September 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH -next] ASoC: SOF: imx8: Fix COMPILE_TEST error
by YueHaibing 05 Sep '19

05 Sep '19

When do compile test, if SND_SOC_SOF_OF is not set, we get: sound/soc/sof/imx/imx8.o: In function `imx8_dsp_handle_request': imx8.c:(.text+0xb0): undefined reference to `snd_sof_ipc_msgs_rx' sound/soc/sof/imx/imx8.o: In function `imx8_ipc_msg_data': imx8.c:(.text+0xf4): undefined reference to `sof_mailbox_read' sound/soc/sof/imx/imx8.o: In function `imx8_dsp_handle_reply': imx8.c:(.text+0x160): undefined reference to `sof_mailbox_read' Make SND_SOC_SOF_IMX_TOPLEVEL always depends on SND_SOC_SOF_OF Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 202acc565a1f ("ASoC: SOF: imx: Add i.MX8 HW support") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sof/imx/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index fd73d84..5acae75 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -2,7 +2,8 @@ config SND_SOC_SOF_IMX_TOPLEVEL bool "SOF support for NXP i.MX audio DSPs" - depends on ARM64 && SND_SOC_SOF_OF || COMPILE_TEST + depends on ARM64|| COMPILE_TEST + depends on SND_SOC_SOF_OF help This adds support for Sound Open Firmware for NXP i.MX platforms. Say Y if you have such a device. -- 2.7.4

4 3

[alsa-devel] [PATCH]Add tas2770 driver code
by shifu0704＠thundersoft.com 05 Sep '19

