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[alsa-devel] [PATCH 2/2] ASoC: cs35l35: Add device tree documentation for CS35L35
by Li Xu 13 Dec '16

13 Dec '16

Add device tree documentation for Cirrus Logic CS35L35 speaker amplifier Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- .../devicetree/bindings/sound/cs35l35.txt | 172 +++++++++++++++++++++ 1 file changed, 172 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l35.txt diff --git a/Documentation/devicetree/bindings/sound/cs35l35.txt b/Documentation/devicetree/bindings/sound/cs35l35.txt new file mode 100644 index 0000000..8b13f67 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l35.txt @@ -0,0 +1,172 @@ +CS35L35 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l35" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - interrupt-parent : Specifies the phandle of the interrupt controller to + which the IRQs from CS35L35 are delivered to. + - interrupts : IRQ line info CS35L35. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + +Optional properties: + - cirrus,reset-gpios : Active low GPIO used to reset the amplifier + + - cirrus,stereo-config : Boolean to determine if there are 2 AMPs for a + Stereo configuration + + - cirrus,audio-channel : Set Location of Audio Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,advisory-channel : Set Location of Advisory Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,shared-boost : Boolean to enable ClassH tracking of Advisory Signal + if 2 Devices share Boost BST_CTL + + - cirrus,sp-drv-strength : Value for setting the Serial Port drive strength + Table 3-10 of the datasheet lists drive-strength specifications + 0 = 1x (Default) + 1 = .5x + + - cirrus,bst-pdn-fet-on : Boolean to determine if the Boost PDN control + powers down with a rectification FET On or Off. If VSPK is supplied + externally then FET is off. + + - cirrus,boost-ctl-millivolt : Boost Converter control word. Step Size of 100mV + 0x00 = (Default) VP + 0x01 = 2600mV + 0x41 = 9000mV + + - cirrus,boost-ipk-milliamp : Boost-converter peak current limit. + Configures the peak current by monitoring the current through the boost FET. + Step size: 112mA + 0x00 = 1680mA + 0x19 = 4480mA + + - cirrus,amp-gain-zc : Boolean to determine if to use Amplifier gain-change + zero-cross + +Optional H/G Algorithm sub-node: + + The cs35l35 node can have a single "cirrus,classh-internal-algo" sub-node + that will disable automatic control of the internal H/G Algorithm. + + It is strongly recommended that the Datasheet be referenced when adjusting + or using these Class H Algorithm controls over the internal Algorithm. + Serious damage can occur to the Device and surrounding components. + + - cirrus,classh-internal-algo : Sub-node for the Internal Class H Algorithm + See Section 4.3 Internal Class H Algorithm in the Datasheet. + If not used, the device manages the ClassH Algorithm internally. + +Optional properties for the "cirrus,classh-internal-algo" Sub-node + + Section 7.29 Class H Control + - cirrus,classh-bst-overide : Boolean + - cirrus,classh-bst-max-limit + - cirrus,classh-mem-depth + + Section 7.30 Class H Headroom Control + - cirrus,classh-headroom + + Section 7.31 Class H Release Rate + - cirrus,classh-release-rate + + Section 7.32 Class H Weak FET Drive Control + - cirrus,classh-wk-fet-disable + - cirrus,classh-wk-fet-delay + - cirrus,classh-wk-fet-thld + + Section 7.34 Class H VP Control + - cirrus,classh-vpch-auto + - cirrus,classh-vpch-rate + - cirrus,classh-vpch-man + +Optional Monitor Signal Format sub-node: + + The cs35l35 node can have a single "cirrus,monitor-signal-format" sub-node + for adjusting the Depth, Location and Frame of the Monitoring Signals + for Algorithms. + + See Sections 4.8.2 through 4.8.4 Serial-Port Control in the Datasheet + + -cirrus,monitor-signal-format : Sub-node for the Monitor Signaling Formating + on the I2S Port. Each of the 3 8 bit values in the array contain the settings + for depth, location, and frame. + + If not used, the defaults for the 6 monitor signals is used. + + Sections 7.44 - 7.53 lists values for the depth, location, and frame + for each monitoring signal. + + - cirrus,imon : 3 8 bit values to set the depth, location, and frame + of the IMON monitor signal. + + - cirrus,vmon : 3 8 bit values to set the depth, location, and frame + of the VMON monitor signal. + + - cirrus,vpmon : 3 8 bit values to set the depth, location, and frame + of the VPMON monitor signal. + + - cirrus,vbstmon : 3 8 bit values to set the depth, location, and frame + of the VBSTMON monitor signal + + - cirrus,vpbrstat : 3 8 bit values to set the depth, location, and frame + of the VPBRSTAT monitor signal + + - cirrus,zerofill : 3 8 bit values to set the depth, location, and frame\ + of the ZEROFILL packet in the monitor signal + +Example: + +cs35l35: audio-codec@20 { + compatible = "cirrus,cs35l35"; + reg = <0x20>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&axi_gpio 54 1>; + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + cirrus,boost-ctl = <0x41>; + + cirrus,stereo-config { + cirrus,audio-channel = <0x00>; + cirrus,advisory-channel = <0x01>; + cirrus,shared-boost; + }; + + cirrus,classh-internal-algo { + cirrus,classh-bst-overide; + cirrus,classh-bst-max-limit = <0x01>; + cirrus,classh-mem-depth = <0x01>; + cirrus,classh-release-rate = <0x08>; + cirrus,classh-headroom-millivolt = <0x0B>; + cirrus,classh-wk-fet-disable = <0x01>; + cirrus,classh-wk-fet-delay = <0x04>; + cirrus,classh-wk-fet-thld = <0x01>; + cirrus,classh-vpch-auto = <0x01>; + cirrus,classh-vpch-rate = <0x02>; + cirrus,classh-vpch-man = <0x05>; + }; + + /* Depth, Location, Frame */ + cirrus,monitor-signal-format { + cirrus,imon = /bits/ 8 <0x03 0x00 0x01>; + cirrus,vmon = /bits/ 8 <0x03 0x00 0x00>; + cirrus,vpmon = /bits/ 8 <0x03 0x04 0x00>; + cirrus,vbstmon = /bits/ 8 <0x03 0x04 0x01>; + cirrus,vpbrstat = /bits/ 8 <0x00 0x04 0x00>; + cirrus,zerofill = /bits/ 8 <0x00 0x00 0x00>; + }; + +}; -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: cs35l35: Add support for Cirrus CS35L35 Amplifier
by Li Xu 13 Dec '16

13 Dec '16

Add driver support for Cirrus Logic CS35L35 boosted speaker amplifier Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- include/sound/cs35l35.h | 104 ++++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l35.c | 1355 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l35.h | 284 ++++++++++ 5 files changed, 1750 insertions(+) create mode 100644 include/sound/cs35l35.h create mode 100644 sound/soc/codecs/cs35l35.c create mode 100644 sound/soc/codecs/cs35l35.h diff --git a/include/sound/cs35l35.h b/include/sound/cs35l35.h new file mode 100644 index 0000000..005a813 --- /dev/null +++ b/include/sound/cs35l35.h @@ -0,0 +1,104 @@ +/* + * linux/sound/cs35l35.h -- Platform data for CS35l35 + * + * Copyright (c) 2016 Cirrus Logic Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __CS35L35_H +#define __CS35L35_H + +struct classh_cfg { + /* + * Class H Algorithm Control Variables + * You can either have it done + * automatically or you can adjust + * these variables for tuning + * + * if you do not enable the internal algorithm + * you will get a set of mixer controls for + * Class H tuning + * + * Section 4.3 of the datasheet + */ + /* Internal ClassH Algorithm */ + bool classh_bst_override; + bool classh_algo_enable; + int classh_bst_max_limit; + int classh_mem_depth; + int classh_release_rate; + int classh_headroom; + int classh_wk_fet_disable; + int classh_wk_fet_delay; + int classh_wk_fet_thld; + int classh_vpch_auto; + int classh_vpch_rate; + int classh_vpch_man; +}; + +struct monitor_cfg { + /* + * Signal Monitor Data + * highly configurable signal monitoring + * data positioning and different types of + * monitoring data. + * + * Section 4.8.2 - 4.8.4 of the datasheet + */ + bool is_present; + bool imon_specs; + bool vmon_specs; + bool vpmon_specs; + bool vbstmon_specs; + bool vpbrstat_specs; + bool zerofill_specs; + u8 imon_dpth; + u8 imon_loc; + u8 imon_frm; + u8 vmon_dpth; + u8 vmon_loc; + u8 vmon_frm; + u8 vpmon_dpth; + u8 vpmon_loc; + u8 vpmon_frm; + u8 vbstmon_dpth; + u8 vbstmon_loc; + u8 vbstmon_frm; + u8 vpbrstat_dpth; + u8 vpbrstat_loc; + u8 vpbrstat_frm; + u8 zerofill_dpth; + u8 zerofill_loc; + u8 zerofill_frm; +}; + +struct cs35l35_platform_data { + + /* Stereo (2 Device) */ + bool stereo; + /* serial port drive strength */ + int sp_drv_str; + /* Boost Power Down with FET */ + bool bst_pdn_fet_on; + /* Boost Voltage : used if ClassH Algo Enabled */ + int bst_vctl; + /* Boost Converter Peak Current CTRL */ + int bst_ipk; + /* Amp Gain Zero Cross */ + bool gain_zc; + /* Audio Input Location */ + int aud_channel; + /* Advisory Input Location */ + int adv_channel; + /* Shared Boost for stereo */ + bool shared_bst; + /* ClassH Algorithm */ + struct classh_cfg classh_algo; + /* Monitor Config */ + struct monitor_cfg mon_cfg; +}; + +#endif /* __CS35L35_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9e1718a..3fd0a08 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -49,6 +49,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS35L32 if I2C select SND_SOC_CS35L33 if I2C select SND_SOC_CS35L34 if I2C + select SND_SOC_CS35L35 if I2C select SND_SOC_CS42L42 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT @@ -407,6 +408,10 @@ config SND_SOC_CS35L34 tristate "Cirrus Logic CS35L34 CODEC" depends on I2C +config SND_SOC_CS35L35 + tristate "Cirrus Logic CS35L35 CODEC" + depends on I2C + config SND_SOC_CS42L42 tristate "Cirrus Logic CS42L42 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7e1dad7..a5622f6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -39,6 +39,7 @@ snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o +snd-soc-cs35l35-objs := cs35l35.o snd-soc-cs42l42-objs := cs42l42.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o @@ -268,6 +269,7 @@ obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o +obj-$(CONFIG_SND_SOC_CS35L35) += snd-soc-cs35l35.o obj-$(CONFIG_SND_SOC_CS42L42) += snd-soc-cs42l42.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c new file mode 100644 index 0000000..b05e900 --- /dev/null +++ b/sound/soc/codecs/cs35l35.c @@ -0,0 +1,1355 @@ +/* + * cs35l35.c -- CS35L35 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Li Xu <li.xu(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs35l35.h> +#include <linux/of_irq.h> +#include <linux/completion.h> + +#include "cs35l35.h" + +static const struct reg_default cs35l35_reg[] = { + {CS35L35_PWRCTL1, 0x01}, + {CS35L35_PWRCTL2, 0x11}, + {CS35L35_PWRCTL3, 0x00}, + {CS35L35_CLK_CTL1, 0x04}, + {CS35L35_CLK_CTL2, 0x10}, + {CS35L35_CLK_CTL3, 0xCF}, + {CS35L35_SP_FMT_CTL1, 0x20}, + {CS35L35_SP_FMT_CTL2, 0x00}, + {CS35L35_SP_FMT_CTL3, 0x02}, + {CS35L35_MAG_COMP_CTL, 0x00}, + {CS35L35_AMP_INP_DRV_CTL, 0x01}, + {CS35L35_AMP_DIG_VOL_CTL, 0x12}, + {CS35L35_AMP_DIG_VOL, 0x00}, + {CS35L35_ADV_DIG_VOL, 0x00}, + {CS35L35_PROTECT_CTL, 0x06}, + {CS35L35_AMP_GAIN_AUD_CTL, 0x13}, + {CS35L35_AMP_GAIN_PDM_CTL, 0x00}, + {CS35L35_AMP_GAIN_ADV_CTL, 0x00}, + {CS35L35_GPI_CTL, 0x00}, + {CS35L35_BST_CVTR_V_CTL, 0x00}, + {CS35L35_BST_PEAK_I, 0x07}, + {CS35L35_BST_RAMP_CTL, 0x85}, + {CS35L35_BST_CONV_COEF_1, 0x20}, + {CS35L35_BST_CONV_COEF_2, 0x20}, + {CS35L35_BST_CONV_SLOPE_COMP, 0x47}, + {CS35L35_BST_CONV_SW_FREQ, 0x04}, + {CS35L35_CLASS_H_CTL, 0x0B}, + {CS35L35_CLASS_H_HEADRM_CTL, 0x0B}, + {CS35L35_CLASS_H_RELEASE_RATE, 0x08}, + {CS35L35_CLASS_H_FET_DRIVE_CTL, 0x41}, + {CS35L35_CLASS_H_VP_CTL, 0xC5}, + {CS35L35_VPBR_CTL, 0x0A}, + {CS35L35_VPBR_VOL_CTL, 0x09}, + {CS35L35_VPBR_TIMING_CTL, 0x6A}, + {CS35L35_VPBR_MODE_VOL_CTL, 0x00}, + {CS35L35_SPKR_MON_CTL, 0xC0}, + {CS35L35_IMON_SCALE_CTL, 0x30}, + {CS35L35_AUDIN_RXLOC_CTL, 0x00}, + {CS35L35_ADVIN_RXLOC_CTL, 0x80}, + {CS35L35_VMON_TXLOC_CTL, 0x00}, + {CS35L35_IMON_TXLOC_CTL, 0x80}, + {CS35L35_VPMON_TXLOC_CTL, 0x04}, + {CS35L35_VBSTMON_TXLOC_CTL, 0x84}, + {CS35L35_VPBR_STATUS_TXLOC_CTL, 0x04}, + {CS35L35_ZERO_FILL_LOC_CTL, 0x00}, + {CS35L35_AUDIN_DEPTH_CTL, 0x0F}, + {CS35L35_SPKMON_DEPTH_CTL, 0x0F}, + {CS35L35_SUPMON_DEPTH_CTL, 0x0F}, + {CS35L35_ZEROFILL_DEPTH_CTL, 0x00}, + {CS35L35_MULT_DEV_SYNCH1, 0x02}, + {CS35L35_MULT_DEV_SYNCH2, 0x80}, + {CS35L35_PROT_RELEASE_CTL, 0x00}, + {CS35L35_DIAG_MODE_REG_LOCK, 0x00}, + {CS35L35_DIAG_MODE_CTL_1, 0x40}, + {CS35L35_DIAG_MODE_CTL_2, 0x00}, + {CS35L35_INT_MASK_1, 0xFF}, + {CS35L35_INT_MASK_2, 0xFF}, + {CS35L35_INT_MASK_3, 0xFF}, + {CS35L35_INT_MASK_4, 0xFF}, + +}; + +static bool cs35l35_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L35_DEVID_AB ... CS35L35_REV_ID: + case CS35L35_INT_STATUS_1: + case CS35L35_INT_STATUS_2: + case CS35L35_INT_STATUS_3: + case CS35L35_INT_STATUS_4: + case CS35L35_PLL_STATUS: + case CS35L35_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l35_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L35_DEVID_AB ... CS35L35_PWRCTL3: + case CS35L35_CLK_CTL1 ... CS35L35_SP_FMT_CTL3: + case CS35L35_MAG_COMP_CTL ... CS35L35_AMP_GAIN_AUD_CTL: + case CS35L35_AMP_GAIN_PDM_CTL ... CS35L35_BST_PEAK_I: + case CS35L35_BST_RAMP_CTL ... CS35L35_BST_CONV_SW_FREQ: + case CS35L35_CLASS_H_CTL ... CS35L35_CLASS_H_VP_CTL: + case CS35L35_CLASS_H_STATUS: + case CS35L35_VPBR_CTL ... CS35L35_VPBR_MODE_VOL_CTL: + case CS35L35_VPBR_ATTEN_STATUS: + case CS35L35_SPKR_MON_CTL: + case CS35L35_IMON_SCALE_CTL ... CS35L35_ZEROFILL_DEPTH_CTL: + case CS35L35_MULT_DEV_SYNCH1 ... CS35L35_PROT_RELEASE_CTL: + case CS35L35_DIAG_MODE_REG_LOCK ... CS35L35_DIAG_MODE_CTL_2: + case CS35L35_INT_MASK_1 ... CS35L35_PLL_STATUS: + case CS35L35_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l35_precious_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L35_INT_STATUS_1: + case CS35L35_INT_STATUS_2: + case CS35L35_INT_STATUS_3: + case CS35L35_INT_STATUS_4: + case CS35L35_PLL_STATUS: + case CS35L35_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static int cs35l35_sdin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MCLK_DIS_MASK, 0 << CS35L35_MCLK_DIS_SHIFT); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_DISCHG_FILT_MASK, 0 << CS35L35_DISCHG_FILT_SHIFT); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL_MASK, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL_MASK, 1); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_DISCHG_FILT_MASK, 1 << CS35L35_DISCHG_FILT_SHIFT); + + ret = wait_for_completion_timeout(&cs35l35->pdn_done, + msecs_to_jiffies(100)); + if (ret == 0) { + pr_err("TIMEOUT PDN_DONE did not complete in 100ms\n"); + ret = -ETIMEDOUT; + } + + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MCLK_DIS_MASK, 1 << CS35L35_MCLK_DIS_SHIFT); + break; + default: + pr_err("Invalid event = 0x%x\n", event); + ret = -EINVAL; + } + return ret; +} + +static int cs35l35_main_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + unsigned int reg[4]; + int i; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (cs35l35->pdata.bst_pdn_fet_on) + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, 0 << CS35L35_PDN_BST_FETON_SHIFT); + else + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, 0 << CS35L35_PDN_BST_FETOFF_SHIFT); + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_MUTE_MASK, 0 << CS35L35_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_POST_PMU: + usleep_range(5000, 5100); + /* If PDM mode we must use VP + * for Voltage control + */ + if (cs35l35->pdm_mode) + regmap_update_bits(cs35l35->regmap, + CS35L35_BST_CVTR_V_CTL, CS35L35_BST_CTL_MASK, + 0 << CS35L35_BST_CTL_SHIFT); + for (i = 0; i < 2; i++) + regmap_bulk_read(cs35l35->regmap, CS35L35_INT_STATUS_1, + &reg, ARRAY_SIZE(reg)); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_MUTE_MASK, 1 << CS35L35_AMP_MUTE_SHIFT); + if (cs35l35->pdata.bst_pdn_fet_on) + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, 1 << CS35L35_PDN_BST_FETON_SHIFT); + else + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, 1 << CS35L35_PDN_BST_FETOFF_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + usleep_range(5000, 5100); + /* If PDM mode we should switch back to pdata value + * for Voltage control when we go down + */ + if (cs35l35->pdm_mode) + regmap_update_bits(cs35l35->regmap, + CS35L35_BST_CVTR_V_CTL, CS35L35_BST_CTL_MASK, + cs35l35->pdata.bst_vctl << CS35L35_BST_CTL_SHIFT); + + break; + default: + pr_err("Invalid event = 0x%x\n", event); + } + return 0; +} + +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 50, 0); + +static const struct snd_kcontrol_new cs35l35_aud_controls[] = { + SOC_SINGLE_SX_TLV("Digital Audio Volume", CS35L35_AMP_DIG_VOL, + 0, 0x34, 0xE4, dig_vol_tlv), + SOC_SINGLE_TLV("AMP Audio Gain", CS35L35_AMP_GAIN_AUD_CTL, 0, 19, 0, + amp_gain_tlv), + SOC_SINGLE_TLV("AMP PDM Gain", CS35L35_AMP_GAIN_PDM_CTL, 0, 19, 0, + amp_gain_tlv), +}; + +static const struct snd_kcontrol_new cs35l35_adv_controls[] = { + SOC_SINGLE_SX_TLV("Digital Advisory Volume", CS35L35_ADV_DIG_VOL, + 0, 0x34, 0xE4, dig_vol_tlv), + SOC_SINGLE_TLV("AMP Advisory Gain", CS35L35_AMP_GAIN_ADV_CTL, 0, 19, 0, + amp_gain_tlv), +}; + +static const struct snd_soc_dapm_widget cs35l35_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN_E("SDIN", NULL, 0, CS35L35_PWRCTL3, 1, 1, + cs35l35_sdin_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_OUT("SDOUT", NULL, 0, CS35L35_PWRCTL3, 2, 1), + + SND_SOC_DAPM_OUTPUT("SPK"), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("VBST"), + SND_SOC_DAPM_INPUT("ISENSE"), + SND_SOC_DAPM_INPUT("VSENSE"), + + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L35_PWRCTL2, 7, 1), + SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L35_PWRCTL2, 6, 1), + SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L35_PWRCTL3, 3, 1), + SND_SOC_DAPM_ADC("VBSTMON ADC", NULL, CS35L35_PWRCTL3, 4, 1), + SND_SOC_DAPM_ADC("CLASS H", NULL, CS35L35_PWRCTL2, 5, 1), + + SND_SOC_DAPM_OUT_DRV_E("Main AMP", CS35L35_PWRCTL2, 0, 1, NULL, 0, + cs35l35_main_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), +}; + +static const struct snd_soc_dapm_route cs35l35_audio_map[] = { + {"VPMON ADC", NULL, "VP"}, + {"VBSTMON ADC", NULL, "VBST"}, + {"IMON ADC", NULL, "ISENSE"}, + {"VMON ADC", NULL, "VSENSE"}, + {"SDOUT", NULL, "IMON ADC"}, + {"SDOUT", NULL, "VMON ADC"}, + {"SDOUT", NULL, "VBSTMON ADC"}, + {"SDOUT", NULL, "VPMON ADC"}, + {"AMP Capture", NULL, "SDOUT"}, + + {"SDIN", NULL, "AMP Playback"}, + {"CLASS H", NULL, "SDIN"}, + {"Main AMP", NULL, "CLASS H"}, + {"SPK", NULL, "Main AMP"}, +}; + +static int cs35l35_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MS_MASK, 1 << CS35L35_MS_SHIFT); + cs35l35->slave_mode = false; + break; + case SND_SOC_DAIFMT_CBS_CFS: + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MS_MASK, 0 << CS35L35_MS_SHIFT); + cs35l35->slave_mode = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cs35l35->i2s_mode = true; + cs35l35->pdm_mode = false; + break; + case SND_SOC_DAIFMT_PDM: + cs35l35->pdm_mode = true; + cs35l35->i2s_mode = false; + break; + default: + return -EINVAL; + } + + return 0; +} + +struct cs35l35_sysclk_config { + int sysclk; + int srate; + u8 clk_cfg; +}; + +static struct cs35l35_sysclk_config cs35l35_clk_ctl[] = { + + /* SYSCLK, Sample Rate, Serial Port Cfg */ + {5644800, 44100, 0x00}, + {5644800, 88200, 0x40}, + {6144000, 48000, 0x10}, + {6144000, 96000, 0x50}, + {11289600, 44100, 0x01}, + {11289600, 88200, 0x41}, + {11289600, 176400, 0x81}, + {12000000, 44100, 0x03}, + {12000000, 48000, 0x13}, + {12000000, 88200, 0x43}, + {12000000, 96000, 0x53}, + {12000000, 176400, 0x83}, + {12000000, 192000, 0x93}, + {12288000, 48000, 0x11}, + {12288000, 96000, 0x51}, + {12288000, 192000, 0x91}, + {13000000, 44100, 0x07}, + {13000000, 48000, 0x17}, + {13000000, 88200, 0x47}, + {13000000, 96000, 0x57}, + {13000000, 176400, 0x87}, + {13000000, 192000, 0x97}, + {22579200, 44100, 0x02}, + {22579200, 88200, 0x42}, + {22579200, 176400, 0x82}, + {24000000, 44100, 0x0B}, + {24000000, 48000, 0x1B}, + {24000000, 88200, 0x4B}, + {24000000, 96000, 0x5B}, + {24000000, 176400, 0x8B}, + {24000000, 192000, 0x9B}, + {24576000, 48000, 0x12}, + {24576000, 96000, 0x52}, + {24576000, 192000, 0x92}, + {26000000, 44100, 0x0F}, + {26000000, 48000, 0x1F}, + {26000000, 88200, 0x4F}, + {26000000, 96000, 0x5F}, + {26000000, 176400, 0x8F}, + {26000000, 192000, 0x9F}, +}; + +static int cs35l35_get_clk_config(int sysclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l35_clk_ctl); i++) { + if (cs35l35_clk_ctl[i].sysclk == sysclk && + cs35l35_clk_ctl[i].srate == srate) + return cs35l35_clk_ctl[i].clk_cfg; + } + return -EINVAL; +} + +static int cs35l35_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + struct classh_cfg *classh = &cs35l35->pdata.classh_algo; + int srate = params_rate(params); + int ret = 0; + u8 sp_sclks; + int audin_format; + int errata_chk; + + int clk_ctl = cs35l35_get_clk_config(cs35l35->sysclk, srate); + + if (clk_ctl < 0) { + dev_err(codec->dev, "Invalid CLK:Rate %d:%d\n", + cs35l35->sysclk, srate); + return -EINVAL; + } + + ret = regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL2, + CS35L35_CLK_CTL2_MASK, clk_ctl); + if (ret != 0) { + dev_err(codec->dev, "Failed to set port config %d\n", ret); + return ret; + } + + /* Rev A0 Errata + * + * When configured for the weak-drive detection path (CH_WKFET_DIS = 0) + * the Class H algorithm does not enable weak-drive operation for + * nonzero values of CH_WKFET_DELAY if SP_RATE = 01 or 10 + * + */ + errata_chk = clk_ctl & CS35L35_SP_RATE_MASK; + + if (classh->classh_wk_fet_disable == 0x00 && + (errata_chk == 0x01 || errata_chk == 0x03)) { + ret = regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, CS35L35_CH_WKFET_DEL_MASK, + 0 << CS35L35_CH_WKFET_DEL_SHIFT); + if (ret != 0) { + dev_err(codec->dev, "Failed to set fet config %d\n", + ret); + return ret; + } + } + +/* + * You can pull more Monitor data from the SDOUT pin than going to SDIN + * Just make sure your SCLK is fast enough to fill the frame + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (params_width(params)) { + case 8: + audin_format = CS35L35_SDIN_DEPTH_8; + break; + case 16: + audin_format = CS35L35_SDIN_DEPTH_16; + break; + case 24: + audin_format = CS35L35_SDIN_DEPTH_24; + break; + default: + dev_err(codec->dev, "Unsupported Width %d\n", + params_width(params)); + } + regmap_update_bits(cs35l35->regmap, + CS35L35_AUDIN_DEPTH_CTL, CS35L35_AUDIN_DEPTH_MASK, + audin_format << CS35L35_AUDIN_DEPTH_SHIFT); + if (cs35l35->pdata.stereo) { + regmap_update_bits(cs35l35->regmap, + CS35L35_AUDIN_DEPTH_CTL, CS35L35_ADVIN_DEPTH_MASK, + audin_format << CS35L35_ADVIN_DEPTH_SHIFT); + } + } +/* We have to take the SCLK to derive num sclks + * to configure the CLOCK_CTL3 register correctly + */ + if ((cs35l35->sclk / srate) % 4) { + dev_err(codec->dev, "Unsupported sclk/fs ratio %d:%d\n", + cs35l35->sclk, srate); + return -EINVAL; + } + sp_sclks = ((cs35l35->sclk / srate) / 4) - 1; + + if (cs35l35->i2s_mode) { + /* Only certain ratios are supported in I2S Slave Mode */ + if (cs35l35->slave_mode) { + switch (sp_sclks) { + case CS35L35_SP_SCLKS_32FS: + case CS35L35_SP_SCLKS_48FS: + case CS35L35_SP_SCLKS_64FS: + break; + default: + dev_err(codec->dev, "ratio not supported\n"); + return -EINVAL; + }; + } else { + /* Only certain ratios supported in I2S MASTER Mode */ + switch (sp_sclks) { + case CS35L35_SP_SCLKS_32FS: + case CS35L35_SP_SCLKS_64FS: + break; + default: + dev_err(codec->dev, "ratio not supported\n"); + return -EINVAL; + }; + } + ret = regmap_update_bits(cs35l35->regmap, + CS35L35_CLK_CTL3, CS35L35_SP_SCLKS_MASK, + sp_sclks << CS35L35_SP_SCLKS_SHIFT); + if (ret != 0) { + dev_err(codec->dev, "Failed to set fsclk %d\n", ret); + return ret; + } + } + if (cs35l35->pdm_mode) { + regmap_update_bits(cs35l35->regmap, CS35L35_AMP_INP_DRV_CTL, + CS35L35_PDM_MODE_MASK, 1 << CS35L35_PDM_MODE_SHIFT); + } else { + regmap_update_bits(cs35l35->regmap, CS35L35_AMP_INP_DRV_CTL, + CS35L35_PDM_MODE_MASK, 0 << CS35L35_PDM_MODE_SHIFT); + } + return ret; +} + +static const unsigned int cs35l35_src_rates[] = { + 44100, 48000, 88200, 96000, 176400, 192000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l35_constraints = { + .count = ARRAY_SIZE(cs35l35_src_rates), + .list = cs35l35_src_rates, +}; + +static int cs35l35_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs35l35_constraints); + return 0; +} + +static const unsigned int cs35l35_pdm_rates[] = { + 44100, 48000, 88200, 96000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l35_pdm_constraints = { + .count = ARRAY_SIZE(cs35l35_pdm_rates), + .list = cs35l35_pdm_rates, +}; + +static int cs35l35_pdm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs35l35_pdm_constraints); + return 0; +} + +static int cs35l35_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + + /* Need the SCLK Frequency */ + cs35l35->sclk = freq; + + return 0; +} + +static const struct snd_soc_dai_ops cs35l35_ops = { + .startup = cs35l35_pcm_startup, + .set_fmt = cs35l35_set_dai_fmt, + .hw_params = cs35l35_pcm_hw_params, + .set_sysclk = cs35l35_dai_set_sysclk, +}; + +static const struct snd_soc_dai_ops cs35l35_pdm_ops = { + .startup = cs35l35_pdm_startup, + .set_fmt = cs35l35_set_dai_fmt, + .hw_params = cs35l35_pcm_hw_params, + .set_sysclk = cs35l35_dai_set_sysclk, +}; + +static struct snd_soc_dai_driver cs35l35_dai[] = { + { + .name = "cs35l35-pcm", + .id = 0, + .playback = { + .stream_name = "AMP Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L35_FORMATS, + }, + .capture = { + .stream_name = "AMP Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L35_FORMATS, + }, + .ops = &cs35l35_ops, + .symmetric_rates = 1, + }, + { + .name = "cs35l35-pdm", + .id = 1, + .playback = { + .stream_name = "PDM Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L35_FORMATS, + }, + .ops = &cs35l35_pdm_ops, + }, +}; + +static int cs35l35_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, + int dir) +{ + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + int clksrc; + int ret = 0; + + switch (clk_id) { + case 0: + clksrc = CS35L35_CLK_SOURCE_MCLK; + break; + case 1: + clksrc = CS35L35_CLK_SOURCE_SCLK; + break; + case 2: + clksrc = CS35L35_CLK_SOURCE_PDM; + break; + default: + dev_err(codec->dev, "Invalid CLK Source\n"); + return -EINVAL; + }; + + switch (freq) { + case 5644800: + case 6144000: + case 11289600: + case 12000000: + case 12288000: + case 13000000: + case 22579200: + case 24000000: + case 24576000: + case 26000000: + cs35l35->sysclk = freq; + break; + default: + dev_err(codec->dev, "Invalid CLK Frequency\n"); + return -EINVAL; + } + + ret = regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_CLK_SOURCE_MASK, clksrc << CS35L35_CLK_SOURCE_SHIFT); + if (ret != 0) { + dev_err(codec->dev, "Failed to set sysclk %d\n", ret); + return ret; + } + + return ret; +} + +static int cs35l35_codec_probe(struct snd_soc_codec *codec) +{ + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + struct classh_cfg *classh = &cs35l35->pdata.classh_algo; + struct monitor_cfg *monitor_config = &cs35l35->pdata.mon_cfg; + int ret; + + /* Set Platform Data */ + if (cs35l35->pdata.bst_vctl) + regmap_update_bits(cs35l35->regmap, CS35L35_BST_CVTR_V_CTL, + CS35L35_BST_CTL_MASK, cs35l35->pdata.bst_vctl); + + if (cs35l35->pdata.bst_ipk) + regmap_update_bits(cs35l35->regmap, CS35L35_BST_PEAK_I, + CS35L35_BST_IPK_MASK, + cs35l35->pdata.bst_ipk << CS35L35_BST_IPK_SHIFT); + + if (cs35l35->pdata.gain_zc) + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_GAIN_ZC_MASK, + cs35l35->pdata.gain_zc << CS35L35_AMP_GAIN_ZC_SHIFT); + + if (cs35l35->pdata.aud_channel) + regmap_update_bits(cs35l35->regmap, + CS35L35_AUDIN_RXLOC_CTL, + CS35L35_AUD_IN_LR_MASK, + cs35l35->pdata.aud_channel << CS35L35_AUD_IN_LR_SHIFT); + + if (cs35l35->pdata.stereo) { + regmap_update_bits(cs35l35->regmap, + CS35L35_ADVIN_RXLOC_CTL, CS35L35_ADV_IN_LR_MASK, + cs35l35->pdata.adv_channel << CS35L35_ADV_IN_LR_SHIFT); + if (cs35l35->pdata.shared_bst) + regmap_update_bits(cs35l35->regmap, CS35L35_CLASS_H_CTL, + CS35L35_CH_STEREO_MASK, 1 << CS35L35_CH_STEREO_SHIFT); + ret = snd_soc_add_codec_controls(codec, cs35l35_adv_controls, + ARRAY_SIZE(cs35l35_adv_controls)); + if (ret) + return ret; + } + + if (cs35l35->pdata.sp_drv_str) + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_SP_DRV_MASK, + cs35l35->pdata.sp_drv_str << CS35L35_SP_DRV_SHIFT); + + if (classh->classh_algo_enable) { + if (classh->classh_bst_override) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_CTL, CS35L35_CH_BST_OVR_MASK, + classh->classh_bst_override << CS35L35_CH_BST_OVR_SHIFT); + if (classh->classh_bst_max_limit) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_CTL, CS35L35_CH_BST_LIM_MASK, + classh->classh_bst_max_limit << CS35L35_CH_BST_LIM_SHIFT); + if (classh->classh_mem_depth) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_CTL, CS35L35_CH_MEM_DEPTH_MASK, + classh->classh_mem_depth << CS35L35_CH_MEM_DEPTH_SHIFT); + if (classh->classh_headroom) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_HEADRM_CTL, CS35L35_CH_HDRM_CTL_MASK, + classh->classh_headroom << CS35L35_CH_HDRM_CTL_SHIFT); + if (classh->classh_release_rate) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_RELEASE_RATE, CS35L35_CH_REL_RATE_MASK, + classh->classh_release_rate << CS35L35_CH_REL_RATE_SHIFT); + if (classh->classh_wk_fet_disable) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, CS35L35_CH_WKFET_DIS_MASK, + classh->classh_wk_fet_disable << CS35L35_CH_WKFET_DIS_SHIFT); + if (classh->classh_wk_fet_delay) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, CS35L35_CH_WKFET_DEL_MASK, + classh->classh_wk_fet_delay << CS35L35_CH_WKFET_DEL_SHIFT); + if (classh->classh_wk_fet_thld) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, CS35L35_CH_WKFET_THLD_MASK, + classh->classh_wk_fet_thld << CS35L35_CH_WKFET_THLD_SHIFT); + if (classh->classh_vpch_auto) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_VP_CTL, CS35L35_CH_VP_AUTO_MASK, + classh->classh_vpch_auto << CS35L35_CH_VP_AUTO_SHIFT); + if (classh->classh_vpch_rate) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_VP_CTL, CS35L35_CH_VP_RATE_MASK, + classh->classh_vpch_rate << CS35L35_CH_VP_RATE_SHIFT); + if (classh->classh_vpch_man) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_VP_CTL, CS35L35_CH_VP_MAN_MASK, + classh->classh_vpch_man << CS35L35_CH_VP_MAN_SHIFT); + } + + if (monitor_config->is_present) { + if (monitor_config->vmon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SPKMON_DEPTH_CTL, CS35L35_VMON_DEPTH_MASK, + monitor_config->vmon_dpth << CS35L35_VMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VMON_TXLOC_CTL, CS35L35_MON_TXLOC_MASK, + monitor_config->vmon_loc << CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VMON_TXLOC_CTL, CS35L35_MON_FRM_MASK, + monitor_config->vmon_frm << CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->imon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SPKMON_DEPTH_CTL, CS35L35_IMON_DEPTH_MASK, + monitor_config->imon_dpth << CS35L35_IMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_IMON_TXLOC_CTL, CS35L35_MON_TXLOC_MASK, + monitor_config->imon_loc << CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_IMON_TXLOC_CTL, CS35L35_MON_FRM_MASK, + monitor_config->imon_frm << CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->vpmon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, CS35L35_VPMON_DEPTH_MASK, + monitor_config->vpmon_dpth << CS35L35_VPMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPMON_TXLOC_CTL, CS35L35_MON_TXLOC_MASK, + monitor_config->vpmon_loc << CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPMON_TXLOC_CTL, CS35L35_MON_FRM_MASK, + monitor_config->vpmon_frm << CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->vbstmon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, CS35L35_VBSTMON_DEPTH_MASK, + monitor_config->vpmon_dpth << CS35L35_VBSTMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VBSTMON_TXLOC_CTL, CS35L35_MON_TXLOC_MASK, + monitor_config->vbstmon_loc << CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VBSTMON_TXLOC_CTL, CS35L35_MON_FRM_MASK, + monitor_config->vbstmon_frm << CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->vpbrstat_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, CS35L35_VPBRSTAT_DEPTH_MASK, + monitor_config->vpbrstat_dpth << CS35L35_VPBRSTAT_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPBR_STATUS_TXLOC_CTL, CS35L35_MON_TXLOC_MASK, + monitor_config->vpbrstat_loc << CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPBR_STATUS_TXLOC_CTL, CS35L35_MON_FRM_MASK, + monitor_config->vpbrstat_frm << CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->zerofill_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, CS35L35_ZEROFILL_DEPTH_MASK, + monitor_config->zerofill_dpth << CS35L35_ZEROFILL_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_ZERO_FILL_LOC_CTL, CS35L35_MON_TXLOC_MASK, + monitor_config->zerofill_loc << CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_ZERO_FILL_LOC_CTL, CS35L35_MON_FRM_MASK, + monitor_config->zerofill_frm << CS35L35_MON_FRM_SHIFT); + } + } + + return ret; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l35 = { + .probe = cs35l35_codec_probe, + .set_sysclk = cs35l35_codec_set_sysclk, + .component_driver = { + .controls = cs35l35_aud_controls, + .num_controls = ARRAY_SIZE(cs35l35_aud_controls), + .dapm_widgets = cs35l35_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l35_dapm_widgets), + + .dapm_routes = cs35l35_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l35_audio_map), + }, +}; + +static struct regmap_config cs35l35_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L35_MAX_REGISTER, + .reg_defaults = cs35l35_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l35_reg), + .volatile_reg = cs35l35_volatile_register, + .readable_reg = cs35l35_readable_register, + .precious_reg = cs35l35_precious_register, + .cache_type = REGCACHE_RBTREE, +}; + +static irqreturn_t cs35l35_irq(int irq, void *data) +{ + struct cs35l35_private *cs35l35 = data; + unsigned int sticky1, sticky2, sticky3, sticky4; + unsigned int mask1, mask2, mask3, mask4, current1; + + /* ack the irq by reading all status registers */ + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_4, &sticky4); + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_3, &sticky3); + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_2, &sticky2); + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_1, &sticky1); + + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_4, &mask4); + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_3, &mask3); + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_2, &mask2); + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_1, &mask1); + + /* Check to see if unmasked bits are active */ + if (!(sticky1 & ~mask1) && !(sticky2 & ~mask2) && !(sticky3 & ~mask3) + && !(sticky4 & ~mask4)) + return IRQ_NONE; + + if (sticky2 & CS35L35_PDN_DONE) + complete(&cs35l35->pdn_done); + + /* read the current values */ + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_1, &current1); + + /* handle the interrupts */ + if (sticky1 & CS35L35_CAL_ERR) { + pr_err("%s : Calibration Error\n", __func__); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_CAL_ERR)) { + pr_debug("%s : Cal error release\n", __func__); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_CAL_ERR_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_CAL_ERR_RLS, + CS35L35_CAL_ERR_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_CAL_ERR_RLS, 0); + } + } + + if (sticky1 & CS35L35_AMP_SHORT) { + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_AMP_SHORT)) { + pr_debug("%s :Amp short error release\n", __func__); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_SHORT_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_SHORT_RLS, + CS35L35_SHORT_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_SHORT_RLS, 0); + } + } + + if (sticky1 & CS35L35_OTW) { + pr_err("%s : Over temperature warning\n", __func__); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_OTW)) { + pr_debug("%s : Over temperature warning release\n", + __func__); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_OTW_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_OTW_RLS, + CS35L35_OTW_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_OTW_RLS, 0); + } + } + + if (sticky1 & CS35L35_OTE) { + pr_crit("%s : Over temperature error\n", __func__); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_OTE)) { + pr_debug("%s : Over temperature error release\n", + __func__); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_OTE_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_OTE_RLS, + CS35L35_OTE_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, CS35L35_OTE_RLS, 0); + } + } + + if (sticky3 & CS35L35_BST_HIGH) { + pr_crit("%s : VBST error: powering off!\n", __func__); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_AMP, CS35L35_PDN_AMP); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL, CS35L35_PDN_ALL); + } + + if (sticky3 & CS35L35_LBST_SHORT) { + pr_crit("%s : LBST error: powering off!\n", __func__); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_AMP, CS35L35_PDN_AMP); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL, CS35L35_PDN_ALL); + } + + if (sticky2 & CS35L35_VPBR_ERR) + pr_err("%s : Error: Reactive Brownout\n", __func__); + + if (sticky4 & CS35L35_VMON_OVFL) + pr_err("%s : Error: VMON overflow\n", __func__); + + if (sticky4 & CS35L35_IMON_OVFL) + pr_err("%s : Error: IMON overflow\n", __func__); + + return IRQ_HANDLED; +} + + +static int cs35l35_handle_of_data(struct i2c_client *i2c_client, + struct cs35l35_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + struct device_node *classh, *signal_format; + struct classh_cfg *classh_config = &pdata->classh_algo; + struct monitor_cfg *monitor_config = &pdata->mon_cfg; + unsigned int val32 = 0; + u8 monitor_array[3]; + int ret = 0; + + if (!np) + return 0; + + pdata->bst_pdn_fet_on = of_property_read_bool(np, + "cirrus,boost-pdn-fet-on"); + + if (of_property_read_u32(np, "cirrus,boost-ctl-millivolt", &val32) >= 0) + pdata->bst_vctl = val32; + + if (of_property_read_u32(np, "cirrus,boost-ipk-milliamp", &val32) >= 0) + pdata->bst_ipk = val32; + + if (of_property_read_u32(np, "cirrus,sp-drv-strength", &val32) >= 0) + pdata->sp_drv_str = val32; + + pdata->stereo = of_property_read_bool(np, "cirrus,stereo-config"); + + if (pdata->stereo) { + if (of_property_read_u32(np, "cirrus,audio-channel", &val32) >= 0) + pdata->aud_channel = val32; + if (of_property_read_u32(np, "cirrus,advisory-channel", + &val32) >= 0) + pdata->adv_channel = val32; + pdata->shared_bst = of_property_read_bool(np, + "cirrus,shared-boost"); + } + + pdata->gain_zc = of_property_read_bool(np, "cirrus,amp-gain-zc"); + + classh = of_get_child_by_name(np, "cirrus,classh-internal-algo"); + classh_config->classh_algo_enable = classh ? true : false; + + if (classh_config->classh_algo_enable) { + classh_config->classh_bst_override = + of_property_read_bool(np, "cirrus,classh-bst-overide"); + + if (of_property_read_u32(classh, "cirrus,classh-bst-max-limit", + &val32) >= 0) + classh_config->classh_bst_max_limit = val32; + if (of_property_read_u32(classh, "cirrus,classh-mem-depth", + &val32) >= 0) + classh_config->classh_mem_depth = val32; + if (of_property_read_u32(classh, "cirrus,classh-release-rate", + &val32) >= 0) + classh_config->classh_release_rate = val32; + if (of_property_read_u32(classh, "cirrus,classh-headroom", + &val32) >= 0) + classh_config->classh_headroom = val32; + if (of_property_read_u32(classh, "cirrus,classh-wk-fet-disable", + &val32) >= 0) + classh_config->classh_wk_fet_disable = val32; + if (of_property_read_u32(classh, "cirrus,classh-wk-fet-delay", + &val32) >= 0) + classh_config->classh_wk_fet_delay = val32; + if (of_property_read_u32(classh, "cirrus,classh-wk-fet-thld", + &val32) >= 0) + classh_config->classh_wk_fet_thld = val32; + if (of_property_read_u32(classh, "cirrus,classh-vpch-auto", + &val32) >= 0) + classh_config->classh_vpch_auto = val32; + if (of_property_read_u32(classh, "cirrus,classh-vpch-rate", + &val32) >= 0) + classh_config->classh_vpch_rate = val32; + if (of_property_read_u32(classh, "cirrus,classh-vpch-man", + &val32) >= 0) + classh_config->classh_vpch_man = val32; + } + of_node_put(classh); + + /* frame depth location */ + signal_format = of_get_child_by_name(np, "cirrus,monitor-signal-format"); + monitor_config->is_present = signal_format ? true : false; + if (monitor_config->is_present) { + ret = of_property_read_u8_array(signal_format, "cirrus,imon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->imon_specs = true; + monitor_config->imon_dpth = monitor_array[0]; + monitor_config->imon_loc = monitor_array[1]; + monitor_config->imon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vmon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vmon_specs = true; + monitor_config->vmon_dpth = monitor_array[0]; + monitor_config->vmon_loc = monitor_array[1]; + monitor_config->vmon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vpmon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vpmon_specs = true; + monitor_config->vpmon_dpth = monitor_array[0]; + monitor_config->vpmon_loc = monitor_array[1]; + monitor_config->vpmon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vbstmon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vbstmon_specs = true; + monitor_config->vbstmon_dpth = monitor_array[0]; + monitor_config->vbstmon_loc = monitor_array[1]; + monitor_config->vbstmon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vpbrstat", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vpbrstat_specs = true; + monitor_config->vpbrstat_dpth = monitor_array[0]; + monitor_config->vpbrstat_loc = monitor_array[1]; + monitor_config->vpbrstat_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,zerofill", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->zerofill_specs = true; + monitor_config->zerofill_dpth = monitor_array[0]; + monitor_config->zerofill_loc = monitor_array[1]; + monitor_config->zerofill_frm = monitor_array[2]; + } + } + of_node_put(signal_format); + + return 0; +} + +/* Errata Rev A0 */ +static const struct reg_sequence cs35l35_errata_patch[] = { + + { 0x7F, 0x99 }, + { 0x00, 0x99 }, + { 0x52, 0x22 }, + { 0x04, 0x14 }, + { 0x6D, 0x44 }, + { 0x24, 0x10 }, + { 0x58, 0xC4 }, + { 0x00, 0x98 }, + { 0x18, 0x08 }, + { 0x00, 0x00 }, + { 0x7F, 0x00 }, +}; + +static int cs35l35_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l35_private *cs35l35; + struct cs35l35_platform_data *pdata = + dev_get_platdata(&i2c_client->dev); + int i; + int ret; + unsigned int devid = 0; + unsigned int reg; + + cs35l35 = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs35l35_private), + GFP_KERNEL); + if (!cs35l35) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs35l35); + cs35l35->regmap = devm_regmap_init_i2c(i2c_client, &cs35l35_regmap); + if (IS_ERR(cs35l35->regmap)) { + ret = PTR_ERR(cs35l35->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(cs35l35_supplies); i++) + cs35l35->supplies[i].supply = cs35l35_supplies[i]; + cs35l35->num_supplies = ARRAY_SIZE(cs35l35_supplies); + + ret = devm_regulator_bulk_get(&i2c_client->dev, + cs35l35->num_supplies, + cs35l35->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request core supplies: %d\n", + ret); + return ret; + } + + if (pdata) { + cs35l35->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs35l35_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&i2c_client->dev, + "could not allocate pdata\n"); + return -ENOMEM; + } + if (i2c_client->dev.of_node) { + ret = cs35l35_handle_of_data(i2c_client, pdata); + if (ret != 0) + return ret; + + } + cs35l35->pdata = *pdata; + } + + ret = regulator_bulk_enable(cs35l35->num_supplies, + cs35l35->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable core supplies: %d\n", + ret); + return ret; + } + + /* returning NULL can be an option if in stereo mode */ + cs35l35->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs35l35->reset_gpio)) + return PTR_ERR(cs35l35->reset_gpio); + + if (cs35l35->reset_gpio) + gpiod_set_value_cansleep(cs35l35->reset_gpio, 1); + + init_completion(&cs35l35->pdn_done); + + ret = regmap_register_patch(cs35l35->regmap, cs35l35_errata_patch, + ARRAY_SIZE(cs35l35_errata_patch)); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to apply errata patch\n"); + return ret; + } + + ret = devm_request_threaded_irq(&i2c_client->dev, i2c_client->irq, NULL, + cs35l35_irq, IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs35l35", cs35l35); + if (ret != 0) { + dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + /* initialize codec */ + ret = regmap_read(cs35l35->regmap, CS35L35_DEVID_AB, &reg); + + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs35l35->regmap, CS35L35_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs35l35->regmap, CS35L35_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L35_CHIP_ID) { + dev_err(&i2c_client->dev, + "CS35L35 Device ID (%X). Expected ID %X\n", + devid, CS35L35_CHIP_ID); + ret = -ENODEV; + goto err; + } + + ret = regmap_read(cs35l35->regmap, CS35L35_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + goto err; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L35 (%x), Revision: %02X\n", devid, + ret & 0xFF); + + /* Set the INT Masks for critical errors */ + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_1, CS35L35_INT1_CRIT_MASK); + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_2, CS35L35_INT2_CRIT_MASK); + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_3, CS35L35_INT3_CRIT_MASK); + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_4, CS35L35_INT4_CRIT_MASK); + + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PWR2_PDN_MASK, CS35L35_PWR2_PDN_MASK); + + if (cs35l35->pdata.bst_pdn_fet_on) + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, 1 << CS35L35_PDN_BST_FETON_SHIFT); + else + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, 1 << CS35L35_PDN_BST_FETOFF_SHIFT); + + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL3, + CS35L35_PWR3_PDN_MASK, CS35L35_PWR3_PDN_MASK); + + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_MUTE_MASK, 1 << CS35L35_AMP_MUTE_SHIFT); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l35, cs35l35_dai, + ARRAY_SIZE(cs35l35_dai)); + if (ret < 0) { + dev_err(&i2c_client->dev, + "%s: Register codec failed\n", __func__); + goto err; + } + +err: + regulator_bulk_disable(cs35l35->num_supplies, + cs35l35->supplies); + return ret; +} + +static int cs35l35_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct of_device_id cs35l35_of_match[] = { + {.compatible = "cirrus,cs35l35"}, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l35_of_match); + +static const struct i2c_device_id cs35l35_id[] = { + {"cs35l35", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l35_id); + +static struct i2c_driver cs35l35_i2c_driver = { + .driver = { + .name = "cs35l35", + .of_match_table = cs35l35_of_match, + }, + .id_table = cs35l35_id, + .probe = cs35l35_i2c_probe, + .remove = cs35l35_i2c_remove, +}; + +module_i2c_driver(cs35l35_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L35 driver"); +MODULE_AUTHOR("Li Xu, Cirrus Logic Inc, <li.xu(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l35.h b/sound/soc/codecs/cs35l35.h new file mode 100644 index 0000000..fb02785 --- /dev/null +++ b/sound/soc/codecs/cs35l35.h @@ -0,0 +1,284 @@ +/* + * cs35l35.h -- CS35L35 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Li Xu <li.xu(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __CS35L35_H__ +#define __CS35L35_H__ + +#define CS35L35_FIRSTREG 0x01 +#define CS35L35_LASTREG 0x7E +#define CS35L35_CHIP_ID 0x00035A35 +#define CS35L35_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L35_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L35_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L35_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L35_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L35_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L35_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L35_PWRCTL3 0x08 /* Power Ctl 3 */ +#define CS35L35_CLK_CTL1 0x0A /* Clocking Ctl 1 */ +#define CS35L35_CLK_CTL2 0x0B /* Clocking Ctl 2 */ +#define CS35L35_CLK_CTL3 0x0C /* Clocking Ctl 3 */ +#define CS35L35_SP_FMT_CTL1 0x0D /* Serial Port Format CTL1 */ +#define CS35L35_SP_FMT_CTL2 0x0E /* Serial Port Format CTL2 */ +#define CS35L35_SP_FMT_CTL3 0x0F /* Serial Port Format CTL3 */ +#define CS35L35_MAG_COMP_CTL 0x13 /* Magnitude Comp CTL */ +#define CS35L35_AMP_INP_DRV_CTL 0x14 /* Amp Input Drive Ctl */ +#define CS35L35_AMP_DIG_VOL_CTL 0x15 /* Amplifier Dig Volume Ctl */ +#define CS35L35_AMP_DIG_VOL 0x16 /* Amplifier Dig Volume */ +#define CS35L35_ADV_DIG_VOL 0x17 /* Advisory Digital Volume */ +#define CS35L35_PROTECT_CTL 0x18 /* Amp Gain - Prot Ctl Param */ +#define CS35L35_AMP_GAIN_AUD_CTL 0x19 /* Amp Serial Port Gain Ctl */ +#define CS35L35_AMP_GAIN_PDM_CTL 0x1A /* Amplifier Gain PDM Ctl */ +#define CS35L35_AMP_GAIN_ADV_CTL 0x1B /* Amplifier Gain Ctl */ +#define CS35L35_GPI_CTL 0x1C /* GPI Ctl */ +#define CS35L35_BST_CVTR_V_CTL 0x1D /* Boost Conv Voltage Ctl */ +#define CS35L35_BST_PEAK_I 0x1E /* Boost Conv Peak Current */ +#define CS35L35_BST_RAMP_CTL 0x20 /* Boost Conv Soft Ramp Ctl */ +#define CS35L35_BST_CONV_COEF_1 0x21 /* Boost Conv Coefficients 1 */ +#define CS35L35_BST_CONV_COEF_2 0x22 /* Boost Conv Coefficients 2 */ +#define CS35L35_BST_CONV_SLOPE_COMP 0x23 /* Boost Conv Slope Comp */ +#define CS35L35_BST_CONV_SW_FREQ 0x24 /* Boost Conv L BST SW Freq */ +#define CS35L35_CLASS_H_CTL 0x30 /* CLS H Control */ +#define CS35L35_CLASS_H_HEADRM_CTL 0x31 /* CLS H Headroom Ctl */ +#define CS35L35_CLASS_H_RELEASE_RATE 0x32 /* CLS H Release Rate */ +#define CS35L35_CLASS_H_FET_DRIVE_CTL 0x33 /* CLS H Weak FET Drive Ctl */ +#define CS35L35_CLASS_H_VP_CTL 0x34 /* CLS H VP Ctl */ +#define CS35L35_CLASS_H_STATUS 0x38 /* CLS H Status */ +#define CS35L35_VPBR_CTL 0x3A /* VPBR Ctl */ +#define CS35L35_VPBR_VOL_CTL 0x3B /* VPBR Volume Ctl */ +#define CS35L35_VPBR_TIMING_CTL 0x3C /* VPBR Timing Ctl */ +#define CS35L35_VPBR_MODE_VOL_CTL 0x3D /* VPBR Mode/Attack Vol Ctl */ +#define CS35L35_VPBR_ATTEN_STATUS 0x4B /* VPBR Attenuation Status */ +#define CS35L35_SPKR_MON_CTL 0x4E /* Speaker Monitoring Ctl */ +#define CS35L35_IMON_SCALE_CTL 0x51 /* IMON Scale Ctl */ +#define CS35L35_AUDIN_RXLOC_CTL 0x52 /* Audio Input RX Loc Ctl */ +#define CS35L35_ADVIN_RXLOC_CTL 0x53 /* Advisory Input RX Loc Ctl */ +#define CS35L35_VMON_TXLOC_CTL 0x54 /* VMON TX Loc Ctl */ +#define CS35L35_IMON_TXLOC_CTL 0x55 /* IMON TX Loc Ctl */ +#define CS35L35_VPMON_TXLOC_CTL 0x56 /* VPMON TX Loc Ctl */ +#define CS35L35_VBSTMON_TXLOC_CTL 0x57 /* VBSTMON TX Loc Ctl */ +#define CS35L35_VPBR_STATUS_TXLOC_CTL 0x58 /* VPBR Status TX Loc Ctl */ +#define CS35L35_ZERO_FILL_LOC_CTL 0x59 /* Zero Fill Loc Ctl */ +#define CS35L35_AUDIN_DEPTH_CTL 0x5A /* Audio Input Depth Ctl */ +#define CS35L35_SPKMON_DEPTH_CTL 0x5B /* SPK Mon Output Depth Ctl */ +#define CS35L35_SUPMON_DEPTH_CTL 0x5C /* Supply Mon Out Depth Ctl */ +#define CS35L35_ZEROFILL_DEPTH_CTL 0x5D /* Zero Fill Mon Output Ctl */ +#define CS35L35_MULT_DEV_SYNCH1 0x62 /* Multidevice Synch */ +#define CS35L35_MULT_DEV_SYNCH2 0x63 /* Multidevice Synch 2 */ +#define CS35L35_PROT_RELEASE_CTL 0x64 /* Protection Release Ctl */ +#define CS35L35_DIAG_MODE_REG_LOCK 0x68 /* Diagnostic Mode Reg Lock */ +#define CS35L35_DIAG_MODE_CTL_1 0x69 /* Diagnostic Mode Ctl 1 */ +#define CS35L35_DIAG_MODE_CTL_2 0x6A /* Diagnostic Mode Ctl 2 */ +#define CS35L35_INT_MASK_1 0x70 /* Interrupt Mask 1 */ +#define CS35L35_INT_MASK_2 0x71 /* Interrupt Mask 2 */ +#define CS35L35_INT_MASK_3 0x72 /* Interrupt Mask 3 */ +#define CS35L35_INT_MASK_4 0x73 /* Interrupt Mask 4 */ +#define CS35L35_INT_STATUS_1 0x74 /* Interrupt Status 1 */ +#define CS35L35_INT_STATUS_2 0x75 /* Interrupt Status 2 */ +#define CS35L35_INT_STATUS_3 0x76 /* Interrupt Status 3 */ +#define CS35L35_INT_STATUS_4 0x77 /* Interrupt Status 4 */ +#define CS35L35_PLL_STATUS 0x78 /* PLL Status */ +#define CS35L35_OTP_TRIM_STATUS 0x7E /* OTP Trim Status */ + +#define CS35L35_MAX_REGISTER 0x7F + +/* CS35L35_PWRCTL1 */ +#define CS35L35_SFT_RST 0x80 +#define CS35L35_DISCHG_FLT 0x02 +#define CS35L35_PDN_ALL 0x01 + +/* CS35L35_PWRCTL2 */ +#define CS35L35_PDN_VMON 0x80 +#define CS35L35_PDN_IMON 0x40 +#define CS35L35_PDN_CLASSH 0x20 +#define CS35L35_PDN_VPBR 0x10 +#define CS35L35_PDN_BST 0x04 +#define CS35L35_PDN_AMP 0x01 + +/* CS35L35_PWRCTL3 */ +#define CS35L35_PDN_VBSTMON_OUT 0x10 +#define CS35L35_PDN_VMON_OUT 0x08 + +#define CS35L35_AUDIN_DEPTH_MASK 0x03 +#define CS35L35_AUDIN_DEPTH_SHIFT 0 +#define CS35L35_ADVIN_DEPTH_MASK 0x12 +#define CS35L35_ADVIN_DEPTH_SHIFT 2 +#define CS35L35_SDIN_DEPTH_8 0x01 +#define CS35L35_SDIN_DEPTH_16 0x02 +#define CS35L35_SDIN_DEPTH_24 0x03 + +#define CS35L35_SDOUT_DEPTH_8 0x01 +#define CS35L35_SDOUT_DEPTH_12 0x02 +#define CS35L35_SDOUT_DEPTH_16 0x03 + +#define CS35L35_AUD_IN_LR_MASK 0x80 +#define CS35L35_AUD_IN_LR_SHIFT 7 +#define CS35L35_ADV_IN_LR_MASK 0x80 +#define CS35L35_ADV_IN_LR_SHIFT 7 +#define CS35L35_AUD_IN_LOC_MASK 0x0F +#define CS35L35_AUD_IN_LOC_SHIFT 0 +#define CS35L35_ADV_IN_LOC_MASK 0x0F +#define CS35L35_ADV_IN_LOC_SHIFT 0 + +#define CS35L35_IMON_DEPTH_MASK 0x03 +#define CS35L35_IMON_DEPTH_SHIFT 0 +#define CS35L35_VMON_DEPTH_MASK 0x0C +#define CS35L35_VMON_DEPTH_SHIFT 2 +#define CS35L35_VBSTMON_DEPTH_MASK 0x03 +#define CS35L35_VBSTMON_DEPTH_SHIFT 0 +#define CS35L35_VPMON_DEPTH_MASK 0x0C +#define CS35L35_VPMON_DEPTH_SHIFT 2 +#define CS35L35_VPBRSTAT_DEPTH_MASK 0x18 +#define CS35L35_VPBRSTAT_DEPTH_SHIFT 4 +#define CS35L35_ZEROFILL_DEPTH_MASK 0x03 +#define CS35L35_ZEROFILL_DEPTH_SHIFT 0x00 + +#define CS35L35_MON_TXLOC_MASK 0x3F +#define CS35L35_MON_TXLOC_SHIFT 0 +#define CS35L35_MON_FRM_MASK 0x80 +#define CS35L35_MON_FRM_SHIFT 7 + +#define CS35L35_MS_MASK 0x80 +#define CS35L35_MS_SHIFT 7 +#define CS35L35_SPMODE_MASK 0x40 +#define CS35L35_SP_DRV_MASK 0x10 +#define CS35L35_SP_DRV_SHIFT 4 +#define CS35L35_CLK_CTL2_MASK 0xFF +#define CS35L35_PDM_MODE_MASK 0x40 +#define CS35L35_PDM_MODE_SHIFT 6 +#define CS35L35_CLK_SOURCE_MASK 0x03 +#define CS35L35_CLK_SOURCE_SHIFT 0 +#define CS35L35_CLK_SOURCE_MCLK 0 +#define CS35L35_CLK_SOURCE_SCLK 1 +#define CS35L35_CLK_SOURCE_PDM 2 + +#define CS35L35_SP_SCLKS_MASK 0x0F +#define CS35L35_SP_SCLKS_SHIFT 0x00 +#define CS35L35_SP_SCLKS_16FS 0x03 +#define CS35L35_SP_SCLKS_32FS 0x07 +#define CS35L35_SP_SCLKS_48FS 0x0B +#define CS35L35_SP_SCLKS_64FS 0x0F +#define CS35L35_SP_RATE_MASK 0xC0 + +#define CS35L35_PDN_BST_MASK 0x06 +#define CS35L35_PDN_BST_FETON_SHIFT 1 +#define CS35L35_PDN_BST_FETOFF_SHIFT 2 +#define CS35L35_PWR2_PDN_MASK 0xE0 +#define CS35L35_PWR3_PDN_MASK 0x1E +#define CS35L35_PDN_ALL_MASK 0x01 +#define CS35L35_DISCHG_FILT_MASK 0x02 +#define CS35L35_DISCHG_FILT_SHIFT 1 +#define CS35L35_MCLK_DIS_MASK 0x04 +#define CS35L35_MCLK_DIS_SHIFT 2 + +#define CS35L35_BST_CTL_MASK 0x7F +#define CS35L35_BST_CTL_SHIFT 0 +#define CS35L35_BST_IPK_MASK 0x1F +#define CS35L35_BST_IPK_SHIFT 0 +#define CS35L35_AMP_MUTE_MASK 0x20 +#define CS35L35_AMP_MUTE_SHIFT 5 +#define CS35L35_AMP_GAIN_ZC_MASK 0x10 +#define CS35L35_AMP_GAIN_ZC_SHIFT 4 + +/* Class H Algorithm Control */ +#define CS35L35_CH_STEREO_MASK 0x40 +#define CS35L35_CH_STEREO_SHIFT 6 +#define CS35L35_CH_BST_OVR_MASK 0x04 +#define CS35L35_CH_BST_OVR_SHIFT 2 +#define CS35L35_CH_BST_LIM_MASK 0x08 +#define CS35L35_CH_BST_LIM_SHIFT 3 +#define CS35L35_CH_MEM_DEPTH_MASK 0x01 +#define CS35L35_CH_MEM_DEPTH_SHIFT 0 +#define CS35L35_CH_HDRM_CTL_MASK 0x3F +#define CS35L35_CH_HDRM_CTL_SHIFT 0 +#define CS35L35_CH_REL_RATE_MASK 0xFF +#define CS35L35_CH_REL_RATE_SHIFT 0 +#define CS35L35_CH_WKFET_DIS_MASK 0x80 +#define CS35L35_CH_WKFET_DIS_SHIFT 7 +#define CS35L35_CH_WKFET_DEL_MASK 0x70 +#define CS35L35_CH_WKFET_DEL_SHIFT 4 +#define CS35L35_CH_WKFET_THLD_MASK 0x0F +#define CS35L35_CH_WKFET_THLD_SHIFT 0 +#define CS35L35_CH_VP_AUTO_MASK 0x80 +#define CS35L35_CH_VP_AUTO_SHIFT 7 +#define CS35L35_CH_VP_RATE_MASK 0x60 +#define CS35L35_CH_VP_RATE_SHIFT 5 +#define CS35L35_CH_VP_MAN_MASK 0x1F +#define CS35L35_CH_VP_MAN_SHIFT 0 + +/* CS35L35_PROT_RELEASE_CTL */ +#define CS35L35_CAL_ERR_RLS 0x80 +#define CS35L35_SHORT_RLS 0x04 +#define CS35L35_OTW_RLS 0x02 +#define CS35L35_OTE_RLS 0x01 + +/* INT Mask Registers */ +#define CS35L35_INT1_CRIT_MASK 0x38 +#define CS35L35_INT2_CRIT_MASK 0xEF +#define CS35L35_INT3_CRIT_MASK 0xEE +#define CS35L35_INT4_CRIT_MASK 0xFF + +/* PDN DONE Masks */ +#define CS35L35_M_PDN_DONE_SHIFT 4 +#define CS35L35_M_PDN_DONE_MASK 0x10 + +/* CS35L35_INT_1 */ +#define CS35L35_CAL_ERR 0x80 +#define CS35L35_OTP_ERR 0x40 +#define CS35L35_LRCLK_ERR 0x20 +#define CS35L35_SPCLK_ERR 0x10 +#define CS35L35_MCLK_ERR 0x08 +#define CS35L35_AMP_SHORT 0x04 +#define CS35L35_OTW 0x02 +#define CS35L35_OTE 0x01 + +/* CS35L35_INT_2 */ +#define CS35L35_PDN_DONE 0x10 +#define CS35L35_VPBR_ERR 0x02 +#define CS35L35_VPBR_CLR 0x01 + +/* CS35L35_INT_3 */ +#define CS35L35_BST_HIGH 0x10 +#define CS35L35_BST_HIGH_FLAG 0x08 +#define CS35L35_BST_IPK_FLAG 0x04 +#define CS35L35_LBST_SHORT 0x01 + +/* CS35L35_INT_4 */ +#define CS35L35_VMON_OVFL 0x08 +#define CS35L35_IMON_OVFL 0x04 + +#define CS35L35_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct cs35l35_private { + struct snd_soc_codec *codec; + struct cs35l35_platform_data pdata; + struct regmap *regmap; + struct regulator_bulk_data supplies[2]; + int num_supplies; + int sysclk; + int sclk; + bool pdm_mode; + bool i2s_mode; + bool slave_mode; + /* GPIO for /RST */ + struct gpio_desc *reset_gpio; + struct completion pdn_done; +}; + +static const char * const cs35l35_supplies[] = { + "VA", + "VP", +}; + +#endif -- 1.9.1

1 0

[alsa-devel] [alsa-lib][PATCH] pcm: fix wrong document references to PCM APIs which perform direct memory access with frame copying
by Takashi Sakamoto 13 Dec '16

13 Dec '16

In a design of ALSA PCM interface, for PCM frame transmission to/from kernel space, applications can select from two options; direct memory access or ioctl(2). Available options are decided depending on device capacity and machine architecture. Applications can get available options by the first entry of 'struct snd_pcm_hw_params.masks'. When the mask includes 'SNDRV_PCM_ACCESS_MMAP_xxx', applications can use direct memory access. For this use case, userspace library has two types of PCM API. One is to expose a pointer over the memory to start reading/writing PCM frames. Another is to copy PCM frames between the memory and a given buffer. Current documentation includes wrong references to these APIs to describe their advantages/disadvantages. This confuses application developers because the references indicate PCM APIs to execute ioctl(2) operation to read/write PCM frames. This commit fixes the bug. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/pcm/pcm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index f2ca02b..0cf740f 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -260,8 +260,9 @@ If you like to use the compatibility functions in mmap mode, there are read / write routines equaling to standard read / write transfers. Using these functions discards the benefits of direct access to memory region. See the #snd_pcm_mmap_readi(), -#snd_pcm_writei(), #snd_pcm_readn() -and #snd_pcm_writen() functions. +#snd_pcm_mmap_writei(), #snd_pcm_mmap_readn() +and #snd_pcm_mmap_writen() functions. These functions use +#snd_pcm_areas_copy() internally. \section pcm_errors Error codes -- 2.9.3

2 6

[alsa-devel] [PATCH] hiface: Fix sample rate changes
by Jussi Laako 12 Dec '16

12 Dec '16

When attempting to change sample rate for the interface after first set-play cycle, the change is not reflected to the hardware, while from user space everything seems to go fine. This patch fixes the issue and the driver now behaves the same way as for example the USB Audio Class driver.

3 5

[alsa-devel] [PATCH] ALSA: usb-audio: Eliminate noise at the start of DSD playback.
by Nobutaka Okabe 12 Dec '16

12 Dec '16

[Problem] In some USB DACs, a terrible pop noise comes to be heard at the start of DSD playback (in the following situations). - play first DSD track - change from PCM track to DSD track - change from DSD64 track to DSD128 track (and etc...) - seek DSD track - Fast-Forward/Rewind DSD track [Cause] At the start of playback, there is a little silence. The silence bit pattern "0x69" is required on DSD mode, but it is not like that. [Solution] This patch adds DSD silence pattern to the endpoint settings. Signed-off-by: Nobutaka Okabe <nob77413(a)gmail.com> --- sound/usb/endpoint.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 57b0d99..a2cdf33 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -638,7 +638,21 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, ep->datainterval = fmt->datainterval; ep->stride = frame_bits >> 3; - ep->silence_value = pcm_format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0; + + switch (pcm_format) { + case SNDRV_PCM_FORMAT_U8: + ep->silence_value = 0x80; + break; + case SNDRV_PCM_FORMAT_DSD_U8: + case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: + case SNDRV_PCM_FORMAT_DSD_U16_BE: + case SNDRV_PCM_FORMAT_DSD_U32_BE: + ep->silence_value = 0x69; + break; + default: + ep->silence_value = 0; + } /* assume max. frequency is 50% higher than nominal */ ep->freqmax = ep->freqn + (ep->freqn >> 1); -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add native DSD support for TEAC 501/503 DAC
by Nobutaka Okabe 12 Dec '16

12 Dec '16

This patch adds native DSD support for the following devices. - TEAC NT-503 - TEAC UD-503 - TEAC UD-501 (1)Add quirks for native DSD support for TEAC devices. (2)A specific vendor command is needed to switch between PCM/DOP and DSD mode, same as Denon/Marantz devices. Signed-off-by: Nobutaka Okabe <nob77413(a)gmail.com> --- sound/usb/quirks.c | 38 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2782155..b3fd238 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1165,6 +1165,18 @@ static bool is_marantz_denon_dac(unsigned int id) return false; } +/* TEAC UD-501/UD-503/NT-503 USB DACs need a vendor cmd to switch + * between PCM/DOP and native DSD mode + */ +static bool is_teac_50X_dac(unsigned int id) +{ + switch (id) { + case USB_ID(0x0644, 0x8043): /* TEAC UD-501/UD-503/NT-503 */ + return true; + } + return false; +} + int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, struct audioformat *fmt) { @@ -1192,6 +1204,26 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, break; } mdelay(20); + } else if (is_teac_50X_dac(subs->stream->chip->usb_id)) { + /* Vendor mode switch cmd is required. */ + switch (fmt->altsetting) { + case 3: /* DSD mode (DSD_U32) requested */ + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0, + USB_DIR_OUT|USB_TYPE_VENDOR|USB_RECIP_INTERFACE, + 1, 1, NULL, 0); + if (err < 0) + return err; + break; + + case 2: /* PCM or DOP mode (S32) requested */ + case 1: /* PCM mode (S16) requested */ + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0, + USB_DIR_OUT|USB_TYPE_VENDOR|USB_RECIP_INTERFACE, + 0, 1, NULL, 0); + if (err < 0) + return err; + break; + } } return 0; } @@ -1337,5 +1369,11 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; } + /* TEAC devices with USB DAC functionality */ + if (is_teac_50X_dac(chip->usb_id)) { + if (fp->altsetting == 3) + return SNDRV_PCM_FMTBIT_DSD_U32_BE; + } + return 0; } -- 2.7.4

2 1

[alsa-devel] [GIT PULL] ASoC updates for v4.10
by Mark Brown 12 Dec '16

12 Dec '16

The following changes since commit 69973b830859bc6529a7a0468ba0d80ee5117826: Linux 4.9 (2016-12-11 11:17:54 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.10 for you to fetch changes up to a5de5b74a50113564a1e0850e2da96c37c35e55d: Merge remote-tracking branches 'asoc/topic/wm9712', 'asoc/topic/wm9713' and 'asoc/topic/zte' into asoc-next (2016-12-12 15:53:32 +0000) ---------------------------------------------------------------- ASoC: Updates for v4.10 There's been a few bits of framework work this time around and quite a lot of cleanups and improvements to existing code: - Support for stereo DAPM controls from Chen-yu Tsai. - Some initial work on the of-graph sound card from Morimoto-san, the main bulk of this is currently in binding review. - Lots of Renesas cleanups from Morimoto-san and sunxi work from Chen-yu Tsai. - regmap conversions of the remaining AC'97 drivers from Lars-Peter Clausen. - A new version of the topology ABI from Mengdong Lin. - New drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and Realtek RT5665. ---------------------------------------------------------------- Adam Thomson (1): ASoC: da7219: Improve pop/click performance for sensitive HPs Alexey Khoroshilov (1): SoC: mxs-saif: check validity of ids in mxs_saif_probe() Arnaud Pouliquen (5): ASoC: sti: fix errors management ASoC: sti: reset refactoring ASoC: sti: clean unused include ASoC: sti-sas: clean legacy in sti-sas ASoC: sti-sas: add missing return in messages strings Arnd Bergmann (2): ASoC: lpass-platform: initialize dma channel number ASoC: topology: avoid uninitialized kcontrol_type Axel Lin (6): regulator: rk808: Use rdev_get_id() to access id of regulator ASoC: cs35l34: Remove CS35L34_CHIP_ID from cs35l34_readable_register ASoC: msm8916-wcd-analog: Update correct register setting for MIC BIAS Internal1 ASoC: rt5665: Fix missing mutex_unlock in rt5665_calibrate ASoC: rt5665: Use devm_gpio_request_one() ASoC: cs35l34: Simplify the logic to set CS35L34_MCLK_CTL setting Bard Liao (10): ASoC: rt5663: rename rt5668 as rt5663 v2 ASoC: rt5660: enable MCLK detection ASoC: rt5640: enable MCLK detection ASoC: rt5670: Enable MCLK detection ASoC: rt5670: increse LDO power ASoC: rt5640: add Mono ADC Capture Switch control ASoC: rt286: remove unnecessary selection in Kconfig ASoC: add rt5665 codec driver ASoC: rl6231: add 19.2M to 4.096M pll preset table ASoC: rt298: disable IRQ when jack is NULL Charles Keepax (18): ASoC: arizona: Add gating for clock when used for direct MCLK ASoC: arizona: Add gating for source clocks of the FLLs ASoC: samsung: i2s: Fixup last IRQ unsafe spin lock call ASoC: cs42xx8: Mark chip ID as volatile and remove cache bypass ASoC: cs42l56: Make ID registers volatile and remove cache bypass ASoC: cs42l73: Remove cache bypass for read of ID registers ASoC: arizona: Move request of speaker IRQs into bus probe ASoC: arizona: Move request of DSP IRQ into bus probe ASoC: cs47l24: Fixup missing variable typo ASoC: arizona: Access driver data through platform from compressed ops ASoC: core: If a platform doesn't have an of_node use parent's node ASoC: wm2200: Correct types of mixer texts and values ASoC: arizona: Move notifier functions to header and make inline ASoC: arizona: Call arizona_init_notifiers for all CODECs ASoC: wm_adsp: Only write shutdown controls for active firmwares ASoC: wm_adsp: Check return value from wm_adsp_buffer_init ASoC: arizona: Remove redundant extern declarations ASoC: wm_adsp: Remove redundant extern declarations Chen-Yu Tsai (19): ASoC: dapm: Support second register for DAPM control updates ASoC: dapm: Implement stereo mixer control support ASoC: dapm: Introduce DAPM_DOUBLE dual channel control type ASoC: dapm: Introduce DAPM_DOUBLE_R dual channel dual register control type ASoC: sun4i-codec: Move data structures to add create_card call to quirks ASoC: sun4i-codec: Revise comments for register definition macros ASoC: sun4i-codec: Expand quirks to handle register offsets and card creation ASoC: sun4i-codec: Increase DMA max burst to 8 ASoC: sun4i-codec: Add support for A31 playback through headphone output ASoC: sun4i-codec: Add support for A31 Line In playback ASoC: sun4i-codec: Add support for A31 Line Out playback ASoC: sun4i-codec: Add support for A31 analog microphone inputs ASoC: sun4i-codec: Add support for A31 board level audio routing ASoC: sun4i-codec: Add support for A31 ADC capture path ASoC: sun4i-codec: Add support for optional reset control to quirks ASoC: sunxi: Add bindings for A23/A33/H3 codec's analog path controls ASoC: sunxi: Add support for A23/A33/H3 codec's analog path controls ASoC: sun4i-codec: Add support for A23 codec ASoC: sun4i-codec: Add support for H3 codec Colin Ian King (2): ASoC: sst-bxt-da7219_max98357a: fix obsoleted initializer for array ASoC: mioa701_wm9713: add missing white space in dev_err message Dan Carpenter (2): ASoC: Intel: Skylake: Fix a shift wrapping bug ASoC: sunxi: Uninitialized variable in probe() Dharageswari R (2): ASoC: Intel: Skylake: Use DPIB to update position for Playback stream ASoC: Intel: Skylake: Use DPIB to update position for Playback stream Florian Vaussard (1): ASoC: cs42l56: Fix misuse of regmap_update_bits Geert Uytterhoeven (1): ASoC: davinci-mcbsp: DT fix s/interrupts-names/interrupt-names/ GuruprasadX Pawse (2): ASoC: Intel: Skylake: Don't use dma I2S config structure in kernel ASoC: Intel: Skylake: Removed the unused I2S blob structure Jack Yu (1): ASoC: Add jd function for rt5663. James Schulman (2): ASoC: Add support for CS42L42 codec ASoC: cs42l42: Add devicetree bindings for CS42L42 Jarkko Nikula (2): ASoC: rl6347a: Use dev_err for I2C communication error prints ASoC: rl6347a: Use dev_err for I2C communication error prints Jayachandran B (3): ASoC: Intel: Skylake: Add D0iX callbacks ASoC: Intel: Skylake: remove pci device enabling calls on suspend ASoC: Intel: Skylake: Flush pending D0i3 request on suspend John Hsu (5): ASoC: nau8825: Disable short Frame Sync detection logic ASoC: nau8825: AD/DA over sampling rate configuration ASoC: nau8825: FLL parameters finetune ASoC: nau8825: lock longer to avoid playback pop upon resume ASoC: nau8825: disable sinc filter for high THD of ADC Julia Lawall (7): ASoC: intel: broadwell: constify snd_soc_ops structures ASoC: atmel_wm8904: constify snd_soc_ops structures ASoC: rockchip: constify snd_soc_ops structures ASoC: qcom: storm: constify snd_soc_ops structures ASoC: tegra: constify snd_soc_ops structures ASoC: constify snd_soc_ops structures ASoC: constify snd_soc_ops structures Krzysztof Kozlowski (4): ASoC: samsung: Remove non-existing MACH dependencies ASoC: samsung: smdk_wm8580: Remove old platforms and drop mach-types usage ASoC: samsung: Enable COMPILE_TEST for entire Samsung ASoc ASoC: samsung: Enable COMPILE_TEST for SmartQ and WM8580 Kuninori Morimoto (47): ASoC: soc.h: use bit field for playback/capture_only ASoC: bunch up bit field for snd_soc_pcm_runtime ASoC: bunch up bit field for snd_soc_dai ASoC: remove component from snd_soc_pcm_runtime ASoC: rsnd: remove duplicate define of rsnd_dvc_of_node() ASoC: rsnd: amend .probe/.remove call for DPCM ASoC: rsnd: add rsnd_mod_next() for for_each_rsnd_mod_xxx() ASoC: rsnd: use for_each_rsnd_mod_xxx() on rsnd_dai_call() ASoC: rsnd: use for_each_rsnd_mod_xxx() on rsnd_rdai_continuance_probe() ASoC: rsnd: add rsnd_parse_of_node() and integrate rsnd_xxx_of_node ASoC: rsnd: remove non DT support for DMA ASoC: rsnd: don't use devm_request_irq() for SSI ASoC: rsnd: remove rsnd_dma_detach() ASoC: rsnd: don't call unneeded of_node_put() on dma.c ASoC: rsnd: add nolock_start/stop callback ASoC: rsnd: depends on OF ASoC: rsnd: enable COMPILE_TEST ASoC: rsnd: remove "Gen2 only" comment ASoC: rsnd: rsnd_reg cleanup for SSIU ASoC: rsnd: fixup SCU_SYS_STATUSx access ASoC: rsnd: clear SSI_SYS_STATUSx every time ASoC: rsnd: enable SRC sync even FIN = FOUT ASoC: max98504: Add missing MAX98504 on SND_SOC_ALL_CODECS ASoC: rsnd: use BRGCKR instead of SSICKR ASoC: simple-scu-card: code sync: follow to simple family style ASoC: simple-scu-card: code sync: rename asoc_simple_card_priv ASoC: simple-scu-card: code sync: tidyup props/link naming ASoC: simple-card-utils: remove unnecessary cpu/codec pointer check ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_routing ASoC: soc-core: snd_soc_get_dai_name() become non static ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_prefix ASoC: soc-core: adjust for graph on snd_soc_of_parse_card_name ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_simple_widgets ASoC: uda1380: Remove #if IS_ENABLED(CONFIG_I2C) ASoC: wm9081: Remove #if IS_ENABLED(CONFIG_I2C) ASoC: wm8523: Remove #if IS_ENABLED(CONFIG_I2C) ASoC: wm8580: Remove #if IS_ENABLED(CONFIG_I2C) ASoC: rsnd: Request/Release DMA channel each time ASoC: rsnd: use dma_sync_single_for_xxx() for IOMMU ASoC: core: replace codec_dev_list to component_dev_list on Card ASoC: core: replace aux_comp_list to component_dev_list ASoC: add Component level suspend/resume ASoC: rsnd: rsnd_get_dalign() needs to care SSIU, not SSI ASoC: simple_card_utils: tidyup file comment/define ASoC: rsnd: tidyup ssi->usrcnt counter check in hw_params ASoC: rsnd: enable/disable ADG when suspend/resume timing ASoC: rsnd: setup BRGCKR/BRRA/BRRB when starting Lars-Peter Clausen (27): ASoC: wm9712: Convert to regmap ASoC: wm9712: Remove ac97_read/ac97_write wrappers ASoC: wm9705: Convert to regmap ASoC: wm9705: Remove ac97_read/ac97_write wrappers ASoC: wm9712: Remove unused DAI ID defines ASoC: wm9713: Remove unused DAI ID defines ASoC: wm8753: Remove unused DAI ID defines ASoC: wm9705: Remove unused DAI ID defines ASoC: pxa2xx-i2s: Remove unused DAI ID defines ASoC: pxa-ssp: Remove unused DAI ID defines ASoC: es8328: Remove some unused defines ASoC: ak4641: Remove unused DAI ID defines ASoC: ak4641: Remove unused AK4641_CACHEREGNUM define ASoC: ak4641: Move register defines to main source ASoC: stac9766: Remove unused DAI ID defines ASoC: stac9766: Remove register paging support ASoC: stac9766: Move register defines to main source file ASoC: stac9766: Convert to regmap ASoC: stac9766: Remove ac97_read/ac97_write wrappers ASoC: samsung: Remove unselectable smdk_wm8580pcm ASoC: pxa: Make static string arrays 'const 'char * const []' ASoC: rt5616: Don't use rtd->codec ASoC: intel: mfld: Make static string arrays 'const 'char * const []' ASoC: Make return type of dpcm_state_string() const char * ASoC: uda1380: Remove unused DAI ID defines ASoC: uda1380: Request GPIOs at bus probe time ASoC: ab8500: Remove explicit initialization of driver callbacks to NULL Mark Brown (31): Merge branch 'topic/dmaengine' of git://git.kernel.org/.../broonie/sound into asoc-samsung Merge remote-tracking branch 'asoc/topic/rl6347a' into asoc-rl6347a Merge branch 'fix/samsung' of git://git.kernel.org/.../broonie/sound into asoc-samsung Merge branch 'fix/qcom' of git://git.kernel.org/.../broonie/sound into asoc-qcom Merge branch 'fix/sunxi' of git://git.kernel.org/.../broonie/sound into asoc-sunxi Merge branch 'topic/dapm' of git://git.kernel.org/.../broonie/sound into asoc-sunxi Merge branch 'topic/dapm-pin' of git://git.kernel.org/.../broonie/sound into asoc-arizona Merge branch 'topic/component' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branches 'asoc/fix/da7219-pops' and 'asoc/fix/qcom' into asoc-linus Merge remote-tracking branch 'asoc/topic/component' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm-pin' into asoc-next Merge remote-tracking branch 'asoc/topic/dmaengine' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/ab8500', 'asoc/topic/arizona', 'asoc/topic/atmel', 'asoc/topic/bcm' and 'asoc/topic/bitfield' into asoc-next Merge remote-tracking branches 'asoc/topic/compress', 'asoc/topic/const' and 'asoc/topic/cs35l34' into asoc-next Merge remote-tracking branches 'asoc/topic/cs42l42', 'asoc/topic/cs42l56', 'asoc/topic/cs42l73' and 'asoc/topic/cs42xx8' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci' and 'asoc/topic/doc' into asoc-next Merge remote-tracking branches 'asoc/topic/dpcm', 'asoc/topic/es8328', 'asoc/topic/extcon' and 'asoc/topic/fsl' into asoc-next Merge remote-tracking branches 'asoc/topic/inntel', 'asoc/topic/input', 'asoc/topic/max98504' and 'asoc/topic/nau8825' into asoc-next Merge remote-tracking branches 'asoc/topic/of-graph', 'asoc/topic/pxa', 'asoc/topic/qcom' and 'asoc/topic/rk808' into asoc-next Merge remote-tracking branches 'asoc/topic/rl6231', 'asoc/topic/rl6347a', 'asoc/topic/rockchip', 'asoc/topic/rt286' and 'asoc/topic/rt298' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5514', 'asoc/topic/rt5616', 'asoc/topic/rt5640', 'asoc/topic/rt5660' and 'asoc/topic/rt5663' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5665', 'asoc/topic/rt5670', 'asoc/topic/rt5677', 'asoc/topic/samsung' and 'asoc/topic/simple' into asoc-next Merge remote-tracking branches 'asoc/topic/stac9766', 'asoc/topic/sti', 'asoc/topic/sti-codec', 'asoc/topic/sunxi' and 'asoc/topic/tegra' into asoc-next Merge remote-tracking branches 'asoc/topic/tlv320aic31xx', 'asoc/topic/topology', 'asoc/topic/uda1380', 'asoc/topic/wm2200' and 'asoc/topic/wm8523' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8580', 'asoc/topic/wm8753', 'asoc/topic/wm8978', 'asoc/topic/wm9081' and 'asoc/topic/wm9705' into asoc-next Merge remote-tracking branches 'asoc/topic/wm9712', 'asoc/topic/wm9713' and 'asoc/topic/zte' into asoc-next Matt Flax (1): ASoC: wm8580: Add the wm8581 codec to the driver Maxime Ripard (3): ASoC: sun4i-i2s: Implement capture support ASoC: sunxi: i2s: Implement set_sysclk ASoC: wm8978: Adjust clock indices so that simple card works Mengdong Lin (17): ASoC: topology: ABI - Define DAI physical PCM data formats ASoC: topology: Make manifest backward compatible from ABI v4 ASoC: topology: Make PCM backward compatible from ABI v4 ASoC: topology: Only use valid names of PCM for the kernel DAI & DAI link ASoC: topology: Support topology file of ABI v4 ASoC: topology: ABI - Add flags and private data to PCM ASoC: topology: ABI - Update physical DAI link configuration for version 5 ASoC: Define API to find a dai link ASoC: topology: Add support to configure existing physical DAI links ASoC: topology: Rename the function to create a FE link ASoC: topology: ABI - Rename struct and type for physical DAIs ASoC: topology: Rename functions & variables for physical DAIs ASoC: topology: ABI - Add voice wake up flag for DAI links ASoC: topology: Check name strings of physical DAI links ASoC: topology: ABI - Rename be_dai_elems to dai_elems in manifest ASoC: topology: Allow a widget to have multiple enum controls ASoC: topology: Only free TLV for volume mixers of a widget Oder Chiou (1): ASoC: rt5514: Add the DMIC initial delay to wait it ready. Pardha Saradhi K (2): ASoC: Intel: Skylake: Add D0iX IPCs ASoC: Intel: Skylake: Add support for programming D0i3C Paul Handrigan (2): ASoC: cs35l34: Add device tree bindings file for cs35l34 ASoC: cs35l34: Initial commit of the cs35l34 CODEC driver. Peter Rosin (3): ASoC: atmel_ssc_dai: if not provided, default to sensible dividers ASoC: tse850: document axentia,tse850-pcm5142 bindings ASoC: atmel: tse850: add ASoC driver for the Axentia TSE-850 Peter Ujfalusi (2): ASoC: tlv320aic31xx: Add missing of_device_id for dac3100 ASoC: tlv320aic31xx: Add support for tlv320dac3101 Pierre-Louis Bossart (3): ASoC: Intel: common: add ACPI package extraction utility ASoC: Intel: detect audio routing with CHAN package ASoC: Intel: bytct_rt5640: change default capture settings Randy Dunlap (1): ASoC: fsl: fix fsl_spdif.c build errors Richard Fitzgerald (12): ASoC: wm_adsp: Remove duplicate set of kcontrol->iface ASoC: wm_adsp: Signal firmware shutdown through event control ASoC: wm_adsp: factor out getting region name from type ASoC: wm_adsp: Support acknowledged controls ASoC: wm_adsp: factor out getting base register for a control ASoC: wm_adsp: Firmware controls should be added as codec controls ASoC: wm_adsp: Remove unused wm_coeff_ctl.kcontrol ASoC: core: Add component pin control functions ASoC: arizona: Use component pin control functions extcon: arizona: Use SoC component pin control functions Input: arizona-haptics - Use SoC component pin control functions ASoC: wm_adsp: wm_adsp_buf_alloc should use kfree in error path Sathyanarayana Nujella (2): ASoC: Intel: report JACK_LINEOUT event ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture Scott Branden (1): ASoC: bcm: add depends on HAS_DMA Sebastien Guiriec (4): ASoC: Intel: atom: Add debug information related to FW version ASoC: Intel: atom: Add sysfs entry in order to store FW version ASoC: Intel: atom: save FW version ASoC: Intel: Add ASoC Intel SST Atom sysfs description Shawn Guo (2): ASoC: zte: spdif and i2s drivers are not zx296702 specific ASoC: zte: spdif: correct ZX_SPDIF_CLK_RAT define Srinivas Kandagatla (7): ASoC: codecs: Add msm8916-wcd analog codec ASoC: codecs: Add msm8916-wcd digital codec ASoC: apq8016-sbc: dt bindings: remove incorrect property ASoC: qcom: apq8016-sbc: Add support to multi codec. ASoC: lpass-platform: use dma_ch instead of rdma_ch/wrdma_ch ASoC: codecs: msm8916-wcd-analog: clean parse_dt() ASoC: codecs: msm8916-wcd-digital: rename parse_dt to get_clks Stuart Henderson (1): ASoC: wm_adsp: Add support for SYSTEM firmware controls Sylwester Nawrocki (11): ASoC: s3c24xx_uda134x: Move global variables to driver's data structure ASoC: samsung: Remove unneeded initialization of chan_name ASoC: Drop SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag ASoC: samsung: s3c24xx-i2s: Don't use platform_data for DMA parameters ASoC: samsung: s3c24xx-i2s: Debug/error trace cleanup ASoC: samsung: pcm: Conversion to use devm_ioremap_resource() ASoC: samsung: s2c24xx-i2s: remove redundant error message ASoC: samsung: Drop AC97 drivers ASoC: samsung: s3c24xx-i2s.c merge fixup ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem ASoC: samsung: Add machine driver for Exynos5433 based TM2 board Takashi Iwai (10): ASoC: rt5514: Remove superfluous linux/kthread.h inclusion ASoC: rt5677: Remove superfluous linux/kthread.h inclusion ASoC: cht_bsw_rt5645: Fix leftover kmalloc ASoC: Intel: atom: Make some messages to debug level ASoC: Intel: Add missing 10EC5672 ACPI ID matching for Cherry Trail ASoC: rt5670: Enable Braswell platform workaround for Dell Wyse 3040 ASoC: rt5670: Add missing 10EC5072 ACPI ID ASoC: intel: Replace kthread with work ASoC: intel: Fix crash at suspend/resume without card registration ASoC: cht_bsw_rt5672: Use HID translation unit Valentin Rothberg (1): ASoC: samsung: Makefile cleanup Vinod Koul (6): ASoC: Intel: Add check_dsp_lp_on callback on IPC ASoC: Intel: Add sst_ipc_tx_message_nopm ASoC: Intel: Skylake: Add support for LPMode ASoC: Intel: Skylake: Add support for specifying D0i3 configuration ASoC: Intel: Skylake: Add D0i3 mode ref counting ASoC: compress: Add support for compress dai ops Wei Yongjun (2): ASoC: rk3399_gru_sound: Fix non static symbol warning ASoC: wm8580: Fix non static symbol warnings anish kumar (1): ASoC: Codec to codec dai link description kbuild test robot (2): ASoC: sun4i-codec: fix semicolon.cocci warnings ASoC: fix platform_no_drv_owner.cocci warnings Documentation/ABI/testing/sysfs-platform-sst-atom | 17 + .../bindings/sound/axentia,tse850-pcm5142.txt | 88 + .../devicetree/bindings/sound/cs35l34.txt | 64 + .../devicetree/bindings/sound/cs42l42.txt | 110 + .../devicetree/bindings/sound/davinci-mcbsp.txt | 2 +- .../devicetree/bindings/sound/qcom,apq8016-sbc.txt | 5 +- .../bindings/sound/qcom,msm8916-wcd-analog.txt | 85 + .../bindings/sound/qcom,msm8916-wcd-digital.txt | 20 + Documentation/devicetree/bindings/sound/rt5514.txt | 3 + Documentation/devicetree/bindings/sound/rt5663.txt | 6 +- Documentation/devicetree/bindings/sound/rt5665.txt | 68 + .../bindings/sound/samsung,tm2-audio.txt | 38 + .../devicetree/bindings/sound/sun4i-codec.txt | 65 +- .../bindings/sound/sun8i-codec-analog.txt | 16 + .../devicetree/bindings/sound/tlv320aic31xx.txt | 1 + Documentation/devicetree/bindings/sound/wm8580.txt | 4 +- Documentation/sound/alsa/soc/codec_to_codec.txt | 103 + MAINTAINERS | 7 + drivers/extcon/extcon-arizona.c | 8 +- drivers/input/misc/arizona-haptics.c | 13 +- drivers/regulator/rk808-regulator.c | 9 +- include/dt-bindings/sound/cs42l42.h | 73 + include/sound/cs35l34.h | 35 + include/sound/dmaengine_pcm.h | 6 - include/sound/rt5514.h | 20 + include/sound/rt5665.h | 47 + include/sound/simple_card_utils.h | 8 +- include/sound/soc-dai.h | 43 +- include/sound/soc-dapm.h | 14 + include/sound/soc-topology.h | 2 +- include/sound/soc.h | 87 +- include/uapi/sound/asoc.h | 90 +- include/uapi/sound/snd_sst_tokens.h | 8 +- sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel_ssc_dai.c | 83 +- sound/soc/atmel/atmel_ssc_dai.h | 1 + sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/tse850-pcm5142.c | 472 ++ sound/soc/bcm/Kconfig | 1 + sound/soc/codecs/Kconfig | 31 +- sound/soc/codecs/Makefile | 11 +- sound/soc/codecs/ab8500-codec.c | 2 - sound/soc/codecs/adau17x1.c | 2 +- sound/soc/codecs/ak4641.c | 22 +- sound/soc/codecs/ak4641.h | 47 - sound/soc/codecs/arizona.c | 153 +- sound/soc/codecs/arizona.h | 109 +- sound/soc/codecs/cs35l34.c | 1251 +++++ sound/soc/codecs/cs35l34.h | 269 ++ sound/soc/codecs/cs42l42.c | 1986 ++++++++ sound/soc/codecs/cs42l42.h | 776 ++++ sound/soc/codecs/cs42l56.c | 18 +- sound/soc/codecs/cs42l73.c | 4 - sound/soc/codecs/cs42xx8.c | 10 - sound/soc/codecs/cs47l24.c | 55 +- sound/soc/codecs/da7219-aad.c | 18 +- sound/soc/codecs/da7219.c | 139 +- sound/soc/codecs/da7219.h | 5 + sound/soc/codecs/es8328.h | 37 - sound/soc/codecs/msm8916-wcd-analog.c | 890 ++++ sound/soc/codecs/msm8916-wcd-digital.c | 923 ++++ sound/soc/codecs/nau8825.c | 142 +- sound/soc/codecs/nau8825.h | 16 + sound/soc/codecs/rl6231.c | 1 + sound/soc/codecs/rl6347a.c | 2 +- sound/soc/codecs/rt298.c | 24 +- sound/soc/codecs/rt5514-spi.c | 1 - sound/soc/codecs/rt5514.c | 17 + sound/soc/codecs/rt5514.h | 2 + sound/soc/codecs/rt5616.c | 3 +- sound/soc/codecs/rt5640.c | 5 + sound/soc/codecs/rt5640.h | 6 + sound/soc/codecs/rt5660.c | 4 + sound/soc/codecs/rt5660.h | 3 + sound/soc/codecs/rt5663.c | 1137 ++--- sound/soc/codecs/rt5663.h | 1162 ++--- sound/soc/codecs/rt5665.c | 4874 ++++++++++++++++++++ sound/soc/codecs/rt5665.h | 1990 ++++++++ sound/soc/codecs/rt5670.c | 16 +- sound/soc/codecs/rt5670.h | 1 + sound/soc/codecs/rt5677-spi.c | 1 - sound/soc/codecs/stac9766.c | 162 +- sound/soc/codecs/stac9766.h | 17 - sound/soc/codecs/sti-sas.c | 179 +- sound/soc/codecs/tlv320aic31xx.c | 3 + sound/soc/codecs/tlv320aic31xx.h | 1 + sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/uda1380.c | 77 +- sound/soc/codecs/uda1380.h | 4 - sound/soc/codecs/wm2200.c | 4 +- sound/soc/codecs/wm5102.c | 59 +- sound/soc/codecs/wm5110.c | 61 +- sound/soc/codecs/wm8523.c | 24 +- sound/soc/codecs/wm8580.c | 123 +- sound/soc/codecs/wm8753.h | 3 - sound/soc/codecs/wm8978.h | 2 +- sound/soc/codecs/wm8997.c | 39 +- sound/soc/codecs/wm8998.c | 38 +- sound/soc/codecs/wm9081.c | 2 - sound/soc/codecs/wm9705.c | 138 +- sound/soc/codecs/wm9705.h | 11 - sound/soc/codecs/wm9712.c | 177 +- sound/soc/codecs/wm9712.h | 11 - sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/wm9713.h | 4 - sound/soc/codecs/wm_adsp.c | 354 +- sound/soc/codecs/wm_adsp.h | 27 +- sound/soc/codecs/wmfw.h | 4 + sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/efika-audio-fabric.c | 1 - sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/fsl/imx-wm8962.c | 2 +- sound/soc/generic/simple-card-utils.c | 5 +- sound/soc/generic/simple-card.c | 2 +- sound/soc/generic/simple-scu-card.c | 115 +- sound/soc/intel/atom/sst-atom-controls.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 + sound/soc/intel/atom/sst/sst.c | 39 + sound/soc/intel/atom/sst/sst.h | 1 + sound/soc/intel/atom/sst/sst_acpi.c | 2 + sound/soc/intel/atom/sst/sst_ipc.c | 11 + sound/soc/intel/atom/sst/sst_stream.c | 4 +- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 3 +- sound/soc/intel/boards/bdw-rt5677.c | 2 +- sound/soc/intel/boards/broadwell.c | 18 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 26 +- sound/soc/intel/boards/bxt_rt298.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 68 +- sound/soc/intel/boards/bytcr_rt5651.c | 4 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 10 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 46 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/mfld_machine.c | 4 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 6 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 6 +- sound/soc/intel/boards/skl_rt286.c | 4 +- sound/soc/intel/common/sst-acpi.h | 17 +- sound/soc/intel/common/sst-ipc.c | 85 +- sound/soc/intel/common/sst-ipc.h | 8 +- sound/soc/intel/common/sst-match-acpi.c | 57 + sound/soc/intel/haswell/sst-haswell-ipc.c | 3 +- sound/soc/intel/skylake/bxt-sst.c | 145 + sound/soc/intel/skylake/skl-messages.c | 39 +- sound/soc/intel/skylake/skl-pcm.c | 28 +- sound/soc/intel/skylake/skl-sst-cldma.c | 1 - sound/soc/intel/skylake/skl-sst-dsp.h | 12 + sound/soc/intel/skylake/skl-sst-ipc.c | 71 +- sound/soc/intel/skylake/skl-sst-ipc.h | 37 +- sound/soc/intel/skylake/skl-sst-utils.c | 2 +- sound/soc/intel/skylake/skl-topology.c | 47 + sound/soc/intel/skylake/skl-topology.h | 28 +- sound/soc/intel/skylake/skl.c | 59 +- sound/soc/intel/skylake/skl.h | 6 + sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/mxs/mxs-saif.c | 13 +- sound/soc/mxs/mxs-sgtl5000.c | 2 +- sound/soc/pxa/corgi.c | 6 +- sound/soc/pxa/e740_wm9705.c | 1 - sound/soc/pxa/e750_wm9705.c | 1 - sound/soc/pxa/e800_wm9712.c | 1 - sound/soc/pxa/em-x270.c | 1 - sound/soc/pxa/hx4700.c | 2 - sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/mioa701_wm9713.c | 2 +- sound/soc/pxa/palm27x.c | 1 - sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/pxa-ssp.h | 6 - sound/soc/pxa/pxa2xx-i2s.h | 3 - sound/soc/pxa/spitz.c | 6 +- sound/soc/pxa/tosa.c | 7 +- sound/soc/qcom/apq8016_sbc.c | 11 +- sound/soc/qcom/lpass-platform.c | 49 +- sound/soc/qcom/storm.c | 2 +- sound/soc/rockchip/rk3399_gru_sound.c | 8 +- sound/soc/rockchip/rockchip_max98090.c | 2 +- sound/soc/rockchip/rockchip_rt5645.c | 2 +- sound/soc/samsung/Kconfig | 58 +- sound/soc/samsung/Makefile | 9 +- sound/soc/samsung/ac97.c | 437 -- sound/soc/samsung/dmaengine.c | 8 +- sound/soc/samsung/i2s.c | 8 +- sound/soc/samsung/ln2440sbc_alc650.c | 72 - sound/soc/samsung/pcm.c | 60 +- sound/soc/samsung/regs-ac97.h | 66 - sound/soc/samsung/s3c2412-i2s.c | 2 - sound/soc/samsung/s3c24xx-i2s.c | 51 +- sound/soc/samsung/s3c24xx_uda134x.c | 79 +- sound/soc/samsung/smdk2443_wm9710.c | 68 - sound/soc/samsung/smdk_wm8580.c | 30 +- sound/soc/samsung/smdk_wm8580pcm.c | 175 - sound/soc/samsung/smdk_wm9713.c | 108 - sound/soc/samsung/tm2_wm5110.c | 552 +++ sound/soc/sh/Kconfig | 3 +- sound/soc/sh/rcar/adg.c | 61 +- sound/soc/sh/rcar/core.c | 175 +- sound/soc/sh/rcar/dma.c | 295 +- sound/soc/sh/rcar/dvc.c | 2 - sound/soc/sh/rcar/gen.c | 12 +- sound/soc/sh/rcar/rsnd.h | 156 +- sound/soc/sh/rcar/src.c | 13 +- sound/soc/sh/rcar/ssi.c | 28 +- sound/soc/sh/rcar/ssiu.c | 20 + sound/soc/soc-compress.c | 98 +- sound/soc/soc-core.c | 181 +- sound/soc/soc-dapm.c | 154 +- sound/soc/soc-generic-dmaengine-pcm.c | 13 +- sound/soc/soc-pcm.c | 2 +- sound/soc/soc-topology.c | 751 ++- sound/soc/soc-utils.c | 199 + sound/soc/sti/sti_uniperif.c | 43 +- sound/soc/sti/uniperif.h | 2 + sound/soc/sti/uniperif_player.c | 91 +- sound/soc/sti/uniperif_reader.c | 41 +- sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun4i-codec.c | 867 +++- sound/soc/sunxi/sun4i-i2s.c | 105 +- sound/soc/sunxi/sun8i-codec-analog.c | 665 +++ sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 2 +- sound/soc/tegra/trimslice.c | 2 +- sound/soc/zte/Kconfig | 16 +- sound/soc/zte/Makefile | 4 +- sound/soc/zte/{zx296702-i2s.c => zx-i2s.c} | 0 sound/soc/zte/{zx296702-spdif.c => zx-spdif.c} | 2 +- 232 files changed, 22048 insertions(+), 4367 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-platform-sst-atom create mode 100644 Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt create mode 100644 Documentation/devicetree/bindings/sound/cs35l34.txt create mode 100644 Documentation/devicetree/bindings/sound/cs42l42.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital.txt create mode 100755 Documentation/devicetree/bindings/sound/rt5665.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt create mode 100644 Documentation/sound/alsa/soc/codec_to_codec.txt create mode 100644 include/dt-bindings/sound/cs42l42.h create mode 100644 include/sound/cs35l34.h create mode 100644 include/sound/rt5514.h create mode 100755 include/sound/rt5665.h create mode 100644 sound/soc/atmel/tse850-pcm5142.c delete mode 100644 sound/soc/codecs/ak4641.h create mode 100644 sound/soc/codecs/cs35l34.c create mode 100644 sound/soc/codecs/cs35l34.h create mode 100644 sound/soc/codecs/cs42l42.c create mode 100644 sound/soc/codecs/cs42l42.h create mode 100644 sound/soc/codecs/msm8916-wcd-analog.c create mode 100644 sound/soc/codecs/msm8916-wcd-digital.c create mode 100644 sound/soc/codecs/rt5665.c create mode 100644 sound/soc/codecs/rt5665.h delete mode 100644 sound/soc/codecs/stac9766.h delete mode 100644 sound/soc/codecs/wm9705.h delete mode 100644 sound/soc/codecs/wm9712.h delete mode 100644 sound/soc/samsung/ac97.c delete mode 100644 sound/soc/samsung/ln2440sbc_alc650.c delete mode 100644 sound/soc/samsung/regs-ac97.h delete mode 100644 sound/soc/samsung/smdk2443_wm9710.c delete mode 100644 sound/soc/samsung/smdk_wm8580pcm.c delete mode 100644 sound/soc/samsung/smdk_wm9713.c create mode 100644 sound/soc/samsung/tm2_wm5110.c create mode 100644 sound/soc/sunxi/sun8i-codec-analog.c rename sound/soc/zte/{zx296702-i2s.c => zx-i2s.c} (100%) rename sound/soc/zte/{zx296702-spdif.c => zx-spdif.c} (99%)

2 1

[alsa-devel] [PATCH 2/2] ASoC: cs43130: Add devicetree bindings for CS43130
by Li Xu 12 Dec '16

12 Dec '16

Add devicetree bindings documentation file for Cirrus Logic CS43130 codec. Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- .../devicetree/bindings/sound/cs43130.txt | 41 ++++++++++++++++++++++ 1 file changed, 41 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs43130.txt diff --git a/Documentation/devicetree/bindings/sound/cs43130.txt b/Documentation/devicetree/bindings/sound/cs43130.txt new file mode 100644 index 0000000..9a2a22a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs43130.txt @@ -0,0 +1,41 @@ +CS43130 DAC + +Required properties: + + - compatible : "cirrus,cs43130" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply, VL-supply, VCP-supply, VD-supply: + power supplies for the device, as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + +Optional properties: + + - reset-gpios : gpio used to reset the Device + + - cirrus,xtal-ibias: + When external MCLK is generated by external crystal + oscillator, CS43130 can be used to provide bias current + for external crystal. Amount of bias current sent is + set as: + 1 = 7.5uA + 2 = 12.5uA + 3 = 15uA + +Example: + +cs43130: cs43130@30 { + compatible = "cirrus,cs43130"; + reg = <0x30>; + reset-gpios = <&axi_gpio 54 0>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + VL-supply = <&dummy_vreg>; + VCP-supply = <&dummy_vreg>; + VD-supply = <&dummy_vreg>; + cirrus,xtal-ibias = <2>; + interrupt-parent = <&gpio0>; + interrupts = <55 8>; +}; -- 1.9.1

2 2

[alsa-devel] [PATCH v2 1/2] ASoC: Add support for CS43130 codec
by Li Xu 12 Dec '16

