: Enable BCI bit if SAI works on synchronous mode with BYP asserted Edson Juliano Drosdeck (1): ASoC: nau8824: Add quirk to active-high jack-detect Herve Codina (1): ASoC: simple-card: Add missing of_node_put() in case of error Mark Brown (1): ASoC: mediatek: fix use-after-free in driver remove Richard Fitzgerald (1): ASoC: cs35l56: Remove NULL check from cs35l56_sdw_dai_set_stream() Robert Hancock (1): ASoC: simple-card-utils: fix PCM constraint error check Ryan Lee (2): ASoC: max98363: Removed 32bit support ASoC: max98363: limit the number of channel to 1 Sicong Jiang (1): ASoC: amd: yc: Add Thinkpad Neo14 to quirks list for acp6x Srinivas Kandagatla (3): ASoC: codecs: wsa883x: do not set can_multi_write flag ASoC: codecs: wsa881x: do not set can_multi_write flag ASoC: codecs: wcd938x-sdw: do not set can_multi_write flag Trevor Wu (2): ASoC: mediatek: mt8188: fix use-after-free in driver remove path ASoC: mediatek: mt8195: fix use-after-free in driver remove path Vijendar Mukunda (1): ASoC: amd: ps: fix for acp_lock access in pdm driver sound/soc/amd/ps/pci-ps.c | 3 +- sound/soc/amd/ps/ps-pdm-dma.c | 10 +++--- sound/soc/amd/yc/acp6x-mach.c | 7 ++++ sound/soc/codecs/cs35l56.c | 3 -- sound/soc/codecs/max98363.c | 4 +-- sound/soc/codecs/nau8824.c | 24 ++++++++++++++ sound/soc/codecs/wcd938x-sdw.c | 1 - sound/soc/codecs/wsa881x.c | 1 - sound/soc/codecs/wsa883x.c | 1 - sound/soc/fsl/fsl_sai.c | 11 +++++-- sound/soc/fsl/fsl_sai.h | 1 + sound/soc/generic/simple-card-utils.c | 2 +- sound/soc/generic/simple-card.c | 1 + sound/soc/mediatek/mt8188/mt8188-afe-clk.c | 7 ---- sound/soc/mediatek/mt8188/mt8188-afe-clk.h | 1 - sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 4 --- sound/soc/mediatek/mt8188/mt8188-audsys-clk.c | 47 ++++++++++++++------------- sound/soc/mediatek/mt8188/mt8188-audsys-clk.h | 1 - sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 5 --- sound/soc/mediatek/mt8195/mt8195-afe-clk.h | 1 - sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 4 --- sound/soc/mediatek/mt8195/mt8195-audsys-clk.c | 47 ++++++++++++++------------- sound/soc/mediatek/mt8195/mt8195-audsys-clk.h | 1 - 23 files changed, 99 insertions(+), 88 deletions(-)

2 1

[bug report] ASoC: starfive: Add JH7110 TDM driver
by Dan Carpenter 06 Jun '23

06 Jun '23

Hello Walker Chen, The patch fd4762b6b5cf: "ASoC: starfive: Add JH7110 TDM driver" from May 26, 2023, leads to the following Smatch static checker warning: sound/soc/starfive/jh7110_tdm.c:584 jh7110_tdm_clk_reset_get() warn: passing zero to 'PTR_ERR' sound/soc/starfive/jh7110_tdm.c 564 static int jh7110_tdm_clk_reset_get(struct platform_device *pdev, 565 struct jh7110_tdm_dev *tdm) 566 { 567 int ret; 568 569 tdm->clks[0].id = "mclk_inner"; 570 tdm->clks[1].id = "tdm_ahb"; 571 tdm->clks[2].id = "tdm_apb"; 572 tdm->clks[3].id = "tdm_internal"; 573 tdm->clks[4].id = "tdm_ext"; 574 tdm->clks[5].id = "tdm"; 575 576 ret = devm_clk_bulk_get(&pdev->dev, ARRAY_SIZE(tdm->clks), tdm->clks); 577 if (ret) { 578 dev_err(&pdev->dev, "Failed to get tdm clocks\n"); 579 return ret; 580 } 581 582 tdm->resets = devm_reset_control_array_get_exclusive(&pdev->dev); The devm_reset_control_array_get() function returns NULL if it's an optional request. But this is not optional. 583 if (IS_ERR_OR_NULL(tdm->resets)) { So that means this should be an if (IS_ERR()) check. --> 584 ret = PTR_ERR(tdm->resets); 585 dev_err(&pdev->dev, "Failed to get tdm resets"); Or if optional was intended then NULL should not be treated as an error case, but as a special kind of success case (no error message). See my blog for a long form of this information: https://staticthinking.wordpress.com/2022/08/01/mixing-error-pointers-and-n… 586 return ret; 587 } 588 589 return 0; 590 } regards, dan carpenter

2 1

[PATCH v3 0/3] Fixes and cleanup for CS35L41 HDA
by Stefan Binding 06 Jun '23

06 Jun '23

Several minor issues were found during additional testing and static analysis. These patches fix these minor issues. CHANGES SINCE V2: Patch 3: - Add missing break CHANGES SINCE V1: Patch 2: - Removed unnecessary cast Stefan Binding (3): ALSA: hda: cs35l41: Clean up Firmware Load Controls ALSA: hda: cs35l41: Fix endian conversions ALSA: hda/realtek: Delete cs35l41 component master during free sound/pci/hda/cs35l41_hda.c | 32 ++++++++++++++------------------ sound/pci/hda/patch_realtek.c | 3 +++ 2 files changed, 17 insertions(+), 18 deletions(-) -- 2.34.1

2 4

[PATCH] ALSA: control: Keep the previous numid at snd_ctl_rename_id()
by Takashi Iwai 06 Jun '23

06 Jun '23

We don't need to change the numid at each time snd_ctl_rename_id() is called, as the control element size itself doesn't change. Let's keep the previous numid value. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 82aa1af1d1d8..f4a90a687ae5 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -730,12 +730,16 @@ EXPORT_SYMBOL_GPL(snd_ctl_activate_id); * Finds the control with the old id from the card, and replaces the * id with the new one. * + * The function tries to keep the already assigned numid while replacing + * the rest. + * * Return: Zero if successful, or a negative error code on failure. */ int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id) { struct snd_kcontrol *kctl; + int saved_numid; down_write(&card->controls_rwsem); kctl = snd_ctl_find_id(card, src_id); @@ -743,10 +747,10 @@ int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id, up_write(&card->controls_rwsem); return -ENOENT; } + saved_numid = kctl->id.numid; remove_hash_entries(card, kctl); kctl->id = *dst_id; - kctl->id.numid = card->last_numid + 1; - card->last_numid += kctl->count; + kctl->id.numid = saved_numid; add_hash_entries(card, kctl); up_write(&card->controls_rwsem); return 0; -- 2.35.3

2 2

[PATCH 0/4] Add Loongson HD Audio support
by Yanteng Si 06 Jun '23

06 Jun '23

* The Loongson HDA controller is compatible with High Definition Audio Specification Revision 1.0a. See <https://loongson.github.io/LoongArch-Documentation/Loongson- 7A1000-usermanual-EN.html#hda-controller> Yanteng Si (4): ALSA: hda: Add Loongson LS7A HD-Audio support ALSA: hda: Using polling mode for loongson controller by default ALSA: hda: Workaround for SDnCTL register on loongson controller ALSA: hda/intel: Workaround for WALLCLK register for loongson controller include/linux/pci_ids.h | 3 +++ sound/hda/hdac_controller.c | 8 +++++++- sound/hda/hdac_device.c | 1 + sound/hda/hdac_stream.c | 8 +++++++- sound/pci/hda/hda_controller.h | 1 + sound/pci/hda/hda_intel.c | 10 ++++++++++ sound/pci/hda/patch_hdmi.c | 1 + 7 files changed, 30 insertions(+), 2 deletions(-) -- 2.31.4

2 8

[PATCH] firmware: cs_dsp: Log correct region name in bin error messages
by Richard Fitzgerald 06 Jun '23

06 Jun '23

In cs_dsp_load_coeff() region_name should be set in the XM/YM/ZM cases otherwise any errors will log the region as "Unknown". While doing this also change one error message that logged the region type ID to log the region_name instead. This makes it consistent with other messages in the same function. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/firmware/cirrus/cs_dsp.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index d7e46a57ecf9..6a9aa97373d3 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -2124,6 +2124,7 @@ static int cs_dsp_load_coeff(struct cs_dsp *dsp, const struct firmware *firmware file, blocks, le32_to_cpu(blk->len), type, le32_to_cpu(blk->id)); + region_name = cs_dsp_mem_region_name(type); mem = cs_dsp_find_region(dsp, type); if (!mem) { cs_dsp_err(dsp, "No base for region %x\n", type); @@ -2147,8 +2148,8 @@ static int cs_dsp_load_coeff(struct cs_dsp *dsp, const struct firmware *firmware reg = dsp->ops->region_to_reg(mem, reg); reg += offset; } else { - cs_dsp_err(dsp, "No %x for algorithm %x\n", - type, le32_to_cpu(blk->id)); + cs_dsp_err(dsp, "No %s for algorithm %x\n", + region_name, le32_to_cpu(blk->id)); } break; -- 2.30.2

2 1

[PATCH 0/4] ALSA: Fix kctl->id initialization
by Takashi Iwai 06 Jun '23

06 Jun '23

Inspired by the recent Jaroslav's fix patch, I skimmed over the tree and found that there are a few other similar bugs. Here is a patch set to address them. Most of them are just shuffling the call order, while the one in HD-audio is changed to use snd_ctl_replace_id() helper. Takashi === Takashi Iwai (4): ALSA: ymfpci: Fix kctl->id initialization ALSA: cmipci: Fix kctl->id initialization ALSA: gus: Fix kctl->id initialization ALSA: hda: Fix kctl->id initialization sound/isa/gus/gus_pcm.c | 2 +- sound/pci/cmipci.c | 6 +++--- sound/pci/hda/hda_codec.c | 6 +++++- sound/pci/ymfpci/ymfpci_main.c | 6 +++--- 4 files changed, 12 insertions(+), 8 deletions(-) -- 2.35.3

2 5

[PATCH] ALSA: ice1712,ice1724: fix the kcontrol->id initialization
by Jaroslav Kysela 06 Jun '23

06 Jun '23

The new xarray lookup code requires to know complete kcontrol->id before snd_ctl_add() call. Reorder the code to make the initialization properly. Cc: stable(a)kernel.org # v5.19+ Reported-by: Martin Zidek <zidek(a)master.cz> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/ice1712/aureon.c | 7 ++++--- sound/pci/ice1712/ice1712.c | 14 +++++++++----- sound/pci/ice1712/ice1724.c | 16 ++++++++++------ 3 files changed, 23 insertions(+), 14 deletions(-) diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 24b978234000..027849329c1b 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -1899,11 +1899,12 @@ static int aureon_add_controls(struct snd_ice1712 *ice) else { for (i = 0; i < ARRAY_SIZE(cs8415_controls); i++) { struct snd_kcontrol *kctl; - err = snd_ctl_add(ice->card, (kctl = snd_ctl_new1(&cs8415_controls[i], ice))); - if (err < 0) - return err; + kctl = snd_ctl_new1(&cs8415_controls[i], ice); if (i > 1) kctl->id.device = ice->pcm->device; + err = snd_ctl_add(ice->card, kctl); + if (err < 0) + return err; } } } diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index a5241a287851..3b0c3e70987b 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2371,22 +2371,26 @@ int snd_ice1712_spdif_build_controls(struct snd_ice1712 *ice) if (snd_BUG_ON(!ice->pcm_pro)) return -EIO; - err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_ice1712_spdif_default, ice)); + kctl = snd_ctl_new1(&snd_ice1712_spdif_default, ice); + kctl->id.device = ice->pcm_pro->device; + err = snd_ctl_add(ice->card, kctl); if (err < 0) return err; + kctl = snd_ctl_new1(&snd_ice1712_spdif_maskc, ice); kctl->id.device = ice->pcm_pro->device; - err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_ice1712_spdif_maskc, ice)); + err = snd_ctl_add(ice->card, kctl); if (err < 0) return err; + kctl = snd_ctl_new1(&snd_ice1712_spdif_maskp, ice); kctl->id.device = ice->pcm_pro->device; - err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_ice1712_spdif_maskp, ice)); + err = snd_ctl_add(ice->card, kctl); if (err < 0) return err; + kctl = snd_ctl_new1(&snd_ice1712_spdif_stream, ice); kctl->id.device = ice->pcm_pro->device; - err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_ice1712_spdif_stream, ice)); + err = snd_ctl_add(ice->card, kctl); if (err < 0) return err; - kctl->id.device = ice->pcm_pro->device; ice->spdif.stream_ctl = kctl; return 0; } diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 6fab2ad85bbe..1dc776acd637 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2392,23 +2392,27 @@ static int snd_vt1724_spdif_build_controls(struct snd_ice1712 *ice) if (err < 0) return err; - err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_vt1724_spdif_default, ice)); + kctl = snd_ctl_new1(&snd_vt1724_spdif_default, ice); + kctl->id.device = ice->pcm->device; + err = snd_ctl_add(ice->card, kctl); if (err < 0) return err; + kctl = snd_ctl_new1(&snd_vt1724_spdif_maskc, ice); kctl->id.device = ice->pcm->device; - err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_vt1724_spdif_maskc, ice)); + err = snd_ctl_add(ice->card, kctl); if (err < 0) return err; + kctl = snd_ctl_new1(&snd_vt1724_spdif_maskp, ice); kctl->id.device = ice->pcm->device; - err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_vt1724_spdif_maskp, ice)); + err = snd_ctl_add(ice->card, kctl); if (err < 0) return err; - kctl->id.device = ice->pcm->device; #if 0 /* use default only */ - err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_vt1724_spdif_stream, ice)); + kctl = snd_ctl_new1(&snd_vt1724_spdif_stream, ice); + kctl->id.device = ice->pcm->device; + err = snd_ctl_add(ice->card, kctl); if (err < 0) return err; - kctl->id.device = ice->pcm->device; ice->spdif.stream_ctl = kctl; #endif return 0; -- 2.39.2

2 1

Add Asus ROG STRIX X670E-F Gaming Wifi to USB-Audio.conf
by GitHub pull_request - opened 06 Jun '23

06 Jun '23

alsa-project/alsa-ucm-conf pull request #320 was opened from ksenchy: Since the id was already present in the list, I have only added a comment denoting this board also uses that id My alsa-info as a proof can be found here: http://alsa-project.org/db/?f=1ad83f467dc312e2ac7743859c647695dbccb6ce Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/320 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/320.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/9] ASoC: SOF: Intel: LunarLake preparation patches
by Pierre-Louis Bossart 05 Jun '23

05 Jun '23

This patchset adds the changes required for the hda-dai extension to deal with SSSP/DMIC/SoundWire starting with LunarLake, as well as the new TLV IPC to provide the DMA stream_tag to the DSP firmware. LunarLake support for SSP/DMIC is ready but is gated by the patch "ASoC: SOF: Intel: shim: add enum for ACE 2.0 IP used in LunarLake" currently only present in the SoundWire tree. Pierre-Louis Bossart (9): ASoC: SOF: Intel: hda-dai: add error checks to prevent static analysis warnings ASoC: SOF: Intel: hda-dai: add codec_dai_set_stream callback ASoC: SOF: Intel: hda-dai: add calc_stream_format callback ASoC: SOF: Intel: hda-dai: add get_hlink callback ASoC: SOF: ipc4-topology: extend ALH-specific data structure ASoC: SOF: ipc4-topology: introduce DMA config TLV ASoC: SOF: ipc4-topology: add DMA config TLV to IPC data ASoC: SOF: Intel: mtl: prepare for code reuse ASoC: SOF: Intel: hda: add helper to extract SoundWire link count sound/soc/sof/intel/hda-dai-ops.c | 58 ++++++++++++++++++++++- sound/soc/sof/intel/hda-dai.c | 64 +++++++++++--------------- sound/soc/sof/intel/hda.c | 25 ++++++++++ sound/soc/sof/intel/hda.h | 19 ++++++++ sound/soc/sof/intel/mtl.c | 12 ++--- sound/soc/sof/intel/mtl.h | 7 +++ sound/soc/sof/ipc4-topology.c | 44 ++++++++++++++---- sound/soc/sof/ipc4-topology.h | 76 +++++++++++++++++++++++++------ 8 files changed, 239 insertions(+), 66 deletions(-) -- 2.37.2

2 10

[PATCH v3 0/7] ASoC: mt8188: add new board support
by Trevor Wu 05 Jun '23

05 Jun '23

In the series, we extend the capability of mt8188-mt6359 driver. The following changes are included. 1. Divide ADDA BE dai into two dais for SOF. 2. Register hdmi/dp jack pins. 3. dai_fmt can be configured from device tree. 4. Add some I2S codecs support. In addition, new compatible string "mediatek,mt8188-nau8825" is included for a new board support. Changes since v2: - unify coding style based on reviewer's suggestion - fix typo Changes since v1: - add "Headphone Jack" for Nau8825 - separate binding patch 2 from patch 6, and rename the subject - clean up debugfs for freed widget Trevor Wu (7): ASoC: mediatek: mt8188: separate ADDA playback dai from capture dai ASoC: dt-bindings: mediatek,mt8188-mt6359: remove ADDA_BE from link-name ASoC: mediatek: mt8188-mt6359: register hdmi/dp jack pins ASoC: mediatek: common: soundcard driver add dai_fmt support ASoC: soc-dapm.c: clean up debugfs for freed widget ASoC: mediatek: mt8188-mt6359: support new board with nau88255 ASoC: dt-bindings: mediatek,mt8188-mt6359: add NAU8825 support .../sound/mediatek,mt8188-mt6359.yaml | 27 +- sound/soc/mediatek/Kconfig | 4 + .../mediatek/common/mtk-soundcard-driver.c | 53 ++- sound/soc/mediatek/mt8188/mt8188-afe-common.h | 3 +- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 73 ++-- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 386 +++++++++++++++++- sound/soc/soc-dapm.c | 16 + 7 files changed, 507 insertions(+), 55 deletions(-) -- 2.18.0

3 10

[PATCH] ALSA: hda/realtek: Add quirk for Clevo NS50AU
by Tim Crawford 05 Jun '23

05 Jun '23

Fixes headset detection on Clevo NS50AU. Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 172ffc2c332b..345847ef8350 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9608,6 +9608,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x5101, "Clevo S510WU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x5157, "Clevo W517GU1", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x51a1, "Clevo NS50MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x51b1, "Clevo NS50AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x5630, "Clevo NP50RNJS", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70a1, "Clevo NB70T[HJK]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70b3, "Clevo NK70SB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.40.1

2 1

[PATCH v2] ALSA: hda/realtek: Add quirks for Asus ROG 2024 laptops using CS35L41
by Stefan Binding 05 Jun '23

05 Jun '23

Add support for Asus ROG 2024 models using CS35L41 SPI with Internal Boost. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7b5f194513c7b..fe6c2b54329c7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9547,6 +9547,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1043, 0x1b93, "ASUS G614JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), @@ -9565,6 +9566,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1f12, "ASUS UM5302", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1f92, "ASUS ROG Flow X16", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), + SND_PCI_QUIRK(0x1043, 0x3a20, "ASUS G614JZR", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x3a30, "ASUS G814JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x3a50, "ASUS G834JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), -- 2.34.1

2 1

[PATCH v2 0/3] Fixes and cleanup for CS35L41 HDA
by Stefan Binding 05 Jun '23

05 Jun '23

Several minor issues were found during additional testing and static analysis. These patches fix these minor issues. CHANGES SINCE V1: Patch 2: - Removed unnecessary cast Stefan Binding (3): ALSA: hda: cs35l41: Clean up Firmware Load Controls ALSA: hda: cs35l41: Fix endian conversions ALSA: hda/realtek: Delete cs35l41 component master during free sound/pci/hda/cs35l41_hda.c | 32 ++++++++++++++------------------ sound/pci/hda/patch_realtek.c | 2 ++ 2 files changed, 16 insertions(+), 18 deletions(-) -- 2.34.1

2 4

A data race between snd_seq_oss_midi_open and snd_seq_oss_midi_filemode
by Gong, Sishuai 05 Jun '23

05 Jun '23

Hi there, We found a data race could happen between snd_seq_oss_midi_open() and snd_seq_oss_midi_filemode() over the variable mdev->opened. When running concurrently, snd_seq_oss_midi_open() makes the return value of snd_seq_oss_midi_filemode() non-deterministic: Thread-1 Thread-2 //snd_seq_oss_midi_filemode() //snd_seq_oss_midi_open() if (mdev->opened & PERM_WRITE) mode |= SNDRV_SEQ_OSS_FILE_WRITE; mdev->opened |= PERM_READ; if (mdev->opened & PERM_READ) mode |= SNDRV_SEQ_OSS_FILE_READ; It is not clear to us that whether this is a serious problem but we would like to report this just in case. Thanks, Sishuai

2 1

[PATCH RESEND v1 0/3] Fixes and cleanup for CS35L41 HDA
by Stefan Binding 05 Jun '23

05 Jun '23

Several minor issues were found during additional testing and static analysis. These patches fix these minor issues. Stefan Binding (3): ALSA: hda: cs35l41: Clean up Firmware Load Controls ALSA: hda: cs35l41: Fix endian conversions ALSA: hda/realtek: Delete cs35l41 component master during free sound/pci/hda/cs35l41_hda.c | 40 ++++++++++++++++------------------- sound/pci/hda/patch_realtek.c | 2 ++ 2 files changed, 20 insertions(+), 22 deletions(-) -- 2.34.1

2 7

[PATCH] [v2] ASoC: amd: vangogh: select CONFIG_SND_AMD_ACP_CONFIG
by Arnd Bergmann 05 Jun '23

05 Jun '23

From: Arnd Bergmann <arnd(a)arndb.de> The vangogh driver just gained a link time dependency that now causes randconfig builds to fail: x86_64-linux-ld: sound/soc/amd/vangogh/pci-acp5x.o: in function `snd_acp5x_probe': pci-acp5x.c:(.text+0xbb): undefined reference to `snd_amd_acp_find_config' Fixes: e89f45edb747e ("ASoC: amd: vangogh: Add check for acp config flags in vangogh platform") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- v2: select CONFIG_SND_AMD_ACP_CONFIG from all ACP5, not just VANGOGH_MACH. --- sound/soc/amd/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 08e42082f5e96..713348508001a 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -71,6 +71,7 @@ config SND_SOC_AMD_RENOIR_MACH config SND_SOC_AMD_ACP5x tristate "AMD Audio Coprocessor-v5.x I2S support" depends on X86 && PCI + select SND_AMD_ACP_CONFIG help This option enables ACP v5.x support on AMD platform -- 2.39.2

3 3

[PATCH 1/2] ALSA: usb-audio: Use __le16 for 16bit USB descriptor fields
by Takashi Iwai 05 Jun '23

05 Jun '23

Use proper notion for 16bit values for fixing the sparse warnings. Fixes: f8ddb0fb3289 ("ALSA: usb-audio: Define USB MIDI 2.0 specs") Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202305260528.wcqjXso8-lkp@intel.com/ Closes: https://lore.kernel.org/oe-kbuild-all/202305270534.odwHL9F0-lkp@intel.com/ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/linux/usb/midi-v2.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/include/linux/usb/midi-v2.h b/include/linux/usb/midi-v2.h index ebbffcae0417..16f09d959a2d 100644 --- a/include/linux/usb/midi-v2.h +++ b/include/linux/usb/midi-v2.h @@ -73,7 +73,7 @@ struct usb_ms20_gr_trm_block_header_descriptor { __u8 bLength; /* 5 */ __u8 bDescriptorType; /* USB_DT_CS_GR_TRM_BLOCK */ __u8 bDescriptorSubtype; /* USB_MS_GR_TRM_BLOCK_HEADER */ - __u16 wTotalLength; /* Total number of bytes */ + __le16 wTotalLength; /* Total number of bytes */ } __packed; /* 5.4.2.1 Group Terminal Block Descriptor */ @@ -87,8 +87,8 @@ struct usb_ms20_gr_trm_block_descriptor { __u8 nNumGroupTrm; /* Number of member Group Terminals spanned */ __u8 iBlockItem; /* String ID of Block item */ __u8 bMIDIProtocol; /* Default MIDI protocol */ - __u16 wMaxInputBandwidth; /* Max input bandwidth capability in 4kB/s */ - __u16 wMaxOutputBandwidth; /* Max output bandwidth capability in 4kB/s */ + __le16 wMaxInputBandwidth; /* Max input bandwidth capability in 4kB/s */ + __le16 wMaxOutputBandwidth; /* Max output bandwidth capability in 4kB/s */ } __packed; #endif /* __LINUX_USB_MIDI_V2_H */ -- 2.35.3

1 1

[PATCH] ALSA: usb-audio: Fix "cannot set freq 48000 to ep 0x3" msgs
by Joakim Tjernlund 05 Jun '23

05 Jun '23

On some USB speaker devices(Jabra/Logitech) we see above error msg when connecting device to computer and kernel is probing the device. Moving the snd_usb_init_sample_rate() to after usb_set_interface() makes the error go away. Signed-off-by: Joakim Tjernlund <joakim.tjernlund(a)infinera.com> Cc: stable(a)vger.kernel.org --- sound/usb/stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index f10f4e6d3fb8..d9ac8663a48b 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1226,8 +1226,8 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, /* try to set the interface... */ usb_set_interface(chip->dev, iface_no, 0); snd_usb_init_pitch(chip, fp); - snd_usb_init_sample_rate(chip, fp, fp->rate_max); usb_set_interface(chip->dev, iface_no, altno); + snd_usb_init_sample_rate(chip, fp, fp->rate_max); } return 0; } -- 2.39.3

2 4

[PATCH v2 0/2] ASoC: mediatek: fix use-after-free in driver remove path
by Trevor Wu 05 Jun '23

05 Jun '23

These patches concern modifications made in mt8186[1]. The clock unregistration mechanism used in mt8188 and mt8195 is similar with mt8186, resulting in the same problem existing within the driver. Therefore, the solution has also been applied to these two platforms. [1] https://lore.kernel.org/all/20230511092437.1.I31cceffc8c45bb1af16eb613e197b… Changes since v1: - remove unnecessary cast Trevor Wu (2): ASoC: mediatek: mt8188: fix use-after-free in driver remove path ASoC: mediatek: mt8195: fix use-after-free in driver remove path sound/soc/mediatek/mt8188/mt8188-afe-clk.c | 7 --- sound/soc/mediatek/mt8188/mt8188-afe-clk.h | 1 - sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 4 -- sound/soc/mediatek/mt8188/mt8188-audsys-clk.c | 47 ++++++++++--------- sound/soc/mediatek/mt8188/mt8188-audsys-clk.h | 1 - sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 5 -- sound/soc/mediatek/mt8195/mt8195-afe-clk.h | 1 - sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 4 -- sound/soc/mediatek/mt8195/mt8195-audsys-clk.c | 47 ++++++++++--------- sound/soc/mediatek/mt8195/mt8195-audsys-clk.h | 1 - 10 files changed, 48 insertions(+), 70 deletions(-) -- 2.18.0

4 7

[PATCH] ASoC: amd: vangogh: select CONFIG_SND_AMD_ACP_CONFIG
by Arnd Bergmann 05 Jun '23

05 Jun '23

From: Arnd Bergmann <arnd(a)arndb.de> The vangogh driver just gained a link time dependency that now causes randconfig builds to fail: x86_64-linux-ld: sound/soc/amd/vangogh/pci-acp5x.o: in function `snd_acp5x_probe': pci-acp5x.c:(.text+0xbb): undefined reference to `snd_amd_acp_find_config' Fixes: e89f45edb747e ("ASoC: amd: vangogh: Add check for acp config flags in vangogh platform") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/amd/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 08e42082f5e96..e724cb3c70b74 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -81,6 +81,7 @@ config SND_SOC_AMD_VANGOGH_MACH tristate "AMD Vangogh support for NAU8821 CS35L41" select SND_SOC_NAU8821 select SND_SOC_CS35L41_SPI + select SND_AMD_ACP_CONFIG depends on SND_SOC_AMD_ACP5x && I2C && SPI_MASTER help This option enables machine driver for Vangogh platform -- 2.39.2

3 3

[PATCH 1/2] ASoC: codecs: wsa883x: do not set can_multi_write flag
by Srinivas Kandagatla 05 Jun '23

05 Jun '23

regmap-sdw does not support multi register writes, so there is no point in setting this flag. This also leads to incorrect programming of WSA codecs with regmap_multi_reg_write() call. This invalid configuration should have been rejected by regmap-sdw. Fixes: 43b8c7dc85a1 ("ASoC: codecs: add wsa883x amplifier support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wsa883x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index b83b5b0d4bab..2faf66c60703 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -946,7 +946,6 @@ static struct regmap_config wsa883x_regmap_config = { .writeable_reg = wsa883x_writeable_register, .reg_format_endian = REGMAP_ENDIAN_NATIVE, .val_format_endian = REGMAP_ENDIAN_NATIVE, - .can_multi_write = true, .use_single_read = true, }; -- 2.21.0

3 12

[PATCH 0/3] Add NAU8825C support
by David Lin 05 Jun '23

05 Jun '23

This series adds nau8825c support. The driver can be used on NAU8825B and NAU8825C. David Lin (3): ASoC: nau8825: Add registers patch for NAU8825C ASoC: nau8825: Update the calculation of FLL for NAU8825C ASoC: nau8825: Update output control for NAU8825C sound/soc/codecs/nau8825.c | 111 +++++++++++++++++++++++++++++++------ sound/soc/codecs/nau8825.h | 4 ++ 2 files changed, 97 insertions(+), 18 deletions(-) -- 2.25.1

2 4

[PATCH V2 0/9] ASoC: amd: ps: add SoundWire support
by Vijendar Mukunda 05 Jun '23

05 Jun '23

This patch series add support for - Platform device creation for SoundWire Manager instances and PDM controller. - SoundWire DMA driver. - Interrupt handling for SoundWire manager related interrupts, SoundWire DMA interrupts and ACP error interrupts. - ACP PCI driver PM ops modification with respect to SoundWire Power modes. Changes since v1: - update "soundwire" as "SoundWire" in code. - add comments for platform device mask and platform device count - remove unused variables in acp pci driver private data structure - refactor dma enable register structures in SoundWire DMA driver - add TODO comments in IRQ handler - update IRQ mask values using bit macros - Fix build error reported in Makefile - rename "sdw_dma_stream_instance" structure name as "acp_sdw_dma_stream" Vijendar Mukunda (9): ASoC: amd: ps: create platform devices based on acp config ASoC: amd: ps: handle SoundWire interrupts in acp pci driver ASoC: amd: ps: add SoundWire dma driver ASoC: amd: ps: add SoundWire dma driver dma ops ASoC: amd: ps: add support for SoundWire DMA interrupts ASoC: amd: ps: add pm ops support for SoundWire dma driver ASoC: amd: ps: enable SoundWire dma driver build ASoC: amd: update comments in Kconfig file ASoC: amd: ps: Add SoundWire specific checks in pci driver in pm ops. sound/soc/amd/Kconfig | 3 +- sound/soc/amd/ps/Makefile | 2 + sound/soc/amd/ps/acp63.h | 180 ++++++++++- sound/soc/amd/ps/pci-ps.c | 396 +++++++++++++++++++++-- sound/soc/amd/ps/ps-sdw-dma.c | 572 ++++++++++++++++++++++++++++++++++ 5 files changed, 1129 insertions(+), 24 deletions(-) create mode 100644 sound/soc/amd/ps/ps-sdw-dma.c -- 2.34.1

3 34

[PATCH] ALSA: hda/realtek: Add "Intel Reference board" and "NUC 13" SSID in the ALC256
by Pierre-Louis Bossart 05 Jun '23

05 Jun '23

From: "Sayed, Karimuddin" <karimuddin.sayed(a)intel.com> Add "Intel Reference boad" and "Intel NUC 13" SSID in the alc256. Enable jack headset volume buttons Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Sayed, Karimuddin <karimuddin.sayed(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7e4765eff80..46ddd4899355 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9588,6 +9588,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x124c, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x1252, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x10ec, 0x12cc, "Intel Reference board", ALC225_FIXUP_HEADSET_JACK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP), @@ -9807,6 +9808,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), + SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC225_FIXUP_HEADSET_JACK), SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), #if 0 -- 2.37.2

2 1

[PATCH v4 00/12] ALSA: hda: Adding support for CS35L56 on HDA systems
by Richard Fitzgerald 05 Jun '23

05 Jun '23

This set of patches adds support for using the CS35L56 boosted smart amplifier on HDA systems. In these systems the CS35L56 audio is routed through a HDA-to-I2S bridge codec. This doesn't include the changes to the Realtek driver to actually hook up the CS35L56 driver, because we don't yet have the QUIRK IDs to associate it with. But we want to publish the driver now so that it is available for bringing up hardware with the CS35L56. The first 9 patches are moving code out of the ASoC driver and into the shared library so that it can be shared with the HDA driver. Patch #10 fixes missing #includes in the HDA headers so that the CS35L56 driver doesn't have to #include headers that it doesn't use. Finally, #11 and #12 actually add the support for CS35L56 on HDA. CHANGES SINCE V3 Patch 01: - Fix double-free of IRQ caused by passing wrong cookie to devm_irq_free() Patch 12: - Fix memory leak of struct firmware object and firmware name strings. - Find and open firmware files before taking the irq_lock mutex and pm_runtime. - Don't bother executing the firmware loading code if no firmware files were found. Richard Fitzgerald (4): ASoC: cs35l56: Move runtime suspend/resume to shared library ASoC: cs35l56: Move cs_dsp init into shared library ASoC: cs35l56: Move part of cs35l56_init() to shared library ALSA: hda: Fix missing header dependencies Simon Trimmer (8): ASoC: cs35l56: Move shared data into a common data structure ASoC: cs35l56: Make cs35l56_system_reset() code more generic ASoC: cs35l56: Convert utility functions to use common data structure ASoC: cs35l56: Move utility functions to shared file ASoC: cs35l56: Make common function for control port wait ASoC: cs35l56: Make a common function to shutdown the DSP ALSA: hda: Add mute_hook to hda_component ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier include/sound/cs35l56.h | 29 +- sound/pci/hda/Kconfig | 31 + sound/pci/hda/Makefile | 6 + sound/pci/hda/cs35l56_hda.c | 1010 +++++++++++++++++++++++++++++ sound/pci/hda/cs35l56_hda.h | 48 ++ sound/pci/hda/cs35l56_hda_i2c.c | 69 ++ sound/pci/hda/cs35l56_hda_spi.c | 68 ++ sound/pci/hda/hda_auto_parser.h | 2 + sound/pci/hda/hda_component.h | 1 + sound/pci/hda/hda_generic.h | 3 + sound/soc/codecs/cs35l56-i2c.c | 14 +- sound/soc/codecs/cs35l56-sdw.c | 72 +- sound/soc/codecs/cs35l56-shared.c | 459 ++++++++++++- sound/soc/codecs/cs35l56-spi.c | 10 +- sound/soc/codecs/cs35l56.c | 648 ++++-------------- sound/soc/codecs/cs35l56.h | 15 +- 16 files changed, 1893 insertions(+), 592 deletions(-) create mode 100644 sound/pci/hda/cs35l56_hda.c create mode 100644 sound/pci/hda/cs35l56_hda.h create mode 100644 sound/pci/hda/cs35l56_hda_i2c.c create mode 100644 sound/pci/hda/cs35l56_hda_spi.c -- 2.30.2

2 14

[PATCH] ALSA: hda/realtek: Add a quirk for HP Slim Desktop S01
by Ai Chao 05 Jun '23

05 Jun '23

Add a quirk for HP Slim Desktop S01 to fixup headset MIC no presence. Signed-off-by: Ai Chao <aichao(a)kylinos.cn> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7e4765eff80..70b1e4b91521 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11694,6 +11694,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8719, "HP", ALC897_FIXUP_HP_HSMIC_VERB), SND_PCI_QUIRK(0x103c, 0x872b, "HP", ALC897_FIXUP_HP_HSMIC_VERB), SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2), + SND_PCI_QUIRK(0x103c, 0x8768, "HP Slim Desktop S01", ALC671_FIXUP_HP_HEADSET_MIC2), SND_PCI_QUIRK(0x103c, 0x877e, "HP 288 Pro G6", ALC671_FIXUP_HP_HEADSET_MIC2), SND_PCI_QUIRK(0x103c, 0x885f, "HP 288 Pro G8", ALC671_FIXUP_HP_HEADSET_MIC2), SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE), -- 2.25.1

2 1

[PATCH v1] ALSA: hda/realtek: Add quirks for Asus ROG 2024 laptops using CS35L41
by Stefan Binding 05 Jun '23

05 Jun '23

Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7b5f194513c7b..6f8e83b45c6f3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9547,6 +9547,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1043, 0x1b93, "ASUS G614JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), @@ -9565,6 +9566,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1f12, "ASUS UM5302", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1f92, "ASUS ROG Flow X16", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), + SND_PCI_QUIRK(0x1043, 0x3A20, "ASUS G614JZR", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x3A30, "ASUS G814JVR/JIR ", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x3A40, "ASUS G814JZR", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x3A50, "ASUS G834JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x3A60, "ASUS G634JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), -- 2.34.1

2 1

[PATCH 0/6] ALSA: emu10k1: improvements related to the driver's procfs
by Oswald Buddenhagen 05 Jun '23

05 Jun '23

FIXME: the last commit is still being tested; something appears fishy. Oswald Buddenhagen (6): ALSA: emu10k1: hide absent 2nd pointer-offset register set from /proc ALSA: emu10k1: fix writing 1st pointer-offset register set through /proc ALSA: emu10k1: actually disassemble DSP instructions in /proc ALSA: emu10k1: include FX send amounts in /proc output ALSA: emu10k1: make E-MU FPGA register dump in /proc more useful ALSA: emu10k1: vastly improve usefulness of info in /proc include/sound/emu10k1.h | 11 + sound/pci/emu10k1/emufx.c | 87 ++++--- sound/pci/emu10k1/emuproc.c | 440 ++++++++++++++++++++++-------------- sound/pci/emu10k1/io.c | 28 ++- 4 files changed, 362 insertions(+), 204 deletions(-) -- 2.40.0.152.g15d061e6df

3 11

[PATCH] sound: Switch i2c drivers back to use .probe()
by Uwe Kleine-König 05 Jun '23

05 Jun '23

After commit b8a1a4cd5a98 ("i2c: Provide a temporary .probe_new() call-back type"), all drivers being converted to .probe_new() and then 03c835f498b5 ("i2c: Switch .probe() to not take an id parameter") convert back to (the new) .probe() to be able to eventually drop .probe_new() from struct i2c_driver. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/aoa/codecs/onyx.c | 2 +- sound/aoa/codecs/tas.c | 2 +- sound/pci/hda/cs35l41_hda_i2c.c | 2 +- sound/ppc/keywest.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 4c75381f5ab8..a8a59d71dcec 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -1048,7 +1048,7 @@ static struct i2c_driver onyx_driver = { .driver = { .name = "aoa_codec_onyx", }, - .probe_new = onyx_i2c_probe, + .probe = onyx_i2c_probe, .remove = onyx_i2c_remove, .id_table = onyx_i2c_id, }; diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index f906e9aaddcf..ab1472390061 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -936,7 +936,7 @@ static struct i2c_driver tas_driver = { .driver = { .name = "aoa_codec_tas", }, - .probe_new = tas_i2c_probe, + .probe = tas_i2c_probe, .remove = tas_i2c_remove, .id_table = tas_i2c_id, }; diff --git a/sound/pci/hda/cs35l41_hda_i2c.c b/sound/pci/hda/cs35l41_hda_i2c.c index 7826b1a12d7d..b44536fbba17 100644 --- a/sound/pci/hda/cs35l41_hda_i2c.c +++ b/sound/pci/hda/cs35l41_hda_i2c.c @@ -58,7 +58,7 @@ static struct i2c_driver cs35l41_i2c_driver = { .pm = &cs35l41_hda_pm_ops, }, .id_table = cs35l41_hda_i2c_id, - .probe_new = cs35l41_hda_i2c_probe, + .probe = cs35l41_hda_i2c_probe, .remove = cs35l41_hda_i2c_remove, }; module_i2c_driver(cs35l41_i2c_driver); diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index 0c4f43963c75..dfc1fc9b701d 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -90,7 +90,7 @@ static struct i2c_driver keywest_driver = { .driver = { .name = "PMac Keywest Audio", }, - .probe_new = keywest_probe, + .probe = keywest_probe, .remove = keywest_remove, .id_table = keywest_i2c_id, }; base-commit: ac9a78681b921877518763ba0e89202254349d1b -- 2.39.2

3 2

[PATCH v1 1/1] selftests: alsa: pcm-test: Fix compiler warnings about the format
by Mirsad Goran Todorovac 05 Jun '23

05 Jun '23

GCC 11.3.0 issues warnings in this module about wrong sizes of format specifiers: pcm-test.c: In function ‘test_pcm_time’: pcm-test.c:384:68: warning: format ‘%ld’ expects argument of type ‘long int’, but argument 5 \ has type ‘unsigned int’ [-Wformat=] 384 | snprintf(msg, sizeof(msg), "rate mismatch %ld != %ld", rate, rrate); pcm-test.c:455:53: warning: format ‘%d’ expects argument of type ‘int’, but argument 4 has \ type ‘long int’ [-Wformat=] 455 | "expected %d, wrote %li", rate, frames); pcm-test.c:462:53: warning: format ‘%d’ expects argument of type ‘int’, but argument 4 has \ type ‘long int’ [-Wformat=] 462 | "expected %d, wrote %li", rate, frames); pcm-test.c:467:53: warning: format ‘%d’ expects argument of type ‘int’, but argument 4 has \ type ‘long int’ [-Wformat=] 467 | "expected %d, wrote %li", rate, frames); Simple fix according to compiler's suggestion removed the warnings. Signed-off-by: Mirsad Goran Todorovac <mirsad.todorovac(a)alu.unizg.hr> --- tools/testing/selftests/alsa/pcm-test.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/tools/testing/selftests/alsa/pcm-test.c b/tools/testing/selftests/alsa/pcm-test.c index 3e390fe67eb9..b7eef32addb4 100644 --- a/tools/testing/selftests/alsa/pcm-test.c +++ b/tools/testing/selftests/alsa/pcm-test.c @@ -381,7 +381,7 @@ static void test_pcm_time(struct pcm_data *data, enum test_class class, goto __close; } if (rrate != rate) { - snprintf(msg, sizeof(msg), "rate mismatch %ld != %ld", rate, rrate); + snprintf(msg, sizeof(msg), "rate mismatch %ld != %d", rate, rrate); goto __close; } rperiod_size = period_size; @@ -447,24 +447,24 @@ static void test_pcm_time(struct pcm_data *data, enum test_class class, frames = snd_pcm_writei(handle, samples, rate); if (frames < 0) { snprintf(msg, sizeof(msg), - "Write failed: expected %d, wrote %li", rate, frames); + "Write failed: expected %ld, wrote %li", rate, frames); goto __close; } if (frames < rate) { snprintf(msg, sizeof(msg), - "expected %d, wrote %li", rate, frames); + "expected %ld, wrote %li", rate, frames); goto __close; } } else { frames = snd_pcm_readi(handle, samples, rate); if (frames < 0) { snprintf(msg, sizeof(msg), - "expected %d, wrote %li", rate, frames); + "expected %ld, wrote %li", rate, frames); goto __close; } if (frames < rate) { snprintf(msg, sizeof(msg), - "expected %d, wrote %li", rate, frames); + "expected %ld, wrote %li", rate, frames); goto __close; } } -- 2.34.1

3 3

[PATCH v2 00/21] ASoC: replace dpcm_playback/capture to playback/capture_only
by Kuninori Morimoto 05 Jun '23

05 Jun '23

Hi Mark This is v2 patch-set of dpcm_playback/capture flag cleanup. Current ASoC DPCM need dpcm_playback/capture flags to use it. But we are using playback/capture_only flag on Normal/Codec2Codec case. I think these are duplicated, we can share same flags for all cases. On v1 patch-set, we noticed that some DPCM BE Codec valid check breaks compatibility. static int soc_get_playback_capture(...) { ... (A) if (dai_link->dynamic || dai_link->no_pcm) { ... if (dai_link->dpcm_playback) { ... (B) for_each_rtd_cpu_dais(rtd, i, cpu_dai) { (C) if (snd_soc_dai_stream_valid(cpu_dai, stream)) { has_playback = 1; break; } } ... } (A) is for DPCM case, and "dai_link->no_pcm" means BE, (B)/(C) means CPU validation check. In many case, DPCM is like this. FE (dynamic) BE (no_pcm) [CPU/dummy-Codec] - [dummy-CPU/Codec] DPCM FE (dynamic) Codec no check is no problem, because it is dummy DAI. DPCM BE (no_pcm) Codec no check is not good, but checking it might breaks compatibility, because some Codec doesn't have necessary settings (= channels_min). Solving this issue seems not easy, because it is using very complex setting timing. v2 ignores DPCM BE Codec check, same as before, but added comment for it. I hope we can valid check for all cases in some day. v1 -> v2 - Add Reviewed-by - Separate Intel patch - tidyup playback/capture_only flags conversion Link: https://lore.kernel.org/r/87353uqjiu.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/ab3f0c0a-62fd-a468-b3cf-0e4b59bac6ae@linux.intel.… Kuninori Morimoto (21): ASoC: soc-pcm.c: indicate error if stream has no playback no capture ASoC: soc-pcm.c: use dai_link on soc_get_playback_capture() ASoC: soc-pcm.c: cleanup soc_get_playback_capture() error ASoC: soc-pcm.c: use temporary variable at soc_get_playback_capture() ASoC: soc-pcm.c: tidyup playback/capture_only at soc_get_playback_capture() ASoC: soc-pcm.c: cleanup normal connection loop at soc_get_playback_capture() part1 ASoC: soc-pcm.c: cleanup normal connection loop at soc_get_playback_capture() part2 ASoC: soc-pcm.c: cleanup soc_get_playback_capture() ASoC: amd: replace dpcm_playback/capture to playback/capture_only ASoC: fsl: replace dpcm_playback/capture to playback/capture_only ASoC: sof: replace dpcm_playback/capture to playback/capture_only ASoC: meson: replace dpcm_playback/capture to playback/capture_only ASoC: Intel: replace dpcm_playback/capture to playback/capture_only ASoC: samsung: replace dpcm_playback/capture to playback/capture_only ASoC: mediatek: replace dpcm_playback/capture to playback/capture_only ASoC: soc-dai.c: replace dpcm_playback/capture to playback/capture_only ASoC: Intel/avs: replace dpcm_playback/capture to playback/capture_only ASoC: soc-core.c: replace dpcm_playback/capture to playback/capture_only ASoC: soc-topology.c: replace dpcm_playback/capture to playback/capture_only ASoC: soc-compress.c: replace dpcm_playback/capture to playback/capture_only ASoC: soc-pcm.c: remove dpcm_playback/capture include/sound/soc.h | 4 - sound/soc/amd/acp-da7219-max98357a.c | 20 +-- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 20 +-- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 3 - sound/soc/fsl/fsl-asoc-card.c | 16 +-- sound/soc/fsl/imx-audmix.c | 6 +- sound/soc/fsl/imx-card.c | 4 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 32 +++-- sound/soc/intel/boards/bxt_rt298.c | 26 ++-- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 14 +-- sound/soc/intel/boards/ehl_rt5660.c | 14 +-- sound/soc/intel/boards/glk_rt5682_max98357a.c | 30 +++-- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 26 ++-- sound/soc/intel/boards/kbl_da7219_max98927.c | 54 ++++----- sound/soc/intel/boards/kbl_rt5660.c | 18 ++- sound/soc/intel/boards/kbl_rt5663_max98927.c | 44 +++---- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 22 ++-- sound/soc/intel/boards/skl_hda_dsp_common.c | 14 +-- .../soc/intel/boards/skl_nau88l25_max98357a.c | 26 ++-- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 26 ++-- sound/soc/intel/boards/skl_rt286.c | 26 ++-- sound/soc/intel/boards/sof_cs42l42.c | 12 +- sound/soc/intel/boards/sof_da7219_max98373.c | 16 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_nau8825.c | 12 +- sound/soc/intel/boards/sof_pcm512x.c | 8 +- sound/soc/intel/boards/sof_rt5682.c | 12 +- sound/soc/intel/boards/sof_sdw.c | 4 +- sound/soc/intel/boards/sof_ssp_amp.c | 11 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +-- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++-- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++--- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++--- .../mt8186/mt8186-mt6366-da7219-max98357.c | 86 +++++-------- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 86 +++++-------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 48 ++++---- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 ++++++------ sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 +++++---- sound/soc/meson/axg-card.c | 8 +- sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/samsung/odroid.c | 10 +- sound/soc/soc-compress.c | 11 +- sound/soc/soc-core.c | 20 +-- sound/soc/soc-dai.c | 6 +- sound/soc/soc-pcm.c | 114 +++++++----------- sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 4 - 84 files changed, 511 insertions(+), 842 deletions(-) -- 2.25.1