05 Sep '19

From 5b539e7dbebebf7e58c38a6b54f222bc116d63f3 Mon Sep 17 00:00:00 2001 From: Shi Fu <shifu0704(a)thundersoft.com> Date: Mon, 1 Jul 2019 11:00:13 +0800 Signed-off-by: Shi Fu <shifu0704(a)thundersoft.com> Subject: [PATCH v5] dt: tas2770: Add tas2770 smart PA dt bindings. Add tas2770 smart PA dt bindings. --- Documentation/devicetree/bindings/tas2770.txt | 38 +++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) create mode 100644 Documentation/devicetree/bindings/tas2770.txt diff --git a/Documentation/devicetree/bindings/tas2770.txt b/Documentation/devicetree/bindings/tas2770.txt new file mode 100644 index 0000000..f70b310 --- /dev/null +++ b/Documentation/devicetree/bindings/tas2770.txt @@ -0,0 +1,38 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,left-slot: - Sets TDM RX left time slots. + - ti,right-slot: - Sets TDM RX right time slots. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + + - reset-gpio: Reset GPIO number of left device. + - irq-gpio: IRQ GPIO number of left device. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + reset-gpio = <&gpio15 1 GPIO_ACTIVE_LOW>; + irq-gpio = <&gpio16 1 GPIO_ACTIVE_LOW>; + ti,asi-format = <0>; + ti,left-slot = <0>; + ti,right-slot = <1>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4 From 167a85aa30a41471ccddd9912b192d92800a4212 Mon Sep 17 00:00:00 2001 From: TS Shi Fu <shifu0704(a)thundersoft.com> Date: Mon, 24 Jun 2019 18:26:09 +0800 Subject: [PATCH v5] Add tas2770 driver code Add tas2770 driver code Date: Wed, 15 May 2019 12:08:42 +0800 From: Shi Fu <shifu0704(a)thundersoft.com> Signed-off-by: Shi Fu <shifu0704(a)thundersoft.com> Change-Id: Ibd5b04b4c0d0021a8057023aa6b43f97bba372b1 --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2770.c | 1103 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2770.h | 173 +++++++ 4 files changed, 1283 insertions(+) create mode 100644 sound/soc/codecs/tas2770.c create mode 100644 sound/soc/codecs/tas2770.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8f3e787..cc92da3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -111,6 +111,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_STI_SAS select SND_SOC_TAS2552 if I2C + select SND_SOC_TAS2770 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TAS571X if I2C select SND_SOC_TFA9879 if I2C @@ -652,6 +653,10 @@ config SND_SOC_TAS2552 tristate "Texas Instruments TAS2552 Mono Audio amplifier" depends on I2C +config SND_SOC_TAS2770 + tristate "Texas Instruments TAS2770 speaker amplifier" + depends on I2C + 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 5305cc6..1b58d08 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -116,6 +116,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-sti-sas-objs := sti-sas.o snd-soc-tas5086-objs := tas5086.o snd-soc-tas571x-objs := tas571x.o +snd-soc-tas2770-objs := tas2770.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o @@ -332,6 +333,7 @@ obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o +obj-$(CONFIG_SND_SOC_TAS2770) += snd-soc-tas2770.o obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c new file mode 100644 index 0000000..9fc0c11 --- /dev/null +++ b/sound/soc/codecs/tas2770.c @@ -0,0 +1,1103 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * ALSA SoC Texas Instruments TAS2770 20-W Digital Input Mono Class-D + * Audio Amplifier with Speaker I/V Sense + * + * Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/ + * Author: Tracy Yi <tracy-yi(a)ti.com> + * Shi Fu <shifu0704(a)thundersoft.com> + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/err.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <linux/firmware.h> +#include <linux/regmap.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "tas2770.h" + +#define TAS2770_MDELAY 0xFFFFFFFE +#define TAS2770_CHECK_PERIOD 5000 /* 5 second */ + +static void tas2770_hw_reset(struct tas2770_priv *p_tas2770) +{ + if (p_tas2770->mn_reset_gpio) { + gpiod_set_value_cansleep(p_tas2770->mn_reset_gpio, 0); + msleep(20); + gpiod_set_value_cansleep(p_tas2770->mn_reset_gpio, 1); + } + + p_tas2770->mn_current_book = -1; + p_tas2770->mn_current_page = -1; +} + +static void tas2770_enable_irq(struct tas2770_priv *p_tas2770, bool enable) +{ + if (enable) { + if (p_tas2770->mb_irq_enable) + return; + + if (gpio_is_valid(p_tas2770->mn_irq)) + enable_irq(p_tas2770->mn_irq); + p_tas2770->mb_irq_enable = true; + } else { + if (gpio_is_valid(p_tas2770->mn_irq)) + disable_irq_nosync(p_tas2770->mn_irq); + p_tas2770->mb_irq_enable = false; + } +} + +static int tas2770_runtime_suspend(struct tas2770_priv *p_tas2770) +{ + p_tas2770->mb_runtime_suspend = true; + + return 0; +} + +static int tas2770_runtime_resume(struct tas2770_priv *p_tas2770) +{ + + p_tas2770->mb_runtime_suspend = false; + + return 0; +} + +static int tas2770_regmap_write(struct tas2770_priv *p_tas2770, + unsigned int reg, unsigned int value) +{ + int nResult = 0; + int retry_count = TAS2770_I2C_RETRY_COUNT; + + while (retry_count--) { + nResult = snd_soc_component_write(p_tas2770->component, reg, + value); + if (!nResult) + break; + msleep(20); + } + if (retry_count == -1) + return ERROR_I2C_FAILED; + else + return 0; +} + +static int tas2770_regmap_read(struct tas2770_priv *p_tas2770, + unsigned int reg, unsigned int *value) +{ + int nResult = 0; + int retry_count = TAS2770_I2C_RETRY_COUNT; + + while (retry_count--) { + nResult = snd_soc_component_read(p_tas2770->component, reg, + value); + if (!nResult) + break; + + msleep(20); + } + if (!retry_count) + return ERROR_I2C_FAILED; + + return 0; +} + +static int tas2770_codec_suspend(struct snd_soc_component *component) +{ + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + + mutex_lock(&p_tas2770->codec_lock); + tas2770_runtime_suspend(p_tas2770); + mutex_unlock(&p_tas2770->codec_lock); + + return 0; +} + +static int tas2770_codec_resume(struct snd_soc_component *component) +{ + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + + mutex_lock(&p_tas2770->codec_lock); + tas2770_runtime_resume(p_tas2770); + mutex_unlock(&p_tas2770->codec_lock); + + return 0; +} + +static const char * const tas2770_ASI1_src[] = { + "I2C offset", "Left", "Right", "LeftRightDiv2", +}; + +static SOC_ENUM_SINGLE_DECL( + tas2770_ASI1_src_enum, TAS2770_TDM_CFG_REG2, + 4, tas2770_ASI1_src); + +static const struct snd_kcontrol_new tas2770_asi1_mux = + SOC_DAPM_ENUM("ASI1 Source", tas2770_ASI1_src_enum); + +static int tas2770_set_power_state(struct tas2770_priv *p_tas2770, int state) +{ + struct snd_soc_component *component = p_tas2770->component; + + switch (state) { + case TAS2770_POWER_ACTIVE: + snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_ACTIVE); + p_tas2770->mn_power_state = TAS2770_POWER_ACTIVE; + tas2770_enable_irq(p_tas2770, true); + break; + + case TAS2770_POWER_MUTE: + snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_MUTE); + p_tas2770->mn_power_state = TAS2770_POWER_MUTE; + break; + + case TAS2770_POWER_SHUTDOWN: + snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_SHUTDOWN); + tas2770_enable_irq(p_tas2770, false); + p_tas2770->mn_power_state = TAS2770_POWER_SHUTDOWN; + break; + + default: + dev_err(p_tas2770->dev, + "wrong power state setting %d\n", state); + return -EINVAL; + } + + return 0; +} + +static int tas2770_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + int ret; + + mutex_lock(&p_tas2770->codec_lock); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + ret = tas2770_set_power_state(p_tas2770, + TAS2770_POWER_MUTE); + if (ret) + goto end; + break; + case SND_SOC_DAPM_PRE_PMD: + ret = tas2770_set_power_state(p_tas2770, + TAS2770_POWER_SHUTDOWN); + if (ret) + goto end; + break; + default: + dev_err(p_tas2770->dev, "Not supported evevt\n"); + ret = -EINVAL; + } + +end: + mutex_unlock(&p_tas2770->codec_lock); + return ret; +} + +static const struct snd_kcontrol_new isense_switch = + SOC_DAPM_SINGLE("Switch", TAS2770_PWR_CTRL, 3, 1, 1); +static const struct snd_kcontrol_new vsense_switch = + SOC_DAPM_SINGLE("Switch", TAS2770_PWR_CTRL, 2, 1, 1); + +static const struct snd_soc_dapm_widget tas2770_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, + &tas2770_asi1_mux), + SND_SOC_DAPM_SWITCH("ISENSE", TAS2770_PWR_CTRL, 3, 1, + &isense_switch), + SND_SOC_DAPM_SWITCH("VSENSE", TAS2770_PWR_CTRL, 2, 1, + &vsense_switch), + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2770_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_SIGGEN("VMON"), + SND_SOC_DAPM_SIGGEN("IMON") +}; + +static const struct snd_soc_dapm_route tas2770_audio_map[] = { + {"ASI1 Sel", "I2C offset", "ASI1"}, + {"ASI1 Sel", "Left", "ASI1"}, + {"ASI1 Sel", "Right", "ASI1"}, + {"ASI1 Sel", "LeftRightDiv2", "ASI1"}, + {"DAC", NULL, "ASI1 Sel"}, + {"OUT", NULL, "DAC"}, + {"ISENSE", "Switch", "IMON"}, + {"VSENSE", "Switch", "VMON"}, +}; + + +static int tas2770_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + int ret; + + + mutex_lock(&p_tas2770->codec_lock); + if (mute) + ret = tas2770_set_power_state(p_tas2770, TAS2770_POWER_MUTE); + else + ret = tas2770_set_power_state(p_tas2770, TAS2770_POWER_ACTIVE); + + mutex_unlock(&p_tas2770->codec_lock); + return ret; +} + +static int tas2770_set_slot(struct snd_soc_component *component, int slot_width) +{ + int ret; + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + + switch (slot_width) { + case 16: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXS_MASK, + TAS2770_TDM_CFG_REG2_RXS_16BITS); + break; + + case 24: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXS_MASK, + TAS2770_TDM_CFG_REG2_RXS_24BITS); + break; + + case 32: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXS_MASK, + TAS2770_TDM_CFG_REG2_RXS_32BITS); + break; + + case 0: + /* Do not change slot width */ + ret = 0; + break; + + default: + ret = -EINVAL; + } + + if (!ret) + p_tas2770->mn_slot_width = slot_width; + + return ret; +} + +static int tas2770_set_bitwidth(struct tas2770_priv *p_tas2770, int bitwidth) +{ + int ret; + struct snd_soc_component *component = p_tas2770->component; + + switch (bitwidth) { + case SNDRV_PCM_FORMAT_S16_LE: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXW_MASK, + TAS2770_TDM_CFG_REG2_RXW_16BITS); + /* If machine driver did not call set slot width */ + if (p_tas2770->mn_slot_width == 0) + tas2770_set_slot(component, TAS2770_SLOT_16BIT); + p_tas2770->mn_v_mon_slot_no = p_tas2770->mn_i_mon_slot_no + 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXW_MASK, + TAS2770_TDM_CFG_REG2_RXW_24BITS); + if (p_tas2770->mn_slot_width == 0) + tas2770_set_slot(component, TAS2770_SLOT_32BIT); + p_tas2770->mn_v_mon_slot_no = p_tas2770->mn_i_mon_slot_no + 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXW_MASK, + TAS2770_TDM_CFG_REG2_RXW_32BITS); + if (p_tas2770->mn_slot_width == 0) + tas2770_set_slot(component, TAS2770_SLOT_32BIT); + p_tas2770->mn_v_mon_slot_no = p_tas2770->mn_i_mon_slot_no + 4; + break; + + default: + return -EINVAL; + } + + p_tas2770->mn_ch_size = bitwidth; + + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG5, + TAS2770_TDM_CFG_REG5_VSNS_MASK | + TAS2770_TDM_CFG_REG5_50_MASK, + TAS2770_TDM_CFG_REG5_VSNS_ENABLE | + p_tas2770->mn_v_mon_slot_no); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG6, + TAS2770_TDM_CFG_REG6_ISNS_MASK | + TAS2770_TDM_CFG_REG6_50_MASK, + TAS2770_TDM_CFG_REG6_ISNS_ENABLE | + p_tas2770->mn_i_mon_slot_no); + +end: + return ret; +} + +static int tas2770_set_samplerate(struct tas2770_priv *p_tas2770, + int samplerate) +{ + int ret; + struct snd_soc_component *component = p_tas2770->component; + + switch (samplerate) { + case 48000: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_48KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); + if (ret) + goto end; + break; + case 44100: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); + if (ret) + goto end; + break; + case 96000: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_48KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); + break; + case 88200: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); + break; + case 19200: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_48KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); + if (ret) + goto end; + break; + case 17640: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); + break; + default: + ret = -EINVAL; + } + +end: + if (!ret) + p_tas2770->mn_sampling_rate = samplerate; + return ret; +} + +static int tas2770_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + int ret; + + mutex_lock(&p_tas2770->codec_lock); + + ret = tas2770_set_bitwidth(p_tas2770, params_format(params)); + if (ret < 0) + goto end; + + + ret = tas2770_set_samplerate(p_tas2770, params_rate(params)); + +end: + mutex_unlock(&p_tas2770->codec_lock); + return ret; +} + +static int tas2770_set_fmt(struct tas2770_priv *p_tas2770, unsigned int fmt) +{ + u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; + int ret; + int value = 0; + struct snd_soc_component *component = p_tas2770->component; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(p_tas2770->dev, "ASI format master is not found\n"); + ret = -EINVAL; + return ret; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + asi_cfg_1 |= TAS2770_TDM_CFG_REG1_RX_RSING; + break; + case SND_SOC_DAIFMT_IB_NF: + asi_cfg_1 |= TAS2770_TDM_CFG_REG1_RX_FALING; + break; + default: + dev_err(p_tas2770->dev, "ASI format Inverse is not found\n"); + ret = -EINVAL; + return ret; + } + + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, + TAS2770_TDM_CFG_REG1_RX_MASK, + asi_cfg_1); + if (ret) + return ret; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case (SND_SOC_DAIFMT_I2S): + tdm_rx_start_slot = 1; + break; + case (SND_SOC_DAIFMT_DSP_A): + case (SND_SOC_DAIFMT_DSP_B): + tdm_rx_start_slot = 1; + break; + case (SND_SOC_DAIFMT_LEFT_J): + tdm_rx_start_slot = 0; + break; + default: + dev_err(p_tas2770->dev, + "DAI Format is not found, fmt=0x%x\n", fmt); + ret = -EINVAL; + return ret; + } + + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, + TAS2770_TDM_CFG_REG1_MASK, + (tdm_rx_start_slot << TAS2770_TDM_CFG_REG1_51_SHIFT)); + if (ret) + return ret; + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, + TAS2770_TDM_CFG_REG3_30_MASK, + (p_tas2770->mn_left_slot << + TAS2770_TDM_CFG_REG3_30_SHIFT)); + if (ret) + return ret; + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, + TAS2770_TDM_CFG_REG3_RXS_MASK, + (p_tas2770->mn_right_slot << + TAS2770_TDM_CFG_REG3_RXS_SHIFT)); + if (ret) + return ret; + + value = snd_soc_component_read32(component, TAS2770_TDM_CFG_REG3); + + p_tas2770->mn_asi_format = fmt; + + return 0; +} + +static int tas2770_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + int ret; + + mutex_lock(&p_tas2770->codec_lock); + + ret = tas2770_set_fmt(p_tas2770, fmt); + + mutex_unlock(&p_tas2770->codec_lock); + return ret; +} + +static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + int ret; + struct snd_soc_component *component = dai->component; + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + + mutex_lock(&p_tas2770->codec_lock); + ret = tas2770_set_slot(component, slot_width); + mutex_unlock(&p_tas2770->codec_lock); + + return ret; +} + +static struct snd_soc_dai_ops tas2770_dai_ops = { + .digital_mute = tas2770_mute, + .hw_params = tas2770_hw_params, + .set_fmt = tas2770_set_dai_fmt, + .set_tdm_slot = tas2770_set_dai_tdm_slot, +}; + +#define TAS2770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +#define TAS2770_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_192000\ + ) + +static struct snd_soc_dai_driver tas2770_dai_driver[] = { + { + .name = "tas2770 ASI1", + .id = 0, + .playback = { + .stream_name = "ASI1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TAS2770_RATES, + .formats = TAS2770_FORMATS, + }, + .capture = { + .stream_name = "ASI1 Capture", + .channels_min = 0, + .channels_max = 2, + .rates = TAS2770_RATES, + .formats = TAS2770_FORMATS, + }, + .ops = &tas2770_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int tas2770_codec_probe(struct snd_soc_component *component) +{ + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + + p_tas2770->component = component; + + return 0; +} + +static void tas2770_codec_remove(struct snd_soc_component *component) +{ + pm_runtime_put(component->dev); +} + +static DECLARE_TLV_DB_SCALE(tas2770_digital_tlv, 1100, 50, 0); +static DECLARE_TLV_DB_SCALE(tas2770_playback_volume, -12750, 50, 0); + +static const struct snd_kcontrol_new tas2770_snd_controls[] = { + SOC_SINGLE_TLV("Amp Output Level", TAS2770_PLAY_CFG_REG0, + 0, 0x14, 0, + tas2770_digital_tlv), + SOC_SINGLE_TLV("Playback Volume", TAS2770_PLAY_CFG_REG2, + 0, TAS2770_PLAY_CFG_REG2_VMAX, 1, + tas2770_playback_volume), +}; + +static const struct snd_soc_component_driver soc_component_driver_tas2770 = { + .probe = tas2770_codec_probe, + .remove = tas2770_codec_remove, + .suspend = tas2770_codec_suspend, + .resume = tas2770_codec_resume, + .controls = tas2770_snd_controls, + .num_controls = ARRAY_SIZE(tas2770_snd_controls), + .dapm_widgets = tas2770_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas2770_dapm_widgets), + .dapm_routes = tas2770_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas2770_audio_map), + .idle_bias_on = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static int tas2770_register_codec(struct tas2770_priv *p_tas2770) +{ + return devm_snd_soc_register_component(p_tas2770->dev, + &soc_component_driver_tas2770, + tas2770_dai_driver, ARRAY_SIZE(tas2770_dai_driver)); +} + +static void tas2770_load_config(struct tas2770_priv *p_tas2770) +{ + int ret; + + tas2770_hw_reset(p_tas2770); + tas2770_regmap_write(p_tas2770, TAS2770_SW_RST, + TAS2770_RST); + + ret = tas2770_set_slot(p_tas2770->component, p_tas2770->mn_slot_width); + if (ret) + goto error_out; + + ret = tas2770_set_fmt(p_tas2770, p_tas2770->mn_asi_format); + if (ret) + goto error_out; + + ret = tas2770_set_bitwidth(p_tas2770, p_tas2770->mn_ch_size); + if (ret) + goto error_out; + + ret = tas2770_set_samplerate(p_tas2770, p_tas2770->mn_sampling_rate); + if (ret) + goto error_out; + + ret = tas2770_set_power_state(p_tas2770, p_tas2770->mn_power_state); + if (ret) + goto error_out; + +error_out: + return; +} + +static const struct reg_default tas2770_reg_defaults[] = { + { TAS2770_PAGE, 0x00 }, + { TAS2770_SW_RST, 0x00 }, + { TAS2770_PWR_CTRL, 0x0e }, + { TAS2770_PLAY_CFG_REG0, 0x10 }, + { TAS2770_PLAY_CFG_REG1, 0x01 }, + { TAS2770_PLAY_CFG_REG2, 0x00 }, + { TAS2770_MSC_CFG_REG0, 0x07 }, + { TAS2770_TDM_CFG_REG1, 0x02 }, + { TAS2770_TDM_CFG_REG2, 0x0a }, + { TAS2770_TDM_CFG_REG3, 0x10 }, + { TAS2770_INT_MASK_REG0, 0xfc }, + { TAS2770_INT_MASK_REG1, 0xb1 }, + { TAS2770_INT_CFG, 0x05 }, + { TAS2770_MISC_IRQ, 0x81 }, + { TAS2770_CLK_CGF, 0x0c }, + +}; + +static bool tas2770_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS2770_PAGE: /* regmap implementation requires this */ + case TAS2770_SW_RST: /* always clears after write */ + case TAS2770_BO_PRV_REG0:/* has a self clearing bit */ + case TAS2770_LVE_INT_REG0: + case TAS2770_LVE_INT_REG1: + case TAS2770_LAT_INT_REG0:/* Sticky interrupt flags */ + case TAS2770_LAT_INT_REG1:/* Sticky interrupt flags */ + case TAS2770_VBAT_MSB: + case TAS2770_VBAT_LSB: + case TAS2770_TEMP_MSB: + case TAS2770_TEMP_LSB: + return true; + } + return false; +} + +static bool tas2770_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS2770_LVE_INT_REG0: + case TAS2770_LVE_INT_REG1: + case TAS2770_LAT_INT_REG0: + case TAS2770_LAT_INT_REG1: + case TAS2770_VBAT_MSB: + case TAS2770_VBAT_LSB: + case TAS2770_TEMP_MSB: + case TAS2770_TEMP_LSB: + case TAS2770_TDM_CLK_DETC: + case TAS2770_REV_AND_GPID: + return false; + } + return true; +} + +static const struct regmap_range_cfg tas2770_regmap_ranges[] = { + { + .range_min = 0, + .range_max = 1 * 128, + .selector_reg = TAS2770_PAGE, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config tas2770_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = tas2770_writeable, + .volatile_reg = tas2770_volatile, + .reg_defaults = tas2770_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2770_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .ranges = tas2770_regmap_ranges, + .num_ranges = ARRAY_SIZE(tas2770_regmap_ranges), + .max_register = 1 * 128, +}; + +static irqreturn_t tas2770_irq_handler(int irq, void *dev_id) +{ + struct tas2770_priv *p_tas2770 = (struct tas2770_priv *)dev_id; + + unsigned int nDevInt1Status = 0, nDevInt2Status = 0; + int n_counter = 2; + int n_result; + + mutex_lock(&p_tas2770->codec_lock); + + tas2770_enable_irq(p_tas2770, false); + + if (p_tas2770->mb_runtime_suspend) + goto end; + + if (p_tas2770->mn_power_state == TAS2770_POWER_SHUTDOWN) + goto end; + + n_result = tas2770_regmap_write(p_tas2770, TAS2770_INT_MASK_REG0, + TAS2770_INT_MASK_REG0_DISABLE); + if (n_result) + goto reload; + n_result = tas2770_regmap_write(p_tas2770, TAS2770_INT_MASK_REG1, + TAS2770_INT_MASK_REG1_DISABLE); + if (n_result) + goto reload; + + n_result = tas2770_regmap_read(p_tas2770, + TAS2770_LAT_INT_REG0, &nDevInt1Status); + if (n_result >= 0) + n_result = tas2770_regmap_read(p_tas2770, + TAS2770_LAT_INT_REG1, &nDevInt2Status); + else + goto reload; + + if (((nDevInt1Status & 0x3) != 0) || ((nDevInt2Status & 0x0f) != 0)) { + /* in case of INT_OC, INT_OT, INT_OVLT, INT_UVLT, INT_BO */ + + if (nDevInt1Status & + TAS2770_LAT_INT_REG0_OCE_FLG) { + p_tas2770->mn_err_code |= ERROR_OVER_CURRENT; + dev_err(p_tas2770->dev, "SPK over current!\n"); + } else { + p_tas2770->mn_err_code &= ~ERROR_OVER_CURRENT; + } + + if (nDevInt1Status & + TAS2770_LAT_INT_REG0_OTE_FLG) { + p_tas2770->mn_err_code |= ERROR_DIE_OVERTEMP; + dev_err(p_tas2770->dev, "die over temperature!\n"); + } else { + p_tas2770->mn_err_code &= ~ERROR_DIE_OVERTEMP; + } + + if (nDevInt2Status & + TAS2770_LAT_INT_REG1_VBA_TOV) { + p_tas2770->mn_err_code |= ERROR_OVER_VOLTAGE; + dev_err(p_tas2770->dev, "SPK over voltage!\n"); + } else { + p_tas2770->mn_err_code &= ~ERROR_OVER_VOLTAGE; + } + + if (nDevInt2Status & + TAS2770_LAT_INT_REG1_VBA_TUV) { + p_tas2770->mn_err_code |= ERROR_UNDER_VOLTAGE; + dev_err(p_tas2770->dev, "SPK under voltage!\n"); + } else { + p_tas2770->mn_err_code &= ~ERROR_UNDER_VOLTAGE; + } + + if (nDevInt2Status & + TAS2770_LAT_INT_REG1_BOUT_FLG) { + p_tas2770->mn_err_code |= ERROR_BROWNOUT; + dev_err(p_tas2770->dev, "brownout!\n"); + } else { + p_tas2770->mn_err_code &= ~ERROR_BROWNOUT; + } + + goto reload; + } else { + n_counter = 2; + + while (n_counter > 0) { + n_result = tas2770_regmap_read(p_tas2770, + TAS2770_PWR_CTRL, &nDevInt1Status); + if (n_result) + goto reload; + + if ((nDevInt1Status & + TAS2770_PWR_CTRL_MASK) + != TAS2770_PWR_CTRL_SHUTDOWN) + break; + + n_counter--; + if (n_counter > 0) + msleep(20); + } + + if ((nDevInt1Status & + TAS2770_PWR_CTRL_MASK) + == TAS2770_PWR_CTRL_SHUTDOWN) { + dev_err(p_tas2770->dev, "%s, Critical ERROR REG[0x%x] = 0x%x\n", + __func__, + TAS2770_PWR_CTRL, + nDevInt1Status); + p_tas2770->mn_err_code |= ERROR_CLASSD_PWR; + goto reload; + } + p_tas2770->mn_err_code &= ~ERROR_CLASSD_PWR; + } + + n_result = tas2770_regmap_write(p_tas2770, + TAS2770_INT_MASK_REG0, 0xfc); + if (n_result) + goto reload; + + n_result = tas2770_regmap_write(p_tas2770, + TAS2770_INT_MASK_REG1, 0xb1); + if (n_result) + goto reload; + + goto end; + +reload: + /* hardware reset and reload */ + tas2770_load_config(p_tas2770); + +end: + mutex_unlock(&p_tas2770->codec_lock); + tas2770_enable_irq(p_tas2770, true); + + return IRQ_HANDLED; +} + +static int tas2770_parse_dt(struct device *dev, struct tas2770_priv *p_tas2770) +{ + int rc = 0; + + rc = fwnode_property_read_u32(dev->fwnode, "ti,asi-format", + &p_tas2770->mn_asi_format); + if (rc) { + dev_err(p_tas2770->dev, "Looking up %s property failed %d\n", + "ti,asi-format", rc); + goto end; + } + + rc = fwnode_property_read_u32(dev->fwnode, "ti,left-slot", + &p_tas2770->mn_left_slot); + if (rc) { + dev_err(p_tas2770->dev, "Looking up %s property failed %d\n", + "ti,left-slot", rc); + goto end; + } + + rc = fwnode_property_read_u32(dev->fwnode, "ti,right-slot", + &p_tas2770->mn_right_slot); + if (rc) { + dev_err(p_tas2770->dev, "Looking up %s property failed %d\n", + "ti,right-slot", rc); + goto end; + } + + rc = fwnode_property_read_u32(dev->fwnode, "ti,imon-slot-no", + &p_tas2770->mn_i_mon_slot_no); + if (rc) { + dev_err(p_tas2770->dev, "Looking up %s property failed %d\n", + "ti,imon-slot-no", rc); + goto end; + } + + rc = fwnode_property_read_u32(dev->fwnode, "ti,vmon-slot-no", + &p_tas2770->mn_v_mon_slot_no); + if (rc) { + dev_err(p_tas2770->dev, "Looking up %s property failed %d\n", + "ti,vmon-slot-no", rc); + goto end; + } + +end: + return rc; +} + +static int tas2770_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct tas2770_priv *p_tas2770; + int n_result; + + p_tas2770 = devm_kzalloc(&client->dev, + sizeof(struct tas2770_priv), GFP_KERNEL); + if (p_tas2770 == NULL) + return -ENOMEM; + p_tas2770->dev = &client->dev; + p_tas2770->mn_irq = client->irq; + + i2c_set_clientdata(client, p_tas2770); + dev_set_drvdata(&client->dev, p_tas2770); + p_tas2770->mn_power_state = TAS2770_POWER_SHUTDOWN; + + p_tas2770->regmap = devm_regmap_init_i2c(client, &tas2770_i2c_regmap); + if (IS_ERR(p_tas2770->regmap)) { + n_result = PTR_ERR(p_tas2770->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + n_result); + goto end; + } + + if (client->dev.of_node) { + n_result = tas2770_parse_dt(&client->dev, p_tas2770); + if (n_result) { + dev_err(p_tas2770->dev, "%s: Failed to parse devicetree\n", + __func__); + goto end; + } + } + + p_tas2770->mn_reset_gpio = devm_gpiod_get_optional(p_tas2770->dev, + "reset-gpio", + GPIOD_OUT_HIGH); + if (IS_ERR(p_tas2770->mn_reset_gpio)) { + if (PTR_ERR(p_tas2770->mn_reset_gpio) == -EPROBE_DEFER) + return -EPROBE_DEFER; + p_tas2770->mn_reset_gpio = NULL; + } + + if (p_tas2770->mn_irq > 0) { + n_result = devm_request_threaded_irq(p_tas2770->dev, + p_tas2770->mn_irq, + NULL, tas2770_irq_handler, + IRQF_TRIGGER_FALLING | IRQF_ONESHOT, client->name, + p_tas2770); + if (n_result) { + dev_err(p_tas2770->dev, "Unable to request IRQ\n"); + return n_result; + } + } + + p_tas2770->mn_ch_size = 0; + p_tas2770->mn_slot_width = 0; + + tas2770_hw_reset(p_tas2770); + tas2770_regmap_write(p_tas2770, TAS2770_SW_RST, + TAS2770_RST); + + mutex_init(&p_tas2770->dev_lock); + + mutex_init(&p_tas2770->codec_lock); + n_result = tas2770_register_codec(p_tas2770); + if (n_result) + dev_err(p_tas2770->dev, "Register codec failed.\n"); + +end: + return n_result; +} + +static int tas2770_i2c_remove(struct i2c_client *client) +{ + struct tas2770_priv *p_tas2770 = i2c_get_clientdata(client); + + mutex_destroy(&p_tas2770->dev_lock); + + return 0; +} + + +static const struct i2c_device_id tas2770_i2c_id[] = { + { "tas2770", 0}, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2770_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id tas2770_of_match[] = { + { .compatible = "ti,tas2770" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2770_of_match); +#endif + +static struct i2c_driver tas2770_i2c_driver = { + .driver = { + .name = "tas2770", + .owner = THIS_MODULE, +#if defined(CONFIG_OF) + .of_match_table = of_match_ptr(tas2770_of_match), +#endif + }, + .probe = tas2770_i2c_probe, + .remove = tas2770_i2c_remove, + .id_table = tas2770_i2c_id, +}; + +module_i2c_driver(tas2770_i2c_driver); + +MODULE_AUTHOR("Shi Fu <shifu0704(a)thundersoft.com>"); +MODULE_DESCRIPTION("TAS2770 I2C Smart Amplifier driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/tas2770.h b/sound/soc/codecs/tas2770.h new file mode 100644 index 0000000..f08e208 --- /dev/null +++ b/sound/soc/codecs/tas2770.h @@ -0,0 +1,173 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC TAS2770 codec driver + * + * Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/ + */ +#ifndef __TAS2770__ +#define __TAS2770__ + +/* Book Control Register (available in page0 of each book) */ +#define TAS2770_BOOKCTL_PAGE 0 +#define TAS2770_BOOKCTL_REG 127 +#define TAS2770_REG(page, reg) ((page * 128) + reg) + /* Page */ +#define TAS2770_PAGE TAS2770_REG(0X0, 0x00) +#define TAS2770_PAGE_PAGE_MASK 255 + /* Software Reset */ +#define TAS2770_SW_RST TAS2770_REG(0X0, 0x01) +#define TAS2770_RST BIT(0) + /* Power Control */ +#define TAS2770_PWR_CTRL TAS2770_REG(0X0, 0x02) +#define TAS2770_PWR_CTRL_MASK 0x3 +#define TAS2770_PWR_CTRL_ACTIVE 0x0 +#define TAS2770_PWR_CTRL_MUTE BIT(0) +#define TAS2770_PWR_CTRL_SHUTDOWN 0x2 + /* Playback Configuration Reg0 */ +#define TAS2770_PLAY_CFG_REG0 TAS2770_REG(0X0, 0x03) + /* Playback Configuration Reg1 */ +#define TAS2770_PLAY_CFG_REG1 TAS2770_REG(0X0, 0x04) + /* Playback Configuration Reg2 */ +#define TAS2770_PLAY_CFG_REG2 TAS2770_REG(0X0, 0x05) +#define TAS2770_PLAY_CFG_REG2_VMAX 0xc9 + /* Misc Configuration Reg0 */ +#define TAS2770_MSC_CFG_REG0 TAS2770_REG(0X0, 0x07) + /* TDM Configuration Reg0 */ +#define TAS2770_TDM_CFG_REG0 TAS2770_REG(0X0, 0x0A) +#define TAS2770_TDM_CFG_REG0_SMP_MASK BIT(5) +#define TAS2770_TDM_CFG_REG0_SMP_48KHZ 0x0 +#define TAS2770_TDM_CFG_REG0_SMP_44_1KHZ BIT(5) +#define TAS2770_TDM_CFG_REG0_31_MASK 0xe +#define TAS2770_TDM_CFG_REG0_31_44_1_48KHZ 0x6 +#define TAS2770_TDM_CFG_REG0_31_88_2_96KHZ 0x8 +#define TAS2770_TDM_CFG_REG0_31_176_4_192KHZ 0xa + /* TDM Configuration Reg1 */ +#define TAS2770_TDM_CFG_REG1 TAS2770_REG(0X0, 0x0B) +#define TAS2770_TDM_CFG_REG1_MASK 0x3e +#define TAS2770_TDM_CFG_REG1_51_SHIFT 1 +#define TAS2770_TDM_CFG_REG1_RX_MASK BIT(0) +#define TAS2770_TDM_CFG_REG1_RX_RSING 0x0 +#define TAS2770_TDM_CFG_REG1_RX_FALING BIT(0) + /* TDM Configuration Reg2 */ +#define TAS2770_TDM_CFG_REG2 TAS2770_REG(0X0, 0x0C) +#define TAS2770_TDM_CFG_REG2_RXW_MASK 0xc +#define TAS2770_TDM_CFG_REG2_RXW_16BITS 0x0 +#define TAS2770_TDM_CFG_REG2_RXW_24BITS 0x8 +#define TAS2770_TDM_CFG_REG2_RXW_32BITS 0xc +#define TAS2770_TDM_CFG_REG2_RXS_MASK 0x3 +#define TAS2770_TDM_CFG_REG2_RXS_16BITS 0x0 +#define TAS2770_TDM_CFG_REG2_RXS_24BITS BIT(0) +#define TAS2770_TDM_CFG_REG2_RXS_32BITS 0x2 + /* TDM Configuration Reg3 */ +#define TAS2770_TDM_CFG_REG3 TAS2770_REG(0X0, 0x0D) +#define TAS2770_TDM_CFG_REG3_RXS_MASK 0xf0 +#define TAS2770_TDM_CFG_REG3_RXS_SHIFT 0x4 +#define TAS2770_TDM_CFG_REG3_30_MASK 0xf +#define TAS2770_TDM_CFG_REG3_30_SHIFT 0 + /* TDM Configuration Reg5 */ +#define TAS2770_TDM_CFG_REG5 TAS2770_REG(0X0, 0x0F) +#define TAS2770_TDM_CFG_REG5_VSNS_MASK BIT(6) +#define TAS2770_TDM_CFG_REG5_VSNS_ENABLE BIT(6) +#define TAS2770_TDM_CFG_REG5_50_MASK 0x3f + /* TDM Configuration Reg6 */ +#define TAS2770_TDM_CFG_REG6 TAS2770_REG(0X0, 0x10) +#define TAS2770_TDM_CFG_REG6_ISNS_MASK BIT(6) +#define TAS2770_TDM_CFG_REG6_ISNS_ENABLE BIT(6) +#define TAS2770_TDM_CFG_REG6_50_MASK 0x3f + /* Brown Out Prevention Reg0 */ +#define TAS2770_BO_PRV_REG0 TAS2770_REG(0X0, 0x1B) + /* Interrupt MASK Reg0 */ +#define TAS2770_INT_MASK_REG0 TAS2770_REG(0X0, 0x20) +#define TAS2770_INT_MASK_REG0_DISABLE 0xff + /* Interrupt MASK Reg1 */ +#define TAS2770_INT_MASK_REG1 TAS2770_REG(0X0, 0x21) +#define TAS2770_INT_MASK_REG1_DISABLE 0xff + /* Live-Interrupt Reg0 */ +#define TAS2770_LVE_INT_REG0 TAS2770_REG(0X0, 0x22) + /* Live-Interrupt Reg1 */ +#define TAS2770_LVE_INT_REG1 TAS2770_REG(0X0, 0x23) + /* Latched-Interrupt Reg0 */ +#define TAS2770_LAT_INT_REG0 TAS2770_REG(0X0, 0x24) +#define TAS2770_LAT_INT_REG0_OCE_FLG BIT(1) +#define TAS2770_LAT_INT_REG0_OTE_FLG BIT(0) + /* Latched-Interrupt Reg1 */ +#define TAS2770_LAT_INT_REG1 TAS2770_REG(0X0, 0x25) +#define TAS2770_LAT_INT_REG1_VBA_TOV BIT(3) +#define TAS2770_LAT_INT_REG1_VBA_TUV BIT(2) +#define TAS2770_LAT_INT_REG1_BOUT_FLG BIT(1) + /* VBAT MSB */ +#define TAS2770_VBAT_MSB TAS2770_REG(0X0, 0x27) + /* VBAT LSB */ +#define TAS2770_VBAT_LSB TAS2770_REG(0X0, 0x28) + /* TEMP MSB */ +#define TAS2770_TEMP_MSB TAS2770_REG(0X0, 0x29) + /* TEMP LSB */ +#define TAS2770_TEMP_LSB TAS2770_REG(0X0, 0x2A) + /* Interrupt Configuration */ +#define TAS2770_INT_CFG TAS2770_REG(0X0, 0x30) + /* Misc IRQ */ +#define TAS2770_MISC_IRQ TAS2770_REG(0X0, 0x32) + /* Clock Configuration */ +#define TAS2770_CLK_CGF TAS2770_REG(0X0, 0x3C) + /* TDM Clock detection monitor */ +#define TAS2770_TDM_CLK_DETC TAS2770_REG(0X0, 0x77) + /* Revision and PG ID */ +#define TAS2770_REV_AND_GPID TAS2770_REG(0X0, 0x7D) + +#define TAS2770_POWER_ACTIVE 0 +#define TAS2770_POWER_MUTE 1 +#define TAS2770_POWER_SHUTDOWN 2 +#define ERROR_OVER_CURRENT 0x0000001 +#define ERROR_DIE_OVERTEMP 0x0000002 +#define ERROR_OVER_VOLTAGE 0x0000004 +#define ERROR_UNDER_VOLTAGE 0x0000008 +#define ERROR_BROWNOUT 0x0000010 +#define ERROR_CLASSD_PWR 0x0000020 +#define TAS2770_SLOT_16BIT 16 +#define TAS2770_SLOT_32BIT 32 +#define TAS2770_I2C_RETRY_COUNT 3 +#define ERROR_I2C_FAILED -2 + +struct tas2770_register { + int book; + int page; + int reg; +}; + +struct tas2770_dai_cfg { + unsigned int dai_fmt; + unsigned int tdm_delay; +}; + +struct tas2770_priv { + struct device *dev; + struct regmap *regmap; + struct snd_soc_codec *codec; + struct snd_soc_component *component; + struct mutex dev_lock; + struct hrtimer mtimer; + int mn_clkin; + int mn_clkid; + int mn_power_state; + int mn_current_book; + int mn_current_page; + int mn_load; + int mn_asi_format; + struct gpio_desc *mn_reset_gpio; + int mn_irq; + bool mb_irq_enable; + int mn_sampling_rate; + int mn_frame_size; + int mn_pll; + int mn_left_slot; + int mn_right_slot; + int mn_ch_size; + int mn_slot_width; + int mn_v_mon_slot_no; + int mn_i_mon_slot_no; + bool mb_runtime_suspend; + unsigned int mn_err_code; + struct mutex codec_lock; +}; + +#endif /* __TAS2770__ */ -- 2.7.4 Frank Shi 石扶 Thunder Software Technology Co., Ltd. 中科创达软件股份有限公司 Mobile: +86-13002957678 | Email: shifu0704(a)thundersoft.com Add: 10F, #1, Digital China Technology Park, No.20, Zhangba Rd. 4th, High-tech Zone, Xi'an City, Shaanxi Province, 710075, P.R. China 西安市高新区丈八四路20号神州数码科技园1号楼10层，710075 www.thundersoft.com CONFIDENTIALITY. This e-mail and its attachments are intended only for the person or entity whose address is listed above and may contain confidential and/or privileged material. Any retransmission, dissemination or other use of this information by persons or entities other than the intended recipient is prohibited. If you received this e-mail in error, please do not disclose the contents to anyone, but notify the sender by email immediately, and delete this e-mail and any attachments from your system. Thank you. 保密提示。本电子邮件及其附件仅适用于上述地址的个人或实体，并可能包含机密和/或特许资料。禁止对目标收件人以外的个人或实体进行此类信息的任何转播，传播或其他使用。如果您误收了此电子邮件，请不要向任何人透露这些内容，而应立即以邮件形式通知发件人，并从您的系统中删除本邮件和任何附件。谢谢。