12 Dec '16

Add support for Cirrus Logic CS43130 codec. Support I2C control and I2S audio playback. Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs43130.c | 1164 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs43130.h | 268 ++++++++++ 4 files changed, 1440 insertions(+) create mode 100644 sound/soc/codecs/cs43130.c create mode 100644 sound/soc/codecs/cs43130.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9e1718a..d6ede2b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -59,6 +59,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_I2C if I2C select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C + select SND_SOC_CS43130 if I2C select SND_SOC_CS4349 if I2C select SND_SOC_CS47L24 if MFD_CS47L24 select SND_SOC_CS53L30 if I2C @@ -473,6 +474,11 @@ config SND_SOC_CS42XX8_I2C select SND_SOC_CS42XX8 select REGMAP_I2C +# Cirrus Logic CS43130 HiFi DAC +config SND_SOC_CS43130 + tristate "Cirrus Logic CS43130 CODEC" + depends on I2C + # Cirrus Logic CS4349 HiFi DAC config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7e1dad7..2f15228 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -52,6 +52,7 @@ snd-soc-cs4271-i2c-objs := cs4271-i2c.o snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o +snd-soc-cs43130-objs := cs43130.o snd-soc-cs4349-objs := cs4349.o snd-soc-cs47l24-objs := cs47l24.o snd-soc-cs53l30-objs := cs53l30.o @@ -281,6 +282,7 @@ obj-$(CONFIG_SND_SOC_CS4271_I2C) += snd-soc-cs4271-i2c.o obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o +obj-$(CONFIG_SND_SOC_CS43130) += snd-soc-cs43130.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CS47L24) += snd-soc-cs47l24.o obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c new file mode 100644 index 0000000..79e4d2a --- /dev/null +++ b/sound/soc/codecs/cs43130.c @@ -0,0 +1,1164 @@ +/* + * cs43130.c -- CS43130 ALSA Soc Audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Authors: Li Xu <li.xu(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/pm_runtime.h> +#include <linux/of_irq.h> + +#include "cs43130.h" + + +static const struct reg_default cs43130_reg_defaults[] = { + { CS43130_SYS_CLK_CTL_1, 0x06 }, + { CS43130_SP_SRATE, 0x01 }, + { CS43130_SP_BITSIZE, 0x05 }, + { CS43130_PAD_INT_CFG, 0x03 }, + { CS43130_PWDN_CTL, 0xFE }, + { CS43130_CRYSTAL_SET, 0x04 }, + { CS43130_PLL_SET_1, 0x00 }, + { CS43130_PLL_SET_2, 0x00 }, + { CS43130_PLL_SET_3, 0x00 }, + { CS43130_PLL_SET_4, 0x00 }, + { CS43130_PLL_SET_5, 0x40 }, + { CS43130_PLL_SET_6, 0x10 }, + { CS43130_PLL_SET_7, 0x80 }, + { CS43130_PLL_SET_8, 0x03 }, + { CS43130_PLL_SET_9, 0x02 }, + { CS43130_PLL_SET_10, 0x02 }, + { CS43130_CLKOUT_CTL, 0x00 }, + { CS43130_ASP_NUM_1, 0x01 }, + { CS43130_ASP_NUM_2, 0x00 }, + { CS43130_ASP_DENOM_1, 0x08 }, + { CS43130_ASP_DENOM_2, 0x00 }, + { CS43130_ASP_LRCK_HI_TIME_1, 0x1F }, + { CS43130_ASP_LRCK_HI_TIME_2, 0x00 }, + { CS43130_ASP_LRCK_PERIOD_1, 0x3F }, + { CS43130_ASP_LRCK_PERIOD_2, 0x00 }, + { CS43130_ASP_CLOCK_CONF, 0x0C }, + { CS43130_ASP_FRAME_CONF, 0x0A }, + { CS43130_XSP_NUM_1, 0x01 }, + { CS43130_XSP_NUM_2, 0x00 }, + { CS43130_XSP_DENOM_1, 0x02 }, + { CS43130_XSP_DENOM_2, 0x00 }, + { CS43130_XSP_LRCK_HI_TIME_1, 0x1F }, + { CS43130_XSP_LRCK_HI_TIME_2, 0x00 }, + { CS43130_XSP_LRCK_PERIOD_1, 0x3F }, + { CS43130_XSP_LRCK_PERIOD_2, 0x00 }, + { CS43130_XSP_CLOCK_CONF, 0x0C }, + { CS43130_XSP_FRAME_CONF, 0x0A }, + { CS43130_ASP_CH_1_LOC, 0x00 }, + { CS43130_ASP_CH_2_LOC, 0x00 }, + { CS43130_ASP_CH_1_SZ_EN, 0x06 }, + { CS43130_ASP_CH_2_SZ_EN, 0x0E }, + { CS43130_XSP_CH_1_LOC, 0x00 }, + { CS43130_XSP_CH_2_LOC, 0x00 }, + { CS43130_XSP_CH_1_SZ_EN, 0x06 }, + { CS43130_XSP_CH_2_SZ_EN, 0x0E }, + { CS43130_DSD_VOL_B, 0x78 }, + { CS43130_DSD_VOL_A, 0x78 }, + { CS43130_DSD_PATH_CTL_1, 0xA8 }, + { CS43130_DSD_INT_CFG, 0x00 }, + { CS43130_DSD_PATH_CTL_2, 0x00 }, + { CS43130_DSD_PCM_MIX_CTL, 0x00 }, + { CS43130_DSD_PATH_CTL_3, 0x40 }, + { CS43130_HP_OUT_CTL_1, 0x30 }, + { CS43130_PCM_FILT_OPT, 0x02 }, + { CS43130_PCM_VOL_B, 0x78 }, + { CS43130_PCM_VOL_A, 0x78 }, + { CS43130_PCM_PATH_CTL_1, 0xA8 }, + { CS43130_PCM_PATH_CTL_2, 0x00 }, + { CS43130_CLASS_H_CTL, 0x1E }, + { CS43130_HP_DETECT, 0x04 }, + { CS43130_HP_LOAD_1, 0x00 }, + { CS43130_HP_MEAS_LOAD_1, 0x00 }, + { CS43130_HP_MEAS_LOAD_2, 0x00 }, + { CS43130_INT_MASK_1, 0xFF }, + { CS43130_INT_MASK_2, 0xFF }, + { CS43130_INT_MASK_3, 0xFF }, + { CS43130_INT_MASK_4, 0xFF }, + { CS43130_INT_MASK_5, 0xFF }, +}; + +static bool cs43130_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5: + return true; + default: + return false; + } +} + +static bool cs43130_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS43130_DEVID_AB ... CS43130_SYS_CLK_CTL_1: + case CS43130_SP_SRATE ... CS43130_PAD_INT_CFG: + case CS43130_DXD1: + case CS43130_PWDN_CTL: + case CS43130_DXD2: + case CS43130_CRYSTAL_SET: + case CS43130_PLL_SET_1 ... CS43130_PLL_SET_5: + case CS43130_PLL_SET_6: + case CS43130_PLL_SET_7: + case CS43130_PLL_SET_8: + case CS43130_PLL_SET_9: + case CS43130_PLL_SET_10: + case CS43130_CLKOUT_CTL: + case CS43130_ASP_NUM_1 ... CS43130_ASP_FRAME_CONF: + case CS43130_XSP_NUM_1 ... CS43130_XSP_FRAME_CONF: + case CS43130_ASP_CH_1_LOC: + case CS43130_ASP_CH_2_LOC: + case CS43130_ASP_CH_1_SZ_EN: + case CS43130_ASP_CH_2_SZ_EN: + case CS43130_XSP_CH_1_LOC: + case CS43130_XSP_CH_2_LOC: + case CS43130_XSP_CH_1_SZ_EN: + case CS43130_XSP_CH_2_SZ_EN: + case CS43130_DSD_VOL_B ... CS43130_DSD_PATH_CTL_3: + case CS43130_HP_OUT_CTL_1: + case CS43130_PCM_FILT_OPT ... CS43130_PCM_PATH_CTL_2: + case CS43130_CLASS_H_CTL: + case CS43130_HP_DETECT: + case CS43130_HP_STATUS: + case CS43130_HP_LOAD_1: + case CS43130_HP_MEAS_LOAD_1: + case CS43130_HP_MEAS_LOAD_2: + case CS43130_HP_DC_STAT_1: + case CS43130_HP_DC_STAT_2: + case CS43130_HP_AC_STAT_1: + case CS43130_HP_AC_STAT_2: + case CS43130_HP_LOAD_STAT: + case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5: + case CS43130_INT_MASK_1 ... CS43130_INT_MASK_5: + return true; + default: + return false; + } +} + +static bool cs43130_precious_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS43130_INT_STATUS_1 ... CS43130_INT_STATUS_5: + return true; + default: + return false; + } +} + +struct cs43130_pll_params { + u32 pll_in; + u8 mclk_int; + u8 sclk_prediv; + u8 pll_div_int; + u32 pll_div_frac; + u8 pll_mode; + u8 pll_divout; + u32 pll_out; + u8 pll_cal_ratio; +}; + +static const struct cs43130_pll_params pll_ratio_table[] = { + { 9600000, 1, 0x02, 0x49, 0x800000, 0x00, 0x08, 22579200, 151 }, + { 9600000, 0, 0x02, 0x50, 0x000000, 0x00, 0x08, 24576000, 164 }, + + { 11289600, 1, 0x02, 0X40, 0, 0x01, 0x08, 22579200, 128 }, + { 11289600, 0, 0x02, 0x44, 0x06F700, 0x0, 0x08, 24576000, 139 }, + + { 12000000, 1, 0x02, 0x49, 0x800000, 0x00, 0x0A, 22579200, 120 }, + { 12000000, 0, 0x02, 0x40, 0x000000, 0x00, 0x08, 24576000, 131 }, + + { 12288000, 1, 0x02, 0x49, 0x800000, 0x01, 0x0A, 22579200, 118 }, + { 12288000, 0, 0x02, 0x40, 0x000000, 0x01, 0x08, 24576000, 128 }, + + { 13000000, 1, 0x02, 0x49, 0x800000, 0x01, 0x0A, 22579200, 118 }, + { 13000000, 0, 0x02, 0x40, 0x000000, 0x01, 0x08, 24576000, 128 }, + + { 19200000, 1, 0x02, 0x45, 0x797680, 0x01, 0x0A, 22579200, 111 }, + { 19200000, 0, 0x02, 0x3C, 0x7EA940, 0x01, 0x08, 24576000, 121 }, + + { 22579200, 1, 0, 0, 0, 0, 0, 22579200, 0 }, + { 22579200, 0, 0x03, 0x44, 0x06F700, 0x00, 0x08, 24576000, 139 }, + + { 24000000, 1, 0x03, 0x49, 0x800000, 0x00, 0x0A, 22579200, 120 }, + { 24000000, 0, 0x03, 0x40, 0x000000, 0x00, 0x08, 24576000, 131 }, + + { 24576000, 1, 0x03, 0x49, 0x800000, 0x01, 0x0A, 22579200, 128 }, + { 24576000, 0, 0, 0, 0, 0, 0, 24576000, 0 }, + + { 26000000, 1, 0x03, 0x45, 0x797680, 0x01, 0x0A, 22579200, 111 }, + { 26000000, 0, 0x03, 0x3C, 0x7EA940, 0x01, 0x08, 24576000, 121 }, +}; + +static int cs43130_pll_config(struct snd_soc_codec *codec) +{ + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + int i; + + dev_dbg(codec->dev, "%s: cs43130->mclk = %d, cs43130->pll_out = %d", + __func__, cs43130->mclk, cs43130->pll_out); + for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) { + if (pll_ratio_table[i].pll_in == cs43130->mclk && + pll_ratio_table[i].pll_out == cs43130->pll_out) { + + cs43130->mclk_int = pll_ratio_table[i].mclk_int; + + if (pll_ratio_table[i].pll_cal_ratio == 0) { + if (cs43130->xtal_ibias > 0) { + usleep_range(1000, 1050); + /*PDN_XTAL = 0,enable*/ + regmap_update_bits(cs43130->regmap, + CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, + 0 << CS43130_PDN_XTAL_SHIFT); + } + + /* PLL_START = 0, disable PLL_START */ + regmap_update_bits(cs43130->regmap, + CS43130_PLL_SET_1, + CS43130_PLL_START_MASK, + 0 << CS43130_PLL_START_MASK); + + cs43130->pll_bypass = true; + return 0; + } + + /* PDN_PLL= 0,enable */ + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_PLL_MASK, + 0 << CS43130_PDN_PLL_SHIFT); + + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_9, + CS43130_PLL_REF_PREDIV_MASK, + pll_ratio_table[i].sclk_prediv); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_5, + pll_ratio_table[i].pll_div_int); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_2, + pll_ratio_table[i].pll_div_frac & + CS43130_7_0_MASK); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_3, + (pll_ratio_table[i].pll_div_frac & + CS43130_15_8_MASK) >> 8); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_4, + (pll_ratio_table[i].pll_div_frac & + CS43130_23_16_MASK) >> 16); + + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_8, + CS43130_PLL_MODE_MASK, + pll_ratio_table[i].pll_mode + << CS43130_PLL_MODE_SHIFT); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_6, + pll_ratio_table[i].pll_divout); + + regmap_write(cs43130->regmap, CS43130_PLL_SET_7, + pll_ratio_table[i].pll_cal_ratio); + + /* PLL_START = 1, enable PLL_START */ + regmap_update_bits(cs43130->regmap, CS43130_PLL_SET_1, + CS43130_PLL_START_MASK, CS43130_PLL_START_MASK); + cs43130->pll_bypass = false; + return 0; + } + } + return -EINVAL; +} + +static int cs43130_format_config(struct snd_soc_codec *codec) +{ + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + int period_size = 0; + int pulse_width = 0; + int asp_fsd; + int asp_stp; + int bick_inv; + int asp_m = 0; + int asp_sprate = 0; + int ret = 0; + unsigned int bitwidth_sclk = (cs43130->sclk / cs43130->fs) / 2; + unsigned int bitwidth_dai = (cs43130->dai_bit + 1) * 8; + + if (cs43130->fs) { + if (bitwidth_sclk < bitwidth_dai) { + dev_err(codec->dev, "Format not supported\n"); + return -EINVAL; + } + period_size = cs43130->sclk / cs43130->fs; + pulse_width = period_size/2; + + if (cs43130->sclk != 0) + asp_m = cs43130->pll_out / cs43130->sclk; + } + dev_dbg(codec->dev, "%s: cs43130->sclk = %d, cs43130->fs = %d, cs43130->dai_bit = %d", + __func__, cs43130->sclk, cs43130->fs, cs43130->dai_bit); + dev_dbg(codec->dev, "%s: period_size = %d, pulse_width = %d, asp_m = %d", + __func__, period_size, pulse_width, asp_m); + + if (cs43130->dai_format) { + /*MSB*/ + bick_inv = 0; + asp_fsd = 0; + asp_stp = 1; + + } else { + /*I2S*/ + bick_inv = 1; + asp_fsd = 2; /* one bick delay */ + asp_stp = 0; + } + dev_dbg(codec->dev, + "%s: cs43130->dai_format = %d, bick_inv = %d, asp_fsd = %d, asp_stp = %d", + __func__, cs43130->dai_format, bick_inv, asp_fsd, asp_stp); + + switch (cs43130->fs) { + case 32000: + asp_sprate = CS43130_ASP_SPRATE_32K; + break; + case 44100: + asp_sprate = CS43130_ASP_SPRATE_44_1K; + break; + case 48000: + asp_sprate = CS43130_ASP_SPRATE_48K; + break; + case 88200: + asp_sprate = CS43130_ASP_SPRATE_88_2K; + break; + case 96000: + asp_sprate = CS43130_ASP_SPRATE_96K; + break; + case 176400: + asp_sprate = CS43130_ASP_SPRATE_176_4K; + break; + case 192000: + asp_sprate = CS43130_ASP_SPRATE_192K; + break; + case 352800: + asp_sprate = CS43130_ASP_SPRATE_352_8K; + break; + case 384000: + asp_sprate = CS43130_ASP_SPRATE_384K; + break; + default: + dev_err(codec->dev, "sample rate(%d) not supported\n", + cs43130->fs); + return -EINVAL; + } + dev_dbg(codec->dev, "%s: asp_sprate = %d, cs43130->asp_size = %d", + __func__, asp_sprate, cs43130->asp_size); + + /* ASP_SPRATE = fs*/ + regmap_write(cs43130->regmap, CS43130_SP_SRATE, asp_sprate); + /*ASP_SPSIZE*/ + regmap_update_bits(cs43130->regmap, CS43130_SP_BITSIZE, + CS43130_SP_BITSIZE_ASP_MASK, cs43130->asp_size); + + + /* Set up slave mode */ + /*BICK = ASP_N/ASP_M * PLL_OUT */ + /* ASP_N = 1 */ + regmap_write(cs43130->regmap, CS43130_ASP_NUM_1, 1); + regmap_write(cs43130->regmap, CS43130_ASP_NUM_2, 0); + + /*ASP_M*/ + regmap_write(cs43130->regmap, CS43130_ASP_DENOM_1, asp_m & 0xff); + regmap_write(cs43130->regmap, CS43130_ASP_DENOM_2, (asp_m >> 8) & 0x3f); + + + /* H / L ratio of LRCK*/ + regmap_write(cs43130->regmap, CS43130_ASP_LRCK_HI_TIME_1, + (pulse_width-1) & 0xff); + regmap_write(cs43130->regmap, CS43130_ASP_LRCK_HI_TIME_2, + ((pulse_width-1) >> 8) & 0xff); + + /* the period of LRCK*/ + regmap_write(cs43130->regmap, CS43130_ASP_LRCK_PERIOD_1, + (period_size-1) & 0xff); + regmap_write(cs43130->regmap, CS43130_ASP_LRCK_PERIOD_2, + ((period_size-1) >> 8) & 0xff); + + /*resolution*/ + regmap_update_bits(cs43130->regmap, CS43130_ASP_CH_1_SZ_EN, + CS43130_SP_BITSIZE_ASP_MASK, cs43130->dai_bit); + regmap_update_bits(cs43130->regmap, CS43130_ASP_CH_2_SZ_EN, + CS43130_SP_BITSIZE_ASP_MASK, cs43130->dai_bit); + + regmap_update_bits(cs43130->regmap, CS43130_ASP_FRAME_CONF, + CS43130_ASP_FSD_MASK, asp_fsd << CS43130_ASP_FSD_SHIFT); + + regmap_update_bits(cs43130->regmap, CS43130_ASP_FRAME_CONF, + CS43130_ASP_STP_MASK, asp_stp << CS43130_ASP_STP_SHIFT); + + /* set clk master/slave */ + dev_dbg(codec->dev, "%s: cs43130->dai_mode = %d", + __func__, cs43130->dai_mode); + regmap_update_bits(cs43130->regmap, CS43130_ASP_CLOCK_CONF, + CS43130_ASP_MODE_MASK, cs43130->dai_mode << CS43130_ASP_MODE_SHIFT); + + return ret; +} + +static int cs43130_change_clksrc(struct snd_soc_codec *codec, + enum cs43130_mclk_src_sel src) +{ + int ret = 0; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_SRC_SEL_MASK, src << CS43130_MCLK_SRC_SEL_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, + CS43130_MCLK_INT_MASK, cs43130->mclk_int << CS43130_MCLK_INT_SHIFT); + + usleep_range(150, 200); + + return ret; +} + +static int cs43130_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + unsigned int bitwidth; + int ret = 0; + + cs43130->fs = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + cs43130->dai_bit = CS43130_SP_BIT_SIZE_8; + cs43130->asp_size = CS43130_SP_BIT_SIZE_32; + break; + case SNDRV_PCM_FORMAT_S16_LE: + cs43130->dai_bit = CS43130_SP_BIT_SIZE_16; + cs43130->asp_size = CS43130_SP_BIT_SIZE_24; + break; + case SNDRV_PCM_FORMAT_S24_LE: + cs43130->dai_bit = CS43130_SP_BIT_SIZE_24; + cs43130->asp_size = CS43130_SP_BIT_SIZE_16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + cs43130->dai_bit = CS43130_SP_BIT_SIZE_32; + cs43130->asp_size = CS43130_SP_BIT_SIZE_8; + break; + default: + dev_err(codec->dev, "Format(%d) not supported", + params_format(params)); + return -EINVAL; + } + + bitwidth = (cs43130->dai_bit+1)*8; + dev_dbg(codec->dev, "(data bit)%d: (rate)%d", + bitwidth, cs43130->fs); + + ret = cs43130_format_config(codec); + return ret; +} + +static const DECLARE_TLV_DB_SCALE(pcm_vol_tlv, -12750, 50, 1); + +static const struct snd_kcontrol_new cs43130_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS43130_PCM_VOL_A, CS43130_PCM_VOL_B, 0, 0xFF, 1, + pcm_vol_tlv), + SOC_SINGLE("Swap L/R", CS43130_PCM_PATH_CTL_2, 1, 1, 0), + SOC_SINGLE("Copy L/R", CS43130_PCM_PATH_CTL_2, 0, 1, 0), +}; + +static int cs43130_aspin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (cs43130->pll_bypass) + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_XTAL); + else + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_PLL); + + usleep_range(10000, 10050); + /* ASP_3ST = 0 in master mode */ + if (cs43130->dai_mode) + regmap_update_bits(cs43130->regmap, CS43130_PAD_INT_CFG, + 0x01, 0x00); + + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_CLKOUT_MASK, 0 + << CS43130_PDN_CLKOUT_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + break; + default: + dev_err(codec->dev, "Invalid ASPOUT event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static int cs43130_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + cs43130_change_clksrc(codec, CS43130_MCLK_SRC_RCO); + + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_XTAL_MASK, 1 + << CS43130_PDN_XTAL_SHIFT); + regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, + CS43130_PDN_PLL_MASK, 1 + << CS43130_PDN_PLL_SHIFT); + break; + default: + dev_err(codec->dev, "Invalid DAC event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static int cs43130_hpin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + regmap_write(cs43130->regmap, CS43130_DXD1, 0x99); + regmap_update_bits(cs43130->regmap, CS43130_HP_OUT_CTL_1, + CS43130_HP_IN_EN_MASK, 0 << CS43130_HP_IN_EN_SHIFT); + regmap_write(cs43130->regmap, CS43130_DXD2, 0x00); + regmap_write(cs43130->regmap, CS43130_DXD1, 0x00); + break; + case SND_SOC_DAPM_POST_PMU: + regmap_write(cs43130->regmap, CS43130_DXD1, 0x99); + regmap_write(cs43130->regmap, CS43130_DXD2, 0x01); + regmap_update_bits(cs43130->regmap, CS43130_HP_OUT_CTL_1, + CS43130_HP_IN_EN_MASK, 1 << CS43130_HP_IN_EN_SHIFT); + regmap_write(cs43130->regmap, CS43130_DXD1, 0x00); + break; + default: + dev_err(codec->dev, "Invalid HPIN event = 0x%x\n", event); + return -EINVAL; + } + return 0; +} + +static const struct snd_soc_dapm_widget cs43130_dapm_widgets[] = { + + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + + SND_SOC_DAPM_AIF_IN_E("ASPIN", NULL, 0, CS43130_PWDN_CTL, + CS43130_PDN_ASP_SHIFT, 1, cs43130_aspin_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + + SND_SOC_DAPM_DAC_E("HiFi DAC", + NULL, CS43130_PWDN_CTL, CS43130_PDN_HP_SHIFT, 1, + cs43130_dac_event, + (SND_SOC_DAPM_PRE_PMD) + ), + + SND_SOC_DAPM_LINE("Analog Playback", cs43130_hpin_event), +}; + +static const struct snd_soc_dapm_route cs43130_routes[] = { + {"ASPIN", NULL, "DAC Playback"}, + {"HiFi DAC", NULL, "ASPIN"}, + + {"HPOUTA", NULL, "HiFi DAC"}, + {"HPOUTB", NULL, "HiFi DAC"}, + {"HPOUTA", NULL, "Analog Playback"}, + {"HPOUTB", NULL, "Analog Playback"}, +}; + +static const unsigned int cs43130_src_rates[] = { + 32000, 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 +}; + +static const struct snd_pcm_hw_constraint_list cs43130_constraints = { + .count = ARRAY_SIZE(cs43130_src_rates), + .list = cs43130_src_rates, +}; + +static int cs43130_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs43130_constraints); + return 0; +} + +static int cs43130_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs43130->dai_mode = CS43130_SLAVE_MODE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs43130->dai_mode = CS43130_MASTER_MODE; + break; + default: + dev_err(codec->dev, "unsupported i2s master mode\n"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cs43130->dai_format = 0; + break; + case SND_SOC_DAIFMT_LEFT_J: + cs43130->dai_format = 1; + break; + default: + dev_err(codec->dev, "unsupported audio format except I2S and MSB\n"); + return -EINVAL; + } + + /* BICK/LRCK pority */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + cs43130->bick_invert = false; + cs43130->lrck_invert = false; + break; + case SND_SOC_DAIFMT_IB_NF: + cs43130->bick_invert = true; + cs43130->lrck_invert = false; + break; + case SND_SOC_DAIFMT_NB_IF: + cs43130->bick_invert = false; + cs43130->lrck_invert = true; + break; + case SND_SOC_DAIFMT_IB_IF: + cs43130->bick_invert = true; + cs43130->lrck_invert = true; + break; + default: + dev_err(codec->dev, "unsupported audio polarity\n"); + return -EINVAL; + } + + return 0; +} + + +static int cs43130_set_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + unsigned int reg; + u8 mute_reg; + + regmap_read(cs43130->regmap, CS43130_PCM_PATH_CTL_1, &reg); + mute_reg = reg & 0xfc; + if (mute) + regmap_write(cs43130->regmap, CS43130_PCM_PATH_CTL_1, + mute_reg | 0x03); + else + regmap_write(cs43130->regmap, CS43130_PCM_PATH_CTL_1, mute_reg); + + return ret; +} + +static int cs43130_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + + switch (div_id) { + case CS43130_AIF_BICK_RATE: + cs43130->bick = div; + break; + default: + dev_err(codec->dev, + "Unsupported divide value: div_id = %d", div_id); + return -EINVAL; + } + return 0; +} + +static int cs43130_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + int ret = 0; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + if (freq_in < 9600000 || freq_in > 26000000) { + dev_err(codec->dev, + "unsupported pll input reference clock:%d\n", freq_in); + return -EINVAL; + } + + switch (freq_in) { + case 9600000: + case 11289600: + case 12000000: + case 12288000: + case 13000000: + case 19200000: + case 22579200: + case 24000000: + case 24576000: + case 26000000: + cs43130->mclk = freq_in; + break; + default: + dev_err(codec->dev, + "unsupported pll input reference clock:%d\n", freq_in); + return -EINVAL; + } + + switch (freq_out) { + case 22579200: + cs43130->pll_out = freq_out; + cs43130->mclk_int = 1; + break; + case 24576000: + cs43130->pll_out = freq_out; + cs43130->mclk_int = 0; + break; + default: + dev_err(codec->dev, + "unsupported pll output reference clock:%d\n", + freq_out); + return -EINVAL; + } + + ret = cs43130_pll_config(codec); + dev_dbg(codec->dev, "%s: cs43130->pll_bypass = %d", + __func__, cs43130->pll_bypass); + return ret; +} + +static int cs43130_dai_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s: clk_id = %d, freq = %d, dir = %d", + __func__, clk_id, freq, dir); + cs43130->sclk = freq; + return 0; +} + +static const struct snd_soc_dai_ops cs43130_dai_ops = { + .startup = cs43130_pcm_startup, + .hw_params = cs43130_pcm_hw_params, + .set_sysclk = cs43130_dai_set_sysclk, + .set_fmt = cs43130_set_dai_fmt, + .digital_mute = cs43130_set_mute, + .set_clkdiv = cs43130_set_clkdiv, +}; + +static struct snd_soc_dai_driver cs43130_dai[] = { + { + .name = "cs43130_hifi", + .id = 0, + .playback = { + .stream_name = "DAC Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS43130_ASP_FORMATS, + }, + .ops = &cs43130_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "cs43130-xsp", + .id = 1, + .playback = { + .stream_name = "XSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS43130_XSP_FORMATS, + }, + .symmetric_rates = 1, + }, +}; + +static int cs43130_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + /* 24576000 is not supported */ + unsigned int mclk_int_freq = 22579200; + + dev_dbg(codec->dev, "%s: clk_id = %d, source = %d, freq = %d, dir = %d\n", + __func__, clk_id, source, freq, dir); + /* + * freq is external mclk freq + * if freq == mclk_int_freq, pll is bypassed + * modify mclk_int_freq as needed for application + */ + cs43130_set_pll(codec, 0, 0, freq, mclk_int_freq); + return 0; +} + +static irqreturn_t cs43130_irq_thread(int irq, void *data) +{ + struct cs43130_private *cs43130 = + (struct cs43130_private *)data; + struct snd_soc_codec *codec = cs43130->codec; + unsigned int stickies[CS43130_NUM_INT]; + unsigned int masks[CS43130_NUM_INT]; + unsigned int i; + + /* Read all INT status and mask reg */ + regmap_bulk_read(cs43130->regmap, CS43130_INT_STATUS_1, + stickies, CS43130_NUM_INT * sizeof(unsigned int)); + regmap_bulk_read(cs43130->regmap, CS43130_INT_MASK_1, + masks, CS43130_NUM_INT * sizeof(unsigned int)); + + for (i = 0; i < ARRAY_SIZE(stickies); i++) + stickies[i] = stickies[i] & (~masks[i]); + + if (stickies[0] & CS43130_XTAL_RDY_INT) + dev_dbg(codec->dev, "%s: Crystal ready", __func__); + + if (stickies[0] & CS43130_XTAL_ERR_INT) + dev_err(codec->dev, "%s: Crystal err", __func__); + + if (stickies[0] & CS43130_HP_PLUG_INT) + dev_dbg(codec->dev, "%s: HP plugged", __func__); + + if (stickies[0] & CS43130_HP_UNPLUG_INT) + dev_dbg(codec->dev, "%s: HP unplugged", __func__); + + return IRQ_HANDLED; +} + +static int cs43130_probe(struct snd_soc_codec *codec) +{ + struct cs43130_private *cs43130 = snd_soc_codec_get_drvdata(codec); + + cs43130->codec = codec; + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs43130 = { + .probe = cs43130_probe, + .component_driver = { + .controls = cs43130_snd_controls, + .num_controls = ARRAY_SIZE(cs43130_snd_controls), + .dapm_widgets = cs43130_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs43130_dapm_widgets), + .dapm_routes = cs43130_routes, + .num_dapm_routes = ARRAY_SIZE(cs43130_routes), + }, + .set_sysclk = cs43130_codec_set_sysclk, + .set_pll = cs43130_set_pll, +}; + +static const struct regmap_config cs43130_regmap = { + .reg_bits = 24, + .pad_bits = 8, + .val_bits = 8, + + .max_register = CS43130_LASTREG, + .reg_defaults = cs43130_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs43130_reg_defaults), + .readable_reg = cs43130_readable_register, + .precious_reg = cs43130_precious_register, + .volatile_reg = cs43130_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs43130_handle_device_data( + struct i2c_client *i2c_client, struct cs43130_private *cs43130) +{ + struct device_node *np = i2c_client->dev.of_node; + unsigned int val; + int ret = 0; + + of_property_read_u32(np, "cirrus,xtal-ibias", &val); + switch (val) { + case 1: + cs43130->xtal_ibias = CS43130_XTAL_IBIAS_7_5UA; + break; + case 2: + cs43130->xtal_ibias = CS43130_XTAL_IBIAS_12_5UA; + break; + case 3: + cs43130->xtal_ibias = CS43130_XTAL_IBIAS_15UA; + break; + default: + dev_info(&i2c_client->dev, + "cirrus,xtal-ibias value or xtal unused %d", + val); + } + return ret; +} + +static int cs43130_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs43130_private *cs43130; + int ret; + unsigned int devid = 0; + unsigned int reg; + int i; + + cs43130 = devm_kzalloc(&client->dev, sizeof(*cs43130), GFP_KERNEL); + if (cs43130 == NULL) + return -ENOMEM; + + i2c_set_clientdata(client, cs43130); + + cs43130->regmap = devm_regmap_init_i2c(client, &cs43130_regmap); + if (IS_ERR(cs43130->regmap)) { + ret = PTR_ERR(cs43130->regmap); + return ret; + } + + if (client->dev.of_node) { + ret = cs43130_handle_device_data(client, cs43130); + if (ret != 0) + return ret; + } + for (i = 0; i < ARRAY_SIZE(cs43130->supplies); i++) + cs43130->supplies[i].supply = cs43130_supply_names[i]; + + ret = devm_regulator_bulk_get(&client->dev, + ARRAY_SIZE(cs43130->supplies), + cs43130->supplies); + if (ret != 0) { + dev_err(&client->dev, + "Failed to request supplies: %d\n", ret); + return ret; + } + ret = regulator_bulk_enable(ARRAY_SIZE(cs43130->supplies), + cs43130->supplies); + if (ret != 0) { + dev_err(&client->dev, + "Failed to enable supplies: %d\n", ret); + return ret; + } + + cs43130->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs43130->reset_gpio)) + return PTR_ERR(cs43130->reset_gpio); + + gpiod_set_value_cansleep(cs43130->reset_gpio, 1); + + usleep_range(2000, 2050); + + /* initialize codec */ + ret = regmap_read(cs43130->regmap, CS43130_DEVID_AB, &reg); + + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs43130->regmap, CS43130_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs43130->regmap, CS43130_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + switch (devid) { + case CS43130_CHIP_ID: + break; + case CS4399_CHIP_ID: + break; + default: + dev_err(&client->dev, + "CS43130 Device ID (%X). Expected ID %X or %X\n", + devid, CS43130_CHIP_ID, CS4399_CHIP_ID); + ret = -ENODEV; + goto err; + } + + cs43130->dev_id = devid; + ret = regmap_read(cs43130->regmap, CS43130_REV_ID, &reg); + if (ret < 0) { + dev_err(&client->dev, "Get Revision ID failed\n"); + goto err; + } + + dev_info(&client->dev, + "Cirrus Logic CS43130 (%x), Revision: %02X\n", devid, + reg & 0xFF); + + /* Enable interrupt handler */ + ret = devm_request_threaded_irq(&client->dev, + client->irq, + NULL, cs43130_irq_thread, + IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs43130", cs43130); + if (ret != 0) { + dev_err(&client->dev, "Failed to request IRQ: %d\n", ret); + return ret; + } + + /* Unmask INT */ + regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, + CS43130_XTAL_RDY_INT | CS43130_XTAL_ERR_INT | + CS43130_HP_PLUG_INT | CS43130_HP_UNPLUG_INT, 0); + + /* Enable HP detect */ + regmap_update_bits(cs43130->regmap, CS43130_HP_DETECT, + CS43130_HP_DETECT_CTRL_MASK, CS43130_HP_DETECT_CTRL_MASK); + + regmap_write(cs43130->regmap, + CS43130_CRYSTAL_SET, cs43130->xtal_ibias); + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_cs43130, cs43130_dai, + ARRAY_SIZE(cs43130_dai)); + + if (ret < 0) { + dev_err(&client->dev, + "%s: snd_soc_register_codec failed with ret = %d\n", + __func__, ret); + goto err; + } + return 0; +err: + return ret; + +} + +static int cs43130_i2c_remove(struct i2c_client *client) +{ + struct cs43130_private *cs43130 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + if (cs43130->reset_gpio) + gpiod_set_value_cansleep(cs43130->reset_gpio, 0); + + pm_runtime_disable(&client->dev); + regulator_bulk_disable(CS43130_NUM_SUPPLIES, + cs43130->supplies); + + return 0; +} + +#ifdef CONFIG_PM +static int cs43130_runtime_suspend(struct device *dev) +{ + struct cs43130_private *cs43130 = dev_get_drvdata(dev); + + regcache_cache_only(cs43130->regmap, true); + regcache_mark_dirty(cs43130->regmap); + + gpiod_set_value_cansleep(cs43130->reset_gpio, 0); + + regulator_bulk_disable(CS43130_NUM_SUPPLIES, + cs43130->supplies); + return 0; +} + +static int cs43130_runtime_resume(struct device *dev) +{ + struct cs43130_private *cs43130 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(CS43130_NUM_SUPPLIES, + cs43130->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + regcache_cache_only(cs43130->regmap, false); + + gpiod_set_value_cansleep(cs43130->reset_gpio, 1); + + usleep_range(2000, 2050); + + ret = regcache_sync(cs43130->regmap); + if (ret != 0) { + dev_err(dev, "Failed to restore register cache\n"); + goto err; + } + return 0; +err: + regcache_cache_only(cs43130->regmap, true); + regulator_bulk_disable(CS43130_NUM_SUPPLIES, + cs43130->supplies); + + return ret; +} +#endif + +static const struct dev_pm_ops cs43130_runtime_pm = { + SET_RUNTIME_PM_OPS(cs43130_runtime_suspend, cs43130_runtime_resume, + NULL) +}; + +static const struct of_device_id cs43130_of_match[] = { + { .compatible = "cirrus,cs43130", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs43130_of_match); + +static const struct i2c_device_id cs43130_i2c_id[] = { + {"cs43130", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs43130_i2c_id); + +static struct i2c_driver cs43130_i2c_driver = { + .driver = { + .name = "cs43130", + .of_match_table = cs43130_of_match, + }, + .id_table = cs43130_i2c_id, + .probe = cs43130_i2c_probe, + .remove = cs43130_i2c_remove, +}; + +module_i2c_driver(cs43130_i2c_driver); + +MODULE_AUTHOR("Li Xu <li.xu(a)cirrus.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS43130 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h new file mode 100644 index 0000000..bceae76 --- /dev/null +++ b/sound/soc/codecs/cs43130.h @@ -0,0 +1,268 @@ +/* + * ALSA SoC CS43130 codec driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Li Xu <li.xu(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#ifndef __CS43130_H__ +#define __CS43130_H__ + +/* CS43130 registers addresses */ +/* all reg address is shifted by a byte for control byte to be LSB */ +#define CS43130_FIRSTREG 0x010000 +#define CS43130_LASTREG 0x0F0014 +#define CS43130_CHIP_ID 0x00043130 +#define CS4399_CHIP_ID 0x00043990 +#define CS43130_DEVID_AB 0x010000 /*Device ID A & B [RO]*/ +#define CS43130_DEVID_CD 0x010001 /*Device ID C & D [RO]*/ +#define CS43130_DEVID_E 0x010002 /*Device ID E [RO]*/ +#define CS43130_FAB_ID 0x010003 /*Fab ID [RO]*/ +#define CS43130_REV_ID 0x010004 /*Revision ID [RO]*/ +#define CS43130_SUBREV_ID 0x010005 /*Subrevision ID*/ +#define CS43130_SYS_CLK_CTL_1 0x010006 /*System Clocking Ctl 1*/ +#define CS43130_SP_SRATE 0x01000B /*Serial Port Sample Rate*/ +#define CS43130_SP_BITSIZE 0x01000C /*Serial Port Bit Size*/ +#define CS43130_PAD_INT_CFG 0x01000D /*Pad Interface Config*/ +#define CS43130_DXD1 0x010010 /*DXD1*/ +#define CS43130_PWDN_CTL 0x020000 /*Power Down Ctl*/ +#define CS43130_DXD2 0x020019 /*DXD2*/ +#define CS43130_CRYSTAL_SET 0x020052 /*Crystal Setting*/ +#define CS43130_PLL_SET_1 0x030001 /*PLL Setting 1*/ +#define CS43130_PLL_SET_2 0x030002 /*PLL Setting 2*/ +#define CS43130_PLL_SET_3 0x030003 /*PLL Setting 3*/ +#define CS43130_PLL_SET_4 0x030004 /*PLL Setting 4*/ +#define CS43130_PLL_SET_5 0x030005 /*PLL Setting 5*/ +#define CS43130_PLL_SET_6 0x030008 /*PLL Setting 6*/ +#define CS43130_PLL_SET_7 0x03000A /*PLL Setting 7*/ +#define CS43130_PLL_SET_8 0x03001B /*PLL Setting 8*/ +#define CS43130_PLL_SET_9 0x040002 /*PLL Setting 9*/ +#define CS43130_PLL_SET_10 0x040003 /*PLL Setting 10*/ +#define CS43130_CLKOUT_CTL 0x040004 /*CLKOUT Ctl*/ +#define CS43130_ASP_NUM_1 0x040010 /*ASP Numerator 1*/ +#define CS43130_ASP_NUM_2 0x040011 /*ASP Numerator 2*/ +#define CS43130_ASP_DENOM_1 0x040012 /*ASP Denominator 1*/ +#define CS43130_ASP_DENOM_2 0x040013 /*ASP Denominator 2*/ +#define CS43130_ASP_LRCK_HI_TIME_1 0x040014 /*ASP LRCK High Time 1*/ +#define CS43130_ASP_LRCK_HI_TIME_2 0x040015 /*ASP LRCK High Time 2*/ +#define CS43130_ASP_LRCK_PERIOD_1 0x040016 /*ASP LRCK Period 1*/ +#define CS43130_ASP_LRCK_PERIOD_2 0x040017 /*ASP LRCK Period 2*/ +#define CS43130_ASP_CLOCK_CONF 0x040018 /*ASP Clock Config*/ +#define CS43130_ASP_FRAME_CONF 0x040019 /*ASP Frame Config*/ +#define CS43130_XSP_NUM_1 0x040020 /*XSP Numerator 1*/ +#define CS43130_XSP_NUM_2 0x040021 /*XSP Numerator 2*/ +#define CS43130_XSP_DENOM_1 0x040022 /*XSP Denominator 1*/ +#define CS43130_XSP_DENOM_2 0x040023 /*XSP Denominator 2*/ +#define CS43130_XSP_LRCK_HI_TIME_1 0x040024 /*XSP LRCK High Time 1*/ +#define CS43130_XSP_LRCK_HI_TIME_2 0x040025 /*XSP LRCK High Time 2*/ +#define CS43130_XSP_LRCK_PERIOD_1 0x040026 /*XSP LRCK Period 1*/ +#define CS43130_XSP_LRCK_PERIOD_2 0x040027 /*XSP LRCK Period 2*/ +#define CS43130_XSP_CLOCK_CONF 0x040028 /*XSP Clock Config*/ +#define CS43130_XSP_FRAME_CONF 0x040029 /*XSP Frame Config*/ +#define CS43130_ASP_CH_1_LOC 0x050000 /*ASP Chan 1 Location*/ +#define CS43130_ASP_CH_2_LOC 0x050001 /*ASP Chan 2 Location*/ +#define CS43130_ASP_CH_1_SZ_EN 0x05000A /*ASP Chan 1 Size, Enable*/ +#define CS43130_ASP_CH_2_SZ_EN 0x05000B /*ASP Chan 2 Size, Enable*/ +#define CS43130_XSP_CH_1_LOC 0x060000 /*XSP Chan 1 Location*/ +#define CS43130_XSP_CH_2_LOC 0x060001 /*XSP Chan 2 Location*/ +#define CS43130_XSP_CH_1_SZ_EN 0x06000A /*XSP Chan 1 Size, Enable*/ +#define CS43130_XSP_CH_2_SZ_EN 0x06000B /*XSP Chan 2 Size, Enable*/ +#define CS43130_DSD_VOL_B 0x070000 /*DSD Volume B*/ +#define CS43130_DSD_VOL_A 0x070001 /*DSD Volume A*/ +#define CS43130_DSD_PATH_CTL_1 0x070002 /*DSD Proc Path Sig Ctl 1*/ +#define CS43130_DSD_INT_CFG 0x070003 /*DSD Interface Config*/ +#define CS43130_DSD_PATH_CTL_2 0x070004 /*DSD Proc Path Sig Ctl 2*/ +#define CS43130_DSD_PCM_MIX_CTL 0x070005 /*DSD and PCM Mixing Ctl*/ +#define CS43130_DSD_PATH_CTL_3 0x070006 /*DSD Proc Path Sig Ctl 3*/ +#define CS43130_HP_OUT_CTL_1 0x080000 /*HP Output Ctl 1*/ +#define CS43130_PCM_FILT_OPT 0x090000 /*PCM Filter Option*/ +#define CS43130_PCM_VOL_B 0x090001 /*PCM Volume B*/ +#define CS43130_PCM_VOL_A 0x090002 /*PCM Volume A*/ +#define CS43130_PCM_PATH_CTL_1 0x090003 /*PCM Path Signal Ctl 1*/ +#define CS43130_PCM_PATH_CTL_2 0x090004 /*PCM Path Signal Ctl 2*/ +#define CS43130_CLASS_H_CTL 0x0B0000 /*Class H Ctl*/ +#define CS43130_HP_DETECT 0x0D0000 /*HP Detect*/ +#define CS43130_HP_STATUS 0x0D0001 /*HP Status [RO]*/ +#define CS43130_HP_LOAD_1 0x0E0000 /*HP Load 1*/ +#define CS43130_HP_MEAS_LOAD_1 0x0E0003 /*HP Load Measurement 1*/ +#define CS43130_HP_MEAS_LOAD_2 0x0E0004 /*HP Load Measurement 2*/ +#define CS43130_HP_DC_STAT_1 0x0E000D /*HP DC Load Status 0 [RO]*/ +#define CS43130_HP_DC_STAT_2 0x0E000E /*HP DC Load Status 1 [RO]*/ +#define CS43130_HP_AC_STAT_1 0x0E0010 /*HP AC Load Status 0 [RO]*/ +#define CS43130_HP_AC_STAT_2 0x0E0011 /*HP AC Load Status 1 [RO]*/ +#define CS43130_HP_LOAD_STAT 0x0E001A /*HP Load Status [RO]*/ +#define CS43130_INT_STATUS_1 0x0F0000 /*Interrupt Status 1*/ +#define CS43130_INT_STATUS_2 0x0F0001 /*Interrupt Status 2*/ +#define CS43130_INT_STATUS_3 0x0F0002 /*Interrupt Status 3*/ +#define CS43130_INT_STATUS_4 0x0F0003 /*Interrupt Status 4*/ +#define CS43130_INT_STATUS_5 0x0F0004 /*Interrupt Status 5*/ +#define CS43130_INT_MASK_1 0x0F0010 /*Interrupt Mask 1*/ +#define CS43130_INT_MASK_2 0x0F0011 /*Interrupt Mask 2*/ +#define CS43130_INT_MASK_3 0x0F0012 /*Interrupt Mask 3*/ +#define CS43130_INT_MASK_4 0x0F0013 /*Interrupt Mask 4*/ +#define CS43130_INT_MASK_5 0x0F0014 /*Interrupt Mask 5*/ + +#define CS43130_MCLK_SRC_SEL_MASK 0x03 +#define CS43130_MCLK_SRC_SEL_SHIFT 0 +#define CS43130_MCLK_INT_MASK 0x04 +#define CS43130_MCLK_INT_SHIFT 2 +#define CS43130_SP_SRATE_MASK 0x0F +#define CS43130_SP_SRATE_SHIFT 0 +#define CS43130_SP_BITSIZE_ASP_MASK 0x03 +#define CS43130_SP_BITSIZE_ASP_SHIFT 0 +#define CS43130_HP_DETECT_CTRL_SHIFT 6 +#define CS43130_HP_DETECT_CTRL_MASK (0x03 << CS43130_HP_DETECT_CTRL_SHIFT) +#define CS43130_HP_DETECT_INV_SHIFT 5 +#define CS43130_HP_DETECT_INV_MASK (1 << CS43130_HP_DETECT_INV_SHIFT) + +/* CS43130_INT_MASK_1 */ +#define CS43130_HP_PLUG_INT_SHIFT 6 +#define CS43130_HP_PLUG_INT (1 << CS43130_HP_PLUG_INT_SHIFT) +#define CS43130_HP_UNPLUG_INT_SHIFT 5 +#define CS43130_HP_UNPLUG_INT (1 << CS43130_HP_UNPLUG_INT_SHIFT) +#define CS43130_XTAL_RDY_INT_SHIFT 4 +#define CS43130_XTAL_RDY_INT (1 << CS43130_XTAL_RDY_INT_SHIFT) +#define CS43130_XTAL_ERR_INT_SHIFT 3 +#define CS43130_XTAL_ERR_INT (1 << CS43130_XTAL_ERR_INT_SHIFT) + +/*Reg CS43130_SP_BITSIZE*/ +#define CS43130_SP_BIT_SIZE_8 0x00 +#define CS43130_SP_BIT_SIZE_16 0x01 +#define CS43130_SP_BIT_SIZE_24 0x02 +#define CS43130_SP_BIT_SIZE_32 0x03 + +/*PLL*/ +#define CS43130_PLL_START_MASK (0x1<<0) +#define CS43130_PLL_MODE_MASK 0x02 +#define CS43130_PLL_MODE_SHIFT 1 + +#define CS43130_PLL_REF_PREDIV_MASK 0x3 + +#define CS43130_ASP_STP_MASK 0x10 +#define CS43130_ASP_STP_SHIFT 4 +#define CS43130_ASP_5050_MASK 0x08 +#define CS43130_ASP_5050_SHIFT 3 +#define CS43130_ASP_FSD_MASK 0x07 +#define CS43130_ASP_FSD_SHIFT 0 + +#define CS43130_ASP_MODE_MASK 0x10 +#define CS43130_ASP_MODE_SHIFT 4 +#define CS43130_ASP_SCPOL_OUT_MASK 0x08 +#define CS43130_ASP_SCPOL_OUT_SHIFT 3 +#define CS43130_ASP_SCPOL_IN_MASK 0x04 +#define CS43130_ASP_SCPOL_IN_SHIFT 2 +#define CS43130_ASP_LCPOL_OUT_MASK 0x02 +#define CS43130_ASP_LCPOL_OUT_SHIFT 1 +#define CS43130_ASP_LCPOL_IN_MASK 0x01 +#define CS43130_ASP_LCPOL_IN_SHIFT 0 + +/*Reg CS43130_PWDN_CTL*/ +#define CS43130_PDN_XSP_MASK 0x80 +#define CS43130_PDN_XSP_SHIFT 7 +#define CS43130_PDN_ASP_MASK 0x40 +#define CS43130_PDN_ASP_SHIFT 6 +#define CS43130_PDN_DSPIF_MASK 0x20 +#define CS43130_PDN_DSDIF_SHIFT 5 +#define CS43130_PDN_HP_MASK 0x10 +#define CS43130_PDN_HP_SHIFT 4 +#define CS43130_PDN_XTAL_MASK 0x08 +#define CS43130_PDN_XTAL_SHIFT 3 +#define CS43130_PDN_PLL_MASK 0x04 +#define CS43130_PDN_PLL_SHIFT 2 +#define CS43130_PDN_CLKOUT_MASK 0x02 +#define CS43130_PDN_CLKOUT_SHIFT 1 + +#define CS43130_7_0_MASK 0xFF +#define CS43130_15_8_MASK 0xFF00 +#define CS43130_23_16_MASK 0xFF0000 + +/* Reg CS43130_HP_OUT_CTL_1 */ +#define CS43130_HP_IN_EN_SHIFT 3 +#define CS43130_HP_IN_EN_MASK 0x08 + +#define CS43130_ASP_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define CS43130_XSP_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +enum cs43130_asp_rate { + CS43130_ASP_SPRATE_32K = 0, + CS43130_ASP_SPRATE_44_1K, + CS43130_ASP_SPRATE_48K, + CS43130_ASP_SPRATE_88_2K, + CS43130_ASP_SPRATE_96K, + CS43130_ASP_SPRATE_176_4K, + CS43130_ASP_SPRATE_192K, + CS43130_ASP_SPRATE_352_8K, + CS43130_ASP_SPRATE_384K, +}; + +enum cs43130_mclk_src_sel { + CS43130_MCLK_SRC_XTAL = 0, + CS43130_MCLK_SRC_PLL, + CS43130_MCLK_SRC_RCO +}; + +enum cs43130_mode { + CS43130_SLAVE_MODE = 0, + CS43130_MASTER_MODE +}; + +enum cs43130_xtal_ibias { + CS43130_XTAL_IBIAS_15UA = 2, + CS43130_XTAL_IBIAS_12_5UA = 4, + CS43130_XTAL_IBIAS_7_5UA = 6, +}; + +#define CS43130_AIF_BICK_RATE 1 +#define CS43130_SYSCLK_MCLK 1 +#define CS43130_NUM_SUPPLIES 5 +static const char *const cs43130_supply_names[CS43130_NUM_SUPPLIES] = { + "VA", + "VP", + "VCP", + "VD", + "VL", +}; + +#define CS43130_NUM_INT 5 /* number of interrupt status reg */ + +struct cs43130_private { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct regulator_bulk_data supplies[CS43130_NUM_SUPPLIES]; + /* codec device ID */ + unsigned int dev_id; + int mclk; + int sclk; + int xtal_ibias; + + bool pll_bypass; + int pll_out; + int mclk_int; + int dai_format; + int dai_mode; + int dai_bit; + int asp_size; + int fs; + bool bick_invert; + bool lrck_invert; + int bick; + struct gpio_desc *reset_gpio; +}; + +#endif /* __CS43130_H__ */ -- 1.9.1

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[alsa-devel] [PATCH] ASoC: topology: avoid uninitialized kcontrol_type
by Arnd Bergmann 12 Dec '16

12 Dec '16

When num_kcontrols is zero, widget->dobj.widget.kcontrol_type gets set to an uninitialized local variable: sound/soc/soc-topology.c: In function 'soc_tplg_dapm_widget_create': sound/soc/soc-topology.c:1566:36: error: 'kcontrol_type' may be used uninitialized in this function [-Werror=maybe-uninitialized] I could not figure out which of the valid types would be appropriate here, so this sets it to '0', which is invalid but at least well-defined here. There is probably a better way to address the issue. Fixes: eea3dd4f1247 ("ASoC: topology: Only free TLV for volume mixers of a widget") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/soc-topology.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 11feb19e9730..65670b2b408c 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1485,6 +1485,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, tplg->pos += (sizeof(struct snd_soc_tplg_dapm_widget) + w->priv.size); if (w->num_kcontrols == 0) { + kcontrol_type = 0; template.num_kcontrols = 0; goto widget; } -- 2.9.0

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Re: [alsa-devel] [PATCH 0/7] Move dell-led to drivers/platform/x86
by Mario.Limonciello＠dell.com 11 Dec '16

11 Dec '16

add Anthony Wong @ Canonical. I'm about to go on break and won't have access to hardware for a while. Anthony, Your team has this hardware more readily available than I do. Would you be able to have someone on the team verify this series?

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[alsa-devel] [asoc:topic/cq93vc 1/1] include/linux/mfd/davinci_voicecodec.h:31:27: fatal error: mach/hardware.h: No such file or directory
by kbuild test robot 11 Dec '16

11 Dec '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/cq93vc head: 4de429888c5b534f43a3d693e3627021b17e900c commit: 4de429888c5b534f43a3d693e3627021b17e900c [1/1] ASoC: cq93vc: remove MFD_DAVINCI_VOICECODEC dependency for CQ0093VC config: xtensa-allmodconfig (attached as .config) compiler: xtensa-linux-gcc (GCC) 4.9.0 reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 4de429888c5b534f43a3d693e3627021b17e900c # save the attached .config to linux build tree make.cross ARCH=xtensa All errors (new ones prefixed by >>): In file included from sound/soc/codecs/cq93vc.c:32:0: >> include/linux/mfd/davinci_voicecodec.h:31:27: fatal error: mach/hardware.h: No such file or directory #include <mach/hardware.h> ^ compilation terminated. vim +31 include/linux/mfd/davinci_voicecodec.h ca26308c Miguel Aguilar 2010-03-11 15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ca26308c Miguel Aguilar 2010-03-11 16 * GNU General Public License for more details. ca26308c Miguel Aguilar 2010-03-11 17 * ca26308c Miguel Aguilar 2010-03-11 18 * You should have received a copy of the GNU General Public License ca26308c Miguel Aguilar 2010-03-11 19 * along with this program; if not, write to the Free Software ca26308c Miguel Aguilar 2010-03-11 20 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA ca26308c Miguel Aguilar 2010-03-11 21 */ ca26308c Miguel Aguilar 2010-03-11 22 ca26308c Miguel Aguilar 2010-03-11 23 #ifndef __LINUX_MFD_DAVINCI_VOICECODEC_H_ c593aca4 Rasmus Villemoes 2014-08-22 24 #define __LINUX_MFD_DAVINCI_VOICECODEC_H_ ca26308c Miguel Aguilar 2010-03-11 25 ca26308c Miguel Aguilar 2010-03-11 26 #include <linux/kernel.h> ca26308c Miguel Aguilar 2010-03-11 27 #include <linux/platform_device.h> ca26308c Miguel Aguilar 2010-03-11 28 #include <linux/mfd/core.h> 3ad7a42d Matt Porter 2013-03-06 29 #include <linux/platform_data/edma.h> ca26308c Miguel Aguilar 2010-03-11 30 99717470 Sachin Kamat 2013-06-18 @31 #include <mach/hardware.h> ca26308c Miguel Aguilar 2010-03-11 32 921a2c87 Mark Brown 2013-08-31 33 struct regmap; 921a2c87 Mark Brown 2013-08-31 34 ca26308c Miguel Aguilar 2010-03-11 35 /* ca26308c Miguel Aguilar 2010-03-11 36 * Register values. ca26308c Miguel Aguilar 2010-03-11 37 */ ca26308c Miguel Aguilar 2010-03-11 38 #define DAVINCI_VC_PID 0x00 ca26308c Miguel Aguilar 2010-03-11 39 #define DAVINCI_VC_CTRL 0x04 :::::: The code at line 31 was first introduced by commit :::::: 997174705458d2abdbc31ba1594bf2a4503cb41a mfd: davinci_voicecodec: Fix build breakage :::::: TO: Sachin Kamat <sachin.kamat(a)linaro.org> :::::: CC: Samuel Ortiz <sameo(a)linux.intel.com> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH SND/USB]: Add QuickCam Communicate Deluxe/S7500 to volume_control_quirks.
by Con Kolivas 09 Dec '16

09 Dec '16

The Logitech QuickCam Communicate Deluxe/S7500 microphone fails with the following warning. [ 6.778995] usb 2-1.2.2.2: Warning! Unlikely big volume range (=3072), cval->res is probably wrong. [ 6.778996] usb 2-1.2.2.2: [5] FU [Mic Capture Volume] ch = 1, val = 4608/7680/1 Adding it to the list of devices in volume_control_quirks makes it work properly, fixing related typo. Signed-off-by: Con Kolivas <kernel(a)kolivas.org> --- sound/usb/mixer.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 2f8c388..4703cae 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -932,9 +932,10 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ case USB_ID(0x046d, 0x08ca): /* Logitech Quickcam Fusion */ case USB_ID(0x046d, 0x0991): + case USB_ID(0x046d, 0x09a2): /* QuickCam Communicate Deluxe/S7500 */ /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384. - * Proboly there is some logitech magic behind this number --fishor + * Probably there is some logitech magic behind this number --fishor */ if (!strcmp(kctl->id.name, "Mic Capture Volume")) { usb_audio_info(chip, -- 2.7.4

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[alsa-devel] [PATCH V3 2/2] ASoC: samsung: add GPIOLIB dependency
by Fabian Frederick 09 Dec '16

09 Dec '16

Both SND_SOC_SMARTQ and SND_SOC_SAMSUNG_TM2_WM5110 use gpio/consumer.h This patch adds GPIOLIB || COMPILE_TEST to Kconfig entries to fix runtime dependency. See commit 638f958baeaf ("extcon: Allow compile test of GPIO consumers if !GPIOLIB") for similar problem and explanations. Reviewed-by: Krzysztof Kozlowski <krzk(a)kernel.org> Reported-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- V3: -Fix commit message (suggested by Krzysztof Kozlowski) -Add reviewed-by sound/soc/samsung/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 7c42315..f1f1d79 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -111,6 +111,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SMARTQ tristate "SoC I2S Audio support for SmartQ board" depends on MACH_SMARTQ || COMPILE_TEST + depends on GPIOLIB || COMPILE_TEST depends on I2C select SND_SAMSUNG_I2S select SND_SOC_WM8750 @@ -193,6 +194,7 @@ config SND_SOC_ARNDALE_RT5631_ALC5631 config SND_SOC_SAMSUNG_TM2_WM5110 tristate "SoC I2S Audio support for WM5110 on TM2 board" depends on SND_SOC_SAMSUNG && MFD_ARIZONA && I2C && SPI_MASTER + depends on GPIOLIB || COMPILE_TEST select SND_SOC_MAX98504 select SND_SOC_WM5110 select SND_SAMSUNG_I2S -- 2.7.4

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[alsa-devel] [PATCH V3 1/2] ASoC: samsung: include gpio consumer.h
by Fabian Frederick 09 Dec '16

09 Dec '16

Fix the following build errors on X86_32 !GPIOLIB sound/soc/samsung/tm2_wm5110.c:220:3: error: implicit declaration of function 'gpiod_set_value_cansleep' [-Werror=implicit-function-declaration] sound/soc/samsung/tm2_wm5110.c:438:24: error: implicit declaration of function 'devm_gpiod_get' [-Werror=implicit-function-declaration] Reviewed-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- V3: -Add reviewed-by V2: -Add configuration settings (suggested by Krzysztof Kozlowski) sound/soc/samsung/tm2_wm5110.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 5cdf7d1..24cc9d6 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -12,6 +12,7 @@ #include <linux/clk.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> #include <sound/pcm_params.h> -- 2.7.4

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[alsa-devel] [PATCH v5 0/2] Add platform clock for BayTrail platforms
by Irina Tirdea 09 Dec '16

09 Dec '16

These patches specifically enable the audio MCLK required by Baytrail CR devices. It is the remaining part of a bigger set of patches (already merged in Mark's tree) that enable sound for Baytrail CR devices (especially Asus T100TAF) [1]. They include the clock driver and a move of the non-architectural pmc_atom driver code into drivers/platform/x86. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/111704.html Changes from v4: - move the pmc_atom driver from arch/x86/platform/atom to drivers/platform/x86 Changes from v3: - replace devm_kzalloc with devm_kcalloc - add x86 architecture maintainers Changes from v2: - move clk platform data structures to a separate include file - store clk_hw pointer for the fixed rate clocks Changes from v1: - register the clk device as child of pmc device - pass iomem pointer from pmc driver to clk driver to avoid using pmc_atom_read()/write() and use readl/writel API instead - use devm_clk_hw_register/clkdev_hw_create instead of clk_register/clkdev_create Irina Tirdea (2): arch/x86/platform/atom: Move pmc_atom to drivers/platform/x86 clk: x86: Add Atom PMC platform clocks arch/x86/Kconfig | 4 - arch/x86/platform/atom/Makefile | 1 - drivers/acpi/acpi_lpss.c | 2 +- drivers/clk/x86/Makefile | 1 + drivers/clk/x86/clk-byt-plt.c | 380 +++++++++++++++++++++ drivers/platform/x86/Kconfig | 6 + drivers/platform/x86/Makefile | 1 + .../atom => drivers/platform/x86}/pmc_atom.c | 81 ++++- include/linux/platform_data/x86/clk-byt-plt.h | 31 ++ .../linux/platform_data/x86}/pmc_atom.h | 3 + 10 files changed, 499 insertions(+), 11 deletions(-) create mode 100644 drivers/clk/x86/clk-byt-plt.c rename {arch/x86/platform/atom => drivers/platform/x86}/pmc_atom.c (88%) create mode 100644 include/linux/platform_data/x86/clk-byt-plt.h rename {arch/x86/include/asm => include/linux/platform_data/x86}/pmc_atom.h (98%) -- 1.9.1

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[alsa-devel] [alsa-utils][PATCH 0/5] Add man page for alsaucm + other fixes
by Antonio Ospite 09 Dec '16

09 Dec '16

Hi, while adding the alsaucm man page (patch 4/5) I noticed some other things worth fixing, so here is the whole patch set. In patch 4/5 the reStructuredText content is exactly the same reviewed by Liam, so I put his ACK in the commit message. Thanks, Antonio Antonio Ospite (5): alsaucm: mention the "list1" command in the usage output configure.ac: fix the check for xmlto availability configure.ac: add a check for rst2man, a reStructuredText man page generator alsaucm: add a man page, generated from reStructuredText INSTALL: document how to configure a build for installation in a local dir .gitignore | 1 + INSTALL | 7 ++ alsaucm/Makefile.am | 7 ++ alsaucm/alsaucm.rst | 235 ++++++++++++++++++++++++++++++++++++++++++++++++++++ alsaucm/usecase.c | 3 +- configure.ac | 15 +++- 6 files changed, 264 insertions(+), 4 deletions(-) create mode 100644 alsaucm/alsaucm.rst -- Antonio Ospite https://ao2.it https://twitter.com/ao2it A: Because it messes up the order in which people normally read text. See http://en.wikipedia.org/wiki/Posting_style Q: Why is top-posting such a bad thing?

2 7

[alsa-devel] [PATCH v6 0/3] Media Device Allocator API
by Shuah Khan 09 Dec '16

09 Dec '16

Media Device Allocator API to allows multiple drivers share a media device. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. Patches 0001 and 0002 are rebased to 4.9-rc7. Patch 0003 for snd-usb-audio is a rebase of the patch that was tested with the original Media Device Allocator patch series. snd-usb-audio patch includes the fixes found during 4.7-rc1 time in the original snd-usb-audio patch. Changes to patches in this series: Changes to patch 0001 since v5: (comments from Mauro and Sakari) - Removed struct device from media_device_instance. mdev.dev is used instead. - Added documentation. Changes to patch 0002: - No changes since patch v2, applies cleanly on top of the following: media: Protect enable_source and disable_source handler code paths https://lkml.org/lkml/2016/11/29/1001 Changes to patch 0003: - Changed to hold graph_mutex to check and call enable_source and disable_source handlers - to match au0828 doing the same in: media: Protect enable_source and disable_source handler code paths https://lkml.org/lkml/2016/11/29/1001 Changes to patch 0001 since v4: - Addressed Sakari's review comments with the exception of opting to not introduce media_device_usb_allocate() macro, and to not add a new routine to find media device instance to avoid a one line check. Changes to patch 0001 since v3: - Fixed undefined reference to `__media_device_usb_init compile error when CONFIG_USB is disabled. - Fixed kernel paging error when accessing /dev/mediaX after rmmod of the module that owns the media_device. The fix bumps the reference count for the owner when second driver comes along to share the media_device. If au0828 owns the media_device, then snd_usb_audio will bump the refcount for au0828, so it won't get deleted and vice versa. Changes to patch 0002 since v2: - Updated media_device_delete() to pass in module name. Changes to patch 0003 since the last version in 4.7-rc1: - Included fixes to bugs found during testing. - Updated to use the Media Allocator API. This patch series has been tested with au0828 and snd-usb-audio drivers. Ran bind and unbind loop tests on each driver with mc_nextgen_test and media_device_test app loop tests while checking lsmod and dmesg. Please refer to tools/testing/selftests/media_tests/regression_test.txt for testing done on this series. Shuah Khan (3): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API sound/usb: Use Media Controller API to share media resources Documentation/media/kapi/mc-core.rst | 37 ++++ drivers/media/Makefile | 3 +- drivers/media/media-dev-allocator.c | 133 ++++++++++++++ drivers/media/usb/au0828/au0828-core.c | 12 +- drivers/media/usb/au0828/au0828.h | 1 + include/media/media-dev-allocator.h | 54 ++++++ sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 326 +++++++++++++++++++++++++++++++++ sound/usb/media.h | 73 ++++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 28 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 17 files changed, 688 insertions(+), 14 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.7.4

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[alsa-devel] [PATCH] alsactl: Fix potential NULL dereferences in daemon mode
by Takashi Iwai 09 Dec '16

09 Dec '16

The code releasing the each card object may access to NULL when a bogus count is given. Add a NULL check just to make sure. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Something I forgot to merge, casually found in the working directory... alsactl/daemon.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/alsactl/daemon.c b/alsactl/daemon.c index 127701efaf0f..4f0a935a3987 100644 --- a/alsactl/daemon.c +++ b/alsactl/daemon.c @@ -451,8 +451,10 @@ save: out: free(pfd); remove(pidfile); - for (i = 0; i < count; i++) - card_free(&cards[i]); - free(cards); + if (cards) { + for (i = 0; i < count; i++) + card_free(&cards[i]); + free(cards); + } return 0; } -- 2.11.0

1 0

[alsa-devel] [PATCH 4/4] hda: thinkpad_helper: Add support for hid-lenovo LED control
by Dennis Wassenberg 09 Dec '16

09 Dec '16

Make the thinkpad_helper able to support not only led control over acpi with thinkpad_acpi driver but also led control over hid-lenovo. The hid-lenovo driver adapted the led control api of thinkpad_acpi. Make the thinkpad_acpi and hid-lenovo able to work combined to support connected Lenovo USB keyboards at Lenovo Thinkpad devices. Signed-off-by: Dennis Wassenberg <dennis.wassenberg(a)secunet.com> --- sound/pci/hda/thinkpad_helper.c | 182 +++++++++++++++++++++++++++++++--------- 1 file changed, 144 insertions(+), 38 deletions(-) diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index f0955fd..0d98624 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -1,83 +1,188 @@ /* Helper functions for Thinkpad LED control; * to be included from codec driver + * + * These helper functions include both LED control over thinkpad_acpi and + * hid-lenovo */ -#if IS_ENABLED(CONFIG_THINKPAD_ACPI) +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) || IS_ENABLED(CONFIG_HID_LENOVO) #include <linux/acpi.h> +#include <linux/hid-lenovo.h> #include <linux/thinkpad_acpi.h> -static int (*led_set_func)(int, bool); +static int (*led_set_func_tpacpi)(int, bool); +static int (*led_set_func_hid_lenovo)(int, bool); static void (*old_vmaster_hook)(void *, int); static bool is_thinkpad(struct hda_codec *codec) { + return (codec->core.subsystem_id >> 16 == 0x17aa); +} + +static bool is_thinkpad_acpi(struct hda_codec *codec) +{ return is_thinkpad(codec) && (acpi_dev_found("LEN0068") || acpi_dev_found("IBM0068")); } -static void update_tpacpi_mute_led(void *private_data, int enabled) +static void update_thinkpad_mute_led(void *private_data, int enabled) { if (old_vmaster_hook) old_vmaster_hook(private_data, enabled); - if (led_set_func) - led_set_func(TPACPI_LED_MUTE, !enabled); +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) + if (led_set_func_tpacpi) + led_set_func_tpacpi(TPACPI_LED_MUTE, !enabled); +#endif +#if IS_ENABLED(CONFIG_HID_LENOVO) + if (led_set_func_hid_lenovo) + led_set_func_hid_lenovo(HID_LENOVO_LED_MUTE, !enabled); +#endif } -static void update_tpacpi_micmute_led(struct hda_codec *codec, + + +static void update_thinkpad_micmute_led(struct hda_codec *codec, struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - if (!ucontrol || !led_set_func) + if (!ucontrol) return; if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { /* TODO: How do I verify if it's a mono or stereo here? */ bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]; - led_set_func(TPACPI_LED_MICMUTE, !val); +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) + if (led_set_func_tpacpi) + led_set_func_tpacpi(TPACPI_LED_MICMUTE, !val); +#endif +#if IS_ENABLED(CONFIG_HID_LENOVO) + if (led_set_func_hid_lenovo) + led_set_func_hid_lenovo(HID_LENOVO_LED_MICMUTE, !val); +#endif } } -static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, - const struct hda_fixup *fix, int action) +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) +static int hda_fixup_thinkpad_acpi_prepare(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; - bool removefunc = false; + int ret = -ENXIO; - if (action == HDA_FIXUP_ACT_PROBE) { - if (!is_thinkpad(codec)) - return; - if (!led_set_func) - led_set_func = symbol_request(tpacpi_led_set); - if (!led_set_func) { - codec_warn(codec, - "Failed to find thinkpad-acpi symbol tpacpi_led_set\n"); - return; - } + if (!is_thinkpad(codec)) + return -ENODEV; + if (!is_thinkpad_acpi(codec)) + return -ENODEV; + if (!led_set_func_tpacpi) + led_set_func_tpacpi = symbol_request(tpacpi_led_set); + if (!led_set_func_tpacpi) { + codec_warn(codec, + "Failed to find thinkpad-acpi symbol tpacpi_led_set\n"); + return -ENOENT; + } - removefunc = true; - if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { - old_vmaster_hook = spec->vmaster_mute.hook; - spec->vmaster_mute.hook = update_tpacpi_mute_led; - removefunc = false; - } - if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { - if (spec->num_adc_nids > 1) - codec_dbg(codec, - "Skipping micmute LED control due to several ADCs"); - else { - spec->cap_sync_hook = update_tpacpi_micmute_led; - removefunc = false; - } + if (led_set_func_tpacpi(TPACPI_LED_MUTE, false) >= 0) { + old_vmaster_hook = spec->vmaster_mute.hook; + spec->vmaster_mute.hook = update_thinkpad_mute_led; + ret = 0; + } + + if (led_set_func_tpacpi(TPACPI_LED_MICMUTE, false) >= 0) { + if (spec->num_adc_nids > 1) + codec_dbg(codec, + "Skipping micmute LED control due to several ADCs"); + else { + spec->cap_sync_hook = update_thinkpad_micmute_led; + ret = 0; } } - if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + return ret; +} + +static void hda_fixup_thinkpad_acpi_setup(struct hda_codec *codec, int action) +{ + int ret = 0; + + if (action == HDA_FIXUP_ACT_PROBE) { + ret = hda_fixup_thinkpad_acpi_prepare(codec); + } + + if (led_set_func_tpacpi && + (action == HDA_FIXUP_ACT_FREE || ret)) { + symbol_put(tpacpi_led_set); - led_set_func = NULL; + led_set_func_tpacpi = NULL; + old_vmaster_hook = NULL; + } +} +#else +static void hda_fixup_thinkpad_acpi_setup(struct hda_codec *codec, int action) +{ + return 0; +} +#endif + +#if IS_ENABLED(CONFIG_HID_LENOVO) +static int hda_fixup_thinkpad_hid_prepare(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + int ret = 0; + + if (!is_thinkpad(codec)) + return -ENODEV; + if (!led_set_func_hid_lenovo) + led_set_func_hid_lenovo = symbol_request(hid_lenovo_led_set); + if (!led_set_func_hid_lenovo) { + codec_warn(codec, + "Failed to find hid-lenovo symbol hid_lenovo_led_set\n"); + return -ENOENT; + } + + if (update_thinkpad_mute_led != spec->vmaster_mute.hook) + old_vmaster_hook = spec->vmaster_mute.hook; + + /* do not remove hook if setting delay does not work currently because + * it is a usb hid devices which is not connected right now + * maybe is will be connected later + */ + led_set_func_hid_lenovo(HID_LENOVO_LED_MUTE, false); + spec->vmaster_mute.hook = update_thinkpad_mute_led; + + led_set_func_hid_lenovo(HID_LENOVO_LED_MICMUTE, false); + spec->cap_sync_hook = update_thinkpad_micmute_led; + + return ret; +} + +static void hda_fixup_thinkpad_hid_setup(struct hda_codec *codec, int action) +{ + int ret = 0; + + if (action == HDA_FIXUP_ACT_PROBE) { + ret = hda_fixup_thinkpad_hid_prepare(codec); + } + + if (led_set_func_hid_lenovo && + (action == HDA_FIXUP_ACT_FREE || ret)) { + + symbol_put(hid_lenovo_led_set); + led_set_func_hid_lenovo = NULL; old_vmaster_hook = NULL; } } +#else +static void hda_fixup_thinkpad_hid_setup(struct hda_codec *codec, int action) +{ +} +#endif + +/** this fixup is not for acpi case only, it will handle hid-lenovo as well */ +static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + hda_fixup_thinkpad_acpi_setup(codec, action); + hda_fixup_thinkpad_hid_setup(codec, action); +} #else /* CONFIG_THINKPAD_ACPI */ @@ -87,3 +192,4 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, } #endif /* CONFIG_THINKPAD_ACPI */ + -- 1.9.1 --

1 0

[alsa-devel] [PATCH 3/4] hid-lenovo: Handle X1 Tablet cover FNLock function appropriately
by Dennis Wassenberg 09 Dec '16

09 Dec '16

The FNLock state is not handled by the X1 tablet cover itself. There is the need to set the FNLock LED which actually enables the FNLock state of the cover. This means activating the FNLock LED will enable FNLock state and FNLock LED. --- drivers/hid/hid-lenovo.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/drivers/hid/hid-lenovo.c b/drivers/hid/hid-lenovo.c index 2efd5b0..3ace8c3 100644 --- a/drivers/hid/hid-lenovo.c +++ b/drivers/hid/hid-lenovo.c @@ -526,6 +526,29 @@ static void lenovo_led_brightness_set_tpx1cover(struct led_classdev *led_cdev, } } +static int lenovo_event_tpx1cover(struct hid_device *hdev, + struct hid_field *field, struct hid_usage *usage, __s32 value) +{ + int ret = 0; + + if (usage->hid == HID_CP_CONSUMERCONTROL) { + if (usage->usage_index == 8 && value == 1) { + struct lenovo_drvdata_tpx1cover *drv_data = hid_get_drvdata(hdev); + + if (drv_data && drv_data->led_present) { + drv_data->fnlock_state = lenovo_led_brightness_get_tpx1cover( + &drv_data->led_fnlock) == LED_OFF ? 1 : 0; + lenovo_led_brightness_set_tpx1cover( + &drv_data->led_fnlock, + drv_data->fnlock_state ? LED_FULL : LED_OFF); + } + } + + } + + return ret; +} + static int lenovo_event(struct hid_device *hdev, struct hid_field *field, struct hid_usage *usage, __s32 value) { @@ -533,6 +556,8 @@ static int lenovo_event(struct hid_device *hdev, struct hid_field *field, case USB_DEVICE_ID_LENOVO_CUSBKBD: case USB_DEVICE_ID_LENOVO_CBTKBD: return lenovo_event_cptkbd(hdev, field, usage, value); + case USB_DEVICE_ID_LENOVO_X1_COVER: + return lenovo_event_tpx1cover(hdev, field, usage, value); default: return 0; } -- 1.9.1

1 0

[alsa-devel] [PATCH 2/4] hid-lenovo: Add support for X1 Tablet cover LED control
by Dennis Wassenberg 09 Dec '16

09 Dec '16

Add support for controlling MUTE, MICMUTE and FNLOCK LEDs. In ordner to enable MUTE and MICMUTE LED control over thinkpad_helper the external interface hid_lenovo_led_set is introduced. This enables setting Lenovo LEDs from external. This is needed because the X1 Tablet Cover is the first thinkpad_helper controlable device which is connected via USB. Additionally the led state is stored inside the hid-lenovo driver to make the driver able to set the led appropriate after device attach. --- drivers/hid/hid-lenovo.c | 262 +++++++++++++++++++++++++++++++++++++++++++++ include/linux/hid-lenovo.h | 15 +++ 2 files changed, 277 insertions(+) create mode 100644 include/linux/hid-lenovo.h diff --git a/drivers/hid/hid-lenovo.c b/drivers/hid/hid-lenovo.c index 4fc332c..2efd5b0 100644 --- a/drivers/hid/hid-lenovo.c +++ b/drivers/hid/hid-lenovo.c @@ -21,11 +21,27 @@ #include <linux/sysfs.h> #include <linux/device.h> #include <linux/hid.h> +#include <linux/hid-lenovo.h> #include <linux/input.h> #include <linux/leds.h> #include "hid-ids.h" +struct lenovo_led_list_entry { + struct led_classdev *dev; + enum hid_lenovo_led_type type; + struct list_head list; +}; + +struct lenovo_led_type_entry { + struct led_classdev *dev; + enum hid_lenovo_led_type type; + struct list_head list; +}; + +static struct list_head hid_lenovo_leds = LIST_HEAD_INIT(hid_lenovo_leds); +static enum led_brightness hid_lenovo_initial_leds[HID_LENOVO_LED_MAX]; + struct lenovo_drvdata_tpkbd { int led_state; struct led_classdev led_mute; @@ -44,6 +60,44 @@ struct lenovo_drvdata_cptkbd { int sensitivity; }; +struct lenovo_drvdata_tpx1cover { + uint16_t led_state; + uint8_t fnlock_state; + uint8_t led_present; + struct led_classdev led_mute; + struct led_classdev led_micmute; + struct led_classdev led_fnlock; +}; + +int hid_lenovo_led_set(enum hid_lenovo_led_type led, bool on) +{ + struct led_classdev *dev = NULL; + struct lenovo_led_list_entry *entry; + + if (led >= HID_LENOVO_LED_MAX) + return -EINVAL; + + hid_lenovo_initial_leds[led] = on ? LED_FULL : LED_OFF; + + list_for_each_entry(entry, &hid_lenovo_leds, list) { + if (entry->type == led) { + dev = entry->dev; + break; + } + } + + if (!dev) + return -ENODEV; + + if (!dev->brightness_set) + return -ENODEV; + + dev->brightness_set(dev, on ? LED_FULL : LED_OFF); + + return 0; +} +EXPORT_SYMBOL_GPL(hid_lenovo_led_set); + #define map_key_clear(c) hid_map_usage_clear(hi, usage, bit, max, EV_KEY, (c)) static const __u8 lenovo_pro_dock_need_fixup_collection[] = { @@ -416,6 +470,62 @@ static int lenovo_event_cptkbd(struct hid_device *hdev, return 0; } +static enum led_brightness lenovo_led_brightness_get_tpx1cover( + struct led_classdev *led_cdev) +{ + struct device *dev = led_cdev->dev->parent; + struct hid_device *hdev = to_hid_device(dev); + struct lenovo_drvdata_tpx1cover *drv_data = hid_get_drvdata(hdev); + enum hid_lenovo_led_type led; + + if (led_cdev == &drv_data->led_mute) + led = HID_LENOVO_LED_MUTE; + else if (led_cdev == &drv_data->led_micmute) + led = HID_LENOVO_LED_MICMUTE; + else if (led_cdev == &drv_data->led_fnlock) + led = HID_LENOVO_LED_FNLOCK; + else + return LED_OFF; + + return drv_data->led_state & (1 << led) + ? LED_FULL + : LED_OFF; +} + +static void lenovo_led_brightness_set_tpx1cover(struct led_classdev *led_cdev, + enum led_brightness value) +{ + struct device *dev = led_cdev->dev->parent; + struct hid_device *hdev = to_hid_device(dev); + struct lenovo_drvdata_tpx1cover *drv_data = hid_get_drvdata(hdev); + struct hid_report *report; + enum hid_lenovo_led_type led; + + if (led_cdev == &drv_data->led_mute) { + led = HID_LENOVO_LED_MUTE; + } else if (led_cdev == &drv_data->led_micmute) { + led = HID_LENOVO_LED_MICMUTE; + } else if (led_cdev == &drv_data->led_fnlock) { + led = HID_LENOVO_LED_FNLOCK; + } else { + hid_warn(hdev, "Invalid LED to set.\n"); + return; + } + + if (value == LED_OFF) + drv_data->led_state &= ~(1 << led); + else + drv_data->led_state |= 1 << led; + + report = hdev->report_enum[HID_OUTPUT_REPORT].report_id_hash[9]; + if (report) { + report->field[0]->value[0] = ((led + 1) << 4) | 0x44; + report->field[0]->value[1] = (drv_data->led_state & (1 << led)) + ? 0x02 : 0x01; + hid_hw_request(hdev, report, HID_REQ_SET_REPORT); + } +} + static int lenovo_event(struct hid_device *hdev, struct hid_field *field, struct hid_usage *usage, __s32 value) { @@ -788,6 +898,7 @@ static int lenovo_probe_tpkbd(struct hid_device *hdev) static int lenovo_probe_tpx1cover_configure(struct hid_device *hdev) { struct hid_report *report = hdev->report_enum[HID_OUTPUT_REPORT].report_id_hash[9]; + struct lenovo_drvdata_tpx1cover *drv_data = hid_get_drvdata(hdev); if (!report) return -ENOENT; @@ -807,14 +918,33 @@ static int lenovo_probe_tpx1cover_configure(struct hid_device *hdev) hid_hw_request(hdev, report, HID_REQ_SET_REPORT); hid_hw_wait(hdev); + lenovo_led_brightness_set_tpx1cover(&drv_data->led_mute, + hid_lenovo_initial_leds[HID_LENOVO_LED_MUTE]); + + hid_hw_wait(hdev); + + lenovo_led_brightness_set_tpx1cover(&drv_data->led_micmute, + hid_lenovo_initial_leds[HID_LENOVO_LED_MICMUTE]); + hid_hw_wait(hdev); + + lenovo_led_brightness_set_tpx1cover(&drv_data->led_fnlock, LED_FULL); + return 0; } static int lenovo_probe_tpx1cover_special_functions(struct hid_device *hdev) { + struct device *dev = &hdev->dev; + struct lenovo_drvdata_tpx1cover *drv_data = NULL; + struct lenovo_led_list_entry *mute_entry = NULL; + struct lenovo_led_list_entry *micmute_entry = NULL; + struct lenovo_led_list_entry *fnlock_entry = NULL; + struct hid_report *report; bool report_match = 1; + int ret = 0; + /* Used for LED control */ report = hid_validate_values(hdev, HID_INPUT_REPORT, 3, 0, 16); report_match &= report ? 1 : 0; @@ -825,7 +955,97 @@ static int lenovo_probe_tpx1cover_special_functions(struct hid_device *hdev) if (!report_match) return -ENODEV; + drv_data = devm_kzalloc(&hdev->dev, + sizeof(struct lenovo_drvdata_tpx1cover), + GFP_KERNEL); + + if (!drv_data) { + hid_err(hdev, + "Could not allocate memory for tpx1cover driver data\n"); + return -ENOMEM; + } + + mute_entry = kzalloc(sizeof(*mute_entry), GFP_KERNEL); + if (!mute_entry) { + hid_err(hdev, "Could not allocate memory for led mute entry\n"); + ret = -ENOMEM; + goto err_cleanup; + } + drv_data->led_mute.name = "MUTE_LED"; + drv_data->led_mute.brightness_get = lenovo_led_brightness_get_tpx1cover; + drv_data->led_mute.brightness_set = lenovo_led_brightness_set_tpx1cover; + devm_led_classdev_register(dev, &drv_data->led_mute); + + INIT_LIST_HEAD(&mute_entry->list); + mute_entry->dev = &drv_data->led_mute; + mute_entry->type = HID_LENOVO_LED_MUTE; + list_add_tail(&mute_entry->list, &hid_lenovo_leds); + + + micmute_entry = kzalloc(sizeof(*micmute_entry), GFP_KERNEL); + if (!micmute_entry) { + hid_err(hdev, "Could not allocate memory for led micmute entry\n"); + ret = -ENOMEM; + goto err_cleanup; + } + + drv_data->led_micmute.name = "MICMUTE_LED"; + drv_data->led_micmute.brightness_get = lenovo_led_brightness_get_tpx1cover; + drv_data->led_micmute.brightness_set = lenovo_led_brightness_set_tpx1cover; + devm_led_classdev_register(dev, &drv_data->led_micmute); + + INIT_LIST_HEAD(&micmute_entry->list); + micmute_entry->dev = &drv_data->led_micmute; + micmute_entry->type = HID_LENOVO_LED_MICMUTE; + list_add_tail(&micmute_entry->list, &hid_lenovo_leds); + + + fnlock_entry = kzalloc(sizeof(*fnlock_entry), GFP_KERNEL); + if (!fnlock_entry) { + hid_err(hdev, "Could not allocate memory for led fnlock entry\n"); + ret = -ENOMEM; + goto err_cleanup; + } + + drv_data->led_fnlock.brightness_get = lenovo_led_brightness_get_tpx1cover; + drv_data->led_fnlock.brightness_set = lenovo_led_brightness_set_tpx1cover; + drv_data->led_fnlock.name = "FNLOCK_LED"; + devm_led_classdev_register(dev, &drv_data->led_fnlock); + + INIT_LIST_HEAD(&fnlock_entry->list); + fnlock_entry->dev = &drv_data->led_fnlock; + fnlock_entry->type = HID_LENOVO_LED_FNLOCK; + list_add_tail(&fnlock_entry->list, &hid_lenovo_leds); + + + drv_data->led_state = 0; + drv_data->fnlock_state = 1; + drv_data->led_present = 1; + + hid_set_drvdata(hdev, drv_data); + return lenovo_probe_tpx1cover_configure(hdev); + +err_cleanup: + if (mute_entry) { + list_del(&mute_entry->list); + kfree (mute_entry); + } + + if (micmute_entry) { + list_del(&micmute_entry->list); + kfree (micmute_entry); + } + + if (fnlock_entry) { + list_del(&fnlock_entry->list); + kfree (fnlock_entry); + } + + if (drv_data) + kfree(drv_data); + + return ret; } static int lenovo_probe_tpx1cover_touch(struct hid_device *hdev) @@ -846,6 +1066,7 @@ static int lenovo_probe_tpx1cover_touch(struct hid_device *hdev) static int lenovo_probe_tpx1cover_keyboard(struct hid_device *hdev) { + struct lenovo_drvdata_tpx1cover *drv_data; struct hid_report *report; bool report_match = 1; int ret = 0; @@ -860,6 +1081,21 @@ static int lenovo_probe_tpx1cover_keyboard(struct hid_device *hdev) if (!report_match) ret = -ENODEV; + drv_data = devm_kzalloc(&hdev->dev, + sizeof(struct lenovo_drvdata_tpx1cover), + GFP_KERNEL); + + if (!drv_data) { + hid_err(hdev, + "Could not allocate memory for tpx1cover driver data\n"); + return -ENOMEM; + } + + drv_data->led_state = 0; + drv_data->led_present = 0; + drv_data->fnlock_state = 0; + hid_set_drvdata(hdev, drv_data); + return ret; } @@ -1001,6 +1237,29 @@ static void lenovo_remove_cptkbd(struct hid_device *hdev) &lenovo_attr_group_cptkbd); } +static void lenovo_remove_tpx1cover(struct hid_device *hdev) +{ + struct lenovo_drvdata_tpx1cover *drv_data = hid_get_drvdata(hdev); + struct lenovo_led_list_entry *entry, *head; + + if (!drv_data) + return; + + if (drv_data->led_present) { + list_for_each_entry_safe(entry, head, &hid_lenovo_leds, list) { + if (entry->dev == &drv_data->led_fnlock + || entry->dev == &drv_data->led_micmute + || entry->dev == &drv_data->led_mute) { + + list_del(&entry->list); + kfree(entry); + } + } + } + + hid_set_drvdata(hdev, NULL); +} + static void lenovo_remove(struct hid_device *hdev) { switch (hdev->product) { @@ -1011,6 +1270,9 @@ static void lenovo_remove(struct hid_device *hdev) case USB_DEVICE_ID_LENOVO_CBTKBD: lenovo_remove_cptkbd(hdev); break; + case USB_DEVICE_ID_LENOVO_X1_COVER: + lenovo_remove_tpx1cover(hdev); + break; } hid_hw_stop(hdev); diff --git a/include/linux/hid-lenovo.h b/include/linux/hid-lenovo.h new file mode 100644 index 0000000..00eb140 --- /dev/null +++ b/include/linux/hid-lenovo.h @@ -0,0 +1,15 @@ + +#ifndef __HID_LENOVO_H__ +#define __HID_LENOVO_H__ + + +enum hid_lenovo_led_type { + HID_LENOVO_LED_FNLOCK = 0, + HID_LENOVO_LED_MUTE, + HID_LENOVO_LED_MICMUTE, + HID_LENOVO_LED_MAX +}; + +int hid_lenovo_led_set(enum hid_lenovo_led_type led, bool on); + +#endif /* __HID_LENOVO_H_ */ -- 1.9.1

1 0

[alsa-devel] [PATCH 1/4] hid-lenovo: Add support for X1 Tablet cover special keys
by Dennis Wassenberg 09 Dec '16

09 Dec '16

The Lenovo X1 Tablet Cover is connected via USB. It constists of 1 usb device with 3 usb interfaces. Interface 0 represents keyboard, interface 1 the function / special keys and LED control, interface 2 is the Synaptics touchpad and pointing stick. This patch will introduce the function / special keys handling of USB interface 1. --- drivers/hid/hid-core.c | 1 + drivers/hid/hid-ids.h | 1 + drivers/hid/hid-lenovo.c | 160 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 162 insertions(+) diff --git a/drivers/hid/hid-core.c b/drivers/hid/hid-core.c index cff060b..5b38afc 100644 --- a/drivers/hid/hid-core.c +++ b/drivers/hid/hid-core.c @@ -1932,6 +1932,7 @@ void hid_disconnect(struct hid_device *hdev) { HID_USB_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_CUSBKBD) }, { HID_BLUETOOTH_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_CBTKBD) }, { HID_USB_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_TPPRODOCK) }, + { HID_USB_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_X1_COVER) }, #endif { HID_USB_DEVICE(USB_VENDOR_ID_LOGITECH, USB_DEVICE_ID_MX3000_RECEIVER) }, { HID_USB_DEVICE(USB_VENDOR_ID_LOGITECH, USB_DEVICE_ID_S510_RECEIVER) }, diff --git a/drivers/hid/hid-ids.h b/drivers/hid/hid-ids.h index ec277b9..961ee4f 100644 --- a/drivers/hid/hid-ids.h +++ b/drivers/hid/hid-ids.h @@ -623,6 +623,7 @@ #define USB_DEVICE_ID_LENOVO_CUSBKBD 0x6047 #define USB_DEVICE_ID_LENOVO_CBTKBD 0x6048 #define USB_DEVICE_ID_LENOVO_TPPRODOCK 0x6067 +#define USB_DEVICE_ID_LENOVO_X1_COVER 0x6085 #define USB_VENDOR_ID_LG 0x1fd2 #define USB_DEVICE_ID_LG_MULTITOUCH 0x0064 diff --git a/drivers/hid/hid-lenovo.c b/drivers/hid/hid-lenovo.c index 1ac4ff4..4fc332c 100644 --- a/drivers/hid/hid-lenovo.c +++ b/drivers/hid/hid-lenovo.c @@ -3,9 +3,11 @@ * - ThinkPad USB Keyboard with TrackPoint (tpkbd) * - ThinkPad Compact Bluetooth Keyboard with TrackPoint (cptkbd) * - ThinkPad Compact USB Keyboard with TrackPoint (cptkbd) + * - ThinkPad X1 Cover USB Keyboard with TrackPoint and Touchpad (tpx1cover) * * Copyright (c) 2012 Bernhard Seibold * Copyright (c) 2014 Jamie Lentin <jm(a)lentin.co.uk> + * Copyright (c) 2016 Dennis Wassenberg <dennis.wassenberg(a)secunet.com> */ /* @@ -86,6 +88,55 @@ static int lenovo_input_mapping_tpkbd(struct hid_device *hdev, return 0; } +static int lenovo_input_mapping_tpx1cover(struct hid_device *hdev, + struct hid_input *hi, struct hid_field *field, + struct hid_usage *usage, unsigned long **bit, int *max) +{ + if (usage->hid == HID_CP_CONSUMERCONTROL) { + /* Unknown keys -> Idenditied by usage index! */ + switch (usage->usage_index) { + case 8: + // FNLock key + return -1; + + case 9: + input_set_capability(hi->input, EV_KEY, KEY_MICMUTE); + map_key_clear(KEY_MICMUTE); + return 1; + + case 10: + input_set_capability(hi->input, EV_KEY, KEY_CONFIG); + map_key_clear(KEY_CONFIG); + return 1; + + case 11: + input_set_capability(hi->input, EV_KEY, KEY_SEARCH); + map_key_clear(KEY_SEARCH); + return 1; + + case 12: + input_set_capability(hi->input, EV_KEY, KEY_SETUP); + map_key_clear(KEY_SETUP); + return 1; + + case 13: + input_set_capability(hi->input, EV_KEY, KEY_SWITCHVIDEOMODE); + map_key_clear(KEY_SWITCHVIDEOMODE); + return 1; + + case 14: + input_set_capability(hi->input, EV_KEY, KEY_RFKILL); + map_key_clear(KEY_RFKILL); + return 1; + + } + + return -1; + } + + return 0; +} + static int lenovo_input_mapping_cptkbd(struct hid_device *hdev, struct hid_input *hi, struct hid_field *field, struct hid_usage *usage, unsigned long **bit, int *max) @@ -172,6 +223,9 @@ static int lenovo_input_mapping(struct hid_device *hdev, case USB_DEVICE_ID_LENOVO_CBTKBD: return lenovo_input_mapping_cptkbd(hdev, hi, field, usage, bit, max); + case USB_DEVICE_ID_LENOVO_X1_COVER: + return lenovo_input_mapping_tpx1cover(hdev, hi, field, + usage, bit, max); default: return 0; } @@ -731,6 +785,107 @@ static int lenovo_probe_tpkbd(struct hid_device *hdev) return ret; } +static int lenovo_probe_tpx1cover_configure(struct hid_device *hdev) +{ + struct hid_report *report = hdev->report_enum[HID_OUTPUT_REPORT].report_id_hash[9]; + + if (!report) + return -ENOENT; + + report->field[0]->value[0] = 0x54; + report->field[0]->value[1] = 0x20; + hid_hw_request(hdev, report, HID_REQ_SET_REPORT); + hid_hw_wait(hdev); + + report->field[0]->value[0] = 0x54; + report->field[0]->value[1] = 0x08; + hid_hw_request(hdev, report, HID_REQ_SET_REPORT); + hid_hw_wait(hdev); + + report->field[0]->value[0] = 0xA0; + report->field[0]->value[1] = 0x02; + hid_hw_request(hdev, report, HID_REQ_SET_REPORT); + hid_hw_wait(hdev); + + return 0; +} + +static int lenovo_probe_tpx1cover_special_functions(struct hid_device *hdev) +{ + struct hid_report *report; + bool report_match = 1; + + /* Used for LED control */ + report = hid_validate_values(hdev, HID_INPUT_REPORT, 3, 0, 16); + report_match &= report ? 1 : 0; + /* Identify USB Intf 1 which is the only one with HID_FEATURE_REPORT 100 */ + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 100, 0, 1); + report_match &= report ? 1 : 0; + + if (!report_match) + return -ENODEV; + + return lenovo_probe_tpx1cover_configure(hdev); +} + +static int lenovo_probe_tpx1cover_touch(struct hid_device *hdev) +{ + struct hid_report *report; + bool report_match = 1; + int ret = 0; + + /* Identify USB Intf 2 which is the only one with HID_FEATURE_REPORT 15 */ + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 15, 0, 3); + report_match &= report ? 1 : 0; + + if (!report_match) + ret = -ENODEV; + + return ret; +} + +static int lenovo_probe_tpx1cover_keyboard(struct hid_device *hdev) +{ + struct hid_report *report; + bool report_match = 1; + int ret = 0; + + /* Identify USB Intf0 which is the only one with HID_INPUT_REPORT 0 + * and HID_OUTPUT_REPORT 0 */ + report = hid_validate_values(hdev, HID_INPUT_REPORT, 0, 0, 1); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_OUTPUT_REPORT, 0, 0, 1); + report_match &= report ? 1 : 0; + + if (!report_match) + ret = -ENODEV; + + return ret; +} + +static int lenovo_probe_tpx1cover(struct hid_device *hdev) +{ + int ret = 0; + + /* + * Probing for special function keys and LED control -> usb intf 1 + * Probing for touch input -> usb intf 2 (handled by rmi4 driver) + * Other (keyboard) -> usb intf 0 + */ + if (!lenovo_probe_tpx1cover_special_functions(hdev)) { + /* special function keys and LED control */ + ret = 0; + } else if (!lenovo_probe_tpx1cover_touch(hdev)) { + /* handled by rmi */ + ret = -ENODEV; + } else { + /* keyboard */ + ret = lenovo_probe_tpx1cover_keyboard(hdev); + } + + return ret; +} + static int lenovo_probe_cptkbd(struct hid_device *hdev) { int ret; @@ -803,6 +958,9 @@ static int lenovo_probe(struct hid_device *hdev, case USB_DEVICE_ID_LENOVO_CBTKBD: ret = lenovo_probe_cptkbd(hdev); break; + case USB_DEVICE_ID_LENOVO_X1_COVER: + ret = lenovo_probe_tpx1cover(hdev); + break; default: ret = 0; break; @@ -883,6 +1041,7 @@ static int lenovo_input_configured(struct hid_device *hdev, { HID_USB_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_CUSBKBD) }, { HID_BLUETOOTH_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_CBTKBD) }, { HID_USB_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_TPPRODOCK) }, + { HID_USB_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_X1_COVER) }, { } }; @@ -902,3 +1061,4 @@ static int lenovo_input_configured(struct hid_device *hdev, module_hid_driver(lenovo_driver); MODULE_LICENSE("GPL"); + -- 1.9.1

1 0

Re: [alsa-devel] Cherry trail and Nau 88l24 codec
by Joseph Wang 09 Dec '16

09 Dec '16

YES!!!! Thank you Intel and Nuvoton. I'm getting beautiful sound in the external speakers. The jack detection is a bit flaky, but I can work on that. The code is available at https://github.com/joequant/nau8824

1 0

[alsa-devel] [PATCH v2 1/2] hid-lenovo: Add support for X1 Tablet special keys and LED control
by Dennis Wassenberg 09 Dec '16

09 Dec '16

The Lenovo X1 Tablet Cover is connected via USB. It constists of 1 device with 3 usb interfaces. Interface 0 represents keyboard, interface 1 the function / special keys and LED control, interface 2 is the Synaptics touchpad and pointing stick. This driver will bind to interfaces 0 and 1 and handle function / special keys including LED control. HID: add device id for Lenovo X1 Tablet Cover Signed-off-by: Dennis Wassenberg <dennis.wassenberg(a)secunet.com> --- Changes v1 -> v2 (suggested by Benjamin Tissoires): * Squashed add of device IDs for Lenovo Thinkpad X1 Tablet Cover together with add of support for Lenovo Thinkpad X1 Tablet Cover into one patch drivers/hid/hid-core.c | 1 + drivers/hid/hid-ids.h | 1 + drivers/hid/hid-lenovo.c | 549 +++++++++++++++++++++++++++++++++++++++++++++ include/linux/hid-lenovo.h | 15 ++ 4 files changed, 566 insertions(+) create mode 100644 include/linux/hid-lenovo.h diff --git a/drivers/hid/hid-core.c b/drivers/hid/hid-core.c index 6add0b6..ba6a200 100644 --- a/drivers/hid/hid-core.c +++ b/drivers/hid/hid-core.c @@ -2111,6 +2111,7 @@ void hid_disconnect(struct hid_device *hdev) { HID_BLUETOOTH_DEVICE(USB_VENDOR_ID_NINTENDO, USB_DEVICE_ID_NINTENDO_WIIMOTE2) }, { HID_USB_DEVICE(USB_VENDOR_ID_RAZER, USB_DEVICE_ID_RAZER_BLADE_14) }, { HID_USB_DEVICE(USB_VENDOR_ID_CMEDIA, USB_DEVICE_ID_CM6533) }, + { HID_USB_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_X1_COVER) }, { } }; diff --git a/drivers/hid/hid-ids.h b/drivers/hid/hid-ids.h index 3466f0d..1f08fb5 100644 --- a/drivers/hid/hid-ids.h +++ b/drivers/hid/hid-ids.h @@ -615,6 +615,7 @@ #define USB_DEVICE_ID_LENOVO_CUSBKBD 0x6047 #define USB_DEVICE_ID_LENOVO_CBTKBD 0x6048 #define USB_DEVICE_ID_LENOVO_TPPRODOCK 0x6067 +#define USB_DEVICE_ID_LENOVO_X1_COVER 0x6085 #define USB_VENDOR_ID_LG 0x1fd2 #define USB_DEVICE_ID_LG_MULTITOUCH 0x0064 diff --git a/drivers/hid/hid-lenovo.c b/drivers/hid/hid-lenovo.c index 1ac4ff4..4251aac 100644 --- a/drivers/hid/hid-lenovo.c +++ b/drivers/hid/hid-lenovo.c @@ -3,9 +3,11 @@ * - ThinkPad USB Keyboard with TrackPoint (tpkbd) * - ThinkPad Compact Bluetooth Keyboard with TrackPoint (cptkbd) * - ThinkPad Compact USB Keyboard with TrackPoint (cptkbd) + * - ThinkPad X1 Cover USB Keyboard with TrackPoint and Touchpad (tpx1cover) * * Copyright (c) 2012 Bernhard Seibold * Copyright (c) 2014 Jamie Lentin <jm(a)lentin.co.uk> + * Copyright (c) 2016 Dennis Wassenberg <dennis.wassenberg(a)secunet.com> */ /* @@ -19,11 +21,19 @@ #include <linux/sysfs.h> #include <linux/device.h> #include <linux/hid.h> +#include <linux/hid-lenovo.h> #include <linux/input.h> #include <linux/leds.h> #include "hid-ids.h" +struct led_table_entry { + struct led_classdev *dev; + uint8_t state; +}; + +static struct led_table_entry hid_lenovo_led_table[HID_LENOVO_LED_MAX]; + struct lenovo_drvdata_tpkbd { int led_state; struct led_classdev led_mute; @@ -42,6 +52,37 @@ struct lenovo_drvdata_cptkbd { int sensitivity; }; +struct lenovo_drvdata_tpx1cover { + uint16_t led_state; + uint8_t fnlock_state; + uint8_t led_present; + struct led_classdev led_mute; + struct led_classdev led_micmute; + struct led_classdev led_fnlock; +}; + +int hid_lenovo_led_set(int led_num, bool on) +{ + struct led_classdev *dev; + + if (led_num >= HID_LENOVO_LED_MAX) + return -EINVAL; + + dev = hid_lenovo_led_table[led_num].dev; + hid_lenovo_led_table[led_num].state = on; + + if (!dev) + return -ENODEV; + + if (!dev->brightness_set) + return -ENODEV; + + dev->brightness_set(dev, on ? LED_FULL : LED_OFF); + + return 0; +} +EXPORT_SYMBOL_GPL(hid_lenovo_led_set); + #define map_key_clear(c) hid_map_usage_clear(hi, usage, bit, max, EV_KEY, (c)) static const __u8 lenovo_pro_dock_need_fixup_collection[] = { @@ -86,6 +127,84 @@ static int lenovo_input_mapping_tpkbd(struct hid_device *hdev, return 0; } +static int lenovo_input_mapping_tpx1cover(struct hid_device *hdev, + struct hid_input *hi, struct hid_field *field, + struct hid_usage *usage, unsigned long **bit, int *max) +{ + if ((usage->hid & HID_USAGE_PAGE) == HID_UP_CONSUMER) { + switch (usage->hid & HID_USAGE) { + case 0x0001: // Unknown keys -> Idenditied by usage index! + map_key_clear(KEY_UNKNOWN); + switch (usage->usage_index) { + case 0x8: + input_set_capability(hi->input, EV_KEY, KEY_FN); + break; + + case 0x9: + input_set_capability(hi->input, EV_KEY, KEY_MICMUTE); + break; + + case 0xa: + input_set_capability(hi->input, EV_KEY, KEY_CONFIG); + break; + + case 0xb: + input_set_capability(hi->input, EV_KEY, KEY_SEARCH); + break; + + case 0xc: + input_set_capability(hi->input, EV_KEY, KEY_SETUP); + break; + + case 0xd: + input_set_capability(hi->input, EV_KEY, KEY_SWITCHVIDEOMODE); + break; + + case 0xe: + input_set_capability(hi->input, EV_KEY, KEY_RFKILL); + break; + + default: + return -1; + } + + return 1; + + case 0x006f: // Consumer.006f ---> Key.BrightnessUp + map_key_clear(KEY_BRIGHTNESSUP); + return 1; + + case 0x0070: // Consumer.0070 ---> Key.BrightnessDown + map_key_clear(KEY_BRIGHTNESSDOWN); + return 1; + + case 0x00b7:// Consumer.00b7 ---> Key.StopCD + map_key_clear(KEY_STOPCD); + return 1; + + case 0x00cd: // Consumer.00cd ---> Key.PlayPause + map_key_clear(KEY_PLAYPAUSE); + return 1; + + case 0x00e0: // Consumer.00e0 ---> Absolute.Volume + return 0; + case 0x00e2: // Consumer.00e2 ---> Key.Mute + map_key_clear(KEY_MUTE); + return 1; + + case 0x00e9: // Consumer.00e9 ---> Key.VolumeUp + map_key_clear(KEY_VOLUMEUP); + return 1; + + case 0x00ea: // Consumer.00ea ---> Key.VolumeDown + map_key_clear(KEY_VOLUMEDOWN); + return 1; + } + } + + return 0; +} + static int lenovo_input_mapping_cptkbd(struct hid_device *hdev, struct hid_input *hi, struct hid_field *field, struct hid_usage *usage, unsigned long **bit, int *max) @@ -172,6 +291,9 @@ static int lenovo_input_mapping(struct hid_device *hdev, case USB_DEVICE_ID_LENOVO_CBTKBD: return lenovo_input_mapping_cptkbd(hdev, hi, field, usage, bit, max); + case USB_DEVICE_ID_LENOVO_X1_COVER: + return lenovo_input_mapping_tpx1cover(hdev, hi, field, + usage, bit, max); default: return 0; } @@ -362,6 +484,143 @@ static int lenovo_event_cptkbd(struct hid_device *hdev, return 0; } +static enum led_brightness lenovo_led_brightness_get_tpx1cover( + struct led_classdev *led_cdev) +{ + struct device *dev = led_cdev->dev->parent; + struct hid_device *hdev = to_hid_device(dev); + struct lenovo_drvdata_tpx1cover *drv_data = hid_get_drvdata(hdev); + int led_nr = 0; + + if (led_cdev == &drv_data->led_mute) + led_nr = 0; + else if (led_cdev == &drv_data->led_micmute) + led_nr = 1; + else if (led_cdev == &drv_data->led_fnlock) + led_nr = 2; + else + return LED_OFF; + + return drv_data->led_state & (1 << led_nr) + ? LED_FULL + : LED_OFF; +} + +static void lenovo_led_brightness_set_tpx1cover(struct led_classdev *led_cdev, + enum led_brightness value) +{ + struct device *dev = led_cdev->dev->parent; + struct hid_device *hdev = to_hid_device(dev); + struct lenovo_drvdata_tpx1cover *drv_data = hid_get_drvdata(hdev); + struct hid_report *report; + int led_nr = -1; + int led_nr_hw = -1; + + if (led_cdev == &drv_data->led_mute) { + led_nr = 0; + led_nr_hw = 0x64; + } else if (led_cdev == &drv_data->led_micmute) { + led_nr = 1; + led_nr_hw = 0x74; + } else if (led_cdev == &drv_data->led_fnlock) { + led_nr = 2; + led_nr_hw = 0x54; + } else { + hid_warn(hdev, "Invalid LED to set.\n"); + return; + } + + if (value == LED_OFF) + drv_data->led_state &= ~(1 << led_nr); + else + drv_data->led_state |= 1 << led_nr; + + report = hdev->report_enum[HID_OUTPUT_REPORT].report_id_hash[9]; + if (report) { + report->field[0]->value[0] = led_nr_hw; + report->field[0]->value[1] = (drv_data->led_state & (1 << led_nr)) + ? 0x02 : 0x01; + hid_hw_request(hdev, report, HID_REQ_SET_REPORT); + } +} + +static int lenovo_event_tpx1cover(struct hid_device *hdev, + struct hid_field *field, struct hid_usage *usage, __s32 value) +{ + int ret = 0; + + if ((usage->hid & HID_USAGE_PAGE) == HID_UP_CONSUMER + && (usage->hid & HID_USAGE) == 0x0001) { + + if (usage->usage_index == 0x8 && value == 1) { + struct lenovo_drvdata_tpx1cover *drv_data = hid_get_drvdata(hdev); + + if (drv_data && drv_data->led_present) { + drv_data->fnlock_state = lenovo_led_brightness_get_tpx1cover( + &drv_data->led_fnlock) == LED_OFF ? 1 : 0; + lenovo_led_brightness_set_tpx1cover( + &drv_data->led_fnlock, + drv_data->fnlock_state ? LED_FULL : LED_OFF); + } + } + + if (usage->usage_index == 0x9 && value == 1) { + input_event(field->hidinput->input, EV_KEY, KEY_MICMUTE, 1); + input_sync(field->hidinput->input); + input_event(field->hidinput->input, EV_KEY, KEY_MICMUTE, 0); + input_sync(field->hidinput->input); + ret = 1; + } + + if (usage->usage_index == 0xa && value == 1) { + input_event(field->hidinput->input, EV_KEY, KEY_CONFIG, 1); + input_sync(field->hidinput->input); + input_event(field->hidinput->input, EV_KEY, KEY_CONFIG, 0); + input_sync(field->hidinput->input); + + ret = 1; + } + + if (usage->usage_index == 0xb && value == 1) { + input_event(field->hidinput->input, EV_KEY, KEY_SEARCH, 1); + input_sync(field->hidinput->input); + input_event(field->hidinput->input, EV_KEY, KEY_SEARCH, 0); + input_sync(field->hidinput->input); + + ret = 1; + } + + if (usage->usage_index == 0xc && value == 1) { + input_event(field->hidinput->input, EV_KEY, KEY_SETUP, 1); + input_sync(field->hidinput->input); + input_event(field->hidinput->input, EV_KEY, KEY_SETUP, 0); + input_sync(field->hidinput->input); + + ret = 1; + } + + if (usage->usage_index == 0xd && value == 1) { + input_event(field->hidinput->input, EV_KEY, KEY_SWITCHVIDEOMODE, 1); + input_sync(field->hidinput->input); + input_event(field->hidinput->input, EV_KEY, KEY_SWITCHVIDEOMODE, 0); + input_sync(field->hidinput->input); + + ret = 1; + } + + if (usage->usage_index == 0xe && value == 1) { + input_event(field->hidinput->input, EV_KEY, KEY_RFKILL, 1); + input_sync(field->hidinput->input); + input_event(field->hidinput->input, EV_KEY, KEY_RFKILL, 0); + input_sync(field->hidinput->input); + + ret = 1; + } + } + + return ret; +} + static int lenovo_event(struct hid_device *hdev, struct hid_field *field, struct hid_usage *usage, __s32 value) { @@ -369,6 +628,8 @@ static int lenovo_event(struct hid_device *hdev, struct hid_field *field, case USB_DEVICE_ID_LENOVO_CUSBKBD: case USB_DEVICE_ID_LENOVO_CBTKBD: return lenovo_event_cptkbd(hdev, field, usage, value); + case USB_DEVICE_ID_LENOVO_X1_COVER: + return lenovo_event_tpx1cover(hdev, field, usage, value); default: return 0; } @@ -731,6 +992,251 @@ static int lenovo_probe_tpkbd(struct hid_device *hdev) return ret; } +static int lenovo_probe_tpx1cover_configure(struct hid_device *hdev) +{ + struct hid_report *report = hdev->report_enum[HID_OUTPUT_REPORT].report_id_hash[9]; + struct lenovo_drvdata_tpx1cover *drv_data = hid_get_drvdata(hdev); + + if (!drv_data) + return -ENODEV; + + if (!report) + return -ENOENT; + + report->field[0]->value[0] = 0x54; + report->field[0]->value[1] = 0x20; + hid_hw_request(hdev, report, HID_REQ_SET_REPORT); + hid_hw_wait(hdev); + + report->field[0]->value[0] = 0x54; + report->field[0]->value[1] = 0x08; + hid_hw_request(hdev, report, HID_REQ_SET_REPORT); + hid_hw_wait(hdev); + + report->field[0]->value[0] = 0xA0; + report->field[0]->value[1] = 0x02; + hid_hw_request(hdev, report, HID_REQ_SET_REPORT); + hid_hw_wait(hdev); + + lenovo_led_brightness_set_tpx1cover(&drv_data->led_mute, + hid_lenovo_led_table[HID_LENOVO_LED_MUTE].state ? LED_FULL : LED_OFF); + hid_hw_wait(hdev); + + lenovo_led_brightness_set_tpx1cover(&drv_data->led_micmute, + hid_lenovo_led_table[HID_LENOVO_LED_MICMUTE].state ? LED_FULL : LED_OFF); + hid_hw_wait(hdev); + + lenovo_led_brightness_set_tpx1cover(&drv_data->led_fnlock, LED_FULL); + + return 0; +} + +static int lenovo_probe_tpx1cover_special_functions(struct hid_device *hdev) +{ + struct device *dev = &hdev->dev; + struct lenovo_drvdata_tpx1cover *drv_data = NULL; + + size_t name_sz = strlen(dev_name(dev)) + 16; + char *name_led = NULL; + + struct hid_report *report; + bool report_match = 1; + + int ret = 0; + + report = hid_validate_values(hdev, HID_INPUT_REPORT, 2, 0, 3); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_INPUT_REPORT, 3, 0, 16); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_OUTPUT_REPORT, 9, 0, 2); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 32, 0, 1); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 84, 0, 1); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 100, 0, 1); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 116, 0, 1); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 132, 0, 1); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 144, 0, 1); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 162, 0, 1); + report_match &= report ? 1 : 0; + + if (!report_match) { + ret = -ENODEV; + goto err; + } + + drv_data = devm_kzalloc(&hdev->dev, + sizeof(struct lenovo_drvdata_tpx1cover), + GFP_KERNEL); + + if (!drv_data) { + hid_err(hdev, + "Could not allocate memory for tpx1cover driver data\n"); + ret = -ENOMEM; + goto err; + } + + name_led = devm_kzalloc(&hdev->dev, name_sz, GFP_KERNEL); + if (!name_led) { + hid_err(hdev, "Could not allocate memory for mute led data\n"); + ret = -ENOMEM; + goto err_cleanup; + } + snprintf(name_led, name_sz, "%s:amber:mute", dev_name(dev)); + + drv_data->led_mute.name = name_led; + drv_data->led_mute.brightness_get = lenovo_led_brightness_get_tpx1cover; + drv_data->led_mute.brightness_set = lenovo_led_brightness_set_tpx1cover; + drv_data->led_mute.dev = dev; + hid_lenovo_led_table[HID_LENOVO_LED_MUTE].dev = &drv_data->led_mute; + led_classdev_register(dev, &drv_data->led_mute); + + + name_led = devm_kzalloc(&hdev->dev, name_sz, GFP_KERNEL); + if (!name_led) { + hid_err(hdev, + "Could not allocate memory for mic mute led data\n"); + ret = -ENOMEM; + goto err_cleanup; + } + snprintf(name_led, name_sz, "%s:amber:micmute", dev_name(dev)); + + drv_data->led_micmute.name = name_led; + drv_data->led_micmute.brightness_get = lenovo_led_brightness_get_tpx1cover; + drv_data->led_micmute.brightness_set = lenovo_led_brightness_set_tpx1cover; + drv_data->led_micmute.dev = dev; + hid_lenovo_led_table[HID_LENOVO_LED_MICMUTE].dev = &drv_data->led_micmute; + led_classdev_register(dev, &drv_data->led_micmute); + + + name_led = devm_kzalloc(&hdev->dev, name_sz, GFP_KERNEL); + if (!name_led) { + hid_err(hdev, + "Could not allocate memory for FN lock led data\n"); + ret = -ENOMEM; + goto err_cleanup; + } + + snprintf(name_led, name_sz, "%s:amber:fnlock", dev_name(dev)); + + drv_data->led_fnlock.name = name_led; + drv_data->led_fnlock.brightness_get = lenovo_led_brightness_get_tpx1cover; + drv_data->led_fnlock.brightness_set = lenovo_led_brightness_set_tpx1cover; + drv_data->led_fnlock.dev = dev; + hid_lenovo_led_table[HID_LENOVO_LED_FNLOCK].dev = &drv_data->led_fnlock; + led_classdev_register(dev, &drv_data->led_fnlock); + + drv_data->led_state = 0; + drv_data->fnlock_state = 1; + drv_data->led_present = 1; + + hid_set_drvdata(hdev, drv_data); + + return lenovo_probe_tpx1cover_configure(hdev); + +err_cleanup: + if (drv_data->led_fnlock.name) { + led_classdev_unregister(&drv_data->led_fnlock); + devm_kfree(&hdev->dev, (void *) drv_data->led_fnlock.name); + } + + if (drv_data->led_micmute.name) { + led_classdev_unregister(&drv_data->led_micmute); + devm_kfree(&hdev->dev, (void *) drv_data->led_micmute.name); + } + + if (drv_data->led_mute.name) { + led_classdev_unregister(&drv_data->led_mute); + devm_kfree(&hdev->dev, (void *) drv_data->led_mute.name); + } + + if (drv_data) + kfree(drv_data); + +err: + return ret; +} + +static int lenovo_probe_tpx1cover_touch(struct hid_device *hdev) +{ + struct hid_report *report; + bool report_match = 1; + int ret = 0; + + report = hid_validate_values(hdev, HID_INPUT_REPORT, 2, 0, 2); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_INPUT_REPORT, 2, 1, 2); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_INPUT_REPORT, 11, 0, 61); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_INPUT_REPORT, 12, 0, 61); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_INPUT_REPORT, 16, 0, 3); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_INPUT_REPORT, 16, 1, 2); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_OUTPUT_REPORT, 9, 0, 20); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_OUTPUT_REPORT, 10, 0, 20); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 14, 0, 1); + report_match &= report ? 1 : 0; + report = hid_validate_values(hdev, HID_FEATURE_REPORT, 15, 0, 3); + report_match &= report ? 1 : 0; + + if (!report_match) + ret = -ENODEV; + + return ret; +} + +static int lenovo_probe_tpx1cover(struct hid_device *hdev) +{ + int ret = 0; + + /* + * Probing for special function keys and LED control -> usb intf 1 + * Probing for touch input -> usb intf 2 (handled by rmi4 driver) + * Other (keyboard) -> usb intf 0 + */ + if (!lenovo_probe_tpx1cover_special_functions(hdev)) { + // special function keys and LED control + ret = 0; + } else if (!lenovo_probe_tpx1cover_touch(hdev)) { + // handled by rmi + ret = -ENODEV; + } else { + // keyboard + struct lenovo_drvdata_tpx1cover *drv_data; + + drv_data = devm_kzalloc(&hdev->dev, + sizeof(struct lenovo_drvdata_tpx1cover), + GFP_KERNEL); + + if (!drv_data) { + hid_err(hdev, + "Could not allocate memory for tpx1cover driver data\n"); + ret = -ENOMEM; + goto out; + } + + drv_data->led_state = 0; + drv_data->led_present = 0; + drv_data->fnlock_state = 0; + hid_set_drvdata(hdev, drv_data); + + ret = 0; + } + +out: + return ret; +} + static int lenovo_probe_cptkbd(struct hid_device *hdev) { int ret; @@ -803,6 +1309,9 @@ static int lenovo_probe(struct hid_device *hdev, case USB_DEVICE_ID_LENOVO_CBTKBD: ret = lenovo_probe_cptkbd(hdev); break; + case USB_DEVICE_ID_LENOVO_X1_COVER: + ret = lenovo_probe_tpx1cover(hdev); + break; default: ret = 0; break; @@ -843,6 +1352,42 @@ static void lenovo_remove_cptkbd(struct hid_device *hdev) &lenovo_attr_group_cptkbd); } +static void lenovo_remove_tpx1cover(struct hid_device *hdev) +{ + struct lenovo_drvdata_tpx1cover *drv_data = hid_get_drvdata(hdev); + + if (!drv_data) + return; + + if (drv_data->led_present) { + if (drv_data->led_fnlock.name) { + hid_lenovo_led_table[HID_LENOVO_LED_FNLOCK].dev = NULL; + + led_classdev_unregister(&drv_data->led_fnlock); + devm_kfree(&hdev->dev, (void *) drv_data->led_fnlock.name); + } + + if (drv_data->led_micmute.name) { + hid_lenovo_led_table[HID_LENOVO_LED_MICMUTE].dev = NULL; + + led_classdev_unregister(&drv_data->led_micmute); + devm_kfree(&hdev->dev, (void *) drv_data->led_micmute.name); + } + + if (drv_data->led_mute.name) { + hid_lenovo_led_table[HID_LENOVO_LED_MUTE].dev = NULL; + + led_classdev_unregister(&drv_data->led_mute); + devm_kfree(&hdev->dev, (void *) drv_data->led_mute.name); + } + } + + if (drv_data) + devm_kfree(&hdev->dev, drv_data); + + hid_set_drvdata(hdev, NULL); +} + static void lenovo_remove(struct hid_device *hdev) { switch (hdev->product) { @@ -853,6 +1398,9 @@ static void lenovo_remove(struct hid_device *hdev) case USB_DEVICE_ID_LENOVO_CBTKBD: lenovo_remove_cptkbd(hdev); break; + case USB_DEVICE_ID_LENOVO_X1_COVER: + lenovo_remove_tpx1cover(hdev); + break; } hid_hw_stop(hdev); @@ -883,6 +1431,7 @@ static int lenovo_input_configured(struct hid_device *hdev, { HID_USB_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_CUSBKBD) }, { HID_BLUETOOTH_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_CBTKBD) }, { HID_USB_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_TPPRODOCK) }, + { HID_USB_DEVICE(USB_VENDOR_ID_LENOVO, USB_DEVICE_ID_LENOVO_X1_COVER) }, { } }; diff --git a/include/linux/hid-lenovo.h b/include/linux/hid-lenovo.h new file mode 100644 index 0000000..d0b0145 --- /dev/null +++ b/include/linux/hid-lenovo.h @@ -0,0 +1,15 @@ + +#ifndef __HID_LENOVO_H__ +#define __HID_LENOVO_H__ + + +enum { + HID_LENOVO_LED_MUTE, + HID_LENOVO_LED_MICMUTE, + HID_LENOVO_LED_FNLOCK, + HID_LENOVO_LED_MAX, +}; + +int hid_lenovo_led_set(int led_num, bool on); + +#endif /* __HID_LENOVO_H_ */ -- 1.9.1