3 28

[PATCH v1 00/43] ep93xx device tree conversion
by Nikita Shubin 04 Jun '23

04 Jun '23

This series aims to convert ep93xx from platform to full device tree support. Alexander, Kris - there are some significant changes in clk and pinctrl so can i ask you to tests all once again. So i am not applying: Tested-by: Michael Peters <mpeters(a)embeddedTS.com> Reviewed-by: Kris Bahnsen <kris(a)embeddedTS.com> Tags yet. Major changes from v0 to v1: - I totally forgot to include dts bindings for USB, they are working, including in this version - retinkered ep93xx keypad, the stange thing about it that it always used zeroed platform data from the very beginning - my first impulse was to remove it entirely, espesially it's ep9307+ variant, which Alexander and me doesn't have - major YAML bindings overhaul according to Krzysztof comments - nand helper converted to LEGACY nand controller - cleanup clk - cleanup pinctrl Sorry if i missed something, first time handling such a big (at least for me) chunk of patches. Next version should be much faster spin. Alexandre Belloni: st,m48t86 is totally trivial, but it has 2 regs instead of one, so dt_binding_check doesn't allow it in trivial.yaml, regs should be increased to "maxItems: 2" Miquel Raynal: Currently made it LEGACY as a more easier way for now, as this series will merge - it will be much easier to cleanup the rest one by one, i hope it's ok. Stephen Boyd: Majory of issues fixed, but: - removing dma from init section requires converting it from half dt/platform monstrosity into fully dt compatible - i would like to have ep93xx_clk_data and ep93xx_map global for now - they can be removed once dma subsys_initcall removed Andrew Lunn: I've tinkered with the preferred way, however this involves turning on - CONFIG_PHYLIB - CONFIG_MDIO_DEVICE And maybe CONFIG_MICREL_PHY (at least for me, unless i can use some common phy driver) which implies a kernel size increase - which is undesirable for us. Can we slip by getting phyid directly from device tree in ep93xx_eth ? Link: https://lore.kernel.org/all/20230424123522.18302-1-nikita.shubin@maquefel.m… Cc: kris(a)embeddedTS.com Alexander Sverdlin (3): ARM: dts: ep93xx: Add I2S and AC97 nodes ARM: dts: ep93xx: Add EDB9302 DT ASoC: cirrus: edb93xx: Delete driver Nikita Shubin (40): gpio: ep93xx: split device in multiple dt-bindings: soc: Add Cirrus EP93xx soc: Add SoC driver for Cirrus ep93xx dt-bindings: clock: Add Cirrus EP93xx clk: ep93xx: add DT support for Cirrus EP93xx dt-bindings: pinctrl: Add Cirrus EP93xx pinctrl: add a Cirrus ep93xx SoC pin controller dt-bindings: timers: Add Cirrus EP93xx clocksource: ep93xx: Add driver for Cirrus Logic EP93xx dt-bindings: rtc: Add Cirrus EP93xx rtc: ep93xx: add DT support for Cirrus EP93xx dt-bindings: watchdog: Add Cirrus EP93x watchdog: ep93xx: add DT support for Cirrus EP93xx power: reset: Add a driver for the ep93xx reset dt-bindings: pwm: Add Cirrus EP93xx pwm: ep93xx: add DT support for Cirrus EP93xx dt-bindings: spi: Add Cirrus EP93xx spi: ep93xx: add DT support for Cirrus EP93xx dt-bindings: net: Add Cirrus EP93xx net: cirrus: add DT support for Cirrus EP93xx dt-bindings: dma: Add Cirrus EP93xx dma: cirrus: add DT support for Cirrus EP93xx dt-bindings: mtd: Add ts7250 nand-controller mtd: nand: add support for ts72xx dt-bindings: ata: Add Cirrus EP93xx pata: cirrus: add DT support for Cirrus EP93xx dt-bindings: input: Add Cirrus EP93xx keypad input: keypad: ep93xx: add DT support for Cirrus EP93xx dt-bindings: rtc: Add ST M48T86 rtc: m48t86: add DT support for m48t86 dt-bindings: wdt: Add ts72xx wdt: ts72xx: add DT support for ts72xx dt-bindings: gpio: Add Cirrus EP93xx gpio: ep93xx: add DT support for gpio-ep93xx ARM: dts: add device tree for ep93xx Soc ARM: ep93xx: DT for the Cirrus ep93xx SoC platforms pwm: ep93xx: drop legacy pinctrl pata: cirrus: drop legacy pinctrl ARM: ep93xx: delete all boardfiles ARM: ep93xx: soc: drop defines .../devicetree/bindings/arm/ep93xx.yaml | 107 ++ .../bindings/ata/cirrus,ep9312-pata.yaml | 44 + .../bindings/clock/cirrus,ep9301.yaml | 64 + .../bindings/dma/cirrus,ep9301-dma-m2m.yaml | 72 + .../bindings/dma/cirrus,ep9301-dma-m2p.yaml | 124 ++ .../devicetree/bindings/gpio/gpio-ep9301.yaml | 154 ++ .../bindings/input/cirrus,ep9307-keypad.yaml | 86 + .../bindings/mtd/technologic,nand.yaml | 47 + .../bindings/net/cirrus,ep9301-eth.yaml | 61 + .../pinctrl/cirrus,ep9301-pinctrl.yaml | 66 + .../bindings/pwm/cirrus,ep9301-pwm.yaml | 48 + .../bindings/rtc/cirrus,ep9301-rtc.yaml | 40 + .../bindings/rtc/st,m48t86-rtc.yaml | 38 + .../devicetree/bindings/spi/spi-ep9301.yaml | 69 + .../bindings/timer/cirrus,ep9301-timer.yaml | 49 + .../bindings/watchdog/cirrus,ep9301-wdt.yaml | 46 + .../watchdog/technologic,ts7200-wdt.yaml | 46 + arch/arm/Makefile | 1 - arch/arm/boot/dts/Makefile | 1 + arch/arm/boot/dts/ep93xx-bk3.dts | 119 ++ arch/arm/boot/dts/ep93xx-edb9302.dts | 160 ++ arch/arm/boot/dts/ep93xx-ts7250.dts | 132 ++ arch/arm/boot/dts/ep93xx.dtsi | 477 +++++ arch/arm/mach-ep93xx/Kconfig | 20 +- arch/arm/mach-ep93xx/Makefile | 11 - arch/arm/mach-ep93xx/clock.c | 733 -------- arch/arm/mach-ep93xx/core.c | 1017 ---------- arch/arm/mach-ep93xx/dma.c | 114 -- arch/arm/mach-ep93xx/edb93xx.c | 344 ---- arch/arm/mach-ep93xx/ep93xx-regs.h | 38 - arch/arm/mach-ep93xx/gpio-ep93xx.h | 111 -- arch/arm/mach-ep93xx/hardware.h | 25 - arch/arm/mach-ep93xx/irqs.h | 76 - arch/arm/mach-ep93xx/platform.h | 42 - arch/arm/mach-ep93xx/soc.h | 212 --- arch/arm/mach-ep93xx/ts72xx.c | 422 ----- arch/arm/mach-ep93xx/ts72xx.h | 94 - arch/arm/mach-ep93xx/vision_ep9307.c | 311 --- drivers/ata/pata_ep93xx.c | 33 +- drivers/clk/Kconfig | 8 + drivers/clk/Makefile | 1 + drivers/clk/clk-ep93xx.c | 850 +++++++++ drivers/clocksource/Kconfig | 11 + drivers/clocksource/Makefile | 1 + .../clocksource}/timer-ep93xx.c | 141 +- drivers/dma/ep93xx_dma.c | 136 +- drivers/gpio/gpio-ep93xx.c | 329 ++-- drivers/input/keyboard/ep93xx_keypad.c | 78 +- drivers/mtd/nand/raw/Kconfig | 7 + drivers/mtd/nand/raw/Makefile | 1 + .../nand/raw/technologic-nand-controller.c | 151 ++ drivers/net/ethernet/cirrus/ep93xx_eth.c | 67 +- drivers/pinctrl/Kconfig | 7 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/pinctrl-ep93xx.c | 1672 +++++++++++++++++ drivers/power/reset/Kconfig | 10 + drivers/power/reset/Makefile | 1 + drivers/power/reset/ep93xx-restart.c | 65 + drivers/pwm/pwm-ep93xx.c | 26 +- drivers/rtc/rtc-ep93xx.c | 8 + drivers/rtc/rtc-m48t86.c | 8 + drivers/soc/Kconfig | 1 + drivers/soc/Makefile | 1 + drivers/soc/cirrus/Kconfig | 11 + drivers/soc/cirrus/Makefile | 2 + drivers/soc/cirrus/soc-ep93xx.c | 134 ++ drivers/spi/spi-ep93xx.c | 31 +- drivers/watchdog/ep93xx_wdt.c | 8 + drivers/watchdog/ts72xx_wdt.c | 8 + .../dt-bindings/clock/cirrus,ep93xx-clock.h | 53 + include/linux/platform_data/dma-ep93xx.h | 3 + include/linux/platform_data/eth-ep93xx.h | 10 - include/linux/platform_data/keypad-ep93xx.h | 32 - include/linux/soc/cirrus/ep93xx.h | 40 +- sound/soc/cirrus/Kconfig | 9 - sound/soc/cirrus/Makefile | 4 - sound/soc/cirrus/edb93xx.c | 117 -- 77 files changed, 5575 insertions(+), 4122 deletions(-) create mode 100644 Documentation/devicetree/bindings/arm/ep93xx.yaml create mode 100644 Documentation/devicetree/bindings/ata/cirrus,ep9312-pata.yaml create mode 100644 Documentation/devicetree/bindings/clock/cirrus,ep9301.yaml create mode 100644 Documentation/devicetree/bindings/dma/cirrus,ep9301-dma-m2m.yaml create mode 100644 Documentation/devicetree/bindings/dma/cirrus,ep9301-dma-m2p.yaml create mode 100644 Documentation/devicetree/bindings/gpio/gpio-ep9301.yaml create mode 100644 Documentation/devicetree/bindings/input/cirrus,ep9307-keypad.yaml create mode 100644 Documentation/devicetree/bindings/mtd/technologic,nand.yaml create mode 100644 Documentation/devicetree/bindings/net/cirrus,ep9301-eth.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/cirrus,ep9301-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pwm/cirrus,ep9301-pwm.yaml create mode 100644 Documentation/devicetree/bindings/rtc/cirrus,ep9301-rtc.yaml create mode 100644 Documentation/devicetree/bindings/rtc/st,m48t86-rtc.yaml create mode 100644 Documentation/devicetree/bindings/spi/spi-ep9301.yaml create mode 100644 Documentation/devicetree/bindings/timer/cirrus,ep9301-timer.yaml create mode 100644 Documentation/devicetree/bindings/watchdog/cirrus,ep9301-wdt.yaml create mode 100644 Documentation/devicetree/bindings/watchdog/technologic,ts7200-wdt.yaml create mode 100644 arch/arm/boot/dts/ep93xx-bk3.dts create mode 100644 arch/arm/boot/dts/ep93xx-edb9302.dts create mode 100644 arch/arm/boot/dts/ep93xx-ts7250.dts create mode 100644 arch/arm/boot/dts/ep93xx.dtsi delete mode 100644 arch/arm/mach-ep93xx/Makefile delete mode 100644 arch/arm/mach-ep93xx/clock.c delete mode 100644 arch/arm/mach-ep93xx/core.c delete mode 100644 arch/arm/mach-ep93xx/dma.c delete mode 100644 arch/arm/mach-ep93xx/edb93xx.c delete mode 100644 arch/arm/mach-ep93xx/ep93xx-regs.h delete mode 100644 arch/arm/mach-ep93xx/gpio-ep93xx.h delete mode 100644 arch/arm/mach-ep93xx/hardware.h delete mode 100644 arch/arm/mach-ep93xx/irqs.h delete mode 100644 arch/arm/mach-ep93xx/platform.h delete mode 100644 arch/arm/mach-ep93xx/soc.h delete mode 100644 arch/arm/mach-ep93xx/ts72xx.c delete mode 100644 arch/arm/mach-ep93xx/ts72xx.h delete mode 100644 arch/arm/mach-ep93xx/vision_ep9307.c create mode 100644 drivers/clk/clk-ep93xx.c rename {arch/arm/mach-ep93xx => drivers/clocksource}/timer-ep93xx.c (52%) create mode 100644 drivers/mtd/nand/raw/technologic-nand-controller.c create mode 100644 drivers/pinctrl/pinctrl-ep93xx.c create mode 100644 drivers/power/reset/ep93xx-restart.c create mode 100644 drivers/soc/cirrus/Kconfig create mode 100644 drivers/soc/cirrus/Makefile create mode 100644 drivers/soc/cirrus/soc-ep93xx.c create mode 100644 include/dt-bindings/clock/cirrus,ep93xx-clock.h delete mode 100644 include/linux/platform_data/eth-ep93xx.h delete mode 100644 include/linux/platform_data/keypad-ep93xx.h delete mode 100644 sound/soc/cirrus/edb93xx.c -- 2.37.4

2 1

[PATCH] ASoC: simple-card-utils: fix PCM constraint error check
by Robert Hancock 02 Jun '23

02 Jun '23

The code in asoc_simple_startup was treating any non-zero return from snd_pcm_hw_constraint_minmax as an error, when this can return 1 in some normal cases and only negative values indicate an error. When this happened, it caused asoc_simple_startup to disable the clocks it just enabled and return 1, which was not treated as an error by the calling code which only checks for negative return values. Then when the PCM is eventually shut down, it causes the clock framework to complain about disabling clocks that were not enabled. Fix the check for snd_pcm_hw_constraint_minmax return value to only treat negative values as an error. Fixes: 5ca2ab459817 ("ASoC: simple-card-utils: Add new system-clock-fixed flag") Signed-off-by: Robert Hancock <robert.hancock(a)calian.com> --- sound/soc/generic/simple-card-utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 467edd96eae5..e5ff61c1e9d1 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -314,7 +314,7 @@ int asoc_simple_startup(struct snd_pcm_substream *substream) } ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, fixed_rate, fixed_rate); - if (ret) + if (ret < 0) goto codec_err; } -- 2.40.1

2 1

[PATCH] ASoC: fsl_sai: Enable BCI bit if SAI works on synchronous mode with BYP asserted
by Chancel Liu 02 Jun '23

02 Jun '23

There's an issue on SAI synchronous mode that TX/RX side can't get BCLK from RX/TX it sync with if BYP bit is asserted. It's a workaround to fix it that enable SION of IOMUX pad control and assert BCI. For example if TX sync with RX which means both TX and RX are using clk form RX and BYP=1. TX can get BCLK only if the following two conditions are valid: 1. SION of RX BCLK IOMUX pad is set to 1 2. BCI of TX is set to 1 Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 11 +++++++++-- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d9344025dc16..5e09f634c61b 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -491,14 +491,21 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) regmap_update_bits(sai->regmap, reg, FSL_SAI_CR2_MSEL_MASK, FSL_SAI_CR2_MSEL(sai->mclk_id[tx])); - if (savediv == 1) + if (savediv == 1) { regmap_update_bits(sai->regmap, reg, FSL_SAI_CR2_DIV_MASK | FSL_SAI_CR2_BYP, FSL_SAI_CR2_BYP); - else + if (fsl_sai_dir_is_synced(sai, adir)) + regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx, ofs), + FSL_SAI_CR2_BCI, FSL_SAI_CR2_BCI); + else + regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx, ofs), + FSL_SAI_CR2_BCI, 0); + } else { regmap_update_bits(sai->regmap, reg, FSL_SAI_CR2_DIV_MASK | FSL_SAI_CR2_BYP, savediv / 2 - 1); + } if (sai->soc_data->max_register >= FSL_SAI_MCTL) { /* SAI is in master mode at this point, so enable MCLK */ diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 3eb994aef36a..8254c3547b87 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -116,6 +116,7 @@ /* SAI Transmit and Receive Configuration 2 Register */ #define FSL_SAI_CR2_SYNC BIT(30) +#define FSL_SAI_CR2_BCI BIT(28) #define FSL_SAI_CR2_MSEL_MASK (0x3 << 26) #define FSL_SAI_CR2_MSEL_BUS 0 #define FSL_SAI_CR2_MSEL_MCLK1 BIT(26) -- 2.25.1

3 2

configure: Update for autoconf 2.71
by GitHub pull_request - opened 02 Jun '23

02 Jun '23

alsa-project/alsa-lib pull request #324 was opened from phodina: Hi, this is an update due to Alpine Linux packaging only `autoconf-2.71-r2` so that the https://github.com/alsa-project/alsa-lib/pull/308 could be tested. Request URL : https://github.com/alsa-project/alsa-lib/pull/324 Patch URL : https://github.com/alsa-project/alsa-lib/pull/324.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: SOF: ipc4-topology: Use size_t for variable passed to kzalloc()
by Christophe JAILLET 02 Jun '23

02 Jun '23

struct_size() checks for overflow, but assigning its result to just a u32 may still overflow after a successful check. Use a size_t instead in order to be cleaner. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Based on analysis from Dan Carpenter on another patch (see [1]). [1]: https://lore.kernel.org/all/00e84595-e2c9-48ea-8737-18da34eaafbf@kili.mount… --- sound/soc/sof/ipc4-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index db64e0cb8663..50faa4c88b97 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -881,7 +881,7 @@ static int sof_ipc4_widget_setup_comp_process(struct snd_sof_widget *swidget) /* allocate memory for base config extension if needed */ if (process->init_config == SOF_IPC4_MODULE_INIT_CONFIG_TYPE_BASE_CFG_WITH_EXT) { struct sof_ipc4_base_module_cfg_ext *base_cfg_ext; - u32 ext_size = struct_size(base_cfg_ext, pin_formats, + size_t ext_size = struct_size(base_cfg_ext, pin_formats, swidget->num_input_pins + swidget->num_output_pins); base_cfg_ext = kzalloc(ext_size, GFP_KERNEL); -- 2.34.1

3 3

[PATCH 1/4] soundwire: stream: Add missing clear of alloc_slave_rt
by Charles Keepax 02 Jun '23

02 Jun '23

The current path that skips allocating the slave runtime does not clear the alloc_slave_rt flag, this is clearly incorrect. Add the missing clear, so the runtime won't be erroneously cleaned up. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- drivers/soundwire/stream.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index c2191c07442b0..379228f221869 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -2021,8 +2021,10 @@ int sdw_stream_add_slave(struct sdw_slave *slave, skip_alloc_master_rt: s_rt = sdw_slave_rt_find(slave, stream); - if (s_rt) + if (s_rt) { + alloc_slave_rt = false; goto skip_alloc_slave_rt; + } s_rt = sdw_slave_rt_alloc(slave, m_rt); if (!s_rt) { -- 2.30.2

2 9

GSequencer enters MIDI awards a second time
by Joël Krähemann 02 Jun '23

02 Jun '23

Hi all, Please check and vote for your favorite sequencer: https://www.midi.org/component/zoo/item/advanced-gtk-sequencer-2?Itemid=1423 cheers, Joël http://gsequencer.org

1 0

[PATCH] ALSA: hda/realtek: Add Lenovo P3 Tower platform
by RenHai 02 Jun '23

02 Jun '23

Headset microphone on this platform does not work without ALC897_FIXUP_HEADSET_MIC_PIN fixup. Signed-off-by: RenHai <kean0048(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7b5f194513c7..d0b44cb6f41f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11715,6 +11715,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x14cd, 0x5003, "USI", ALC662_FIXUP_USI_HEADSET_MODE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC662_FIXUP_LENOVO_MULTI_CODECS), SND_PCI_QUIRK(0x17aa, 0x1057, "Lenovo P360", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x1064, "Lenovo P3 Tower", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32ca, "Lenovo ThinkCentre M80", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32cb, "Lenovo ThinkCentre M70", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x32cf, "Lenovo ThinkCentre M950", ALC897_FIXUP_HEADSET_MIC_PIN), -- 2.25.1

1 0

[PATCH 1/2] ASoC: max98363: Removed 32bit support
by “Ryan 01 Jun '23

01 Jun '23

From: Ryan Lee <ryans.lee(a)analog.com> MAX98363 does not support 32bit depth audio. Removed 32bit from the supported format list. Instead, added 16bit and 24bit to the list. Signed-off-by: Ryan Lee <ryans.lee(a)analog.com> --- sound/soc/codecs/max98363.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98363.c b/sound/soc/codecs/max98363.c index 4585ebb1e82c..8aa9d9c67aa2 100644 --- a/sound/soc/codecs/max98363.c +++ b/sound/soc/codecs/max98363.c @@ -211,7 +211,7 @@ static int max98363_io_init(struct sdw_slave *slave) } #define MAX98363_RATES SNDRV_PCM_RATE_8000_192000 -#define MAX98363_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) +#define MAX98363_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) static int max98363_sdw_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, -- 2.34.1

2 2

[PATCH] ASoC: cs35l56: Remove NULL check from cs35l56_sdw_dai_set_stream()
by Richard Fitzgerald 01 Jun '23

01 Jun '23

The dma pointer must be set to the passed stream pointer, even if that pointer is NULL. Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 3c07bd1e959e..c03f9d3c9a13 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -704,9 +704,6 @@ static int cs35l56_sdw_dai_hw_free(struct snd_pcm_substream *substream, static int cs35l56_sdw_dai_set_stream(struct snd_soc_dai *dai, void *sdw_stream, int direction) { - if (!sdw_stream) - return 0; - snd_soc_dai_dma_data_set(dai, direction, sdw_stream); return 0; -- 2.30.2

3 2

[PATCH v3 0/1] ASoC: SOF: add mt8188 audio support
by Trevor Wu 01 Jun '23

01 Jun '23

This series adds mt8188 audio support and dbg_dump callback for mt8186 and mt8188. Changes since v2: - resend the correct patch for adsp debug dump Changes since v1: - fix typo - adopt reviewer's suggestion Trevor Wu (1): ASoC: SOF: mediatek: add adsp debug dump sound/soc/sof/mediatek/mt8186/mt8186.c | 22 ++++++++++++++++++++++ sound/soc/sof/mediatek/mt8186/mt8186.h | 5 +++++ 2 files changed, 27 insertions(+) -- 2.18.0

3 3

[PATCH] firmware: cs_dsp: Log that a bin file was loaded
by Richard Fitzgerald 01 Jun '23

01 Jun '23

Change the message at the start of bin file loading from cs_dsp_dbg() to cs_dsp_info() so that there is confirmation in the kernel log that a bin file was loaded, and the name of the file. Signed-off-by: Richard Fitzgerald <rf(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 e4ccfb6a8fa5..d7e46a57ecf9 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -2059,10 +2059,10 @@ static int cs_dsp_load_coeff(struct cs_dsp *dsp, const struct firmware *firmware goto out_fw; } - cs_dsp_dbg(dsp, "%s: v%d.%d.%d\n", file, - (le32_to_cpu(hdr->ver) >> 16) & 0xff, - (le32_to_cpu(hdr->ver) >> 8) & 0xff, - le32_to_cpu(hdr->ver) & 0xff); + cs_dsp_info(dsp, "%s: v%d.%d.%d\n", file, + (le32_to_cpu(hdr->ver) >> 16) & 0xff, + (le32_to_cpu(hdr->ver) >> 8) & 0xff, + le32_to_cpu(hdr->ver) & 0xff); pos = le32_to_cpu(hdr->len); -- 2.30.2

2 1

CSC3551 and devices missing related _DSD bits
by Luke Jones 01 Jun '23

01 Jun '23

Hi Takashi (and all others), We're beginning to encounter more and more laptops from ASUS in the ROG range where the bass speakers are connected to a cirrus amp, and the driver for cirrus is unable to pick this up due to being missing a _DSD section in the ACPI (SSDT). I've emailed engineers at ASUS I have contact with about this but it is unlikely that we will see corrections for each and every model + variant they have. The models we have seen so far are: - GX650P - GU604V - GA402X And there have been numerous forum postings in various places with the exact same issue for other models from ASUS that are not in the ROG group I deal with - oh and also some Lenovo and HP devices also. Some of the logs are: [ 5.542312] Serial bus multi instantiate pseudo device driver CSC3551:00: Instantiated 2 I2C devices. [ 5.717261] cs35l41-hda i2c-CSC3551:00-cs35l41-hda.0: Error: ACPI _DSD Properties are missing for HID CSC3551. [ 5.717269] cs35l41-hda i2c-CSC3551:00-cs35l41-hda.0: error -EINVAL: Platform not supported [ 5.717286] cs35l41-hda: probe of i2c-CSC3551:00-cs35l41-hda.0 failed with error -22 [ 5.717734] cs35l41-hda i2c-CSC3551:00-cs35l41-hda.1: Error: ACPI _DSD Properties are missing for HID CSC3551. [ 5.717737] cs35l41-hda i2c-CSC3551:00-cs35l41-hda.1: error -EINVAL: Platform not supported [ 5.717750] cs35l41-hda: probe of i2c-CSC3551:00-cs35l41-hda.1 failed with error -22 and: [ 3.343851] cs35l41-hda spi1-CSC3551:00-cs35l41-hda.0: Error: ACPI _DSD Properties are missing for HID CSC3551. [ 3.343854] cs35l41-hda spi1-CSC3551:00-cs35l41-hda.0: error -EINVAL: Platform not supported [ 3.343855] cs35l41-hda: probe of spi1-CSC3551:00-cs35l41-hda.0 failed with error -22 [ 3.345544] cs35l41-hda spi1-CSC3551:00-cs35l41-hda.1: Error: ACPI _DSD Properties are missing for HID CSC3551. [ 3.345546] cs35l41-hda spi1-CSC3551:00-cs35l41-hda.1: error -EINVAL: Platform not supported [ 3.345548] cs35l41-hda: probe of spi1-CSC3551:00-cs35l41-hda.1 failed with error -22 So there are SPI connected, and I2C connected. Alex has already submitted the related patch_realtek work for the GU604V and no doubt there will be some follow up patches from myself or others for the other models. Alex has attempted to patch the Cirrus related parts in kernel direct with: 1. https://github.com/bno1/linux-xanmod-gu604/blob/acpi-hack/ patch03_gu604v_speaker_acpi_hack.patch 2. https://github.com/bno1/linux-xanmod-gu604/blob/acpi-hack/ patch04_gu604v_spi_cs_gpio_hack.patch both of these patches have been tested and working for over a week. From what I see the patch04_gu604v_spi_cs_gpio_hack patch needs to shift to the cirrus driver sound/pci/hda/cs35l41_hda.c, and that driver must now have a quirk table - unless we can somehow add the cirrus related bits to the realtek code? I've personally not played with this beyond keeping an eye on it all and am reluctant to begin work on a solution until I've had some input on the direction it should take. In any case, this is not an isolated issue and we could benefit from putting in place some quirk or similar to accommodate for laptops where the vendor for some reason does not include the _DSD parts for use with the cirrus driver. The trend is looking so far like this practice will continue regardless of the vendor. Known reports: - https://bugzilla.kernel.org/show_bug.cgi?id=216194 - https://www.spinics.net/lists/alsa-devel/msg146434.html - https://gist.github.com/lamperez/862763881c0e1c812392b5574727f6ff - https://forums.linuxmint.com/viewtopic.php?t=392762 - https://h30434.www3.hp.com/t5/Notebook-Operating-System-and-Recovery/Envy-x… Regards, Luke

4 7

[PATCH 00/43] ep93xx device tree conversion
by Nikita Shubin 01 Jun '23

01 Jun '23

This series aims to convert ep93xx from platform to full device tree support. Tested on ts7250 64 RAM/128 MiB Nand flash, edb9302. Thank you Linus and Arnd for your support, review and comments, sorry if i missed something - these series are quite big for me. Big thanks to Alexander Sverdlin for his testing, support, review, fixes and patches. Alexander Sverdlin (4): ARM: dts: ep93xx: Add ADC node ARM: dts: ep93xx: Add I2S and AC97 nodes ARM: dts: ep93xx: Add EDB9302 DT ASoC: cirrus: edb93xx: Delete driver Nikita Shubin (39): gpio: ep93xx: split device in multiple soc: Add SoC driver for Cirrus ep93xx dt-bindings: pinctrl: Add DT bindings ep93xx pinctrl pinctrl: add a Cirrus ep93xx SoC pin controller dt-bindings: timers: add DT bindings for Cirrus EP93xx clocksource: ep93xx: Add driver for Cirrus Logic EP93xx dt-bindings: rtc: add DT bindings for Cirrus EP93xx rtc: ep93xx: add DT support for Cirrus EP93xx dt-bindings: watchdog: add DT bindings for Cirrus EP93x watchdog: ep93xx: add DT support for Cirrus EP93xx dt-bindings: clock: add DT bindings for Cirrus EP93xx clk: ep93xx: add DT support for Cirrus EP93xx power: reset: Add a driver for the ep93xx reset dt-bindings: pwm: Add DT bindings ep93xx PWM pwm: ep93xx: add DT support for Cirrus EP93xx dt-bindings: spi: Add DT bindings ep93xx spi spi: ep93xx: add DT support for Cirrus EP93xx dt-bindings: net: Add DT bindings ep93xx eth net: cirrus: add DT support for Cirrus EP93xx dt-bindings: dma: Add DT bindings ep93xx dma dma: cirrus: add DT support for Cirrus EP93xx dt-bindings: mtd: add DT bindings for ts7250 nand mtd: ts72xx_nand: add platform helper dt-bindings: ata: Add DT bindings ep93xx pata pata: cirrus: add DT support for Cirrus EP93xx dt-bindings: input: Add DT bindings ep93xx keypad input: keypad: ep93xx: add DT support for Cirrus EP93xx dt-bindings: rtc: Add DT binding m48t86 rtc rtc: m48t86: add DT support for m48t86 dt-bindings: wdt: Add DT binding ts72xx wdt wdt: ts72xx: add DT support for ts72xx dt-bindings: gpio: Add DT bindings ep93xx gpio gpio: ep93xx: add DT support for gpio-ep93xx ARM: dts: add device tree for ep93xx Soc ARM: ep93xx: DT for the Cirrus ep93xx SoC platforms pwm: ep93xx: drop legacy pinctrl input: keypad: ep93xx: drop legacy pinctrl ARM: ep93xx: soc: drop defines ARM: ep93xx: delete all boardfiles .../devicetree/bindings/arm/ep93xx.yaml | 99 + .../bindings/ata/cirrus,ep93xx-pata.yaml | 40 + .../bindings/dma/cirrus,ep93xx-dma-m2m.yaml | 66 + .../bindings/dma/cirrus,ep93xx-dma-m2p.yaml | 102 + .../devicetree/bindings/gpio/gpio-ep93xx.yaml | 161 ++ .../bindings/input/cirrus,ep93xx-keypad.yaml | 123 ++ .../bindings/mtd/technologic,nand.yaml | 56 + .../bindings/net/cirrus,ep93xx_eth.yaml | 51 + .../pinctrl/cirrus,ep93xx-pinctrl.yaml | 66 + .../bindings/pwm/cirrus,ep93xx-pwm.yaml | 45 + .../bindings/rtc/cirrus,ep93xx-rtc.yaml | 32 + .../bindings/rtc/dallas,rtc-m48t86.yaml | 33 + .../devicetree/bindings/spi/spi-ep93xx.yaml | 68 + .../bindings/timer/cirrus,ep93xx-timer.yaml | 41 + .../bindings/watchdog/cirrus,ep93xx-wdt.yaml | 38 + .../watchdog/technologic,ts72xx-wdt.yaml | 39 + arch/arm/Makefile | 1 - arch/arm/boot/dts/Makefile | 1 + arch/arm/boot/dts/ep93xx-bk3.dts | 96 + arch/arm/boot/dts/ep93xx-edb9302.dts | 150 ++ arch/arm/boot/dts/ep93xx-ts7250.dts | 113 ++ arch/arm/boot/dts/ep93xx.dtsi | 466 +++++ arch/arm/mach-ep93xx/Kconfig | 20 +- arch/arm/mach-ep93xx/Makefile | 11 - arch/arm/mach-ep93xx/core.c | 1017 ---------- arch/arm/mach-ep93xx/dma.c | 114 -- arch/arm/mach-ep93xx/edb93xx.c | 344 ---- arch/arm/mach-ep93xx/ep93xx-regs.h | 38 - arch/arm/mach-ep93xx/gpio-ep93xx.h | 111 -- arch/arm/mach-ep93xx/hardware.h | 25 - arch/arm/mach-ep93xx/irqs.h | 76 - arch/arm/mach-ep93xx/platform.h | 42 - arch/arm/mach-ep93xx/soc.h | 212 -- arch/arm/mach-ep93xx/ts72xx.c | 422 ---- arch/arm/mach-ep93xx/ts72xx.h | 94 - arch/arm/mach-ep93xx/vision_ep9307.c | 311 --- drivers/ata/pata_ep93xx.c | 9 + drivers/clk/Kconfig | 8 + drivers/clk/Makefile | 1 + .../clock.c => drivers/clk/clk-ep93xx.c | 491 +++-- drivers/clocksource/Kconfig | 11 + drivers/clocksource/Makefile | 1 + .../clocksource}/timer-ep93xx.c | 143 +- drivers/dma/ep93xx_dma.c | 119 +- drivers/gpio/gpio-ep93xx.c | 329 ++-- drivers/input/keyboard/ep93xx_keypad.c | 25 +- drivers/mtd/nand/raw/Kconfig | 8 + drivers/mtd/nand/raw/Makefile | 1 + drivers/mtd/nand/raw/ts72xx_nand.c | 94 + drivers/net/ethernet/cirrus/ep93xx_eth.c | 49 +- drivers/pinctrl/Kconfig | 7 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/pinctrl-ep93xx.c | 1698 +++++++++++++++++ drivers/power/reset/Kconfig | 10 + drivers/power/reset/Makefile | 1 + drivers/power/reset/ep93xx-restart.c | 65 + drivers/pwm/pwm-ep93xx.c | 24 +- drivers/rtc/rtc-ep93xx.c | 8 + drivers/rtc/rtc-m48t86.c | 10 + drivers/soc/Kconfig | 1 + drivers/soc/Makefile | 1 + drivers/soc/cirrus/Kconfig | 11 + drivers/soc/cirrus/Makefile | 2 + drivers/soc/cirrus/soc-ep93xx.c | 134 ++ drivers/spi/spi-ep93xx.c | 31 +- drivers/watchdog/ep93xx_wdt.c | 8 + drivers/watchdog/ts72xx_wdt.c | 8 + .../dt-bindings/clock/cirrus,ep93xx-clock.h | 53 + include/linux/platform_data/dma-ep93xx.h | 3 + include/linux/soc/cirrus/ep93xx.h | 28 +- sound/soc/cirrus/Kconfig | 9 - sound/soc/cirrus/Makefile | 4 - sound/soc/cirrus/edb93xx.c | 119 -- 73 files changed, 4796 insertions(+), 3453 deletions(-) create mode 100644 Documentation/devicetree/bindings/arm/ep93xx.yaml create mode 100644 Documentation/devicetree/bindings/ata/cirrus,ep93xx-pata.yaml create mode 100644 Documentation/devicetree/bindings/dma/cirrus,ep93xx-dma-m2m.yaml create mode 100644 Documentation/devicetree/bindings/dma/cirrus,ep93xx-dma-m2p.yaml create mode 100644 Documentation/devicetree/bindings/gpio/gpio-ep93xx.yaml create mode 100644 Documentation/devicetree/bindings/input/cirrus,ep93xx-keypad.yaml create mode 100644 Documentation/devicetree/bindings/mtd/technologic,nand.yaml create mode 100644 Documentation/devicetree/bindings/net/cirrus,ep93xx_eth.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/cirrus,ep93xx-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pwm/cirrus,ep93xx-pwm.yaml create mode 100644 Documentation/devicetree/bindings/rtc/cirrus,ep93xx-rtc.yaml create mode 100644 Documentation/devicetree/bindings/rtc/dallas,rtc-m48t86.yaml create mode 100644 Documentation/devicetree/bindings/spi/spi-ep93xx.yaml create mode 100644 Documentation/devicetree/bindings/timer/cirrus,ep93xx-timer.yaml create mode 100644 Documentation/devicetree/bindings/watchdog/cirrus,ep93xx-wdt.yaml create mode 100644 Documentation/devicetree/bindings/watchdog/technologic,ts72xx-wdt.yaml create mode 100644 arch/arm/boot/dts/ep93xx-bk3.dts create mode 100644 arch/arm/boot/dts/ep93xx-edb9302.dts create mode 100644 arch/arm/boot/dts/ep93xx-ts7250.dts create mode 100644 arch/arm/boot/dts/ep93xx.dtsi delete mode 100644 arch/arm/mach-ep93xx/Makefile delete mode 100644 arch/arm/mach-ep93xx/core.c delete mode 100644 arch/arm/mach-ep93xx/dma.c delete mode 100644 arch/arm/mach-ep93xx/edb93xx.c delete mode 100644 arch/arm/mach-ep93xx/ep93xx-regs.h delete mode 100644 arch/arm/mach-ep93xx/gpio-ep93xx.h delete mode 100644 arch/arm/mach-ep93xx/hardware.h delete mode 100644 arch/arm/mach-ep93xx/irqs.h delete mode 100644 arch/arm/mach-ep93xx/platform.h delete mode 100644 arch/arm/mach-ep93xx/soc.h delete mode 100644 arch/arm/mach-ep93xx/ts72xx.c delete mode 100644 arch/arm/mach-ep93xx/ts72xx.h delete mode 100644 arch/arm/mach-ep93xx/vision_ep9307.c rename arch/arm/mach-ep93xx/clock.c => drivers/clk/clk-ep93xx.c (60%) rename {arch/arm/mach-ep93xx => drivers/clocksource}/timer-ep93xx.c (51%) create mode 100644 drivers/mtd/nand/raw/ts72xx_nand.c create mode 100644 drivers/pinctrl/pinctrl-ep93xx.c create mode 100644 drivers/power/reset/ep93xx-restart.c create mode 100644 drivers/soc/cirrus/Kconfig create mode 100644 drivers/soc/cirrus/Makefile create mode 100644 drivers/soc/cirrus/soc-ep93xx.c create mode 100644 include/dt-bindings/clock/cirrus,ep93xx-clock.h delete mode 100644 sound/soc/cirrus/edb93xx.c -- 2.39.2

7 11

[PATCH v2 0/2] ASoC: SOF: add mt8188 audio support
by Trevor Wu 01 Jun '23

01 Jun '23

This series adds mt8188 audio support and dbg_dump callback for mt8186 and mt8188. Changes since v1: - fix typo - adopt reviewer's suggestion Trevor Wu (2): ASoC: SOF: mediatek: add mt8188 audio support ASoC: SOF: mediatek: add adsp debug dump sound/soc/sof/mediatek/mt8186/mt8186.c | 83 +++++++++++++++++++++++++- 1 file changed, 82 insertions(+), 1 deletion(-) -- 2.18.0

3 5

[PATCH 0/2] ASoC: mediatek: fix use-after-free in driver remove path
by Trevor Wu 01 Jun '23

01 Jun '23

These patches concern modifications made in mt8186[1]. The clock unregistration mechanism used in mt8188 and mt8195 is similar with mt8186, resulting in the same problem existing within the driver. Therefore, the solution has also been applied to these two platforms. [1] https://lore.kernel.org/all/20230511092437.1.I31cceffc8c45bb1af16eb613e197b… Trevor Wu (2): ASoC: mediatek: mt8188: fix use-after-free in driver remove path ASoC: mediatek: mt8195: fix use-after-free in driver remove path sound/soc/mediatek/mt8188/mt8188-afe-clk.c | 7 --- sound/soc/mediatek/mt8188/mt8188-afe-clk.h | 1 - sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 4 -- sound/soc/mediatek/mt8188/mt8188-audsys-clk.c | 47 ++++++++++--------- sound/soc/mediatek/mt8188/mt8188-audsys-clk.h | 1 - sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 5 -- sound/soc/mediatek/mt8195/mt8195-afe-clk.h | 1 - sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 4 -- sound/soc/mediatek/mt8195/mt8195-audsys-clk.c | 47 ++++++++++--------- sound/soc/mediatek/mt8195/mt8195-audsys-clk.h | 1 - 10 files changed, 48 insertions(+), 70 deletions(-) -- 2.18.0

3 6

[PATCH 0/2] ASoC: SOF: add mt8188 audio support
by Trevor Wu 01 Jun '23

01 Jun '23

This series adds mt8188 audio support and dbg_dump callback for mt8186 and mt8188. Trevor Wu (2): ASoC: SOF: mediatek: add mt8188 audio support ASoC: SOF: mediatek: add adsp debug dump sound/soc/sof/mediatek/mt8186/mt8186.c | 83 +++++++++++++++++++++++++- sound/soc/sof/mediatek/mt8186/mt8186.h | 5 ++ 2 files changed, 87 insertions(+), 1 deletion(-) -- 2.18.0

5 7

Linux 6.4-rc4 - audio doesn't work
by Shuah Khan 01 Jun '23

01 Jun '23

All, I updated my laptop to Linux 6.4-rc4 and audio stopped working. I haven't done any debugging yet since this is my primary system. Here are the audio hardware details: lspci | grep Audio 05:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Renoir Radeon High Definition Audio Controller 05:00.5 Multimedia controller: Advanced Micro Devices, Inc. [AMD] ACP/ACP3X/ACP6x Audio Coprocessor (rev 01) 05:00.6 Audio device: Advanced Micro Devices, Inc. [AMD] Family 17h/19h HD Audio Controller I see just the Dummy Output device in sound settings. I will start poking around later on today. thanks, -- Shuah

1 0

[PATCH 0/5] ASoC: minor cleanup for soc_get_playback_capture()
by Kuninori Morimoto 31 May '23

31 May '23

Hi Mark This is minor cleanup patches for soc_get_playback_capture(). Link: https://lore.kernel.org/r/87ttw1gqgn.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87353uqjiu.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/ab3f0c0a-62fd-a468-b3cf-0e4b59bac6ae@linux.intel.… Kuninori Morimoto (5): ASoC: soc-pcm.c: indicate error if stream has no playback no capture ASoC: soc-pcm.c: use dai_link on soc_get_playback_capture() ASoC: soc-pcm.c: cleanup soc_get_playback_capture() error ASoC: soc-pcm.c: use temporary variable at soc_get_playback_capture() ASoC: soc-pcm.c: tidyup playback/capture_only at soc_get_playback_capture() sound/soc/soc-pcm.c | 58 ++++++++++++++++++++++++++------------------- 1 file changed, 33 insertions(+), 25 deletions(-) -- 2.25.1

4 8

[PATCH] ASoC: amd: yc: Add Thinkpad Neo14 to quirks list for acp6x
by Sicong Jiang 31 May '23

31 May '23

Thinkpad Neo14 Ryzen Edition uses Ryzen 6800H processor, and adding to quirks list for acp6x will enable internal mic. Signed-off-by: Sicong Jiang <kevin.jiangsc(a)gmail.com> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 4406a5def..246299a17 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -171,6 +171,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21CL"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21EF"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.38.0.83.gd420dda057

2 1

[PATCH AUTOSEL 5.10 19/21] ASoC: dwc: move DMA init to snd_soc_dai_driver probe()
by Sasha Levin 31 May '23

31 May '23

From: Maxim Kochetkov <fido_max(a)inbox.ru> [ Upstream commit 011a8719d6105dcb48077ea7a6a88ac019d4aa50 ] When using DMA mode we are facing with Oops: [ 396.458157] Unable to handle kernel access to user memory without uaccess routines at virtual address 000000000000000c [ 396.469374] Oops [#1] [ 396.471839] Modules linked in: [ 396.475144] CPU: 0 PID: 114 Comm: arecord Not tainted 6.0.0-00164-g9a8eccdaf2be-dirty #68 [ 396.483619] Hardware name: YMP ELCT FPGA (DT) [ 396.488156] epc : dmaengine_pcm_open+0x1d2/0x342 [ 396.493227] ra : dmaengine_pcm_open+0x1d2/0x342 [ 396.498140] epc : ffffffff807fe346 ra : ffffffff807fe346 sp : ffffffc804e138f0 [ 396.505602] gp : ffffffff817bf730 tp : ffffffd8042c8ac0 t0 : 6500000000000000 [ 396.513045] t1 : 0000000000000064 t2 : 656e69676e65616d s0 : ffffffc804e13990 [ 396.520477] s1 : ffffffd801b86a18 a0 : 0000000000000026 a1 : ffffffff816920f8 [ 396.527897] a2 : 0000000000000010 a3 : fffffffffffffffe a4 : 0000000000000000 [ 396.535319] a5 : 0000000000000000 a6 : ffffffd801b87040 a7 : 0000000000000038 [ 396.542740] s2 : ffffffd801b94a00 s3 : 0000000000000000 s4 : ffffffd80427f5e8 [ 396.550153] s5 : ffffffd80427f5e8 s6 : ffffffd801b44410 s7 : fffffffffffffff5 [ 396.557569] s8 : 0000000000000800 s9 : 0000000000000001 s10: ffffffff8066d254 [ 396.564978] s11: ffffffd8059cf768 t3 : ffffffff817d5577 t4 : ffffffff817d5577 [ 396.572391] t5 : ffffffff817d5578 t6 : ffffffc804e136e8 [ 396.577876] status: 0000000200000120 badaddr: 000000000000000c cause: 000000000000000d [ 396.586007] [<ffffffff806839f4>] snd_soc_component_open+0x1a/0x68 [ 396.592439] [<ffffffff807fdd62>] __soc_pcm_open+0xf0/0x502 [ 396.598217] [<ffffffff80685d86>] soc_pcm_open+0x2e/0x4e [ 396.603741] [<ffffffff8066cea4>] snd_pcm_open_substream+0x442/0x68e [ 396.610313] [<ffffffff8066d1ea>] snd_pcm_open+0xfa/0x212 [ 396.615868] [<ffffffff8066d39c>] snd_pcm_capture_open+0x3a/0x60 [ 396.622048] [<ffffffff8065b35a>] snd_open+0xa8/0x17a [ 396.627421] [<ffffffff801ae036>] chrdev_open+0xa0/0x218 [ 396.632893] [<ffffffff801a5a28>] do_dentry_open+0x17c/0x2a6 [ 396.638713] [<ffffffff801a6d9a>] vfs_open+0x1e/0x26 [ 396.643850] [<ffffffff801b8544>] path_openat+0x96e/0xc96 [ 396.649518] [<ffffffff801b9390>] do_filp_open+0x7c/0xf6 [ 396.655034] [<ffffffff801a6ff2>] do_sys_openat2+0x8a/0x11e [ 396.660765] [<ffffffff801a735a>] sys_openat+0x50/0x7c [ 396.666068] [<ffffffff80003aca>] ret_from_syscall+0x0/0x2 [ 396.674964] ---[ end trace 0000000000000000 ]--- It happens because of play_dma_data/capture_dma_data pointers are NULL. Current implementation assigns these pointers at snd_soc_dai_driver startup() callback and reset them back to NULL at shutdown(). But soc_pcm_open() sequence uses DMA pointers in dmaengine_pcm_open() before snd_soc_dai_driver startup(). Most generic DMA capable I2S drivers use snd_soc_dai_driver probe() callback to init DMA pointers only once at probe. So move DMA init to dw_i2s_dai_probe and drop shutdown() and startup() callbacks. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> Link: https://lore.kernel.org/r/20230512110343.66664-1-fido_max@inbox.ru Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/dwc/dwc-i2s.c | 41 +++++++++-------------------------------- 1 file changed, 9 insertions(+), 32 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 36da0f01571a1..3e14fac6666e9 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -183,30 +183,6 @@ static void i2s_stop(struct dw_i2s_dev *dev, } } -static int dw_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); - union dw_i2s_snd_dma_data *dma_data = NULL; - - if (!(dev->capability & DWC_I2S_RECORD) && - (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) - return -EINVAL; - - if (!(dev->capability & DWC_I2S_PLAY) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) - return -EINVAL; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &dev->play_dma_data; - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - dma_data = &dev->capture_dma_data; - - snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)dma_data); - - return 0; -} - static void dw_i2s_config(struct dw_i2s_dev *dev, int stream) { u32 ch_reg; @@ -305,12 +281,6 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static void dw_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - snd_soc_dai_set_dma_data(dai, substream, NULL); -} - static int dw_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -382,8 +352,6 @@ static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) } static const struct snd_soc_dai_ops dw_i2s_dai_ops = { - .startup = dw_i2s_startup, - .shutdown = dw_i2s_shutdown, .hw_params = dw_i2s_hw_params, .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, @@ -624,6 +592,14 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, } +static int dw_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &dev->play_dma_data, &dev->capture_dma_data); + return 0; +} + static int dw_i2s_probe(struct platform_device *pdev) { const struct i2s_platform_data *pdata = pdev->dev.platform_data; @@ -642,6 +618,7 @@ static int dw_i2s_probe(struct platform_device *pdev) return -ENOMEM; dw_i2s_dai->ops = &dw_i2s_dai_ops; + dw_i2s_dai->probe = dw_i2s_dai_probe; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); dev->i2s_base = devm_ioremap_resource(&pdev->dev, res); -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 21/24] ASoC: dwc: move DMA init to snd_soc_dai_driver probe()
by Sasha Levin 31 May '23

31 May '23

From: Maxim Kochetkov <fido_max(a)inbox.ru> [ Upstream commit 011a8719d6105dcb48077ea7a6a88ac019d4aa50 ] When using DMA mode we are facing with Oops: [ 396.458157] Unable to handle kernel access to user memory without uaccess routines at virtual address 000000000000000c [ 396.469374] Oops [#1] [ 396.471839] Modules linked in: [ 396.475144] CPU: 0 PID: 114 Comm: arecord Not tainted 6.0.0-00164-g9a8eccdaf2be-dirty #68 [ 396.483619] Hardware name: YMP ELCT FPGA (DT) [ 396.488156] epc : dmaengine_pcm_open+0x1d2/0x342 [ 396.493227] ra : dmaengine_pcm_open+0x1d2/0x342 [ 396.498140] epc : ffffffff807fe346 ra : ffffffff807fe346 sp : ffffffc804e138f0 [ 396.505602] gp : ffffffff817bf730 tp : ffffffd8042c8ac0 t0 : 6500000000000000 [ 396.513045] t1 : 0000000000000064 t2 : 656e69676e65616d s0 : ffffffc804e13990 [ 396.520477] s1 : ffffffd801b86a18 a0 : 0000000000000026 a1 : ffffffff816920f8 [ 396.527897] a2 : 0000000000000010 a3 : fffffffffffffffe a4 : 0000000000000000 [ 396.535319] a5 : 0000000000000000 a6 : ffffffd801b87040 a7 : 0000000000000038 [ 396.542740] s2 : ffffffd801b94a00 s3 : 0000000000000000 s4 : ffffffd80427f5e8 [ 396.550153] s5 : ffffffd80427f5e8 s6 : ffffffd801b44410 s7 : fffffffffffffff5 [ 396.557569] s8 : 0000000000000800 s9 : 0000000000000001 s10: ffffffff8066d254 [ 396.564978] s11: ffffffd8059cf768 t3 : ffffffff817d5577 t4 : ffffffff817d5577 [ 396.572391] t5 : ffffffff817d5578 t6 : ffffffc804e136e8 [ 396.577876] status: 0000000200000120 badaddr: 000000000000000c cause: 000000000000000d [ 396.586007] [<ffffffff806839f4>] snd_soc_component_open+0x1a/0x68 [ 396.592439] [<ffffffff807fdd62>] __soc_pcm_open+0xf0/0x502 [ 396.598217] [<ffffffff80685d86>] soc_pcm_open+0x2e/0x4e [ 396.603741] [<ffffffff8066cea4>] snd_pcm_open_substream+0x442/0x68e [ 396.610313] [<ffffffff8066d1ea>] snd_pcm_open+0xfa/0x212 [ 396.615868] [<ffffffff8066d39c>] snd_pcm_capture_open+0x3a/0x60 [ 396.622048] [<ffffffff8065b35a>] snd_open+0xa8/0x17a [ 396.627421] [<ffffffff801ae036>] chrdev_open+0xa0/0x218 [ 396.632893] [<ffffffff801a5a28>] do_dentry_open+0x17c/0x2a6 [ 396.638713] [<ffffffff801a6d9a>] vfs_open+0x1e/0x26 [ 396.643850] [<ffffffff801b8544>] path_openat+0x96e/0xc96 [ 396.649518] [<ffffffff801b9390>] do_filp_open+0x7c/0xf6 [ 396.655034] [<ffffffff801a6ff2>] do_sys_openat2+0x8a/0x11e [ 396.660765] [<ffffffff801a735a>] sys_openat+0x50/0x7c [ 396.666068] [<ffffffff80003aca>] ret_from_syscall+0x0/0x2 [ 396.674964] ---[ end trace 0000000000000000 ]--- It happens because of play_dma_data/capture_dma_data pointers are NULL. Current implementation assigns these pointers at snd_soc_dai_driver startup() callback and reset them back to NULL at shutdown(). But soc_pcm_open() sequence uses DMA pointers in dmaengine_pcm_open() before snd_soc_dai_driver startup(). Most generic DMA capable I2S drivers use snd_soc_dai_driver probe() callback to init DMA pointers only once at probe. So move DMA init to dw_i2s_dai_probe and drop shutdown() and startup() callbacks. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> Link: https://lore.kernel.org/r/20230512110343.66664-1-fido_max@inbox.ru Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/dwc/dwc-i2s.c | 41 +++++++++-------------------------------- 1 file changed, 9 insertions(+), 32 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 315ca5c4b057a..0dddd83f099c9 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -183,30 +183,6 @@ static void i2s_stop(struct dw_i2s_dev *dev, } } -static int dw_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); - union dw_i2s_snd_dma_data *dma_data = NULL; - - if (!(dev->capability & DWC_I2S_RECORD) && - (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) - return -EINVAL; - - if (!(dev->capability & DWC_I2S_PLAY) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) - return -EINVAL; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &dev->play_dma_data; - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - dma_data = &dev->capture_dma_data; - - snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)dma_data); - - return 0; -} - static void dw_i2s_config(struct dw_i2s_dev *dev, int stream) { u32 ch_reg; @@ -305,12 +281,6 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static void dw_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - snd_soc_dai_set_dma_data(dai, substream, NULL); -} - static int dw_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -382,8 +352,6 @@ static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) } static const struct snd_soc_dai_ops dw_i2s_dai_ops = { - .startup = dw_i2s_startup, - .shutdown = dw_i2s_shutdown, .hw_params = dw_i2s_hw_params, .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, @@ -624,6 +592,14 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, } +static int dw_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &dev->play_dma_data, &dev->capture_dma_data); + return 0; +} + static int dw_i2s_probe(struct platform_device *pdev) { const struct i2s_platform_data *pdata = pdev->dev.platform_data; @@ -642,6 +618,7 @@ static int dw_i2s_probe(struct platform_device *pdev) return -ENOMEM; dw_i2s_dai->ops = &dw_i2s_dai_ops; + dw_i2s_dai->probe = dw_i2s_dai_probe; dev->i2s_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(dev->i2s_base)) -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 14/24] ASoC: soc-pcm: test if a BE can be prepared
by Sasha Levin 31 May '23

31 May '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> [ Upstream commit e123036be377ddf628226a7c6d4f9af5efd113d3 ] In the BE hw_params configuration, the existing code checks if any of the existing FEs are prepared, running, paused or suspended - and skips the configuration in those cases. This allows multiple calls of hw_params which the ALSA state machine supports. This check is not handled for the prepare stage, which can lead to the same BE being prepared multiple times. This patch adds a check similar to that of the hw_params, with the main difference being that the suspended state is allowed: the ALSA state machine allows a transition from suspended to prepared with hw_params skipped. This problem was detected on Intel IPC4/SoundWire devices, where the BE dailink .prepare stage is used to configure the SoundWire stream with a bank switch. Multiple .prepare calls lead to conflicts with the .trigger operation with IPC4 configurations. This problem was not detected earlier on Intel devices, HDaudio BE dailinks detect that the link is already prepared and skip the configuration, and for IPC3 devices there is no BE trigger. Link: https://github.com/thesofproject/sof/issues/7596 Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Link: https://lore.kernel.org/r/20230517185731.487124-1-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc-dpcm.h | 4 ++++ sound/soc/soc-pcm.c | 20 ++++++++++++++++++++ 2 files changed, 24 insertions(+) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index d963f3b608489..4bb3ebfdaa45e 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -123,6 +123,10 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream); +/* can this BE perform prepare */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + /* is the current PCM operation for this FE ? */ int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 76c807372ef50..6ec248778e2fc 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2398,6 +2398,9 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) if (!snd_soc_dpcm_be_can_update(fe, be, stream)) continue; + if (!snd_soc_dpcm_can_be_prepared(fe, be, stream)) + continue; + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) && @@ -3047,3 +3050,20 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); + +/* + * We can only prepare a BE DAI if any of it's FE are not prepared, + * running or paused for the specified stream direction. + */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_PREPARE, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared); -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 11/24] ASoC: lpass: Fix for KASAN use_after_free out of bounds
by Sasha Levin 31 May '23

31 May '23

From: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com> [ Upstream commit 75e5fab7db0cecb6e16b22c34608f0b40a4c7cd1 ] When we run syzkaller we get below Out of Bounds error. "KASAN: slab-out-of-bounds Read in regcache_flat_read" Below is the backtrace of the issue: BUG: KASAN: slab-out-of-bounds in regcache_flat_read+0x10c/0x110 Read of size 4 at addr ffffff8088fbf714 by task syz-executor.4/14144 CPU: 6 PID: 14144 Comm: syz-executor.4 Tainted: G W Hardware name: Qualcomm Technologies, Inc. sc7280 CRD platform (rev5+) (DT) Call trace: dump_backtrace+0x0/0x4ec show_stack+0x34/0x50 dump_stack_lvl+0xdc/0x11c print_address_description+0x30/0x2d8 kasan_report+0x178/0x1e4 __asan_report_load4_noabort+0x44/0x50 regcache_flat_read+0x10c/0x110 regcache_read+0xf8/0x5a0 _regmap_read+0x45c/0x86c _regmap_update_bits+0x128/0x290 regmap_update_bits_base+0xc0/0x15c snd_soc_component_update_bits+0xa8/0x22c snd_soc_component_write_field+0x68/0xd4 tx_macro_put_dec_enum+0x1d0/0x268 snd_ctl_elem_write+0x288/0x474 By Error checking and checking valid values issue gets rectifies. Signed-off-by: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com Link: https://lore.kernel.org/r/20230511112532.16106-1-quic_visr@quicinc.com Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/lpass-tx-macro.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 35d148e60c334..4377e91733b87 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -743,6 +743,8 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, struct tx_macro *tx = snd_soc_component_get_drvdata(component); val = ucontrol->value.enumerated.item[0]; + if (val >= e->items) + return -EINVAL; switch (e->reg) { case CDC_TX_INP_MUX_ADC_MUX0_CFG0: @@ -769,6 +771,9 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, case CDC_TX_INP_MUX_ADC_MUX7_CFG0: mic_sel_reg = CDC_TX7_TX_PATH_CFG0; break; + default: + dev_err(component->dev, "Error in configuration!!\n"); + return -EINVAL; } if (val != 0) { -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 29/33] ASoC: dwc: move DMA init to snd_soc_dai_driver probe()
by Sasha Levin 31 May '23