2 1

[alsa-devel] [PATCH -next] ASoC: amd: use devm_platform_ioremap_resource() to simplify code
by YueHaibing 04 Sep '19

04 Sep '19

Use devm_platform_ioremap_resource() to simplify the code a bit. This is detected by coccinelle. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/amd/acp-pcm-dma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index d26653f..52225b4 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1251,8 +1251,7 @@ static int acp_audio_probe(struct platform_device *pdev) if (!audio_drv_data) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - audio_drv_data->acp_mmio = devm_ioremap_resource(&pdev->dev, res); + audio_drv_data->acp_mmio = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(audio_drv_data->acp_mmio)) return PTR_ERR(audio_drv_data->acp_mmio); -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: tegra: use devm_platform_ioremap_resource() to simplify code
by YueHaibing 04 Sep '19

04 Sep '19

Use devm_platform_ioremap_resource() to simplify the code a bit. This is detected by coccinelle. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/tegra/tegra30_ahub.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 9523812..635eacb 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -511,7 +511,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) const struct tegra30_ahub_soc_data *soc_data; struct reset_control *rst; int i; - struct resource *res0, *res1; + struct resource *res0; void __iomem *regs_apbif, *regs_ahub; int ret = 0; @@ -587,8 +587,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) } regcache_cache_only(ahub->regmap_apbif, true); - res1 = platform_get_resource(pdev, IORESOURCE_MEM, 1); - regs_ahub = devm_ioremap_resource(&pdev->dev, res1); + regs_ahub = devm_platform_ioremap_resource(pdev, 1); if (IS_ERR(regs_ahub)) return PTR_ERR(regs_ahub); -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: sirf-audio: use devm_platform_ioremap_resource() to simplify code
by YueHaibing 04 Sep '19

04 Sep '19

Use devm_platform_ioremap_resource() to simplify the code a bit. This is detected by coccinelle. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/sirf-audio-codec.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 9009a74..a061d78 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -459,7 +459,6 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev) int ret; struct sirf_audio_codec *sirf_audio_codec; void __iomem *base; - struct resource *mem_res; sirf_audio_codec = devm_kzalloc(&pdev->dev, sizeof(struct sirf_audio_codec), GFP_KERNEL); @@ -468,8 +467,7 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev) platform_set_drvdata(pdev, sirf_audio_codec); - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(&pdev->dev, mem_res); + base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(base)) return PTR_ERR(base); -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: ep93xx: use devm_platform_ioremap_resource() to simplify code
by YueHaibing 04 Sep '19

04 Sep '19

Use devm_platform_ioremap_resource() to simplify the code a bit. This is detected by coccinelle. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/cirrus/ep93xx-ac97.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index 84c967f..e21eaa1 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -362,7 +362,6 @@ static const struct snd_soc_component_driver ep93xx_ac97_component = { static int ep93xx_ac97_probe(struct platform_device *pdev) { struct ep93xx_ac97_info *info; - struct resource *res; int irq; int ret; @@ -370,8 +369,7 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) if (!info) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - info->regs = devm_ioremap_resource(&pdev->dev, res); + info->regs = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(info->regs)) return PTR_ERR(info->regs); -- 2.7.4

2 1

[alsa-devel] [RFC PATCH 0/5] ASoC: SOF: Intel: SoundWire initial integration
by Pierre-Louis Bossart 04 Sep '19

04 Sep '19

This RFC is the companion of the other RFC 'soundwire: intel: simplify DAI/PDI handling'. Our purpose at this point is to gather feedback on the interfaces between the Intel SOF parts and the SoundWire code. The suggested solution is a simple init/release inserted at probe/remove and resume/suspend, as well as two callbacks for the SOF driver to generate IPC configurations with the firmware. That level of separation completely hides the details of the SoundWire DAIs and will allow for 'transparent' multi-cpu DAI support, which will be handled in the machine driver and the soundwire DAIs. This solution was tested on IceLake and CometLake, and captures the feedback from SOF contributors on an initial integration that was deemed too complicated (and rightly so). Pierre-Louis Bossart (5): ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add helper to initialize SoundWire IP ASoC: SOF: Intel: hda: add SoundWire IP support ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: add support for SoundWire suspend/resume include/sound/sof/dai-intel.h | 18 ++-- sound/soc/sof/intel/hda-dsp.c | 11 +++ sound/soc/sof/intel/hda.c | 157 ++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 11 +++ 4 files changed, 188 insertions(+), 9 deletions(-) base-commit: 3b3aaa017e8072b1bfddda92be296b3463d870be -- 2.20.1

3 19

[alsa-devel] [PATCH v3 1/4] ASoC: es8316: judge PCM rate at later timing
by Katsuhiro Suzuki 04 Sep '19

04 Sep '19

This patch change the judge timing about playing/capturing PCM rate. Original code set constraints list of PCM rate limits at set_sysclk. This strategy works well if system is using fixed rate clock. But some boards and SoC (such as RockPro64 and RockChip I2S) has connected SoC MCLK out to ES8316 MCLK in. In this case, SoC side I2S will choose suitable frequency of MCLK such as fs * mclk-fs when user starts playing or capturing. Bad scenario as follows (mclk-fs = 256): - Initialize sysclk by correct value (Ex. 12.288MHz) - ES8316 set constraints of PCM rate by sysclk 48kHz (1/256), 32kHz (1/384), 30.720kHz (1/400), 24kHz (1/512), 16kHz (1/768), 12kHz (1/1024) - Play 48kHz sound, it's acceptable - Sysclk is not changed - Play 32kHz sound, it's acceptable - Set sysclk by 8.192MHz (= fs * mclk-fs = 32k * 256) - ES8316 set constraints of PCM rate by sysclk 32kHz (1/256), 21.33kHz (1/384), 20.48kHz (1/400), 16kHz (1/512), 10.66kHz (1/768), 8kHz (1/1024) - Play 48kHz again, but it's NOT acceptable because constraints list does not allow 48kHz Root cause of this strange behavior is changing constraints list at set_sysclk timing. It seems that is too early to determine. So this patch does not use constraints list and check PCM rate limit more later timing at hw_params. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/codecs/es8316.c | 37 +++++++++++++++---------------------- 1 file changed, 15 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index ed2959dbe1fb..229808fa627c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -363,27 +363,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); - int i; - int count = 0; es8316->sysclk = freq; if (freq == 0) return 0; - /* Limit supported sample rates to ones that can be autodetected - * by the codec running in slave mode. - */ - for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { - const unsigned int ratio = supported_mclk_lrck_ratios[i]; - - if (freq % ratio == 0) - es8316->allowed_rates[count++] = freq / ratio; - } - - es8316->sysclk_constraints.list = es8316->allowed_rates; - es8316->sysclk_constraints.count = count; - return 0; } @@ -449,13 +434,6 @@ static int es8316_pcm_startup(struct snd_pcm_substream *substream, return -EINVAL; } - /* The set of sample rates that can be supported depends on the - * MCLK supplied to the CODEC. - */ - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &es8316->sysclk_constraints); - return 0; } @@ -466,12 +444,27 @@ static int es8316_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); u8 wordlen = 0; + int i; if (!es8316->sysclk) { dev_err(component->dev, "No MCLK configured\n"); return -EINVAL; } + /* Limit supported sample rates to ones that can be autodetected + * by the codec running in slave mode. + */ + for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { + const unsigned int ratio = supported_mclk_lrck_ratios[i]; + + if (es8316->sysclk % ratio != 0) + continue; + if (es8316->sysclk / ratio == params_rate(params)) + break; + } + if (i == NR_SUPPORTED_MCLK_LRCK_RATIOS) + return -EINVAL; + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: wordlen = ES8316_SERDATA2_LEN_16; -- 2.23.0.rc1

3 12

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix the problem of two front mics on a ThinkCentre
by Hui Wang 04 Sep '19

04 Sep '19

This ThinkCentre machine has a new realtek codec alc222, it is not in the support list, we add it in the realtek.c then this machine can apply FIXUPs for the realtek codec. And this machine has two front mics which can't be handled by PA so far, it uses the pin 0x18 and 0x19 as the front mics, as a result the existing FIXUP ALC294_FIXUP_LENOVO_MIC_LOCATION doesn't work on this machine. Fortunately another FIXUP ALC283_FIXUP_HEADSET_MIC also can change the location for one of the two mics on this machine. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.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 3b9583aa63c1..74697c1dd1d5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7151,6 +7151,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x312a, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), + SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), 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), @@ -9038,6 +9039,7 @@ static int patch_alc680(struct hda_codec *codec) static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0215, "ALC215", patch_alc269), HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0222, "ALC222", patch_alc269), HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269), HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269), HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269), -- 2.17.1

3 3

[alsa-devel] [PATCH] drm: bridge/dw_hdmi: add audio sample channel status setting
by Cheng-Yi Chiang 04 Sep '19