2 4

[alsa-devel] [PATCH v2 0/8] ASoC: sunxi: Add support for audio codec in A23/H3 SoCs
by Chen-Yu Tsai 09 Dec '16

09 Dec '16

Hi everyone, This is v2 of my Allwinner A23 and H3 audio codec support series. Changes since v1: - Use DEFINE_RES_MEM for the analog path controls block resources. - Added Rob's ack. This series adds support for the audio codec found in Allwinner A23 and H3 SoCs. The design and data paths are similar to the audio codec found in earlier SoCs such as the A31. The analog audio paths are symmetrical with left/right channels and down-mix selectors for mono differential output. What deviates from previous SoCs is that the analog path controls have been moved to a separate control bus, accessed through a message box like register interface in the PRCM block. This necessitates writing a separate component driver for it, which is then tied into the sound card as an ASoC auxiliary device. Patch 1 adds the analog path controls block to the sun6i-prcm driver as a sub-device, for the A23. The H3 currently does not use the PRCM driver. Patch 2 adds PCM and card support for the A23 codec to the sun4i-codec driver. Patch 3 adds a device node for the analog path controls block to the A23 dtsi. Patch 4 adds a device node for the audio codec, and the phandle for the analog path controls block to the A23 dtsi. Patch 5 enables the audio codec for the A23 Q8 tablets. On these tablets the headphone output is driven in DC coupled, or "direct drive", mode. Patch 6 adds PCM and card support for the H3 codec to the sun4i-codec driver. Patch 7 adds device nodes for the audio codec and analog path controls block to the H3 dtsi. Patch 8 enables the audio codec on the Orange Pi PC. The audio output jack on the board is tied to the line out pins on the SoC. Please take a look and let me know what you think. In addition, the sun4i-codec driver is getting pretty large. Maybe we want to split the different parts into different files? Regards ChenYu Chen-Yu Tsai (8): mfd: sun6i-prcm: Add codec analog controls sub-device for Allwinner A23 ASoC: sun4i-codec: Add support for A23 codec ARM: dts: sun8i: Add codec analog path controls node in PRCM for A23/A33 ARM: dts: sun8i-a23: Add device node for internal audio codec ARM: dts: sun8i-a23: q8-tablet: Enable internal audio codec ASoC: sun4i-codec: Add support for H3 codec ARM: dts: sun8i-h3: Add device nodes for audio codec and its analog controls ARM: dts: sun8i-h3: orange-pi-pc: Enable audio codec .../devicetree/bindings/sound/sun4i-codec.txt | 14 +- arch/arm/boot/dts/sun8i-a23-a33.dtsi | 4 + arch/arm/boot/dts/sun8i-a23-q8-tablet.dts | 23 +++ arch/arm/boot/dts/sun8i-a23.dtsi | 16 ++ arch/arm/boot/dts/sun8i-h3-orangepi-pc.dts | 8 + arch/arm/boot/dts/sun8i-h3.dtsi | 19 +++ drivers/mfd/sun6i-prcm.c | 13 ++ sound/soc/sunxi/sun4i-codec.c | 179 +++++++++++++++++++++ 8 files changed, 274 insertions(+), 2 deletions(-) -- 2.10.2

3 12

[alsa-devel] [PATCH] ALSA: usb-audio: add implicit fb quirk for Axe-Fx II
by Alberto Aguirre 09 Dec '16

09 Dec '16

The Axe-Fx II implicit feedback end point and the data sync endpoint are in different interface descriptors. Add quirk to ensure a sync endpoint is properly configured. Signed-off-by: Alberto Aguirre <albaguirre(a)gmail.com> --- sound/usb/pcm.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 44d178e..0fe2d56 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -348,6 +348,16 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, alts = &iface->altsetting[1]; goto add_sync_ep; + case USB_ID(0x2466, 0x8003): + ep = 0x86; + iface = usb_ifnum_to_if(dev, 2); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; + } if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && -- 2.9.3

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: wm_adsp_buf_alloc should use kfree in error path
by Richard Fitzgerald 09 Dec '16

09 Dec '16

buf was allocated by kzalloc() so it should be passed to kfree() Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ff111a82..593b7d1 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -183,15 +183,15 @@ static struct wm_adsp_buf *wm_adsp_buf_alloc(const void *src, size_t len, struct wm_adsp_buf *buf = kzalloc(sizeof(*buf), GFP_KERNEL); if (buf == NULL) return NULL; buf->buf = vmalloc(len); if (!buf->buf) { - vfree(buf); + kfree(buf); return NULL; } memcpy(buf->buf, src, len); if (list) list_add_tail(&buf->list, list); -- 1.9.1

1 0

[alsa-devel] [PATCH 0/2] alsa-lib: dynamically adapt the avail_min on the slave
by Jiada Wang 09 Dec '16

09 Dec '16

When configuring avail_min to multiple of slave period size it can happen that user waits one slave period longer than needed for available data. Root cause is implicit grabbing of slave samples in avail_update operation. On next entering poll, the slave will wait for the avail_min threshold reached again, as he is not aware that there are already pending samples in the above layer which are not yet provided to user. Solution is to dynamically adapt the avail_min on the slave. Andreas Pape (2): plugin: dynamically update avail_min on slave rate: dynamic update avail_min on slave src/pcm/pcm_plugin.c | 48 +++++++++++++++++++++++++++++++++++++++++++++++- src/pcm/pcm_plugin.h | 3 +++ src/pcm/pcm_rate.c | 2 +- 3 files changed, 51 insertions(+), 2 deletions(-) -- 2.9.3

2 5

[alsa-devel] [PATCH] ALSA: usb-audio: add implicit fb quirk for Axe-Fx II
by Alberto Aguirre 08 Dec '16

08 Dec '16

The Axe-Fx II implicit feedback end point and the data sink endpoint are in different interface descriptors. Add quirk to ensure a sync endpoint is properly configured. Signed-off-by: Alberto Aguirre <albaguirre(a)gmail.com> --- sound/usb/pcm.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 44d178e..584f2a6 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -348,6 +348,16 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, alts = &iface->altsetting[1]; goto add_sync_ep; + case USB_ID(0x2466, 0x8003): /* Fractal Audio Axe-Fx II */ + ep = 0x86; + iface = usb_ifnum_to_if(dev, 2); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; + } if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && -- 2.9.3

1 0

[alsa-devel] warnings with clk_disable_unprepare()
by Pierre-Louis Bossart 08 Dec '16

08 Dec '16

Hi, In the modified cht-bsw_rt5645 machine driver for Asus X205TA and T100HA, I disable the audio mclk with a call to clk_disable_unprepare() when I see a SND_SOC_DAPM_EVENT_OFF event in the platform_clock_control hook. For some reason I seem to get multiple SND_SOC_DAPM_EVENT_OFF events and the clock framework complains with dmesg warnings that the refcount is already zero (which has no functional side effect). WARNING: CPU: 0 PID: 2002 at drivers/clk/clk.c:476 clk_unprepare+0x1a/0x21 WARNING: CPU: 2 PID: 2095 at drivers/clk/clk.c:594 clk_core_disable_lock+0x12/0x1b I could fix this in three different ways: - remove multiple SND_SOC_DAPM_EVENT_OFF events (not sure what the root cause is, maybe issues with jack detection?) - call a clk_is_enabled() or clk_refcount() function before calling clk_disable_unprepare() (but they don't seem to exist in include/linux/clk.h) - count the references myself (which does exactly the same as what the clk framework already does so it's not terribly elegant) Any suggestions on what the least-bad solution might be? Thanks -Pierre

2 1

[alsa-devel] [bug report] ASoC: topology: Only free TLV for volume mixers of a widget
by Dan Carpenter 08 Dec '16

08 Dec '16

Hello Mengdong Lin, The patch eea3dd4f1247: "ASoC: topology: Only free TLV for volume mixers of a widget" from Nov 25, 2016, leads to the following static checker warning: sound/soc/soc-topology.c:1566 soc_tplg_dapm_widget_create() error: uninitialized symbol 'kcontrol_type'. sound/soc/soc-topology.c 1440 static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, 1441 struct snd_soc_tplg_dapm_widget *w) 1442 { 1443 struct snd_soc_dapm_context *dapm = &tplg->comp->dapm; 1444 struct snd_soc_dapm_widget template, *widget; 1445 struct snd_soc_tplg_ctl_hdr *control_hdr; 1446 struct snd_soc_card *card = tplg->comp->card; 1447 unsigned int kcontrol_type; ^^^^^^^^^^^^^ 1448 int ret = 0; 1449 1450 if (strnlen(w->name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 1451 SNDRV_CTL_ELEM_ID_NAME_MAXLEN) 1452 return -EINVAL; 1453 if (strnlen(w->sname, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 1454 SNDRV_CTL_ELEM_ID_NAME_MAXLEN) 1455 return -EINVAL; 1456 1457 dev_dbg(tplg->dev, "ASoC: creating DAPM widget %s id %d\n", 1458 w->name, w->id); 1459 1460 memset(&template, 0, sizeof(template)); 1461 1462 /* map user to kernel widget ID */ 1463 template.id = get_widget_id(w->id); 1464 if (template.id < 0) 1465 return template.id; 1466 1467 template.name = kstrdup(w->name, GFP_KERNEL); 1468 if (!template.name) 1469 return -ENOMEM; 1470 template.sname = kstrdup(w->sname, GFP_KERNEL); 1471 if (!template.sname) { 1472 ret = -ENOMEM; 1473 goto err; 1474 } 1475 template.reg = w->reg; 1476 template.shift = w->shift; 1477 template.mask = w->mask; 1478 template.subseq = w->subseq; 1479 template.on_val = w->invert ? 0 : 1; 1480 template.off_val = w->invert ? 1 : 0; 1481 template.ignore_suspend = w->ignore_suspend; 1482 template.event_flags = w->event_flags; 1483 template.dobj.index = tplg->index; 1484 1485 tplg->pos += 1486 (sizeof(struct snd_soc_tplg_dapm_widget) + w->priv.size); 1487 if (w->num_kcontrols == 0) { 1488 template.num_kcontrols = 0; 1489 goto widget; Assume we hit this goto. 1490 } 1491 1492 control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; 1493 dev_dbg(tplg->dev, "ASoC: template %s has %d controls of type %x\n", 1494 w->name, w->num_kcontrols, control_hdr->type); [ snip ] 1546 1547 widget: 1548 ret = soc_tplg_widget_load(tplg, &template, w); 1549 if (ret < 0) 1550 goto hdr_err; 1551 1552 /* card dapm mutex is held by the core if we are loading topology 1553 * data during sound card init. */ 1554 if (card->instantiated) 1555 widget = snd_soc_dapm_new_control(dapm, &template); 1556 else 1557 widget = snd_soc_dapm_new_control_unlocked(dapm, &template); 1558 if (widget == NULL) { 1559 dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n", 1560 w->name); 1561 ret = -ENOMEM; 1562 goto hdr_err; 1563 } 1564 1565 widget->dobj.type = SND_SOC_DOBJ_WIDGET; 1566 widget->dobj.widget.kcontrol_type = kcontrol_type; Then it's not initialized here. I don't know if this causes an issue beyond the static checker warning. 1567 widget->dobj.ops = tplg->ops; 1568 widget->dobj.index = tplg->index; 1569 kfree(template.sname); regards, dan carpenter

2 1

[alsa-devel] [PATCH V2 1/2] ASoC: samsung: include gpio consumer.h
by Fabian Frederick 08 Dec '16

08 Dec '16

Fix the following build errors on X86_32 !GPIOLIB sound/soc/samsung/tm2_wm5110.c:220:3: error: implicit declaration of function 'gpiod_set_value_cansleep' [-Werror=implicit-function-declaration] sound/soc/samsung/tm2_wm5110.c:438:24: error: implicit declaration of function 'devm_gpiod_get' [-Werror=implicit-function-declaration] Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- V2: -Add configuration settings (suggested by Krzysztof Kozlowski) sound/soc/samsung/tm2_wm5110.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 5cdf7d1..24cc9d6 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -12,6 +12,7 @@ #include <linux/clk.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> #include <sound/pcm_params.h> -- 2.7.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: zte: spdif and i2s drivers are not zx296702 specific
by Shawn Guo 08 Dec '16

08 Dec '16

From: Shawn Guo <shawn.guo(a)linaro.org> ZTE ZX SPDIF and I2S drivers can work on not only ZX296702 but also other ZTE ZX family SoCs like ZX296718, which is an arm64 platform. Let's make a few renaming and tweak the Kconfig a bit to get the drivers available for other ZTE ZX platforms. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/zte/Kconfig | 16 ++++++++-------- sound/soc/zte/Makefile | 4 ++-- sound/soc/zte/{zx296702-i2s.c => zx-i2s.c} | 0 sound/soc/zte/{zx296702-spdif.c => zx-spdif.c} | 0 4 files changed, 10 insertions(+), 10 deletions(-) rename sound/soc/zte/{zx296702-i2s.c => zx-i2s.c} (100%) rename sound/soc/zte/{zx296702-spdif.c => zx-spdif.c} (100%) diff --git a/sound/soc/zte/Kconfig b/sound/soc/zte/Kconfig index c47eb25e441f..6d8a90d36315 100644 --- a/sound/soc/zte/Kconfig +++ b/sound/soc/zte/Kconfig @@ -1,17 +1,17 @@ -config ZX296702_SPDIF - tristate "ZX296702 spdif" - depends on SOC_ZX296702 || COMPILE_TEST +config ZX_SPDIF + tristate "ZTE ZX SPDIF Driver Support" + depends on ARCH_ZX || COMPILE_TEST depends on COMMON_CLK select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the - zx296702 spdif interface + ZTE ZX SPDIF interface -config ZX296702_I2S - tristate "ZX296702 i2s" - depends on SOC_ZX296702 || COMPILE_TEST +config ZX_I2S + tristate "ZTE ZX I2S Driver Support" + depends on ARCH_ZX || COMPILE_TEST depends on COMMON_CLK select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the - zx296702 i2s interface + ZTE ZX I2S interface diff --git a/sound/soc/zte/Makefile b/sound/soc/zte/Makefile index 254ed2c8c1a0..77768f5fd10c 100644 --- a/sound/soc/zte/Makefile +++ b/sound/soc/zte/Makefile @@ -1,2 +1,2 @@ -obj-$(CONFIG_ZX296702_SPDIF) += zx296702-spdif.o -obj-$(CONFIG_ZX296702_I2S) += zx296702-i2s.o +obj-$(CONFIG_ZX_SPDIF) += zx-spdif.o +obj-$(CONFIG_ZX_I2S) += zx-i2s.o diff --git a/sound/soc/zte/zx296702-i2s.c b/sound/soc/zte/zx-i2s.c similarity index 100% rename from sound/soc/zte/zx296702-i2s.c rename to sound/soc/zte/zx-i2s.c diff --git a/sound/soc/zte/zx296702-spdif.c b/sound/soc/zte/zx-spdif.c similarity index 100% rename from sound/soc/zte/zx296702-spdif.c rename to sound/soc/zte/zx-spdif.c -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: remove unused 'out_fmt' variable
by Kirtika Ruchandani 08 Dec '16

08 Dec '16

Commit 4cd9899f0d16 introduced struct skl_module_fmt* out_fmt without using it. Compiling with W=1 gives the below warning, fix it. sound/soc/intel/skylake/skl-topology.c: In function ‘skl_tplg_update_buffer_size’: sound/soc/intel/skylake/skl-topology.c:301:34: warning: variable ‘out_fmt’ set but not used [-Wunused-but-set-variable] This is a harmless warning and is only being fixed to reduce the noise with W=1 in the kernel. Fixes: 4cd9899f0d16 ("ASoC: Intel: Skylake: Add multiple pin formats") Cc: Hardik T Shah <hardik.t.shah(a)intel.com> Cc: Omair M Abdullah <omair.m.abdullah(a)intel.com> Cc: Jeeja KP <jeeja.kp(a)intel.com> Cc: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Kirtika Ruchandani <kirtika(a)chromium.org> --- sound/soc/intel/skylake/skl-topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index bd313c9..4352236a 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -298,7 +298,7 @@ static void skl_tplg_update_buffer_size(struct skl_sst *ctx, struct skl_module_cfg *mcfg) { int multiplier = 1; - struct skl_module_fmt *in_fmt, *out_fmt; + struct skl_module_fmt *in_fmt; int in_rate, out_rate; @@ -306,7 +306,6 @@ static void skl_tplg_update_buffer_size(struct skl_sst *ctx, * change for pin 0 only */ in_fmt = &mcfg->in_fmt[0]; - out_fmt = &mcfg->out_fmt[0]; if (mcfg->m_type == SKL_MODULE_TYPE_SRCINT) multiplier = 5; -- 2.8.0.rc3.226.g39d4020

3 3

[alsa-devel] [PATCH] ASoC: Intel: Skylake: remove unused 'runtime' variable
by Kirtika Ruchandani 08 Dec '16

08 Dec '16

skl_platform_open() defines and sets 'struct snd_pcm_runtime* runtime' but does not use it. Compiling with W=1 gives the following warning, fix it. sound/soc/intel/skylake/skl-pcm.c: In function ‘skl_platform_open’: sound/soc/intel/skylake/skl-pcm.c:941:26: warning: variable ‘runtime’ set but not used [-Wunused-but-set-variable] This was introduced with the initial driver commit a40e693c7f5e ("ASoC: Intel: Add Skylake HDA platform driver"). This is a harmless warning and is only being fixed to reduce the noise with W=1 in the kernel. Fixes: a40e693c7f5e ("ASoC: Intel: Add Skylake HDA platform driver") Cc: Jeeja KP <jeeja.kp(a)intel.com> Cc: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Kirtika Ruchandani <kirtika(a)chromium.org> --- sound/soc/intel/skylake/skl-pcm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 84b5101..3254507 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -938,14 +938,12 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { static int skl_platform_open(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_link *dai_link = rtd->dai_link; dev_dbg(rtd->cpu_dai->dev, "In %s:%s\n", __func__, dai_link->cpu_dai_name); - runtime = substream->runtime; snd_soc_set_runtime_hwparams(substream, &azx_pcm_hw); return 0; -- 2.8.0.rc3.226.g39d4020

2 1

[alsa-devel] [PATCH v5 00/14] ASoC: add OF graph base simple-card
by Kuninori Morimoto 08 Dec '16

08 Dec '16

Hi Rob, Mark These are v5 of OF graph base simple-card patch-set. I removed new "type" property on v4 patch-set, but I noticed that it is necessary for ALSA SoC binding purpose. Thus, this v5 has it again. For example HDMI case, its DT will has video and sound ports. This DT will be used from HDMI Video driver and HDMI sound driver. HDMI video side can handle all somehow, because it is fully under its control. HDMI sound side will just references it. But it can't know total Video/Sound ports number which is necessary for ALSA SoC side. For example, if HDMI video had 4 ports, sound had 2 ports, this case, HDMI sound ports will be port@4, port@5. Here, ALSA SoC side needs to know total 2 sound port, and, it should be handled as 1st / 2nd port. It is impossible without "type" property. 1) - 7) : OF graph new feature 8) - 14) : OF graph base simple-card Kuninori Morimoto (14): 1) Documentation: of: add type property 2) of_graph: add of_graph_get_remote_endpoint() 3) of_graph: add of_graph_port_type_is() 4) of_graph: add of_graph_get_port_parent() 5) of_graph: add of_graph_get_top_port() 6) of_graph: add for_each_of_port() / for_each_of_endpoint_in_port() 7) of_graph: add of_graph_get_endpoint_count() 8) ASoC: simple-card-utils: adjust for graph on asoc_simple_card_parse_card_name 9) ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() 10) ASoC: simple-card-utils: add asoc_simple_card_try_to_probe_graph_card() 11) ASoC: add simple-graph-card document 12) ASoC: add simple-graph-card support 13) ASoC: add simple-graph-scu-card document 14) ASoC: add simple-graph-scu-card support Documentation/devicetree/bindings/graph.txt | 21 + .../bindings/sound/simple-graph-card.txt | 67 +++ .../bindings/sound/simple-graph-scu-card.txt | 69 +++ drivers/of/base.c | 166 +++++++- include/linux/of_graph.h | 67 +++ include/sound/simple_card_utils.h | 13 + sound/soc/generic/Kconfig | 15 + sound/soc/generic/Makefile | 4 + sound/soc/generic/simple-card-utils.c | 98 ++++- sound/soc/generic/simple-card.c | 2 +- sound/soc/generic/simple-graph-card.c | 461 +++++++++++++++++++++ sound/soc/generic/simple-graph-scu-card.c | 441 ++++++++++++++++++++ sound/soc/generic/simple-scu-card.c | 2 +- 13 files changed, 1413 insertions(+), 13 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/simple-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-graph-scu-card.txt create mode 100644 sound/soc/generic/simple-graph-card.c create mode 100644 sound/soc/generic/simple-graph-scu-card.c -- 1.9.1

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[alsa-devel] [PATCH v4 0/2] ALSA controls management using index/device/sub-devices fields
by Arnaud Pouliquen 07 Dec '16

07 Dec '16

Previous discussions summary: - Topic of the patchset: Patch concerns ASoc DAI drivers. Aim is to be able to instantiate PCM control using device field according to the associated PCM char device. For this, a relationship between DAIs PCM control and PCM char device needs to be establish. => Proposal is to add field in DAI driver struct to declare PCM controls that need to be linked to PCM character device on DAI link probing. - Limitation of the patchset: This patchset seems only a first step as 2 concerns are been highlighted: - Patch is limited to "static" DAI-link. This not responds to backend (no_pcm dai links) and some none-DAI codecs, that could register PCm controls. In theses cases it is not possible to establish relationship during probe. => No solution proposed for time beeing. - This patchset does not fix conflict of 2 "identical" PCM controls declared by 2 drivers. For instance, a conflict can exist between a PCM control created in DAI driver and the same control declared in a codec driver. As index field is forced to 0, prefix should be used in codec. => To discuss solution in a separate thread as issue already present without this patchset several approaches seem possible like using index field ( control auto indexation in ASoC) or prefix in Naming. V4: - [PATCH v4 1/2] ASoC: core: allow DAI PCM controls bound to PCM device Fix and rework error management in soc_link_dai_pcm_controls function. - [PATCH v4 2/2] ASoC: sti: bind pcm controls to pcm device. No change since V2. V3: - [PATCH v3 1/2] ASoC: core: allow DAI PCM controls bound to PCM device Code optimization based on Takashi Sakamoto suggestion - [RFC V2 3/3] ASoC: hdmi-codec: Example of PCM control bound to PCM device for multi patch suppressed as it was send as example in previous version. Patch is valid but can have impact on existing drivers as control device field value is updated - [PATCH v3 2/2] ASoC: sti: bind pcm controls to pcm device. No change vs V2 V2: http://www.spinics.net/lists/alsa-devel/msg57045.html Aim of this version is to continue discussion on DAI PCM control focused on ASoC drivers. In this V2 implementation in Soc-core is simplified to limit impact on existing code. Update of the RFC V1 based on discussions: - [RFC 4/4] iecset: allow to select control with device and sub-device numbers no more part of the RFC V2, will be discussed in a separate thread - [RFC 2/4] ALSA: control: increment index field for duplicated control. no more part of the RFC V2, no more need as RFC subject is PCM controls - [RFC V2 1/3] ASoC: core: allow DAI PCM controls bound to PCM device Patch reworked from V1 to simplify implementation - Binding is not done for Dai links tagged with no_pcm (DPCM). - no more possibility to add the controls after the DAI link probing. - [RFC V2 2/3] ASoC: sti: bind PCM controls to PCM device. - [RFC V2 3/3] ASoC: hdmi-codec: Example of PCM control bound to PCM device for multi Example of implementation in STI DAI driver and HDMI-codec drivers V1: http://www.spinics.net/lists/alsa-devel/msg56479.html 1) Alsa-utils patch - iecset: allow to select control with device and sub-device numbers This patch allows to access to 2 iec controls differentiated by device/sub-devices numbers => For me, this patch is mandatory to be able to address the ASoC IEC controls, in case of no fix is implemented to allows index field update in ASoC. 2) Alsa driver patches - ASoC: core: allow PCM control binding to PCM device Add relationship between DAIs PCM controls and PCM device. - ALSA: control: increment index field for duplicated control. Generic implementation of the patch proposed in HDA driver (http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=ea9b…) - ASoC: sti: use bind_pcm_ctl implementation of bind_pcm_ctl for sti driver. Arnaud Pouliquen (2): ASoC: core: allow DAI PCM controls bound to PCM device ASoC: sti: bind pcm controls to pcm device. include/sound/soc-dai.h | 4 ++++ sound/soc/soc-core.c | 35 +++++++++++++++++++++++++++++++++++ sound/soc/sti/sti_uniperif.c | 33 ++++----------------------------- 3 files changed, 43 insertions(+), 29 deletions(-) -- 1.9.1

3 5

[alsa-devel] [PATCH] speaker-test: Fix chmap wav file selection.
by Arnaud Pouliquen 07 Dec '16

07 Dec '16

The channel selection currently does not work properly when a channel map control is provided but no manual channel map was explicitly requested with "-m". For example, the CEA/HDMI 6ch (surround 5.1) map is: FL, FR, LFE, FC, RL, RR. Tested command: speaker-test -D hdmi -c 6 -t wav Speaker-test tries to play channels in this following order: 0 - Front Left 3 - Front Center 1 - Front Right 5 - Rear Right 4 - Rear Left 2 - LFE But wav file played on associated speakers are not aligned. Here are the real files played: 0- /usr/share/sounds/alsa/Front_Left.wav => OK 3- /usr/share/sounds/alsa/Rear_Right.wav => OK 1- /usr/share/sounds/alsa/Front_Right.wav => OK 5- /usr/share/sounds/alsa/Rear_Center.wav => KO 4- found file /usr/share/sounds/alsa/Front_Center.wav => KO 2- /usr/share/sounds/alsa/Rear_Left.wav => KO Issue is that associated wav files ordering is reworked only if channel_map_set variable is set. Fix consists in allowing wavs re-ordering if a channel mapping as been get or set, i.e. channel_map is not null. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- speaker-test/speaker-test.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index 5b9cbec..46b0589 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -843,7 +843,7 @@ static int setup_wav_file(int chn) return check_wav_file(chn, given_test_wav_file); #ifdef CONFIG_SUPPORT_CHMAP - if (channel_map_set && chn < channel_map->channels) { + if (channel_map && chn < channel_map->channels) { int channel = channel_map->pos[chn] - SND_CHMAP_FL; if (channel >= 0 && channel < MAX_CHANNELS) return check_wav_file(chn, wavs[channel]); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: tidyup ssi->usrcnt counter check in hw_params
by Kuninori Morimoto 07 Dec '16

07 Dec '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ssi->usrcnt will be updated on snd_soc_dai_ops::trigger, but snd_pcm_ops::hw_params will be called *before* it. Thus, ssi->usrcnt is still 0 when 1st call. rsnd_ssi_hw_params() needs to check its called count, this means trigger should be if (ssi->usrcnt) instead of if (ssi->usrcnt > 1). Reported-by: Nguyen Viet Dung <nv-dung(a)jinso.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 932047a..dcb3d0a 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -433,11 +433,14 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod, int chan = params_channels(params); /* - * Already working. - * It will happen if SSI has parent/child connection. + * snd_pcm_ops::hw_params will be called *before* + * snd_soc_dai_ops::trigger. Thus, ssi->usrcnt is 0 + * in 1st call. */ - if (ssi->usrcnt > 1) { + if (ssi->usrcnt) { /* + * Already working. + * It will happen if SSI has parent/child connection. * it is error if child <-> parent SSI uses * different channels. */ -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: rsnd: fixup Suspend/Resume clock issue
by Kuninori Morimoto 07 Dec '16

07 Dec '16

Hi Mark These patches fixup Renesas sound's clock stopping issue when Suspend/Resume. These are tested on R-Car Gen3 Salvator-X board Kuninori Morimoto (2): ASoC: rsnd: setup BRGCKR/BRRA/BRRB when starting ASoC: rsnd: enable/disable ADG when suspend/resume timing sound/soc/sh/rcar/adg.c | 59 +++++++++++++++++++++++++++++------------------- sound/soc/sh/rcar/core.c | 24 ++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 3 +++ 3 files changed, 63 insertions(+), 23 deletions(-) -- 2.7.4

2 4

[alsa-devel] Cherry trail and Nau 88l24 codec
by Joseph Wang 07 Dec '16

07 Dec '16

Hi all, I've been trying to get sound working on my W9S pipo device. The people at Nuvoton have been extraordinarily helpful, but now the issue is writing the machine driver. My work is at https://github.com/joequant/nau8824 I've gotten the drivers to load, but getting timeouts with the firmware.... [ 102.447840] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 102.447872] intel_sst_acpi 808622A8:00: fw returned err -16 [ 103.471448] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 and then [ 116.820782] intel_sst_acpi 808622A8:00: fw returned err -16 [ 116.938921] intel_sst_acpi 808622A8:00: FW sent error response 0x5000e [ 116.939065] intel_sst_acpi 808622A8:00: FW sent error response 0x5000e [ 116.939579] intel_sst_acpi 808622A8:00: FW sent async error msg: [ 116.939614] (null)00040021 00000000 [ 116.939685] (null)00040021 00000000 [ 116.953260] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 116.953704] intel_sst_acpi 808622A8:00: Alloc for str 3 pipe 0xe [ 116.954062] sst-mfld-sst-mfld-platform: SET_SWM cmd max inputs reached One other path is that ICube has gotten this codec to work on their iwork8 Ultimate android device. They are required to release the source under GPL2, but haven't done so, so if all else fails, it will be a matter of tracking down someone in that company and explaining the issues to them. Since I'm in Hong Kong, I might just find where the things are manufactured, and just show up at their office. But it appears that there are some "magic bits" that are missing. One question that I have is what is the proper DAI to use (i.e. PCM or I2S) for cherry trail. The second question is that looking at the windows driver for nau88l24, there appears to be a test mode that runs the driver against a Realtek rt5645. Should I assume that this means that the initialization is similar? Also I'm quite interested in getting this to work. The W9S Pipo is a perfect netbook that has a big screen that is half the weight of my old laptop. The big thing is that someone realized that instead of putting in a SATA hard disk, that they can use an SD card as main storage, and that cuts down on the weight.

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[alsa-devel] [patch] ALSA: cs46xx: add a new line
by Dan Carpenter 07 Dec '16

07 Dec '16

We accidentally deleted a newline so now the "nreallocated++;" statement is hanging out way off to the right of the screen. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 4a0cbd2241d8..aa61615288ff 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -107,7 +107,8 @@ static int shadow_and_reallocate_code (struct snd_cs46xx * chip, u32 * data, u32 dev_dbg(chip->card->dev, "handle_wideop:[2] %05x:%05x addr %04x\n", - hival, loval, address); nreallocated++; + hival, loval, address); + nreallocated++; } /* wide_opcodes[j] == wide_op */ } /* for */ } /* mod_type == 0 ... */

2 1

[alsa-devel] [PATCH v7 0/3] Media Device Allocator API
by Shuah Khan 07 Dec '16

07 Dec '16

Media Device Allocator API to allows multiple drivers share a media device. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. Patches 0001 and 0002 are rebased to 4.9-rc7. Patch 0003 for snd-usb-audio is a rebase of the patch that was tested with the original Media Device Allocator patch series. snd-usb-audio patch includes the fixes found during 4.7-rc1 time in the original snd-usb-audio patch. Changes to patches in this series: Changes to patch 0001 since v6: - No changes Changes to patch 0002 since v6: - No changes Changes to patch 0003 since v6: - Addressed Takashi's review comments on patch v6 Depends on: https://lkml.org/lkml/2016/11/29/1001 Changes to patch 0001 since v5: (comments from Mauro and Sakari) - Removed struct device from media_device_instance. mdev.dev is used instead. - Added documentation. Changes to patch 0002: - No changes since patch v2, applies cleanly on top of the following: media: Protect enable_source and disable_source handler code paths https://lkml.org/lkml/2016/11/29/1001 Changes to patch 0003: - Changed to hold graph_mutex to check and call enable_source and disable_source handlers - to match au0828 doing the same in: media: Protect enable_source and disable_source handler code paths https://lkml.org/lkml/2016/11/29/1001 Changes to patch 0001 since v4: - Addressed Sakari's review comments with the exception of opting to not introduce media_device_usb_allocate() macro, and to not add a new routine to find media device instance to avoid a one line check. Changes to patch 0001 since v3: - Fixed undefined reference to `__media_device_usb_init compile error when CONFIG_USB is disabled. - Fixed kernel paging error when accessing /dev/mediaX after rmmod of the module that owns the media_device. The fix bumps the reference count for the owner when second driver comes along to share the media_device. If au0828 owns the media_device, then snd_usb_audio will bump the refcount for au0828, so it won't get deleted and vice versa. Changes to patch 0002 since v2: - Updated media_device_delete() to pass in module name. Changes to patch 0003 since the last version in 4.7-rc1: - Included fixes to bugs found during testing. - Updated to use the Media Allocator API. This patch series has been tested with au0828 and snd-usb-audio drivers. Ran bind and unbind loop tests on each driver with mc_nextgen_test and media_device_test app loop tests while checking lsmod and dmesg. Please refer to tools/testing/selftests/media_tests/regression_test.txt for testing done on this series. Shuah Khan (3): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API sound/usb: Use Media Controller API to share media resources Documentation/media/kapi/mc-core.rst | 37 ++++ drivers/media/Makefile | 3 +- drivers/media/media-dev-allocator.c | 133 ++++++++++++++ drivers/media/usb/au0828/au0828-core.c | 12 +- drivers/media/usb/au0828/au0828.h | 1 + include/media/media-dev-allocator.h | 54 ++++++ sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 321 +++++++++++++++++++++++++++++++++ sound/usb/media.h | 73 ++++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 17 files changed, 684 insertions(+), 14 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.7.4

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[alsa-devel] [patch] dbri: move dereference after check for NULL
by Dan Carpenter 06 Dec '16

06 Dec '16

We accidentally introduced a dereference before the NULL check in xmit_descs() as part of silencing a GCC warning. Fixes: 16f46050e709 ("dbri: Fix compiler warning") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Goes into the sparc tree. diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 3fe4468..52063b2 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -1702,7 +1702,7 @@ interrupts are disabled. static void xmit_descs(struct snd_dbri *dbri) { struct dbri_streaminfo *info; - u32 dvma_addr = (u32)dbri->dma_dvma; + u32 dvma_addr; s32 *cmd; unsigned long flags; int first_td; @@ -1710,6 +1710,7 @@ static void xmit_descs(struct snd_dbri *dbri) if (dbri == NULL) return; /* Disabled */ + dvma_addr = (u32)dbri->dma_dvma; info = &dbri->stream_info[DBRI_REC]; spin_lock_irqsave(&dbri->lock, flags);

2 1

[alsa-devel] Applied "ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture" to the asoc tree
by Mark Brown 06 Dec '16

06 Dec '16

The patch ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c1aee1d828caa86a663c57a56eead7fd5f22d80f Mon Sep 17 00:00:00 2001 From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Date: Tue, 6 Dec 2016 07:08:36 -0800 Subject: [PATCH] ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture This patch updates FE channel constraints & BE fixup to support quad channel DMIC capture. DMIC pin's BE fixup is configured based on channel input, i.e. either stereo or quad. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index bff80b467262..3a8c24a59020 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -30,6 +30,7 @@ #define BXT_DIALOG_CODEC_DAI "da7219-hifi" #define BXT_MAXIM_CODEC_DAI "HiFi" #define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 static struct snd_soc_jack broxton_headset; @@ -182,6 +183,16 @@ static struct snd_pcm_hw_constraint_list constraints_channels = { .mask = 0, }; +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + static int bxt_fe_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -258,7 +269,10 @@ static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - channels->min = channels->max = DUAL_CHANNEL; + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; return 0; } @@ -267,9 +281,9 @@ static int broxton_dmic_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - runtime->hw.channels_max = DUAL_CHANNEL; + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); + &constraints_channels_quad); return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -- 2.10.2

1 0

[alsa-devel] [PATCH] ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture
by sathyanarayana.nujella＠intel.com 06 Dec '16

06 Dec '16

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> This patch updates FE channel constraints & BE fixup to support quad channel DMIC capture. DMIC pin's BE fixup is configured based on channel input, i.e. either stereo or quad. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index bff80b4..3a8c24a 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -30,6 +30,7 @@ #define BXT_DIALOG_CODEC_DAI "da7219-hifi" #define BXT_MAXIM_CODEC_DAI "HiFi" #define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 static struct snd_soc_jack broxton_headset; @@ -182,6 +183,16 @@ static struct snd_pcm_hw_constraint_list constraints_channels = { .mask = 0, }; +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + static int bxt_fe_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -258,7 +269,10 @@ static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - channels->min = channels->max = DUAL_CHANNEL; + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; return 0; } @@ -267,9 +281,9 @@ static int broxton_dmic_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - runtime->hw.channels_max = DUAL_CHANNEL; + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); + &constraints_channels_quad); return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -- 2.8.4

1 0

[alsa-devel] [PATCH 0/2 v2] usb-audio misc fix
by Jiada Wang 06 Dec '16

06 Dec '16

This patch set contains the following patches Andreas Pape (1): ALSA: usb-audio: more tolerant packetsize Daniel Girnus (1): ALSA: usb-audio: avoid setting of sample rate multiple times on bus sound/usb/endpoint.c | 4 ++-- sound/usb/pcm.c | 21 +++++++++++---------- 2 files changed, 13 insertions(+), 12 deletions(-) -- 2.9.3

3 5

[alsa-devel] [PATCH] ASoC: nau8825: disable sinc filter for high THD of ADC
by John Hsu 06 Dec '16

06 Dec '16

This bit will enable 4th order SINC filter. =1, filter will enable; but it consumes higher power. =0, the sinc filter is disable, and it should always keep 0 value to get high THD. Therefor, disable the filter when codec initiation for better performance when recording. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 3 ++- sound/soc/codecs/nau8825.h | 2 ++ 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 2b89569..efe3a44 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1846,7 +1846,8 @@ static void nau8825_init_regs(struct nau8825 *nau8825) * (audible hiss). Set it to something better. */ regmap_update_bits(regmap, NAU8825_REG_ADC_RATE, - NAU8825_ADC_SYNC_DOWN_MASK, NAU8825_ADC_SYNC_DOWN_64); + NAU8825_ADC_SYNC_DOWN_MASK | NAU8825_ADC_SINC4_EN, + NAU8825_ADC_SYNC_DOWN_64); regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1, NAU8825_DAC_OVERSAMPLE_MASK, NAU8825_DAC_OVERSAMPLE_64); /* Disable DACR/L power */ diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 0672a25..5d1704e 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -269,6 +269,8 @@ #define NAU8825_BIQ_PATH_DAC (1 << NAU8825_BIQ_PATH_SFT) /* ADC_RATE (0x2b) */ +#define NAU8825_ADC_SINC4_SFT 4 +#define NAU8825_ADC_SINC4_EN (1 << NAU8825_ADC_SINC4_SFT) #define NAU8825_ADC_SYNC_DOWN_SFT 0 #define NAU8825_ADC_SYNC_DOWN_MASK 0x3 #define NAU8825_ADC_SYNC_DOWN_32 0 -- 2.6.4

2 3

[alsa-devel] [PATCH] ASoC: cs35l34: Simplify the logic to set CS35L34_MCLK_CTL setting
by Axel Lin 06 Dec '16

06 Dec '16

The logic of "value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_XXXXXX;" is unnecessary complex. By setting CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_MASK as the mask for regmap_update_bits() call, what the code does is exactly the same as setting value = CS35L34_MCLK_RATE_XXXXXX. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l34.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index e0f672a..7c5d151 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -610,15 +610,15 @@ static int cs35l34_dai_set_sysclk(struct snd_soc_dai *dai, switch (freq) { case CS35L34_MCLK_5644: - value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_5P6448; + value = CS35L34_MCLK_RATE_5P6448; cs35l34->mclk_int = freq; break; case CS35L34_MCLK_6: - value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_6P0000; + value = CS35L34_MCLK_RATE_6P0000; cs35l34->mclk_int = freq; break; case CS35L34_MCLK_6144: - value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_6P1440; + value = CS35L34_MCLK_RATE_6P1440; cs35l34->mclk_int = freq; break; case CS35L34_MCLK_11289: -- 2.9.3

3 2

[alsa-devel] [PATCH] ALSA: hda - Gate the mic jack on HP Z1 Gen3 AiO
by Takashi Iwai 06 Dec '16

06 Dec '16

HP Z1 Gen3 AiO with Conexant codec doesn't give an unsolicited event to the headset mic pin upon the jack plugging, it reports only to the headphone pin. It results in the missing mic switching. Let's fix up by simply gating the jack event. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ed62748a6d55..c15c51bea26d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -262,6 +262,7 @@ enum { CXT_FIXUP_CAP_MIX_AMP_5047, CXT_FIXUP_MUTE_LED_EAPD, CXT_FIXUP_HP_SPECTRE, + CXT_FIXUP_HP_GATE_MIC, }; /* for hda_fixup_thinkpad_acpi() */ @@ -633,6 +634,17 @@ static void cxt_fixup_cap_mix_amp_5047(struct hda_codec *codec, (1 << AC_AMPCAP_MUTE_SHIFT)); } +static void cxt_fixup_hp_gate_mic_jack(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* the mic pin (0x19) doesn't give an unsolicited event; + * probe the mic pin together with the headphone pin (0x16) + */ + if (action == HDA_FIXUP_ACT_PROBE) + snd_hda_jack_set_gating_jack(codec, 0x19, 0x16); +} + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ @@ -774,6 +786,10 @@ static const struct hda_fixup cxt_fixups[] = { { } } }, + [CXT_FIXUP_HP_GATE_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_hp_gate_mic_jack, + }, }; static const struct snd_pci_quirk cxt5045_fixups[] = { @@ -824,6 +840,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), + SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), -- 2.11.0

1 0

[alsa-devel] [PATCH v2] ALSA: hda: when comparing pin configurations, ignore assoc in addition to seq
by Kai-Heng Feng 06 Dec '16

06 Dec '16

Commit [64047d7f4912 ALSA: hda - ignore the assoc and seq when comparing pin configurations] intented to ignore both seq and assoc at pin comparing, but it only ignored seq. So that commit may still fail to match pins on some machines. Change the bitmask to also ignore assoc. v2: Use macro to do bit masking. Thanks to Hui Wang for the analysis. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 4ad29f8..a03cf68 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -884,6 +884,8 @@ void snd_hda_apply_fixup(struct hda_codec *codec, int action) } EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); +#define IGNORE_SEQ_ASSOC (~(AC_DEFCFG_SEQUENCE | AC_DEFCFG_DEF_ASSOC)) + static bool pin_config_match(struct hda_codec *codec, const struct hda_pintbl *pins) { @@ -901,7 +903,7 @@ static bool pin_config_match(struct hda_codec *codec, for (; t_pins->nid; t_pins++) { if (t_pins->nid == nid) { found = 1; - if ((t_pins->val & 0xfffffff0) == (cfg & 0xfffffff0)) + if ((t_pins->val & IGNORE_SEQ_ASSOC) == (cfg & IGNORE_SEQ_ASSOC)) break; else if ((cfg & 0xf0000000) == 0x40000000 && (t_pins->val & 0xf0000000) == 0x40000000) break; -- 2.10.2

2 1

[alsa-devel] [PATCH] ALSA: hda: ignore assoc when comparing pin configurations
by Kai-Heng Feng 06 Dec '16

06 Dec '16

Commit [64047d7f4912 ALSA: hda - ignore the assoc and seq when comparing pin configurations] may still fail to match pins on some machines, because the bitmask it used only ignore seq but not assoc. Change the bitmask to also ignore assoc. Thanks to Hui Wang for the analysis. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 4ad29f8..fdc30d0 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -901,7 +901,7 @@ static bool pin_config_match(struct hda_codec *codec, for (; t_pins->nid; t_pins++) { if (t_pins->nid == nid) { found = 1; - if ((t_pins->val & 0xfffffff0) == (cfg & 0xfffffff0)) + if ((t_pins->val & 0xffffff00) == (cfg & 0xffffff00)) break; else if ((cfg & 0xf0000000) == 0x40000000 && (t_pins->val & 0xf0000000) == 0x40000000) break; -- 2.10.2

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[alsa-devel] [PATCH v2 0/3] ASoC: intel: atom: Add FW version information
by Sebastien Guiriec 05 Dec '16

05 Dec '16

This patch series is adding FW version information of SST. Version is both log as dev_info and sysfs entry. Update since V1: - sysfs entry is set to "FW not yet loaded" in case we do not have loaded the FW once Sebastien Guiriec (3): ASoC: intel: atom: Add debug information related to FW version ASoC: Intel: atom: save FW version ASoC: Intel: atom: Add sysfs entry in order to store FW version sound/soc/intel/atom/sst/sst.c | 39 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/atom/sst/sst.h | 1 + sound/soc/intel/atom/sst/sst_ipc.c | 10 +++++++++- 3 files changed, 49 insertions(+), 1 deletion(-) -- 2.9.3 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

3 5

[alsa-devel] Applied "ASoC: Intel: atom: save FW version" to the asoc tree
by Mark Brown 05 Dec '16

05 Dec '16

The patch ASoC: Intel: atom: save FW version has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c7da1e9d2641f67e8b20703118ac02fe79604f10 Mon Sep 17 00:00:00 2001 From: Sebastien Guiriec <sebastien.guiriec(a)intel.com> Date: Mon, 5 Dec 2016 17:45:01 +0100 Subject: [PATCH] ASoC: Intel: atom: save FW version After the boot of the SST FW the firmware version is send back to the driver. This patch is saving the FW version inside the driver. Signed-off-by: Sebastien Guiriec <sebastien.guiriec(a)intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst.h | 1 + sound/soc/intel/atom/sst/sst_ipc.c | 6 ++++++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index 3f493862e98d..5c9a51cc77aa 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -436,6 +436,7 @@ struct intel_sst_drv { */ char firmware_name[FW_NAME_SIZE]; + struct snd_sst_fw_version fw_version; struct sst_fw_save *fw_save; }; diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 7934a0047384..374bb61c596d 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -242,6 +242,12 @@ static void process_fw_init(struct intel_sst_drv *sst_drv_ctx, dev_dbg(sst_drv_ctx->dev, "Build date %s Time %s\n", init->build_info.date, init->build_info.time); + /* Save FW version */ + sst_drv_ctx->fw_version.type = init->fw_version.type; + sst_drv_ctx->fw_version.major = init->fw_version.major; + sst_drv_ctx->fw_version.minor = init->fw_version.minor; + sst_drv_ctx->fw_version.build = init->fw_version.build; + ret: sst_wake_up_block(sst_drv_ctx, retval, FW_DWNL_ID, 0 , NULL, 0); } -- 2.10.2

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[alsa-devel] [PATCH v5 0/4] ASoC: intel: atom: Add FW version information
by Sebastien Guiriec 05 Dec '16

05 Dec '16

This patch series is adding FW version information of SST. Version is both log as dev_info and sysfs entry. Update since V1: - sysfs entry is set to "FW not yet loaded" in case we do not have loaded the FW once Update since V2: - Add Vinod ack. https://www.spinics.net/lists/alsa-devel/msg56521.html Update since V3: - Add sysfs entry description. Update since V4: - Add ack from vinod on documentation - Resend on top of v4.9-rc1. Sebastien Guiriec (4): ASoC: intel: atom: Add debug information related to FW version ASoC: Intel: atom: save FW version ASoC: Intel: atom: Add sysfs entry in order to store FW version Documentation: ABI: Add ASoC Intel SST Atom sysfs description Documentation/ABI/testing/sysfs-platform-sst-atom | 17 ++++++++++ sound/soc/intel/atom/sst/sst.c | 39 +++++++++++++++++++++++ sound/soc/intel/atom/sst/sst.h | 1 + sound/soc/intel/atom/sst/sst_ipc.c | 10 +++++- 4 files changed, 66 insertions(+), 1 deletion(-) create mode 100644 Documentation/ABI/testing/sysfs-platform-sst-atom -- 2.9.3 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

2 6

[alsa-devel] [PATCH v6 0/3] ASoC: intel: atom: Add FW version information
by Sebastien Guiriec 05 Dec '16

05 Dec '16

This patch series is adding FW version information of SST. Version is both log as dev_info and sysfs entry. Update since V1: - sysfs entry is set to "FW not yet loaded" in case we do not have loaded the FW once Update since V2: - Add Vinod ack. https://www.spinics.net/lists/alsa-devel/msg56521.html Update since V3: - Add sysfs entry description. Update since V4: - Add ack from vinod on documentation - Resend on top of v4.9-rc1. Update since V5: - Resend on top of topic/intel - Remove first patch as the first version was already applied inside topic/intel without getting "Applied" notifiaction. https://www.spinics.net/lists/alsa-devel/msg56122.html Sebastien Guiriec (3): ASoC: Intel: atom: save FW version ASoC: Intel: atom: Add sysfs entry in order to store FW version Documentation: ABI: Add ASoC Intel SST Atom sysfs description Documentation/ABI/testing/sysfs-platform-sst-atom | 17 ++++++++++ sound/soc/intel/atom/sst/sst.c | 39 +++++++++++++++++++++++ sound/soc/intel/atom/sst/sst.h | 1 + sound/soc/intel/atom/sst/sst_ipc.c | 6 ++++ 4 files changed, 63 insertions(+) create mode 100644 Documentation/ABI/testing/sysfs-platform-sst-atom -- 2.9.3 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

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Re: [alsa-devel] [PATCH v3] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches
by Peter Ujfalusi 05 Dec '16

05 Dec '16

Hi, On 12/01/2016 03:19 AM, Matt Ranostay wrote: > We can get audio errors if hitting deeper idle states on omaps: > > [alsa.c:230] error: Fatal problem with alsa output, error -5. > [audio.c:614] error: Error in writing audio (Input/output error?)! > > This seems to happen with off mode idle enabled as power for the > whole SoC may get cut off between filling the McBSP fifo using DMA. > While active DMA blocks deeper idle states in hardware, McBSP > activity does not seem to do so. > > Basing the QoS latency on the FIFO size with the approximate delay > required being 2.3 milliseconds per buffer location. > > Note that this is different from snd_pcm_hw_constraint_step() as > that can configure things based on the buffer and period size. > However, that does not help to block idle between the fifo fills. > > Note that in theory some omaps are capable of playing audio while > hitting deeper idle states. If somebody needs that and gets it > working, we can set the latency requirements accordingly. > > Based on the original patch by Tony Lindgren > Link: https://patchwork.kernel.org/patch/9305867/ > > Cc: Tony Lindgren <tony(a)atomide.com> > Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> > Signed-off-by: Matt Ranostay <matt(a)ranostay.consulting> > --- > Changes from v1: > * add calculations for latency per number of FIFO locations > > Changes from v2: > * add missing mcbsp.h header change > > sound/soc/omap/mcbsp.c | 15 +++++++++++++++ > sound/soc/omap/mcbsp.h | 2 ++ > 2 files changed, 17 insertions(+) > > diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c > index 06fec5699cc8..b4a773ca2bac 100644 > --- a/sound/soc/omap/mcbsp.c > +++ b/sound/soc/omap/mcbsp.c > @@ -25,6 +25,7 @@ > #include <linux/io.h> > #include <linux/slab.h> > #include <linux/pm_runtime.h> > +#include <linux/pm_qos.h> > > #include <linux/platform_data/asoc-ti-mcbsp.h> > > @@ -637,12 +638,21 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) > * Here we start the McBSP, by enabling transmitter, receiver or both. > * If no transmitter or receiver is active prior calling, then sample-rate > * generator and frame sync are started. > + * > + * Also setting of the QoS latency for the FIFO which varies upon the buffer > + * size. Approximately 2.3 milliseconds per FIFO location. > */ > void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) > { > int enable_srg = 0; > + int latency = mcbsp->pdata->buffer_size * 23; I think this is not correct. The McBSP FIFO time depends on the sample rate and on the number of channels the audio is using. With 8KHz mono you have 12 times more time per FIFO element compared to 48KHz stereo. As it has been discussed with Tony we should calculate the QoS latency in hw_params: latency_ms = ((FIFOsize - FIFOthreshold) / channels) * 1000/sampling-rate On OMAP3.McBSP2 for example (44.1KHz, stereo): FIFO threshold 128 - DMA request will be triggered when 128 slots are free in the FIFO - at that point we have still 1152 words in the FIFO. - if the C wakeup latency is longer then what it takes to play out the samples from the FIFO (13.06ms), we will drain the FIFO and got underflow. - in this case the QOS should be set as 13.06ms FIFO threshold 1024 - DMA request will be triggered when 1024 slots are free in the FIFO - at that point we have still 256 words in the FIFO. - if the C wakeup latency is longer then what it takes to play out the samples from the FIFO (2.9ms), we will drain the FIFO and got underflow. - in this case the QOS should be 2.9ms On other McBSPs with 128 word FIFO the required latency is shorter to ensure we don't drain the FIFO. > u16 w; > > + /* Prevent omap hardware from hitting off between fifo fills */ > + if (latency) > + pm_qos_add_request(&mcbsp->pm_qos_req, > + PM_QOS_CPU_DMA_LATENCY, latency); > + > if (mcbsp->st_data) > omap_st_start(mcbsp); > > @@ -731,6 +741,8 @@ void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) > > if (mcbsp->st_data) > omap_st_stop(mcbsp); > + > + pm_qos_remove_request(&mcbsp->pm_qos_req); > } > > int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) > @@ -1098,6 +1110,9 @@ int omap_mcbsp_init(struct platform_device *pdev) > > void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) > { > + if (pm_qos_request_active(&mcbsp->pm_qos_req)) > + pm_qos_remove_request(&mcbsp->pm_qos_req); > + > if (mcbsp->pdata->buffer_size) > sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); > > diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h > index 61e93b1c185d..e603f33f4082 100644 > --- a/sound/soc/omap/mcbsp.h > +++ b/sound/soc/omap/mcbsp.h > @@ -325,6 +325,8 @@ struct omap_mcbsp { > unsigned int in_freq; > int clk_div; > int wlen; > + > + struct pm_qos_request pm_qos_req; > }; > > void omap_mcbsp_config(struct omap_mcbsp *mcbsp, > -- Péter

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[alsa-devel] [PATCH] ASoC: cs42l56: Fix misuse of regmap_update_bits
by Florian Vaussard 05 Dec '16

05 Dec '16

Using regmap_update_bits(..., mask, 1) with 'mask' following (1 << k) and k greater than 0 is wrong. Indeed, _regmap_update_bits will perform (mask & 1), which results in 0 if LSB of mask is 0. Thus the call regmap_update_bits(..., mask, 1) is in reality equivalent to regmap_update_bits(..., mask, 0). In such a case, the correct use is regmap_update_bits(..., mask, mask). This driver is performing such a mistake with the CS42L56_AIN*_REF_MASK masks, which equal 0x10, 0x20, 0x40 and 0x80. Fix the driver to make it consistent with the API. Please note that this change is untested, as I do not have this piece of hardware. Testers are welcome! Signed-off-by: Florian Vaussard <florian.vaussard(a)heig-vd.ch> --- sound/soc/codecs/cs42l56.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 3e2c046..cb6ca85 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1277,19 +1277,23 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client, if (cs42l56->pdata.ain1a_ref_cfg) regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, - CS42L56_AIN1A_REF_MASK, 1); + CS42L56_AIN1A_REF_MASK, + CS42L56_AIN1A_REF_MASK); if (cs42l56->pdata.ain1b_ref_cfg) regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, - CS42L56_AIN1B_REF_MASK, 1); + CS42L56_AIN1B_REF_MASK, + CS42L56_AIN1B_REF_MASK); if (cs42l56->pdata.ain2a_ref_cfg) regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, - CS42L56_AIN2A_REF_MASK, 1); + CS42L56_AIN2A_REF_MASK, + CS42L56_AIN2A_REF_MASK); if (cs42l56->pdata.ain2b_ref_cfg) regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, - CS42L56_AIN2B_REF_MASK, 1); + CS42L56_AIN2B_REF_MASK, + CS42L56_AIN2B_REF_MASK); if (cs42l56->pdata.micbias_lvl) regmap_update_bits(cs42l56->regmap, CS42L56_GAIN_BIAS_CTL, -- 2.5.5

4 3

[alsa-devel] [PATCH 1/2] ASoC: topology: Allow a widget to have multiple enum controls
by mengdong.lin＠linux.intel.com 05 Dec '16

05 Dec '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This patch can create multiple enumerated mixer controls for a widget. Previously topology kernel driver assumes a widget can have only one emumerated mixer control. We need to remove this restriction for Broxton. Its firmware modules (widgets) may need multiple enum controls based on the channel and MIC combination. No ABI change is needed. The ABI allows a widget to embed multiple controls. Reported-by: G Kranthi <gudishax.kranthikumar(a)intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4dfdc65..63e1a50 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -486,21 +486,24 @@ static void remove_widget(struct snd_soc_component *comp, dobj->ops->widget_unload(comp, dobj); /* - * Dynamic Widgets either have 1 enum kcontrol or 1..N mixers. + * Dynamic Widgets either have 1..N enum kcontrols or mixers. * The enum may either have an array of values or strings. */ if (dobj->widget.kcontrol_enum) { /* enumerated widget mixer */ - struct soc_enum *se = - (struct soc_enum *)w->kcontrols[0]->private_value; + for (i = 0; i < w->num_kcontrols; i++) { + struct snd_kcontrol *kcontrol = w->kcontrols[i]; + struct soc_enum *se = + (struct soc_enum *)kcontrol->private_value; - snd_ctl_remove(card, w->kcontrols[0]); + snd_ctl_remove(card, kcontrol); - kfree(se->dobj.control.dvalues); - for (i = 0; i < se->items; i++) - kfree(se->dobj.control.dtexts[i]); + kfree(se->dobj.control.dvalues); + for (i = 0; i < se->items; i++) + kfree(se->dobj.control.dtexts[i]); - kfree(se); + kfree(se); + } kfree(w->kcontrol_news); } else { /* non enumerated widget mixer */ @@ -1256,98 +1259,105 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( } static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( - struct soc_tplg *tplg) + struct soc_tplg *tplg, int num_kcontrols) { struct snd_kcontrol_new *kc; struct snd_soc_tplg_enum_control *ec; struct soc_enum *se; - int i, err; - - ec = (struct snd_soc_tplg_enum_control *)tplg->pos; - tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + - ec->priv.size); - - /* validate kcontrol */ - if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == - SNDRV_CTL_ELEM_ID_NAME_MAXLEN) - return NULL; + int i, j, err; - kc = kzalloc(sizeof(*kc), GFP_KERNEL); + kc = kcalloc(num_kcontrols, sizeof(*kc), GFP_KERNEL); if (kc == NULL) return NULL; - se = kzalloc(sizeof(*se), GFP_KERNEL); - if (se == NULL) - goto err; + for (i = 0; i < num_kcontrols; i++) { + ec = (struct snd_soc_tplg_enum_control *)tplg->pos; + /* validate kcontrol */ + if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return NULL; + + se = kzalloc(sizeof(*se), GFP_KERNEL); + if (se == NULL) + goto err; - dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", - ec->hdr.name); + dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", + ec->hdr.name); - kc->name = ec->hdr.name; - kc->private_value = (long)se; - kc->iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc->access = ec->hdr.access; + kc[i].name = ec->hdr.name; + kc[i].private_value = (long)se; + kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc[i].access = ec->hdr.access; - /* we only support FL/FR channel mapping atm */ - se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); - se->shift_l = tplc_chan_get_shift(tplg, ec->channel, SNDRV_CHMAP_FL); - se->shift_r = tplc_chan_get_shift(tplg, ec->channel, SNDRV_CHMAP_FR); + /* we only support FL/FR channel mapping atm */ + se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); + se->shift_l = tplc_chan_get_shift(tplg, ec->channel, + SNDRV_CHMAP_FL); + se->shift_r = tplc_chan_get_shift(tplg, ec->channel, + SNDRV_CHMAP_FR); - se->items = ec->items; - se->mask = ec->mask; - se->dobj.index = tplg->index; + se->items = ec->items; + se->mask = ec->mask; + se->dobj.index = tplg->index; - switch (ec->hdr.ops.info) { - case SND_SOC_TPLG_CTL_ENUM_VALUE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - err = soc_tplg_denum_create_values(se, ec); - if (err < 0) { - dev_err(tplg->dev, "ASoC: could not create values for %s\n", - ec->hdr.name); + switch (ec->hdr.ops.info) { + case SND_SOC_TPLG_CTL_ENUM_VALUE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: + err = soc_tplg_denum_create_values(se, ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: could not create values for %s\n", + ec->hdr.name); + goto err_se; + } + /* fall through to create texts */ + case SND_SOC_TPLG_CTL_ENUM: + case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: + err = soc_tplg_denum_create_texts(se, ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: could not create texts for %s\n", + ec->hdr.name); + goto err_se; + } + break; + default: + dev_err(tplg->dev, "ASoC: invalid enum control type %d for %s\n", + ec->hdr.ops.info, ec->hdr.name); goto err_se; } - /* fall through to create texts */ - case SND_SOC_TPLG_CTL_ENUM: - case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: - err = soc_tplg_denum_create_texts(se, ec); + + /* map io handlers */ + err = soc_tplg_kcontrol_bind_io(&ec->hdr, &kc[i], tplg); + if (err) { + soc_control_err(tplg, &ec->hdr, ec->hdr.name); + goto err_se; + } + + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, &kc[i], + (struct snd_soc_tplg_ctl_hdr *)ec); if (err < 0) { - dev_err(tplg->dev, "ASoC: could not create texts for %s\n", + dev_err(tplg->dev, "ASoC: failed to init %s\n", ec->hdr.name); goto err_se; } - break; - default: - dev_err(tplg->dev, "ASoC: invalid enum control type %d for %s\n", - ec->hdr.ops.info, ec->hdr.name); - goto err_se; - } - /* map io handlers */ - err = soc_tplg_kcontrol_bind_io(&ec->hdr, kc, tplg); - if (err) { - soc_control_err(tplg, &ec->hdr, ec->hdr.name); - goto err_se; - } - - /* pass control to driver for optional further init */ - err = soc_tplg_init_kcontrol(tplg, kc, - (struct snd_soc_tplg_ctl_hdr *)ec); - if (err < 0) { - dev_err(tplg->dev, "ASoC: failed to init %s\n", - ec->hdr.name); - goto err_se; + tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + + ec->priv.size); } return kc; err_se: - /* free values and texts */ - kfree(se->dobj.control.dvalues); - for (i = 0; i < ec->items; i++) - kfree(se->dobj.control.dtexts[i]); + for (; i >= 0; i--) { + /* free values and texts */ + se = (struct soc_enum *)kc[i].private_value; + kfree(se->dobj.control.dvalues); + for (j = 0; j < ec->items; j++) + kfree(se->dobj.control.dtexts[j]); - kfree(se); + kfree(se); + } err: kfree(kc); @@ -1499,9 +1509,10 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: template.dobj.widget.kcontrol_enum = 1; - template.num_kcontrols = 1; + template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = - soc_tplg_dapm_widget_denum_create(tplg); + soc_tplg_dapm_widget_denum_create(tplg, + template.num_kcontrols); if (!template.kcontrol_news) { ret = -ENOMEM; goto hdr_err; -- 2.5.0

2 1

[alsa-devel] [PATCH 2/2] ASoC: topology: Only free TLV for volume mixers of a widget
by mengdong.lin＠linux.intel.com 05 Dec '16

05 Dec '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This patch will check the type of embedded controls for a widget, and only free the TLV of volume mixers. Bytes controls don't have TLV. Just free the private value which is used as struct soc_mixer_control for volume mixers or soc_bytes_ext for bytes controls. No need to cast to these types before freeing it. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index d318fe4..86c884a 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -53,7 +53,7 @@ struct snd_soc_dobj_control { /* dynamic widget object */ struct snd_soc_dobj_widget { - unsigned int kcontrol_enum:1; /* this widget is an enum kcontrol */ + unsigned int kcontrol_type; /* kcontrol type: mixer, enum, bytes */ }; /* generic dynamic object - all dynamic objects belong to this struct */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 63e1a50..11feb19 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -489,7 +489,7 @@ static void remove_widget(struct snd_soc_component *comp, * Dynamic Widgets either have 1..N enum kcontrols or mixers. * The enum may either have an array of values or strings. */ - if (dobj->widget.kcontrol_enum) { + if (dobj->widget.kcontrol_type == SND_SOC_TPLG_TYPE_ENUM) { /* enumerated widget mixer */ for (i = 0; i < w->num_kcontrols; i++) { struct snd_kcontrol *kcontrol = w->kcontrols[i]; @@ -506,16 +506,21 @@ static void remove_widget(struct snd_soc_component *comp, } kfree(w->kcontrol_news); } else { - /* non enumerated widget mixer */ + /* volume mixer or bytes controls */ for (i = 0; i < w->num_kcontrols; i++) { struct snd_kcontrol *kcontrol = w->kcontrols[i]; - struct soc_mixer_control *sm = - (struct soc_mixer_control *) kcontrol->private_value; - kfree(w->kcontrols[i]->tlv.p); + if (dobj->widget.kcontrol_type + == SND_SOC_TPLG_TYPE_MIXER) + kfree(kcontrol->tlv.p); - snd_ctl_remove(card, w->kcontrols[i]); - kfree(sm); + snd_ctl_remove(card, kcontrol); + + /* Private value is used as struct soc_mixer_control + * for volume mixers or soc_bytes_ext for bytes + * controls. + */ + kfree((void *)kcontrol->private_value); } kfree(w->kcontrol_news); } @@ -1439,6 +1444,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, struct snd_soc_dapm_widget template, *widget; struct snd_soc_tplg_ctl_hdr *control_hdr; struct snd_soc_card *card = tplg->comp->card; + unsigned int kcontrol_type; int ret = 0; if (strnlen(w->name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == @@ -1494,6 +1500,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, case SND_SOC_TPLG_CTL_VOLSW_XR_SX: case SND_SOC_TPLG_CTL_RANGE: case SND_SOC_TPLG_DAPM_CTL_VOLSW: + kcontrol_type = SND_SOC_TPLG_TYPE_MIXER; /* volume mixer */ template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = soc_tplg_dapm_widget_dmixer_create(tplg, @@ -1508,7 +1515,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - template.dobj.widget.kcontrol_enum = 1; + kcontrol_type = SND_SOC_TPLG_TYPE_ENUM; /* enumerated mixer */ template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = soc_tplg_dapm_widget_denum_create(tplg, @@ -1519,6 +1526,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, } break; case SND_SOC_TPLG_CTL_BYTES: + kcontrol_type = SND_SOC_TPLG_TYPE_BYTES; /* bytes control */ template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = soc_tplg_dapm_widget_dbytes_create(tplg, @@ -1555,6 +1563,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, } widget->dobj.type = SND_SOC_DOBJ_WIDGET; + widget->dobj.widget.kcontrol_type = kcontrol_type; widget->dobj.ops = tplg->ops; widget->dobj.index = tplg->index; kfree(template.sname); -- 2.5.0

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Re: [alsa-devel] ProjectMix IO Feedback
by Takashi Sakamoto 05 Dec '16

05 Dec '16

\Hi Darren, (C.C.ed to alsa-devel and ffado-user) On Dec 1 2016 19:47, Darren Anderson wrote: > Hi Takashi, > > Just wanted to give you some feedback on how the ProjectMix IO is working > with your changes that were merged into the Linux Kernel. > > Out of the box, on Ubuntu with kernel version 4.4.0-31, on boot, the > interface makes its distinctive popping sound as it sets the clock rate, > showing that the driver is loading fine. It is also now detected by > PulseAudio (without using a jack sink, as I previously had to). I'm then > able to use ffado-mixer to control the device. Thanks for your feedback to ALSA bebob driver. But it's better to post this kind of report to public mailing list, instead of personal address. There might be people who need the information. > There are, however, three issues. > > Firstly, the sample rate seems to be set every time an audio stream is > about to begin. This can be quite irritating at times, due to the > aforementioned distinctive popping sound and lag in the starting of audio > streams. Would it be possible to set the sample rate once at boot time, and > leave it at this unless the user specifically instructs otherwise? Applications can set sampling rate as they want, by ALSA PCM interface. I think it better to keep it what it is unless there's few special cases. For example, ALSA dice driver performs what you expected, due to hardware restriction. Application can't set sampling rate by ALSA PCM interface and they require the other application for this purpose. > Secondly, the main audio seems to be routed from Pulse through the "Aux > 1/2" channel, as opposed to "Mixer 1/2". The device will output no audio > until the "Analog 1/2" output is set to use "Aux 1/2" in FFADO mixer. > However in fixing this issue, a separate issue may be created. As most > distributions play some sort of "start-up sound", and the device seems to > initialize to full volume unless instructed otherwise, the fact that the > output channel needs to be changed may actually save some ear drums. It may > be wise to ask the device to reduce to something reasonable like 25% volume > when the driver is loaded. Or perhaps just mute all outputs by default. Just after booting up, your device has no routing configuration, therefore it generates no sound till configured with use space applications. Default settings of gain/volume are 0dB, perhaps. But this can also be configured from user space applications. What I can do is to produce a way to control devices for applications in user land, and actual way to use the device is left to application developers. If users have something to do, they would do it. > And finally, I've have found that the quality of the audio output from the > device is of a somewhat lower fidelity than the equivalent output on > Windows. I'm currently testing the interface using my KRK Rokit monitor > speakers and finding that the output is lacking in low frequencies. It > sounds as if there is some high pass filtering going on. ALSA bebob driver allows applications to use the device with both of 16 bit sample and 24bit sample with adjusted 8 bit, for playbacking. I guess that your trial is to playback 16 bit PCM sample and looks with highpass filter. Please check the property of PCM substream. > I hope that you find this feedback useful to your ongoing efforts. You're > doing great work. > > If I can do anything else to help, don't hesitate to ask. Cheers Takashi Sakamoto

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[alsa-devel] [PATCH] ASoC: rsnd: rsnd_get_dalign() needs to care SSIU, not SSI
by Kuninori Morimoto 05 Dec '16

05 Dec '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> SSIU was controlled by SSI before, but commit c7f69ab53("ASoC: rsnd: use mod base common method on SSIU") separated it into ssiu.c But, it didn't care about rsnd_get_dalign() for judging SSI_BUSIF_DALIGN register value which changes the stream data order. This function will be called from cmd/src/ssiu now, but current code still cares ssi, not ssiu. This patch fix it up Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 6 +++--- sound/soc/sh/rcar/rsnd.h | 1 + 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index f181410..28ef9ab 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -306,7 +306,7 @@ u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io) */ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { - struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *ssiu = rsnd_io_to_mod_ssiu(io); struct rsnd_mod *target; struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); u32 val = 0x76543210; @@ -315,11 +315,11 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) if (rsnd_io_is_play(io)) { struct rsnd_mod *src = rsnd_io_to_mod_src(io); - target = src ? src : ssi; + target = src ? src : ssiu; } else { struct rsnd_mod *cmd = rsnd_io_to_mod_cmd(io); - target = cmd ? cmd : ssi; + target = cmd ? cmd : ssiu; } mask <<= runtime->channels * 4; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index a8f61d7..8ca1705 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -382,6 +382,7 @@ struct rsnd_dai_stream { }; #define rsnd_io_to_mod(io, i) ((i) < RSND_MOD_MAX ? (io)->mod[(i)] : NULL) #define rsnd_io_to_mod_ssi(io) rsnd_io_to_mod((io), RSND_MOD_SSI) +#define rsnd_io_to_mod_ssiu(io) rsnd_io_to_mod((io), RSND_MOD_SSIU) #define rsnd_io_to_mod_ssip(io) rsnd_io_to_mod((io), RSND_MOD_SSIP) #define rsnd_io_to_mod_src(io) rsnd_io_to_mod((io), RSND_MOD_SRC) #define rsnd_io_to_mod_ctu(io) rsnd_io_to_mod((io), RSND_MOD_CTU) -- 2.7.4

2 1

[alsa-devel] [PATCH 0/2] ASoC: cq93vc: enable compiling on COMPILE_TEST
by Kuninori Morimoto 05 Dec '16

05 Dec '16

Hi Lee, Mark Current cq93vc sound codec indicates that it has MFD_DAVINCI_VOICECODEC dependency, but it doesn't. This means we can compile it from non DAVINCE environment by COMPILE_TEST. These patches solve this issue. I tested this on ARCH=x86 + make allyesconfig, and ARCH=ARM + shmobile_defconfig + COMPILE_TEST. Kuninori Morimoto (2): mfd: davinci_voicecodec: tidyup header difinitions ASoC: cq93vc: remove MFD_DAVINCI_VOICECODEC dependency for CQ0093VC drivers/mfd/davinci_voicecodec.c | 1 + include/linux/mfd/davinci_voicecodec.h | 4 ---- sound/soc/codecs/Kconfig | 2 +- 3 files changed, 2 insertions(+), 5 deletions(-) -- 1.9.1

3 9

[alsa-devel] [PATCH] ASoC: simple_card_utils: tidyup file comment/define
by Kuninori Morimoto 05 Dec '16

05 Dec '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> simple_card_utils was created as simple_card_core in 1st prototype, and current code still have it. Let's tidyup Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/simple_card_utils.h | 8 ++++---- sound/soc/generic/simple-card-utils.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index fd641255..64e90ca 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -1,5 +1,5 @@ /* - * simple_card_core.h + * simple_card_utils.h * * Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> * @@ -7,8 +7,8 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ -#ifndef __SIMPLE_CARD_CORE_H -#define __SIMPLE_CARD_CORE_H +#ifndef __SIMPLE_CARD_UTILS_H +#define __SIMPLE_CARD_UTILS_H #include <sound/soc.h> @@ -68,4 +68,4 @@ void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, int asoc_simple_card_clean_reference(struct snd_soc_card *card); -#endif /* __SIMPLE_CARD_CORE_H */ +#endif /* __SIMPLE_CARD_UTILS_H */ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index e5b80f5..cf02625 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -1,5 +1,5 @@ /* - * simple-card-core.c + * simple-card-utils.c * * Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> * -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: Intel: Add pre & post suspend calls in bxt machine
by Vinod Koul 05 Dec '16

05 Dec '16

We need to disable jack detection before entering suspend and enable it after resume. This was essential as headphone jack widget will always remain ON if a jack is inserted and this results in no audio after system resumes from a suspend. Signed-off-by: Senthilnathan Veppur <senthilnathanx.veppur(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/boards/bxt_rt298.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 1309405b3808..b272b72e6a77 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -432,6 +432,33 @@ static int bxt_fe_startup(struct snd_pcm_substream *substream) }, }; +static int bxt_suspend_pre(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + rt298_mic_detect(codec, NULL); + break; + } + } + return 0; +} + +static int bxt_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + rt298_mic_detect(codec, &broxton_headset); + break; + } + } + return 0; +} + + /* broxton audio machine driver for SPT + RT298S */ static struct snd_soc_card broxton_rt298 = { .name = "broxton-rt298", @@ -445,6 +472,9 @@ static int bxt_fe_startup(struct snd_pcm_substream *substream) .dapm_routes = broxton_rt298_map, .num_dapm_routes = ARRAY_SIZE(broxton_rt298_map), .fully_routed = true, + .suspend_pre = bxt_suspend_pre, + .resume_post = bxt_resume_post, + }; static int broxton_audio_probe(struct platform_device *pdev) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: cq93vc: enable compiling on COMPILE_TEST
by Kuninori Morimoto 05 Dec '16

05 Dec '16

Hi Mark I posted cq93vc dependency fixup (= no depend to MFD_DAVINCI_VOICECODEC) patch which were 2 patches. [1/2] was tidyup patch for mfd side, and [2/2] was tidyup patch for ASoC Kconfig. [1/2] was accepted by Lee, but [2/2] wasn't. This is for it. This patch is based on Lee's git://git.kernel.org/pub/scm/linux/kernel/git/lee/mfd.git :: for-mfd-next Especially this commit 8ca9edc837932469b81b8b47ea43a074b6add970 ("mfd: davinci_voicecodec: Tidyup header difinitions") Kuninori Morimoto (1): ASoC: cq93vc: remove MFD_DAVINCI_VOICECODEC dependency for CQ0093VC drivers/mfd/davinci_voicecodec.c | 1 + include/linux/mfd/davinci_voicecodec.h | 4 ---- sound/soc/codecs/Kconfig | 2 +- 3 files changed, 2 insertions(+), 5 deletions(-) -- 1.9.1

2 2

[alsa-devel] [PATCH 0/3 v1] usb-misc fix
by Jiada Wang 05 Dec '16

05 Dec '16

This patch set contains the following patches Andreas Pape (1): ALSA: usb-audio: more tolerant packetsize Daniel Girnus (1): ALSA: usb-audio: avoid setting of sample rate multiple times on bus Mark Craske (1): ALSA: usb-audio: fix race in snd_usb_endpoint_stop sound/usb/endpoint.c | 24 ++++++++++++++++-------- sound/usb/pcm.c | 21 +++++++++++---------- 2 files changed, 27 insertions(+), 18 deletions(-) -- 2.9.3

4 22

[alsa-devel] [RFC PATCH v4 7/7] ALSA: x86: hdmi: continue playback even when display resolution changes
by Jerome Anand 02 Dec '16

02 Dec '16

When the display resolution changes, the drm disables the display pipes due to which audio rendering stops. At this time, we need to ensure the existing audio pointers and buffers are cleared out so that the playback can restarted once the display pipe is enabled with a different N/CTS values Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Jerome Anand <jerome.anand(a)intel.com> --- sound/x86/intel_hdmi_audio.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 9249521..d6fd638 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -43,6 +43,7 @@ static DEFINE_MUTEX(had_mutex); static int hdmi_card_index = SNDRV_DEFAULT_IDX1; static char *hdmi_card_id = SNDRV_DEFAULT_STR1; static struct snd_intelhad *had_data; +static int underrun_count; module_param(hdmi_card_index, int, 0444); MODULE_PARM_DESC(hdmi_card_index, @@ -1114,6 +1115,7 @@ static int snd_intelhad_open(struct snd_pcm_substream *substream) intelhaddata = snd_pcm_substream_chip(substream); had_stream = intelhaddata->private_data; runtime = substream->runtime; + underrun_count = 0; pm_runtime_get(intelhaddata->dev); @@ -1506,10 +1508,23 @@ static snd_pcm_uframes_t snd_intelhad_pcm_pointer( buf_id = intelhaddata->curr_buf % 4; had_read_register(AUD_BUF_A_LENGTH + (buf_id * HAD_REG_WIDTH), &t); - if (t == 0) { - pr_debug("discovered buffer done for buf %d\n", buf_id); - /* had_process_buffer_done(intelhaddata); */ + + if ((t == 0) || (t == ((u32)-1L))) { + underrun_count++; + pr_debug("discovered buffer done for buf %d, count = %d\n", + buf_id, underrun_count); + + if (underrun_count > (HAD_MIN_PERIODS/2)) { + pr_debug("assume audio_codec_reset, underrun = %d - do xrun\n", + underrun_count); + underrun_count = 0; + return SNDRV_PCM_POS_XRUN; + } + } else { + /* Reset Counter */ + underrun_count = 0; } + t = intelhaddata->buf_info[buf_id].buf_size - t; if (intelhaddata->stream_info.buffer_rendered) -- 2.9.3

1 0

[alsa-devel] [RFC PATCH v4 6/7] ALSA: x86: hdmi: Fixup some monitor
by Jerome Anand 02 Dec '16

02 Dec '16

This change was given to Canonical apparently to fix an issue with on some monitor brand. It's not clear what this patch does but it doesn't seem to have side effects. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Jerome Anand <jerome.anand(a)intel.com> --- sound/x86/intel_hdmi_audio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index d9ce750..9249521 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -337,6 +337,7 @@ static void snd_intelhad_reset_audio_v2(u8 reset) static int had_prog_status_reg(struct snd_pcm_substream *substream, struct snd_intelhad *intelhaddata) { + union aud_cfg cfg_val = {.cfg_regval = 0}; union aud_ch_status_0 ch_stat0 = {.status_0_regval = 0}; union aud_ch_status_1 ch_stat1 = {.status_1_regval = 0}; int format; @@ -347,6 +348,7 @@ static int had_prog_status_reg(struct snd_pcm_substream *substream, IEC958_AES0_NONAUDIO)>>1; ch_stat0.status_0_regx.clk_acc = (intelhaddata->aes_bits & IEC958_AES3_CON_CLOCK)>>4; + cfg_val.cfg_regx.val_bit = ch_stat0.status_0_regx.lpcm_id; switch (substream->runtime->rate) { case AUD_SAMPLE_RATE_32: @@ -426,7 +428,6 @@ int snd_intelhad_prog_audio_ctrl_v2(struct snd_pcm_substream *substream, else cfg_val.cfg_regx_v2.layout = LAYOUT1; - cfg_val.cfg_regx_v2.val_bit = 1; had_write_register(AUD_CONFIG, cfg_val.cfg_regval); return 0; } @@ -482,7 +483,6 @@ int snd_intelhad_prog_audio_ctrl_v1(struct snd_pcm_substream *substream, } - cfg_val.cfg_regx.val_bit = 1; had_write_register(AUD_CONFIG, cfg_val.cfg_regval); return 0; } -- 2.9.3

1 0

[alsa-devel] [RFC PATCH v4 5/7] ALSA: x86: hdmi: Improve position reporting
by Jerome Anand 02 Dec '16

02 Dec '16

Use a hw register to calculate sub-period position reports. This makes PulseAudio happier. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Jerome Anand <jerome.anand(a)intel.com> --- sound/x86/intel_hdmi_audio.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 461b7d7..d9ce750 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1492,6 +1492,8 @@ static snd_pcm_uframes_t snd_intelhad_pcm_pointer( { struct snd_intelhad *intelhaddata; u32 bytes_rendered = 0; + u32 t; + int buf_id; /* pr_debug("snd_intelhad_pcm_pointer called\n"); */ @@ -1502,6 +1504,14 @@ static snd_pcm_uframes_t snd_intelhad_pcm_pointer( return SNDRV_PCM_POS_XRUN; } + buf_id = intelhaddata->curr_buf % 4; + had_read_register(AUD_BUF_A_LENGTH + (buf_id * HAD_REG_WIDTH), &t); + if (t == 0) { + pr_debug("discovered buffer done for buf %d\n", buf_id); + /* had_process_buffer_done(intelhaddata); */ + } + t = intelhaddata->buf_info[buf_id].buf_size - t; + if (intelhaddata->stream_info.buffer_rendered) div_u64_rem(intelhaddata->stream_info.buffer_rendered, intelhaddata->stream_info.ring_buf_size, @@ -1509,7 +1519,7 @@ static snd_pcm_uframes_t snd_intelhad_pcm_pointer( intelhaddata->stream_info.buffer_ptr = bytes_to_frames( substream->runtime, - bytes_rendered); + bytes_rendered + t); return intelhaddata->stream_info.buffer_ptr; } -- 2.9.3

1 0

[alsa-devel] [RFC PATCH v4 4/7] ALSA: x86: hdmi: Add audio support for BYT and CHT
by Jerome Anand 02 Dec '16