31 May '23

From: Maxim Kochetkov <fido_max(a)inbox.ru> [ Upstream commit 011a8719d6105dcb48077ea7a6a88ac019d4aa50 ] When using DMA mode we are facing with Oops: [ 396.458157] Unable to handle kernel access to user memory without uaccess routines at virtual address 000000000000000c [ 396.469374] Oops [#1] [ 396.471839] Modules linked in: [ 396.475144] CPU: 0 PID: 114 Comm: arecord Not tainted 6.0.0-00164-g9a8eccdaf2be-dirty #68 [ 396.483619] Hardware name: YMP ELCT FPGA (DT) [ 396.488156] epc : dmaengine_pcm_open+0x1d2/0x342 [ 396.493227] ra : dmaengine_pcm_open+0x1d2/0x342 [ 396.498140] epc : ffffffff807fe346 ra : ffffffff807fe346 sp : ffffffc804e138f0 [ 396.505602] gp : ffffffff817bf730 tp : ffffffd8042c8ac0 t0 : 6500000000000000 [ 396.513045] t1 : 0000000000000064 t2 : 656e69676e65616d s0 : ffffffc804e13990 [ 396.520477] s1 : ffffffd801b86a18 a0 : 0000000000000026 a1 : ffffffff816920f8 [ 396.527897] a2 : 0000000000000010 a3 : fffffffffffffffe a4 : 0000000000000000 [ 396.535319] a5 : 0000000000000000 a6 : ffffffd801b87040 a7 : 0000000000000038 [ 396.542740] s2 : ffffffd801b94a00 s3 : 0000000000000000 s4 : ffffffd80427f5e8 [ 396.550153] s5 : ffffffd80427f5e8 s6 : ffffffd801b44410 s7 : fffffffffffffff5 [ 396.557569] s8 : 0000000000000800 s9 : 0000000000000001 s10: ffffffff8066d254 [ 396.564978] s11: ffffffd8059cf768 t3 : ffffffff817d5577 t4 : ffffffff817d5577 [ 396.572391] t5 : ffffffff817d5578 t6 : ffffffc804e136e8 [ 396.577876] status: 0000000200000120 badaddr: 000000000000000c cause: 000000000000000d [ 396.586007] [<ffffffff806839f4>] snd_soc_component_open+0x1a/0x68 [ 396.592439] [<ffffffff807fdd62>] __soc_pcm_open+0xf0/0x502 [ 396.598217] [<ffffffff80685d86>] soc_pcm_open+0x2e/0x4e [ 396.603741] [<ffffffff8066cea4>] snd_pcm_open_substream+0x442/0x68e [ 396.610313] [<ffffffff8066d1ea>] snd_pcm_open+0xfa/0x212 [ 396.615868] [<ffffffff8066d39c>] snd_pcm_capture_open+0x3a/0x60 [ 396.622048] [<ffffffff8065b35a>] snd_open+0xa8/0x17a [ 396.627421] [<ffffffff801ae036>] chrdev_open+0xa0/0x218 [ 396.632893] [<ffffffff801a5a28>] do_dentry_open+0x17c/0x2a6 [ 396.638713] [<ffffffff801a6d9a>] vfs_open+0x1e/0x26 [ 396.643850] [<ffffffff801b8544>] path_openat+0x96e/0xc96 [ 396.649518] [<ffffffff801b9390>] do_filp_open+0x7c/0xf6 [ 396.655034] [<ffffffff801a6ff2>] do_sys_openat2+0x8a/0x11e [ 396.660765] [<ffffffff801a735a>] sys_openat+0x50/0x7c [ 396.666068] [<ffffffff80003aca>] ret_from_syscall+0x0/0x2 [ 396.674964] ---[ end trace 0000000000000000 ]--- It happens because of play_dma_data/capture_dma_data pointers are NULL. Current implementation assigns these pointers at snd_soc_dai_driver startup() callback and reset them back to NULL at shutdown(). But soc_pcm_open() sequence uses DMA pointers in dmaengine_pcm_open() before snd_soc_dai_driver startup(). Most generic DMA capable I2S drivers use snd_soc_dai_driver probe() callback to init DMA pointers only once at probe. So move DMA init to dw_i2s_dai_probe and drop shutdown() and startup() callbacks. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> Link: https://lore.kernel.org/r/20230512110343.66664-1-fido_max@inbox.ru Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/dwc/dwc-i2s.c | 41 +++++++++-------------------------------- 1 file changed, 9 insertions(+), 32 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 7f7dd07c63b2f..d2dd865e5ad89 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -183,30 +183,6 @@ static void i2s_stop(struct dw_i2s_dev *dev, } } -static int dw_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); - union dw_i2s_snd_dma_data *dma_data = NULL; - - if (!(dev->capability & DWC_I2S_RECORD) && - (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) - return -EINVAL; - - if (!(dev->capability & DWC_I2S_PLAY) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) - return -EINVAL; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &dev->play_dma_data; - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - dma_data = &dev->capture_dma_data; - - snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)dma_data); - - return 0; -} - static void dw_i2s_config(struct dw_i2s_dev *dev, int stream) { u32 ch_reg; @@ -305,12 +281,6 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static void dw_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - snd_soc_dai_set_dma_data(dai, substream, NULL); -} - static int dw_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -382,8 +352,6 @@ static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) } static const struct snd_soc_dai_ops dw_i2s_dai_ops = { - .startup = dw_i2s_startup, - .shutdown = dw_i2s_shutdown, .hw_params = dw_i2s_hw_params, .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, @@ -625,6 +593,14 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, } +static int dw_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &dev->play_dma_data, &dev->capture_dma_data); + return 0; +} + static int dw_i2s_probe(struct platform_device *pdev) { const struct i2s_platform_data *pdata = pdev->dev.platform_data; @@ -643,6 +619,7 @@ static int dw_i2s_probe(struct platform_device *pdev) return -ENOMEM; dw_i2s_dai->ops = &dw_i2s_dai_ops; + dw_i2s_dai->probe = dw_i2s_dai_probe; dev->i2s_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(dev->i2s_base)) -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 28/33] ASoC: cs35l41: Fix default regmap values for some registers
by Sasha Levin 31 May '23

31 May '23

From: Stefan Binding <sbinding(a)opensource.cirrus.com> [ Upstream commit e2d035f5a7d597bbabc268e236ec6c0408c4af0e ] Several values do not match the defaults of CS35L41, fix them. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Acked-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20230414152552.574502-4-sbinding@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l41-lib.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs35l41-lib.c b/sound/soc/codecs/cs35l41-lib.c index 04be71435491e..c2c56e5608094 100644 --- a/sound/soc/codecs/cs35l41-lib.c +++ b/sound/soc/codecs/cs35l41-lib.c @@ -46,7 +46,7 @@ static const struct reg_default cs35l41_reg[] = { { CS35L41_DSP1_RX5_SRC, 0x00000020 }, { CS35L41_DSP1_RX6_SRC, 0x00000021 }, { CS35L41_DSP1_RX7_SRC, 0x0000003A }, - { CS35L41_DSP1_RX8_SRC, 0x00000001 }, + { CS35L41_DSP1_RX8_SRC, 0x0000003B }, { CS35L41_NGATE1_SRC, 0x00000008 }, { CS35L41_NGATE2_SRC, 0x00000009 }, { CS35L41_AMP_DIG_VOL_CTRL, 0x00008000 }, @@ -58,8 +58,8 @@ static const struct reg_default cs35l41_reg[] = { { CS35L41_IRQ1_MASK2, 0xFFFFFFFF }, { CS35L41_IRQ1_MASK3, 0xFFFF87FF }, { CS35L41_IRQ1_MASK4, 0xFEFFFFFF }, - { CS35L41_GPIO1_CTRL1, 0xE1000001 }, - { CS35L41_GPIO2_CTRL1, 0xE1000001 }, + { CS35L41_GPIO1_CTRL1, 0x81000001 }, + { CS35L41_GPIO2_CTRL1, 0x81000001 }, { CS35L41_MIXER_NGATE_CFG, 0x00000000 }, { CS35L41_MIXER_NGATE_CH1_CFG, 0x00000303 }, { CS35L41_MIXER_NGATE_CH2_CFG, 0x00000303 }, -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 21/33] ASoC: Intel: avs: Add missing checks on FE startup
by Sasha Levin 31 May '23

31 May '23

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 25148f57a2a6d157779bae494852e172952ba980 ] Constraint functions have return values, they should be checked for potential errors. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20230519201711.4073845-8-amadeuszx.slawinski@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/pcm.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 8fe5917b1e263..22f21f3f612d1 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -424,21 +424,34 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so host_stream = snd_hdac_ext_stream_assign(bus, substream, HDAC_EXT_STREAM_TYPE_HOST); if (!host_stream) { - kfree(data); - return -EBUSY; + ret = -EBUSY; + goto err; } data->host_stream = host_stream; - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto err; + /* avoid wrap-around with wall-clock */ - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, 20, 178000000); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates); + ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, 20, 178000000); + if (ret < 0) + goto err; + + ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates); + if (ret < 0) + goto err; + snd_pcm_set_sync(substream); dev_dbg(dai->dev, "%s fe STARTUP tag %d str %p", __func__, hdac_stream(host_stream)->stream_tag, substream); return 0; + +err: + kfree(data); + return ret; } static void avs_dai_fe_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 18/33] ASoC: soc-pcm: test if a BE can be prepared
by Sasha Levin 31 May '23

31 May '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> [ Upstream commit e123036be377ddf628226a7c6d4f9af5efd113d3 ] In the BE hw_params configuration, the existing code checks if any of the existing FEs are prepared, running, paused or suspended - and skips the configuration in those cases. This allows multiple calls of hw_params which the ALSA state machine supports. This check is not handled for the prepare stage, which can lead to the same BE being prepared multiple times. This patch adds a check similar to that of the hw_params, with the main difference being that the suspended state is allowed: the ALSA state machine allows a transition from suspended to prepared with hw_params skipped. This problem was detected on Intel IPC4/SoundWire devices, where the BE dailink .prepare stage is used to configure the SoundWire stream with a bank switch. Multiple .prepare calls lead to conflicts with the .trigger operation with IPC4 configurations. This problem was not detected earlier on Intel devices, HDaudio BE dailinks detect that the link is already prepared and skip the configuration, and for IPC3 devices there is no BE trigger. Link: https://github.com/thesofproject/sof/issues/7596 Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Link: https://lore.kernel.org/r/20230517185731.487124-1-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc-dpcm.h | 4 ++++ sound/soc/soc-pcm.c | 20 ++++++++++++++++++++ 2 files changed, 24 insertions(+) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 5b689c663290f..27a5642f07cda 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -125,6 +125,10 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream); +/* can this BE perform prepare */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + /* is the current PCM operation for this FE ? */ int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7a486ca9633c1..f3964060a0447 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2403,6 +2403,9 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) if (!snd_soc_dpcm_be_can_update(fe, be, stream)) continue; + if (!snd_soc_dpcm_can_be_prepared(fe, be, stream)) + continue; + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) && @@ -3043,3 +3046,20 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); + +/* + * We can only prepare a BE DAI if any of it's FE are not prepared, + * running or paused for the specified stream direction. + */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_PREPARE, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared); -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 14/33] ASoC: lpass: Fix for KASAN use_after_free out of bounds
by Sasha Levin 31 May '23

31 May '23

From: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com> [ Upstream commit 75e5fab7db0cecb6e16b22c34608f0b40a4c7cd1 ] When we run syzkaller we get below Out of Bounds error. "KASAN: slab-out-of-bounds Read in regcache_flat_read" Below is the backtrace of the issue: BUG: KASAN: slab-out-of-bounds in regcache_flat_read+0x10c/0x110 Read of size 4 at addr ffffff8088fbf714 by task syz-executor.4/14144 CPU: 6 PID: 14144 Comm: syz-executor.4 Tainted: G W Hardware name: Qualcomm Technologies, Inc. sc7280 CRD platform (rev5+) (DT) Call trace: dump_backtrace+0x0/0x4ec show_stack+0x34/0x50 dump_stack_lvl+0xdc/0x11c print_address_description+0x30/0x2d8 kasan_report+0x178/0x1e4 __asan_report_load4_noabort+0x44/0x50 regcache_flat_read+0x10c/0x110 regcache_read+0xf8/0x5a0 _regmap_read+0x45c/0x86c _regmap_update_bits+0x128/0x290 regmap_update_bits_base+0xc0/0x15c snd_soc_component_update_bits+0xa8/0x22c snd_soc_component_write_field+0x68/0xd4 tx_macro_put_dec_enum+0x1d0/0x268 snd_ctl_elem_write+0x288/0x474 By Error checking and checking valid values issue gets rectifies. Signed-off-by: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com Link: https://lore.kernel.org/r/20230511112532.16106-1-quic_visr@quicinc.com Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/lpass-tx-macro.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index d1d9d8d2df2d2..9f59518005a5f 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -745,6 +745,8 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, struct tx_macro *tx = snd_soc_component_get_drvdata(component); val = ucontrol->value.enumerated.item[0]; + if (val >= e->items) + return -EINVAL; switch (e->reg) { case CDC_TX_INP_MUX_ADC_MUX0_CFG0: @@ -771,6 +773,9 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, case CDC_TX_INP_MUX_ADC_MUX7_CFG0: mic_sel_reg = CDC_TX7_TX_PATH_CFG0; break; + default: + dev_err(component->dev, "Error in configuration!!\n"); + return -EINVAL; } if (val != 0) { -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 33/37] ASoC: dwc: move DMA init to snd_soc_dai_driver probe()
by Sasha Levin 31 May '23

31 May '23

From: Maxim Kochetkov <fido_max(a)inbox.ru> [ Upstream commit 011a8719d6105dcb48077ea7a6a88ac019d4aa50 ] When using DMA mode we are facing with Oops: [ 396.458157] Unable to handle kernel access to user memory without uaccess routines at virtual address 000000000000000c [ 396.469374] Oops [#1] [ 396.471839] Modules linked in: [ 396.475144] CPU: 0 PID: 114 Comm: arecord Not tainted 6.0.0-00164-g9a8eccdaf2be-dirty #68 [ 396.483619] Hardware name: YMP ELCT FPGA (DT) [ 396.488156] epc : dmaengine_pcm_open+0x1d2/0x342 [ 396.493227] ra : dmaengine_pcm_open+0x1d2/0x342 [ 396.498140] epc : ffffffff807fe346 ra : ffffffff807fe346 sp : ffffffc804e138f0 [ 396.505602] gp : ffffffff817bf730 tp : ffffffd8042c8ac0 t0 : 6500000000000000 [ 396.513045] t1 : 0000000000000064 t2 : 656e69676e65616d s0 : ffffffc804e13990 [ 396.520477] s1 : ffffffd801b86a18 a0 : 0000000000000026 a1 : ffffffff816920f8 [ 396.527897] a2 : 0000000000000010 a3 : fffffffffffffffe a4 : 0000000000000000 [ 396.535319] a5 : 0000000000000000 a6 : ffffffd801b87040 a7 : 0000000000000038 [ 396.542740] s2 : ffffffd801b94a00 s3 : 0000000000000000 s4 : ffffffd80427f5e8 [ 396.550153] s5 : ffffffd80427f5e8 s6 : ffffffd801b44410 s7 : fffffffffffffff5 [ 396.557569] s8 : 0000000000000800 s9 : 0000000000000001 s10: ffffffff8066d254 [ 396.564978] s11: ffffffd8059cf768 t3 : ffffffff817d5577 t4 : ffffffff817d5577 [ 396.572391] t5 : ffffffff817d5578 t6 : ffffffc804e136e8 [ 396.577876] status: 0000000200000120 badaddr: 000000000000000c cause: 000000000000000d [ 396.586007] [<ffffffff806839f4>] snd_soc_component_open+0x1a/0x68 [ 396.592439] [<ffffffff807fdd62>] __soc_pcm_open+0xf0/0x502 [ 396.598217] [<ffffffff80685d86>] soc_pcm_open+0x2e/0x4e [ 396.603741] [<ffffffff8066cea4>] snd_pcm_open_substream+0x442/0x68e [ 396.610313] [<ffffffff8066d1ea>] snd_pcm_open+0xfa/0x212 [ 396.615868] [<ffffffff8066d39c>] snd_pcm_capture_open+0x3a/0x60 [ 396.622048] [<ffffffff8065b35a>] snd_open+0xa8/0x17a [ 396.627421] [<ffffffff801ae036>] chrdev_open+0xa0/0x218 [ 396.632893] [<ffffffff801a5a28>] do_dentry_open+0x17c/0x2a6 [ 396.638713] [<ffffffff801a6d9a>] vfs_open+0x1e/0x26 [ 396.643850] [<ffffffff801b8544>] path_openat+0x96e/0xc96 [ 396.649518] [<ffffffff801b9390>] do_filp_open+0x7c/0xf6 [ 396.655034] [<ffffffff801a6ff2>] do_sys_openat2+0x8a/0x11e [ 396.660765] [<ffffffff801a735a>] sys_openat+0x50/0x7c [ 396.666068] [<ffffffff80003aca>] ret_from_syscall+0x0/0x2 [ 396.674964] ---[ end trace 0000000000000000 ]--- It happens because of play_dma_data/capture_dma_data pointers are NULL. Current implementation assigns these pointers at snd_soc_dai_driver startup() callback and reset them back to NULL at shutdown(). But soc_pcm_open() sequence uses DMA pointers in dmaengine_pcm_open() before snd_soc_dai_driver startup(). Most generic DMA capable I2S drivers use snd_soc_dai_driver probe() callback to init DMA pointers only once at probe. So move DMA init to dw_i2s_dai_probe and drop shutdown() and startup() callbacks. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> Link: https://lore.kernel.org/r/20230512110343.66664-1-fido_max@inbox.ru Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/dwc/dwc-i2s.c | 41 +++++++++-------------------------------- 1 file changed, 9 insertions(+), 32 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 7f7dd07c63b2f..d2dd865e5ad89 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -183,30 +183,6 @@ static void i2s_stop(struct dw_i2s_dev *dev, } } -static int dw_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); - union dw_i2s_snd_dma_data *dma_data = NULL; - - if (!(dev->capability & DWC_I2S_RECORD) && - (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) - return -EINVAL; - - if (!(dev->capability & DWC_I2S_PLAY) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) - return -EINVAL; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &dev->play_dma_data; - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - dma_data = &dev->capture_dma_data; - - snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)dma_data); - - return 0; -} - static void dw_i2s_config(struct dw_i2s_dev *dev, int stream) { u32 ch_reg; @@ -305,12 +281,6 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static void dw_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - snd_soc_dai_set_dma_data(dai, substream, NULL); -} - static int dw_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -382,8 +352,6 @@ static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) } static const struct snd_soc_dai_ops dw_i2s_dai_ops = { - .startup = dw_i2s_startup, - .shutdown = dw_i2s_shutdown, .hw_params = dw_i2s_hw_params, .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, @@ -625,6 +593,14 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, } +static int dw_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &dev->play_dma_data, &dev->capture_dma_data); + return 0; +} + static int dw_i2s_probe(struct platform_device *pdev) { const struct i2s_platform_data *pdata = pdev->dev.platform_data; @@ -643,6 +619,7 @@ static int dw_i2s_probe(struct platform_device *pdev) return -ENOMEM; dw_i2s_dai->ops = &dw_i2s_dai_ops; + dw_i2s_dai->probe = dw_i2s_dai_probe; dev->i2s_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res); if (IS_ERR(dev->i2s_base)) -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 32/37] ASoC: cs35l41: Fix default regmap values for some registers
by Sasha Levin 31 May '23

31 May '23

From: Stefan Binding <sbinding(a)opensource.cirrus.com> [ Upstream commit e2d035f5a7d597bbabc268e236ec6c0408c4af0e ] Several values do not match the defaults of CS35L41, fix them. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Acked-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20230414152552.574502-4-sbinding@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l41-lib.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs35l41-lib.c b/sound/soc/codecs/cs35l41-lib.c index 04be71435491e..c2c56e5608094 100644 --- a/sound/soc/codecs/cs35l41-lib.c +++ b/sound/soc/codecs/cs35l41-lib.c @@ -46,7 +46,7 @@ static const struct reg_default cs35l41_reg[] = { { CS35L41_DSP1_RX5_SRC, 0x00000020 }, { CS35L41_DSP1_RX6_SRC, 0x00000021 }, { CS35L41_DSP1_RX7_SRC, 0x0000003A }, - { CS35L41_DSP1_RX8_SRC, 0x00000001 }, + { CS35L41_DSP1_RX8_SRC, 0x0000003B }, { CS35L41_NGATE1_SRC, 0x00000008 }, { CS35L41_NGATE2_SRC, 0x00000009 }, { CS35L41_AMP_DIG_VOL_CTRL, 0x00008000 }, @@ -58,8 +58,8 @@ static const struct reg_default cs35l41_reg[] = { { CS35L41_IRQ1_MASK2, 0xFFFFFFFF }, { CS35L41_IRQ1_MASK3, 0xFFFF87FF }, { CS35L41_IRQ1_MASK4, 0xFEFFFFFF }, - { CS35L41_GPIO1_CTRL1, 0xE1000001 }, - { CS35L41_GPIO2_CTRL1, 0xE1000001 }, + { CS35L41_GPIO1_CTRL1, 0x81000001 }, + { CS35L41_GPIO2_CTRL1, 0x81000001 }, { CS35L41_MIXER_NGATE_CFG, 0x00000000 }, { CS35L41_MIXER_NGATE_CH1_CFG, 0x00000303 }, { CS35L41_MIXER_NGATE_CH2_CFG, 0x00000303 }, -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 24/37] ASoC: Intel: avs: Add missing checks on FE startup
by Sasha Levin 31 May '23

31 May '23

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 25148f57a2a6d157779bae494852e172952ba980 ] Constraint functions have return values, they should be checked for potential errors. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20230519201711.4073845-8-amadeuszx.slawinski@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/pcm.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 31c032a0f7e4b..1fbb2c2fadb55 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -468,21 +468,34 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so host_stream = snd_hdac_ext_stream_assign(bus, substream, HDAC_EXT_STREAM_TYPE_HOST); if (!host_stream) { - kfree(data); - return -EBUSY; + ret = -EBUSY; + goto err; } data->host_stream = host_stream; - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto err; + /* avoid wrap-around with wall-clock */ - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, 20, 178000000); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates); + ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, 20, 178000000); + if (ret < 0) + goto err; + + ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates); + if (ret < 0) + goto err; + snd_pcm_set_sync(substream); dev_dbg(dai->dev, "%s fe STARTUP tag %d str %p", __func__, hdac_stream(host_stream)->stream_tag, substream); return 0; + +err: + kfree(data); + return ret; } static void avs_dai_fe_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 23/37] ASoC: Intel: avs: Fix avs_path_module::instance_id size
by Sasha Levin 31 May '23

31 May '23

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 320f4d868b83a804e3a4bd61a5b7d0f1db66380e ] All IPCs using instance_id use 8 bit value. Original commit used 16 bit value because FW reports possible max value in 16 bit field, but in practice FW limits the value to 8 bits. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20230519201711.4073845-7-amadeuszx.slawinski@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/avs.h | 4 ++-- sound/soc/intel/avs/dsp.c | 4 ++-- sound/soc/intel/avs/path.h | 2 +- sound/soc/intel/avs/probes.c | 2 +- 4 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index d7fccdcb9c167..0cf38c9e768e7 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -283,8 +283,8 @@ void avs_release_firmwares(struct avs_dev *adev); int avs_dsp_init_module(struct avs_dev *adev, u16 module_id, u8 ppl_instance_id, u8 core_id, u8 domain, void *param, u32 param_size, - u16 *instance_id); -void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u16 instance_id, + u8 *instance_id); +void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u8 instance_id, u8 ppl_instance_id, u8 core_id); int avs_dsp_create_pipeline(struct avs_dev *adev, u16 req_size, u8 priority, bool lp, u16 attributes, u8 *instance_id); diff --git a/sound/soc/intel/avs/dsp.c b/sound/soc/intel/avs/dsp.c index b881100d3e02a..aa03af4473e94 100644 --- a/sound/soc/intel/avs/dsp.c +++ b/sound/soc/intel/avs/dsp.c @@ -225,7 +225,7 @@ static int avs_dsp_put_core(struct avs_dev *adev, u32 core_id) int avs_dsp_init_module(struct avs_dev *adev, u16 module_id, u8 ppl_instance_id, u8 core_id, u8 domain, void *param, u32 param_size, - u16 *instance_id) + u8 *instance_id) { struct avs_module_entry mentry; bool was_loaded = false; @@ -272,7 +272,7 @@ int avs_dsp_init_module(struct avs_dev *adev, u16 module_id, u8 ppl_instance_id, return ret; } -void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u16 instance_id, +void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u8 instance_id, u8 ppl_instance_id, u8 core_id) { struct avs_module_entry mentry; diff --git a/sound/soc/intel/avs/path.h b/sound/soc/intel/avs/path.h index 197222c5e008e..657f7b093e805 100644 --- a/sound/soc/intel/avs/path.h +++ b/sound/soc/intel/avs/path.h @@ -37,7 +37,7 @@ struct avs_path_pipeline { struct avs_path_module { u16 module_id; - u16 instance_id; + u8 instance_id; union avs_gtw_attributes gtw_attrs; struct avs_tplg_module *template; diff --git a/sound/soc/intel/avs/probes.c b/sound/soc/intel/avs/probes.c index 70a94201d6a56..275928281c6c6 100644 --- a/sound/soc/intel/avs/probes.c +++ b/sound/soc/intel/avs/probes.c @@ -18,7 +18,7 @@ static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id { struct avs_probe_cfg cfg = {{0}}; struct avs_module_entry mentry; - u16 dummy; + u8 dummy; avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry); -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 22/37] ASoC: Intel: avs: Account for UID of ACPI device
by Sasha Levin 31 May '23

31 May '23

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit 836855100b87b4dd7a82546131779dc255c18b67 ] Configurations with multiple codecs attached to the platform are supported but only if each from the set is different. Add new field representing the 'Unique ID' so that codecs that share Vendor and Part IDs can be differentiated and thus enabling support for such configurations. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20230519201711.4073845-6-amadeuszx.slawinski@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc-acpi.h | 1 + sound/soc/intel/avs/board_selection.c | 2 +- 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index b38fd25c57295..528279056b3ab 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -170,6 +170,7 @@ struct snd_soc_acpi_link_adr { /* Descriptor for SST ASoC machine driver */ struct snd_soc_acpi_mach { u8 id[ACPI_ID_LEN]; + const char *uid; const struct snd_soc_acpi_codecs *comp_ids; const u32 link_mask; const struct snd_soc_acpi_link_adr *links; diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c index b2823c2107f77..60f8fb0bff95b 100644 --- a/sound/soc/intel/avs/board_selection.c +++ b/sound/soc/intel/avs/board_selection.c @@ -443,7 +443,7 @@ static int avs_register_i2s_boards(struct avs_dev *adev) } for (mach = boards->machs; mach->id[0]; mach++) { - if (!acpi_dev_present(mach->id, NULL, -1)) + if (!acpi_dev_present(mach->id, mach->uid, -1)) continue; if (mach->machine_quirk) -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 19/37] ASoC: soc-pcm: test if a BE can be prepared
by Sasha Levin 31 May '23

31 May '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> [ Upstream commit e123036be377ddf628226a7c6d4f9af5efd113d3 ] In the BE hw_params configuration, the existing code checks if any of the existing FEs are prepared, running, paused or suspended - and skips the configuration in those cases. This allows multiple calls of hw_params which the ALSA state machine supports. This check is not handled for the prepare stage, which can lead to the same BE being prepared multiple times. This patch adds a check similar to that of the hw_params, with the main difference being that the suspended state is allowed: the ALSA state machine allows a transition from suspended to prepared with hw_params skipped. This problem was detected on Intel IPC4/SoundWire devices, where the BE dailink .prepare stage is used to configure the SoundWire stream with a bank switch. Multiple .prepare calls lead to conflicts with the .trigger operation with IPC4 configurations. This problem was not detected earlier on Intel devices, HDaudio BE dailinks detect that the link is already prepared and skip the configuration, and for IPC3 devices there is no BE trigger. Link: https://github.com/thesofproject/sof/issues/7596 Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Link: https://lore.kernel.org/r/20230517185731.487124-1-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc-dpcm.h | 4 ++++ sound/soc/soc-pcm.c | 20 ++++++++++++++++++++ 2 files changed, 24 insertions(+) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 1e7d09556fe3e..b7bc1865b9e4a 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -123,6 +123,10 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream); +/* can this BE perform prepare */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + /* is the current PCM operation for this FE ? */ int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7958c9defd492..1db82501fec18 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2417,6 +2417,9 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) if (!snd_soc_dpcm_be_can_update(fe, be, stream)) continue; + if (!snd_soc_dpcm_can_be_prepared(fe, be, stream)) + continue; + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) && @@ -3057,3 +3060,20 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); + +/* + * We can only prepare a BE DAI if any of it's FE are not prepared, + * running or paused for the specified stream direction. + */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_PREPARE, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared); -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 15/37] ASoC: lpass: Fix for KASAN use_after_free out of bounds
by Sasha Levin 31 May '23

31 May '23

From: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com> [ Upstream commit 75e5fab7db0cecb6e16b22c34608f0b40a4c7cd1 ] When we run syzkaller we get below Out of Bounds error. "KASAN: slab-out-of-bounds Read in regcache_flat_read" Below is the backtrace of the issue: BUG: KASAN: slab-out-of-bounds in regcache_flat_read+0x10c/0x110 Read of size 4 at addr ffffff8088fbf714 by task syz-executor.4/14144 CPU: 6 PID: 14144 Comm: syz-executor.4 Tainted: G W Hardware name: Qualcomm Technologies, Inc. sc7280 CRD platform (rev5+) (DT) Call trace: dump_backtrace+0x0/0x4ec show_stack+0x34/0x50 dump_stack_lvl+0xdc/0x11c print_address_description+0x30/0x2d8 kasan_report+0x178/0x1e4 __asan_report_load4_noabort+0x44/0x50 regcache_flat_read+0x10c/0x110 regcache_read+0xf8/0x5a0 _regmap_read+0x45c/0x86c _regmap_update_bits+0x128/0x290 regmap_update_bits_base+0xc0/0x15c snd_soc_component_update_bits+0xa8/0x22c snd_soc_component_write_field+0x68/0xd4 tx_macro_put_dec_enum+0x1d0/0x268 snd_ctl_elem_write+0x288/0x474 By Error checking and checking valid values issue gets rectifies. Signed-off-by: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com Link: https://lore.kernel.org/r/20230511112532.16106-1-quic_visr@quicinc.com Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/lpass-tx-macro.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 589c490a8c487..88d8fa58ec1eb 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -746,6 +746,8 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, struct tx_macro *tx = snd_soc_component_get_drvdata(component); val = ucontrol->value.enumerated.item[0]; + if (val >= e->items) + return -EINVAL; switch (e->reg) { case CDC_TX_INP_MUX_ADC_MUX0_CFG0: @@ -772,6 +774,9 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, case CDC_TX_INP_MUX_ADC_MUX7_CFG0: mic_sel_reg = CDC_TX7_TX_PATH_CFG0; break; + default: + dev_err(component->dev, "Error in configuration!!\n"); + return -EINVAL; } if (val != 0) { -- 2.39.2

1 0

[PATCH] ASoC: nau8825: Add the management of headset detection for power saving
by David Lin 31 May '23

31 May '23

The patch is to manage HSD feature for power saving. The detail is to disable HSD feature after the headset detection is done. When the jack is inserted, the HSD feature will be enabled again. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index cc3e18207c42..f6dd84b32e0b 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1704,6 +1704,10 @@ static void nau8825_setup_auto_irq(struct nau8825 *nau8825) { struct regmap *regmap = nau8825->regmap; + /* Enable HSD function */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, + NAU8825_HSD_AUTO_MODE, NAU8825_HSD_AUTO_MODE); + /* Enable headset jack type detection complete interruption and * jack ejection interruption. */ @@ -1955,6 +1959,9 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_MCLK_SRC_MASK, 0xf); + /* Disable HSD function */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, NAU8825_HSD_AUTO_MODE, 0); + /* Leaving HPOL/R grounded after jack insert by default. They will be * ungrounded as part of the widget power up sequence at the beginning * of playback to reduce pop. -- 2.25.1

2 1

[PATCH] dirvers/gpio: remove redundant reset gpio suffix
by Lizhe 31 May '23

31 May '23

no need to add the suffix "gpio" or "gpios" after "reset", as they will be automatically added, see of_find_gpio(). Signed-off-by: Lizhe <sensor1010(a)163.com> --- sound/soc/codecs/cs35l34.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index b3f98023e6a7..cbf0f2f4f9df 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -1061,7 +1061,7 @@ static int cs35l34_i2c_probe(struct i2c_client *i2c_client) dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); cs35l34->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, - "reset-gpios", GPIOD_OUT_LOW); + "reset", GPIOD_OUT_LOW); if (IS_ERR(cs35l34->reset_gpio)) { ret = PTR_ERR(cs35l34->reset_gpio); goto err_regulator; -- 2.34.1

2 1

[PATCH v2 0/6] Add cs42l43 PC focused SoundWire CODEC
by Charles Keepax 31 May '23

31 May '23

This patch chain adds support for the Cirrus Logic cs42l43 PC focused SoundWire CODEC. The chain is currently based of Lee's for-mfd-next branch. Change notes are included with each patch, most of the changes are trivial, the notable ones are moving the IRQs out of irqchip and into the MFD, and moving the DT binding to sound. Thanks, Charles Charles Keepax (4): dt-bindings: mfd: cirrus,cs42l43: Add initial DT binding mfd: cs42l43: Add support for cs42l43 core driver pinctrl: cs42l43: Add support for the cs42l43 ASoC: cs42l43: Add support for the cs42l43 Lucas Tanure (2): soundwire: bus: Allow SoundWire peripherals to register IRQ handlers spi: cs42l43: Add SPI controller support .../bindings/sound/cirrus,cs42l43.yaml | 320 +++ MAINTAINERS | 5 + drivers/mfd/Kconfig | 23 + drivers/mfd/Makefile | 3 + drivers/mfd/cs42l43-i2c.c | 86 + drivers/mfd/cs42l43-sdw.c | 213 ++ drivers/mfd/cs42l43.c | 1141 +++++++++ drivers/mfd/cs42l43.h | 23 + drivers/pinctrl/cirrus/Kconfig | 11 + drivers/pinctrl/cirrus/Makefile | 2 + drivers/pinctrl/cirrus/pinctrl-cs42l43.c | 609 +++++ drivers/soundwire/bus.c | 31 + drivers/soundwire/bus_type.c | 12 + drivers/spi/Kconfig | 7 + drivers/spi/Makefile | 1 + drivers/spi/spi-cs42l43.c | 279 ++ include/linux/mfd/cs42l43-regs.h | 1172 +++++++++ include/linux/mfd/cs42l43.h | 102 + include/linux/soundwire/sdw.h | 9 + include/sound/cs42l43.h | 17 + sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42l43-jack.c | 951 +++++++ sound/soc/codecs/cs42l43-sdw.c | 75 + sound/soc/codecs/cs42l43.c | 2275 +++++++++++++++++ sound/soc/codecs/cs42l43.h | 126 + 26 files changed, 7513 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml create mode 100644 drivers/mfd/cs42l43-i2c.c create mode 100644 drivers/mfd/cs42l43-sdw.c create mode 100644 drivers/mfd/cs42l43.c create mode 100644 drivers/mfd/cs42l43.h create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs42l43.c create mode 100644 drivers/spi/spi-cs42l43.c create mode 100644 include/linux/mfd/cs42l43-regs.h create mode 100644 include/linux/mfd/cs42l43.h create mode 100644 include/sound/cs42l43.h create mode 100644 sound/soc/codecs/cs42l43-jack.c create mode 100644 sound/soc/codecs/cs42l43-sdw.c create mode 100644 sound/soc/codecs/cs42l43.c create mode 100644 sound/soc/codecs/cs42l43.h -- 2.30.2

5 14

[PATCH v5 0/3] Add TDM audio on StarFive JH7110
by Walker Chen 31 May '23

31 May '23

This patchset adds TDM audio driver for the StarFive JH7110 SoC. The first patch adds device tree binding for TDM module. The second patch adds tdm driver support for JH7110 SoC. The last patch adds device tree node and pins configuration of tdm to JH7110 dts. The series has been tested on the VisionFive 2 board by plugging an audio expansion board. For more information of audio expansion board, you can take a look at the following webpage: https://wiki.seeedstudio.com/ReSpeaker_2_Mics_Pi_HAT/ The last patch should be applied after the following patch: https://lore.kernel.org/all/20230322094820.24738-5-walker.chen@starfivetech… Changes since v4: - Rebased on Linux 6.4-rc2. - Adjusted code indentation in jh7110_tdm_syncdiv(). - Removed '__maybe_unused' for jh7110_tdm_system_suspend() and jh7110_tdm_system_resume(). - Removed initialization for variable 'ret' in jh7110_tdm_hw_params(). - Return immedially when entering unsurpported case in jh7110_tdm_set_dai_fmt(). - Check the return value with IS_ERR_OR_NULL() for the call to devm_reset_control_array_get_exclusive(). - Dropped the jh7110_tdm_clk_enable() in probe and rely on runtime PM check if CONFIG_PM is enabled. - Used RUNTIME_PM_OPS() instead of SET_RUNTIME_PM_OPS() and SYSTEM_SLEEP_PM_OPS() instead of SET_SYSTEM_SLEEP_PM_OPS() in structure jh7110_tdm_pm_ops. Changes since v3: - Modified the subject of patch [1/3] to reflect which subsystem it is. - Use the set of functions with 'clk_bulk_' to get/enable/disable clocks. - Dropped the overlay from patch [3/3]. - Dropped the redundant macro 'CONFIG_PM' and 'CONFIG_PM_SLEEP' around suspend() and resume(). Changes since v2: - Use dt-overlay to describe sound card because need to plug the audio expansion board into the VisionFive2 board. - Modified the coding style for driver. - Moved assignment of stop_dma_first to startup function of dai_driver. - Dropped some useless macro definition. - Use loops to get/enable/disable clocks. Changes since v1: - Rebased on Linux 6.3-rc4. - Added the dts file dedicated to describe audio card. - Added the item for JH7110 audio board to the dt-binding of StarFive SoC-based boards. --- v4: https://lore.kernel.org/all/20230511091549.28003-1-walker.chen@starfivetech… v3: https://lore.kernel.org/all/20230506090116.9206-1-walker.chen@starfivetech.… v2: https://lore.kernel.org/all/20230420024118.22677-1-walker.chen@starfivetech… v1: https://lore.kernel.org/all/20230329153320.31390-1-walker.chen@starfivetech… Walker Chen (3): ASoC: dt-bindings: Add TDM controller bindings for StarFive JH7110 ASoC: starfive: Add JH7110 TDM driver riscv: dts: starfive: add the node and pins configuration for tdm .../bindings/sound/starfive,jh7110-tdm.yaml | 98 +++ MAINTAINERS | 6 + .../jh7110-starfive-visionfive-2.dtsi | 40 ++ arch/riscv/boot/dts/starfive/jh7110.dtsi | 21 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/starfive/Kconfig | 15 + sound/soc/starfive/Makefile | 2 + sound/soc/starfive/jh7110_tdm.c | 679 ++++++++++++++++++ 9 files changed, 863 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh7110-tdm.yaml create mode 100644 sound/soc/starfive/Kconfig create mode 100644 sound/soc/starfive/Makefile create mode 100644 sound/soc/starfive/jh7110_tdm.c base-commit: f1fcbaa18b28dec10281551dfe6ed3a3ed80e3d6 prerequisite-patch-id: 398744c61913c76a35754de867c4f820ca7a8d99 prerequisite-patch-id: 07986c6e98ce85d3ee84f38c780e72145f098fc8 prerequisite-patch-id: 1babe83d6bf999bad17584dc595480f9070a5369 prerequisite-patch-id: 63c1f70ca2bcf827429542e60d74411b7666ceff prerequisite-patch-id: 9fbb7ad1dd258bb8ff5946c4a0e59de4bfd82a04 prerequisite-patch-id: a7f61973ccb6bc425daa8a86c048fd0f5a915b76 prerequisite-patch-id: 39e1be2a3d1593577ab997f55f59367cba665aa7 prerequisite-patch-id: 0159f09bb0a1ff711a00ae17ef5b12662c9c7d3d prerequisite-patch-id: 2ddada18ab6ea5cd1da14212aaf59632f5203d40 prerequisite-patch-id: 584c256c9acb52ee2773d0c81c3f4977fc18155a prerequisite-patch-id: b37ac15032973e1fcd918f157c82a0606775c9e9 prerequisite-patch-id: 6abf359fa445f4104432ddee27044dfbfb128417 -- 2.17.1

4 10

speaker-test: allow controlling how long the noise is heard by the user
by GitHub issues - opened 30 May '23

30 May '23

alsa-project/alsa-utils issue #219 was opened from jidanni: $ speaker-test --nloops 1 makes a pink noise for about one second. On the man page mention how to make this into say a half second, as some users would like to listen to shorter test sounds, and no matter if the user does ``` $ speaker-test --device hdmi:HDMI -c 2 --nloops 1 --period 5 $ speaker-test --device hdmi:HDMI -c 2 --nloops 1 --period 50 $ speaker-test --device hdmi:HDMI -c 2 --nloops 1 --period 500 $ speaker-test --device hdmi:HDMI -c 2 --nloops 1 --period 5000 ``` what the user hears is exactly the same, therefore --period is not what controls this item. I am saying that some users can confirm their speakers work without listing to a full second of pink noise, and thus would like a way to listen to a shorter length. As far as ``` -b | --buffer TIME Use buffer size of TIME microseconds. When 0 is given, use the maximal buffer size. The default value is 0. -p | --period TIME Use period size of TIME microseconds. When 0 is given, the pe‐ riods given by -P option is used. The default value is 0. ``` are concerned, they must be documenting something completely different. Debian Package: alsa-utils Version: 1.2.8-1 Issue URL : https://github.com/alsa-project/alsa-utils/issues/219 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 0/3] ASoC: topology: allow for partial dailink stream_name match
by Pierre-Louis Bossart 30 May '23

30 May '23

The topology file and the machine driver rely on common definitions for the dailink stream_name. To avoid any backwards-compatibility problems, the machine driver stream names are set in stone and cannot be modified. This is problematic when we try to name some of the topology widgets after the stream_name, since the widget name is limited to 44 characters tools/include/uapi/sound/asound.h:#define SNDRV_CTL_ELEM_ID_NAME_MAXLEN 44 Existing examples include "Analog Playback and Capture" for HDaudio dailinks, which leaves less than 20 chars to identify widgets/controls with a meaningful name. Since the 44-char limit is part of the UAPI definitions, we assumed there is no way to increase it. This patchset suggests instead a partial match which allows topology files to use a shorter stream_name, which in turn allows for self-explanatory widget names that comply with the 44-char limit. This should not break any existing setup but with the introduction of a partial match new dailinks should be named carefully to avoid confusions between e.g. 'link1' and 'link10'. The last patch fixes such an issue in the 'nocodec' test topology used by Intel. Ranjani Sridharan (3): ASoC: topology: Allow partial matching when finding DAI link ASoC: SOF: topology: Use partial match for connecting DAI link and DAI widget ASoC: SOF: Intel: HDA: Limit the number of dai drivers for nocodec mode sound/soc/soc-topology.c | 6 +++--- sound/soc/sof/intel/hda.c | 6 +++++- sound/soc/sof/intel/hda.h | 4 +++- sound/soc/sof/topology.c | 2 +- 4 files changed, 12 insertions(+), 6 deletions(-) -- 2.37.2

2 4

[PATCH v2] SoC: ti: davinci-mcasp: Use pcm_for_each_format() macro
by Min-Hua Chen 30 May '23

30 May '23

Use pcm_for_each_format for the PCM format iteration and fix the following sparse warnings. sound/soc/ti/davinci-mcasp.c:1336:26: sparse: warning: restricted snd_pcm_format_t degrades to integer sound/soc/ti/davinci-mcasp.c:1358:26: sparse: warning: restricted snd_pcm_format_t degrades to integer sound/soc/ti/davinci-mcasp.c:1438:26: sparse: warning: restricted snd_pcm_format_t degrades to integer No functional changes. Signed-off-by: Min-Hua Chen <minhuadotchen(a)gmail.com> --- Change since v1: use clear subject format --- sound/soc/ti/davinci-mcasp.c | 27 +++++++++++++++------------ 1 file changed, 15 insertions(+), 12 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index c0892be2992b..172fea764a31 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1328,15 +1328,16 @@ static int davinci_mcasp_hw_rule_slot_width(struct snd_pcm_hw_params *params, struct davinci_mcasp_ruledata *rd = rule->private; struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); struct snd_mask nfmt; - int i, slot_width; + int slot_width; + snd_pcm_format_t i; snd_mask_none(&nfmt); slot_width = rd->mcasp->slot_width; - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - if (snd_mask_test(fmt, i)) { + pcm_for_each_format(i) { + if (snd_mask_test_format(fmt, i)) { if (snd_pcm_format_width(i) <= slot_width) { - snd_mask_set(&nfmt, i); + snd_mask_set_format(&nfmt, i); } } } @@ -1350,15 +1351,16 @@ static int davinci_mcasp_hw_rule_format_width(struct snd_pcm_hw_params *params, struct davinci_mcasp_ruledata *rd = rule->private; struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); struct snd_mask nfmt; - int i, format_width; + int format_width; + snd_pcm_format_t i; snd_mask_none(&nfmt); format_width = rd->mcasp->max_format_width; - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - if (snd_mask_test(fmt, i)) { + pcm_for_each_format(i) { + if (snd_mask_test_format(fmt, i)) { if (snd_pcm_format_width(i) == format_width) { - snd_mask_set(&nfmt, i); + snd_mask_set_format(&nfmt, i); } } } @@ -1431,12 +1433,13 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, struct snd_mask nfmt; int rate = params_rate(params); int slots = rd->mcasp->tdm_slots; - int i, count = 0; + int count = 0; + snd_pcm_format_t i; snd_mask_none(&nfmt); - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - if (snd_mask_test(fmt, i)) { + pcm_for_each_format(i) { + if (snd_mask_test_format(fmt, i)) { uint sbits = snd_pcm_format_width(i); unsigned int sysclk_freq; int ppm; @@ -1454,7 +1457,7 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, sbits * slots * rate, false); if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { - snd_mask_set(&nfmt, i); + snd_mask_set_format(&nfmt, i); count++; } } -- 2.34.1

2 1

[PATCH] ASoC: amd: vangogh: Add check for acp config flags in vangogh platform
by Venkata Prasad Potturu 30 May '23

30 May '23

We have SOF and generic ACP support enabled for Vangogh platform on some machines. Since we have same PCI id used for probing, add check for machine configuration flag to avoid conflict with newer pci drivers. Such machine flag has been initialized via dmi match on few Vangogh based machines. If no flag is specified probe and register older platform device. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/vangogh/acp5x.h | 2 ++ sound/soc/amd/vangogh/pci-acp5x.c | 7 ++++++- 2 files changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/vangogh/acp5x.h b/sound/soc/amd/vangogh/acp5x.h index bd9f1c5684d1..ac1936a8c43f 100644 --- a/sound/soc/amd/vangogh/acp5x.h +++ b/sound/soc/amd/vangogh/acp5x.h @@ -147,6 +147,8 @@ static inline void acp_writel(u32 val, void __iomem *base_addr) writel(val, base_addr - ACP5x_PHY_BASE_ADDRESS); } +int snd_amd_acp_find_config(struct pci_dev *pci); + static inline u64 acp_get_byte_count(struct i2s_stream_instance *rtd, int direction) { diff --git a/sound/soc/amd/vangogh/pci-acp5x.c b/sound/soc/amd/vangogh/pci-acp5x.c index e0df17c88e8e..c4634a8a17cd 100644 --- a/sound/soc/amd/vangogh/pci-acp5x.c +++ b/sound/soc/amd/vangogh/pci-acp5x.c @@ -125,10 +125,15 @@ static int snd_acp5x_probe(struct pci_dev *pci, { struct acp5x_dev_data *adata; struct platform_device_info pdevinfo[ACP5x_DEVS]; - unsigned int irqflags; + unsigned int irqflags, flag; int ret, i; u32 addr, val; + /* Return if acp config flag is defined */ + flag = snd_amd_acp_find_config(pci); + if (flag) + return -ENODEV; + irqflags = IRQF_SHARED; if (pci->revision != 0x50) return -ENODEV; -- 2.25.1

2 1

[PATCH] ASoC: amd: ps: fix for acp_lock access in pdm driver
by Vijendar Mukunda 30 May '23

30 May '23

Sending the mutex address(acp_lock) as platform data during ACP PDM platform driver register sequence, its creating copy of the platform data. Referencing this platform data in ACP PDM driver results incorrect reference to the common lock usage. Instead of directly passing the lock address as platform data, retrieve it from parent driver data structure and use the same lock reference in ACP PDM driver. Fixes: 45aa83cb9388 ("ASoC: amd: ps: use acp_lock to protect common registers in pdm driver") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 3 +-- sound/soc/amd/ps/ps-pdm-dma.c | 10 +++++----- 2 files changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index c957718abefc..54752d6040d6 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -198,8 +198,7 @@ static int create_acp63_platform_devs(struct pci_dev *pci, struct acp63_dev_data case ACP63_PDM_DEV_MASK: adata->pdm_dev_index = 0; acp63_fill_platform_dev_info(&pdevinfo[0], parent, NULL, "acp_ps_pdm_dma", - 0, adata->res, 1, &adata->acp_lock, - sizeof(adata->acp_lock)); + 0, adata->res, 1, NULL, 0); acp63_fill_platform_dev_info(&pdevinfo[1], parent, NULL, "dmic-codec", 0, NULL, 0, NULL, 0); acp63_fill_platform_dev_info(&pdevinfo[2], parent, NULL, "acp_ps_mach", diff --git a/sound/soc/amd/ps/ps-pdm-dma.c b/sound/soc/amd/ps/ps-pdm-dma.c index 3ecc6cf3fd34..bdbbb797c74d 100644 --- a/sound/soc/amd/ps/ps-pdm-dma.c +++ b/sound/soc/amd/ps/ps-pdm-dma.c @@ -358,12 +358,12 @@ static int acp63_pdm_audio_probe(struct platform_device *pdev) { struct resource *res; struct pdm_dev_data *adata; + struct acp63_dev_data *acp_data; + struct device *parent; int status; - if (!pdev->dev.platform_data) { - dev_err(&pdev->dev, "platform_data not retrieved\n"); - return -ENODEV; - } + parent = pdev->dev.parent; + acp_data = dev_get_drvdata(parent); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { dev_err(&pdev->dev, "IORESOURCE_MEM FAILED\n"); @@ -379,7 +379,7 @@ static int acp63_pdm_audio_probe(struct platform_device *pdev) return -ENOMEM; adata->capture_stream = NULL; - adata->acp_lock = pdev->dev.platform_data; + adata->acp_lock = &acp_data->acp_lock; dev_set_drvdata(&pdev->dev, adata); status = devm_snd_soc_register_component(&pdev->dev, &acp63_pdm_component, -- 2.34.1

2 1

[PATCH] ASoC: nau8824: Add quirk to active-high jack-detect
by drosdeck 30 May '23

30 May '23

Add entries for Positivo laptops: CW14Q01P, K1424G, N14ZP74G to the DMI table, so that active-high jack-detect will work properly on these laptops. Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> --- sound/soc/codecs/nau8824.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 4f19fd9b65d1..5a4db8944d06 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1903,6 +1903,30 @@ static const struct dmi_system_id nau8824_quirk_table[] = { }, .driver_data = (void *)(NAU8824_MONO_SPEAKER), }, + { + /* Positivo CW14Q01P */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "CW14Q01P"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo K1424G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "K1424G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo N14ZP74G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "N14ZP74G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, {} }; -- 2.39.2

2 1

[PATCH] sound: Fix incorrect calculation of object size by sizeof
by Lu Hongfei 30 May '23

30 May '23

What we need to calculate is the size of the object, not the size of the pointer. This patch can fix this error. Signed-off-by: Lu Hongfei <luhongfei(a)vivo.com> --- sound/soc/sof/ipc4-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) mode change 100644 => 100755 sound/soc/sof/ipc4-topology.c diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 5abe616a2054..29b6dede53b0 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -220,7 +220,7 @@ static int sof_ipc4_get_audio_fmt(struct snd_soc_component *scomp, ret = sof_update_ipc_object(scomp, available_fmt, SOF_AUDIO_FMT_NUM_TOKENS, swidget->tuples, - swidget->num_tuples, sizeof(available_fmt), 1); + swidget->num_tuples, sizeof(*available_fmt), 1); if (ret) { dev_err(scomp->dev, "Failed to parse audio format token count\n"); return ret; -- 2.39.0

2 1

[PATCH v4 6/6] ASoC: dt-bindings: Add tas2781 amplifier
by Shenghao Ding 30 May '23