04 Sep '19

From: Yakir Yang <ykk(a)rock-chips.com> When transmitting IEC60985 linear PCM audio, we configure the Audio Sample Channel Status information of all the channel status bits in the IEC60958 frame. Refer to 60958-3 page 10 for frequency, original frequency, and wordlength setting. This fix the issue that audio does not come out on some monitors (e.g. LG 22CV241) Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 59 +++++++++++++++++++++++ drivers/gpu/drm/bridge/synopsys/dw-hdmi.h | 20 ++++++++ 2 files changed, 79 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index bd65d0479683..34d46e25d610 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -582,6 +582,63 @@ static unsigned int hdmi_compute_n(unsigned int freq, unsigned long pixel_clk) return n; } +static void hdmi_set_schnl(struct dw_hdmi *hdmi) +{ + u8 aud_schnl_samplerate; + u8 aud_schnl_8; + + /* These registers are on RK3288 using version 2.0a. */ + if (hdmi->version != 0x200a) + return; + + switch (hdmi->sample_rate) { + case 32000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_32K; + break; + case 44100: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_44K1; + break; + case 48000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_48K; + break; + case 88200: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_88K2; + break; + case 96000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_96K; + break; + case 176400: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_176K4; + break; + case 192000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_192K; + break; + case 768000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_768K; + break; + default: + dev_warn(hdmi->dev, "Unsupported audio sample rate (%u)\n", + hdmi->sample_rate); + return; + } + + /* set channel status register */ + hdmi_modb(hdmi, aud_schnl_samplerate, HDMI_FC_AUDSCHNLS7_SMPRATE_MASK, + HDMI_FC_AUDSCHNLS7); + + /* + * Set original frequency to be the same as frequency. + * Use one-complement value as stated in IEC60958-3 page 13. + */ + aud_schnl_8 = (~aud_schnl_samplerate) << + HDMI_FC_AUDSCHNLS8_ORIGSAMPFREQ_OFFSET; + + /* This means word length is 16 bit. Refer to IEC60958-3 page 12. */ + aud_schnl_8 |= 2 << HDMI_FC_AUDSCHNLS8_WORDLEGNTH_OFFSET; + + hdmi_writeb(hdmi, aud_schnl_8, HDMI_FC_AUDSCHNLS8); +} + static void hdmi_set_clk_regenerator(struct dw_hdmi *hdmi, unsigned long pixel_clk, unsigned int sample_rate) { @@ -620,6 +677,8 @@ static void hdmi_set_clk_regenerator(struct dw_hdmi *hdmi, hdmi->audio_cts = cts; hdmi_set_cts_n(hdmi, cts, hdmi->audio_enable ? n : 0); spin_unlock_irq(&hdmi->audio_lock); + + hdmi_set_schnl(hdmi); } static void hdmi_init_clk_regenerator(struct dw_hdmi *hdmi) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h index 6988f12d89d9..619ebc1c8354 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h @@ -158,6 +158,17 @@ #define HDMI_FC_SPDDEVICEINF 0x1062 #define HDMI_FC_AUDSCONF 0x1063 #define HDMI_FC_AUDSSTAT 0x1064 +#define HDMI_FC_AUDSV 0x1065 +#define HDMI_FC_AUDSU 0x1066 +#define HDMI_FC_AUDSCHNLS0 0x1067 +#define HDMI_FC_AUDSCHNLS1 0x1068 +#define HDMI_FC_AUDSCHNLS2 0x1069 +#define HDMI_FC_AUDSCHNLS3 0x106a +#define HDMI_FC_AUDSCHNLS4 0x106b +#define HDMI_FC_AUDSCHNLS5 0x106c +#define HDMI_FC_AUDSCHNLS6 0x106d +#define HDMI_FC_AUDSCHNLS7 0x106e +#define HDMI_FC_AUDSCHNLS8 0x106f #define HDMI_FC_DATACH0FILL 0x1070 #define HDMI_FC_DATACH1FILL 0x1071 #define HDMI_FC_DATACH2FILL 0x1072 @@ -706,6 +717,15 @@ enum { /* HDMI_FC_AUDSCHNLS7 field values */ HDMI_FC_AUDSCHNLS7_ACCURACY_OFFSET = 4, HDMI_FC_AUDSCHNLS7_ACCURACY_MASK = 0x30, + HDMI_FC_AUDSCHNLS7_SMPRATE_MASK = 0x0f, + HDMI_FC_AUDSCHNLS7_SMPRATE_192K = 0xe, + HDMI_FC_AUDSCHNLS7_SMPRATE_176K4 = 0xc, + HDMI_FC_AUDSCHNLS7_SMPRATE_96K = 0xa, + HDMI_FC_AUDSCHNLS7_SMPRATE_768K = 0x9, + HDMI_FC_AUDSCHNLS7_SMPRATE_88K2 = 0x8, + HDMI_FC_AUDSCHNLS7_SMPRATE_32K = 0x3, + HDMI_FC_AUDSCHNLS7_SMPRATE_48K = 0x2, + HDMI_FC_AUDSCHNLS7_SMPRATE_44K1 = 0x0, /* HDMI_FC_AUDSCHNLS8 field values */ HDMI_FC_AUDSCHNLS8_ORIGSAMPFREQ_MASK = 0xf0, -- 2.23.0.187.g17f5b7556c-goog

5 13

[alsa-devel] [PATCH] soundwire: bus: set initial value to port_status
by Bard liao 04 Sep '19

04 Sep '19

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> port_status[port_num] are assigned for each port_num in some if conditions. So some of the port_status may not be initialized. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index f6a1e4b4813d..33f41b3e642e 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -835,7 +835,7 @@ static int sdw_handle_port_interrupt(struct sdw_slave *slave, static int sdw_handle_slave_alerts(struct sdw_slave *slave) { struct sdw_slave_intr_status slave_intr; - u8 clear = 0, bit, port_status[15]; + u8 clear = 0, bit, port_status[15] = {0}; int port_num, stat, ret, count = 0; unsigned long port; bool slave_notify = false; -- 2.17.1

3 2

[alsa-devel] some lock needed for flush_delayed_work ?
by Kuninori Morimoto 04 Sep '19

04 Sep '19

Hi ALSA ML I got below error when I unbind/bind CPU DAI. It seems we need to have some kind of lock ? I think => message came from linux/kernel/locking/lockdep.c :: assign_lock_key ... [ 6.392640] asoc-audio-graph-card sound: ASoC: CPU DAI not registered => [ 6.399430] INFO: trying to register non-static key. => [ 6.404393] the code is fine but needs lockdep annotation. => [ 6.409875] turning off the locking correctness validator. [ 6.415362] CPU: 2 PID: 193 Comm: test-bind-unbin Not tainted 5.3.0-rc1+ #625 [ 6.422495] Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT) [ 6.430497] Call trace: [ 6.432949] dump_backtrace+0x0/0x140 [ 6.436609] show_stack+0x24/0x30 [ 6.439927] dump_stack+0xc4/0x10c [ 6.443328] register_lock_class+0x4e0/0x618 [ 6.447594] __lock_acquire+0x7c/0x1de8 [ 6.451426] lock_acquire+0xdc/0x1e0 [ 6.455000] del_timer_sync+0x60/0xe8 [ 6.458660] flush_delayed_work+0x2c/0x70 [ 6.462671] soc_cleanup_card_resources+0x50/0x348 [ 6.467459] snd_soc_bind_card+0x1e4/0x990 [ 6.471551] snd_soc_add_component+0x248/0x310 [ 6.475992] snd_soc_register_component+0x54/0x70 [ 6.480696] devm_snd_soc_register_component+0x60/0xb8 [ 6.485833] rsnd_probe+0x164/0x448 [ 6.489321] platform_drv_probe+0x58/0xa8 [ 6.493330] really_probe+0xec/0x2a0 [ 6.496902] driver_probe_device+0x5c/0xf0 [ 6.500996] device_driver_attach+0x74/0x80 [ 6.505177] bind_store+0xdc/0x120 [ 6.508575] drv_attr_store+0x40/0x58 [ 6.512237] sysfs_kf_write+0x50/0x78 [ 6.515896] kernfs_fop_write+0xf0/0x1e8 [ 6.519819] __vfs_write+0x48/0x90 [ 6.523218] vfs_write+0xac/0x1b8 [ 6.526529] ksys_write+0x74/0xf8 [ 6.529840] __arm64_sys_write+0x24/0x30 [ 6.533762] el0_svc_common.constprop.0+0x98/0x170 [ 6.538550] el0_svc_compat_handler+0x2c/0x38 [ 6.542904] el0_svc_compat+0x8/0x10 Thank you for your help !! Best regards --- Kuninori Morimoto

2 2

[alsa-devel] snd_usb_audio - ALSA woke us up to write...was actually nothing to write
by Nathan Royce 03 Sep '19

03 Sep '19

***** ... Sep 03 14:32:58 username kernel: perf: interrupt took too long (4174 > 4156), lowering kernel.perf_event_max_sample_rate to 47700 Sep 03 14:33:03 username plasmashell[1057]: qml: temp unit: 0 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Releasing inhibition with cookie 14 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Releasing inhibition with cookie 15 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Restoring DPMS features after inhibition release Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Can't contact ck Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Releasing inhibition with cookie 16 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Can't contact ck Sep 03 14:33:15 username plasmashell[1057]: qml: temp unit: 0 ... Sep 03 14:33:23 username pulseaudio[1081]: E: [alsa-sink-USB Audio] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Sep 03 14:33:23 username pulseaudio[1081]: E: [alsa-sink-USB Audio] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Sep 03 14:33:23 username pulseaudio[1081]: E: [alsa-sink-USB Audio] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Sep 03 14:33:24 username plasmashell[1057]: qml: temp unit: 0 ... ***** During this time period, I do believe I was starting an android emulator and also watching a twitter video that had audio.

1 0

[alsa-devel] [PATCH] ASoC: soc-core: add comment to jack at soc_remove_component()
by Kuninori Morimoto 03 Sep '19

03 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Basically, driver which setups snd_soc_component_set_jack() need to release it by themselves. But, as framework level robustness, soc_remove_component() also releases it. To avoid code reader confuse, this patch makes it clarify. This patch makes it clarify. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 659b90a..60f7e8b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1171,7 +1171,9 @@ static void soc_remove_component(struct snd_soc_component *component) if (component->probed) snd_soc_component_remove(component); + /* For framework level robustness */ snd_soc_component_set_jack(component, NULL, NULL); + list_del(&component->card_list); snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); -- 2.7.4

2 1

[alsa-devel] sound/usb kernel modules as a DKMS package.
by Hin-Tak Leung 03 Sep '19

03 Sep '19

Hi all, I (We... this is kind of my 9-5 job) have been having some reliability issue with a Focusrite Scarlett Gen 2 18i20, and I was almost tempted to start looking at the windows driver, when I came upon Geoffrey's work. So I made the bleeding-edge sound/usb modules (for the 5.4 time frame) from Takashi's tree built as dkms modules against main line 5.2 . I hope this is useful for other people using USB sound devices. https://github.com/HinTak/sound-usb-dkms This make it easier / faster to try out new code, without building the whole kernel. I would appreciate tips and tricks about the Scarlett 18i20, as well as general recommendation for new professional gear of the same sort to buy. Incidentally, Tobias Hoffmann (who wrote the first gen driver about 5 years ago) was my GSoC student working on printer-related matters more than a decade ago; I did some wifi/USB related work in the kernel (am listed in linux/MAINTAINERS) so you can assume that I am relatively happy with driver-related matters. Cheers, Hin-Tak

2 3

[alsa-devel] [PATCH 1/2] conf/ucm: Add UCM profile for cht-bsw-rt5672 boards
by Hans de Goede 03 Sep '19

03 Sep '19

Add an UCM profile for Intel boards with a RT5672 codec. Re-use the existing platform enable and disable sequences for BYT/CHT SST support and add a codecs/rt5672 dir with codec specific enable / disable sequences for the various inputs and outputs. This is partly based on earlier work done here: https://github.com/plbossart/UCM/tree/master/cht-bsw-rt5672 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- configure.ac | 2 + src/conf/ucm/Makefile.am | 1 + src/conf/ucm/cht-bsw-rt5672/HiFi.conf | 27 +++++++++++ src/conf/ucm/cht-bsw-rt5672/Makefile.am | 4 ++ .../ucm/cht-bsw-rt5672/cht-bsw-rt5672.conf | 6 +++ src/conf/ucm/codecs/Makefile.am | 1 + src/conf/ucm/codecs/rt5672/DMIC1.conf | 28 +++++++++++ src/conf/ucm/codecs/rt5672/DMIC2.conf | 28 +++++++++++ src/conf/ucm/codecs/rt5672/EnableSeq.conf | 48 +++++++++++++++++++ src/conf/ucm/codecs/rt5672/HeadPhones.conf | 23 +++++++++ src/conf/ucm/codecs/rt5672/HeadsetMic.conf | 38 +++++++++++++++ src/conf/ucm/codecs/rt5672/Makefile.am | 5 ++ src/conf/ucm/codecs/rt5672/MonoSpeaker.conf | 26 ++++++++++ src/conf/ucm/codecs/rt5672/Speaker.conf | 22 +++++++++ 14 files changed, 259 insertions(+) create mode 100644 src/conf/ucm/cht-bsw-rt5672/HiFi.conf create mode 100644 src/conf/ucm/cht-bsw-rt5672/Makefile.am create mode 100644 src/conf/ucm/cht-bsw-rt5672/cht-bsw-rt5672.conf create mode 100644 src/conf/ucm/codecs/rt5672/DMIC1.conf create mode 100644 src/conf/ucm/codecs/rt5672/DMIC2.conf create mode 100644 src/conf/ucm/codecs/rt5672/EnableSeq.conf create mode 100644 src/conf/ucm/codecs/rt5672/HeadPhones.conf create mode 100644 src/conf/ucm/codecs/rt5672/HeadsetMic.conf create mode 100644 src/conf/ucm/codecs/rt5672/Makefile.am create mode 100644 src/conf/ucm/codecs/rt5672/MonoSpeaker.conf create mode 100644 src/conf/ucm/codecs/rt5672/Speaker.conf diff --git a/configure.ac b/configure.ac index 2e955760..9ccca7aa 100644 --- a/configure.ac +++ b/configure.ac @@ -741,6 +741,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/Makefile \ src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/Makefile \ src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/Makefile \ + src/conf/ucm/cht-bsw-rt5672/Makefile \ src/conf/ucm/chtnau8824/Makefile \ src/conf/ucm/chtrt5645/Makefile \ src/conf/ucm/chtrt5645-mono-speaker-analog-mic/Makefile \ @@ -772,6 +773,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/codecs/rt5640/Makefile \ src/conf/ucm/codecs/rt5645/Makefile \ src/conf/ucm/codecs/rt5651/Makefile \ + src/conf/ucm/codecs/rt5672/Makefile \ src/conf/ucm/platforms/Makefile \ src/conf/ucm/platforms/bytcr/Makefile \ src/conf/topology/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 02257048..2ed4e1a3 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -25,6 +25,7 @@ bytcr-rt5651-stereo-spk-dmic-mic \ bytcr-rt5651-stereo-spk-in1-mic \ bytcr-rt5651-stereo-spk-in2-mic \ bytcr-rt5651-stereo-spk-in12-mic \ +cht-bsw-rt5672 \ chtnau8824 \ chtrt5645 \ chtrt5645-mono-speaker-analog-mic \ diff --git a/src/conf/ucm/cht-bsw-rt5672/HiFi.conf b/src/conf/ucm/cht-bsw-rt5672/HiFi.conf new file mode 100644 index 00000000..b83099d8 --- /dev/null +++ b/src/conf/ucm/cht-bsw-rt5672/HiFi.conf @@ -0,0 +1,27 @@ +# Adapted from https://github.com/plbossart/UCM/tree/master/cht-bsw-rt5672 + +SectionVerb { + EnableSequence [ + cdev "hw:chtbswrt5672" + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5672/EnableSeq.conf> + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + <platforms/bytcr/PlatformDisableSeq.conf> + ] + + Value { + PlaybackPCM "hw:chtbswrt5672" + CapturePCM "hw:chtbswrt5672" + } +} + +<codecs/rt5672/Speaker.conf> +<codecs/rt5672/MonoSpeaker.conf> +<codecs/rt5672/HeadPhones.conf> + +<codecs/rt5672/DMIC1.conf> +<codecs/rt5672/DMIC2.conf> +<codecs/rt5672/HeadsetMic.conf> diff --git a/src/conf/ucm/cht-bsw-rt5672/Makefile.am b/src/conf/ucm/cht-bsw-rt5672/Makefile.am new file mode 100644 index 00000000..c3d557fa --- /dev/null +++ b/src/conf/ucm/cht-bsw-rt5672/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/cht-bsw-rt5672 +ucm_DATA = cht-bsw-rt5672.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/cht-bsw-rt5672/cht-bsw-rt5672.conf b/src/conf/ucm/cht-bsw-rt5672/cht-bsw-rt5672.conf new file mode 100644 index 00000000..c79024d9 --- /dev/null +++ b/src/conf/ucm/cht-bsw-rt5672/cht-bsw-rt5672.conf @@ -0,0 +1,6 @@ +# Adapted from https://github.com/plbossart/UCM/tree/master/cht-bsw-rt5672 + +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Play HiFi quality Music" +} diff --git a/src/conf/ucm/codecs/Makefile.am b/src/conf/ucm/codecs/Makefile.am index 5987b9cf..17c4ebf9 100644 --- a/src/conf/ucm/codecs/Makefile.am +++ b/src/conf/ucm/codecs/Makefile.am @@ -4,4 +4,5 @@ es8316 \ rt5640 \ rt5645 \ rt5651 \ +rt5672 \ nau8824 diff --git a/src/conf/ucm/codecs/rt5672/DMIC1.conf b/src/conf/ucm/codecs/rt5672/DMIC1.conf new file mode 100644 index 00000000..9a8db1a5 --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/DMIC1.conf @@ -0,0 +1,28 @@ +SectionDevice."DigitalMic-DMIC1" { + Comment "Internal Digital Microphone on DMIC1" + + ConflictingDevice [ + "DigitalMic-DMIC2" + "HeadsetMic" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Int Mic Switch' on" + cset "name='Stereo1 DMIC Mux' DMIC1" + cset "name='Stereo1 ADC 2 Mux' DMIC" + cset "name='Sto1 ADC MIXL ADC2 Switch' on" + cset "name='Sto1 ADC MIXR ADC2 Switch' on" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Int Mic Switch' off" + cset "name='Sto1 ADC MIXL ADC2 Switch' off" + cset "name='Sto1 ADC MIXR ADC2 Switch' off" + ] + + Value { + CaptureChannels 2 + } +} diff --git a/src/conf/ucm/codecs/rt5672/DMIC2.conf b/src/conf/ucm/codecs/rt5672/DMIC2.conf new file mode 100644 index 00000000..b960436e --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/DMIC2.conf @@ -0,0 +1,28 @@ +SectionDevice."DigitalMic-DMIC2" { + Comment "Internal Digital Microphone on DMIC2" + + ConflictingDevice [ + "DigitalMic-DMIC1" + "HeadsetMic" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Int Mic Switch' on" + cset "name='Stereo1 DMIC Mux' DMIC2" + cset "name='Stereo1 ADC 2 Mux' DMIC" + cset "name='Sto1 ADC MIXL ADC2 Switch' on" + cset "name='Sto1 ADC MIXR ADC2 Switch' on" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Int Mic Switch' off" + cset "name='Sto1 ADC MIXL ADC2 Switch' off" + cset "name='Sto1 ADC MIXR ADC2 Switch' off" + ] + + Value { + CaptureChannels 2 + } +} diff --git a/src/conf/ucm/codecs/rt5672/EnableSeq.conf b/src/conf/ucm/codecs/rt5672/EnableSeq.conf new file mode 100644 index 00000000..bd4f357c --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/EnableSeq.conf @@ -0,0 +1,48 @@ +# Playback (I2S1) + +# common +cset "name='DAC1 L Mux' IF1 DAC" +cset "name='DAC1 R Mux' IF1 DAC" +cset "name='DAC1 MIXL DAC1 Switch' on" +cset "name='DAC1 MIXR DAC1 Switch' on" +cset "name='Stereo DAC MIXL DAC L1 Switch' on" +cset "name='Stereo DAC MIXL DAC R1 Switch' off" +cset "name='Stereo DAC MIXR DAC R1 Switch' on" +cset "name='Stereo DAC MIXR DAC L1 Switch' off" + +# speaker +cset "name='PDM1 L Mux' Stereo DAC" +cset "name='PDM1 R Mux' Stereo DAC" + +# headphone +cset "name='HPOVOL MIXL DAC1 Switch' on" +cset "name='HPOVOL MIXR DAC1 Switch' on" +cset "name='HPO MIX HPVOL Switch' on" + +# line out +# cset "name='OUT MIXL DAC L1 Switch' on" +# cset "name='OUT MIXR DAC R1 Switch' on" +# cset "name='LOUT MIX OUTMIX L Switch' on" +# cset "name='LOUT MIX OUTMIX R Switch' on" + +# 47=0dB, 0.375 dB/step +cset "name='ADC Capture Volume' 47" +cset "name='ADC Capture Switch' on" + +# Headset mic is quite soft, boost it a bit, 1 = 20dB which is the first +# available boost step +cset "name='IN1 Boost Volume' 1" + +# Start with all outputs / inputs disabled +cset "name='Ext Spk Switch' off" +cset "name='Headphone Switch' off" +cset "name='Int Mic Switch' off" +cset "name='Headset Mic Switch' off" +cset "name='Sto1 ADC MIXL ADC1 Switch' off" +cset "name='Sto1 ADC MIXR ADC1 Switch' off" +cset "name='Sto1 ADC MIXL ADC2 Switch' off" +cset "name='Sto1 ADC MIXR ADC2 Switch' off" +cset "name='RECMIXL BST1 Switch' off" +cset "name='RECMIXR BST1 Switch' off" +cset "name='RECMIXL BST2 Switch' off" +cset "name='RECMIXR BST2 Switch' off" diff --git a/src/conf/ucm/codecs/rt5672/HeadPhones.conf b/src/conf/ucm/codecs/rt5672/HeadPhones.conf new file mode 100644 index 00000000..cb8aad2d --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/HeadPhones.conf @@ -0,0 +1,23 @@ +SectionDevice."Headphones" { + Comment "Headphones" + + ConflictingDevice [ + "Speaker" + "MonoSpeaker" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Headphone Switch' on" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Headphone Switch' off" + ] + + Value { + PlaybackChannels "2" + JackControl "Headphone Jack" + } +} diff --git a/src/conf/ucm/codecs/rt5672/HeadsetMic.conf b/src/conf/ucm/codecs/rt5672/HeadsetMic.conf new file mode 100644 index 00000000..cc6d8d56 --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/HeadsetMic.conf @@ -0,0 +1,38 @@ +SectionDevice."HeadsetMic" { + Comment "Headset Microphone" + + ConflictingDevice [ + "DigitalMic-DMIC1" + "DigitalMic-DMIC2" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Headset Mic Switch' on" + cset "name='RECMIXL BST1 Switch' on" + cset "name='RECMIXR BST1 Switch' on" + cset "name='Sto1 ADC MIXL ADC1 Switch' on" + cset "name='Sto1 ADC MIXR ADC1 Switch' on" + # For unknown reasons the headset mic sound is only recorded + # on the left channel, we tweak the data-stream reception to + # use the left samples for both slots, so that we get a stereo + # stream with the mic sound on both channels + cset "name='codec_in rx deinterleaver codec_in0_1' 1" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Headset Mic Switch' off" + cset "name='RECMIXL BST1 Switch' off" + cset "name='RECMIXR BST1 Switch' off" + cset "name='Sto1 ADC MIXL ADC1 Switch' off" + cset "name='Sto1 ADC MIXR ADC1 Switch' off" + # Undo data-stream reception mono-mix workaround + cset "name='codec_in rx deinterleaver codec_in0_1' 2" + ] + + Value { + CaptureChannels "2" + JackControl "Headset Mic Jack" + } +} diff --git a/src/conf/ucm/codecs/rt5672/Makefile.am b/src/conf/ucm/codecs/rt5672/Makefile.am new file mode 100644 index 00000000..feba0b76 --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/Makefile.am @@ -0,0 +1,5 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/codecs/rt5672 +ucm_DATA = DMIC1.conf DMIC2.conf EnableSeq.conf HeadPhones.conf \ + HeadsetMic.conf MonoSpeaker.conf Speaker.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/codecs/rt5672/MonoSpeaker.conf b/src/conf/ucm/codecs/rt5672/MonoSpeaker.conf new file mode 100644 index 00000000..828a1154 --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/MonoSpeaker.conf @@ -0,0 +1,26 @@ +SectionDevice."MonoSpeaker" { + Comment "Mono Speaker" + + ConflictingDevice [ + "Speaker" + "Headphones" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Stereo DAC MIXR DAC R1 Switch' off" + cset "name='Stereo DAC MIXL DAC R1 Switch' on" + cset "name='Ext Spk Switch' on" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Ext Spk Switch' off" + cset "name='Stereo DAC MIXL DAC R1 Switch' off" + cset "name='Stereo DAC MIXR DAC R1 Switch' on" + ] + + Value { + PlaybackChannels "2" + } +} diff --git a/src/conf/ucm/codecs/rt5672/Speaker.conf b/src/conf/ucm/codecs/rt5672/Speaker.conf new file mode 100644 index 00000000..f9291e53 --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/Speaker.conf @@ -0,0 +1,22 @@ +SectionDevice."Speaker" { + Comment "Speakers" + + ConflictingDevice [ + "MonoSpeaker" + "Headphones" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Ext Spk Switch' on" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Ext Spk Switch' off" + ] + + Value { + PlaybackChannels "2" + } +} -- 2.23.0