02 Dec '16

Hdmi audio driver based on the child platform device created by gfx driver is implemented. This audio driver is derived from legacy intel hdmi audio driver. The interfaces for interaction between gfx and audio are updated and the driver implementation updated to derive interrupts in its own address space based on irq chip framework Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Jerome Anand <jerome.anand(a)intel.com> --- sound/x86/Makefile | 2 + sound/x86/intel_hdmi_audio.c | 1907 ++++++++++++++++++++++++++++++++++++++ sound/x86/intel_hdmi_audio.h | 201 ++++ sound/x86/intel_hdmi_audio_if.c | 551 +++++++++++ sound/x86/intel_hdmi_lpe_audio.c | 16 +- 5 files changed, 2671 insertions(+), 6 deletions(-) create mode 100644 sound/x86/intel_hdmi_audio.c create mode 100644 sound/x86/intel_hdmi_audio.h create mode 100644 sound/x86/intel_hdmi_audio_if.c diff --git a/sound/x86/Makefile b/sound/x86/Makefile index 78b2ae1..bc074d0 100644 --- a/sound/x86/Makefile +++ b/sound/x86/Makefile @@ -3,6 +3,8 @@ DRIVER_NAME := hdmi_lpe_audio ccflags-y += -Idrivers/gpu/drm/i915 $(DRIVER_NAME)-objs += \ + intel_hdmi_audio.o \ + intel_hdmi_audio_if.o \ intel_hdmi_lpe_audio.o obj-$(CONFIG_HDMI_LPE_AUDIO) += $(DRIVER_NAME).o diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c new file mode 100644 index 0000000..461b7d7 --- /dev/null +++ b/sound/x86/intel_hdmi_audio.c @@ -0,0 +1,1907 @@ +/* + * intel_hdmi_audio.c - Intel HDMI audio driver + * + * Copyright (C) 2016 Intel Corp + * Authors: Sailaja Bandarupalli <sailaja.bandarupalli(a)intel.com> + * Ramesh Babu K V <ramesh.babu(a)intel.com> + * Vaibhav Agarwal <vaibhav.agarwal(a)intel.com> + * Jerome Anand <jerome.anand(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * ALSA driver for Intel HDMI audio + */ + +#define pr_fmt(fmt) "had: " fmt + +#include <linux/platform_device.h> +#include <linux/io.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/acpi.h> +#include <asm/cacheflush.h> +#include <sound/pcm.h> +#include <sound/core.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/control.h> +#include <sound/initval.h> +#include "intel_hdmi_audio.h" + +static DEFINE_MUTEX(had_mutex); + +/*standard module options for ALSA. This module supports only one card*/ +static int hdmi_card_index = SNDRV_DEFAULT_IDX1; +static char *hdmi_card_id = SNDRV_DEFAULT_STR1; +static struct snd_intelhad *had_data; + +module_param(hdmi_card_index, int, 0444); +MODULE_PARM_DESC(hdmi_card_index, + "Index value for INTEL Intel HDMI Audio controller."); +module_param(hdmi_card_id, charp, 0444); +MODULE_PARM_DESC(hdmi_card_id, + "ID string for INTEL Intel HDMI Audio controller."); + +/* + * ELD SA bits in the CEA Speaker Allocation data block +*/ +static int eld_speaker_allocation_bits[] = { + [0] = FL | FR, + [1] = LFE, + [2] = FC, + [3] = RL | RR, + [4] = RC, + [5] = FLC | FRC, + [6] = RLC | RRC, + /* the following are not defined in ELD yet */ + [7] = 0, +}; + +/* + * This is an ordered list! + * + * The preceding ones have better chances to be selected by + * hdmi_channel_allocation(). + */ +static struct cea_channel_speaker_allocation channel_allocations[] = { +/* channel: 7 6 5 4 3 2 1 0 */ +{ .ca_index = 0x00, .speakers = { 0, 0, 0, 0, 0, 0, FR, FL } }, + /* 2.1 */ +{ .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } }, + /* Dolby Surround */ +{ .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } }, + /* surround40 */ +{ .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } }, + /* surround41 */ +{ .ca_index = 0x09, .speakers = { 0, 0, RR, RL, 0, LFE, FR, FL } }, + /* surround50 */ +{ .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } }, + /* surround51 */ +{ .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, + /* 6.1 */ +{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, + /* surround71 */ +{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, + +{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, +{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, +{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, +{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, +{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, +{ .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } }, +{ .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } }, +{ .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } }, +{ .ca_index = 0x17, .speakers = { FRC, FLC, 0, 0, FC, LFE, FR, FL } }, +{ .ca_index = 0x18, .speakers = { FRC, FLC, 0, RC, 0, 0, FR, FL } }, +{ .ca_index = 0x19, .speakers = { FRC, FLC, 0, RC, 0, LFE, FR, FL } }, +{ .ca_index = 0x1a, .speakers = { FRC, FLC, 0, RC, FC, 0, FR, FL } }, +{ .ca_index = 0x1b, .speakers = { FRC, FLC, 0, RC, FC, LFE, FR, FL } }, +{ .ca_index = 0x1c, .speakers = { FRC, FLC, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x1d, .speakers = { FRC, FLC, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x1e, .speakers = { FRC, FLC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x1f, .speakers = { FRC, FLC, RR, RL, FC, LFE, FR, FL } }, +}; + +static struct channel_map_table map_tables[] = { + { SNDRV_CHMAP_FL, 0x00, FL }, + { SNDRV_CHMAP_FR, 0x01, FR }, + { SNDRV_CHMAP_RL, 0x04, RL }, + { SNDRV_CHMAP_RR, 0x05, RR }, + { SNDRV_CHMAP_LFE, 0x02, LFE }, + { SNDRV_CHMAP_FC, 0x03, FC }, + { SNDRV_CHMAP_RLC, 0x06, RLC }, + { SNDRV_CHMAP_RRC, 0x07, RRC }, + {} /* terminator */ +}; + +/* hardware capability structure */ +static const struct snd_pcm_hardware snd_intel_hadstream = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_DOUBLE | + SNDRV_PCM_INFO_MMAP| + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH), + .formats = (SNDRV_PCM_FMTBIT_S24 | + SNDRV_PCM_FMTBIT_U24), + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .rate_min = HAD_MIN_RATE, + .rate_max = HAD_MAX_RATE, + .channels_min = HAD_MIN_CHANNEL, + .channels_max = HAD_MAX_CHANNEL, + .buffer_bytes_max = HAD_MAX_BUFFER, + .period_bytes_min = HAD_MIN_PERIOD_BYTES, + .period_bytes_max = HAD_MAX_PERIOD_BYTES, + .periods_min = HAD_MIN_PERIODS, + .periods_max = HAD_MAX_PERIODS, + .fifo_size = HAD_FIFO_SIZE, +}; + +/* Register access functions */ + +inline int had_get_hwstate(struct snd_intelhad *intelhaddata) +{ + /* Check for device presence -SW state */ + if (intelhaddata->drv_status == HAD_DRV_DISCONNECTED) { + pr_debug("%s:Device not connected:%d\n", __func__, + intelhaddata->drv_status); + return -ENODEV; + } + + return 0; +} + +inline int had_get_caps(enum had_caps_list query, void *caps) +{ + int retval; + struct snd_intelhad *intelhaddata = had_data; + + retval = had_get_hwstate(intelhaddata); + if (!retval) + retval = intelhaddata->query_ops.hdmi_audio_get_caps(query, + caps); + + return retval; +} + +inline int had_set_caps(enum had_caps_list set_element, void *caps) +{ + int retval; + struct snd_intelhad *intelhaddata = had_data; + + retval = had_get_hwstate(intelhaddata); + if (!retval) + retval = intelhaddata->query_ops.hdmi_audio_set_caps( + set_element, caps); + + return retval; +} + +inline int had_read_register(uint32_t offset, uint32_t *data) +{ + int retval; + struct snd_intelhad *intelhaddata = had_data; + + retval = had_get_hwstate(intelhaddata); + if (!retval) + retval = intelhaddata->reg_ops.hdmi_audio_read_register( + offset + intelhaddata->audio_cfg_offset, data); + + return retval; +} + +inline int had_write_register(uint32_t offset, uint32_t data) +{ + int retval; + struct snd_intelhad *intelhaddata = had_data; + + retval = had_get_hwstate(intelhaddata); + if (!retval) + retval = intelhaddata->reg_ops.hdmi_audio_write_register( + offset + intelhaddata->audio_cfg_offset, data); + + return retval; +} + +inline int had_read_modify(uint32_t offset, uint32_t data, uint32_t mask) +{ + int retval; + struct snd_intelhad *intelhaddata = had_data; + + retval = had_get_hwstate(intelhaddata); + if (!retval) + retval = intelhaddata->reg_ops.hdmi_audio_read_modify( + offset + intelhaddata->audio_cfg_offset, + data, mask); + + return retval; +} +/** + * had_read_modify_aud_config_v2 - Specific function to read-modify + * AUD_CONFIG register on VLV2.The had_read_modify() function should not + * directly be used on VLV2 for updating AUD_CONFIG register. + * This is because: + * Bit6 of AUD_CONFIG register is writeonly due to a silicon bug on VLV2 + * HDMI IP. As a result a read-modify of AUD_CONFIG regiter will always + * clear bit6. AUD_CONFIG[6:4] represents the "channels" field of the + * register. This field should be 1xy binary for configuration with 6 or + * more channels. Read-modify of AUD_CONFIG (Eg. for enabling audio) + * causes the "channels" field to be updated as 0xy binary resulting in + * bad audio. The fix is to always write the AUD_CONFIG[6:4] with + * appropriate value when doing read-modify of AUD_CONFIG register. + * + * @substream: the current substream or NULL if no active substream + * @data : data to be written + * @mask : mask + * + */ +inline int had_read_modify_aud_config_v2(struct snd_pcm_substream *substream, + uint32_t data, uint32_t mask) +{ + union aud_cfg cfg_val = {.cfg_regval = 0}; + u8 channels; + + /* + * If substream is NULL, there is no active stream. + * In this case just set channels to 2 + */ + if (substream) + channels = substream->runtime->channels; + else + channels = 2; + cfg_val.cfg_regx_v2.num_ch = channels - 2; + + data = data | cfg_val.cfg_regval; + mask = mask | AUD_CONFIG_CH_MASK_V2; + + pr_debug("%s : data = %x, mask =%x\n", __func__, data, mask); + + return had_read_modify(AUD_CONFIG, data, mask); +} + +/** + * snd_intelhad_enable_audio_v1 - to enable audio + * + * @substream: Current substream or NULL if no active substream. + * @enable: 1 if audio is to be enabled; 0 if audio is to be disabled. + * + */ +static void snd_intelhad_enable_audio_v1(struct snd_pcm_substream *substream, + u8 enable) +{ + had_read_modify(AUD_CONFIG, enable, BIT(0)); +} + +/** + * snd_intelhad_enable_audio_v2 - to enable audio + * + * @substream: Current substream or NULL if no active substream. + * @enable: 1 if audio is to be enabled; 0 if audio is to be disabled. + */ +static void snd_intelhad_enable_audio_v2(struct snd_pcm_substream *substream, + u8 enable) +{ + had_read_modify_aud_config_v2(substream, enable, BIT(0)); +} + +/** + * snd_intelhad_reset_audio_v1 - to reset audio subsystem + * + * @reset: 1 to reset audio; 0 to bring audio out of reset. + * + */ +static void snd_intelhad_reset_audio_v1(u8 reset) +{ + had_write_register(AUD_HDMI_STATUS, reset); +} + +/** + * snd_intelhad_reset_audio_v2 - to reset audio subsystem + * + * @reset: 1 to reset audio; 0 to bring audio out of reset. + * + */ +static void snd_intelhad_reset_audio_v2(u8 reset) +{ + had_write_register(AUD_HDMI_STATUS_v2, reset); +} + +/** + * had_prog_status_reg - to initialize audio channel status registers + * + * @substream:substream for which the prepare function is called + * @intelhaddata:substream private data + * + * This function is called in the prepare callback + */ +static int had_prog_status_reg(struct snd_pcm_substream *substream, + struct snd_intelhad *intelhaddata) +{ + union aud_ch_status_0 ch_stat0 = {.status_0_regval = 0}; + union aud_ch_status_1 ch_stat1 = {.status_1_regval = 0}; + int format; + + pr_debug("Entry %s\n", __func__); + + ch_stat0.status_0_regx.lpcm_id = (intelhaddata->aes_bits & + IEC958_AES0_NONAUDIO)>>1; + ch_stat0.status_0_regx.clk_acc = (intelhaddata->aes_bits & + IEC958_AES3_CON_CLOCK)>>4; + + switch (substream->runtime->rate) { + case AUD_SAMPLE_RATE_32: + ch_stat0.status_0_regx.samp_freq = CH_STATUS_MAP_32KHZ; + break; + + case AUD_SAMPLE_RATE_44_1: + ch_stat0.status_0_regx.samp_freq = CH_STATUS_MAP_44KHZ; + break; + case AUD_SAMPLE_RATE_48: + ch_stat0.status_0_regx.samp_freq = CH_STATUS_MAP_48KHZ; + break; + case AUD_SAMPLE_RATE_88_2: + ch_stat0.status_0_regx.samp_freq = CH_STATUS_MAP_88KHZ; + break; + case AUD_SAMPLE_RATE_96: + ch_stat0.status_0_regx.samp_freq = CH_STATUS_MAP_96KHZ; + break; + case AUD_SAMPLE_RATE_176_4: + ch_stat0.status_0_regx.samp_freq = CH_STATUS_MAP_176KHZ; + break; + case AUD_SAMPLE_RATE_192: + ch_stat0.status_0_regx.samp_freq = CH_STATUS_MAP_192KHZ; + break; + + default: + /* control should never come here */ + return -EINVAL; + break; + + } + had_write_register(AUD_CH_STATUS_0, ch_stat0.status_0_regval); + + format = substream->runtime->format; + + if (format == SNDRV_PCM_FORMAT_S16_LE) { + ch_stat1.status_1_regx.max_wrd_len = MAX_SMPL_WIDTH_20; + ch_stat1.status_1_regx.wrd_len = SMPL_WIDTH_16BITS; + } else if (format == SNDRV_PCM_FORMAT_S24_LE) { + ch_stat1.status_1_regx.max_wrd_len = MAX_SMPL_WIDTH_24; + ch_stat1.status_1_regx.wrd_len = SMPL_WIDTH_24BITS; + } else { + ch_stat1.status_1_regx.max_wrd_len = 0; + ch_stat1.status_1_regx.wrd_len = 0; + } + had_write_register(AUD_CH_STATUS_1, ch_stat1.status_1_regval); + return 0; +} + +/** + * snd_intelhad_prog_audio_ctrl_v2 - to initialize audio + * registers and buffer confgiuration registers + * + * @substream:substream for which the prepare function is called + * @intelhaddata:substream private data + * + * This function is called in the prepare callback + */ +int snd_intelhad_prog_audio_ctrl_v2(struct snd_pcm_substream *substream, + struct snd_intelhad *intelhaddata) +{ + union aud_cfg cfg_val = {.cfg_regval = 0}; + union aud_buf_config buf_cfg = {.buf_cfgval = 0}; + u8 channels; + + had_prog_status_reg(substream, intelhaddata); + + buf_cfg.buf_cfg_regx_v2.audio_fifo_watermark = FIFO_THRESHOLD; + buf_cfg.buf_cfg_regx_v2.dma_fifo_watermark = DMA_FIFO_THRESHOLD; + buf_cfg.buf_cfg_regx_v2.aud_delay = 0; + had_write_register(AUD_BUF_CONFIG, buf_cfg.buf_cfgval); + + channels = substream->runtime->channels; + cfg_val.cfg_regx_v2.num_ch = channels - 2; + if (channels <= 2) + cfg_val.cfg_regx_v2.layout = LAYOUT0; + else + cfg_val.cfg_regx_v2.layout = LAYOUT1; + + cfg_val.cfg_regx_v2.val_bit = 1; + had_write_register(AUD_CONFIG, cfg_val.cfg_regval); + return 0; +} + +/** + * snd_intelhad_prog_audio_ctrl_v1 - to initialize audio + * registers and buffer confgiuration registers + * + * @substream:substream for which the prepare function is called + * @intelhaddata:substream private data + * + * This function is called in the prepare callback + */ +int snd_intelhad_prog_audio_ctrl_v1(struct snd_pcm_substream *substream, + struct snd_intelhad *intelhaddata) +{ + union aud_cfg cfg_val = {.cfg_regval = 0}; + union aud_buf_config buf_cfg = {.buf_cfgval = 0}; + u8 channels; + + had_prog_status_reg(substream, intelhaddata); + + buf_cfg.buf_cfg_regx.fifo_width = FIFO_THRESHOLD; + buf_cfg.buf_cfg_regx.aud_delay = 0; + had_write_register(AUD_BUF_CONFIG, buf_cfg.buf_cfgval); + + channels = substream->runtime->channels; + + switch (channels) { + case 1: + case 2: + cfg_val.cfg_regx.num_ch = CH_STEREO; + cfg_val.cfg_regx.layout = LAYOUT0; + break; + + case 3: + case 4: + cfg_val.cfg_regx.num_ch = CH_THREE_FOUR; + cfg_val.cfg_regx.layout = LAYOUT1; + break; + + case 5: + case 6: + cfg_val.cfg_regx.num_ch = CH_FIVE_SIX; + cfg_val.cfg_regx.layout = LAYOUT1; + break; + + case 7: + case 8: + cfg_val.cfg_regx.num_ch = CH_SEVEN_EIGHT; + cfg_val.cfg_regx.layout = LAYOUT1; + break; + + } + + cfg_val.cfg_regx.val_bit = 1; + had_write_register(AUD_CONFIG, cfg_val.cfg_regval); + return 0; +} +/* + * Compute derived values in channel_allocations[]. + */ +static void init_channel_allocations(void) +{ + int i, j; + struct cea_channel_speaker_allocation *p; + + pr_debug("%s: Enter\n", __func__); + + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + p = channel_allocations + i; + p->channels = 0; + p->spk_mask = 0; + for (j = 0; j < ARRAY_SIZE(p->speakers); j++) + if (p->speakers[j]) { + p->channels++; + p->spk_mask |= p->speakers[j]; + } + } +} + +/* + * The transformation takes two steps: + * + * eld->spk_alloc => (eld_speaker_allocation_bits[]) => spk_mask + * spk_mask => (channel_allocations[]) => ai->CA + * + * TODO: it could select the wrong CA from multiple candidates. +*/ +static int snd_intelhad_channel_allocation(struct snd_intelhad *intelhaddata, + int channels) +{ + int i; + int ca = 0; + int spk_mask = 0; + + /* + * CA defaults to 0 for basic stereo audio + */ + if (channels <= 2) + return 0; + + /* + * expand ELD's speaker allocation mask + * + * ELD tells the speaker mask in a compact(paired) form, + * expand ELD's notions to match the ones used by Audio InfoFrame. + */ + + for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { + if (intelhaddata->eeld.speaker_allocation_block & (1 << i)) + spk_mask |= eld_speaker_allocation_bits[i]; + } + + /* search for the first working match in the CA table */ + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if (channels == channel_allocations[i].channels && + (spk_mask & channel_allocations[i].spk_mask) == + channel_allocations[i].spk_mask) { + ca = channel_allocations[i].ca_index; + break; + } + } + + pr_debug("HDMI: select CA 0x%x for %d\n", ca, channels); + + return ca; +} + +/* from speaker bit mask to ALSA API channel position */ +static int spk_to_chmap(int spk) +{ + struct channel_map_table *t = map_tables; + + for (; t->map; t++) { + if (t->spk_mask == spk) + return t->map; + } + return 0; +} + +void had_build_channel_allocation_map(struct snd_intelhad *intelhaddata) +{ + int i = 0, c = 0; + int spk_mask = 0; + struct snd_pcm_chmap_elem *chmap; + uint8_t eld_high, eld_high_mask = 0xF0; + uint8_t high_msb; + + chmap = kzalloc(sizeof(*chmap), GFP_KERNEL); + if (chmap == NULL) { + intelhaddata->chmap->chmap = NULL; + return; + } + + had_get_caps(HAD_GET_ELD, &intelhaddata->eeld); + + pr_debug("eeld.speaker_allocation_block = %x\n", + intelhaddata->eeld.speaker_allocation_block); + + /* WA: Fix the max channel supported to 8 */ + + /* + * Sink may support more than 8 channels, if eld_high has more than + * one bit set. SOC supports max 8 channels. + * Refer eld_speaker_allocation_bits, for sink speaker allocation + */ + + /* if 0x2F < eld < 0x4F fall back to 0x2f, else fall back to 0x4F */ + eld_high = intelhaddata->eeld.speaker_allocation_block & eld_high_mask; + if ((eld_high & (eld_high-1)) && (eld_high > 0x1F)) { + /* eld_high & (eld_high-1): if more than 1 bit set */ + /* 0x1F: 7 channels */ + for (i = 1; i < 4; i++) { + high_msb = eld_high & (0x80 >> i); + if (high_msb) { + intelhaddata->eeld.speaker_allocation_block &= + high_msb | 0xF; + break; + } + } + } + + for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { + if (intelhaddata->eeld.speaker_allocation_block & (1 << i)) + spk_mask |= eld_speaker_allocation_bits[i]; + } + + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if (spk_mask == channel_allocations[i].spk_mask) { + for (c = 0; c < channel_allocations[i].channels; c++) { + chmap->map[c] = spk_to_chmap( + channel_allocations[i].speakers[ + (MAX_SPEAKERS - 1)-c]); + } + chmap->channels = channel_allocations[i].channels; + intelhaddata->chmap->chmap = chmap; + break; + } + } + if (i >= ARRAY_SIZE(channel_allocations)) { + intelhaddata->chmap->chmap = NULL; + kfree(chmap); + } +} + +/* + ** ALSA API channel-map control callbacks + **/ +static int had_chmap_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct snd_intelhad *intelhaddata = info->private_data; + + if (intelhaddata->drv_status == HAD_DRV_DISCONNECTED) + return -ENODEV; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = HAD_MAX_CHANNEL; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SNDRV_CHMAP_LAST; + return 0; +} + +#ifndef USE_ALSA_DEFAULT_TLV +static int had_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct snd_intelhad *intelhaddata = info->private_data; + + if (intelhaddata->drv_status == HAD_DRV_DISCONNECTED) + return -ENODEV; + + /* TODO: Fix for query channel map */ + return -EPERM; +} +#endif + +static int had_chmap_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct snd_intelhad *intelhaddata = info->private_data; + int i = 0; + const struct snd_pcm_chmap_elem *chmap; + + if (intelhaddata->drv_status == HAD_DRV_DISCONNECTED) + return -ENODEV; + if (intelhaddata->chmap->chmap == NULL) + return -ENODATA; + chmap = intelhaddata->chmap->chmap; + for (i = 0; i < chmap->channels; i++) { + ucontrol->value.integer.value[i] = chmap->map[i]; + pr_debug("chmap->map[%d] = %d\n", i, chmap->map[i]); + } + + return 0; +} + +static int had_chmap_ctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + /* TODO: Get channel map and set swap register */ + return -EPERM; +} + +static int had_register_chmap_ctls(struct snd_intelhad *intelhaddata, + struct snd_pcm *pcm) +{ + int err = 0; + + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + NULL, 0, (unsigned long)intelhaddata, + &intelhaddata->chmap); + if (err < 0) + return err; + + intelhaddata->chmap->private_data = intelhaddata; + intelhaddata->kctl = intelhaddata->chmap->kctl; + intelhaddata->kctl->info = had_chmap_ctl_info; + intelhaddata->kctl->get = had_chmap_ctl_get; + intelhaddata->kctl->put = had_chmap_ctl_put; +#ifndef USE_ALSA_DEFAULT_TLV + intelhaddata->kctl->tlv.c = had_chmap_ctl_tlv; +#endif + intelhaddata->chmap->chmap = NULL; + return 0; +} + +/** + * snd_intelhad_prog_dip_v1 - to initialize Data Island Packets registers + * + * @substream:substream for which the prepare function is called + * @intelhaddata:substream private data + * + * This function is called in the prepare callback + */ +static void snd_intelhad_prog_dip_v1(struct snd_pcm_substream *substream, + struct snd_intelhad *intelhaddata) +{ + int i; + union aud_ctrl_st ctrl_state = {.ctrl_val = 0}; + union aud_info_frame2 frame2 = {.fr2_val = 0}; + union aud_info_frame3 frame3 = {.fr3_val = 0}; + u8 checksum = 0; + int channels; + + channels = substream->runtime->channels; + + had_write_register(AUD_CNTL_ST, ctrl_state.ctrl_val); + + frame2.fr2_regx.chnl_cnt = substream->runtime->channels - 1; + + frame3.fr3_regx.chnl_alloc = snd_intelhad_channel_allocation( + intelhaddata, channels); + + /*Calculte the byte wide checksum for all valid DIP words*/ + for (i = 0; i < BYTES_PER_WORD; i++) + checksum += (INFO_FRAME_WORD1 >> i*BITS_PER_BYTE) & MASK_BYTE0; + for (i = 0; i < BYTES_PER_WORD; i++) + checksum += (frame2.fr2_val >> i*BITS_PER_BYTE) & MASK_BYTE0; + for (i = 0; i < BYTES_PER_WORD; i++) + checksum += (frame3.fr3_val >> i*BITS_PER_BYTE) & MASK_BYTE0; + + frame2.fr2_regx.chksum = -(checksum); + + had_write_register(AUD_HDMIW_INFOFR, INFO_FRAME_WORD1); + had_write_register(AUD_HDMIW_INFOFR, frame2.fr2_val); + had_write_register(AUD_HDMIW_INFOFR, frame3.fr3_val); + + /* program remaining DIP words with zero */ + for (i = 0; i < HAD_MAX_DIP_WORDS-VALID_DIP_WORDS; i++) + had_write_register(AUD_HDMIW_INFOFR, 0x0); + + ctrl_state.ctrl_regx.dip_freq = 1; + ctrl_state.ctrl_regx.dip_en_sta = 1; + had_write_register(AUD_CNTL_ST, ctrl_state.ctrl_val); +} + +/** + * snd_intelhad_prog_dip_v2 - to initialize Data Island Packets registers + * + * @substream:substream for which the prepare function is called + * @intelhaddata:substream private data + * + * This function is called in the prepare callback + */ +static void snd_intelhad_prog_dip_v2(struct snd_pcm_substream *substream, + struct snd_intelhad *intelhaddata) +{ + int i; + union aud_ctrl_st ctrl_state = {.ctrl_val = 0}; + union aud_info_frame2 frame2 = {.fr2_val = 0}; + union aud_info_frame3 frame3 = {.fr3_val = 0}; + u8 checksum = 0; + int channels; + + channels = substream->runtime->channels; + + had_write_register(AUD_CNTL_ST, ctrl_state.ctrl_val); + + frame2.fr2_regx.chnl_cnt = substream->runtime->channels - 1; + + frame3.fr3_regx.chnl_alloc = snd_intelhad_channel_allocation( + intelhaddata, channels); + + /*Calculte the byte wide checksum for all valid DIP words*/ + for (i = 0; i < BYTES_PER_WORD; i++) + checksum += (INFO_FRAME_WORD1 >> i*BITS_PER_BYTE) & MASK_BYTE0; + for (i = 0; i < BYTES_PER_WORD; i++) + checksum += (frame2.fr2_val >> i*BITS_PER_BYTE) & MASK_BYTE0; + for (i = 0; i < BYTES_PER_WORD; i++) + checksum += (frame3.fr3_val >> i*BITS_PER_BYTE) & MASK_BYTE0; + + frame2.fr2_regx.chksum = -(checksum); + + had_write_register(AUD_HDMIW_INFOFR_v2, INFO_FRAME_WORD1); + had_write_register(AUD_HDMIW_INFOFR_v2, frame2.fr2_val); + had_write_register(AUD_HDMIW_INFOFR_v2, frame3.fr3_val); + + /* program remaining DIP words with zero */ + for (i = 0; i < HAD_MAX_DIP_WORDS-VALID_DIP_WORDS; i++) + had_write_register(AUD_HDMIW_INFOFR_v2, 0x0); + + ctrl_state.ctrl_regx.dip_freq = 1; + ctrl_state.ctrl_regx.dip_en_sta = 1; + had_write_register(AUD_CNTL_ST, ctrl_state.ctrl_val); +} + +/** + * snd_intelhad_prog_buffer - programs buffer + * address and length registers + * + * @substream:substream for which the prepare function is called + * @intelhaddata:substream private data + * + * This function programs ring buffer address and length into registers. + */ +int snd_intelhad_prog_buffer(struct snd_intelhad *intelhaddata, + int start, int end) +{ + u32 ring_buf_addr, ring_buf_size, period_bytes; + u8 i, num_periods; + struct snd_pcm_substream *substream; + + substream = intelhaddata->stream_info.had_substream; + if (!substream) { + pr_err("substream is NULL\n"); + dump_stack(); + return 0; + } + + ring_buf_addr = substream->runtime->dma_addr; + ring_buf_size = snd_pcm_lib_buffer_bytes(substream); + intelhaddata->stream_info.ring_buf_size = ring_buf_size; + period_bytes = frames_to_bytes(substream->runtime, + substream->runtime->period_size); + num_periods = substream->runtime->periods; + + /* + * buffer addr should be 64 byte aligned, period bytes + * will be used to calculate addr offset + */ + period_bytes &= ~0x3F; + + /* Hardware supports MAX_PERIODS buffers */ + if (end >= HAD_MAX_PERIODS) + return -EINVAL; + + for (i = start; i <= end; i++) { + /* Program the buf registers with addr and len */ + intelhaddata->buf_info[i].buf_addr = ring_buf_addr + + (i * period_bytes); + if (i < num_periods-1) + intelhaddata->buf_info[i].buf_size = period_bytes; + else + intelhaddata->buf_info[i].buf_size = ring_buf_size - + (period_bytes*i); + + had_write_register(AUD_BUF_A_ADDR + (i * HAD_REG_WIDTH), + intelhaddata->buf_info[i].buf_addr | + BIT(0) | BIT(1)); + had_write_register(AUD_BUF_A_LENGTH + (i * HAD_REG_WIDTH), + period_bytes); + intelhaddata->buf_info[i].is_valid = true; + } + pr_debug("%s:buf[%d-%d] addr=%#x and size=%d\n", __func__, start, end, + intelhaddata->buf_info[start].buf_addr, + intelhaddata->buf_info[start].buf_size); + intelhaddata->valid_buf_cnt = num_periods; + return 0; +} + +inline int snd_intelhad_read_len(struct snd_intelhad *intelhaddata) +{ + int i, retval = 0; + u32 len[4]; + + for (i = 0; i < 4 ; i++) { + had_read_register(AUD_BUF_A_LENGTH + (i * HAD_REG_WIDTH), + &len[i]); + if (!len[i]) + retval++; + } + if (retval != 1) { + for (i = 0; i < 4 ; i++) + pr_debug("buf[%d] size=%d\n", i, len[i]); + } + + return retval; +} + +/** + * snd_intelhad_prog_cts_v1 - Program HDMI audio CTS value + * + * @aud_samp_freq: sampling frequency of audio data + * @tmds: sampling frequency of the display data + * @n_param: N value, depends on aud_samp_freq + * @intelhaddata:substream private data + * + * Program CTS register based on the audio and display sampling frequency + */ +static void snd_intelhad_prog_cts_v1(u32 aud_samp_freq, u32 tmds, u32 n_param, + struct snd_intelhad *intelhaddata) +{ + u32 cts_val; + u64 dividend, divisor; + + /* Calculate CTS according to HDMI 1.3a spec*/ + dividend = (u64)tmds * n_param*1000; + divisor = 128 * aud_samp_freq; + cts_val = div64_u64(dividend, divisor); + pr_debug("TMDS value=%d, N value=%d, CTS Value=%d\n", + tmds, n_param, cts_val); + had_write_register(AUD_HDMI_CTS, (BIT(20) | cts_val)); +} + +/** + * snd_intelhad_prog_cts_v2 - Program HDMI audio CTS value + * + * @aud_samp_freq: sampling frequency of audio data + * @tmds: sampling frequency of the display data + * @n_param: N value, depends on aud_samp_freq + * @intelhaddata:substream private data + * + * Program CTS register based on the audio and display sampling frequency + */ +static void snd_intelhad_prog_cts_v2(u32 aud_samp_freq, u32 tmds, u32 n_param, + struct snd_intelhad *intelhaddata) +{ + u32 cts_val; + u64 dividend, divisor; + + /* Calculate CTS according to HDMI 1.3a spec*/ + dividend = (u64)tmds * n_param*1000; + divisor = 128 * aud_samp_freq; + cts_val = div64_u64(dividend, divisor); + pr_debug("TMDS value=%d, N value=%d, CTS Value=%d\n", + tmds, n_param, cts_val); + had_write_register(AUD_HDMI_CTS, (BIT(24) | cts_val)); +} + +static int had_calculate_n_value(u32 aud_samp_freq) +{ + s32 n_val; + + /* Select N according to HDMI 1.3a spec*/ + switch (aud_samp_freq) { + case AUD_SAMPLE_RATE_32: + n_val = 4096; + break; + + case AUD_SAMPLE_RATE_44_1: + n_val = 6272; + break; + + case AUD_SAMPLE_RATE_48: + n_val = 6144; + break; + + case AUD_SAMPLE_RATE_88_2: + n_val = 12544; + break; + + case AUD_SAMPLE_RATE_96: + n_val = 12288; + break; + + case AUD_SAMPLE_RATE_176_4: + n_val = 25088; + break; + + case HAD_MAX_RATE: + n_val = 24576; + break; + + default: + n_val = -EINVAL; + break; + } + return n_val; +} + +/** + * snd_intelhad_prog_n_v1 - Program HDMI audio N value + * + * @aud_samp_freq: sampling frequency of audio data + * @n_param: N value, depends on aud_samp_freq + * @intelhaddata:substream private data + * + * This function is called in the prepare callback. + * It programs based on the audio and display sampling frequency + */ +static int snd_intelhad_prog_n_v1(u32 aud_samp_freq, u32 *n_param, + struct snd_intelhad *intelhaddata) +{ + s32 n_val; + + n_val = had_calculate_n_value(aud_samp_freq); + + if (n_val < 0) + return n_val; + + had_write_register(AUD_N_ENABLE, (BIT(20) | n_val)); + *n_param = n_val; + return 0; +} + +/** + * snd_intelhad_prog_n_v2 - Program HDMI audio N value + * + * @aud_samp_freq: sampling frequency of audio data + * @n_param: N value, depends on aud_samp_freq + * @intelhaddata:substream private data + * + * This function is called in the prepare callback. + * It programs based on the audio and display sampling frequency + */ +static int snd_intelhad_prog_n_v2(u32 aud_samp_freq, u32 *n_param, + struct snd_intelhad *intelhaddata) +{ + s32 n_val; + + n_val = had_calculate_n_value(aud_samp_freq); + + if (n_val < 0) + return n_val; + + had_write_register(AUD_N_ENABLE, (BIT(24) | n_val)); + *n_param = n_val; + return 0; +} + +static void had_clear_underrun_intr_v1(struct snd_intelhad *intelhaddata) +{ + u32 hdmi_status, i = 0; + + /* Handle Underrun interrupt within Audio Unit */ + had_write_register(AUD_CONFIG, 0); + /* Reset buffer pointers */ + had_write_register(AUD_HDMI_STATUS, 1); + had_write_register(AUD_HDMI_STATUS, 0); + /** + * The interrupt status 'sticky' bits might not be cleared by + * setting '1' to that bit once... + */ + do { /* clear bit30, 31 AUD_HDMI_STATUS */ + had_read_register(AUD_HDMI_STATUS, &hdmi_status); + pr_debug("HDMI status =0x%x\n", hdmi_status); + if (hdmi_status & AUD_CONFIG_MASK_UNDERRUN) { + i++; + hdmi_status &= (AUD_CONFIG_MASK_SRDBG | + AUD_CONFIG_MASK_FUNCRST); + hdmi_status |= ~AUD_CONFIG_MASK_UNDERRUN; + had_write_register(AUD_HDMI_STATUS, hdmi_status); + } else + break; + } while (i < MAX_CNT); + if (i >= MAX_CNT) + pr_err("Unable to clear UNDERRUN bits\n"); +} + +static void had_clear_underrun_intr_v2(struct snd_intelhad *intelhaddata) +{ + u32 hdmi_status, i = 0; + + /* Handle Underrun interrupt within Audio Unit */ + had_write_register(AUD_CONFIG, 0); + /* Reset buffer pointers */ + had_write_register(AUD_HDMI_STATUS_v2, 1); + had_write_register(AUD_HDMI_STATUS_v2, 0); + /** + * The interrupt status 'sticky' bits might not be cleared by + * setting '1' to that bit once... + */ + do { /* clear bit30, 31 AUD_HDMI_STATUS */ + had_read_register(AUD_HDMI_STATUS_v2, &hdmi_status); + pr_debug("HDMI status =0x%x\n", hdmi_status); + if (hdmi_status & AUD_CONFIG_MASK_UNDERRUN) { + i++; + had_write_register(AUD_HDMI_STATUS_v2, hdmi_status); + } else + break; + } while (i < MAX_CNT); + if (i >= MAX_CNT) + pr_err("Unable to clear UNDERRUN bits\n"); +} + +/** +* snd_intelhad_open - stream initializations are done here +* @substream:substream for which the stream function is called +* +* This function is called whenever a PCM stream is opened +*/ +static int snd_intelhad_open(struct snd_pcm_substream *substream) +{ + struct snd_intelhad *intelhaddata; + struct snd_pcm_runtime *runtime; + struct had_stream_pvt *stream; + struct had_pvt_data *had_stream; + int retval; + + pr_debug("snd_intelhad_open called\n"); + intelhaddata = snd_pcm_substream_chip(substream); + had_stream = intelhaddata->private_data; + runtime = substream->runtime; + + pm_runtime_get(intelhaddata->dev); + + if (had_get_hwstate(intelhaddata)) { + pr_err("%s: HDMI cable plugged-out\n", __func__); + retval = -ENODEV; + goto exit_put_handle; + } + + /* Check, if device already in use */ + if (runtime->private_data) { + pr_err("Device already in use\n"); + retval = -EBUSY; + goto exit_put_handle; + } + + /* set the runtime hw parameter with local snd_pcm_hardware struct */ + runtime->hw = snd_intel_hadstream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) { + retval = -ENOMEM; + goto exit_put_handle; + } + stream->stream_status = STREAM_INIT; + runtime->private_data = stream; + + retval = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (retval < 0) + goto exit_err; + + /* Make sure, that the period size is always aligned + * 64byte boundary + */ + retval = snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64); + if (retval < 0) { + pr_err("%s:step_size=64 failed,err=%d\n", __func__, retval); + goto exit_err; + } + + return retval; +exit_err: + kfree(stream); +exit_put_handle: + pm_runtime_put(intelhaddata->dev); + runtime->private_data = NULL; + return retval; +} + +/** +* had_period_elapsed - updates the hardware pointer status +* @had_substream:substream for which the stream function is called +* +*/ +static void had_period_elapsed(void *had_substream) +{ + struct snd_pcm_substream *substream = had_substream; + struct had_stream_pvt *stream; + + /* pr_debug("had_period_elapsed called\n"); */ + + if (!substream || !substream->runtime) + return; + stream = substream->runtime->private_data; + if (!stream) + return; + + if (stream->stream_status != STREAM_RUNNING) + return; + snd_pcm_period_elapsed(substream); +} + +/** +* snd_intelhad_init_stream - internal function to initialize stream info +* @substream:substream for which the stream function is called +* +*/ +static int snd_intelhad_init_stream(struct snd_pcm_substream *substream) +{ + struct snd_intelhad *intelhaddata = snd_pcm_substream_chip(substream); + + pr_debug("snd_intelhad_init_stream called\n"); + + pr_debug("setting buffer ptr param\n"); + intelhaddata->stream_info.period_elapsed = had_period_elapsed; + intelhaddata->stream_info.had_substream = substream; + intelhaddata->stream_info.buffer_ptr = 0; + intelhaddata->stream_info.buffer_rendered = 0; + intelhaddata->stream_info.sfreq = substream->runtime->rate; + return 0; +} + +/** + * snd_intelhad_close- to free parameteres when stream is stopped + * + * @substream: substream for which the function is called + * + * This function is called by ALSA framework when stream is stopped + */ +static int snd_intelhad_close(struct snd_pcm_substream *substream) +{ + struct snd_intelhad *intelhaddata; + struct snd_pcm_runtime *runtime; + + pr_debug("snd_intelhad_close called\n"); + + intelhaddata = snd_pcm_substream_chip(substream); + runtime = substream->runtime; + + if (!runtime->private_data) { + pr_debug("close() might have called after failed open"); + return 0; + } + + intelhaddata->stream_info.buffer_rendered = 0; + intelhaddata->stream_info.buffer_ptr = 0; + intelhaddata->stream_info.str_id = 0; + intelhaddata->stream_info.had_substream = NULL; + + /* Check if following drv_status modification is required - VA */ + if (intelhaddata->drv_status != HAD_DRV_DISCONNECTED) { + intelhaddata->drv_status = HAD_DRV_CONNECTED; + pr_debug("%s @ %d:DEBUG PLUG/UNPLUG : HAD_DRV_CONNECTED\n", + __func__, __LINE__); + } + kfree(runtime->private_data); + runtime->private_data = NULL; + pm_runtime_put(intelhaddata->dev); + return 0; +} + +/** + * snd_intelhad_hw_params- to setup the hardware parameters + * like allocating the buffers + * + * @substream: substream for which the function is called + * @hw_params: hardware parameters + * + * This function is called by ALSA framework when hardware params are set + */ +static int snd_intelhad_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + unsigned long addr; + int pages, buf_size, retval; + + pr_debug("snd_intelhad_hw_params called\n"); + + if (!hw_params) + return -EINVAL; + + buf_size = params_buffer_bytes(hw_params); + retval = snd_pcm_lib_malloc_pages(substream, buf_size); + if (retval < 0) + return retval; + pr_debug("%s:allocated memory = %d\n", __func__, buf_size); + /* mark the pages as uncached region */ + addr = (unsigned long) substream->runtime->dma_area; + pages = (substream->runtime->dma_bytes + PAGE_SIZE - 1) / PAGE_SIZE; + retval = set_memory_uc(addr, pages); + if (retval) { + pr_err("set_memory_uc failed.Error:%d\n", retval); + return retval; + } + memset(substream->runtime->dma_area, 0, buf_size); + + return retval; +} + +/** + * snd_intelhad_hw_free- to release the resources allocated during + * hardware params setup + * + * @substream: substream for which the function is called + * + * This function is called by ALSA framework before close callback. + * + */ +static int snd_intelhad_hw_free(struct snd_pcm_substream *substream) +{ + unsigned long addr; + u32 pages; + + pr_debug("snd_intelhad_hw_free called\n"); + + /* mark back the pages as cached/writeback region before the free */ + if (substream->runtime->dma_area != NULL) { + addr = (unsigned long) substream->runtime->dma_area; + pages = (substream->runtime->dma_bytes + PAGE_SIZE - 1) / + PAGE_SIZE; + set_memory_wb(addr, pages); + return snd_pcm_lib_free_pages(substream); + } + return 0; +} + +/** +* snd_intelhad_pcm_trigger - stream activities are handled here +* @substream:substream for which the stream function is called +* @cmd:the stream commamd thats requested from upper layer +* This function is called whenever an a stream activity is invoked +*/ +static int snd_intelhad_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + int caps, retval = 0; + unsigned long flag_irq; + struct snd_intelhad *intelhaddata; + struct had_stream_pvt *stream; + struct had_pvt_data *had_stream; + + pr_debug("snd_intelhad_pcm_trigger called\n"); + + intelhaddata = snd_pcm_substream_chip(substream); + stream = substream->runtime->private_data; + had_stream = intelhaddata->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + pr_debug("Trigger Start\n"); + + /* Disable local INTRs till register prgmng is done */ + if (had_get_hwstate(intelhaddata)) { + pr_err("_START: HDMI cable plugged-out\n"); + retval = -ENODEV; + break; + } + stream->stream_status = STREAM_RUNNING; + + spin_lock_irqsave(&intelhaddata->had_spinlock, flag_irq); + had_stream->stream_type = HAD_RUNNING_STREAM; + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irq); + + /* Enable Audio */ + /* + * ToDo: Need to enable UNDERRUN interrupts as well + * caps = HDMI_AUDIO_UNDERRUN | HDMI_AUDIO_BUFFER_DONE; + */ + caps = HDMI_AUDIO_BUFFER_DONE; + retval = had_set_caps(HAD_SET_ENABLE_AUDIO_INT, &caps); + retval = had_set_caps(HAD_SET_ENABLE_AUDIO, NULL); + intelhaddata->ops->enable_audio(substream, 1); + + pr_debug("Processed _Start\n"); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + pr_debug("Trigger Stop\n"); + spin_lock_irqsave(&intelhaddata->had_spinlock, flag_irq); + intelhaddata->stream_info.str_id = 0; + intelhaddata->curr_buf = 0; + + /* Stop reporting BUFFER_DONE/UNDERRUN to above layers*/ + + had_stream->stream_type = HAD_INIT; + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irq); + /* Disable Audio */ + /* + * ToDo: Need to disable UNDERRUN interrupts as well + * caps = HDMI_AUDIO_UNDERRUN | HDMI_AUDIO_BUFFER_DONE; + */ + caps = HDMI_AUDIO_BUFFER_DONE; + had_set_caps(HAD_SET_DISABLE_AUDIO_INT, &caps); + intelhaddata->ops->enable_audio(substream, 0); + /* Reset buffer pointers */ + intelhaddata->ops->reset_audio(1); + intelhaddata->ops->reset_audio(0); + stream->stream_status = STREAM_DROPPED; + had_set_caps(HAD_SET_DISABLE_AUDIO, NULL); + break; + + default: + retval = -EINVAL; + } + return retval; +} + +/** +* snd_intelhad_pcm_prepare- internal preparation before starting a stream +* +* @substream: substream for which the function is called +* +* This function is called when a stream is started for internal preparation. +*/ +static int snd_intelhad_pcm_prepare(struct snd_pcm_substream *substream) +{ + int retval; + u32 disp_samp_freq, n_param; + struct snd_intelhad *intelhaddata; + struct snd_pcm_runtime *runtime; + struct had_pvt_data *had_stream; + + pr_debug("snd_intelhad_pcm_prepare called\n"); + + intelhaddata = snd_pcm_substream_chip(substream); + runtime = substream->runtime; + had_stream = intelhaddata->private_data; + + if (had_get_hwstate(intelhaddata)) { + pr_err("%s: HDMI cable plugged-out\n", __func__); + snd_pcm_stop(substream, SNDRV_PCM_STATE_DISCONNECTED); + retval = -ENODEV; + goto prep_end; + } + + pr_debug("period_size=%d\n", + (int)frames_to_bytes(runtime, runtime->period_size)); + pr_debug("periods=%d\n", runtime->periods); + pr_debug("buffer_size=%d\n", (int)snd_pcm_lib_buffer_bytes(substream)); + pr_debug("rate=%d\n", runtime->rate); + pr_debug("channels=%d\n", runtime->channels); + + if (intelhaddata->stream_info.str_id) { + pr_debug("_prepare is called for existing str_id#%d\n", + intelhaddata->stream_info.str_id); + retval = snd_intelhad_pcm_trigger(substream, + SNDRV_PCM_TRIGGER_STOP); + return retval; + } + + retval = snd_intelhad_init_stream(substream); + if (retval) + goto prep_end; + + + /* Get N value in KHz */ + retval = had_get_caps(HAD_GET_DISPLAY_RATE, &disp_samp_freq); + if (retval) { + pr_err("querying display sampling freq failed %#x\n", retval); + goto prep_end; + } + + had_get_caps(HAD_GET_ELD, &intelhaddata->eeld); + + retval = intelhaddata->ops->prog_n(substream->runtime->rate, &n_param, + intelhaddata); + if (retval) { + pr_err("programming N value failed %#x\n", retval); + goto prep_end; + } + intelhaddata->ops->prog_cts(substream->runtime->rate, + disp_samp_freq, n_param, intelhaddata); + + intelhaddata->ops->prog_dip(substream, intelhaddata); + + retval = intelhaddata->ops->audio_ctrl(substream, intelhaddata); + + /* Prog buffer address */ + retval = snd_intelhad_prog_buffer(intelhaddata, + HAD_BUF_TYPE_A, HAD_BUF_TYPE_D); + + /* + * Program channel mapping in following order: + * FL, FR, C, LFE, RL, RR + */ + + had_write_register(AUD_BUF_CH_SWAP, SWAP_LFE_CENTER); + +prep_end: + return retval; +} + +/** + * snd_intelhad_pcm_pointer- to send the current buffer pointerprocessed by hw + * + * @substream: substream for which the function is called + * + * This function is called by ALSA framework to get the current hw buffer ptr + * when a period is elapsed + */ +static snd_pcm_uframes_t snd_intelhad_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_intelhad *intelhaddata; + u32 bytes_rendered = 0; + + /* pr_debug("snd_intelhad_pcm_pointer called\n"); */ + + intelhaddata = snd_pcm_substream_chip(substream); + + if (intelhaddata->flag_underrun) { + intelhaddata->flag_underrun = 0; + return SNDRV_PCM_POS_XRUN; + } + + if (intelhaddata->stream_info.buffer_rendered) + div_u64_rem(intelhaddata->stream_info.buffer_rendered, + intelhaddata->stream_info.ring_buf_size, + &(bytes_rendered)); + + intelhaddata->stream_info.buffer_ptr = bytes_to_frames( + substream->runtime, + bytes_rendered); + return intelhaddata->stream_info.buffer_ptr; +} + +/** +* snd_intelhad_pcm_mmap- mmaps a kernel buffer to user space for copying data +* +* @substream: substream for which the function is called +* @vma: struct instance of memory VMM memory area +* +* This function is called by OS when a user space component +* tries to get mmap memory from driver +*/ +static int snd_intelhad_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + + pr_debug("snd_intelhad_pcm_mmap called\n"); + + pr_debug("entry with prot:%s\n", __func__); + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +int hdmi_audio_mode_change(struct snd_pcm_substream *substream) +{ + int retval = 0; + u32 disp_samp_freq, n_param; + struct snd_intelhad *intelhaddata; + + intelhaddata = snd_pcm_substream_chip(substream); + + /* Disable Audio */ + intelhaddata->ops->enable_audio(substream, 0); + + /* Update CTS value */ + retval = had_get_caps(HAD_GET_DISPLAY_RATE, &disp_samp_freq); + if (retval) { + pr_err("querying display sampling freq failed %#x\n", retval); + goto out; + } + + retval = intelhaddata->ops->prog_n(substream->runtime->rate, &n_param, + intelhaddata); + if (retval) { + pr_err("programming N value failed %#x\n", retval); + goto out; + } + intelhaddata->ops->prog_cts(substream->runtime->rate, + disp_samp_freq, n_param, intelhaddata); + + /* Enable Audio */ + intelhaddata->ops->enable_audio(substream, 1); + +out: + return retval; +} + +/*PCM operations structure and the calls back for the same */ +struct snd_pcm_ops snd_intelhad_playback_ops = { + .open = snd_intelhad_open, + .close = snd_intelhad_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_intelhad_hw_params, + .hw_free = snd_intelhad_hw_free, + .prepare = snd_intelhad_pcm_prepare, + .trigger = snd_intelhad_pcm_trigger, + .pointer = snd_intelhad_pcm_pointer, + .mmap = snd_intelhad_pcm_mmap, +}; + +/** + * snd_intelhad_create - to crete alsa card instance + * + * @intelhaddata: pointer to internal context + * @card: pointer to card + * + * This function is called when the hdmi cable is plugged in + */ +static int snd_intelhad_create( + struct snd_intelhad *intelhaddata, + struct snd_card *card) +{ + int retval; + static struct snd_device_ops ops = { + }; + + pr_debug("snd_intelhad_create called\n"); + + if (!intelhaddata) + return -EINVAL; + + /* ALSA api to register the device */ + retval = snd_device_new(card, SNDRV_DEV_LOWLEVEL, intelhaddata, &ops); + return retval; +} +/** + * snd_intelhad_pcm_free - to free the memory allocated + * + * @pcm: pointer to pcm instance + * This function is called when the device is removed + */ +static void snd_intelhad_pcm_free(struct snd_pcm *pcm) +{ + pr_debug("Freeing PCM preallocated pages\n"); + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int had_iec958_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static int had_iec958_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_intelhad *intelhaddata = snd_kcontrol_chip(kcontrol); + + ucontrol->value.iec958.status[0] = (intelhaddata->aes_bits >> 0) & 0xff; + ucontrol->value.iec958.status[1] = (intelhaddata->aes_bits >> 8) & 0xff; + ucontrol->value.iec958.status[2] = + (intelhaddata->aes_bits >> 16) & 0xff; + ucontrol->value.iec958.status[3] = + (intelhaddata->aes_bits >> 24) & 0xff; + return 0; +} +static int had_iec958_mask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.iec958.status[0] = 0xff; + ucontrol->value.iec958.status[1] = 0xff; + ucontrol->value.iec958.status[2] = 0xff; + ucontrol->value.iec958.status[3] = 0xff; + return 0; +} +static int had_iec958_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned int val; + struct snd_intelhad *intelhaddata = snd_kcontrol_chip(kcontrol); + + pr_debug("entered had_iec958_put\n"); + val = (ucontrol->value.iec958.status[0] << 0) | + (ucontrol->value.iec958.status[1] << 8) | + (ucontrol->value.iec958.status[2] << 16) | + (ucontrol->value.iec958.status[3] << 24); + if (intelhaddata->aes_bits != val) { + intelhaddata->aes_bits = val; + return 1; + } + return 1; +} + +static struct snd_kcontrol_new had_control_iec958_mask = { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, MASK), + .info = had_iec958_info, /* shared */ + .get = had_iec958_mask_get, +}; + +static struct snd_kcontrol_new had_control_iec958 = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = had_iec958_info, + .get = had_iec958_get, + .put = had_iec958_put +}; + +static struct snd_intel_had_interface had_interface = { + .name = "hdmi-audio", + .query = hdmi_audio_query, + .suspend = hdmi_audio_suspend, + .resume = hdmi_audio_resume, +}; + +static struct had_ops had_ops_v1 = { + .enable_audio = snd_intelhad_enable_audio_v1, + .reset_audio = snd_intelhad_reset_audio_v1, + .prog_n = snd_intelhad_prog_n_v1, + .prog_cts = snd_intelhad_prog_cts_v1, + .audio_ctrl = snd_intelhad_prog_audio_ctrl_v1, + .prog_dip = snd_intelhad_prog_dip_v1, + .handle_underrun = had_clear_underrun_intr_v1, +}; + +static struct had_ops had_ops_v2 = { + .enable_audio = snd_intelhad_enable_audio_v2, + .reset_audio = snd_intelhad_reset_audio_v2, + .prog_n = snd_intelhad_prog_n_v2, + .prog_cts = snd_intelhad_prog_cts_v2, + .audio_ctrl = snd_intelhad_prog_audio_ctrl_v2, + .prog_dip = snd_intelhad_prog_dip_v2, + .handle_underrun = had_clear_underrun_intr_v2, +}; +/** + * hdmi_audio_probe - to create sound card instance for HDMI audio playabck + * + *@haddata: pointer to HAD private data + *@card_id: card for which probe is called + * + * This function is called when the hdmi cable is plugged in. This function + * creates and registers the sound card with ALSA + */ +int hdmi_audio_probe(void *deviceptr) +{ + int retval; + struct snd_pcm *pcm; + struct snd_card *card; + struct had_callback_ops ops_cb; + struct snd_intelhad *intelhaddata; + struct had_pvt_data *had_stream; + struct platform_device *devptr = deviceptr; + + pr_debug("Enter %s\n", __func__); + + pr_debug("hdmi_audio_probe dma_mask: %p\n", devptr->dev.dma_mask); + + /* allocate memory for saving internal context and working */ + intelhaddata = kzalloc(sizeof(*intelhaddata), GFP_KERNEL); + if (!intelhaddata) + return -ENOMEM; + + had_stream = kzalloc(sizeof(*had_stream), GFP_KERNEL); + if (!had_stream) { + retval = -ENOMEM; + goto free_haddata; + } + + had_data = intelhaddata; + ops_cb.intel_had_event_call_back = had_event_handler; + + /* registering with display driver to get access to display APIs */ + + retval = mid_hdmi_audio_setup( + ops_cb.intel_had_event_call_back, + &(intelhaddata->reg_ops), + &(intelhaddata->query_ops)); + if (retval) { + pr_err("querying display driver APIs failed %#x\n", retval); + goto free_hadstream; + } + mutex_lock(&had_mutex); + spin_lock_init(&intelhaddata->had_spinlock); + intelhaddata->drv_status = HAD_DRV_DISCONNECTED; + pr_debug("%s @ %d:DEBUG PLUG/UNPLUG : HAD_DRV_DISCONNECTED\n", + __func__, __LINE__); + + /* create a card instance with ALSA framework */ + retval = snd_card_new(&devptr->dev, hdmi_card_index, hdmi_card_id, + THIS_MODULE, 0, &card); + + if (retval) + goto unlock_mutex; + intelhaddata->card = card; + intelhaddata->card_id = hdmi_card_id; + intelhaddata->card_index = card->number; + intelhaddata->private_data = had_stream; + intelhaddata->flag_underrun = 0; + intelhaddata->aes_bits = SNDRV_PCM_DEFAULT_CON_SPDIF; + strncpy(card->driver, INTEL_HAD, strlen(INTEL_HAD)); + strncpy(card->shortname, INTEL_HAD, strlen(INTEL_HAD)); + + retval = snd_pcm_new(card, INTEL_HAD, PCM_INDEX, MAX_PB_STREAMS, + MAX_CAP_STREAMS, &pcm); + if (retval) + goto err; + + /* setup private data which can be retrieved when required */ + pcm->private_data = intelhaddata; + pcm->private_free = snd_intelhad_pcm_free; + pcm->info_flags = 0; + strncpy(pcm->name, card->shortname, strlen(card->shortname)); + /* setup the ops for palyabck */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_intelhad_playback_ops); + /* allocate dma pages for ALSA stream operations + * memory allocated is based on size, not max value + * thus using same argument for max & size + */ + retval = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_DEV, NULL, + HAD_MAX_BUFFER, HAD_MAX_BUFFER); + + if (card->dev == NULL) + pr_debug("card->dev is NULL!!!!! Should not be this case\n"); + else if (card->dev->dma_mask == NULL) + pr_debug("hdmi_audio_probe dma_mask is NULL!!!!!\n"); + else + pr_debug("hdmi_audio_probe dma_mask is : %p\n", + card->dev->dma_mask); + + if (retval) + goto err; + + /* internal function call to register device with ALSA */ + retval = snd_intelhad_create(intelhaddata, card); + if (retval) + goto err; + + card->private_data = &intelhaddata; + retval = snd_card_register(card); + if (retval) + goto err; + + /* IEC958 controls */ + retval = snd_ctl_add(card, snd_ctl_new1(&had_control_iec958_mask, + intelhaddata)); + if (retval < 0) + goto err; + retval = snd_ctl_add(card, snd_ctl_new1(&had_control_iec958, + intelhaddata)); + if (retval < 0) + goto err; + + init_channel_allocations(); + + /* Register channel map controls */ + retval = had_register_chmap_ctls(intelhaddata, pcm); + if (retval < 0) + goto err; + + intelhaddata->dev = &devptr->dev; + pm_runtime_set_active(intelhaddata->dev); + pm_runtime_enable(intelhaddata->dev); + + mutex_unlock(&had_mutex); + retval = mid_hdmi_audio_register(&had_interface, intelhaddata); + if (retval) { + pr_err("registering with display driver failed %#x\n", retval); + snd_card_free(card); + goto free_hadstream; + } + + intelhaddata->hw_silence = 1; + had_ops_v1 = had_ops_v1; /* unused */ + intelhaddata->ops = &had_ops_v2; + + return retval; +err: + snd_card_free(card); +unlock_mutex: + mutex_unlock(&had_mutex); +free_hadstream: + kfree(had_stream); + pm_runtime_disable(intelhaddata->dev); + intelhaddata->dev = NULL; +free_haddata: + kfree(intelhaddata); + intelhaddata = NULL; + pr_err("Error returned from %s api %#x\n", __func__, retval); + return retval; +} + +/** + * hdmi_audio_remove - removes the alsa card + * + *@haddata: pointer to HAD private data + * + * This function is called when the hdmi cable is un-plugged. This function + * free the sound card. + */ +int hdmi_audio_remove(void *pdevptr) +{ + struct snd_intelhad *intelhaddata = had_data; + int caps; + + pr_debug("Enter %s\n", __func__); + + if (!intelhaddata) + return 0; + + if (intelhaddata->drv_status != HAD_DRV_DISCONNECTED) { + caps = HDMI_AUDIO_UNDERRUN | HDMI_AUDIO_BUFFER_DONE; + had_set_caps(HAD_SET_DISABLE_AUDIO_INT, &caps); + had_set_caps(HAD_SET_DISABLE_AUDIO, NULL); + } + snd_card_free(intelhaddata->card); + kfree(intelhaddata->private_data); + kfree(intelhaddata); + return 0; +} + +MODULE_AUTHOR("Sailaja Bandarupalli <sailaja.bandarupalli(a)intel.com>"); +MODULE_AUTHOR("Ramesh Babu K V <ramesh.babu(a)intel.com>"); +MODULE_AUTHOR("Vaibhav Agarwal <vaibhav.agarwal(a)intel.com>"); +MODULE_AUTHOR("Jerome Anand <jerome.anand(a)intel.com>"); +MODULE_DESCRIPTION("Intel HDMI Audio driver"); +MODULE_LICENSE("GPL v2"); +MODULE_SUPPORTED_DEVICE("{Intel,Intel_HAD}"); +MODULE_VERSION(HAD_DRIVER_VERSION); diff --git a/sound/x86/intel_hdmi_audio.h b/sound/x86/intel_hdmi_audio.h new file mode 100644 index 0000000..1ef25b6 --- /dev/null +++ b/sound/x86/intel_hdmi_audio.h @@ -0,0 +1,201 @@ +/* + * Copyright (C) 2016 Intel Corporation + * Authors: Sailaja Bandarupalli <sailaja.bandarupalli(a)intel.com> + * Ramesh Babu K V <ramesh.babu(a)intel.com> + * Vaibhav Agarwal <vaibhav.agarwal(a)intel.com> + * Jerome Anand <jerome.anand(a)intel.com> + * + * Permission is hereby granted, free of charge, to any person obtaining + * a copy of this software and associated documentation files + * (the "Software"), to deal in the Software without restriction, + * including without limitation the rights to use, copy, modify, merge, + * publish, distribute, sublicense, and/or sell copies of the Software, + * and to permit persons to whom the Software is furnished to do so, + * subject to the following conditions: + * + * The above copyright notice and this permission notice (including the + * next paragraph) shall be included in all copies or substantial + * portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND + * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS + * BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN + * ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN + * CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE + * SOFTWARE. + */ + +#ifndef _INTEL_HDMI_AUDIO_H_ +#define _INTEL_HDMI_AUDIO_H_ + +#include <linux/types.h> +#include <sound/initval.h> +#include <linux/version.h> +#include <linux/pm_runtime.h> +#include <sound/asoundef.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include "intel_hdmi_lpe_audio.h" + +#define PCM_INDEX 0 +#define MAX_PB_STREAMS 1 +#define MAX_CAP_STREAMS 0 +#define HDMI_AUDIO_DRIVER "hdmi-audio" + +#define INFO_FRAME_WORD1 0x000a0184 +#define FIFO_THRESHOLD 0xFE +#define DMA_FIFO_THRESHOLD 0x7 +#define BYTES_PER_WORD 0x4 + +/* Sampling rate as per IEC60958 Ver 3 */ +#define CH_STATUS_MAP_32KHZ 0x3 +#define CH_STATUS_MAP_44KHZ 0x0 +#define CH_STATUS_MAP_48KHZ 0x2 +#define CH_STATUS_MAP_88KHZ 0x8 +#define CH_STATUS_MAP_96KHZ 0xA +#define CH_STATUS_MAP_176KHZ 0xC +#define CH_STATUS_MAP_192KHZ 0xE + +#define MAX_SMPL_WIDTH_20 0x0 +#define MAX_SMPL_WIDTH_24 0x1 +#define SMPL_WIDTH_16BITS 0x1 +#define SMPL_WIDTH_24BITS 0x5 +#define CHANNEL_ALLOCATION 0x1F +#define MASK_BYTE0 0x000000FF +#define VALID_DIP_WORDS 3 +#define LAYOUT0 0 +#define LAYOUT1 1 +#define SWAP_LFE_CENTER 0x00fac4c8 +#define AUD_CONFIG_CH_MASK_V2 0x70 + +struct pcm_stream_info { + int str_id; + void *had_substream; + void (*period_elapsed)(void *had_substream); + u32 buffer_ptr; + u64 buffer_rendered; + u32 ring_buf_size; + int sfreq; +}; + +struct ring_buf_info { + uint32_t buf_addr; + uint32_t buf_size; + uint8_t is_valid; +}; + +struct had_stream_pvt { + enum had_stream_status stream_status; + int stream_ops; + ssize_t dbg_cum_bytes; +}; + +struct had_pvt_data { + enum had_status_stream stream_type; +}; + +struct had_callback_ops { + had_event_call_back intel_had_event_call_back; +}; + +/** + * struct snd_intelhad - intelhad driver structure + * + * @card: ptr to hold card details + * @card_index: sound card index + * @card_id: detected sound card id + * @reg_ops: register operations to program registers + * @query_ops: caps call backs for get/set operations + * @drv_status: driver status + * @buf_info: ring buffer info + * @stream_info: stream information + * @eeld: holds EELD info + * @curr_buf: pointer to hold current active ring buf + * @valid_buf_cnt: ring buffer count for stream + * @had_spinlock: driver lock + * @aes_bits: IEC958 status bits + * @buff_done: id of current buffer done intr + * @dev: platoform device handle + * @kctl: holds kctl ptrs used for channel map + * @chmap: holds channel map info + * @audio_reg_base: hdmi audio register base offset + * @hw_silence: flag indicates SoC support for HW silence/Keep alive + * @ops: holds ops functions based on platform + */ +struct snd_intelhad { + struct snd_card *card; + int card_index; + char *card_id; + struct hdmi_audio_registers_ops reg_ops; + struct hdmi_audio_query_set_ops query_ops; + enum had_drv_status drv_status; + struct ring_buf_info buf_info[HAD_NUM_OF_RING_BUFS]; + struct pcm_stream_info stream_info; + union otm_hdmi_eld_t eeld; + enum intel_had_aud_buf_type curr_buf; + int valid_buf_cnt; + unsigned int aes_bits; + int flag_underrun; + struct had_pvt_data *private_data; + spinlock_t had_spinlock; + enum intel_had_aud_buf_type buff_done; + struct device *dev; + struct snd_kcontrol *kctl; + struct snd_pcm_chmap *chmap; + unsigned int *audio_reg_base; + unsigned int audio_cfg_offset; + bool hw_silence; + struct had_ops *ops; +}; + +struct had_ops { + void (*enable_audio)(struct snd_pcm_substream *substream, + u8 enable); + void (*reset_audio)(u8 reset); + int (*prog_n)(u32 aud_samp_freq, u32 *n_param, + struct snd_intelhad *intelhaddata); + void (*prog_cts)(u32 aud_samp_freq, u32 tmds, u32 n_param, + struct snd_intelhad *intelhaddata); + int (*audio_ctrl)(struct snd_pcm_substream *substream, + struct snd_intelhad *intelhaddata); + void (*prog_dip)(struct snd_pcm_substream *substream, + struct snd_intelhad *intelhaddata); + void (*handle_underrun)(struct snd_intelhad *intelhaddata); +}; + + +int had_event_handler(enum had_event_type event_type, void *data); + +int hdmi_audio_query(void *drv_data, struct hdmi_audio_event event); +int hdmi_audio_suspend(void *drv_data, struct hdmi_audio_event event); +int hdmi_audio_resume(void *drv_data); +int hdmi_audio_mode_change(struct snd_pcm_substream *substream); +extern struct snd_pcm_ops snd_intelhad_playback_ops; + +int snd_intelhad_init_audio_ctrl(struct snd_pcm_substream *substream, + struct snd_intelhad *intelhaddata, + int flag_silence); +int snd_intelhad_prog_buffer(struct snd_intelhad *intelhaddata, + int start, int end); +int snd_intelhad_invd_buffer(int start, int end); +inline int snd_intelhad_read_len(struct snd_intelhad *intelhaddata); +void had_build_channel_allocation_map(struct snd_intelhad *intelhaddata); + +/* Register access functions */ +inline int had_get_hwstate(struct snd_intelhad *intelhaddata); +inline int had_get_caps(enum had_caps_list query_element, + void *capabilties); +inline int had_set_caps(enum had_caps_list set_element, + void *capabilties); +inline int had_read_register(uint32_t reg_addr, uint32_t *data); +inline int had_write_register(uint32_t reg_addr, uint32_t data); +inline int had_read_modify(uint32_t reg_addr, uint32_t data, + uint32_t mask); + +/**/ +int hdmi_audio_probe(void *devptr); +int hdmi_audio_remove(void *pdev); + +#endif /* _INTEL_HDMI_AUDIO_ */ diff --git a/sound/x86/intel_hdmi_audio_if.c b/sound/x86/intel_hdmi_audio_if.c new file mode 100644 index 0000000..c650ba4 --- /dev/null +++ b/sound/x86/intel_hdmi_audio_if.c @@ -0,0 +1,551 @@ +/* + * intel_hdmi_audio_if.c - Intel HDMI audio driver for MID + * + * Copyright (C) 2016 Intel Corp + * Authors: Sailaja Bandarupalli <sailaja.bandarupalli(a)intel.com> + * Ramesh Babu K V <ramesh.babu(a)intel.com> + * Vaibhav Agarwal <vaibhav.agarwal(a)intel.com> + * Jerome Anand <jerome.anand(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * ALSA driver for Intel MID HDMI audio controller. This file contains + * interface functions exposed to HDMI Display driver and code to register + * with ALSA framework.. + */ + +#define pr_fmt(fmt) "had: " fmt + +#include <linux/io.h> +#include <linux/jiffies.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/core.h> +#include "intel_hdmi_audio.h" +#include "intel_hdmi_lpe_audio.h" + +/** + * hdmi_audio_query - hdmi audio query function + * + *@haddata: pointer to HAD private data + *@event: audio event for which this method is invoked + * + * This function is called by client driver to query the + * hdmi audio. + */ +int hdmi_audio_query(void *haddata, struct hdmi_audio_event event) +{ + struct snd_pcm_substream *substream = NULL; + struct had_pvt_data *had_stream; + unsigned long flag_irqs; + struct snd_intelhad *intelhaddata = (struct snd_intelhad *)haddata; + + if (intelhaddata->stream_info.had_substream) + substream = intelhaddata->stream_info.had_substream; + had_stream = intelhaddata->private_data; + switch (event.type) { + case HAD_EVENT_QUERY_IS_AUDIO_BUSY: + spin_lock_irqsave(&intelhaddata->had_spinlock, flag_irqs); + + if ((had_stream->stream_type == HAD_RUNNING_STREAM) || + substream) { + spin_unlock_irqrestore(&intelhaddata->had_spinlock, + flag_irqs); + pr_debug("Audio stream active\n"); + return -EBUSY; + } + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + break; + + case HAD_EVENT_QUERY_IS_AUDIO_SUSPENDED: + spin_lock_irqsave(&intelhaddata->had_spinlock, flag_irqs); + if (intelhaddata->drv_status == HAD_DRV_SUSPENDED) { + spin_unlock_irqrestore(&intelhaddata->had_spinlock, + flag_irqs); + pr_debug("Audio is suspended\n"); + return 1; + } + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + break; + + default: + pr_debug("error un-handled event !!\n"); + return -EINVAL; + break; + + } + + return 0; +} + +/** + * hdmi_audio_suspend - power management suspend function + * + *@haddata: pointer to HAD private data + *@event: pm event for which this method is invoked + * + * This function is called by client driver to suspend the + * hdmi audio. + */ +int hdmi_audio_suspend(void *haddata, struct hdmi_audio_event event) +{ + int caps, retval = 0; + struct had_pvt_data *had_stream; + unsigned long flag_irqs; + struct snd_pcm_substream *substream; + struct snd_intelhad *intelhaddata = (struct snd_intelhad *)haddata; + + pr_debug("Enter:%s\n", __func__); + + had_stream = intelhaddata->private_data; + substream = intelhaddata->stream_info.had_substream; + + if (intelhaddata->dev->power.runtime_status != RPM_SUSPENDED) { + pr_err("audio stream is active\n"); + return -EAGAIN; + } + + + spin_lock_irqsave(&intelhaddata->had_spinlock, flag_irqs); + if (intelhaddata->drv_status == HAD_DRV_DISCONNECTED) { + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + pr_debug("had not connected\n"); + return retval; + } + + if (intelhaddata->drv_status == HAD_DRV_SUSPENDED) { + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + pr_debug("had already suspended\n"); + return retval; + } + + intelhaddata->drv_status = HAD_DRV_SUSPENDED; + pr_debug("%s @ %d:DEBUG PLUG/UNPLUG : HAD_DRV_SUSPENDED\n", + __func__, __LINE__); + + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + /* + * ToDo: Need to disable UNDERRUN interrupts as well + * caps = HDMI_AUDIO_UNDERRUN | HDMI_AUDIO_BUFFER_DONE; + */ + caps = HDMI_AUDIO_BUFFER_DONE; + had_set_caps(HAD_SET_DISABLE_AUDIO_INT, &caps); + had_set_caps(HAD_SET_DISABLE_AUDIO, NULL); + pr_debug("Exit:%s", __func__); + return retval; +} + +/** + * hdmi_audio_resume - power management resume function + * + *@haddata: pointer to HAD private data + * + * This function is called by client driver to resume the + * hdmi audio. + */ +int hdmi_audio_resume(void *haddata) +{ + int caps, retval = 0; + struct snd_intelhad *intelhaddata = (struct snd_intelhad *)haddata; + unsigned long flag_irqs; + + pr_debug("Enter:%s\n", __func__); + + spin_lock_irqsave(&intelhaddata->had_spinlock, flag_irqs); + if (intelhaddata->drv_status == HAD_DRV_DISCONNECTED) { + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + pr_debug("had not connected\n"); + return 0; + } + + if (intelhaddata->drv_status != HAD_DRV_SUSPENDED) { + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + pr_err("had is not in suspended state\n"); + return 0; + } + + if (had_get_hwstate(intelhaddata)) { + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + pr_err("Failed to resume. Device not accessible\n"); + return -ENODEV; + } + + intelhaddata->drv_status = HAD_DRV_CONNECTED; + pr_debug("%s @ %d:DEBUG PLUG/UNPLUG : HAD_DRV_DISCONNECTED\n", + __func__, __LINE__); + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + /* + * ToDo: Need to enable UNDERRUN interrupts as well + * caps = HDMI_AUDIO_UNDERRUN | HDMI_AUDIO_BUFFER_DONE; + */ + caps = HDMI_AUDIO_BUFFER_DONE; + retval = had_set_caps(HAD_SET_ENABLE_AUDIO_INT, &caps); + retval = had_set_caps(HAD_SET_ENABLE_AUDIO, NULL); + pr_debug("Exit:%s", __func__); + return retval; +} + +static inline int had_chk_intrmiss(struct snd_intelhad *intelhaddata, + enum intel_had_aud_buf_type buf_id) +{ + int i, intr_count = 0; + enum intel_had_aud_buf_type buff_done; + u32 buf_size, buf_addr; + struct had_pvt_data *had_stream; + unsigned long flag_irqs; + + had_stream = intelhaddata->private_data; + + buff_done = buf_id; + + intr_count = snd_intelhad_read_len(intelhaddata); + if (intr_count > 1) { + /* In case of active playback */ + pr_err("Driver detected %d missed buffer done interrupt(s)!!!!\n", + (intr_count - 1)); + if (intr_count > 3) + return intr_count; + + buf_id += (intr_count - 1); + /* Reprogram registers*/ + for (i = buff_done; i < buf_id; i++) { + int j = i % 4; + + buf_size = intelhaddata->buf_info[j].buf_size; + buf_addr = intelhaddata->buf_info[j].buf_addr; + had_write_register(AUD_BUF_A_LENGTH + + (j * HAD_REG_WIDTH), buf_size); + had_write_register( + AUD_BUF_A_ADDR+(j * HAD_REG_WIDTH), + (buf_addr | BIT(0) | BIT(1))); + } + buf_id = buf_id % 4; + spin_lock_irqsave(&intelhaddata->had_spinlock, flag_irqs); + intelhaddata->buff_done = buf_id; + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + } + + return intr_count; +} + +int had_process_buffer_done(struct snd_intelhad *intelhaddata) +{ + int retval = 0; + u32 len = 1; + enum intel_had_aud_buf_type buf_id; + enum intel_had_aud_buf_type buff_done; + struct pcm_stream_info *stream; + u32 buf_size; + struct had_pvt_data *had_stream; + int intr_count; + enum had_status_stream stream_type; + unsigned long flag_irqs; + + had_stream = intelhaddata->private_data; + stream = &intelhaddata->stream_info; + intr_count = 1; + + spin_lock_irqsave(&intelhaddata->had_spinlock, flag_irqs); + if (intelhaddata->drv_status == HAD_DRV_DISCONNECTED) { + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + pr_err("%s:Device already disconnected\n", __func__); + return retval; + } + buf_id = intelhaddata->curr_buf; + intelhaddata->buff_done = buf_id; + buff_done = intelhaddata->buff_done; + buf_size = intelhaddata->buf_info[buf_id].buf_size; + stream_type = had_stream->stream_type; + + pr_debug("Enter:%s buf_id=%d\n", __func__, buf_id); + + /* Every debug statement has an implication + * of ~5msec. Thus, avoid having >3 debug statements + * for each buffer_done handling. + */ + + /* Check for any intr_miss in case of active playback */ + if (had_stream->stream_type == HAD_RUNNING_STREAM) { + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + intr_count = had_chk_intrmiss(intelhaddata, buf_id); + if (!intr_count || (intr_count > 3)) { + pr_err("HAD SW state in non-recoverable!!! mode\n"); + pr_err("Already played stale data\n"); + return retval; + } + buf_id += (intr_count - 1); + buf_id = buf_id % 4; + spin_lock_irqsave(&intelhaddata->had_spinlock, flag_irqs); + } + + intelhaddata->buf_info[buf_id].is_valid = true; + if (intelhaddata->valid_buf_cnt-1 == buf_id) { + if (had_stream->stream_type >= HAD_RUNNING_STREAM) + intelhaddata->curr_buf = HAD_BUF_TYPE_A; + } else + intelhaddata->curr_buf = buf_id + 1; + + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + + if (had_get_hwstate(intelhaddata)) { + pr_err("HDMI cable plugged-out\n"); + return retval; + } + + /*Reprogram the registers with addr and length*/ + had_write_register(AUD_BUF_A_LENGTH + + (buf_id * HAD_REG_WIDTH), buf_size); + had_write_register(AUD_BUF_A_ADDR+(buf_id * HAD_REG_WIDTH), + intelhaddata->buf_info[buf_id].buf_addr| + BIT(0) | BIT(1)); + + had_read_register(AUD_BUF_A_LENGTH + (buf_id * HAD_REG_WIDTH), + &len); + pr_debug("%s:Enabled buf[%d]\n", __func__, buf_id); + + /* In case of actual data, + * report buffer_done to above ALSA layer + */ + buf_size = intelhaddata->buf_info[buf_id].buf_size; + if (stream_type >= HAD_RUNNING_STREAM) { + intelhaddata->stream_info.buffer_rendered += + (intr_count * buf_size); + stream->period_elapsed(stream->had_substream); + } + + return retval; +} + +int had_process_buffer_underrun(struct snd_intelhad *intelhaddata) +{ + int retval = 0; + enum intel_had_aud_buf_type buf_id; + struct pcm_stream_info *stream; + struct had_pvt_data *had_stream; + enum had_status_stream stream_type; + unsigned long flag_irqs; + int drv_status; + + had_stream = intelhaddata->private_data; + stream = &intelhaddata->stream_info; + + spin_lock_irqsave(&intelhaddata->had_spinlock, flag_irqs); + buf_id = intelhaddata->curr_buf; + stream_type = had_stream->stream_type; + intelhaddata->buff_done = buf_id; + drv_status = intelhaddata->drv_status; + if (stream_type == HAD_RUNNING_STREAM) + intelhaddata->curr_buf = HAD_BUF_TYPE_A; + + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + + pr_debug("Enter:%s buf_id=%d, stream_type=%d\n", + __func__, buf_id, stream_type); + + intelhaddata->ops->handle_underrun(intelhaddata); + + if (drv_status == HAD_DRV_DISCONNECTED) { + pr_err("%s:Device already disconnected\n", __func__); + return retval; + } + + if (stream_type == HAD_RUNNING_STREAM) { + /* Report UNDERRUN error to above layers */ + intelhaddata->flag_underrun = 1; + stream->period_elapsed(stream->had_substream); + } + + return retval; +} + +int had_process_hot_plug(struct snd_intelhad *intelhaddata) +{ + int retval = 0; + enum intel_had_aud_buf_type buf_id; + struct snd_pcm_substream *substream; + struct had_pvt_data *had_stream; + unsigned long flag_irqs; + + pr_debug("Enter:%s\n", __func__); + + substream = intelhaddata->stream_info.had_substream; + had_stream = intelhaddata->private_data; + + spin_lock_irqsave(&intelhaddata->had_spinlock, flag_irqs); + if (intelhaddata->drv_status == HAD_DRV_CONNECTED) { + pr_debug("Device already connected\n"); + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + return retval; + } + buf_id = intelhaddata->curr_buf; + intelhaddata->buff_done = buf_id; + intelhaddata->drv_status = HAD_DRV_CONNECTED; + pr_debug("%s @ %d:DEBUG PLUG/UNPLUG : HAD_DRV_CONNECTED\n", + __func__, __LINE__); + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + + pr_debug("Processing HOT_PLUG, buf_id = %d\n", buf_id); + + /* Query display driver for audio register base */ + if (intelhaddata->reg_ops.hdmi_audio_get_register_base( + &intelhaddata->audio_reg_base, + &intelhaddata->audio_cfg_offset)) { + pr_err("Unable to get audio reg base from Display driver\n"); + goto err; + } + + if (intelhaddata->audio_reg_base == NULL) { + pr_err("audio reg base value is NULL\n"); + goto err; + } + + pr_debug("%s audio_reg_base = 0x%p\n", __func__, + intelhaddata->audio_reg_base); + + /* Safety check */ + if (substream) { + pr_debug("There should not be active PB from ALSA\n"); + pr_debug("Signifies, cable is plugged-in even before\n"); + pr_debug("processing snd_pcm_disconnect\n"); + /* Set runtime->state to hw_params done */ + snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); + } + + had_build_channel_allocation_map(intelhaddata); + + return retval; + +err: + pm_runtime_disable(intelhaddata->dev); + intelhaddata->dev = NULL; + return retval; +} + +int had_process_hot_unplug(struct snd_intelhad *intelhaddata) +{ + int caps, retval = 0; + enum intel_had_aud_buf_type buf_id; + struct had_pvt_data *had_stream; + unsigned long flag_irqs; + + pr_debug("Enter:%s\n", __func__); + + had_stream = intelhaddata->private_data; + buf_id = intelhaddata->curr_buf; + + spin_lock_irqsave(&intelhaddata->had_spinlock, flag_irqs); + + if (intelhaddata->drv_status == HAD_DRV_DISCONNECTED) { + pr_debug("Device already disconnected\n"); + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + return retval; + + } else { + /* Disable Audio */ + caps = HDMI_AUDIO_BUFFER_DONE; + retval = had_set_caps(HAD_SET_DISABLE_AUDIO_INT, &caps); + retval = had_set_caps(HAD_SET_DISABLE_AUDIO, NULL); + intelhaddata->ops->enable_audio( + intelhaddata->stream_info.had_substream, 0); + } + + intelhaddata->drv_status = HAD_DRV_DISCONNECTED; + pr_debug("%s @ %d:DEBUG PLUG/UNPLUG : HAD_DRV_DISCONNECTED\n", + __func__, __LINE__); + + /* Report to above ALSA layer */ + if (intelhaddata->stream_info.had_substream != NULL) { + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + pr_debug("%s: unlock -> sending pcm_stop -> lock\n", __func__); + snd_pcm_stop(intelhaddata->stream_info.had_substream, + SNDRV_PCM_STATE_DISCONNECTED); + spin_lock_irqsave(&intelhaddata->had_spinlock, flag_irqs); + } + + had_stream->stream_type = HAD_INIT; + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + kfree(intelhaddata->chmap->chmap); + intelhaddata->chmap->chmap = NULL; + intelhaddata->audio_reg_base = NULL; + pr_debug("%s: unlocked -> returned\n", __func__); + + return retval; +} + +/** + * had_event_handler - Call back function to handle events + * + * @event_type: Event type to handle + * @data: data related to the event_type + * + * This function is invoked to handle HDMI events from client driver. + */ +int had_event_handler(enum had_event_type event_type, void *data) +{ + int retval = 0; + struct snd_intelhad *intelhaddata = data; + enum intel_had_aud_buf_type buf_id; + struct snd_pcm_substream *substream; + struct had_pvt_data *had_stream; + unsigned long flag_irqs; + + buf_id = intelhaddata->curr_buf; + had_stream = intelhaddata->private_data; + + /* Switching to a function can drop atomicity even in INTR context. + * Thus, a big lock is acquired to maintain atomicity. + * This can be optimized later. + * Currently, only buffer_done/_underrun executes in INTR context. + * Also, locking is implemented separately to avoid real contention + * of data(struct intelhaddata) between IRQ/SOFT_IRQ/PROCESS context. + */ + substream = intelhaddata->stream_info.had_substream; + switch (event_type) { + case HAD_EVENT_AUDIO_BUFFER_DONE: + retval = had_process_buffer_done(intelhaddata); + break; + + case HAD_EVENT_AUDIO_BUFFER_UNDERRUN: + retval = had_process_buffer_underrun(intelhaddata); + break; + + case HAD_EVENT_HOT_PLUG: + retval = had_process_hot_plug(intelhaddata); + break; + + case HAD_EVENT_HOT_UNPLUG: + retval = had_process_hot_unplug(intelhaddata); + break; + + case HAD_EVENT_MODE_CHANGING: + pr_debug(" called _event_handler with _MODE_CHANGE event\n"); + /* Process only if stream is active & cable Plugged-in */ + spin_lock_irqsave(&intelhaddata->had_spinlock, flag_irqs); + if (intelhaddata->drv_status >= HAD_DRV_DISCONNECTED) { + spin_unlock_irqrestore(&intelhaddata->had_spinlock, + flag_irqs); + break; + } + spin_unlock_irqrestore(&intelhaddata->had_spinlock, flag_irqs); + if ((had_stream->stream_type == HAD_RUNNING_STREAM) + && substream) + retval = hdmi_audio_mode_change(substream); + break; + + default: + pr_debug("error un-handled event !!\n"); + retval = -EINVAL; + break; + + } + return retval; +} diff --git a/sound/x86/intel_hdmi_lpe_audio.c b/sound/x86/intel_hdmi_lpe_audio.c index f31ab72..16194c6 100644 --- a/sound/x86/intel_hdmi_lpe_audio.c +++ b/sound/x86/intel_hdmi_lpe_audio.c @@ -36,6 +36,7 @@ #include <sound/initval.h> #include <drm/intel_lpe_audio.h> #include "intel_hdmi_lpe_audio.h" +#include "intel_hdmi_audio.h" /* globals*/ struct platform_device *gpdev; @@ -461,9 +462,9 @@ static void notify_audio_lpe(void *audio_ptr) /** * hdmi_lpe_audio_probe - start bridge with i915 * - * This function is called when the i915 driver creates the hdmi-lpe-audio - * platform device. Card creation is deferred until a hot plug event is - * received + * This function is called when the i915 driver creates the + * hdmi-lpe-audio platform device. Card creation is deferred until a + * hot plug event is received */ static int hdmi_lpe_audio_probe(struct platform_device *pdev) { @@ -505,8 +506,8 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) (unsigned int)res_mmio->start, (unsigned int)res_mmio->end); mmio_start = ioremap_nocache(res_mmio->start, - (size_t)((res_mmio->end - res_mmio->start) - + 1)); + (size_t)((res_mmio->end - + res_mmio->start) + 1)); if (!mmio_start) { pr_err("Could not get ioremap\n"); return -EACCES; @@ -556,11 +557,12 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) platform_set_drvdata(pdev, ctx); + ret = hdmi_audio_probe((void *)pdev); + pr_debug("hdmi lpe audio: setting pin eld notify callback\n"); spin_lock_irqsave(&pdata->lpe_audio_slock, flag_irq); pdata->notify_audio_lpe = notify_audio_lpe; - if (pdata->notify_pending) { pr_debug("%s: handle pending notification\n", __func__); @@ -584,6 +586,8 @@ static int hdmi_lpe_audio_remove(struct platform_device *pdev) pr_debug("Enter %s\n", __func__); + hdmi_audio_remove(pdev); + /* get context, release resources */ ctx = platform_get_drvdata(pdev); iounmap(ctx->mmio_start); -- 2.9.3

1 0

[alsa-devel] [RFC PATCH v4 3/7] ALSA: add shell for Intel HDMI LPE audio driver
by Jerome Anand 02 Dec '16

02 Dec '16

On Baytrail and Cherrytrail, HDaudio may be fused out or disabled by the BIOS. This driver enables an alternate path to the i915 display registers and DMA. Although there is no hardware path between i915 display and LPE/SST audio clusters, this HDMI capability is referred to in the documentation as "HDMI LPE Audio" so we keep the name for consistency. There is no hardware path or control dependencies with the LPE/SST DSP functionality. The hdmi-lpe-audio driver will be probed when the i915 driver creates a child platform device. Since this driver is neither SoC nor PCI, a new x86 folder is added Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Jerome Anand <jerome.anand(a)intel.com> --- sound/Kconfig | 2 + sound/Makefile | 2 +- sound/x86/Kconfig | 16 + sound/x86/Makefile | 8 + sound/x86/intel_hdmi_lpe_audio.c | 622 +++++++++++++++++++++++++++++++++++ sound/x86/intel_hdmi_lpe_audio.h | 692 +++++++++++++++++++++++++++++++++++++++ 6 files changed, 1341 insertions(+), 1 deletion(-) create mode 100644 sound/x86/Kconfig create mode 100644 sound/x86/Makefile create mode 100644 sound/x86/intel_hdmi_lpe_audio.c create mode 100644 sound/x86/intel_hdmi_lpe_audio.h diff --git a/sound/Kconfig b/sound/Kconfig index 5a240e0..ee2e69a 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -108,6 +108,8 @@ source "sound/parisc/Kconfig" source "sound/soc/Kconfig" +source "sound/x86/Kconfig" + endif # SND menuconfig SOUND_PRIME diff --git a/sound/Makefile b/sound/Makefile index c41bdf5..6de45d2 100644 --- a/sound/Makefile +++ b/sound/Makefile @@ -5,7 +5,7 @@ obj-$(CONFIG_SOUND) += soundcore.o obj-$(CONFIG_SOUND_PRIME) += oss/ obj-$(CONFIG_DMASOUND) += oss/ obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \ - firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ hda/ + firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ hda/ x86/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig new file mode 100644 index 0000000..182adf3 --- /dev/null +++ b/sound/x86/Kconfig @@ -0,0 +1,16 @@ +menuconfig SND_X86 + tristate "X86 sound devices" + ---help--- + + X86 sound devices that don't fall under SoC or PCI categories + +if SND_X86 + +config HDMI_LPE_AUDIO + tristate "HDMI audio without HDaudio on Intel Atom platforms" + depends on DRM_I915 + default n + help + Choose this option to support HDMI LPE Audio mode + +endif # SND_X86 diff --git a/sound/x86/Makefile b/sound/x86/Makefile new file mode 100644 index 0000000..78b2ae1 --- /dev/null +++ b/sound/x86/Makefile @@ -0,0 +1,8 @@ +DRIVER_NAME := hdmi_lpe_audio + +ccflags-y += -Idrivers/gpu/drm/i915 + +$(DRIVER_NAME)-objs += \ + intel_hdmi_lpe_audio.o + +obj-$(CONFIG_HDMI_LPE_AUDIO) += $(DRIVER_NAME).o diff --git a/sound/x86/intel_hdmi_lpe_audio.c b/sound/x86/intel_hdmi_lpe_audio.c new file mode 100644 index 0000000..f31ab72 --- /dev/null +++ b/sound/x86/intel_hdmi_lpe_audio.c @@ -0,0 +1,622 @@ +/* + * intel_hdmi_lpe_audio.c - Intel HDMI LPE audio driver for Atom platforms + * + * Copyright (C) 2016 Intel Corp + * Authors: + * Jerome Anand <jerome.anand(a)intel.com> + * Aravind Siddappaji <aravindx.siddappaji(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#define pr_fmt(fmt) "hdmi_lpe_audio: " fmt + +#include <linux/platform_device.h> +#include <linux/irqreturn.h> +#include <linux/interrupt.h> +#include <linux/io.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <sound/pcm.h> +#include <sound/core.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/control.h> +#include <sound/initval.h> +#include <drm/intel_lpe_audio.h> +#include "intel_hdmi_lpe_audio.h" + +/* globals*/ +struct platform_device *gpdev; +int _hdmi_state; +union otm_hdmi_eld_t hdmi_eld; + +struct hdmi_lpe_audio_ctx { + int irq; + void __iomem *mmio_start; + had_event_call_back had_event_callbacks; + struct snd_intel_had_interface *had_interface; + void *had_pvt_data; + int tmds_clock_speed; + unsigned int had_config_offset; + int hdmi_audio_interrupt_mask; + struct work_struct hdmi_audio_wq; +}; + +static inline void hdmi_set_eld(void *eld) +{ + int size = (sizeof(hdmi_eld)) > HDMI_MAX_ELD_BYTES ? + HDMI_MAX_ELD_BYTES : + (sizeof(hdmi_eld)); + + memcpy((void *)&hdmi_eld, eld, size); +} + +static inline int hdmi_get_eld(void *eld) +{ + memcpy(eld, (void *)&hdmi_eld, sizeof(hdmi_eld)); + + { + int i; + uint8_t *eld_data = (uint8_t *)&hdmi_eld; + + pr_debug("hdmi_get_eld:\n{{"); + + for (i = 0; i < sizeof(hdmi_eld); i++) + pr_debug("0x%x, ", eld_data[i]); + + pr_debug("}}\n"); + } + return HAD_SUCCESS; +} + + +static inline struct hdmi_lpe_audio_ctx *get_hdmi_context(void) +{ + struct hdmi_lpe_audio_ctx *ctx; + + ctx = platform_get_drvdata(gpdev); + return ctx; +} + +/* + * return whether HDMI audio device is busy. + */ +bool mid_hdmi_audio_is_busy(void *ddev) +{ + struct hdmi_lpe_audio_ctx *ctx; + int hdmi_audio_busy = 0; + struct hdmi_audio_event hdmi_audio_event; + + pr_debug("%s: Enter", __func__); + + ctx = platform_get_drvdata(gpdev); + + if (_hdmi_state == hdmi_connector_status_disconnected) { + /* HDMI is not connected, assuming audio device is idle. */ + return false; + } + + if (ctx->had_interface) { + hdmi_audio_event.type = HAD_EVENT_QUERY_IS_AUDIO_BUSY; + hdmi_audio_busy = ctx->had_interface->query( + ctx->had_pvt_data, + hdmi_audio_event); + return hdmi_audio_busy != 0; + } + return false; +} + +/* + * return whether HDMI audio device is suspended. + */ +bool mid_hdmi_audio_suspend(void *ddev) +{ + struct hdmi_lpe_audio_ctx *ctx; + struct hdmi_audio_event hdmi_audio_event; + int ret = 0; + + ctx = platform_get_drvdata(gpdev); + + if (_hdmi_state == hdmi_connector_status_disconnected) { + /* HDMI is not connected, assuming audio device + * is suspended already. + */ + return true; + } + + pr_debug("%s: _hdmi_state %d", __func__, + _hdmi_state); + + if (ctx->had_interface) { + hdmi_audio_event.type = 0; + ret = ctx->had_interface->suspend(ctx->had_pvt_data, + hdmi_audio_event); + return (ret == 0) ? true : false; + } + return true; +} + +void mid_hdmi_audio_resume(void *ddev) +{ + struct hdmi_lpe_audio_ctx *ctx; + + ctx = platform_get_drvdata(gpdev); + + if (_hdmi_state == hdmi_connector_status_disconnected) { + /* HDMI is not connected, there is no need + * to resume audio device. + */ + return; + } + + pr_debug("%s: _hdmi_state %d", __func__, _hdmi_state); + + if (ctx->had_interface) + ctx->had_interface->resume(ctx->had_pvt_data); +} + +void mid_hdmi_audio_signal_event(enum had_event_type event) +{ + struct hdmi_lpe_audio_ctx *ctx; + + pr_debug("%s: Enter\n", __func__); + + ctx = platform_get_drvdata(gpdev); + + if (ctx->had_event_callbacks) + (*ctx->had_event_callbacks)(event, + ctx->had_pvt_data); +} + +/** + * hdmi_audio_write: + * used to write into display controller HDMI audio registers. + * + */ +static int hdmi_audio_write(uint32_t reg, uint32_t val) +{ + struct hdmi_lpe_audio_ctx *ctx; + + ctx = platform_get_drvdata(gpdev); + + pr_debug("%s: reg[0x%x] = 0x%x\n", __func__, reg, val); + + iowrite32(val, (ctx->mmio_start+reg)); + + return HAD_SUCCESS; +} + +/** + * hdmi_audio_read: + * used to get the register value read from + * display controller HDMI audio registers. + */ +static int hdmi_audio_read(uint32_t reg, uint32_t *val) +{ + struct hdmi_lpe_audio_ctx *ctx; + + ctx = platform_get_drvdata(gpdev); + *val = ioread32(ctx->mmio_start+reg); + pr_debug("%s: reg[0x%x] = 0x%x\n", __func__, reg, *val); + return HAD_SUCCESS; +} + +/** + * hdmi_audio_rmw: + * used to update the masked bits in display controller HDMI + * audio registers. + */ +static int hdmi_audio_rmw(uint32_t reg, uint32_t val, uint32_t mask) +{ + struct hdmi_lpe_audio_ctx *ctx; + uint32_t val_tmp = 0; + + ctx = platform_get_drvdata(gpdev); + + val_tmp = (val & mask) | + ((ioread32(ctx->mmio_start + reg)) & ~mask); + + iowrite32(val_tmp, (ctx->mmio_start+reg)); + pr_debug("%s: reg[0x%x] = 0x%x\n", __func__, reg, val_tmp); + + return HAD_SUCCESS; +} + +/** + * hdmi_audio_get_caps: + * used to return the HDMI audio capabilities. + * e.g. resolution, frame rate. + */ +static int hdmi_audio_get_caps(enum had_caps_list get_element, + void *capabilities) +{ + struct hdmi_lpe_audio_ctx *ctx; + int ret = HAD_SUCCESS; + + ctx = get_hdmi_context(); + + pr_debug("%s: Enter\n", __func__); + + switch (get_element) { + case HAD_GET_ELD: + ret = hdmi_get_eld(capabilities); + break; + case HAD_GET_DISPLAY_RATE: + /* ToDo: Verify if sampling freq logic is correct */ + memcpy(capabilities, &(ctx->tmds_clock_speed), + sizeof(uint32_t)); + pr_debug("%s: tmds_clock_speed = 0x%x\n", __func__, + ctx->tmds_clock_speed); + break; + default: + break; + } + + return ret; +} + +/** + * hdmi_audio_get_register_base + * used to get the current hdmi base address + */ +int hdmi_audio_get_register_base(uint32_t **reg_base, + uint32_t *config_offset) +{ + struct hdmi_lpe_audio_ctx *ctx; + + ctx = platform_get_drvdata(gpdev); + *reg_base = (uint32_t *)(ctx->mmio_start); + *config_offset = ctx->had_config_offset; + pr_debug("%s: reg_base = 0x%p, cfg_off = 0x%x\n", __func__, + *reg_base, *config_offset); + return HAD_SUCCESS; +} + +/** + * hdmi_audio_set_caps: + * used to set the HDMI audio capabilities. + * e.g. Audio INT. + */ +int hdmi_audio_set_caps(enum had_caps_list set_element, + void *capabilties) +{ + struct hdmi_lpe_audio_ctx *ctx; + + ctx = platform_get_drvdata(gpdev); + + pr_debug("%s: cap_id = 0x%x\n", __func__, set_element); + + switch (set_element) { + case HAD_SET_ENABLE_AUDIO_INT: + { + uint32_t status_reg; + + hdmi_audio_read(AUD_HDMI_STATUS_v2 + + ctx->had_config_offset, &status_reg); + status_reg |= + HDMI_AUDIO_BUFFER_DONE | HDMI_AUDIO_UNDERRUN; + hdmi_audio_write(AUD_HDMI_STATUS_v2 + + ctx->had_config_offset, status_reg); + hdmi_audio_read(AUD_HDMI_STATUS_v2 + + ctx->had_config_offset, &status_reg); + + } + break; + default: + break; + } + + return HAD_SUCCESS; +} + +static struct hdmi_audio_registers_ops hdmi_audio_reg_ops = { + .hdmi_audio_get_register_base = hdmi_audio_get_register_base, + .hdmi_audio_read_register = hdmi_audio_read, + .hdmi_audio_write_register = hdmi_audio_write, + .hdmi_audio_read_modify = hdmi_audio_rmw, +}; + +static struct hdmi_audio_query_set_ops hdmi_audio_get_set_ops = { + .hdmi_audio_get_caps = hdmi_audio_get_caps, + .hdmi_audio_set_caps = hdmi_audio_set_caps, +}; + +int mid_hdmi_audio_setup( + had_event_call_back audio_callbacks, + struct hdmi_audio_registers_ops *reg_ops, + struct hdmi_audio_query_set_ops *query_ops) +{ + struct hdmi_lpe_audio_ctx *ctx; + + ctx = platform_get_drvdata(gpdev); + + pr_debug("%s: called\n", __func__); + + reg_ops->hdmi_audio_get_register_base = + (hdmi_audio_reg_ops.hdmi_audio_get_register_base); + reg_ops->hdmi_audio_read_register = + (hdmi_audio_reg_ops.hdmi_audio_read_register); + reg_ops->hdmi_audio_write_register = + (hdmi_audio_reg_ops.hdmi_audio_write_register); + reg_ops->hdmi_audio_read_modify = + (hdmi_audio_reg_ops.hdmi_audio_read_modify); + query_ops->hdmi_audio_get_caps = + hdmi_audio_get_set_ops.hdmi_audio_get_caps; + query_ops->hdmi_audio_set_caps = + hdmi_audio_get_set_ops.hdmi_audio_set_caps; + + ctx->had_event_callbacks = audio_callbacks; + + return HAD_SUCCESS; +} + +void _had_wq(struct work_struct *work) +{ + mid_hdmi_audio_signal_event(HAD_EVENT_HOT_PLUG); +} + +int mid_hdmi_audio_register(struct snd_intel_had_interface *driver, + void *had_data) +{ + struct hdmi_lpe_audio_ctx *ctx; + + ctx = platform_get_drvdata(gpdev); + + pr_debug("%s: called\n", __func__); + + ctx->had_pvt_data = had_data; + ctx->had_interface = driver; + + /* The Audio driver is loading now and we need to notify + * it if there is an HDMI device attached + */ + INIT_WORK(&ctx->hdmi_audio_wq, _had_wq); + pr_debug("%s: Scheduling HDMI audio work queue\n", __func__); + schedule_work(&ctx->hdmi_audio_wq); + + return HAD_SUCCESS; +} + +static irqreturn_t display_pipe_interrupt_handler(int irq, void *dev_id) +{ + u32 audio_stat, audio_reg; + + struct hdmi_lpe_audio_ctx *ctx; + + pr_debug("%s: Enter\n", __func__); + + ctx = platform_get_drvdata(gpdev); + + audio_reg = ctx->had_config_offset + AUD_HDMI_STATUS_v2; + hdmi_audio_read(audio_reg, &audio_stat); + + if (audio_stat & HDMI_AUDIO_UNDERRUN) { + hdmi_audio_write(audio_reg, HDMI_AUDIO_UNDERRUN); + mid_hdmi_audio_signal_event( + HAD_EVENT_AUDIO_BUFFER_UNDERRUN); + } + + if (audio_stat & HDMI_AUDIO_BUFFER_DONE) { + hdmi_audio_write(audio_reg, HDMI_AUDIO_BUFFER_DONE); + mid_hdmi_audio_signal_event( + HAD_EVENT_AUDIO_BUFFER_DONE); + } + + return IRQ_HANDLED; +} + +static void notify_audio_lpe(void *audio_ptr) +{ + struct hdmi_lpe_audio_ctx *ctx = get_hdmi_context(); + struct intel_hdmi_lpe_audio_pdata *pdata = gpdev->dev.platform_data; + struct intel_hdmi_lpe_audio_eld *eld = audio_ptr; + + if (pdata->hdmi_connected != true) { + + pr_debug("%s: Event: HAD_NOTIFY_HOT_UNPLUG\n", + __func__); + + if (_hdmi_state == hdmi_connector_status_connected) { + + _hdmi_state = + hdmi_connector_status_disconnected; + + mid_hdmi_audio_signal_event( + HAD_EVENT_HOT_UNPLUG); + } else + pr_debug("%s: Already Unplugged!\n", __func__); + + } else if (eld != NULL) { + + hdmi_set_eld(eld->eld_data); + + mid_hdmi_audio_signal_event(HAD_EVENT_HOT_PLUG); + + _hdmi_state = hdmi_connector_status_connected; + + pr_debug("%s: HAD_NOTIFY_ELD : port = %d, tmds = %d\n", + __func__, eld->port_id, + pdata->tmds_clock_speed); + + if (pdata->tmds_clock_speed) { + ctx->tmds_clock_speed = pdata->tmds_clock_speed; + mid_hdmi_audio_signal_event(HAD_EVENT_MODE_CHANGING); + } + } else + pr_debug("%s: Event: NULL EDID!!\n", __func__); +} + +/** + * hdmi_lpe_audio_probe - start bridge with i915 + * + * This function is called when the i915 driver creates the hdmi-lpe-audio + * platform device. Card creation is deferred until a hot plug event is + * received + */ +static int hdmi_lpe_audio_probe(struct platform_device *pdev) +{ + struct hdmi_lpe_audio_ctx *ctx; + struct intel_hdmi_lpe_audio_pdata *pdata; + int irq; + struct resource *res_mmio; + void __iomem *mmio_start; + int ret = 0; + unsigned long flag_irq; + const struct pci_device_id cherryview_ids[] = { + {PCI_DEVICE(0x8086, 0x22b0)}, + {PCI_DEVICE(0x8086, 0x22b1)}, + {PCI_DEVICE(0x8086, 0x22b2)}, + {PCI_DEVICE(0x8086, 0x22b3)}, + {} + }; + + pr_debug("Enter %s\n", __func__); + + /*TBD:remove globals*/ + gpdev = pdev; + _hdmi_state = hdmi_connector_status_disconnected; + + /* get resources */ + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + pr_err("Could not get irq resource\n"); + return -ENODEV; + } + + res_mmio = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res_mmio) { + pr_err("Could not get IO_MEM resources\n"); + return -ENXIO; + } + + pr_debug("%s: mmio_start = 0x%x, mmio_end = 0x%x\n", __func__, + (unsigned int)res_mmio->start, (unsigned int)res_mmio->end); + + mmio_start = ioremap_nocache(res_mmio->start, + (size_t)((res_mmio->end - res_mmio->start) + + 1)); + if (!mmio_start) { + pr_err("Could not get ioremap\n"); + return -EACCES; + } + + /* setup interrupt handler */ + ret = request_irq(irq, display_pipe_interrupt_handler, + 0, /* FIXME: is IRQF_SHARED needed ? */ + pdev->name, + NULL); + if (ret < 0) { + pr_err("request_irq failed\n"); + iounmap(mmio_start); + return -ENODEV; + } + + /* alloc and save context */ + ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); + if (ctx == NULL) { + free_irq(irq, NULL); + iounmap(mmio_start); + return -ENOMEM; + } + + ctx->irq = irq; + pr_debug("hdmi lpe audio: irq num = %d\n", irq); + ctx->mmio_start = mmio_start; + ctx->tmds_clock_speed = DIS_SAMPLE_RATE_148_5; + + if (pci_dev_present(cherryview_ids)) { + pr_debug("%s: Cherrytrail LPE - Detected\n", __func__); + ctx->had_config_offset = AUDIO_HDMI_CONFIG_C; + } else { + pr_debug("%s: Baytrail LPE - Assume\n", __func__); + ctx->had_config_offset = AUDIO_HDMI_CONFIG_A; + } + + pdata = pdev->dev.platform_data; + + if (pdata == NULL) { + pr_err("%s: quit: pdata not allocated by i915!!\n", __func__); + kfree(ctx); + free_irq(irq, NULL); + iounmap(mmio_start); + return -ENOMEM; + } + + platform_set_drvdata(pdev, ctx); + + pr_debug("hdmi lpe audio: setting pin eld notify callback\n"); + + spin_lock_irqsave(&pdata->lpe_audio_slock, flag_irq); + pdata->notify_audio_lpe = notify_audio_lpe; + + if (pdata->notify_pending) { + + pr_debug("%s: handle pending notification\n", __func__); + notify_audio_lpe(&pdata->eld); + pdata->notify_pending = false; + } + spin_unlock_irqrestore(&pdata->lpe_audio_slock, flag_irq); + + return ret; +} + +/** + * hdmi_lpe_audio_remove - stop bridge with i915 + * + * This function is called when the platform device is destroyed. The sound + * card should have been removed on hot plug event. + */ +static int hdmi_lpe_audio_remove(struct platform_device *pdev) +{ + struct hdmi_lpe_audio_ctx *ctx; + + pr_debug("Enter %s\n", __func__); + + /* get context, release resources */ + ctx = platform_get_drvdata(pdev); + iounmap(ctx->mmio_start); + free_irq(ctx->irq, NULL); + kfree(ctx); + return HAD_SUCCESS; +} + +static int hdmi_lpe_audio_suspend(struct platform_device *pt_dev, + pm_message_t state) +{ + pr_debug("Enter %s\n", __func__); + mid_hdmi_audio_suspend(NULL); + return HAD_SUCCESS; +} + +static int hdmi_lpe_audio_resume(struct platform_device *pt_dev) +{ + pr_debug("Enter %s\n", __func__); + mid_hdmi_audio_resume(NULL); + return HAD_SUCCESS; +} + +static struct platform_driver hdmi_lpe_audio_driver = { + .driver = { + .name = "hdmi-lpe-audio", + }, + .probe = hdmi_lpe_audio_probe, + .remove = hdmi_lpe_audio_remove, + .suspend = hdmi_lpe_audio_suspend, + .resume = hdmi_lpe_audio_resume +}; + +module_platform_driver(hdmi_lpe_audio_driver); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:hdmi_lpe_audio"); diff --git a/sound/x86/intel_hdmi_lpe_audio.h b/sound/x86/intel_hdmi_lpe_audio.h new file mode 100644 index 0000000..d4b94f0a --- /dev/null +++ b/sound/x86/intel_hdmi_lpe_audio.h @@ -0,0 +1,692 @@ +/* + * intel_hdmi_lpe_audio.h - Intel HDMI LPE audio driver + * + * Copyright (C) 2016 Intel Corp + * Authors: Sailaja Bandarupalli <sailaja.bandarupalli(a)intel.com> + * Ramesh Babu K V <ramesh.babu(a)intel.com> + * Vaibhav Agarwal <vaibhav.agarwal(a)intel.com> + * Jerome Anand <jerome.anand(a)intel.com> + * Aravind Siddappaji <aravindx.siddappaji(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#ifndef __INTEL_HDMI_LPE_AUDIO_H +#define __INTEL_HDMI_LPE_AUDIO_H + +#include <linux/types.h> +#include <sound/initval.h> +#include <linux/version.h> +#include <linux/pm_runtime.h> +#include <sound/asoundef.h> +#include <sound/control.h> +#include <sound/pcm.h> + +#define HMDI_LPE_AUDIO_DRIVER_NAME "intel-hdmi-lpe-audio" +#define HAD_DRIVER_VERSION "0.01.003" +#define HAD_MAX_DEVICES 1 +#define HAD_MIN_CHANNEL 2 +#define HAD_MAX_CHANNEL 8 +#define HAD_NUM_OF_RING_BUFS 4 + +/* HDMI Audio LPE Error Codes */ +#define HAD_SUCCESS 0 +#define HAD_FAIL 1 + +/* Assume 192KHz, 8channel, 25msec period */ +#define HAD_MAX_BUFFER (600*1024) +#define HAD_MIN_BUFFER (32*1024) +#define HAD_MAX_PERIODS 4 +#define HAD_MIN_PERIODS 4 +#define HAD_MAX_PERIOD_BYTES (HAD_MAX_BUFFER/HAD_MIN_PERIODS) +#define HAD_MIN_PERIOD_BYTES 256 +#define HAD_FIFO_SIZE 0 /* fifo not being used */ +#define MAX_SPEAKERS 8 +/* TODO: Add own tlv when channel map is ported for user space */ +#define USE_ALSA_DEFAULT_TLV + +#define AUD_SAMPLE_RATE_32 32000 +#define AUD_SAMPLE_RATE_44_1 44100 +#define AUD_SAMPLE_RATE_48 48000 +#define AUD_SAMPLE_RATE_88_2 88200 +#define AUD_SAMPLE_RATE_96 96000 +#define AUD_SAMPLE_RATE_176_4 176400 +#define AUD_SAMPLE_RATE_192 192000 + +#define HAD_MIN_RATE AUD_SAMPLE_RATE_32 +#define HAD_MAX_RATE AUD_SAMPLE_RATE_192 + +#define DIS_SAMPLE_RATE_25_2 25200 +#define DIS_SAMPLE_RATE_27 27000 +#define DIS_SAMPLE_RATE_54 54000 +#define DIS_SAMPLE_RATE_74_25 74250 +#define DIS_SAMPLE_RATE_148_5 148500 +#define HAD_REG_WIDTH 0x08 +#define HAD_MAX_HW_BUFS 0x04 +#define HAD_MAX_DIP_WORDS 16 +#define INTEL_HAD "IntelHdmiLpeAudio" + +/* _AUD_CONFIG register MASK */ +#define AUD_CONFIG_MASK_UNDERRUN 0xC0000000 +#define AUD_CONFIG_MASK_SRDBG 0x00000002 +#define AUD_CONFIG_MASK_FUNCRST 0x00000001 + +#define MAX_CNT 0xFF +#define HAD_SUSPEND_DELAY 1000 + +#define OTM_HDMI_ELD_SIZE 128 + +union otm_hdmi_eld_t { + uint8_t eld_data[OTM_HDMI_ELD_SIZE]; + #pragma pack(1) + struct { + /* Byte[0] = ELD Version Number */ + union { + uint8_t byte0; + struct { + uint8_t reserved:3; /* Reserf */ + uint8_t eld_ver:5; /* ELD Version Number */ + /* 00000b - reserved + * 00001b - first rev, obsoleted + * 00010b - version 2, supporting CEA version + * 861D or below + * 00011b:11111b - reserved + * for future + */ + }; + }; + + /* Byte[1] = Vendor Version Field */ + union { + uint8_t vendor_version; + struct { + uint8_t reserved1:3; + uint8_t veld_ver:5; /* Version number of the ELD + * extension. This value is + * provisioned and unique to + * each vendor. + */ + }; + }; + + /* Byte[2] = Baseline Length field */ + uint8_t baseline_eld_length; /* Length of the Baseline structure + * divided by Four. + */ + + /* Byte [3] = Reserved for future use */ + uint8_t byte3; + + /* Starting of the BaseLine EELD structure + * Byte[4] = Monitor Name Length + */ + union { + uint8_t byte4; + struct { + uint8_t mnl:5; + uint8_t cea_edid_rev_id:3; + }; + }; + + /* Byte[5] = Capabilities */ + union { + uint8_t capabilities; + struct { + uint8_t hdcp:1; /* HDCP support */ + uint8_t ai_support:1; /* AI support */ + uint8_t connection_type:2; /* Connection type + * 00 - HDMI + * 01 - DP + * 10 -11 Reserved + * for future + * connection types + */ + uint8_t sadc:4; /* Indicates number of 3 bytes + * Short Audio Descriptors. + */ + }; + }; + + /* Byte[6] = Audio Synch Delay */ + uint8_t audio_synch_delay; /* Amount of time reported by the + * sink that the video trails audio + * in milliseconds. + */ + + /* Byte[7] = Speaker Allocation Block */ + union { + uint8_t speaker_allocation_block; + struct { + uint8_t flr:1; /*Front Left and Right channels*/ + uint8_t lfe:1; /*Low Frequency Effect channel*/ + uint8_t fc:1; /*Center transmission channel*/ + uint8_t rlr:1; /*Rear Left and Right channels*/ + uint8_t rc:1; /*Rear Center channel*/ + uint8_t flrc:1; /*Front left and Right of Center + *transmission channels + */ + uint8_t rlrc:1; /*Rear left and Right of Center + *transmission channels + */ + uint8_t reserved3:1; /* Reserved */ + }; + }; + + /* Byte[8 - 15] - 8 Byte port identification value */ + uint8_t port_id_value[8]; + + /* Byte[16 - 17] - 2 Byte Manufacturer ID */ + uint8_t manufacturer_id[2]; + + /* Byte[18 - 19] - 2 Byte Product ID */ + uint8_t product_id[2]; + + /* Byte [20-83] - 64 Bytes of BaseLine Data */ + uint8_t mn_sand_sads[64]; /* This will include + * - ASCII string of Monitor name + * - List of 3 byte SADs + * - Zero padding + */ + + /* Vendor ELD Block should continue here! + * No Vendor ELD block defined as of now. + */ + }; + #pragma pack() +}; + +/** + * enum had_status - Audio stream states + * + * @STREAM_INIT: Stream initialized + * @STREAM_RUNNING: Stream running + * @STREAM_PAUSED: Stream paused + * @STREAM_DROPPED: Stream dropped + */ +enum had_stream_status { + STREAM_INIT = 0, + STREAM_RUNNING = 1, + STREAM_PAUSED = 2, + STREAM_DROPPED = 3 +}; + +/** + * enum had_status_stream - HAD stream states + */ +enum had_status_stream { + HAD_INIT = 0, + HAD_RUNNING_STREAM, +}; + +enum had_drv_status { + HAD_DRV_CONNECTED, + HAD_DRV_RUNNING, + HAD_DRV_DISCONNECTED, + HAD_DRV_SUSPENDED, + HAD_DRV_ERR, +}; + +/* enum intel_had_aud_buf_type - HDMI controller ring buffer types */ +enum intel_had_aud_buf_type { + HAD_BUF_TYPE_A = 0, + HAD_BUF_TYPE_B = 1, + HAD_BUF_TYPE_C = 2, + HAD_BUF_TYPE_D = 3, +}; + +enum num_aud_ch { + CH_STEREO = 0, + CH_THREE_FOUR = 1, + CH_FIVE_SIX = 2, + CH_SEVEN_EIGHT = 3 +}; + +/* HDMI Controller register offsets - audio domain common */ +/* Base address for below regs = 0x65000 */ +enum hdmi_ctrl_reg_offset_common { + AUDIO_HDMI_CONFIG_A = 0x000, + AUDIO_HDMI_CONFIG_B = 0x800, + AUDIO_HDMI_CONFIG_C = 0x900, +}; +/* HDMI controller register offsets */ +enum hdmi_ctrl_reg_offset_v1 { + AUD_CONFIG = 0x0, + AUD_CH_STATUS_0 = 0x08, + AUD_CH_STATUS_1 = 0x0C, + AUD_HDMI_CTS = 0x10, + AUD_N_ENABLE = 0x14, + AUD_SAMPLE_RATE = 0x18, + AUD_BUF_CONFIG = 0x20, + AUD_BUF_CH_SWAP = 0x24, + AUD_BUF_A_ADDR = 0x40, + AUD_BUF_A_LENGTH = 0x44, + AUD_BUF_B_ADDR = 0x48, + AUD_BUF_B_LENGTH = 0x4c, + AUD_BUF_C_ADDR = 0x50, + AUD_BUF_C_LENGTH = 0x54, + AUD_BUF_D_ADDR = 0x58, + AUD_BUF_D_LENGTH = 0x5c, + AUD_CNTL_ST = 0x60, + AUD_HDMI_STATUS = 0x68, + AUD_HDMIW_INFOFR = 0x114, +}; + +/* + * Delta changes in HDMI controller register offsets + * compare to v1 version + */ + +enum hdmi_ctrl_reg_offset_v2 { + AUD_HDMI_STATUS_v2 = 0x64, + AUD_HDMIW_INFOFR_v2 = 0x68, +}; + +/* + * CEA speaker placement: + * + * FL FLC FC FRC FR + * + * LFE + * + * RL RLC RC RRC RR + * + * The Left/Right Surround channel _notions_ LS/RS in SMPTE 320M + * corresponds to CEA RL/RR; The SMPTE channel _assignment_ C/LFE is + * swapped to CEA LFE/FC. + */ +enum cea_speaker_placement { + FL = (1 << 0), /* Front Left */ + FC = (1 << 1), /* Front Center */ + FR = (1 << 2), /* Front Right */ + FLC = (1 << 3), /* Front Left Center */ + FRC = (1 << 4), /* Front Right Center */ + RL = (1 << 5), /* Rear Left */ + RC = (1 << 6), /* Rear Center */ + RR = (1 << 7), /* Rear Right */ + RLC = (1 << 8), /* Rear Left Center */ + RRC = (1 << 9), /* Rear Right Center */ + LFE = (1 << 10), /* Low Frequency Effect */ +}; + +struct cea_channel_speaker_allocation { + int ca_index; + int speakers[8]; + + /* derived values, just for convenience */ + int channels; + int spk_mask; +}; + +struct channel_map_table { + unsigned char map; /* ALSA API channel map position */ + unsigned char cea_slot; /* CEA slot value */ + int spk_mask; /* speaker position bit mask */ +}; + +/** + * union aud_cfg - Audio configuration + * + * @cfg_regx: individual register bits + * @cfg_regval: full register value + * + */ +union aud_cfg { + struct { + u32 aud_en:1; + u32 layout:1; + u32 fmt:2; + u32 num_ch:2; + u32 rsvd0:1; + u32 set:1; + u32 flat:1; + u32 val_bit:1; + u32 user_bit:1; + u32 underrun:1; + u32 rsvd1:20; + } cfg_regx; + struct { + u32 aud_en:1; + u32 layout:1; + u32 fmt:2; + u32 num_ch:3; + u32 set:1; + u32 flat:1; + u32 val_bit:1; + u32 user_bit:1; + u32 underrun:1; + u32 packet_mode:1; + u32 left_align:1; + u32 bogus_sample:1; + u32 dp_modei:1; + u32 rsvd:16; + } cfg_regx_v2; + u32 cfg_regval; +}; + +/** + * union aud_ch_status_0 - Audio Channel Status 0 Attributes + * + * @status_0_regx:individual register bits + * @status_0_regval:full register value + * + */ +union aud_ch_status_0 { + struct { + u32 ch_status:1; + u32 lpcm_id:1; + u32 cp_info:1; + u32 format:3; + u32 mode:2; + u32 ctg_code:8; + u32 src_num:4; + u32 ch_num:4; + u32 samp_freq:4; + u32 clk_acc:2; + u32 rsvd:2; + } status_0_regx; + u32 status_0_regval; +}; + +/** + * union aud_ch_status_1 - Audio Channel Status 1 Attributes + * + * @status_1_regx: individual register bits + * @status_1_regval: full register value + * + */ +union aud_ch_status_1 { + struct { + u32 max_wrd_len:1; + u32 wrd_len:3; + u32 rsvd:28; + } status_1_regx; + u32 status_1_regval; +}; + +/** + * union aud_hdmi_cts - CTS register + * + * @cts_regx: individual register bits + * @cts_regval: full register value + * + */ +union aud_hdmi_cts { + struct { + u32 cts_val:20; + u32 en_cts_prog:1; + u32 rsvd:11; + } cts_regx; + struct { + u32 cts_val:24; + u32 en_cts_prog:1; + u32 rsvd:7; + } cts_regx_v2; + u32 cts_regval; +}; + +/** + * union aud_hdmi_n_enable - N register + * + * @n_regx: individual register bits + * @n_regval: full register value + * + */ +union aud_hdmi_n_enable { + struct { + u32 n_val:20; + u32 en_n_prog:1; + u32 rsvd:11; + } n_regx; + struct { + u32 n_val:24; + u32 en_n_prog:1; + u32 rsvd:7; + } n_regx_v2; + u32 n_regval; +}; + +/** + * union aud_buf_config - Audio Buffer configurations + * + * @buf_cfg_regx: individual register bits + * @buf_cfgval: full register value + * + */ +union aud_buf_config { + struct { + u32 fifo_width:8; + u32 rsvd0:8; + u32 aud_delay:8; + u32 rsvd1:8; + } buf_cfg_regx; + struct { + u32 audio_fifo_watermark:8; + u32 dma_fifo_watermark:3; + u32 rsvd0:5; + u32 aud_delay:8; + u32 rsvd1:8; + } buf_cfg_regx_v2; + u32 buf_cfgval; +}; + +/** + * union aud_buf_ch_swap - Audio Sample Swapping offset + * + * @buf_ch_swap_regx: individual register bits + * @buf_ch_swap_val: full register value + * + */ +union aud_buf_ch_swap { + struct { + u32 first_0:3; + u32 second_0:3; + u32 first_1:3; + u32 second_1:3; + u32 first_2:3; + u32 second_2:3; + u32 first_3:3; + u32 second_3:3; + u32 rsvd:8; + } buf_ch_swap_regx; + u32 buf_ch_swap_val; +}; + +/** + * union aud_buf_addr - Address for Audio Buffer + * + * @buf_addr_regx: individual register bits + * @buf_addr_val: full register value + * + */ +union aud_buf_addr { + struct { + u32 valid:1; + u32 intr_en:1; + u32 rsvd:4; + u32 addr:26; + } buf_addr_regx; + u32 buf_addr_val; +}; + +/** + * union aud_buf_len - Length of Audio Buffer + * + * @buf_len_regx: individual register bits + * @buf_len_val: full register value + * + */ +union aud_buf_len { + struct { + u32 buf_len:20; + u32 rsvd:12; + } buf_len_regx; + u32 buf_len_val; +}; + +/** + * union aud_ctrl_st - Audio Control State Register offset + * + * @ctrl_regx: individual register bits + * @ctrl_val: full register value + * + */ +union aud_ctrl_st { + struct { + u32 ram_addr:4; + u32 eld_ack:1; + u32 eld_addr:4; + u32 eld_buf_size:5; + u32 eld_valid:1; + u32 cp_ready:1; + u32 dip_freq:2; + u32 dip_idx:3; + u32 dip_en_sta:4; + u32 rsvd:7; + } ctrl_regx; + u32 ctrl_val; +}; + +/** + * union aud_info_frame1 - Audio HDMI Widget Data Island Packet offset + * + * @fr1_regx: individual register bits + * @fr1_val: full register value + * + */ +union aud_info_frame1 { + struct { + u32 pkt_type:8; + u32 ver_num:8; + u32 len:5; + u32 rsvd:11; + } fr1_regx; + u32 fr1_val; +}; + +/** + * union aud_info_frame2 - DIP frame 2 + * + * @fr2_regx: individual register bits + * @fr2_val: full register value + * + */ +union aud_info_frame2 { + struct { + u32 chksum:8; + u32 chnl_cnt:3; + u32 rsvd0:1; + u32 coding_type:4; + u32 smpl_size:2; + u32 smpl_freq:3; + u32 rsvd1:3; + u32 format:8; + } fr2_regx; + u32 fr2_val; +}; + +/** + * union aud_info_frame3 - DIP frame 3 + * + * @fr3_regx: individual register bits + * @fr3_val: full register value + * + */ +union aud_info_frame3 { + struct { + u32 chnl_alloc:8; + u32 rsvd0:3; + u32 lsv:4; + u32 dm_inh:1; + u32 rsvd1:16; + } fr3_regx; + u32 fr3_val; +}; + +enum hdmi_connector_status { + hdmi_connector_status_connected = 1, + hdmi_connector_status_disconnected = 2, + hdmi_connector_status_unknown = 3, +}; + +#define HDMI_AUDIO_UNDERRUN (1UL<<31) +#define HDMI_AUDIO_BUFFER_DONE (1UL<<29) + + +#define PORT_ENABLE (1 << 31) +#define SDVO_AUDIO_ENABLE (1 << 6) + +enum had_caps_list { + HAD_GET_ELD = 1, + HAD_GET_DISPLAY_RATE, + HAD_SET_ENABLE_AUDIO, + HAD_SET_DISABLE_AUDIO, + HAD_SET_ENABLE_AUDIO_INT, + HAD_SET_DISABLE_AUDIO_INT, +}; + +enum had_event_type { + HAD_EVENT_HOT_PLUG = 1, + HAD_EVENT_HOT_UNPLUG, + HAD_EVENT_MODE_CHANGING, + HAD_EVENT_AUDIO_BUFFER_DONE, + HAD_EVENT_AUDIO_BUFFER_UNDERRUN, + HAD_EVENT_QUERY_IS_AUDIO_BUSY, + HAD_EVENT_QUERY_IS_AUDIO_SUSPENDED, +}; + +/* + * HDMI Display Controller Audio Interface + * + */ +typedef int (*had_event_call_back) (enum had_event_type event_type, + void *ctxt_info); + +struct hdmi_audio_registers_ops { + int (*hdmi_audio_get_register_base)(uint32_t **reg_base, + uint32_t *config_offset); + int (*hdmi_audio_read_register)(uint32_t reg_addr, uint32_t *data); + int (*hdmi_audio_write_register)(uint32_t reg_addr, uint32_t data); + int (*hdmi_audio_read_modify)(uint32_t reg_addr, uint32_t data, + uint32_t mask); +}; + +struct hdmi_audio_query_set_ops { + int (*hdmi_audio_get_caps)(enum had_caps_list query_element, + void *capabilties); + int (*hdmi_audio_set_caps)(enum had_caps_list set_element, + void *capabilties); +}; + +struct hdmi_audio_event { + int type; +}; + +struct snd_intel_had_interface { + const char *name; + int (*query)(void *had_data, struct hdmi_audio_event event); + int (*suspend)(void *had_data, struct hdmi_audio_event event); + int (*resume)(void *had_data); +}; + +bool mid_hdmi_audio_is_busy(void *dev); +bool mid_hdmi_audio_suspend(void *dev); +void mid_hdmi_audio_resume(void *dev); +void mid_hdmi_audio_signal_event(enum had_event_type event); +int mid_hdmi_audio_setup( + had_event_call_back audio_callbacks, + struct hdmi_audio_registers_ops *reg_ops, + struct hdmi_audio_query_set_ops *query_ops); +int mid_hdmi_audio_register( + struct snd_intel_had_interface *driver, + void *had_data); + +#endif -- 2.9.3

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[alsa-devel] [RFC PATCH v4 2/7] drm/i915: Add support for audio driver notifications
by Jerome Anand 02 Dec '16

02 Dec '16

Notifiations like mode change, hot plug and edid to the audio driver are added. This is inturn used by the audio driver for its functionality. A new interface file capturing the notifications needed by the audio driver is added Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Jerome Anand <jerome.anand(a)intel.com> --- drivers/gpu/drm/i915/i915_drv.h | 3 +++ drivers/gpu/drm/i915/intel_audio.c | 8 ++++++ drivers/gpu/drm/i915/intel_hdmi.c | 1 + drivers/gpu/drm/i915/intel_lpe_audio.c | 46 ++++++++++++++++++++++++++++++++++ include/drm/intel_lpe_audio.h | 1 + 5 files changed, 59 insertions(+) diff --git a/drivers/gpu/drm/i915/i915_drv.h b/drivers/gpu/drm/i915/i915_drv.h index 2a79048..33bc44c 100644 --- a/drivers/gpu/drm/i915/i915_drv.h +++ b/drivers/gpu/drm/i915/i915_drv.h @@ -3561,6 +3561,9 @@ int intel_lpe_audio_setup(struct drm_i915_private *dev_priv); void intel_lpe_audio_teardown(struct drm_i915_private *dev_priv); void intel_lpe_audio_irq_handler(struct drm_i915_private *dev_priv); bool intel_lpe_audio_detect(struct drm_i915_private *dev_priv); +void intel_lpe_audio_notify(struct drm_i915_private *dev_priv, + void *eld, int port, int tmds_clk_speed, + bool connected); /* intel_i2c.c */ extern int intel_setup_gmbus(struct drm_device *dev); diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index 1c509f7..55a6831 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -24,6 +24,7 @@ #include <linux/kernel.h> #include <linux/component.h> #include <drm/i915_component.h> +#include <drm/intel_lpe_audio.h> #include "intel_drv.h" #include <drm/drmP.h> @@ -627,6 +628,10 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder, if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, (int) port, (int) pipe); + + if (HAS_LPE_AUDIO(dev_priv)) + intel_lpe_audio_notify(dev_priv, connector->eld, port, + crtc_state->port_clock, true); } /** @@ -660,6 +665,9 @@ void intel_audio_codec_disable(struct intel_encoder *intel_encoder) if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, (int) port, (int) pipe); + + if (HAS_LPE_AUDIO(dev_priv)) + intel_lpe_audio_notify(dev_priv, NULL, port, 0, true); } /** diff --git a/drivers/gpu/drm/i915/intel_hdmi.c b/drivers/gpu/drm/i915/intel_hdmi.c index fb88e32..02d50e3 100644 --- a/drivers/gpu/drm/i915/intel_hdmi.c +++ b/drivers/gpu/drm/i915/intel_hdmi.c @@ -36,6 +36,7 @@ #include <drm/drm_edid.h> #include "intel_drv.h" #include <drm/i915_drm.h> +#include <drm/intel_lpe_audio.h> #include "i915_drv.h" static struct drm_device *intel_hdmi_to_dev(struct intel_hdmi *intel_hdmi) diff --git a/drivers/gpu/drm/i915/intel_lpe_audio.c b/drivers/gpu/drm/i915/intel_lpe_audio.c index e12e5f7..a141a9c 100644 --- a/drivers/gpu/drm/i915/intel_lpe_audio.c +++ b/drivers/gpu/drm/i915/intel_lpe_audio.c @@ -361,3 +361,49 @@ void intel_lpe_audio_teardown(struct drm_i915_private *dev_priv) spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); } + + +/** + * intel_lpe_audio_notify() - notify lpe audio event + * audio driver and i915 + * @dev_priv: the i915 drm device private data + * @eld : ELD data + * @port: port id + * @tmds_clk_speed: tmds clock frequency in Hz + * @connected: hdmi connected/disconnected + * + * Notify lpe audio driver of eld change. + */ +void intel_lpe_audio_notify(struct drm_i915_private *dev_priv, + void *eld, int port, int tmds_clk_speed, + bool connected) +{ + unsigned long irq_flags; + + if (HAS_LPE_AUDIO(dev_priv)) { + struct intel_hdmi_lpe_audio_pdata *pdata = dev_get_platdata( + &(dev_priv->lpe_audio.platdev->dev)); + + spin_lock_irqsave(&pdata->lpe_audio_slock, + irq_flags); + + if (eld != NULL) { + memcpy(pdata->eld.eld_data, eld, + HDMI_MAX_ELD_BYTES); + pdata->eld.port_id = port; + + if (tmds_clk_speed) + pdata->tmds_clock_speed = + tmds_clk_speed; + } + pdata->hdmi_connected = connected; + if (pdata->notify_audio_lpe) + pdata->notify_audio_lpe( + (eld != NULL) ? &pdata->eld : NULL); + else + pdata->notify_pending = true; + + spin_unlock_irqrestore(&pdata->lpe_audio_slock, + irq_flags); + } +} diff --git a/include/drm/intel_lpe_audio.h b/include/drm/intel_lpe_audio.h index a64c449..952de05 100644 --- a/include/drm/intel_lpe_audio.h +++ b/include/drm/intel_lpe_audio.h @@ -25,6 +25,7 @@ #define _INTEL_LPE_AUDIO_H_ #include <linux/types.h> +#include <linux/spinlock_types.h> #define HDMI_MAX_ELD_BYTES 128 -- 2.9.3

1 0

[alsa-devel] [RFC PATCH v4 1/7] drm/i915: setup bridge for HDMI LPE audio driver
by Jerome Anand 02 Dec '16