30 May '23

Create tas2781.yaml for tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v4: - remove '\t' in the file Changes to be committed: new file: Documentation/devicetree/bindings/sound/ti,tas2781.yaml --- .../devicetree/bindings/sound/ti,tas2781.yaml | 88 +++++++++++++++++++ 1 file changed, 88 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas2781.yaml diff --git a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml new file mode 100644 index 000000000000..b3dcd7b18f5d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml @@ -0,0 +1,88 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2022 - 2023 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tas2781.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TAS2781 SmartAMP + +maintainers: + - Shenghao Ding <shenghao-ding(a)ti.com> + +description: + The TAS2781 is a mono, digital input Class-D audio amplifier + optimized for efficiently driving high peak power into small + loudspeakers. Integrated an on-chip DSP supports Texas Instruments + Smart Amp speaker protection algorithm. The integrated speaker + voltage and current sense provides for real time + monitoring of loudspeaker behavior. + +allOf: + - $ref: dai-common.yaml# + +properties: + '#address-cells': + const: 1 + '#size-cells': + const: 0 + + compatible: + enum: + - ti,tas2781 + + reg: + description: + I2C address, in multiple tas2781s case, all the i2c address + aggreate as one Audio Device to support multiple audio slots. + maxItems: 4 + items: + minimum: 0x38 + maximum: 0x3f + + reset-gpios: + maxItems: 1 + + interrupts: + maxItems: 1 + + ti,broadcast-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Generic I2C address for all the tas2781 devices in + purpose of I2C broadcast during the multi-device + writes, useless in mono case or remove this item to + disable broadcast mode. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + /* example with quad support, such as tablet or pad device */ + #address-cells = <1>; + #size-cells = <0>; + quad: codec@38 { + compatible = "ti,tas2781"; + reg = < 0x38 /* Audio slot 0 */ + 0x3a /* Audio slot 1 */ + 0x39 /* Audio slot 2 */ + 0x3b /* Audio slot 3 */ + >; + #sound-dai-cells = <1>; + reset-gpios = < &gpio1 10 GPIO_ACTIVE_HIGH >; + interrupt-parent = <&gpio1>; + interrupts = <15>; + /* Generic I2C addr among all the tas2781s for i2c data broadcast */ + ti,broadcast-addr = <0x40>; + }; + }; +... -- 2.34.1

2 1

[PATCH 0/3] ES8316 audio codec fixes on Rock5B
by Cristian Ciocaltea 30 May '23

30 May '23

This patch series handles a few issues related to the ES8316 audio codec, discovered while doing some testing on the Rock 5B board. Cristian Ciocaltea (3): ASoC: es8316: Increment max value for ALC Capture Target Volume control ASoC: es8316: Do not set rate constraints for unsupported MCLKs arm64: dts: rockchip: Assign ES8316 MCLK rate on rk3588-rock-5b arch/arm64/boot/dts/rockchip/rk3588-rock-5b.dts | 2 ++ sound/soc/codecs/es8316.c | 4 ++-- 2 files changed, 4 insertions(+), 2 deletions(-) -- 2.40.1

2 13

Fwd: Apple USB-C 3.5mm dongle cannot output 41000hz
by Bagas Sanjaya 30 May '23

30 May '23

Hi, I notice a regression report on Bugzilla [1]. Quoting from it: > Overview: > With an A2049 Apple USB-C to 3.5mm DAC dongle, I have the same issue as outlined in this link: ALSA won't detect 44100Hz as an available sample rate: > https://github.com/mikebrady/shairport-sync/issues/1504 > > > Steps to Reproduce: > Play a 44.1Khz audio file. > > > Actual Results: > Pipewire reports that the output is locked to 48000hz, and ALSA reports that this is the only detected supported rate: > > $ cat /proc/asound/card0/stream0 > Playback: > Status: Running > Interface = 1 > Altset = 1 > Packet Size = 288 > Momentary freq = 48000 Hz (0x30.0000) > Interface 1 > Altset 1 > Format: S24_3LE > Channels: 2 > Endpoint: 0x02 (2 OUT) (SYNC) > Rates: 48000 - 48000 (continuous) > Bits: 0 > Channel map: FL FR > Interface 1 > Altset 2 > Format: S16_LE > Channels: 2 > Endpoint: 0x02 (2 OUT) (SYNC) > Rates: 48000 - 48000 (continuous) > Bits: 0 > Channel map: FL FR > > > Expected Result: > It should be switching between 44100-48000hz, the only two rates it supports. > > > Build Date and Hardware: > Running Arch Linux with kernel 6.3.4-arch1-1, alsa-lib 1.2.9-1. > AMD Ryzen 3600 on a MSI B450M Mortar Max motherboard > > > Additional Information: > It seems like a regression between Linux 4.x and 5.x, and halfway through the above thread the user "quantonian" added the device to the ALSA quirks table and has posted a patch: > > https://github.com/mikebrady/shairport-sync/issues/1504#issuecomment-119335… > > Thanks if this can be fixed in the kernel. The dongle in question is US $9: > https://www.apple.com/shop/product/MU7E2AM/A/usb-c-to-35-mm-headphone-jack-… See bugzilla for the full thread. Unfortunately, the reporter said that he can only run kernel as old as 5.6, as older than that, he experiences unrelated GPU crashes, so he can't bisect this regression. Anyway, I'm adding it to regzbot: #regzbot introduced: v4.19..v5.10 https://bugzilla.kernel.org/show_bug.cgi?id=217501 #regzbot title: 44100hz is mistaken as 48000hz on Apple Headphone Jack Adapter #regzbot link: https://github.com/mikebrady/shairport-sync/issues/1504 #regzbot link: https://www.audiosciencereview.com/forum/index.php?threads/review-apple-vs-… Thanks. [1]: https://bugzilla.kernel.org/show_bug.cgi?id=217501 -- An old man doll... just what I always wanted! - Clara

1 0

[PATCH 0/3] ASoC: SOF: Intel: mtl: Enable multicore support
by Peter Ujfalusi 30 May '23

30 May '23

Hi, The following series will enable multicore support on MTL platforms similarly to other Intel platforms. The TGL patch is included to simplify the core_put implementation. Multicore support can be enabled by updated topologies, with current set of tplg files this series is not introducing any runtime change. Regards, Peter --- Rander Wang (3): ASoC: SOF: Intel: mtl: setup primary core info on MeteorLake platform ASoC: SOF: Intel: mtl: add core_get & put support on MeterLake platforms ASoC: SOF: Intel: tgl: unify core_put on IPC3 & IPC4 path sound/soc/sof/intel/mtl.c | 52 +++++++++++++++++++++++++++++++++++---- sound/soc/sof/intel/tgl.c | 10 +++++--- 2 files changed, 54 insertions(+), 8 deletions(-) -- 2.40.1

3 6

[PATCH v3 00/12] ALSA: hda: Adding support for CS35L56 on HDA systems
by Richard Fitzgerald 30 May '23

30 May '23

This set of patches adds support for using the CS35L56 boosted smart amplifier on HDA systems. In these systems the CS35L56 audio is routed through a HDA-to-I2S bridge codec. This doesn't include the changes to the Realtek driver to actually hook up the CS35L56 driver, because we don't yet have the QUIRK IDs to associate it with. But we want to publish the driver now so that it is available for bringing up hardware with the CS35L56. The first 9 patches are moving code out of the ASoC driver and into the shared library so that it can be shared with the HDA driver. Patch #10 fixes missing #includes in the HDA headers so that the CS35L56 driver doesn't have to #include headers that it doesn't use. Finally, #11 and #12 actually add the support for CS35L56 on HDA. CHANGES SINCE V2 All in patch #12: - Add __maybe_unused to runtime suspend/resume callbacks to prevent build error if CONFIG_PM is not defined - Removed __maybe_unused from system suspend/resume functions - Change i2c .probe_new to .probe - Removed unnecessary returns in cs35l56_hda_create_controls() - Change if (ret != 0) to if (ret) Richard Fitzgerald (4): ASoC: cs35l56: Move runtime suspend/resume to shared library ASoC: cs35l56: Move cs_dsp init into shared library ASoC: cs35l56: Move part of cs35l56_init() to shared library ALSA: hda: Fix missing header dependencies Simon Trimmer (8): ASoC: cs35l56: Move shared data into a common data structure ASoC: cs35l56: Make cs35l56_system_reset() code more generic ASoC: cs35l56: Convert utility functions to use common data structure ASoC: cs35l56: Move utility functions to shared file ASoC: cs35l56: Make common function for control port wait ASoC: cs35l56: Make a common function to shutdown the DSP ALSA: hda: Add mute_hook to hda_component ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier include/sound/cs35l56.h | 29 +- sound/pci/hda/Kconfig | 31 + sound/pci/hda/Makefile | 6 + sound/pci/hda/cs35l56_hda.c | 990 ++++++++++++++++++++++++++++++ sound/pci/hda/cs35l56_hda.h | 48 ++ sound/pci/hda/cs35l56_hda_i2c.c | 69 +++ sound/pci/hda/cs35l56_hda_spi.c | 68 ++ sound/pci/hda/hda_auto_parser.h | 2 + sound/pci/hda/hda_component.h | 1 + sound/pci/hda/hda_generic.h | 3 + sound/soc/codecs/cs35l56-i2c.c | 14 +- sound/soc/codecs/cs35l56-sdw.c | 72 +-- sound/soc/codecs/cs35l56-shared.c | 459 +++++++++++++- sound/soc/codecs/cs35l56-spi.c | 10 +- sound/soc/codecs/cs35l56.c | 648 ++++--------------- sound/soc/codecs/cs35l56.h | 15 +- 16 files changed, 1873 insertions(+), 592 deletions(-) create mode 100644 sound/pci/hda/cs35l56_hda.c create mode 100644 sound/pci/hda/cs35l56_hda.h create mode 100644 sound/pci/hda/cs35l56_hda_i2c.c create mode 100644 sound/pci/hda/cs35l56_hda_spi.c -- 2.30.2

2 14

[PATCH v3 00/21] ASoC: replace dpcm_playback/capture to playback/capture_only
by Kuninori Morimoto 30 May '23

30 May '23

Hi Mark This is v3 patch-set of dpcm_playback/capture flag cleanup. Current ASoC DPCM need dpcm_playback/capture flags to use it. But we are using playback/capture_only flag on Normal/Codec2Codec case. I think these are duplicated, we can share same flags for all cases. On v1 patch-set, we have noticed that it has been don't check DPCM BE Codec valid before. Therefore, it breaks Intel sound card probing if it strictly check all cases validation. To solve this issue, we need to update Intel driver first, but it is very complex and I can't test it. Thus, this patch-set keep not to check DPCM BE Codec to keep compatibility. But it should be updated in the future. v2 patch-set had error handling mistake. CPU:Codec = 1:N has been possible to handling, but v2 patch-set handled it as error. v3 solved it. v2 -> v3 - fixup error handling v1 -> v2 - Add Reviewed-by - Separate Intel patch - tidyup playback/capture_only flags conversion Link: https://lore.kernel.org/r/87ttw1gqgn.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87353uqjiu.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/ab3f0c0a-62fd-a468-b3cf-0e4b59bac6ae@linux.intel.… Kuninori Morimoto (21): ASoC: soc-pcm.c: indicate error if stream has no playback no capture ASoC: soc-pcm.c: use dai_link on soc_get_playback_capture() ASoC: soc-pcm.c: cleanup soc_get_playback_capture() error ASoC: soc-pcm.c: use temporary variable at soc_get_playback_capture() ASoC: soc-pcm.c: tidyup playback/capture_only at soc_get_playback_capture() ASoC: soc-pcm.c: cleanup normal connection loop at soc_get_playback_capture() part1 ASoC: soc-pcm.c: cleanup normal connection loop at soc_get_playback_capture() part2 ASoC: soc-pcm.c: cleanup soc_get_playback_capture() ASoC: amd: replace dpcm_playback/capture to playback/capture_only ASoC: fsl: replace dpcm_playback/capture to playback/capture_only ASoC: sof: replace dpcm_playback/capture to playback/capture_only ASoC: meson: replace dpcm_playback/capture to playback/capture_only ASoC: Intel: replace dpcm_playback/capture to playback/capture_only ASoC: samsung: replace dpcm_playback/capture to playback/capture_only ASoC: mediatek: replace dpcm_playback/capture to playback/capture_only ASoC: soc-dai.c: replace dpcm_playback/capture to playback/capture_only ASoC: Intel/avs: replace dpcm_playback/capture to playback/capture_only ASoC: soc-core.c: replace dpcm_playback/capture to playback/capture_only ASoC: soc-topology.c: replace dpcm_playback/capture to playback/capture_only ASoC: soc-compress.c: replace dpcm_playback/capture to playback/capture_only ASoC: soc-pcm.c: remove dpcm_playback/capture include/sound/soc.h | 4 - sound/soc/amd/acp-da7219-max98357a.c | 20 +-- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 20 +-- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 3 - sound/soc/fsl/fsl-asoc-card.c | 16 +-- sound/soc/fsl/imx-audmix.c | 6 +- sound/soc/fsl/imx-card.c | 4 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 32 +++-- sound/soc/intel/boards/bxt_rt298.c | 26 ++-- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 14 +-- sound/soc/intel/boards/ehl_rt5660.c | 14 +-- sound/soc/intel/boards/glk_rt5682_max98357a.c | 30 +++-- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 26 ++-- sound/soc/intel/boards/kbl_da7219_max98927.c | 54 ++++---- sound/soc/intel/boards/kbl_rt5660.c | 18 ++- sound/soc/intel/boards/kbl_rt5663_max98927.c | 44 +++---- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 22 ++-- sound/soc/intel/boards/skl_hda_dsp_common.c | 14 +-- .../soc/intel/boards/skl_nau88l25_max98357a.c | 26 ++-- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 26 ++-- sound/soc/intel/boards/skl_rt286.c | 26 ++-- sound/soc/intel/boards/sof_cs42l42.c | 12 +- sound/soc/intel/boards/sof_da7219_max98373.c | 16 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_nau8825.c | 12 +- sound/soc/intel/boards/sof_pcm512x.c | 8 +- sound/soc/intel/boards/sof_rt5682.c | 12 +- sound/soc/intel/boards/sof_sdw.c | 4 +- sound/soc/intel/boards/sof_ssp_amp.c | 11 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +-- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++-- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++--- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++--- .../mt8186/mt8186-mt6366-da7219-max98357.c | 86 +++++-------- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 86 +++++-------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 48 ++++---- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 ++++++------ sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 +++++---- sound/soc/meson/axg-card.c | 8 +- sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/samsung/odroid.c | 10 +- sound/soc/soc-compress.c | 11 +- sound/soc/soc-core.c | 20 +-- sound/soc/soc-dai.c | 6 +- sound/soc/soc-pcm.c | 115 +++++++----------- sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 4 - 84 files changed, 512 insertions(+), 842 deletions(-) -- 2.25.1

2 25

[PATCH v2] ASoC: SOF: imx: Add OF machine descriptors for i.MX platforms
by Daniel Baluta 29 May '23

29 May '23

From: Paul Olaru <paul.olaru(a)nxp.com> Covered platforms: i.MX8QXP, i.MX8QM, i.MX8MP, i.MX8ULP. These descriptors are used to locate the machine driver and topology filenames. Note: because the compatible values for the boards do not change between the cs42888 and the wm8960 cases, I have selected a common topology filename, which is intended to be a symlink to either the wm8960 or the cs42888 topology file. For uniformity I did the same for i.MX8ULP. Signed-off-by: Paul Olaru <paul.olaru(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- Changes since v1: - rebased patch on Mark's tree. sound/soc/sof/imx/imx8.c | 16 ++++++++++++++++ sound/soc/sof/imx/imx8m.c | 10 ++++++++++ sound/soc/sof/imx/imx8ulp.c | 10 ++++++++++ 3 files changed, 36 insertions(+) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 2844d9a8040a..3c9f0cfd8218 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -608,7 +608,22 @@ static struct snd_sof_dsp_ops sof_imx8x_ops = { SNDRV_PCM_INFO_NO_PERIOD_WAKEUP }; +static struct snd_sof_of_mach sof_imx8_machs[] = { + { + .compatible = "fsl,imx8qxp", + .drv_name = "asoc-simple-card", + .sof_tplg_filename = "sof-imx8.tplg", + }, + { + .compatible = "fsl,imx8qm", + .drv_name = "asoc-simple-card", + .sof_tplg_filename = "sof-imx8.tplg", + }, + {} +}; + static struct sof_dev_desc sof_of_imx8qxp_desc = { + .of_machines = sof_imx8_machs, .ipc_supported_mask = BIT(SOF_IPC), .ipc_default = SOF_IPC, .default_fw_path = { @@ -625,6 +640,7 @@ static struct sof_dev_desc sof_of_imx8qxp_desc = { }; static struct sof_dev_desc sof_of_imx8qm_desc = { + .of_machines = sof_imx8_machs, .ipc_supported_mask = BIT(SOF_IPC), .ipc_default = SOF_IPC, .default_fw_path = { diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index 1243f8a6141e..a4050f6ec263 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -470,7 +470,17 @@ static struct snd_sof_dsp_ops sof_imx8m_ops = { SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, }; +static struct snd_sof_of_mach sof_imx8m_machs[] = { + { + .compatible = "fsl,imx8mp", + .drv_name = "asoc-simple-card", + .sof_tplg_filename = "sof-imx8m.tplg", + }, + {} +}; + static struct sof_dev_desc sof_of_imx8mp_desc = { + .of_machines = sof_imx8m_machs, .ipc_supported_mask = BIT(SOF_IPC), .ipc_default = SOF_IPC, .default_fw_path = { diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c index 4a562c9856e9..82fcc672bb1c 100644 --- a/sound/soc/sof/imx/imx8ulp.c +++ b/sound/soc/sof/imx/imx8ulp.c @@ -477,7 +477,17 @@ static struct snd_sof_dsp_ops sof_imx8ulp_ops = { .set_power_state = imx8ulp_dsp_set_power_state, }; +static struct snd_sof_of_mach sof_imx8ulp_machs[] = { + { + .compatible = "fsl,imx8ulp", + .drv_name = "asoc-simple-card", + .sof_tplg_filename = "sof-imx8ulp.tplg", + }, + {} +}; + static struct sof_dev_desc sof_of_imx8ulp_desc = { + .of_machines = sof_imx8ulp_machs, .ipc_supported_mask = BIT(SOF_IPC), .ipc_default = SOF_IPC, .default_fw_path = { -- 2.25.1

3 4

[PATCH 1/2] soundwire: qcom: fix unbalanced pm_runtime_put()
by Krzysztof Kozlowski 29 May '23

29 May '23

This reverts commit 57ed510b0547 ("soundwire: qcom: use pm_runtime_resume_and_get()") which introduced unbalanced pm_runtime_put(), when device did not have runtime PM enabled. If pm_runtime_resume_and_get() failed with -EACCES, the driver continued execution and finally called pm_runtime_put_autosuspend(). Since pm_runtime_resume_and_get() drops the usage counter on every error, this lead to double decrement of that counter visible in certain debugfs actions on unattached devices (still in reset state): $ cat /sys/kernel/debug/soundwire/master-0-0/sdw:0:0217:f001:00:0/registers qcom-soundwire 3210000.soundwire-controller: swrm_wait_for_wr_fifo_avail err write overflow soundwire sdw-master-0: trf on Slave 1 failed:-5 read addr e36 count 1 soundwire sdw:0:0217:f001:00:0: Runtime PM usage count underflow! Fixes: 57ed510b0547 ("soundwire: qcom: use pm_runtime_resume_and_get()") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/soundwire/qcom.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index aad5942e5980..f442280af9d3 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -614,11 +614,12 @@ static irqreturn_t qcom_swrm_wake_irq_handler(int irq, void *dev_id) struct qcom_swrm_ctrl *ctrl = dev_id; int ret; - ret = pm_runtime_resume_and_get(ctrl->dev); + ret = pm_runtime_get_sync(ctrl->dev); if (ret < 0 && ret != -EACCES) { dev_err_ratelimited(ctrl->dev, - "pm_runtime_resume_and_get failed in %s, ret %d\n", + "pm_runtime_get_sync failed in %s, ret %d\n", __func__, ret); + pm_runtime_put_noidle(ctrl->dev); return ret; } @@ -1197,11 +1198,12 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int ret, i; - ret = pm_runtime_resume_and_get(ctrl->dev); + ret = pm_runtime_get_sync(ctrl->dev); if (ret < 0 && ret != -EACCES) { dev_err_ratelimited(ctrl->dev, - "pm_runtime_resume_and_get failed in %s, ret %d\n", + "pm_runtime_get_sync failed in %s, ret %d\n", __func__, ret); + pm_runtime_put_noidle(ctrl->dev); return ret; } @@ -1402,11 +1404,12 @@ static int swrm_reg_show(struct seq_file *s_file, void *data) struct qcom_swrm_ctrl *ctrl = s_file->private; int reg, reg_val, ret; - ret = pm_runtime_resume_and_get(ctrl->dev); + ret = pm_runtime_get_sync(ctrl->dev); if (ret < 0 && ret != -EACCES) { dev_err_ratelimited(ctrl->dev, - "pm_runtime_resume_and_get failed in %s, ret %d\n", + "pm_runtime_get_sync failed in %s, ret %d\n", __func__, ret); + pm_runtime_put_noidle(ctrl->dev); return ret; } -- 2.34.1

3 4

[PATCH 0/3] soundwire: improve bus reset
by Bard Liao 29 May '23

29 May '23

Improve bus reset sequence by updating programing sequence. Pierre-Louis Bossart (3): soundwire: intel_bus_common: enable interrupts last soundwire: intel/cadence: update hardware reset sequence soundwire: cadence: revisit parity injection drivers/soundwire/cadence_master.c | 50 ++++++++++----- drivers/soundwire/cadence_master.h | 3 + drivers/soundwire/intel_bus_common.c | 93 ++++++++++++++++------------ 3 files changed, 89 insertions(+), 57 deletions(-) -- 2.25.1

2 4

[PATCH] soundwire: intel_bus_common: update error log
by Bard Liao 29 May '23

29 May '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> We use __func__ in all calls of sdw_cdns_check_self_clearing_bits(), except in one case. Likely an editing miss when the code was refactored. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/intel_bus_common.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/drivers/soundwire/intel_bus_common.c b/drivers/soundwire/intel_bus_common.c index be0c93106fc8..e5ac3cc7cb79 100644 --- a/drivers/soundwire/intel_bus_common.c +++ b/drivers/soundwire/intel_bus_common.c @@ -182,8 +182,7 @@ int intel_start_bus_after_clock_stop(struct sdw_intel *sdw) return ret; } - sdw_cdns_check_self_clearing_bits(cdns, "intel_resume_runtime no_quirks", - true, INTEL_MASTER_RESET_ITERATIONS); + sdw_cdns_check_self_clearing_bits(cdns, __func__, true, INTEL_MASTER_RESET_ITERATIONS); return 0; } -- 2.25.1

2 1

[PATCH] soundwire: amd: Improve error message in remove callback
by Uwe Kleine-König 29 May '23

29 May '23

Returning an error code in the remove callback yields to an error message remove callback returned a non-zero value. This will be ignored. After that the device is removed anyhow. Improve the error message to at least say what the actual problem is. While touching that code, convert the driver to the .remove_new() callback which returns no value with the same effect as returning zero in a .remove() callback. As the return value is ignored by the core the only effect of this patch is to improve the error message. (And the motivating effect is that there is one less driver using .remove().) Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- drivers/soundwire/amd_manager.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 9fb7f91ca182..08aeb7ed00e1 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -972,15 +972,18 @@ static int amd_sdw_manager_probe(struct platform_device *pdev) return 0; } -static int amd_sdw_manager_remove(struct platform_device *pdev) +static void amd_sdw_manager_remove(struct platform_device *pdev) { struct amd_sdw_manager *amd_manager = dev_get_drvdata(&pdev->dev); + int ret; pm_runtime_disable(&pdev->dev); cancel_work_sync(&amd_manager->probe_work); amd_disable_sdw_interrupts(amd_manager); sdw_bus_master_delete(&amd_manager->bus); - return amd_disable_sdw_manager(amd_manager); + ret = amd_disable_sdw_manager(amd_manager); + if (ret) + dev_err(&pdev->dev, "Failed to disable device (%pe)\n", ERR_PTR(ret)); } static int amd_sdw_clock_stop(struct amd_sdw_manager *amd_manager) @@ -1194,7 +1197,7 @@ static const struct dev_pm_ops amd_pm = { static struct platform_driver amd_sdw_driver = { .probe = &amd_sdw_manager_probe, - .remove = &amd_sdw_manager_remove, + .remove_new = &amd_sdw_manager_remove, .driver = { .name = "amd_sdw_manager", .pm = &amd_pm, base-commit: ac9a78681b921877518763ba0e89202254349d1b -- 2.39.2

2 1

[PATCH v4 3/6] ASoC: tas2781: Add tas2781 driver
by Shenghao Ding 28 May '23

28 May '23

Create tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v4: - fixed issue| Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202305192322.gBZ4JIyr-lkp@intel.com/ Changes to be committed: modified: sound/soc/codecs/Kconfig modified: sound/soc/codecs/Makefile new file: sound/soc/codecs/tas2781-comlib.c new file: sound/soc/codecs/tas2781-i2c.c --- sound/soc/codecs/Kconfig | 25 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/tas2781-comlib.c | 581 +++++++++++++++++++++++ sound/soc/codecs/tas2781-i2c.c | 736 ++++++++++++++++++++++++++++++ 4 files changed, 1348 insertions(+) create mode 100644 sound/soc/codecs/tas2781-comlib.c create mode 100644 sound/soc/codecs/tas2781-i2c.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8020097d4e4c..0481ed52e1c3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -234,6 +234,9 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2764 imply SND_SOC_TAS2770 imply SND_SOC_TAS2780 + imply SND_SOC_TAS2781_COMLIB + imply SND_SOC_TAS2781_FMWLIB + imply SND_SOC_TAS2781 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1699,6 +1702,28 @@ config SND_SOC_TAS2780 Enable support for Texas Instruments TAS2780 high-efficiency digital input mono Class-D audio power amplifiers. +config SND_SOC_TAS2781_COMLIB + depends on I2C + select CRC8 + select REGMAP_I2C + tristate + +config SND_SOC_TAS2781_FMWLIB + tristate + default n + +config SND_SOC_TAS2781 + tristate "Texas Instruments TAS2781 speaker amplifier" + depends on I2C + select SND_SOC_TAS2781_COMLIB + select SND_SOC_TAS2781_FMWLIB + help + Enable support for Texas Instruments TAS2781 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + Note the TAS2781 driver implements a flexible and configurable + algo coefficient setting, for one, two or even multiple TAS2781 + chips. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5cdbae88e6e3..97c3881ed587 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -269,6 +269,9 @@ snd-soc-tas5805m-objs := tas5805m.o snd-soc-tas6424-objs := tas6424.o snd-soc-tda7419-objs := tda7419.o snd-soc-tas2770-objs := tas2770.o +snd-soc-tas2781-comlib-objs := tas2781-comlib.o +snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o +snd-soc-tas2781-objs := tas2781-i2c.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -633,6 +636,9 @@ obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS2562) += snd-soc-tas2562.o obj-$(CONFIG_SND_SOC_TAS2764) += snd-soc-tas2764.o obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o +obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o +obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o +obj-$(CONFIG_SND_SOC_TAS2781) += snd-soc-tas2781.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c new file mode 100644 index 000000000000..5994698326b1 --- /dev/null +++ b/sound/soc/codecs/tas2781-comlib.c @@ -0,0 +1,581 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// tas2781-lib.c -- TAS2781 Common functions for HDA and ASoC Audio drivers +// +// Copyright 2023 Texas Instruments, Inc. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> + +#include <linux/crc8.h> +#include <linux/firmware.h> +#include <linux/gpio/consumer.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_irq.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tas2781.h> + +#define TASDEVICE_CRC8_POLYNOMIAL 0x4d + +static const struct regmap_range_cfg tasdevice_ranges[] = { + { + .range_min = 0, + .range_max = 256 * 128, + .selector_reg = TASDEVICE_PAGE_SELECT, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 8, + .val_bits = 8, + .cache_type = REGCACHE_RBTREE, + .ranges = tasdevice_ranges, + .num_ranges = ARRAY_SIZE(tasdevice_ranges), + .max_register = 256 * 128, +}; + +static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, + enum channel chn, int book) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + int ret = 0; + int i; + + if (chn < tas_priv->ndev) { + struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; + + if (tas_priv->glb_addr.ref_cnt != 0) { + tas_priv->glb_addr.ref_cnt = 0; + tas_priv->glb_addr.cur_book = -1; + } + client->addr = tasdev->dev_addr; + if (tasdev->cur_book != book) { + ret = regmap_write(tas_priv->regmap, + TASDEVICE_BOOKCTL_REG, book); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", + __func__, ret); + goto out; + } + tasdev->cur_book = book; + } + } else if (chn == tas_priv->ndev) { + struct global_addr *glb_addr = &tas_priv->glb_addr; + + /* Chn == device number will enable global broadcast mode*/ + if (tas_priv->glb_addr.ref_cnt == 0) + for (i = 0; i < tas_priv->ndev; i++) + tas_priv->tasdevice[i].cur_book = -1; + client->addr = glb_addr->dev_addr; + if (glb_addr->cur_book != book) { + ret = regmap_write(tas_priv->regmap, + TASDEVICE_BOOKCTL_REG, book); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, book%x, E=%d\n", + __func__, book, ret); + goto out; + } + glb_addr->cur_book = book; + } + + glb_addr->ref_cnt++; + } else { + ret = -EINVAL; + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + } + +out: + return ret; +} + +int tasdevice_dev_read(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned int *val) +{ + int ret = 0; + + if (chn < tas_priv->ndev) { + ret = tasdevice_change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_read(tas_priv->regmap, TASDEVICE_PGRG(reg), val); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else { + ret = -EINVAL; + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + } + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_read); + +int tasdevice_dev_write(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned int value) +{ + int ret = 0; + + /* chn <= tas_dev->ndev will support global broadcast mode*/ + if (chn <= tas_priv->ndev) { + ret = tasdevice_change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_write(tas_priv->regmap, TASDEVICE_PGRG(reg), + value); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else { + ret = -EINVAL; + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + } + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_write); + +int tasdevice_dev_bulk_write( + struct tasdevice_priv *tas_priv, enum channel chn, + unsigned int reg, unsigned char *data, + unsigned int len) +{ + int ret = 0; + + /* chn <= tas_dev->ndev will support global broadcast mode*/ + if (chn <= tas_priv->ndev) { + ret = tasdevice_change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_bulk_write(tas_priv->regmap, TASDEVICE_PGRG(reg), + data, len); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else { + ret = -EINVAL; + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + } + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_bulk_write); + +int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned char *data, + unsigned int len) +{ + int ret = 0; + + if (chn < tas_priv->ndev) { + ret = tasdevice_change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_bulk_read(tas_priv->regmap, TASDEVICE_PGRG(reg), + data, len); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_bulk_read); + +int tasdevice_dev_update_bits( + struct tasdevice_priv *tas_priv, enum channel chn, + unsigned int reg, unsigned int mask, unsigned int value) +{ + int ret = 0; + + /* update-bits can't support global broadcast mode, + * because this operation contains a reading operation + * for register value. In global mode, reading register + * from all of the tas2781 will cause confusion. + */ + if (chn < tas_priv->ndev) { + ret = tasdevice_change_chn_book(tas_priv, chn, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) + goto out; + + ret = regmap_update_bits(tas_priv->regmap, + TASDEVICE_PGRG(reg), mask, value); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + } else { + dev_err(tas_priv->dev, "%s, no such channel(%d)\n", __func__, + chn); + ret = -EINVAL; + } + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_dev_update_bits); + +/* Only register write and bulk write support global mode + * Other cases, such as bits update, register read and bulk + * only support device-by-device operation + */ +static void tas2781_set_global_mode(struct tasdevice_priv *tas_priv) +{ + int i, ret; + + for (i = 0; i < tas_priv->ndev; i++) { + ret = tasdevice_dev_update_bits(tas_priv, i, + TAS2781_MISC_CFG2, TAS2781_GLOBAL_ADDR_MASK, + TAS2781_GLOBAL_ADDR_ENABLE); + if (ret < 0) { + dev_err(tas_priv->dev, "%s: chn %d set glb fail, %d\n", + __func__, i, ret); + continue; + } + } +} + +struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c) +{ + struct tasdevice_priv *tas_priv; + + tas_priv = devm_kzalloc(&i2c->dev, sizeof(*tas_priv), GFP_KERNEL); + if (!tas_priv) + return NULL; + tas_priv->dev = &i2c->dev; + tas_priv->client = (void *)i2c; + + return tas_priv; +} +EXPORT_SYMBOL_GPL(tasdevice_kzalloc); + +void tas2781_reset(struct tasdevice_priv *tas_dev) +{ + int ret, i; + + if (tas_dev->reset) { + gpiod_set_value_cansleep(tas_dev->reset, 0); + usleep_range(500, 1000); + gpiod_set_value_cansleep(tas_dev->reset, 1); + } else { + for (i = 0; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_write(tas_dev, i, + TAS2781_REG_SWRESET, + TAS2781_REG_SWRESET_RESET); + if (ret < 0) + dev_err(tas_dev->dev, + "dev %d swreset fail, %d\n", + i, ret); + } + } + usleep_range(1000, 1050); +} +EXPORT_SYMBOL_GPL(tas2781_reset); + +int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, + void (*cont)(const struct firmware *fw, void *context)) +{ + int ret = 0; + + /* Codec Lock Hold to ensure that codec_probe and firmware parsing and + * loading do not simultaneously execute. + */ + mutex_lock(&tas_priv->codec_lock); + + scnprintf(tas_priv->rca_binaryname, 64, "%srca%d.bin", + tas_priv->dev_name, tas_priv->ndev); + crc8_populate_msb(tas_priv->crc8_lkp_tbl, TASDEVICE_CRC8_POLYNOMIAL); + tas_priv->codec = codec; + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_priv->rca_binaryname, tas_priv->dev, GFP_KERNEL, tas_priv, + cont); + if (ret) + dev_err(tas_priv->dev, "request_firmware_nowait err:0x%08x\n", + ret); + + tas2781_reset(tas_priv); + if (tas_priv->set_global_mode) + tas_priv->set_global_mode(tas_priv); + + /* Codec Lock Release*/ + mutex_unlock(&tas_priv->codec_lock); + return ret; +} +EXPORT_SYMBOL_GPL(tascodec_init); + +int tasdevice_init(struct tasdevice_priv *tas_priv) +{ + int ret = 0; + int i; + + tas_priv->regmap = devm_regmap_init_i2c(tas_priv->client, + &tasdevice_regmap); + if (IS_ERR(tas_priv->regmap)) { + ret = PTR_ERR(tas_priv->regmap); + dev_err(tas_priv->dev, "Failed to allocate register map: %d\n", + ret); + goto out; + } + + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + for (i = 0; i < tas_priv->ndev; i++) { + tas_priv->tasdevice[i].cur_book = -1; + tas_priv->tasdevice[i].cur_prog = -1; + tas_priv->tasdevice[i].cur_conf = -1; + } + + if (tas_priv->glb_addr.dev_addr != 0 + && tas_priv->glb_addr.dev_addr < 0x7F) + tas_priv->set_global_mode = tas2781_set_global_mode; + dev_set_drvdata(tas_priv->dev, tas_priv); + + mutex_init(&tas_priv->codec_lock); + +out: + return ret; +} +EXPORT_SYMBOL_GPL(tasdevice_init); + +static void tasdev_dsp_prog_blk_remove(struct tasdevice_prog *prog) +{ + struct tasdevice_data *im; + struct tasdev_blk *blk; + unsigned int nr_blk; + + if (!prog) + return; + + im = &(prog->dev_data); + + if (!im->dev_blks) + return; + + for (nr_blk = 0; nr_blk < im->nr_blk; nr_blk++) { + blk = &(im->dev_blks[nr_blk]); + kfree(blk->data); + } + kfree(im->dev_blks); +} + +static void tasdev_dsp_prog_remove(struct tasdevice_prog *prog, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_prog_blk_remove(&prog[i]); + kfree(prog); +} + +static void tasdev_dsp_cfg_blk_remove(struct tasdevice_config *cfg) +{ + struct tasdevice_data *im; + struct tasdev_blk *blk; + unsigned int nr_blk; + + if (cfg) { + im = &(cfg->dev_data); + + if (!im->dev_blks) + return; + + for (nr_blk = 0; nr_blk < im->nr_blk; nr_blk++) { + blk = &(im->dev_blks[nr_blk]); + kfree(blk->data); + } + kfree(im->dev_blks); + } +} + +static void tasdev_dsp_cfg_remove(struct tasdevice_config *config, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_cfg_blk_remove(&config[i]); + kfree(config); +} + +void tasdevice_dsp_remove(void *context) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + struct tasdevice_fw *tas_fmw = tas_dev->fmw; + + if (!tas_dev->fmw) + return; + + if (tas_fmw->programs) + tasdev_dsp_prog_remove(tas_fmw->programs, + tas_fmw->nr_programs); + if (tas_fmw->configs) + tasdev_dsp_cfg_remove(tas_fmw->configs, + tas_fmw->nr_configurations); + kfree(tas_fmw); + tas_dev->fmw = NULL; +} +EXPORT_SYMBOL_GPL(tasdevice_dsp_remove); + +void tasdevice_remove(struct tasdevice_priv *tas_priv) +{ + if (gpio_is_valid(tas_priv->irq_info.irq_gpio)) + gpio_free(tas_priv->irq_info.irq_gpio); + kfree(tas_priv->acpi_subsystem_id); + mutex_destroy(&tas_priv->codec_lock); +} +EXPORT_SYMBOL_GPL(tasdevice_remove); + +static int tasdevice_clamp(int val, int max, unsigned int invert) +{ + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + return val; +} + +int tasdevice_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask; + int max = mc->max; + int err_cnt = 0; + int val, i, ret; + + mask = (1 << fls(max)) - 1; + mask <<= mc->shift; + val = tasdevice_clamp(ucontrol->value.integer.value[0], max, invert); + for (i = 0; i < tas_priv->ndev; i++) { + ret = tasdevice_dev_update_bits(tas_priv, i, + mc->reg, mask, (unsigned int)(val << mc->shift)); + if (!ret) + continue; + err_cnt++; + dev_err(tas_priv->dev, "set AMP vol error in dev %d\n", i); + } + + /* All the devices set error, return 0 */ + return (err_cnt == tas_priv->ndev) ? 0 : 1; +} +EXPORT_SYMBOL_GPL(tasdevice_amp_putvol); + +int tasdevice_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask = 0; + int max = mc->max; + int ret = 0; + int val; + + /* Read the primary device */ + ret = tasdevice_dev_read(tas_priv, 0, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + mask = (1 << fls(max)) - 1; + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + val = tasdevice_clamp(val, max, invert); + ucontrol->value.integer.value[0] = val; + +out: + return ret; + +} +EXPORT_SYMBOL_GPL(tasdevice_amp_getvol); + +int tasdevice_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int err_cnt = 0; + int ret = 1; + int val, i; + + val = tasdevice_clamp(ucontrol->value.integer.value[0], max, invert); + + if (tas_priv->set_global_mode) { + ret = tasdevice_dev_write(tas_priv, tas_priv->ndev, + mc->reg, (unsigned int)val); + if (ret) + dev_err(tas_priv->dev, "%s, error in global mode\n", + __func__); + } + + if (!tas_priv->set_global_mode || ret != 0) { + for (i = 0; i < tas_priv->ndev; i++) { + ret = tasdevice_dev_write(tas_priv, i, mc->reg, + (unsigned int)val); + if (!ret) + continue; + err_cnt++; + dev_err(tas_priv->dev, + "set digital vol err in dev %d\n", i); + } + } + + /* All the devices set error, return 0 */ + return (err_cnt == tas_priv->ndev) ? 0 : 1; + +} +EXPORT_SYMBOL_GPL(tasdevice_digital_putvol); + +int tasdevice_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int ret, val; + + /* Read the primary device as the whole */ + ret = tasdevice_dev_read(tas_priv, 0, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get digital vol error\n", + __func__); + goto out; + } + + val = tasdevice_clamp(val, max, invert); + ucontrol->value.integer.value[0] = val; + +out: + return ret; + +} +EXPORT_SYMBOL_GPL(tasdevice_digital_getvol); + +MODULE_DESCRIPTION("TAS2781 common library"); +MODULE_AUTHOR("Shenghao Ding, TI, <shenghao-ding(a)ti.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c new file mode 100644 index 000000000000..0443014def3d --- /dev/null +++ b/sound/soc/codecs/tas2781-i2c.c @@ -0,0 +1,736 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#include <linux/crc8.h> +#include <linux/firmware.h> +#include <linux/gpio/consumer.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_irq.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tas2781.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +static const struct i2c_device_id tasdevice_id[] = { + { "tas2781", TAS2781 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, tasdevice_id); + +#ifdef CONFIG_OF +static const struct of_device_id tasdevice_of_match[] = { + { .compatible = "ti,tas2781" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tasdevice_of_match); +#endif + +/** + * tas2781_digital_getvol - get the volum control + * @kcontrol: control pointer + * @ucontrol: User data + * Customer Kcontrol for tas2781 is primarily for regmap booking, paging + * depends on internal regmap mechanism. + * tas2781 contains book and page two-level register map, especially + * book switching will set the register BXXP00R7F, after switching to the + * correct book, then leverage the mechanism for paging to access the + * register. + */ +static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_digital_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_digital_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_amp_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_amp_putvol(tas_priv, ucontrol, mc); +} + +static const struct snd_kcontrol_new tas2781_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Digital Volume", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), +}; + +static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int ret = 0; + + if (tas_priv->rcabin.profile_cfg_id != + ucontrol->value.integer.value[0]) { + tas_priv->rcabin.profile_cfg_id = + ucontrol->value.integer.value[0]; + ret = 0; + } + + return ret; +} + +static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)tas_fw->nr_programs; + + return 0; +} + +static int tasdevice_info_configurations( + struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = + snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)tas_fw->nr_configurations - 1; + + return 0; +} + +static int tasdevice_info_profile(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->rcabin.ncfgs - 1; + + return 0; +} + +static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + + return 0; +} + +static int tasdevice_create_control(struct tasdevice_priv *tas_priv) +{ + struct snd_kcontrol_new *prof_ctrls; + int nr_controls = 1; + int mix_index = 0; + int ret; + char *name; + + prof_ctrls = devm_kcalloc(tas_priv->dev, nr_controls, + sizeof(prof_ctrls[0]), GFP_KERNEL); + if (!prof_ctrls) { + ret = -ENOMEM; + goto out; + } + + /* Create a mixer item for selecting the active profile */ + name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + if (!name) { + ret = -ENOMEM; + goto out; + } + scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "spk-profile-id"); + prof_ctrls[mix_index].name = name; + prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + prof_ctrls[mix_index].info = tasdevice_info_profile; + prof_ctrls[mix_index].get = tasdevice_get_profile_id; + prof_ctrls[mix_index].put = tasdevice_set_profile_id; + mix_index++; + + ret = snd_soc_add_component_controls(tas_priv->codec, + prof_ctrls, nr_controls < mix_index ? nr_controls : mix_index); + +out: + return ret; +} + +static int tasdevice_program_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + + return 0; +} + +static int tasdevice_program_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + unsigned int nr_program = ucontrol->value.integer.value[0]; + int ret = 0; + + if (tas_priv->cur_prog != nr_program) { + tas_priv->cur_prog = nr_program; + ret = 1; + } + + return ret; +} + +static int tasdevice_configuration_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + + return 0; +} + +static int tasdevice_configuration_put( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + unsigned int nr_configuration = ucontrol->value.integer.value[0]; + int ret = 0; + + if (tas_priv->cur_conf != nr_configuration) { + tas_priv->cur_conf = nr_configuration; + ret = 1; + } + + return ret; +} + +static int tasdevice_dsp_create_ctrls( + struct tasdevice_priv *tas_priv) +{ + struct snd_kcontrol_new *dsp_ctrls; + char *prog_name, *conf_name; + int nr_controls = 2; + int mix_index = 0; + int ret; + + /* Alloc kcontrol via devm_kzalloc, which don't manually + * free the kcontrol + */ + dsp_ctrls = devm_kcalloc(tas_priv->dev, nr_controls, + sizeof(dsp_ctrls[0]), GFP_KERNEL); + if (!dsp_ctrls) { + ret = -ENOMEM; + goto out; + } + + /* Create a mixer item for selecting the active profile */ + prog_name = devm_kzalloc(tas_priv->dev, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); + conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + if (!prog_name || !conf_name) { + ret = -ENOMEM; + goto out; + } + + scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "spk-prog-id"); + dsp_ctrls[mix_index].name = prog_name; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_programs; + dsp_ctrls[mix_index].get = tasdevice_program_get; + dsp_ctrls[mix_index].put = tasdevice_program_put; + mix_index++; + + scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "spk-conf-id"); + dsp_ctrls[mix_index].name = conf_name; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_configurations; + dsp_ctrls[mix_index].get = tasdevice_configuration_get; + dsp_ctrls[mix_index].put = tasdevice_configuration_put; + mix_index++; + + ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, + nr_controls < mix_index ? nr_controls : mix_index); + +out: + return ret; +} + +static void tasdevice_fw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; + int ret = 0; + int i; + + mutex_lock(&tas_priv->codec_lock); + + ret = tasdevice_rca_parser(tas_priv, fmw); + if (ret) + goto out; + tasdevice_create_control(tas_priv); + + tasdevice_dsp_remove(tas_priv); + tasdevice_calbin_remove(tas_priv); + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", + tas_priv->dev_name); + ret = tasdevice_dsp_parser(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dspfw load %s error\n", + tas_priv->coef_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + tasdevice_dsp_create_ctrls(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; + + /* If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + for (i = 0; i < tas_priv->ndev; i++) { + scnprintf(tas_priv->cal_binaryname[i], 64, "%s_cal_0x%02x.bin", + tas_priv->dev_name, tas_priv->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_priv, + tas_priv->cal_binaryname[i], i); + if (ret != 0) + dev_err(tas_priv->dev, + "%s: load %s error, default will effect\n", + __func__, tas_priv->cal_binaryname[i]); + } + + tasdevice_prmg_calibdata_load(tas_priv, 0); + tas_priv->cur_prog = 0; +out: + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + /*If DSP FW fail, kcontrol won't be created */ + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + } + mutex_unlock(&tas_priv->codec_lock); + if (fmw) + release_firmware(fmw); +} + +static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *codec = snd_soc_dapm_to_component(w->dapm); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int state = 0; + + /* Codec Lock Hold */ + mutex_lock(&tas_priv->codec_lock); + if (event == SND_SOC_DAPM_PRE_PMD) + state = 1; + tasdevice_tuning_switch(tas_priv, state); + /* Codec Lock Release*/ + mutex_unlock(&tas_priv->codec_lock); + + return 0; +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT_E("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, + 0, 0, tasdevice_dapm_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC") +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"SPK", NULL, "ASI"}, + {"OUT", NULL, "SPK"}, + {"ASI OUT", NULL, "DMIC"} +}; + +static int tasdevice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *codec = dai->component; + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int ret = 0; + + if (tas_priv->fw_state != TASDEVICE_DSP_FW_ALL_OK) { + dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + ret = -EINVAL; + } + + return ret; +} + +static int tasdevice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct tasdevice_priv *tas_priv = snd_soc_dai_get_drvdata(dai); + unsigned int slot_width; + unsigned int fsrate; + int bclk_rate; + int rc = 0; + + fsrate = params_rate(params); + switch (fsrate) { + case 48000: + case 44100: + break; + default: + dev_err(tas_priv->dev, "%s: incorrect sample rate = %u\n", + __func__, fsrate); + rc = -EINVAL; + goto out; + } + + slot_width = params_width(params); + switch (slot_width) { + case 16: + case 20: + case 24: + case 32: + break; + default: + dev_err(tas_priv->dev, "%s: incorrect slot width = %u\n", + __func__, slot_width); + rc = -EINVAL; + goto out; + } + + bclk_rate = snd_soc_params_to_bclk(params); + if (bclk_rate < 0) { + dev_err(tas_priv->dev, "%s: incorrect bclk rate = %d\n", + __func__, bclk_rate); + rc = bclk_rate; + goto out; + } + +out: + return rc; +} + +static int tasdevice_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct tasdevice_priv *tas_priv = snd_soc_dai_get_drvdata(codec_dai); + + tas_priv->sysclk = freq; + + return 0; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .startup = tasdevice_startup, + .hw_params = tasdevice_hw_params, + .set_sysclk = tasdevice_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2781_codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_codec_probe(struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + return tascodec_init(tas_priv, codec, tasdevice_fw_ready); +} + +static void tasdevice_deinit(void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + tasdevice_calbin_remove(tas_priv); + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; +} + +static void tasdevice_codec_remove( + struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + tasdevice_deinit(tas_priv); +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_codec_probe, + .remove = tasdevice_codec_remove, + .controls = tas2781_snd_controls, + .num_controls = ARRAY_SIZE(tas2781_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + unsigned int dev_addrs[max_chn]; + int rc, i, ndev = 0; + + if (tas_priv->isacpi) { + ndev = device_property_read_u32_array(&client->dev, + "ti,audio-slots", NULL, 0); + if (ndev <= 0) { + ndev = 1; + dev_addrs[0] = client->addr; + } else { + ndev = (ndev < ARRAY_SIZE(dev_addrs)) + ? ndev : ARRAY_SIZE(dev_addrs); + ndev = device_property_read_u32_array(&client->dev, + "ti,audio-slots", dev_addrs, ndev); + } + + tas_priv->irq_info.irq_gpio = + acpi_dev_gpio_irq_get(ACPI_COMPANION(&client->dev), 0); + } else { + struct device_node *np = tas_priv->dev->of_node; +#ifdef CONFIG_OF + const __be32 *reg, *reg_end; + int len, sw, aw; + + aw = of_n_addr_cells(np); + sw = of_n_size_cells(np); + if (sw == 0) { + reg = (const __be32 *)of_get_property(np, + "reg", &len); + reg_end = reg + len/sizeof(*reg); + ndev = 0; + do { + dev_addrs[ndev] = of_read_number(reg, aw); + reg += aw; + ndev++; + } while (reg < reg_end); + } else { + ndev = 1; + dev_addrs[0] = client->addr; + } +#else + ndev = 1; + dev_addrs[0] = client->addr; +#endif + tas_priv->irq_info.irq_gpio = of_irq_get(np, 0); + } + tas_priv->ndev = ndev; + for (i = 0; i < ndev; i++) + tas_priv->tasdevice[i].dev_addr = dev_addrs[i]; + + if (ndev > 1) { + rc = device_property_read_u32(&client->dev, + "ti,broadcast-addr", + &(tas_priv->glb_addr.dev_addr)); + if (rc) { + dev_err(tas_priv->dev, + "Looking up broadcast-addr failed %d\n", rc); + tas_priv->glb_addr.dev_addr = 0; + } + } + + tas_priv->reset = devm_gpiod_get_optional(&client->dev, + "reset-gpios", GPIOD_OUT_HIGH); + if (IS_ERR(tas_priv->reset)) + dev_err(tas_priv->dev, "%s Can't get reset GPIO\n", + __func__); + + strcpy(tas_priv->dev_name, tasdevice_id[tas_priv->chip_id].name); + + if (gpio_is_valid(tas_priv->irq_info.irq_gpio)) { + rc = gpio_request(tas_priv->irq_info.irq_gpio, + "AUDEV-IRQ"); + if (!rc) { + gpio_direction_input( + tas_priv->irq_info.irq_gpio); + + tas_priv->irq_info.irq = + gpio_to_irq(tas_priv->irq_info.irq_gpio); + } else + dev_err(tas_priv->dev, "%s: GPIO %d request error\n", + __func__, tas_priv->irq_info.irq_gpio); + } else + dev_err(tas_priv->dev, + "Looking up irq-gpio property failed %d\n", + tas_priv->irq_info.irq_gpio); +} + +static int tasdevice_i2c_probe(struct i2c_client *i2c) +{ + const struct i2c_device_id *id = i2c_match_id(tasdevice_id, i2c); + const struct acpi_device_id *acpi_id; + struct tasdevice_priv *tas_priv; + int ret; + + tas_priv = tasdevice_kzalloc(i2c); + if (!tas_priv) + return -ENOMEM; + + if (ACPI_HANDLE(&i2c->dev)) { + acpi_id = acpi_match_device(i2c->dev.driver->acpi_match_table, + &i2c->dev); + if (!acpi_id) { + dev_err(&i2c->dev, "No driver data\n"); + ret = -EINVAL; + goto err; + } + tas_priv->chip_id = acpi_id->driver_data; + tas_priv->isacpi = true; + } else { + tas_priv->chip_id = id ? id->driver_data : 0; + tas_priv->isacpi = false; + } + + tasdevice_parse_dt(tas_priv); + + ret = tasdevice_init(tas_priv); + if (ret) + goto err; + + ret = devm_snd_soc_register_component(tas_priv->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_priv->dev, "%s: codec register error:0x%08x\n", + __func__, ret); + goto err; + } +err: + if (ret < 0) + tasdevice_remove(tas_priv); + return ret; +} + +static void tasdevice_i2c_remove(struct i2c_client *client) +{ + struct tasdevice_priv *tas_priv = i2c_get_clientdata(client); + + tasdevice_remove(tas_priv); +} + +#ifdef CONFIG_ACPI +static const struct acpi_device_id tasdevice_acpi_match[] = { + { "TAS2781", TAS2781 }, + {}, +}; + +MODULE_DEVICE_TABLE(acpi, tasdevice_acpi_match); +#endif + +static struct i2c_driver tasdevice_i2c_driver = { + .driver = { + .name = "tas2781-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tasdevice_of_match), +#ifdef CONFIG_ACPI + .acpi_match_table = ACPI_PTR(tasdevice_acpi_match), +#endif + }, + .probe = tasdevice_i2c_probe, + .remove = tasdevice_i2c_remove, + .id_table = tasdevice_id, +}; + +module_i2c_driver(tasdevice_i2c_driver); + +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_AUTHOR("Kevin Lu <kevin-lu(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2781 Driver"); +MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS(SND_SOC_TAS2781_FMWLIB); -- 2.34.1

2 2

[PATCH v4 1/6] ASoC: tas2781: Add Header file for tas2781 driver
by Shenghao Ding 28 May '23