4 9

[alsa-devel] [RFC PATCH 0/7] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 03 Sep '19

03 Sep '19

Hi all, here's a RFC patch series that adapts SOF (and one example machine driver) to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. Main points I'd like your input on: 1) Is the high-level approach ok? - SOF already uses pci/hda/ for all others codecs, so HDMI has been the sole exception where code has been duplicated. - I've tried to keep changes to hda/hdmi minimal. - This series implements logic to parse the PCM topology in a dynamic fashion, so we do not have to change all existing (and future) DSP topologis to use fixed PCM device numbers for HDMI, and we avoid need to hardcode PCM device numbers in machine driver code. 2) Can we drop hdac_hdmi and its support from machine drivers, or do we need to make it optional and keep it around? - Current series does not add any Kconfig options, but simply switches SOF to use HDA codecs for all, including HDMI/DP. This means hdac_hdmi is never used with SOF and could be dropped (if SST is ok as well). - This may break some usage with SST (input is welcome!) - The change is visible to applications. The ALSA mixer interface is different (OTOH with the new driver, playback works out-of-the-box while with hdac_hdmi you needed to set the multiple controls first, to have any audio out). - Alternatively I could add a KConfig option and we could have a deprecation period for hdac_hdmi, allowing people to compile with the old driver during transition time. This will require #ifdef'ing in all the machine drivers. .. once these are addressed, I can proceed to extend the patchset with all affected machine drivers. Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) - Pre-reviewed at: https://github.com/thesofproject/linux/pull/1155 Kai Vehmanen (7): ALSA: hda - add mst_no_extra_pcms flag to hda_codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ALSA: hda/hdmi - allow control creation without a linked pcm ALSA: hda/hdmi - implement mst_no_extra_pcms flag ALSA: hda/hdmi - complete pcm_setup_pin without snd_pcm link ASoC: SOF: Intel: load hda codec module also for HDMI/DP include/sound/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 31 ++++--- sound/soc/codecs/hdac_hda.c | 95 +++++++++++++++++--- sound/soc/codecs/hdac_hda.h | 10 ++- sound/soc/intel/boards/skl_hda_dsp_common.c | 27 ++---- sound/soc/intel/boards/skl_hda_dsp_common.h | 61 +++++++++++++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 7 -- sound/soc/sof/intel/hda-codec.c | 11 +-- sound/soc/sof/intel/hda.h | 5 +- 9 files changed, 189 insertions(+), 59 deletions(-) -- 2.17.1

2 14

[alsa-devel] [PATCH v2 1/3] ASoC: es8316: judge PCM rate at later timing
by Katsuhiro Suzuki 03 Sep '19

03 Sep '19

This patch change the judge timing about playing/capturing PCM rate. Original code set constraints list of PCM rate limits at set_sysclk. This strategy works well if system is using fixed rate clock. But some boards and SoC (such as RockPro64 and RockChip I2S) has connected SoC MCLK out to ES8316 MCLK in. In this case, SoC side I2S will choose suitable frequency of MCLK such as fs * mclk-fs when user starts playing or capturing. Bad scenario as follows (mclk-fs = 256): - Initialize sysclk by correct value (Ex. 12.288MHz) - ES8316 set constraints of PCM rate by sysclk 48kHz (1/256), 32kHz (1/384), 30.720kHz (1/400), 24kHz (1/512), 16kHz (1/768), 12kHz (1/1024) - Play 48kHz sound, it's acceptable - Sysclk is not changed - Play 32kHz sound, it's acceptable - Set sysclk by 8.192MHz (= fs * mclk-fs = 32k * 256) - ES8316 set constraints of PCM rate by sysclk 32kHz (1/256), 21.33kHz (1/384), 20.48kHz (1/400), 16kHz (1/512), 10.66kHz (1/768), 8kHz (1/1024) - Play 48kHz again, but it's NOT acceptable because constraints list does not allow 48kHz Root cause of this strange behavior is changing constraints list at set_sysclk timing. It seems that is too early to determine. So this patch does not use constraints list and check PCM rate limit more later timing at hw_params. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/codecs/es8316.c | 37 +++++++++++++++---------------------- 1 file changed, 15 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index ed2959dbe1fb..229808fa627c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -363,27 +363,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); - int i; - int count = 0; es8316->sysclk = freq; if (freq == 0) return 0; - /* Limit supported sample rates to ones that can be autodetected - * by the codec running in slave mode. - */ - for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { - const unsigned int ratio = supported_mclk_lrck_ratios[i]; - - if (freq % ratio == 0) - es8316->allowed_rates[count++] = freq / ratio; - } - - es8316->sysclk_constraints.list = es8316->allowed_rates; - es8316->sysclk_constraints.count = count; - return 0; } @@ -449,13 +434,6 @@ static int es8316_pcm_startup(struct snd_pcm_substream *substream, return -EINVAL; } - /* The set of sample rates that can be supported depends on the - * MCLK supplied to the CODEC. - */ - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &es8316->sysclk_constraints); - return 0; } @@ -466,12 +444,27 @@ static int es8316_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); u8 wordlen = 0; + int i; if (!es8316->sysclk) { dev_err(component->dev, "No MCLK configured\n"); return -EINVAL; } + /* Limit supported sample rates to ones that can be autodetected + * by the codec running in slave mode. + */ + for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { + const unsigned int ratio = supported_mclk_lrck_ratios[i]; + + if (es8316->sysclk % ratio != 0) + continue; + if (es8316->sysclk / ratio == params_rate(params)) + break; + } + if (i == NR_SUPPORTED_MCLK_LRCK_RATIOS) + return -EINVAL; + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: wordlen = ES8316_SERDATA2_LEN_16; -- 2.23.0.rc1

2 6

[alsa-devel] [PATCH] MAINTAINERS: update entry for firewire audio drivers with UAPI header
by Takashi Sakamoto 03 Sep '19

03 Sep '19

The most of drivers in ALSA firewire stack supports common ioctl commands to enable/disable packet streaming as well as some ioctl commands for model-specific features. An UAPI header is exported to userspace. This commit adds supplement for entry of ALSA firewire stack with a path of the UAPI header. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 4200194e69ea..6bd54ce6dc66 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6271,6 +6271,7 @@ L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained F: sound/firewire/ +F: include/uapi/sound/firewire.h FIREWIRE MEDIA DRIVERS (firedtv) M: Stefan Richter <stefanr(a)s5r6.in-berlin.de> -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: simple-card: indicate rebind issue
by Kuninori Morimoto 03 Sep '19

03 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ALSA SoC try to rebind Sound Card if Card/CPU/Codec/Platform were unbinded and re-binded again. But, Simple Card might can't rebind again if user do for example unbind CPU or Codec driver bind CPU or Codec driver Because Simple Card is still pointing old/unbinded CPU or Codec driver's DAI name at dlc->dai_name. To avoid this issue, it needs to alloc memory and keep DAI name even though if CPU or Codec driver was unbinded. Or, always do unbind/bind at Sound Card. For now, this patch indicates this issue as FIXME. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 20 +++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 0eac3bc..fc9c753 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -46,7 +46,25 @@ static int asoc_simple_parse_dai(struct device_node *node, if (ret) return ret; - /* Get dai->name */ + /* + * FIXME + * + * Here, dlc->dai_name is pointer to CPU/Codec DAI name. + * If user unbinded CPU or Codec driver, but not for Sound Card, + * dlc->dai_name is keeping unbinded CPU or Codec + * driver's pointer. + * + * If user re-bind CPU or Codec driver again, ALSA SoC will try + * to rebind Card via snd_soc_try_rebind_card(), but because of + * above reason, it might can't bind Sound Card. + * Because Sound Card is pointing to released dai_name pointer. + * + * To avoid this rebind Card issue, + * 1) It needs to alloc memory to keep dai_name eventhough + * CPU or Codec driver was unbinded, or + * 2) user need to rebind Sound Card everytime + * if he unbinded CPU or Codec. + */ ret = snd_soc_of_get_dai_name(node, &dlc->dai_name); if (ret < 0) return ret; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: audio-graph: indicate rebind issue
by Kuninori Morimoto 03 Sep '19

03 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ALSA SoC try to rebind Sound Card if Card/CPU/Codec/Platform were unbinded and re-binded again. But, Audio Graph Card might can't rebind again if user do for example unbind CPU or Codec driver bind CPU or Codec driver Because Audio Graph Card is still pointing old/unbinded CPU or Codec driver's DAI name at dlc->dai_name. To avoid this issue, it needs to alloc memory and keep DAI name even though if CPU or Codec driver was unbinded. Or, always do unbind/bind at Sound Card. For now, this patch indicates this issue as FIXME. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 288df24..6007e63 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -129,6 +129,25 @@ static int asoc_simple_parse_dai(struct device_node *ep, args.args[0] = graph_get_dai_id(ep); args.args_count = (of_graph_get_endpoint_count(node) > 1); + /* + * FIXME + * + * Here, dlc->dai_name is pointer to CPU/Codec DAI name. + * If user unbinded CPU or Codec driver, but not for Sound Card, + * dlc->dai_name is keeping unbinded CPU or Codec + * driver's pointer. + * + * If user re-bind CPU or Codec driver again, ALSA SoC will try + * to rebind Card via snd_soc_try_rebind_card(), but because of + * above reason, it might can't bind Sound Card. + * Because Sound Card is pointing to released dai_name pointer. + * + * To avoid this rebind Card issue, + * 1) It needs to alloc memory to keep dai_name eventhough + * CPU or Codec driver was unbinded, or + * 2) user need to rebind Sound Card everytime + * if he unbinded CPU or Codec. + */ ret = snd_soc_get_dai_name(&args, &dlc->dai_name); if (ret < 0) return ret; -- 2.7.4

2 1

[alsa-devel] [PATCH] alsamixer: Remove exp10 usage
by Rosen Penev 03 Sep '19

03 Sep '19

exp10 is a GNU extension and not available everywhere (eg. uClibc-ng). Signed-off-by: Rosen Penev <rosenp(a)gmail.com> --- alsamixer/volume_mapping.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/alsamixer/volume_mapping.c b/alsamixer/volume_mapping.c index 48cfbe2..29ab061 100644 --- a/alsamixer/volume_mapping.c +++ b/alsamixer/volume_mapping.c @@ -108,9 +108,9 @@ static double get_normalized_volume(snd_mixer_elem_t *elem, if (use_linear_dB_scale(min, max)) return (value - min) / (double)(max - min); - normalized = exp10((value - max) / 6000.0); + normalized = pow(10, (value - max) / 6000.0); if (min != SND_CTL_TLV_DB_GAIN_MUTE) { - min_norm = exp10((min - max) / 6000.0); + min_norm = pow(10, (min - max) / 6000.0); normalized = (normalized - min_norm) / (1 - min_norm); } @@ -143,7 +143,7 @@ static int set_normalized_volume(snd_mixer_elem_t *elem, } if (min != SND_CTL_TLV_DB_GAIN_MUTE) { - min_norm = exp10((min - max) / 6000.0); + min_norm = pow(10, (min - max) / 6000.0); volume = volume * (1 - min_norm) + min_norm; } value = lrint_dir(6000.0 * log10(volume), dir) + max; -- 2.17.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: xlnx: add Xilinx logicPD-I2S FPGA IP support
by Miquel Raynal 03 Sep '19