02 Dec '16

Enable support for HDMI LPE audio mode on Baytrail and Cherrytrail when HDaudio controller is not detected Setup minimum required resources during i915_driver_load: 1. Create a platform device to share MMIO/IRQ resources 2. Make the platform device child of i915 device for runtime PM. 3. Create IRQ chip to forward HDMI LPE audio irqs. HDMI LPE audio driver (a standalone sound driver) probes the LPE audio device and creates a new sound card. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Jerome Anand <jerome.anand(a)intel.com> --- Documentation/gpu/i915.rst | 9 + drivers/gpu/drm/i915/Makefile | 3 + drivers/gpu/drm/i915/i915_drv.c | 8 +- drivers/gpu/drm/i915/i915_drv.h | 15 ++ drivers/gpu/drm/i915/i915_irq.c | 27 +++ drivers/gpu/drm/i915/i915_reg.h | 3 + drivers/gpu/drm/i915/intel_lpe_audio.c | 363 +++++++++++++++++++++++++++++++++ include/drm/intel_lpe_audio.h | 45 ++++ 8 files changed, 471 insertions(+), 2 deletions(-) create mode 100644 drivers/gpu/drm/i915/intel_lpe_audio.c create mode 100644 include/drm/intel_lpe_audio.h diff --git a/Documentation/gpu/i915.rst b/Documentation/gpu/i915.rst index 117d2ab..a671eee 100644 --- a/Documentation/gpu/i915.rst +++ b/Documentation/gpu/i915.rst @@ -213,6 +213,15 @@ Video BIOS Table (VBT) .. kernel-doc:: drivers/gpu/drm/i915/intel_vbt_defs.h :internal: +intel hdmi lpe audio support +---------------------------- + +.. kernel-doc:: drivers/gpu/drm/i915/intel_lpe_audio.c + :doc: LPE Audio integration for HDMI or DP playback + +.. kernel-doc:: drivers/gpu/drm/i915/intel_lpe_audio.c + :internal: + Memory Management and Command Submission ======================================== diff --git a/drivers/gpu/drm/i915/Makefile b/drivers/gpu/drm/i915/Makefile index 580602d..3a2369c 100644 --- a/drivers/gpu/drm/i915/Makefile +++ b/drivers/gpu/drm/i915/Makefile @@ -126,6 +126,9 @@ i915-y += intel_gvt.o include $(src)/gvt/Makefile endif +# LPE Audio for VLV and CHT +i915-y += intel_lpe_audio.o + obj-$(CONFIG_DRM_I915) += i915.o CFLAGS_i915_trace_points.o := -I$(src) diff --git a/drivers/gpu/drm/i915/i915_drv.c b/drivers/gpu/drm/i915/i915_drv.c index b893e67..93811ed 100644 --- a/drivers/gpu/drm/i915/i915_drv.c +++ b/drivers/gpu/drm/i915/i915_drv.c @@ -1144,7 +1144,8 @@ static void i915_driver_register(struct drm_i915_private *dev_priv) if (IS_GEN5(dev_priv)) intel_gpu_ips_init(dev_priv); - i915_audio_component_init(dev_priv); + if (intel_lpe_audio_init(dev_priv) < 0) + i915_audio_component_init(dev_priv); /* * Some ports require correctly set-up hpd registers for detection to @@ -1162,7 +1163,10 @@ static void i915_driver_register(struct drm_i915_private *dev_priv) */ static void i915_driver_unregister(struct drm_i915_private *dev_priv) { - i915_audio_component_cleanup(dev_priv); + if (HAS_LPE_AUDIO(dev_priv)) + intel_lpe_audio_teardown(dev_priv); + else + i915_audio_component_cleanup(dev_priv); intel_gpu_ips_teardown(); acpi_video_unregister(); diff --git a/drivers/gpu/drm/i915/i915_drv.h b/drivers/gpu/drm/i915/i915_drv.h index 970e50b..2a79048 100644 --- a/drivers/gpu/drm/i915/i915_drv.h +++ b/drivers/gpu/drm/i915/i915_drv.h @@ -2284,6 +2284,12 @@ struct drm_i915_private { /* Used to save the pipe-to-encoder mapping for audio */ struct intel_encoder *av_enc_map[I915_MAX_PIPES]; + /* necessary resource sharing with HDMI LPE audio driver. */ + struct { + struct platform_device *platdev; + int irq; + } lpe_audio; + /* * NOTE: This is the dri1/ums dungeon, don't add stuff here. Your patch * will be rejected. Instead look for a better place. @@ -2773,6 +2779,8 @@ intel_info(const struct drm_i915_private *dev_priv) #define HAS_POOLED_EU(dev_priv) ((dev_priv)->info.has_pooled_eu) +#define HAS_LPE_AUDIO(dev_priv) ((dev_priv)->lpe_audio.platdev != NULL) + #define INTEL_PCH_DEVICE_ID_MASK 0xff00 #define INTEL_PCH_IBX_DEVICE_ID_TYPE 0x3b00 #define INTEL_PCH_CPT_DEVICE_ID_TYPE 0x1c00 @@ -3547,6 +3555,13 @@ extern int i915_restore_state(struct drm_device *dev); void i915_setup_sysfs(struct drm_i915_private *dev_priv); void i915_teardown_sysfs(struct drm_i915_private *dev_priv); +/* i915_lpe_audio.c */ +int intel_lpe_audio_init(struct drm_i915_private *dev_priv); +int intel_lpe_audio_setup(struct drm_i915_private *dev_priv); +void intel_lpe_audio_teardown(struct drm_i915_private *dev_priv); +void intel_lpe_audio_irq_handler(struct drm_i915_private *dev_priv); +bool intel_lpe_audio_detect(struct drm_i915_private *dev_priv); + /* intel_i2c.c */ extern int intel_setup_gmbus(struct drm_device *dev); extern void intel_teardown_gmbus(struct drm_device *dev); diff --git a/drivers/gpu/drm/i915/i915_irq.c b/drivers/gpu/drm/i915/i915_irq.c index 07ca71c..6f710da 100644 --- a/drivers/gpu/drm/i915/i915_irq.c +++ b/drivers/gpu/drm/i915/i915_irq.c @@ -1894,6 +1894,14 @@ static irqreturn_t valleyview_irq_handler(int irq, void *arg) valleyview_pipestat_irq_ack(dev_priv, iir, pipe_stats); /* + * LPE audio interrupts are only enabled on Baytrail and + * CherryView platforms without HDaudio + */ + if (iir & (I915_LPE_PIPE_A_INTERRUPT | + I915_LPE_PIPE_B_INTERRUPT)) + intel_lpe_audio_irq_handler(dev_priv); + + /* * VLV_IIR is single buffered, and reflects the level * from PIPESTAT/PORT_HOTPLUG_STAT, hence clear it last. */ @@ -1974,6 +1982,15 @@ static irqreturn_t cherryview_irq_handler(int irq, void *arg) valleyview_pipestat_irq_ack(dev_priv, iir, pipe_stats); /* + * LPE audio interrupts are only enabled on Baytrail and + * CherryView platforms without HDaudio + */ + if (iir & (I915_LPE_PIPE_A_INTERRUPT | + I915_LPE_PIPE_B_INTERRUPT | + I915_LPE_PIPE_C_INTERRUPT)) + intel_lpe_audio_irq_handler(dev_priv); + + /* * VLV_IIR is single buffered, and reflects the level * from PIPESTAT/PORT_HOTPLUG_STAT, hence clear it last. */ @@ -2914,6 +2931,7 @@ static void vlv_display_irq_postinstall(struct drm_i915_private *dev_priv) u32 pipestat_mask; u32 enable_mask; enum pipe pipe; + u32 val; pipestat_mask = PLANE_FLIP_DONE_INT_STATUS_VLV | PIPE_CRC_DONE_INTERRUPT_STATUS; @@ -2930,6 +2948,15 @@ static void vlv_display_irq_postinstall(struct drm_i915_private *dev_priv) WARN_ON(dev_priv->irq_mask != ~0); + /* add interrupt masks unconditially here, the actual unmask + * will take place only if the LPE_AUDIO mode is detected + */ + val = (I915_LPE_PIPE_A_INTERRUPT | + I915_LPE_PIPE_B_INTERRUPT | + I915_LPE_PIPE_C_INTERRUPT); + + enable_mask |= val; + dev_priv->irq_mask = ~enable_mask; GEN5_IRQ_INIT(VLV_, dev_priv->irq_mask, enable_mask); diff --git a/drivers/gpu/drm/i915/i915_reg.h b/drivers/gpu/drm/i915/i915_reg.h index fcad8fa..6b16ed0 100644 --- a/drivers/gpu/drm/i915/i915_reg.h +++ b/drivers/gpu/drm/i915/i915_reg.h @@ -2395,6 +2395,9 @@ enum skl_disp_power_wells { #define I915_ASLE_INTERRUPT (1<<0) #define I915_BSD_USER_INTERRUPT (1<<25) +#define I915_HDMI_LPE_AUDIO_BASE (VLV_DISPLAY_BASE + 0x65000) +#define I915_HDMI_LPE_AUDIO_SIZE 0x1000 + #define GEN6_BSD_RNCID _MMIO(0x12198) #define GEN7_FF_THREAD_MODE _MMIO(0x20a0) diff --git a/drivers/gpu/drm/i915/intel_lpe_audio.c b/drivers/gpu/drm/i915/intel_lpe_audio.c new file mode 100644 index 0000000..e12e5f7 --- /dev/null +++ b/drivers/gpu/drm/i915/intel_lpe_audio.c @@ -0,0 +1,363 @@ +/* + * Copyright © 2016 Intel Corporation + * + * Permission is hereby granted, free of charge, to any person obtaining a + * copy of this software and associated documentation files (the "Software"), + * to deal in the Software without restriction, including without limitation + * the rights to use, copy, modify, merge, publish, distribute, sublicense, + * and/or sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice (including the next + * paragraph) shall be included in all copies or substantial portions of the + * Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS + * IN THE SOFTWARE. + * + * Authors: + * Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> + * Jerome Anand <jerome.anand(a)intel.com> + * based on VED patches + * + */ + +/** + * DOC: LPE Audio integration for HDMI or DP playback + * + * Motivation: + * Atom platforms (e.g. valleyview and cherryTrail) integrates a DMA-based + * interface as an alternative to the traditional HDaudio path. While this + * mode is unrelated to the LPE aka SST audio engine, the documentation refers + * to this mode as LPE so we keep this notation for the sake of consistency. + * + * The interface is handled by a separate standalone driver maintained in the + * ALSA subsystem for simplicity. To minimize the interaction between the two + * subsystems, a bridge is setup between the hdmi-lpe-audio and i915: + * 1. Create a platform device to share MMIO/IRQ resources + * 2. Make the platform device child of i915 device for runtime PM. + * 3. Create IRQ chip to forward the LPE audio irqs. + * the hdmi-lpe-audio driver probes the lpe audio device and creates a new + * sound card + * + * Threats: + * Due to the restriction in Linux platform device model, user need manually + * uninstall the hdmi-lpe-audio driver before uninstalling i915 module, + * otherwise we might run into use-after-free issues after i915 removes the + * platform device: even though hdmi-lpe-audio driver is released, the modules + * is still in "installed" status. + * + * Implementation: + * The MMIO/REG platform resources are created according to the registers + * specification. + * When forwarding LPE audio irqs, the flow control handler selection depends + * on the platform, for example on valleyview handle_simple_irq is enough. + * + */ + +#include <linux/acpi.h> +#include <linux/device.h> +#include <linux/pci.h> + +#include "i915_drv.h" +#include <linux/delay.h> +#include <drm/intel_lpe_audio.h> + +static struct platform_device* +lpe_audio_platdev_create(struct drm_i915_private *dev_priv) +{ + struct drm_device *dev = &dev_priv->drm; + int ret; + struct resource rsc[2] = {}; + struct platform_device *platdev; + u64 *dma_mask; + struct intel_hdmi_lpe_audio_pdata *pdata; + + if (dev_priv->lpe_audio.irq < 0) + return ERR_PTR(-EINVAL); + + platdev = platform_device_alloc("hdmi-lpe-audio", -1); + if (!platdev) { + ret = -ENOMEM; + DRM_ERROR("Failed to allocate LPE audio platform device\n"); + goto err; + } + + /* to work-around check_addr in nommu_map_sg() */ + dma_mask = kmalloc(sizeof(*platdev->dev.dma_mask), GFP_KERNEL); + if (!dma_mask) { + ret = -ENOMEM; + DRM_ERROR("Failed to allocate dma_mask\n"); + goto err_put_dev; + } + *dma_mask = DMA_BIT_MASK(32); + platdev->dev.dma_mask = dma_mask; + platdev->dev.coherent_dma_mask = *dma_mask; + + rsc[0].start = rsc[0].end = dev_priv->lpe_audio.irq; + rsc[0].flags = IORESOURCE_IRQ; + rsc[0].name = "hdmi-lpe-audio-irq"; + + rsc[1].start = pci_resource_start(dev->pdev, 0) + + I915_HDMI_LPE_AUDIO_BASE; + rsc[1].end = pci_resource_start(dev->pdev, 0) + + I915_HDMI_LPE_AUDIO_BASE + I915_HDMI_LPE_AUDIO_SIZE - 1; + rsc[1].flags = IORESOURCE_MEM; + rsc[1].name = "hdmi-lpe-audio-mmio"; + + ret = platform_device_add_resources(platdev, rsc, 2); + if (ret) { + DRM_ERROR("Failed to add resource for platform device: %d\n", + ret); + goto err_put_dev; + } + + pdata = kzalloc(sizeof(*pdata), GFP_KERNEL); + + if (pdata == NULL) { + ret = -ENOMEM; + goto err_put_dev; + } + platdev->dev.platform_data = pdata; + spin_lock_init(&pdata->lpe_audio_slock); + + /* for LPE audio driver's runtime-PM */ + platdev->dev.parent = dev->dev; + ret = platform_device_add(platdev); + if (ret) { + DRM_ERROR("Failed to add LPE audio platform device: %d\n", + ret); + goto err_put_dev; + } + + return platdev; +err_put_dev: + platform_device_put(platdev); + kfree(dma_mask); +err: + return ERR_PTR(ret); +} + +static void lpe_audio_platdev_destroy(struct drm_i915_private *dev_priv) +{ + platform_device_unregister(dev_priv->lpe_audio.platdev); + kfree(dev_priv->lpe_audio.platdev->dev.dma_mask); +} + +static void lpe_audio_irq_unmask(struct irq_data *d) +{ + struct drm_i915_private *dev_priv = d->chip_data; + unsigned long irqflags; + u32 val = (I915_LPE_PIPE_A_INTERRUPT | + I915_LPE_PIPE_B_INTERRUPT); + + spin_lock_irqsave(&dev_priv->irq_lock, irqflags); + + /* + * VLV_IER is already set in the vlv_display_postinstall(), + * we only change VLV_IIR and VLV_IMR + */ + dev_priv->irq_mask &= ~val; + I915_WRITE(VLV_IIR, val); + I915_WRITE(VLV_IIR, val); + I915_WRITE(VLV_IMR, dev_priv->irq_mask); + POSTING_READ(VLV_IMR); + + spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); +} + +static void lpe_audio_irq_mask(struct irq_data *d) +{ + struct drm_i915_private *dev_priv = d->chip_data; + unsigned long irqflags; + u32 val = (I915_LPE_PIPE_A_INTERRUPT | + I915_LPE_PIPE_B_INTERRUPT); + + spin_lock_irqsave(&dev_priv->irq_lock, irqflags); + + /* + * VLV_IER is already set in the vlv_display_postinstall(), + * we only change VLV_IIR and VLV_IMR + */ + dev_priv->irq_mask |= val; + I915_WRITE(VLV_IMR, dev_priv->irq_mask); + I915_WRITE(VLV_IIR, val); + I915_WRITE(VLV_IIR, val); + POSTING_READ(VLV_IIR); + + spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); +} + +static struct irq_chip lpe_audio_irqchip = { + .name = "hdmi_lpe_audio_irqchip", + .irq_mask = lpe_audio_irq_mask, + .irq_unmask = lpe_audio_irq_unmask, +}; + +static int lpe_audio_irq_init(struct drm_i915_private *dev_priv) +{ + int irq = dev_priv->lpe_audio.irq; + + WARN_ON(!intel_irqs_enabled(dev_priv)); + irq_set_chip_and_handler_name(irq, + &lpe_audio_irqchip, + handle_simple_irq, + "hdmi_lpe_audio_irq_handler"); + + return irq_set_chip_data(irq, dev_priv); +} + +/** + * intel_lpe_audio_irq_handler() - forwards the LPE audio irq + * @dev_priv: the i915 drm device private data + * + * the LPE Audio irq is forwarded to the irq handler registered by LPE audio + * driver. + */ +void intel_lpe_audio_irq_handler(struct drm_i915_private *dev_priv) +{ + int ret; + + ret = generic_handle_irq(dev_priv->lpe_audio.irq); + if (ret) + DRM_ERROR_RATELIMITED("error handling LPE audio irq: %d\n", + ret); +} + +/** + * intel_lpe_audio_detect() - check & setup lpe audio if present + * @dev_priv: the i915 drm device private data + * + * Detect if lpe audio is present + * + * Return: true if lpe audio present else Return = false + */ +bool intel_lpe_audio_detect(struct drm_i915_private *dev_priv) +{ + int lpe_present = false; + + if (IS_VALLEYVIEW(dev_priv) || IS_CHERRYVIEW(dev_priv)) { + static const struct pci_device_id atom_hdaudio_ids[] = { + /* Baytrail */ + {PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x0f04)}, + /* Braswell */ + {PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x2284)}, + {} + }; + + if (!pci_dev_present(atom_hdaudio_ids)) { + DRM_INFO("%s%s\n", "HDaudio controller not detected,", + "using LPE audio instead\n"); + lpe_present = true; + } + } + return lpe_present; +} + +/** + * intel_lpe_audio_setup() - setup the bridge between HDMI LPE Audio + * driver and i915 + * @dev_priv: the i915 drm device private data + * + * set up the minimum required resources for the bridge: irq chip, + * platform resource and platform device. i915 device is set as parent + * of the new platform device. + * + * Return: 0 if successful. non-zero if allocation/initialization fails + */ +int intel_lpe_audio_setup(struct drm_i915_private *dev_priv) +{ + int ret; + + dev_priv->lpe_audio.irq = irq_alloc_desc(0); + if (dev_priv->lpe_audio.irq < 0) { + DRM_ERROR("Failed to allocate IRQ desc: %d\n", + dev_priv->lpe_audio.irq); + ret = dev_priv->lpe_audio.irq; + goto err; + } + + DRM_DEBUG("irq = %d\n", dev_priv->lpe_audio.irq); + + ret = lpe_audio_irq_init(dev_priv); + + if (ret) { + DRM_ERROR("Failed to initialize irqchip for lpe audio: %d\n", + ret); + goto err_free_irq; + } + + dev_priv->lpe_audio.platdev = lpe_audio_platdev_create(dev_priv); + + if (IS_ERR(dev_priv->lpe_audio.platdev)) { + ret = PTR_ERR(dev_priv->lpe_audio.platdev); + DRM_ERROR("Failed to create lpe audio platform device: %d\n", + ret); + goto err_free_irq; + } + + return 0; +err_free_irq: + irq_free_desc(dev_priv->lpe_audio.irq); +err: + dev_priv->lpe_audio.irq = -1; + dev_priv->lpe_audio.platdev = NULL; + return ret; +} + + +/** + * intel_lpe_audio_init() - detect and setup the bridge between HDMI LPE Audio + * driver and i915 + * @dev_priv: the i915 drm device private data + * + * Return: 0 if successful. non-zero if detection or + * llocation/initialization fails + */ +int intel_lpe_audio_init(struct drm_i915_private *dev_priv) +{ + int ret = -ENODEV; + + if (intel_lpe_audio_detect(dev_priv)) { + ret = intel_lpe_audio_setup(dev_priv); + if (ret < 0) + DRM_ERROR("failed to setup LPE Audio bridge\n"); + } + return ret; +} + +/** + * intel_lpe_audio_teardown() - destroy the bridge between HDMI LPE + * audio driver and i915 + * @dev_priv: the i915 drm device private data + * + * release all the resources for LPE audio <-> i915 bridge. + */ +void intel_lpe_audio_teardown(struct drm_i915_private *dev_priv) +{ + unsigned long irqflags; + struct irq_desc *desc; + + if (!HAS_LPE_AUDIO(dev_priv)) + return; + + desc = irq_to_desc(dev_priv->lpe_audio.irq); + + /** + * mask LPE audio irq before destroying + */ + lpe_audio_irq_mask(&desc->irq_data); + + spin_lock_irqsave(&dev_priv->irq_lock, irqflags); + + lpe_audio_platdev_destroy(dev_priv); + + irq_free_desc(dev_priv->lpe_audio.irq); + + spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); +} diff --git a/include/drm/intel_lpe_audio.h b/include/drm/intel_lpe_audio.h new file mode 100644 index 0000000..a64c449 --- /dev/null +++ b/include/drm/intel_lpe_audio.h @@ -0,0 +1,45 @@ +/* + * Copyright © 2016 Intel Corporation + * + * Permission is hereby granted, free of charge, to any person obtaining a + * copy of this software and associated documentation files (the "Software"), + * to deal in the Software without restriction, including without limitation + * the rights to use, copy, modify, merge, publish, distribute, sublicense, + * and/or sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice (including the next + * paragraph) shall be included in all copies or substantial portions of the + * Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS + * IN THE SOFTWARE. + */ + +#ifndef _INTEL_LPE_AUDIO_H_ +#define _INTEL_LPE_AUDIO_H_ + +#include <linux/types.h> + +#define HDMI_MAX_ELD_BYTES 128 + +struct intel_hdmi_lpe_audio_eld { + int port_id; + unsigned char eld_data[HDMI_MAX_ELD_BYTES]; +}; + +struct intel_hdmi_lpe_audio_pdata { + bool notify_pending; + int tmds_clock_speed; + bool hdmi_connected; + struct intel_hdmi_lpe_audio_eld eld; + void (*notify_audio_lpe)(void *audio_ptr); + spinlock_t lpe_audio_slock; +}; + +#endif /* _I915_LPE_AUDIO_H_ */ -- 2.9.3

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[alsa-devel] [RFC PATCH v4 0/7] Add support for Legacy HDMI audio drivers
by Jerome Anand 02 Dec '16

02 Dec '16

Legacy (CherryTrail/ Baytrail) HDMI audio drivers added Legacy hdmi audio-Gfx interaction/ interfacing is updated to use irq chip framework Jerome Anand (7): drm/i915: setup bridge for HDMI LPE audio driver drm/i915: Add support for audio driver notifications ALSA: add shell for Intel HDMI LPE audio driver ALSA: x86: hdmi: Add audio support for BYT and CHT ALSA: x86: hdmi: Improve position reporting ALSA: x86: hdmi: Fixup some monitor ALSA: x86: hdmi: continue playback even when display resolution changes Documentation/gpu/i915.rst | 9 + drivers/gpu/drm/i915/Makefile | 3 + drivers/gpu/drm/i915/i915_drv.c | 8 +- drivers/gpu/drm/i915/i915_drv.h | 18 + drivers/gpu/drm/i915/i915_irq.c | 27 + drivers/gpu/drm/i915/i915_reg.h | 3 + drivers/gpu/drm/i915/intel_audio.c | 8 + drivers/gpu/drm/i915/intel_hdmi.c | 1 + drivers/gpu/drm/i915/intel_lpe_audio.c | 409 +++++++ include/drm/intel_lpe_audio.h | 46 + sound/Kconfig | 2 + sound/Makefile | 2 +- sound/x86/Kconfig | 16 + sound/x86/Makefile | 10 + sound/x86/intel_hdmi_audio.c | 1932 ++++++++++++++++++++++++++++++++ sound/x86/intel_hdmi_audio.h | 201 ++++ sound/x86/intel_hdmi_audio_if.c | 551 +++++++++ sound/x86/intel_hdmi_lpe_audio.c | 626 +++++++++++ sound/x86/intel_hdmi_lpe_audio.h | 692 ++++++++++++ 19 files changed, 4561 insertions(+), 3 deletions(-) create mode 100644 drivers/gpu/drm/i915/intel_lpe_audio.c create mode 100644 include/drm/intel_lpe_audio.h create mode 100644 sound/x86/Kconfig create mode 100644 sound/x86/Makefile create mode 100644 sound/x86/intel_hdmi_audio.c create mode 100644 sound/x86/intel_hdmi_audio.h create mode 100644 sound/x86/intel_hdmi_audio_if.c create mode 100644 sound/x86/intel_hdmi_lpe_audio.c create mode 100644 sound/x86/intel_hdmi_lpe_audio.h -- 2.9.3 base-commit: b34f27e00c9a21d2078062da760e24fd4a620391

1 0

[alsa-devel] [PATCH] ALSA: core: update supported sample rates
by Li Xu 02 Dec '16

02 Dec '16

Add 12KHz, 24KHz, 352.8KHz and 384KHz audio sample rates Signed-off-by: Li Xu <li.xu(a)cirrus.com> --- include/sound/pcm.h | 27 +++++++++++++++++---------- sound/core/pcm_native.c | 7 ++++--- 2 files changed, 21 insertions(+), 13 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index af1fb37..3ce4ad9 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -119,16 +119,20 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_5512 (1<<0) /* 5512Hz */ #define SNDRV_PCM_RATE_8000 (1<<1) /* 8000Hz */ #define SNDRV_PCM_RATE_11025 (1<<2) /* 11025Hz */ -#define SNDRV_PCM_RATE_16000 (1<<3) /* 16000Hz */ -#define SNDRV_PCM_RATE_22050 (1<<4) /* 22050Hz */ -#define SNDRV_PCM_RATE_32000 (1<<5) /* 32000Hz */ -#define SNDRV_PCM_RATE_44100 (1<<6) /* 44100Hz */ -#define SNDRV_PCM_RATE_48000 (1<<7) /* 48000Hz */ -#define SNDRV_PCM_RATE_64000 (1<<8) /* 64000Hz */ -#define SNDRV_PCM_RATE_88200 (1<<9) /* 88200Hz */ -#define SNDRV_PCM_RATE_96000 (1<<10) /* 96000Hz */ -#define SNDRV_PCM_RATE_176400 (1<<11) /* 176400Hz */ -#define SNDRV_PCM_RATE_192000 (1<<12) /* 192000Hz */ +#define SNDRV_PCM_RATE_12000 (1<<3) /* 12000Hz */ +#define SNDRV_PCM_RATE_16000 (1<<4) /* 16000Hz */ +#define SNDRV_PCM_RATE_22050 (1<<5) /* 22050Hz */ +#define SNDRV_PCM_RATE_24000 (1<<6) /* 24000Hz */ +#define SNDRV_PCM_RATE_32000 (1<<7) /* 32000Hz */ +#define SNDRV_PCM_RATE_44100 (1<<8) /* 44100Hz */ +#define SNDRV_PCM_RATE_48000 (1<<9) /* 48000Hz */ +#define SNDRV_PCM_RATE_64000 (1<<10) /* 64000Hz */ +#define SNDRV_PCM_RATE_88200 (1<<11) /* 88200Hz */ +#define SNDRV_PCM_RATE_96000 (1<<12) /* 96000Hz */ +#define SNDRV_PCM_RATE_176400 (1<<13) /* 176400Hz */ +#define SNDRV_PCM_RATE_192000 (1<<14) /* 192000Hz */ +#define SNDRV_PCM_RATE_352800 (1<<15) /* 352800Hz */ +#define SNDRV_PCM_RATE_384000 (1<<16) /* 384000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1<<30) /* continuous range */ #define SNDRV_PCM_RATE_KNOT (1<<31) /* supports more non-continuos rates */ @@ -141,6 +145,9 @@ struct snd_pcm_ops { SNDRV_PCM_RATE_88200|SNDRV_PCM_RATE_96000) #define SNDRV_PCM_RATE_8000_192000 (SNDRV_PCM_RATE_8000_96000|SNDRV_PCM_RATE_176400|\ SNDRV_PCM_RATE_192000) +#define SNDRV_PCM_RATE_8000_384000 (SNDRV_PCM_RATE_8000_192000|\ + SNDRV_PCM_RATE_384000) + #define _SNDRV_PCM_FMTBIT(fmt) (1ULL << (__force int)SNDRV_PCM_FORMAT_##fmt) #define SNDRV_PCM_FMTBIT_S8 _SNDRV_PCM_FMTBIT(S8) #define SNDRV_PCM_FMTBIT_U8 _SNDRV_PCM_FMTBIT(U8) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 9d33c1e..9bc1011 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1981,12 +1981,13 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 +#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_384000 != 1 << 16 #error "Change this table" #endif -static unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 }; +static unsigned int rates[] = { 5512, 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000, 64000, 88200, + 96000, 176400, 192000, 352800, 384000 }; const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { .count = ARRAY_SIZE(rates), -- 1.9.1

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[alsa-devel] [PATCH v3 0/3] ucm: Add support for component devices
by mengdong.lin＠linux.intel.com 02 Dec '16

02 Dec '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Sound cards are defined by machines. And off-soc codecs and DSPs embedded in DSP can be taken as the components and resued by different machines/ sound cards. This series allows codec and SOC vendors to install UCM configuration files for a specific codec or DSP, and sound card verb files can include these conf files of components. Component devices will not be exposed to applications for backward compatibility. So audio servers like PulseAudio and CRAS still only see the machine devices. History: v2: Use an array to define component directories. Document syntax to enable/disable a component device. Hide cdev defined by the parent device in ucm manager for executing a component sequence. v3: No longer use macros for UCM manager to enter/exit component domain, to make code simpler. Mengdong Lin (3): ucm: Skip component directories when scanning sound card configuration files ucm: Parse sequence of component devices ucm: Execute sequence of component devices src/ucm/main.c | 78 +++++++++++++++++++++++++- src/ucm/parser.c | 154 +++++++++++++++++++++++++++++++++++++++++++++++++++- src/ucm/ucm_local.h | 19 +++++++ src/ucm/utils.c | 1 + 4 files changed, 249 insertions(+), 3 deletions(-) -- 2.7.4

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[alsa-devel] [alsa-utils] alsaucm should come with a man page
by Antonio Ospite 02 Dec '16

02 Dec '16

Hi, I used alsaucm to test mixer use cases for an Intel Baytrail tablet, I wanted to use bare alsa before bringing pulseaudio in, and it took some time to get my head around it. The output from the "alsaucm --help" is too terse, I had to look at the alsa-lib code[1] to see the possible IDENTIFIERs, and even then it was not clear to me that some operations, like for instance "list _devices", can only work after setting a _verb in the same _execution_, for instance this sequence doesn't work: # alsaucm -c bytcr-rt5640 set _verb HiFi # alsaucm -c bytcr-rt5640 list _devices But this command does: # alsaucm -c bytcr-rt5640 -b - <<EOM set _verb HiFi list _devces EOM Now that I know something more it makes sense: UCM just operates on mixer settings, it does not have its own external state. However, my point is that a man page explaining these things can make the life easier to new users. It would be better if alsa/UCM developers wrote the man page, but if no one steps up I guess I could draft a first version myself. Thanks, Antonio [1] http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=include/use-case.h -- Antonio Ospite http://ao2.it A: Because it messes up the order in which people normally read text. See http://en.wikipedia.org/wiki/Posting_style Q: Why is top-posting such a bad thing?

3 9

[alsa-devel] [PATCH v5 1/2] ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem
by Sylwester Nawrocki 01 Dec '16

01 Dec '16

This patch adds DT binding documentation for Exnos5433 based TM2 and TM2E boards sound subsystem. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Changes since v4: - indentation changes. Changes since v2: - none. Changes since initial version: - dropped clocks, clock-names properties, instead properties from the CODEC node will be used, - property renames: 'samsung,model' -> 'model', 'samsung,i2s-controller' -> 'i2s-controller', 'samsung,speaker-amplifier' -> 'audio-amplifier', - added 'audio-codec' property. --- .../bindings/sound/samsung,tm2-audio.txt | 38 ++++++++++++++++++++++ 1 file changed, 38 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt new file mode 100644 index 0000000..94442e5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt @@ -0,0 +1,38 @@ +Samsung Exynos5433 TM2(E) audio complex with WM5110 codec + +Required properties: + + - compatible : "samsung,tm2-audio" + - model : the user-visible name of this sound complex + - audio-codec : the phandle of the wm5110 audio codec node, + as described in ../mfd/arizona.txt + - i2s-controller : the phandle of the I2S controller + - audio-amplifier : the phandle of the MAX98504 amplifier + - samsung,audio-routing : a list of the connections between audio components; + each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source; valid names for sources and sinks are the + WM5110's and MAX98504's pins and the jacks on the + board: HP, SPK, Main Mic, Sub Mic, Third Mic, + Headset Mic + - mic-bias-gpios : GPIO pin that enables the Main Mic bias regulator + + +Example: + +sound { + compatible = "samsung,tm2-audio"; + audio-codec = <&wm5110>; + i2s-controller = <&i2s0>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3 2 0>; + model = "wm5110"; + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "Main Mic", "MICBIAS2", + "IN1R", "Main Mic"; +}; -- 1.9.1

4 8

[alsa-devel] [PATCH v3 0/3][RFC] ASoC: Component has suspend/resume support
by Kuninori Morimoto 01 Dec '16

01 Dec '16

Hi Mark, Lars-Peter These patches are v3+RFC of Component level suspend/resume support. I'm waiting Vinod's cleanup patch which will remove "codec_dev_list" from Intel drivers. But until then, I would like to get review about v3 patch. This v3 patch-set is still modifing Intel's driver at this point, but final/official v3 patchset will doesn't have it, because it will be based on Vinod's cleanup patch. Kuninori Morimoto (3): ASoC: core: replace codec_dev_list to component_dev_list on Card ASoC: core: replace aux_comp_list to component_dev_list ASoC: add Component level suspend/resume include/sound/soc.h | 16 +++--- sound/soc/intel/boards/broadwell.c | 16 +++--- sound/soc/intel/boards/cht_bsw_rt5672.c | 16 +++--- sound/soc/soc-core.c | 90 ++++++++++++++++++++------------- 4 files changed, 85 insertions(+), 53 deletions(-) -- 1.9.1

2 7

[alsa-devel] [PATCH v4 0/4] ASoC: intel: atom: Add FW version information
by Sebastien Guiriec 01 Dec '16

01 Dec '16

This patch series is adding FW version information of SST. Version is both log as dev_info and sysfs entry. Update since V1: - sysfs entry is set to "FW not yet loaded" in case we do not have loaded the FW once Update since V2: - Add Vinod ack. https://www.spinics.net/lists/alsa-devel/msg56521.html Update since V3: - Add sysfs entry description. Sebastien Guiriec (4): ASoC: intel: atom: Add debug information related to FW version ASoC: Intel: atom: save FW version ASoC: Intel: atom: Add sysfs entry in order to store FW version Documentation: ABI: Add ASoC Intel SST Atom sysfs description Documentation/ABI/testing/sysfs-platform-sst-atom | 18 +++++++++++ sound/soc/intel/atom/sst/sst.c | 39 +++++++++++++++++++++++ sound/soc/intel/atom/sst/sst.h | 1 + sound/soc/intel/atom/sst/sst_ipc.c | 10 +++++- 4 files changed, 67 insertions(+), 1 deletion(-) create mode 100644 Documentation/ABI/testing/sysfs-platform-sst-atom -- 2.9.3 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

3 8

[alsa-devel] [PATCH v3 0/2] ALSA controls management using index/device/sub-devices fields
by Arnaud Pouliquen 01 Dec '16

01 Dec '16

V3: I changed "RFC" prefix to "PATCH", hoping that patchset is enough mature in term of code and discussions, to be considered as a potential patch... Previous discussions summary: - topic of the patchset: Patch concerns ASoc DAI drivers. Aim is to be able to instantiate PCM control using device field according to the associated PCM char device. For this, a relationship between DAIs PCM control and PCM char device needs to be establish. => Proposal is to add field in DAI driver struct to declare PCM controls that need to be linked to PCM character device on DAI link probing. - Limitation of the patchset: This patchset seems only a first step as 2 concerns are been highlighted: - Patch is limited to "static" DAI-link. This not responds to backend (no_pcm dai links) and some none-DAI codecs, that could register PCm controls. In theses cases it is not possible to establish relationship during probe. => No solution proposed for time beeing. - This patchset does not fix conflict of 2 "identical" PCM controls declared by 2 drivers. For instance, a conflict can exist between a PCM control created in DAI driver and the same control declared in a codec driver. As index field is forced to 0, prefix should be used in codec. => To discuss solution in a separate thread as issue already present without this patchset several approaches seem possible like using index field ( control auto indexation in ASoC) or prefix in Naming. Changes: - [PATCH v3 1/2] ASoC: core: allow DAI PCM controls bound to PCM device Code optimization based on Takashi Sakamoto suggestion - [RFC V2 3/3] ASoC: hdmi-codec: Example of PCM control bound to PCM device for multi patch suppressed as it was send as example in previous version. Patch is valid but can have impact on existing drivers as control device field value is updated - [PATCH v3 2/2] ASoC: sti: bind pcm controls to pcm device. No change vs V2 V2: http://www.spinics.net/lists/alsa-devel/msg57045.html Aim of this version is to continue discussion on DAI PCM control focused on ASoC drivers. In this V2 implementation in Soc-core is simplified to limit impact on existing code. Update of the RFC V1 based on discussions: - [RFC 4/4] iecset: allow to select control with device and sub-device numbers no more part of the RFC V2, will be discussed in a separate thread - [RFC 2/4] ALSA: control: increment index field for duplicated control. no more part of the RFC V2, no more need as RFC subject is PCM controls - [RFC V2 1/3] ASoC: core: allow DAI PCM controls bound to PCM device Patch reworked from V1 to simplify implementation - Binding is not done for Dai links tagged with no_pcm (DPCM). - no more possibility to add the controls after the DAI link probing. - [RFC V2 2/3] ASoC: sti: bind PCM controls to PCM device. - [RFC V2 3/3] ASoC: hdmi-codec: Example of PCM control bound to PCM device for multi Example of implementation in STI DAI driver and HDMI-codec drivers V1: http://www.spinics.net/lists/alsa-devel/msg56479.html 1) Alsa-utils patch - iecset: allow to select control with device and sub-device numbers This patch allows to access to 2 iec controls differentiated by device/sub-devices numbers => For me, this patch is mandatory to be able to address the ASoC IEC controls, in case of no fix is implemented to allows index field update in ASoC. 2) Alsa driver patches - ASoC: core: allow PCM control binding to PCM device Add relationship between DAIs PCM controls and PCM device. - ALSA: control: increment index field for duplicated control. Generic implementation of the patch proposed in HDA driver (http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=ea9b…) - ASoC: sti: use bind_pcm_ctl implementation of bind_pcm_ctl for sti driver. Arnaud Pouliquen (2): ASoC: core: allow DAI PCM controls bound to PCM device ASoC: sti: bind pcm controls to pcm device. include/sound/soc-dai.h | 4 ++++ sound/soc/soc-core.c | 28 ++++++++++++++++++++++++++++ sound/soc/sti/sti_uniperif.c | 33 ++++----------------------------- 3 files changed, 36 insertions(+), 29 deletions(-) -- 1.9.1

4 11

[alsa-devel] [PATCH v2 1/1] ALSA: SOC: DMA: access buffer pointer atomically
by Jiada Wang 01 Dec '16

01 Dec '16

From: Andreas Pape <apape(a)de.adit-jv.com> Setting pointer and afterwards check for wrap around leads to the possibility of returning the inconsistent pointer position. This patch accesses buffer pointer atomically to avoid this issue. v2: use READ_ONCE and WRITE_ONCE to ensure read/write of buffer pointer in only one access also updates snd_dmaengine_pcm_pointer_no_residue to ensure atomic access Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/core/pcm_dmaengine.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 8eb58c7..b6255fd 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -139,12 +139,14 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_config_from_dai_data); static void dmaengine_pcm_dma_complete(void *arg) { + unsigned int new_pos; struct snd_pcm_substream *substream = arg; struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - prtd->pos += snd_pcm_lib_period_bytes(substream); - if (prtd->pos >= snd_pcm_lib_buffer_bytes(substream)) - prtd->pos = 0; + new_pos = prtd->pos + snd_pcm_lib_period_bytes(substream); + if (new_pos >= snd_pcm_lib_buffer_bytes(substream)) + new_pos = 0; + WRITE_ONCE(prtd->pos, new_pos); snd_pcm_period_elapsed(substream); } @@ -235,7 +237,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger); snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - return bytes_to_frames(substream->runtime, prtd->pos); + return bytes_to_frames(substream->runtime, READ_ONCE(prtd->pos)); } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue); -- 2.9.3

2 1

[alsa-devel] [PATCH 39/39] Annotate hardware config module parameters in sound/pci/
by David Howells 01 Dec '16

01 Dec '16

When the kernel is running in secure boot mode, we lock down the kernel to prevent userspace from modifying the running kernel image. Whilst this includes prohibiting access to things like /dev/mem, it must also prevent access by means of configuring driver modules in such a way as to cause a device to access or modify the kernel image. To this end, annotate module_param* statements that refer to hardware configuration and indicate for future reference what type of parameter they specify. The parameter parser in the core sees this information and can skip such parameters with an error message if the kernel is locked down. The module initialisation then runs as normal, but just sees whatever the default values for those parameters is. Note that we do still need to do the module initialisation because some drivers have viable defaults set in case parameters aren't specified and some drivers support automatic configuration (e.g. PNP or PCI) in addition to manually coded parameters. This patch annotates drivers in sound/pci/. Suggested-by: One Thousand Gnomes <gnomes(a)lxorguk.ukuu.org.uk> Signed-off-by: David Howells <dhowells(a)redhat.com> cc: Jaroslav Kysela <perex(a)perex.cz> cc: Takashi Iwai <tiwai(a)suse.com> cc: alsa-devel(a)alsa-project.org --- sound/pci/als4000.c | 2 +- sound/pci/cmipci.c | 6 +++--- sound/pci/ens1370.c | 2 +- sound/pci/riptide/riptide.c | 6 +++--- sound/pci/sonicvibes.c | 2 +- sound/pci/via82xx.c | 2 +- sound/pci/ymfpci/ymfpci.c | 6 +++--- 7 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index edabe1371660..e17d1742057d 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -102,7 +102,7 @@ MODULE_PARM_DESC(id, "ID string for ALS4000 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable ALS4000 soundcard."); #ifdef SUPPORT_JOYSTICK -module_param_array(joystick_port, int, NULL, 0444); +module_param_hw_array(joystick_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port address for ALS4000 soundcard. (0 = disabled)"); #endif diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 73f593526b2d..8a217752a5f8 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -68,14 +68,14 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for C-Media PCI soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable C-Media PCI soundcard."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port."); module_param_array(soft_ac3, bool, NULL, 0444); MODULE_PARM_DESC(soft_ac3, "Software-conversion of raw SPDIF packets (model 033 only)."); #ifdef SUPPORT_JOYSTICK -module_param_array(joystick_port, int, NULL, 0444); +module_param_hw_array(joystick_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port address."); #endif diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 7e760fed0728..cdb3637a2bb1 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -106,7 +106,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Ensoniq AudioPCI soundcard."); #ifdef SUPPORT_JOYSTICK #ifdef CHIP1371 -module_param_array(joystick_port, int, NULL, 0444); +module_param_hw_array(joystick_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port address."); #else module_param_array(joystick, bool, NULL, 0444); diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index ada5f01d479c..8b39f03f2144 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -137,12 +137,12 @@ MODULE_PARM_DESC(id, "ID string for Riptide soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Riptide soundcard."); #ifdef SUPPORT_JOYSTICK -module_param_array(joystick_port, int, NULL, 0444); +module_param_hw_array(joystick_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port # for Riptide soundcard."); #endif -module_param_array(mpu_port, int, NULL, 0444); +module_param_hw_array(mpu_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU401 port # for Riptide driver."); -module_param_array(opl3_port, int, NULL, 0444); +module_param_hw_array(opl3_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(opl3_port, "OPL3 port # for Riptide driver."); /* diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index e1a13870bb80..01ace45d9405 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -66,7 +66,7 @@ module_param_array(reverb, bool, NULL, 0444); MODULE_PARM_DESC(reverb, "Enable reverb (SRAM is present) for S3 SonicVibes soundcard."); module_param_array(mge, bool, NULL, 0444); MODULE_PARM_DESC(mge, "MIC Gain Enable for S3 SonicVibes soundcard."); -module_param(dmaio, uint, 0444); +module_param_hw(dmaio, uint, ioport, 0444); MODULE_PARM_DESC(dmaio, "DDMA i/o base address for S3 SonicVibes soundcard."); /* diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 38a17b4342a6..2dc8f1a86aa2 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -92,7 +92,7 @@ module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for VIA 82xx bridge."); module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for VIA 82xx bridge."); -module_param(mpu_port, long, 0444); +module_param_hw(mpu_port, long, ioport, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port. (VT82C686x only)"); #ifdef SUPPORT_JOYSTICK module_param(joystick, bool, 0444); diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 812e27a1bcbc..4faf3e1ed06a 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -55,12 +55,12 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for the Yamaha DS-1 PCI soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Yamaha DS-1 soundcard."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 Port."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM OPL-3 Port."); #ifdef SUPPORT_JOYSTICK -module_param_array(joystick_port, long, NULL, 0444); +module_param_hw_array(joystick_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port address"); #endif module_param_array(rear_switch, bool, NULL, 0444);

1 0

[alsa-devel] [PATCH 38/39] Annotate hardware config module parameters in sound/oss/
by David Howells 01 Dec '16

01 Dec '16

When the kernel is running in secure boot mode, we lock down the kernel to prevent userspace from modifying the running kernel image. Whilst this includes prohibiting access to things like /dev/mem, it must also prevent access by means of configuring driver modules in such a way as to cause a device to access or modify the kernel image. To this end, annotate module_param* statements that refer to hardware configuration and indicate for future reference what type of parameter they specify. The parameter parser in the core sees this information and can skip such parameters with an error message if the kernel is locked down. The module initialisation then runs as normal, but just sees whatever the default values for those parameters is. Note that we do still need to do the module initialisation because some drivers have viable defaults set in case parameters aren't specified and some drivers support automatic configuration (e.g. PNP or PCI) in addition to manually coded parameters. This patch annotates drivers in sound/oss/. Suggested-by: One Thousand Gnomes <gnomes(a)lxorguk.ukuu.org.uk> Signed-off-by: David Howells <dhowells(a)redhat.com> cc: Jaroslav Kysela <perex(a)perex.cz> cc: Takashi Iwai <tiwai(a)suse.com> cc: Riccardo Facchetti <fizban(a)tin.it> cc: Andrew Veliath <andrewtv(a)usa.net> cc: alsa-devel(a)alsa-project.org --- sound/oss/ad1848.c | 8 ++++---- sound/oss/aedsp16.c | 12 ++++++------ sound/oss/mpu401.c | 4 ++-- sound/oss/msnd_pinnacle.c | 20 ++++++++++---------- sound/oss/opl3.c | 2 +- sound/oss/pas2_card.c | 18 +++++++++--------- sound/oss/pss.c | 14 +++++++------- sound/oss/sb_card.c | 10 +++++----- sound/oss/trix.c | 18 +++++++++--------- sound/oss/uart401.c | 4 ++-- sound/oss/uart6850.c | 4 ++-- sound/oss/waveartist.c | 8 ++++---- 12 files changed, 61 insertions(+), 61 deletions(-) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 6368e5c7d0ba..6cae59c4e3c4 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -2810,10 +2810,10 @@ static int __initdata dma = -1; static int __initdata dma2 = -1; static int __initdata type = 0; -module_param(io, int, 0); /* I/O for a raw AD1848 card */ -module_param(irq, int, 0); /* IRQ to use */ -module_param(dma, int, 0); /* First DMA channel */ -module_param(dma2, int, 0); /* Second DMA channel */ +module_param_hw(io, int, ioport, 0); /* I/O for a raw AD1848 card */ +module_param_hw(irq, int, irq, 0); /* IRQ to use */ +module_param_hw(dma, int, dma, 0); /* First DMA channel */ +module_param_hw(dma2, int, dma, 0); /* Second DMA channel */ module_param(type, int, 0); /* Card type */ module_param(deskpro_xl, bool, 0); /* Special magic for Deskpro XL boxen */ module_param(deskpro_m, bool, 0); /* Special magic for Deskpro M box */ diff --git a/sound/oss/aedsp16.c b/sound/oss/aedsp16.c index bb477d5c8528..f058ed6bdb69 100644 --- a/sound/oss/aedsp16.c +++ b/sound/oss/aedsp16.c @@ -1303,17 +1303,17 @@ static int __initdata mpu_irq = -1; static int __initdata mss_base = -1; static int __initdata mpu_base = -1; -module_param(io, int, 0); +module_param_hw(io, int, ioport, 0); MODULE_PARM_DESC(io, "I/O base address (0x220 0x240)"); -module_param(irq, int, 0); +module_param_hw(irq, int, irq, 0); MODULE_PARM_DESC(irq, "IRQ line (5 7 9 10 11)"); -module_param(dma, int, 0); +module_param_hw(dma, int, dma, 0); MODULE_PARM_DESC(dma, "dma line (0 1 3)"); -module_param(mpu_irq, int, 0); +module_param_hw(mpu_irq, int, irq, 0); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ line (5 7 9 10 0)"); -module_param(mss_base, int, 0); +module_param_hw(mss_base, int, ioport, 0); MODULE_PARM_DESC(mss_base, "MSS emulation I/O base address (0x530 0xE80)"); -module_param(mpu_base, int, 0); +module_param_hw(mpu_base, int, ioport, 0); MODULE_PARM_DESC(mpu_base,"MPU-401 I/O base address (0x300 0x310 0x320 0x330)"); MODULE_AUTHOR("Riccardo Facchetti <fizban(a)tin.it>"); MODULE_DESCRIPTION("Audio Excel DSP 16 Driver Version " VERSION); diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 862735005b43..20e8fa46f647 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -1748,8 +1748,8 @@ static struct address_info cfg; static int io = -1; static int irq = -1; -module_param(irq, int, 0); -module_param(io, int, 0); +module_param_hw(irq, int, irq, 0); +module_param_hw(io, int, ioport, 0); static int __init init_mpu401(void) { diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c index a8bb4a06ba6f..8e5221d15066 100644 --- a/sound/oss/msnd_pinnacle.c +++ b/sound/oss/msnd_pinnacle.c @@ -1725,22 +1725,22 @@ static int calibrate_signal __initdata = CONFIG_MSND_CALSIGNAL; #endif /* MODULE */ -module_param (io, int, 0); -module_param (irq, int, 0); -module_param (mem, int, 0); +module_param_hw (io, int, ioport, 0); +module_param_hw (irq, int, irq, 0); +module_param_hw (mem, int, iomem, 0); module_param (write_ndelay, int, 0); module_param (fifosize, int, 0); module_param (calibrate_signal, int, 0); #ifndef MSND_CLASSIC module_param (digital, bool, 0); -module_param (cfg, int, 0); +module_param_hw (cfg, int, ioport, 0); module_param (reset, int, 0); -module_param (mpu_io, int, 0); -module_param (mpu_irq, int, 0); -module_param (ide_io0, int, 0); -module_param (ide_io1, int, 0); -module_param (ide_irq, int, 0); -module_param (joystick_io, int, 0); +module_param_hw (mpu_io, int, ioport, 0); +module_param_hw (mpu_irq, int, irq, 0); +module_param_hw (ide_io0, int, ioport, 0); +module_param_hw (ide_io1, int, ioport, 0); +module_param_hw (ide_irq, int, irq, 0); +module_param_hw (joystick_io, int, ioport, 0); #endif static int __init msnd_init(void) diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index b6d19adf8f41..f0f5b5be6314 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -1200,7 +1200,7 @@ static int me; static int io = -1; -module_param(io, int, 0); +module_param_hw(io, int, ioport, 0); static int __init init_opl3 (void) { diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c index b07954a79536..769fca692d2a 100644 --- a/sound/oss/pas2_card.c +++ b/sound/oss/pas2_card.c @@ -383,15 +383,15 @@ static int __initdata sb_irq = -1; static int __initdata sb_dma = -1; static int __initdata sb_dma16 = -1; -module_param(io, int, 0); -module_param(irq, int, 0); -module_param(dma, int, 0); -module_param(dma16, int, 0); - -module_param(sb_io, int, 0); -module_param(sb_irq, int, 0); -module_param(sb_dma, int, 0); -module_param(sb_dma16, int, 0); +module_param_hw(io, int, ioport, 0); +module_param_hw(irq, int, irq, 0); +module_param_hw(dma, int, dma, 0); +module_param_hw(dma16, int, dma, 0); + +module_param_hw(sb_io, int, ioport, 0); +module_param_hw(sb_irq, int, irq, 0); +module_param_hw(sb_dma, int, dma, 0); +module_param_hw(sb_dma16, int, dma, 0); module_param(joystick, bool, 0); module_param(symphony, bool, 0); diff --git a/sound/oss/pss.c b/sound/oss/pss.c index 81314f9e2ccb..33c3a442e162 100644 --- a/sound/oss/pss.c +++ b/sound/oss/pss.c @@ -1139,19 +1139,19 @@ static bool pss_no_sound = 0; /* Just configure non-sound components */ static bool pss_keep_settings = 1; /* Keep hardware settings at module exit */ static char *pss_firmware = "/etc/sound/pss_synth"; -module_param(pss_io, int, 0); +module_param_hw(pss_io, int, ioport, 0); MODULE_PARM_DESC(pss_io, "Set i/o base of PSS card (probably 0x220 or 0x240)"); -module_param(mss_io, int, 0); +module_param_hw(mss_io, int, ioport, 0); MODULE_PARM_DESC(mss_io, "Set WSS (audio) i/o base (0x530, 0x604, 0xE80, 0xF40, or other. Address must end in 0 or 4 and must be from 0x100 to 0xFF4)"); -module_param(mss_irq, int, 0); +module_param_hw(mss_irq, int, irq, 0); MODULE_PARM_DESC(mss_irq, "Set WSS (audio) IRQ (3, 5, 7, 9, 10, 11, 12)"); -module_param(mss_dma, int, 0); +module_param_hw(mss_dma, int, dma, 0); MODULE_PARM_DESC(mss_dma, "Set WSS (audio) DMA (0, 1, 3)"); -module_param(mpu_io, int, 0); +module_param_hw(mpu_io, int, ioport, 0); MODULE_PARM_DESC(mpu_io, "Set MIDI i/o base (0x330 or other. Address must be on 4 location boundaries and must be from 0x100 to 0xFFC)"); -module_param(mpu_irq, int, 0); +module_param_hw(mpu_irq, int, irq, 0); MODULE_PARM_DESC(mpu_irq, "Set MIDI IRQ (3, 5, 7, 9, 10, 11, 12)"); -module_param(pss_cdrom_port, int, 0); +module_param_hw(pss_cdrom_port, int, ioport, 0); MODULE_PARM_DESC(pss_cdrom_port, "Set the PSS CDROM port i/o base (0x340 or other)"); module_param(pss_enable_joystick, bool, 0); MODULE_PARM_DESC(pss_enable_joystick, "Enables the PSS joystick port (1 to enable, 0 to disable)"); diff --git a/sound/oss/sb_card.c b/sound/oss/sb_card.c index fb5d7250de38..2a92cfe6cfe9 100644 --- a/sound/oss/sb_card.c +++ b/sound/oss/sb_card.c @@ -61,15 +61,15 @@ static int __initdata uart401 = 0; static int __initdata pnp = 0; #endif -module_param(io, int, 000); +module_param_hw(io, int, ioport, 000); MODULE_PARM_DESC(io, "Soundblaster i/o base address (0x220,0x240,0x260,0x280)"); -module_param(irq, int, 000); +module_param_hw(irq, int, irq, 000); MODULE_PARM_DESC(irq, "IRQ (5,7,9,10)"); -module_param(dma, int, 000); +module_param_hw(dma, int, dma, 000); MODULE_PARM_DESC(dma, "8-bit DMA channel (0,1,3)"); -module_param(dma16, int, 000); +module_param_hw(dma16, int, dma, 000); MODULE_PARM_DESC(dma16, "16-bit DMA channel (5,6,7)"); -module_param(mpu_io, int, 000); +module_param_hw(mpu_io, int, ioport, 000); MODULE_PARM_DESC(mpu_io, "MPU base address"); module_param(type, int, 000); MODULE_PARM_DESC(type, "You can set this to specific card type (doesn't " \ diff --git a/sound/oss/trix.c b/sound/oss/trix.c index 3c494dc93b93..a57bc635d758 100644 --- a/sound/oss/trix.c +++ b/sound/oss/trix.c @@ -413,15 +413,15 @@ static int __initdata sb_irq = -1; static int __initdata mpu_io = -1; static int __initdata mpu_irq = -1; -module_param(io, int, 0); -module_param(irq, int, 0); -module_param(dma, int, 0); -module_param(dma2, int, 0); -module_param(sb_io, int, 0); -module_param(sb_dma, int, 0); -module_param(sb_irq, int, 0); -module_param(mpu_io, int, 0); -module_param(mpu_irq, int, 0); +module_param_hw(io, int, ioport, 0); +module_param_hw(irq, int, irq, 0); +module_param_hw(dma, int, dma, 0); +module_param_hw(dma2, int, dma, 0); +module_param_hw(sb_io, int, ioport, 0); +module_param_hw(sb_dma, int, dma, 0); +module_param_hw(sb_irq, int, irq, 0); +module_param_hw(mpu_io, int, ioport, 0); +module_param_hw(mpu_irq, int, irq, 0); module_param(joystick, bool, 0); static int __init init_trix(void) diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c index dae4d4344407..83dcc85b8688 100644 --- a/sound/oss/uart401.c +++ b/sound/oss/uart401.c @@ -429,8 +429,8 @@ static struct address_info cfg_mpu; static int io = -1; static int irq = -1; -module_param(io, int, 0444); -module_param(irq, int, 0444); +module_param_hw(io, int, ioport, 0444); +module_param_hw(irq, int, irq, 0444); static int __init init_uart401(void) diff --git a/sound/oss/uart6850.c b/sound/oss/uart6850.c index 1079133dd6ab..eda32d7eddbd 100644 --- a/sound/oss/uart6850.c +++ b/sound/oss/uart6850.c @@ -315,8 +315,8 @@ static struct address_info cfg_mpu; static int __initdata io = -1; static int __initdata irq = -1; -module_param(io, int, 0); -module_param(irq, int, 0); +module_param_hw(io, int, ioport, 0); +module_param_hw(irq, int, irq, 0); static int __init init_uart6850(void) { diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c index 0b8d0de87273..4f0c3a232e41 100644 --- a/sound/oss/waveartist.c +++ b/sound/oss/waveartist.c @@ -2036,8 +2036,8 @@ __setup("waveartist=", setup_waveartist); #endif MODULE_DESCRIPTION("Rockwell WaveArtist RWA-010 sound driver"); -module_param(io, int, 0); /* IO base */ -module_param(irq, int, 0); /* IRQ */ -module_param(dma, int, 0); /* DMA */ -module_param(dma2, int, 0); /* DMA2 */ +module_param_hw(io, int, ioport, 0); /* IO base */ +module_param_hw(irq, int, irq, 0); /* IRQ */ +module_param_hw(dma, int, dma, 0); /* DMA */ +module_param_hw(dma2, int, dma, 0); /* DMA2 */ MODULE_LICENSE("GPL");

1 0

[alsa-devel] [PATCH 37/39] Annotate hardware config module parameters in sound/isa/
by David Howells 01 Dec '16

01 Dec '16

When the kernel is running in secure boot mode, we lock down the kernel to prevent userspace from modifying the running kernel image. Whilst this includes prohibiting access to things like /dev/mem, it must also prevent access by means of configuring driver modules in such a way as to cause a device to access or modify the kernel image. To this end, annotate module_param* statements that refer to hardware configuration and indicate for future reference what type of parameter they specify. The parameter parser in the core sees this information and can skip such parameters with an error message if the kernel is locked down. The module initialisation then runs as normal, but just sees whatever the default values for those parameters is. Note that we do still need to do the module initialisation because some drivers have viable defaults set in case parameters aren't specified and some drivers support automatic configuration (e.g. PNP or PCI) in addition to manually coded parameters. This patch annotates drivers in sound/isa/. Suggested-by: One Thousand Gnomes <gnomes(a)lxorguk.ukuu.org.uk> Signed-off-by: David Howells <dhowells(a)redhat.com> cc: Jaroslav Kysela <perex(a)perex.cz> cc: Takashi Iwai <tiwai(a)suse.com> cc: alsa-devel(a)alsa-project.org --- sound/isa/ad1848/ad1848.c | 6 +++--- sound/isa/adlib.c | 2 +- sound/isa/cmi8328.c | 12 ++++++------ sound/isa/cmi8330.c | 20 ++++++++++---------- sound/isa/cs423x/cs4231.c | 12 ++++++------ sound/isa/cs423x/cs4236.c | 18 +++++++++--------- sound/isa/es1688/es1688.c | 12 ++++++------ sound/isa/es18xx.c | 12 ++++++------ sound/isa/galaxy/galaxy.c | 16 ++++++++-------- sound/isa/gus/gusclassic.c | 8 ++++---- sound/isa/gus/gusextreme.c | 16 ++++++++-------- sound/isa/gus/gusmax.c | 8 ++++---- sound/isa/gus/interwave.c | 10 +++++----- sound/isa/msnd/msnd_pinnacle.c | 20 ++++++++++---------- sound/isa/opl3sa2.c | 16 ++++++++-------- sound/isa/opti9xx/miro.c | 14 +++++++------- sound/isa/opti9xx/opti92x-ad1848.c | 14 +++++++------- sound/isa/sb/jazz16.c | 12 ++++++------ sound/isa/sb/sb16.c | 14 +++++++------- sound/isa/sb/sb8.c | 6 +++--- sound/isa/sc6000.c | 12 ++++++------ sound/isa/sscape.c | 12 ++++++------ sound/isa/wavefront/wavefront.c | 18 +++++++++--------- 23 files changed, 145 insertions(+), 145 deletions(-) diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c index a302d1f8d14f..e739b1c85c25 100644 --- a/sound/isa/ad1848/ad1848.c +++ b/sound/isa/ad1848/ad1848.c @@ -55,11 +55,11 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for " CRD_NAME " driver."); module_param_array(thinkpad, bool, NULL, 0444); MODULE_PARM_DESC(thinkpad, "Enable only for the onboard CS4248 of IBM Thinkpad 360/750/755 series."); diff --git a/sound/isa/adlib.c b/sound/isa/adlib.c index 8d3060fd7ad7..5fb619eca5c8 100644 --- a/sound/isa/adlib.c +++ b/sound/isa/adlib.c @@ -27,7 +27,7 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); static int snd_adlib_match(struct device *dev, unsigned int n) diff --git a/sound/isa/cmi8328.c b/sound/isa/cmi8328.c index 787475084f46..8e1756c3b9bb 100644 --- a/sound/isa/cmi8328.c +++ b/sound/isa/cmi8328.c @@ -51,18 +51,18 @@ MODULE_PARM_DESC(index, "Index value for CMI8328 soundcard."); module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for CMI8328 soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for CMI8328 driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for CMI8328 driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 for CMI8328 driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 for CMI8328 driver."); -module_param_array(mpuport, long, NULL, 0444); +module_param_hw_array(mpuport, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8328 driver."); -module_param_array(mpuirq, int, NULL, 0444); +module_param_hw_array(mpuirq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8328 MPU-401 port."); #ifdef SUPPORT_JOYSTICK module_param_array(gameport, bool, NULL, 0444); diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index dfedfd85f205..f64b29ab5cc7 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -95,27 +95,27 @@ module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); #endif -module_param_array(sbport, long, NULL, 0444); +module_param_hw_array(sbport, long, ioport, NULL, 0444); MODULE_PARM_DESC(sbport, "Port # for CMI8330/CMI8329 SB driver."); -module_param_array(sbirq, int, NULL, 0444); +module_param_hw_array(sbirq, int, irq, NULL, 0444); MODULE_PARM_DESC(sbirq, "IRQ # for CMI8330/CMI8329 SB driver."); -module_param_array(sbdma8, int, NULL, 0444); +module_param_hw_array(sbdma8, int, dma, NULL, 0444); MODULE_PARM_DESC(sbdma8, "DMA8 for CMI8330/CMI8329 SB driver."); -module_param_array(sbdma16, int, NULL, 0444); +module_param_hw_array(sbdma16, int, dma, NULL, 0444); MODULE_PARM_DESC(sbdma16, "DMA16 for CMI8330/CMI8329 SB driver."); -module_param_array(wssport, long, NULL, 0444); +module_param_hw_array(wssport, long, ioport, NULL, 0444); MODULE_PARM_DESC(wssport, "Port # for CMI8330/CMI8329 WSS driver."); -module_param_array(wssirq, int, NULL, 0444); +module_param_hw_array(wssirq, int, irq, NULL, 0444); MODULE_PARM_DESC(wssirq, "IRQ # for CMI8330/CMI8329 WSS driver."); -module_param_array(wssdma, int, NULL, 0444); +module_param_hw_array(wssdma, int, dma, NULL, 0444); MODULE_PARM_DESC(wssdma, "DMA for CMI8330/CMI8329 WSS driver."); -module_param_array(fmport, long, NULL, 0444); +module_param_hw_array(fmport, long, ioport, NULL, 0444); MODULE_PARM_DESC(fmport, "FM port # for CMI8330/CMI8329 driver."); -module_param_array(mpuport, long, NULL, 0444); +module_param_hw_array(mpuport, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8330/CMI8329 driver."); -module_param_array(mpuirq, int, NULL, 0444); +module_param_hw_array(mpuirq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8330/CMI8329 MPU-401 port."); #ifdef CONFIG_PNP static int isa_registered; diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index ef7448e9f813..e8edd9017a2f 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -55,17 +55,17 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for " CRD_NAME " driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for " CRD_NAME " driver."); static int snd_cs4231_match(struct device *dev, unsigned int n) diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 9d7582c90a95..1f9a3b2be7a1 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -98,23 +98,23 @@ MODULE_PARM_DESC(enable, "Enable " IDENT " soundcard."); module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "ISA PnP detection for specified soundcard."); #endif -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " IDENT " driver."); -module_param_array(cport, long, NULL, 0444); +module_param_hw_array(cport, long, ioport, NULL, 0444); MODULE_PARM_DESC(cport, "Control port # for " IDENT " driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " IDENT " driver."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port # for " IDENT " driver."); -module_param_array(sb_port, long, NULL, 0444); +module_param_hw_array(sb_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(sb_port, "SB port # for " IDENT " driver (optional)."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for " IDENT " driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " IDENT " driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for " IDENT " driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for " IDENT " driver."); #ifdef CONFIG_PNP diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 1901c2bb6c3b..36320e7f2789 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -71,17 +71,17 @@ module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); #endif MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); -module_param_array(irq, int, NULL, 0444); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port # for ES1688 driver."); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); -module_param_array(dma8, int, NULL, 0444); +module_param_hw_array(dma8, int, dma, NULL, 0444); MODULE_PARM_DESC(dma8, "8-bit DMA # for " CRD_NAME " driver."); #ifdef CONFIG_PNP diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 5094b62d8f77..0cabe2b8974f 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1999,17 +1999,17 @@ MODULE_PARM_DESC(enable, "Enable ES18xx soundcard."); module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); #endif -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for ES18xx driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for ES18xx driver."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port # for ES18xx driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for ES18xx driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA 1 # for ES18xx driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA 2 # for ES18xx driver."); #ifdef CONFIG_PNP diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c index 379abe2cbeb2..b9994cc9f5fb 100644 --- a/sound/isa/galaxy/galaxy.c +++ b/sound/isa/galaxy/galaxy.c @@ -53,21 +53,21 @@ static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); -module_param_array(wss_port, long, NULL, 0444); +module_param_hw_array(wss_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(wss_port, "WSS port # for " CRD_NAME " driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port # for " CRD_NAME " driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "Playback DMA # for " CRD_NAME " driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "Capture DMA # for " CRD_NAME " driver."); /* diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c index c169be49ed71..92a997ab1229 100644 --- a/sound/isa/gus/gusclassic.c +++ b/sound/isa/gus/gusclassic.c @@ -58,13 +58,13 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for " CRD_NAME " driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for " CRD_NAME " driver."); module_param_array(joystick_dac, int, NULL, 0444); MODULE_PARM_DESC(joystick_dac, "Joystick DAC level 0.59V-4.52V or 0.389V-2.98V for " CRD_NAME " driver."); diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 77ac2fd723b4..beb52c0f70ea 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -66,21 +66,21 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); -module_param_array(gf1_port, long, NULL, 0444); +module_param_hw_array(gf1_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(gf1_port, "GF1 port # for " CRD_NAME " driver (optional)."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); -module_param_array(gf1_irq, int, NULL, 0444); +module_param_hw_array(gf1_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(gf1_irq, "GF1 IRQ # for " CRD_NAME " driver."); -module_param_array(dma8, int, NULL, 0444); +module_param_hw_array(dma8, int, dma, NULL, 0444); MODULE_PARM_DESC(dma8, "8-bit DMA # for " CRD_NAME " driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "GF1 DMA # for " CRD_NAME " driver."); module_param_array(joystick_dac, int, NULL, 0444); MODULE_PARM_DESC(joystick_dac, "Joystick DAC level 0.59V-4.52V or 0.389V-2.98V for " CRD_NAME " driver."); diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index dd88c9d33492..63309a453140 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -56,13 +56,13 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for GUS MAX soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable GUS MAX soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for GUS MAX driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for GUS MAX driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for GUS MAX driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for GUS MAX driver."); module_param_array(joystick_dac, int, NULL, 0444); MODULE_PARM_DESC(joystick_dac, "Joystick DAC level 0.59V-4.52V or 0.389V-2.98V for GUS MAX driver."); diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 70d0040484c8..0687b7ef3e53 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -92,17 +92,17 @@ MODULE_PARM_DESC(enable, "Enable InterWave soundcard."); module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "ISA PnP detection for specified soundcard."); #endif -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for InterWave driver."); #ifdef SNDRV_STB -module_param_array(port_tc, long, NULL, 0444); +module_param_hw_array(port_tc, long, ioport, NULL, 0444); MODULE_PARM_DESC(port_tc, "Tone control (TEA6330T - i2c bus) port # for InterWave driver."); #endif -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for InterWave driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for InterWave driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for InterWave driver."); module_param_array(joystick_dac, int, NULL, 0444); MODULE_PARM_DESC(joystick_dac, "Joystick DAC level 0.59V-4.52V or 0.389V-2.98V for InterWave driver."); diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 4c072666115d..ad4897337df5 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -800,22 +800,22 @@ MODULE_LICENSE("GPL"); MODULE_FIRMWARE(INITCODEFILE); MODULE_FIRMWARE(PERMCODEFILE); -module_param_array(io, long, NULL, S_IRUGO); +module_param_hw_array(io, long, ioport, NULL, S_IRUGO); MODULE_PARM_DESC(io, "IO port #"); -module_param_array(irq, int, NULL, S_IRUGO); -module_param_array(mem, long, NULL, S_IRUGO); +module_param_hw_array(irq, int, irq, NULL, S_IRUGO); +module_param_hw_array(mem, long, iomem, NULL, S_IRUGO); module_param_array(write_ndelay, int, NULL, S_IRUGO); module_param(calibrate_signal, int, S_IRUGO); #ifndef MSND_CLASSIC module_param_array(digital, int, NULL, S_IRUGO); -module_param_array(cfg, long, NULL, S_IRUGO); +module_param_hw_array(cfg, long, ioport, NULL, S_IRUGO); module_param_array(reset, int, 0, S_IRUGO); -module_param_array(mpu_io, long, NULL, S_IRUGO); -module_param_array(mpu_irq, int, NULL, S_IRUGO); -module_param_array(ide_io0, long, NULL, S_IRUGO); -module_param_array(ide_io1, long, NULL, S_IRUGO); -module_param_array(ide_irq, int, NULL, S_IRUGO); -module_param_array(joystick_io, long, NULL, S_IRUGO); +module_param_hw_array(mpu_io, long, ioport, NULL, S_IRUGO); +module_param_hw_array(mpu_irq, int, irq, NULL, S_IRUGO); +module_param_hw_array(ide_io0, long, ioport, NULL, S_IRUGO); +module_param_hw_array(ide_io1, long, ioport, NULL, S_IRUGO); +module_param_hw_array(ide_irq, int, irq, NULL, S_IRUGO); +module_param_hw_array(joystick_io, long, ioport, NULL, S_IRUGO); #endif diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index ae133633a420..4098e3e0353d 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -69,21 +69,21 @@ MODULE_PARM_DESC(enable, "Enable OPL3-SA soundcard."); module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); #endif -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for OPL3-SA driver."); -module_param_array(sb_port, long, NULL, 0444); +module_param_hw_array(sb_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(sb_port, "SB port # for OPL3-SA driver."); -module_param_array(wss_port, long, NULL, 0444); +module_param_hw_array(wss_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(wss_port, "WSS port # for OPL3-SA driver."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port # for OPL3-SA driver."); -module_param_array(midi_port, long, NULL, 0444); +module_param_hw_array(midi_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(midi_port, "MIDI port # for OPL3-SA driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for OPL3-SA driver."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for OPL3-SA driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for OPL3-SA driver."); module_param_array(opl3sa3_ymode, int, NULL, 0444); MODULE_PARM_DESC(opl3sa3_ymode, "Speaker size selection for 3D Enhancement mode: Desktop/Large Notebook/Small Notebook/HiFi."); diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 3a9067db1a84..bcbff56f060d 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -69,19 +69,19 @@ module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for miro soundcard."); module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for miro soundcard."); -module_param(port, long, 0444); +module_param_hw(port, long, ioport, 0444); MODULE_PARM_DESC(port, "WSS port # for miro driver."); -module_param(mpu_port, long, 0444); +module_param_hw(mpu_port, long, ioport, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for miro driver."); -module_param(fm_port, long, 0444); +module_param_hw(fm_port, long, ioport, 0444); MODULE_PARM_DESC(fm_port, "FM Port # for miro driver."); -module_param(irq, int, 0444); +module_param_hw(irq, int, irq, 0444); MODULE_PARM_DESC(irq, "WSS irq # for miro driver."); -module_param(mpu_irq, int, 0444); +module_param_hw(mpu_irq, int, irq, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 irq # for miro driver."); -module_param(dma1, int, 0444); +module_param_hw(dma1, int, dma, 0444); MODULE_PARM_DESC(dma1, "1st dma # for miro driver."); -module_param(dma2, int, 0444); +module_param_hw(dma2, int, dma, 0444); MODULE_PARM_DESC(dma2, "2nd dma # for miro driver."); module_param(wss, int, 0444); MODULE_PARM_DESC(wss, "wss mode"); diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 0a5266003786..ceddb392b1e3 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -88,20 +88,20 @@ MODULE_PARM_DESC(id, "ID string for opti9xx based soundcard."); module_param(isapnp, bool, 0444); MODULE_PARM_DESC(isapnp, "Enable ISA PnP detection for specified soundcard."); #endif -module_param(port, long, 0444); +module_param_hw(port, long, ioport, 0444); MODULE_PARM_DESC(port, "WSS port # for opti9xx driver."); -module_param(mpu_port, long, 0444); +module_param_hw(mpu_port, long, ioport, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for opti9xx driver."); -module_param(fm_port, long, 0444); +module_param_hw(fm_port, long, ioport, 0444); MODULE_PARM_DESC(fm_port, "FM port # for opti9xx driver."); -module_param(irq, int, 0444); +module_param_hw(irq, int, irq, 0444); MODULE_PARM_DESC(irq, "WSS irq # for opti9xx driver."); -module_param(mpu_irq, int, 0444); +module_param_hw(mpu_irq, int, irq, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 irq # for opti9xx driver."); -module_param(dma1, int, 0444); +module_param_hw(dma1, int, dma, 0444); MODULE_PARM_DESC(dma1, "1st dma # for opti9xx driver."); #if defined(CS4231) || defined(OPTi93X) -module_param(dma2, int, 0444); +module_param_hw(dma2, int, dma, 0444); MODULE_PARM_DESC(dma2, "2nd dma # for opti9xx driver."); #endif /* CS4231 || OPTi93X */ diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index 4d909971eedb..bfa0055e1fd6 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -50,17 +50,17 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for Media Vision Jazz16 based soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Media Vision Jazz16 based soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for jazz16 driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for jazz16 driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for jazz16 driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for jazz16 driver."); -module_param_array(dma8, int, NULL, 0444); +module_param_hw_array(dma8, int, dma, NULL, 0444); MODULE_PARM_DESC(dma8, "DMA8 # for jazz16 driver."); -module_param_array(dma16, int, NULL, 0444); +module_param_hw_array(dma16, int, dma, NULL, 0444); MODULE_PARM_DESC(dma16, "DMA16 # for jazz16 driver."); #define SB_JAZZ16_WAKEUP 0xaf diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 4a7d7c89808f..3b2e4f405ff2 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -99,21 +99,21 @@ MODULE_PARM_DESC(enable, "Enable SoundBlaster 16 soundcard."); module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); #endif -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for SB16 driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for SB16 driver."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port # for SB16 PnP driver."); #ifdef SNDRV_SBAWE_EMU8000 -module_param_array(awe_port, long, NULL, 0444); +module_param_hw_array(awe_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(awe_port, "AWE port # for SB16 PnP driver."); #endif -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for SB16 driver."); -module_param_array(dma8, int, NULL, 0444); +module_param_hw_array(dma8, int, dma, NULL, 0444); MODULE_PARM_DESC(dma8, "8-bit DMA # for SB16 driver."); -module_param_array(dma16, int, NULL, 0444); +module_param_hw_array(dma16, int, dma, NULL, 0444); MODULE_PARM_DESC(dma16, "16-bit DMA # for SB16 driver."); module_param_array(mic_agc, int, NULL, 0444); MODULE_PARM_DESC(mic_agc, "Mic Auto-Gain-Control switch."); diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index ad42d2364199..d77dcba276b5 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -47,11 +47,11 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for Sound Blaster soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Sound Blaster soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for SB8 driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for SB8 driver."); -module_param_array(dma8, int, NULL, 0444); +module_param_hw_array(dma8, int, dma, NULL, 0444); MODULE_PARM_DESC(dma8, "8-bit DMA # for SB8 driver."); struct snd_sb8 { diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index b61a6633d8f2..c09d9b914efe 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -64,17 +64,17 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for sc-6000 based soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable sc-6000 based soundcard."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for sc-6000 driver."); -module_param_array(mss_port, long, NULL, 0444); +module_param_hw_array(mss_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mss_port, "MSS Port # for sc-6000 driver."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for sc-6000 driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for sc-6000 driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for sc-6000 driver."); -module_param_array(dma, int, NULL, 0444); +module_param_hw_array(dma, int, dma, NULL, 0444); MODULE_PARM_DESC(dma, "DMA # for sc-6000 driver."); module_param_array(joystick, bool, NULL, 0444); MODULE_PARM_DESC(joystick, "Enable gameport."); diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index fdcfa29e2205..54f5758a1bb3 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -63,22 +63,22 @@ MODULE_PARM_DESC(index, "Index number for SoundScape soundcard"); module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "Description for SoundScape card"); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for SoundScape driver."); -module_param_array(wss_port, long, NULL, 0444); +module_param_hw_array(wss_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(wss_port, "WSS Port # for SoundScape driver."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for SoundScape driver."); -module_param_array(mpu_irq, int, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU401 IRQ # for SoundScape driver."); -module_param_array(dma, int, NULL, 0444); +module_param_hw_array(dma, int, dma, NULL, 0444); MODULE_PARM_DESC(dma, "DMA # for SoundScape driver."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for SoundScape driver."); module_param_array(joystick, bool, NULL, 0444); diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index a0987a57c8a9..da4e9a85f0af 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -63,23 +63,23 @@ MODULE_PARM_DESC(enable, "Enable WaveFront soundcard."); module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "ISA PnP detection for WaveFront soundcards."); #endif -module_param_array(cs4232_pcm_port, long, NULL, 0444); +module_param_hw_array(cs4232_pcm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(cs4232_pcm_port, "Port # for CS4232 PCM interface."); -module_param_array(cs4232_pcm_irq, int, NULL, 0444); +module_param_hw_array(cs4232_pcm_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(cs4232_pcm_irq, "IRQ # for CS4232 PCM interface."); -module_param_array(dma1, int, NULL, 0444); +module_param_hw_array(dma1, int, dma, NULL, 0444); MODULE_PARM_DESC(dma1, "DMA1 # for CS4232 PCM interface."); -module_param_array(dma2, int, NULL, 0444); +module_param_hw_array(dma2, int, dma, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for CS4232 PCM interface."); -module_param_array(cs4232_mpu_port, long, NULL, 0444); +module_param_hw_array(cs4232_mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(cs4232_mpu_port, "port # for CS4232 MPU-401 interface."); -module_param_array(cs4232_mpu_irq, int, NULL, 0444); +module_param_hw_array(cs4232_mpu_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(cs4232_mpu_irq, "IRQ # for CS4232 MPU-401 interface."); -module_param_array(ics2115_irq, int, NULL, 0444); +module_param_hw_array(ics2115_irq, int, irq, NULL, 0444); MODULE_PARM_DESC(ics2115_irq, "IRQ # for ICS2115."); -module_param_array(ics2115_port, long, NULL, 0444); +module_param_hw_array(ics2115_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(ics2115_port, "Port # for ICS2115."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port #."); module_param_array(use_cs4232_midi, bool, NULL, 0444); MODULE_PARM_DESC(use_cs4232_midi, "Use CS4232 MPU-401 interface (inaccessibly located inside your computer)");

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[alsa-devel] [PATCH 36/39] Annotate hardware config module parameters in sound/drivers/
by David Howells 01 Dec '16

01 Dec '16

When the kernel is running in secure boot mode, we lock down the kernel to prevent userspace from modifying the running kernel image. Whilst this includes prohibiting access to things like /dev/mem, it must also prevent access by means of configuring driver modules in such a way as to cause a device to access or modify the kernel image. To this end, annotate module_param* statements that refer to hardware configuration and indicate for future reference what type of parameter they specify. The parameter parser in the core sees this information and can skip such parameters with an error message if the kernel is locked down. The module initialisation then runs as normal, but just sees whatever the default values for those parameters is. Note that we do still need to do the module initialisation because some drivers have viable defaults set in case parameters aren't specified and some drivers support automatic configuration (e.g. PNP or PCI) in addition to manually coded parameters. This patch annotates drivers in sound/drivers/. Suggested-by: One Thousand Gnomes <gnomes(a)lxorguk.ukuu.org.uk> Signed-off-by: David Howells <dhowells(a)redhat.com> cc: Jaroslav Kysela <perex(a)perex.cz> cc: Takashi Iwai <tiwai(a)suse.com> cc: alsa-devel(a)alsa-project.org --- sound/drivers/mpu401/mpu401.c | 4 ++-- sound/drivers/mtpav.c | 4 ++-- sound/drivers/serial-u16550.c | 4 ++-- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index fed7e7e2177b..9b86e00d7d95 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -53,9 +53,9 @@ MODULE_PARM_DESC(enable, "Enable MPU-401 device."); module_param_array(pnp, bool, NULL, 0444); MODULE_PARM_DESC(pnp, "PnP detection for MPU-401 device."); #endif -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for MPU-401 device."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for MPU-401 device."); module_param_array(uart_enter, bool, NULL, 0444); MODULE_PARM_DESC(uart_enter, "Issue UART_ENTER command at open."); diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 30e8a1d5bc87..c6bab7cf4fe7 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -86,9 +86,9 @@ module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for MotuMTPAV MIDI."); module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for MotuMTPAV MIDI."); -module_param(port, long, 0444); +module_param_hw(port, long, ioport, 0444); MODULE_PARM_DESC(port, "Parallel port # for MotuMTPAV MIDI."); -module_param(irq, int, 0444); +module_param_hw(irq, int, irq, 0444); MODULE_PARM_DESC(irq, "Parallel IRQ # for MotuMTPAV MIDI."); module_param(hwports, int, 0444); MODULE_PARM_DESC(hwports, "Hardware ports # for MotuMTPAV MIDI."); diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 1927b89e1d1f..04be126fe4e6 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -84,9 +84,9 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for Serial MIDI."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable UART16550A chip."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for UART16550A chip."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for UART16550A chip."); module_param_array(speed, int, NULL, 0444); MODULE_PARM_DESC(speed, "Speed in bauds.");

1 0

[alsa-devel] [PATCH] ASoC: rt5670: Enable Braswell platform workaround for Dell Wyse 3040
by Takashi Iwai 01 Dec '16

01 Dec '16

Dell Wyse 3040 requires the platform workaround for enabling dmic and jack detection as defined for Braswell CRB. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- I guess the platform data should be applied generically to all CHT machines instead of DMI matching? DMI matching doesn't scale, obviously. sound/soc/codecs/rt5670.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 98c7efbc6ee0..97bafac3bc15 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2827,6 +2827,13 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_BOARD_NAME, "Braswell CRB"), }, }, + { + .ident = "Dell Wyse 3040", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), + DMI_MATCH(DMI_PRODUCT_NAME, "Wyse 3040"), + }, + }, {} }; -- 2.10.2

3 5

Re: [alsa-devel] [PATCH v2] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches
by kbuild test robot 01 Dec '16

01 Dec '16

Hi Matt, [auto build test ERROR on asoc/for-next] [also build test ERROR on v4.9-rc7 next-20161130] [if your patch is applied to the wrong git tree, please drop us a note to help improve the system] url: https://github.com/0day-ci/linux/commits/Matt-Ranostay/ASoC-omap-mcbsp-Add-… base: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next config: arm-allmodconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (Debian 6.1.1-9) 6.1.1 20160705 reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross # save the attached .config to linux build tree make.cross ARCH=arm All errors (new ones prefixed by >>): sound/soc/omap/mcbsp.c: In function 'omap_mcbsp_start': >> sound/soc/omap/mcbsp.c:653:28: error: 'struct omap_mcbsp' has no member named 'pm_qos_req'; did you mean 'dma_req'? pm_qos_add_request(&mcbsp->pm_qos_req, ^~ sound/soc/omap/mcbsp.c: In function 'omap_mcbsp_stop': sound/soc/omap/mcbsp.c:745:30: error: 'struct omap_mcbsp' has no member named 'pm_qos_req'; did you mean 'dma_req'? pm_qos_remove_request(&mcbsp->pm_qos_req); ^~ sound/soc/omap/mcbsp.c: In function 'omap_mcbsp_cleanup': sound/soc/omap/mcbsp.c:1113:34: error: 'struct omap_mcbsp' has no member named 'pm_qos_req'; did you mean 'dma_req'? if (pm_qos_request_active(&mcbsp->pm_qos_req)) ^~ sound/soc/omap/mcbsp.c:1114:31: error: 'struct omap_mcbsp' has no member named 'pm_qos_req'; did you mean 'dma_req'? pm_qos_remove_request(&mcbsp->pm_qos_req); ^~ vim +653 sound/soc/omap/mcbsp.c 647 int enable_srg = 0; 648 int latency = mcbsp->pdata->buffer_size * 23; 649 u16 w; 650 651 /* Prevent omap hardware from hitting off between fifo fills */ 652 if (latency) > 653 pm_qos_add_request(&mcbsp->pm_qos_req, 654 PM_QOS_CPU_DMA_LATENCY, latency); 655 656 if (mcbsp->st_data) --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH V2 0/5] ASoC/arizona: Ensure pin searches use widget name prefix
by Richard Fitzgerald 01 Dec '16

01 Dec '16

The name of a codec pin can have an optional prefix string, which is defined by the audio machine driver. The snd_soc_dapm_x_pin functions take the fully-specified name including the prefix and so the existing code would fail to find the pin if the audio machine driver had added a prefix. This patch chain adds new helper functions that take a non-prefixed name for a specific ASoC component and internally add that component's prefix. The other patches update the arizona drivers to use these new functions. Richard Fitzgerald (5): ASoC: core: Add component pin control functions ASoC: arizona: Use component pin control functions regulator: arizona-micsupp: Use SoC component pin control functions extcon: arizona: Use SoC component pin control functions Input: arizona-haptics - Use SoC component pin control functions drivers/extcon/extcon-arizona.c | 8 +- drivers/input/misc/arizona-haptics.c | 13 ++- drivers/regulator/arizona-micsupp.c | 6 +- include/sound/soc.h | 20 ++++ sound/soc/codecs/arizona.c | 12 ++- sound/soc/codecs/cs47l24.c | 3 +- sound/soc/codecs/wm5102.c | 3 +- sound/soc/codecs/wm5110.c | 3 +- sound/soc/codecs/wm8997.c | 3 +- sound/soc/codecs/wm8998.c | 3 +- sound/soc/soc-utils.c | 199 +++++++++++++++++++++++++++++++++++ 11 files changed, 255 insertions(+), 18 deletions(-) -- 1.9.1

4 13

[alsa-devel] [PATCH v4] clkdev: add devm_of_clk_get()
by Kuninori Morimoto 01 Dec '16

01 Dec '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current Linux has of_clk_get(), but doesn't have devm_of_clk_get(). This patch adds it. It is implemeted in clk-devres.c to share devm_clk_release(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v3 -> v4 - git log explain why it is implemeted in clk-devres - it is related to CONFIG_HAVE_CLK drivers/clk/clk-devres.c | 21 +++++++++++++++++++++ include/linux/clk.h | 27 +++++++++++++++++++++++---- 2 files changed, 44 insertions(+), 4 deletions(-) diff --git a/drivers/clk/clk-devres.c b/drivers/clk/clk-devres.c index 8f57154..2449b25 100644 --- a/drivers/clk/clk-devres.c +++ b/drivers/clk/clk-devres.c @@ -53,3 +53,24 @@ void devm_clk_put(struct device *dev, struct clk *clk) WARN_ON(ret); } EXPORT_SYMBOL(devm_clk_put); + +struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index) +{ + struct clk **ptr, *clk; + + ptr = devres_alloc(devm_clk_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return ERR_PTR(-ENOMEM); + + clk = of_clk_get(np, index); + if (!IS_ERR(clk)) { + *ptr = clk; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return clk; +} +EXPORT_SYMBOL(devm_of_clk_get); diff --git a/include/linux/clk.h b/include/linux/clk.h index 123c027..7f50c5f 100644 --- a/include/linux/clk.h +++ b/include/linux/clk.h @@ -17,8 +17,9 @@ #include <linux/notifier.h> struct device; - struct clk; +struct device_node; +struct of_phandle_args; /** * DOC: clk notifier callback types @@ -249,6 +250,21 @@ static inline void clk_unprepare(struct clk *clk) struct clk *devm_clk_get(struct device *dev, const char *id); /** + * devm_clk_get - lookup and obtain a managed reference to a clock producer. + * @dev: device for clock "consumer" + * @np: pointer to clock consumer node + * @index: clock index + * + * This function parses the clocks, and uses them to look up the + * struct clk from the registered list of clock providers by using + * @np and @index. + * + * The clock will automatically be freed when the device is unbound + * from the bus. + */ +struct clk *devm_of_clk_get(struct device *dev, struct device_node *np, int index); + +/** * clk_enable - inform the system when the clock source should be running. * @clk: clock source * @@ -432,6 +448,12 @@ static inline struct clk *devm_clk_get(struct device *dev, const char *id) return NULL; } +static inline struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index) +{ + return NULL; +} + static inline void clk_put(struct clk *clk) {} static inline void devm_clk_put(struct device *dev, struct clk *clk) {} @@ -501,9 +523,6 @@ static inline void clk_disable_unprepare(struct clk *clk) clk_unprepare(clk); } -struct device_node; -struct of_phandle_args; - #if defined(CONFIG_OF) && defined(CONFIG_COMMON_CLK) struct clk *of_clk_get(struct device_node *np, int index); struct clk *of_clk_get_by_name(struct device_node *np, const char *name); -- 1.9.1

2 2

[alsa-devel] [PATCH v2] clkdev: add devm_of_clk_get()
by Kuninori Morimoto 01 Dec '16

01 Dec '16

This is based on devm_clk_get() Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - added "static" and "inline" on header drivers/clk/clkdev.c | 26 ++++++++++++++++++++++++++ include/linux/clk.h | 7 +++++++ 2 files changed, 33 insertions(+) diff --git a/drivers/clk/clkdev.c b/drivers/clk/clkdev.c index 89cc700..93a613b 100644 --- a/drivers/clk/clkdev.c +++ b/drivers/clk/clkdev.c @@ -55,6 +55,32 @@ struct clk *of_clk_get(struct device_node *np, int index) } EXPORT_SYMBOL(of_clk_get); +static void devm_of_clk_release(struct device *dev, void *res) +{ + clk_put(*(struct clk **)res); +} + +struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index) +{ + struct clk **ptr, *clk; + + ptr = devres_alloc(devm_of_clk_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return ERR_PTR(-ENOMEM); + + clk = of_clk_get(np, index); + if (!IS_ERR(clk)) { + *ptr = clk; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return clk; +} +EXPORT_SYMBOL(devm_of_clk_get); + static struct clk *__of_clk_get_by_name(struct device_node *np, const char *dev_id, const char *name) diff --git a/include/linux/clk.h b/include/linux/clk.h index a89ba4e..33cd540 100644 --- a/include/linux/clk.h +++ b/include/linux/clk.h @@ -502,6 +502,8 @@ struct of_phandle_args; #if defined(CONFIG_OF) && defined(CONFIG_COMMON_CLK) struct clk *of_clk_get(struct device_node *np, int index); +struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index); struct clk *of_clk_get_by_name(struct device_node *np, const char *name); struct clk *of_clk_get_from_provider(struct of_phandle_args *clkspec); #else @@ -509,6 +511,11 @@ static inline struct clk *of_clk_get(struct device_node *np, int index) { return ERR_PTR(-ENOENT); } +static inline struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index) +{ + return ERR_PTR(-ENOENT); +} static inline struct clk *of_clk_get_by_name(struct device_node *np, const char *name) { -- 1.9.1

4 15

[alsa-devel] Modifying PCM buffer before DMA
by Ricard Wanderlof 30 Nov '16

30 Nov '16

Working with a SoC which has an I2S interface which reuquires 16-bit-swapped 32 bit samples (i.e. the first and 16 halfwords are swapped, but within the halfwords the bytes themselves are little endian), I'm trying to figure out a way to adjust the data in the PCM driver so that ALSA sees standard S32_LE samples. My approach is to use a tight loop to halfword-swap the data. This works fine for capture, where the swap routine can be run triggered by the DMA interrupt as each period arrives from the I2S interface, before calling snd_pcm_period_elapsed(). The problem I have is for playback. I want to have SNDRV_PCM_INFO_MMAP set in snd_pcm_hardware.info in order to be use plugins such as dmix, which in practice means that the PCM driver doesn't really know exactly how much data the application has written before the DMA starts writing it out to the I2S interface. As far as I can tell, the application layer asks the driver for the hw_ptr but there is no way the driver can actually know how much data the application has in fact written. Contrary to the capture case, once the DMA interrupt for a previous period has been triggered, the data for the current period will already be on its way out to the I2S interface, courtesy of the DMA. Disabling MMAP seems to be one way to get the PCM driver to take greater control of the transfer (for instance via snd_pcm_ops->copy() ), but I want to avoid that as mentioned above. My question is, is there another way? /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

Re: [alsa-devel] [PATCH v2] ALSA SoC MAX98927 driver - Revision
by Mark Brown 30 Nov '16

30 Nov '16

On Fri, Nov 25, 2016 at 02:02:47PM +0900, Ryan Lee wrote: > Signed-off-by: Ryan Lee <RyanS.Lee(a)maximintegrated.com> > --- > .../devicetree/bindings/sound/max98927.txt | 33 +- > sound/soc/codecs/max98927.c | 839 +++++------ > sound/soc/codecs/max98927.h | 1463 ++++---------------- This is modifying existing code, you need to at least attempt to supply a description of the changes you are trying to make and most likely send a series of individual, split out changes. Please, review the patch submission process in SubmittingPatches before sending anything further.

1 0

[alsa-devel] [PATCH] ASoC: intel: Replace kthread with work
by Takashi Iwai 30 Nov '16

30 Nov '16

The usage pattern of kthread worker in Intel SST drivers can be replaced gracefully with the normal workqueue, which is more light- weight and easier to manage in general. Let's do it. While in the replacement, move the schedule_work() call inside the spinlock for excluding the race, too. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Vinod, I tested this only lightly, so it'd be helpful if the patch is checked on the affected devices before merging. sound/soc/intel/baytrail/sst-baytrail-ipc.c | 3 +- sound/soc/intel/common/sst-ipc.c | 58 ++++++++++------------------- sound/soc/intel/common/sst-ipc.h | 4 +- sound/soc/intel/haswell/sst-haswell-ipc.c | 3 +- sound/soc/intel/skylake/skl-sst-cldma.c | 1 - sound/soc/intel/skylake/skl-sst-ipc.c | 2 +- sound/soc/intel/skylake/skl-sst-ipc.h | 1 - 7 files changed, 24 insertions(+), 48 deletions(-) diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c index 7ab14ce65a73..260447da32b8 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -23,7 +23,6 @@ #include <linux/slab.h> #include <linux/delay.h> #include <linux/platform_device.h> -#include <linux/kthread.h> #include <linux/firmware.h> #include <linux/io.h> #include <asm/div64.h> @@ -338,7 +337,7 @@ static irqreturn_t sst_byt_irq_thread(int irq, void *context) spin_unlock_irqrestore(&sst->spinlock, flags); /* continue to send any remaining messages... */ - kthread_queue_work(&ipc->kworker, &ipc->kwork); + schedule_work(&ipc->kwork); return IRQ_HANDLED; } diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c index 6c672ac79cce..d20356255d0c 100644 --- a/sound/soc/intel/common/sst-ipc.c +++ b/sound/soc/intel/common/sst-ipc.c @@ -26,7 +26,6 @@ #include <linux/sched.h> #include <linux/delay.h> #include <linux/platform_device.h> -#include <linux/kthread.h> #include <sound/asound.h> #include "sst-dsp.h" @@ -109,10 +108,9 @@ static int ipc_tx_message(struct sst_generic_ipc *ipc, u64 header, ipc->ops.tx_data_copy(msg, tx_data, tx_bytes); list_add_tail(&msg->list, &ipc->tx_list); + schedule_work(&ipc->kwork); spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); - kthread_queue_work(&ipc->kworker, &ipc->kwork); - if (wait) return tx_wait_done(ipc, msg, rx_data); else @@ -156,35 +154,32 @@ static int msg_empty_list_init(struct sst_generic_ipc *ipc) return -ENOMEM; } -static void ipc_tx_msgs(struct kthread_work *work) +static void ipc_tx_msgs(struct work_struct *work) { struct sst_generic_ipc *ipc = container_of(work, struct sst_generic_ipc, kwork); struct ipc_message *msg; - unsigned long flags; - spin_lock_irqsave(&ipc->dsp->spinlock, flags); + spin_lock_irq(&ipc->dsp->spinlock); - if (list_empty(&ipc->tx_list) || ipc->pending) { - spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); - return; - } - - /* if the DSP is busy, we will TX messages after IRQ. - * also postpone if we are in the middle of procesing completion irq*/ - if (ipc->ops.is_dsp_busy && ipc->ops.is_dsp_busy(ipc->dsp)) { - dev_dbg(ipc->dev, "ipc_tx_msgs dsp busy\n"); - spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); - return; - } + while (!list_empty(&ipc->tx_list) && !ipc->pending) { + /* if the DSP is busy, we will TX messages after IRQ. + * also postpone if we are in the middle of processing + * completion irq + */ + if (ipc->ops.is_dsp_busy && ipc->ops.is_dsp_busy(ipc->dsp)) { + dev_dbg(ipc->dev, "ipc_tx_msgs dsp busy\n"); + break; + } - msg = list_first_entry(&ipc->tx_list, struct ipc_message, list); - list_move(&msg->list, &ipc->rx_list); + msg = list_first_entry(&ipc->tx_list, struct ipc_message, list); + list_move(&msg->list, &ipc->rx_list); - if (ipc->ops.tx_msg != NULL) - ipc->ops.tx_msg(ipc, msg); + if (ipc->ops.tx_msg != NULL) + ipc->ops.tx_msg(ipc, msg); + } - spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); + spin_unlock_irq(&ipc->dsp->spinlock); } int sst_ipc_tx_message_wait(struct sst_generic_ipc *ipc, u64 header, @@ -280,19 +275,7 @@ int sst_ipc_init(struct sst_generic_ipc *ipc) if (ret < 0) return -ENOMEM; - /* start the IPC message thread */ - kthread_init_worker(&ipc->kworker); - ipc->tx_thread = kthread_run(kthread_worker_fn, - &ipc->kworker, "%s", - dev_name(ipc->dev)); - if (IS_ERR(ipc->tx_thread)) { - dev_err(ipc->dev, "error: failed to create message TX task\n"); - ret = PTR_ERR(ipc->tx_thread); - kfree(ipc->msg); - return ret; - } - - kthread_init_work(&ipc->kwork, ipc_tx_msgs); + INIT_WORK(&ipc->kwork, ipc_tx_msgs); return 0; } EXPORT_SYMBOL_GPL(sst_ipc_init); @@ -301,8 +284,7 @@ void sst_ipc_fini(struct sst_generic_ipc *ipc) { int i; - if (ipc->tx_thread) - kthread_stop(ipc->tx_thread); + cancel_work_sync(&ipc->kwork); if (ipc->msg) { for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) { diff --git a/sound/soc/intel/common/sst-ipc.h b/sound/soc/intel/common/sst-ipc.h index ceb7e468a3fa..ea79cff5c6cc 100644 --- a/sound/soc/intel/common/sst-ipc.h +++ b/sound/soc/intel/common/sst-ipc.h @@ -23,7 +23,6 @@ #include <linux/list.h> #include <linux/workqueue.h> #include <linux/sched.h> -#include <linux/kthread.h> #define IPC_MAX_MAILBOX_BYTES 256 @@ -65,8 +64,7 @@ struct sst_generic_ipc { struct list_head empty_list; wait_queue_head_t wait_txq; struct task_struct *tx_thread; - struct kthread_worker kworker; - struct kthread_work kwork; + struct work_struct kwork; bool pending; struct ipc_message *msg; int tx_data_max_size; diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index e432a31fd9f2..a3459d1682a6 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -26,7 +26,6 @@ #include <linux/delay.h> #include <linux/sched.h> #include <linux/platform_device.h> -#include <linux/kthread.h> #include <linux/firmware.h> #include <linux/dma-mapping.h> #include <linux/debugfs.h> @@ -818,7 +817,7 @@ static irqreturn_t hsw_irq_thread(int irq, void *context) spin_unlock_irqrestore(&sst->spinlock, flags); /* continue to send any remaining messages... */ - kthread_queue_work(&ipc->kworker, &ipc->kwork); + schedule_work(&ipc->kwork); return IRQ_HANDLED; } diff --git a/sound/soc/intel/skylake/skl-sst-cldma.c b/sound/soc/intel/skylake/skl-sst-cldma.c index efa2532114ba..c9f6d87381db 100644 --- a/sound/soc/intel/skylake/skl-sst-cldma.c +++ b/sound/soc/intel/skylake/skl-sst-cldma.c @@ -17,7 +17,6 @@ #include <linux/device.h> #include <linux/mm.h> -#include <linux/kthread.h> #include <linux/delay.h> #include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 797cf4053235..499c0b15e5d0 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -464,7 +464,7 @@ irqreturn_t skl_dsp_irq_thread_handler(int irq, void *context) skl_ipc_int_enable(dsp); /* continue to send any remaining messages... */ - kthread_queue_work(&ipc->kworker, &ipc->kwork); + schedule_work(&ipc->kwork); return IRQ_HANDLED; } diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h index 0334ed4af031..5c37442a6db4 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.h +++ b/sound/soc/intel/skylake/skl-sst-ipc.h @@ -16,7 +16,6 @@ #ifndef __SKL_IPC_H #define __SKL_IPC_H -#include <linux/kthread.h> #include <linux/irqreturn.h> #include "../common/sst-ipc.h" -- 2.10.2

3 3

[alsa-devel] [PATCH] ASoC: nau8825: lock longer to avoid playback pop upon resume
by John Hsu 30 Nov '16

30 Nov '16

There is an issue about pop noise in NAU88L25 as follows. Issue 54078: Chell_headphone pop back from S3 (1)Play directly to hw, bypassing CRAS: sox -b 16 -n -t alsa hw:0,0 synth sine 200 sine 200 (2)Close lid or powerd_dbus_suspend, then press a key to resume. (3)no audio after resume (4)Audio will be back after close then reopen the pcm device. After verification, we find one defect is that semaphone lock is not long enough and expired. In this situation, the playback comes back early but pauses a while to wait for the crosstalk detection done. But the detection spends too much time and lock time is up. Therefore, the playback and jack detection sequence interfere to each other. That breaks sequence and makes noise. The driver extends the lock time for the issue. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 3f9137f..2b89569 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1233,7 +1233,7 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); unsigned int val_len = 0, osr; - nau8825_sema_acquire(nau8825, 2 * HZ); + nau8825_sema_acquire(nau8825, 3 * HZ); /* CLK_DAC or CLK_ADC = OSR * FS * DAC or ADC clock frequency is defined as Over Sampling Rate (OSR) @@ -1293,7 +1293,7 @@ static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); unsigned int ctrl1_val = 0, ctrl2_val = 0; - nau8825_sema_acquire(nau8825, 2 * HZ); + nau8825_sema_acquire(nau8825, 3 * HZ); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -2121,7 +2121,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, * fered by cross talk process, the driver make the playback * preparation halted until cross talk process finish. */ - nau8825_sema_acquire(nau8825, 2 * HZ); + nau8825_sema_acquire(nau8825, 3 * HZ); nau8825_configure_mclk_as_sysclk(regmap); /* MCLK not changed by clock tree */ regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, @@ -2169,7 +2169,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, * fered by cross talk process, the driver make the playback * preparation halted until cross talk process finish. */ - nau8825_sema_acquire(nau8825, 2 * HZ); + nau8825_sema_acquire(nau8825, 3 * HZ); /* Higher FLL reference input frequency can only set lower * gain error, such as 0000 for input reference from MCLK * 12.288Mhz. @@ -2191,7 +2191,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, * fered by cross talk process, the driver make the playback * preparation halted until cross talk process finish. */ - nau8825_sema_acquire(nau8825, 2 * HZ); + nau8825_sema_acquire(nau8825, 3 * HZ); /* If FLL reference input is from low frequency source, * higher error gain can apply such as 0xf which has * the most sensitive gain error correction threshold, @@ -2217,7 +2217,7 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, * fered by cross talk process, the driver make the playback * preparation halted until cross talk process finish. */ - nau8825_sema_acquire(nau8825, 2 * HZ); + nau8825_sema_acquire(nau8825, 3 * HZ); /* If FLL reference input is from low frequency source, * higher error gain can apply such as 0xf which has * the most sensitive gain error correction threshold, -- 2.6.4

2 1

[alsa-devel] [PATCH v2] ASoC: intel: Fix crash at suspend/resume without card registration
by Takashi Iwai 30 Nov '16

30 Nov '16

When ASoC Intel SST Medfield driver is probed but without codec / card assigned, it causes an Oops and freezes the kernel at suspend/resume, PM: Suspending system (freeze) Suspending console(s) (use no_console_suspend to debug) BUG: unable to handle kernel NULL pointer dereference at 0000000000000018 IP: [<ffffffffc09d9409>] sst_soc_prepare+0x19/0xa0 [snd_soc_sst_mfld_platform] Oops: 0000 [#1] PREEMPT SMP CPU: 0 PID: 1552 Comm: systemd-sleep Tainted: G W 4.9.0-rc6-1.g5f5c2ad-default #1 Call Trace: [<ffffffffb45318f9>] dpm_prepare+0x209/0x460 [<ffffffffb4531b61>] dpm_suspend_start+0x11/0x60 [<ffffffffb40d3cc2>] suspend_devices_and_enter+0xb2/0x710 [<ffffffffb40d462e>] pm_suspend+0x30e/0x390 [<ffffffffb40d2eba>] state_store+0x8a/0x90 [<ffffffffb43c670f>] kobj_attr_store+0xf/0x20 [<ffffffffb42b0d97>] sysfs_kf_write+0x37/0x40 [<ffffffffb42b02bc>] kernfs_fop_write+0x11c/0x1b0 [<ffffffffb422be68>] __vfs_write+0x28/0x140 [<ffffffffb43728a8>] ? apparmor_file_permission+0x18/0x20 [<ffffffffb433b2ab>] ? security_file_permission+0x3b/0xc0 [<ffffffffb422d095>] vfs_write+0xb5/0x1a0 [<ffffffffb422e3d6>] SyS_write+0x46/0xa0 [<ffffffffb4719fbb>] entry_SYSCALL_64_fastpath+0x1e/0xad Add proper NULL checks in the PM code of mdfld driver. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: drop the bogus return code from sst_soc_complete(). sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 25c6d87c818e..76bcb947b31c 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -771,6 +771,9 @@ static int sst_soc_prepare(struct device *dev) struct sst_data *drv = dev_get_drvdata(dev); struct snd_soc_pcm_runtime *rtd; + if (!drv->soc_card) + return 0; + /* suspend all pcms first */ snd_soc_suspend(drv->soc_card->dev); snd_soc_poweroff(drv->soc_card->dev); @@ -793,6 +796,9 @@ static void sst_soc_complete(struct device *dev) struct sst_data *drv = dev_get_drvdata(dev); struct snd_soc_pcm_runtime *rtd; + if (!drv->soc_card) + return; + /* restart SSPs */ list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) { struct snd_soc_dai *dai = rtd->cpu_dai; -- 2.10.2

3 2

[alsa-devel] [PATCH] ASoC: cht_bsw_rt5672: Use HID translation unit
by Takashi Iwai 30 Nov '16

30 Nov '16

Instead of hard-coded "i2c-10EC5670:00", use the translation helper to avoid the mismatch between i2c-codec and ACPI strings just like what we've done for bytcr_rt5640. This gives more robust binding on funky devices like Dell Wyse 3040. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/boards/cht_bsw_rt5672.c | 30 ++++++++++++++++++++++++++++-- 1 file changed, 28 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index df9d254baa18..fe607d8701a0 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -25,12 +25,14 @@ #include <sound/jack.h> #include "../../codecs/rt5670.h" #include "../atom/sst-atom-controls.h" +#include "../common/sst-acpi.h" /* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */ #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI "rt5670-aif1" static struct snd_soc_jack cht_bsw_headset; +static char cht_bsw_codec_name[16]; /* Headset jack detection DAPM pins */ static struct snd_soc_jack_pin cht_bsw_headset_pins[] = { @@ -295,7 +297,7 @@ static int cht_suspend_pre(struct snd_soc_card *card) struct snd_soc_codec *codec; list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + if (!strcmp(codec->component.name, cht_bsw_codec_name)) { dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); rt5670_jack_suspend(codec); break; @@ -309,7 +311,7 @@ static int cht_resume_post(struct snd_soc_card *card) struct snd_soc_codec *codec; list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + if (!strcmp(codec->component.name, cht_bsw_codec_name)) { dev_dbg(codec->dev, "enabling jack detect for resume.\n"); rt5670_jack_resume(codec); break; @@ -335,9 +337,33 @@ static struct snd_soc_card snd_soc_card_cht = { .resume_post = cht_resume_post, }; +#define RT5672_I2C_DEFAULT "i2c-10EC5670:00" + static int snd_cht_mc_probe(struct platform_device *pdev) { int ret_val = 0; + struct sst_acpi_mach *mach = pdev->dev.platform_data; + const char *i2c_name; + int i; + + strcpy(cht_bsw_codec_name, RT5672_I2C_DEFAULT); + + /* fixup codec name based on HID */ + if (mach) { + i2c_name = sst_acpi_find_name_from_hid(mach->id); + if (i2c_name) { + snprintf(cht_bsw_codec_name, sizeof(cht_bsw_codec_name), + "i2c-%s", i2c_name); + for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) { + if (!strcmp(cht_dailink[i].codec_name, + RT5672_I2C_DEFAULT)) { + cht_dailink[i].codec_name = + cht_bsw_codec_name; + break; + } + } + } + } /* register the soc card */ snd_soc_card_cht.dev = &pdev->dev; -- 2.10.2

4 4

[alsa-devel] [PATCH 00/10] ASoC: sunxi: Add support for audio codec in A23/H3 SoCs
by Chen-Yu Tsai 30 Nov '16

30 Nov '16

Hi everyone, This series adds support for the audio codec found in Allwinner A23 and H3 SoCs. The design and data paths are similar to the audio codec found in earlier SoCs such as the A31. The analog audio paths are symmetrical with left/right channels and down-mix selectors for mono differential output. What deviates from previous SoCs is that the analog path controls have been moved to a separate control bus, accessed through a message box like register interface in the PRCM block. This necessitates writing a separate component driver for it, which is then tied into the sound card as an ASoC auxiliary device. Patches 1 and 2 add the binding and driver for the analog path control block. This is the more complete version. Previously I provided a not fully tested version to Mylene Josserand from Free Electrons to use with the A33. Their version was then trimmed down and switched to SOC_DAPM_SINGLE controls. Mine implements all the commonly used paths, and uses the new stereo SOC_DAPM_DOUBLE controls. I also have a separate patch for "Phone In" in case anyone wants to test them. It is not included as my hardware does not use that input. As for the A33, the analog controls are exactly the same as the A23, so this driver can be reused, but the PCM and DAI bits are completely different. Patch 3 adds the analog path controls block to the sun6i-prcm driver as a sub-device, for the A23. The H3 currently does not use the PRCM driver. Patch 4 adds PCM and card support for the A23 codec to the sun4i-codec driver. Patch 5 adds a device node for the analog path controls block to the A23 dtsi. Patch 6 adds a device node for the audio codec, and the phandle for the analog path controls block to the A23 dtsi. Patch 7 enables the audio codec for the A23 Q8 tablets. On these tablets the headphone output is driven in DC coupled, or "direct drive", mode. Patch 8 adds PCM and card support for the H3 codec to the sun4i-codec driver. Patch 9 adds device nodes for the audio codec and analog path controls block to the H3 dtsi. Patch 10 enables the audio codec on the Orange Pi PC. The audio output jack on the board is tied to the line out pins on the SoC. Please take a look and let me know what you think. In addition, the sun4i-codec driver is getting pretty large. Maybe we want to split the different parts into different files? Regards ChenYu Chen-Yu Tsai (10): ASoC: sunxi: Add bindings for A23/A33/H3 codec's analog path controls ASoC: sunxi: Add support for A23/A33/H3 codec's analog path controls mfd: sun6i-prcm: Add codec analog controls sub-device for Allwinner A23 ASoC: sun4i-codec: Add support for A23 codec ARM: dts: sun8i: Add codec analog path controls node in PRCM for A23/A33 ARM: dts: sun8i-a23: Add device node for internal audio codec ARM: dts: sun8i-a23: q8-tablet: Enable internal audio codec ASoC: sun4i-codec: Add support for H3 codec ARM: dts: sun8i-h3: Add device nodes for audio codec and its analog controls ARM: dts: sun8i-h3: orange-pi-pc: Enable audio codec .../devicetree/bindings/sound/sun4i-codec.txt | 14 +- .../bindings/sound/sun8i-codec-analog.txt | 16 + arch/arm/boot/dts/sun8i-a23-a33.dtsi | 4 + arch/arm/boot/dts/sun8i-a23-q8-tablet.dts | 23 + arch/arm/boot/dts/sun8i-a23.dtsi | 16 + arch/arm/boot/dts/sun8i-h3-orangepi-pc.dts | 8 + arch/arm/boot/dts/sun8i-h3.dtsi | 19 + drivers/mfd/sun6i-prcm.c | 17 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun4i-codec.c | 179 ++++++ sound/soc/sunxi/sun8i-codec-analog.c | 665 +++++++++++++++++++++ 12 files changed, 968 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt create mode 100644 sound/soc/sunxi/sun8i-codec-analog.c -- 2.10.2

5 21

[alsa-devel] [PATCH resend] ASoC: Intel: Add missing 10EC5672 ACPI ID matching for Cherry Trail
by Takashi Iwai 30 Nov '16

30 Nov '16

Add the missing ACPI ID 10EC5672 for Cherry Trail, which bounds with cht-bsw-rt5672 driver. This combination was found on Dell Wyse 3040. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Resent, as I forgot Cc to alsa-devel ML, sorry! sound/soc/intel/atom/sst/sst_acpi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 0a88537ca58a..f4d92bbc5373 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -452,6 +452,8 @@ static struct sst_acpi_mach sst_acpi_bytcr[] = { static struct sst_acpi_mach sst_acpi_chv[] = { {"10EC5670", "cht-bsw-rt5672", "intel/fw_sst_22a8.bin", "cht-bsw", NULL, &chv_platform_data }, + {"10EC5672", "cht-bsw-rt5672", "intel/fw_sst_22a8.bin", "cht-bsw", NULL, + &chv_platform_data }, {"10EC5645", "cht-bsw-rt5645", "intel/fw_sst_22a8.bin", "cht-bsw", NULL, &chv_platform_data }, {"10EC5650", "cht-bsw-rt5645", "intel/fw_sst_22a8.bin", "cht-bsw", NULL, -- 2.10.2

3 2

[alsa-devel] [PATCH] ASoC: rt5670: Add missing 10EC5072 ACPI ID
by Takashi Iwai 30 Nov '16

30 Nov '16

rt5670 driver supports also RT5672 codec, but its ACPI ID is missing. This was found on Dell Wyse 3040 box. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/rt5670.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 12d3bc14a558..98c7efbc6ee0 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2813,6 +2813,7 @@ MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); #ifdef CONFIG_ACPI static const struct acpi_device_id rt5670_acpi_match[] = { { "10EC5670", 0}, + { "10EC5672", 0}, { }, }; MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match); -- 2.10.2

2 1

[alsa-devel] [PATCH 1/1] ALSA: SOC: DMA: increment buffer pointer atomically
by Jiada Wang 30 Nov '16

30 Nov '16

From: Andreas Pape <apape(a)de.adit-jv.com> Setting pointer and afterwards check for wrap around leads to the possibility of returning the inconsistent pointer position. This patch increments buffer pointer atomically to avoid this issue. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/core/pcm_dmaengine.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 8eb58c7..6f6da11 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -139,12 +139,14 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_config_from_dai_data); static void dmaengine_pcm_dma_complete(void *arg) { + unsigned int new_pos; struct snd_pcm_substream *substream = arg; struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - prtd->pos += snd_pcm_lib_period_bytes(substream); - if (prtd->pos >= snd_pcm_lib_buffer_bytes(substream)) - prtd->pos = 0; + new_pos = prtd->pos + snd_pcm_lib_period_bytes(substream); + if (new_pos >= snd_pcm_lib_buffer_bytes(substream)) + new_pos = 0; + prtd->pos = new_pos; snd_pcm_period_elapsed(substream); } -- 2.9.3

2 3

[alsa-devel] [PATCH v4 0/3] ASoC: Component has suspend/resume support
by Kuninori Morimoto 30 Nov '16

30 Nov '16

Hi Mark, Lars-Peter These patches are v4+RFC of Component level suspend/resume support. I had waited Vinod's cleanup patch which will remove "codec_dev_list" from Intel drivers. But it seems nothing happen at this point. So, this v4 patch-set expects it will be reviwed/accepted *before* Vinod's patch. Kuninori Morimoto (3): ASoC: core: replace codec_dev_list to component_dev_list on Card ASoC: core: replace aux_comp_list to component_dev_list ASoC: add Component level suspend/resume include/sound/soc.h | 16 +++--- sound/soc/intel/boards/broadwell.c | 16 +++--- sound/soc/intel/boards/cht_bsw_rt5672.c | 16 +++--- sound/soc/soc-core.c | 90 ++++++++++++++++++++------------- 4 files changed, 85 insertions(+), 53 deletions(-) -- 1.9.1

1 3

[alsa-devel] TEST
by Jaechul Lee 30 Nov '16

30 Nov '16

TEST?