28 May '23

Create Header file for tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v4: - correct some enums have capitalized Chn, chn to all Changes to be committed: new file: include/sound/tas2781-dsp.h new file: include/sound/tas2781-tlv.h new file: include/sound/tas2781.h --- include/sound/tas2781-dsp.h | 190 ++++++++++++++++++++++++++++++++++++ include/sound/tas2781-tlv.h | 22 +++++ include/sound/tas2781.h | 182 ++++++++++++++++++++++++++++++++++ 3 files changed, 394 insertions(+) create mode 100644 include/sound/tas2781-dsp.h create mode 100644 include/sound/tas2781-tlv.h create mode 100644 include/sound/tas2781.h diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h new file mode 100644 index 000000000000..b05f20bc4a1f --- /dev/null +++ b/include/sound/tas2781-dsp.h @@ -0,0 +1,190 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TASDEVICE_DSP_H__ +#define __TASDEVICE_DSP_H__ + +#define MAIN_ALL_DEVICES 0x0d +#define MAIN_DEVICE_A 0x01 +#define MAIN_DEVICE_B 0x08 +#define MAIN_DEVICE_C 0x10 +#define MAIN_DEVICE_D 0x14 +#define COEFF_DEVICE_A 0x03 +#define COEFF_DEVICE_B 0x0a +#define COEFF_DEVICE_C 0x11 +#define COEFF_DEVICE_D 0x15 +#define PRE_DEVICE_A 0x04 +#define PRE_DEVICE_B 0x0b +#define PRE_DEVICE_C 0x12 +#define PRE_DEVICE_D 0x16 + +#define PPC3_VERSION 0x4100 +#define PPC3_VERSION_TAS2781 0x14c00 +#define RCA_CONFIGID_BYPASS_ALL 0 +#define TASDEVICE_DEVICE_SUM 8 +#define TASDEVICE_CONFIG_SUM 64 + +enum channel { + top_left_Chn, + top_right_chn, + bottom_left_Chn, + bottom_right_chn, + max_chn, +}; + +enum tasdevice_dsp_dev_idx { + TASDEVICE_DSP_TAS_2555 = 0, + TASDEVICE_DSP_TAS_2555_STEREO, + TASDEVICE_DSP_TAS_2557_MONO, + TASDEVICE_DSP_TAS_2557_DUAL_MONO, + TASDEVICE_DSP_TAS_2559, + TASDEVICE_DSP_TAS_2563, + TASDEVICE_DSP_TAS_2563_DUAL_MONO = 7, + TASDEVICE_DSP_TAS_2563_QUAD, + TASDEVICE_DSP_TAS_2563_21, + TASDEVICE_DSP_TAS_2781, + TASDEVICE_DSP_TAS_2781_DUAL_MONO, + TASDEVICE_DSP_TAS_2781_21, + TASDEVICE_DSP_TAS_2781_QUAD, + TASDEVICE_DSP_TAS_MAX_DEVICE +}; + +struct tasdevice_fw_fixed_hdr { + unsigned int fwsize; + unsigned int ppcver; + unsigned int drv_ver; +}; + +struct tasdevice_dspfw_hdr { + struct tasdevice_fw_fixed_hdr fixed_hdr; + unsigned short device_family; + unsigned short device; + unsigned char ndev; +}; + +struct tasdev_blk { + int nr_retry; + unsigned int type; + unsigned char is_pchksum_present; + unsigned char pchksum; + unsigned char is_ychksum_present; + unsigned char ychksum; + unsigned int nr_cmds; + unsigned int blk_size; + unsigned int nr_subblocks; + unsigned char *data; +}; + +struct tasdevice_data { + char name[64]; + unsigned int nr_blk; + struct tasdev_blk *dev_blks; +}; + +struct tasdevice_prog { + unsigned int prog_size; + struct tasdevice_data dev_data; +}; + +struct tasdevice_config { + unsigned int cfg_size; + char name[64]; + struct tasdevice_data dev_data; +}; + +struct tasdevice_calibration { + struct tasdevice_data dev_data; +}; + +struct tasdevice_fw { + struct tasdevice_dspfw_hdr fw_hdr; + unsigned short nr_programs; + struct tasdevice_prog *programs; + unsigned short nr_configurations; + struct tasdevice_config *configs; + unsigned short nr_calibrations; + struct tasdevice_calibration *calibrations; + struct device *dev; +}; + +enum tasdevice_dsp_fw_state { + TASDEVICE_DSP_FW_NONE = 0, + TASDEVICE_DSP_FW_PENDING, + TASDEVICE_DSP_FW_FAIL, + TASDEVICE_DSP_FW_ALL_OK, +}; + +enum tasdevice_bin_blk_type { + TASDEVICE_BIN_BLK_COEFF = 1, + TASDEVICE_BIN_BLK_POST_POWER_UP, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN, + TASDEVICE_BIN_BLK_PRE_POWER_UP, + TASDEVICE_BIN_BLK_POST_SHUTDOWN +}; + +struct tasdevice_rca_hdr { + unsigned int img_sz; + unsigned int checksum; + unsigned int binary_version_num; + unsigned int drv_fw_version; + unsigned char plat_type; + unsigned char dev_family; + unsigned char reserve; + unsigned char ndev; + unsigned char devs[TASDEVICE_DEVICE_SUM]; + unsigned int nconfig; + unsigned int config_size[TASDEVICE_CONFIG_SUM]; +}; + +struct tasdev_blk_data { + unsigned char dev_idx; + unsigned char block_type; + unsigned short yram_checksum; + unsigned int block_size; + unsigned int n_subblks; + unsigned char *regdata; +}; + +struct tasdevice_config_info { + unsigned int nblocks; + unsigned int real_nblocks; + unsigned char active_dev; + struct tasdev_blk_data **blk_data; +}; + +struct tasdevice_rca { + struct tasdevice_rca_hdr fw_hdr; + int ncfgs; + struct tasdevice_config_info **cfg_info; + int profile_cfg_id; +}; + +void tasdevice_select_cfg_blk(void *context, int conf_no, + unsigned char block_type); +void tasdevice_config_info_remove(void *context); +void tasdevice_dsp_remove(void *context); +int tasdevice_dsp_parser(void *context); +int tasdevice_rca_parser(void *context, const struct firmware *fmw); +void tasdevice_dsp_remove(void *context); +void tasdevice_calbin_remove(void *context); +int tasdevice_select_tuningprm_cfg(void *context, int prm, + int cfg_no, int rca_conf_no); +int tasdevice_prmg_load(void *context, int prm_no); +int tasdevice_prmg_calibdata_load(void *context, int prm_no); +void tasdevice_tuning_switch(void *context, int state); +int tas2781_load_calibration(void *context, char *file_name, + enum channel i); + +#endif diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h new file mode 100644 index 000000000000..f4310dce655a --- /dev/null +++ b/include/sound/tas2781-tlv.h @@ -0,0 +1,22 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TAS2781_TLV_H__ +#define __TAS2781_TLV_H__ + +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); +static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); + +#endif /* __TAS2781_LIB_H__ */ diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h new file mode 100644 index 000000000000..399bb8f9b54a --- /dev/null +++ b/include/sound/tas2781.h @@ -0,0 +1,182 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TAS2781_H__ +#define __TAS2781_H__ + +#include <linux/kernel.h> +#include "tas2781-dsp.h" + +/* version number */ +#define TAS2781_DRV_VER 1 +#define SMARTAMP_MODULE_NAME "tas2781" +#define TAS2781_GLOBAL_ADDR 0x40 +#define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_88200) +#define TASDEVICE_MAX_CHANNELS 8 + +#define TASDEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/*PAGE Control Register (available in page0 of each book) */ +#define TASDEVICE_PAGE_SELECT 0x00 +#define TASDEVICE_BOOKCTL_PAGE 0x00 +#define TASDEVICE_BOOKCTL_REG 127 +#define TASDEVICE_BOOK_ID(reg) (reg / (256 * 128)) +#define TASDEVICE_PAGE_ID(reg) ((reg % (256 * 128)) / 128) +#define TASDEVICE_PAGE_REG(reg) ((reg % (256 * 128)) % 128) +#define TASDEVICE_PGRG(reg) (reg % (256 * 128)) +#define TASDEVICE_REG(book, page, reg) (((book * 256 * 128) + \ + (page * 128)) + reg) + +/*Software Reset */ +#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) +#define TAS2781_REG_SWRESET_RESET BIT(0) + +/* Enable Global addresses */ +#define TAS2781_MISC_CFG2 TASDEVICE_REG(0x0, 0X0, 0x07) +#define TAS2781_GLOBAL_ADDR_MASK BIT(1) +#define TAS2781_GLOBAL_ADDR_ENABLE BIT(1) + +/*I2C Checksum */ +#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) + +/* Volume control */ +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) + +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + +enum audio_device { + TAS2781 = 0, +}; + +enum device_catlog_id { + LENOVO = 0, + OTHERS +}; + +struct tasdevice { + struct tasdevice_fw *cali_data_fmw; + unsigned int dev_addr; + unsigned int err_code; + unsigned char cur_book; + short cur_prog; + short cur_conf; + bool is_loading; + bool is_loaderr; +}; + +/* + * This item is used to store the generic i2c address of + * all the tas2781 devices for I2C broadcast during the multi-device + * writes, useless in mono case. + */ +struct global_addr { + int ref_cnt; + unsigned int dev_addr; + unsigned char cur_book; +}; + +struct tasdevice_irqinfo { + int irq_gpio; + int irq; +}; + +struct calidata { + unsigned char *data; + unsigned long total_sz; +}; + +struct tasdevice_priv { + struct tasdevice tasdevice[max_chn]; + struct tasdevice_irqinfo irq_info; + struct global_addr glb_addr; + struct tasdevice_rca rcabin; + struct tasdevice_fw *fmw; + struct regmap *regmap; + struct mutex codec_lock; + struct calidata cali_data; + struct gpio_desc *reset; + struct device *dev; + struct tm tm; + + enum device_catlog_id catlog_id; + unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; + unsigned char cal_binaryname[max_chn][64]; + unsigned char coef_binaryname[64]; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + unsigned char ndev; + unsigned int magic_num; + unsigned int chip_id; + unsigned int sysclk; + const char *acpi_subsystem_id; + int index; + int cur_prog; + int cur_conf; + int fw_state; + void *client; + void *codec; + bool isacpi; + bool playback_started; + void (*set_global_mode)(struct tasdevice_priv *tas_priv); + int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, int offset); + int (*fw_parse_program_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block); +}; + +void tas2781_reset(struct tasdevice_priv *tas_dev); +int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, + void (*cont)(const struct firmware *fw, void *context)); +struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c); +int tasdevice_init(struct tasdevice_priv *tas_priv); +void tasdevice_remove(struct tasdevice_priv *tas_priv); +int tasdevice_dev_read(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned int *value); +int tasdevice_dev_write(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned int value); +int tasdevice_dev_bulk_write( + struct tasdevice_priv *tas_priv, enum channel chn, + unsigned int reg, unsigned char *p_data, unsigned int n_length); +int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned char *p_data, + unsigned int n_length); +int tasdevice_dev_update_bits( + struct tasdevice_priv *tasdevice, enum channel chn, + unsigned int reg, unsigned int mask, unsigned int value); +int tasdevice_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); + +#endif /* __TAS2781_H__ */ -- 2.34.1

2 1

[PATCH v4 4/6] ALSA: hda/tas2781: Add tas2781 HDA driver
by Shenghao Ding 28 May '23

28 May '23

Add tas2781 configs for Lenovo Laptops. All of the tas2781s in the laptop will be aggregated as one codec. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v4: - Add tiwai(a)suse.de into Cc list - remove useless index - combine ALC287_FIXUP_TAS2781_I2C_2 and ALC287_FIXUP_TAS2781_I2C_4 together as ALC287_FIXUP_TAS2781_I2C, The code view all the tas2781s in the laptop as one instance. Changes to be committed: modified: sound/pci/hda/patch_realtek.c --- sound/pci/hda/patch_realtek.c | 98 +++++++++++++++++++++++++++++++++-- 1 file changed, 95 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7b5f194513c7..7a333443296f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6705,7 +6705,7 @@ static void comp_generic_playback_hook(struct hda_pcm_stream *hinfo, struct hda_ } } -struct cs35l41_dev_name { +struct scodec_dev_name { const char *bus; const char *hid; int index; @@ -6714,7 +6714,7 @@ struct cs35l41_dev_name { /* match the device name in a slightly relaxed manner */ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) { - struct cs35l41_dev_name *p = data; + struct scodec_dev_name *p = data; const char *d = dev_name(dev); int n = strlen(p->bus); char tmp[32]; @@ -6730,12 +6730,32 @@ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) return !strcmp(d + n, tmp); } +static int comp_match_tas2781_dev_name(struct device *dev, + void *data) +{ + struct scodec_dev_name *p = data; + const char *d = dev_name(dev); + int n = strlen(p->bus); + char tmp[32]; + + /* check the bus name */ + if (strncmp(d, p->bus, n)) + return 0; + /* skip the bus number */ + if (isdigit(d[n])) + n++; + /* the rest must be exact matching */ + snprintf(tmp, sizeof(tmp), "-%s:00", p->hid); + + return !strcmp(d + n, tmp); +} + static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char *bus, const char *hid, int count) { struct device *dev = hda_codec_dev(cdc); struct alc_spec *spec = cdc->spec; - struct cs35l41_dev_name *rec; + struct scodec_dev_name *rec; int ret, i; switch (action) { @@ -6760,6 +6780,38 @@ static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char } } +static void tas2781_generic_fixup(struct hda_codec *cdc, int action, + const char *bus, const char *hid) +{ + struct device *dev = hda_codec_dev(cdc); + struct alc_spec *spec = cdc->spec; + struct scodec_dev_name *rec; + int ret; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + rec = devm_kmalloc(dev, sizeof(*rec), GFP_KERNEL); + if (!rec) + return; + rec->bus = bus; + rec->hid = hid; + rec->index = 0; + spec->comps[0].codec = cdc; + component_match_add(dev, &spec->match, + comp_match_tas2781_dev_name, rec); + ret = component_master_add_with_match(dev, &comp_master_ops, + spec->match); + if (ret) + codec_err(cdc, + "Fail to register component aggregator %d\n", + ret); + else + spec->gen.pcm_playback_hook = + comp_generic_playback_hook; + break; + } +} + static void cs35l41_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { cs35l41_generic_fixup(cdc, action, "i2c", "CSC3551", 2); @@ -6787,6 +6839,12 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st cs35l41_generic_fixup(cdc, action, "i2c", "CLSA0101", 2); } +static void tas2781_fixup_i2c(struct hda_codec *cdc, + const struct hda_fixup *fix, int action) +{ + tas2781_generic_fixup(cdc, action, "i2c", "TIAS2781"); +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -7203,6 +7261,7 @@ enum { ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, ALC236_FIXUP_DELL_DUAL_CODECS, + ALC287_FIXUP_TAS2781_I2C, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9207,6 +9266,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, }, + [ALC287_FIXUP_TAS2781_I2C] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9754,6 +9819,33 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual powe mode2 YC", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bf, "Yoga S980-14.5 proX LX Dual", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38c3, "Y980 DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", + ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.34.1

1 0

[PATCH v4 2/6] ASoC: tas2781: firmware lib
by Shenghao Ding 28 May '23

28 May '23

Create tas2781 firmware lib. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v4: - fixed issue | Reported-by: kernel test robot <lkp(a)intel.com> | Link: https://lore.kernel.org/oe-kbuild-all/202305022033.LiI7Ojm4-lkp@intel.com/ - Move back to codecs folder, this firmware lib is special for the audio cases - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202305221045.K3Bx6bts-lkp@intel.com/ Changes to be committed: new file: sound/soc/codecs/tas2781-fmwlib.c --- sound/soc/codecs/tas2781-fmwlib.c | 2439 +++++++++++++++++++++++++++++ 1 file changed, 2439 insertions(+) create mode 100644 sound/soc/codecs/tas2781-fmwlib.c diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c new file mode 100644 index 000000000000..a17ed68ad7db --- /dev/null +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -0,0 +1,2439 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// tasdevice-fmw.c -- TASDEVICE firmware support +// +// Copyright 2023 Texas Instruments, Inc. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> + +#include <linux/crc8.h> +#include <linux/firmware.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_irq.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/tas2781.h> + + +#define ERROR_PRAM_CRCCHK 0x0000000 +#define ERROR_YRAM_CRCCHK 0x0000001 +#define PPC_DRIVER_CRCCHK 0x00000200 + +#define TAS2781_SA_COEFF_SWAP_REG TASDEVICE_REG(0, 0x35, 0x2c) +#define TAS2781_YRAM_BOOK1 140 +#define TAS2781_YRAM1_PAGE 42 +#define TAS2781_YRAM1_START_REG 88 + +#define TAS2781_YRAM2_START_PAGE 43 +#define TAS2781_YRAM2_END_PAGE 49 +#define TAS2781_YRAM2_START_REG 8 +#define TAS2781_YRAM2_END_REG 127 + +#define TAS2781_YRAM3_PAGE 50 +#define TAS2781_YRAM3_START_REG 8 +#define TAS2781_YRAM3_END_REG 27 + +/*should not include B0_P53_R44-R47 */ +#define TAS2781_YRAM_BOOK2 0 +#define TAS2781_YRAM4_START_PAGE 50 +#define TAS2781_YRAM4_END_PAGE 60 + +#define TAS2781_YRAM5_PAGE 61 +#define TAS2781_YRAM5_START_REG TAS2781_YRAM3_START_REG +#define TAS2781_YRAM5_END_REG TAS2781_YRAM3_END_REG + +#define TASDEVICE_MAXPROGRAM_NUM_KERNEL 5 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS 64 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL 10 +#define MAIN_ALL_DEVICES_1X 0x01 +#define MAIN_DEVICE_A_1X 0x02 +#define MAIN_DEVICE_B_1X 0x03 +#define MAIN_DEVICE_C_1X 0x04 +#define MAIN_DEVICE_D_1X 0x05 +#define COEFF_DEVICE_A_1X 0x12 +#define COEFF_DEVICE_B_1X 0x13 +#define COEFF_DEVICE_C_1X 0x14 +#define COEFF_DEVICE_D_1X 0x15 +#define PRE_DEVICE_A_1X 0x22 +#define PRE_DEVICE_B_1X 0x23 +#define PRE_DEVICE_C_1X 0x24 +#define PRE_DEVICE_D_1X 0x25 +#define PRE_SOFTWARE_RESET_DEVICE_A 0x41 +#define PRE_SOFTWARE_RESET_DEVICE_B 0x42 +#define PRE_SOFTWARE_RESET_DEVICE_C 0x43 +#define PRE_SOFTWARE_RESET_DEVICE_D 0x44 +#define POST_SOFTWARE_RESET_DEVICE_A 0x45 +#define POST_SOFTWARE_RESET_DEVICE_B 0x46 +#define POST_SOFTWARE_RESET_DEVICE_C 0x47 +#define POST_SOFTWARE_RESET_DEVICE_D 0x48 + +struct tas_crc { + unsigned char offset; + unsigned char len; +}; + +static const char deviceNumber[TASDEVICE_DSP_TAS_MAX_DEVICE] = { + 1, 2, 1, 2, 1, 1, 0, 2, 4, 3, 1, 2, 3, 4 +}; + +static struct tasdevice_config_info *tasdevice_add_config( + struct tasdevice_priv *tas_priv, unsigned char *config_data, + unsigned int config_size, int *status) +{ + struct tasdevice_config_info *cfg_info; + struct tasdev_blk_data **bk_da; + unsigned int config_offset = 0; + unsigned int i; + + /* In most projects are many audio cases, such as music, handfree, + * receiver, games, audio-to-haptics, PMIC record, bypass mode, + * portrait, landscape, etc. Even in multiple audios, one or + * two of the chips will work for the special case, such as + * ultrasonic application. In order to support these variable-numbers + * of audio cases, flexible configs have been introduced in the + * dsp firmware. + */ + cfg_info = kzalloc(sizeof(struct tasdevice_config_info), GFP_KERNEL); + if (!cfg_info) { + *status = -ENOMEM; + goto out; + } + + if (tas_priv->rcabin.fw_hdr.binary_version_num >= 0x105) { + if (config_offset + 64 > (int)config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add conf: Out of boundary\n"); + goto out; + } + config_offset += 64; + } + + if (config_offset + 4 > (int)config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add config: Out of boundary\n"); + goto out; + } + + /* convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + cfg_info->nblocks = + be32_to_cpup((__be32 *)&config_data[config_offset]); + config_offset += 4; + + /* Several kinds of dsp/algorithm firmwares can run on tas2781, + * the number and size of blk are not fixed and different among + * these firmwares. + */ + bk_da = cfg_info->blk_data = kcalloc(cfg_info->nblocks, + sizeof(struct tasdev_blk_data *), GFP_KERNEL); + if (!bk_da) { + *status = -ENOMEM; + goto out; + } + cfg_info->real_nblocks = 0; + for (i = 0; i < cfg_info->nblocks; i++) { + if (config_offset + 12 > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d nblocks = %u!\n", + __func__, i, cfg_info->nblocks); + break; + } + bk_da[i] = kzalloc(sizeof(struct tasdev_blk_data), GFP_KERNEL); + if (!bk_da[i]) { + *status = -ENOMEM; + break; + } + + bk_da[i]->dev_idx = config_data[config_offset]; + config_offset++; + + bk_da[i]->block_type = config_data[config_offset]; + config_offset++; + + if (bk_da[i]->block_type == TASDEVICE_BIN_BLK_PRE_POWER_UP) { + if (bk_da[i]->dev_idx == 0) + cfg_info->active_dev = 1; + else + cfg_info->active_dev |= 1 << + (bk_da[i]->dev_idx - 1); + + } + bk_da[i]->yram_checksum = + be16_to_cpup((__be16 *)&config_data[config_offset]); + config_offset += 2; + bk_da[i]->block_size = + be32_to_cpup((__be32 *)&config_data[config_offset]); + config_offset += 4; + + bk_da[i]->n_subblks = + be32_to_cpup((__be32 *)&config_data[config_offset]); + + config_offset += 4; + + if (config_offset + bk_da[i]->block_size > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d blks = %u!\n", + __func__, i, cfg_info->nblocks); + break; + } + /* instead of kzalloc+memcpy */ + bk_da[i]->regdata = kmemdup(&config_data[config_offset], + bk_da[i]->block_size, GFP_KERNEL); + if (!bk_da[i]->regdata) { + *status = -ENOMEM; + goto out; + } + + config_offset += bk_da[i]->block_size; + cfg_info->real_nblocks += 1; + } + +out: + return cfg_info; +} + +int tasdevice_rca_parser(void *context, const struct firmware *fmw) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_config_info **cfg_info; + struct tasdevice_rca_hdr *fw_hdr; + struct tasdevice_rca *rca; + unsigned int total_config_sz = 0; + unsigned char *buf; + int offset = 0; + int ret = 0; + int i; + + rca = &(tas_priv->rcabin); + fw_hdr = &(rca->fw_hdr); + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "Failed to read %s\n", + tas_priv->rca_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + ret = -EINVAL; + goto out; + } + buf = (unsigned char *)fmw->data; + + fw_hdr->img_sz = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + if (fw_hdr->img_sz != fmw->size) { + dev_err(tas_priv->dev, + "File size not match, %d %u", (int)fmw->size, + fw_hdr->img_sz); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + ret = -EINVAL; + goto out; + } + + fw_hdr->checksum = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + fw_hdr->binary_version_num = be32_to_cpup((__be32 *)&buf[offset]); + if (fw_hdr->binary_version_num < 0x103) { + dev_err(tas_priv->dev, "File version 0x%04x is too low", + fw_hdr->binary_version_num); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + ret = -EINVAL; + goto out; + } + offset += 4; + fw_hdr->drv_fw_version = be32_to_cpup((__be32 *)&buf[offset]); + offset += 8; + fw_hdr->plat_type = buf[offset]; + offset += 1; + fw_hdr->dev_family = buf[offset]; + offset += 1; + fw_hdr->reserve = buf[offset]; + offset += 1; + fw_hdr->ndev = buf[offset]; + offset += 1; + if (fw_hdr->ndev != tas_priv->ndev) { + dev_err(tas_priv->dev, + "ndev(%u) in rcabin mismatch ndev(%u) in DTS\n", + fw_hdr->ndev, tas_priv->ndev); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + ret = -EINVAL; + goto out; + } + if (offset + TASDEVICE_DEVICE_SUM > fw_hdr->img_sz) { + dev_err(tas_priv->dev, "rca_ready: Out of boundary!\n"); + ret = -EINVAL; + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + + for (i = 0; i < TASDEVICE_DEVICE_SUM; i++, offset++) + fw_hdr->devs[i] = buf[offset]; + + fw_hdr->nconfig = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + + for (i = 0; i < TASDEVICE_CONFIG_SUM; i++) { + fw_hdr->config_size[i] = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + total_config_sz += fw_hdr->config_size[i]; + } + + if (fw_hdr->img_sz - total_config_sz != (unsigned int)offset) { + dev_err(tas_priv->dev, "Bin file error!\n"); + ret = -EINVAL; + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + + cfg_info = kcalloc(fw_hdr->nconfig, sizeof(*cfg_info), GFP_KERNEL); + if (!cfg_info) { + ret = -ENOMEM; + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + rca->cfg_info = cfg_info; + rca->ncfgs = 0; + for (i = 0; i < (int)fw_hdr->nconfig; i++) { + rca->ncfgs += 1; + cfg_info[i] = tasdevice_add_config(tas_priv, &buf[offset], + fw_hdr->config_size[i], &ret); + if (ret) { + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + offset += (int)fw_hdr->config_size[i]; + } +out: + return ret; +} +EXPORT_SYMBOL_NS_GPL(tasdevice_rca_parser, SND_SOC_TAS2781_FMWLIB); + +static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + + if (offset + 16 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + + /* convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + block->type = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + block->is_pchksum_present = data[offset]; + offset++; + + block->pchksum = data[offset]; + offset++; + + block->is_ychksum_present = data[offset]; + offset++; + + block->ychksum = data[offset]; + offset++; + + block->blk_size = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + block->nr_subblocks = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + if (offset + block->blk_size > fmw->size) { + dev_err(tas_fmw->dev, "%s: nSublocks error\n", __func__); + offset = -EINVAL; + goto out; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], block->blk_size, GFP_KERNEL); + if (!block->data) { + offset = -ENOMEM; + goto out; + } + offset += block->blk_size; + +out: + return offset; +} + +static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = fmw->data; + struct tasdev_blk *blk; + unsigned int i; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + img_data->nr_blk = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(struct tasdev_blk), GFP_KERNEL); + if (!img_data->dev_blks) { + offset = -ENOMEM; + goto out; + } + + for (i = 0; i < img_data->nr_blk; i++) { + blk = &(img_data->dev_blks[i]); + offset = fw_parse_block_data_kernel(tas_fmw, blk, fmw, offset); + if (offset < 0) { + offset = -EINVAL; + break; + } + } + +out: + return offset; +} + +static int fw_parse_program_data_kernel( + struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + struct tasdevice_prog *program; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &(tas_fmw->programs[i]); + if (offset + 72 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + offset = -EINVAL; + goto out; + } + /*skip 72 unused byts*/ + offset += 72; + + offset = fw_parse_data_kernel(tas_fmw, &(program->dev_data), + fmw, offset); + if (offset < 0) + goto out; + } + +out: + return offset; +} + +static int fw_parse_configuration_data_kernel( + struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + struct tasdevice_config *config; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &(tas_fmw->configs[i]); + if (offset + 80 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + offset = -EINVAL; + goto out; + } + memcpy(config->name, &data[offset], 64); + /*skip extra 16 bytes*/ + offset += 80; + + offset = fw_parse_data_kernel(tas_fmw, &(config->dev_data), + fmw, offset); + if (offset < 0) + goto out; + } + +out: + return offset; +} + +static int fw_parse_variable_header_kernel( + struct tasdevice_priv *tas_priv, const struct firmware *fmw, + int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + struct tasdevice_prog *program; + struct tasdevice_config *config; + const unsigned char *buf = fmw->data; + unsigned short max_confs; + unsigned int i; + + if (offset + 12 + 4 * TASDEVICE_MAXPROGRAM_NUM_KERNEL > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + fw_hdr->device_family = be16_to_cpup((__be16 *)&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s:not TAS device\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 2; + fw_hdr->device = be16_to_cpup((__be16 *)&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + offset = -EINVAL; + goto out; + } + offset += 2; + fw_hdr->ndev = deviceNumber[fw_hdr->device]; + + if (fw_hdr->ndev != tas_priv->ndev) { + dev_err(tas_priv->dev, + "%s: ndev(%u) in dspbin mismatch ndev(%u) in DTS\n", + __func__, fw_hdr->ndev, tas_priv->ndev); + offset = -EINVAL; + goto out; + } + + tas_fmw->nr_programs = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + + if (tas_fmw->nr_programs == 0 || tas_fmw->nr_programs > + TASDEVICE_MAXPROGRAM_NUM_KERNEL) { + dev_err(tas_priv->dev, "mnPrograms is invalid\n"); + offset = -EINVAL; + goto out; + } + + tas_fmw->programs = kcalloc(tas_fmw->nr_programs, + sizeof(struct tasdevice_prog), GFP_KERNEL); + if (!tas_fmw->programs) { + offset = -ENOMEM; + goto out; + } + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &(tas_fmw->programs[i]); + program->prog_size = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + } + + /* Skip the unused prog_size */ + offset += 4 * (TASDEVICE_MAXPROGRAM_NUM_KERNEL - tas_fmw->nr_programs); + + tas_fmw->nr_configurations = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + + /* The max number of config in firmware greater than 4 pieces of + * tas2781s is different from the one lower than 4 pieces of + * tas2781s. + */ + max_confs = (fw_hdr->ndev >= 4) ? + TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS : + TASDEVICE_MAXCONFIG_NUM_KERNEL; + if (tas_fmw->nr_configurations == 0 || + tas_fmw->nr_configurations > max_confs) { + dev_err(tas_priv->dev, "%s: Conf is invalid\n", __func__); + offset = -EINVAL; + goto out; + } + + if (offset + 4 * max_confs > fmw->size) { + dev_err(tas_priv->dev, "%s: mpConfigurations err\n", __func__); + offset = -EINVAL; + goto out; + } + + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(struct tasdevice_config), GFP_KERNEL); + if (!tas_fmw->configs) { + offset = -ENOMEM; + goto out; + } + + for (i = 0; i < tas_fmw->nr_programs; i++) { + config = &(tas_fmw->configs[i]); + config->cfg_size = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + } + + /* Skip the unused configs */ + offset += 4 * (max_confs - tas_fmw->nr_programs); + +out: + return offset; +} + +static int tasdevice_process_block(void *context, unsigned char *data, + unsigned char dev_idx, int sublocksize) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; + int subblk_offset, chn, chnend, rc; + unsigned char subblk_typ = data[1]; + int blktyp = dev_idx & 0xC0; + int idx = dev_idx & 0x3F; + bool is_err = false; + + if (idx) { + chn = idx - 1; + chnend = idx; + } else { + if (tas_priv->set_global_mode) { + chn = tas_priv->ndev; + chnend = tas_priv->ndev + 1; + } else { + chn = 0; + chnend = tas_priv->ndev; + } + } + + for (; chn < chnend; chn++) { + if (!tas_priv->set_global_mode && + tas_priv->tasdevice[chn].is_loading == false) + continue; + + is_err = false; + subblk_offset = 2; + switch (subblk_typ) { + case TASDEVICE_CMD_SING_W: { + int i; + unsigned short len = be16_to_cpup((__be16 *)&data[2]); + + subblk_offset += 2; + if (subblk_offset + 4 * len > sublocksize) { + dev_err(tas_priv->dev, + "process_block: Out of boundary\n"); + is_err = true; + break; + } + + for (i = 0; i < len; i++) { + rc = tasdevice_dev_write(tas_priv, chn, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + data[subblk_offset + 3]); + if (rc < 0) { + is_err = true; + dev_err(tas_priv->dev, + "process_block: single write error\n"); + } + subblk_offset += 4; + } + } + break; + case TASDEVICE_CMD_BURST: { + unsigned short len = be16_to_cpup((__be16 *)&data[2]); + + subblk_offset += 2; + if (subblk_offset + 4 + len > sublocksize) { + dev_err(tas_priv->dev, + "%s: BST Out of boundary\n", + __func__); + is_err = true; + break; + } + if (len % 4) { + dev_err(tas_priv->dev, + "%s:Bst-len(%u)not div by 4\n", + __func__, len); + break; + } + + rc = tasdevice_dev_bulk_write(tas_priv, chn, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + &(data[subblk_offset + 4]), len); + if (rc < 0) { + is_err = true; + dev_err(tas_priv->dev, + "%s: bulk_write error = %d\n", + __func__, rc); + } + subblk_offset += (len + 4); + } + break; + case TASDEVICE_CMD_DELAY: { + unsigned int sleep_time = 0; + + if (subblk_offset + 2 > sublocksize) { + dev_err(tas_priv->dev, + "%s: delay Out of boundary\n", + __func__); + is_err = true; + break; + } + sleep_time = be16_to_cpup((__be16 *)&data[2]) * 1000; + usleep_range(sleep_time, sleep_time + 50); + subblk_offset += 2; + } + break; + case TASDEVICE_CMD_FIELD_W: + if (subblk_offset + 6 > sublocksize) { + dev_err(tas_priv->dev, + "%s: bit write Out of boundary\n", + __func__); + is_err = true; + break; + } + rc = tasdevice_dev_update_bits(tas_priv, chn, + TASDEVICE_REG(data[subblk_offset + 2], + data[subblk_offset + 3], + data[subblk_offset + 4]), + data[subblk_offset + 1], + data[subblk_offset + 5]); + if (rc < 0) { + is_err = true; + dev_err(tas_priv->dev, + "%s: update_bits error = %d\n", + __func__, rc); + } + subblk_offset += 6; + break; + default: + break; + } + if (is_err == true && blktyp != 0) { + if (blktyp == 0x80) { + tas_priv->tasdevice[chn].cur_prog = -1; + tas_priv->tasdevice[chn].cur_conf = -1; + } else + tas_priv->tasdevice[chn].cur_conf = -1; + } + } + + return subblk_offset; +} + +void tasdevice_select_cfg_blk(void *pContext, int conf_no, + unsigned char block_type) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) pContext; + struct tasdevice_rca *rca = &(tas_priv->rcabin); + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdev_blk_data **blk_data; + int j, k, chn, chnend; + + if (conf_no >= rca->ncfgs || conf_no < 0 || !cfg_info) { + dev_err(tas_priv->dev, "conf_no should be not more than %u\n", + rca->ncfgs); + return; + } + blk_data = cfg_info[conf_no]->blk_data; + + for (j = 0; j < (int)cfg_info[conf_no]->real_nblocks; j++) { + unsigned int length = 0, rc = 0; + + if (block_type > 5 || block_type < 2) { + dev_err(tas_priv->dev, + "block_type should be in range from 2 to 5\n"); + break; + } + if (block_type != blk_data[j]->block_type) + continue; + + for (k = 0; k < (int)blk_data[j]->n_subblks; k++) { + if (blk_data[j]->dev_idx) { + chn = blk_data[j]->dev_idx - 1; + chnend = blk_data[j]->dev_idx; + } else { + chn = 0; + chnend = tas_priv->ndev; + } + for (; chn < chnend; chn++) + tas_priv->tasdevice[chn].is_loading = true; + + rc = tasdevice_process_block(tas_priv, + blk_data[j]->regdata + length, + blk_data[j]->dev_idx, + blk_data[j]->block_size - length); + length += rc; + if (blk_data[j]->block_size < length) { + dev_err(tas_priv->dev, + "%s: %u %u out of boundary\n", + __func__, length, + blk_data[j]->block_size); + break; + } + } + if (length != blk_data[j]->block_size) + dev_err(tas_priv->dev, "%s: %u %u size is not same\n", + __func__, length, blk_data[j]->block_size); + } +} +EXPORT_SYMBOL_NS_GPL(tasdevice_select_cfg_blk, SND_SOC_TAS2781_FMWLIB); + +static int tasdevice_load_block_kernel( + struct tasdevice_priv *tasdevice, struct tasdev_blk *block) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &(tasdevice->fmw->fw_hdr); + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &(fw_hdr->fixed_hdr); + const unsigned int blk_size = block->blk_size; + unsigned int i, length; + unsigned char *data = block->data; + unsigned char dev_idx = 0; + + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + switch (block->type) { + case MAIN_ALL_DEVICES_1X: + dev_idx = 0x80; + break; + case MAIN_DEVICE_A_1X: + dev_idx = 0x81; + break; + case COEFF_DEVICE_A_1X: + case PRE_DEVICE_A_1X: + case PRE_SOFTWARE_RESET_DEVICE_A: + case POST_SOFTWARE_RESET_DEVICE_A: + dev_idx = 0xC1; + break; + case MAIN_DEVICE_B_1X: + dev_idx = 0x82; + break; + case COEFF_DEVICE_B_1X: + case PRE_DEVICE_B_1X: + case PRE_SOFTWARE_RESET_DEVICE_B: + case POST_SOFTWARE_RESET_DEVICE_B: + dev_idx = 0xC2; + break; + case MAIN_DEVICE_C_1X: + dev_idx = 0x83; + break; + case COEFF_DEVICE_C_1X: + case PRE_DEVICE_C_1X: + case PRE_SOFTWARE_RESET_DEVICE_C: + case POST_SOFTWARE_RESET_DEVICE_C: + dev_idx = 0xC3; + break; + case MAIN_DEVICE_D_1X: + dev_idx = 0x84; + break; + case COEFF_DEVICE_D_1X: + case PRE_DEVICE_D_1X: + case PRE_SOFTWARE_RESET_DEVICE_D: + case POST_SOFTWARE_RESET_DEVICE_D: + dev_idx = 0xC4; + break; + default: + dev_info(tasdevice->dev, + "%s: load block: Other Type = 0x%02x\n", + __func__, block->type); + break; + } + } else if (fw_fixed_hdr->ppcver >= + PPC3_VERSION) { + switch (block->type) { + case MAIN_ALL_DEVICES_1X: + dev_idx = 0x80; + break; + case MAIN_DEVICE_A_1X: + dev_idx = 0x81; + break; + case COEFF_DEVICE_A_1X: + case PRE_DEVICE_A_1X: + dev_idx = 0xC1; + break; + case MAIN_DEVICE_B_1X: + dev_idx = 0x82; + break; + case COEFF_DEVICE_B_1X: + case PRE_DEVICE_B_1X: + dev_idx = 0xC2; + break; + case MAIN_DEVICE_C_1X: + dev_idx = 0x83; + break; + case COEFF_DEVICE_C_1X: + case PRE_DEVICE_C_1X: + dev_idx = 0xC3; + break; + case MAIN_DEVICE_D_1X: + dev_idx = 0x84; + break; + case COEFF_DEVICE_D_1X: + case PRE_DEVICE_D_1X: + dev_idx = 0xC4; + break; + default: + dev_info(tasdevice->dev, + "%s: load block: Other Type = 0x%02x\n", + __func__, block->type); + break; + } + } else { + switch (block->type) { + case MAIN_ALL_DEVICES: + dev_idx = 0|0x80; + break; + case MAIN_DEVICE_A: + dev_idx = 0x81; + break; + case COEFF_DEVICE_A: + case PRE_DEVICE_A: + dev_idx = 0xC1; + break; + case MAIN_DEVICE_B: + dev_idx = 0x82; + break; + case COEFF_DEVICE_B: + case PRE_DEVICE_B: + dev_idx = 0xC2; + break; + case MAIN_DEVICE_C: + dev_idx = 0x83; + break; + case COEFF_DEVICE_C: + case PRE_DEVICE_C: + dev_idx = 0xC3; + break; + case MAIN_DEVICE_D: + dev_idx = 0x84; + break; + case COEFF_DEVICE_D: + case PRE_DEVICE_D: + dev_idx = 0xC4; + break; + default: + dev_info(tasdevice->dev, + "%s: load block: Other Type = 0x%02x\n", + __func__, block->type); + break; + } + } + + for (i = 0, length = 0; i < block->nr_subblocks; i++) { + int rc = tasdevice_process_block(tasdevice, data + length, + dev_idx, blk_size - length); + if (rc < 0) { + dev_err(tasdevice->dev, + "%s: %u %u sublock write error\n", + __func__, length, blk_size); + break; + } + length += (unsigned int)rc; + if (blk_size < length) { + dev_err(tasdevice->dev, "%s: %u %u out of boundary\n", + __func__, length, blk_size); + break; + } + } + + return 0; +} + +static int fw_parse_variable_hdr(struct tasdevice_priv + *tas_priv, struct tasdevice_dspfw_hdr *fw_hdr, + const struct firmware *fmw, int offset) +{ + const unsigned char *buf = fmw->data; + int len = strlen((char *)&buf[offset]); + + len++; + + if (offset + len + 8 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + + offset += len; + + fw_hdr->device_family = be32_to_cpup((__be32 *)&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s: not TAS device\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 4; + + fw_hdr->device = be32_to_cpup((__be32 *)&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + offset = -EINVAL; + goto out; + } + offset += 4; + fw_hdr->ndev = deviceNumber[fw_hdr->device]; + +out: + return offset; +} + +static int fw_parse_variable_header_git(struct tasdevice_priv + *tas_priv, const struct firmware *fmw, int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + + offset = fw_parse_variable_hdr(tas_priv, fw_hdr, fmw, offset); + if (offset < 0) + goto out; + if (fw_hdr->ndev != tas_priv->ndev) { + dev_err(tas_priv->dev, + "%s: ndev(%u) in dspbin mismatch ndev(%u) in DTS\n", + __func__, fw_hdr->ndev, tas_priv->ndev); + offset = -EINVAL; + } + +out: + return offset; +} + +static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + int n; + + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Type error\n", __func__); + offset = -EINVAL; + goto out; + } + block->type = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + if (tas_fmw->fw_hdr.fixed_hdr.drv_ver >= PPC_DRIVER_CRCCHK) { + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "PChkSumPresent error\n"); + offset = -EINVAL; + goto out; + } + block->is_pchksum_present = data[offset]; + offset++; + + block->pchksum = data[offset]; + offset++; + + block->is_ychksum_present = data[offset]; + offset++; + + block->ychksum = data[offset]; + offset++; + } else { + block->is_pchksum_present = 0; + block->is_ychksum_present = 0; + } + + block->nr_cmds = be32_to_cpup((__be32 *)&data[offset]); + offset += 4; + + n = block->nr_cmds * 4; + if (offset + n > fmw->size) { + dev_err(tas_fmw->dev, + "%s: File Size(%lu) error offset = %d n = %d\n", + __func__, (unsigned long)fmw->size, offset, n); + offset = -EINVAL; + goto out; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], n, GFP_KERNEL); + if (!block->data) { + offset = -ENOMEM; + goto out; + } + offset += n; + +out: + return offset; +} + +/* When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_data(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = (unsigned char *)fmw->data; + struct tasdev_blk *blk; + unsigned int i; + int n; + + if (offset + 64 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Name error\n", __func__); + offset = -EINVAL; + goto out; + } + memcpy(img_data->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n++; + if (offset + n + 2 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Description error\n", __func__); + offset = -EINVAL; + goto out; + } + offset += n; + img_data->nr_blk = be16_to_cpup((__be16 *)&data[offset]); + offset += 2; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(struct tasdev_blk), GFP_KERNEL); + if (!img_data->dev_blks) { + offset = -ENOMEM; + goto out; + } + for (i = 0; i < img_data->nr_blk; i++) { + blk = &(img_data->dev_blks[i]); + offset = fw_parse_block_data(tas_fmw, blk, fmw, offset); + if (offset < 0) { + offset = -EINVAL; + goto out; + } + } + +out: + return offset; +} + +/* When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_program_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + unsigned char *buf = (unsigned char *)fmw->data; + struct tasdevice_prog *program; + int i; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + tas_fmw->nr_programs = be16_to_cpup((__be16 *)&buf[offset]); + offset += 2; + + if (tas_fmw->nr_programs == 0) { + /*Not error in calibration Data file, return directly*/ + dev_info(tas_priv->dev, "%s: No Programs data, maybe calbin\n", + __func__); + goto out; + } + + tas_fmw->programs = + kcalloc(tas_fmw->nr_programs, sizeof(struct tasdevice_prog), + GFP_KERNEL); + if (!tas_fmw->programs) { + offset = -ENOMEM; + goto out; + } + for (i = 0; i < tas_fmw->nr_programs; i++) { + int n = 0; + + program = &(tas_fmw->programs[i]); + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 64; + + n = strlen((char *)&buf[offset]); + /* skip '\0' and 5 unused bytes */ + n += 6; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + offset = -EINVAL; + goto out; + } + + offset += n; + + offset = fw_parse_data(tas_fmw, &(program->dev_data), fmw, + offset); + if (offset < 0) + goto out; + } + +out: + return offset; +} + +/* When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_configuration_data( + struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + struct tasdevice_config *config; + unsigned int i; + int n; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + tas_fmw->nr_configurations = be16_to_cpup((__be16 *)&data[offset]); + offset += 2; + + if (tas_fmw->nr_configurations == 0) { + dev_err(tas_priv->dev, "%s: Conf is zero\n", __func__); + /*Not error for calibration Data file, return directly*/ + goto out; + } + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(struct tasdevice_config), GFP_KERNEL); + if (!tas_fmw->configs) { + offset = -ENOMEM; + goto out; + } + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &(tas_fmw->configs[i]); + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "File Size err\n"); + offset = -EINVAL; + goto out; + } + memcpy(config->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n += 15; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + offset = -EINVAL; + goto out; + } + + offset += n; + + offset = fw_parse_data(tas_fmw, &(config->dev_data), + fmw, offset); + if (offset < 0) + goto out; + } + +out: + return offset; +} + +static bool check_inpage_yram_rg(struct tas_crc *cd, + unsigned char reg, unsigned char len) +{ + bool in = false; + + + if (reg <= TAS2781_YRAM5_END_REG && + reg >= TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_END_REG) + cd->len = TAS2781_YRAM5_END_REG - reg + 1; + else + cd->len = len; + cd->offset = reg; + in = true; + } else if (reg < TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_START_REG) { + cd->offset = TAS2781_YRAM5_START_REG; + cd->len = len - TAS2781_YRAM5_START_REG + reg; + in = true; + } + } + + return in; +} + +static bool check_inpage_yram_bk1(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (page == TAS2781_YRAM1_PAGE) { + if (reg >= TAS2781_YRAM1_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg + len > TAS2781_YRAM1_START_REG) { + cd->offset = TAS2781_YRAM1_START_REG; + cd->len = len - TAS2781_YRAM1_START_REG + reg; + in = true; + } + } else if (page == TAS2781_YRAM3_PAGE) + in = check_inpage_yram_rg(cd, reg, len); + + return in; +} + +/* Return Code: + * true -- the registers are in the inpage yram + * false -- the registers are NOT in the inpage yram + */ +static bool check_inpage_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1) { + in = check_inpage_yram_bk1(cd, page, reg, len); + goto end; + } + if (book == TAS2781_YRAM_BOOK2 && page == TAS2781_YRAM5_PAGE) + in = check_inpage_yram_rg(cd, reg, len); + +end: + return in; +} + +static bool check_inblock_yram_bk(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if ((page >= TAS2781_YRAM4_START_PAGE && + page <= TAS2781_YRAM4_END_PAGE) || + (page >= TAS2781_YRAM2_START_PAGE && + page <= TAS2781_YRAM2_END_PAGE)) { + if (reg <= TAS2781_YRAM2_END_REG && + reg >= TAS2781_YRAM2_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg < TAS2781_YRAM2_START_REG) { + if (reg + len - 1 >= TAS2781_YRAM2_START_REG) { + cd->offset = TAS2781_YRAM2_START_REG; + cd->len = reg + len - TAS2781_YRAM2_START_REG; + in = true; + } + } + } + + return in; +} + +/* Return Code: + * true -- the registers are in the inblock yram + * false -- the registers are NOT in the inblock yram + */ +static bool check_inblock_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1 || book == TAS2781_YRAM_BOOK2) + in = check_inblock_yram_bk(cd, page, reg, len); + + return in; +} + +static bool check_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in; + + in = check_inpage_yram(cd, book, page, reg, len); + if (in) + goto end; + in = check_inblock_yram(cd, book, page, reg, len); + +end: + return in; +} + +static int tasdev_multibytes_chksum(struct tasdevice_priv *tasdevice, + enum channel chn, unsigned char book, unsigned char page, + unsigned char reg, unsigned int len) +{ + struct tas_crc crc_data; + unsigned char crc_chksum = 0; + unsigned char nBuf1[128]; + int ret = 0; + int i; + bool in; + + if ((reg + len - 1) > 127) { + ret = -EINVAL; + dev_err(tasdevice->dev, "firmware error\n"); + goto end; + } + + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (reg == TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG)) + && (len == 4)) { + /*DSP swap command, pass */ + ret = 0; + goto end; + } + + in = check_yram(&crc_data, book, page, reg, len); + if (!in) + goto end; + + if (len == 1) { + dev_err(tasdevice->dev, "firmware error\n"); + ret = -EINVAL; + goto end; + } + + ret = tasdevice_dev_bulk_read(tasdevice, chn, + TASDEVICE_REG(book, page, crc_data.offset), + nBuf1, crc_data.len); + if (ret < 0) + goto end; + + for (i = 0; i < crc_data.len; i++) { + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (page == TASDEVICE_PAGE_ID( + TAS2781_SA_COEFF_SWAP_REG)) + && ((i + crc_data.offset) + >= TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG)) + && ((i + crc_data.offset) + <= (TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG) + + 4))) + /*DSP swap command, bypass */ + continue; + else + crc_chksum += crc8(tasdevice->crc8_lkp_tbl, &nBuf1[i], + 1, 0); + } + + ret = crc_chksum; + +end: + return ret; +} + +static int do_singlereg_checksum(struct tasdevice_priv *tasdevice, + enum channel chl, unsigned char book, unsigned char page, + unsigned char reg, unsigned char val) +{ + struct tas_crc crc_data; + unsigned int nData1; + int ret = 0; + bool in; + + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (reg >= TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG)) + && (reg <= (TASDEVICE_PAGE_REG( + TAS2781_SA_COEFF_SWAP_REG) + 4))) { + /*DSP swap command, pass */ + ret = 0; + goto end; + } + + in = check_yram(&crc_data, book, page, reg, 1); + if (!in) + goto end; + ret = tasdevice_dev_read(tasdevice, chl, + TASDEVICE_REG(book, page, reg), &nData1); + if (ret < 0) + goto end; + + if (nData1 != val) { + dev_err(tasdevice->dev, + "B[0x%x]P[0x%x]R[0x%x] W[0x%x], R[0x%x]\n", + book, page, reg, val, nData1); + tasdevice->tasdevice[chl].err_code |= ERROR_YRAM_CRCCHK; + ret = -EAGAIN; + goto end; + } + + ret = crc8(tasdevice->crc8_lkp_tbl, &val, 1, 0); + +end: + return ret; +} + +static void set_err_prg_cfg(unsigned int type, struct tasdevice *dev) +{ + if ((type == MAIN_ALL_DEVICES) || (type == MAIN_DEVICE_A) + || (type == MAIN_DEVICE_B) || (type == MAIN_DEVICE_C) + || (type == MAIN_DEVICE_D)) + dev->cur_prog = -1; + else + dev->cur_conf = -1; +} + +static int tasdev_bytes_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, int chn, unsigned char book, + unsigned char page, unsigned char reg, unsigned int len, + unsigned char val, unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) + ret = tasdev_multibytes_chksum(tas_priv, chn, book, page, reg, + len); + else + ret = do_singlereg_checksum(tas_priv, chn, book, page, reg, + val); + + if (ret > 0) { + *crc_chksum += (unsigned char)ret; + goto end; + } + + if (ret != -EAGAIN) + goto end; + + block->nr_retry--; + if (block->nr_retry > 0) + goto end; + + set_err_prg_cfg(block->type, &tas_priv->tasdevice[chn]); + +end: + return ret; +} + +static int tasdev_multibytes_wr(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, int chn, unsigned char book, + unsigned char page, unsigned char reg, unsigned char *data, + unsigned int len, unsigned int *nr_cmds, + unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) { + ret = tasdevice_dev_bulk_write(tas_priv, chn, + TASDEVICE_REG(book, page, reg), data + 3, len); + if (ret < 0) + goto end; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, chn, + book, page, reg, len, 0, crc_chksum); + } else { + ret = tasdevice_dev_write(tas_priv, chn, + TASDEVICE_REG(book, page, reg), data[3]); + if (ret < 0) + goto end; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, chn, book, + page, reg, 1, data[3], crc_chksum); + } + + if (!block->is_ychksum_present || ret >= 0) { + *nr_cmds += 1; + if (len >= 2) + *nr_cmds += ((len - 2) / 4) + 1; + } + +end: + return ret; +} + +static int tasdev_block_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, int chn) +{ + unsigned int nr_value; + int ret; + + ret = tasdevice_dev_read(tas_priv, chn, TASDEVICE_I2CChecksum, + &nr_value); + if (ret < 0) { + dev_err(tas_priv->dev, "%s: Chn %d\n", __func__, chn); + set_err_prg_cfg(block->type, &tas_priv->tasdevice[chn]); + goto end; + } + + if ((nr_value & 0xff) != block->pchksum) { + dev_err(tas_priv->dev, "%s: Blk PChkSum Chn %d ", __func__, + chn); + dev_err(tas_priv->dev, "PChkSum = 0x%x, Reg = 0x%x\n", + block->pchksum, (nr_value & 0xff)); + tas_priv->tasdevice[chn].err_code |= ERROR_PRAM_CRCCHK; + ret = -EAGAIN; + block->nr_retry--; + + if (block->nr_retry <= 0) + set_err_prg_cfg(block->type, + &tas_priv->tasdevice[chn]); + } else + tas_priv->tasdevice[chn].err_code &= ~ERROR_PRAM_CRCCHK; + +end: + return ret; +} + +static int tasdev_load_blk(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, int chn) +{ + unsigned int sleep_time; + unsigned int len; + unsigned int nr_cmds; + unsigned char *data = block->data; + unsigned char crc_chksum = 0; + unsigned char offset; + unsigned char book; + unsigned char page; + unsigned char val; + int ret = 0; + + while (block->nr_retry > 0) { + if (block->is_pchksum_present) { + ret = tasdevice_dev_write(tas_priv, chn, + TASDEVICE_I2CChecksum, 0); + if (ret < 0) + break; + } + + if (block->is_ychksum_present) + crc_chksum = 0; + + nr_cmds = 0; + + while (nr_cmds < block->nr_cmds) { + data = block->data + nr_cmds * 4; + + book = data[0]; + page = data[1]; + offset = data[2]; + val = data[3]; + + nr_cmds++; + /*Single byte write*/ + if (offset <= 0x7F) { + ret = tasdevice_dev_write(tas_priv, chn, + TASDEVICE_REG(book, page, offset), + val); + if (ret < 0) + goto end; + if (block->is_ychksum_present) { + ret = tasdev_bytes_chksum(tas_priv, + block, chn, book, page, offset, + 1, val, &crc_chksum); + if (ret < 0) + break; + } + continue; + } + /*sleep command*/ + if (offset == 0x81) { + /*book -- data[0] page -- data[1]*/ + sleep_time = ((book << 8) + page)*1000; + usleep_range(sleep_time, sleep_time + 50); + continue; + } + /*Multiple bytes write*/ + if (offset == 0x85) { + data += 4; + len = (book << 8) + page; + book = data[0]; + page = data[1]; + offset = data[2]; + ret = tasdev_multibytes_wr(tas_priv, + block, chn, book, page, offset, data, + len, &nr_cmds, &crc_chksum); + if (ret < 0) + break; + } + } + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) /*err in current device, skip it*/ + break; + + if (block->is_pchksum_present) { + ret = tasdev_block_chksum(tas_priv, block, chn); + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) /*err in current device, skip it*/ + break; + } + + if (block->is_ychksum_present) { + /* TBD, open it when FW ready */ + dev_err(tas_priv->dev, + "Blk YChkSum: FW = 0x%x, YCRC = 0x%x\n", + block->ychksum, crc_chksum); + + tas_priv->tasdevice[chn].err_code &= + ~ERROR_YRAM_CRCCHK; + ret = 0; + } + /*skip current blk*/ + break; + } + +end: + return ret; +} + +static int tasdevice_load_block(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + int chnend = 0; + int ret = 0; + int chn = 0; + int rc = 0; + + switch (block->type) { + case MAIN_ALL_DEVICES: + chn = 0; + chnend = tas_priv->ndev; + break; + case MAIN_DEVICE_A: + case COEFF_DEVICE_A: + case PRE_DEVICE_A: + chn = 0; + chnend = 1; + break; + case MAIN_DEVICE_B: + case COEFF_DEVICE_B: + case PRE_DEVICE_B: + chn = 1; + chnend = 2; + break; + case MAIN_DEVICE_C: + case COEFF_DEVICE_C: + case PRE_DEVICE_C: + chn = 2; + chnend = 3; + break; + case MAIN_DEVICE_D: + case COEFF_DEVICE_D: + case PRE_DEVICE_D: + chn = 3; + chnend = 4; + break; + default: + dev_dbg(tas_priv->dev, "load blk: Other Type = 0x%02x\n", + block->type); + break; + } + + for (; chn < chnend; chn++) { + block->nr_retry = 6; + if (tas_priv->tasdevice[chn].is_loading == false) + continue; + ret = tasdev_load_blk(tas_priv, block, chn); + if (ret < 0) + dev_err(tas_priv->dev, "dev %d, Blk (%d) load error\n", + chn, block->type); + rc |= ret; + } + + return rc; +} + +static int dspfw_default_callback(struct tasdevice_priv *tas_priv, + unsigned int drv_ver, unsigned int ppcver) +{ + int rc = 0; + + if (drv_ver == 0x100) { + if (ppcver >= PPC3_VERSION) { + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + } else { + switch (ppcver) { + case 0x00: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + break; + default: + dev_err(tas_priv->dev, + "%s: PPCVer must be 0x0 or 0x%02x", + __func__, PPC3_VERSION); + dev_err(tas_priv->dev, " Current:0x%02x\n", + ppcver); + rc = -EINVAL; + break; + } + } + } else { + dev_err(tas_priv->dev, + "DrvVer must be 0x0, 0x230 or above 0x230 "); + dev_err(tas_priv->dev, "current is 0x%02x\n", drv_ver); + rc = -EINVAL; + } + + return rc; +} + +static int load_calib_data(struct tasdevice_priv *tas_priv, + struct tasdevice_data *dev_data) +{ + struct tasdev_blk *block; + unsigned int i; + int ret = 0; + + for (i = 0; i < dev_data->nr_blk; i++) { + block = &(dev_data->dev_blks[i]); + ret = tasdevice_load_block(tas_priv, block); + if (ret < 0) + break; + } + + return ret; +} + +static int fw_parse_header(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &(fw_hdr->fixed_hdr); + const unsigned char magic_number[] = { 0x35, 0x35, 0x35, 0x32 }; + const unsigned char *buf = (unsigned char *)fmw->data; + + if (offset + 92 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + if (memcmp(&buf[offset], magic_number, 4)) { + dev_err(tas_priv->dev, "%s: Magic num NOT match\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 4; + + /* Convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + fw_fixed_hdr->fwsize = be32_to_cpup((__be32 *)&buf[offset]); + offset += 4; + if (fw_fixed_hdr->fwsize != fmw->size) { + dev_err(tas_priv->dev, "File size not match, %lu %u", + (unsigned long)fmw->size, fw_fixed_hdr->fwsize); + offset = -EINVAL; + goto out; + } + offset += 4; + fw_fixed_hdr->ppcver = be32_to_cpup((__be32 *)&buf[offset]); + offset += 8; + fw_fixed_hdr->drv_ver = be32_to_cpup((__be32 *)&buf[offset]); + offset += 72; + + out: + return offset; +} + +static int fw_parse_variable_hdr_cal(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + + offset = fw_parse_variable_hdr(tas_priv, fw_hdr, fmw, offset); + if (offset < 0) + goto out; + if (fw_hdr->ndev != 1) { + dev_err(tas_priv->dev, + "%s: calbin must be 1, but currently ndev(%u)\n", + __func__, fw_hdr->ndev); + offset = -EINVAL; + } + +out: + return offset; +} + +/* When calibrated data parsing error occurs, DSP can still work with default + * calibrated data, memory resource related to calibrated data will be + * released in the tasdevice_codec_remove. + */ +static int fw_parse_calibration_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_calibration *calibration; + unsigned char *data = (unsigned char *)fmw->data; + unsigned int i, n; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: Calibrations error\n", __func__); + offset = -EINVAL; + goto out; + } + tas_fmw->nr_calibrations = be16_to_cpup((__be16 *)&data[offset]); + offset += 2; + + if (tas_fmw->nr_calibrations != 1) { + dev_err(tas_priv->dev, + "%s: only support one calibraiton(%d)!\n", + __func__, tas_fmw->nr_calibrations); + goto out; + } + + tas_fmw->calibrations = kcalloc(tas_fmw->nr_calibrations, + sizeof(struct tasdevice_calibration), GFP_KERNEL); + if (!tas_fmw->calibrations) { + offset = -ENOMEM; + goto out; + } + for (i = 0; i < tas_fmw->nr_calibrations; i++) { + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "Calibrations error\n"); + offset = -EINVAL; + goto out; + } + calibration = &(tas_fmw->calibrations[i]); + offset += 64; + + n = strlen((char *)&data[offset]); + /* skip '\0' and 2 unused bytes */ + n += 3; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + offset = -EINVAL; + goto out; + } + offset += n; + + offset = fw_parse_data(tas_fmw, &(calibration->dev_data), fmw, + offset); + if (offset < 0) + goto out; + } + +out: + return offset; +} + +int tas2781_load_calibration(void *context, char *file_name, + enum channel i) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; + struct tasdevice *tasdev = &(tas_priv->tasdevice[i]); + const struct firmware *fw_entry; + struct tasdevice_fw *tas_fmw; + struct firmware fmw; + int offset = 0; + int ret; + + ret = request_firmware(&fw_entry, file_name, tas_priv->dev); + if (ret) { + dev_info(tas_priv->dev, "%s: Request firmware %s failed\n", + __func__, file_name); + goto out; + } + + if (!fw_entry->size) { + dev_err(tas_priv->dev, "%s: file read error: size = %lu\n", + __func__, (unsigned long)fw_entry->size); + goto out; + } + fmw.size = fw_entry->size; + fmw.data = fw_entry->data; + + tas_fmw = tasdev->cali_data_fmw = kzalloc(sizeof(struct tasdevice_fw), + GFP_KERNEL); + if (!tasdev->cali_data_fmw) { + ret = -ENOMEM; + goto out; + } + tas_fmw->dev = tas_priv->dev; + offset = fw_parse_header(tas_priv, tas_fmw, &fmw, offset); + if (offset == -EINVAL) { + dev_err(tas_priv->dev, "fw_parse_header EXIT!\n"); + ret = offset; + goto out; + } + offset = fw_parse_variable_hdr_cal(tas_priv, tas_fmw, &fmw, offset); + if (offset == -EINVAL) { + dev_err(tas_priv->dev, + "%s: fw_parse_variable_header_cal EXIT!\n", __func__); + ret = offset; + goto out; + } + offset = fw_parse_program_data(tas_priv, tas_fmw, &fmw, offset); + if (offset < 0) { + dev_err(tas_priv->dev, "fw_parse_program_data EXIT!\n"); + ret = offset; + goto out; + } + offset = fw_parse_configuration_data(tas_priv, tas_fmw, &fmw, offset); + if (offset < 0) { + dev_err(tas_priv->dev, "fw_parse_configuration_data EXIT!\n"); + ret = offset; + goto out; + } + offset = fw_parse_calibration_data(tas_priv, tas_fmw, &fmw, offset); + if (offset < 0) { + dev_err(tas_priv->dev, "fw_parse_calibration_data EXIT!\n"); + ret = offset; + goto out; + } + +out: + if (fw_entry) + release_firmware(fw_entry); + + return ret; +} +EXPORT_SYMBOL_NS_GPL(tas2781_load_calibration, SND_SOC_TAS2781_FMWLIB); + +static int tasdevice_dspfw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr; + struct tasdevice_fw *tas_fmw; + int offset = 0; + int ret = 0; + + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "%s: Failed to read firmware %s\n", + __func__, tas_priv->coef_binaryname); + ret = -EINVAL; + goto out; + } + + tas_priv->fmw = kzalloc(sizeof(struct tasdevice_fw), GFP_KERNEL); + if (!tas_priv->fmw) { + ret = -ENOMEM; + goto out; + } + tas_fmw = tas_priv->fmw; + tas_fmw->dev = tas_priv->dev; + offset = fw_parse_header(tas_priv, tas_fmw, fmw, offset); + + if (offset == -EINVAL) { + ret = -EINVAL; + goto out; + } + fw_fixed_hdr = &(tas_fmw->fw_hdr.fixed_hdr); + /* Support different versions of firmware */ + switch (fw_fixed_hdr->drv_ver) { + case 0x301: + case 0x302: + case 0x502: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + break; + case 0x202: + case 0x400: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + break; + default: + ret = dspfw_default_callback(tas_priv, + fw_fixed_hdr->drv_ver, fw_fixed_hdr->ppcver); + if (ret) + goto out; + break; + } + + offset = tas_priv->fw_parse_variable_header(tas_priv, fmw, offset); + if (offset < 0) { + ret = offset; + goto out; + } + offset = tas_priv->fw_parse_program_data(tas_priv, tas_fmw, fmw, + offset); + if (offset < 0) { + ret = offset; + goto out; + } + offset = tas_priv->fw_parse_configuration_data(tas_priv, + tas_fmw, fmw, offset); + if (offset < 0) + ret = offset; + +out: + return ret; +} + +int tasdevice_dsp_parser(void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; + const struct firmware *fw_entry; + int ret; + + ret = request_firmware(&fw_entry, tas_priv->coef_binaryname, + tas_priv->dev); + if (ret) { + dev_err(tas_priv->dev, "%s: load %s error\n", __func__, + tas_priv->coef_binaryname); + goto out; + } + + ret = tasdevice_dspfw_ready(fw_entry, tas_priv); + release_firmware(fw_entry); + fw_entry = NULL; + +out: + return ret; +} +EXPORT_SYMBOL_NS_GPL(tasdevice_dsp_parser, SND_SOC_TAS2781_FMWLIB); + +static void tas2781_clear_calfirmware(struct tasdevice_fw *tas_fmw) +{ + struct tasdevice_calibration *calibration; + struct tasdev_blk *block; + struct tasdevice_data *im; + unsigned int blks; + int i; + + if (!tas_fmw->calibrations) + goto out; + + for (i = 0; i < tas_fmw->nr_calibrations; i++) { + calibration = &(tas_fmw->calibrations[i]); + if (!calibration) + continue; + + im = &(calibration->dev_data); + + if (!im->dev_blks) + continue; + + for (blks = 0; blks < im->nr_blk; blks++) { + block = &(im->dev_blks[blks]); + if (!block) + continue; + kfree(block->data); + } + kfree(im->dev_blks); + } + kfree(tas_fmw->calibrations); +out: + kfree(tas_fmw); +} + +void tasdevice_calbin_remove(void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice *tasdev; + int i; + + if (!tas_priv) + return; + + for (i = 0; i < tas_priv->ndev; i++) { + tasdev = &(tas_priv->tasdevice[i]); + if (!tasdev->cali_data_fmw) + continue; + tas2781_clear_calfirmware(tasdev->cali_data_fmw); + tasdev->cali_data_fmw = NULL; + } +} +EXPORT_SYMBOL_NS_GPL(tasdevice_calbin_remove, SND_SOC_TAS2781_FMWLIB); + +void tasdevice_config_info_remove(void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_rca *rca = &(tas_priv->rcabin); + struct tasdevice_config_info **ci = rca->cfg_info; + int i, j; + + if (!ci) + return; + for (i = 0; i < rca->ncfgs; i++) { + if (!ci[i]) + continue; + if (ci[i]->blk_data) { + for (j = 0; j < (int)ci[i]->real_nblocks; j++) { + if (!ci[i]->blk_data[j]) + continue; + kfree(ci[i]->blk_data[j]->regdata); + kfree(ci[i]->blk_data[j]); + } + kfree(ci[i]->blk_data); + } + kfree(ci[i]); + } + kfree(ci); +} +EXPORT_SYMBOL_NS_GPL(tasdevice_config_info_remove, SND_SOC_TAS2781_FMWLIB); + +static int tasdevice_load_data(struct tasdevice_priv *tas_priv, + struct tasdevice_data *dev_data) +{ + struct tasdev_blk *block; + unsigned int i; + int ret = 0; + + for (i = 0; i < dev_data->nr_blk; i++) { + block = &(dev_data->dev_blks[i]); + ret = tas_priv->tasdevice_load_block(tas_priv, block); + if (ret < 0) + break; + } + + return ret; +} + +int tasdevice_select_tuningprm_cfg(void *context, int prm_no, + int cfg_no, int rca_conf_no) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_rca *rca = &(tas_priv->rcabin); + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + struct tasdevice_config *conf; + int prog_status = 0; + int status, i; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + goto out; + } + + if (cfg_no >= tas_fmw->nr_configurations) { + dev_err(tas_priv->dev, + "%s: cfg(%d) is not in range of conf %u\n", + __func__, cfg_no, tas_fmw->nr_configurations); + goto out; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + goto out; + } + + if (rca_conf_no > rca->ncfgs || rca_conf_no <= 0 || + !cfg_info) { + dev_err(tas_priv->dev, + "conf_no:%d should be in range from 0 to %u\n", + rca_conf_no, rca->ncfgs-1); + goto out; + } + + conf = &(tas_fmw->configs[cfg_no]); + for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { + if (cfg_info[rca_conf_no]->active_dev & (1 << i)) { + if (tas_priv->tasdevice[i].cur_prog != prm_no) { + tas_priv->tasdevice[i].cur_conf = -1; + tas_priv->tasdevice[i].is_loading = true; + prog_status++; + } + } else + tas_priv->tasdevice[i].is_loading = false; + tas_priv->tasdevice[i].is_loaderr = false; + } + + if (prog_status) { + program = &(tas_fmw->programs[prm_no]); + tasdevice_load_data(tas_priv, &(program->dev_data)); + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].is_loaderr == true) + continue; + else if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) { + struct tasdevice_fw *cal_fmw = + tas_priv->tasdevice[i].cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration + *cal = cal_fmw->calibrations; + + if (cal) + load_calib_data(tas_priv, + &(cal->dev_data)); + } + tas_priv->tasdevice[i].cur_prog = prm_no; + } + } + } + + for (i = 0, status = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].cur_conf != cfg_no + && (cfg_info[rca_conf_no]->active_dev & (1 << i)) + && (tas_priv->tasdevice[i].is_loaderr == false)) { + status++; + tas_priv->tasdevice[i].is_loading = true; + } else + tas_priv->tasdevice[i].is_loading = false; + } + + if (status) { + status = 0; + tasdevice_load_data(tas_priv, &(conf->dev_data)); + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].is_loaderr == true) { + status |= 1 << (i + 4); + continue; + } else if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) + tas_priv->tasdevice[i].cur_conf = cfg_no; + } + } else + dev_err(tas_priv->dev, + "%s: No device is in active in conf %d\n", + __func__, rca_conf_no); + + status |= cfg_info[rca_conf_no]->active_dev; + +out: + return prog_status; +} +EXPORT_SYMBOL_NS_GPL(tasdevice_select_tuningprm_cfg, + SND_SOC_TAS2781_FMWLIB); + +int tasdevice_prmg_load(void *context, int prm_no) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + int prog_status = 0; + int i; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + goto out; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + goto out; + } + + for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].cur_prog != prm_no) { + tas_priv->tasdevice[i].cur_conf = -1; + tas_priv->tasdevice[i].is_loading = true; + prog_status++; + } + } + + if (prog_status) { + program = &(tas_fmw->programs[prm_no]); + tasdevice_load_data(tas_priv, &(program->dev_data)); + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].is_loaderr == true) + continue; + else if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) + tas_priv->tasdevice[i].cur_prog = prm_no; + } + } + if (tas_priv->set_global_mode) + tas_priv->set_global_mode(tas_priv); +out: + return prog_status; +} +EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_load, SND_SOC_TAS2781_FMWLIB); + +int tasdevice_prmg_calibdata_load(void *context, int prm_no) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + int prog_status = 0; + int i; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + goto out; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + goto out; + } + + for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].cur_prog != prm_no) { + tas_priv->tasdevice[i].cur_conf = -1; + tas_priv->tasdevice[i].is_loading = true; + prog_status++; + } + tas_priv->tasdevice[i].is_loaderr = false; + } + + if (prog_status) { + program = &(tas_fmw->programs[prm_no]); + tasdevice_load_data(tas_priv, &(program->dev_data)); + for (i = 0; i < tas_priv->ndev; i++) { + if (tas_priv->tasdevice[i].is_loaderr == true) + continue; + else if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) { + struct tasdevice_fw *cal_fmw = + tas_priv->tasdevice[i].cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration *cal = + cal_fmw->calibrations; + + if (cal) + load_calib_data(tas_priv, + &(cal->dev_data)); + } + tas_priv->tasdevice[i].cur_prog = prm_no; + } + } + } + +out: + return prog_status; +} +EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_calibdata_load, + SND_SOC_TAS2781_FMWLIB); + +void tasdevice_tuning_switch(void *context, int state) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + int profile_cfg_id = 0; + + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + return; + } + + if (state == 0) { + if (tas_priv->cur_prog >= tas_fmw->nr_programs) + /*bypass all in rca is profile id 0*/ + profile_cfg_id = RCA_CONFIGID_BYPASS_ALL; + else { + /*dsp mode or tuning mode*/ + profile_cfg_id = tas_priv->rcabin.profile_cfg_id; + tasdevice_select_tuningprm_cfg(tas_priv, + tas_priv->cur_prog, tas_priv->cur_conf, + profile_cfg_id); + if (tas_priv->set_global_mode) + tas_priv->set_global_mode(tas_priv); + } + + tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + } else + tasdevice_select_cfg_blk(tas_priv, + tas_priv->rcabin.profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN); +} +EXPORT_SYMBOL_NS_GPL(tasdevice_tuning_switch, + SND_SOC_TAS2781_FMWLIB); + +MODULE_DESCRIPTION("Texas Firmware Support"); +MODULE_AUTHOR("Shenghao Ding, TI, <shenghao-ding(a)ti.com>"); +MODULE_LICENSE("GPL"); -- 2.34.1