03 Sep '19

This IP is very simple so this driver manage both the DAI and the PCM streams, hence the presence of both components in this driver. There are plenty available interruptions when capturing or playing back audio that can be triggered but the only one that fits the ALSA sound system is the XFER_DONE which is used to bound sound periods. Other interrupts are masked. Please note that capture and playback are not possible at the same time though. Capture seems to work (at least it creates a file with something inside) but I have no capture mechanism on the board to actually test that it works correctly. Signed-off-by: Miquel Raynal <miquel.raynal(a)bootlin.com> --- Hello, This is my first contribution in the sound subsystem, I hope I've understood the core but I might be entirely wrong as well, so please do not hesitate to be critical on my choices. Thanks, Miquèl sound/soc/xilinx/Kconfig | 7 + sound/soc/xilinx/Makefile | 2 + sound/soc/xilinx/xlnx-logicpd-i2s.c | 468 ++++++++++++++++++++++++++++ 3 files changed, 477 insertions(+) create mode 100644 sound/soc/xilinx/xlnx-logicpd-i2s.c diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 47f606b924e4..b62cae6750b9 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -7,6 +7,13 @@ config SND_SOC_XILINX_I2S PCM data. In receiver mode, IP receives PCM audio and encapsulates PCM in AES format and sends AES data. +config SND_SOC_XILINX_LOGICPD_I2S + tristate "Audio support for the Xilinx logicPD I2S" + help + Select this option to enable Xilinx logicPD I2S slave + transceiver. This enables I2S playback and capture using + Xilinx/logicPD IP. + config SND_SOC_XILINX_AUDIO_FORMATTER tristate "Audio support for the the Xilinx audio formatter" help diff --git a/sound/soc/xilinx/Makefile b/sound/soc/xilinx/Makefile index d79fd38b094b..d127c30f8fe2 100644 --- a/sound/soc/xilinx/Makefile +++ b/sound/soc/xilinx/Makefile @@ -1,5 +1,7 @@ snd-soc-xlnx-i2s-objs := xlnx_i2s.o obj-$(CONFIG_SND_SOC_XILINX_I2S) += snd-soc-xlnx-i2s.o +snd-soc-xlnx-logicpd-i2s-objs := xlnx-logicpd-i2s.o +obj-$(CONFIG_SND_SOC_XILINX_LOGICPD_I2S) += snd-soc-xlnx-logicpd-i2s.o snd-soc-xlnx-formatter-pcm-objs := xlnx_formatter_pcm.o obj-$(CONFIG_SND_SOC_XILINX_AUDIO_FORMATTER) += snd-soc-xlnx-formatter-pcm.o snd-soc-xlnx-spdif-objs := xlnx_spdif.o diff --git a/sound/soc/xilinx/xlnx-logicpd-i2s.c b/sound/soc/xilinx/xlnx-logicpd-i2s.c new file mode 100644 index 000000000000..325a5bb6978a --- /dev/null +++ b/sound/soc/xilinx/xlnx-logicpd-i2s.c @@ -0,0 +1,468 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Xilinx logicPD logiI2S - I2S slave transceiver v2 support + * + * Copyright (C) 2019 Bootlin + * + * Author: Miquel Raynal <miquel.raynal(a)bootlin.com> + */ + +#include <linux/dma-mapping.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/platform_device.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define DRV_NAME "xlnx_logicpd_i2s" + +#define IP_VERSION 0x0 +#define PATCH_LEVEL(reg) (((reg) & GENMASK(4, 0)) + 'a') +#define MINOR_REV(reg) (((reg) & GENMASK(10, 5)) >> 5) +#define MAJOR_REV(reg) (((reg) & GENMASK(16, 11)) >> 11) +#define LICENSE_TYPE(reg) (((reg) & GENMASK(18, 17)) >> 17) +#define CONTROL_REG(s) ((s) == SNDRV_PCM_STREAM_PLAYBACK ? 0x4 : 0x24) +#define ENGINE_EN BIT(0) +#define XFER_DONE BIT(1) +#define BUFF_BASE_ADDR_REG(s) ((s) == SNDRV_PCM_STREAM_PLAYBACK ? 0x8 : 0x28) +#define BUFF_LEN_REG(s) ((s) == SNDRV_PCM_STREAM_PLAYBACK ? 0xC : 0x2C) +#define FIFO_STAT_REG(s) ((s) == SNDRV_PCM_STREAM_PLAYBACK ? 0x10 : 0x30) +#define INTR_MASK_REG(s) ((s) == SNDRV_PCM_STREAM_PLAYBACK ? 0x14 : 0x34) +#define XFER_DONE_INTR BIT(31) +#define INTR_STAT_REG(s) ((s) == SNDRV_PCM_STREAM_PLAYBACK ? 0x18 : 0x38) +#define FIFO_COUNT(reg) ((reg) >> 20) + +#define BYTES_TO_WORDS(n) ((n) / 4) + +/* Arbitrarily chosen period size */ +#define PCM_PERIOD_WORDS SZ_8K +#define PCM_PERIOD_BYTES (PCM_PERIOD_WORDS * 4) +/* This is the actual maximum size that can actually be moved in one chunk */ +#define PCM_BUF_WORDS (SZ_64K - 1) +#define PCM_BUF_BYTES (PCM_BUF_WORDS * 4) + +struct xlnx_logicpd_i2s; + +/** + * struct xlnx_logicpd_stream - Internal stream representation + * + * @i2s: Chip data + * @substream: Core substream structure + * @period_idx: Index of the period within the circular buffer + */ +struct xlnx_logicpd_stream { + struct xlnx_logicpd_i2s *i2s; + struct snd_pcm_substream *substream; + unsigned int period_idx; +}; + +/** + * struct xlnx_logicpd_i2s - Chip structure + * + * @base: Registers base address + * @streams: Playback and capture streams in an array + */ +struct xlnx_logicpd_i2s { + void __iomem *base; + struct xlnx_logicpd_stream streams[2]; +}; + +static struct xlnx_logicpd_i2s *substream_to_cpu_dai_chip(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + + return snd_soc_dai_get_drvdata(rtd->cpu_dai); +} + +/* PCM methods */ + +static const struct snd_pcm_hardware xlnx_logicpd_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_HALF_DUPLEX, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 2, + .period_bytes_min = 0, + .period_bytes_max = PCM_PERIOD_BYTES, + .periods_min = 0, + .periods_max = -1, + .buffer_bytes_max = PCM_BUF_BYTES, +}; + +static int xlnx_logicpd_pcm_open(struct snd_pcm_substream *substream) +{ + struct xlnx_logicpd_i2s *i2s = substream_to_cpu_dai_chip(substream); + unsigned int dir = substream->stream; + + snd_soc_set_runtime_hwparams(substream, &xlnx_logicpd_pcm_hardware); + + i2s->streams[dir].substream = substream; + + return 0; +} + +static int xlnx_logicpd_pcm_close(struct snd_pcm_substream *substream) +{ + struct xlnx_logicpd_i2s *i2s = substream_to_cpu_dai_chip(substream); + unsigned int dir = substream->stream; + + i2s->streams[dir].substream = NULL; + + return 0; +} + +static int xlnx_logicpd_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + return 0; +} + +static snd_pcm_uframes_t xlnx_logicpd_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct xlnx_logicpd_i2s *i2s = substream_to_cpu_dai_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int period_sz = snd_pcm_lib_period_bytes(substream); + unsigned int dir = substream->stream; + + return bytes_to_frames(runtime, + i2s->streams[dir].period_idx * period_sz); +} + +static int xlnx_logicpd_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +static const struct snd_pcm_ops xlnx_logicpd_pcm_ops = { + .open = xlnx_logicpd_pcm_open, + .close = xlnx_logicpd_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = xlnx_logicpd_pcm_hw_params, + .pointer = xlnx_logicpd_pcm_pointer, + .mmap = xlnx_logicpd_pcm_mmap, +}; + +static int xlnx_logicpd_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int dir; + + for (dir = SNDRV_PCM_STREAM_PLAYBACK; + dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + substream = pcm->streams[dir].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + buf->area = dma_alloc_coherent(pcm->card->dev, PCM_BUF_BYTES, + &buf->addr, GFP_KERNEL); + buf->bytes = PCM_BUF_BYTES; + if (!buf->area) + return -ENOMEM; + } + + return 0; +} + +static void xlnx_logicpd_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int dir; + + for (dir = SNDRV_PCM_STREAM_PLAYBACK; + dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + substream = pcm->streams[dir].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static const struct snd_soc_component_driver xlnx_logicpd_pcm_component = { + .name = "xlnx-logicp-pcm", + .ops = &xlnx_logicpd_pcm_ops, + .pcm_new = xlnx_logicpd_pcm_new, + .pcm_free = xlnx_logicpd_pcm_free, +}; + +/* DAI methods */ + +static void xlnx_logicpd_dai_int_en(struct xlnx_logicpd_i2s *i2s, int dir) +{ + u32 reg; + + reg = readl_relaxed(i2s->base + INTR_MASK_REG(dir)); + reg &= ~XFER_DONE_INTR; + writel(reg, i2s->base + INTR_MASK_REG(dir)); +} + +static void xlnx_logicpd_dai_int_dis(struct xlnx_logicpd_i2s *i2s, int dir) +{ + u32 reg; + + reg = readl_relaxed(i2s->base + INTR_MASK_REG(dir)); + reg |= XFER_DONE_INTR; + writel_relaxed(reg, i2s->base + INTR_MASK_REG(dir)); +} + +static irqreturn_t xlnx_logicpd_dai_isr(int irq, void *dev_id) +{ + struct xlnx_logicpd_stream *stream = dev_id; + struct xlnx_logicpd_i2s *i2s = stream->i2s; + struct snd_pcm_substream *substream = stream->substream; + unsigned int period_sz = snd_pcm_lib_period_bytes(substream); + unsigned int buf_sz = snd_pcm_lib_buffer_bytes(substream); + dma_addr_t buf_addr = substream->dma_buffer.addr; + unsigned int dir = substream->stream; + u32 reg; + + /* Reading INTR_STAT deasserts the host interrupt */ + reg = readl_relaxed(i2s->base + INTR_STAT_REG(dir)); + + /* + * When the XFER_DONE interrupt is triggered, it means the period has + * been entirely shifted into the FIFO. At this point, we can move the + * buffer pointer to the next period and ask to transfer another chunk + * of data. Whenever the FIFO will be at its "almost full" state (4096 + * words minus the threshold of 100 words) the internal DMA engine will + * automatically restart shifting data to the FIFO until its full state. + * Hence, the host has up to 3996 words (in our case, 3996 frames) to + * serve the interrupt before an underrun that would happen, at eg. + * 44100Hz, after 90ms. + */ + if (reg & XFER_DONE_INTR) { + unsigned int offset_in_buf = ++stream->period_idx * period_sz; + + if (offset_in_buf >= buf_sz) { + stream->period_idx = 0; + offset_in_buf = stream->period_idx * period_sz; + } + + /* Move on to the next period in the overall buffer */ + writel_relaxed(buf_addr + offset_in_buf, + i2s->base + BUFF_BASE_ADDR_REG(dir)); + /* The last period might be smaller, update length if needed */ + period_sz = min(period_sz, buf_sz - offset_in_buf); + writel_relaxed(BYTES_TO_WORDS(period_sz), + i2s->base + BUFF_LEN_REG(dir)); + + /* Inform the PCM middle-layer */ + snd_pcm_period_elapsed(substream); + } + + return IRQ_HANDLED; +} + +static int xlnx_logicpd_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct xlnx_logicpd_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int period_sz = snd_pcm_lib_period_bytes(substream); + dma_addr_t buf_addr = substream->dma_buffer.addr; + unsigned int dir = substream->stream; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + i2s->streams[dir].period_idx = 0; + /* Disable the other engine if enabled */ + if (readl(i2s->base + CONTROL_REG(!dir)) & ENGINE_EN) + writel(0, i2s->base + CONTROL_REG(!dir)); + /* Enable the desired engine */ + writel_relaxed(ENGINE_EN, i2s->base + CONTROL_REG(dir)); + /* Set the buffer start address */ + writel_relaxed(buf_addr, i2s->base + BUFF_BASE_ADDR_REG(dir)); + /* Enable the XFER_DONE IRQ, signaling the end of the period */ + xlnx_logicpd_dai_int_en(i2s, dir); + /* Actually start the internal DMA engine */ + writel(BYTES_TO_WORDS(period_sz), + i2s->base + BUFF_LEN_REG(dir)); + break; + case SNDRV_PCM_TRIGGER_STOP: + /* Disable the interrupts */ + xlnx_logicpd_dai_int_dis(i2s, dir); + /* Ensure the host IRQ is deasserted */ + readl_relaxed(i2s->base + INTR_STAT_REG(dir)); + break; + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops xlnx_logicpd_dai_ops = { + .trigger = xlnx_logicpd_dai_trigger, +}; + +static int xlnx_logicpd_dai_probe(struct snd_soc_dai *dai) +{ + struct xlnx_logicpd_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int dir; + + for (dir = SNDRV_PCM_STREAM_PLAYBACK; + dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + i2s->streams[dir].i2s = i2s; + + /* Reset the transmitter/receiver engine */ + writel_relaxed(0, i2s->base + CONTROL_REG(dir)); + /* Mask all interrupts */ + writel_relaxed(GENMASK(31, 0), i2s->base + INTR_MASK_REG(dir)); + } + + return 0; +} + +struct snd_soc_dai_driver xlnx_logicpd_dai = { + .name = "xylinx-logicpd-dai", + .probe = xlnx_logicpd_dai_probe, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = 8000, + .rate_max = 192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = 8000, + .rate_max = 192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .symmetric_rates = 1, + .ops = &xlnx_logicpd_dai_ops, +}; + +static const struct snd_soc_component_driver xlnx_logicpd_i2s_component = { + .name = DRV_NAME, + .ops = &xlnx_logicpd_pcm_ops, +}; + +static const struct of_device_id xlnx_logicpd_i2s_of_match[] = { + { + .compatible = "xlnx,logicpd-i2s-dai", + }, + {}, +}; +MODULE_DEVICE_TABLE(of, xlnx_logicpd_i2s_of_match); + +static int xlnx_logicpd_i2s_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct xlnx_logicpd_i2s *i2s; + struct xlnx_logicpd_stream *stream; + int tx_irq, rx_irq, ret; + u32 reg; + + i2s = devm_kzalloc(dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) + return -ENOMEM; + + dev_set_drvdata(dev, i2s); + + i2s->base = devm_platform_ioremap_resource(pdev, 0); + if (IS_ERR(i2s->base)) + return PTR_ERR(i2s->base); + + stream = &i2s->streams[SNDRV_PCM_STREAM_PLAYBACK]; + tx_irq = platform_get_irq_byname(pdev, "tx"); + if (tx_irq > 0) { + ret = devm_request_irq(dev, tx_irq, xlnx_logicpd_dai_isr, + 0, "logicpd-i2s-tx", stream); + if (ret) + return ret; + } else { + dev_err(dev, "TX IRQ not available (%d), disabling playback\n", + tx_irq); + tx_irq = 0; + } + + stream = &i2s->streams[SNDRV_PCM_STREAM_CAPTURE]; + rx_irq = platform_get_irq_byname(pdev, "rx"); + if (rx_irq > 0) { + ret = devm_request_irq(dev, rx_irq, xlnx_logicpd_dai_isr, + 0, "logicpd-i2s-rx", stream); + if (ret) + return ret; + } else { + dev_err(dev, "RX IRQ not available (%d), disabling capture\n", + rx_irq); + rx_irq = 0; + } + + if (!tx_irq && !rx_irq) + return -EINVAL; + + ret = devm_snd_soc_register_component(dev, &xlnx_logicpd_pcm_component, + NULL, 0); + if (ret) { + dev_err(dev, "cannot register PCM component (%d)\n", ret); + return ret; + } + + ret = devm_snd_soc_register_component(dev, &xlnx_logicpd_i2s_component, + &xlnx_logicpd_dai, 1); + if (ret) { + dev_err(dev, "cannot register I2S component (%d)\n", ret); + return ret; + } + + reg = readl_relaxed(i2s->base + IP_VERSION); + dev_info(dev, "%s DAI version %u.%u.%c (license: %s) registered\n", + xlnx_logicpd_dai.name, + (unsigned int)MAJOR_REV(reg), + (unsigned int)MINOR_REV(reg), + (char)PATCH_LEVEL(reg), + LICENSE_TYPE(reg) == 0 ? "source" : + (LICENSE_TYPE(reg) == 1 ? "eval" : "release")); + + return ret; +} + +static struct platform_driver xlnx_logicpd_i2s_driver = { + .driver = { + .name = DRV_NAME, + .of_match_table = xlnx_logicpd_i2s_of_match, + }, + .probe = xlnx_logicpd_i2s_probe, +}; + +module_platform_driver(xlnx_logicpd_i2s_driver); + +MODULE_AUTHOR("Miquel Raynal <miquel.raynal(a)bootlin.com>"); +MODULE_DESCRIPTION("Xilinx logicPD I2S module"); +MODULE_LICENSE("GPL v2"); -- 2.20.1

3 8

[alsa-devel] [PATCH] ALSA: hda/realtek - Enable internal speaker & headset mic of ASUS UX431FL
by Jian-Hong Pan 03 Sep '19

03 Sep '19

Original pin node values of ASUS UX431FL with ALC294: 0x12 0xb7a60140 0x13 0x40000000 0x14 0x90170110 0x15 0x411111f0 0x16 0x411111f0 0x17 0x90170111 0x18 0x411111f0 0x19 0x411111f0 0x1a 0x411111f0 0x1b 0x411111f0 0x1d 0x4066852d 0x1e 0x411111f0 0x1f 0x411111f0 0x21 0x04211020 1. Has duplicated internal speakers (0x14 & 0x17) which makes the output route become confused. So, the output volume cannot be changed by setting. 2. Misses the headset mic pin node. This patch disables the confusing speaker (NID 0x14) and enables the headset mic (NID 0x19). Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e333b3e30e31..0a1fa99a6723 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5797,6 +5797,7 @@ enum { ALC286_FIXUP_ACER_AIO_HEADSET_MIC, ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, ALC299_FIXUP_PREDATOR_SPK, + ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -6837,6 +6838,16 @@ static const struct hda_fixup alc269_fixups[] = { { } } }, + [ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x411111f0 }, /* disable confusing internal speaker */ + { 0x19, 0x04a11150 }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6995,6 +7006,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), -- 2.20.1

2 3

[alsa-devel] [PATCH 2/2] Fix a stack buffer overflow bug in check_input_term
by Hui Peng 02 Sep '19

02 Sep '19

`check_input_term` recursively calls itself with input from device side (e.g., uac_input_terminal_descriptor.bCSourceID) as argument (id). In `check_input_term`, if `check_input_term` is called with the same `id` argument as the caller, it triggers endless recursive call, resulting kernel space stack overflow. This patch fixes the bug by adding a bitmap to `struct mixer_build` to keep track of the checked ids and stop the execution if some id has been checked (similar to how parse_audio_unit handles unitid argument). CVE: CVE-2018-15118 Reported-by: Hui Peng <benquike(a)gmail.com> Reported-by: Mathias Payer <mathias.payer(a)nebelwelt.net> Signed-off-by: Hui Peng <benquike(a)gmail.com> --- sound/usb/mixer.c | 29 ++++++++++++++++++++++++----- 1 file changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 10ddec76f906..e24572fd6e30 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -81,6 +81,7 @@ struct mixer_build { unsigned char *buffer; unsigned int buflen; DECLARE_BITMAP(unitbitmap, MAX_ID_ELEMS); + DECLARE_BITMAP(termbitmap, MAX_ID_ELEMS); struct usb_audio_term oterm; const struct usbmix_name_map *map; const struct usbmix_selector_map *selector_map; @@ -709,15 +710,24 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm * parse the source unit recursively until it reaches to a terminal * or a branched unit. */ -static int check_input_term(struct mixer_build *state, int id, +static int __check_input_term(struct mixer_build *state, int id, struct usb_audio_term *term) { int err; void *p1; + unsigned char *hdr; memset(term, 0, sizeof(*term)); - while ((p1 = find_audio_control_unit(state, id)) != NULL) { - unsigned char *hdr = p1; + for (;;) { + /* a loop in the terminal chain? */ + if (test_and_set_bit(id, state->termbitmap)) + return -EINVAL; + + p1 = find_audio_control_unit(state, id); + if (!p1) + break; + + hdr = p1; term->id = id; switch (hdr[2]) { case UAC_INPUT_TERMINAL: @@ -732,7 +742,7 @@ static int check_input_term(struct mixer_build *state, int id, /* call recursively to verify that the * referenced clock entity is valid */ - err = check_input_term(state, d->bCSourceID, term); + err = __check_input_term(state, d->bCSourceID, term); if (err < 0) return err; @@ -764,7 +774,7 @@ static int check_input_term(struct mixer_build *state, int id, case UAC2_CLOCK_SELECTOR: { struct uac_selector_unit_descriptor *d = p1; /* call recursively to retrieve the channel info */ - err = check_input_term(state, d->baSourceID[0], term); + err = __check_input_term(state, d->baSourceID[0], term); if (err < 0) return err; term->type = d->bDescriptorSubtype << 16; /* virtual type */ @@ -811,6 +821,15 @@ static int check_input_term(struct mixer_build *state, int id, return -ENODEV; } + +static int check_input_term(struct mixer_build *state, int id, + struct usb_audio_term *term) +{ + memset(term, 0, sizeof(*term)); + memset(state->termbitmap, 0, sizeof(state->termbitmap)); + return __check_input_term(state, id, term); +} + /* * Feature Unit */ -- 2.17.1