1 0

[alsa-devel] [PATCH] ALSA: line6: Claim POD X3 USB data interface
by Andrej Krutak 29 Nov '16

29 Nov '16

Deny userspace from using the interface concurrently. Signed-off-by: Andrej Krutak <dev(a)andree.sk> --- sound/usb/line6/driver.h | 1 + sound/usb/line6/podhd.c | 24 ++++++++++++++++++++++++ 2 files changed, 25 insertions(+) diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index 7e3a3aa..5d18957 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -98,6 +98,7 @@ struct line6_properties { int altsetting; + unsigned ctrl_if; unsigned ep_ctrl_r; unsigned ep_ctrl_w; unsigned ep_audio_r; diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index 49cd4a6..6c53b87 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -153,6 +153,7 @@ struct usb_line6_podhd { .rats = &podhd_ratden}, .bytes_per_channel = 3 /* SNDRV_PCM_FMTBIT_S24_3LE */ }; +static struct usb_driver podhd_driver; static void podhd_startup_start_workqueue(unsigned long data); static void podhd_startup_workqueue(struct work_struct *work); @@ -291,8 +292,13 @@ static void podhd_disconnect(struct usb_line6 *line6) struct usb_line6_podhd *pod = (struct usb_line6_podhd *)line6; if (pod->line6.properties->capabilities & LINE6_CAP_CONTROL) { + struct usb_interface *intf; + del_timer_sync(&pod->startup_timer); cancel_work_sync(&pod->startup_work); + + intf = usb_ifnum_to_if(line6->usbdev, pod->line6.properties->ctrl_if); + usb_driver_release_interface(&podhd_driver, intf); } } @@ -304,10 +310,26 @@ static int podhd_init(struct usb_line6 *line6, { int err; struct usb_line6_podhd *pod = (struct usb_line6_podhd *) line6; + struct usb_interface *intf; line6->disconnect = podhd_disconnect; if (pod->line6.properties->capabilities & LINE6_CAP_CONTROL) { + /* claim the data interface */ + intf = usb_ifnum_to_if(line6->usbdev, pod->line6.properties->ctrl_if); + if (!intf) { + dev_err(pod->line6.ifcdev, "interface %d not found\n", + pod->line6.properties->ctrl_if); + return -ENODEV; + } + + err = usb_driver_claim_interface(&podhd_driver, intf, NULL); + if (err != 0) { + dev_err(pod->line6.ifcdev, "can't claim interface %d, error %d\n", + pod->line6.properties->ctrl_if, err); + return err; + } + /* create sysfs entries: */ err = snd_card_add_dev_attr(line6->card, &podhd_dev_attr_group); if (err < 0) @@ -406,6 +428,7 @@ static int podhd_init(struct usb_line6 *line6, .altsetting = 1, .ep_ctrl_r = 0x81, .ep_ctrl_w = 0x01, + .ctrl_if = 1, .ep_audio_r = 0x86, .ep_audio_w = 0x02, }, @@ -417,6 +440,7 @@ static int podhd_init(struct usb_line6 *line6, .altsetting = 1, .ep_ctrl_r = 0x81, .ep_ctrl_w = 0x01, + .ctrl_if = 1, .ep_audio_r = 0x86, .ep_audio_w = 0x02, }, -- 1.9.1

2 6

[alsa-devel] [PATCH] ALSA: usb-audio: Fix bogus error return in snd_usb_create_stream()
by Takashi Iwai 29 Nov '16

29 Nov '16

The function returns -EINVAL even if it builds the stream properly. The bogus error code sneaked in during the code refactoring, but it wasn't noticed until now since the returned error code itself is ignored in anyway. Kill it here, but there is no behavior change by this patch, obviously. Fixes: e5779998bf8b ('ALSA: usb-audio: refactor code') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 2ddc034673a8..f36cb068dad3 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -206,7 +206,6 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int if (! snd_usb_parse_audio_interface(chip, interface)) { usb_set_interface(dev, interface, 0); /* reset the current interface */ usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - return -EINVAL; } return 0; -- 2.10.2

1 0

[alsa-devel] [PATCH] ALSA: echoaudio: Fix improper return value in function load_asic
by Takashi Iwai 29 Nov '16

29 Nov '16

From: Pan Bian <bianpan2016(a)163.com> When the second call to load_asic_generic() fails in function load_asic(), "false" is returned. The real value of "false" is 0, which indicates success in the context. As a result, the execution status and the return value may be inconsistent. This patch fixes the bug. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=188761 Signed-off-by: Pan Bian <bianpan2016(a)163.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/echoaudio/layla24_dsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index df28e5117359..c02bc1dcc170 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -135,7 +135,7 @@ static int load_asic(struct echoaudio *chip) err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC, FW_LAYLA24_2S_ASIC); if (err < 0) - return false; + return err; /* Now give the external ASIC a little time to set up */ mdelay(10); -- 2.10.2

1 0

[alsa-devel] --max-file-time producing incorrect results
by Rob Nertney 29 Nov '16

29 Nov '16

I have a diagnostic mode for my hardware, which gives me a known pattern in 1 second increments (for my predetermined frame size/rate). If I simply pass a -d duration flag over and over with arecord, every capture is perfect when I inspect it. When I use the --max-file-time=1, my first PCM is perfect, then the following ones aren't. It appears that in between these 1-second PCM files, I am losing 128 frames between the 1-second files. I'm not sure what could be causing this type of error; any thoughts? Thanks, Rob

1 1

Re: [alsa-devel] [RFC PATCH v3 2/7] drm/i915: Add support for audio driver notifications
by Ville Syrjälä 29 Nov '16

29 Nov '16

On Fri, Nov 25, 2016 at 05:25:43AM +0530, Jerome Anand wrote: > Notifiations like mode change, hot plug and edid to > the audio driver are added. This is inturn used by the > audio driver for its functionality. > > A new interface file capturing the notifications needed by the > audio driver is added > > Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> > Signed-off-by: Jerome Anand <jerome.anand(a)intel.com> > --- > drivers/gpu/drm/i915/i915_drv.h | 3 +++ > drivers/gpu/drm/i915/intel_audio.c | 8 ++++++ > drivers/gpu/drm/i915/intel_hdmi.c | 1 + > drivers/gpu/drm/i915/intel_lpe_audio.c | 49 ++++++++++++++++++++++++++++++++++ > include/drm/intel_lpe_audio.h | 1 + > 5 files changed, 62 insertions(+) > > diff --git a/drivers/gpu/drm/i915/i915_drv.h b/drivers/gpu/drm/i915/i915_drv.h > index 2a79048..33bc44c 100644 > --- a/drivers/gpu/drm/i915/i915_drv.h > +++ b/drivers/gpu/drm/i915/i915_drv.h > @@ -3561,6 +3561,9 @@ int intel_lpe_audio_setup(struct drm_i915_private *dev_priv); > void intel_lpe_audio_teardown(struct drm_i915_private *dev_priv); > void intel_lpe_audio_irq_handler(struct drm_i915_private *dev_priv); > bool intel_lpe_audio_detect(struct drm_i915_private *dev_priv); > +void intel_lpe_audio_notify(struct drm_i915_private *dev_priv, > + void *eld, int port, int tmds_clk_speed, > + bool connected); > > /* intel_i2c.c */ > extern int intel_setup_gmbus(struct drm_device *dev); > diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c > index 1c509f7..55a6831 100644 > --- a/drivers/gpu/drm/i915/intel_audio.c > +++ b/drivers/gpu/drm/i915/intel_audio.c > @@ -24,6 +24,7 @@ > #include <linux/kernel.h> > #include <linux/component.h> > #include <drm/i915_component.h> > +#include <drm/intel_lpe_audio.h> > #include "intel_drv.h" > > #include <drm/drmP.h> > @@ -627,6 +628,10 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder, > if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) > acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, > (int) port, (int) pipe); > + > + if (HAS_LPE_AUDIO(dev_priv)) > + intel_lpe_audio_notify(dev_priv, connector->eld, port, > + crtc_state->port_clock, true); > } > > /** > @@ -660,6 +665,9 @@ void intel_audio_codec_disable(struct intel_encoder *intel_encoder) > if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) > acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, > (int) port, (int) pipe); > + > + if (HAS_LPE_AUDIO(dev_priv)) > + intel_lpe_audio_notify(dev_priv, NULL, port, 0, true); > } > > /** > diff --git a/drivers/gpu/drm/i915/intel_hdmi.c b/drivers/gpu/drm/i915/intel_hdmi.c > index fb88e32..02d50e3 100644 > --- a/drivers/gpu/drm/i915/intel_hdmi.c > +++ b/drivers/gpu/drm/i915/intel_hdmi.c > @@ -36,6 +36,7 @@ > #include <drm/drm_edid.h> > #include "intel_drv.h" > #include <drm/i915_drm.h> > +#include <drm/intel_lpe_audio.h> > #include "i915_drv.h" > > static struct drm_device *intel_hdmi_to_dev(struct intel_hdmi *intel_hdmi) > diff --git a/drivers/gpu/drm/i915/intel_lpe_audio.c b/drivers/gpu/drm/i915/intel_lpe_audio.c > index 5335fc6..93f83cb 100644 > --- a/drivers/gpu/drm/i915/intel_lpe_audio.c > +++ b/drivers/gpu/drm/i915/intel_lpe_audio.c > @@ -367,3 +367,52 @@ void intel_lpe_audio_teardown(struct drm_i915_private *dev_priv) > > spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); > } > + > + > +/** > + * intel_lpe_audio_notify() - notify lpe audio event > + * audio driver and i915 > + * @dev_priv: the i915 drm device private data > + * @eld : ELD data > + * @port: port id > + * @tmds_clk_speed: tmds clock frequency in Hz > + * @connected: hdmi connected/disconnected > + * > + * Notify lpe audio driver of eld change. > + */ > +void intel_lpe_audio_notify(struct drm_i915_private *dev_priv, > + void *eld, int port, int tmds_clk_speed, > + bool connected) > +{ > + unsigned long irq_flags; > + > + if (HAS_LPE_AUDIO(dev_priv)) { > + struct intel_hdmi_lpe_audio_pdata *pdata = dev_get_platdata( > + &(dev_priv->lpe_audio.platdev->dev)); > + > + if (pdata) { How could the !pdata case happen? If it can't we shouldn't cater for it. > + spin_lock_irqsave(&pdata->lpe_audio_slock, > + irq_flags); > + > + if (eld != NULL) { > + memcpy(pdata->eld.eld_data, eld, > + HDMI_MAX_ELD_BYTES); > + pdata->eld.port_id = port; > + > + if (tmds_clk_speed) Why the if? > + pdata->tmds_clock_speed = > + tmds_clk_speed; > + } > + pdata->hdmi_connected = connected; Also can't really see much need for this. Any one of the port/tmds clock/eld should be sufficient. > + if (pdata->notify_audio_lpe) > + pdata->notify_audio_lpe( > + (eld != NULL) ? &pdata->eld : NULL); > + else > + pdata->notify_pending = true; Still not sure why the "pending" thing is useful. Can't the audio driver just do its thing (whatever it is) unconditionally? When disabling just clear the port to INVALID, eld to zero, and tmds clock to 0, and it should all be fine no? > + > + spin_unlock_irqrestore(&pdata->lpe_audio_slock, > + irq_flags); > + } else > + DRM_DEBUG_DRIVER("no audio notification\n"); > + } > +} > diff --git a/include/drm/intel_lpe_audio.h b/include/drm/intel_lpe_audio.h > index a64c449..952de05 100644 > --- a/include/drm/intel_lpe_audio.h > +++ b/include/drm/intel_lpe_audio.h > @@ -25,6 +25,7 @@ > #define _INTEL_LPE_AUDIO_H_ > > #include <linux/types.h> > +#include <linux/spinlock_types.h> > > #define HDMI_MAX_ELD_BYTES 128 > > -- > 2.9.3 -- Ville Syrjälä Intel OTC

4 9

[alsa-devel] [PATCH][alsa-lib] topology: fix unused-variable warnings introduced to build_link()
by Takashi Sakamoto 29 Nov '16

29 Nov '16

A function with unused automatic variables was recently introduced to this library, then compiler generates below warnings. pcm.c: In function ‘build_link’: pcm.c:213:6: warning: unused variable ‘i’ [-Wunused-variable] int i, num_hw_configs = 0, err = 0; ^ pcm.c:210:43: warning: unused variable ‘cmpnt’ [-Wunused-variable] struct snd_soc_tplg_link_cmpnt *codec, *cmpnt; ^~~~~ pcm.c:210:35: warning: unused variable ‘codec’ [-Wunused-variable] struct snd_soc_tplg_link_cmpnt *codec, *cmpnt; ^~~~~ pcm.c:209:20: warning: unused variable ‘ref_elem’ [-Wunused-variable] struct tplg_elem *ref_elem = NULL; This commit removes them. Fixes: 53f7711769d4("topology: Define a function to build a single physical DAI link") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/topology/pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/src/topology/pcm.c b/src/topology/pcm.c index 2e0a120..8f8a703 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -206,11 +206,9 @@ static int tplg_build_stream_cfg(snd_tplg_t *tplg, static int build_link(snd_tplg_t *tplg, struct tplg_elem *elem) { struct snd_soc_tplg_link_config *link = elem->link; - struct tplg_elem *ref_elem = NULL; - struct snd_soc_tplg_link_cmpnt *codec, *cmpnt; struct tplg_ref *ref; struct list_head *base, *pos; - int i, num_hw_configs = 0, err = 0; + int num_hw_configs = 0, err = 0; err = tplg_build_stream_cfg(tplg, link->stream, link->num_streams); -- 2.9.3

1 0

[alsa-devel] [PATCH] pcm: dmix: Do not discard slave reported delay in status result
by Takashi Iwai 29 Nov '16

29 Nov '16

Like the change done for dshare plugin, we can calculate the delay more precisely from the slave PCM. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_dmix.c | 45 ++++++++++++++++++++++++++++----------------- 1 file changed, 28 insertions(+), 17 deletions(-) diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 2fedb1c14a3b..825677ffeb3f 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -391,23 +391,14 @@ static void snd_pcm_dmix_sync_area(snd_pcm_t *pcm) /* * synchronize hardware pointer (hw_ptr) with ours */ -static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) +static int snd_pcm_dmix_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_ptr) { snd_pcm_direct_t *dmix = pcm->private_data; - snd_pcm_uframes_t slave_hw_ptr, old_slave_hw_ptr, avail; + snd_pcm_uframes_t old_slave_hw_ptr, avail; snd_pcm_sframes_t diff; - switch (snd_pcm_state(dmix->spcm)) { - case SND_PCM_STATE_DISCONNECTED: - dmix->state = SND_PCM_STATE_DISCONNECTED; - return -ENODEV; - default: - break; - } - if (dmix->slowptr) - snd_pcm_hwsync(dmix->spcm); old_slave_hw_ptr = dmix->slave_hw_ptr; - slave_hw_ptr = dmix->slave_hw_ptr = *dmix->spcm->hw.ptr; + dmix->slave_hw_ptr = *dmix->spcm->hw.ptr; diff = slave_hw_ptr - old_slave_hw_ptr; if (diff == 0) /* fast path */ return 0; @@ -440,6 +431,24 @@ static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) return 0; } +static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) +{ + snd_pcm_direct_t *dmix = pcm->private_data; + + switch (snd_pcm_state(dmix->spcm)) { + case SND_PCM_STATE_DISCONNECTED: + dmix->state = SND_PCM_STATE_DISCONNECTED; + return -ENODEV; + default: + break; + } + + if (dmix->slowptr) + snd_pcm_hwsync(dmix->spcm); + + return snd_pcm_dmix_sync_ptr0(pcm, *dmix->spcm->hw.ptr); +} + /* * plugin implementation */ @@ -467,22 +476,24 @@ static int snd_pcm_dmix_status(snd_pcm_t *pcm, snd_pcm_status_t * status) { snd_pcm_direct_t *dmix = pcm->private_data; + memset(status, 0, sizeof(*status)); + snd_pcm_status(dmix->spcm, status); + switch (dmix->state) { case SNDRV_PCM_STATE_DRAINING: case SNDRV_PCM_STATE_RUNNING: - snd_pcm_dmix_sync_ptr(pcm); + snd_pcm_dmix_sync_ptr0(pcm, status->hw_ptr); + status->delay += snd_pcm_mmap_playback_delay(pcm) + + status->avail - dmix->spcm->buffer_size; break; default: break; } - memset(status, 0, sizeof(*status)); - snd_pcm_status(dmix->spcm, status); - status->state = snd_pcm_dmix_state(pcm); + status->trigger_tstamp = dmix->trigger_tstamp; status->avail = snd_pcm_mmap_playback_avail(pcm); status->avail_max = status->avail > dmix->avail_max ? status->avail : dmix->avail_max; dmix->avail_max = 0; - status->delay = snd_pcm_mmap_playback_delay(pcm); return 0; } -- 2.10.2

1 0

[alsa-devel] [PATCH alsa-lib] pcm: Add the PCM state checks to plugins
by Takashi Iwai 29 Nov '16

29 Nov '16

I noticed that some plugin codes have no proper PCM state checks and it results in expected outcomes. For example, when snd_pcm_drain() is called for a dmix PCM after calling snd_pcm_drop(), it stalls unexpectedly. It's just because its drain callback doesn't expect the SND_PCM_SETUP state. We can fix such a bug in each place one by one, but a safer way would be to filter out all such cases commonly in the PCM API functions themselves. This patch adds the PCM state sanity checks to major API functions so that they return -EBADFD when called in the unexpected PCM states. As well as for the thread-safety extension, it'd be a question of the performance; again at this time, the hw PCM is considered as an exception, and it has pcm->own_state_check flag set, which means that the common PCM state checks are skipped. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm.c | 48 ++++++++++++++++++++++++++++++++++++++++++++++++ src/pcm/pcm_hw.c | 1 + src/pcm/pcm_local.h | 1 + 3 files changed, 50 insertions(+) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index cd87bc759ded..f2ca02b55973 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -655,6 +655,23 @@ playback devices. #include <limits.h> #include "pcm_local.h" +#ifndef DOC_HIDDEN +#define P_STATE(x) (1U << SND_PCM_STATE_ ## x) +#define P_STATE_RUNNABLE (P_STATE(PREPARED) | \ + P_STATE(RUNNING) | \ + P_STATE(XRUN) | \ + P_STATE(PAUSED) | \ + P_STATE(DRAINING)) + +/* check whether the PCM is in the unexpected state */ +static int bad_pcm_state(snd_pcm_t *pcm, unsigned int supported_states) +{ + if (pcm->own_state_check) + return 0; /* don't care, the plugin checks by itself */ + return !(supported_states & (1U << snd_pcm_state(pcm))); +} +#endif + /** * \brief get identifier of PCM handle * \param pcm PCM handle @@ -1120,6 +1137,8 @@ int snd_pcm_prepare(snd_pcm_t *pcm) SNDMSG("PCM not set up"); return -EIO; } + if (bad_pcm_state(pcm, ~P_STATE(DISCONNECTED))) + return -EBADFD; snd_pcm_lock(pcm); err = pcm->fast_ops->prepare(pcm->fast_op_arg); snd_pcm_unlock(pcm); @@ -1166,6 +1185,8 @@ int snd_pcm_start(snd_pcm_t *pcm) SNDMSG("PCM not set up"); return -EIO; } + if (bad_pcm_state(pcm, P_STATE(PREPARED))) + return -EBADFD; snd_pcm_lock(pcm); err = __snd_pcm_start(pcm); snd_pcm_unlock(pcm); @@ -1194,6 +1215,9 @@ int snd_pcm_drop(snd_pcm_t *pcm) SNDMSG("PCM not set up"); return -EIO; } + if (bad_pcm_state(pcm, P_STATE_RUNNABLE | P_STATE(SETUP) | + P_STATE(SUSPENDED))) + return -EBADFD; snd_pcm_lock(pcm); err = pcm->fast_ops->drop(pcm->fast_op_arg); snd_pcm_unlock(pcm); @@ -1222,6 +1246,8 @@ int snd_pcm_drain(snd_pcm_t *pcm) SNDMSG("PCM not set up"); return -EIO; } + if (bad_pcm_state(pcm, P_STATE_RUNNABLE)) + return -EBADFD; /* lock handled in the callback */ return pcm->fast_ops->drain(pcm->fast_op_arg); } @@ -1247,6 +1273,8 @@ int snd_pcm_pause(snd_pcm_t *pcm, int enable) SNDMSG("PCM not set up"); return -EIO; } + if (bad_pcm_state(pcm, P_STATE_RUNNABLE)) + return -EBADFD; snd_pcm_lock(pcm); err = pcm->fast_ops->pause(pcm->fast_op_arg, enable); snd_pcm_unlock(pcm); @@ -1273,6 +1301,8 @@ snd_pcm_sframes_t snd_pcm_rewindable(snd_pcm_t *pcm) SNDMSG("PCM not set up"); return -EIO; } + if (bad_pcm_state(pcm, P_STATE_RUNNABLE)) + return -EBADFD; snd_pcm_lock(pcm); result = pcm->fast_ops->rewindable(pcm->fast_op_arg); snd_pcm_unlock(pcm); @@ -1299,6 +1329,8 @@ snd_pcm_sframes_t snd_pcm_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t frames) } if (frames == 0) return 0; + if (bad_pcm_state(pcm, P_STATE_RUNNABLE)) + return -EBADFD; snd_pcm_lock(pcm); result = pcm->fast_ops->rewind(pcm->fast_op_arg, frames); snd_pcm_unlock(pcm); @@ -1325,6 +1357,8 @@ snd_pcm_sframes_t snd_pcm_forwardable(snd_pcm_t *pcm) SNDMSG("PCM not set up"); return -EIO; } + if (bad_pcm_state(pcm, P_STATE_RUNNABLE)) + return -EBADFD; snd_pcm_lock(pcm); result = pcm->fast_ops->forwardable(pcm->fast_op_arg); snd_pcm_unlock(pcm); @@ -1355,6 +1389,8 @@ snd_pcm_sframes_t snd_pcm_forward(snd_pcm_t *pcm, snd_pcm_uframes_t frames) } if (frames == 0) return 0; + if (bad_pcm_state(pcm, P_STATE_RUNNABLE)) + return -EBADFD; snd_pcm_lock(pcm); result = pcm->fast_ops->forward(pcm->fast_op_arg, frames); snd_pcm_unlock(pcm); @@ -1393,6 +1429,8 @@ snd_pcm_sframes_t snd_pcm_writei(snd_pcm_t *pcm, const void *buffer, snd_pcm_ufr SNDMSG("invalid access type %s", snd_pcm_access_name(pcm->access)); return -EINVAL; } + if (bad_pcm_state(pcm, P_STATE_RUNNABLE)) + return -EBADFD; return _snd_pcm_writei(pcm, buffer, size); } @@ -1427,6 +1465,8 @@ snd_pcm_sframes_t snd_pcm_writen(snd_pcm_t *pcm, void **bufs, snd_pcm_uframes_t SNDMSG("invalid access type %s", snd_pcm_access_name(pcm->access)); return -EINVAL; } + if (bad_pcm_state(pcm, P_STATE_RUNNABLE)) + return -EBADFD; return _snd_pcm_writen(pcm, bufs, size); } @@ -1461,6 +1501,8 @@ snd_pcm_sframes_t snd_pcm_readi(snd_pcm_t *pcm, void *buffer, snd_pcm_uframes_t SNDMSG("invalid access type %s", snd_pcm_access_name(pcm->access)); return -EINVAL; } + if (bad_pcm_state(pcm, P_STATE_RUNNABLE)) + return -EBADFD; return _snd_pcm_readi(pcm, buffer, size); } @@ -1495,6 +1537,8 @@ snd_pcm_sframes_t snd_pcm_readn(snd_pcm_t *pcm, void **bufs, snd_pcm_uframes_t s SNDMSG("invalid access type %s", snd_pcm_access_name(pcm->access)); return -EINVAL; } + if (bad_pcm_state(pcm, P_STATE_RUNNABLE)) + return -EBADFD; return _snd_pcm_readn(pcm, bufs, size); } @@ -6938,6 +6982,8 @@ int snd_pcm_mmap_begin(snd_pcm_t *pcm, { int err; + if (bad_pcm_state(pcm, P_STATE_RUNNABLE)) + return -EBADFD; snd_pcm_lock(pcm); err = __snd_pcm_mmap_begin(pcm, areas, offset, frames); snd_pcm_unlock(pcm); @@ -7033,6 +7079,8 @@ snd_pcm_sframes_t snd_pcm_mmap_commit(snd_pcm_t *pcm, { snd_pcm_sframes_t result; + if (bad_pcm_state(pcm, P_STATE_RUNNABLE)) + return -EBADFD; snd_pcm_lock(pcm); result = __snd_pcm_mmap_commit(pcm, offset, frames); snd_pcm_unlock(pcm); diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 56e88b6bf6c0..30cd5d0f6503 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1516,6 +1516,7 @@ int snd_pcm_hw_open_fd(snd_pcm_t **pcmp, const char *name, #ifdef THREAD_SAFE_API pcm->need_lock = 0; /* hw plugin is thread-safe */ #endif + pcm->own_state_check = 1; /* skip the common state check */ ret = snd_pcm_hw_mmap_status(pcm); if (ret < 0) { diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index bba2f15ac463..32e6dcdf6fcd 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -234,6 +234,7 @@ struct _snd_pcm { * use the mmaped buffer of the slave */ unsigned int donot_close: 1; /* don't close this PCM */ + unsigned int own_state_check:1; /* plugin has own PCM state check */ snd_pcm_channel_info_t *mmap_channels; snd_pcm_channel_area_t *running_areas; snd_pcm_channel_area_t *stopped_areas; -- 2.10.2

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Remove redundant extern declarations
by Charles Keepax 29 Nov '16

29 Nov '16

Functions are given external linkage by default making the extern's unnecessary, as such remove them. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.h | 25 ++++++++++++------------- 1 file changed, 12 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 3f7c8ee..411d062 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -110,18 +110,17 @@ int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -extern int wm_adsp_compr_open(struct wm_adsp *dsp, - struct snd_compr_stream *stream); -extern int wm_adsp_compr_free(struct snd_compr_stream *stream); -extern int wm_adsp_compr_set_params(struct snd_compr_stream *stream, - struct snd_compr_params *params); -extern int wm_adsp_compr_get_caps(struct snd_compr_stream *stream, - struct snd_compr_caps *caps); -extern int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd); -extern int wm_adsp_compr_handle_irq(struct wm_adsp *dsp); -extern int wm_adsp_compr_pointer(struct snd_compr_stream *stream, - struct snd_compr_tstamp *tstamp); -extern int wm_adsp_compr_copy(struct snd_compr_stream *stream, - char __user *buf, size_t count); +int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream); +int wm_adsp_compr_free(struct snd_compr_stream *stream); +int wm_adsp_compr_set_params(struct snd_compr_stream *stream, + struct snd_compr_params *params); +int wm_adsp_compr_get_caps(struct snd_compr_stream *stream, + struct snd_compr_caps *caps); +int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd); +int wm_adsp_compr_handle_irq(struct wm_adsp *dsp); +int wm_adsp_compr_pointer(struct snd_compr_stream *stream, + struct snd_compr_tstamp *tstamp); +int wm_adsp_compr_copy(struct snd_compr_stream *stream, + char __user *buf, size_t count); #endif -- 2.1.4

2 3

[alsa-devel] [PATCH] ASoC: Intel: atom: Make some messages to debug level
by Takashi Iwai 29 Nov '16

29 Nov '16

Intel sst driver spews lots of kernel messages in INFO level; typically, it gives a kernel message at each time it allocates or frees a stream, or at each time when a stream is started or stopped. This will piles up quickly, and it's almost useless for normal users, so let's hide them to KERN_DEBUG level. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/atom/sst-atom-controls.c | 2 +- sound/soc/intel/atom/sst/sst_stream.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 0838478c4c3f..c7b3cbf92faf 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -937,7 +937,7 @@ int send_ssp_cmd(struct snd_soc_dai *dai, const char *id, bool enable) struct sst_data *drv = snd_soc_dai_get_drvdata(dai); int ssp_id; - dev_info(dai->dev, "Enter: enable=%d port_name=%s\n", enable, id); + dev_dbg(dai->dev, "Enter: enable=%d port_name=%s\n", enable, id); if (strcmp(id, "ssp0-port") == 0) ssp_id = SSP_MODEM; diff --git a/sound/soc/intel/atom/sst/sst_stream.c b/sound/soc/intel/atom/sst/sst_stream.c index 4ccc80e5e8cc..51bdeeecb7c8 100644 --- a/sound/soc/intel/atom/sst/sst_stream.c +++ b/sound/soc/intel/atom/sst/sst_stream.c @@ -104,7 +104,7 @@ int sst_alloc_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, void *params) sst_init_stream(&sst_drv_ctx->streams[str_id], alloc_param.codec_type, str_id, alloc_param.operation, 0); - dev_info(sst_drv_ctx->dev, "Alloc for str %d pipe %#x\n", + dev_dbg(sst_drv_ctx->dev, "Alloc for str %d pipe %#x\n", str_id, pipe_id); ret = sst_prepare_and_post_msg(sst_drv_ctx, task_id, IPC_CMD, IPC_IA_ALLOC_STREAM_MRFLD, pipe_id, sizeof(alloc_param), @@ -415,7 +415,7 @@ int sst_free_stream(struct intel_sst_drv *sst_drv_ctx, int str_id) str_info->status = STREAM_UN_INIT; mutex_unlock(&str_info->lock); - dev_info(sst_drv_ctx->dev, "Free for str %d pipe %#x\n", + dev_dbg(sst_drv_ctx->dev, "Free for str %d pipe %#x\n", str_id, str_info->pipe_id); retval = sst_prepare_and_post_msg(sst_drv_ctx, str_info->task_id, IPC_CMD, IPC_IA_FREE_STREAM_MRFLD, str_info->pipe_id, 0, -- 2.10.2

3 2

[alsa-devel] [PATCH 1/2] ASoC: rt298: fix jack type detect error
by Bard Liao 29 Nov '16

29 Nov '16

rt298_jack_detect may be called before card is instantiated. And snd_soc_dapm_force_enable_pin will not work in that case. So, update bit manually by regmap_update_bits. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index f80cfe4..dceffe9 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -249,6 +249,11 @@ static int rt298_jack_detect(struct rt298_priv *rt298, bool *hp, bool *mic) snd_soc_dapm_force_enable_pin(dapm, "LDO1"); snd_soc_dapm_sync(dapm); + regmap_update_bits(rt298->regmap, + RT298_POWER_CTRL1, 0x1001, 0); + regmap_update_bits(rt298->regmap, + RT298_POWER_CTRL2, 0x4, 0x4); + regmap_write(rt298->regmap, RT298_SET_MIC1, 0x24); msleep(50); -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] Add headset Mic support
by Kailang 29 Nov '16

29 Nov '16

Hi Takashi, This patch was turn on support Headset Mic for ALC622. It is USI ODM platform. Many Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH] ALSA SoC MAX98927 driver - Initial release
by Ryan Lee 29 Nov '16

29 Nov '16

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- .../devicetree/bindings/sound/max98927.txt | 32 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98927.c | 954 +++++++++++++++ sound/soc/codecs/max98927.h | 1253 ++++++++++++++++++++ 5 files changed, 2246 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/max98927.txt mode change 100644 => 100755 sound/soc/codecs/Kconfig mode change 100644 => 100755 sound/soc/codecs/Makefile create mode 100755 sound/soc/codecs/max98927.c create mode 100755 sound/soc/codecs/max98927.h diff --git a/Documentation/devicetree/bindings/sound/max98927.txt b/Documentation/devicetree/bindings/sound/max98927.txt new file mode 100755 index 0000000..ddcd332 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98927.txt @@ -0,0 +1,32 @@ +max98927 audio CODEC + +This device supports I2C. + +Required properties: + + - compatible : "maxim,max98927" + + - vmon-slot-no : slot number used to send voltage information + or in inteleave mode this will be used as + interleave slot. + + - imon-slot-no : slot number used to send current information + + - interleave-mode : When using two MAX98927 in a system it is + possible to create ADC data that that will + overflow the frame size. Digital Audio Interleave + mode provides a means to output VMON and IMON data + from two devices on a single DOUT line when running + smaller frames sizes such as 32 BCLKS per LRCLK or + 48 BCLKS per LRCLK. + + - reg : the I2C address of the device for I2C + +Example: + +codec: max98927@3a { + compatible = "maxim,max98927"; + vmon-slot-no = <1>; + imon-slot-no = <0>; + reg = <0x3a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig old mode 100644 new mode 100755 index c67667b..45f21ca --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -86,6 +86,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C + select SND_SOC_MAX98927 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9860 if I2C select SND_SOC_MAX9768 if I2C @@ -573,6 +574,10 @@ config SND_SOC_MAX98925 config SND_SOC_MAX98926 tristate +config SND_SOC_MAX98927 + tristate "Maxim Integrated MAX98927 Speaker Amplifier" + depends on I2C + config SND_SOC_MAX9850 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile old mode 100644 new mode 100755 index 958cd49..1f5fe2c --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -82,6 +82,7 @@ snd-soc-max98371-objs := max98371.o snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o +snd-soc-max98927-objs := max98927.o snd-soc-max9850-objs := max9850.o snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o @@ -306,6 +307,7 @@ obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o +obj-$(CONFIG_SND_SOC_MAX98927) += snd-soc-max98927.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c new file mode 100755 index 0000000..d85c84f --- /dev/null +++ b/sound/soc/codecs/max98927.c @@ -0,0 +1,954 @@ +/* + * max98927.c -- MAX98927 ALSA Soc Audio driver + * + * Copyright 2013-15 Maxim Integrated Products + * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <sound/tlv.h> +#include "max98927.h" + +static struct reg_default max98927_reg_map[] = { + {0x0014, 0x78}, + {0x0015, 0xFF}, + {0x0043, 0x04}, + {0x0017, 0x55}, + /* For mono driver we are just enabling one channel*/ + {MAX98927_PCM_Rx_Enables_A, 0x03}, + {MAX98927_PCM_Tx_HiZ_Control_A, 0xfc}, + {MAX98927_PCM_Tx_HiZ_Control_B, 0xff}, + {MAX98927_PCM_Tx_Channel_Sources_A, 0x01}, + {MAX98927_PCM_Tx_Channel_Sources_B, 0x01}, + {MAX98927_Measurement_DSP_Config, 0xf7}, + {0x0025, 0x80}, + {0x0026, 0x01}, + {0x0035, 0x40}, + {0x0036, 0x40}, + {0x0037, 0x02}, + {0x0039, 0x01}, + {0x003c, 0x44}, + {0x003d, 0x04}, + {0x0040, 0x10}, + {0x0042, 0x3f}, + {0x0044, 0x00}, + {0x0045, 0x24}, + {0x007f, 0x06}, + {0x0087, 0x1c}, + {0x0089, 0x03}, + {0x009f, 0x01}, +}; + +void max98927_wrapper_write(struct max98927_priv *max98927, + unsigned int reg, unsigned int val) +{ + if (max98927->regmap) + regmap_write(max98927->regmap, reg, val); + if (max98927->sub_regmap) + regmap_write(max98927->sub_regmap, reg, val); +} + +void max98927_wrap_update_bits(struct max98927_priv *max98927, + unsigned int reg, unsigned int mask, unsigned int val) +{ + if (max98927->regmap) + regmap_update_bits(max98927->regmap, reg, mask, val); + if (max98927->sub_regmap) + regmap_update_bits(max98927->sub_regmap, reg, mask, val); +} + +static int max98927_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, unsigned int reg, + unsigned int mask, unsigned int shift) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int data; + + regmap_read(max98927->regmap, reg, &data); + ucontrol->value.integer.value[0] = + (data & mask) >> shift; + return 0; +} + +static int max98927_reg_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, unsigned int reg, + unsigned int mask, unsigned int shift) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + + max98927_wrap_update_bits(max98927, reg, mask, sel << shift); + dev_dbg(codec->dev, "%s: register 0x%02X, value 0x%02X\n", + __func__, reg, sel); + return 0; +} + +static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + unsigned int invert = 0; + + dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + max98927_wrap_update_bits(max98927, MAX98927_PCM_Master_Mode, + MAX98927_PCM_Master_Mode_PCM_MSTR_MODE_Mask, + MAX98927_PCM_Master_Mode_PCM_MSTR_MODE_SLAVE); + break; + case SND_SOC_DAIFMT_CBM_CFM: + max98927->master = true; + max98927_wrap_update_bits(max98927, MAX98927_PCM_Master_Mode, + MAX98927_PCM_Master_Mode_PCM_MSTR_MODE_Mask, + MAX98927_PCM_Master_Mode_PCM_MSTR_MODE_MASTER); + break; + case SND_SOC_DAIFMT_CBS_CFM: + max98927_wrap_update_bits(max98927, MAX98927_PCM_Master_Mode, + MAX98927_PCM_Master_Mode_PCM_MSTR_MODE_Mask, + MAX98927_PCM_Master_Mode_PCM_MSTR_MODE_HYBRID); + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + invert = MAX98927_PCM_Mode_Config_PCM_BCLKEDGE; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + max98927->iface |= SND_SOC_DAIFMT_I2S; + max98927_wrap_update_bits(max98927, + MAX98927_PCM_Mode_Config, + max98927->iface, max98927->iface); + break; + case SND_SOC_DAIFMT_LEFT_J: + max98927->iface |= SND_SOC_DAIFMT_LEFT_J; + max98927_wrap_update_bits(max98927, + MAX98927_PCM_Mode_Config, + max98927->iface, max98927->iface); + break; + default: + return -EINVAL; + } + + /* pcm channel configuration */ + if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + max98927_wrapper_write(max98927, + MAX98927_PCM_Rx_Enables_A, + MAX98927_PCM_Rx_Enables_A_PCM_RX_CH0_EN| + MAX98927_PCM_Rx_Enables_A_PCM_RX_CH1_EN); + max98927_wrapper_write(max98927, + MAX98927_PCM_Tx_Enables_A, + MAX98927_PCM_Tx_Enables_A_PCM_TX_CH0_EN| + MAX98927_PCM_Tx_Enables_A_PCM_TX_CH1_EN); + } + max98927_wrap_update_bits(max98927, MAX98927_PCM_Mode_Config, + MAX98927_PCM_Mode_Config_PCM_BCLKEDGE, invert); + return 0; +} + +/* codec MCLK rate in master mode */ +static const int rate_table[] = { + 5644800, 6000000, 6144000, 6500000, + 9600000, 11289600, 12000000, 12288000, + 13000000, 19200000, +}; + +static int max98927_set_clock(struct max98927_priv *max98927, + struct snd_pcm_hw_params *params) +{ + /* BCLK/LRCLK ratio calculation */ + int blr_clk_ratio = params_channels(params) * max98927->ch_size; + int reg = MAX98927_PCM_Clock_setup; + int mask = MAX98927_PCM_Clock_setup_PCM_BSEL_Mask; + int value; + + if (max98927->master) { + int i; + /* match rate to closest value */ + for (i = 0; i < ARRAY_SIZE(rate_table); i++) { + if (rate_table[i] >= max98927->sysclk) + break; + } + if (i == ARRAY_SIZE(rate_table)) { + pr_err("%s couldn't get the MCLK to match codec\n", + __func__); + return -EINVAL; + } + max98927_wrap_update_bits(max98927, MAX98927_PCM_Master_Mode, + MAX98927_PCM_Master_Mode_PCM_MCLK_RATE_Mask, + i << MAX98927_PCM_Master_Mode_PCM_MCLK_RATE_SHIFT); + } + + switch (blr_clk_ratio) { + case 32: + value = 2; + break; + case 48: + value = 3; + break; + case 64: + value = 4; + break; + default: + return -EINVAL; + } + max98927_wrap_update_bits(max98927, + reg, mask, value); + return 0; +} + +static int max98927_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int sampling_rate = 0; + + /* pcm mode configuration */ + switch (snd_pcm_format_width(params_format(params))) { + case 16: + max98927_wrap_update_bits(max98927, + MAX98927_PCM_Mode_Config, + MAX98927_PCM_Mode_Config_PCM_CHANSZ_16, + MAX98927_PCM_Mode_Config_PCM_CHANSZ_16); + max98927->ch_size = 16; + break; + case 24: + max98927_wrap_update_bits(max98927, + MAX98927_PCM_Mode_Config, + MAX98927_PCM_Mode_Config_PCM_CHANSZ_24, + MAX98927_PCM_Mode_Config_PCM_CHANSZ_24); + max98927->ch_size = 24; + break; + case 32: + max98927_wrap_update_bits(max98927, + MAX98927_PCM_Mode_Config, + MAX98927_PCM_Mode_Config_PCM_CHANSZ_32, + MAX98927_PCM_Mode_Config_PCM_CHANSZ_32); + max98927->ch_size = 32; + break; + default: + pr_err("%s: format unsupported %d", + __func__, params_format(params)); + goto err; + } + dev_dbg(codec->dev, "%s: format supported %d", + __func__, params_format(params)); + + /* sampling rate configuration */ + switch (params_rate(params)) { + case 8000: + sampling_rate |= + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_8000; + break; + case 11025: + sampling_rate |= + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_11025; + break; + case 12000: + sampling_rate |= + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_12000; + break; + case 16000: + sampling_rate |= + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_16000; + break; + case 22050: + sampling_rate |= + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_22050; + break; + case 24000: + sampling_rate |= + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_24000; + break; + case 32000: + sampling_rate |= + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_32000; + break; + case 44100: + sampling_rate |= + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_44100; + break; + case 48000: + sampling_rate |= + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_48000; + break; + default: + pr_err("%s rate %d not supported\n", + __func__, params_rate(params)); + goto err; + } + /* set DAI_SR to correct LRCLK frequency */ + max98927_wrap_update_bits(max98927, MAX98927_PCM_Sample_rate_setup_1, + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_Mask, sampling_rate); + max98927_wrap_update_bits(max98927, MAX98927_PCM_Sample_rate_setup_2, + MAX98927_PCM_Sample_rate_setup_2_SPK_SR_Mask, sampling_rate<<4); + max98927_wrap_update_bits(max98927, MAX98927_PCM_Sample_rate_setup_2, + MAX98927_PCM_Sample_rate_setup_2_IVADC_SR_Mask, sampling_rate); + return max98927_set_clock(max98927, params); +err: + return -EINVAL; +} + +#define MAX98927_RATES SNDRV_PCM_RATE_8000_48000 + +#define MAX98927_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static int max98927_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + max98927->sysclk = freq; + return 0; +} + +static const struct snd_soc_dai_ops max98927_dai_ops = { + .set_sysclk = max98927_dai_set_sysclk, + .set_fmt = max98927_dai_set_fmt, + .hw_params = max98927_dai_hw_params, +}; + +static void max98927_handle_pdata(struct snd_soc_codec *codec) +{ + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + struct max98927_reg_default *regInfo; + int cfg_size = 0; + int x; + + if (max98927->regcfg != NULL) + cfg_size = max98927->regcfg_sz / sizeof(uint32_t); + + if (cfg_size <= 0) { + dev_dbg(codec->dev, + "Register configuration is not required.\n"); + return; + } + + /* direct configuration from device tree */ + for (x = 0; x < cfg_size; x += 3) { + regInfo = (struct max98927_reg_default *)&max98927->regcfg[x]; + dev_info(codec->dev, "CH:%d, reg:0x%02x, value:0x%02x\n", + be32_to_cpu(regInfo->ch), + be32_to_cpu(regInfo->reg), + be32_to_cpu(regInfo->def)); + if (be32_to_cpu(regInfo->ch) == PRI_MAX98927 + && max98927->regmap) + regmap_write(max98927->regmap, + be32_to_cpu(regInfo->reg), + be32_to_cpu(regInfo->def)); + else if (be32_to_cpu(regInfo->ch) == SEC_MAX98927 + && max98927->sub_regmap) + regmap_write(max98927->sub_regmap, + be32_to_cpu(regInfo->reg), + be32_to_cpu(regInfo->def)); + } +} + +static int max98927_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + max98927_wrap_update_bits(max98927, + MAX98927_AMP_enables, 1, 1); + /* enable the v and i for vi feedback */ + max98927_wrap_update_bits(max98927, + MAX98927_Measurement_enables, + MAX98927_Measurement_enables_IVADC_V_EN, + MAX98927_Measurement_enables_IVADC_V_EN); + max98927_wrap_update_bits(max98927, + MAX98927_Measurement_enables, + MAX98927_Measurement_enables_IVADC_I_EN, + MAX98927_Measurement_enables_IVADC_I_EN); + max98927_wrap_update_bits(max98927, + MAX98927_Global_Enable, 1, 1); + break; + case SND_SOC_DAPM_POST_PMD: + max98927_wrap_update_bits(max98927, + MAX98927_Global_Enable, 1, 0); + max98927_wrap_update_bits(max98927, + MAX98927_AMP_enables, 1, 0); + /* disable the v and i for vi feedback */ + max98927_wrap_update_bits(max98927, + MAX98927_Measurement_enables, + MAX98927_Measurement_enables_IVADC_V_EN, + 0); + max98927_wrap_update_bits(max98927, + MAX98927_Measurement_enables, + MAX98927_Measurement_enables_IVADC_I_EN, + 0); + break; + default: + return 0; + } + return 0; +} + +static const struct snd_soc_dapm_widget max98927_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", MAX98927_AMP_enables, + 0, 0, max98927_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("BE_OUT"), +}; + +static DECLARE_TLV_DB_SCALE(max98927_spk_tlv, 300, 300, 0); +static DECLARE_TLV_DB_SCALE(max98927_digital_tlv, -1600, 25, 0); + +static int max98927_spk_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = max98927->spk_gain; + dev_dbg(codec->dev, "%s: spk_gain setting returned %d\n", __func__, + (int) ucontrol->value.integer.value[0]); + return 0; +} + +static int max98927_spk_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + + if (sel < ((1 << MAX98927_Speaker_Gain_Width) - 1)) { + max98927_wrap_update_bits(max98927, MAX98927_Speaker_Gain, + MAX98927_Speaker_Gain_SPK_PCM_GAIN_Mask, sel); + max98927->spk_gain = sel; + } + return 0; +} + +static int max98927_digital_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = max98927->digital_gain; + dev_dbg(codec->dev, "%s: spk_gain setting returned %d\n", __func__, + (int) ucontrol->value.integer.value[0]); + return 0; +} + +static int max98927_digital_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + + if (sel < ((1 << MAX98927_AMP_VOL_WIDTH) - 1)) { + max98927_wrap_update_bits(max98927, MAX98927_AMP_volume_control, + MAX98927_AMP_volume_control_AMP_VOL_Mask, sel); + max98927->digital_gain = sel; + } + return 0; +} + +static int max98927_boost_voltage_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_get(kcontrol, ucontrol, MAX98927_Boost_Control_0, + MAX98927_Boost_Control_0_BST_VOUT_Mask, 0); +} + +static int max98927_boost_voltage_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_put(kcontrol, ucontrol, MAX98927_Boost_Control_0, + MAX98927_Boost_Control_0_BST_VOUT_Mask, 0); +} + +static int max98927_amp_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_get(kcontrol, ucontrol, MAX98927_Boost_Control_0, + MAX98927_Boost_Control_0_BST_VOUT_Mask, + MAX98927_AMP_VOL_LOCATION_SHIFT); +} + +static int max98927_amp_dsp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_put(kcontrol, ucontrol, MAX98927_Brownout_enables, + MAX98927_Brownout_enables_AMP_DSP_EN, MAX98927_BDE_DSP_SHIFT); +} + +static int max98927_amp_dsp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_get(kcontrol, ucontrol, MAX98927_Brownout_enables, + MAX98927_Brownout_enables_AMP_DSP_EN, MAX98927_BDE_DSP_SHIFT); +} + +static int max98927_ramp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_put(kcontrol, ucontrol, MAX98927_AMP_DSP_Config, + MAX98927_AMP_DSP_Config_AMP_VOL_RMP_BYPASS, + MAX98927_SPK_RMP_EN_SHIFT); +} +static int max98927_ramp_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_get(kcontrol, ucontrol, MAX98927_AMP_DSP_Config, + MAX98927_AMP_DSP_Config_AMP_VOL_RMP_BYPASS, + MAX98927_SPK_RMP_EN_SHIFT); +} + +static int max98927_dre_en_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_put(kcontrol, ucontrol, MAX98927_DRE_Control, + MAX98927_DRE_Control_DRE_EN, 0); +} +static int max98927_dre_en_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_get(kcontrol, ucontrol, MAX98927_DRE_Control, + MAX98927_DRE_Control_DRE_EN, 0); +} +static int max98927_amp_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_put(kcontrol, ucontrol, + MAX98927_AMP_volume_control, + MAX98927_AMP_volume_control_AMP_VOL_SEL, + MAX98927_AMP_VOL_LOCATION_SHIFT); +} +static int max98927_spk_src_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_get(kcontrol, ucontrol, + MAX98927_Speaker_source_select, + MAX98927_Speaker_source_select_SPK_SOURCE_Mask, 0); +} + +static int max98927_spk_src_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_put(kcontrol, ucontrol, + MAX98927_Speaker_source_select, + MAX98927_Speaker_source_select_SPK_SOURCE_Mask, 0); +} + +static int max98927_mono_out_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_get(kcontrol, ucontrol, + MAX98927_PCM_to_speaker_monomix_A, + MAX98927_PCM_to_spkmonomix_A_DMONOMIX_CFG_Mask, + MAX98927_PCM_to_speaker_monomix_A_SHIFT); +} + +static int max98927_mono_out_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98927_reg_put(kcontrol, ucontrol, + MAX98927_PCM_to_speaker_monomix_A, + MAX98927_PCM_to_spkmonomix_A_DMONOMIX_CFG_Mask, + MAX98927_PCM_to_speaker_monomix_A_SHIFT); +} + +static bool max98927_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x0001 ... 0x0028: + case 0x002B ... 0x004E: + case 0x0051 ... 0x0055: + case 0x005A ... 0x0061: + case 0x0072 ... 0x0087: + case 0x00FF: + case 0x0100: + case 0x01FF: + return true; + } + return false; +}; + +static const char * const max98927_boost_voltage_text[] = { + "6.5V", "6.625V", "6.75V", "6.875V", "7V", "7.125V", "7.25V", "7.375V", + "7.5V", "7.625V", "7.75V", "7.875V", "8V", "8.125V", "8.25V", "8.375V", + "8.5V", "8.625V", "8.75V", "8.875V", "9V", "9.125V", "9.25V", "9.375V", + "9.5V", "9.625V", "9.75V", "9.875V", "10V" +}; + +static const char * const max98927_speaker_source_text[] = { + "i2s", "reserved", "tone", "pdm" +}; + +static const char * const max98927_monomix_output_text[] = { + "ch_0", "ch_1", "ch_1_2_div" +}; + +static const struct soc_enum max98927_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max98927_monomix_output_text), + max98927_monomix_output_text), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max98927_speaker_source_text), + max98927_speaker_source_text), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max98927_boost_voltage_text), + max98927_boost_voltage_text), +}; + +static const struct snd_kcontrol_new max98927_snd_controls[] = { + SOC_SINGLE_EXT_TLV("Speaker Volume", MAX98927_Speaker_Gain, + 0, (1<<MAX98927_Speaker_Gain_Width)-1, 0, + max98927_spk_gain_get, max98927_spk_gain_put, + max98927_spk_tlv), + SOC_SINGLE_EXT_TLV("Digital Gain", MAX98927_AMP_volume_control, + 0, (1<<MAX98927_AMP_VOL_WIDTH)-1, 0, + max98927_digital_gain_get, max98927_digital_gain_put, + max98927_digital_tlv), + SOC_SINGLE_EXT("Amp DSP Enable", MAX98927_Brownout_enables, + MAX98927_BDE_DSP_SHIFT, 1, 0, + max98927_amp_dsp_get, max98927_amp_dsp_put), + SOC_SINGLE_EXT("Ramp Switch", MAX98927_AMP_DSP_Config, + MAX98927_SPK_RMP_EN_SHIFT, 1, 1, + max98927_ramp_switch_get, max98927_ramp_switch_put), + SOC_SINGLE_EXT("DRE EN", MAX98927_DRE_Control, + MAX98927_DRE_Control_DRE_SHIFT, 1, 0, + max98927_dre_en_get, max98927_dre_en_put), + SOC_SINGLE_EXT("Amp Volume Location", MAX98927_AMP_volume_control, + MAX98927_AMP_VOL_LOCATION_SHIFT, 1, 0, + max98927_amp_vol_get, max98927_amp_vol_put), + + SOC_ENUM_EXT("Boost Output Voltage", max98927_enum[2], + max98927_boost_voltage_get, max98927_boost_voltage_put), + SOC_ENUM_EXT("Speaker Source", max98927_enum[1], + max98927_spk_src_get, max98927_spk_src_put), + SOC_ENUM_EXT("Monomix Output", max98927_enum[0], + max98927_mono_out_get, max98927_mono_out_put), +}; + +static const struct snd_soc_dapm_route max98927_audio_map[] = { + {"BE_OUT", NULL, "Amp Enable"}, +}; + +static struct snd_soc_dai_driver max98927_dai[] = { + { + .name = "max98927-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98927_RATES, + .formats = MAX98927_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98927_RATES, + .formats = MAX98927_FORMATS, + }, + .ops = &max98927_dai_ops, + } +}; + +static int max98927_probe(struct snd_soc_codec *codec) +{ + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int ret = 0, reg = 0, i; + + max98927->codec = codec; + codec->control_data = max98927->regmap; + codec->cache_bypass = 1; + + /* Software Reset */ + max98927_wrapper_write(max98927, + MAX98927_Software_Reset, MAX98927_Software_Reset_RST); + + /* Check Revision ID for the primary MAX98927*/ + ret = regmap_read(max98927->regmap, MAX98927_REV_ID, &reg); + if (ret < 0) + dev_err(codec->dev, + "Failed to read: 0x%02X\n", MAX98927_REV_ID); + else + dev_info(codec->dev, + "MAX98927 revisionID: 0x%02X\n", reg); + + /* Check Revision ID for the secondary MAX98927*/ + if (max98927->sub_regmap) { + ret = regmap_read(max98927->sub_regmap, MAX98927_REV_ID, &reg); + if (ret < 0) + dev_err(codec->dev, + "Failed to read: 0x%02X from secodnary device\n" + , MAX98927_REV_ID); + else + dev_info(codec->dev, + "Secondary device revisionID: 0x%02X\n", reg); + } + + /* Register initialization */ + for (i = 0; i < sizeof(max98927_reg_map)/ + sizeof(max98927_reg_map[0]); i++) + max98927_wrapper_write(max98927, + max98927_reg_map[i].reg, + max98927_reg_map[i].def); + + if (max98927->regmap) + regmap_write(max98927->regmap, + MAX98927_PCM_Tx_Channel_Sources_A, + (max98927->i_l_slot + <<MAX98927_PCM_Tx_Ch_Sources_A_I_SHIFT| + max98927->v_l_slot)&0xFF); + if (max98927->sub_regmap) + regmap_write(max98927->sub_regmap, + MAX98927_PCM_Tx_Channel_Sources_A, + (max98927->i_r_slot + <<MAX98927_PCM_Tx_Ch_Sources_A_I_SHIFT| + max98927->v_r_slot)&0xFF); + + /* Set interleave mode */ + if (max98927->interleave_mode) + max98927_wrap_update_bits(max98927, + MAX98927_PCM_Tx_Channel_Sources_B, + MAX98927_PCM_Tx_Channel_Src_INTERLEAVE_Mask, + MAX98927_PCM_Tx_Channel_Src_INTERLEAVE_Mask); + + max98927_handle_pdata(codec); + + return ret; +} + +static const struct snd_soc_codec_driver soc_codec_dev_max98927 = { + .probe = max98927_probe, + .dapm_routes = max98927_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98927_audio_map), + .dapm_widgets = max98927_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98927_dapm_widgets), + .controls = max98927_snd_controls, + .num_controls = ARRAY_SIZE(max98927_snd_controls), +}; + +static const struct regmap_config max98927_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = MAX98927_REV_ID, + .reg_defaults = max98927_reg_map, + .num_reg_defaults = ARRAY_SIZE(max98927_reg_map), + .readable_reg = max98927_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static struct i2c_board_info max98927_i2c_sub_board[] = { + { + I2C_BOARD_INFO("max98927_sub", 0x39), + } +}; + +static struct i2c_driver max98927_i2c_sub_driver = { + .driver = { + .name = "max98927_sub", + .owner = THIS_MODULE, + }, +}; + +struct i2c_client *max98927_add_sub_device(int bus_id, int slave_addr) +{ + struct i2c_client *i2c = NULL; + struct i2c_adapter *adapter; + + max98927_i2c_sub_board[0].addr = slave_addr; + + adapter = i2c_get_adapter(bus_id); + if (adapter) { + i2c = i2c_new_device(adapter, max98927_i2c_sub_board); + if (i2c) + i2c->dev.driver = &max98927_i2c_sub_driver.driver; + } + + return i2c; +} + +int probe_common(struct i2c_client *i2c, struct max98927_priv *max98927) +{ + int ret = 0, value; + + if (!of_property_read_u32(i2c->dev.of_node, "vmon-l-slot", &value)) + max98927->v_l_slot = value & 0xF; + else + max98927->v_l_slot = 0; + if (!of_property_read_u32(i2c->dev.of_node, "imon-l-slot", &value)) + max98927->i_l_slot = value & 0xF; + else + max98927->i_l_slot = 1; + if (!of_property_read_u32(i2c->dev.of_node, "vmon-r-slot", &value)) + max98927->v_r_slot = value & 0xF; + else + max98927->v_r_slot = 2; + if (!of_property_read_u32(i2c->dev.of_node, "imon-r-slot", &value)) + max98927->i_r_slot = value & 0xF; + else + max98927->i_r_slot = 3; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98927, + max98927_dai, ARRAY_SIZE(max98927_dai)); + if (ret < 0) + dev_err(&i2c->dev, + "Failed to register codec: %d\n", ret); + return ret; +} + +static int max98927_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + + int ret = 0, value; + struct max98927_priv *max98927 = NULL; + + max98927 = devm_kzalloc(&i2c->dev, + sizeof(*max98927), GFP_KERNEL); + + if (!max98927) { + ret = -ENOMEM; + goto err; + } + i2c_set_clientdata(i2c, max98927); + + /* update interleave mode info */ + if (!of_property_read_u32(i2c->dev.of_node, + "interleave_mode", &value)) { + if (value > 0) + max98927->interleave_mode = 1; + else + max98927->interleave_mode = 0; + } else + max98927->interleave_mode = 0; + + /* update direct configuration info */ + max98927->regcfg = of_get_property(i2c->dev.of_node, + "maxim,regcfg", &max98927->regcfg_sz); + + /* check for secondary MAX98927 */ + ret = of_property_read_u32(i2c->dev.of_node, + "maxim,sub_reg", &max98927->sub_reg); + if (ret) { + dev_err(&i2c->dev, "Sub-device slave address was not found.\n"); + max98927->sub_reg = -1; + } + ret = of_property_read_u32(i2c->dev.of_node, + "maxim,sub_bus", &max98927->sub_bus); + if (ret) { + dev_err(&i2c->dev, "Sub-device bus information was not found.\n"); + max98927->sub_bus = i2c->adapter->nr; + } + + /* regmap initialization for primary device */ + max98927->regmap + = devm_regmap_init_i2c(i2c, &max98927_regmap); + if (IS_ERR(max98927->regmap)) { + ret = PTR_ERR(max98927->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + goto err; + } + + /* regmap initialization for secondary device */ + if (max98927->sub_reg > 0) { + max98927->sub_i2c = max98927_add_sub_device(max98927->sub_bus, + max98927->sub_reg); + if (IS_ERR(max98927->sub_i2c)) { + dev_err(&max98927->sub_i2c->dev, + "Second MAX98927 was not found\n"); + ret = PTR_ERR(max98927->regmap); + goto err; + } else { + max98927->sub_regmap = regmap_init_i2c( + max98927->sub_i2c, &max98927_regmap); + if (IS_ERR(max98927->sub_regmap)) { + ret = PTR_ERR(max98927->sub_regmap); + dev_err(&max98927->sub_i2c->dev, + "Failed to allocate sub_regmap: %d\n", + ret); + goto err; + } + } + } + + /* codec registeration */ + ret = probe_common(i2c, max98927); + + return ret; + +err: + if (max98927) + devm_kfree(&i2c->dev, max98927); + return ret; +} + +static int max98927_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98927_i2c_id[] = { + { "max98927", 0}, + { }, +}; + +MODULE_DEVICE_TABLE(i2c, max98927_i2c_id); + +static const struct of_device_id max98927_of_match[] = { + { .compatible = "maxim,max98927", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98927_of_match); + +static struct i2c_driver max98927_i2c_driver = { + .driver = { + .name = "max98927", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(max98927_of_match), + .pm = NULL, + }, + .probe = max98927_i2c_probe, + .remove = max98927_i2c_remove, + .id_table = max98927_i2c_id, +}; + +module_i2c_driver(max98927_i2c_driver) + +MODULE_DESCRIPTION("ALSA SoC MAX98927 driver"); +MODULE_AUTHOR("Ryan Lee <ryans.lee(a)maximintegrated.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h new file mode 100755 index 0000000..2305185 --- /dev/null +++ b/sound/soc/codecs/max98927.h @@ -0,0 +1,1253 @@ +/* + * max98927.c -- MAX98927 ALSA Soc Audio driver + * + * Copyright 2008-11 Wolfson Microelectronics PLC. + * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ +#ifndef __MAX98927_REGISTERDEFS_H +#define __MAX98927_REGISTERDEFS_H +#ifdef CONFIG_SND_SOC_MAXIM_DSM +#include <sound/maxim_dsm.h> +#endif /* CONFIG_SND_SOC_MAXIM_DSM */ + +enum { + PRI_MAX98927 = 0, + SEC_MAX98927 = 1, + MAX_DEV_ID_MAX98927, +} MAX98927deviceID; + +enum { + /*Interrupt Raw 1 (Address 0x0001)*/ + MAX98927_Interrupt_Raw_1 = 0x0001, + MAX98927_Interrupt_Raw_1_BDE_ACTIVE_END_RAW = (0x1 << 0), + MAX98927_Interrupt_Raw_1_BDE_ACTIVE_BGN_RAW = (0x1 << 1), + MAX98927_Interrupt_Raw_1_BDE_LEVEL_CHANGE_RAW = (0x1 << 2), + MAX98927_Interrupt_Raw_1_BDE_L8_RAW = (0x1 << 3), + MAX98927_Interrupt_Raw_1_THERMWARN_END_RAW = (0x1 << 4), + MAX98927_Interrupt_Raw_1_THERMWARN_START_RAW = (0x1 << 5), + MAX98927_Interrupt_Raw_1_THERMSHDN_END_RAW = (0x1 << 6), + MAX98927_Interrupt_Raw_1_THERMSHDN_START_RAW = (0x1 << 7), + + /* Interrupt Raw 2 (Address 0x0002)*/ + MAX98927_Interrupt_Raw_2 = 0x0002, + MAX98927_Interrupt_Raw_2_WATCHDOGWARN_RAW = (0x1 << 0), + MAX98927_Interrupt_Raw_2_WATCHDOGFAIL_RAW = (0x1 << 1), + MAX98927_Interrupt_Raw_2_BOOSTCURRLIM_RAW = (0x1 << 2), + MAX98927_Interrupt_Raw_2_CLKSTOP_RAW = (0x1 << 3), + MAX98927_Interrupt_Raw_2_CLKSTART_RAW = (0x1 << 4), + MAX98927_Interrupt_Raw_2_MEASADC_END_RAW = (0x1 << 5), + MAX98927_Interrupt_Raw_2_PWRDN_DONE_RAW = (0x1 << 6), + MAX98927_Interrupt_Raw_2_PWRUP_DONE_RAW = (0x1 << 7), + + /* Interrupt Raw 3 (Address 0x0003)*/ + MAX98927_Interrupt_Raw_3 = 0x0003, + MAX98927_Interrupt_Raw_3_PWRUP_FAIL_RAW = (0x1 << 0), + MAX98927_Interrupt_Raw_3_AUTH_DONE_RAW = (0x1 << 1), + MAX98927_Interrupt_Raw_3_SPK_OVC_RAW = (0x1 << 2), + MAX98927_Interrupt_Raw_3_BST_UVLO_RAW = (0x1 << 3), + + /* Interrupt State 1 (Address 0x0004)*/ + MAX98927_Interrupt_State_1 = 0x0004, + MAX98927_Interrupt_State_1_BDE_ACTIVE_END_STATE = (0x1 << 0), + MAX98927_Interrupt_State_1_BDE_ACTIVE_BGN_STATE = (0x1 << 1), + MAX98927_Interrupt_State_1_BDE_LEVEL_CHANGE_STATE = (0x1 << 2), + MAX98927_Interrupt_State_1_BDE_L8_STATE = (0x1 << 3), + MAX98927_Interrupt_State_1_THERMWARN_END_STATE = (0x1 << 4), + MAX98927_Interrupt_State_1_THERMWARN_START_STATE = (0x1 << 5), + MAX98927_Interrupt_State_1_THERMSHDN_END_STATE = (0x1 << 6), + MAX98927_Interrupt_State_1_THERMSHDN_START_STATE = (0x1 << 7), + + /* Interrupt State 2 (Address 0x0005)*/ + MAX98927_Interrupt_State_2 = 0x0005, + MAX98927_Interrupt_State_2_WATCHDOGWARN_STATE = (0x1 << 0), + MAX98927_Interrupt_State_2_WATCHDOGFAIL_STATE = (0x1 << 1), + MAX98927_Interrupt_State_2_BOOSTCURRLIM_STATE = (0x1 << 2), + MAX98927_Interrupt_State_2_CLKSTOP_STATE = (0x1 << 3), + MAX98927_Interrupt_State_2_CLKSTART_STATE = (0x1 << 4), + MAX98927_Interrupt_State_2_MEASADC_END_STATE = (0x1 << 5), + MAX98927_Interrupt_State_2_PWRDN_DONE_STATE = (0x1 << 6), + MAX98927_Interrupt_State_2_PWRUP_DONE_STATE = (0x1 << 7), + + /* Interrupt State 3 (Address 0x0006)*/ + MAX98927_Interrupt_State_3 = 0x0006, + MAX98927_Interrupt_State_3_PWRUP_FAIL_STATE = (0x1 << 0), + MAX98927_Interrupt_State_3_AUTH_DONE_STATE = (0x1 << 1), + MAX98927_Interrupt_State_3_SPK_OVC_STATE = (0x1 << 2), + MAX98927_Interrupt_State_3_BST_UVLO_STATE = (0x1 << 3), + + /* Interrupt Flag 1 (Address 0x0007)*/ + MAX98927_Interrupt_Flag_1 = 0x0007, + MAX98927_Interrupt_Flag_1_BDE_ACTIVE_END_FLAG = (0x1 << 0), + MAX98927_Interrupt_Flag_1_BDE_ACTIVE_BGN_FLAG = (0x1 << 1), + MAX98927_Interrupt_Flag_1_BDE_LEVEL_CHANGE_FLAG = (0x1 << 2), + MAX98927_Interrupt_Flag_1_BDE_L8_FLAG = (0x1 << 3), + MAX98927_Interrupt_Flag_1_THERMWARN_END_FLAG = (0x1 << 4), + MAX98927_Interrupt_Flag_1_THERMWARN_START_FLAG = (0x1 << 5), + MAX98927_Interrupt_Flag_1_THERMSHDN_END_FLAG = (0x1 << 6), + MAX98927_Interrupt_Flag_1_THERMSHDN_START_FLAG = (0x1 << 7), + + /* Interrupt Flag 2 (Address 0x0008)*/ + MAX98927_Interrupt_Flag_2 = 0x0008, + MAX98927_Interrupt_Flag_2_WATCHDOGWARN_FLAG = (0x1 << 0), + MAX98927_Interrupt_Flag_2_WATCHDOGFAIL_FLAG = (0x1 << 1), + MAX98927_Interrupt_Flag_2_BOOSTCURRLIM_FLAG = (0x1 << 2), + MAX98927_Interrupt_Flag_2_CLKSTOP_FLAG = (0x1 << 3), + MAX98927_Interrupt_Flag_2_CLKSTART_FLAG = (0x1 << 4), + MAX98927_Interrupt_Flag_2_MEASADC_END_FLAG = (0x1 << 5), + MAX98927_Interrupt_Flag_2_PWRDN_DONE_FLAG = (0x1 << 6), + MAX98927_Interrupt_Flag_2_PWRUP_DONE_FLAG = (0x1 << 7), + + /* Interrupt Flag 3 (Address 0x0009)*/ + MAX98927_Interrupt_Flag_3 = 0x0009, + MAX98927_Interrupt_Flag_3_PWRUP_FAIL_FLAG = (0x1 << 0), + MAX98927_Interrupt_Flag_3_AUTH_DONE_FLAG = (0x1 << 1), + MAX98927_Interrupt_Flag_3_SPK_OVC_FLAG = (0x1 << 2), + MAX98927_Interrupt_Flag_3_BST_UVLO_FLAG = (0x1 << 3), + + /* Interrupt Enable 1 (Address 0x000a)*/ + MAX98927_Interrupt_Enable_1 = 0x000a, + MAX98927_Interrupt_Enable_1_BDE_ACTIVE_END_EN = (0x1 << 0), + MAX98927_Interrupt_Enable_1_BDE_ACTIVE_BGN_EN = (0x1 << 1), + MAX98927_Interrupt_Enable_1_BDE_LEVEL_CHANGE_EN = (0x1 << 2), + MAX98927_Interrupt_Enable_1_BDE_L8_EN = (0x1 << 3), + MAX98927_Interrupt_Enable_1_THERMWARN_END_EN = (0x1 << 4), + MAX98927_Interrupt_Enable_1_THERMWARN_START_EN = (0x1 << 5), + MAX98927_Interrupt_Enable_1_THERMSHDN_END_EN = (0x1 << 6), + MAX98927_Interrupt_Enable_1_THERMSHDN_START_EN = (0x1 << 7), + + /* Interrupt Enable 2 (Address 0x000b)*/ + MAX98927_Interrupt_Enable_2 = 0x000b, + MAX98927_Interrupt_Enable_2_WATCHDOGWARN_EN = (0x1 << 0), + MAX98927_Interrupt_Enable_2_WATCHDOGFAIL_EN = (0x1 << 1), + MAX98927_Interrupt_Enable_2_BOOSTCURRLIM_EN = (0x1 << 2), + MAX98927_Interrupt_Enable_2_CLKSTOP_EN = (0x1 << 3), + MAX98927_Interrupt_Enable_2_CLKSTART_EN = (0x1 << 4), + MAX98927_Interrupt_Enable_2_MEASADC_END_EN = (0x1 << 5), + MAX98927_Interrupt_Enable_2_PWRDN_DONE_EN = (0x1 << 6), + MAX98927_Interrupt_Enable_2_PWRUP_DONE_EN = (0x1 << 7), + + /* Interrupt Enable 3 (Address 0x000c)*/ + MAX98927_Interrupt_Enable_3 = 0x000c, + MAX98927_Interrupt_Enable_3_PWRUP_FAIL_EN = (0x1 << 0), + MAX98927_Interrupt_Enable_3_AUTH_DONE_EN = (0x1 << 1), + MAX98927_Interrupt_Enable_3_SPK_OVC_EN = (0x1 << 2), + MAX98927_Interrupt_Enable_3_BST_UVLO_EN = (0x1 << 3), + + /* Interrupt Flag Clear 1 (Address 0x000d)*/ + MAX98927_Interrupt_Flag_Clear_1 = 0x000d, + MAX98927_Interrupt_Flag_Clear_1_BDE_ACTIVE_END_CLR = (0x1 << 0), + MAX98927_Interrupt_Flag_Clear_1_BDE_ACTIVE_BGN_CLR = (0x1 << 1), + MAX98927_Interrupt_Flag_Clear_1_BDE_LEVEL_CHANGE_CLR = (0x1 << 2), + MAX98927_Interrupt_Flag_Clear_1_BDE_L8_CLR = (0x1 << 3), + MAX98927_Interrupt_Flag_Clear_1_THERMWARN_END_CLR = (0x1 << 4), + MAX98927_Interrupt_Flag_Clear_1_THERMWARN_START_CLR = (0x1 << 5), + MAX98927_Interrupt_Flag_Clear_1_THERMSHDN_END_CLR = (0x1 << 6), + MAX98927_Interrupt_Flag_Clear_1_THERMSHDN_START_CLR = (0x1 << 7), + + /* Interrupt Flag Clear 2 (Address 0x000e)*/ + MAX98927_Interrupt_Flag_Clear_2 = 0x000e, + MAX98927_Interrupt_Flag_Clear_2_WATCHDOGWARN_CLR = (0x1 << 0), + MAX98927_Interrupt_Flag_Clear_2_WATCHDOGFAIL_CLR = (0x1 << 1), + MAX98927_Interrupt_Flag_Clear_2_BOOSTCURRLIM_CLR = (0x1 << 2), + MAX98927_Interrupt_Flag_Clear_2_CLKSTOP_CLR = (0x1 << 3), + MAX98927_Interrupt_Flag_Clear_2_CLKSTART_CLR = (0x1 << 4), + MAX98927_Interrupt_Flag_Clear_2_MEASADC_END_CLR = (0x1 << 5), + MAX98927_Interrupt_Flag_Clear_2_PWRDN_DONE_CLR = (0x1 << 6), + MAX98927_Interrupt_Flag_Clear_2_PWRUP_DONE_CLR = (0x1 << 7), + + /* Interrupt Flag Clear 3 (Address 0x000f)*/ + MAX98927_Interrupt_Flag_Clear_3 = 0x000f, + MAX98927_Interrupt_Flag_Clear_3_PWRUP_FAIL_CLR = (0x1 << 0), + MAX98927_Interrupt_Flag_Clear_3_AUTH_DONE_CLR = (0x1 << 1), + MAX98927_Interrupt_Flag_Clear_3_SPK_OVC_CLR = (0x1 << 2), + MAX98927_Interrupt_Flag_Clear_3_BST_UVLO_CLR = (0x1 << 3), + + /* IRQ Control (Address 0x0010)*/ + MAX98927_IRQ_Control = 0x0010, + MAX98927_IRQ_Control_IRQ_EN = (0x1 << 0), + MAX98927_IRQ_Control_IRQ_POL = (0x1 << 1), + MAX98927_IRQ_Control_IRQ_MODE = (0x1 << 2), + + /* Clock monitor enable (Address 0x0011)*/ + MAX98927_Clock_monitor_enable = 0x0011, + MAX98927_Clock_monitor_enable_CMON_ENA = (0x1 << 0), + MAX98927_Clock_monitor_enable_CMON_AUTORESTART_ENA = (0x1 << 1), + + /* Watchdog Control (Address 0x0012)*/ + MAX98927_Watchdog_Control = 0x0012, + MAX98927_Watchdog_Control_WDT_ENA = (0x1 << 0), + MAX98927_Watchdog_Control_WDT_MODE = (0x1 << 1), + MAX98927_Watchdog_Control_WDT_TO_SEL_Mask = (0x3 << 2), + MAX98927_Watchdog_Control_WDT_TO_SEL_5 = (0x0 << 2), + MAX98927_Watchdog_Control_WDT_TO_SEL_10 = (0x1 << 2), + MAX98927_Watchdog_Control_WDT_TO_SEL_35 = (0x2 << 2), + MAX98927_Watchdog_Control_WDT_TO_SEL_50 = (0x3 << 2), + MAX98927_Watchdog_Control_WDT_HW_SOURCE = (0x1 << 4), + + /* Watchdog SW Reset (Address 0x0013)*/ + MAX98927_Watchdog_SW_Reset = 0x0013, + MAX98927_Watchdog_SW_Reset_WDT_SW_RST_Mask = (0xff << 0), + + /* Meas ADC Thermal Warning Threshhold (Address 0x0014)*/ + MAX98927_Meas_ADC_TW_Threshhold = 0x0014, + MAX98927_Meas_ADC_TW_Threshhold_MEAS_ADC_WARN_THRESH_Mask + = (0xff << 0), + + /* Meas ADC Thermal Shutdown Threshhold (Address 0x0015)*/ + MAX98927_Meas_ADC_TS_Threshhold = 0x0015, + MAX98927_Meas_ADC_TS_Threshhold_MEAS_ADC_SHDN_THRESH_Mask + = (0xff << 0), + + /* Meas ADC Thermal Hysteresis (Address 0x0016)*/ + MAX98927_Meas_ADC_Thermal_Hysteresis = 0x0016, + MAX98927_Meas_ADC_TH_MEAS_ADC_THERM_HYST_Mask = (0x1f << 0), + + /* Pin Config (Address 0x0017)*/ + MAX98927_Pin_Config = 0x0017, + MAX98927_Pin_Config_DOUT_DRV_Mask = (0x3 << 0), + MAX98927_Pin_Config_DOUT_DRV_01 = (0x0 << 0), + MAX98927_Pin_Config_DOUT_DRV_11 = (0x2 << 0), + MAX98927_Pin_Config_BCLK_DRV_Mask = (0x3 << 2), + MAX98927_Pin_Config_BCLK_DRV_01 = (0x0 << 2), + MAX98927_Pin_Config_BCLK_DRV_11 = (0x2 << 2), + MAX98927_Pin_Config_LRCLK_DRV_Mask = (0x3 << 4), + MAX98927_Pin_Config_LRCLK_DRV_01 = (0x0 << 4), + MAX98927_Pin_Config_LRCLK_DRV_11 = (0x2 << 4), + MAX98927_Pin_Config_ICC_DRV_Mask = (0x3 << 6), + MAX98927_Pin_Config_ICC_DRV_01 = (0x0 << 6), + MAX98927_Pin_Config_ICC_DRV_11 = (0x2 << 6), + + /* PCM Rx Enables A (Address 0x0018)*/ + MAX98927_PCM_Rx_Enables_A = 0x0018, + MAX98927_PCM_Rx_Enables_A_PCM_RX_CH0_EN = (0x1 << 0), + MAX98927_PCM_Rx_Enables_A_PCM_RX_CH1_EN = (0x1 << 1), + MAX98927_PCM_Rx_Enables_A_PCM_RX_CH2_EN = (0x1 << 2), + MAX98927_PCM_Rx_Enables_A_PCM_RX_CH3_EN = (0x1 << 3), + MAX98927_PCM_Rx_Enables_A_PCM_RX_CH4_EN = (0x1 << 4), + MAX98927_PCM_Rx_Enables_A_PCM_RX_CH5_EN = (0x1 << 5), + MAX98927_PCM_Rx_Enables_A_PCM_RX_CH6_EN = (0x1 << 6), + MAX98927_PCM_Rx_Enables_A_PCM_RX_CH7_EN = (0x1 << 7), + + /* PCM Rx Enables B (Address 0x0019)*/ + MAX98927_PCM_Rx_Enables_B = 0x0019, + MAX98927_PCM_Rx_Enables_B_PCM_RX_CH8_EN = (0x1 << 0), + MAX98927_PCM_Rx_Enables_B_PCM_RX_CH9_EN = (0x1 << 1), + MAX98927_PCM_Rx_Enables_B_PCM_RX_CH10_EN = (0x1 << 2), + MAX98927_PCM_Rx_Enables_B_PCM_RX_CH11_EN = (0x1 << 3), + MAX98927_PCM_Rx_Enables_B_PCM_RX_CH12_EN = (0x1 << 4), + MAX98927_PCM_Rx_Enables_B_PCM_RX_CH13_EN = (0x1 << 5), + MAX98927_PCM_Rx_Enables_B_PCM_RX_CH14_EN = (0x1 << 6), + MAX98927_PCM_Rx_Enables_B_PCM_RX_CH15_EN = (0x1 << 7), + + /* PCM Tx Enables A (Address 0x001a)*/ + MAX98927_PCM_Tx_Enables_A = 0x001a, + MAX98927_PCM_Tx_Enables_A_PCM_TX_CH0_EN = (0x1 << 0), + MAX98927_PCM_Tx_Enables_A_PCM_TX_CH1_EN = (0x1 << 1), + MAX98927_PCM_Tx_Enables_A_PCM_TX_CH2_EN = (0x1 << 2), + MAX98927_PCM_Tx_Enables_A_PCM_TX_CH3_EN = (0x1 << 3), + MAX98927_PCM_Tx_Enables_A_PCM_TX_CH4_EN = (0x1 << 4), + MAX98927_PCM_Tx_Enables_A_PCM_TX_CH5_EN = (0x1 << 5), + MAX98927_PCM_Tx_Enables_A_PCM_TX_CH6_EN = (0x1 << 6), + MAX98927_PCM_Tx_Enables_A_PCM_TX_CH7_EN = (0x1 << 7), + + /* PCM Tx Enables B (Address 0x001b)*/ + MAX98927_PCM_Tx_Enables_B = 0x001b, + MAX98927_PCM_Tx_Enables_B_PCM_TX_CH8_EN = (0x1 << 0), + MAX98927_PCM_Tx_Enables_B_PCM_TX_CH9_EN = (0x1 << 1), + MAX98927_PCM_Tx_Enables_B_PCM_TX_CH10_EN = (0x1 << 2), + MAX98927_PCM_Tx_Enables_B_PCM_TX_CH11_EN = (0x1 << 3), + MAX98927_PCM_Tx_Enables_B_PCM_TX_CH12_EN = (0x1 << 4), + MAX98927_PCM_Tx_Enables_B_PCM_TX_CH13_EN = (0x1 << 5), + MAX98927_PCM_Tx_Enables_B_PCM_TX_CH14_EN = (0x1 << 6), + MAX98927_PCM_Tx_Enables_B_PCM_TX_CH15_EN = (0x1 << 7), + + /* PCM Tx HiZ Control A (Address 0x001c)*/ + MAX98927_PCM_Tx_HiZ_Control_A = 0x001c, + MAX98927_PCM_Tx_HiZ_Control_A_PCM_TX_CH0_HIZ = (0x1 << 0), + MAX98927_PCM_Tx_HiZ_Control_A_PCM_TX_CH1_HIZ = (0x1 << 1), + MAX98927_PCM_Tx_HiZ_Control_A_PCM_TX_CH2_HIZ = (0x1 << 2), + MAX98927_PCM_Tx_HiZ_Control_A_PCM_TX_CH3_HIZ = (0x1 << 3), + MAX98927_PCM_Tx_HiZ_Control_A_PCM_TX_CH4_HIZ = (0x1 << 4), + MAX98927_PCM_Tx_HiZ_Control_A_PCM_TX_CH5_HIZ = (0x1 << 5), + MAX98927_PCM_Tx_HiZ_Control_A_PCM_TX_CH6_HIZ = (0x1 << 6), + MAX98927_PCM_Tx_HiZ_Control_A_PCM_TX_CH7_HIZ = (0x1 << 7), + + /* PCM Tx HiZ Control B (Address 0x001d)*/ + MAX98927_PCM_Tx_HiZ_Control_B = 0x001d, + MAX98927_PCM_Tx_HiZ_Control_B_PCM_TX_CH8_HIZ = (0x1 << 0), + MAX98927_PCM_Tx_HiZ_Control_B_PCM_TX_CH9_HIZ = (0x1 << 1), + MAX98927_PCM_Tx_HiZ_Control_B_PCM_TX_CH10_HIZ = (0x1 << 2), + MAX98927_PCM_Tx_HiZ_Control_B_PCM_TX_CH11_HIZ = (0x1 << 3), + MAX98927_PCM_Tx_HiZ_Control_B_PCM_TX_CH12_HIZ = (0x1 << 4), + MAX98927_PCM_Tx_HiZ_Control_B_PCM_TX_CH13_HIZ = (0x1 << 5), + MAX98927_PCM_Tx_HiZ_Control_B_PCM_TX_CH14_HIZ = (0x1 << 6), + MAX98927_PCM_Tx_HiZ_Control_B_PCM_TX_CH15_HIZ = (0x1 << 7), + + /* PCM Tx Channel Sources A (Address 0x001e)*/ + MAX98927_PCM_Tx_Channel_Sources_A = 0x001e, + MAX98927_PCM_Tx_Channel_Sources_A_PCM_IVADC_V_DEST_Mask = (0xf << 0), + MAX98927_PCM_Tx_Channel_Sources_A_PCM_IVADC_I_DEST_Mask = (0xf << 4), + + /* PCM Tx Channel Sources B (Address 0x001f)*/ + MAX98927_PCM_Tx_Channel_Sources_B = 0x001f, + MAX98927_PCM_Tx_Channel_Sources_B_PCM_AMP_DSP_DEST_Mask = (0xf << 0), + MAX98927_PCM_Tx_Channel_Src_INTERLEAVE_Mask = (0x1 << 5), + + /* PCM Mode Config (Address 0x0020)*/ + MAX98927_PCM_Mode_Config = 0x0020, + MAX98927_PCM_Mode_Config_PCM_TX_EXTRA_HIZ = (0x1 << 0), + MAX98927_PCM_Mode_Config_PCM_CHANSEL = (0x1 << 1), + MAX98927_PCM_Mode_Config_PCM_BCLKEDGE = (0x1 << 2), + MAX98927_PCM_Mode_Config_PCM_FORMAT_Mask = (0x7 << 3), + MAX98927_PCM_Mode_Config_PCM_FORMAT_I2S = (0x0 << 3), + MAX98927_PCM_Mode_Config_PCM_FORMAT_LEFT = (0x1 << 3), + MAX98927_PCM_Mode_Config_PCM_FORMAT_TDM_0 = (0x3 << 3), + MAX98927_PCM_Mode_Config_PCM_FORMAT_TDM_1 = (0x4 << 3), + MAX98927_PCM_Mode_Config_PCM_FORMAT_TDM_2 = (0x5 << 3), + MAX98927_PCM_Mode_Config_PCM_FORMAT_ = (0x6 << 3), + MAX98927_PCM_Mode_Config_PCM_CHANSZ_Mask = (0x3 << 6), + MAX98927_PCM_Mode_Config_PCM_CHANSZ_16 = (0x1 << 6), + MAX98927_PCM_Mode_Config_PCM_CHANSZ_24 = (0x2 << 6), + MAX98927_PCM_Mode_Config_PCM_CHANSZ_32 = (0x3 << 6), + + /* PCM Master Mode (Address 0x0021)*/ + MAX98927_PCM_Master_Mode = 0x0021, + MAX98927_PCM_Master_Mode_PCM_MSTR_MODE_Mask = (0x3 << 0), + MAX98927_PCM_Master_Mode_PCM_MSTR_MODE_SLAVE = (0x0 << 0), + MAX98927_PCM_Master_Mode_PCM_MSTR_MODE_MASTER = (0x3 << 0), + MAX98927_PCM_Master_Mode_PCM_MSTR_MODE_HYBRID = (0x1 << 0), + MAX98927_PCM_Master_Mode_PCM_MCLK_RATE_Mask = (0xf << 2), + MAX98927_PCM_Master_Mode_PCM_CLK_SOURCE = (0x1 << 6), + + /* PCM Clock setup (Address 0x0022)*/ + MAX98927_PCM_Clock_setup = 0x0022, + MAX98927_PCM_Clock_setup_PCM_BSEL_Mask = (0xf << 0), + MAX98927_PCM_Clock_setup_PCM_MSEL_Mask = (0xf << 4), + + /* PCM Sample rate setup 1 (Address 0x0023)*/ + MAX98927_PCM_Sample_rate_setup_1 = 0x0023, + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_Mask = (0xf << 0), + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_8000 = (0x0 << 0), + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_11025 = (0x1 << 0), + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_12000 = (0x2 << 0), + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_16000 = (0x3 << 0), + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_22050 = (0x4 << 0), + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_24000 = (0x5 << 0), + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_32000 = (0x6 << 0), + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_44100 = (0x7 << 0), + MAX98927_PCM_Sample_rate_setup_1_DIG_IF_SR_48000 = (0x8 << 0), + + /* PCM Sample rate setup 1 (Address 0x0024)*/ + MAX98927_PCM_Sample_rate_setup_2 = 0x0024, + MAX98927_PCM_Sample_rate_setup_2_IVADC_SR_Mask = (0xf << 0), + MAX98927_PCM_Sample_rate_setup_2_SPK_SR_Mask = (0xf << 4), + MAX98927_PCM_Sample_rate_setup_2_SPK_SR_0001 = (0x0 << 4), + MAX98927_PCM_Sample_rate_setup_2_SPK_SR_0011 = (0x2 << 4), + MAX98927_PCM_Sample_rate_setup_2_SPK_SR_0101 = (0x4 << 4), + MAX98927_PCM_Sample_rate_setup_2_SPK_SR_0111 = (0x6 << 4), + MAX98927_PCM_Sample_rate_setup_2_SPK_SR_1001 = (0x8 << 4), + MAX98927_PCM_Sample_rate_setup_2_SPK_SR_1011 = (0xa << 4), + MAX98927_PCM_Sample_rate_setup_2_SPK_SR_1101 = (0xc << 4), + MAX98927_PCM_Sample_rate_setup_2_SPK_SR_ = (0xf << 4), + + /* PCM to speaker monomix A (Address 0x0025)*/ + MAX98927_PCM_to_speaker_monomix_A = 0x0025, + MAX98927_PCM_to_spkmonomix_A_DMONOMIX_CH0_SOURCE_Mask = (0xf << 0), + MAX98927_PCM_to_spkmonomix_A_DMONOMIX_CFG_Mask = (0x3 << 6), + MAX98927_PCM_to_spkmonomix_A_DMONOMIX_CFG_1 = (0x0 << 6), + MAX98927_PCM_to_spkmonomix_A_DMONOMIX_CFG_3 = (0x0 << 6), + + /* PCM to speaker monomix B (Address 0x0026)*/ + MAX98927_PCM_to_spkmonomix_B = 0x0026, + MAX98927_PCM_to_spkmonomix_B_DMONOMIX_CH1_SOURCE_Mask = (0xf << 0), + + /* ICC RX Enables A (Address 0x0027)*/ + MAX98927_ICC_RX_Enables_A = 0x0027, + MAX98927_ICC_RX_Enables_A_ICC_RX_CH0_EN = (0x1 << 0), + MAX98927_ICC_RX_Enables_A_ICC_RX_CH1_EN = (0x1 << 1), + MAX98927_ICC_RX_Enables_A_ICC_RX_CH2_EN = (0x1 << 2), + MAX98927_ICC_RX_Enables_A_ICC_RX_CH3_EN = (0x1 << 3), + MAX98927_ICC_RX_Enables_A_ICC_RX_CH4_EN = (0x1 << 4), + MAX98927_ICC_RX_Enables_A_ICC_RX_CH5_EN = (0x1 << 5), + MAX98927_ICC_RX_Enables_A_ICC_RX_CH6_EN = (0x1 << 6), + MAX98927_ICC_RX_Enables_A_ICC_RX_CH7_EN = (0x1 << 7), + + /* ICC RX Enables B (Address 0x0028)*/ + MAX98927_ICC_RX_Enables_B = 0x0028, + MAX98927_ICC_RX_Enables_B_ICC_RX_CH8_EN = (0x1 << 0), + MAX98927_ICC_RX_Enables_B_ICC_RX_CH9_EN = (0x1 << 1), + MAX98927_ICC_RX_Enables_B_ICC_RX_CH10_EN = (0x1 << 2), + MAX98927_ICC_RX_Enables_B_ICC_RX_CH11_EN = (0x1 << 3), + MAX98927_ICC_RX_Enables_B_ICC_RX_CH12_EN = (0x1 << 4), + MAX98927_ICC_RX_Enables_B_ICC_RX_CH13_EN = (0x1 << 5), + MAX98927_ICC_RX_Enables_B_ICC_RX_CH14_EN = (0x1 << 6), + MAX98927_ICC_RX_Enables_B_ICC_RX_CH15_EN = (0x1 << 7), + + /* ICC TX Enables A (Address 0x002b)*/ + MAX98927_ICC_TX_Enables_A = 0x002b, + MAX98927_ICC_TX_Enables_A_ICC_TX_CH0_EN = (0x1 << 0), + MAX98927_ICC_TX_Enables_A_ICC_TX_CH1_EN = (0x1 << 1), + MAX98927_ICC_TX_Enables_A_ICC_TX_CH2_EN = (0x1 << 2), + MAX98927_ICC_TX_Enables_A_ICC_TX_CH3_EN = (0x1 << 3), + MAX98927_ICC_TX_Enables_A_ICC_TX_CH4_EN = (0x1 << 4), + MAX98927_ICC_TX_Enables_A_ICC_TX_CH5_EN = (0x1 << 5), + MAX98927_ICC_TX_Enables_A_ICC_TX_CH6_EN = (0x1 << 6), + MAX98927_ICC_TX_Enables_A_ICC_TX_CH7_EN = (0x1 << 7), + + /* ICC TX Enables B (Address 0x002c)*/ + MAX98927_ICC_TX_Enables_B = 0x002c, + MAX98927_ICC_TX_Enables_B_ICC_TX_CH8_EN = (0x1 << 0), + MAX98927_ICC_TX_Enables_B_ICC_TX_CH9_EN = (0x1 << 1), + MAX98927_ICC_TX_Enables_B_ICC_TX_CH10_EN = (0x1 << 2), + MAX98927_ICC_TX_Enables_B_ICC_TX_CH11_EN = (0x1 << 3), + MAX98927_ICC_TX_Enables_B_ICC_TX_CH12_EN = (0x1 << 4), + MAX98927_ICC_TX_Enables_B_ICC_TX_CH13_EN = (0x1 << 5), + MAX98927_ICC_TX_Enables_B_ICC_TX_CH14_EN = (0x1 << 6), + MAX98927_ICC_TX_Enables_B_ICC_TX_CH15_EN = (0x1 << 7), + + /* ICC Data Order Select (Address 0x002d)*/ + MAX98927_ICC_Data_Order_Select = 0x002d, + MAX98927_ICC_Data_Order_Select_ICC_DRIVE_MODE = (0x1 << 3), + + /* ICC HiZ Manual Mode (Address 0x002e)*/ + MAX98927_ICC_HiZ_Manual_Mode = 0x002e, + MAX98927_ICC_HiZ_Manual_Mode_ICC_TX_HIZ_MANUAL = (0x1 << 0), + MAX98927_ICC_HiZ_Manual_Mode_ICC_TX_EXTRA_HIZ = (0x1 << 1), + + /* ICC TX HiZ Enables A (Address 0x002f)*/ + MAX98927_ICC_TX_HiZ_Enables_A = 0x002f, + MAX98927_ICC_TX_HiZ_Enables_A_ICC_TX_CH0_HIZ = (0x1 << 0), + MAX98927_ICC_TX_HiZ_Enables_A_ICC_TX_CH1_HIZ = (0x1 << 1), + MAX98927_ICC_TX_HiZ_Enables_A_ICC_TX_CH2_HIZ = (0x1 << 2), + MAX98927_ICC_TX_HiZ_Enables_A_ICC_TX_CH3_HIZ = (0x1 << 3), + MAX98927_ICC_TX_HiZ_Enables_A_ICC_TX_CH4_HIZ = (0x1 << 4), + MAX98927_ICC_TX_HiZ_Enables_A_ICC_TX_CH5_HIZ = (0x1 << 5), + MAX98927_ICC_TX_HiZ_Enables_A_ICC_TX_CH6_HIZ = (0x1 << 6), + MAX98927_ICC_TX_HiZ_Enables_A_ICC_TX_CH7_HIZ = (0x1 << 7), + + /* ICC TX HiZ Enables B (Address 0x0030)*/ + MAX98927_ICC_TX_HiZ_Enables_B = 0x0030, + MAX98927_ICC_TX_HiZ_Enables_B_ICC_TX_CH8_HIZ = (0x1 << 0), + MAX98927_ICC_TX_HiZ_Enables_B_ICC_TX_CH9_HIZ = (0x1 << 1), + MAX98927_ICC_TX_HiZ_Enables_B_ICC_TX_CH10_HIZ = (0x1 << 2), + MAX98927_ICC_TX_HiZ_Enables_B_ICC_TX_CH11_HIZ = (0x1 << 3), + MAX98927_ICC_TX_HiZ_Enables_B_ICC_TX_CH12_HIZ = (0x1 << 4), + MAX98927_ICC_TX_HiZ_Enables_B_ICC_TX_CH13_HIZ = (0x1 << 5), + MAX98927_ICC_TX_HiZ_Enables_B_ICC_TX_CH14_HIZ = (0x1 << 6), + MAX98927_ICC_TX_HiZ_Enables_B_ICC_TX_CH15_HIZ = (0x1 << 7), + + /* ICC Link Enables (Address 0x0031)*/ + MAX98927_ICC_Link_Enables = 0x0031, + MAX98927_ICC_Link_Enables_ICC_LINK_EN = (0x1 << 1), + + /* PDM Tx Enables (Address 0x0032)*/ + MAX98927_PDM_Tx_Enables = 0x0032, + MAX98927_PDM_Tx_Enables_PDM_TX_EN = (0x1 << 0), + MAX98927_PDM_Tx_Enables_PDM_TX_CLK_DIV2 = (0x1 << 1), + + /* PDM Tx HiZ Control (Address 0x0033)*/ + MAX98927_PDM_Tx_HiZ_Control = 0x0033, + MAX98927_PDM_Tx_HiZ_Control_PDM_TX_HIZ = (0x1 << 0), + + /* PDM Tx Control (Address 0x0034)*/ + MAX98927_PDM_Tx_Control = 0x0034, + MAX98927_PDM_Tx_Control_PDM_TX_CH0_SOURCE = (0x1 << 0), + MAX98927_PDM_Tx_Control_PDM_TX_CH1_SOURCE = (0x1 << 1), + + /* PDM Rx Enable (Address 0x0034)*/ + MAX98927_PDM_Rx_Enable = 0x0035, + MAX98927_PDM_Rx_Enable_PDM_RX_EN = (0x1 << 0), + MAX98927_PDM_Rx_Enable_PDM_DSP_EN = (0x1 << 1), + MAX98927_PDM_Rx_Enable_PDM_DITH_EN = (0x1 << 2), + MAX98927_PDM_Rx_Enable_PDM_RX_CH_SEL = (0x1 << 3), + MAX98927_PDM_Rx_Enable_PDM_FIFO_RDY_LVL_Mask = (0xf << 4), + + /* AMP volume control (Address 0x0036)*/ + MAX98927_AMP_volume_control = 0x0036, + MAX98927_AMP_volume_control_AMP_VOL_Mask = (0x7f << 0), + MAX98927_AMP_volume_control_AMP_VOL_SEL = (0x1 << 7), + + /* AMP DSP Config (Address 0x0037)*/ + MAX98927_AMP_DSP_Config = 0x0037, + MAX98927_AMP_DSP_Config_AMP_DCBLK_EN = (0x1 << 0), + MAX98927_AMP_DSP_Config_AMP_DITH_EN = (0x1 << 1), + MAX98927_AMP_DSP_Config_DAC_HALF_REF_CURRENT = (0x1 << 2), + MAX98927_AMP_DSP_Config_DAC_DOUBLE_RFB = (0x1 << 3), + MAX98927_AMP_DSP_Config_AMP_VOL_RMP_BYPASS = (0x1 << 4), + MAX98927_AMP_DSP_Config_DAC_INVERT = (0x1 << 5), + + /* Tone Generator and DC Config (Address 0x0038)*/ + MAX98927_Tone_Generator_and_DC_Config = 0x0038, + MAX98927_Tone_Generator_and_DC_Config_TONE_CONFIG_Mask = (0xf << 0), + + /* DRE Control (Address 0x0039)*/ + MAX98927_DRE_Control = 0x0039, + MAX98927_DRE_Control_DRE_EN = (0x1 << 0), + + /* AMP enables (Address 0x003a)*/ + MAX98927_AMP_enables = 0x003a, + MAX98927_AMP_enables_SPK_EN = (0x1 << 0), + + /* Speaker source select (Address 0x003b)*/ + MAX98927_Speaker_source_select = 0x003b, + MAX98927_Speaker_source_select_SPK_SOURCE_Mask = (0x3 << 0), + MAX98927_Speaker_source_select_SPK_SOURCE_01 = (0x0 << 0), + MAX98927_Speaker_source_select_SPK_SOURCE_11 = (0x2 << 0), + + /* Speaker Gain (Address 0x003c)*/ + MAX98927_Speaker_Gain = 0x003c, + MAX98927_Speaker_Gain_SPK_PCM_GAIN_Mask = (0x7 << 0), + MAX98927_Speaker_Gain_SPK_PCM_GAIN_001 = (0x0 << 0), + MAX98927_Speaker_Gain_SPK_PCM_GAIN_011 = (0x2 << 0), + MAX98927_Speaker_Gain_SPK_PCM_GAIN_101 = (0x4 << 0), + MAX98927_Speaker_Gain_SPK_PCM_GAIN_111 = (0x6 << 0), + MAX98927_Speaker_Gain_SPK_PDM_GAIN_Mask = (0x7 << 4), + MAX98927_Speaker_Gain_SPK_PDM_GAIN_001 = (0x0 << 4), + MAX98927_Speaker_Gain_SPK_PDM_GAIN_011 = (0x2 << 4), + MAX98927_Speaker_Gain_SPK_PDM_GAIN_101 = (0x4 << 4), + MAX98927_Speaker_Gain_SPK_PDM_GAIN_111 = (0x6 << 4), + + /* SSM Configuration (Address 0x003d)*/ + MAX98927_SSM_Configuration = 0x003d, + MAX98927_SSM_Configuration_SSM_MOD_INDEX_Mask = (0x7 << 0), + MAX98927_SSM_Configuration_SSM_MOD_INDEX_001 = (0x0 << 0), + MAX98927_SSM_Configuration_SSM_MOD_INDEX_011 = (0x2 << 0), + MAX98927_SSM_Configuration_SSM_MOD_INDEX_101 = (0x4 << 0), + MAX98927_SSM_Configuration_SSM_MOD_INDEX_ = (0x6 << 0), + MAX98927_SSM_Configuration_SPK_FSW_SEL = (0x1 << 3), + MAX98927_SSM_Configuration_SSM_ENA = (0x1 << 7), + + /* Measurement enables (Address 0x003e)*/ + MAX98927_Measurement_enables = 0x003e, + MAX98927_Measurement_enables_IVADC_V_EN = (0x1 << 0), + MAX98927_Measurement_enables_IVADC_I_EN = (0x1 << 1), + + /* Measurement DSP Config (Address 0x003f)*/ + MAX98927_Measurement_DSP_Config = 0x003f, + MAX98927_Measurement_DSP_Config_MEAS_V_DCBLK_EN = (0x1 << 0), + MAX98927_Measurement_DSP_Config_MEAS_I_DCBLK_EN = (0x1 << 1), + MAX98927_Measurement_DSP_Config_MEAS_DITH_EN = (0x1 << 2), + MAX98927_Measurement_DSP_Config_MEAS_V_DCBLK_Mask = (0x3 << 4), + MAX98927_Measurement_DSP_Config_MEAS_V_DCBLK_01 = (0x0 << 4), + MAX98927_Measurement_DSP_Config_MEAS_V_DCBLK_11 = (0x2 << 4), + MAX98927_Measurement_DSP_Config_MEAS_I_DCBLK_Mask = (0x3 << 6), + MAX98927_Measurement_DSP_Config_MEAS_I_DCBLK_01 = (0x0 << 6), + MAX98927_Measurement_DSP_Config_MEAS_I_DCBLK_11 = (0x2 << 6), + + /* Boost Control 0 (Address 0x0040)*/ + MAX98927_Boost_Control_0 = 0x0040, + MAX98927_Boost_Control_0_BST_VOUT_Mask = (0x1f << 0), + MAX98927_Boost_Control_0_EXT_PVDD_EN = (0x1 << 7), + + /* Boost Control 3 (Address 0x0041)*/ + MAX98927_Boost_Control_3 = 0x0041, + MAX98927_Boost_Control_3_BST_SKIPLOAD_Mask = (0x3 << 0), + MAX98927_Boost_Control_3_BST_SKIPLOAD_01 = (0x0 << 0), + MAX98927_Boost_Control_3_BST_SKIPLOAD_11 = (0x2 << 0), + MAX98927_Boost_Control_3_BST_PHASE_Mask = (0x7 << 2), + MAX98927_Boost_Control_3_BST_PHASE_001 = (0x0 << 2), + MAX98927_Boost_Control_3_BST_PHASE_011 = (0x2 << 2), + MAX98927_Boost_Control_3_BST_PHASE_ = (0x1 << 2), + MAX98927_Boost_Control_3_BST_SLOWSTART = (0x1 << 5), + + /* Boost Control 1 (Address 0x0042)*/ + MAX98927_Boost_Control_1 = 0x0042, + /*#BYHAND width >= 5:*/ + MAX98927_Boost_Control_1_BST_ILIM_Mask = (0x3f << 0), + + /* Meas ADC Config (Address 0x0043)*/ + MAX98927_Meas_ADC_Config = 0x0043, + MAX98927_Meas_ADC_Config_MEAS_ADC_CH0_EN = (0x1 << 0), + MAX98927_Meas_ADC_Config_MEAS_ADC_CH1_EN = (0x1 << 1), + MAX98927_Meas_ADC_Config_MEAS_ADC_CH2_EN = (0x1 << 2), + + /* Meas ADC Base Divide MSByte (Address 0x0044)*/ + MAX98927_Meas_ADC_Base_Divide_MSByte = 0x0044, + /*#BYHAND width >= 5:*/ + MAX98927_Meas_ADC_Base_Divide_MSByte_MEAS_ADC_BASE_DIV_Mask + = (0xff << 0), + + /* Meas ADC Base Divide LSByte (Address 0x0045)*/ + MAX98927_Meas_ADC_Base_Divide_LSByte = 0x0045, + /*#BYHAND width >= 5:*/ + MAX98927_Meas_ADC_Base_Divide_LSByte_MEAS_ADC_BASE_DIV_Mask + = (0xff << 0), + + /* Meas ADC Chan 0 Divide (Address 0x0046)*/ + MAX98927_Meas_ADC_Chan_0_Divide = 0x0046, + /*#BYHAND width >= 5:*/ + MAX98927_Meas_ADC_Chan_0_Divide_MEAS_ADC_CH0_DIV_Mask = (0xff << 0), + + /* Meas ADC Chan 1 Divide (Address 0x0047)*/ + MAX98927_Meas_ADC_Chan_1_Divide = 0x0047, + /*#BYHAND width >= 5:*/ + MAX98927_Meas_ADC_Chan_1_Divide_MEAS_ADC_CH1_DIV_Mask = (0xff << 0), + + /* Meas ADC Chan 2 Divide (Address 0x0048)*/ + MAX98927_Meas_ADC_Chan_2_Divide = 0x0048, + /*#BYHAND width >= 5:*/ + MAX98927_Meas_ADC_Chan_2_Divide_MEAS_ADC_CH2_DIV_Mask = (0xff << 0), + + /* Meas ADC Chan 0 Filt Config (Address 0x0049)*/ + MAX98927_Meas_ADC_Chan_0_Filt_Config = 0x0049, + MAX98927_Meas_ADC_Chan_0_Filt_Config_MEAS_ADC_CH0_FILT_AVG_Mask + = (0x7 << 0), + MAX98927_Meas_ADC_Chan_0_Filt_Config_MEAS_ADC_CH0_FILT_AVG_001 + = (0x0 << 0), + MAX98927_Meas_ADC_Chan_0_Filt_Config_MEAS_ADC_CH0_FILT_AVG_011 + = (0x2 << 0), + MAX98927_Meas_ADC_Chan_0_Filt_Config_MEAS_ADC_CH0_FILT_AVG_101 + = (0x4 << 0), + MAX98927_Meas_ADC_Chan_0_Filt_Config_MEAS_ADC_CH0_FILT_EN + = (0x1 << 3), + + /* Meas ADC Chan 1 Filt Config (Address 0x004a)*/ + MAX98927_Meas_ADC_Chan_1_Filt_Config = 0x004a, + MAX98927_Meas_ADC_Chan_1_Filt_Config_MEAS_ADC_CH1_FILT_AVG_Mask + = (0x7 << 0), + MAX98927_Meas_ADC_Chan_1_Filt_Config_MEAS_ADC_CH1_FILT_AVG_001 + = (0x0 << 0), + MAX98927_Meas_ADC_Chan_1_Filt_Config_MEAS_ADC_CH1_FILT_AVG_011 + = (0x2 << 0), + MAX98927_Meas_ADC_Chan_1_Filt_Config_MEAS_ADC_CH1_FILT_AVG_101 + = (0x4 << 0), + MAX98927_Meas_ADC_Chan_1_Filt_Config_MEAS_ADC_CH1_FILT_EN + = (0x1 << 3), + + /* Meas ADC Chan 2 Filt Config (Address 0x004b)*/ + MAX98927_Meas_ADC_Chan_2_Filt_Config = 0x004b, + MAX98927_Meas_ADC_Chan_2_Filt_Config_MEAS_ADC_CH2_FILT_AVG_Mask + = (0x7 << 0), + MAX98927_Meas_ADC_Chan_2_Filt_Config_MEAS_ADC_CH2_FILT_AVG_001 + = (0x0 << 0), + MAX98927_Meas_ADC_Chan_2_Filt_Config_MEAS_ADC_CH2_FILT_AVG_011 + = (0x2 << 0), + MAX98927_Meas_ADC_Chan_2_Filt_Config_MEAS_ADC_CH2_FILT_AVG_101 + = (0x4 << 0), + MAX98927_Meas_ADC_Chan_2_Filt_Config_MEAS_ADC_CH2_FILT_EN + = (0x1 << 3), + + /* Meas ADC Chan 0 Readback (Address 0x004c)*/ + MAX98927_Meas_ADC_Chan_0_Readback = 0x004c, + /*#BYHAND width >= 5:*/ + MAX98927_Meas_ADC_Chan_0_Readback_MEAS_ADC_CH0_DATA_Mask + = (0xff << 0), + + /* Meas ADC Chan 1 Readback (Address 0x004d)*/ + MAX98927_Meas_ADC_Chan_1_Readback = 0x004d, + /*#BYHAND width >= 5:*/ + MAX98927_Meas_ADC_Chan_1_Readback_MEAS_ADC_CH1_DATA_Mask + = (0xff << 0), + + /* Meas ADC Chan 2 Readback (Address 0x004e)*/ + MAX98927_Meas_ADC_Chan_2_Readback = 0x004e, + /*#BYHAND width >= 5:*/ + MAX98927_Meas_ADC_Chan_2_Readback_MEAS_ADC_CH2_DATA_Mask + = (0xff << 0), + + /* Brownout status (Address 0x0051)*/ + MAX98927_Brownout_status = 0x0051, + MAX98927_Brownout_status_BDE_STATE_Mask = (0xf << 0), + + /* Brownout enables (Address 0x0052)*/ + MAX98927_Brownout_enables = 0x0052, + MAX98927_Brownout_enables_BDE_EN = (0x1 << 0), + MAX98927_Brownout_enables_BDE_AMP_EN = (0x1 << 1), + MAX98927_Brownout_enables_AMP_DSP_EN = (0x1 << 2), + + /* Brownout level infinite hold (Address 0x0053)*/ + MAX98927_Brownout_level_infinite_hold = 0x0053, + MAX98927_Brownout_level_infinite_hold_BDE_L8_INF_HLD = (0x1 << 1), + + /* Brownout level infinite hold clear (Address 0x0054)*/ + MAX98927_Brownout_level_infinite_hold_clear = 0x0054, + MAX98927_Brownout_level_infinite_hold_clear_BDE_L8_HLD_RLS + = (0x1 << 1), + + /* Brownout level hold (Address 0x0055)*/ + MAX98927_Brownout_level_hold = 0x0055, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout_level_hold_BDE_HLD_Mask = (0xff << 0), + + /* Brownout level 1 threshold (Address 0x0056)*/ + MAX98927_Brownout__level_1_threshold = 0x0056, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_1_threshold_BDE_L1_VTHRESH_Mask = (0xff << 0), + + /* Brownout level 2 threshold (Address 0x0057)*/ + MAX98927_Brownout__level_2_threshold = 0x0057, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_2_threshold_BDE_L2_VTHRESH_Mask = (0xff << 0), + + /* Brownout level 3 threshold (Address 0x0058)*/ + MAX98927_Brownout__level_3_threshold = 0x0058, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_3_threshold_BDE_L3_VTHRESH_Mask = (0xff << 0), + + /* Brownout level 4 threshold (Address 0x0059)*/ + MAX98927_Brownout__level_4_threshold = 0x0059, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_4_threshold_BDE_L4_VTHRESH_Mask = (0xff << 0), + + /* Brownout level 5 threshold (Address 0x005a)*/ + MAX98927_Brownout__level_5_threshold = 0x005a, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_5_threshold_BDE_L5_VTHRESH_Mask = (0xff << 0), + + /* Brownout level 6 threshold (Address 0x005b)*/ + MAX98927_Brownout__level_6_threshold = 0x005b, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_6_threshold_BDE_L6_VTHRESH_Mask = (0xff << 0), + + /* Brownout level 7 threshold (Address 0x005c)*/ + MAX98927_Brownout__level_7_threshold = 0x005c, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_7_threshold_BDE_L7_VTHRESH_Mask = (0xff << 0), + + /* Brownout level 8 threshold (Address 0x005d)*/ + MAX98927_Brownout__level_8_threshold = 0x005d, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_8_threshold_BDE_L8_VTHRESH_Mask = (0xff << 0), + + /* Brownout threshold hysterysis (Address 0x005e)*/ + MAX98927_Brownout_threshold_hysterysis = 0x005e, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout_threshold_hysterysis_BDE_VTHRESH_HYST_Mask + = (0xff << 0), + /* Brownout AMP limiter attack/release (Address 0x005f)*/ + MAX98927_Brownout_AMP_limiter_attack_release = 0x005f, + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_RLS_Mask + = (0xf << 0), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_RLS_0001 + = (0x0 << 0), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_RLS_0011 + = (0x2 << 0), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_RLS_0101 + = (0x4 << 0), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_RLS_0111 + = (0x6 << 0), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_RLS_1001 + = (0x8 << 0), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_RLS_1011 + = (0xa << 0), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_RLS_1101 + = (0xc << 0), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_RLS_1111 + = (0xe << 0), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_ATK_Mask + = (0xf << 4), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_ATK_0001 + = (0x0 << 4), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_ATK_0011 + = (0x2 << 4), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_ATK_0101 + = (0x4 << 4), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_ATK_0111 + = (0x6 << 4), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_ATK_1001 + = (0x8 << 4), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_ATK_1011 + = (0xa << 4), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_ATK_1101 + = (0xc << 4), + MAX98927_Brownout_AMP_limiter_attack_release_AMP_LIM_ATK_1111 + = (0xe << 4), + + /* Brownout AMP gain attack/release (Address 0x0060)*/ + MAX98927_Brownout_AMP_gain_attack_release = 0x0060, + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_RLS_Mask + = (0xf << 0), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_RLS_0001 + = (0x0 << 0), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_RLS_0011 + = (0x2 << 0), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_RLS_0101 + = (0x4 << 0), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_RLS_0111 + = (0x6 << 0), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_RLS_1001 + = (0x8 << 0), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_RLS_1011 + = (0xa << 0), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_RLS_1101 + = (0xc << 0), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_RLS_1111 + = (0xe << 0), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_ATK_Mask + = (0xf << 4), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_ATK_0001 + = (0x0 << 4), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_ATK_0011 + = (0x2 << 4), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_ATK_0101 + = (0x4 << 4), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_ATK_0111 + = (0x6 << 4), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_ATK_1001 + = (0x8 << 4), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_ATK_1011 + = (0xa << 4), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_ATK_1101 + = (0xc << 4), + MAX98927_Brownout_AMP_gain_attack_release_AMP_GAIN_ATK_1111 + = (0xe << 4), + + /* Brownout AMP1 clip mode (Address 0x0061)*/ + MAX98927_Brownout_AMP1_clip_mode = 0x0061, + MAX98927_Brownout_AMP1_clip_mode_AMP_CLIP_MODE = (0x1 << 0), + + /* Brownout level 1 current limit (Address 0x0062)*/ + MAX98927_Brownout__level_1_current_limit = 0x0062, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_1_current_limit_BDE_L1_ILIM_Mask + = (0x3f << 0), + + /* Brownout level 1 amp 1 control 1 (Address 0x0063)*/ + MAX98927_Brownout__level_1_amp_1_control_1 = 0x0063, + MAX98927_Brownout__level_1_amp_1_control_1_BDE_L1_AMP1_LIM_Mask + = (0xf << 0), + + /* Brownout level 1 amp 1 control 2 (Address 0x0064)*/ + MAX98927_Brownout__level_1_amp_1_control_2 = 0x0064, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_1_amp_1_control_2_BDE_L1_AMP1_CLIP_Mask + = (0x3f << 0), + + /* Brownout level 1 amp 1 control 3 (Address 0x0065)*/ + MAX98927_Brownout__level_1_amp_1_control_3 = 0x0065, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_1_amp_1_control_3_BDE_L1_AMP1_GAIN_Mask + = (0x3f << 0), + + /* Brownout level 2 current limit (Address 0x0066)*/ + MAX98927_Brownout__level_2_current_limit = 0x0066, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_2_current_limit_BDE_L2_ILIM_Mask = (0x3f << 0), + + /* Brownout level 2 amp 1 control 1 (Address 0x0067)*/ + MAX98927_Brownout__level_2_amp_1_control_1 = 0x0067, + MAX98927_Brownout__level_2_amp_1_control_1_BDE_L2_AMP1_LIM_Mask + = (0xf << 0), + + /* Brownout level 2 amp 1 control 2 (Address 0x0068)*/ + MAX98927_Brownout__level_2_amp_1_control_2 = 0x0068, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_2_amp_1_control_2_BDE_L2_AMP1_CLIP_Mask + = (0x3f << 0), + + /* Brownout level 2 amp 1 control 3 (Address 0x0069)*/ + MAX98927_Brownout__level_2_amp_1_control_3 = 0x0069, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_2_amp_1_control_3_BDE_L2_AMP1_GAIN_Mask + = (0x3f << 0), + + /* Brownout level 3 current limit (Address 0x006a)*/ + MAX98927_Brownout__level_3_current_limit = 0x006a, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_3_current_limit_BDE_L3_ILIM_Mask = (0x3f << 0), + + /* Brownout level 3 amp 1 control 1 (Address 0x006b)*/ + MAX98927_Brownout__level_3_amp_1_control_1 = 0x006b, + MAX98927_Brownout__level_3_amp_1_control_1_BDE_L3_AMP1_LIM_Mask + = (0xf << 0), + + /* Brownout level 3 amp 1 control 2 (Address 0x006c)*/ + MAX98927_Brownout__level_3_amp_1_control_2 = 0x006c, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_3_amp_1_control_2_BDE_L3_AMP1_CLIP_Mask + = (0x3f << 0), + + /* Brownout level 3 amp 1 control 3 (Address 0x006d)*/ + MAX98927_Brownout__level_3_amp_1_control_3 = 0x006d, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_3_amp_1_control_3_BDE_L3_AMP1_GAIN_Mask + = (0x3f << 0), + + /* Brownout level 4 current limit (Address 0x006e)*/ + MAX98927_Brownout__level_4_current_limit = 0x006e, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_4_current_limit_BDE_L4_ILIM_Mask = (0x3f << 0), + + /* Brownout level 4 amp 1 control 1 (Address 0x006f)*/ + MAX98927_Brownout__level_4_amp_1_control_1 = 0x006f, + MAX98927_Brownout__level_4_amp_1_control_1_BDE_L4_AMP1_LIM_Mask + = (0xf << 0), + + /* Brownout level 4 amp 1 control 2 (Address 0x0070)*/ + MAX98927_Brownout__level_4_amp_1_control_2 = 0x0070, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_4_amp_1_control_2_BDE_L4_AMP1_CLIP_Mask + = (0x3f << 0), + + /* Brownout level 4 amp 1 control 3 (Address 0x0071)*/ + MAX98927_Brownout__level_4_amp_1_control_3 = 0x0071, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_4_amp_1_control_3_BDE_L4_AMP1_GAIN_Mask + = (0x3f << 0), + + /* Brownout level 5 current limit (Address 0x0072)*/ + MAX98927_Brownout__level_5_current_limit = 0x0072, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_5_current_limit_BDE_L5_ILIM_Mask = (0x3f << 0), + + /* Brownout level 5 amp 1 control 1 (Address 0x0073)*/ + MAX98927_Brownout__level_5_amp_1_control_1 = 0x0073, + MAX98927_Brownout__level_5_amp_1_control_1_BDE_L5_AMP1_LIM_Mask + = (0xf << 0), + + /* Brownout level 5 amp 1 control 2 (Address 0x0074)*/ + MAX98927_Brownout__level_5_amp_1_control_2 = 0x0074, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_5_amp_1_control_2_BDE_L5_AMP1_CLIP_Mask + = (0x3f << 0), + + /* Brownout level 5 amp 1 control 3 (Address 0x0075)*/ + MAX98927_Brownout__level_5_amp_1_control_3 = 0x0075, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_5_amp_1_control_3_BDE_L5_AMP1_GAIN_Mask + = (0x3f << 0), + + /* Brownout level 6 current limit (Address 0x0076)*/ + MAX98927_Brownout__level_6_current_limit = 0x0076, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_6_current_limit_BDE_L6_ILIM_Mask = (0x3f << 0), + + /* Brownout level 6 amp 1 control 1 (Address 0x0077)*/ + MAX98927_Brownout__level_6_amp_1_control_1 = 0x0077, + MAX98927_Brownout__level_6_amp_1_control_1_BDE_L6_AMP1_LIM_Mask + = (0xf << 0), + + /* Brownout level 6 amp 1 control 2 (Address 0x0078)*/ + MAX98927_Brownout__level_6_amp_1_control_2 = 0x0078, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_6_amp_1_control_2_BDE_L6_AMP1_CLIP_Mask + = (0x3f << 0), + + /* Brownout level 6 amp 1 control 3 (Address 0x0079)*/ + MAX98927_Brownout__level_6_amp_1_control_3 = 0x0079, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_6_amp_1_control_3_BDE_L6_AMP1_GAIN_Mask + = (0x3f << 0), + + /* Brownout level 7 current limit (Address 0x007a)*/ + MAX98927_Brownout__level_7_current_limit = 0x007a, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_7_current_limit_BDE_L7_ILIM_Mask = (0x3f << 0), + + /* Brownout level 7 amp 1 control 1 (Address 0x007b)*/ + MAX98927_Brownout__level_7_amp_1_control_1 = 0x007b, + MAX98927_Brownout__level_7_amp_1_control_1_BDE_L7_AMP1_LIM_Mask + = (0xf << 0), + + /* Brownout level 7 amp 1 control 2 (Address 0x007c)*/ + MAX98927_Brownout__level_7_amp_1_control_2 = 0x007c, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_7_amp_1_control_2_BDE_L7_AMP1_CLIP_Mask + = (0x3f << 0), + + /* Brownout level 7 amp 1 control 3 (Address 0x007d)*/ + MAX98927_Brownout__level_7_amp_1_control_3 = 0x007d, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_7_amp_1_control_3_BDE_L7_AMP1_GAIN_Mask + = (0x3f << 0), + + /* Brownout level 8 current limit (Address 0x007e)*/ + MAX98927_Brownout__level_8_current_limit = 0x007e, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_8_current_limit_BDE_L8_ILIM_Mask = (0x3f << 0), + + /* Brownout level 8 amp 1 control 1 (Address 0x007f)*/ + MAX98927_Brownout__level_8_amp_1_control_1 = 0x007f, + MAX98927_Brownout__level_8_amp_1_control_1_BDE_L8_AMP1_LIM_Mask + = (0xf << 0), + + /* Brownout level 8 amp 1 control 2 (Address 0x0080)*/ + MAX98927_Brownout__lvl_8_amp_1_control_2 = 0x0080, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__lvl_8_amp_1_control_2_BDE_L8_AMP1_CLIP_Mask + = (0x3f << 0), + MAX98927_Brownout__lvl_8_amp_1_control_2_BDE_L8_AMP1_MUTE + = (0x1 << 7), + + /* Brownout level 8 amp 1 control 3 (Address 0x0081)*/ + MAX98927_Brownout__level_8_amp_1_control_3 = 0x0081, + /*#BYHAND width >= 5:*/ + MAX98927_Brownout__level_8_amp_1_control_3_BDE_L8_AMP1_GAIN_Mask + = (0x3f << 0), + + /* Env Tracker Vout Headroom (Address 0x0082)*/ + MAX98927_Env_Tracker_Vout_Headroom = 0x0082, + /*#BYHAND width >= 5:*/ + MAX98927_Env_Tracker_Vout_Head_ENV_TRACKER_BST_VOUT_HEADROOM_Mask + = (0x1f << 0), + + /* Env Tracker Boost Vout Delay (Address 0x0083)*/ + MAX98927_Env_Tracker_Boost_Vout_Delay = 0x0083, + /*#BYHAND width >= 5:*/ + MAX98927_Env_Tracker_Boost_V_Delay_ENV_TRACKER_BST_VOUT_DELAY_Mask + = (0x1f << 0), + MAX98927_Env_Tracker_Boost_Vout_Delay_ENV_TRACKER_BDE_MODE + = (0x1 << 7), + + /* Env Tracker Release Rate (Address 0x0084)*/ + MAX98927_Env_Tracker_Release_Rate = 0x0084, + MAX98927_Env_Tracker_Release_Rate_ENV_TRACKER_RLS_RATE_Mask + = (0x7 << 0), + MAX98927_Env_Tracker_Release_Rate_ENV_TRACKER_RLS_RATE_001 + = (0x0 << 0), + MAX98927_Env_Tracker_Release_Rate_ENV_TRACKER_RLS_RATE_011 + = (0x2 << 0), + MAX98927_Env_Tracker_Release_Rate_ENV_TRACKER_RLS_RATE_101 + = (0x4 << 0), + MAX98927_Env_Tracker_Release_Rate_ENV_TRACKER_RLS_RATE_111 + = (0x6 << 0), + MAX98927_Env_Tracker_Release_Rate_ENV_TRACKER_PEAK_DET_LPF_BYP_EN + = (0x1 << 3), + MAX98927_Env_Tracker_Release_Rate_ENV_TRACKER_RLS_RATE_SCALE_Mask + = (0x3 << 4), + MAX98927_Env_Tracker_Release_Rate_ENV_TRACKER_RLS_RATE_SCALE_01 + = (0x0 << 4), + MAX98927_Env_Tracker_Release_Rate_ENV_TRACKER_RLS_RATE_SCALE_11 + = (0x2 << 4), + + /* Env Tracker Hold Rate (Address 0x0085)*/ + MAX98927_Env_Tracker_Hold_Rate = 0x0085, + MAX98927_Env_Tracker_Hold_Rate_ENV_TRACKER_HOLD_RATE_Mask + = (0x7 << 0), + MAX98927_Env_Tracker_Hold_Rate_ENV_TRACKER_HOLD_RATE_001 + = (0x0 << 0), + MAX98927_Env_Tracker_Hold_Rate_ENV_TRACKER_HOLD_RATE_011 + = (0x2 << 0), + MAX98927_Env_Tracker_Hold_Rate_ENV_TRACKER_HOLD_RATE_101 + = (0x4 << 0), + MAX98927_Env_Tracker_Hold_Rate_ENV_TRACKER_HOLD_RATE_111 + = (0x6 << 0), + + /* Env Tracker Control (Address 0x0086)*/ + MAX98927_Env_Tracker_Control = 0x0086, + MAX98927_Env_Tracker_Control_ENV_TRACKER_EN = (0x1 << 0), + + /* Env Tracker Boost Vout ReadBack (Address 0x0087)*/ + MAX98927_Env_Tracker__Boost_Vout_ReadBack = 0x0087, + /*#BYHAND width >= 5:*/ + MAX98927_Env_Tracker__Boost_Vout_RB_ENV_TRACKER_BST_VOUT_RD_Mask + = (0x1f << 0), + + /* Advanced Settings (Address 0x0089)*/ + MAX98927_Advanced_Settings = 0x0089, + MAX98927_Advanced_Settings_DAC_HALF_FIR = (0x1 << 0), + MAX98927_Advanced_Settings_PDM_MOD_SEL = (0x1 << 1), + MAX98927_Advanced_Settings_ISOCH_EN = (0x1 << 2), + + /* DAC Test 1 (Address 0x009f)*/ + MAX98927_DAC_Test_1 = 0x009f, + MAX98927_DAC_Test_1_DAC_PCM_TIMING = (0x1 << 0), + MAX98927_DAC_Test_1_DAC_HALFI_AMP = (0x1 << 1), + MAX98927_DAC_Test_1_DAC_LONG_HOLD = (0x1 << 3), + MAX98927_DAC_Test_1_DAC_DISABLE_CHOP = (0x1 << 4), + MAX98927_DAC_Test_1_DAC_TM = (0x1 << 5), + MAX98927_DAC_Test_1_DAC_INVERT_DACCLK = (0x1 << 6), + + /* Authentication key 0 (Address 0x00ea)*/ + MAX98927_Authentication_key_0 = 0x00ea, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_key_0_AUTH_KEY_Mask = (0xff << 0), + + /* Authentication key 1 (Address 0x00eb)*/ + MAX98927_Authentication_key_1 = 0x00eb, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_key_1_AUTH_KEY_Mask = (0xff << 0), + + /* Authentication key 2 (Address 0x00ec)*/ + MAX98927_Authentication_key_2 = 0x00ec, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_key_2_AUTH_KEY_Mask = (0xff << 0), + + /* Authentication key 3 (Address 0x00ed)*/ + MAX98927_Authentication_key_3 = 0x00ed, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_key_3_AUTH_KEY_Mask = (0xff << 0), + + /* Authentication enable (Address 0x00ee)*/ + MAX98927_Authentication_enable = 0x00ee, + MAX98927_Authentication_enable_AUTH_EN = (0x1 << 0), + + /* Authentication result 0 (Address 0x00ef)*/ + MAX98927_Authentication_result_0 = 0x00ef, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_0_AUTH_RESULT_Mask = (0xff << 0), + + /* Authentication result 1 (Address 0x00f0)*/ + MAX98927_Authentication_result_1 = 0x00f0, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_1_AUTH_RESULT_Mask = (0xff << 0), + + /* Authentication result 2 (Address 0x00f1)*/ + MAX98927_Authentication_result_2 = 0x00f1, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_2_AUTH_RESULT_Mask = (0xff << 0), + + /* Authentication result 3 (Address 0x00f2)*/ + MAX98927_Authentication_result_3 = 0x00f2, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_3_AUTH_RESULT_Mask = (0xff << 0), + + /* Authentication result 4 (Address 0x00f3)*/ + MAX98927_Authentication_result_4 = 0x00f3, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_4_AUTH_RESULT_Mask = (0xff << 0), + + /* Authentication result 5 (Address 0x00f4)*/ + MAX98927_Authentication_result_5 = 0x00f4, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_5_AUTH_RESULT_Mask = (0xff << 0), + + /* Authentication result 6 (Address 0x00f5)*/ + MAX98927_Authentication_result_6 = 0x00f5, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_6_AUTH_RESULT_Mask = (0xff << 0), + + /* Authentication result 7 (Address 0x00f6)*/ + MAX98927_Authentication_result_7 = 0x00f6, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_7_AUTH_RESULT_Mask = (0xff << 0), + + /* Authentication result 8 (Address 0x00f7)*/ + MAX98927_Authentication_result_8 = 0x00f7, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_8_AUTH_RESULT_Mask = (0xff << 0), + + /* Authentication result 9 (Address 0x00f8)*/ + MAX98927_Authentication_result_9 = 0x00f8, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_9_AUTH_RESULT_Mask = (0xff << 0), + + /* Authentication result 10 (Address 0x00f9)*/ + MAX98927_Authentication_result_10 = 0x00f9, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_10_AUTH_RESULT_Mask = (0xff << 0), + + /* Authentication result 11 (Address 0x00fa)*/ + MAX98927_Authentication_result_11 = 0x00fa, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_11_AUTH_RESULT_Mask = (0xff << 0), + + /* Authentication result 12 (Address 0x00fb)*/ + MAX98927_Authentication_result_12 = 0x00fb, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_12_AUTH_RESULT_Mask = (0xff << 0), + + /* Authentication result 13 (Address 0x00fc)*/ + MAX98927_Authentication_result_13 = 0x00fc, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_13_AUTH_RESULT_Mask = (0xff << 0), + + /* Authentication result 14 (Address 0x00fd)*/ + MAX98927_Authentication_result_14 = 0x00fd, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_14_AUTH_RESULT_Mask = (0xff << 0), + + /* Authentication result 15 (Address 0x00fe)*/ + MAX98927_Authentication_result_15 = 0x00fe, + /*#BYHAND width >= 5:*/ + MAX98927_Authentication_result_15_AUTH_RESULT_Mask = (0xff << 0), + + /* Global Enable (Address 0x00ff)*/ + MAX98927_Global_Enable = 0x00ff, + MAX98927_Global_Enable_EN = (0x1 << 0), + /* Software Reset (Address 0x0100)*/ + MAX98927_Software_Reset = 0x0100, + MAX98927_Software_Reset_RST = (0x1 << 0), + + /* REV ID (Address 0x01ff)*/ + MAX98927_REV_ID = 0x01ff, + /*#BYHAND width >= 5:*/ + MAX98927_REV_ID_REV_ID_Mask = (0xff << 0), +} MAX98927Registers; + +struct max98927_reg_default { + unsigned int ch; + unsigned int reg; + unsigned int def; +}; +struct max98927_priv { + struct regmap *regmap; + struct regmap *sub_regmap; + struct snd_soc_codec *codec; + struct max98927_pdata *pdata; + const uint32_t *regcfg; + uint32_t regcfg_sz; + unsigned int spk_gain; + unsigned int sysclk; + unsigned int v_l_slot; + unsigned int i_l_slot; + unsigned int v_r_slot; + unsigned int i_r_slot; + bool interleave_mode; + unsigned int ch_size; + unsigned int rate; + unsigned int iface; + unsigned int master; + unsigned int thres_hyste; + unsigned int level5_hold; + unsigned int level6_hold; + unsigned int level7_hold; + unsigned int level8_hold; + unsigned int amp_limit; + unsigned int amp_limit_rel; + unsigned int amp1_level; + unsigned int amp2_level; + unsigned int amp3_level; + unsigned int amp1_level8; + unsigned int amp2_level8; + unsigned int amp3_level8; + unsigned int amp1_level7; + unsigned int amp2_level7; + unsigned int amp3_level7; + unsigned int amp1_level6; + unsigned int amp2_level6; + unsigned int amp3_level6; + unsigned int amp1_level5; + unsigned int amp2_level5; + unsigned int amp3_level5; + unsigned int digital_gain; + unsigned int pdm_gain; + unsigned int level_hold; + struct i2c_client *sub_i2c; + int sub_reg; + int sub_bus; +}; + +#define MAX98927_GLOBAL_SHIFT 0 +#define M98927_DAI_MSEL_SHIFT 4 +#define M98927_DAI_BSEL_SHIFT 0 +#define M98927_DAI_BSEL_32 (2 << M98927_DAI_BSEL_SHIFT) +#define M98927_DAI_BSEL_48 (3 << M98927_DAI_BSEL_SHIFT) +#define M98927_DAI_BSEL_64 (4 << M98927_DAI_BSEL_SHIFT) +#define M98927_DAI_MSEL_32 (2 << M98927_DAI_MSEL_SHIFT) +#define M98927_DAI_MSEL_48 (3 << M98927_DAI_MSEL_SHIFT) +#define M98927_DAI_MSEL_64 (4 << M98927_DAI_MSEL_SHIFT) +#define MAX98927_Speaker_Gain_Width 3 +#define MAX98927_SPK_RMP_EN_SHIFT 4 +#define MAX98927_PDM_GAIN_SHIFT 4 +#define MAX98927_pdm_Gain_Width 3 +#define MAX98927_AMP_VOL_WIDTH 7 +#define MAX98927_AMP_VOL_LOCATION_SHIFT 7 +#define MAX98927_PDM_Rx_Enable_PDM_CH_SHIFT 3 +#define MAX98927_PCM_to_speaker_monomix_A_SHIFT 6 +#define MAX98927_PCM_Sample_rate_setup_2_DIG_IF_SR_48000 (0x8 << 4) +#define MAX98927_PCM_FORMAT_DSP_A (0x3 << 3) +#define MAX98927_DRE_Control_DRE_SHIFT 0x1 +#define MAX98927_PCM_Master_Mode_PCM_MCLK_RATE_SHIFT (2) +#define MAX98927_Brownout_AMP_limiter_attack_release_shift 4 +#define MAX98927_BDE_DSP_SHIFT 2 +#define MAX98927_Speaker_Gain_SPK_PDM_GAIN_SHIFT (4) +#define MAX98927_BDE_AMP_SHIFT (1) +#define MAX98927_PCM_Tx_Ch_Sources_A_I_SHIFT (4) +#endif -- 1.9.1