1 0

Asus ROG Zephyrus GX701L sound problem
by Kernel.org Bugbot 27 May '23

27 May '23

rakic.iwan writes via Kernel.org Bugzilla: Hello, I must mention right at the beginning that I tried all the solutions I found on the forums, but without success. The problem exists only with the speakers on the laptop, the microphone works, as do the headphones. I extracted the audio dump from windows and compared it with the dump from Linux and thus created this pinfcg below. Please, can someone add a kernel fix for ROG Zephyrus S17 GX701LWS_GX701LWS, Subsystem Id: 0x10431f01? ``` [codec] 0x10ec0294 0x10431f01 0 [pincfg] 0x19 0x03A11050 0x1a 0x03A11C30 ``` This is what a quirk should look like: +SND_PCI_QUIRK(0x1043, 0x1f01, “ASUS GX701L”, ALC294_FIXUP_ASUS_SPK) [ 2.396344] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC294: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 2.396348] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 2.396349] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 2.396350] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 2.396351] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 2.396352] snd_hda_codec_realtek hdaudioC0D0: Headset Mic=0x19 [ 2.396353] snd_hda_codec_realtek hdaudioC0D0: Internal Mic=0x12 hwinfo --sound  :heavy_check_mark: 22: PCI 1f.3: 0403 Audio device [Created at pci.386] Unique ID: nS1_.Qb7ky1jjQRD SysFS ID: /devices/pci0000:00/0000:00:1f.3 SysFS BusID: 0000:00:1f.3 Hardware Class: sound Model: “Intel Comet Lake PCH cAVS” Vendor: pci 0x8086 “Intel Corporation” Device: pci 0x06c8 “Comet Lake PCH cAVS” SubVendor: pci 0x1043 “ASUSTeK Computer Inc.” SubDevice: pci 0x1f01 Driver: “snd_hda_intel” Driver Modules: “snd_hda_intel” Memory Range: 0x94300000-0x94303fff (rw,non-prefetchable) Memory Range: 0x94100000-0x941fffff (rw,non-prefetchable) IRQ: 157 (817 events) Module Alias: “pci:v00008086d000006C8sv00001043sd00001F01bc04sc03i80” Driver Info #0: Driver Status: snd_hda_intel is active Driver Activation Cmd: “modprobe snd_hda_intel” Driver Info #1: Driver Status: snd_soc_skl is active Driver Activation Cmd: “modprobe snd_soc_skl” Driver Info #2: Driver Status: snd_sof_pci_intel_cnl is active Driver Activation Cmd: “modprobe snd_sof_pci_intel_cnl” Config Status: cfg=new, avail=yes, need=no, active=unknown 29: PCI 100.1: 0403 Audio device [Created at pci.386] Unique ID: NXNs.SkelvpPm387 Parent ID: vSkL.ZFPYIgTFUyC SysFS ID: /devices/pci0000:00/0000:00:01.0/0000:01:00.1 SysFS BusID: 0000:01:00.1 Hardware Class: sound Model: “nVidia TU104 HD Audio Controller” Vendor: pci 0x10de “nVidia Corporation” Device: pci 0x10f8 “TU104 HD Audio Controller” SubVendor: pci 0x1043 “ASUSTeK Computer Inc.” SubDevice: pci 0x186f Revision: 0xa1 Driver: “snd_hda_intel” Driver Modules: “snd_hda_intel” Memory Range: 0x94080000-0x94083fff (rw,non-prefetchable) IRQ: 17 (1698 events) Module Alias: “pci:v000010DEd000010F8sv00001043sd0000186Fbc04sc03i00” Driver Info #0: Driver Status: snd_hda_intel is active Driver Activation Cmd: “modprobe snd_hda_intel” Config Status: cfg=new, avail=yes, need=no, active=unknown Attached to: #13 (PCI bridge) If you need any more data, or smth just say so. View: https://bugzilla.kernel.org/show_bug.cgi?id=216392#c0 You can reply to this message to join the discussion. -- Deet-doot-dot, I am a bot. Kernel.org Bugzilla (peebz 0.1)

1 2

Matebook 11 Intel 2021; always need to manually set config in amixer
by GitHub issues - edited 27 May '23

27 May '23

alsa-project/alsa-ucm-conf issue #319 was edited from TigerbyteDev: Hey, the drivers in sof got fixed but I still need to manually set the alsa config. My microphone also isn't working, not sure why. This got investigated a bit in the other issue too. Config: ```bash amixer -c 0 cset name='Headphone Switch' on amixer -c 0 cset name='Headphone Playback Volume' 3,3 amixer -c 0 cset name='Right Headphone Mixer Right DAC Switch' on amixer -c 0 cset name='Left Headphone Mixer Left DAC Switch' on amixer -c 0 cset name='DAC Playback Volume' 999,999 amixer -c 0 cset name='Headphone Mixer Volume' 999,999 amixer -c 0 cset name='Differential Mux' lin1-rin1 amixer -c 0 cset name='Differential Mux' lin2-rin2 amixer -c 0 cset name='DAC Soft Ramp Switch' off ``` Issue in SOF: https://github.com/thesofproject/sof-bin/issues/121 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/319 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Matebook 11 Intel 2021; always need to manually set config in amixer
by GitHub issues - opened 27 May '23

27 May '23

alsa-project/alsa-ucm-conf issue #319 was opened from TigerbyteDev: Hey, the drivers in sof got fixed but I still need to manually set the alsa config. My microphone also isn't working, not sure why. This got investigated a bit in the other issue too. Config: ```bash amixer -c 0 cset name='Headphone Switch' on amixer -c 0 cset name='Headphone Playback Volume' 3,3 amixer -c 0 cset name='Right Headphone Mixer Right DAC Switch' on amixer -c 0 cset name='Left Headphone Mixer Left DAC Switch' on amixer -c 0 cset name='DAC Playback Volume' 999,999 amixer -c 0 cset name='Headphone Mixer Volume' 999,999 amixer -c 0 cset name='Differential Mux' lin1-rin1 amixer -c 0 cset name='Differential Mux' lin2-rin2 amixer -c 0 cset name='DAC Soft Ramp Switch' off ``` Issue in SOF: https://github.com/thesofproject/sof-bin/issues/121 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/319 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] soundwire: qcom: add proper error paths in qcom_swrm_startup()
by Krzysztof Kozlowski 27 May '23

27 May '23

Reverse actions in qcom_swrm_startup() error paths to avoid leaking stream memory and keeping runtime PM unbalanced. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/soundwire/qcom.c | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index f442280af9d3..baaa75320ae3 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1208,8 +1208,10 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, } sruntime = sdw_alloc_stream(dai->name); - if (!sruntime) - return -ENOMEM; + if (!sruntime) { + ret = -ENOMEM; + goto err_alloc; + } ctrl->sruntime[dai->id] = sruntime; @@ -1219,12 +1221,19 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, if (ret < 0 && ret != -ENOTSUPP) { dev_err(dai->dev, "Failed to set sdw stream on %s\n", codec_dai->name); - sdw_release_stream(sruntime); - return ret; + goto err_set_stream; } } return 0; + +err_set_stream: + sdw_release_stream(sruntime); +err_alloc: + pm_runtime_mark_last_busy(ctrl->dev); + pm_runtime_put_autosuspend(ctrl->dev); + + return ret; } static void qcom_swrm_shutdown(struct snd_pcm_substream *substream, -- 2.34.1

3 2

[RESEND PATCH] soundwire: qcom: drop unused struct qcom_swrm_ctrl members
by Krzysztof Kozlowski 27 May '23

27 May '23

Drop unused members from the driver state container: struct qcom_swrm_ctrl. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/soundwire/qcom.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index aad5942e5980..ab1098ec56e6 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -187,12 +187,9 @@ struct qcom_swrm_ctrl { #endif struct completion broadcast; struct completion enumeration; - struct work_struct slave_work; /* Port alloc/free lock */ struct mutex port_lock; struct clk *hclk; - u8 wr_cmd_id; - u8 rd_cmd_id; int irq; unsigned int version; int wake_irq; -- 2.34.1

3 2

[PATCH] soundwire: intel: read AC timing control register before updating it
by Bard Liao 27 May '23

27 May '23

From: Chao Song <chao.song(a)linux.intel.com> Start from ACE1.x, DOAISE is added to AC timing control register bit 5, it combines with DOAIS to get effective timing, and has the default value 1. The current code fills DOAIS, DACTQE and DODS bits to a variable initialized to zero, and updates the variable to AC timing control register. With this operation, We change DOAISE to 0, and force a much more aggressive timing. The timing is even unable to form a working waveform on SDA pin on Meteorlake. This patch uses read-modify-write operation for the AC timing control register access, thus makes sure those bits not supposed and intended to change are not touched. Signed-off-by: Chao Song <chao.song(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 238acf5c97a9..8cd2c73ac66f 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -260,7 +260,7 @@ static void intel_shim_init(struct sdw_intel *sdw) { void __iomem *shim = sdw->link_res->shim; unsigned int link_id = sdw->instance; - u16 ioctl = 0, act = 0; + u16 ioctl = 0, act; /* Initialize Shim */ ioctl |= SDW_SHIM_IOCTL_BKE; @@ -281,6 +281,7 @@ static void intel_shim_init(struct sdw_intel *sdw) intel_shim_glue_to_master_ip(sdw); + act = intel_readw(shim, SDW_SHIM_CTMCTL(link_id)); u16p_replace_bits(&act, 0x1, SDW_SHIM_CTMCTL_DOAIS); act |= SDW_SHIM_CTMCTL_DACTQE; act |= SDW_SHIM_CTMCTL_DODS; -- 2.25.1

2 1

[PATCH] soundwire: dmi-quirks: add new mapping for HP Spectre x360
by Bard Liao 27 May '23

27 May '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> A BIOS/DMI update seems to have broken some devices, let's add a new mapping. Link: https://github.com/thesofproject/linux/issues/4323 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/dmi-quirks.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 58ea013fa918..2a1096dab63d 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -99,6 +99,13 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_tgl_bios, }, + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "HP"), + DMI_MATCH(DMI_BOARD_NAME, "8709"), + }, + .driver_data = (void *)intel_tgl_bios, + }, { /* quirk used for NUC15 'Bishop County' LAPBC510 and LAPBC710 skews */ .matches = { -- 2.25.1

2 1

[tiwai-sound:topic/midi20 9/40] sound/usb/midi2.c:548:16: sparse: sparse: cast to restricted __le16
by kernel test robot 27 May '23

27 May '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/midi20 head: 77700b81bd0e47d89d50eb4b3f2f323492f79998 commit: ff49d1df79aef7580fe3ac99d17c3f886655d080 [9/40] ALSA: usb-audio: USB MIDI 2.0 UMP support config: microblaze-randconfig-s032-20230526 (https://download.01.org/0day-ci/archive/20230527/202305270534.odwHL9F0-lkp@…) compiler: microblaze-linux-gcc (GCC) 12.1.0 reproduce: mkdir -p ~/bin wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # apt-get install sparse # sparse version: v0.6.4-39-gce1a6720-dirty # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add tiwai-sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags tiwai-sound topic/midi20 git checkout ff49d1df79aef7580fe3ac99d17c3f886655d080 # save the config file mkdir build_dir && cp config build_dir/.config COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-12.1.0 ~/bin/make.cross C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' O=build_dir ARCH=microblaze olddefconfig COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-12.1.0 ~/bin/make.cross C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' O=build_dir ARCH=microblaze SHELL=/bin/bash sound/usb/ If you fix the issue, kindly add following tag where applicable | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202305270534.odwHL9F0-lkp@intel.com/ sparse warnings: (new ones prefixed by >>) >> sound/usb/midi2.c:548:16: sparse: sparse: cast to restricted __le16 >> sound/usb/midi2.c:548:16: sparse: sparse: cast to restricted __le16 >> sound/usb/midi2.c:548:16: sparse: sparse: cast to restricted __le16 >> sound/usb/midi2.c:548:16: sparse: sparse: cast to restricted __le16 vim +548 sound/usb/midi2.c 530 531 /* get the full group terminal block descriptors and return the size */ 532 static int get_group_terminal_block_descs(struct snd_usb_midi2_interface *umidi) 533 { 534 struct usb_host_interface *hostif = umidi->hostif; 535 struct usb_device *dev = umidi->chip->dev; 536 struct usb_ms20_gr_trm_block_header_descriptor header = { 0 }; 537 unsigned char *data; 538 int err, size; 539 540 err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 541 USB_REQ_GET_DESCRIPTOR, 542 USB_RECIP_INTERFACE | USB_TYPE_STANDARD | USB_DIR_IN, 543 USB_DT_CS_GR_TRM_BLOCK << 8 | hostif->desc.bAlternateSetting, 544 hostif->desc.bInterfaceNumber, 545 &header, sizeof(header)); 546 if (err < 0) 547 return err; > 548 size = __le16_to_cpu(header.wTotalLength); 549 if (!size) { 550 dev_err(&dev->dev, "Failed to get GTB descriptors for %d:%d\n", 551 hostif->desc.bInterfaceNumber, hostif->desc.bAlternateSetting); 552 return -EINVAL; 553 } 554 555 data = kzalloc(size, GFP_KERNEL); 556 if (!data) 557 return -ENOMEM; 558 559 err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 560 USB_REQ_GET_DESCRIPTOR, 561 USB_RECIP_INTERFACE | USB_TYPE_STANDARD | USB_DIR_IN, 562 USB_DT_CS_GR_TRM_BLOCK << 8 | hostif->desc.bAlternateSetting, 563 hostif->desc.bInterfaceNumber, data, size); 564 if (err < 0) { 565 kfree(data); 566 return err; 567 } 568 569 umidi->blk_descs = data; 570 umidi->blk_desc_size = size; 571 return 0; 572 } 573 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

alsa-lib fails test with clang-16
by GitHub issues - opened 27 May '23

27 May '23

alsa-project/alsa-lib issue #323 was opened from fsvm88: Hello, While building is fine, testing alsa-lib on Gentoo fails since clang-16, due to clang-16 defaulting to c99. The problem is at `alsa-lib-1.2.8/test/oldapi.c`, in the typedef declaration, which leads to the following error: ``` alsa-lib-1.2.8/test/oldapi.c:40:6: error: incompatible function pointer types assigning to 'myfcn *' (aka 'void (*)(void *)') from 'int (*)(const snd_pcm_hw_params_t *)' (aka 'int (*)(const struct _snd_pcm_hw_params *)') [-Werror,-Wincompatible-func fcn = &snd_pcm_hw_params_get_access; ^ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ``` [This is the Gentoo bug with all the information](https://bugs.gentoo.org/885551) [I have created a patch based on the compiler's suggestion](https://885551.bugs.gentoo.org/attachment.cgi?id=861055), but I'm not sure if it's the right way to solve this. Issue URL : https://github.com/alsa-project/alsa-lib/issues/323 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 00/13] ALSA: hda: Adding support for CS35L56 on HDA systems
by Richard Fitzgerald 26 May '23

26 May '23

This set of patches adds support for using the CS35L56 boosted smart amplifier on HDA systems. In these systems the CS35L56 audio is routed through a HDA-to-I2S bridge codec. This doesn't include the changes to the Realtek driver to actually hook up the CS35L56 driver, because we don't yet have the QUIRK IDs to associate it with. But we want to publish the driver now so that it is available for bringing up hardware with the CS35L56. The first 10 patches are moving code out of the ASoC driver and into the shared library so that it can be shared with the HDA driver. Patch #11 fixes missing #includes in the HDA headers so that the CS35L56 driver doesn't have to #include headers that it doesn't use. Finally, #12 and #13 actually add the support for CS35L56 on HDA. Richard Fitzgerald (5): ASoC: cs35l56: Move runtime suspend/resume to shared library ASoC: cs35l56: Move cs_dsp init into shared library ASoC: cs35l56: Move part of cs35l56_init() to shared library ASoC: cs35l56: Pass correct pointer to cs35l56_irq() ALSA: hda: Fix missing header dependencies Simon Trimmer (8): ASoC: cs35l56: Move shared data into a common data structure ASoC: cs35l56: Make cs35l56_system_reset() code more generic ASoC: cs35l56: Convert utility functions to use common data structure ASoC: cs35l56: Move utility functions to shared file ASoC: cs35l56: Make common function for control port wait ASoC: cs35l56: Make a common function to shutdown the DSP ALSA: hda: Add mute_hook to hda_component ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier include/sound/cs35l56.h | 29 +- sound/pci/hda/Kconfig | 31 + sound/pci/hda/Makefile | 6 + sound/pci/hda/cs35l56_hda.c | 995 ++++++++++++++++++++++++++++++ sound/pci/hda/cs35l56_hda.h | 48 ++ sound/pci/hda/cs35l56_hda_i2c.c | 69 +++ sound/pci/hda/cs35l56_hda_spi.c | 68 ++ sound/pci/hda/hda_auto_parser.h | 2 + sound/pci/hda/hda_component.h | 1 + sound/pci/hda/hda_generic.h | 3 + sound/soc/codecs/cs35l56-i2c.c | 14 +- sound/soc/codecs/cs35l56-sdw.c | 72 +-- sound/soc/codecs/cs35l56-shared.c | 459 +++++++++++++- sound/soc/codecs/cs35l56-spi.c | 10 +- sound/soc/codecs/cs35l56.c | 648 ++++--------------- sound/soc/codecs/cs35l56.h | 15 +- 16 files changed, 1878 insertions(+), 592 deletions(-) create mode 100644 sound/pci/hda/cs35l56_hda.c create mode 100644 sound/pci/hda/cs35l56_hda.h create mode 100644 sound/pci/hda/cs35l56_hda_i2c.c create mode 100644 sound/pci/hda/cs35l56_hda_spi.c -- 2.30.2

3 19

Failed to import use case configuration
by GitHub issues - opened 26 May '23

26 May '23

alsa-project/alsa-utils issue #217 was opened from jesusjimsa: Hello, I have stored the configuration for my sound card in an asound.state file and `alsactl` is reading it fine, but during boot and whenever I try running the command manually, this message appears in the log: ``` alsa-lib ../../../git/src/ucm/main.c:1412:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -2 ``` I don't understand what is going wrong and I haven't found much online. Any ideas for how I could fix this error or what might be the origin of it? This is what I get if I run `aplay -l`: ``` ~# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mycard [mycard], device 0: 44004000.audio-controller-tlv320aic31xx-hifi tlv320aic31xx-hifi [44004000.audio-controller-tlv320aic31xx-hifi tlv320aic31xx-hifi] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` Thank you. Issue URL : https://github.com/alsa-project/alsa-utils/issues/217 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v6 0/8] Add SC7280 audioreach device tree nodes
by Mohammad Rafi Shaik 26 May '23

26 May '23

Add SC7280 audioreach device tree nodes and extract audio specific dtsi nodes and add them in new file. This patch series depends on: -- https://patchwork.kernel.org/project/linux-clk/list/?series=717985 Corresponding dt-bindings not mainlined yet. -- https://git.kernel.org/pub/scm/linux/kernel/git/remoteproc/linux.git/commit… Changes Since v5: -- Re-arrange the patch list, driver changes should come after binding. -- Remove incorrect dai cells property in "Add sound node for crd-rev3 board" patch. -- Remove newlines in "Add LPASS PIL node" patch. -- Update the commit message in "Update VA/RX/TX macro clock nodes" patch. -- Update the commit message in "Update lpass_tlmm node" patch. Changes Since v4: -- Modify lpasscc clock controller node name. -- Disable lpass_core node. -- Modify Model name in sound node in "Add sound node for crd-rev3 board" patch. -- Remove protection domain property in "Add LPASS PIL node". Changes Since v3: -- Remove deleting digital codecs in crd-rev3 board specific dtsi and upadate them using phandle. -- Update commit message in "Update lpass_tlmm node" patch. -- Change the position of status property in LPASS PIL node. -- Update commit message in "Add sound node" patch. Changes Since v2: -- Remove Patch related to Add CGCR reset property. -- Remove Patch related to Disable legacy path clock nodes. -- Add dt-bindings for missing properties. -- Change the order of nodes. -- Move digictal codec macro nodes to root node from soc node. -- Add adsp-pil-mode property in required clock nodes. Changes Since v1: -- Move remoteproc node to soc dtsi file. -- Add qcom, adsp-pil-mode reg property in lpasscc node. -- Fix typo errors. -- Remove redundant status properties. Srinivasa Rao Mandadapu (8): arm64: dts: qcom: sc7280: Modify lpasscc node name dt-bindings: remoteproc: qcom: sc7280-adsp-pil: Add missing properties arm64: dts: qcom: sc7280: Extract audio nodes from common idp dtsi file arm64: dts: qcom: sc7280: Add sound node for crd-rev3 board arm64: dts: qcom: sc7280: Add LPASS PIL node arm64: dts: qcom: sc7280: Modify VA/RX/TX macro clock nodes for audioreach solution arm64: dts: qcom: sc7280: Modify LPASS_MCC reg region size in the lpass_tlmm node arm64: dts: qcom: sc7280: Add qcom,adsp-pil-mode property in clock nodes .../remoteproc/qcom,sc7280-adsp-pil.yaml | 30 ++- .../arm64/boot/dts/qcom/sc7280-audio-idp.dtsi | 131 +++++++++ arch/arm64/boot/dts/qcom/sc7280-crd-r3.dts | 1 + .../sc7280-herobrine-audioreach-wcd9385.dtsi | 253 ++++++++++++++++++ arch/arm64/boot/dts/qcom/sc7280-idp.dtsi | 122 --------- arch/arm64/boot/dts/qcom/sc7280.dtsi | 4 +- 6 files changed, 413 insertions(+), 128 deletions(-) create mode 100644 arch/arm64/boot/dts/qcom/sc7280-audio-idp.dtsi create mode 100644 arch/arm64/boot/dts/qcom/sc7280-herobrine-audioreach-wcd9385.dtsi -- 2.25.1

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[tiwai-sound:topic/midi20 25/40] sound/core/seq/seq_clientmgr.c:482 snd_seq_read() error: copy_to_user() '&cell->event' too small (28 vs 32)
by Dan Carpenter 26 May '23

26 May '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/midi20 head: 77700b81bd0e47d89d50eb4b3f2f323492f79998 commit: 46397622a3fa8372b8fda0f04b33d16923b03b1b [25/40] ALSA: seq: Add UMP support config: i386-randconfig-m021-20230525 (https://download.01.org/0day-ci/archive/20230526/202305261415.NY0vapZK-lkp@…) compiler: gcc-11 (Debian 11.3.0-12) 11.3.0 If you fix the issue, kindly add following tag where applicable | Reported-by: kernel test robot <lkp(a)intel.com> | Reported-by: Dan Carpenter <error27(a)gmail.com> | Closes: https://lore.kernel.org/r/202305261415.NY0vapZK-lkp@intel.com/ smatch warnings: sound/core/seq/seq_clientmgr.c:482 snd_seq_read() error: copy_to_user() '&cell->event' too small (28 vs 32) vim +482 sound/core/seq/seq_clientmgr.c c7e0b5bf9fff1b Takashi Iwai 2005-11-17 407 static ssize_t snd_seq_read(struct file *file, char __user *buf, size_t count, c7e0b5bf9fff1b Takashi Iwai 2005-11-17 408 loff_t *offset) ^1da177e4c3f41 Linus Torvalds 2005-04-16 409 { c7e0b5bf9fff1b Takashi Iwai 2005-11-17 410 struct snd_seq_client *client = file->private_data; c7e0b5bf9fff1b Takashi Iwai 2005-11-17 411 struct snd_seq_fifo *fifo; 46397622a3fa83 Takashi Iwai 2023-05-23 412 size_t aligned_size; ^1da177e4c3f41 Linus Torvalds 2005-04-16 413 int err; ^1da177e4c3f41 Linus Torvalds 2005-04-16 414 long result = 0; c7e0b5bf9fff1b Takashi Iwai 2005-11-17 415 struct snd_seq_event_cell *cell; ^1da177e4c3f41 Linus Torvalds 2005-04-16 416 ^1da177e4c3f41 Linus Torvalds 2005-04-16 417 if (!(snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_INPUT)) ^1da177e4c3f41 Linus Torvalds 2005-04-16 418 return -ENXIO; ^1da177e4c3f41 Linus Torvalds 2005-04-16 419 96d4f267e40f95 Linus Torvalds 2019-01-03 420 if (!access_ok(buf, count)) ^1da177e4c3f41 Linus Torvalds 2005-04-16 421 return -EFAULT; ^1da177e4c3f41 Linus Torvalds 2005-04-16 422 ^1da177e4c3f41 Linus Torvalds 2005-04-16 423 /* check client structures are in place */ 7eaa943c8ed8e9 Takashi Iwai 2008-08-08 424 if (snd_BUG_ON(!client)) 7eaa943c8ed8e9 Takashi Iwai 2008-08-08 425 return -ENXIO; ^1da177e4c3f41 Linus Torvalds 2005-04-16 426 f9a6bb841f7370 Takashi Iwai 2021-06-08 427 if (!client->accept_input) f9a6bb841f7370 Takashi Iwai 2021-06-08 428 return -ENXIO; f9a6bb841f7370 Takashi Iwai 2021-06-08 429 fifo = client->data.user.fifo; f9a6bb841f7370 Takashi Iwai 2021-06-08 430 if (!fifo) ^1da177e4c3f41 Linus Torvalds 2005-04-16 431 return -ENXIO; ^1da177e4c3f41 Linus Torvalds 2005-04-16 432 ^1da177e4c3f41 Linus Torvalds 2005-04-16 433 if (atomic_read(&fifo->overflow) > 0) { ^1da177e4c3f41 Linus Torvalds 2005-04-16 434 /* buffer overflow is detected */ ^1da177e4c3f41 Linus Torvalds 2005-04-16 435 snd_seq_fifo_clear(fifo); ^1da177e4c3f41 Linus Torvalds 2005-04-16 436 /* return error code */ ^1da177e4c3f41 Linus Torvalds 2005-04-16 437 return -ENOSPC; ^1da177e4c3f41 Linus Torvalds 2005-04-16 438 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 439 ^1da177e4c3f41 Linus Torvalds 2005-04-16 440 cell = NULL; ^1da177e4c3f41 Linus Torvalds 2005-04-16 441 err = 0; ^1da177e4c3f41 Linus Torvalds 2005-04-16 442 snd_seq_fifo_lock(fifo); ^1da177e4c3f41 Linus Torvalds 2005-04-16 443 46397622a3fa83 Takashi Iwai 2023-05-23 444 if (client->midi_version > 0) 46397622a3fa83 Takashi Iwai 2023-05-23 445 aligned_size = sizeof(struct snd_seq_ump_event); 46397622a3fa83 Takashi Iwai 2023-05-23 446 else 46397622a3fa83 Takashi Iwai 2023-05-23 447 aligned_size = sizeof(struct snd_seq_event); 46397622a3fa83 Takashi Iwai 2023-05-23 448 ^1da177e4c3f41 Linus Torvalds 2005-04-16 449 /* while data available in queue */ 46397622a3fa83 Takashi Iwai 2023-05-23 450 while (count >= aligned_size) { ^1da177e4c3f41 Linus Torvalds 2005-04-16 451 int nonblock; ^1da177e4c3f41 Linus Torvalds 2005-04-16 452 ^1da177e4c3f41 Linus Torvalds 2005-04-16 453 nonblock = (file->f_flags & O_NONBLOCK) || result > 0; f9a6bb841f7370 Takashi Iwai 2021-06-08 454 err = snd_seq_fifo_cell_out(fifo, &cell, nonblock); f9a6bb841f7370 Takashi Iwai 2021-06-08 455 if (err < 0) ^1da177e4c3f41 Linus Torvalds 2005-04-16 456 break; 46397622a3fa83 Takashi Iwai 2023-05-23 457 if (!event_is_compatible(client, &cell->event)) { 46397622a3fa83 Takashi Iwai 2023-05-23 458 snd_seq_cell_free(cell); 46397622a3fa83 Takashi Iwai 2023-05-23 459 cell = NULL; 46397622a3fa83 Takashi Iwai 2023-05-23 460 continue; 46397622a3fa83 Takashi Iwai 2023-05-23 461 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 462 if (snd_seq_ev_is_variable(&cell->event)) { Smatch isn't clever enough to know that if snd_seq_ev_is_variable() is false that means client->midi_version is zero. 46397622a3fa83 Takashi Iwai 2023-05-23 463 struct snd_seq_ump_event tmpev; 46397622a3fa83 Takashi Iwai 2023-05-23 464 46397622a3fa83 Takashi Iwai 2023-05-23 465 memcpy(&tmpev, &cell->event, aligned_size); ^1da177e4c3f41 Linus Torvalds 2005-04-16 466 tmpev.data.ext.len &= ~SNDRV_SEQ_EXT_MASK; 46397622a3fa83 Takashi Iwai 2023-05-23 467 if (copy_to_user(buf, &tmpev, aligned_size)) { ^1da177e4c3f41 Linus Torvalds 2005-04-16 468 err = -EFAULT; ^1da177e4c3f41 Linus Torvalds 2005-04-16 469 break; ^1da177e4c3f41 Linus Torvalds 2005-04-16 470 } 46397622a3fa83 Takashi Iwai 2023-05-23 471 count -= aligned_size; 46397622a3fa83 Takashi Iwai 2023-05-23 472 buf += aligned_size; 4d23359b7ec8b0 Clemens Ladisch 2005-09-05 473 err = snd_seq_expand_var_event(&cell->event, count, 4d23359b7ec8b0 Clemens Ladisch 2005-09-05 474 (char __force *)buf, 0, 46397622a3fa83 Takashi Iwai 2023-05-23 475 aligned_size); ^1da177e4c3f41 Linus Torvalds 2005-04-16 476 if (err < 0) ^1da177e4c3f41 Linus Torvalds 2005-04-16 477 break; ^1da177e4c3f41 Linus Torvalds 2005-04-16 478 result += err; ^1da177e4c3f41 Linus Torvalds 2005-04-16 479 count -= err; ^1da177e4c3f41 Linus Torvalds 2005-04-16 480 buf += err; ^1da177e4c3f41 Linus Torvalds 2005-04-16 481 } else { 46397622a3fa83 Takashi Iwai 2023-05-23 @482 if (copy_to_user(buf, &cell->event, aligned_size)) { So smatch complains that this might be a read overflow. ^1da177e4c3f41 Linus Torvalds 2005-04-16 483 err = -EFAULT; ^1da177e4c3f41 Linus Torvalds 2005-04-16 484 break; ^1da177e4c3f41 Linus Torvalds 2005-04-16 485 } 46397622a3fa83 Takashi Iwai 2023-05-23 486 count -= aligned_size; 46397622a3fa83 Takashi Iwai 2023-05-23 487 buf += aligned_size; ^1da177e4c3f41 Linus Torvalds 2005-04-16 488 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 489 snd_seq_cell_free(cell); ^1da177e4c3f41 Linus Torvalds 2005-04-16 490 cell = NULL; /* to be sure */ 46397622a3fa83 Takashi Iwai 2023-05-23 491 result += aligned_size; ^1da177e4c3f41 Linus Torvalds 2005-04-16 492 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 493 ^1da177e4c3f41 Linus Torvalds 2005-04-16 494 if (err < 0) { ^1da177e4c3f41 Linus Torvalds 2005-04-16 495 if (cell) ^1da177e4c3f41 Linus Torvalds 2005-04-16 496 snd_seq_fifo_cell_putback(fifo, cell); ^1da177e4c3f41 Linus Torvalds 2005-04-16 497 if (err == -EAGAIN && result > 0) ^1da177e4c3f41 Linus Torvalds 2005-04-16 498 err = 0; ^1da177e4c3f41 Linus Torvalds 2005-04-16 499 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 500 snd_seq_fifo_unlock(fifo); ^1da177e4c3f41 Linus Torvalds 2005-04-16 501 ^1da177e4c3f41 Linus Torvalds 2005-04-16 502 return (err < 0) ? err : result; ^1da177e4c3f41 Linus Torvalds 2005-04-16 503 } -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[tiwai-sound:topic/midi20 9/40] sound/usb/midi2.c:548:16: sparse: sparse: cast to restricted __le16
by kernel test robot 26 May '23