3 4

[alsa-devel] [PATCH 1/2] Fix an OOB bug in parse_audio_mixer_unit
by Hui Peng 02 Sep '19

02 Sep '19

The `uac_mixer_unit_descriptor` shown as below is read from the device side. In `parse_audio_mixer_unit`, `baSourceID` field is accessed from index 0 to `bNrInPins` - 1, the current implementation assumes that descriptor is always valid (the length of descriptor is no shorter than 5 + `bNrInPins`). If a descriptor read from the device side is invalid, it may trigger out-of-bound memory access. ``` struct uac_mixer_unit_descriptor { __u8 bLength; __u8 bDescriptorType; __u8 bDescriptorSubtype; __u8 bUnitID; __u8 bNrInPins; __u8 baSourceID[]; } ``` This patch fixes the bug by add a sanity check on the length of the descriptor. CVE: CVE-2018-15117 Reported-by: Hui Peng <benquike(a)gmail.com> Reported-by: Mathias Payer <mathias.payer(a)nebelwelt.net> Signed-off-by: Hui Peng <benquike(a)gmail.com> --- sound/usb/mixer.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 1f7eb3816cd7..10ddec76f906 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1628,6 +1628,7 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, int pin, ich, err; if (desc->bLength < 11 || !(input_pins = desc->bNrInPins) || + desc->bLength < sizeof(*desc) + desc->bNrInPins || !(num_outs = uac_mixer_unit_bNrChannels(desc))) { usb_audio_err(state->chip, "invalid MIXER UNIT descriptor %d\n", -- 2.17.1

3 4

[alsa-devel] [PATCH] ASoC: fsl_sai: Set SAI Channel Mode to Output Mode
by Daniel Baluta 02 Sep '19

02 Sep '19

>From SAI datasheet: CHMOD, configures if transmit data pins are configured for TDM mode or Output mode. * (0) TDM mode, transmit data pins are tri-stated when slots are masked or channels are disabled. * (1) Output mode, transmit data pins are never tri-stated and will output zero when slots are masked or channels are disabled. When data pins are tri-stated, there is noise on some channels when FS clock value is high and data is read while fsclk is transitioning from high to low. Fix this by setting CHMOD to Output Mode so that pins will output zero when slots are masked or channels are disabled. Cc: NXP Linux Team <linux-imx(a)nxp.com> Signed-off-by: Cosmin-Gabriel Samoila <cosmin.samoila(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 15 ++++++++++++--- sound/soc/fsl/fsl_sai.h | 2 ++ 2 files changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index e896b577b1f7..b9daab0eb6eb 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -467,6 +467,12 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_FRSZ(slots); + /* + * set CHMOD to Output Mode so that transmit data pins will + * output zero when slots are masked or channels are disabled + */ + val_cr4 |= FSL_SAI_CR4_CHMOD; + /* * For SAI master mode, when Tx(Rx) sync with Rx(Tx) clock, Rx(Tx) will * generate bclk and frame clock for Tx(Rx), we should set RCR4(TCR4), @@ -477,7 +483,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, if (!sai->is_slave_mode) { if (!sai->synchronous[TX] && sai->synchronous[RX] && !tx) { regmap_update_bits(sai->regmap, FSL_SAI_TCR4(ofs), - FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, + FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | + FSL_SAI_CR4_CHMOD_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_TCR5(ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | @@ -486,7 +493,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, ~0UL - ((1 << channels) - 1)); } else if (!sai->synchronous[RX] && sai->synchronous[TX] && tx) { regmap_update_bits(sai->regmap, FSL_SAI_RCR4(ofs), - FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, + FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | + FSL_SAI_CR4_CHMOD_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_RCR5(ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | @@ -497,7 +505,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, } regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), - FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, + FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | + FSL_SAI_CR4_CHMOD_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index f96f8d97489d..1e3b4a6889a8 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -119,6 +119,8 @@ #define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) #define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) #define FSL_SAI_CR4_SYWD_MASK (0x1f << 8) +#define FSL_SAI_CR4_CHMOD BIT(5) +#define FSL_SAI_CR4_CHMOD_MASK GENMASK(5, 5) #define FSL_SAI_CR4_MF BIT(4) #define FSL_SAI_CR4_FSE BIT(3) #define FSL_SAI_CR4_FSP BIT(1) -- 2.17.1

4 4

[alsa-devel] [PATCH] ALSA: hda/realtek - Add quirk for HP Pavilion 15
by Sam Bazley 02 Sep '19

02 Sep '19

HP Pavilion 15 (AMD Ryzen-based model) with 103c:84e7 needs the same quirk like HP Envy/Spectre x360 for enabling the mute LED over Mic3 pin. Signed-off-by: Sam Bazley <sambazley(a)fastmail.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 e333b3e30e31..31ba8d68c16d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6973,6 +6973,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x802e, "HP Z240 SFF", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE), -- 2.23.0

2 1

Re: [alsa-devel] sound/usb kernel modules as a DKMS package. (Re: Alsa-devel Digest, Vol 151, Issue 10)
by Hin-Tak Leung 02 Sep '19

02 Sep '19

> Date: Mon, 2 Sep 2019 22:06:07 +0900 > From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> > As a quick glance, the repository is just for deb-based distributions. > >From my experience, users who eager to use the latest source codes tend > to use Arch Linux and Gentoo Linux. It's helpful for the users to > include some instructions just to use the code with DKMS system. Then > you could get any feedback from them as well. Yes, that's correct. It appears that the "dkms mkrpm ..." functionality is not on Ubuntu. I am mostly a Fedora person privately (Slackware before Fedora). The Ubuntu thing is a work-requirement. Have never used Arch nor Gentoo. I'll see if I could get it done on Fedora too. The dkms packaging is mostly done for junior colleagues - I had removed all of work-related material before posting, so it is a bit bare at the moment. > For your information, I maintain out-of-tree codes to develop ALSA > firewire stack for a couple of years, as you start: > https://github.com/takaswie/snd-firewire-improve Yes, thanks. I came across that, and a few other repos which try to package up other part of bleeding-edge sound driver code (like HDA), but not sound/usb yet. Hence this posting.

1 0

[alsa-devel] [PATCH 00/16] ASoC: soc-core cleanup
by Kuninori Morimoto 02 Sep '19

02 Sep '19

Hi Mark I want to add multi CPU DAI support to ALSA SoC. But I noticed that we want to cleanup ALSA SoC before that. These patches will do nothing from "technical" point of view. Just for cleanup Kuninori Morimoto (16): ASoC: soc-core: rename soc_post_component_init() to soc_rtd_init() ASoC: soc-core: add soc_rtd_free() ASoC: soc-core: move soc_probe_component() position ASoC: soc-core: dapm related setup at one place ASoC: soc-core: add snd_soc_dapm_init() ASoC: soc-core: move soc_probe_link_components() position ASoC: soc-core: self contained soc_probe_link_components() ASoC: soc-core: self contained soc_remove_link_components() ASoC: soc-core: self contained soc_remove_link_dais() ASoC: soc-core: move soc_probe_dai() next to soc_remove_dai() ASoC: soc-core: add new soc_link_init() ASoC: soc-core: self contained soc_probe_link_dais() ASoC: soc-core: move soc_probe_link_dais() next to soc_remove_link_dais() ASoC: soc-core: self contained soc_bind_aux_dev() ASoC: soc-core: add soc_unbind_aux_dev() ASoC: soc-core: self contained soc_unbind_aux_dev() include/sound/soc-dapm.h | 3 + sound/soc/soc-core.c | 568 ++++++++++++++++++++++++----------------------- sound/soc/soc-dapm.c | 21 ++ 3 files changed, 310 insertions(+), 282 deletions(-) -- 2.7.4

2 22

[alsa-devel] [PATCH v2] drm: dw-hdmi-i2s: enable audio clock in audio_startup
by Cheng-Yi Chiang 02 Sep '19

02 Sep '19

In the designware databook, the sequence of enabling audio clock and setting format is not clearly specified. Currently, audio clock is enabled in the end of hw_param ops after setting format. On some monitors, there is a possibility that audio does not come out. Fix this by enabling audio clock in audio_startup ops before hw_param ops setting format. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Reviewed-by: Douglas Anderson <dianders(a)chromium.org> Reviewed-by: Jonas Karlman <jonas(a)kwiboo.se> Tested-by: Douglas Anderson <dianders(a)chromium.org> --- Changes from v1: 1. Move audio_startup to the front of audio_shutdown. 2. Fix the indentation of audio_startup equal sign using tab. 3. Rebase the patch on linux-next/master. 4. Add Reviewed-by and Tested-by fields from dianders(a)chromium.org. 5. Add Reviewed-by field from jonas(a)kwiboo.se. drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c index 1d15cf9b6821..34d8e837555f 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -109,6 +109,14 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + return 0; +} + +static int dw_hdmi_i2s_audio_startup(struct device *dev, void *data) +{ + struct dw_hdmi_i2s_audio_data *audio = data; + struct dw_hdmi *hdmi = audio->hdmi; + dw_hdmi_audio_enable(hdmi); return 0; @@ -153,6 +161,7 @@ static int dw_hdmi_i2s_get_dai_id(struct snd_soc_component *component, static struct hdmi_codec_ops dw_hdmi_i2s_ops = { .hw_params = dw_hdmi_i2s_hw_params, + .audio_startup = dw_hdmi_i2s_audio_startup, .audio_shutdown = dw_hdmi_i2s_audio_shutdown, .get_eld = dw_hdmi_i2s_get_eld, .get_dai_id = dw_hdmi_i2s_get_dai_id, -- 2.23.0.187.g17f5b7556c-goog

3 3

[alsa-devel] [PATCH] ASoC: sun4i-i2s: incorrect regmap for A83t
by codekipper＠gmail.com 02 Sep '19

02 Sep '19

From: Marcus Cooper <codekipper(a)gmail.com> Fixes: 21faaea1343f ("ASoC: sun4i-i2s: Add support for A83T") Signed-off-by: Marcus Cooper <codekipper(a)gmail.com> --- sound/soc/sunxi/sun4i-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index cbc6e59aa089..056a299c03fb 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1109,7 +1109,7 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = { static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .has_reset = true, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, - .sun4i_i2s_regmap = &sun4i_i2s_regmap_config, + .sun4i_i2s_regmap = &sun8i_i2s_regmap_config, .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8), .field_fmt_wss = REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2), .field_fmt_sr = REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6), -- 2.23.0

4 3

[alsa-devel] [PATCH] MAINTAINERS: update entry of firewire audio drivers
by Takashi Sakamoto 02 Sep '19

02 Sep '19

This commit adds myself as one of maintainers for firewire audio drivers and IEC 61883-1/6 packet streaming engine. I call them ALSA firewire stack as a whole. 6 years ago I joined in development for this category of drivers with heavy reverse-engineering tasks and over 100 models are now available from ALSA applications. IEEE 1394 bus itself and units on the bus are enough legacy but the development still continues. I have a plan to add drastic enhancement in kernel v5.5 and v5.6 period. This commit adds myself into MAINTAINERS so that developers and users can easily find active developer to post their issues, especially for regression. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- MAINTAINERS | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 24e29b2e53c9..8929a2ec75f7 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6264,8 +6264,9 @@ S: Maintained F: drivers/hwmon/f75375s.c F: include/linux/f75375s.h -FIREWIRE AUDIO DRIVERS +FIREWIRE AUDIO DRIVERS and IEC 61883-1/6 PACKET STREAMING ENGINE M: Clemens Ladisch <clemens(a)ladisch.de> ++M: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained -- 2.20.1

2 2

[alsa-devel] [PATCH] alsa-utils: Remove usleep usage
by Rosen Penev 02 Sep '19

02 Sep '19

usleep was deprecated in POSIX 2008 and is not available without deprecated APIs enabled in uClibc-ng. Signed-off-by: Rosen Penev <rosenp(a)gmail.com> --- alsactl/daemon.c | 3 ++- alsaloop/pcmjob.c | 3 ++- amidi/amidi.c | 6 ++++-- bat/bat.c | 5 +++-- 4 files changed, 11 insertions(+), 6 deletions(-) diff --git a/alsactl/daemon.c b/alsactl/daemon.c index 1d98caa..3b84cf6 100644 --- a/alsactl/daemon.c +++ b/alsactl/daemon.c @@ -348,7 +348,8 @@ static int check_another_instance(const char *pidfile) if (pid >= 0) { /* invoke new card rescan */ if (kill(pid, SIGUSR1) >= 0) { - usleep(1000); + struct timespec req = { 0, 1 * 1000 * 1000 }; + nanosleep(&req, NULL); pid = read_pid_file(pidfile); if (pid >= 0) return 1; diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index 29d1aba..d0347dc 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -662,10 +662,11 @@ static int xrun(struct loopback_handle *lhandle) static int suspend(struct loopback_handle *lhandle) { + struct timespec req = { 0, 1000}; int err; while ((err = snd_pcm_resume(lhandle->handle)) == -EAGAIN) - usleep(1); + nanosleep(&req, NULL); if (err < 0) return xrun(lhandle); return 0; diff --git a/amidi/amidi.c b/amidi/amidi.c index b6e05e1..3b2953d 100644 --- a/amidi/amidi.c +++ b/amidi/amidi.c @@ -254,10 +254,12 @@ static int send_midi_interleaved(void) snd_rawmidi_status(output, st); do { /* 320 µs per byte as noted in Page 1 of MIDI spec */ - usleep((buffer_size - snd_rawmidi_status_get_avail(st)) * 320); + struct timespec req = { 0, (buffer_size - snd_rawmidi_status_get_avail(st)) * 320 * 1000}; + nanosleep(&req, NULL); snd_rawmidi_status(output, st); } while(snd_rawmidi_status_get_avail(st) < buffer_size); - usleep(sysex_interval * 1000); + struct timespec sysex = { 0, sysex_interval * 1000 * 1000}; + nanosleep(&sysex, NULL); } /* find end of SysEx */ diff --git a/bat/bat.c b/bat/bat.c index 8645770..27e7a34 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -176,6 +176,7 @@ static void test_loopback(struct bat *bat) pthread_t capture_id, playback_id; int err; int *thread_result_capture, *thread_result_playback; + struct timespec capture_delay = { 0, CAPTURE_DELAY * 1000 * 1000 }; /* start playback */ err = pthread_create(&playback_id, NULL, @@ -188,7 +189,7 @@ static void test_loopback(struct bat *bat) /* TODO: use a pipe to signal stream start etc - i.e. to sync threads */ /* Let some time for playing something before capturing */ - usleep(CAPTURE_DELAY * 1000); + nanosleep(&capture_delay, NULL); /* start capture */ err = pthread_create(&capture_id, NULL, (void *) bat->capture.fct, bat); @@ -691,7 +692,7 @@ int main(int argc, char *argv[]) } /* Waiting 500ms and start the next round */ - usleep(CAPTURE_DELAY * 1000); + nanosleep(&capture_delay, NULL); } goto out; } -- 2.17.1

2 1

Re: [alsa-devel] [PATCH 2/2] ASoC: rt1011: ADCDAT pin config modification
by Shuming [范書銘] 02 Sep '19

02 Sep '19

> > > So really this is setting up a loopback for testing? > > > We would not like to change ADCDAT pin to input mode in normal case. > > That's why the driver enables ADCDAT pin to the output mode in default. > > The rt1011 supports the feedback signal which could be playback data or I/V > data, etc. > > If the system wants the AEC reference data, rt1011 could feedback the > playback data. > > > The product could connect 2/4/6/8 rt1011 chips on the same I2S bus. > > In a test or debug mode, we could toggle ADCDAT pin to input mode that > > also helps HW engineer check the slot of feedback signal for each rt1011. > > I think this needs more than just a straight userspace control on one device, > these use cases make sense but they'll need to be configured over multiple > chips simultaneously otherwise there's some possibility of hardware damage > (eg, if two chips try to drive the signal at the same time). If this really can be > usefully varied at runtime then the driver bit of this should probably be an API > that the machine driver can call, the machine driver can then expose a control > that sets all the chips involved up together. I understand your concerns and comments. In a design-in project, we will provide the proper control settings to arrange the slot location of ADCDAT signal for each rt1011. HW engineer will confirm the ADCDAT signal, too. So, the customer should not make two chips drive the signal at the same slot and at the same time. But, there is a chance to happen if the customer sets the wrong control settings. We will take your suggestion to make it as API call. The machine driver knows how many rt1011 chips connect and make pin config change together. Thanks. > ------Please consider the environment before printing this e-mail.