4 6

[alsa-devel] [PATCH v3] dbri: Fix compiler warning
by Tushar Dave 28 Nov '16

28 Nov '16

dbri uses 'u32' for dma handle while invoking kernel DMA APIs, instead of using dma_addr_t. This hasn't caused any 'incompatible pointer type' warning on SPARC because until now dma_addr_t is of type u32. However, recent changes in SPARC ATU (iommu) enabled 64bit DMA and therefore dma_addr_t became of type u64. This makes 'incompatible pointer type' warnings inevitable. e.g. sound/sparc/dbri.c: In function ‘snd_dbri_create’: sound/sparc/dbri.c:2538: warning: passing argument 3 of ‘dma_zalloc_coherent’ from incompatible pointer type ./include/linux/dma-mapping.h:608: note: expected ‘dma_addr_t *’ but argument is of type ‘u32 *’ For the record, dbri(sbus) driver never executes on sun4v. Therefore even though 64bit DMA is enabled on SPARC, dbri continues to use legacy iommu that guarantees DMA address is always in 32bit range. This patch resolves above compiler warning. Signed-off-by: Tushar Dave <tushar.n.dave(a)oracle.com> Reviewed-by: thomas tai <thomas.tai(a)oracle.com> --- v2->v3: - Use %pad to print dma_addr_t. Comment by Takashi Iwai. sound/sparc/dbri.c | 26 ++++++++++++++++---------- 1 file changed, 16 insertions(+), 10 deletions(-) diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 0190cb6..3fe4468 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -304,7 +304,7 @@ struct snd_dbri { spinlock_t lock; struct dbri_dma *dma; /* Pointer to our DMA block */ - u32 dma_dvma; /* DBRI visible DMA address */ + dma_addr_t dma_dvma; /* DBRI visible DMA address */ void __iomem *regs; /* dbri HW regs */ int dbri_irqp; /* intr queue pointer */ @@ -657,12 +657,14 @@ static void dbri_cmdwait(struct snd_dbri *dbri) */ static s32 *dbri_cmdlock(struct snd_dbri *dbri, int len) { + u32 dvma_addr = (u32)dbri->dma_dvma; + /* Space for 2 WAIT cmds (replaced later by 1 JUMP cmd) */ len += 2; spin_lock(&dbri->cmdlock); if (dbri->cmdptr - dbri->dma->cmd + len < DBRI_NO_CMDS - 2) return dbri->cmdptr + 2; - else if (len < sbus_readl(dbri->regs + REG8) - dbri->dma_dvma) + else if (len < sbus_readl(dbri->regs + REG8) - dvma_addr) return dbri->dma->cmd; else printk(KERN_ERR "DBRI: no space for commands."); @@ -680,6 +682,7 @@ static s32 *dbri_cmdlock(struct snd_dbri *dbri, int len) */ static void dbri_cmdsend(struct snd_dbri *dbri, s32 *cmd, int len) { + u32 dvma_addr = (u32)dbri->dma_dvma; s32 tmp, addr; static int wait_id = 0; @@ -689,7 +692,7 @@ static void dbri_cmdsend(struct snd_dbri *dbri, s32 *cmd, int len) *(cmd+1) = DBRI_CMD(D_WAIT, 1, wait_id); /* Replace the last command with JUMP */ - addr = dbri->dma_dvma + (cmd - len - dbri->dma->cmd) * sizeof(s32); + addr = dvma_addr + (cmd - len - dbri->dma->cmd) * sizeof(s32); *(dbri->cmdptr+1) = addr; *(dbri->cmdptr) = DBRI_CMD(D_JUMP, 0, 0); @@ -747,6 +750,7 @@ static void dbri_reset(struct snd_dbri *dbri) /* Lock must not be held before calling this */ static void dbri_initialize(struct snd_dbri *dbri) { + u32 dvma_addr = (u32)dbri->dma_dvma; s32 *cmd; u32 dma_addr; unsigned long flags; @@ -764,7 +768,7 @@ static void dbri_initialize(struct snd_dbri *dbri) /* * Initialize the interrupt ring buffer. */ - dma_addr = dbri->dma_dvma + dbri_dma_off(intr, 0); + dma_addr = dvma_addr + dbri_dma_off(intr, 0); dbri->dma->intr[0] = dma_addr; dbri->dbri_irqp = 1; /* @@ -778,7 +782,7 @@ static void dbri_initialize(struct snd_dbri *dbri) dbri->cmdptr = cmd; *(cmd++) = DBRI_CMD(D_WAIT, 1, 0); *(cmd++) = DBRI_CMD(D_WAIT, 1, 0); - dma_addr = dbri->dma_dvma + dbri_dma_off(cmd, 0); + dma_addr = dvma_addr + dbri_dma_off(cmd, 0); sbus_writel(dma_addr, dbri->regs + REG8); spin_unlock(&dbri->cmdlock); @@ -1077,6 +1081,7 @@ static void recv_fixed(struct snd_dbri *dbri, int pipe, volatile __u32 *ptr) static int setup_descs(struct snd_dbri *dbri, int streamno, unsigned int period) { struct dbri_streaminfo *info = &dbri->stream_info[streamno]; + u32 dvma_addr = (u32)dbri->dma_dvma; __u32 dvma_buffer; int desc; int len; @@ -1177,7 +1182,7 @@ static int setup_descs(struct snd_dbri *dbri, int streamno, unsigned int period) else { dbri->next_desc[last_desc] = desc; dbri->dma->desc[last_desc].nda = - dbri->dma_dvma + dbri_dma_off(desc, desc); + dvma_addr + dbri_dma_off(desc, desc); } last_desc = desc; @@ -1192,7 +1197,7 @@ static int setup_descs(struct snd_dbri *dbri, int streamno, unsigned int period) } dbri->dma->desc[last_desc].nda = - dbri->dma_dvma + dbri_dma_off(desc, first_desc); + dvma_addr + dbri_dma_off(desc, first_desc); dbri->next_desc[last_desc] = first_desc; dbri->pipes[info->pipe].first_desc = first_desc; dbri->pipes[info->pipe].desc = first_desc; @@ -1697,6 +1702,7 @@ calling dbri_process_interrupt_buffer(), which works even if the CPU's static void xmit_descs(struct snd_dbri *dbri) { struct dbri_streaminfo *info; + u32 dvma_addr = (u32)dbri->dma_dvma; s32 *cmd; unsigned long flags; int first_td; @@ -1718,7 +1724,7 @@ static void xmit_descs(struct snd_dbri *dbri) *(cmd++) = DBRI_CMD(D_SDP, 0, dbri->pipes[info->pipe].sdp | D_SDP_P | D_SDP_EVERY | D_SDP_C); - *(cmd++) = dbri->dma_dvma + + *(cmd++) = dvma_addr + dbri_dma_off(desc, first_td); dbri_cmdsend(dbri, cmd, 2); @@ -1740,7 +1746,7 @@ static void xmit_descs(struct snd_dbri *dbri) *(cmd++) = DBRI_CMD(D_SDP, 0, dbri->pipes[info->pipe].sdp | D_SDP_P | D_SDP_EVERY | D_SDP_C); - *(cmd++) = dbri->dma_dvma + + *(cmd++) = dvma_addr + dbri_dma_off(desc, first_td); dbri_cmdsend(dbri, cmd, 2); @@ -2539,7 +2545,7 @@ static int snd_dbri_create(struct snd_card *card, if (!dbri->dma) return -ENOMEM; - dprintk(D_GEN, "DMA Cmd Block 0x%p (0x%08x)\n", + dprintk(D_GEN, "DMA Cmd Block 0x%p (%pad)\n", dbri->dma, dbri->dma_dvma); /* Map the registers into memory. */ -- 1.9.1

4 6

[alsa-devel] [PATCH] aplay: Fix to handle pause when system is suspended/Resumed
by jeeja.kp＠intel.com 28 Nov '16

28 Nov '16

From: Jeeja KP <jeeja.kp(a)intel.com> If PCM is paused and then we do system supend-resume, the stream throws error(EBADF) when stream is paused released. Check the pcm state before pause/release and if stream is suspended, call snd_pcm_resume to resume the stream. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> --- aplay/aplay.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/aplay/aplay.c b/aplay/aplay.c index 2da7dda..ee480f2 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -166,6 +166,8 @@ static void end_wave(int fd); static void begin_au(int fd, size_t count); static void end_au(int fd); +static void suspend(void); + static const struct fmt_capture { void (*start) (int fd, size_t count); void (*end) (int fd); @@ -1487,6 +1489,9 @@ static void do_pause(void) fprintf(stderr, _("\rPAUSE command ignored (no hw support)\n")); return; } + if (snd_pcm_state(handle) == SND_PCM_STATE_SUSPENDED) + suspend(); + err = snd_pcm_pause(handle, 1); if (err < 0) { error(_("pause push error: %s"), snd_strerror(err)); @@ -1496,6 +1501,8 @@ static void do_pause(void) while (read(fileno(stdin), &b, 1) != 1); if (b == ' ' || b == '\r') { while (read(fileno(stdin), &b, 1) == 1); + if (snd_pcm_state(handle) == SND_PCM_STATE_SUSPENDED) + suspend(); err = snd_pcm_pause(handle, 0); if (err < 0) error(_("pause release error: %s"), snd_strerror(err)); -- 2.5.0

2 1

[alsa-devel] [PATCH] alsa-lib: alsa dshare endless playback of buffer
by sutar.mounesh＠gmail.com 28 Nov '16

28 Nov '16

From: Anant Agrawal <Anant_Agrawal(a)mentor.com> On snd_pcm_drain() the slave PCM driven via plugin DSHARE is not filled with silence. Result is endless playback of buffer content until pcm is closed. In ALSA pcm dshare plugin, called do_silence method to fix the issue. Signed-off-by: Anant Agrawal <Anant_Agrawal(a)mentor.com> --- --- alsa-lib-1.0.29.orig/src/pcm/pcm_dshare.c 2015-04-28 08:20:44.662995477 -0700 +++ alsa-lib-1.0.29/src/pcm/pcm_dshare.c 2015-04-28 08:20:59.603077419 -0700 @@ -195,6 +195,7 @@ dshare->avail_max = avail; if (avail >= pcm->stop_threshold) { snd_timer_stop(dshare->timer); + do_silence(pcm); gettimestamp(&dshare->trigger_tstamp, pcm->tstamp_type); if (dshare->state == SND_PCM_STATE_RUNNING) { dshare->state = SND_PCM_STATE_XRUN;

2 1

[alsa-devel] [PATCH] alsa-lib: Protect from freeing semaphore when already in use
by sutar.mounesh＠gmail.com 28 Nov '16

28 Nov '16

From: Joshua Frkuska <joshua_frkuska(a)mentor.com> In the case of dshare, dsnoop, and dmix when a device is opened twice and fails the second time, the semaphore is completely discarded. This creates dangling semaphore data. This patch removes the possibility for the semaphore to be destroyed during a typical open failure by first checking if the shared memory can be destroyed or not. If the shared memory cannot be released it means both it and the semaphore are still in use and therefore the semaphore is just released. Signed-off-by: Joshua Frkuska <joshua_frkuska(a)mentor.com> --- src/pcm/pcm_dmix.c | 7 ++++--- src/pcm/pcm_dshare.c | 7 ++++--- src/pcm/pcm_dsnoop.c | 8 +++++--- 3 files changed, 13 insertions(+), 9 deletions(-) diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 2bd5d39..b17f759 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -1128,9 +1128,10 @@ int snd_pcm_dmix_open(snd_pcm_t **pcmp, const char *name, snd_pcm_close(spcm); if (dmix->u.dmix.shmid_sum >= 0) shm_sum_discard(dmix); - if (dmix->shmid >= 0) - snd_pcm_direct_shm_discard(dmix); - if (snd_pcm_direct_semaphore_discard(dmix) < 0) + if ((dmix->shmid >= 0) && (snd_pcm_direct_shm_discard(dmix))) { + if (snd_pcm_direct_semaphore_discard(dmix)) + snd_pcm_direct_semaphore_final(dmix, DIRECT_IPC_SEM_CLIENT); + } else snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); _err_nosem: if (dmix) { diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 5b32951..c080e7a 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -813,9 +813,10 @@ int snd_pcm_dshare_open(snd_pcm_t **pcmp, const char *name, snd_pcm_direct_client_discard(dshare); if (spcm) snd_pcm_close(spcm); - if (dshare->shmid >= 0) - snd_pcm_direct_shm_discard(dshare); - if (snd_pcm_direct_semaphore_discard(dshare) < 0) + if ((dshare->shmid >= 0) && (snd_pcm_direct_shm_discard(dshare))) { + if (snd_pcm_direct_semaphore_discard(dshare)) + snd_pcm_direct_semaphore_final(dshare, DIRECT_IPC_SEM_CLIENT); + } else snd_pcm_direct_semaphore_up(dshare, DIRECT_IPC_SEM_CLIENT); _err_nosem: if (dshare) { diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 055e4f4..74a3531 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -712,10 +712,12 @@ int snd_pcm_dsnoop_open(snd_pcm_t **pcmp, const char *name, snd_pcm_direct_client_discard(dsnoop); if (spcm) snd_pcm_close(spcm); - if (dsnoop->shmid >= 0) - snd_pcm_direct_shm_discard(dsnoop); - if (snd_pcm_direct_semaphore_discard(dsnoop) < 0) + if ((dsnoop->shmid >= 0) && (snd_pcm_direct_shm_discard(dsnoop))) { + if (snd_pcm_direct_semaphore_discard(dsnoop)) + snd_pcm_direct_semaphore_final(dsnoop, DIRECT_IPC_SEM_CLIENT); + } else snd_pcm_direct_semaphore_up(dsnoop, DIRECT_IPC_SEM_CLIENT); + _err_nosem: if (dsnoop) { free(dsnoop->bindings); -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH RESEND] linux-firmware: update BYT audio firmware
by Kyle McMartin 28 Nov '16

28 Nov '16

On Wed, Nov 23, 2016 at 08:59:19AM +0100, Sebastien Guiriec wrote: > From: Vinod Koul <vinod.koul(a)intel.com> > > This updates BYT audio firmware binary to latest version 01_0C_00_01 and > also adds binary for SSP0 port used in many BYT variants > > Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> > Signed-off-by: Sebastien Guiriec <sebastien.guiriec(a)intel.com> applied, thanks. regards, --kyle

1 0

[alsa-devel] [PATCH] ASoC: rt298: disable IRQ when jack is NULL
by Bard Liao 28 Nov '16

28 Nov '16

Disable IRQ when jack is NULL in rt298_mic_detect. So machine driver can use it in suspend/resume. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 2db8179..7150a40 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -326,11 +326,31 @@ static void rt298_jack_detect_work(struct work_struct *work) int rt298_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) { struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm; + bool hp = false; + bool mic = false; + int status = 0; + + /* If jack in NULL, disable HS jack */ + if (!jack) { + regmap_update_bits(rt298->regmap, RT298_IRQ_CTRL, 0x2, 0x0); + dapm = snd_soc_codec_get_dapm(codec); + snd_soc_dapm_disable_pin(dapm, "LDO1"); + snd_soc_dapm_sync(dapm); + return 0; + } rt298->jack = jack; + regmap_update_bits(rt298->regmap, RT298_IRQ_CTRL, 0x2, 0x2); - /* Send an initial empty report */ - snd_soc_jack_report(rt298->jack, 0, + rt298_jack_detect(rt298, &hp, &mic); + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt298->jack, status, SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); return 0; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] crec: Add option to specify codec ID
by Richard Fitzgerald 28 Nov '16

28 Nov '16

This patch adds a -I command line option to set the codec ID, either from a defined set of string values or as a number. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- src/utils/crec.c | 66 ++++++++++++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 59 insertions(+), 7 deletions(-) diff --git a/src/utils/crec.c b/src/utils/crec.c index 8d5b7b0..a586fc4 100644 --- a/src/utils/crec.c +++ b/src/utils/crec.c @@ -83,6 +83,27 @@ static bool streamed; static const unsigned int DEFAULT_CHANNELS = 1; static const unsigned int DEFAULT_RATE = 44100; static const unsigned int DEFAULT_FORMAT = SNDRV_PCM_FORMAT_S16_LE; +static const unsigned int DEFAULT_CODEC_ID = SND_AUDIOCODEC_PCM; + +static const struct { + const char *name; + unsigned int id; +} codec_ids[] = { + { "PCM", SND_AUDIOCODEC_PCM }, + { "MP3", SND_AUDIOCODEC_MP3 }, + { "AMR", SND_AUDIOCODEC_AMR }, + { "AMRWB", SND_AUDIOCODEC_AMRWB }, + { "ARMWBPLUS", SND_AUDIOCODEC_AMRWBPLUS }, + { "AAC", SND_AUDIOCODEC_AAC }, + { "WMA", SND_AUDIOCODEC_WMA }, + { "REAL", SND_AUDIOCODEC_REAL }, + { "VORBIS", SND_AUDIOCODEC_VORBIS }, + { "FLAC", SND_AUDIOCODEC_FLAC }, + { "IEC61937", SND_AUDIOCODEC_IEC61937 }, + { "G723_1", SND_AUDIOCODEC_G723_1 }, + { "G729", SND_AUDIOCODEC_G729 }, +}; +#define CREC_NUM_CODEC_IDS (sizeof(codec_ids) / sizeof(codec_ids[0])) struct riff_chunk { char desc[4]; @@ -153,6 +174,8 @@ static void size_wave_header(struct wave_header *header, uint32_t size) static void usage(void) { + int i; + fprintf(stderr, "usage: crec [OPTIONS] [filename]\n" "-c\tcard number\n" "-d\tdevice node\n" @@ -163,14 +186,22 @@ static void usage(void) "-h\tPrints this help list\n\n" "-C\tSpecify the number of channels (default %u)\n" "-R\tSpecify the sample rate (default %u)\n" - "-F\tSpecify the format: S16_LE, S32_LE (default S16_LE)\n\n" + "-F\tSpecify the format: S16_LE, S32_LE (default S16_LE)\n" + "-I\tSpecify codec ID (default PCM)\n\n" "If filename is not given the output is\n" "written to stdout\n\n" "Example:\n" "\tcrec -c 1 -d 2 test.wav\n" - "\tcrec -f 5 test.wav\n", + "\tcrec -f 5 test.wav\n\n" + "Valid codec IDs:\n", DEFAULT_CHANNELS, DEFAULT_RATE); + for (i = 0; i < CREC_NUM_CODEC_IDS; ++i) + fprintf(stderr, "%s%c", codec_ids[i].name, + (i % 8) ? ' ' : '\n'); + + fprintf(stderr, "\nor the value in decimal or hex\n"); + exit(EXIT_FAILURE); } @@ -239,7 +270,8 @@ static int finish_record(void) static void capture_samples(char *name, unsigned int card, unsigned int device, unsigned long buffer_size, unsigned int frag, unsigned int length, unsigned int rate, - unsigned int channels, unsigned int format) + unsigned int channels, unsigned int format, + unsigned int codec_id) { struct compr_config config; struct snd_codec codec; @@ -288,7 +320,7 @@ static void capture_samples(char *name, unsigned int card, unsigned int device, memset(&codec, 0, sizeof(codec)); memset(&config, 0, sizeof(config)); - codec.id = SND_AUDIOCODEC_PCM; + codec.id = codec_id; codec.ch_in = channels; codec.ch_out = channels; codec.sample_rate = rate; @@ -408,10 +440,11 @@ int main(int argc, char **argv) { char *file; unsigned long buffer_size = 0; - int c; + int c, i; unsigned int card = 0, device = 0, frag = 0, length = 0; unsigned int rate = DEFAULT_RATE, channels = DEFAULT_CHANNELS; unsigned int format = DEFAULT_FORMAT; + unsigned int codec_id = DEFAULT_CODEC_ID; if (signal(SIGINT, sig_handler) == SIG_ERR) { fprintf(stderr, "Error registering signal handler\n"); @@ -422,7 +455,7 @@ int main(int argc, char **argv) usage(); verbose = 0; - while ((c = getopt(argc, argv, "hvl:R:C:F:b:f:c:d:")) != -1) { + while ((c = getopt(argc, argv, "hvl:R:C:F:I:b:f:c:d:")) != -1) { switch (c) { case 'h': usage(); @@ -462,6 +495,25 @@ int main(int argc, char **argv) usage(); } break; + case 'I': + if (optarg[0] == '0') { + codec_id = strtol(optarg, NULL, 0); + } else { + for (i = 0; i < CREC_NUM_CODEC_IDS; ++i) { + if (strcmp(optarg, + codec_ids[i].name) == 0) { + codec_id = codec_ids[i].id; + break; + } + } + + if (i == CREC_NUM_CODEC_IDS) { + fprintf(stderr, "Unrecognised ID: %s\n", + optarg); + usage(); + } + } + break; default: exit(EXIT_FAILURE); } @@ -477,7 +529,7 @@ int main(int argc, char **argv) } capture_samples(file, card, device, buffer_size, frag, length, - rate, channels, format); + rate, channels, format, codec_id); fprintf(finfo, "Finish capturing... Close Normally\n"); -- 1.9.1

4 15

Re: [alsa-devel] [RFC PATCH v3 1/7] drm/i915: setup bridge for HDMI LPE audio driver
by Ville Syrjälä 28 Nov '16

28 Nov '16

On Fri, Nov 25, 2016 at 05:25:42AM +0530, Jerome Anand wrote: > Enable support for HDMI LPE audio mode on Baytrail and > Cherrytrail when HDaudio controller is not detected > > Setup minimum required resources during i915_driver_load: > 1. Create a platform device to share MMIO/IRQ resources > 2. Make the platform device child of i915 device for runtime PM. > 3. Create IRQ chip to forward HDMI LPE audio irqs. > > HDMI LPE audio driver (a standalone sound driver) probes the > LPE audio device and creates a new sound card. > > Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> > Signed-off-by: Jerome Anand <jerome.anand(a)intel.com> > --- > Documentation/gpu/i915.rst | 9 + > drivers/gpu/drm/i915/Makefile | 3 + > drivers/gpu/drm/i915/i915_drv.c | 8 +- > drivers/gpu/drm/i915/i915_drv.h | 15 ++ > drivers/gpu/drm/i915/i915_irq.c | 29 +++ > drivers/gpu/drm/i915/i915_reg.h | 3 + > drivers/gpu/drm/i915/intel_lpe_audio.c | 369 +++++++++++++++++++++++++++++++++ > include/drm/intel_lpe_audio.h | 45 ++++ > 8 files changed, 479 insertions(+), 2 deletions(-) > create mode 100644 drivers/gpu/drm/i915/intel_lpe_audio.c > create mode 100644 include/drm/intel_lpe_audio.h > > diff --git a/Documentation/gpu/i915.rst b/Documentation/gpu/i915.rst > index 117d2ab..a671eee 100644 > --- a/Documentation/gpu/i915.rst > +++ b/Documentation/gpu/i915.rst > @@ -213,6 +213,15 @@ Video BIOS Table (VBT) > .. kernel-doc:: drivers/gpu/drm/i915/intel_vbt_defs.h > :internal: > > +intel hdmi lpe audio support > +---------------------------- > + > +.. kernel-doc:: drivers/gpu/drm/i915/intel_lpe_audio.c > + :doc: LPE Audio integration for HDMI or DP playback > + > +.. kernel-doc:: drivers/gpu/drm/i915/intel_lpe_audio.c > + :internal: > + > Memory Management and Command Submission > ======================================== > > diff --git a/drivers/gpu/drm/i915/Makefile b/drivers/gpu/drm/i915/Makefile > index 580602d..3a2369c 100644 > --- a/drivers/gpu/drm/i915/Makefile > +++ b/drivers/gpu/drm/i915/Makefile > @@ -126,6 +126,9 @@ i915-y += intel_gvt.o > include $(src)/gvt/Makefile > endif > > +# LPE Audio for VLV and CHT > +i915-y += intel_lpe_audio.o > + > obj-$(CONFIG_DRM_I915) += i915.o > > CFLAGS_i915_trace_points.o := -I$(src) > diff --git a/drivers/gpu/drm/i915/i915_drv.c b/drivers/gpu/drm/i915/i915_drv.c > index b893e67..93811ed 100644 > --- a/drivers/gpu/drm/i915/i915_drv.c > +++ b/drivers/gpu/drm/i915/i915_drv.c > @@ -1144,7 +1144,8 @@ static void i915_driver_register(struct drm_i915_private *dev_priv) > if (IS_GEN5(dev_priv)) > intel_gpu_ips_init(dev_priv); > > - i915_audio_component_init(dev_priv); > + if (intel_lpe_audio_init(dev_priv) < 0) > + i915_audio_component_init(dev_priv); > > /* > * Some ports require correctly set-up hpd registers for detection to > @@ -1162,7 +1163,10 @@ static void i915_driver_register(struct drm_i915_private *dev_priv) > */ > static void i915_driver_unregister(struct drm_i915_private *dev_priv) > { > - i915_audio_component_cleanup(dev_priv); > + if (HAS_LPE_AUDIO(dev_priv)) > + intel_lpe_audio_teardown(dev_priv); > + else > + i915_audio_component_cleanup(dev_priv); > > intel_gpu_ips_teardown(); > acpi_video_unregister(); > diff --git a/drivers/gpu/drm/i915/i915_drv.h b/drivers/gpu/drm/i915/i915_drv.h > index 970e50b..2a79048 100644 > --- a/drivers/gpu/drm/i915/i915_drv.h > +++ b/drivers/gpu/drm/i915/i915_drv.h > @@ -2284,6 +2284,12 @@ struct drm_i915_private { > /* Used to save the pipe-to-encoder mapping for audio */ > struct intel_encoder *av_enc_map[I915_MAX_PIPES]; > > + /* necessary resource sharing with HDMI LPE audio driver. */ > + struct { > + struct platform_device *platdev; > + int irq; > + } lpe_audio; > + > /* > * NOTE: This is the dri1/ums dungeon, don't add stuff here. Your patch > * will be rejected. Instead look for a better place. > @@ -2773,6 +2779,8 @@ intel_info(const struct drm_i915_private *dev_priv) > > #define HAS_POOLED_EU(dev_priv) ((dev_priv)->info.has_pooled_eu) > > +#define HAS_LPE_AUDIO(dev_priv) ((dev_priv)->lpe_audio.platdev != NULL) > + > #define INTEL_PCH_DEVICE_ID_MASK 0xff00 > #define INTEL_PCH_IBX_DEVICE_ID_TYPE 0x3b00 > #define INTEL_PCH_CPT_DEVICE_ID_TYPE 0x1c00 > @@ -3547,6 +3555,13 @@ extern int i915_restore_state(struct drm_device *dev); > void i915_setup_sysfs(struct drm_i915_private *dev_priv); > void i915_teardown_sysfs(struct drm_i915_private *dev_priv); > > +/* i915_lpe_audio.c */ > +int intel_lpe_audio_init(struct drm_i915_private *dev_priv); > +int intel_lpe_audio_setup(struct drm_i915_private *dev_priv); > +void intel_lpe_audio_teardown(struct drm_i915_private *dev_priv); > +void intel_lpe_audio_irq_handler(struct drm_i915_private *dev_priv); > +bool intel_lpe_audio_detect(struct drm_i915_private *dev_priv); > + > /* intel_i2c.c */ > extern int intel_setup_gmbus(struct drm_device *dev); > extern void intel_teardown_gmbus(struct drm_device *dev); > diff --git a/drivers/gpu/drm/i915/i915_irq.c b/drivers/gpu/drm/i915/i915_irq.c > index 07ca71c..6c98b28 100644 > --- a/drivers/gpu/drm/i915/i915_irq.c > +++ b/drivers/gpu/drm/i915/i915_irq.c > @@ -1894,6 +1894,15 @@ static irqreturn_t valleyview_irq_handler(int irq, void *arg) > valleyview_pipestat_irq_ack(dev_priv, iir, pipe_stats); > > /* > + * LPE audio interrupts are only enabled on Baytrail and > + * CherryView platforms without HDaudio > + */ > + if (iir & (I915_LPE_PIPE_A_INTERRUPT | > + I915_LPE_PIPE_B_INTERRUPT | > + I915_LPE_PIPE_C_INTERRUPT)) No pipe C on vlv. > + intel_lpe_audio_irq_handler(dev_priv); > + > + /* > * VLV_IIR is single buffered, and reflects the level > * from PIPESTAT/PORT_HOTPLUG_STAT, hence clear it last. > */ > @@ -1974,6 +1983,15 @@ static irqreturn_t cherryview_irq_handler(int irq, void *arg) > valleyview_pipestat_irq_ack(dev_priv, iir, pipe_stats); > > /* > + * LPE audio interrupts are only enabled on Baytrail and > + * CherryView platforms without HDaudio > + */ > + if (iir & (I915_LPE_PIPE_A_INTERRUPT | > + I915_LPE_PIPE_B_INTERRUPT | > + I915_LPE_PIPE_C_INTERRUPT)) > + intel_lpe_audio_irq_handler(dev_priv); > + > + /* > * VLV_IIR is single buffered, and reflects the level > * from PIPESTAT/PORT_HOTPLUG_STAT, hence clear it last. > */ > @@ -2930,6 +2948,17 @@ static void vlv_display_irq_postinstall(struct drm_i915_private *dev_priv) > > WARN_ON(dev_priv->irq_mask != ~0); > > + if (IS_VALLEYVIEW(dev_priv) || IS_CHERRYVIEW(dev_priv)) { This is vlv/chv specific code, so no need to check for it. > + /* add interrupt masks unconditially here, the actual unmask > + * will take place only if the LPE_AUDIO mode is detected > + */ > + u32 val = (I915_LPE_PIPE_A_INTERRUPT | > + I915_LPE_PIPE_B_INTERRUPT | > + I915_LPE_PIPE_C_INTERRUPT); > + > + enable_mask |= val; > + } > + > dev_priv->irq_mask = ~enable_mask; > > GEN5_IRQ_INIT(VLV_, dev_priv->irq_mask, enable_mask); > diff --git a/drivers/gpu/drm/i915/i915_reg.h b/drivers/gpu/drm/i915/i915_reg.h > index fcad8fa..6b16ed0 100644 > --- a/drivers/gpu/drm/i915/i915_reg.h > +++ b/drivers/gpu/drm/i915/i915_reg.h > @@ -2395,6 +2395,9 @@ enum skl_disp_power_wells { > #define I915_ASLE_INTERRUPT (1<<0) > #define I915_BSD_USER_INTERRUPT (1<<25) > > +#define I915_HDMI_LPE_AUDIO_BASE (VLV_DISPLAY_BASE + 0x65000) > +#define I915_HDMI_LPE_AUDIO_SIZE 0x1000 > + > #define GEN6_BSD_RNCID _MMIO(0x12198) > > #define GEN7_FF_THREAD_MODE _MMIO(0x20a0) > diff --git a/drivers/gpu/drm/i915/intel_lpe_audio.c b/drivers/gpu/drm/i915/intel_lpe_audio.c > new file mode 100644 > index 0000000..5335fc6 > --- /dev/null > +++ b/drivers/gpu/drm/i915/intel_lpe_audio.c > @@ -0,0 +1,369 @@ > +/* > + * Copyright © 2016 Intel Corporation > + * > + * Permission is hereby granted, free of charge, to any person obtaining a > + * copy of this software and associated documentation files (the "Software"), > + * to deal in the Software without restriction, including without limitation > + * the rights to use, copy, modify, merge, publish, distribute, sublicense, > + * and/or sell copies of the Software, and to permit persons to whom the > + * Software is furnished to do so, subject to the following conditions: > + * > + * The above copyright notice and this permission notice (including the next > + * paragraph) shall be included in all copies or substantial portions of the > + * Software. > + * > + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR > + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, > + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL > + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER > + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING > + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS > + * IN THE SOFTWARE. > + * > + * Authors: > + * Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> > + * Jerome Anand <jerome.anand(a)intel.com> > + * based on VED patches > + * > + */ > + > +/** > + * DOC: LPE Audio integration for HDMI or DP playback > + * > + * Motivation: > + * Atom platforms (e.g. valleyview and cherryTrail) integrates a DMA-based > + * interface as an alternative to the traditional HDaudio path. While this > + * mode is unrelated to the LPE aka SST audio engine, the documentation refers > + * to this mode as LPE so we keep this notation for the sake of consistency. > + * > + * The interface is handled by a separate standalone driver maintained in the > + * ALSA subsystem for simplicity. To minimize the interaction between the two > + * subsystems, a bridge is setup between the hdmi-lpe-audio and i915: > + * 1. Create a platform device to share MMIO/IRQ resources > + * 2. Make the platform device child of i915 device for runtime PM. > + * 3. Create IRQ chip to forward the LPE audio irqs. > + * the hdmi-lpe-audio driver probes the lpe audio device and creates a new > + * sound card > + * > + * Threats: > + * Due to the restriction in Linux platform device model, user need manually > + * uninstall the hdmi-lpe-audio driver before uninstalling i915 module, > + * otherwise we might run into use-after-free issues after i915 removes the > + * platform device: even though hdmi-lpe-audio driver is released, the modules > + * is still in "installed" status. > + * > + * Implementation: > + * The MMIO/REG platform resources are created according to the registers > + * specification. > + * When forwarding LPE audio irqs, the flow control handler selection depends > + * on the platform, for example on valleyview handle_simple_irq is enough. > + * > + */ > + > +#include <linux/acpi.h> > +#include <linux/device.h> > +#include <linux/pci.h> > + > +#include "i915_drv.h" > +#include <linux/delay.h> > +#include <drm/intel_lpe_audio.h> > + > +static struct platform_device* > +lpe_audio_platdev_create(struct drm_i915_private *dev_priv) > +{ > + struct drm_device *dev = &dev_priv->drm; > + int ret; > + struct resource rsc[2] = {}; > + struct platform_device *platdev; > + u64 *dma_mask; > + struct intel_hdmi_lpe_audio_pdata *pdata; > + > + if (dev_priv->lpe_audio.irq < 0) > + return ERR_PTR(-EINVAL); > + > + platdev = platform_device_alloc("hdmi-lpe-audio", -1); > + if (!platdev) { > + ret = -ENOMEM; > + DRM_ERROR("Failed to allocate LPE audio platform device\n"); > + goto err; > + } > + > + /* to work-around check_addr in nommu_map_sg() */ > + dma_mask = kmalloc(sizeof(*platdev->dev.dma_mask), GFP_KERNEL); > + if (!dma_mask) { > + ret = -ENOMEM; > + DRM_ERROR("Failed to allocate dma_mask\n"); > + goto err_put_dev; > + } > + *dma_mask = DMA_BIT_MASK(32); > + platdev->dev.dma_mask = dma_mask; > + platdev->dev.coherent_dma_mask = *dma_mask; > + > + rsc[0].start = rsc[0].end = dev_priv->lpe_audio.irq; > + rsc[0].flags = IORESOURCE_IRQ; > + rsc[0].name = "hdmi-lpe-audio-irq"; > + > + rsc[1].start = pci_resource_start(dev->pdev, 0) + > + I915_HDMI_LPE_AUDIO_BASE; > + rsc[1].end = pci_resource_start(dev->pdev, 0) + > + I915_HDMI_LPE_AUDIO_BASE + I915_HDMI_LPE_AUDIO_SIZE - 1; > + rsc[1].flags = IORESOURCE_MEM; > + rsc[1].name = "hdmi-lpe-audio-mmio"; > + > + ret = platform_device_add_resources(platdev, rsc, 2); > + if (ret) { > + DRM_ERROR("Failed to add resource for platform device: %d\n", > + ret); > + goto err_put_dev; > + } > + > + pdata = kzalloc(sizeof(*pdata), GFP_KERNEL); > + > + if (pdata == NULL) { > + ret = -ENOMEM; > + goto err_put_dev; > + } > + platdev->dev.platform_data = pdata; > + spin_lock_init(&pdata->lpe_audio_slock); > + > + /* for LPE audio driver's runtime-PM */ > + platdev->dev.parent = dev->dev; > + ret = platform_device_add(platdev); > + if (ret) { > + DRM_ERROR("Failed to add LPE audio platform device: %d\n", > + ret); > + goto err_put_dev; > + } > + > + return platdev; > +err_put_dev: > + platform_device_put(platdev); > + kfree(dma_mask); > +err: > + return ERR_PTR(ret); > +} > + > +static void lpe_audio_platdev_destroy(struct drm_i915_private *dev_priv) > +{ > + if (!HAS_LPE_AUDIO(dev_priv)) > + return; > + platform_device_unregister(dev_priv->lpe_audio.platdev); > + kfree(dev_priv->lpe_audio.platdev->dev.dma_mask); > +} > + > +static void lpe_audio_irq_unmask(struct irq_data *d) > +{ > + struct drm_device *dev = d->chip_data; > + struct drm_i915_private *dev_priv = to_i915(dev); > + unsigned long irqflags; > + u32 val = (I915_LPE_PIPE_A_INTERRUPT | > + I915_LPE_PIPE_B_INTERRUPT | > + I915_LPE_PIPE_C_INTERRUPT); No pipe C on vlv. > + > + spin_lock_irqsave(&dev_priv->irq_lock, irqflags); > + > + /* > + * VLV_IER is already set in the vlv_display_postinstall(), > + * we only change VLV_IIR and VLV_IMR > + */ > + dev_priv->irq_mask &= ~val; > + I915_WRITE(VLV_IIR, val); > + I915_WRITE(VLV_IIR, val); > + I915_WRITE(VLV_IMR, dev_priv->irq_mask); > + POSTING_READ(VLV_IMR); > + > + spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); > +} > + > +static void lpe_audio_irq_mask(struct irq_data *d) > +{ > + struct drm_device *dev = d->chip_data; > + struct drm_i915_private *dev_priv = to_i915(dev); > + unsigned long irqflags; > + u32 val = (I915_LPE_PIPE_A_INTERRUPT | > + I915_LPE_PIPE_B_INTERRUPT | > + I915_LPE_PIPE_C_INTERRUPT); ditto > + > + spin_lock_irqsave(&dev_priv->irq_lock, irqflags); > + > + /* > + * VLV_IER is already set in the vlv_display_postinstall(), > + * we only change VLV_IIR and VLV_IMR > + */ > + dev_priv->irq_mask |= val; > + I915_WRITE(VLV_IMR, dev_priv->irq_mask); > + I915_WRITE(VLV_IIR, val); > + I915_WRITE(VLV_IIR, val); > + POSTING_READ(VLV_IIR); > + > + spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); > +} > + > +static struct irq_chip lpe_audio_irqchip = { > + .name = "hdmi_lpe_audio_irqchip", > + .irq_mask = lpe_audio_irq_mask, > + .irq_unmask = lpe_audio_irq_unmask, > +}; > + > +static int lpe_audio_irq_init(struct drm_i915_private *dev_priv) > +{ > + int irq = dev_priv->lpe_audio.irq; > + > + WARN_ON(!intel_irqs_enabled(dev_priv)); > + irq_set_chip_and_handler_name(irq, > + &lpe_audio_irqchip, > + handle_simple_irq, > + "hdmi_lpe_audio_irq_handler"); > + > + return irq_set_chip_data(irq, &dev_priv->drm); Just pass dev_priv as the chip data. > +} > + > +/** > + * intel_lpe_audio_irq_handler() - forwards the LPE audio irq > + * @dev_priv: the i915 drm device private data > + * > + * the LPE Audio irq is forwarded to the irq handler registered by LPE audio > + * driver. > + */ > +void intel_lpe_audio_irq_handler(struct drm_i915_private *dev_priv) > +{ > + int ret; > + > + ret = generic_handle_irq(dev_priv->lpe_audio.irq); > + if (ret) > + DRM_ERROR_RATELIMITED("error handling LPE audio irq: %d\n", > + ret); > +} > + > +/** > + * intel_lpe_audio_detect() - check & setup lpe audio if present > + * @dev_priv: the i915 drm device private data > + * > + * Detect if lpe audio is present > + * > + * Return: true if lpe audio present else Return = false > + */ > +bool intel_lpe_audio_detect(struct drm_i915_private *dev_priv) > +{ > + int lpe_present = false; > + > + if (IS_VALLEYVIEW(dev_priv) || IS_CHERRYVIEW(dev_priv)) { Could avoid indenting the whole thig by reversing this test and doing an early return. > + static const struct pci_device_id atom_hdaudio_ids[] = { > + /* Baytrail */ > + {PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x0f04)}, > + /* Braswell */ > + {PCI_DEVICE(PCI_VENDOR_ID_INTEL, 0x2284)}, > + {} > + }; IIRC I asked if we could use a HDA class match instead. But I forget what the answer was. No presumably? > + > + if (!pci_dev_present(atom_hdaudio_ids)) { > + DRM_INFO("%s%s\n", "HDaudio controller not detected,", > + "using LPE audio instead\n"); > + lpe_present = true; > + } > + } > + return lpe_present; > +} > + > +/** > + * intel_lpe_audio_setup() - setup the bridge between HDMI LPE Audio > + * driver and i915 > + * @dev_priv: the i915 drm device private data > + * > + * set up the minimum required resources for the bridge: irq chip, > + * platform resource and platform device. i915 device is set as parent > + * of the new platform device. > + * > + * Return: 0 if successful. non-zero if allocation/initialization fails > + */ > +int intel_lpe_audio_setup(struct drm_i915_private *dev_priv) > +{ > + int ret; > + > + dev_priv->lpe_audio.irq = irq_alloc_desc(0); > + if (dev_priv->lpe_audio.irq < 0) { > + DRM_ERROR("Failed to allocate IRQ desc: %d\n", > + dev_priv->lpe_audio.irq); > + ret = dev_priv->lpe_audio.irq; > + goto err; > + } > + > + DRM_DEBUG("irq = %d\n", dev_priv->lpe_audio.irq); > + > + ret = lpe_audio_irq_init(dev_priv); > + > + if (ret) { > + DRM_ERROR("Failed to initialize irqchip for lpe audio: %d\n", > + ret); > + goto err_free_irq; > + } > + > + dev_priv->lpe_audio.platdev = lpe_audio_platdev_create(dev_priv); > + > + if (IS_ERR(dev_priv->lpe_audio.platdev)) { > + ret = PTR_ERR(dev_priv->lpe_audio.platdev); > + DRM_ERROR("Failed to create lpe audio platform device: %d\n", > + ret); > + goto err_free_irq; > + } > + > + return 0; > +err_free_irq: > + irq_free_desc(dev_priv->lpe_audio.irq); > +err: > + dev_priv->lpe_audio.irq = -1; > + dev_priv->lpe_audio.platdev = NULL; > + return ret; > +} > + > + > +/** > + * intel_lpe_audio_init() - detect and setup the bridge between HDMI LPE Audio > + * driver and i915 > + * @dev_priv: the i915 drm device private data > + * > + * Return: 0 if successful. non-zero if detection or > + * llocation/initialization fails > + */ > +int intel_lpe_audio_init(struct drm_i915_private *dev_priv) > +{ > + int ret = -ENODEV; > + > + if (intel_lpe_audio_detect(dev_priv)) { > + ret = intel_lpe_audio_setup(dev_priv); > + if (ret < 0) > + DRM_ERROR("failed to setup LPE Audio bridge\n"); > + } > + return ret; > +} > + > +/** > + * intel_lpe_audio_teardown() - destroy the bridge between HDMI LPE > + * audio driver and i915 > + * @dev_priv: the i915 drm device private data > + * > + * release all the resources for LPE audio <-> i915 bridge. > + */ > +void intel_lpe_audio_teardown(struct drm_i915_private *dev_priv) > +{ > + unsigned long irqflags; > + struct irq_desc *desc; > + > + desc = (dev_priv->lpe_audio.irq >= 0) ? > + irq_to_desc(dev_priv->lpe_audio.irq) : NULL; Why not just "if (lpe not initialized) return;" ? > + > + /** > + * mask LPE audio irq before destroying > + */ > + if (desc) > + lpe_audio_irq_mask(&desc->irq_data); This seems racy. Should unregister the platdev first I think, > + > + spin_lock_irqsave(&dev_priv->irq_lock, irqflags); > + > + lpe_audio_platdev_destroy(dev_priv); > + > + if (desc) > + irq_free_desc(dev_priv->lpe_audio.irq); > + > + spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); > +} > diff --git a/include/drm/intel_lpe_audio.h b/include/drm/intel_lpe_audio.h > new file mode 100644 > index 0000000..a64c449 > --- /dev/null > +++ b/include/drm/intel_lpe_audio.h > @@ -0,0 +1,45 @@ > +/* > + * Copyright © 2016 Intel Corporation > + * > + * Permission is hereby granted, free of charge, to any person obtaining a > + * copy of this software and associated documentation files (the "Software"), > + * to deal in the Software without restriction, including without limitation > + * the rights to use, copy, modify, merge, publish, distribute, sublicense, > + * and/or sell copies of the Software, and to permit persons to whom the > + * Software is furnished to do so, subject to the following conditions: > + * > + * The above copyright notice and this permission notice (including the next > + * paragraph) shall be included in all copies or substantial portions of the > + * Software. > + * > + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR > + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, > + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL > + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER > + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING > + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS > + * IN THE SOFTWARE. > + */ > + > +#ifndef _INTEL_LPE_AUDIO_H_ > +#define _INTEL_LPE_AUDIO_H_ > + > +#include <linux/types.h> > + > +#define HDMI_MAX_ELD_BYTES 128 > + > +struct intel_hdmi_lpe_audio_eld { > + int port_id; > + unsigned char eld_data[HDMI_MAX_ELD_BYTES]; > +}; > + > +struct intel_hdmi_lpe_audio_pdata { > + bool notify_pending; > + int tmds_clock_speed; > + bool hdmi_connected; > + struct intel_hdmi_lpe_audio_eld eld; > + void (*notify_audio_lpe)(void *audio_ptr); > + spinlock_t lpe_audio_slock; > +}; > + > +#endif /* _I915_LPE_AUDIO_H_ */ > -- > 2.9.3 -- Ville Syrjälä Intel OTC

4 7

[alsa-devel] [PATCH v3] clkdev: add devm_of_clk_get()
by Kuninori Morimoto 28 Nov '16

28 Nov '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current Linux has of_clk_get(), but doesn't have devm_of_clk_get(). This patch adds it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 - implement in clk-devres.c, and reused existing devm_clk_release() drivers/clk/clk-devres.c | 21 +++++++++++++++++++++ include/linux/clk.h | 7 +++++++ 2 files changed, 28 insertions(+) diff --git a/drivers/clk/clk-devres.c b/drivers/clk/clk-devres.c index 8f57154..2449b25 100644 --- a/drivers/clk/clk-devres.c +++ b/drivers/clk/clk-devres.c @@ -53,3 +53,24 @@ void devm_clk_put(struct device *dev, struct clk *clk) WARN_ON(ret); } EXPORT_SYMBOL(devm_clk_put); + +struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index) +{ + struct clk **ptr, *clk; + + ptr = devres_alloc(devm_clk_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return ERR_PTR(-ENOMEM); + + clk = of_clk_get(np, index); + if (!IS_ERR(clk)) { + *ptr = clk; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return clk; +} +EXPORT_SYMBOL(devm_of_clk_get); diff --git a/include/linux/clk.h b/include/linux/clk.h index 123c027..1b713db 100644 --- a/include/linux/clk.h +++ b/include/linux/clk.h @@ -506,6 +506,8 @@ static inline void clk_disable_unprepare(struct clk *clk) #if defined(CONFIG_OF) && defined(CONFIG_COMMON_CLK) struct clk *of_clk_get(struct device_node *np, int index); +struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index); struct clk *of_clk_get_by_name(struct device_node *np, const char *name); struct clk *of_clk_get_from_provider(struct of_phandle_args *clkspec); #else @@ -513,6 +515,11 @@ static inline struct clk *of_clk_get(struct device_node *np, int index) { return ERR_PTR(-ENOENT); } +static inline struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index) +{ + return ERR_PTR(-ENOENT); +} static inline struct clk *of_clk_get_by_name(struct device_node *np, const char *name) { -- 1.9.1

3 2

[alsa-devel] [PATCH v2] clkdev: add devm_of_clk_get()
by Kuninori Morimoto 28 Nov '16

28 Nov '16

Current Linux has of_clk_get(), but doesn't have devm_of_clk_get(). This patch adds it. This is based on devm_clk_get() Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - update git log drivers/clk/clkdev.c | 26 ++++++++++++++++++++++++++ include/linux/clk.h | 7 +++++++ 2 files changed, 33 insertions(+) diff --git a/drivers/clk/clkdev.c b/drivers/clk/clkdev.c index 89cc700..93a613b 100644 --- a/drivers/clk/clkdev.c +++ b/drivers/clk/clkdev.c @@ -55,6 +55,32 @@ struct clk *of_clk_get(struct device_node *np, int index) } EXPORT_SYMBOL(of_clk_get); +static void devm_of_clk_release(struct device *dev, void *res) +{ + clk_put(*(struct clk **)res); +} + +struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index) +{ + struct clk **ptr, *clk; + + ptr = devres_alloc(devm_of_clk_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return ERR_PTR(-ENOMEM); + + clk = of_clk_get(np, index); + if (!IS_ERR(clk)) { + *ptr = clk; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return clk; +} +EXPORT_SYMBOL(devm_of_clk_get); + static struct clk *__of_clk_get_by_name(struct device_node *np, const char *dev_id, const char *name) diff --git a/include/linux/clk.h b/include/linux/clk.h index a89ba4e..33cd540 100644 --- a/include/linux/clk.h +++ b/include/linux/clk.h @@ -502,6 +502,8 @@ struct of_phandle_args; #if defined(CONFIG_OF) && defined(CONFIG_COMMON_CLK) struct clk *of_clk_get(struct device_node *np, int index); +struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index); struct clk *of_clk_get_by_name(struct device_node *np, const char *name); struct clk *of_clk_get_from_provider(struct of_phandle_args *clkspec); #else @@ -509,6 +511,11 @@ static inline struct clk *of_clk_get(struct device_node *np, int index) { return ERR_PTR(-ENOENT); } +static inline struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index) +{ + return ERR_PTR(-ENOENT); +} static inline struct clk *of_clk_get_by_name(struct device_node *np, const char *name) { -- 1.9.1

2 2

[alsa-devel] [PATCH v2 2/2] ASoC: topology: Only free TLV for volume mixers of a widget
by mengdong.lin＠linux.intel.com 28 Nov '16

28 Nov '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Replace flag 'kcontrol_enum' with a type value of embedded controls for a widget. So topology can check the type of its embedded controls when removing the widget, and only free the TLV of volume mixers. Bytes controls don't have TLV. And when removing a widget created by topology, just free the private value which is used as struct soc_mixer_control for volume mixers or soc_bytes_ext for bytes controls. No need to cast the private data to control-specific struct pointers before freeing it. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> --- v2: fix lkp warning: 'kcontrol_type' may be used uninitialized in this function [-Wmaybe-uninitialized] widget->dobj.widget.kcontrol_type = kcontrol_type; --- diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index d318fe4..86c884a 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -53,7 +53,7 @@ struct snd_soc_dobj_control { /* dynamic widget object */ struct snd_soc_dobj_widget { - unsigned int kcontrol_enum:1; /* this widget is an enum kcontrol */ + unsigned int kcontrol_type; /* kcontrol type: mixer, enum, bytes */ }; /* generic dynamic object - all dynamic objects belong to this struct */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 63e1a50..6f7c7a4 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -489,7 +489,7 @@ static void remove_widget(struct snd_soc_component *comp, * Dynamic Widgets either have 1..N enum kcontrols or mixers. * The enum may either have an array of values or strings. */ - if (dobj->widget.kcontrol_enum) { + if (dobj->widget.kcontrol_type == SND_SOC_TPLG_TYPE_ENUM) { /* enumerated widget mixer */ for (i = 0; i < w->num_kcontrols; i++) { struct snd_kcontrol *kcontrol = w->kcontrols[i]; @@ -506,16 +506,21 @@ static void remove_widget(struct snd_soc_component *comp, } kfree(w->kcontrol_news); } else { - /* non enumerated widget mixer */ + /* volume mixer or bytes controls */ for (i = 0; i < w->num_kcontrols; i++) { struct snd_kcontrol *kcontrol = w->kcontrols[i]; - struct soc_mixer_control *sm = - (struct soc_mixer_control *) kcontrol->private_value; - kfree(w->kcontrols[i]->tlv.p); + if (dobj->widget.kcontrol_type + == SND_SOC_TPLG_TYPE_MIXER) + kfree(kcontrol->tlv.p); - snd_ctl_remove(card, w->kcontrols[i]); - kfree(sm); + snd_ctl_remove(card, kcontrol); + + /* Private value is used as struct soc_mixer_control + * for volume mixers or soc_bytes_ext for bytes + * controls. + */ + kfree((void *)kcontrol->private_value); } kfree(w->kcontrol_news); } @@ -1439,6 +1444,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, struct snd_soc_dapm_widget template, *widget; struct snd_soc_tplg_ctl_hdr *control_hdr; struct snd_soc_card *card = tplg->comp->card; + unsigned int kcontrol_type = SND_SOC_TPLG_TYPE_MIXER; int ret = 0; if (strnlen(w->name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == @@ -1494,6 +1500,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, case SND_SOC_TPLG_CTL_VOLSW_XR_SX: case SND_SOC_TPLG_CTL_RANGE: case SND_SOC_TPLG_DAPM_CTL_VOLSW: + kcontrol_type = SND_SOC_TPLG_TYPE_MIXER; /* volume mixer */ template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = soc_tplg_dapm_widget_dmixer_create(tplg, @@ -1508,7 +1515,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - template.dobj.widget.kcontrol_enum = 1; + kcontrol_type = SND_SOC_TPLG_TYPE_ENUM; /* enumerated mixer */ template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = soc_tplg_dapm_widget_denum_create(tplg, @@ -1519,6 +1526,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, } break; case SND_SOC_TPLG_CTL_BYTES: + kcontrol_type = SND_SOC_TPLG_TYPE_BYTES; /* bytes control */ template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = soc_tplg_dapm_widget_dbytes_create(tplg, @@ -1555,6 +1563,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, } widget->dobj.type = SND_SOC_DOBJ_WIDGET; + widget->dobj.widget.kcontrol_type = kcontrol_type; widget->dobj.ops = tplg->ops; widget->dobj.index = tplg->index; kfree(template.sname); -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: nau8825: erase pop noise by soft mute
by John Hsu 28 Nov '16

28 Nov '16

The patch is a solution to fix the following issue. Chrome OS Issue 54078: Chell_headphone pop back from S3 When the system resume from S3 and playing music, there is a pop noise happened. It looks like system changes the sound immediately from maximum volume to the volume configured by user in playback beginning. The sound signal changes sharply, and there is a DC offset in the signal to make pop noise. Therefore, the driver introduces the soft mute to avoid the side effect of the unstable signal. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 52 +++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/nau8825.h | 3 +++ 2 files changed, 54 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index e643be9..3bcba14 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -754,8 +754,9 @@ static void nau8825_xtalk_measure(struct nau8825 *nau8825) nau8825->imp_rms[NAU8825_XTALK_HPR_R2L], nau8825->imp_rms[NAU8825_XTALK_HPL_R2L]); dev_dbg(nau8825->dev, "cross talk sidetone: %x\n", sidetone); + nau8825->sidetone = (sidetone << 8) | sidetone; regmap_write(nau8825->regmap, NAU8825_REG_DAC_DGAIN_CTRL, - (sidetone << 8) | sidetone); + nau8825->sidetone); nau8825_xtalk_clean(nau8825); nau8825->xtalk_state = NAU8825_XTALK_DONE; break; @@ -1334,6 +1335,50 @@ int nau8825_enable_jack_detect(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); +/** + * nau8825_soft_mute - provide soft mute function + * + * @component: codec component + * @mute: 1 for mute; 0 for unmute + * + * Enable soft mute to gradually lower DAC volume to zero; + * Soft unmute will gradually increase DAC volume to volume setting. + * We found that signal at beginning is not stable in some application. + * That signal changes sharply will make pop noise; but the steps of soft + * mute is not enough in codec to cover the change period. Thus, the driver + * deffers 20ms to do soft unmute for the pop noise issue. + */ +int nau8825_soft_mute(struct snd_soc_codec *codec, int mute) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + if (mute) { + cancel_delayed_work(&nau8825->softmute_work); + /* Disable sidetone to avoid it affect the mute function */ + regmap_write(nau8825->regmap, NAU8825_REG_DAC_DGAIN_CTRL, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_MUTE_CTRL, + NAU8825_DAC_SOFT_MUTE, NAU8825_DAC_SOFT_MUTE); + } else { + /* Defer 20ms to do soft unmute to skip the unstable signal */ + schedule_delayed_work(&nau8825->softmute_work, + msecs_to_jiffies(20)); + } + + return 0; +} +EXPORT_SYMBOL_GPL(nau8825_soft_mute); + +static void nau8825_unmute_deferred(struct work_struct *work) +{ + struct nau8825 *nau8825 = container_of(work, + struct nau8825, softmute_work.work); + + /* Restore sidetone */ + regmap_write(nau8825->regmap, NAU8825_REG_DAC_DGAIN_CTRL, + nau8825->sidetone); + regmap_update_bits(nau8825->regmap, NAU8825_REG_MUTE_CTRL, + NAU8825_DAC_SOFT_MUTE, 0); +} static bool nau8825_is_jack_inserted(struct regmap *regmap) { @@ -2150,6 +2195,7 @@ static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, static int nau8825_resume_setup(struct nau8825 *nau8825) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(nau8825->dapm); struct regmap *regmap = nau8825->regmap; /* Close clock when jack type detection at manual mode */ @@ -2169,6 +2215,8 @@ static int nau8825_resume_setup(struct nau8825 *nau8825) NAU8825_JACK_DET_DB_BYPASS, NAU8825_JACK_DET_DB_BYPASS); regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, NAU8825_IRQ_INSERT_DIS | NAU8825_IRQ_EJECT_DIS, 0); + /* Enable codec soft mute */ + nau8825_soft_mute(codec, 1); return 0; } @@ -2393,8 +2441,10 @@ static int nau8825_i2c_probe(struct i2c_client *i2c, */ nau8825->xtalk_state = NAU8825_XTALK_DONE; nau8825->xtalk_protect = false; + nau8825->sidetone = 0; sema_init(&nau8825->xtalk_sem, 1); INIT_WORK(&nau8825->xtalk_work, nau8825_xtalk_work); + INIT_DELAYED_WORK(&nau8825->softmute_work, nau8825_unmute_deferred); nau8825_print_device_properties(nau8825); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 1c63e2a..e15ea5c 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -433,6 +433,7 @@ struct nau8825 { struct snd_soc_dapm_context *dapm; struct snd_soc_jack *jack; struct clk *mclk; + struct delayed_work softmute_work; struct work_struct xtalk_work; struct semaphore xtalk_sem; int irq; @@ -459,10 +460,12 @@ struct nau8825 { int xtalk_event_mask; bool xtalk_protect; int imp_rms[NAU8825_XTALK_IMM]; + int sidetone; }; int nau8825_enable_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); +int nau8825_soft_mute(struct snd_soc_codec *codec, int mute); #endif /* __NAU8825_H__ */ -- 2.6.4

2 3

[alsa-devel] [PATCH v2 1/2] ASoC: topology: Allow a widget to have multiple enum controls
by mengdong.lin＠linux.intel.com 28 Nov '16

28 Nov '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This patch can create multiple enumerated mixer controls for a widget. Previously topology kernel driver assumes a widget can have only one emumerated mixer control. We need to remove this restriction for Broxton. Its firmware modules (widgets) may need multiple enum controls based on the channel and MIC combination. No ABI change is needed. The ABI allows a widget to embed multiple controls. Reported-by: G Kranthi <gudishax.kranthikumar(a)intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4dfdc65..63e1a50 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -486,21 +486,24 @@ static void remove_widget(struct snd_soc_component *comp, dobj->ops->widget_unload(comp, dobj); /* - * Dynamic Widgets either have 1 enum kcontrol or 1..N mixers. + * Dynamic Widgets either have 1..N enum kcontrols or mixers. * The enum may either have an array of values or strings. */ if (dobj->widget.kcontrol_enum) { /* enumerated widget mixer */ - struct soc_enum *se = - (struct soc_enum *)w->kcontrols[0]->private_value; + for (i = 0; i < w->num_kcontrols; i++) { + struct snd_kcontrol *kcontrol = w->kcontrols[i]; + struct soc_enum *se = + (struct soc_enum *)kcontrol->private_value; - snd_ctl_remove(card, w->kcontrols[0]); + snd_ctl_remove(card, kcontrol); - kfree(se->dobj.control.dvalues); - for (i = 0; i < se->items; i++) - kfree(se->dobj.control.dtexts[i]); + kfree(se->dobj.control.dvalues); + for (i = 0; i < se->items; i++) + kfree(se->dobj.control.dtexts[i]); - kfree(se); + kfree(se); + } kfree(w->kcontrol_news); } else { /* non enumerated widget mixer */ @@ -1256,98 +1259,105 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( } static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( - struct soc_tplg *tplg) + struct soc_tplg *tplg, int num_kcontrols) { struct snd_kcontrol_new *kc; struct snd_soc_tplg_enum_control *ec; struct soc_enum *se; - int i, err; - - ec = (struct snd_soc_tplg_enum_control *)tplg->pos; - tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + - ec->priv.size); - - /* validate kcontrol */ - if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == - SNDRV_CTL_ELEM_ID_NAME_MAXLEN) - return NULL; + int i, j, err; - kc = kzalloc(sizeof(*kc), GFP_KERNEL); + kc = kcalloc(num_kcontrols, sizeof(*kc), GFP_KERNEL); if (kc == NULL) return NULL; - se = kzalloc(sizeof(*se), GFP_KERNEL); - if (se == NULL) - goto err; + for (i = 0; i < num_kcontrols; i++) { + ec = (struct snd_soc_tplg_enum_control *)tplg->pos; + /* validate kcontrol */ + if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return NULL; + + se = kzalloc(sizeof(*se), GFP_KERNEL); + if (se == NULL) + goto err; - dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", - ec->hdr.name); + dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", + ec->hdr.name); - kc->name = ec->hdr.name; - kc->private_value = (long)se; - kc->iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc->access = ec->hdr.access; + kc[i].name = ec->hdr.name; + kc[i].private_value = (long)se; + kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc[i].access = ec->hdr.access; - /* we only support FL/FR channel mapping atm */ - se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); - se->shift_l = tplc_chan_get_shift(tplg, ec->channel, SNDRV_CHMAP_FL); - se->shift_r = tplc_chan_get_shift(tplg, ec->channel, SNDRV_CHMAP_FR); + /* we only support FL/FR channel mapping atm */ + se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); + se->shift_l = tplc_chan_get_shift(tplg, ec->channel, + SNDRV_CHMAP_FL); + se->shift_r = tplc_chan_get_shift(tplg, ec->channel, + SNDRV_CHMAP_FR); - se->items = ec->items; - se->mask = ec->mask; - se->dobj.index = tplg->index; + se->items = ec->items; + se->mask = ec->mask; + se->dobj.index = tplg->index; - switch (ec->hdr.ops.info) { - case SND_SOC_TPLG_CTL_ENUM_VALUE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - err = soc_tplg_denum_create_values(se, ec); - if (err < 0) { - dev_err(tplg->dev, "ASoC: could not create values for %s\n", - ec->hdr.name); + switch (ec->hdr.ops.info) { + case SND_SOC_TPLG_CTL_ENUM_VALUE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: + err = soc_tplg_denum_create_values(se, ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: could not create values for %s\n", + ec->hdr.name); + goto err_se; + } + /* fall through to create texts */ + case SND_SOC_TPLG_CTL_ENUM: + case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: + err = soc_tplg_denum_create_texts(se, ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: could not create texts for %s\n", + ec->hdr.name); + goto err_se; + } + break; + default: + dev_err(tplg->dev, "ASoC: invalid enum control type %d for %s\n", + ec->hdr.ops.info, ec->hdr.name); goto err_se; } - /* fall through to create texts */ - case SND_SOC_TPLG_CTL_ENUM: - case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: - err = soc_tplg_denum_create_texts(se, ec); + + /* map io handlers */ + err = soc_tplg_kcontrol_bind_io(&ec->hdr, &kc[i], tplg); + if (err) { + soc_control_err(tplg, &ec->hdr, ec->hdr.name); + goto err_se; + } + + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, &kc[i], + (struct snd_soc_tplg_ctl_hdr *)ec); if (err < 0) { - dev_err(tplg->dev, "ASoC: could not create texts for %s\n", + dev_err(tplg->dev, "ASoC: failed to init %s\n", ec->hdr.name); goto err_se; } - break; - default: - dev_err(tplg->dev, "ASoC: invalid enum control type %d for %s\n", - ec->hdr.ops.info, ec->hdr.name); - goto err_se; - } - /* map io handlers */ - err = soc_tplg_kcontrol_bind_io(&ec->hdr, kc, tplg); - if (err) { - soc_control_err(tplg, &ec->hdr, ec->hdr.name); - goto err_se; - } - - /* pass control to driver for optional further init */ - err = soc_tplg_init_kcontrol(tplg, kc, - (struct snd_soc_tplg_ctl_hdr *)ec); - if (err < 0) { - dev_err(tplg->dev, "ASoC: failed to init %s\n", - ec->hdr.name); - goto err_se; + tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + + ec->priv.size); } return kc; err_se: - /* free values and texts */ - kfree(se->dobj.control.dvalues); - for (i = 0; i < ec->items; i++) - kfree(se->dobj.control.dtexts[i]); + for (; i >= 0; i--) { + /* free values and texts */ + se = (struct soc_enum *)kc[i].private_value; + kfree(se->dobj.control.dvalues); + for (j = 0; j < ec->items; j++) + kfree(se->dobj.control.dtexts[j]); - kfree(se); + kfree(se); + } err: kfree(kc); @@ -1499,9 +1509,10 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: template.dobj.widget.kcontrol_enum = 1; - template.num_kcontrols = 1; + template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = - soc_tplg_dapm_widget_denum_create(tplg); + soc_tplg_dapm_widget_denum_create(tplg, + template.num_kcontrols); if (!template.kcontrol_news) { ret = -ENOMEM; goto hdr_err; -- 2.5.0

1 0

[alsa-devel] ASoC: Intel: Enabling LPE audio over I2S on Intel Atom E3815
by Shrirang Bagul 27 Nov '16

27 Nov '16

Hi, I'm attempting to enable audio on Intel Atom E3815. I have a board with RT5660 codec from Realtek connected to the SoC over I2S2. The codec can be configured over I2C bus. The board supports External Mic In and Line Out connections for audio. I am running Ubuntu 16.04 LTS with kernel v4.4. To support RT5660 codec, I have backported patches - d01580c ASoC: rt5660: enable MCLK detection 2b26dd4 ASoC: rt5660: add rt5660 codec driver from https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rt5660 branch To support I2S_CLK using MCLK I've also backported the patch to support PMC platform clocks - [13/17] clk: x86: Add Atom PMC platform clocks (https://patchwork.kernel.org/pat ch/9277989/) To understand the I2S over LPE architecture for audio I'm referring to the Intel Atom Processor E3800: Datasheet (http://www.intel.com/content/dam/www/public/us/ en/documents/datasheets/atom-e3800-family-datasheet.pdf) and discussions on alsa-devel mailing list thread: http://mailman.alsa-project.org/pipermail/alsa-d evel/2015-November/100303.html After implementing the machine driver byt-rt5660.c (by referring to byt- rt5640.c), the CPU DAI baytrail-pcm-audio is registered successfully and the fw_sst_0f28.bin-48kHz_i2s_master firmware also is loaded without any errors. It is possible to configure the codec controls using alsamixer. However, an attempt to record results in an empty PCM file and no audio is heard if I try to playback using aplay. I'm clueless on how to proceed from this point onward in getting the audio to work on this board. The ouput of alsa-info.sh, clk_summary & gpio config from debugfs and dmesg log is attached with this message. I have published the source code here: http://kernel.ubuntu.com/git/snb/ubuntu-xenial.git/log/?h=x-master-au dio-test I would highly appreciate your help! -Shrirang

3 2

[alsa-devel] problem with the driver for ALC3260 sound chip
by marve 26 Nov '16

26 Nov '16

Dear all, Here a rephrase of my issue with TOSLINK SPDIF output of my Dell XPS One 2710 system running Linux. First: I have tried other hardware with my DTS receiver and the receiver worked fine. Also I have checked the HDMI output of my computer, also working! Second: The SPDIF output worked fine on a Windows 7 installation with PCM and DTS. I stopped using Windows privately. Most success I got with debugging the SPDIF TOSLINK is: - With the HDA Intel sound cards Jack Retasking tool: - Check Showing unconnected pins. Actually, the TOSLINK is connected but shows "not connected" in the tool. - Manually override Pin ID 0x1e - Select "SPDIF Out" and "Apply Now" - Then with PulseAudio Volume Control: - In the Configuration Tab, select "Profile: Digital Stereo (IEC958) Output" - In the Output Devices "Digital Output (S/PDIF)" appears and is the only option and in the advanced tab PCM, DTS, AC3, MPEG, EAC3, AAC are selected - I check if in alsamixer tool that the SPDIF channel is unmuted When I play a sound, the vumeter of PulseAudio Volume Control does react, see png attachment (https://postimg.org/image/5tu4rax9t/), but my external DTS receiver doesn't get input! I really got stuck here! Best, Marijn

2 3

[alsa-devel] [PATCH] ASoC: intel: Fix crash at suspend/resume without card registration
by Takashi Iwai 26 Nov '16

26 Nov '16

When ASoC Intel SST Medfield driver is probed but without codec / card assigned, it causes an Oops and freezes the kernel at suspend/resume, PM: Suspending system (freeze) Suspending console(s) (use no_console_suspend to debug) BUG: unable to handle kernel NULL pointer dereference at 0000000000000018 IP: [<ffffffffc09d9409>] sst_soc_prepare+0x19/0xa0 [snd_soc_sst_mfld_platform] Oops: 0000 [#1] PREEMPT SMP CPU: 0 PID: 1552 Comm: systemd-sleep Tainted: G W 4.9.0-rc6-1.g5f5c2ad-default #1 Call Trace: [<ffffffffb45318f9>] dpm_prepare+0x209/0x460 [<ffffffffb4531b61>] dpm_suspend_start+0x11/0x60 [<ffffffffb40d3cc2>] suspend_devices_and_enter+0xb2/0x710 [<ffffffffb40d462e>] pm_suspend+0x30e/0x390 [<ffffffffb40d2eba>] state_store+0x8a/0x90 [<ffffffffb43c670f>] kobj_attr_store+0xf/0x20 [<ffffffffb42b0d97>] sysfs_kf_write+0x37/0x40 [<ffffffffb42b02bc>] kernfs_fop_write+0x11c/0x1b0 [<ffffffffb422be68>] __vfs_write+0x28/0x140 [<ffffffffb43728a8>] ? apparmor_file_permission+0x18/0x20 [<ffffffffb433b2ab>] ? security_file_permission+0x3b/0xc0 [<ffffffffb422d095>] vfs_write+0xb5/0x1a0 [<ffffffffb422e3d6>] SyS_write+0x46/0xa0 [<ffffffffb4719fbb>] entry_SYSCALL_64_fastpath+0x1e/0xad Add proper NULL checks in the PM code of mdfld driver. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 25c6d87c818e..76bcb947b31c 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -771,6 +771,9 @@ static int sst_soc_prepare(struct device *dev) struct sst_data *drv = dev_get_drvdata(dev); struct snd_soc_pcm_runtime *rtd; + if (!drv->soc_card) + return 0; + /* suspend all pcms first */ snd_soc_suspend(drv->soc_card->dev); snd_soc_poweroff(drv->soc_card->dev); @@ -793,6 +796,9 @@ static void sst_soc_complete(struct device *dev) struct sst_data *drv = dev_get_drvdata(dev); struct snd_soc_pcm_runtime *rtd; + if (!drv->soc_card) + return 0; + /* restart SSPs */ list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) { struct snd_soc_dai *dai = rtd->cpu_dai; -- 2.10.2

2 2

[alsa-devel] [PATCH] ALSA: hda/ca0132 - Add quirk for Alienware 15 R2 2016
by Takashi Iwai 25 Nov '16

25 Nov '16

From: Sven Hahne <hahne(a)zeitkunst.eu> I'm using an Alienware 15 R2 and had to use the alienware quirks to get my headphone output working. I fixed it by adding, SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE) to the patch. Signed-off-by: Sven Hahne <hahne(a)zeitkunst.eu> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index ad06866d7c69..11b9b2f17a2e 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -780,6 +780,7 @@ static const struct hda_pintbl alienware_pincfgs[] = { static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE), + SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), {} }; -- 2.10.2

1 0

[alsa-devel] [patch] ASoC: sunxi: Uninitialized variable in probe()
by Dan Carpenter 25 Nov '16

25 Nov '16

Oddly enough, my version of GCC misses this uninitialized variable. Fixes: ba2ff3027b5a ("ASoC: sunxi: Add support for A23/A33/H3 codec's analog path controls") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c index 222bbd4..af02290 100644 --- a/sound/soc/sunxi/sun8i-codec-analog.c +++ b/sound/soc/sunxi/sun8i-codec-analog.c @@ -589,7 +589,7 @@ static int sun8i_codec_analog_cmpnt_probe(struct snd_soc_component *cmpnt) } if (quirks->has_hmic) { - sun8i_codec_add_hmic(cmpnt); + ret = sun8i_codec_add_hmic(cmpnt); if (ret) return ret; }

4 3

[alsa-devel] [PATCH 1/2] ASoC: Intel: Skylake: Don't use dma I2S config structure in kernel
by jeeja.kp＠intel.com 25 Nov '16

25 Nov '16

From: GuruprasadX Pawse <guruprasadx.pawse(a)intel.com> I2S blob config structure can change between DSP copier module versions. We are not modifying these structures anyway and NHLT blobs specify them properly, so use the blob directly to send the dma_control ipc instead of using the predefined I2S blob config structure. Signed-off-by: GuruprasadX Pawse <guruprasadx.pawse(a)intel.com> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> --- sound/soc/intel/skylake/skl-messages.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 4ae021a..e79cbcf 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -527,16 +527,14 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx, int skl_dsp_set_dma_control(struct skl_sst *ctx, struct skl_module_cfg *mconfig) { struct skl_dma_control *dma_ctrl; - struct skl_i2s_config_blob config_blob; struct skl_ipc_large_config_msg msg = {0}; int err = 0; /* - * if blob size is same as capablity size, then no dma control - * present so return + * if blob size zero, then return */ - if (mconfig->formats_config.caps_size == sizeof(config_blob)) + if (mconfig->formats_config.caps_size == 0) return 0; msg.large_param_id = DMA_CONTROL_ID; @@ -550,7 +548,7 @@ int skl_dsp_set_dma_control(struct skl_sst *ctx, struct skl_module_cfg *mconfig) dma_ctrl->node_id = skl_get_node_id(ctx, mconfig); /* size in dwords */ - dma_ctrl->config_length = sizeof(config_blob) / 4; + dma_ctrl->config_length = mconfig->formats_config.caps_size / 4; memcpy(dma_ctrl->config_data, mconfig->formats_config.caps, mconfig->formats_config.caps_size); @@ -558,7 +556,6 @@ int skl_dsp_set_dma_control(struct skl_sst *ctx, struct skl_module_cfg *mconfig) err = skl_ipc_set_large_config(&ctx->ipc, &msg, (u32 *)dma_ctrl); kfree(dma_ctrl); - return err; } -- 2.5.0

3 4

[alsa-devel] [PATCH v4 0/3] Media Device Allocator API
by Shuah Khan 25 Nov '16

25 Nov '16

Media Device Allocator API to allows multiple drivers share a media device. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. Patches 0001 and 0002 are rebased to 4.9-rc4. Patch 0003 for snd-usb-audio is a rebase of the patch that was tested with the original Media Device Allocator patch series. snd-usb-audio patch includes the fixes found during 4.7-rc1 time in the original snd-usb-audio patch. Changes to patch 0001 since v3: - Fixed undefined reference to `__media_device_usb_init compile error when CONFIG_USB is disabled. - Fixed kernel paging error when accessing /dev/mediaX after rmmod of the module that owns the media_device. The fix bumps the reference count for the owner when second driver comes along to share the media_device. If au0828 owns the media_device, then snd_usb_audio will bump the refcount for au0828, so it won't get deleted and vice versa. Changes to patch 0002 since v2: - Updated media_device_delete() to pass in module name. Changes to patch 0003 since the last version in 4.7-rc1: - Included fixes to bugs found during testing. - Updated to use the Media Allocator API. This patch series has been tested with au0828 and snd-usb-audio drivers. Ran bind and unbind loop tests on each driver with mc_nextgen_test and media_device_test app loop tests while checking lsmod and dmesg. Please refer to tools/testing/selftests/media_tests/regression_test.txt for testing done on this series. Shuah Khan (3): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API sound/usb: Use Media Controller API to share media resources drivers/media/Makefile | 3 +- drivers/media/media-dev-allocator.c | 146 +++++++++++++++ drivers/media/usb/au0828/au0828-core.c | 12 +- drivers/media/usb/au0828/au0828.h | 1 + include/media/media-dev-allocator.h | 87 +++++++++ sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 314 +++++++++++++++++++++++++++++++++ sound/usb/media.h | 73 ++++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 28 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 16 files changed, 685 insertions(+), 14 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.7.4

3 11

[alsa-devel] [PATCH] topology: Update physical link configurations in Broadwell text conf file
by mengdong.lin＠linux.intel.com 25 Nov '16

25 Nov '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> To make this conf file a better example, add configurations for the physical link "Codec", same as that defined by Intel Broadwell upstream machine driver. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/src/conf/topology/broadwell/broadwell.conf b/src/conf/topology/broadwell/broadwell.conf index eb89377..b8405d9 100644 --- a/src/conf/topology/broadwell/broadwell.conf +++ b/src/conf/topology/broadwell/broadwell.conf @@ -389,3 +389,22 @@ SectionGraph."dsp" { "Analog Capture, , SSP0 CODEC IN" ] } + +SectionHWConfig."CodecHWConfig" { + id "1" + format "I2S" # physical audio format. + bclk "master" # Platform is master of bit clock + fsync "master" # platform is master of fsync +} + +SectionLink."Codec" { + + # used for binding to the physical link + id "0" + + hw_configs [ + "CodecHWConfig" + ] + + default_hw_conf_id "1" +} -- 2.5.0

2 1

[alsa-devel] [PATCH v5 1/3] media: Media Device Allocator API
by Shuah Khan 25 Nov '16

25 Nov '16

Media Device Allocator API to allows multiple drivers share a media device. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> Reviewed-by: Hans Verkuil <hans.verkuil(a)cisco.com> --- Changes to patch 0001 since v4: - Addressed Sakari's review comments with the exception of opting to not introduce media_device_usb_allocate() macro, and to not add a new routine to find media device instance to avoid a one line check. Documentation/media/kapi/mc-core.rst | 2 + drivers/media/Makefile | 3 +- drivers/media/media-dev-allocator.c | 135 +++++++++++++++++++++++++++++++++++ include/media/media-dev-allocator.h | 88 +++++++++++++++++++++++ 4 files changed, 227 insertions(+), 1 deletion(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h diff --git a/Documentation/media/kapi/mc-core.rst b/Documentation/media/kapi/mc-core.rst index 1a738e5..0b9090d 100644 --- a/Documentation/media/kapi/mc-core.rst +++ b/Documentation/media/kapi/mc-core.rst @@ -262,3 +262,5 @@ in the end provide a way to use driver-specific callbacks. .. kernel-doc:: include/media/media-devnode.h .. kernel-doc:: include/media/media-entity.h + +.. kernel-doc:: include/media/media-dev-allocator.h diff --git a/drivers/media/Makefile b/drivers/media/Makefile index 0deaa93..7c0701d 100644 --- a/drivers/media/Makefile +++ b/drivers/media/Makefile @@ -6,7 +6,8 @@ ifeq ($(CONFIG_MEDIA_CEC_EDID),y) obj-$(CONFIG_MEDIA_SUPPORT) += cec-edid.o endif -media-objs := media-device.o media-devnode.o media-entity.o +media-objs := media-device.o media-devnode.o media-entity.o \ + media-dev-allocator.o # # I2C drivers should come before other drivers, otherwise they'll fail diff --git a/drivers/media/media-dev-allocator.c b/drivers/media/media-dev-allocator.c new file mode 100644 index 0000000..581e50e --- /dev/null +++ b/drivers/media/media-dev-allocator.c @@ -0,0 +1,135 @@ +/* + * media-dev-allocator.c - Media Controller Device Allocator API + * + * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> + * Copyright (c) 2016 Samsung Electronics Co., Ltd. + * + * This file is released under the GPLv2. + * Credits: Suggested by Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> + */ + +/* + * This file adds a global refcounted Media Controller Device Instance API. + * A system wide global media device list is managed and each media device + * includes a kref count. The last put on the media device releases the media + * device instance. + * + */ + +#include <linux/kref.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/usb.h> + +#include <media/media-device.h> + +static LIST_HEAD(media_device_list); +static DEFINE_MUTEX(media_device_lock); + +struct media_device_instance { + struct media_device mdev; + struct module *owner; + struct list_head list; + struct device *dev; + struct kref refcount; +}; + +static inline struct media_device_instance * +to_media_device_instance(struct media_device *mdev) +{ + return container_of(mdev, struct media_device_instance, mdev); +} + +static void media_device_instance_release(struct kref *kref) +{ + struct media_device_instance *mdi = + container_of(kref, struct media_device_instance, refcount); + + dev_dbg(mdi->mdev.dev, "%s: mdev=%p\n", __func__, &mdi->mdev); + + mutex_lock(&media_device_lock); + + media_device_unregister(&mdi->mdev); + media_device_cleanup(&mdi->mdev); + + list_del(&mdi->list); + mutex_unlock(&media_device_lock); + + kfree(mdi); +} + +/* Callers should hold media_device_lock when calling this function */ +static struct media_device *__media_device_get(struct device *dev, + char *module_name) +{ + struct media_device_instance *mdi; + + list_for_each_entry(mdi, &media_device_list, list) { + if (mdi->dev != dev) + continue; + + kref_get(&mdi->refcount); + /* get module reference for the media_device owner */ + if (find_module(module_name) != mdi->owner && + !try_module_get(mdi->owner)) + dev_err(dev, "%s: try_module_get() error\n", __func__); + dev_dbg(dev, "%s: get mdev=%p module_name %s\n", + __func__, &mdi->mdev, module_name); + return &mdi->mdev; + } + + mdi = kzalloc(sizeof(*mdi), GFP_KERNEL); + if (!mdi) + return NULL; + + mdi->dev = dev; + mdi->owner = find_module(module_name); + kref_init(&mdi->refcount); + list_add_tail(&mdi->list, &media_device_list); + + dev_dbg(dev, "%s: alloc mdev=%p module_name %s\n", __func__, + &mdi->mdev, module_name); + return &mdi->mdev; +} + +#if IS_ENABLED(CONFIG_USB) +struct media_device *media_device_usb_allocate(struct usb_device *udev, + char *module_name) +{ + struct media_device *mdev; + + mutex_lock(&media_device_lock); + mdev = __media_device_get(&udev->dev, module_name); + if (!mdev) { + mutex_unlock(&media_device_lock); + return ERR_PTR(-ENOMEM); + } + + /* check if media device is already initialized */ + if (!mdev->dev) + __media_device_usb_init(mdev, udev, udev->product, + module_name); + mutex_unlock(&media_device_lock); + return mdev; +} +EXPORT_SYMBOL_GPL(media_device_usb_allocate); +#endif + +void media_device_delete(struct media_device *mdev, char *module_name) +{ + struct media_device_instance *mdi = to_media_device_instance(mdev); + + dev_dbg(mdi->mdev.dev, "%s: mdev=%p module_name %s\n", + __func__, &mdi->mdev, module_name); + + mutex_lock(&media_device_lock); + /* put module reference if media_device owner is not THIS_MODULE */ + if (mdi->owner != find_module(module_name)) { + module_put(mdi->owner); + dev_dbg(mdi->mdev.dev, + "%s decremented owner module reference\n", __func__); + } + mutex_unlock(&media_device_lock); + kref_put(&mdi->refcount, media_device_instance_release); +} +EXPORT_SYMBOL_GPL(media_device_delete); diff --git a/include/media/media-dev-allocator.h b/include/media/media-dev-allocator.h new file mode 100644 index 0000000..fad8cb9 --- /dev/null +++ b/include/media/media-dev-allocator.h @@ -0,0 +1,88 @@ +/* + * media-dev-allocator.h - Media Controller Device Allocator API + * + * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> + * Copyright (c) 2016 Samsung Electronics Co., Ltd. + * + * This file is released under the GPLv2. + * Credits: Suggested by Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> + */ + +/* + * This file adds a global ref-counted Media Controller Device Instance API. + * A system wide global media device list is managed and each media device + * includes a kref count. The last put on the media device releases the media + * device instance. + */ + +#ifndef _MEDIA_DEV_ALLOCTOR_H +#define _MEDIA_DEV_ALLOCTOR_H + +struct usb_device; + +#ifdef CONFIG_MEDIA_CONTROLLER +/** + * Media Controller Device Allocator API + * + * When media device belongs to more than one driver, the shared media device + * is allocated with the shared struct device as the key for look ups. + * + * Shared media device should stay in registered state until the last driver + * unregisters it. In addition, media device should be released when all the + * references are released. Each driver gets a reference to the media device + * during probe, when it allocates the media device. If media device is already + * allocated, allocate API bumps up the refcount and return the existing media + * device. Driver puts the reference back from its disconnect routine when it + * calls media_device_delete(). + * + * Media device is unregistered and cleaned up from the kref put handler to + * ensure that the media device stays in registered state until the last driver + * unregisters the media device. + * + * Driver Usage: + * + * Drivers should use the media-core routines to get register reference and + * call media_device_delete() routine to make sure the shared media device + * delete is handled correctly. + * + * driver probe: + * Call media_device_usb_allocate() to allocate or get a reference + * Call media_device_register(), if media devnode isn't registered + * + * driver disconnect: + * Call media_device_delete() to free the media_device. Free'ing is + * handled by the put handler. + * + */ + +/** + * media_device_usb_allocate() - Allocate and return media device + * + * @udev: struct usb_device pointer + * @module_name: should be filled with KBUILD_MODNAME + * + * This interface should be called to allocate a media device when multiple + * drivers share usb_device and the media device. This interface allocates + * &media_device structure and calls media_device_usb_init() to initialize + * it. + * + */ +struct media_device *media_device_usb_allocate(struct usb_device *udev, + char *module_name); +/** + * media_device_delete() - Release media device. Calls kref_put(). + * + * @mdev: struct media_device pointer + * @module_name: should be filled with KBUILD_MODNAME + * + * This interface should be called to put Media Device Instance kref. + */ +void media_device_delete(struct media_device *mdev, char *driver_name); +#else +static inline struct media_device *media_device_usb_allocate( + struct usb_device *udev, char *driver_name) + { return NULL; } +static inline void media_device_delete( + struct media_device *mdev, char *module_name) { } +#endif /* CONFIG_MEDIA_CONTROLLER */ +#endif -- 2.7.4

2 1

[alsa-devel] [PATCH resend v3] ASoC: sun4i-codec: Add "Right Mixer" to "Line Out Mono Diff." route
by Chen-Yu Tsai 25 Nov '16

25 Nov '16

The mono differential output for "Line Out" downmixes the stereo audio from the mixer, instead of just taking the left channel. Add a route from the "Right Mixer" to "Line Out Source Playback Route" through the "Mono Differential" path, so DAPM doesn't shut down everything if the left channel is muted. Fixes: 0f909f98d7cb ("ASoC: sun4i-codec: Add support for A31 Line Out playback") Signed-off-by: Chen-Yu Tsai <wens(a)csie.org> --- This patch seems to have fallen through the cracks. --- sound/soc/sunxi/sun4i-codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 092fdcf6de95..b11b0ad4dfbe 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1047,6 +1047,7 @@ static const struct snd_soc_dapm_route sun6i_codec_codec_dapm_routes[] = { { "Line Out Source Playback Route", "Stereo", "Left Mixer" }, { "Line Out Source Playback Route", "Stereo", "Right Mixer" }, { "Line Out Source Playback Route", "Mono Differential", "Left Mixer" }, + { "Line Out Source Playback Route", "Mono Differential", "Right Mixer" }, { "LINEOUT", NULL, "Line Out Source Playback Route" }, /* ADC Routes */ -- 2.10.2