26 May '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/midi20 head: 77700b81bd0e47d89d50eb4b3f2f323492f79998 commit: ff49d1df79aef7580fe3ac99d17c3f886655d080 [9/40] ALSA: usb-audio: USB MIDI 2.0 UMP support config: openrisc-randconfig-s051-20230525 compiler: or1k-linux-gcc (GCC) 12.1.0 reproduce: mkdir -p ~/bin wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # apt-get install sparse # sparse version: v0.6.4-39-gce1a6720-dirty # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add tiwai-sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags tiwai-sound topic/midi20 git checkout ff49d1df79aef7580fe3ac99d17c3f886655d080 # save the config file mkdir build_dir && cp config build_dir/.config COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-12.1.0 ~/bin/make.cross C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' O=build_dir ARCH=openrisc olddefconfig COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-12.1.0 ~/bin/make.cross C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' O=build_dir ARCH=openrisc SHELL=/bin/bash sound/usb/ If you fix the issue, kindly add following tag where applicable | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202305260528.wcqjXso8-lkp@intel.com/ sparse warnings: (new ones prefixed by >>) >> sound/usb/midi2.c:548:16: sparse: sparse: cast to restricted __le16 >> sound/usb/midi2.c:548:16: sparse: sparse: cast to restricted __le16 >> sound/usb/midi2.c:548:16: sparse: sparse: cast to restricted __le16 >> sound/usb/midi2.c:548:16: sparse: sparse: cast to restricted __le16 vim +548 sound/usb/midi2.c 530 531 /* get the full group terminal block descriptors and return the size */ 532 static int get_group_terminal_block_descs(struct snd_usb_midi2_interface *umidi) 533 { 534 struct usb_host_interface *hostif = umidi->hostif; 535 struct usb_device *dev = umidi->chip->dev; 536 struct usb_ms20_gr_trm_block_header_descriptor header = { 0 }; 537 unsigned char *data; 538 int err, size; 539 540 err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 541 USB_REQ_GET_DESCRIPTOR, 542 USB_RECIP_INTERFACE | USB_TYPE_STANDARD | USB_DIR_IN, 543 USB_DT_CS_GR_TRM_BLOCK << 8 | hostif->desc.bAlternateSetting, 544 hostif->desc.bInterfaceNumber, 545 &header, sizeof(header)); 546 if (err < 0) 547 return err; > 548 size = __le16_to_cpu(header.wTotalLength); 549 if (!size) { 550 dev_err(&dev->dev, "Failed to get GTB descriptors for %d:%d\n", 551 hostif->desc.bInterfaceNumber, hostif->desc.bAlternateSetting); 552 return -EINVAL; 553 } 554 555 data = kzalloc(size, GFP_KERNEL); 556 if (!data) 557 return -ENOMEM; 558 559 err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 560 USB_REQ_GET_DESCRIPTOR, 561 USB_RECIP_INTERFACE | USB_TYPE_STANDARD | USB_DIR_IN, 562 USB_DT_CS_GR_TRM_BLOCK << 8 | hostif->desc.bAlternateSetting, 563 hostif->desc.bInterfaceNumber, data, size); 564 if (err < 0) { 565 kfree(data); 566 return err; 567 } 568 569 umidi->blk_descs = data; 570 umidi->blk_desc_size = size; 571 return 0; 572 } 573 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH AUTOSEL 4.14 01/20] ASoC: dwc: limit the number of overrun messages
by Sasha Levin 25 May '23

25 May '23

From: Maxim Kochetkov <fido_max(a)inbox.ru> [ Upstream commit ab6ecfbf40fccf74b6ec2ba7ed6dd2fc024c3af2 ] On slow CPU (FPGA/QEMU emulated) printing overrun messages from interrupt handler to uart console may leads to more overrun errors. So use dev_err_ratelimited to limit the number of error messages. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru Link: https://lore.kernel.org/r/20230505062820.21840-1-fido_max@inbox.ru Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/dwc/dwc-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index e27e21f8569a0..e6a0ec3c0e764 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -132,13 +132,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) /* Error Handling: TX */ if (isr[i] & ISR_TXFO) { - dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "TX overrun (ch_id=%d)\n", i); irq_valid = true; } /* Error Handling: TX */ if (isr[i] & ISR_RXFO) { - dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i); irq_valid = true; } } -- 2.39.2

1 2

[PATCH AUTOSEL 4.19 01/27] ASoC: dwc: limit the number of overrun messages
by Sasha Levin 25 May '23

25 May '23

From: Maxim Kochetkov <fido_max(a)inbox.ru> [ Upstream commit ab6ecfbf40fccf74b6ec2ba7ed6dd2fc024c3af2 ] On slow CPU (FPGA/QEMU emulated) printing overrun messages from interrupt handler to uart console may leads to more overrun errors. So use dev_err_ratelimited to limit the number of error messages. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru Link: https://lore.kernel.org/r/20230505062820.21840-1-fido_max@inbox.ru Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/dwc/dwc-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 65112b9d8588a..90b8814d7506a 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -132,13 +132,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) /* Error Handling: TX */ if (isr[i] & ISR_TXFO) { - dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "TX overrun (ch_id=%d)\n", i); irq_valid = true; } /* Error Handling: TX */ if (isr[i] & ISR_RXFO) { - dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i); irq_valid = true; } } -- 2.39.2

1 2

[PATCH AUTOSEL 5.4 01/27] ASoC: dwc: limit the number of overrun messages
by Sasha Levin 25 May '23

25 May '23

From: Maxim Kochetkov <fido_max(a)inbox.ru> [ Upstream commit ab6ecfbf40fccf74b6ec2ba7ed6dd2fc024c3af2 ] On slow CPU (FPGA/QEMU emulated) printing overrun messages from interrupt handler to uart console may leads to more overrun errors. So use dev_err_ratelimited to limit the number of error messages. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru Link: https://lore.kernel.org/r/20230505062820.21840-1-fido_max@inbox.ru Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/dwc/dwc-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 65112b9d8588a..90b8814d7506a 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -132,13 +132,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) /* Error Handling: TX */ if (isr[i] & ISR_TXFO) { - dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "TX overrun (ch_id=%d)\n", i); irq_valid = true; } /* Error Handling: TX */ if (isr[i] & ISR_RXFO) { - dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i); irq_valid = true; } } -- 2.39.2

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[PATCH AUTOSEL 5.10 01/31] ASoC: dwc: limit the number of overrun messages
by Sasha Levin 25 May '23

25 May '23

From: Maxim Kochetkov <fido_max(a)inbox.ru> [ Upstream commit ab6ecfbf40fccf74b6ec2ba7ed6dd2fc024c3af2 ] On slow CPU (FPGA/QEMU emulated) printing overrun messages from interrupt handler to uart console may leads to more overrun errors. So use dev_err_ratelimited to limit the number of error messages. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru Link: https://lore.kernel.org/r/20230505062820.21840-1-fido_max@inbox.ru Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/dwc/dwc-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 36da0f01571a1..5469399abcb44 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -132,13 +132,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) /* Error Handling: TX */ if (isr[i] & ISR_TXFO) { - dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "TX overrun (ch_id=%d)\n", i); irq_valid = true; } /* Error Handling: TX */ if (isr[i] & ISR_RXFO) { - dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i); irq_valid = true; } } -- 2.39.2

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[PATCH AUTOSEL 5.15 38/43] ALSA: oss: avoid missing-prototype warnings
by Sasha Levin 25 May '23

25 May '23

From: Arnd Bergmann <arnd(a)arndb.de> [ Upstream commit 040b5a046a9e18098580d3ccd029e2318fca7859 ] Two functions are defined and used in pcm_oss.c but also optionally used from io.c, with an optional prototype. If CONFIG_SND_PCM_OSS_PLUGINS is disabled, this causes a warning as the functions are not static and have no prototype: sound/core/oss/pcm_oss.c:1235:19: error: no previous prototype for 'snd_pcm_oss_write3' [-Werror=missing-prototypes] sound/core/oss/pcm_oss.c:1266:19: error: no previous prototype for 'snd_pcm_oss_read3' [-Werror=missing-prototypes] Avoid this by making the prototypes unconditional. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Link: https://lore.kernel.org/r/20230516195046.550584-2-arnd@kernel.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/oss/pcm_plugin.h | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index 46e273bd4a786..50a6b50f5db4c 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -141,6 +141,14 @@ int snd_pcm_area_copy(const struct snd_pcm_channel_area *src_channel, void *snd_pcm_plug_buf_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t size); void snd_pcm_plug_buf_unlock(struct snd_pcm_substream *plug, void *ptr); +#else + +static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; } +static inline snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size) { return clt_size; } +static inline int snd_pcm_plug_slave_format(int format, const struct snd_mask *format_mask) { return format; } + +#endif + snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const char *ptr, snd_pcm_uframes_t size, int in_kernel); @@ -151,14 +159,6 @@ snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream, snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream, void **bufs, snd_pcm_uframes_t frames); -#else - -static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; } -static inline snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size) { return clt_size; } -static inline int snd_pcm_plug_slave_format(int format, const struct snd_mask *format_mask) { return format; } - -#endif - #ifdef PLUGIN_DEBUG #define pdprintf(fmt, args...) printk(KERN_DEBUG "plugin: " fmt, ##args) #else -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 08/43] ASoC: ssm2602: Add workaround for playback distortions
by Sasha Levin 25 May '23

25 May '23

From: Paweł Anikiel <pan(a)semihalf.com> [ Upstream commit f63550e2b165208a2f382afcaf5551df9569e1d4 ] Apply a workaround for what appears to be a hardware quirk. The problem seems to happen when enabling "whole chip power" (bit D7 register R6) for the very first time after the chip receives power. If either "output" (D4) or "DAC" (D3) aren't powered on at that time, playback becomes very distorted later on. This happens on the Google Chameleon v3, as well as on a ZYBO Z7-10: https://ez.analog.com/audio/f/q-a/543726/solved-ssm2603-right-output-offset… I suspect this happens only when using an external MCLK signal (which is the case for both of these boards). Here are some experiments run on a Google Chameleon v3. These were run in userspace using a wrapper around the i2cset utility: ssmset() { i2cset -y 0 0x1a $(($1*2)) $2 } For each of the following sequences, we apply power to the ssm2603 chip, set the configuration registers R0-R5 and R7-R8, run the selected sequence, and check for distortions on playback. ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac OK ssmset 0x09 0x01 # core ssmset 0x06 0x87 # out, dac ssmset 0x06 0x07 # chip OK (disable MCLK) ssmset 0x09 0x01 # core ssmset 0x06 0x1f # chip ssmset 0x06 0x07 # out, dac (enable MCLK) OK ssmset 0x09 0x01 # core ssmset 0x06 0x1f # chip ssmset 0x06 0x07 # out, dac NOT OK ssmset 0x06 0x1f # chip ssmset 0x09 0x01 # core ssmset 0x06 0x07 # out, dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x0f # chip, out ssmset 0x06 0x07 # dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x17 # chip, dac ssmset 0x06 0x07 # out NOT OK For each of the following sequences, we apply power to the ssm2603 chip, run the selected sequence, issue a reset with R15, configure R0-R5 and R7-R8, run one of the NOT OK sequences from above, and check for distortions. ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac OK (disable MCLK) ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac (enable MCLK after reset) NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x17 # chip, dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x0f # chip, out NOT OK ssmset 0x06 0x07 # chip, out, dac NOT OK Signed-off-by: Paweł Anikiel <pan(a)semihalf.com Link: https://lore.kernel.org/r/20230508113037.137627-8-pan@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/ssm2602.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 7964e922b07f6..0a3540c5cdb5c 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -53,6 +53,18 @@ static const struct reg_default ssm2602_reg[SSM2602_CACHEREGNUM] = { { .reg = 0x09, .def = 0x0000 } }; +/* + * ssm2602 register patch + * Workaround for playback distortions after power up: activates digital + * core, and then powers on output, DAC, and whole chip at the same time + */ + +static const struct reg_sequence ssm2602_patch[] = { + { SSM2602_ACTIVE, 0x01 }, + { SSM2602_PWR, 0x07 }, + { SSM2602_RESET, 0x00 }, +}; + /*Appending several "None"s just for OSS mixer use*/ static const char *ssm2602_input_select[] = { @@ -589,6 +601,9 @@ static int ssm260x_component_probe(struct snd_soc_component *component) return ret; } + regmap_register_patch(ssm2602->regmap, ssm2602_patch, + ARRAY_SIZE(ssm2602_patch)); + /* set the update bits */ regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL, LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH); -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 07/43] ASoC: dt-bindings: Adjust #sound-dai-cells on TI's single-DAI codecs
by Sasha Levin 25 May '23

25 May '23

From: Martin Povišer <povik+lin(a)cutebit.org> [ Upstream commit efb2bfd7b3d210c479b9361c176d7426e5eb8663 ] A bunch of TI's codecs have binding schemas which force #sound-dai-cells to one despite those codecs only having a single DAI. Allow for bindings with zero DAI cells and deprecate the former non-zero value. Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org Link: https://lore.kernel.org/r/20230509153412.62847-1-povik+lin@cutebit.org Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- Documentation/devicetree/bindings/sound/tas2562.yaml | 6 ++++-- Documentation/devicetree/bindings/sound/tas2764.yaml | 6 ++++-- Documentation/devicetree/bindings/sound/tas2770.yaml | 6 ++++-- 3 files changed, 12 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml index acd4bbe697315..4adaf92233c8e 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.yaml +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -52,7 +52,9 @@ properties: description: TDM TX current sense time slot. '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -69,7 +71,7 @@ examples: codec: codec@4c { compatible = "ti,tas2562"; reg = <0x4c>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; shutdown-gpios = <&gpio1 15 0>; diff --git a/Documentation/devicetree/bindings/sound/tas2764.yaml b/Documentation/devicetree/bindings/sound/tas2764.yaml index 5bf8c76ecda11..1ffe1a01668fe 100644 --- a/Documentation/devicetree/bindings/sound/tas2764.yaml +++ b/Documentation/devicetree/bindings/sound/tas2764.yaml @@ -46,7 +46,9 @@ properties: description: TDM TX voltage sense time slot. '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -63,7 +65,7 @@ examples: codec: codec@38 { compatible = "ti,tas2764"; reg = <0x38>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; reset-gpios = <&gpio1 15 0>; diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml index 027bebf4e8cf5..aceba9ed813ef 100644 --- a/Documentation/devicetree/bindings/sound/tas2770.yaml +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -54,7 +54,9 @@ properties: - 1 # Falling edge '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -71,7 +73,7 @@ examples: codec: codec@41 { compatible = "ti,tas2770"; reg = <0x41>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; reset-gpio = <&gpio1 15 0>; -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 03/43] ASoC: dwc: limit the number of overrun messages
by Sasha Levin 25 May '23

25 May '23

From: Maxim Kochetkov <fido_max(a)inbox.ru> [ Upstream commit ab6ecfbf40fccf74b6ec2ba7ed6dd2fc024c3af2 ] On slow CPU (FPGA/QEMU emulated) printing overrun messages from interrupt handler to uart console may leads to more overrun errors. So use dev_err_ratelimited to limit the number of error messages. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru Link: https://lore.kernel.org/r/20230505062820.21840-1-fido_max@inbox.ru Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/dwc/dwc-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 315ca5c4b057a..14ad8023fcdcb 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -132,13 +132,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) /* Error Handling: TX */ if (isr[i] & ISR_TXFO) { - dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "TX overrun (ch_id=%d)\n", i); irq_valid = true; } /* Error Handling: TX */ if (isr[i] & ISR_RXFO) { - dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i); irq_valid = true; } } -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 49/57] ALSA: oss: avoid missing-prototype warnings
by Sasha Levin 25 May '23

25 May '23

From: Arnd Bergmann <arnd(a)arndb.de> [ Upstream commit 040b5a046a9e18098580d3ccd029e2318fca7859 ] Two functions are defined and used in pcm_oss.c but also optionally used from io.c, with an optional prototype. If CONFIG_SND_PCM_OSS_PLUGINS is disabled, this causes a warning as the functions are not static and have no prototype: sound/core/oss/pcm_oss.c:1235:19: error: no previous prototype for 'snd_pcm_oss_write3' [-Werror=missing-prototypes] sound/core/oss/pcm_oss.c:1266:19: error: no previous prototype for 'snd_pcm_oss_read3' [-Werror=missing-prototypes] Avoid this by making the prototypes unconditional. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Link: https://lore.kernel.org/r/20230516195046.550584-2-arnd@kernel.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/oss/pcm_plugin.h | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index 46e273bd4a786..50a6b50f5db4c 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -141,6 +141,14 @@ int snd_pcm_area_copy(const struct snd_pcm_channel_area *src_channel, void *snd_pcm_plug_buf_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t size); void snd_pcm_plug_buf_unlock(struct snd_pcm_substream *plug, void *ptr); +#else + +static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; } +static inline snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size) { return clt_size; } +static inline int snd_pcm_plug_slave_format(int format, const struct snd_mask *format_mask) { return format; } + +#endif + snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const char *ptr, snd_pcm_uframes_t size, int in_kernel); @@ -151,14 +159,6 @@ snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream, snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream, void **bufs, snd_pcm_uframes_t frames); -#else - -static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; } -static inline snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size) { return clt_size; } -static inline int snd_pcm_plug_slave_format(int format, const struct snd_mask *format_mask) { return format; } - -#endif - #ifdef PLUGIN_DEBUG #define pdprintf(fmt, args...) printk(KERN_DEBUG "plugin: " fmt, ##args) #else -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 29/57] ASoC: SOF: pm: save io region state in case of errors in resume
by Sasha Levin 25 May '23

25 May '23

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 171b53be635ac15d4feafeb33946035649b1ca14 ] If there are failures in DSP runtime resume, the device state will not reach active and this makes it impossible e.g. to retrieve a possible DSP panic dump via "exception" debugfs node. If CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE=y is set, the data in cache is stale. If debugfs cache is not used, the region simply cannot be read. To allow debugging these scenarios, update the debugfs cache contents in resume error handler. User-space can then later retrieve DSP panic and other state via debugfs (requires SOF debugfs cache to be enabled in build). Reported-by: Curtis Malainey <cujomalainey(a)chromium.org Link: https://github.com/thesofproject/linux/issues/4274 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Reviewed-by: Curtis Malainey <cujomalainey(a)chromium.org Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com Link: https://lore.kernel.org/r/20230512104638.21376-1-peter.ujfalusi@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/pm.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 26ffcbb6e30f4..a1bfa5a37e2ad 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -159,7 +159,7 @@ static int sof_resume(struct device *dev, bool runtime_resume) ret = tplg_ops->set_up_all_pipelines(sdev, false); if (ret < 0) { dev_err(sdev->dev, "Failed to restore pipeline after resume %d\n", ret); - return ret; + goto setup_fail; } } @@ -173,6 +173,18 @@ static int sof_resume(struct device *dev, bool runtime_resume) dev_err(sdev->dev, "ctx_restore IPC error during resume: %d\n", ret); } +setup_fail: +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) + if (ret < 0) { + /* + * Debugfs cannot be read in runtime suspend, so cache + * the contents upon failure. This allows to capture + * possible DSP coredump information. + */ + sof_cache_debugfs(sdev); + } +#endif + return ret; } -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 27/57] ASoC: SOF: pcm: fix pm_runtime imbalance in error handling
by Sasha Levin 25 May '23

25 May '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit da0fe8fd515a471d373acc3682bfb5522cca4d55 ] When an error occurs, we need to make sure the device can pm_runtime suspend instead of keeping it active. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-3-peter.ujfalusi@linux.intel.… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/pcm.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 14571b821ecac..be6f38af37b5d 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -619,16 +619,17 @@ static int sof_pcm_probe(struct snd_soc_component *component) "%s/%s", plat_data->tplg_filename_prefix, plat_data->tplg_filename); - if (!tplg_filename) - return -ENOMEM; + if (!tplg_filename) { + ret = -ENOMEM; + goto pm_error; + } ret = snd_sof_load_topology(component, tplg_filename); - if (ret < 0) { + if (ret < 0) dev_err(component->dev, "error: failed to load DSP topology %d\n", ret); - return ret; - } +pm_error: pm_runtime_mark_last_busy(component->dev); pm_runtime_put_autosuspend(component->dev); -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 26/57] ASoC: SOF: debug: conditionally bump runtime_pm counter on exceptions
by Sasha Levin 25 May '23

25 May '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 3de975862f985f1c9e225a0d13aa3d501373f7c3 ] When a firmware IPC error happens during a pm_runtime suspend, we ignore the error and suspend anyways. However, the code unconditionally increases the runtime_pm counter. This results in a confusing configuration where the code will suspend, resume but never suspend again due to the use of pm_runtime_get_noresume(). The intent of the counter increase was to prevent entry in D3, but if that transition to D3 is already started it cannot be stopped. In addition, there's no point in that case in trying to prevent anything, the firmware error is handled and the next resume will re-initialize the firmware completely. This patch changes the logic to prevent suspend when the device is pm_runtime active and has a use_count > 0. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-2-peter.ujfalusi@linux.intel.… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/debug.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index ade0507328af4..5042312b1b98d 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -437,8 +437,8 @@ void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev, const char *msg) /* should we prevent DSP entering D3 ? */ if (!sdev->ipc_dump_printed) dev_info(sdev->dev, - "preventing DSP entering D3 state to preserve context\n"); - pm_runtime_get_noresume(sdev->dev); + "Attempting to prevent DSP from entering D3 state to preserve context\n"); + pm_runtime_get_if_in_use(sdev->dev); } /* dump vital information to the logs */ -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 10/57] ASoC: ssm2602: Add workaround for playback distortions
by Sasha Levin 25 May '23

25 May '23

From: Paweł Anikiel <pan(a)semihalf.com> [ Upstream commit f63550e2b165208a2f382afcaf5551df9569e1d4 ] Apply a workaround for what appears to be a hardware quirk. The problem seems to happen when enabling "whole chip power" (bit D7 register R6) for the very first time after the chip receives power. If either "output" (D4) or "DAC" (D3) aren't powered on at that time, playback becomes very distorted later on. This happens on the Google Chameleon v3, as well as on a ZYBO Z7-10: https://ez.analog.com/audio/f/q-a/543726/solved-ssm2603-right-output-offset… I suspect this happens only when using an external MCLK signal (which is the case for both of these boards). Here are some experiments run on a Google Chameleon v3. These were run in userspace using a wrapper around the i2cset utility: ssmset() { i2cset -y 0 0x1a $(($1*2)) $2 } For each of the following sequences, we apply power to the ssm2603 chip, set the configuration registers R0-R5 and R7-R8, run the selected sequence, and check for distortions on playback. ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac OK ssmset 0x09 0x01 # core ssmset 0x06 0x87 # out, dac ssmset 0x06 0x07 # chip OK (disable MCLK) ssmset 0x09 0x01 # core ssmset 0x06 0x1f # chip ssmset 0x06 0x07 # out, dac (enable MCLK) OK ssmset 0x09 0x01 # core ssmset 0x06 0x1f # chip ssmset 0x06 0x07 # out, dac NOT OK ssmset 0x06 0x1f # chip ssmset 0x09 0x01 # core ssmset 0x06 0x07 # out, dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x0f # chip, out ssmset 0x06 0x07 # dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x17 # chip, dac ssmset 0x06 0x07 # out NOT OK For each of the following sequences, we apply power to the ssm2603 chip, run the selected sequence, issue a reset with R15, configure R0-R5 and R7-R8, run one of the NOT OK sequences from above, and check for distortions. ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac OK (disable MCLK) ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac (enable MCLK after reset) NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x17 # chip, dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x0f # chip, out NOT OK ssmset 0x06 0x07 # chip, out, dac NOT OK Signed-off-by: Paweł Anikiel <pan(a)semihalf.com Link: https://lore.kernel.org/r/20230508113037.137627-8-pan@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/ssm2602.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index cbbe83b85adaf..cf7927222be15 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -53,6 +53,18 @@ static const struct reg_default ssm2602_reg[SSM2602_CACHEREGNUM] = { { .reg = 0x09, .def = 0x0000 } }; +/* + * ssm2602 register patch + * Workaround for playback distortions after power up: activates digital + * core, and then powers on output, DAC, and whole chip at the same time + */ + +static const struct reg_sequence ssm2602_patch[] = { + { SSM2602_ACTIVE, 0x01 }, + { SSM2602_PWR, 0x07 }, + { SSM2602_RESET, 0x00 }, +}; + /*Appending several "None"s just for OSS mixer use*/ static const char *ssm2602_input_select[] = { @@ -589,6 +601,9 @@ static int ssm260x_component_probe(struct snd_soc_component *component) return ret; } + regmap_register_patch(ssm2602->regmap, ssm2602_patch, + ARRAY_SIZE(ssm2602_patch)); + /* set the update bits */ regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL, LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH); -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 09/57] ALSA: hda/realtek: Add quirks for ASUS GU604V and GU603V
by Sasha Levin 25 May '23

25 May '23

From: Alexandru Sorodoc <ealex95(a)gmail.com> [ Upstream commit 4b963ae1df6426f0e51de64133d379d9bde50c48 ] These models use 2 CS35L41 amplifiers using SPI for down-facing speakers. alc285_fixup_speaker2_to_dac1 is needed to fix volume control of the down-facing speakers. Pin configs are needed to enable headset mic detection. Note that these models lack the ACPI _DSD properties needed to initialize the amplifiers. They can be added during boot to get working sound out of the speakers: https://gist.github.com/lamperez/862763881c0e1c812392b5574727f6ff Signed-off-by: Alexandru Sorodoc <ealex95(a)gmail.com> Link: https://lore.kernel.org/r/20230511161510.315170-1-ealex95@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 172ffc2c332b7..543e0adde3faf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7063,6 +7063,8 @@ enum { ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, ALC285_FIXUP_SPEAKER2_TO_DAC1, + ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1, + ALC285_FIXUP_ASUS_HEADSET_MIC, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, @@ -8033,6 +8035,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, + [ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + .chained = true, + .chain_id = ALC245_FIXUP_CS35L41_SPI_2 + }, + [ALC285_FIXUP_ASUS_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, + { 0x1b, 0x03a11c30 }, + { } + }, + .chained = true, + .chain_id = ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1 + }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -9500,6 +9518,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), + SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604V", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 08/57] ASoC: dt-bindings: Adjust #sound-dai-cells on TI's single-DAI codecs
by Sasha Levin 25 May '23

25 May '23

From: Martin Povišer <povik+lin(a)cutebit.org> [ Upstream commit efb2bfd7b3d210c479b9361c176d7426e5eb8663 ] A bunch of TI's codecs have binding schemas which force #sound-dai-cells to one despite those codecs only having a single DAI. Allow for bindings with zero DAI cells and deprecate the former non-zero value. Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org Link: https://lore.kernel.org/r/20230509153412.62847-1-povik+lin@cutebit.org Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- Documentation/devicetree/bindings/sound/tas2562.yaml | 6 ++++-- Documentation/devicetree/bindings/sound/tas2770.yaml | 6 ++++-- Documentation/devicetree/bindings/sound/tas27xx.yaml | 6 ++++-- 3 files changed, 12 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml index 30f6b029ac085..cb519a4b6e710 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.yaml +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -52,7 +52,9 @@ properties: description: TDM TX current sense time slot. '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -69,7 +71,7 @@ examples: codec: codec@4c { compatible = "ti,tas2562"; reg = <0x4c>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; shutdown-gpios = <&gpio1 15 0>; diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml index bc90e72bf7cf9..1859fbe1cdf17 100644 --- a/Documentation/devicetree/bindings/sound/tas2770.yaml +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -54,7 +54,9 @@ properties: - 1 # Falling edge '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -71,7 +73,7 @@ examples: codec: codec@41 { compatible = "ti,tas2770"; reg = <0x41>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; reset-gpio = <&gpio1 15 0>; diff --git a/Documentation/devicetree/bindings/sound/tas27xx.yaml b/Documentation/devicetree/bindings/sound/tas27xx.yaml index 66a0df8850ea6..079cb6f8d4474 100644 --- a/Documentation/devicetree/bindings/sound/tas27xx.yaml +++ b/Documentation/devicetree/bindings/sound/tas27xx.yaml @@ -47,7 +47,9 @@ properties: description: TDM TX voltage sense time slot. '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -64,7 +66,7 @@ examples: codec: codec@38 { compatible = "ti,tas2764"; reg = <0x38>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; reset-gpios = <&gpio1 15 0>; -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 04/57] ASoC: dwc: limit the number of overrun messages
by Sasha Levin 25 May '23

25 May '23

From: Maxim Kochetkov <fido_max(a)inbox.ru> [ Upstream commit ab6ecfbf40fccf74b6ec2ba7ed6dd2fc024c3af2 ] On slow CPU (FPGA/QEMU emulated) printing overrun messages from interrupt handler to uart console may leads to more overrun errors. So use dev_err_ratelimited to limit the number of error messages. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru Link: https://lore.kernel.org/r/20230505062820.21840-1-fido_max@inbox.ru Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/dwc/dwc-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 7f7dd07c63b2f..3496301582b22 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -132,13 +132,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) /* Error Handling: TX */ if (isr[i] & ISR_TXFO) { - dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "TX overrun (ch_id=%d)\n", i); irq_valid = true; } /* Error Handling: TX */ if (isr[i] & ISR_RXFO) { - dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i); irq_valid = true; } } -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 03/57] ASoC: amd: yc: Add DMI entry to support System76 Pangolin 12
by Sasha Levin 25 May '23

25 May '23

From: Jeremy Soller <jeremy(a)system76.com> [ Upstream commit 7b9891ad25246b18b5ccc19518da7abc7763aa0a ] Add pang12 quirk to enable the internal microphone. Signed-off-by: Jeremy Soller <jeremy(a)system76.com Signed-off-by: Tim Crawford <tcrawford(a)system76.com Link: https://lore.kernel.org/r/20230505161458.19676-1-tcrawford@system76.com Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 0acdf0156f075..b844f321d139e 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -276,6 +276,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_BOARD_NAME, "8A22"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "System76"), + DMI_MATCH(DMI_PRODUCT_VERSION, "pang12"), + } + }, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 59/67] ALSA: oss: avoid missing-prototype warnings
by Sasha Levin 25 May '23

25 May '23

From: Arnd Bergmann <arnd(a)arndb.de> [ Upstream commit 040b5a046a9e18098580d3ccd029e2318fca7859 ] Two functions are defined and used in pcm_oss.c but also optionally used from io.c, with an optional prototype. If CONFIG_SND_PCM_OSS_PLUGINS is disabled, this causes a warning as the functions are not static and have no prototype: sound/core/oss/pcm_oss.c:1235:19: error: no previous prototype for 'snd_pcm_oss_write3' [-Werror=missing-prototypes] sound/core/oss/pcm_oss.c:1266:19: error: no previous prototype for 'snd_pcm_oss_read3' [-Werror=missing-prototypes] Avoid this by making the prototypes unconditional. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Link: https://lore.kernel.org/r/20230516195046.550584-2-arnd@kernel.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/oss/pcm_plugin.h | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index 46e273bd4a786..50a6b50f5db4c 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -141,6 +141,14 @@ int snd_pcm_area_copy(const struct snd_pcm_channel_area *src_channel, void *snd_pcm_plug_buf_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t size); void snd_pcm_plug_buf_unlock(struct snd_pcm_substream *plug, void *ptr); +#else + +static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; } +static inline snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size) { return clt_size; } +static inline int snd_pcm_plug_slave_format(int format, const struct snd_mask *format_mask) { return format; } + +#endif + snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const char *ptr, snd_pcm_uframes_t size, int in_kernel); @@ -151,14 +159,6 @@ snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream, snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream, void **bufs, snd_pcm_uframes_t frames); -#else - -static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; } -static inline snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size) { return clt_size; } -static inline int snd_pcm_plug_slave_format(int format, const struct snd_mask *format_mask) { return format; } - -#endif - #ifdef PLUGIN_DEBUG #define pdprintf(fmt, args...) printk(KERN_DEBUG "plugin: " fmt, ##args) #else -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 34/67] ASoC: SOF: pm: save io region state in case of errors in resume
by Sasha Levin 25 May '23

25 May '23

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 171b53be635ac15d4feafeb33946035649b1ca14 ] If there are failures in DSP runtime resume, the device state will not reach active and this makes it impossible e.g. to retrieve a possible DSP panic dump via "exception" debugfs node. If CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE=y is set, the data in cache is stale. If debugfs cache is not used, the region simply cannot be read. To allow debugging these scenarios, update the debugfs cache contents in resume error handler. User-space can then later retrieve DSP panic and other state via debugfs (requires SOF debugfs cache to be enabled in build). Reported-by: Curtis Malainey <cujomalainey(a)chromium.org Link: https://github.com/thesofproject/linux/issues/4274 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Reviewed-by: Curtis Malainey <cujomalainey(a)chromium.org Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com Link: https://lore.kernel.org/r/20230512104638.21376-1-peter.ujfalusi@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/pm.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 85412aeb1ca16..40f392efd8246 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -159,7 +159,7 @@ static int sof_resume(struct device *dev, bool runtime_resume) ret = tplg_ops->set_up_all_pipelines(sdev, false); if (ret < 0) { dev_err(sdev->dev, "Failed to restore pipeline after resume %d\n", ret); - return ret; + goto setup_fail; } } @@ -173,6 +173,18 @@ static int sof_resume(struct device *dev, bool runtime_resume) dev_err(sdev->dev, "ctx_restore IPC error during resume: %d\n", ret); } +setup_fail: +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) + if (ret < 0) { + /* + * Debugfs cannot be read in runtime suspend, so cache + * the contents upon failure. This allows to capture + * possible DSP coredump information. + */ + sof_cache_debugfs(sdev); + } +#endif + return ret; } -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 33/67] ASoC: SOF: sof-client-probes: fix pm_runtime imbalance in error handling
by Sasha Levin 25 May '23

25 May '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit bc424273c74c1565c459c8f2a6ed95caee368d0a ] When an error occurs, we need to make sure the device can pm_runtime suspend instead of keeping it active. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-4-peter.ujfalusi@linux.intel.… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/sof-client-probes.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c index fff126808bc04..8d9e9d5f40e45 100644 --- a/sound/soc/sof/sof-client-probes.c +++ b/sound/soc/sof/sof-client-probes.c @@ -218,12 +218,7 @@ static ssize_t sof_probes_dfs_points_read(struct file *file, char __user *to, ret = ipc->points_info(cdev, &desc, &num_desc); if (ret < 0) - goto exit; - - pm_runtime_mark_last_busy(dev); - err = pm_runtime_put_autosuspend(dev); - if (err < 0) - dev_err_ratelimited(dev, "debugfs read failed to idle %d\n", err); + goto pm_error; for (i = 0; i < num_desc; i++) { offset = strlen(buf); @@ -241,6 +236,13 @@ static ssize_t sof_probes_dfs_points_read(struct file *file, char __user *to, ret = simple_read_from_buffer(to, count, ppos, buf, strlen(buf)); kfree(desc); + +pm_error: + pm_runtime_mark_last_busy(dev); + err = pm_runtime_put_autosuspend(dev); + if (err < 0) + dev_err_ratelimited(dev, "debugfs read failed to idle %d\n", err); + exit: kfree(buf); return ret; -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 32/67] ASoC: SOF: pcm: fix pm_runtime imbalance in error handling
by Sasha Levin 25 May '23

25 May '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit da0fe8fd515a471d373acc3682bfb5522cca4d55 ] When an error occurs, we need to make sure the device can pm_runtime suspend instead of keeping it active. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-3-peter.ujfalusi@linux.intel.… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/pcm.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 445acb5c3a21b..2570f33db9f3e 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -616,16 +616,17 @@ static int sof_pcm_probe(struct snd_soc_component *component) "%s/%s", plat_data->tplg_filename_prefix, plat_data->tplg_filename); - if (!tplg_filename) - return -ENOMEM; + if (!tplg_filename) { + ret = -ENOMEM; + goto pm_error; + } ret = snd_sof_load_topology(component, tplg_filename); - if (ret < 0) { + if (ret < 0) dev_err(component->dev, "error: failed to load DSP topology %d\n", ret); - return ret; - } +pm_error: pm_runtime_mark_last_busy(component->dev); pm_runtime_put_autosuspend(component->dev); -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 31/67] ASoC: SOF: debug: conditionally bump runtime_pm counter on exceptions
by Sasha Levin 25 May '23

25 May '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 3de975862f985f1c9e225a0d13aa3d501373f7c3 ] When a firmware IPC error happens during a pm_runtime suspend, we ignore the error and suspend anyways. However, the code unconditionally increases the runtime_pm counter. This results in a confusing configuration where the code will suspend, resume but never suspend again due to the use of pm_runtime_get_noresume(). The intent of the counter increase was to prevent entry in D3, but if that transition to D3 is already started it cannot be stopped. In addition, there's no point in that case in trying to prevent anything, the firmware error is handled and the next resume will re-initialize the firmware completely. This patch changes the logic to prevent suspend when the device is pm_runtime active and has a use_count > 0. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-2-peter.ujfalusi@linux.intel.… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/debug.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index ade0507328af4..5042312b1b98d 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -437,8 +437,8 @@ void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev, const char *msg) /* should we prevent DSP entering D3 ? */ if (!sdev->ipc_dump_printed) dev_info(sdev->dev, - "preventing DSP entering D3 state to preserve context\n"); - pm_runtime_get_noresume(sdev->dev); + "Attempting to prevent DSP from entering D3 state to preserve context\n"); + pm_runtime_get_if_in_use(sdev->dev); } /* dump vital information to the logs */ -- 2.39.2

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[PATCH AUTOSEL 6.3 14/67] ASoC: ssm2602: Add workaround for playback distortions
by Sasha Levin 25 May '23

25 May '23

From: Paweł Anikiel <pan(a)semihalf.com> [ Upstream commit f63550e2b165208a2f382afcaf5551df9569e1d4 ] Apply a workaround for what appears to be a hardware quirk. The problem seems to happen when enabling "whole chip power" (bit D7 register R6) for the very first time after the chip receives power. If either "output" (D4) or "DAC" (D3) aren't powered on at that time, playback becomes very distorted later on. This happens on the Google Chameleon v3, as well as on a ZYBO Z7-10: https://ez.analog.com/audio/f/q-a/543726/solved-ssm2603-right-output-offset… I suspect this happens only when using an external MCLK signal (which is the case for both of these boards). Here are some experiments run on a Google Chameleon v3. These were run in userspace using a wrapper around the i2cset utility: ssmset() { i2cset -y 0 0x1a $(($1*2)) $2 } For each of the following sequences, we apply power to the ssm2603 chip, set the configuration registers R0-R5 and R7-R8, run the selected sequence, and check for distortions on playback. ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac OK ssmset 0x09 0x01 # core ssmset 0x06 0x87 # out, dac ssmset 0x06 0x07 # chip OK (disable MCLK) ssmset 0x09 0x01 # core ssmset 0x06 0x1f # chip ssmset 0x06 0x07 # out, dac (enable MCLK) OK ssmset 0x09 0x01 # core ssmset 0x06 0x1f # chip ssmset 0x06 0x07 # out, dac NOT OK ssmset 0x06 0x1f # chip ssmset 0x09 0x01 # core ssmset 0x06 0x07 # out, dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x0f # chip, out ssmset 0x06 0x07 # dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x17 # chip, dac ssmset 0x06 0x07 # out NOT OK For each of the following sequences, we apply power to the ssm2603 chip, run the selected sequence, issue a reset with R15, configure R0-R5 and R7-R8, run one of the NOT OK sequences from above, and check for distortions. ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac OK (disable MCLK) ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac (enable MCLK after reset) NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x17 # chip, dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x0f # chip, out NOT OK ssmset 0x06 0x07 # chip, out, dac NOT OK Signed-off-by: Paweł Anikiel <pan(a)semihalf.com Link: https://lore.kernel.org/r/20230508113037.137627-8-pan@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/ssm2602.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index cbbe83b85adaf..cf7927222be15 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -53,6 +53,18 @@ static const struct reg_default ssm2602_reg[SSM2602_CACHEREGNUM] = { { .reg = 0x09, .def = 0x0000 } }; +/* + * ssm2602 register patch + * Workaround for playback distortions after power up: activates digital + * core, and then powers on output, DAC, and whole chip at the same time + */ + +static const struct reg_sequence ssm2602_patch[] = { + { SSM2602_ACTIVE, 0x01 }, + { SSM2602_PWR, 0x07 }, + { SSM2602_RESET, 0x00 }, +}; + /*Appending several "None"s just for OSS mixer use*/ static const char *ssm2602_input_select[] = { @@ -589,6 +601,9 @@ static int ssm260x_component_probe(struct snd_soc_component *component) return ret; } + regmap_register_patch(ssm2602->regmap, ssm2602_patch, + ARRAY_SIZE(ssm2602_patch)); + /* set the update bits */ regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL, LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH); -- 2.39.2

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[PATCH AUTOSEL 6.3 13/67] ALSA: hda/realtek: Add quirks for ASUS GU604V and GU603V
by Sasha Levin 25 May '23

25 May '23

From: Alexandru Sorodoc <ealex95(a)gmail.com> [ Upstream commit 4b963ae1df6426f0e51de64133d379d9bde50c48 ] These models use 2 CS35L41 amplifiers using SPI for down-facing speakers. alc285_fixup_speaker2_to_dac1 is needed to fix volume control of the down-facing speakers. Pin configs are needed to enable headset mic detection. Note that these models lack the ACPI _DSD properties needed to initialize the amplifiers. They can be added during boot to get working sound out of the speakers: https://gist.github.com/lamperez/862763881c0e1c812392b5574727f6ff Signed-off-by: Alexandru Sorodoc <ealex95(a)gmail.com> Link: https://lore.kernel.org/r/20230511161510.315170-1-ealex95@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 172ffc2c332b7..543e0adde3faf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7063,6 +7063,8 @@ enum { ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, ALC285_FIXUP_SPEAKER2_TO_DAC1, + ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1, + ALC285_FIXUP_ASUS_HEADSET_MIC, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, @@ -8033,6 +8035,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, + [ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + .chained = true, + .chain_id = ALC245_FIXUP_CS35L41_SPI_2 + }, + [ALC285_FIXUP_ASUS_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, + { 0x1b, 0x03a11c30 }, + { } + }, + .chained = true, + .chain_id = ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1 + }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -9500,6 +9518,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), + SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604V", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), -- 2.39.2

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[PATCH AUTOSEL 6.3 12/67] ASoC: dt-bindings: Adjust #sound-dai-cells on TI's single-DAI codecs
by Sasha Levin 25 May '23

25 May '23

From: Martin Povišer <povik+lin(a)cutebit.org> [ Upstream commit efb2bfd7b3d210c479b9361c176d7426e5eb8663 ] A bunch of TI's codecs have binding schemas which force #sound-dai-cells to one despite those codecs only having a single DAI. Allow for bindings with zero DAI cells and deprecate the former non-zero value. Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org Link: https://lore.kernel.org/r/20230509153412.62847-1-povik+lin@cutebit.org Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- Documentation/devicetree/bindings/sound/tas2562.yaml | 6 ++++-- Documentation/devicetree/bindings/sound/tas2770.yaml | 6 ++++-- Documentation/devicetree/bindings/sound/tas27xx.yaml | 6 ++++-- 3 files changed, 12 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml index 1085592cefccc..81218c07079a8 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.yaml +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -55,7 +55,9 @@ properties: description: TDM TX current sense time slot. '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -72,7 +74,7 @@ examples: codec: codec@4c { compatible = "ti,tas2562"; reg = <0x4c>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; shutdown-gpios = <&gpio1 15 0>; diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml index 982949ba8a4be..cdb493db47f9b 100644 --- a/Documentation/devicetree/bindings/sound/tas2770.yaml +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -57,7 +57,9 @@ properties: - 1 # Falling edge '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -74,7 +76,7 @@ examples: codec: codec@41 { compatible = "ti,tas2770"; reg = <0x41>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; reset-gpio = <&gpio1 15 0>; diff --git a/Documentation/devicetree/bindings/sound/tas27xx.yaml b/Documentation/devicetree/bindings/sound/tas27xx.yaml index 0957dd435bb4b..2ef05aacc167a 100644 --- a/Documentation/devicetree/bindings/sound/tas27xx.yaml +++ b/Documentation/devicetree/bindings/sound/tas27xx.yaml @@ -50,7 +50,9 @@ properties: description: TDM TX voltage sense time slot. '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -67,7 +69,7 @@ examples: codec: codec@38 { compatible = "ti,tas2764"; reg = <0x38>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; reset-gpios = <&gpio1 15 0>; -- 2.39.2

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[PATCH AUTOSEL 6.3 11/67] ASoC: jz4740-i2s: Make I2S divider calculations more robust
by Sasha Levin 25 May '23

25 May '23

From: Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com> [ Upstream commit ad721bc919edfd8b4b06977458a412011e2f0c50 ] When the CPU supplies bit/frame clocks, the system clock (clk_i2s) is divided to produce the bit clock. This is a simple 1/N divider with a fairly limited range, so for a given system clock frequency only a few sample rates can be produced. Usually a wider range of sample rates is supported by varying the system clock frequency. The old calculation method was not very robust and could easily produce the wrong clock rate, especially with non-standard rates. For example, if the system clock is 1.99x the target bit clock rate, the divider would be calculated as 1 instead of the more accurate 2. Instead, use a more accurate method that considers two adjacent divider settings and selects the one that produces the least error versus the requested rate. If the error is 5% or higher then the rate setting is rejected to prevent garbled audio. Skip divider calculation when the codec is supplying both the bit and frame clock; in that case, the divider outputs are unused and we don't want to constrain the sample rate. Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com Link: https://lore.kernel.org/r/20230509125134.208129-1-aidanmacdonald.0x0@gmail.… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/jz4740/jz4740-i2s.c | 54 ++++++++++++++++++++++++++++++++--- 1 file changed, 50 insertions(+), 4 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 6d9cfe0a50411..d0f6c945d9aee 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -218,18 +218,48 @@ static int jz4740_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static int jz4740_i2s_get_i2sdiv(unsigned long mclk, unsigned long rate, + unsigned long i2sdiv_max) +{ + unsigned long div, rate1, rate2, err1, err2; + + div = mclk / (64 * rate); + if (div == 0) + div = 1; + + rate1 = mclk / (64 * div); + rate2 = mclk / (64 * (div + 1)); + + err1 = abs(rate1 - rate); + err2 = abs(rate2 - rate); + + /* + * Choose the divider that produces the smallest error in the + * output rate and reject dividers with a 5% or higher error. + * In the event that both dividers are outside the acceptable + * error margin, reject the rate to prevent distorted audio. + * (The number 5% is arbitrary.) + */ + if (div <= i2sdiv_max && err1 <= err2 && err1 < rate/20) + return div; + if (div < i2sdiv_max && err2 < rate/20) + return div + 1; + + return -EINVAL; +} + static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); struct regmap_field *div_field; + unsigned long i2sdiv_max; unsigned int sample_size; - uint32_t ctrl; - int div; + uint32_t ctrl, conf; + int div = 1; regmap_read(i2s->regmap, JZ_REG_AIC_CTRL, &ctrl); - - div = clk_get_rate(i2s->clk_i2s) / (64 * params_rate(params)); + regmap_read(i2s->regmap, JZ_REG_AIC_CONF, &conf); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: @@ -258,11 +288,27 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO; div_field = i2s->field_i2sdiv_playback; + i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_playback.msb, + i2s->soc_info->field_i2sdiv_playback.lsb); } else { ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE; ctrl |= FIELD_PREP(JZ_AIC_CTRL_INPUT_SAMPLE_SIZE, sample_size); div_field = i2s->field_i2sdiv_capture; + i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_capture.msb, + i2s->soc_info->field_i2sdiv_capture.lsb); + } + + /* + * Only calculate I2SDIV if we're supplying the bit or frame clock. + * If the codec is supplying both clocks then the divider output is + * unused, and we don't want it to limit the allowed sample rates. + */ + if (conf & (JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER)) { + div = jz4740_i2s_get_i2sdiv(clk_get_rate(i2s->clk_i2s), + params_rate(params), i2sdiv_max); + if (div < 0) + return div; } regmap_write(i2s->regmap, JZ_REG_AIC_CTRL, ctrl); -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 08/67] ASoC: SOF: amd: Fix NULL pointer crash in acp_sof_ipc_msg_data function
by Sasha Levin 25 May '23

25 May '23

From: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> [ Upstream commit 051d71e073614a72ad423d6dacba37a7eeff274d ] Check substream and runtime variables before assigning. Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com Link: https://lore.kernel.org/r/20230508070510.6100-1-Vsujithkumar.Reddy@amd.com Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/amd/acp-ipc.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/acp-ipc.c b/sound/soc/sof/amd/acp-ipc.c index 4e0c48a361599..749e856dc6011 100644 --- a/sound/soc/sof/amd/acp-ipc.c +++ b/sound/soc/sof/amd/acp-ipc.c @@ -209,7 +209,12 @@ int acp_sof_ipc_msg_data(struct snd_sof_dev *sdev, struct snd_sof_pcm_stream *sp acp_mailbox_read(sdev, offset, p, sz); } else { struct snd_pcm_substream *substream = sps->substream; - struct acp_dsp_stream *stream = substream->runtime->private_data; + struct acp_dsp_stream *stream; + + if (!substream || !substream->runtime) + return -ESTRPIPE; + + stream = substream->runtime->private_data; if (!stream) return -ESTRPIPE; -- 2.39.2

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[PATCH AUTOSEL 6.3 05/67] ASoC: dwc: limit the number of overrun messages
by Sasha Levin 25 May '23

25 May '23

From: Maxim Kochetkov <fido_max(a)inbox.ru> [ Upstream commit ab6ecfbf40fccf74b6ec2ba7ed6dd2fc024c3af2 ] On slow CPU (FPGA/QEMU emulated) printing overrun messages from interrupt handler to uart console may leads to more overrun errors. So use dev_err_ratelimited to limit the number of error messages. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru Link: https://lore.kernel.org/r/20230505062820.21840-1-fido_max@inbox.ru Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/dwc/dwc-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 7f7dd07c63b2f..3496301582b22 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -132,13 +132,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) /* Error Handling: TX */ if (isr[i] & ISR_TXFO) { - dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "TX overrun (ch_id=%d)\n", i); irq_valid = true; } /* Error Handling: TX */ if (isr[i] & ISR_RXFO) { - dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i); irq_valid = true; } } -- 2.39.2

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[PATCH AUTOSEL 6.3 04/67] ASoC: amd: yc: Add DMI entry to support System76 Pangolin 12
by Sasha Levin 25 May '23

25 May '23

From: Jeremy Soller <jeremy(a)system76.com> [ Upstream commit 7b9891ad25246b18b5ccc19518da7abc7763aa0a ] Add pang12 quirk to enable the internal microphone. Signed-off-by: Jeremy Soller <jeremy(a)system76.com Signed-off-by: Tim Crawford <tcrawford(a)system76.com Link: https://lore.kernel.org/r/20230505161458.19676-1-tcrawford@system76.com Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 0acdf0156f075..b844f321d139e 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -276,6 +276,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_BOARD_NAME, "8A22"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "System76"), + DMI_MATCH(DMI_PRODUCT_VERSION, "pang12"), + } + }, {} }; -- 2.39.2

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[PATCH v2 00/12] ALSA: hda: Adding support for CS35L56 on HDA systems
by Richard Fitzgerald 25 May '23

25 May '23

This set of patches adds support for using the CS35L56 boosted smart amplifier on HDA systems. In these systems the CS35L56 audio is routed through a HDA-to-I2S bridge codec. This doesn't include the changes to the Realtek driver to actually hook up the CS35L56 driver, because we don't yet have the QUIRK IDs to associate it with. But we want to publish the driver now so that it is available for bringing up hardware with the CS35L56. The first 9 patches are moving code out of the ASoC driver and into the shared library so that it can be shared with the HDA driver. Patch #10 fixes missing #includes in the HDA headers so that the CS35L56 driver doesn't have to #include headers that it doesn't use. Finally, #11 and #12 actually add the support for CS35L56 on HDA. CHANGES SINCE V1 Squash the fix for passing the correct pointer to cs35l56_irq() into patch #1, where the pointer change to cs35l56_irq() was made. Richard Fitzgerald (4): ASoC: cs35l56: Move runtime suspend/resume to shared library ASoC: cs35l56: Move cs_dsp init into shared library ASoC: cs35l56: Move part of cs35l56_init() to shared library ALSA: hda: Fix missing header dependencies Simon Trimmer (8): ASoC: cs35l56: Move shared data into a common data structure ASoC: cs35l56: Make cs35l56_system_reset() code more generic ASoC: cs35l56: Convert utility functions to use common data structure ASoC: cs35l56: Move utility functions to shared file ASoC: cs35l56: Make common function for control port wait ASoC: cs35l56: Make a common function to shutdown the DSP ALSA: hda: Add mute_hook to hda_component ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier include/sound/cs35l56.h | 29 +- sound/pci/hda/Kconfig | 31 + sound/pci/hda/Makefile | 6 + sound/pci/hda/cs35l56_hda.c | 995 ++++++++++++++++++++++++++++++ sound/pci/hda/cs35l56_hda.h | 48 ++ sound/pci/hda/cs35l56_hda_i2c.c | 69 +++ sound/pci/hda/cs35l56_hda_spi.c | 68 ++ sound/pci/hda/hda_auto_parser.h | 2 + sound/pci/hda/hda_component.h | 1 + sound/pci/hda/hda_generic.h | 3 + sound/soc/codecs/cs35l56-i2c.c | 14 +- sound/soc/codecs/cs35l56-sdw.c | 72 +-- sound/soc/codecs/cs35l56-shared.c | 459 +++++++++++++- sound/soc/codecs/cs35l56-spi.c | 10 +- sound/soc/codecs/cs35l56.c | 648 ++++--------------- sound/soc/codecs/cs35l56.h | 15 +- 16 files changed, 1878 insertions(+), 592 deletions(-) create mode 100644 sound/pci/hda/cs35l56_hda.c create mode 100644 sound/pci/hda/cs35l56_hda.h create mode 100644 sound/pci/hda/cs35l56_hda_i2c.c create mode 100644 sound/pci/hda/cs35l56_hda_spi.c -- 2.30.2