1 0

[alsa-devel] bug
by liebrecht＠grossmann-venter.com 02 Sep '19

02 Sep '19

Clearly a problem with Alsa. Alsa reports "Most likely this is a bug in the ALSA driver 'snd_usb_audio'." Alsa used to work great. howevere recently it takes 1/2-1hour for pulseaudio to convince alsa to make the 1818VSL available. Jack is not involved at all. Pulseaudio connects straight to alsa. Pulseaudio is my saving grace roight now as it perpetually tries to initialize and get alsa to present the audio interface, which it eventually after an hour does "waking up" Also sorts of decides to "wake up" and then suddenly the 1818vsl is found. How do I "wake up" alsa so that I can use soundcards. I cannot wait an hour after reboot or logout to get alsa to work again. See below What happens after starting pulseaudio is that it cycles the 1818 for about 1/2 hour to get a link to the 181vsl soundcard. That is the first paragraph (1) below which is perpetually cycled by pulseaudio. Eventually it succeeds which gives the BUG comment as below in paragraph (2) ------------------------------------------------------- 1) E: [pulseaudio] udev-util.c: Failed to get card object. E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="3" name="usb-PreSonus_AudioBox_1818_VSL_2209-00" card_name="alsa_card.usb-PreSonus_Audio.Box_1818_VSL_2209-00" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1""): initialization failed. (**Note this message is cycled for an hour**) 2) (**Then miraculously....Alsa wakes up and pulseaudio can connect to 1818vsl **) E: [alsa-sink-USB Audio] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. E: [alsa-sink-USB Audio] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. E: [alsa-sink-USB Audio] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ------------------------------------------------------ Dont know what happened top Also to behave this way. Maybe someone has an idea ? upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.64 !!################################ !!Script ran on: Mon Sep 2 03:27:26 UTC 2019 !!Linux Distribution !!------------------ [1;36mWelcome to [1;33mMX Linux[1;36m! Powered by [1;33mDebian[1;36m.[0m PRETTY_NAME="Debian GNU/Linux 9 (stretch)" NAME="Debian GNU/Linux" ID=debian HOME_URL="https://www.debian.org/" SUPPORT_URL="https://www.debian.org/support" BUG_REPORT_URL="https://bugs.debian.org/" MX-18.3_x64 Continuum Dec 20, 2018 MX-18.3_x64 Continuum Dec 20, 2018 !!DMI Information !!--------------- Manufacturer: Supermicro Product Name: H8DGU Product Version: 1234567890 Firmware Version: 3.5c Board Vendor: Supermicro Board Name: H8DGU !!ACPI Device Status Information !!--------------- /sys/bus/acpi/devices/PNP0103:00/status 15 /sys/bus/acpi/devices/PNP0501:00/status 15 /sys/bus/acpi/devices/PNP0501:01/status 15 /sys/bus/acpi/devices/PNP0C0C:00/status 11 /sys/bus/acpi/devices/PNP0C0F:00/status 9 /sys/bus/acpi/devices/PNP0C0F:01/status 9 /sys/bus/acpi/devices/PNP0C0F:02/status 9 /sys/bus/acpi/devices/PNP0C0F:03/status 9 /sys/bus/acpi/devices/PNP0C0F:04/status 9 /sys/bus/acpi/devices/PNP0C0F:05/status 9 /sys/bus/acpi/devices/PNP0C0F:06/status 9 /sys/bus/acpi/devices/PNP0C0F:07/status 9 /sys/bus/acpi/devices/device:1e/status 15 /sys/bus/acpi/devices/device:1f/status 15 /sys/bus/acpi/devices/device:21/status 15 /sys/bus/acpi/devices/device:22/status 15 !!Kernel Information !!------------------ Kernel release: 4.19.0-5-amd64 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k4.19.0-5-amd64 Library version: 1.1.9 Utilities version: 1.1.3 !!Loaded ALSA modules !!------------------- snd_hda_intel snd_usb_audio snd_usb_audio !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfebfc000 irq 46 1 [Interface ]: USB-Audio - USB Uno MIDI Interface M-Audio USB Uno MIDI Interface at usb-0000:00:12.2-1.2.3, full speed 2 [M1x1 ]: USB-Audio - MidiSport 1x1 M-Audio MidiSport 1x1 at usb-0000:00:12.2-1.2.2, full speed !!PCI Soundcards installed in the system !!-------------------------------------- 03:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Device aae0 !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 03:00.1 0403: 1002:aae0 Subsystem: 1462:aae0 !!Modprobe options (Sound related) !!-------------------------------- snd_pcsp: index=-2 snd_usb_audio: index=-2 snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) pm_blacklist : Y position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : N probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : -1 snoop : -1 !!Module: snd_usb_audio autoclock : Y device_setup : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) ignore_ctl_error : N index : -2,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 pid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 quirk_alias : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) use_vmalloc : Y vid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 !!Module: snd_usb_audio autoclock : Y device_setup : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) ignore_ctl_error : N index : -2,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 pid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 quirk_alias : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) use_vmalloc : Y vid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 Node 0x03 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Connection: 1 0x02 Node 0x04 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=02, enabled=1 Connection: 1 0x04 Node 0x06 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=03, enabled=1 Connection: 1 0x06 Node 0x08 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="IEC958 Playback Con Mask", index=3, device=0 Control: name="IEC958 Playback Pro Mask", index=3, device=0 Control: name="IEC958 Playback Default", index=3, device=0 Control: name="IEC958 Playback Switch", index=3, device=0 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=04, enabled=1 Connection: 1 0x08 Node 0x0a [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=1, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="IEC958 Playback Con Mask", index=4, device=0 Control: name="IEC958 Playback Pro Mask", index=4, device=0 Control: name="IEC958 Playback Default", index=4, device=0 Control: name="IEC958 Playback Switch", index=4, device=0 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=05, enabled=1 Connection: 1 0x0a Node 0x0c [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000094: OUT Detect HDMI Pin Default 0x585600f0: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x0c --endcollapse-- !!USB Mixer information !!--------------------- --startcollapse-- USB Mixer: usb_id=0x07630150, ctrlif=0, ctlerr=0 Card: M-Audio USB Uno MIDI Interface at usb-0000:00:12.2-1.2.3, full speed --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 8 Sep 1 04:44 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 10 Sep 1 06:36 /dev/snd/controlC1 crw-rw---- 1 root audio 116, 12 Sep 1 06:36 /dev/snd/controlC2 crw-rw---- 1 root audio 116, 7 Sep 1 04:44 /dev/snd/hwC0D0 crw-rw---- 1 root audio 116, 9 Sep 1 06:36 /dev/snd/midiC1D0 crw-rw---- 1 root audio 116, 11 Sep 1 06:36 /dev/snd/midiC2D0 crw-rw---- 1 root audio 116, 6 Sep 1 23:17 /dev/snd/pcmC0D10p crw-rw---- 1 root audio 116, 2 Sep 1 23:17 /dev/snd/pcmC0D3p crw-rw---- 1 root audio 116, 3 Sep 1 23:17 /dev/snd/pcmC0D7p crw-rw---- 1 root audio 116, 4 Sep 1 23:17 /dev/snd/pcmC0D8p crw-rw---- 1 root audio 116, 5 Sep 1 23:17 /dev/snd/pcmC0D9p crw-rw---- 1 root audio 116, 1 Sep 1 05:13 /dev/snd/seq crw-rw---- 1 root audio 116, 33 Sep 1 04:44 /dev/snd/timer /dev/snd/by-id: total 0 drwxr-xr-x 2 root root 80 Sep 1 22:40 . drwxr-xr-x 4 root root 340 Sep 1 22:40 .. lrwxrwxrwx 1 root root 12 Sep 1 06:36 usb-0763_1011-00 -> ../controlC2 lrwxrwxrwx 1 root root 12 Sep 1 06:36 usb-M-Audio_USB_Uno_MIDI_Interface-00 -> ../controlC1 /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 100 Sep 1 22:40 . drwxr-xr-x 4 root root 340 Sep 1 22:40 .. lrwxrwxrwx 1 root root 12 Sep 1 06:36 pci-0000:00:12.2-usb-0:1.2.2:1.0 -> ../controlC2 lrwxrwxrwx 1 root root 12 Sep 1 06:36 pci-0000:00:12.2-usb-0:1.2.3:1.0 -> ../controlC1 lrwxrwxrwx 1 root root 12 Sep 1 04:44 pci-0000:03:00.1 -> ../controlC0 !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) # v4.0 # Requires alsa-utils libasound2-plugin-equal # ***** Defaults ******************************************************* # Audio card/chip and device to use and control # In most cases these settings should remain commented (#) # When commented, card and device 0 are usually automatically selected # When uncommented, the item corresponding to the number is selected #defaults.pcm.card 0 #defaults.pcm.device 0 #defaults.ctl.card 0 # ***** Reset ********************************************************** # override the existing definition of the default device pcm.!default { type plug # point output to the equalizer device slave.pcm plugequal } # ***** Equalizer ****************************************************** # set up the control interface of the equalizer device ctl.equalizer { type equal } # set up the equalizer device pcm.plugequal { type equal # point output to device named preamp slave.pcm "plug:preamp" } # ***** Pre-Amp ******************************************************** # set up the preamp device pcm.preamp { type softvol # name of slider control to display in alsamixer interface control.name Pre-Amp # minimum dB when slider is at 0% min_dB -5.0 # maximum dB when slider is at 100% max_dB 40.0 # point output to device named duplex slave.pcm "duplex" } # ***** Playback and Capture ******************************************* # set up the playback/capture device pcm.duplex { type asym # point playback output to use dmix playback.pcm "dmix" # point capture input record to use dsnoop capture.pcm "dsnoop" } !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA ATI HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA ATI HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA ATI HDMI], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA ATI HDMI], device 10: HDMI 4 [HDMI 4] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** !!Amixer output !!------------- !!-------Mixer controls for card 0 [HDMI] Card hw:0 'HDMI'/'HDA ATI HDMI at 0xfebfc000 irq 46' Mixer name : 'ATI R6xx HDMI' Components : 'HDA:1002aa01,00aa0100,00100700' Controls : 35 Simple ctrls : 5 Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',1 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',2 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',3 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',4 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] !!-------Mixer controls for card 1 [Interface] Card hw:1 'Interface'/'M-Audio USB Uno MIDI Interface at usb-0000:00:12.2-1.2.3, full speed' Mixer name : 'USB Mixer' Components : 'USB0763:0150' Controls : 1 Simple ctrls : 0 !!-------Mixer controls for card 2 [M1x1] Card hw:2 'M1x1'/'M-Audio MidiSport 1x1 at usb-0000:00:12.2-1.2.2, full speed' Mixer name : '' Components : 'USB0763:1011' Controls : 0 Simple ctrls : 0 !!Alsactl output !!-------------- --startcollapse-- state.HDMI { control.1 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.2 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.3 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.4 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.5 { iface MIXER name 'IEC958 Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.6 { iface PCM device 3 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.7 { iface CARD name 'HDMI/DP,pcm=7 Jack' value false comment { access read type BOOLEAN count 1 } } control.8 { iface MIXER name 'IEC958 Playback Con Mask' index 1 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.9 { iface MIXER name 'IEC958 Playback Pro Mask' index 1 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.10 { iface MIXER name 'IEC958 Playback Default' index 1 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.11 { iface MIXER name 'IEC958 Playback Switch' index 1 value true comment { access 'read write' type BOOLEAN count 1 } } control.12 { iface PCM device 7 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.13 { iface CARD name 'HDMI/DP,pcm=8 Jack' value false comment { access read type BOOLEAN count 1 } } control.14 { iface MIXER name 'IEC958 Playback Con Mask' index 2 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.15 { iface MIXER name 'IEC958 Playback Pro Mask' index 2 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.16 { iface MIXER name 'IEC958 Playback Default' index 2 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.17 { iface MIXER name 'IEC958 Playback Switch' index 2 value true comment { access 'read write' type BOOLEAN count 1 } } control.18 { iface PCM device 8 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.19 { iface CARD name 'HDMI/DP,pcm=9 Jack' value false comment { access read type BOOLEAN count 1 } } control.20 { iface MIXER name 'IEC958 Playback Con Mask' index 3 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.21 { iface MIXER name 'IEC958 Playback Pro Mask' index 3 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.22 { iface MIXER name 'IEC958 Playback Default' index 3 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.23 { iface MIXER name 'IEC958 Playback Switch' index 3 value true comment { access 'read write' type BOOLEAN count 1 } } control.24 { iface PCM device 9 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.25 { iface CARD name 'HDMI/DP,pcm=10 Jack' value false comment { access read type BOOLEAN count 1 } } control.26 { iface MIXER name 'IEC958 Playback Con Mask' index 4 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.27 { iface MIXER name 'IEC958 Playback Pro Mask' index 4 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.28 { iface MIXER name 'IEC958 Playback Default' index 4 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.29 { iface MIXER name 'IEC958 Playback Switch' index 4 value true comment { access 'read write' type BOOLEAN count 1 } } control.30 { iface PCM device 10 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.31 { iface PCM device 3 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.32 { iface PCM device 7 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.33 { iface PCM device 8 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.34 { iface PCM device 9 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.35 { iface PCM device 10 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } } state.Interface { control.1 { iface CARD name 'Keep Interface' value false comment { access 'read write' type BOOLEAN count 1 } } } state.M1x1 { control { } } --endcollapse-- !!All Loaded Modules !!------------------ Module snd_seq_dummy snd_seq_midi snd_seq_midi_event snd_seq bnep rpcsec_gss_krb5 nfsv4 dns_resolver nfs fscache cpufreq_userspace cpufreq_conservative cpufreq_powersave bluetooth pci_stub vboxpci drbg ansi_cprng vboxnetadp ecdh_generic rfkill vboxnetflt vboxdrv ftdi_sio usbserial binfmt_misc snd_usb_audio snd_usbmidi_lib snd_rawmidi snd_seq_device joydev snd_hda_codec_hdmi snd_hda_intel snd_hda_codec amd64_edac_mod edac_mce_amd amdgpu snd_hda_core snd_hwdep snd_pcm_oss kvm_amd ccp mgag200 rng_core snd_mixer_oss kvm irqbypass chash snd_pcm gpu_sched ipmi_si crct10dif_pclmul evdev crc32_pclmul serio_raw ghash_clmulni_intel ipmi_devintf snd_timer ttm ipmi_msghandler drm_kms_helper k10temp sg snd fam15h_power sp5100_tco drm soundcore button nfsd auth_rpcgss nfs_acl lockd grace sunrpc loop vhba ecryptfs parport_pc ppdev lp parport ip_tables x_tables autofs4 sr_mod cdrom ext4 crc16 mbcache jbd2 fscrypto ecb uas usb_storage hid_generic usbhid hid btrfs xor zstd_decompress zstd_compress xxhash sd_mod raid6_pq libcrc32c crc32c_generic vmd crc32c_intel ohci_pci ata_generic aesni_intel aes_x86_64 ahci crypto_simd ehci_pci ohci_hcd pata_atiixp libahci cryptd glue_helper ehci_hcd psmouse libata igb usbcore i2c_piix4 i2c_algo_bit scsi_mod usb_common dca !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x03 0x185600f0 0x05 0x185600f0 0x07 0x185600f0 0x09 0x185600f0 0x0b 0x185600f0 0x0d 0x585600f0 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: !!ALSA/HDA dmesg !!--------------

1 0

[alsa-devel] [PATCH] drm: dw-hdmi-i2s: enable audio clock in audio_startup
by Cheng-Yi Chiang 02 Sep '19

02 Sep '19

In the designware databook, the sequence of enabling audio clock and setting format is not clearly specified. Currently, audio clock is enabled in the end of hw_param ops after setting format. On some monitors, there is a possibility that audio does not come out. Fix this by enabling audio clock in audio_startup ops before hw_param ops setting format. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c index 5cbb71a866d5..08b4adbb1ddc 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -69,6 +69,14 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + return 0; +} + +static int dw_hdmi_i2s_audio_startup(struct device *dev, void *data) +{ + struct dw_hdmi_i2s_audio_data *audio = data; + struct dw_hdmi *hdmi = audio->hdmi; + dw_hdmi_audio_enable(hdmi); return 0; @@ -105,6 +113,7 @@ static int dw_hdmi_i2s_get_dai_id(struct snd_soc_component *component, } static struct hdmi_codec_ops dw_hdmi_i2s_ops = { + .audio_startup = dw_hdmi_i2s_audio_startup, .hw_params = dw_hdmi_i2s_hw_params, .audio_shutdown = dw_hdmi_i2s_audio_shutdown, .get_dai_id = dw_hdmi_i2s_get_dai_id, -- 2.23.0.187.g17f5b7556c-goog

3 3

[alsa-devel] [PATCH] alsa-lib: Remove usleep usage
by Rosen Penev 02 Sep '19

02 Sep '19

usleep was deprecated by POSIX 2008 and is unavailable without deprecated APIs enabled in uClibc-ng. Signed-off-by: Rosen Penev <rosenp(a)gmail.com> --- src/pcm/pcm_shm.c | 3 ++- src/ucm/main.c | 3 ++- test/mixtest.c | 2 +- 3 files changed, 5 insertions(+), 3 deletions(-) diff --git a/src/pcm/pcm_shm.c b/src/pcm/pcm_shm.c index 26a27a57..30a77cd4 100644 --- a/src/pcm/pcm_shm.c +++ b/src/pcm/pcm_shm.c @@ -485,12 +485,13 @@ static int snd_pcm_shm_drain(snd_pcm_t *pcm) snd_pcm_shm_t *shm = pcm->private_data; volatile snd_pcm_shm_ctrl_t *ctrl = shm->ctrl; int err; + struct timespec req = { 0, 10 * 1000 * 1000 }; do { ctrl->cmd = SNDRV_PCM_IOCTL_DRAIN; err = snd_pcm_shm_action(pcm); if (err != -EAGAIN) break; - usleep(10000); + nanosleep(&req, NULL); } while (1); if (err < 0) return err; diff --git a/src/ucm/main.c b/src/ucm/main.c index 99b1cd08..0e5236cb 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -359,6 +359,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, { struct list_head *pos; struct sequence_element *s; + struct timespec sleep = { 0, s->data.sleep * 1000 }; char *cdev = NULL; snd_ctl_t *ctl = NULL; int err = 0; @@ -440,7 +441,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, } break; case SEQUENCE_ELEMENT_TYPE_SLEEP: - usleep(s->data.sleep); + nanosleep(&sleep, NULL); break; case SEQUENCE_ELEMENT_TYPE_EXEC: err = system(s->data.exec); diff --git a/test/mixtest.c b/test/mixtest.c index 57ca5d53..f0940344 100644 --- a/test/mixtest.c +++ b/test/mixtest.c @@ -77,7 +77,7 @@ static double detect_cpu_clock() rdtscll(tsc_begin); clock_gettime(CLOCK_MONOTONIC, &tm_begin); - usleep(1000000); + sleep(1); rdtscll(tsc_end); clock_gettime(CLOCK_MONOTONIC, &tm_end); -- 2.17.1

1 0