1 12

[PATCH] ASoC: cs35l45: Relicense to GPL only
by Charles Keepax 25 May '23

25 May '23

Cirrus never intended to upstream dual licensed code, convert to GPL only. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Apologies for this but this was never our intention and I believe that it can be a little legally involved maintaining dual licensed kernel code. Fortunately not that many people outside of Cirrus have contributed to the files in question. I have CCed everyone who has contributed both internally and externally. Thanks, Charles sound/soc/codecs/cs35l45-i2c.c | 4 ++-- sound/soc/codecs/cs35l45-spi.c | 4 ++-- sound/soc/codecs/cs35l45-tables.c | 2 +- sound/soc/codecs/cs35l45.c | 4 ++-- sound/soc/codecs/cs35l45.h | 2 +- 5 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs35l45-i2c.c b/sound/soc/codecs/cs35l45-i2c.c index 5832ebb90c2b9..77e0f8750f375 100644 --- a/sound/soc/codecs/cs35l45-i2c.c +++ b/sound/soc/codecs/cs35l45-i2c.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0 OR BSD-3-Clause +// SPDX-License-Identifier: GPL-2.0 // // cs35l45-i2c.c -- CS35L45 I2C driver // @@ -72,5 +72,5 @@ module_i2c_driver(cs35l45_i2c_driver); MODULE_DESCRIPTION("I2C CS35L45 driver"); MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <james.schulman(a)cirrus.com>"); -MODULE_LICENSE("Dual BSD/GPL"); +MODULE_LICENSE("GPL"); MODULE_IMPORT_NS(SND_SOC_CS35L45); diff --git a/sound/soc/codecs/cs35l45-spi.c b/sound/soc/codecs/cs35l45-spi.c index a00b23b4180c5..5efb77530cc33 100644 --- a/sound/soc/codecs/cs35l45-spi.c +++ b/sound/soc/codecs/cs35l45-spi.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0 OR BSD-3-Clause +// SPDX-License-Identifier: GPL-2.0 // // cs35l45-spi.c -- CS35L45 SPI driver // @@ -74,5 +74,5 @@ module_spi_driver(cs35l45_spi_driver); MODULE_DESCRIPTION("SPI CS35L45 driver"); MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <james.schulman(a)cirrus.com>"); -MODULE_LICENSE("Dual BSD/GPL"); +MODULE_LICENSE("GPL"); MODULE_IMPORT_NS(SND_SOC_CS35L45); diff --git a/sound/soc/codecs/cs35l45-tables.c b/sound/soc/codecs/cs35l45-tables.c index 46610e64e8188..066f83c0c7ac2 100644 --- a/sound/soc/codecs/cs35l45-tables.c +++ b/sound/soc/codecs/cs35l45-tables.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0 OR BSD-3-Clause +// SPDX-License-Identifier: GPL-2.0 // // cs35l45-tables.c -- CS35L45 ALSA SoC audio driver // diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index c31597f6bfae1..d1edb9876c10f 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0 OR BSD-3-Clause +// SPDX-License-Identifier: GPL-2.0 // // cs35l45.c - CS35L45 ALSA SoC audio driver // @@ -1296,4 +1296,4 @@ EXPORT_SYMBOL_NS_GPL(cs35l45_pm_ops, SND_SOC_CS35L45); MODULE_DESCRIPTION("ASoC CS35L45 driver"); MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <james.schulman(a)cirrus.com>"); MODULE_AUTHOR("Richard Fitzgerald <rf(a)opensource.cirrus.com>"); -MODULE_LICENSE("Dual BSD/GPL"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l45.h b/sound/soc/codecs/cs35l45.h index 0da28439f628b..61135a316df33 100644 --- a/sound/soc/codecs/cs35l45.h +++ b/sound/soc/codecs/cs35l45.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 OR BSD-3-Clause */ +/* SPDX-License-Identifier: GPL-2.0 */ /* * cs35l45.h - CS35L45 ALSA SoC audio driver * -- 2.30.2

6 6

[PATCH] ASoC: cs42l42: Add PLL ratio table values
by Vitaly Rodionov 25 May '23

25 May '23

Add 4.8Mhz 9.6Mhz and 19.2MHz SCLK values for MCLK 12MHz and 12.288MHz requested by Intel. Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index e3edaa1a2761..8aa6af21e52c 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -646,12 +646,19 @@ static const struct cs42l42_pll_params pll_ratio_table[] = { { 3072000, 1, 0x00, 0x3E, 0x800000, 0x03, 0x10, 12000000, 125, 1}, { 4000000, 1, 0x00, 0x30, 0x800000, 0x03, 0x10, 12000000, 96, 1}, { 4096000, 1, 0x00, 0x2E, 0xE00000, 0x03, 0x10, 12000000, 94, 1}, + { 4800000, 1, 0x01, 0x50, 0x000000, 0x03, 0x10, 12000000, 80, 2}, + { 4800000, 1, 0x01, 0x50, 0x000000, 0x01, 0x10, 12288000, 82, 2}, { 5644800, 1, 0x01, 0x40, 0x000000, 0x03, 0x10, 11289600, 128, 1}, { 6000000, 1, 0x01, 0x40, 0x000000, 0x03, 0x10, 12000000, 128, 1}, { 6144000, 1, 0x01, 0x3E, 0x800000, 0x03, 0x10, 12000000, 125, 1}, + { 6144000, 1, 0x01, 0x40, 0x000000, 0x03, 0x10, 12288000, 128, 1}, + { 9600000, 1, 0x02, 0x50, 0x000000, 0x03, 0x10, 12000000, 80, 2}, + { 9600000, 1, 0x02, 0x50, 0x000000, 0x01, 0x10, 12288000, 82, 2}, { 11289600, 0, 0, 0, 0, 0, 0, 11289600, 0, 1}, { 12000000, 0, 0, 0, 0, 0, 0, 12000000, 0, 1}, { 12288000, 0, 0, 0, 0, 0, 0, 12288000, 0, 1}, + { 19200000, 1, 0x03, 0x50, 0x000000, 0x03, 0x10, 12000000, 80, 2}, + { 19200000, 1, 0x03, 0x50, 0x000000, 0x01, 0x10, 12288000, 82, 2}, { 22579200, 1, 0x03, 0x40, 0x000000, 0x03, 0x10, 11289600, 128, 1}, { 24000000, 1, 0x03, 0x40, 0x000000, 0x03, 0x10, 12000000, 128, 1}, { 24576000, 1, 0x03, 0x40, 0x000000, 0x03, 0x10, 12288000, 128, 1} -- 2.34.1

3 2

[PATCH 0/6] Add Support for RK3588s Indiedroid Nova
by Chris Morgan 25 May '23

25 May '23

From: Chris Morgan <macromorgan(a)hotmail.com> Add support for the RK3588s based Indiedroid Nova. Note that this series is dependent on regulator support for the RK806 being upstreamed. https://lore.kernel.org/linux-rockchip/20230515152044.GT10825@google.com/ Chris Morgan (6): ASoC: es8328: Enabling support for 12Mhz sysclk arm64: dts: rockchip: add default pinctrl for rk3588 emmc arm64: dts: rockchip: Add sdio node to rk3588 dt-bindings: vendor-prefixes: add Indiedroid dt-bindings: arm: rockchip: Add Indiedroid Nova arm64: dts: rockchip: Add Indiedroid Nova board .../devicetree/bindings/arm/rockchip.yaml | 5 + .../devicetree/bindings/vendor-prefixes.yaml | 2 + arch/arm64/boot/dts/rockchip/Makefile | 1 + .../dts/rockchip/rk3588s-indiedroid-nova.dts | 761 ++++++++++++++++++ arch/arm64/boot/dts/rockchip/rk3588s.dtsi | 18 + sound/soc/codecs/es8328.c | 17 + 6 files changed, 804 insertions(+) create mode 100644 arch/arm64/boot/dts/rockchip/rk3588s-indiedroid-nova.dts -- 2.34.1

3 13

[PATCH v3 4/5] ALSA: hda/tas2781: Add tas2781 HDA driver
by Shenghao Ding 25 May '23

25 May '23

Create tas2781 HDA driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v3: - fixed issue | Reported-by: kernel test robot <lkp(a)intel.com> | Link: https://lore.kernel.org/oe-kbuild-all/202305022033.LiI7Ojm4-lkp@intel.com/ Changes to be committed: modified: sound/pci/hda/Kconfig modified: sound/pci/hda/Makefile modified: sound/pci/hda/patch_realtek.c new file: sound/pci/hda/tas2781_hda_i2c.c --- sound/pci/hda/Kconfig | 15 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_realtek.c | 106 ++++- sound/pci/hda/tas2781_hda_i2c.c | 806 ++++++++++++++++++++++++++++++++ 4 files changed, 926 insertions(+), 3 deletions(-) create mode 100644 sound/pci/hda/tas2781_hda_i2c.c diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 886255a03e8b..e66257277492 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -130,6 +130,21 @@ config SND_HDA_SCODEC_CS35L41_SPI comment "Set to Y if you want auto-loading the side codec driver" depends on SND_HDA=y && SND_HDA_SCODEC_CS35L41_SPI=m +config SND_HDA_SCODEC_TAS2781_I2C + tristate "Build TAS2781 HD-audio side codec support for I2C Bus" + depends on I2C + depends on ACPI + depends on SND_SOC + select SND_SOC_TAS2781_COMLIB + select SND_SOC_TAS2781_FMWLIB + select CRC32_SARWATE + help + Say Y or M here to include TAS2781 I2C HD-audio side codec support + in snd-hda-intel driver, such as ALC287. + +comment "Set to Y if you want auto-loading the side codec driver" + depends on SND_HDA=y && SND_HDA_SCODEC_TAS2781_I2C=m + config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 00d306104484..1c76609690d6 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -32,6 +32,7 @@ snd-hda-scodec-cs35l41-objs := cs35l41_hda.o snd-hda-scodec-cs35l41-i2c-objs := cs35l41_hda_i2c.o snd-hda-scodec-cs35l41-spi-objs := cs35l41_hda_spi.o snd-hda-cs-dsp-ctls-objs := hda_cs_dsp_ctl.o +snd-hda-scodec-tas2781-i2c-objs := tas2781_hda_i2c.o # common driver obj-$(CONFIG_SND_HDA) := snd-hda-codec.o @@ -56,6 +57,7 @@ obj-$(CONFIG_SND_HDA_SCODEC_CS35L41) += snd-hda-scodec-cs35l41.o obj-$(CONFIG_SND_HDA_SCODEC_CS35L41_I2C) += snd-hda-scodec-cs35l41-i2c.o obj-$(CONFIG_SND_HDA_SCODEC_CS35L41_SPI) += snd-hda-scodec-cs35l41-spi.o obj-$(CONFIG_SND_HDA_CS_DSP_CONTROLS) += snd-hda-cs-dsp-ctls.o +obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_I2C) += snd-hda-scodec-tas2781-i2c.o # this must be the last entry after codec drivers; # otherwise the codec patches won't be hooked before the PCI probe diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 172ffc2c332b..f5b912f90018 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6705,7 +6705,7 @@ static void comp_generic_playback_hook(struct hda_pcm_stream *hinfo, struct hda_ } } -struct cs35l41_dev_name { +struct scodec_dev_name { const char *bus; const char *hid; int index; @@ -6714,7 +6714,7 @@ struct cs35l41_dev_name { /* match the device name in a slightly relaxed manner */ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) { - struct cs35l41_dev_name *p = data; + struct scodec_dev_name *p = data; const char *d = dev_name(dev); int n = strlen(p->bus); char tmp[32]; @@ -6730,12 +6730,31 @@ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) return !strcmp(d + n, tmp); } +static int comp_match_tas2781_dev_name(struct device *dev, void *data) +{ + struct scodec_dev_name *p = data; + const char *d = dev_name(dev); + int n = strlen(p->bus); + char tmp[32]; + + /* check the bus name */ + if (strncmp(d, p->bus, n)) + return 0; + /* skip the bus number */ + if (isdigit(d[n])) + n++; + /* the rest must be exact matching */ + snprintf(tmp, sizeof(tmp), "-%s:00", p->hid); + + return !strcmp(d + n, tmp); +} + static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char *bus, const char *hid, int count) { struct device *dev = hda_codec_dev(cdc); struct alc_spec *spec = cdc->spec; - struct cs35l41_dev_name *rec; + struct scodec_dev_name *rec; int ret, i; switch (action) { @@ -6760,6 +6779,40 @@ static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char } } +static void tas2781_generic_fixup(struct hda_codec *cdc, int action, + const char *bus, const char *hid, int count) +{ + struct device *dev = hda_codec_dev(cdc); + struct alc_spec *spec = cdc->spec; + struct scodec_dev_name *rec; + int ret, i; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + for (i = 0; i < count; i++) { + rec = devm_kmalloc(dev, sizeof(*rec), GFP_KERNEL); + if (!rec) + return; + rec->bus = bus; + rec->hid = hid; + rec->index = i; + spec->comps[i].codec = cdc; + component_match_add(dev, &spec->match, + comp_match_tas2781_dev_name, rec); + } + ret = component_master_add_with_match(dev, &comp_master_ops, + spec->match); + if (ret) + codec_err(cdc, + "Fail to register component aggregator %d\n", + ret); + else + spec->gen.pcm_playback_hook = + comp_generic_playback_hook; + break; + } +} + static void cs35l41_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { cs35l41_generic_fixup(cdc, action, "i2c", "CSC3551", 2); @@ -6787,6 +6840,12 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st cs35l41_generic_fixup(cdc, action, "i2c", "CLSA0101", 2); } +static void tas2781_fixup_i2c(struct hda_codec *cdc, + const struct hda_fixup *fix, int action) +{ + tas2781_generic_fixup(cdc, action, "i2c", "TIAS2781", 1); +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -7201,6 +7260,8 @@ enum { ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, ALC236_FIXUP_DELL_DUAL_CODECS, + ALC287_FIXUP_TAS2781_I2C_2, + ALC287_FIXUP_TAS2781_I2C_4 }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9189,6 +9250,18 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, }, + [ALC287_FIXUP_TAS2781_I2C_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, + [ALC287_FIXUP_TAS2781_I2C_4] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9725,6 +9798,33 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", + ALC287_FIXUP_TAS2781_I2C_4), + SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", + ALC287_FIXUP_TAS2781_I2C_4), + SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual powe mode2 YC", + ALC287_FIXUP_TAS2781_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", + ALC287_FIXUP_TAS2781_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", + ALC287_FIXUP_TAS2781_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", + ALC287_FIXUP_TAS2781_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", + ALC287_FIXUP_TAS2781_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", + ALC287_FIXUP_TAS2781_I2C_4), + SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", + ALC287_FIXUP_TAS2781_I2C_4), + SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", + ALC287_FIXUP_TAS2781_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38bf, "Yoga S980-14.5 proX LX Dual", + ALC287_FIXUP_TAS2781_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38c3, "Y980 DUAL", ALC287_FIXUP_TAS2781_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", + ALC287_FIXUP_TAS2781_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", + ALC287_FIXUP_TAS2781_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c new file mode 100644 index 000000000000..a742805b749b --- /dev/null +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -0,0 +1,806 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2781 HDA I2C driver +// +// Copyright 2023 Texas Instruments, Inc. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> + +#include <linux/acpi.h> +#include <linux/crc8.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/firmware.h> +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <sound/hda_codec.h> +#include <sound/soc.h> +#include <sound/tas2781.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "hda_local.h" +#include "hda_auto_parser.h" +#include "hda_component.h" +#include "hda_jack.h" +#include "hda_generic.h" + +#define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 + +#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .invert = xinvert} } + +enum calib_data { + R0_VAL = 0, + INV_R0, + R0LOW, + POWER, + TLIM, + CALIB_MAX +}; + +static int tas2781_acpi_get_i2c_resource(struct acpi_resource + *ares, void *data) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)data; + struct acpi_resource_i2c_serialbus *sb; + + if (i2c_acpi_get_i2c_resource(ares, &sb)) { + if (sb->slave_address != TAS2781_GLOBAL_ADDR) { + tas_priv->tasdevice[tas_priv->ndev].dev_addr = + (unsigned int) sb->slave_address; + tas_priv->ndev++; + } else + tas_priv->glb_addr.dev_addr = TAS2781_GLOBAL_ADDR; + + } + + return 1; +} + +static int tas2781_hda_read_acpi(struct tasdevice_priv *tas_priv, + const char *hid) +{ + struct acpi_device *adev; + struct device *physdev; + LIST_HEAD(resources); + const char *sub; + int ret; + + adev = acpi_dev_get_first_match_dev(hid, NULL, -1); + if (!adev) { + dev_err(tas_priv->dev, + "Failed to find an ACPI device for %s\n", hid); + return -ENODEV; + } + strcpy(tas_priv->dev_name, hid); + physdev = get_device(acpi_get_first_physical_node(adev)); + acpi_dev_put(adev); + + sub = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); + if (IS_ERR(sub)) + sub = NULL; + dev_info(tas_priv->dev, "subid = %s\n", sub); + + tas_priv->acpi_subsystem_id = sub; + + ret = acpi_dev_get_resources(adev, &resources, + tas2781_acpi_get_i2c_resource, tas_priv); + if (ret < 0) + goto err; + acpi_dev_free_resource_list(&resources); + + put_device(physdev); + dev_info(tas_priv->dev, "global addr: 0x%02x devcnt = %d\n", + tas_priv->glb_addr.dev_addr, tas_priv->ndev); + + return 0; + +err: + dev_err(tas_priv->dev, "Failed acpi ret: %d\n", ret); + put_device(physdev); + + return ret; +} + +static void tas2781_hda_playback_hook(struct device *dev, int action) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret = 0; + + dev_info(tas_priv->dev, "%s: action = %d\n", __func__, action); + switch (action) { + case HDA_GEN_PCM_ACT_OPEN: + pm_runtime_get_sync(dev); + mutex_lock(&tas_priv->codec_lock); + tas_priv->cur_conf = 0; + tas_priv->rcabin.profile_cfg_id = 1; + tasdevice_tuning_switch(tas_priv, 0); + mutex_unlock(&tas_priv->codec_lock); + break; + case HDA_GEN_PCM_ACT_CLOSE: + mutex_lock(&tas_priv->codec_lock); + tasdevice_tuning_switch(tas_priv, 1); + mutex_unlock(&tas_priv->codec_lock); + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + break; + default: + dev_warn(tas_priv->dev, "Playback action not supported: %d\n", + action); + break; + } + + if (ret) + dev_err(tas_priv->dev, "Regmap access fail: %d\n", ret); +} + +static int tasdevice_info_profile(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 1; + uinfo->value.integer.max = max(0, tas_priv->rcabin.ncfgs); + + return 0; +} + +static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + + return 0; +} + +static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + tas_priv->rcabin.profile_cfg_id = ucontrol->value.integer.value[0]; + + return 1; +} + +static int tasdevice_create_control(struct tasdevice_priv *tas_priv) +{ + char prof_ctrl_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + struct hda_codec *codec = tas_priv->codec; + struct snd_kcontrol_new prof_ctrl = { + .name = prof_ctrl_name, + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_profile, + .get = tasdevice_get_profile_id, + .put = tasdevice_set_profile_id, + }; + int ret; + + /* Create a mixer item for selecting the active profile */ + scnprintf(prof_ctrl_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "tasdev-profile-id"); + ret = snd_ctl_add(codec->card, snd_ctl_new1(&prof_ctrl, tas_priv)); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + prof_ctrl.name, ret); + goto out; + } + + dev_dbg(tas_priv->dev, "Added Control %s\n", prof_ctrl.name); + +out: + return ret; +} + +static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)tas_fw->nr_programs; + + return 0; +} + +static int tasdevice_info_configurations( + struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)tas_fw->nr_configurations - 1; + + return 0; +} + +static int tasdevice_program_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + + return 0; +} + +static int tasdevice_program_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + unsigned int nr_program = ucontrol->value.integer.value[0]; + + tas_priv->cur_prog = nr_program; + + return 1; +} + +static int tasdevice_config_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + + return 0; +} + +static int tasdevice_config_put( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + unsigned int nr_configuration = ucontrol->value.integer.value[0]; + + tas_priv->cur_conf = nr_configuration; + + return 1; +} + +/** + * tas2781_digital_getvol - get the volum control + * @kcontrol: control pointer + * @ucontrol: User data + * Customer Kcontrol for tas2781 is primarily for regmap booking, paging + * depends on internal regmap mechanism. + * tas2781 contains book and page two-level register map, especially + * book switching will set the register BXXP00R7F, after switching to the + * correct book, then leverage the mechanism for paging to access the + * register. + */ +static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_digital_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_amp_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_digital_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_amp_putvol(tas_priv, ucontrol, mc); +} + +static const struct snd_kcontrol_new tas2781_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Digital Volume", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), +}; + +static int tasdevice_dsp_create_ctrls(struct tasdevice_priv + *tas_priv) +{ + char prog_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char conf_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + struct hda_codec *codec = tas_priv->codec; + struct snd_kcontrol_new prog_ctl = { + .name = prog_name, + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_programs, + .get = tasdevice_program_get, + .put = tasdevice_program_put, + }; + struct snd_kcontrol_new conf_ctl = { + .name = conf_name, + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_configurations, + .get = tasdevice_config_get, + .put = tasdevice_config_put, + }; + int ret; + + scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "tasdev-prog-id"); + scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "tasdev-conf-id"); + + ret = snd_ctl_add(codec->card, snd_ctl_new1(&prog_ctl, tas_priv)); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + prog_ctl.name, ret); + goto out; + } + + dev_dbg(tas_priv->dev, "Added Control %s\n", prog_ctl.name); + + ret = snd_ctl_add(codec->card, snd_ctl_new1(&conf_ctl, tas_priv)); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + conf_ctl.name, ret); + goto out; + } + + dev_dbg(tas_priv->dev, "Added Control %s\n", conf_ctl.name); +out: + return ret; +} + +static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) +{ + unsigned char page_array[CALIB_MAX] = {0x17, 0x18, 0x18, 0x0d, 0x18}; + unsigned char rgno_array[CALIB_MAX] = {0x74, 0x0c, 0x14, 0x3c, 0x7c}; + unsigned char *data; + int i, j, rc; + + for (i = 0; i < tas_priv->ndev; i++) { + data = tas_priv->cali_data.data + + i * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + for (j = 0; j < CALIB_MAX; j++) { + rc = tasdevice_dev_bulk_write(tas_priv, i, + TASDEVICE_REG(0, page_array[j], rgno_array[j]), + &(data[4 * j]), 4); + if (rc < 0) + dev_err(tas_priv->dev, + "chn %d calib %d bulk_wr err = %d\n", + i, j, rc); + } + } +} + +static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) +{ + efi_guid_t efi_guid = EFI_GUID(0x02f9af02, 0x7734, 0x4233, 0xb4, 0x3d, + 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct hda_codec *codec = tas_priv->codec; + unsigned int subid = codec->core.subsystem_id & 0xFFFF; + struct tm *tm = &tas_priv->tm; + unsigned int attr, crc; + unsigned int *tmp_val; + efi_status_t status; + int ret = 0; + + //Lenovo devices + if ((subid == 0x387d) || (subid == 0x387e) || (subid == 0x3881) + || (subid == 0x3884) || (subid == 0x3886) || (subid == 0x38a7) + || (subid == 0x38a8) || (subid == 0x38ba) || (subid == 0x38bb) + || (subid == 0x38be) || (subid == 0x38bf) || (subid == 0x38c3) + || (subid == 0x38cb) || (subid == 0x38cd)) + efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, 0x09, + 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + tas_priv->cali_data.total_sz = 0; + /* Get real size of UEFI variable */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, tas_priv->cali_data.data); + if (status == EFI_BUFFER_TOO_SMALL) { + ret = -ENODEV; + /* Allocate data buffer of data_size bytes */ + tas_priv->cali_data.data = devm_kzalloc(tas_priv->dev, + tas_priv->cali_data.total_sz, GFP_KERNEL); + if (!tas_priv->cali_data.data) + return -ENOMEM; + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, + tas_priv->cali_data.data); + if (status != EFI_SUCCESS) { + ret = -EINVAL; + goto out; + } + } + + tmp_val = (unsigned int *)tas_priv->cali_data.data; + + crc = crc32(~0, tas_priv->cali_data.data, 84) ^ ~0; + dev_info(tas_priv->dev, "cali crc 0x%08x PK tmp_val 0x%08x\n", + crc, tmp_val[21]); + + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + dev_info(tas_priv->dev, "%4ld-%2d-%2d, %2d:%2d:%2d\n", + tm->tm_year, tm->tm_mon, tm->tm_mday, + tm->tm_hour, tm->tm_min, tm->tm_sec); + tas2781_apply_calib(tas_priv); + } +out: + return ret; +} + +static void tasdevice_fw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; + struct hda_codec *codec = tas_priv->codec; + int i, ret = 0; + + pm_runtime_get_sync(tas_priv->dev); + mutex_lock(&tas_priv->codec_lock); + + ret = tasdevice_rca_parser(tas_priv, fmw); + if (ret) + goto out; + tasdevice_create_control(tas_priv); + + for (i = 0; i < ARRAY_SIZE(tas2781_snd_controls); i++) { + ret = snd_ctl_add(codec->card, + snd_ctl_new1(&tas2781_snd_controls[i], tas_priv)); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_snd_controls[i].name, ret); + goto out; + } + } + + tasdevice_dsp_remove(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%04X.bin", + codec->core.subsystem_id & 0xffff); + ret = tasdevice_dsp_parser(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dspfw load %s error\n", + tas_priv->coef_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + tasdevice_dsp_create_ctrls(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; + tasdevice_prmg_load(tas_priv, 0); + tas_priv->cur_prog = 0; + + /* If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + tas2781_save_calibration(tas_priv); + +out: + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + /*If DSP FW fail, kcontrol won't be created */ + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + } + mutex_unlock(&tas_priv->codec_lock); + if (fmw) + release_firmware(fmw); + pm_runtime_mark_last_busy(tas_priv->dev); + pm_runtime_put_autosuspend(tas_priv->dev); +} + +static int tas2781_hda_bind(struct device *dev, struct device *master, + void *master_data) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct hda_component *comps = master_data; + int ret = 0; + + if (!comps || tas_priv->index < 0 || + tas_priv->index >= HDA_MAX_COMPONENTS) + return -EINVAL; + + comps = &comps[tas_priv->index]; + if (comps->dev) + return -EBUSY; + + pm_runtime_get_sync(dev); + + comps->dev = dev; + + strscpy(comps->name, dev_name(dev), sizeof(comps->name)); + + ret = tascodec_init(tas_priv, comps->codec, tasdevice_fw_ready); + + comps->playback_hook = tas2781_hda_playback_hook; + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + + return ret; +} + +static void tas2781_hda_unbind(struct device *dev, + struct device *master, void *master_data) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct hda_component *comps = master_data; + + if (comps[tas_priv->index].dev == dev) + memset(&comps[tas_priv->index], 0, sizeof(*comps)); + + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; +} + +static const struct component_ops tas2781_hda_comp_ops = { + .bind = tas2781_hda_bind, + .unbind = tas2781_hda_unbind, +}; + +static void tas2781_hda_remove(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + pm_runtime_get_sync(tas_priv->dev); + pm_runtime_disable(tas_priv->dev); + + + component_del(tas_priv->dev, &tas2781_hda_comp_ops); + + pm_runtime_put_noidle(tas_priv->dev); + + tasdevice_remove(tas_priv); +} + +static int tas2781_hda_i2c_probe(struct i2c_client *clt) +{ + struct tasdevice_priv *tas_priv; + const char *device_name; + int ret; + + if (strstr(dev_name(&clt->dev), "TIAS2781")) + device_name = "TIAS2781"; + else + return -ENODEV; + + tas_priv = tasdevice_kzalloc(clt); + if (!tas_priv) + return -ENOMEM; + + tas_priv->irq_info.irq = clt->irq; + ret = tas2781_hda_read_acpi(tas_priv, device_name); + if (ret) + return dev_err_probe(tas_priv->dev, ret, + "Platform not supported\n"); + + ret = tasdevice_init(tas_priv); + if (ret) + goto err; + + pm_runtime_set_autosuspend_delay(tas_priv->dev, 3000); + pm_runtime_use_autosuspend(tas_priv->dev); + pm_runtime_mark_last_busy(tas_priv->dev); + pm_runtime_set_active(tas_priv->dev); + pm_runtime_get_noresume(tas_priv->dev); + pm_runtime_enable(tas_priv->dev); + + pm_runtime_put_autosuspend(tas_priv->dev); + + ret = component_add(tas_priv->dev, &tas2781_hda_comp_ops); + if (ret) { + dev_err(tas_priv->dev, "Register component failed: %d\n", ret); + pm_runtime_disable(tas_priv->dev); + } + +err: + if (ret) + tas2781_hda_remove(&clt->dev); + return ret; +} + +static void tas2781_hda_i2c_remove(struct i2c_client *clt) +{ + tas2781_hda_remove(&clt->dev); +} + +static int tas2781_runtime_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int i, ret = 0; + + dev_info(tas_priv->dev, "Runtime Suspend\n"); + + mutex_lock(&tas_priv->codec_lock); + + if (tas_priv->playback_started) { + tasdevice_tuning_switch(tas_priv, 1); + tas_priv->playback_started = false; + } + + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + for (i = 0; i < tas_priv->ndev; i++) { + tas_priv->tasdevice[i].cur_book = -1; + tas_priv->tasdevice[i].cur_prog = -1; + tas_priv->tasdevice[i].cur_conf = -1; + } + + regcache_cache_only(tas_priv->regmap, true); + regcache_mark_dirty(tas_priv->regmap); + + mutex_unlock(&tas_priv->codec_lock); + + return ret; +} + +static int tas2781_runtime_resume(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long calib_data_sz = + tas_priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + int ret = 0; + + dev_info(tas_priv->dev, "Runtime Resume\n"); + + mutex_lock(&tas_priv->codec_lock); + + regcache_cache_only(tas_priv->regmap, false); + ret = regcache_sync(tas_priv->regmap); + if (ret) { + dev_err(tas_priv->dev, + "Failed to restore register cache: %d\n", ret); + goto out; + } + + tasdevice_prmg_load(tas_priv, 0); + tas_priv->cur_prog = 0; + + /* If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + if (tas_priv->cali_data.total_sz > calib_data_sz) + tas2781_apply_calib(tas_priv); + +out: + mutex_unlock(&tas_priv->codec_lock); + + return ret; +} + +static int tas2781_system_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + dev_info(tas_priv->dev, "System Suspend\n"); + + ret = pm_runtime_force_suspend(dev); + if (ret) + return ret; + + /* Shutdown chip before system suspend */ + regcache_cache_only(tas_priv->regmap, false); + tasdevice_tuning_switch(tas_priv, 1); + regcache_cache_only(tas_priv->regmap, true); + regcache_mark_dirty(tas_priv->regmap); + /* + * Reset GPIO may be shared, so cannot reset here. + * However beyond this point, amps may be powered down. + */ + return 0; +} + +static int tas2781_system_resume(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long calib_data_sz = + tas_priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + int i, ret; + + dev_info(tas_priv->dev, "System Resume\n"); + + ret = pm_runtime_force_resume(dev); + + mutex_lock(&tas_priv->codec_lock); + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + for (i = 0; i < tas_priv->ndev; i++) { + tas_priv->tasdevice[i].cur_book = -1; + tas_priv->tasdevice[i].cur_prog = -1; + tas_priv->tasdevice[i].cur_conf = -1; + } + tas2781_reset(tas_priv); + tasdevice_prmg_load(tas_priv, 0); + tas_priv->cur_prog = 0; + + /* If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + if (tas_priv->cali_data.total_sz > calib_data_sz) + tas2781_apply_calib(tas_priv); + mutex_unlock(&tas_priv->codec_lock); + + return ret; +} + +static const struct dev_pm_ops tas2781_hda_pm_ops = { + RUNTIME_PM_OPS(tas2781_runtime_suspend, tas2781_runtime_resume, + NULL) + SYSTEM_SLEEP_PM_OPS(tas2781_system_suspend, tas2781_system_resume) +}; + +static const struct i2c_device_id tas2781_hda_i2c_id[] = { + { "tas2781-hda", 0 }, + {} +}; + +static const struct acpi_device_id tas2781_acpi_hda_match[] = { + {"TIAS2781", 0 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); + +static struct i2c_driver tas2781_hda_i2c_driver = { + .driver = { + .name = "tas2781-hda", + .acpi_match_table = tas2781_acpi_hda_match, + .pm = &tas2781_hda_pm_ops, + }, + .id_table = tas2781_hda_i2c_id, + .probe_new = tas2781_hda_i2c_probe, + .remove = tas2781_hda_i2c_remove, +}; +module_i2c_driver(tas2781_hda_i2c_driver); + +MODULE_DESCRIPTION("TAS2781 HDA Driver"); +MODULE_AUTHOR("Shenghao Ding, TI, <shenghao-ding(a)ti.com>"); +MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS(SND_SOC_TAS2781_FMWLIB); -- 2.34.1

3 4

[PATCH linux-nxp] ASoC: SOF: imx: Add OF machine descriptors for i.MX platforms
by Daniel Baluta 25 May '23

25 May '23

From: Paul Olaru <paul.olaru(a)nxp.com> Covered platforms: i.MX8QXP, i.MX8QM, i.MX8MP, i.MX8ULP. These descriptors are used to locate the machine driver and topology filenames. Note: because the compatible values for the boards do not change between the cs42888 and the wm8960 cases, I have selected a common topology filename, which is intended to be a symlink to either the wm8960 or the cs42888 topology file. For uniformity I did the same for i.MX8ULP. Signed-off-by: Paul Olaru <paul.olaru(a)nxp.com> --- sound/soc/sof/imx/imx8.c | 16 ++++++++++++++++ sound/soc/sof/imx/imx8m.c | 10 ++++++++++ sound/soc/sof/imx/imx8ulp.c | 10 ++++++++++ 3 files changed, 36 insertions(+) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 7e7d1f5c1147..2b91bfc8fb5b 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -624,7 +624,22 @@ static struct snd_sof_dsp_ops sof_imx8x_ops = { SNDRV_PCM_INFO_NO_PERIOD_WAKEUP }; +static struct snd_sof_of_mach sof_imx8_machs[] = { + { + .compatible = "fsl,imx8qxp", + .drv_name = "asoc-simple-card", + .sof_tplg_filename = "sof-imx8.tplg", + }, + { + .compatible = "fsl,imx8qm", + .drv_name = "asoc-simple-card", + .sof_tplg_filename = "sof-imx8.tplg", + }, + {} +}; + static struct sof_dev_desc sof_of_imx8qxp_desc = { + .of_machines = sof_imx8_machs, .ipc_supported_mask = BIT(SOF_IPC), .ipc_default = SOF_IPC, .default_fw_path = { @@ -641,6 +656,7 @@ static struct sof_dev_desc sof_of_imx8qxp_desc = { }; static struct sof_dev_desc sof_of_imx8qm_desc = { + .of_machines = sof_imx8_machs, .ipc_supported_mask = BIT(SOF_IPC), .ipc_default = SOF_IPC, .default_fw_path = { diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index d20011349b43..a05497eb8d89 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -487,7 +487,17 @@ static struct snd_sof_dsp_ops sof_imx8m_ops = { SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, }; +static struct snd_sof_of_mach sof_imx8m_machs[] = { + { + .compatible = "fsl,imx8mp", + .drv_name = "asoc-simple-card", + .sof_tplg_filename = "sof-imx8m.tplg", + }, + {} +}; + static struct sof_dev_desc sof_of_imx8mp_desc = { + .of_machines = sof_imx8m_machs, .ipc_supported_mask = BIT(SOF_IPC), .ipc_default = SOF_IPC, .default_fw_path = { diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c index 52c357ba570e..0c9878c2c420 100644 --- a/sound/soc/sof/imx/imx8ulp.c +++ b/sound/soc/sof/imx/imx8ulp.c @@ -496,7 +496,17 @@ static struct snd_sof_dsp_ops sof_imx8ulp_ops = { .set_power_state = imx8ulp_dsp_set_power_state, }; +static struct snd_sof_of_mach sof_imx8ulp_machs[] = { + { + .compatible = "fsl,imx8ulp", + .drv_name = "asoc-simple-card", + .sof_tplg_filename = "sof-imx8ulp.tplg", + }, + {} +}; + static struct sof_dev_desc sof_of_imx8ulp_desc = { + .of_machines = sof_imx8ulp_machs, .ipc_supported_mask = BIT(SOF_IPC), .ipc_default = SOF_IPC, .default_fw_path = { -- 2.40.1

2 1

Linux Mint 21 MSI Z790-A WIFI (ALC4080) no audio from spdif
by GitHub issues - edited 25 May '23

25 May '23

alsa-project/alsa-ucm-conf issue #318 was edited from jasonzliang: Hi, I been troubleshooting problems with sound not coming out from my optical port and I narrowed it down to Linux having problems with the ALC4080 chipset on my MSI Z790-A WIFI motherboard (id: 0db0:d1d7). The version of alsa-ucm-conf installed on my system is 1.2.6.3-1ubuntu1.5. My question is, what changes do I need to make to the alsa configuration files in order to get audio working again? Thanks Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/318 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Linux Mint 21 MSI Z790-A WIFI (ALC4080) no audio from spdif
by GitHub issues - edited 25 May '23

25 May '23

alsa-project/alsa-ucm-conf issue #318 was edited from jasonzliang: Hi, I been troubleshooting problems with sound not coming out from my optical port and I narrowed it down to Linux having problems with the ALC4080 chipset on my MSI Z790-A WIFI motherboard. The version of alsa-ucm-conf installed on my system is 1.2.6.3-1ubuntu1.5. My question is, what changes do I need to make to the alsa configuration files in order to get audio working again? Thanks Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/318 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Linux Mint 21 MSI Z790-A WIFI (ALC4080) no audio from spdif
by GitHub issues - opened 25 May '23

25 May '23

alsa-project/alsa-ucm-conf issue #318 was opened from jasonzliang: Hi, I been troubleshooting problems with sound not coming out from my optical port and I narrowed it down to Linux having problems with the ALC4080 chipset on my MSI Z790-A WIFI motherboard. My question is, what changes do I need to make to the alsa configuration files in order to get audio working again? Thanks Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/318 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 1/2] ALSA: ump: Drop redundant check of note-on with zero velocity
by Takashi Iwai 25 May '23

25 May '23

The check of a note-on event with zero velocity is done twice, and the latter one is superfluous. Let's drop it. Fixes: 0b5288f5fe63 ("ALSA: ump: Add legacy raw MIDI support") Reported-by: Dan Carpenter <dan.carpenter(a)linaro.org> Closes: https://lore.kernel.org/r/4683198a-84f6-4238-9e87-c70667d84523@kili.mountain Suggested-by: Dan Carpenter <dan.carpenter(a)linaro.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/ump_convert.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/core/ump_convert.c b/sound/core/ump_convert.c index cb7c2f959a27..164829d3e305 100644 --- a/sound/core/ump_convert.c +++ b/sound/core/ump_convert.c @@ -340,9 +340,6 @@ static int cvt_legacy_cmd_to_ump(struct snd_ump_endpoint *ump, switch (status) { case UMP_MSG_STATUS_NOTE_ON: - if (!buf[2]) - status = UMP_MSG_STATUS_NOTE_OFF; - fallthrough; case UMP_MSG_STATUS_NOTE_OFF: midi2->note.note = buf[1]; midi2->note.velocity = upscale_7_to_16bit(buf[2]); -- 2.35.3

1 1

[bug report] ALSA: ump: Add legacy raw MIDI support
by Dan Carpenter 25 May '23

25 May '23

Hello Takashi Iwai, The patch 0b5288f5fe63: "ALSA: ump: Add legacy raw MIDI support" from May 23, 2023, leads to the following Smatch static checker warning: sound/core/ump_convert.c:460 do_convert_to_ump() warn: right shifting more than type allows 8 vs 8 sound/core/ump_convert.c 419 static int do_convert_to_ump(struct snd_ump_endpoint *ump, 420 unsigned char group, unsigned char c, u32 *data) 421 { 422 /* bytes for 0x80-0xf0 */ 423 static unsigned char cmd_bytes[8] = { 424 3, 3, 3, 3, 2, 2, 3, 0 425 }; 426 /* bytes for 0xf0-0xff */ 427 static unsigned char system_bytes[16] = { 428 0, 2, 3, 2, 0, 0, 1, 0, 1, 1, 1, 1, 0, 0, 1, 1 429 }; 430 struct ump_cvt_to_ump *cvt = &ump->out_cvts[group]; 431 unsigned char bytes; 432 433 if (c == UMP_MIDI1_MSG_SYSEX_START) { 434 cvt->in_sysex = 1; 435 cvt->len = 0; 436 return 0; 437 } 438 if (c == UMP_MIDI1_MSG_SYSEX_END) { 439 if (!cvt->in_sysex) 440 return 0; /* skip */ 441 return cvt_legacy_sysex_to_ump(cvt, group, data, true); 442 } 443 444 if ((c & 0xf0) == UMP_MIDI1_MSG_REALTIME) { 445 bytes = system_bytes[c & 0x0f]; 446 if (!bytes) 447 return 0; /* skip */ 448 if (bytes == 1) { 449 data[0] = ump_compose(UMP_MSG_TYPE_SYSTEM, group, 0, c); 450 return 4; 451 } 452 cvt->buf[0] = c; 453 cvt->len = 1; 454 cvt->cmd_bytes = bytes; 455 cvt->in_sysex = 0; /* abort SysEx */ 456 return 0; 457 } 458 459 if (c & 0x80) { --> 460 bytes = cmd_bytes[(c >> 8) & 7]; ^^^^^^ Based on the if statement, it looks like c >> 4 was intended. 461 cvt->buf[0] = c; 462 cvt->len = 1; 463 cvt->cmd_bytes = bytes; 464 cvt->in_sysex = 0; /* abort SysEx */ 465 return 0; 466 } 467 468 if (cvt->in_sysex) { 469 cvt->buf[cvt->len++] = c; 470 if (cvt->len == 6) 471 return cvt_legacy_sysex_to_ump(cvt, group, data, false); 472 return 0; 473 } 474 475 if (!cvt->len) 476 return 0; 477 478 cvt->buf[cvt->len++] = c; 479 if (cvt->len < cvt->cmd_bytes) 480 return 0; 481 cvt->len = 1; 482 if ((cvt->buf[0] & 0xf0) == UMP_MIDI1_MSG_REALTIME) 483 return cvt_legacy_system_to_ump(cvt, group, data); 484 return cvt_legacy_cmd_to_ump(ump, cvt, group, data, cvt->cmd_bytes); 485 } regards, dan carpenter

2 1

[bug report] ALSA: ump: Add legacy raw MIDI support
by Dan Carpenter 25 May '23

25 May '23

Hello Takashi Iwai, The patch 0b5288f5fe63: "ALSA: ump: Add legacy raw MIDI support" from May 23, 2023, leads to the following Smatch static checker warning: sound/core/ump_convert.c:343 cvt_legacy_cmd_to_ump() warn: duplicate check 'buf[2]' (previous on line 333) sound/core/ump_convert.c 305 static int cvt_legacy_cmd_to_ump(struct snd_ump_endpoint *ump, 306 struct ump_cvt_to_ump *cvt, 307 unsigned char group, u32 *data, 308 unsigned char bytes) 309 { 310 const unsigned char *buf = cvt->buf; 311 struct ump_cvt_to_ump_bank *cc; 312 union snd_ump_midi2_msg *midi2 = (union snd_ump_midi2_msg *)data; 313 unsigned char status, channel; 314 315 BUILD_BUG_ON(sizeof(union snd_ump_midi1_msg) != 4); 316 BUILD_BUG_ON(sizeof(union snd_ump_midi2_msg) != 8); 317 318 /* for MIDI 1.0 UMP, it's easy, just pack it into UMP */ 319 if (ump->info.protocol & SNDRV_UMP_EP_INFO_PROTO_MIDI1) { 320 data[0] = ump_compose(UMP_MSG_TYPE_MIDI1_CHANNEL_VOICE, 321 group, 0, buf[0]); 322 data[0] |= buf[1] << 8; 323 if (bytes > 2) 324 data[0] |= buf[2]; 325 return 4; 326 } 327 328 status = *buf >> 4; 329 channel = *buf & 0x0f; 330 cc = &cvt->bank[channel]; 331 332 /* special handling: treat note-on with 0 velocity as note-off */ 333 if (status == UMP_MSG_STATUS_NOTE_ON && !buf[2]) 334 status = UMP_MSG_STATUS_NOTE_OFF; This if statment 335 336 /* initialize the packet */ 337 data[0] = ump_compose(UMP_MSG_TYPE_MIDI2_CHANNEL_VOICE, 338 group, status, channel); 339 data[1] = 0; 340 341 switch (status) { 342 case UMP_MSG_STATUS_NOTE_ON: --> 343 if (!buf[2]) 344 status = UMP_MSG_STATUS_NOTE_OFF; and this one are the same. One could be deleted. 345 fallthrough; 346 case UMP_MSG_STATUS_NOTE_OFF: 347 midi2->note.note = buf[1]; 348 midi2->note.velocity = upscale_7_to_16bit(buf[2]); 349 break; 350 case UMP_MSG_STATUS_POLY_PRESSURE: 351 midi2->paf.note = buf[1]; 352 midi2->paf.data = upscale_7_to_32bit(buf[2]); 353 break; 354 case UMP_MSG_STATUS_CC: 355 switch (buf[1]) { 356 case UMP_CC_RPN_MSB: 357 cc->rpn_set = 1; 358 cc->cc_rpn_msb = buf[2]; 359 return 0; // skip 360 case UMP_CC_RPN_LSB: 361 cc->rpn_set = 1; 362 cc->cc_rpn_lsb = buf[2]; 363 return 0; // skip 364 case UMP_CC_NRPN_MSB: 365 cc->nrpn_set = 1; 366 cc->cc_nrpn_msb = buf[2]; 367 return 0; // skip 368 case UMP_CC_NRPN_LSB: 369 cc->nrpn_set = 1; 370 cc->cc_nrpn_lsb = buf[2]; 371 return 0; // skip 372 case UMP_CC_DATA: 373 cc->cc_data_msb = buf[2]; 374 return 0; // skip 375 case UMP_CC_BANK_SELECT: 376 cc->bank_set = 1; 377 cc->cc_bank_msb = buf[2]; 378 return 0; // skip 379 case UMP_CC_BANK_SELECT_LSB: 380 cc->bank_set = 1; 381 cc->cc_bank_lsb = buf[2]; 382 return 0; // skip 383 case UMP_CC_DATA_LSB: 384 cc->cc_data_lsb = buf[2]; 385 if (cc->rpn_set || cc->nrpn_set) 386 fill_rpn(cc, midi2); 387 else 388 return 0; // skip 389 break; 390 default: 391 midi2->cc.index = buf[1]; 392 midi2->cc.data = upscale_7_to_32bit(buf[2]); 393 break; 394 } 395 break; 396 case UMP_MSG_STATUS_PROGRAM: 397 midi2->pg.program = buf[1]; 398 if (cc->bank_set) { 399 midi2->pg.bank_valid = 1; 400 midi2->pg.bank_msb = cc->cc_bank_msb; 401 midi2->pg.bank_lsb = cc->cc_bank_lsb; 402 cc->bank_set = 0; 403 cc->cc_bank_msb = cc->cc_bank_lsb = 0; 404 } 405 break; 406 case UMP_MSG_STATUS_CHANNEL_PRESSURE: 407 midi2->caf.data = upscale_7_to_32bit(buf[1]); 408 break; 409 case UMP_MSG_STATUS_PITCH_BEND: 410 midi2->pb.data = upscale_14_to_32bit(buf[1] | (buf[2] << 7)); 411 break; 412 default: 413 return 0; 414 } 415 416 return 8; 417 } regards, dan carpenter

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[PATCH v2 0/7] ASoC: mt8188: add new board support
by Trevor Wu 25 May '23

25 May '23

In the series, we extend the capability of mt8188-mt6359 driver. The following changes are included. 1. Divide ADDA BE dai into two dais for SOF. 2. Register hdmi/dp jack pins. 3. dai_fmt can be configured from device tree. 4. Add some I2S codecs support. In addition, new compatible string "mediatek,mt8188-nau8825" is included for a new board support. Changes since v1: - add "Headphone Jack" for Nau8825 - separate binding patch 2 from patch 6, and rename the subject - clean up debugfs for freed widget Trevor Wu (7): ASoC: mediatek: mt8188: separate ADDA playback dai from capture dai ASoC: dt-bindings: mediatek,mt8188-mt6359: remove ADDA_BE from link-name ASoC: mediatek: mt8188-mt6359: register hdmi/dp jack pins ASoC: mediatek: common: soundcard driver add dai_fmt support ASoC: soc-dapm.c: clean up debugfs for freed widget ASoC: mediatek: mt8188-mt6359: support new board with nau88255 ASoC: dt-bindings: mediatek,mt8188-mt6359: add NAU8825 support .../sound/mediatek,mt8188-mt6359.yaml | 27 +- sound/soc/mediatek/Kconfig | 4 + .../mediatek/common/mtk-soundcard-driver.c | 49 ++- sound/soc/mediatek/mt8188/mt8188-afe-common.h | 3 +- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 73 ++-- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 389 +++++++++++++++++- sound/soc/soc-dapm.c | 16 + 7 files changed, 507 insertions(+), 54 deletions(-) -- 2.18.0

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Fwd: XPS 17 - Intel® Smart Sound Technology for MIPI SoundWire® Audio (device not recognized)
by Bagas Sanjaya 25 May '23

25 May '23

Hi, I notice a bug report on Bugzilla [1]. As many developers don't keep an eye on it, I decide to forward it by email. Quoting from it: > Ubuntu (Lobster) 23.04 not recognizing Intel® Smart Sound Technology for MIPI SoundWire® Audio. > > gnome-control-center showing dummy-output. > > I tried with the latest mainline kernel 6.4.00-RC2, still sound not working. See bugzilla for the full thread. The reporter said that he had problem only with integrated sound card on his laptop, and he also attached dmesg and working audio capture from Windows. Thanks. [1]: https://bugzilla.kernel.org/show_bug.cgi?id=217461 -- An old man doll... just what I always wanted! - Clara

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[PATCH 00/20] ASoC: replace dpcm_playback/capture to playback/capture_only
by Kuninori Morimoto 25 May '23

25 May '23

Hi Mark Current ASoC can use DPCM, and we need to set dpcm_playback/capture to use it. OTOH, we are using playback/capture_only flag on Normal/Codec2Codec case. I think it is very duplicated, we can share same rules for all cases. This patch-set cleanup these. 1 - 7 : cleanup functions. 8 : main patch 9 - 19 : replace flags on each drivers 20 : remove old flags Kuninori Morimoto (20): 1 ASoC: soc-pcm.c: indicate error if stream has no playback no capture 2 ASoC: soc-pcm.c: use dai_link on soc_get_playback_capture() 3 ASoC: soc-pcm.c: cleanup soc_get_playback_capture() error 4 ASoC: soc-pcm.c: use temporary variable at soc_get_playback_capture() 5 ASoC: soc-pcm.c: tidyup playback/capture_only at soc_get_playback_capture() 6 ASoC: soc-pcm.c: cleanup normal connection loop at soc_get_playback_capture() part1 7 ASoC: soc-pcm.c: cleanup normal connection loop at soc_get_playback_capture() part2 8 ASoC: soc-pcm.c: cleanup soc_get_playback_capture() 9 ASoC: amd: replace dpcm_playback/capture to playback/capture_only 10 ASoC: fsl: replace dpcm_playback/capture to playback/capture_only 11 ASoC: sof: replace dpcm_playback/capture to playback/capture_only 12 ASoC: meson: replace dpcm_playback/capture to playback/capture_only 13 ASoC: Intel: replace dpcm_playback/capture to playback/capture_only 14 ASoC: samsung: replace dpcm_playback/capture to playback/capture_only 15 ASoC: mediatek: replace dpcm_playback/capture to playback/capture_only 16 ASoC: soc-dai.c: replace dpcm_playback/capture to playback/capture_only 17 ASoC: soc-core.c: replace dpcm_playback/capture to playback/capture_only 18 ASoC: soc-topology.c: replace dpcm_playback/capture to playback/capture_only 19 ASoC: soc-compress.c: replace dpcm_playback/capture to playback/capture_only 20 ASoC: soc-pcm.c: remove dpcm_playback/capture include/sound/soc.h | 4 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++-- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 20 +--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 3 - sound/soc/fsl/fsl-asoc-card.c | 16 +-- sound/soc/fsl/imx-audmix.c | 6 +- sound/soc/fsl/imx-card.c | 4 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 32 +++--- sound/soc/intel/boards/bxt_rt298.c | 26 ++--- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 14 +-- sound/soc/intel/boards/ehl_rt5660.c | 14 +-- sound/soc/intel/boards/glk_rt5682_max98357a.c | 30 +++-- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 26 ++--- sound/soc/intel/boards/kbl_da7219_max98927.c | 54 ++++----- sound/soc/intel/boards/kbl_rt5660.c | 18 ++- sound/soc/intel/boards/kbl_rt5663_max98927.c | 44 ++++--- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 22 ++-- sound/soc/intel/boards/skl_hda_dsp_common.c | 14 +-- .../soc/intel/boards/skl_nau88l25_max98357a.c | 26 ++--- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 26 ++--- sound/soc/intel/boards/skl_rt286.c | 26 ++--- sound/soc/intel/boards/sof_cs42l42.c | 12 +- sound/soc/intel/boards/sof_da7219_max98373.c | 16 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_nau8825.c | 12 +- sound/soc/intel/boards/sof_pcm512x.c | 8 +- sound/soc/intel/boards/sof_rt5682.c | 12 +- sound/soc/intel/boards/sof_sdw.c | 4 +- sound/soc/intel/boards/sof_ssp_amp.c | 11 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++-- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++--- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++--- .../mt8186/mt8186-mt6366-da7219-max98357.c | 86 ++++++-------- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 86 ++++++-------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 48 ++++---- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 ++++++------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 +++++----- sound/soc/meson/axg-card.c | 8 +- sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/samsung/odroid.c | 10 +- sound/soc/soc-compress.c | 11 +- sound/soc/soc-core.c | 20 +--- sound/soc/soc-dai.c | 4 +- sound/soc/soc-pcm.c | 107 +++++++----------- sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 4 - 84 files changed, 502 insertions(+), 842 deletions(-) -- 2.25.1

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How fix microphone for ALC4080
by GitHub issues - opened 24 May '23

24 May '23

alsa-project/alsa-ucm-conf issue #317 was opened from h4kuna: Hello, i'am running on Ubuntu 22.04 where is `lsa-ucm-conf 1.2.6.3-1ubuntu1.5`. The sound card does't work. I have motherboard [MAG X570S TOMAHAWK MAX WIFI](https://www.msi.com/Motherboard/MAG-X570S-TOMAHAWK-MAX-WIFI/Specifica… where is `Realtek® ALC4080`. I saw, it added later. I try to use live CD ubuntu 23.04, where sound works, but microphone still doesn't work. ### Ubuntu 22.04 Output from [alsa-info](http://alsa-project.org/db/?f=d2e40705662fc6a0f9e64af8e64bfd0ded… ``` lsusb | grep 'USB Audio' Bus 003 Device 002: ID 0db0:a073 Micro Star International USB Audio ``` What can i to do for fix microphone? What do you need information? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/317 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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