3 2

[alsa-devel] [PATCH v2] dbri: Fix compiler warning
by Tushar Dave 24 Nov '16

24 Nov '16

dbri uses 'u32' for dma handle while invoking kernel DMA APIs, instead of using dma_addr_t. This hasn't caused any 'incompatible pointer type' warning on SPARC because until now dma_addr_t is of type u32. However, recent changes in SPARC ATU (iommu) enabled 64bit DMA and therefore dma_addr_t became of type u64. This makes 'incompatible pointer type' warnings inevitable. e.g. sound/sparc/dbri.c: In function ‘snd_dbri_create’: sound/sparc/dbri.c:2538: warning: passing argument 3 of ‘dma_zalloc_coherent’ from incompatible pointer type ./include/linux/dma-mapping.h:608: note: expected ‘dma_addr_t *’ but argument is of type ‘u32 *’ For the record, dbri(sbus) driver never executes on sun4v. Therefore even though 64bit DMA is enabled on SPARC, dbri continues to use legacy iommu that guarantees DMA address is always in 32bit range. This patch resolves above compiler warning. Signed-off-by: Tushar Dave <tushar.n.dave(a)oracle.com> Reviewed-by: thomas tai <thomas.tai(a)oracle.com> --- v1->v2: - commit message updated. sound/sparc/dbri.c | 26 ++++++++++++++++---------- 1 file changed, 16 insertions(+), 10 deletions(-) diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 0190cb6..542c1b5 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -304,7 +304,7 @@ struct snd_dbri { spinlock_t lock; struct dbri_dma *dma; /* Pointer to our DMA block */ - u32 dma_dvma; /* DBRI visible DMA address */ + dma_addr_t dma_dvma; /* DBRI visible DMA address */ void __iomem *regs; /* dbri HW regs */ int dbri_irqp; /* intr queue pointer */ @@ -657,12 +657,14 @@ static void dbri_cmdwait(struct snd_dbri *dbri) */ static s32 *dbri_cmdlock(struct snd_dbri *dbri, int len) { + u32 dvma_addr = (u32)dbri->dma_dvma; + /* Space for 2 WAIT cmds (replaced later by 1 JUMP cmd) */ len += 2; spin_lock(&dbri->cmdlock); if (dbri->cmdptr - dbri->dma->cmd + len < DBRI_NO_CMDS - 2) return dbri->cmdptr + 2; - else if (len < sbus_readl(dbri->regs + REG8) - dbri->dma_dvma) + else if (len < sbus_readl(dbri->regs + REG8) - dvma_addr) return dbri->dma->cmd; else printk(KERN_ERR "DBRI: no space for commands."); @@ -680,6 +682,7 @@ static s32 *dbri_cmdlock(struct snd_dbri *dbri, int len) */ static void dbri_cmdsend(struct snd_dbri *dbri, s32 *cmd, int len) { + u32 dvma_addr = (u32)dbri->dma_dvma; s32 tmp, addr; static int wait_id = 0; @@ -689,7 +692,7 @@ static void dbri_cmdsend(struct snd_dbri *dbri, s32 *cmd, int len) *(cmd+1) = DBRI_CMD(D_WAIT, 1, wait_id); /* Replace the last command with JUMP */ - addr = dbri->dma_dvma + (cmd - len - dbri->dma->cmd) * sizeof(s32); + addr = dvma_addr + (cmd - len - dbri->dma->cmd) * sizeof(s32); *(dbri->cmdptr+1) = addr; *(dbri->cmdptr) = DBRI_CMD(D_JUMP, 0, 0); @@ -747,6 +750,7 @@ static void dbri_reset(struct snd_dbri *dbri) /* Lock must not be held before calling this */ static void dbri_initialize(struct snd_dbri *dbri) { + u32 dvma_addr = (u32)dbri->dma_dvma; s32 *cmd; u32 dma_addr; unsigned long flags; @@ -764,7 +768,7 @@ static void dbri_initialize(struct snd_dbri *dbri) /* * Initialize the interrupt ring buffer. */ - dma_addr = dbri->dma_dvma + dbri_dma_off(intr, 0); + dma_addr = dvma_addr + dbri_dma_off(intr, 0); dbri->dma->intr[0] = dma_addr; dbri->dbri_irqp = 1; /* @@ -778,7 +782,7 @@ static void dbri_initialize(struct snd_dbri *dbri) dbri->cmdptr = cmd; *(cmd++) = DBRI_CMD(D_WAIT, 1, 0); *(cmd++) = DBRI_CMD(D_WAIT, 1, 0); - dma_addr = dbri->dma_dvma + dbri_dma_off(cmd, 0); + dma_addr = dvma_addr + dbri_dma_off(cmd, 0); sbus_writel(dma_addr, dbri->regs + REG8); spin_unlock(&dbri->cmdlock); @@ -1077,6 +1081,7 @@ static void recv_fixed(struct snd_dbri *dbri, int pipe, volatile __u32 *ptr) static int setup_descs(struct snd_dbri *dbri, int streamno, unsigned int period) { struct dbri_streaminfo *info = &dbri->stream_info[streamno]; + u32 dvma_addr = (u32)dbri->dma_dvma; __u32 dvma_buffer; int desc; int len; @@ -1177,7 +1182,7 @@ static int setup_descs(struct snd_dbri *dbri, int streamno, unsigned int period) else { dbri->next_desc[last_desc] = desc; dbri->dma->desc[last_desc].nda = - dbri->dma_dvma + dbri_dma_off(desc, desc); + dvma_addr + dbri_dma_off(desc, desc); } last_desc = desc; @@ -1192,7 +1197,7 @@ static int setup_descs(struct snd_dbri *dbri, int streamno, unsigned int period) } dbri->dma->desc[last_desc].nda = - dbri->dma_dvma + dbri_dma_off(desc, first_desc); + dvma_addr + dbri_dma_off(desc, first_desc); dbri->next_desc[last_desc] = first_desc; dbri->pipes[info->pipe].first_desc = first_desc; dbri->pipes[info->pipe].desc = first_desc; @@ -1697,6 +1702,7 @@ calling dbri_process_interrupt_buffer(), which works even if the CPU's static void xmit_descs(struct snd_dbri *dbri) { struct dbri_streaminfo *info; + u32 dvma_addr = (u32)dbri->dma_dvma; s32 *cmd; unsigned long flags; int first_td; @@ -1718,7 +1724,7 @@ static void xmit_descs(struct snd_dbri *dbri) *(cmd++) = DBRI_CMD(D_SDP, 0, dbri->pipes[info->pipe].sdp | D_SDP_P | D_SDP_EVERY | D_SDP_C); - *(cmd++) = dbri->dma_dvma + + *(cmd++) = dvma_addr + dbri_dma_off(desc, first_td); dbri_cmdsend(dbri, cmd, 2); @@ -1740,7 +1746,7 @@ static void xmit_descs(struct snd_dbri *dbri) *(cmd++) = DBRI_CMD(D_SDP, 0, dbri->pipes[info->pipe].sdp | D_SDP_P | D_SDP_EVERY | D_SDP_C); - *(cmd++) = dbri->dma_dvma + + *(cmd++) = dvma_addr + dbri_dma_off(desc, first_td); dbri_cmdsend(dbri, cmd, 2); @@ -2539,7 +2545,7 @@ static int snd_dbri_create(struct snd_card *card, if (!dbri->dma) return -ENOMEM; - dprintk(D_GEN, "DMA Cmd Block 0x%p (0x%08x)\n", + dprintk(D_GEN, "DMA Cmd Block 0x%p (0x%08llx)\n", dbri->dma, dbri->dma_dvma); /* Map the registers into memory. */ -- 1.9.1

3 2

[alsa-devel] [PATCH 1/2] topology: Merge private data before embedded controls of a widget
by mengdong.lin＠linux.intel.com 24 Nov '16

24 Nov '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> For a widget ABI object, its private data sits before the embedded controls, so we should merge the private data blocks at first, and then the embedded controls. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/src/topology/dapm.c b/src/topology/dapm.c index 9fa0aac..e830751 100644 --- a/src/topology/dapm.c +++ b/src/topology/dapm.c @@ -160,7 +160,21 @@ static int tplg_build_widget(snd_tplg_t *tplg, base = &elem->ref_list; - /* for each ref in this control elem */ + /* A widget's private data sits before the embedded controls. + * So merge the private data blocks at first + */ + list_for_each(pos, base) { + ref = list_entry(pos, struct tplg_ref, list); + + if (ref->type != SND_TPLG_TYPE_DATA) + continue; + + err = tplg_copy_data(tplg, elem, ref); + if (err < 0) + return err; + } + + /* Merge the embedded controls */ list_for_each(pos, base) { ref = list_entry(pos, struct tplg_ref, list); @@ -190,12 +204,6 @@ static int tplg_build_widget(snd_tplg_t *tplg, err = copy_dapm_control(elem, ref->elem); break; - case SND_TPLG_TYPE_DATA: - err = tplg_copy_data(tplg, elem, ref); - if (err < 0) - return err; - break; - default: break; } -- 2.5.0

3 2

[alsa-devel] [patch] ALSA: emu10k1: shift wrapping bug in snd_emu10k1_ptr_read()
by Dan Carpenter 24 Nov '16

24 Nov '16

Static analysis says "size" is a number 0-63. So we really want to be doing the shift as a u64 and not a int type. Presumably "size" is never actually more than 31 otherwise the shift wrap would have been detected in testing. The "mask" is a u32 so we only care about the bottom 32 bits which also implies that "size" is less than 32. This code pre-dates git. I haven't tested this change, it's to fix a static analysis warning. I can't think that shift wrapping is the correct behavior so presumably this change is harmless but it definitely changes how the code works when size is larger than 32. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index 706b4f0..fd204f3 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -46,7 +46,7 @@ unsigned int snd_emu10k1_ptr_read(struct snd_emu10k1 * emu, unsigned int reg, un size = (reg >> 24) & 0x3f; offset = (reg >> 16) & 0x1f; - mask = ((1 << size) - 1) << offset; + mask = ((1ULL << size) - 1) << offset; spin_lock_irqsave(&emu->emu_lock, flags); outl(regptr, emu->port + PTR); @@ -81,7 +81,7 @@ void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned i size = (reg >> 24) & 0x3f; offset = (reg >> 16) & 0x1f; - mask = ((1 << size) - 1) << offset; + mask = ((1ULL << size) - 1) << offset; data = (data << offset) & mask; spin_lock_irqsave(&emu->emu_lock, flags);

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - ignore the assoc and seq when comparing pin configurations
by Hui Wang 24 Nov '16

24 Nov '16

More and more pin configurations have been adding to the pin quirk table, lots of them are only different from assoc and seq, but they all apply to the same QUIRK_FIXUP, if we don't compare assoc and seq when matching pin configurations, it will greatly reduce the pin quirk table size. We have tested this change on a couple of Dell laptops, it worked well. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- No hurry to cleanup the existing pin quirk table, if this patch run well for a period of time, then I will do the cleanup little by little. If this patch introduce some problem, we just need to revert this patch. sound/pci/hda/hda_auto_parser.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 7f57a14..4ad29f8 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -901,7 +901,7 @@ static bool pin_config_match(struct hda_codec *codec, for (; t_pins->nid; t_pins++) { if (t_pins->nid == nid) { found = 1; - if (t_pins->val == cfg) + if ((t_pins->val & 0xfffffff0) == (cfg & 0xfffffff0)) break; else if ((cfg & 0xf0000000) == 0x40000000 && (t_pins->val & 0xf0000000) == 0x40000000) break; -- 1.9.1

2 2

[alsa-devel] [RFC v2 0/3] ALSA controls management using index/device/sub-devices fields
by Arnaud Pouliquen 24 Nov '16

24 Nov '16

V2: Aim of this version is to continue discussion on DAI PCM control focused on ASoC drivers. In this V2 implementation in Soc-core is simplified to limit impact on existing code. Proposal is to add field in DAI driver struct to declare PCM controls that need to be linked to PCM character device on DAI link probing. Update of the RFC V1 based on discussions: - [RFC 4/4] iecset: allow to select control with device and sub-device numbers no more part of the RFC V2, will be discussed in a separate thread - [RFC 2/4] ALSA: control: increment index field for duplicated control. no more part of the RFC V2, no more need as RFC subject is PCM controls - [RFC V2 1/3] ASoC: core: allow DAI PCM controls bound to PCM device Patch reworked from V1 to simplify implementation - Binding is not done for Dai links tagged with no_pcm (DPCM). - no more possibility to add the controls after the DAI link probing. - [RFC V2 2/3] ASoC: sti: bind PCM controls to PCM device. - [RFC V2 3/3] ASoC: hdmi-codec: Example of PCM control bound to PCM device for multi Example of implementation in STI DAI driver and HDMI-codec drivers V1: http://www.spinics.net/lists/alsa-devel/msg56479.html 1) Alsa-utils patch - iecset: allow to select control with device and sub-device numbers This patch allows to access to 2 iec controls differentiated by device/sub-devices numbers => For me, this patch is mandatory to be able to address the ASoC IEC controls, in case of no fix is implemented to allows index field update in ASoC. 2) Alsa driver patches - ASoC: core: allow PCM control binding to PCM device Add relationship between DAIs PCM controls and PCM device. - ALSA: control: increment index field for duplicated control. Generic implementation of the patch proposed in HDA driver (http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=ea9b…) - ASoC: sti: use bind_pcm_ctl implementation of bind_pcm_ctl for sti driver. Arnaud Pouliquen (3): ASoC: core: allow DAI PCM controls bound to PCM device ASoC: sti: bind pcm controls to pcm device. ASoC: hdmi-codec: Example of PCM control bound to PCM device for multi HDMI DAIs. Regards, Arnaud include/sound/soc-dai.h | 4 ++++ sound/soc/codecs/hdmi-codec.c | 4 ++-- sound/soc/soc-core.c | 37 +++++++++++++++++++++++++++++++++++++ sound/soc/sti/sti_uniperif.c | 33 ++++----------------------------- 4 files changed, 47 insertions(+), 31 deletions(-) -- 1.9.1

3 6

[alsa-devel] [PATCH v4] drm: bridge: add DesignWare HDMI I2S audio support
by Kuninori Morimoto 24 Nov '16

24 Nov '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current dw-hdmi is supporting sound via AHB bus, but it has I2S audio feature too. This patch adds I2S audio support to dw-hdmi. This HDMI I2S is supported by using ALSA SoC common HDMI encoder driver. Tested-by: Jose Abreu <joabreu(a)synopsys.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v3 -> v4 - use IS_ERR() instead of IS_ERR_OR_NULL() on probe() drivers/gpu/drm/bridge/Kconfig | 8 ++ drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw-hdmi-audio.h | 7 ++ drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c | 142 +++++++++++++++++++++++++++++ drivers/gpu/drm/bridge/dw-hdmi.c | 22 ++++- drivers/gpu/drm/bridge/dw-hdmi.h | 21 +++++ 6 files changed, 199 insertions(+), 2 deletions(-) create mode 100644 drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c diff --git a/drivers/gpu/drm/bridge/Kconfig b/drivers/gpu/drm/bridge/Kconfig index 10e12e7..3129f8d 100644 --- a/drivers/gpu/drm/bridge/Kconfig +++ b/drivers/gpu/drm/bridge/Kconfig @@ -39,6 +39,14 @@ config DRM_DW_HDMI_AHB_AUDIO Designware HDMI block. This is used in conjunction with the i.MX6 HDMI driver. +config DRM_DW_HDMI_I2S_AUDIO + tristate "Synopsis Designware I2S Audio interface" + depends on DRM_DW_HDMI + select SND_SOC_HDMI_CODEC + help + Support the I2S Audio interface which is part of the Synopsis + Designware HDMI block. + config DRM_NXP_PTN3460 tristate "NXP PTN3460 DP/LVDS bridge" depends on OF diff --git a/drivers/gpu/drm/bridge/Makefile b/drivers/gpu/drm/bridge/Makefile index cdf3a3c..9a54f2a 100644 --- a/drivers/gpu/drm/bridge/Makefile +++ b/drivers/gpu/drm/bridge/Makefile @@ -4,6 +4,7 @@ obj-$(CONFIG_DRM_ANALOGIX_ANX78XX) += analogix-anx78xx.o obj-$(CONFIG_DRM_DUMB_VGA_DAC) += dumb-vga-dac.o obj-$(CONFIG_DRM_DW_HDMI) += dw-hdmi.o obj-$(CONFIG_DRM_DW_HDMI_AHB_AUDIO) += dw-hdmi-ahb-audio.o +obj-$(CONFIG_DRM_DW_HDMI_I2S_AUDIO) += dw-hdmi-i2s-audio.o obj-$(CONFIG_DRM_NXP_PTN3460) += nxp-ptn3460.o obj-$(CONFIG_DRM_PARADE_PS8622) += parade-ps8622.o obj-$(CONFIG_DRM_SII902X) += sii902x.o diff --git a/drivers/gpu/drm/bridge/dw-hdmi-audio.h b/drivers/gpu/drm/bridge/dw-hdmi-audio.h index 91f631b..fd1f745 100644 --- a/drivers/gpu/drm/bridge/dw-hdmi-audio.h +++ b/drivers/gpu/drm/bridge/dw-hdmi-audio.h @@ -11,4 +11,11 @@ struct dw_hdmi_audio_data { u8 *eld; }; +struct dw_hdmi_i2s_audio_data { + struct dw_hdmi *hdmi; + + void (*write)(struct dw_hdmi *hdmi, u8 val, int offset); + u8 (*read)(struct dw_hdmi *hdmi, int offset); +}; + #endif diff --git a/drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c new file mode 100644 index 0000000..79391c0 --- /dev/null +++ b/drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c @@ -0,0 +1,142 @@ +/* + * dw-hdmi-i2s-audio.c + * + * Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include <drm/bridge/dw_hdmi.h> + +#include <sound/hdmi-codec.h> + +#include "dw-hdmi.h" +#include "dw-hdmi-audio.h" + +#define DRIVER_NAME "dw-hdmi-i2s-audio" + +static inline void hdmi_write(struct dw_hdmi_i2s_audio_data *audio, + u8 val, int offset) +{ + struct dw_hdmi *hdmi = audio->hdmi; + + audio->write(hdmi, val, offset); +} + +static inline u8 hdmi_read(struct dw_hdmi_i2s_audio_data *audio, int offset) +{ + struct dw_hdmi *hdmi = audio->hdmi; + + return audio->read(hdmi, offset); +} + +static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, + struct hdmi_codec_daifmt *fmt, + struct hdmi_codec_params *hparms) +{ + struct dw_hdmi_i2s_audio_data *audio = data; + struct dw_hdmi *hdmi = audio->hdmi; + u8 conf0 = 0; + u8 conf1 = 0; + u8 inputclkfs = 0; + + /* it cares I2S only */ + if ((fmt->fmt != HDMI_I2S) || + (fmt->bit_clk_master | fmt->frame_clk_master)) { + dev_err(dev, "unsupported format/settings\n"); + return -EINVAL; + } + + inputclkfs = HDMI_AUD_INPUTCLKFS_64FS; + conf0 = HDMI_AUD_CONF0_I2S_ALL_ENABLE; + + switch (hparms->sample_width) { + case 16: + conf1 = HDMI_AUD_CONF1_WIDTH_16; + break; + case 24: + case 32: + conf1 = HDMI_AUD_CONF1_WIDTH_24; + break; + } + + dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); + + hdmi_write(audio, inputclkfs, HDMI_AUD_INPUTCLKFS); + hdmi_write(audio, conf0, HDMI_AUD_CONF0); + hdmi_write(audio, conf1, HDMI_AUD_CONF1); + + dw_hdmi_audio_enable(hdmi); + + return 0; +} + +static void dw_hdmi_i2s_audio_shutdown(struct device *dev, void *data) +{ + struct dw_hdmi_i2s_audio_data *audio = data; + struct dw_hdmi *hdmi = audio->hdmi; + + dw_hdmi_audio_disable(hdmi); + + hdmi_write(audio, HDMI_AUD_CONF0_SW_RESET, HDMI_AUD_CONF0); +} + +static struct hdmi_codec_ops dw_hdmi_i2s_ops = { + .hw_params = dw_hdmi_i2s_hw_params, + .audio_shutdown = dw_hdmi_i2s_audio_shutdown, +}; + +static int snd_dw_hdmi_probe(struct platform_device *pdev) +{ + struct dw_hdmi_i2s_audio_data *audio = pdev->dev.platform_data; + struct platform_device_info pdevinfo; + struct hdmi_codec_pdata pdata; + struct platform_device *platform; + + + pdata.ops = &dw_hdmi_i2s_ops; + pdata.i2s = 1; + pdata.max_i2s_channels = 6; + pdata.data = audio; + + memset(&pdevinfo, 0, sizeof(pdevinfo)); + pdevinfo.parent = pdev->dev.parent; + pdevinfo.id = PLATFORM_DEVID_AUTO; + pdevinfo.name = HDMI_CODEC_DRV_NAME; + pdevinfo.data = &pdata; + pdevinfo.size_data = sizeof(pdata); + pdevinfo.dma_mask = DMA_BIT_MASK(32); + + platform = platform_device_register_full(&pdevinfo); + if (IS_ERR(platform)) + return PTR_ERR(platform); + + dev_set_drvdata(&pdev->dev, platform); + + return 0; +} + +static int snd_dw_hdmi_remove(struct platform_device *pdev) +{ + struct platform_device *platform = dev_get_drvdata(&pdev->dev); + + platform_device_unregister(platform); + + return 0; +} + +static struct platform_driver snd_dw_hdmi_driver = { + .probe = snd_dw_hdmi_probe, + .remove = snd_dw_hdmi_remove, + .driver = { + .name = DRIVER_NAME, + .owner = THIS_MODULE, + }, +}; +module_platform_driver(snd_dw_hdmi_driver); + +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); +MODULE_DESCRIPTION("Synopsis Designware HDMI I2S ALSA SoC interface"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRIVER_NAME); diff --git a/drivers/gpu/drm/bridge/dw-hdmi.c b/drivers/gpu/drm/bridge/dw-hdmi.c index ab7023e..30a2e16 100644 --- a/drivers/gpu/drm/bridge/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/dw-hdmi.c @@ -1639,10 +1639,11 @@ int dw_hdmi_bind(struct device *dev, struct device *master, struct device_node *np = dev->of_node; struct platform_device_info pdevinfo; struct device_node *ddc_node; - struct dw_hdmi_audio_data audio; struct dw_hdmi *hdmi; int ret; u32 val = 1; + u8 config0; + u8 config1; hdmi = devm_kzalloc(dev, sizeof(*hdmi), GFP_KERNEL); if (!hdmi) @@ -1770,7 +1771,12 @@ int dw_hdmi_bind(struct device *dev, struct device *master, pdevinfo.parent = dev; pdevinfo.id = PLATFORM_DEVID_AUTO; - if (hdmi_readb(hdmi, HDMI_CONFIG1_ID) & HDMI_CONFIG1_AHB) { + config0 = hdmi_readb(hdmi, HDMI_CONFIG0_ID); + config1 = hdmi_readb(hdmi, HDMI_CONFIG1_ID); + + if (config1 & HDMI_CONFIG1_AHB) { + struct dw_hdmi_audio_data audio; + audio.phys = iores->start; audio.base = hdmi->regs; audio.irq = irq; @@ -1782,6 +1788,18 @@ int dw_hdmi_bind(struct device *dev, struct device *master, pdevinfo.size_data = sizeof(audio); pdevinfo.dma_mask = DMA_BIT_MASK(32); hdmi->audio = platform_device_register_full(&pdevinfo); + } else if (config0 & HDMI_CONFIG0_I2S) { + struct dw_hdmi_i2s_audio_data audio; + + audio.hdmi = hdmi; + audio.write = hdmi_writeb; + audio.read = hdmi_readb; + + pdevinfo.name = "dw-hdmi-i2s-audio"; + pdevinfo.data = &audio; + pdevinfo.size_data = sizeof(audio); + pdevinfo.dma_mask = DMA_BIT_MASK(32); + hdmi->audio = platform_device_register_full(&pdevinfo); } dev_set_drvdata(dev, hdmi); diff --git a/drivers/gpu/drm/bridge/dw-hdmi.h b/drivers/gpu/drm/bridge/dw-hdmi.h index fc9a560..c8bdbf2 100644 --- a/drivers/gpu/drm/bridge/dw-hdmi.h +++ b/drivers/gpu/drm/bridge/dw-hdmi.h @@ -545,6 +545,10 @@ #define HDMI_I2CM_FS_SCL_LCNT_0_ADDR 0x7E12 enum { + +/* CONFIG0_ID field values */ + HDMI_CONFIG0_I2S = 0x10, + /* CONFIG1_ID field values */ HDMI_CONFIG1_AHB = 0x01, @@ -887,6 +891,17 @@ enum { HDMI_PHY_I2CM_CTLINT_ADDR_ARBITRATION_POL = 0x08, HDMI_PHY_I2CM_CTLINT_ADDR_ARBITRATION_MASK = 0x04, +/* AUD_CONF0 field values */ + HDMI_AUD_CONF0_SW_RESET = 0x80, + HDMI_AUD_CONF0_I2S_ALL_ENABLE = 0x2F, + +/* AUD_CONF1 field values */ + HDMI_AUD_CONF1_MODE_I2S = 0x00, + HDMI_AUD_CONF1_MODE_RIGHT_J = 0x02, + HDMI_AUD_CONF1_MODE_LEFT_J = 0x04, + HDMI_AUD_CONF1_WIDTH_16 = 0x10, + HDMI_AUD_CONF1_WIDTH_24 = 0x18, + /* AUD_CTS3 field values */ HDMI_AUD_CTS3_N_SHIFT_OFFSET = 5, HDMI_AUD_CTS3_N_SHIFT_MASK = 0xe0, @@ -901,6 +916,12 @@ enum { HDMI_AUD_CTS3_CTS_MANUAL = 0x10, HDMI_AUD_CTS3_AUDCTS19_16_MASK = 0x0f, +/* HDMI_AUD_INPUTCLKFS field values */ + HDMI_AUD_INPUTCLKFS_128FS = 0, + HDMI_AUD_INPUTCLKFS_256FS = 1, + HDMI_AUD_INPUTCLKFS_512FS = 2, + HDMI_AUD_INPUTCLKFS_64FS = 4, + /* AHB_DMA_CONF0 field values */ HDMI_AHB_DMA_CONF0_SW_FIFO_RST_OFFSET = 7, HDMI_AHB_DMA_CONF0_SW_FIFO_RST_MASK = 0x80, -- 1.9.1

3 5

[alsa-devel] [PATCH 2/2] topology: Store number of strings of a text section
by mengdong.lin＠linux.intel.com 24 Nov '16

24 Nov '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Enum controls may use text elements and need the number of strings. This patch adds a text child object for a generic element. When parsing a text section from the configuation file, store the text strings and number of strings (num_items) in the text child object of the element. Reported-by: G Kranthi <gudishax.kranthikumar(a)intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/src/topology/ctl.c b/src/topology/ctl.c index 54f8e44..907a97f 100644 --- a/src/topology/ctl.c +++ b/src/topology/ctl.c @@ -159,9 +159,11 @@ static void copy_enum_texts(struct tplg_elem *enum_elem, struct tplg_elem *ref_elem) { struct snd_soc_tplg_enum_control *ec = enum_elem->enum_ctrl; + struct tplg_texts *texts = ref_elem->texts; - memcpy(ec->texts, ref_elem->texts, + memcpy(ec->texts, texts->items, SND_SOC_TPLG_NUM_TEXTS * SNDRV_CTL_ELEM_ID_NAME_MAXLEN); + ec->items += texts->num_items; } /* check referenced text for a enum control */ diff --git a/src/topology/elem.c b/src/topology/elem.c index 1a5ac84..db26483 100644 --- a/src/topology/elem.c +++ b/src/topology/elem.c @@ -159,6 +159,7 @@ struct tplg_elem* tplg_elem_new_common(snd_tplg_t *tplg, break; case SND_TPLG_TYPE_TEXT: list_add_tail(&elem->list, &tplg->text_list); + obj_size = sizeof(struct tplg_texts); break; case SND_TPLG_TYPE_TLV: list_add_tail(&elem->list, &tplg->tlv_list); diff --git a/src/topology/text.c b/src/topology/text.c index 0c6594a..303fbeb 100644 --- a/src/topology/text.c +++ b/src/topology/text.c @@ -25,6 +25,7 @@ static int parse_text_values(snd_config_t *cfg, struct tplg_elem *elem) { + struct tplg_texts *texts = elem->texts; snd_config_iterator_t i, next; snd_config_t *n; const char *value = NULL; @@ -44,13 +45,14 @@ static int parse_text_values(snd_config_t *cfg, struct tplg_elem *elem) if (snd_config_get_string(n, &value) < 0) continue; - elem_copy_text(&elem->texts[j][0], value, + elem_copy_text(&texts->items[j][0], value, SNDRV_CTL_ELEM_ID_NAME_MAXLEN); - tplg_dbg("\t%s\n", &elem->texts[j][0]); + tplg_dbg("\t%s\n", &texts->items[j][0]); j++; } + texts->num_items = j; return 0; } diff --git a/src/topology/tplg_local.h b/src/topology/tplg_local.h index f913563..eb2f7bd 100644 --- a/src/topology/tplg_local.h +++ b/src/topology/tplg_local.h @@ -91,6 +91,11 @@ struct tplg_ref { struct list_head list; }; +struct tplg_texts { + unsigned int num_items; + char items[SND_SOC_TPLG_NUM_TEXTS][SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; +}; + /* element for vendor tokens */ struct tplg_token { char id[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; @@ -128,9 +133,6 @@ struct tplg_elem { char id[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - /* storage for texts and data if this is text or data elem*/ - char texts[SND_SOC_TPLG_NUM_TEXTS][SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - int index; enum snd_tplg_type type; @@ -155,6 +157,7 @@ struct tplg_elem { /* these do not map to UAPI structs but are internal only */ struct snd_soc_tplg_ctl_tlv *tlv; + struct tplg_texts *texts; struct snd_soc_tplg_private *data; struct tplg_vendor_tokens *tokens; struct tplg_vendor_tuples *tuples; -- 2.5.0

1 0

[alsa-devel] [PATCH v2 0/4] ASoC: Convert to module_i2c/spi_driver()
by Kuninori Morimoto 24 Nov '16

24 Nov '16

Hi Mark These are v2 of "ASoC: Convert to module_i2c/spi_driver()" patch series. Mainly, it updated Subject and commit comment, and doesn't have I2C/SPI side patches. It includes I2C side only. Kuninori Morimoto (4): ASoC: uda1380: Remove #if IS_ENABLED(CONFIG_I2C) ASoC: wm9081: Remove #if IS_ENABLED(CONFIG_I2C) ASoC: wm8580: Remove #if IS_ENABLED(CONFIG_I2C) ASoC: wm8523: Remove #if IS_ENABLED(CONFIG_I2C) sound/soc/codecs/Kconfig | 2 ++ sound/soc/codecs/uda1380.c | 22 +--------------------- sound/soc/codecs/wm8523.c | 24 +----------------------- sound/soc/codecs/wm8580.c | 25 +------------------------ sound/soc/codecs/wm9081.c | 2 -- 5 files changed, 5 insertions(+), 70 deletions(-) -- 1.9.1

3 10

[alsa-devel] [PATCH] dbri: Fix compiler warning
by Tushar Dave 23 Nov '16

23 Nov '16

dbri uses 'u32' for dma handle while invoking kernel DMA APIs, instead of using dma_addr_t. This hasn't caused any 'incompatible pointer type' warning on SPARC because until now dma_addr_t is of type u32. However, recent changes in SPARC ATU (iommu) enabled 64bit DMA and therefore dma_addr_t became of type u64. This makes 'incompatible pointer type' warnings inevitable. e.g. sound/sparc/dbri.c: In function ‘snd_dbri_create’: sound/sparc/dbri.c:2538: warning: passing argument 3 of ‘dma_zalloc_coherent’ from incompatible pointer type ./include/linux/dma-mapping.h:608: note: expected ‘dma_addr_t *’ but argument is of type ‘u32 *’ This patch resolves above compiler warning. Signed-off-by: Tushar Dave <tushar.n.dave(a)oracle.com> Reviewed-by: thomas tai <thomas.tai(a)oracle.com> --- sound/sparc/dbri.c | 26 ++++++++++++++++---------- 1 file changed, 16 insertions(+), 10 deletions(-) diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 0190cb6..542c1b5 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -304,7 +304,7 @@ struct snd_dbri { spinlock_t lock; struct dbri_dma *dma; /* Pointer to our DMA block */ - u32 dma_dvma; /* DBRI visible DMA address */ + dma_addr_t dma_dvma; /* DBRI visible DMA address */ void __iomem *regs; /* dbri HW regs */ int dbri_irqp; /* intr queue pointer */ @@ -657,12 +657,14 @@ static void dbri_cmdwait(struct snd_dbri *dbri) */ static s32 *dbri_cmdlock(struct snd_dbri *dbri, int len) { + u32 dvma_addr = (u32)dbri->dma_dvma; + /* Space for 2 WAIT cmds (replaced later by 1 JUMP cmd) */ len += 2; spin_lock(&dbri->cmdlock); if (dbri->cmdptr - dbri->dma->cmd + len < DBRI_NO_CMDS - 2) return dbri->cmdptr + 2; - else if (len < sbus_readl(dbri->regs + REG8) - dbri->dma_dvma) + else if (len < sbus_readl(dbri->regs + REG8) - dvma_addr) return dbri->dma->cmd; else printk(KERN_ERR "DBRI: no space for commands."); @@ -680,6 +682,7 @@ static s32 *dbri_cmdlock(struct snd_dbri *dbri, int len) */ static void dbri_cmdsend(struct snd_dbri *dbri, s32 *cmd, int len) { + u32 dvma_addr = (u32)dbri->dma_dvma; s32 tmp, addr; static int wait_id = 0; @@ -689,7 +692,7 @@ static void dbri_cmdsend(struct snd_dbri *dbri, s32 *cmd, int len) *(cmd+1) = DBRI_CMD(D_WAIT, 1, wait_id); /* Replace the last command with JUMP */ - addr = dbri->dma_dvma + (cmd - len - dbri->dma->cmd) * sizeof(s32); + addr = dvma_addr + (cmd - len - dbri->dma->cmd) * sizeof(s32); *(dbri->cmdptr+1) = addr; *(dbri->cmdptr) = DBRI_CMD(D_JUMP, 0, 0); @@ -747,6 +750,7 @@ static void dbri_reset(struct snd_dbri *dbri) /* Lock must not be held before calling this */ static void dbri_initialize(struct snd_dbri *dbri) { + u32 dvma_addr = (u32)dbri->dma_dvma; s32 *cmd; u32 dma_addr; unsigned long flags; @@ -764,7 +768,7 @@ static void dbri_initialize(struct snd_dbri *dbri) /* * Initialize the interrupt ring buffer. */ - dma_addr = dbri->dma_dvma + dbri_dma_off(intr, 0); + dma_addr = dvma_addr + dbri_dma_off(intr, 0); dbri->dma->intr[0] = dma_addr; dbri->dbri_irqp = 1; /* @@ -778,7 +782,7 @@ static void dbri_initialize(struct snd_dbri *dbri) dbri->cmdptr = cmd; *(cmd++) = DBRI_CMD(D_WAIT, 1, 0); *(cmd++) = DBRI_CMD(D_WAIT, 1, 0); - dma_addr = dbri->dma_dvma + dbri_dma_off(cmd, 0); + dma_addr = dvma_addr + dbri_dma_off(cmd, 0); sbus_writel(dma_addr, dbri->regs + REG8); spin_unlock(&dbri->cmdlock); @@ -1077,6 +1081,7 @@ static void recv_fixed(struct snd_dbri *dbri, int pipe, volatile __u32 *ptr) static int setup_descs(struct snd_dbri *dbri, int streamno, unsigned int period) { struct dbri_streaminfo *info = &dbri->stream_info[streamno]; + u32 dvma_addr = (u32)dbri->dma_dvma; __u32 dvma_buffer; int desc; int len; @@ -1177,7 +1182,7 @@ static int setup_descs(struct snd_dbri *dbri, int streamno, unsigned int period) else { dbri->next_desc[last_desc] = desc; dbri->dma->desc[last_desc].nda = - dbri->dma_dvma + dbri_dma_off(desc, desc); + dvma_addr + dbri_dma_off(desc, desc); } last_desc = desc; @@ -1192,7 +1197,7 @@ static int setup_descs(struct snd_dbri *dbri, int streamno, unsigned int period) } dbri->dma->desc[last_desc].nda = - dbri->dma_dvma + dbri_dma_off(desc, first_desc); + dvma_addr + dbri_dma_off(desc, first_desc); dbri->next_desc[last_desc] = first_desc; dbri->pipes[info->pipe].first_desc = first_desc; dbri->pipes[info->pipe].desc = first_desc; @@ -1697,6 +1702,7 @@ calling dbri_process_interrupt_buffer(), which works even if the CPU's static void xmit_descs(struct snd_dbri *dbri) { struct dbri_streaminfo *info; + u32 dvma_addr = (u32)dbri->dma_dvma; s32 *cmd; unsigned long flags; int first_td; @@ -1718,7 +1724,7 @@ static void xmit_descs(struct snd_dbri *dbri) *(cmd++) = DBRI_CMD(D_SDP, 0, dbri->pipes[info->pipe].sdp | D_SDP_P | D_SDP_EVERY | D_SDP_C); - *(cmd++) = dbri->dma_dvma + + *(cmd++) = dvma_addr + dbri_dma_off(desc, first_td); dbri_cmdsend(dbri, cmd, 2); @@ -1740,7 +1746,7 @@ static void xmit_descs(struct snd_dbri *dbri) *(cmd++) = DBRI_CMD(D_SDP, 0, dbri->pipes[info->pipe].sdp | D_SDP_P | D_SDP_EVERY | D_SDP_C); - *(cmd++) = dbri->dma_dvma + + *(cmd++) = dvma_addr + dbri_dma_off(desc, first_td); dbri_cmdsend(dbri, cmd, 2); @@ -2539,7 +2545,7 @@ static int snd_dbri_create(struct snd_card *card, if (!dbri->dma) return -ENOMEM; - dprintk(D_GEN, "DMA Cmd Block 0x%p (0x%08x)\n", + dprintk(D_GEN, "DMA Cmd Block 0x%p (0x%08llx)\n", dbri->dma, dbri->dma_dvma); /* Map the registers into memory. */ -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: ab8500: Remove explicit initialization of driver callbacks to NULL
by Lars-Peter Clausen 23 Nov '16

23 Nov '16

Fields of structs with global storage are implicitly initialized to 0/NULL, there is usually no need to do this explicitly. Removing the initialization of the legacy suspend/resume callback fields also gets the driver ready for the day when they are eventually removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ab8500-codec.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 935ff7c..312b2a1 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2587,8 +2587,6 @@ static struct platform_driver ab8500_codec_platform_driver = { }, .probe = ab8500_codec_driver_probe, .remove = ab8500_codec_driver_remove, - .suspend = NULL, - .resume = NULL, }; module_platform_driver(ab8500_codec_platform_driver); -- 2.1.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: uda1380: Remove unused DAI ID defines
by Lars-Peter Clausen 23 Nov '16

23 Nov '16

The DAI ID defines are back from the time when DAIs were referenced by a numerical ID. These days a string is used for matching instead and the defines are unused. The last user of these defines was removed in commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support"). So remove the defines as well. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/uda1380.h | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h index 942e392..69a326a 100644 --- a/sound/soc/codecs/uda1380.h +++ b/sound/soc/codecs/uda1380.h @@ -72,8 +72,4 @@ #define R22_SKIP_DCFIL 0x0002 #define R23_AGC_EN 0x0001 -#define UDA1380_DAI_DUPLEX 0 /* playback and capture on single DAI */ -#define UDA1380_DAI_PLAYBACK 1 /* playback DAI */ -#define UDA1380_DAI_CAPTURE 2 /* capture DAI */ - #endif /* _UDA1380_H */ -- 2.1.4

2 3

[alsa-devel] [PATCH v3 0/3] ASoC: intel: atom: Add FW version information
by Sebastien Guiriec 23 Nov '16

23 Nov '16

This patch series is adding FW version information of SST. Version is both log as dev_info and sysfs entry. Update since V1: - sysfs entry is set to "FW not yet loaded" in case we do not have loaded the FW once Update since V2: - Add Vinod ack. https://www.spinics.net/lists/alsa-devel/msg56521.html Sebastien Guiriec (3): ASoC: intel: atom: Add debug information related to FW version ASoC: Intel: atom: save FW version ASoC: Intel: atom: Add sysfs entry in order to store FW version sound/soc/intel/atom/sst/sst.c | 39 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/atom/sst/sst.h | 1 + sound/soc/intel/atom/sst/sst_ipc.c | 10 +++++++++- 3 files changed, 49 insertions(+), 1 deletion(-) -- 2.9.3 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

2 4

[alsa-devel] [PATCH] ASoC: cht_bsw_rt5645: Fix leftover kmalloc
by Takashi Iwai 23 Nov '16

23 Nov '16

cht_bsw_rt5645 driver allocates the own codec_id string but doesn't release it. For simplicity, put the string in cht_mc_private; then the string is allocated in a shot and released altogether. Fixes: c8560b7c917f ("ASoC: cht_bsw_rt5645: Fix writing to string literal") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 56056ed7fcfd..16c94c45ce50 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -44,6 +44,7 @@ struct cht_acpi_card { struct cht_mc_private { struct snd_soc_jack jack; struct cht_acpi_card *acpi_card; + char codec_name[16]; }; static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) @@ -354,7 +355,6 @@ static int snd_cht_mc_probe(struct platform_device *pdev) int i; struct cht_mc_private *drv; struct snd_soc_card *card = snd_soc_cards[0].soc_card; - char codec_name[16]; struct sst_acpi_mach *mach; const char *i2c_name = NULL; int dai_index = 0; @@ -374,12 +374,12 @@ static int snd_cht_mc_probe(struct platform_device *pdev) } card->dev = &pdev->dev; mach = card->dev->platform_data; - sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); + sprintf(drv->codec_name, "i2c-%s:00", drv->acpi_card->codec_id); /* set correct codec name */ for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) if (!strcmp(card->dai_link[i].codec_name, "i2c-10EC5645:00")) { - card->dai_link[i].codec_name = kstrdup(codec_name, GFP_KERNEL); + card->dai_link[i].codec_name = drv->codec_name; dai_index = i; } -- 2.10.2

2 1

[alsa-devel] question about sound/soc/intel/boards/mfld_machine.c
by Julia Lawall 23 Nov '16

23 Nov '16

Hello, I wonder if the file sound/soc/intel/boards/mfld_machine.c is useful? Until: commit e56c72d5f201044b14191c5b83a25e17f2d68ccf Author: Jie Yang <yang.jie(a)intel.com> Date: Thu Apr 2 15:37:02 2015 +0800 ASoC: Intel: create boards folder and move sst boards files in it was in the directory sound/soc/intel and mentioned in sound/soc/intel/Makefile. The above patch moved it to sound/soc/intel/boards/, but didn't adjust its Makefile entry. The patch: commit b97169da06992ef04081e66ed22bbdb23dbf6610 Author: Jie Yang <yang.jie(a)intel.com> Date: Thu Apr 2 15:37:04 2015 +0800 ASoC: Intel: create atom folder and move atom platform files in removed it from sound/soc/intel/Makefile, but didn't move it anywhere. As far as I can tell, there is currently no make information for the file in the kernel. Should the file be dropped? julia

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[alsa-devel] [asoc:topic/samsung 15/16] sound/soc/samsung/i2s.c:1270:2-4: ERROR: test of a variable/field address
by kbuild test robot 22 Nov '16

22 Nov '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/samsung head: 95f5609d223d661419061bd6231da01a317c30d9 commit: a41dcdeee5d87cf3852b857cc3a8507832ec8b42 [15/16] ASoC: samsung: Enable COMPILE_TEST for entire Samsung ASoc coccinelle warnings: (new ones prefixed by >>) >> sound/soc/samsung/i2s.c:1270:2-4: ERROR: test of a variable/field address vim +1270 sound/soc/samsung/i2s.c 40476f61 sound/soc/samsung/i2s.c Padmavathi Venna 2013-01-18 1254 return -ENOMEM; 1c7ac018 sound/soc/s3c24xx/i2s.c Jassi Brar 2010-11-22 1255 } 1c7ac018 sound/soc/s3c24xx/i2s.c Jassi Brar 2010-11-22 1256 f3670536 sound/soc/samsung/i2s.c Sylwester Nawrocki 2015-01-14 1257 spin_lock_init(&pri_dai->spinlock); f3670536 sound/soc/samsung/i2s.c Sylwester Nawrocki 2015-01-14 1258 pri_dai->lock = &pri_dai->spinlock; f3670536 sound/soc/samsung/i2s.c Sylwester Nawrocki 2015-01-14 1259 40476f61 sound/soc/samsung/i2s.c Padmavathi Venna 2013-01-18 1260 if (!np) { 40476f61 sound/soc/samsung/i2s.c Padmavathi Venna 2013-01-18 1261 if (i2s_pdata == NULL) { 40476f61 sound/soc/samsung/i2s.c Padmavathi Venna 2013-01-18 1262 dev_err(&pdev->dev, "Can't work without s3c_audio_pdata\n"); 40476f61 sound/soc/samsung/i2s.c Padmavathi Venna 2013-01-18 1263 return -EINVAL; 40476f61 sound/soc/samsung/i2s.c Padmavathi Venna 2013-01-18 1264 } 40476f61 sound/soc/samsung/i2s.c Padmavathi Venna 2013-01-18 1265 69e7a69a sound/soc/samsung/i2s.c Sylwester Nawrocki 2016-08-04 1266 pri_dai->dma_playback.filter_data = i2s_pdata->dma_playback; 69e7a69a sound/soc/samsung/i2s.c Sylwester Nawrocki 2016-08-04 1267 pri_dai->dma_capture.filter_data = i2s_pdata->dma_capture; 9bdca822 sound/soc/samsung/i2s.c Arnd Bergmann 2015-11-18 1268 pri_dai->filter = i2s_pdata->dma_filter; b9a1a743 sound/soc/samsung/i2s.c Arnd Bergmann 2015-11-18 1269 40476f61 sound/soc/samsung/i2s.c Padmavathi Venna 2013-01-18 @1270 if (&i2s_pdata->type) 40476f61 sound/soc/samsung/i2s.c Padmavathi Venna 2013-01-18 1271 i2s_cfg = &i2s_pdata->type.i2s; 40476f61 sound/soc/samsung/i2s.c Padmavathi Venna 2013-01-18 1272 40476f61 sound/soc/samsung/i2s.c Padmavathi Venna 2013-01-18 1273 if (i2s_cfg) { 40476f61 sound/soc/samsung/i2s.c Padmavathi Venna 2013-01-18 1274 quirks = i2s_cfg->quirks; 40476f61 sound/soc/samsung/i2s.c Padmavathi Venna 2013-01-18 1275 idma_addr = i2s_cfg->idma_addr; 40476f61 sound/soc/samsung/i2s.c Padmavathi Venna 2013-01-18 1276 } 40476f61 sound/soc/samsung/i2s.c Padmavathi Venna 2013-01-18 1277 } else { 7da493e9 sound/soc/samsung/i2s.c Padmavathi Venna 2013-08-12 1278 quirks = i2s_dai_data->quirks; :::::: The code at line 1270 was first introduced by commit :::::: 40476f61897933d524b7069a6df65629a469d922 ASoC: samsung: Add DT support for i2s :::::: TO: Padmavathi Venna <padma.v(a)samsung.com> :::::: CC: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v2] ASoC: wm_adsp: Check return value from wm_adsp_buffer_init
by Charles Keepax 22 Nov '16

22 Nov '16

We are not checking the return value from a call to wm_adsp_buffer_init it looks like this used to be returned at the bottom of the function but probably got missed when more error paths were added. This patch adds the appropriate error check. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Add unlock of the mutex on the error path Thanks, Charles sound/soc/codecs/wm_adsp.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6ccb731..e00ccfc 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2580,8 +2580,13 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, mutex_lock(&dsp->pwr_lock); - if (wm_adsp_fw[dsp->fw].num_caps != 0) + if (wm_adsp_fw[dsp->fw].num_caps != 0) { ret = wm_adsp_buffer_init(dsp); + if (ret < 0) { + mutex_unlock(&dsp->pwr_lock); + goto err; + } + } mutex_unlock(&dsp->pwr_lock); -- 2.1.4

3 2

[alsa-devel] [PATCH] ASoC: Make return type of dpcm_state_string() const char *
by Lars-Peter Clausen 22 Nov '16

22 Nov '16

dpcm_state_string() returns a pointer to a string literal. Modifying a string literal causes undefined behaviour. So make the return type of the function const char * to make it explicit that the returned value should not be modified. This patch is purely cosmetic, none of the users of dpcm_state_string() attempt to modify the returned content. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d56a16a..e7a1eaa 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2882,7 +2882,7 @@ int snd_soc_platform_trigger(struct snd_pcm_substream *substream, EXPORT_SYMBOL_GPL(snd_soc_platform_trigger); #ifdef CONFIG_DEBUG_FS -static char *dpcm_state_string(enum snd_soc_dpcm_state state) +static const char *dpcm_state_string(enum snd_soc_dpcm_state state) { switch (state) { case SND_SOC_DPCM_STATE_NEW: -- 2.1.4

2 1

[alsa-devel] [RFT v2 0/5] ASoC: samsung: Minor cleanup for old machines
by Krzysztof Kozlowski 22 Nov '16

22 Nov '16

Hi, Few patches removing dead code (machines not supported). Changes since v1: 1. Squash two smdk_wm8580 changes into patch #2. Now num_dai_links is always equal to two, so remove also SEC_PLAYBACK dai_link. (suggested by Lars-Peter Clausen). The second patch ([RFT v2 2/5] ASoC: samsung: smdk_wm8580: Remove old platforms and drop mach-types usage) requires testing. I hope I understood the code correctly. The last ARM patch is independent. I will take it through samsung-soc tree. I put it here for reference. Best regards, Krzysztof Krzysztof Kozlowski (5): ASoC: samsung: Remove non-existing MACH dependencies ASoC: samsung: smdk_wm8580: Remove old platforms and drop mach-types usage ASoC: samsung: Enable COMPILE_TEST for SmartQ and WM8580 ASoC: samsung: Enable COMPILE_TEST for entire Samsung ASoc ARM: s5pv210_defconfig: Remove old MACHs arch/arm/configs/s5pv210_defconfig | 4 ---- sound/soc/samsung/Kconfig | 8 +++++--- sound/soc/samsung/smdk_wm8580.c | 28 ++-------------------------- 3 files changed, 7 insertions(+), 33 deletions(-) -- 2.7.4

4 9

[alsa-devel] [PATCH 0/6] ASoC: samsung: Minor cleanup for old machines
by Krzysztof Kozlowski 22 Nov '16

22 Nov '16

Hi, Few patches removing dead code (machines not supported). The third patch ([RFT 3/6] ASoC: samsung: smdk_wm8580: Remove machine specific quirks) requires testing. I hope I understood the code correctly. The last ARM patch is independent. I will take it through samsung-soc tree. I put it here for reference. Best regards, Krzysztof Krzysztof Kozlowski (6): ASoC: samsung: Remove non-existing MACH dependencies ASoC: samsung: smdk_wm8580: Remove old platforms and drop mach-types usage ASoC: samsung: smdk_wm8580: Remove machine specific quirks ASoC: samsung: Enable COMPILE_TEST for SmartQ and WM8580 ASoC: samsung: Enable COMPILE_TEST for entire Samsung ASoc ARM: s5pv210_defconfig: Remove old MACHs arch/arm/configs/s5pv210_defconfig | 4 ---- sound/soc/samsung/Kconfig | 8 +++++--- sound/soc/samsung/smdk_wm8580.c | 17 ++--------------- 3 files changed, 7 insertions(+), 22 deletions(-) -- 2.7.4

4 14

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Remove unused wm_coeff_ctl.kcontrol
by Richard Fitzgerald 22 Nov '16

22 Nov '16

The kcontrol pointer in wm_coeff_ctl is not used now. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6ccb731..7698273 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -451,7 +451,6 @@ struct wm_coeff_ctl { unsigned int offset; size_t len; unsigned int set:1; - struct snd_kcontrol *kcontrol; struct soc_bytes_ext bytes_ext; unsigned int flags; unsigned int type; @@ -1138,8 +1137,6 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) kfree(kcontrol); - ctl->kcontrol = snd_soc_card_get_kcontrol(dsp->card, ctl->name); - return 0; err_kcontrol: -- 1.9.1

2 2

[alsa-devel] [PATCH v3 0/2] ASoC: rsnd: DMA power fix and IOMMU support
by Kuninori Morimoto 22 Nov '16

22 Nov '16

Hi Mark Cc Laurent These are v3 of Renesas sound DMA patches. 1) patch controls Audio-DMAC power. Current driver gets DMAEngile handle when .probe() timing, and release it on .remove() timing. But Audio-DMAC power ON/OFF are related to this timing. This means Audio-DMAC always ON Now. Basically, this kind of DMAC power control should be handled by DMAEngile, but current DMAEngile can't handle pm_runtime_xxx on atomic context. Because of this DMAEngile API issue, the *better* solution is get DMAEngile handle on every playback timing. Kuninori Morimoto (2): ASoC: rsnd: Request/Release DMA channel each time ASoC: rsnd: use dma_sync_single_for_xxx() for IOMMU sound/soc/sh/rcar/core.c | 4 +- sound/soc/sh/rcar/dma.c | 267 +++++++++++++++++++++++++++++++++-------------- 2 files changed, 188 insertions(+), 83 deletions(-) -- 2.7.4

2 3

[alsa-devel] [PATCH 0/8] ASoC: rsnd: DMA power fix and IOMMU support
by Kuninori Morimoto 22 Nov '16

22 Nov '16

Hi Mark These patches are for Renesas sound DMA. 1st feature is that it can power save for Audio DMAC. 2nd feature is that it can support IOMMU 1) - 5) : cleanup for 6) - 8) 6) - 7) : Audio DMAC power save 8) : IOMMU support Kuninori Morimoto (8): 1) ASoC: rsnd: depends on OF 2) ASoC: rsnd: remove non DT support for DMA 3) ASoC: rsnd: don't use devm_request_irq() for SSI 4) ASoC: rsnd: remove rsnd_dma_detach() 5) ASoC: rsnd: don't call unneeded of_node_put() on dma.c 6) ASoC: rsnd: add nolock_start/stop callback 7) ASoC: rsnd: Request/Release DMA channel each time 8) ASoC: rsnd: use dma_sync_single_for_xxx() for IOMMU sound/soc/sh/Kconfig | 1 + sound/soc/sh/rcar/core.c | 30 +++++++++++++- sound/soc/sh/rcar/dma.c | 304 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++------------------------------------------------- sound/soc/sh/rcar/rsnd.h | 18 ++++++-- sound/soc/sh/rcar/src.c | 2 +- sound/soc/sh/rcar/ssi.c | 22 +++++----- 6 files changed, 248 insertions(+), 129 deletions(-) Best regards --- Kuninori Morimoto

2 19

[alsa-devel] [PATCH] ASoC: wm_adsp: Check return value from wm_adsp_buffer_init
by Charles Keepax 22 Nov '16

22 Nov '16

We are not checking the return value from a call to wm_adsp_buffer_init it looks like this used to be returned at the bottom of the function but probably got missed when more error paths were added. This patch adds the appropriate error check. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6ccb731..7e4f8dd 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2580,8 +2580,11 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, mutex_lock(&dsp->pwr_lock); - if (wm_adsp_fw[dsp->fw].num_caps != 0) + if (wm_adsp_fw[dsp->fw].num_caps != 0) { ret = wm_adsp_buffer_init(dsp); + if (ret < 0) + goto err; + } mutex_unlock(&dsp->pwr_lock); -- 2.1.4

2 2

[alsa-devel] [PATCH 0/7] topology: Add support for physical DAIs
by mengdong.lin＠linux.intel.com 22 Nov '16

22 Nov '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This is the last series for topology user space atm. This series adds support for physical DAIs and includes a few minor fixes. All ABI updates here can be supported by kernel in a backward compatible way. Guneshwor Singh (3): topology: ABI - Define new types for physical DAI topology: Support configuring physical DAIs by C API topology: Export physical DAIs to the binary for kernel Mengdong Lin (4): topology: ABI - Update manifest and pump ABI version to 5 topology: Parse physical DAIs in text conf file topology: ABI - Add voice wake up flag for DAI links topology: Revise document and comments for ABI v5 include/sound/asoc.h | 37 +++++++- include/topology.h | 94 ++++++++++++------- src/topology/builder.c | 11 +++ src/topology/data.c | 4 + src/topology/elem.c | 4 + src/topology/parser.c | 16 ++++ src/topology/pcm.c | 223 +++++++++++++++++++++++++++++++++++++++++++++- src/topology/tplg_local.h | 7 ++ 8 files changed, 362 insertions(+), 34 deletions(-) -- 2.5.0

2 8

[alsa-devel] [PATCH v4 00/12] ASoC: add OF graph base simple-card
by Kuninori Morimoto 22 Nov '16

22 Nov '16

Hi Rob, Mark These are v4 of OF graph base simple-card patch-set. v3 patch had ALSA SoC side prepare patches as [1/xx] - [5/xx], but these are already accepted by Mark, thus, this v4 doesn't include these. Mainly, v4 solved non-OF case compile error which was reported by kbuild. 1) - 5) : OF graph new feature 6) - 12) : OF graph base simple-card (depends on above 2 patch-set) Kuninori Morimoto (12): 1) of_graph: add of_graph_get_remote_endpoint() 2) of_graph: add of_graph_get_port_parent() 3) of_graph: add of_graph_get_top_port() 4) of_graph: add for_each_of_port() / for_each_of_endpoint_in_port() 5) of_graph: add of_graph_get_port/endpoint_count() 6) ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() 7) ASoC: simple-card-utils: add asoc_simple_card_try_to_probe_graph_card() 8) ASoC: simple-card-utils: adjust for graph on asoc_simple_card_parse_card_name 9) ASoC: add simple-graph-card document 10) ASoC: add simple-graph-card support 11) ASoC: add simple-graph-scu-card document 12) ASoC: add simple-graph-scu-card support .../bindings/sound/simple-graph-card.txt | 65 +++ .../bindings/sound/simple-graph-scu-card.txt | 65 +++ drivers/of/base.c | 160 ++++++- include/linux/of_graph.h | 59 +++ include/sound/simple_card_utils.h | 19 + sound/soc/generic/Kconfig | 15 + sound/soc/generic/Makefile | 4 + sound/soc/generic/simple-card-utils.c | 98 ++++- sound/soc/generic/simple-card.c | 2 +- sound/soc/generic/simple-graph-card.c | 462 +++++++++++++++++++++ sound/soc/generic/simple-graph-scu-card.c | 417 +++++++++++++++++++ sound/soc/generic/simple-scu-card.c | 2 +- 12 files changed, 1355 insertions(+), 13 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/simple-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-graph-scu-card.txt create mode 100644 sound/soc/generic/simple-graph-card.c create mode 100644 sound/soc/generic/simple-graph-scu-card.c -- 1.9.1

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[alsa-devel] [PATCH 00/19] ASoC: Convert to module_i2c/spi_driver()
by Kuninori Morimoto 21 Nov '16

21 Nov '16

Hi Mark These patches uses module_i2c/spi_driver() on each driver to cleanup driver entries. I confirmed compile test only. 1) - 4): remove #ifdef CONFIG_I2C, add depends on CONFIG_I2C, and uses module_i2c_driver 5) - 19): uses module_i2c/spi_driver() and removed module_init/exit() Kuninori Morimoto (19): 1) ASoC: uda1380: Convert to module_i2c_driver() 2) ASoC: wm9081: Convert to module_i2c_driver() 3) ASoC: wm8580: Convert to module_i2c_driver() 4) ASoC: wm8523: Convert to module_i2c_driver() 5) ASoC: wm8510: Convert to module_i2c/spi_driver() 6) ASoC: wm8995: Convert to module_i2c/spi_driver() 7) ASoC: wm8988: Convert to module_i2c/spi_driver() 8) ASoC: wm8985: Convert to module_i2c/spi_driver() 9) ASoC: wm8983: Convert to module_i2c/spi_driver() 10) ASoC: wm8900: Convert to module_i2c/spi_driver() 11) ASoC: wm8776: Convert to module_i2c/spi_driver() 12) ASoC: wm8753: Convert to module_i2c/spi_driver() 13) ASoC: wm8750: Convert to module_i2c/spi_driver() 14) ASoC: wm8741: Convert to module_i2c/spi_driver() 15) ASoC: wm8731: Convert to module_i2c/spi_driver() 16) ASoC: wm8737: Convert to module_i2c/spi_driver() 17) ASoC: wm8728: Convert to module_i2c/spi_driver() 18) ASoC: wm8711: Convert to module_i2c/spi_driver() 19) ASoC: da7210: Convert to module_i2c/spi_driver() sound/soc/codecs/Kconfig | 2 ++ sound/soc/codecs/da7210.c | 30 ++---------------------------- sound/soc/codecs/uda1380.c | 22 +--------------------- sound/soc/codecs/wm8510.c | 34 ++-------------------------------- sound/soc/codecs/wm8523.c | 24 +----------------------- sound/soc/codecs/wm8580.c | 25 +------------------------ sound/soc/codecs/wm8711.c | 34 ++-------------------------------- sound/soc/codecs/wm8728.c | 34 ++-------------------------------- sound/soc/codecs/wm8731.c | 34 ++-------------------------------- sound/soc/codecs/wm8737.c | 34 ++-------------------------------- sound/soc/codecs/wm8741.c | 34 ++-------------------------------- sound/soc/codecs/wm8750.c | 34 ++-------------------------------- sound/soc/codecs/wm8753.c | 34 ++-------------------------------- sound/soc/codecs/wm8776.c | 34 ++-------------------------------- sound/soc/codecs/wm8900.c | 34 ++-------------------------------- sound/soc/codecs/wm8983.c | 35 ++--------------------------------- sound/soc/codecs/wm8985.c | 35 ++--------------------------------- sound/soc/codecs/wm8988.c | 35 ++--------------------------------- sound/soc/codecs/wm8995.c | 37 ++----------------------------------- sound/soc/codecs/wm9081.c | 2 -- 20 files changed, 35 insertions(+), 552 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: wm_adsp: Only write shutdown controls for active firmwares
by Charles Keepax 21 Nov '16

21 Nov '16

The control list may contain shutdown controls for firmwares that are not currently active, attempting to write this will at best fail. To avoid this issue we skip any control that is not active. Fixes: commit f4f0c4c60c39 ("ASoC: wm_adsp: Signal firmware shutdown through event control") Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4fb6e2f..6ccb731 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1194,6 +1194,9 @@ static void wm_adsp_signal_event_controls(struct wm_adsp *dsp, if (ctl->type != WMFW_CTL_TYPE_HOSTEVENT) continue; + if (!ctl->enabled) + continue; + ret = wm_coeff_write_acked_control(ctl, event); if (ret) adsp_warn(dsp, -- 2.1.4

2 1

[alsa-devel] [RFC 0/4] ALSA controls management using index/device/sub-devices fields
by Arnaud Pouliquen 21 Nov '16

21 Nov '16

I tried to summarize my observation based on discussions concerning the instantiation of the ALSA controls, initiate in following thread: [PATCH] ASoC: core: allow control index different from 0 https://www.spinics.net/lists/ALSA-devel/msg55415.html My test environment: sti platform, 3 playback devices with associated instances of controls card 0: STIB2120 [STI-B2120], device 0: Uni Player #0 (HDMI)-i2s-hifi numid=1,iface=PCM,name='ELD' numid=3,iface=PCM,name='IEC958 Playback Default' numid=2,iface=PCM,name='PCM Playback Oversampling Freq. Adjustment' card 0: STIB2120 [STI-B2120], device 1: Uni Player #2 (DAC)-sas-dai-dac numid=4,iface=PCM,name='PCM Playback Oversampling Freq. Adjustment',device=2 card 0: STIB2120 [STI-B2120], device 2: Uni Player #3 (SPDIF)-sas-dai- spdif-out numid=6,iface=PCM,name='IEC958 Playback Default',device=2 numid=5,iface=PCM,name='PCM Playback Oversampling Freq. Adjustment',device=3 1) From user land point of view 1-1) How to Instantiate generic ALSA control The point is the handling of multi instance of generic ALSA controls. In this case "prefix" can not be used. Controls have to be identified by a combination of device/sub-device/index Examples of controls that seem generics: iface=PCM,name='Capture Channel Map' iface=PCM,name='Playback Channel Map' iface=PCM,name='ELD' iface=MIXER,name='Master Playback Switch' iface=MIXER,name='Master Playback Volume' iface=MIXER,name='Capture Switch' iface=MIXER,name='Capture Volume' iface=MIXER/PCM,name='IEC958 Playback Con Mask' iface=MIXER/PCM,name='IEC958 Playback Pro Mask' iface=MIXER/PCM,name='IEC958 Playback Default' iface=MIXER/PCM,name='IEC958 Playback Switch' 1-2) Different ways of using control instantiation Here is a short benchmark on applications using ALSA controls: - iecset and "amixer scontrols" base control instantiation on "index". - GStreamer and pulseaudio seems based on "device" (through the ALSA card conf files). - "amixer controls" support both. I supposse that this is the reason why "device" and "index fields are both fixed in HDA-intel.cong aligned with driver. 1-3) Questions: - If i consider description provided by Takashi Sakamoto in thread: https://www.spinics.net/lists/ALSA-devel/msg55451.html Should we always apply this rules? - MIXER control type: as it is not linked to PCM device but card, "index" is used to instantiate control. - PCM control type: as it is linked to PCM device, "device/subdevice" is used to instantiate control. Or could we consider that a control can be instantiated using device/subdevice fields and/or index fields? - Concerning IEC controls, should we consider them as PCM or MIXER type controls? In current implementations both are used... 2) From driver point of view 2-1) None SOC drivers: Today solution seems to base indexation on both "index" and "device" number. Device indexation seems not linked to PCM device number Example in ea9b43addc4d ("ALSA: hda - Fix broken workaround for HDMI/SPDIF conflicts"). http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=ea9b… => The drawback is that each driver has to implement it. Could be nice to have in generic code.... 2-2) SOC drivers: - it is not possible to use "index" using helper functions as snd_soc_cnew forces index to 0. Solution could be to implement in soc_core as done in HDA driver. => duplication of the code in ASoC and None ASoC drivers. - Relation chip between control and PCM device. if rules mentioned in 1-3 should be respected, need to link control to PCM device. This point seems quite tricky... At least DAI controls can be simply linked to PCM device during DAI-link probing. 3) Patches proposed: Based on these observations, here are some patches that i tested in my environment. There are complementary, and could answer to some limitations mentioned above. 3-1) Alsa-utils patch - iecset: allow to select control with device and sub-device numbers This patch allows to access to 2 iec controls differentiated by device/sub-devices numbers => For me, this patch is mandatory to be able to address the ASoC IEC controls, in case of no fix is implemented to allows index field update in ASoC. 3-2) Alsa driver patches - ASoC: core: allow PCM control binding to PCM device Add relationship between DAIs PCM controls and PCM device. - ALSA: control: increment index field for duplicated control. Generic implementation of the patch proposed in HDA driver (http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=ea9b…) - ASoC: sti: use bind_pcm_ctl implementation of bind_pcm_ctl for sti driver. Best Regards, Arnaud -- 1.9.1

4 19

[alsa-devel] [PATCH] ASoC: mioa701_wm9713: add missing white space in dev_err message
by Colin King 19 Nov '16

19 Nov '16

From: Colin Ian King <colin.king(a)canonical.com> There is a missing whitespace in the dev_err message between "will" and "lead". Add the whitespace. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/pxa/mioa701_wm9713.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index d1661fa..0fe0abe 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -187,7 +187,7 @@ static int mioa701_wm9713_probe(struct platform_device *pdev) mioa701.dev = &pdev->dev; rc = devm_snd_soc_register_card(&pdev->dev, &mioa701); if (!rc) - dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" + dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will " "lead to overheating and possible destruction of your device." " Do not use without a good knowledge of mio's board design!\n"); return rc; -- 2.10.2

4 4

[alsa-devel] [PATCH] ASoC: intel: mfld: Make static string arrays 'const 'char * const []'
by Lars-Peter Clausen 18 Nov '16

18 Nov '16

const char * const [] is the preferred type for static string arrays since this states explicitly that the individual entries are not going to be changed. Due to limitations in the ASoC API it was not possible to use it for enum text arrays. Commit 87023ff74b23 ("ASoC: Declare const properly for enum texts") changed this, but most drivers still use 'const char * []' as the type for their enum text arrays. Change these occurrences of 'static * const char * []' to 'static const char * const []'. The conversion was done automatically using the following coccinelle semantic patch: // <smpl> @disable optional_qualifier@ identifier s; @@ static -const char * +const char * const s[] = ...; // </smpl> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/intel/boards/mfld_machine.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c index 34f46c7..4e08885 100644 --- a/sound/soc/intel/boards/mfld_machine.c +++ b/sound/soc/intel/boards/mfld_machine.c @@ -81,9 +81,9 @@ static struct snd_soc_jack_zone mfld_zones[] = { }; /* sound card controls */ -static const char *headset_switch_text[] = {"Earpiece", "Headset"}; +static const char * const headset_switch_text[] = {"Earpiece", "Headset"}; -static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"}; +static const char * const lo_text[] = {"Vibra", "Headset", "IHF", "None"}; static const struct soc_enum headset_enum = SOC_ENUM_SINGLE_EXT(2, headset_switch_text); -- 2.1.4

3 2

[alsa-devel] [PATCH v3 0/3] ASoC driver for the TSE-850
by Peter Rosin 18 Nov '16

18 Nov '16

Hi! v2 -> v3 changes: - update the preparatory atmel_ssc_dai patch with error checking and fallback handling of the CMR_DIV divider in addition to the new fallback handling of TCMR_PERIOD/RCMR_PERIOD from v2. v1 -> v2 changes: - new subject for the bindings patch to make it easier to find - move the driver to the sound/soc/atmel directory - add cached values for add/loop1/loop2 to avoid gets from output gpios - use _cansleep when updating gpios - add comment on how the regulator voltage is mapped to the ana enum - drop the .hw_params hook that did set the cpu dai divider and... - ...add a preparatory patch that does this in the cpu dai driver instead - drop .init and set the dapm routes directly in the card struct instead The TSE-850 is an FM Transmitter Station Equipment, designed to generate baseband signals for FM, mainly the DARC subcarrier, but other signals are also possible. This adds a driver for the "sound" bits of the device (quoted since it is normally not used for normal sound output, but that works too of course). I have not provided a patch to add axentia as a devicetree vendor prefix, since such a patch is already pending in an IIO series [1] that seems close to being accepted. However, there are a couple of points that I'm not 100% satisfied with for this driver. First, I do not know how to describe the relays that control if the IN1/IN2 signals are directly routed towards OUT1/OUT2 or if they are routed to the "add" switch. The dapm routing treats this as if the IN1/IN2 signals are always routed to both the "add" switch and to the muxes feeding OUT1/OUT2. This is fine with me since nothing is powered in those sections anyway, so what dapm thinks does not really matter. But it is a wart all the same. Second, there's my comment in tse850_put_mix() when the "add" switch is updated. I believe this update should really happen as a side effect of the call to snd_soc_dapm_mixer_update_power(), so that it happens at the right point compared to other stuff that is powered. But I do not know how to hook that up and instead I flip the switch before the call since it doesn't really matter. I.e., any noise resulting from this badness is negligeble in practice. Cheers, Peter [1] http://lwn.net/Articles/705931 Peter Rosin (3): ASoC: atmel_ssc_dai: if not provided, default to sensible dividers ASoC: tse850: document axentia,tse850-pcm5142 bindings ASoC: atmel: tse850: add ASoC driver for the Axentia TSE-850 .../bindings/sound/axentia,tse850-pcm5142.txt | 88 ++++ MAINTAINERS | 7 + sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel_ssc_dai.c | 83 +++- sound/soc/atmel/atmel_ssc_dai.h | 1 + sound/soc/atmel/tse850-pcm5142.c | 472 +++++++++++++++++++++ 7 files changed, 656 insertions(+), 7 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt create mode 100644 sound/soc/atmel/tse850-pcm5142.c -- 2.1.4

3 6

[alsa-devel] [PATCH v4 0/3] ASoC driver for the TSE-850
by Peter Rosin 18 Nov '16

18 Nov '16

Hi! v3 -> v4 changes: - changed heading of the devicetree bindings as requested by Rob - add ack from Rob on patch 2/3 - rebased onto v4.9-rc5 v2 -> v3 changes: - update the preparatory atmel_ssc_dai patch with error checking and fallback handling of the CMR_DIV divider in addition to the new fallback handling of TCMR_PERIOD/RCMR_PERIOD from v2. v1 -> v2 changes: - new subject for the bindings patch to make it easier to find - move the driver to the sound/soc/atmel directory - add cached values for add/loop1/loop2 to avoid gets from output gpios - use _cansleep when updating gpios - add comment on how the regulator voltage is mapped to the ana enum - drop the .hw_params hook that did set the cpu dai divider and... - ...add a preparatory patch that does this in the cpu dai driver instead - drop .init and set the dapm routes directly in the card struct instead The TSE-850 is an FM Transmitter Station Equipment, designed to generate baseband signals for FM, mainly the DARC subcarrier, but other signals are also possible. This adds a driver for the "sound" bits of the device (quoted since it is normally not used for normal sound output, but that works too of course). I have not provided a patch to add axentia as a devicetree vendor prefix, since such a patch is already pending in an IIO series [1] that is on its merry way into -next. However, there are a couple of points that I'm not 100% satisfied with for this driver. First, I do not know how to describe the relays that control if the IN1/IN2 signals are directly routed towards OUT1/OUT2 or if they are routed to the "add" switch. The dapm routing treats this as if the IN1/IN2 signals are always routed to both the "add" switch and to the muxes feeding OUT1/OUT2. This is fine with me since nothing is powered in those sections anyway, so what dapm thinks does not really matter. But it is a wart all the same. Second, there's my comment in tse850_put_mix() when the "add" switch is updated. I believe this update should really happen as a side effect of the call to snd_soc_dapm_mixer_update_power(), so that it happens at the right point compared to other stuff that is powered. But I do not know how to hook that up and instead I flip the switch before the call since it doesn't really matter. I.e., any noise resulting from this badness is negligeble in practice. Cheers, Peter [1] http://lwn.net/Articles/705931 Peter Rosin (3): ASoC: atmel_ssc_dai: if not provided, default to sensible dividers ASoC: tse850: document axentia,tse850-pcm5142 bindings ASoC: atmel: tse850: add ASoC driver for the Axentia TSE-850 .../bindings/sound/axentia,tse850-pcm5142.txt | 88 ++++ MAINTAINERS | 7 + sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel_ssc_dai.c | 83 +++- sound/soc/atmel/atmel_ssc_dai.h | 1 + sound/soc/atmel/tse850-pcm5142.c | 472 +++++++++++++++++++++ 7 files changed, 656 insertions(+), 7 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt create mode 100644 sound/soc/atmel/tse850-pcm5142.c -- 2.1.4

2 4

[alsa-devel] [PATCH v2 0/3] ASoC driver for the TSE-850
by Peter Rosin 18 Nov '16

18 Nov '16

Hi! v1 -> v2 changes: - new subject for the bindings patch to make it easier to find - move the driver to the sound/soc/atmel directory - add cached values for add/loop1/loop2 to avoid gets from output gpios - use _cansleep when updating gpios - add comment on how the regulator voltage is mapped to the ana enum - drop the .hw_params hook that did set the cpu dai divider and... - ...add a preparatory patch that does this in the cpu dai driver instead - drop .init and set the dapm routes directly in the card struct instead The TSE-850 is an FM Transmitter Station Equipment, designed to generate baseband signals for FM, mainly the DARC subcarrier, but other signals are also possible. This adds a driver for the "sound" bits of the device (quoted since it is normally not used for normal sound output, but that works too of course). I have not provided a patch to add axentia as a devicetree vendor prefix, since such a patch is already pending in an IIO series [1] that seems close to being accepted. However, there are a couple of points that I'm not 100% satisfied with for this driver. First, I do not know how to describe the relays that control if the IN1/IN2 signals are directly routed towards OUT1/OUT2 or if they are routed to the "add" switch. The dapm routing treats this as if the IN1/IN2 signals are always routed to both the "add" switch and to the muxes feeding OUT1/OUT2. This is fine with me since nothing is powered in those sections anyway, so what dapm thinks does not really matter. But it is a wart all the same. Second, there's my comment in tse850_put_mix() when the "add" switch is updated. I believe this update should really happen as a side effect of the call to snd_soc_dapm_mixer_update_power(), so that it happens at the right point compared to other stuff that is powered. But I do not know how to hook that up and instead I flip the switch before the call since it doesn't really matter. I.e., any noise resulting from this badness is negligeble in practice. Cheers, Peter [1] http://lwn.net/Articles/705931 Peter Rosin (3): ASoC: atmel_ssc_dai: if not provided, default to sensible TCMR/RCMR periods ASoC: tse850: document axentia,tse850-pcm5142 bindings ASoC: atmel: tse850: add ASoC driver for the Axentia TSE-850 .../bindings/sound/axentia,tse850-pcm5142.txt | 88 ++++ MAINTAINERS | 7 + sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel_ssc_dai.c | 29 +- sound/soc/atmel/atmel_ssc_dai.h | 1 + sound/soc/atmel/tse850-pcm5142.c | 472 +++++++++++++++++++++ 7 files changed, 603 insertions(+), 6 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt create mode 100644 sound/soc/atmel/tse850-pcm5142.c -- 2.1.4

2 4

[alsa-devel] [PATCH 1/2] ASoC: rt5665: Fix missing mutex_unlock in rt5665_calibrate
by Axel Lin 18 Nov '16

18 Nov '16

Reported-by: Julia Lawall <julia.lawall(a)lip6.fr> Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5665.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 34254fd..7dfaaf1 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4587,7 +4587,7 @@ static void rt5665_calibrate(struct rt5665_priv *rt5665) pr_err("HP Calibration Failure\n"); regmap_write(rt5665->regmap, RT5665_RESET, 0); regcache_cache_bypass(rt5665->regmap, false); - return; + goto out_unlock; } count++; @@ -4606,7 +4606,7 @@ static void rt5665_calibrate(struct rt5665_priv *rt5665) pr_err("MONO Calibration Failure\n"); regmap_write(rt5665->regmap, RT5665_RESET, 0); regcache_cache_bypass(rt5665->regmap, false); - return; + goto out_unlock; } count++; @@ -4621,6 +4621,7 @@ static void rt5665_calibrate(struct rt5665_priv *rt5665) regmap_write(rt5665->regmap, RT5665_BIAS_CUR_CTRL_8, 0xa602); regmap_write(rt5665->regmap, RT5665_ASRC_8, 0x0120); +out_unlock: mutex_unlock(&rt5665->calibrate_mutex); } -- 2.9.3

2 3

[alsa-devel] [PATCH 1/2] ASoC: rt5514: Remove superfluous linux/kthread.h inclusion
by Takashi Iwai 18 Nov '16

18 Nov '16

It's nowhere used in this driver code. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/rt5514-spi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 09103aab0cb2..0901e25d6db6 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -20,7 +20,6 @@ #include <linux/slab.h> #include <linux/gpio.h> #include <linux/sched.h> -#include <linux/kthread.h> #include <linux/uaccess.h> #include <linux/miscdevice.h> #include <linux/regulator/consumer.h> -- 2.10.2

2 3

[alsa-devel] [patch] ASoC: rt5665: Unlock on error in rt5665_calibrate()
by Dan Carpenter 18 Nov '16

18 Nov '16

We should probably unlock before returning here. Fixes: 33ada14a26c8 ("ASoC: add rt5665 codec driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 34254fd..25bd01b 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4587,7 +4587,7 @@ static void rt5665_calibrate(struct rt5665_priv *rt5665) pr_err("HP Calibration Failure\n"); regmap_write(rt5665->regmap, RT5665_RESET, 0); regcache_cache_bypass(rt5665->regmap, false); - return; + goto unlock; } count++; @@ -4606,7 +4606,7 @@ static void rt5665_calibrate(struct rt5665_priv *rt5665) pr_err("MONO Calibration Failure\n"); regmap_write(rt5665->regmap, RT5665_RESET, 0); regcache_cache_bypass(rt5665->regmap, false); - return; + goto unlock; } count++; @@ -4620,7 +4620,7 @@ static void rt5665_calibrate(struct rt5665_priv *rt5665) regmap_write(rt5665->regmap, RT5665_BIAS_CUR_CTRL_8, 0xa602); regmap_write(rt5665->regmap, RT5665_ASRC_8, 0x0120); - +unlock: mutex_unlock(&rt5665->calibrate_mutex); }

2 1

[alsa-devel] [PATCH] ASoC: Intel: boards: Use FS as nau8825 sysclk in nau88l25_max98357a machine
by Grace Kao 18 Nov '16

18 Nov '16

256FS sysclk gives the best audio quality per nau8825 datasheet. Loud headphone pop happens if there is no sysclk during nau8825 playback power up sequence. Currently Skylake does not output MCLK/FS when the back-end DAI op hw_param is called, so we cannot switch to MCLK/FS in hw_param. This patch reduces pop by leting nau8825 keep using its internal VCO clock during widget power up sequence, until SNDRV_PCM_TRIGGER_START when MCLK/FS is available. --- diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 0f89602..6ad8f52 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -70,14 +70,7 @@ return -EIO; } - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { + if (!SND_SOC_DAPM_EVENT_ON(event)) { ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); if (ret < 0) { @@ -272,25 +265,7 @@ .startup = skl_fe_startup, }; -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); - - return ret; -} - -static struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, -}; static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) @@ -306,6 +281,33 @@ return 0; } +static int skylake_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + } + + return ret; +} + +static struct snd_soc_ops skylake_nau8825_ops = { + .trigger = skylake_nau8825_trigger, +}; + static unsigned int channels_dmic[] = { 2, 4, };

2 1

[alsa-devel] [PATCH] crec: Correct name of output binary to crec
by Richard Fitzgerald 18 Nov '16

18 Nov '16

For some reason when the build system was converted to automake by this patch "Convert the build system to autotools" the name of the output binary for crec was changed from 'crec' to 'crecord'. This patch corrects it back to 'crec' Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- src/utils/Makefile.am | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/src/utils/Makefile.am b/src/utils/Makefile.am index 5f685e3..1967d92 100644 --- a/src/utils/Makefile.am +++ b/src/utils/Makefile.am @@ -1,11 +1,11 @@ -bin_PROGRAMS = cplay crecord +bin_PROGRAMS = cplay crec cplay_SOURCES = cplay.c -crecord_SOURCES = crec.c +crec_SOURCES = crec.c cplay_CFLAGS = -I$(top_srcdir)/include -crecord_CFLAGS = -I$(top_srcdir)/include +crec_CFLAGS = -I$(top_srcdir)/include cplay_LDADD = $(top_builddir)/src/lib/libtinycompress.la -crecord_LDADD = $(top_builddir)/src/lib/libtinycompress.la +crec_LDADD = $(top_builddir)/src/lib/libtinycompress.la -- 2.1.4

2 5

[alsa-devel] [PATCH] CHROMIUM: ASoC: Intel: boards: Use FS as nau8825 sysclk in nau88l25_max98357a machine
by Grace Kao 18 Nov '16

18 Nov '16

256FS sysclk gives the best audio quality per nau8825 datasheet. Loud headphone pop happens if there is no sysclk during nau8825 playback power up sequence. Currently Skylake does not output MCLK/FS when the back-end DAI op hw_param is called, so we cannot switch to MCLK/FS in hw_param. This patch reduces pop by leting nau8825 keep using its internal VCO clock during widget power up sequence, until SNDRV_PCM_TRIGGER_START when MCLK/FS is available. This is a temporary fix before Intel ADSP can output MCLK/FS early. BUG=chrome-os-partner:50471 TEST=`emerge-sentry sys-kernel/chromeos-kernel-3_18`. Verified a local build that could reduce the pop sound. Change-Id: I4943fffa8eb1c874cc66b50f0897092bf48dd579 Signed-off-by: Grace Kao <grace.kao(a)intel.com> Reviewed-on: https://chromium-review.googlesource.com/333180 Reviewed-by: Ben Zhang <benzh(a)chromium.org> Series-to: alsa-devel(a)alsa-project.org Series-cc: Vinod.koul(a)intel.com --- diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 0f89602..6ad8f52 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -70,14 +70,7 @@ return -EIO; } - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { + if (!SND_SOC_DAPM_EVENT_ON(event)) { ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); if (ret < 0) { @@ -272,25 +265,7 @@ .startup = skl_fe_startup, }; -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); - - return ret; -} - -static struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, -}; static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) @@ -306,6 +281,33 @@ return 0; } +static int skylake_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + } + + return ret; +} + +static struct snd_soc_ops skylake_nau8825_ops = { + .trigger = skylake_nau8825_trigger, +}; + static unsigned int channels_dmic[] = { 2, 4, };

3 3

[alsa-devel] [PATCH] ALSA: bebob: compare whole string for model name so that readers are not confused
by Takashi Sakamoto 18 Nov '16

18 Nov '16

A modalias of 'ieee1394:ven00000D6Cmo00010060sp' hits units for M-Audio FireWire Audiophile only. However the unit has two states relevant to loaded firmware. Initial firmware returns 'FW Audiophile Bootloader', while functional firmware returns 'FW Audiophile'. ALSA bebob driver compares the units' model name to strings of 24 characters by the first 15 characters. This is shorter way to differentiate these two states but confusing to readers. This commit improves the code for this point. Kernel stack is consumed more in call of check_audiophile_booted() by a bit. With initial firmware: $ ./linux-firewire-utils/src/crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 04276da6 bus_info_length 4, crc_length 39, crc 28070 404 31333934 bus_name "1394" 408 006481f2 irmc 0, cmc 0, isc 0, bmc 0, pmc 0, cyc_clk_acc 100, max_rec 8 (512), max_rom 1, gen 15, spd 2 (S400) 40c 000d6c03 company_id 000d6c | 410 102b7e2e device_id 03102b7e2e | EUI-64 000d6c03102b7e2e root directory ----------------------------------------------------------------- 414 00067414 directory_length 6, crc 29716 418 0c0083c0 node capabilities per IEEE 1394 41c 03000d6c vendor 420 81000009 --> descriptor leaf at 444 424 17010060 model 428 8100000c --> descriptor leaf at 458 42c d1000001 --> unit directory at 430 unit directory at 430 ----------------------------------------------------------------- 430 00049da1 directory_length 4, crc 40353 (should be 48611) 434 1200a02d specifier id: 1394 TA 438 13014001 version: Vender Unique and AV/C 43c 17010060 model 440 8100000f --> descriptor leaf at 47c descriptor leaf at 444 ----------------------------------------------------------------- 444 0004073e leaf_length 4, crc 1854 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 4d2d4155 "M-AU" 454 44494f00 "DIO" descriptor leaf at 458 ----------------------------------------------------------------- 458 00086f21 leaf_length 8, crc 28449 45c 00000000 textual descriptor 460 00000000 minimal ASCII 464 46572041 "FW A" 468 7564696f "udio" 46c 7068696c "phil" 470 6520426f "e Bo" 474 6f746c6f "otlo" 478 61646572 "ader" descriptor leaf at 47c ----------------------------------------------------------------- 47c 00086f21 leaf_length 8, crc 28449 480 00000000 textual descriptor 484 00000000 minimal ASCII 488 46572041 "FW A" 48c 7564696f "udio" 490 7068696c "phil" 494 6520426f "e Bo" 498 6f746c6f "otlo" 49c 61646572 "ader" With functional firmware: $ ./linux-firewire-utils/src/crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 042de16f bus_info_length 4, crc_length 45, crc 57711 404 31333934 bus_name "1394" 408 f0648122 irmc 1, cmc 1, isc 1, bmc 1, pmc 0, cyc_clk_acc 100, max_rec 8 (512), max_rom 1, gen 2, spd 2 (S400) 40c 000d6c03 company_id 000d6c | 410 002b7e2e device_id 03002b7e2e | EUI-64 000d6c03002b7e2e root directory ----------------------------------------------------------------- 414 0009dac4 directory_length 9, crc 56004 418 0400000d hardware version 41c 0c0083c0 node capabilities per IEEE 1394 420 03000d6c vendor 424 81000012 --> descriptor leaf at 46c 428 17010060 model 42c 81000015 --> descriptor leaf at 480 430 13ffffff version 434 d1000002 --> unit directory at 43c 438 d4000006 --> dependent info directory at 450 unit directory at 43c ----------------------------------------------------------------- 43c 00041eb9 directory_length 4, crc 7865 440 1200a02d specifier id: 1394 TA 444 13014001 version: Vender Unique and AV/C 448 17010060 model 44c 81000014 --> descriptor leaf at 49c dependent info directory at 450 ----------------------------------------------------------------- 450 000637c7 directory_length 6, crc 14279 454 120007f5 specifier id 458 13000001 version 45c 3affffc7 (immediate value) 460 3b100000 (immediate value) 464 3cffffc7 (immediate value) 468 3d600000 (immediate value) descriptor leaf at 46c ----------------------------------------------------------------- 46c 0004b8e4 leaf_length 4, crc 47332 470 00000000 textual descriptor 474 00000000 minimal ASCII 478 4d2d4175 "M-Au" 47c 64696f00 "dio" descriptor leaf at 480 ----------------------------------------------------------------- 480 0006194b leaf_length 6, crc 6475 484 00000000 textual descriptor 488 00000000 minimal ASCII 48c 46572041 "FW A" 490 7564696f "udio" 494 7068696c "phil" 498 65000000 "e" descriptor leaf at 49c ----------------------------------------------------------------- 49c 0006194b leaf_length 6, crc 6475 4a0 00000000 textual descriptor 4a4 00000000 minimal ASCII 4a8 46572041 "FW A" 4ac 7564696f "udio" 4b0 7068696c "phil" 4b4 65000000 "e" Reported-by: Nicolas Iooss <nicolas.iooss_linux(a)m4x.org> Reported-by: Joe Perches <joe(a)perches.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 3469ac14..730ea91 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -172,12 +172,12 @@ get_saffire_spec(struct fw_unit *unit) static bool check_audiophile_booted(struct fw_unit *unit) { - char name[24] = {0}; + char name[28] = {0}; if (fw_csr_string(unit->directory, CSR_MODEL, name, sizeof(name)) < 0) return false; - return strncmp(name, "FW Audiophile Bootloader", 15) != 0; + return strncmp(name, "FW Audiophile Bootloader", 24) != 0; } static void -- 2.9.3

2 1

[alsa-devel] [PATCH] sound: Add dependency on DMA
by Florian Fainelli 18 Nov '16

18 Nov '16

Architectures like m32r do not have a proper DMA-API implementation, fixes COMPILE_TEST linking failures for the sounds subsystem. Signed-off-by: Florian Fainelli <f.fainelli(a)gmail.com> --- sound/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/Kconfig b/sound/Kconfig index 5a240e050ae6..a768704d2f47 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -60,6 +60,7 @@ if !M68K && !UML menuconfig SND tristate "Advanced Linux Sound Architecture" + depends on HAS_DMA help Say 'Y' or 'M' to enable ALSA (Advanced Linux Sound Architecture), the new base sound system. -- 2.9.3

2 4

[alsa-devel] [PATCH] CHROMIUM: ASoC: Intel: boards: Use FS as nau8825 sysclk in nau88l25_max98357a machine
by Grace Kao 18 Nov '16

18 Nov '16

256FS sysclk gives the best audio quality per nau8825 datasheet. Loud headphone pop happens if there is no sysclk during nau8825 playback power up sequence. Currently Skylake does not output MCLK/FS when the back-end DAI op hw_param is called, so we cannot switch to MCLK/FS in hw_param. This patch reduces pop by leting nau8825 keep using its internal VCO clock during widget power up sequence, until SNDRV_PCM_TRIGGER_START when MCLK/FS is available. This is a temporary fix before Intel ADSP can output MCLK/FS early. BUG=chrome-os-partner:50471 TEST=`emerge-sentry sys-kernel/chromeos-kernel-3_18`. Verified a local build that could reduce the pop sound. Change-Id: I4943fffa8eb1c874cc66b50f0897092bf48dd579 Signed-off-by: Grace Kao <grace.kao(a)intel.com> Reviewed-on: https://chromium-review.googlesource.com/333180 Reviewed-by: Ben Zhang <benzh(a)chromium.org> Series-to: alsa-devel(a)alsa-project.org Series-cc: Vinod.koul(a)intel.com --- diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 0f89602..6ad8f52 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -70,14 +70,7 @@ return -EIO; } - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { + if (!SND_SOC_DAPM_EVENT_ON(event)) { ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); if (ret < 0) { @@ -272,25 +265,7 @@ .startup = skl_fe_startup, }; -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); - - return ret; -} - -static struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, -}; static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) @@ -306,6 +281,33 @@ return 0; } +static int skylake_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + } + + return ret; +} + +static struct snd_soc_ops skylake_nau8825_ops = { + .trigger = skylake_nau8825_trigger, +}; + static unsigned int channels_dmic[] = { 2, 4, };

1 0

[alsa-devel] sound: bebop: strncmp length oddity
by Joe Perches 17 Nov '16

17 Nov '16

15 isn't the length of the string, is that really what's desired? linux/next/sound/firewire/bebob/bebop.c ----------------------------- static bool check_audiophile_booted(struct fw_unit *unit) { char name[24] = {0}; if (fw_csr_string(unit->directory, CSR_MODEL, name, sizeof(name)) < 0) return false; return strncmp(name, "FW Audiophile Bootloader", 15) != 0; }

4 3

[alsa-devel] [PATCH] sound: Add dependency on DMA
by Florian Fainelli 17 Nov '16

17 Nov '16

Architectures like m32r do not have a proper DMA-API implementation, fixes COMPILE_TEST linking failures for the sounds subsystem. Signed-off-by: Florian Fainelli <f.fainelli(a)gmail.com> --- sound/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/Kconfig b/sound/Kconfig index 5a240e050ae6..a768704d2f47 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -60,6 +60,7 @@ if !M68K && !UML menuconfig SND tristate "Advanced Linux Sound Architecture" + depends on HAS_DMA help Say 'Y' or 'M' to enable ALSA (Advanced Linux Sound Architecture), the new base sound system. -- 2.9.3

1 0

[alsa-devel] Accurate delay reporting from dshare
by Alan Young 17 Nov '16

17 Nov '16

Hi, When the kernel reports current (playback) delay via a call to snd_pcm_status() or snd_pcm_delay() for a normal hardware PCM, then the delay value reported is the sum of space used in the ring buffer plus any delay reported from the underlying runtime driver. snd_pcm_dshare_status() and snd_pcm_dshare_delay() discard this refinement and simply report the use of some ring buffer. Why does it do this and how could the reporting be improved? In particular I am struggling to understand the relationship between a dshare instance's (view of a) ring buffer and the slave (hw) PCM's ring buffer. I understand that the *slowptr* configuration item can be set to get more accurate position updates as part of this reporting but I get the feeling that this will not on its own solve the problem. If it did then one could simply use the delay value reported by the slave PCM. The context of these questions is using dshare to expose 4 logical stereo devices on a platform using an ARM-based SoC with a single 8-channel hardware driver. I need to get the same level of delay report accuracy as I would do without using dshare. Alan.

2 11

[alsa-devel] [PATCH v5 0/7] ARM: ASoC: drm: sun8i: Add DE2 HDMI audio and video
by Jean-Francois Moine 17 Nov '16

17 Nov '16

This patchset series adds HDMI audio and video support to the Allwinner sun8i SoCs which include the display engine 2 (DE2). A first submission in January for video on the H3 could not enter into the mainline kernel due to the lack of license headers in Allwinner's sources. Recently, an announce about Tina OS for the R series https://www.youtube.com/watch?v=h7KD-6HblAU was followed by the upload of a new linux-3.4 source tree https://github.com/tinalinux/linux-3.4 with files containing GPL headers. Well, I don't know if these sources are really from Allwinner, but anyway, this is the opportunity to propose a new version of my DRM HDMI driver. v5: - add overlay plane - add audio support - add support for the A83T - add back the HDMI driver - many bug fixes v4: - drivers/clk/sunxi/Makefile was missing (Emil Velikov) v3: - add the hardware cursor - simplify and fix the DE2 init sequences - generation for all SUNXI SoCs (Andre Przywara) v2: - remove the HDMI driver - remarks from Chen-Yu Tsai and Russell King - DT documentation added Jean-Francois Moine (7): drm: sunxi: Add a basic DRM driver for Allwinner DE2 ASoC: sunxi: Add a simple HDMI CODEC drm: sunxi: add DE2 HDMI support ASoC: sunxi: Add sun8i I2S driver ARM: dts: sun8i-h3: add HDMI audio and video nodes ARM: dts: sun8i-h3: Add HDMI audio and video to the Banana Pi M2+ ARM: dts: sun8i-h3: Add HDMI audio and video to the Orange PI 2 .../devicetree/bindings/display/sunxi/hdmi.txt | 52 ++ .../bindings/display/sunxi/sunxi-de2.txt | 83 ++ .../devicetree/bindings/sound/sun4i-i2s.txt | 38 +- arch/arm/boot/dts/sun8i-h3-bananapi-m2-plus.dts | 17 + arch/arm/boot/dts/sun8i-h3-orangepi-2.dts | 17 + arch/arm/boot/dts/sun8i-h3.dtsi | 67 ++ drivers/gpu/drm/Kconfig | 2 + drivers/gpu/drm/Makefile | 1 + drivers/gpu/drm/sunxi/Kconfig | 29 + drivers/gpu/drm/sunxi/Makefile | 9 + drivers/gpu/drm/sunxi/de2_crtc.c | 475 +++++++++++ drivers/gpu/drm/sunxi/de2_crtc.h | 63 ++ drivers/gpu/drm/sunxi/de2_de.c | 591 +++++++++++++ drivers/gpu/drm/sunxi/de2_drm.h | 47 ++ drivers/gpu/drm/sunxi/de2_drv.c | 378 +++++++++ drivers/gpu/drm/sunxi/de2_hdmi.c | 396 +++++++++ drivers/gpu/drm/sunxi/de2_hdmi.h | 40 + drivers/gpu/drm/sunxi/de2_hdmi_io.c | 927 +++++++++++++++++++++ drivers/gpu/drm/sunxi/de2_hdmi_io.h | 25 + drivers/gpu/drm/sunxi/de2_plane.c | 119 +++ include/sound/sunxi_hdmi.h | 23 + sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sunxi-hdmi.c | 106 +++ sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 3 + sound/soc/sunxi/sun8i-i2s.c | 700 ++++++++++++++++ 27 files changed, 4222 insertions(+), 5 deletions(-) create mode 100644 Documentation/devicetree/bindings/display/sunxi/hdmi.txt create mode 100644 Documentation/devicetree/bindings/display/sunxi/sunxi-de2.txt create mode 100644 drivers/gpu/drm/sunxi/Kconfig create mode 100644 drivers/gpu/drm/sunxi/Makefile create mode 100644 drivers/gpu/drm/sunxi/de2_crtc.c create mode 100644 drivers/gpu/drm/sunxi/de2_crtc.h create mode 100644 drivers/gpu/drm/sunxi/de2_de.c create mode 100644 drivers/gpu/drm/sunxi/de2_drm.h create mode 100644 drivers/gpu/drm/sunxi/de2_drv.c create mode 100644 drivers/gpu/drm/sunxi/de2_hdmi.c create mode 100644 drivers/gpu/drm/sunxi/de2_hdmi.h create mode 100644 drivers/gpu/drm/sunxi/de2_hdmi_io.c create mode 100644 drivers/gpu/drm/sunxi/de2_hdmi_io.h create mode 100644 drivers/gpu/drm/sunxi/de2_plane.c create mode 100644 include/sound/sunxi_hdmi.h create mode 100644 sound/soc/codecs/sunxi-hdmi.c create mode 100644 sound/soc/sunxi/sun8i-i2s.c -- 2.10.1

7 37

[alsa-devel] [patch] ASoC: sh: dma-sh7760: fix a debug printf
by Dan Carpenter 17 Nov '16

17 Nov '16

We intended to print a hex here, but the code will print a base 10 with an x on the end. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 8fad444..c28ee1a 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -204,7 +204,7 @@ static int camelot_prepare(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; - pr_debug("PCM data: addr 0x%08ulx len %d\n", + pr_debug("PCM data: addr 0x%08lx len %d\n", (u32)runtime->dma_addr, runtime->dma_bytes); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {

2 2

[alsa-devel] Mic detection on Mackbook Pro 12,1 (CS4208)
by Paulo Fidalgo 17 Nov '16

17 Nov '16

Hi, After reporting the issue on bugzilla (https://bugzilla.kernel.org/show_bug.cgi?id=155601) I felt I could do more in order to try to get a patch to fix the issue. Basically when I plug an headset(mic+headphone), it only detects the headphone not the mic. Since I assumed the "memory address" of the 0x18 pin is wrong, I've tried to change some values in order to get it working. I have tries both ways: 1. By changing the kernel souce(sound/pci/hda/patch_cirrus): in the function static void cs4208_fixup_mac(struct hda_codec *codec, const struct hda_fixup *fix, int action) I've added an configuration to set the address of the 0x18. Then create a script to remove the module, compile, load and test via arecord if it's working. 2. use the user_pin_configs basically I've iterated for the range of memory addresses found in the card config and try to apply them: for (( num=0x002b4020; num <= 0x90a60100; num+=1 )) do pulseaudio -k ADDRESS=$(printf "0x%08x\n" $num) echo "0x18 $ADDRESS" | sudo tee /sys/class/sound/hwC1D0/user_pin_configs echo "0x1c 0x400000f0" | sudo tee /sys/class/sound/hwC1D0/user_pin_configs echo 1 | sudo tee /sys/class/sound/hwC1D0/reconfig pulseaudio -D arecord -d 5 -vvv -f dat /dev/null 2>&1 | grep -Po " 1\d+%" done but since I need to use the computer, I'm doing this in batches. Both address we're found by alsa-config as the pin addresses. Now my question is, is there a better way (without brute force) to get the right address? Could be anything else, other than the address? I also have MacOS installed, but from what I've found there's no way to know the address in MacOS, if there's a way, please point me ;) Best regards, Paulo Fidalgo

2 4

[alsa-devel] Sound Blaster Z works but ports are a mess
by Frederique 17 Nov '16

17 Nov '16

Dear mailing list individuals, I currently own a Sound Blaster Z from Creative. Testing on Linux Mint 18 64-bit with the stock Linux 4.4 kernel, ALSA 1.0.25+dfsg-0ubuntu5 and ALSA utilities 1.1.0-0ubuntu5. Output only works from the front panel headphone port when HP/Speaker is enabled. All audio enhancements available work as well. Input only works via the microphone port on the card itself, and all audio enhancements like voice clarity and voice transformation work as well. It seems only the ports are a mess. I do not know how to do any audio driver development so I not sure if it would require a lot of work to fix. Does the team have any of these cards? They come in Z, Zx and ZxR editions, with the last card being significantly different port-wise. Would it help if this hardware would be donated possibly by Creative themselves? A screenshot of alsamixer: http://i.imgur.com/OosqrER.png Currently I only have the headset connected to the front panel headphones port as I am awaiting an extension cable to arrive to be able to connect it to the microphone port in the back. A user called voron00 has been attempting development for the card series if it might help at all: http://forums.creative.com/showthread.php?t=741501 Sincerely yours, Frederique

1 0

[alsa-devel] [RFC 00/14] SoundWire bus driver
by Hardik Shah 17 Nov '16

17 Nov '16

Following RFC series adds SoundWire bus driver interface based on the MIPI SoundWire specification 1.1 The SoundWire protocol is a robust, scalable, low complexity, low power, low latency, two-pin (clock and data) multi-drop bus that allows for the transfer of multiple audio streams and embedded control/commands. SoundWire provides synchronization capabilities and supports both PCM and PDM, multichannel data, isochronous and asynchronous modes. SoundWire does borrow a number of concepts from existing interfaces such as HDAudio, AC97, SLIMbus, which already provide control/audio on the same wires, or legacy interfaces such as I2C/I2S, TDM, PDM. The capabilities of SoundWire make it unique however in that it can be implemented in peripherals such as microphones or amplifiers, mix PCM and PDM formats and enable clock scaling to reduce power consumption. More details about the SoundWire protocol can be obtained from MIPI website as listed below (accessible to members only). 1. http://mipi.org/learning-center/webinars 2. https://members.mipi.org/wg/All-Members/document/download/65078 3. https://members.mipi.org/wg/Contributors/document/70055 Hardik Shah (12): SoundWire: Add SoundWire bus driver documentation SoundWire: Add SoundWire stream documentation SoundWire: Add error handling and locking documentation SoundWire: Add device_id table for SoundWire bus SoundWire: Add SoundWire bus driver interfaces SoundWire: Add register/unregister APIs SoundWire: Add SoundWire Slaves register definitions SoundWire: Add API for Slave registers read/write SoundWire: Add support to handle Slave status change SoundWire: Add support for clock stop SoundWire: Add tracing for Slave register read/write regmap: SoundWire: Add regmap support for SoundWire bus Sanyog Kale (2): SoundWire: Add stream and port configuration SoundWire: Add support for SoundWire stream management Documentation/sound/alsa/sdw/error_handling.txt | 71 + Documentation/sound/alsa/sdw/locking.txt | 64 + Documentation/sound/alsa/sdw/stream.txt | 346 +++ Documentation/sound/alsa/sdw/summary.txt | 253 ++ MAINTAINERS | 12 + drivers/base/regmap/Kconfig | 3 + drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw.c | 240 ++ include/linux/mod_devicetable.h | 13 + include/linux/regmap.h | 37 + include/sound/sdw/sdw_registers.h | 277 ++ include/sound/sdw_bus.h | 905 ++++++ include/sound/sdw_master.h | 627 ++++ include/sound/sdw_slave.h | 563 ++++ include/trace/events/sdw.h | 209 ++ sound/Kconfig | 2 + sound/Makefile | 1 + sound/sdw/Kconfig | 6 + sound/sdw/Makefile | 1 + sound/sdw/sdw.c | 3749 +++++++++++++++++++++++ sound/sdw/sdw_priv.h | 811 +++++ sound/sdw/sdw_runtime.c | 2807 +++++++++++++++++ 22 files changed, 10998 insertions(+) create mode 100644 Documentation/sound/alsa/sdw/error_handling.txt create mode 100644 Documentation/sound/alsa/sdw/locking.txt create mode 100644 Documentation/sound/alsa/sdw/stream.txt create mode 100644 Documentation/sound/alsa/sdw/summary.txt create mode 100644 drivers/base/regmap/regmap-sdw.c create mode 100644 include/sound/sdw/sdw_registers.h create mode 100644 include/sound/sdw_bus.h create mode 100644 include/sound/sdw_master.h create mode 100644 include/sound/sdw_slave.h create mode 100644 include/trace/events/sdw.h create mode 100644 sound/sdw/Kconfig create mode 100644 sound/sdw/Makefile create mode 100644 sound/sdw/sdw.c create mode 100644 sound/sdw/sdw_priv.h create mode 100644 sound/sdw/sdw_runtime.c -- 1.7.9.5

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[alsa-devel] Lenovo Dock + Thinkpad X260 -> no HDMI audio output
by Norbert Preining 17 Nov '16

17 Nov '16

Dear all, (please cc) I am trying to get HDMI audio output via a Lenovo Dock Ultra in combination with a Lenovo Thinkpad X260. When plugging the HDMI device (screen) directly into the laptop, audio output can be selected in pavucontrol / Configuration, but when the laptop is connected via the dock all the HDMI devices appear as unplugged. I have googled a bit and found a similar case for T430: http://permalink.gmane.org/gmane.linux.alsa.user/36896 and checking the source code of snd_hda_intel it seems that options snd-hda-intel model=lenovo-dock should help, but indeed it didn't. I am running Debian/sid, on today's git kernel (4.9.0-rc5+). I attach the output of alsa-info. Please let me know whether I can try some more tweaks. Thanks a lot Norbert -- PREINING Norbert + TeX Live & Debian Developer + http://www.preining.info GPG: 0x860CDC13 fp: F7D8 A928 26E3 16A1 9FA0 ACF0 6CAC A448 860C DC13

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[alsa-devel] [MINUTES] Linux Plumbers Audio Mini Conference - Santa Fe 2016
by Liam Girdwood 16 Nov '16

16 Nov '16

Minutes below. Comments, corrections and omissions welcomed. We are missing some links to some of the slides presented. Please reply with a link if you have one. Audio Mini Conference Minutes - Santa Fe 2016 ============================================= Present: Mark, Rakesh, Liam, Vinod, Mengdong, Pierre, Patrick, Takashi S, Javilars, Takashi I, Eric, Dylan, Mauro, Shuah, Charles, Matthew, Lars, Sagar and others. Android USB Audio - Eric ======================== USB audio is becoming more important as analog jack can be removed. USB stack is difficult to optimise for low power/latency audio (too many wake ups with USB 1ms tick). Aim to add no_irq (no IRQ for each period) and mmap support in USB 3.0 stack to improve power and latency. USB playback and capture with synchronised start API is missing from stack/ALSA. Some patches were posted for synchronised endpoint but never made it to the list, Pierre is going to resend. USB currently defines simple audio endpoints. Support is needed to define more complex USB audio endpoints rather than just "playback" and "capture". Sub 10ms latency (round trip ?) is difficult to achieve on USB due to stack and drivers. Mmap may help reduce latency by removing buffer copies. Inter Device Dependencies ========================= Kernel Summit joined for this discussion. There is complexity between devices, more than just a parent/child relationship for some devices making it difficult to order probing and PM ops. Rafael, has created patch set to solve dependences for PM ops in Gregs tree, the patches could also be used for probing. The patches add an API to allow device relationships to be established as a graph. This API supports non parent/child relationships to and allows PM ops to run in a deterministic sequence. Relationship dependency discovery needs to come from DT/ACPI and will need standardised bindings. Mark described the current audio workarounds where the machine driver specifies relationships and deferred probe is used to manage the probing sequence. Some people do not like the high number deferred probe messages in dmesg. Relation graphs must be ordered so that cycles to do not exist. Next steps are to use Rafaels patches for probe ordering. AV Sync - Eric =============== A few frame counters are 32 bit (compressed API) and should be 64 bit. Vinod will update compressed API to 64 bit. Timers should also consider boot time and suspend/resume time as it's currently difficult to keep AV syn over suspend/resume cycles. AV sync on TV's hacky atm, we need AV sync API in ALSA. Takashi noted this was long time problem that needs solution. Some options for AV sync were pseudo transport streams (like existing mpeg format) but these were difficult for mmap(). Other devices had used runtime PLL updating with patches sent in the past ? Compressed API could be used to send transport stream for AV sync. HDMI/DP glue to GFX - Vinod =========================== Currently several interlaces exist to pass DP/HDMI events and information between audio and GFX stacks/drivers. ARM devices using pdata interface, i915 has another solution, BYT uses another interface. Needs to be consistent for user space ELD notification, jack detection/mapping. Currently HDA drivers have hack to assist userspace (PCM5). There are also difficulties mapping PCMs, channels and jacks to multiple HDMI/DP devices. Tikashi and Vinod to investigate best solutions. DP-MST - Vinod =============== Jack notifications more difficult with MST (extra complexity). How do we map multiple jacks from multiple monitors, disable unused pins when not rendering ? The 5.1 use case for gaming has to split channels to multiple monitors. Decision is that new devices would export topology to userpace so that userspace know the jack, channel mappings. Soundwire Support - Rakesh and Pierre ===================================== Soundwire new hardware interface for audio devices. Needs new subsystem support for soundwire master and slave devices. Hardik has posted initial RFC to the list for review. Rakesh presented slides . Link to the slides ? Some people had reviewed the RFC. Some changes to strct device/driver usage required to make it more like other subsystems upstream. There is also need to deal with devices that can power off after or before enumeration. Some confusion over the terminology used too, compared to other bus systems like I2C. Master should be device and not driver. Top level feedback is that it does not look like bus code. There is also a need to consider objects in DT/ACPI. AR Vinod, to look at STIO. Topology and UCM file deployment - Vinod, Mengdong, Liam ======================================================== Difficult to release and deploy topology/ucm updates with alsa-lib cadence. Distros to be contacted for best deployment where cadence can be faster than alsa-lib. Liam and Vinod to contact distro package maintainers. UCM - Liam, Mengdong ==================== Which driver card name string to change, there are 3 - long, short and driver. CRAS uses the string in the []. Not sure about PA. Liam to create patch that includes machine name in card name string. Mengdong has added #include and #define like support to topology. Mengdong will port this to UCM too so that a library of codecs can be built up and included by the machine ucm files. e.g. Asus T100 UCM would include rt5640 codec and baytrail DSP UCM configs. Tracepoint for audio - Lars, Takashi S ====================================== Lars and Takashi S have similar patchsets. Easy to make tooling for parsing traceout. This could replace printk and snd_printk, but builds bigger binary. Option to disable can be added to reduce size. Uses ftrace output and timestamp. Used in the PCM operations. Patches are still WiP and will be posted as RFC when ready. Dropping Dice Driver - Takashi S ================================ Dice hardware has serious problems making it unusable. Takashi S to discuss with Clemens regarding any potential work around for HW issue. If no work around is found then it could be marked as broken or moved to staging ? Sound Open Firmware - Liam ========================== Liam introduced SOF and took QA. Liam is also working on an ELF executable firmware image that can run on the developers host PC. http://alsa-project.org/main/index.php/Firmware Paravirtualisation of Audio Drivers - Rakesh ============================================ Safety critical systems have requirements to play safety critcal sounds (like alarms, warnings) and regular non safety critical sound like media. Rakesh proposed a Xen hypervisor method to guarantee the safety critical audio would always take precedence over the non safety critical audio. Xen would run a privileged OS (for safety critical) and unprivileged OS (for non safety critical audio). A para virtualised audio driver would allow the Unprivileged OS to play audio on the privileged OS. Lars suggested Libvirt and KVM may make this task far easier and less restricted than using Xen. Rakesh to investigate further. Userspace driver for Firewire - Takashi S ========================================= Need to represent mixers for some USB and FW devices in userspace. Why ?. Possibility of using topology for userspace. Nobody keen on exposing kernel APIs as userspavce ABIs. Possibility of looking at libvirt interfaces for userspace drivers many similarities. Kernel component, already use FW sound card. Mixer API not suitable, many bus so use control API. ALSA Stream Configuration Domains - Lars ======================================== ALSA has powerful but complex system to configure and constrain stream configurations at runtime but it does not scale well on modern ASoC based sound devices with multiple components and DAIs. Lars presented a solution that can break complex sound cards down to simple domains where configuration can be set by userspace to exactly match each use case. Userspace would also be able to read the configuration in each domain. New APIs calls would be needed in alsa-lib to achieve this. Lars also requested that ASoC become a first class citizen with ALSA since most new sound cards are ASoC based. Proposal was accepted.

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[alsa-devel] Alsa io plugin crash question
by Hermes Zhang 16 Nov '16

16 Nov '16

Hi all, I am try to write a plugin in alsa io plugin mode static snd_pcm_sframes_t my_read (snd_pcm_ioplug_t * io, const snd_pcm_channel_area_t * areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size) { } In above function, I will copy the audio data to areas with calculate the offset like this: memcpy((char *) areas->addr + (areas->first + areas->step * offset) / 8, data, size); But after serveral calls, the application crash, and I print the "offset" value, it seem always increase, when reach one number (e.g. 66560) it crash. So my question is: is the "offset" value always increase is expected? Or does it should like a ring buffer, so it should start from begin again? I see there is hw_pointer and app_pointer, but they are both read_only, right? How could we adjust the read pointer? Thanks. BTW, I test the plugin in this way: arecord -D my_plug -c 1 -f S16_LE raw Best Regards, Hermes

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[alsa-devel] [RFC PATCH 0/3] Add support for component devices
by mengdong.lin＠linux.intel.com 16 Nov '16

16 Nov '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Sound cards are defined by machines. And off-soc codecs and DSPs embedded in DSP can be taken as the components and resued by different machines/ sound cards. This series allows codec and SOC vendors to install UCM configuration files for a specific codec or DSP, and sound card verb files can just include these conf files of components, and trigger enable/ disable sequence defined by the component devices. Component devices will not be exposed to applications for backward compatibility. So audio servers like PulseAudio and CRAS still only see the machine devices. Mengdong Lin (3): ucm: Skip component directories when scanning sound card configuration files ucm: Parse sequence of component devices ucm: Execute sequence of component devices src/ucm/main.c | 98 ++++++++++++++++++++++++++++++++++++----- src/ucm/parser.c | 122 +++++++++++++++++++++++++++++++++++++++++++++++++++- src/ucm/ucm_local.h | 11 +++++ src/ucm/utils.c | 1 + 4 files changed, 220 insertions(+), 12 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v9 0/3] ASoC: MT8173 HDMI jack detection
by Philipp Zabel 16 Nov '16

16 Nov '16

Hi, now that a new HDMI state notifier RFC has been posted [1], I've updated this series to use it instead of the previous prototype. I've also sent the drm/mediatek part of this series separately [2], which is needed to actually generate the events the hdmi-codec driver listens for. Changes since v8: - Dropped the HDMI notification prototype patch - Rebased onto the "video: add HDMI state notifier support" patch by Hans Verkuil, based on Russell King's earlier version. - Dropped the drm/mediatek patch regards Philipp [1] https://patchwork.linuxtv.org/patch/38109/ [2] https://patchwork.kernel.org/patch/9430337/ Philipp Zabel (3): ASoC: hdmi-codec: use HDMI state notifier to add jack support ASoC: mediatek: Add jack detection support to mt8173-rt5650-rt5676 machine driver ASoC: mediatek: Add jack detection support to the mt8173-rt5650 machine driver include/sound/hdmi-codec.h | 7 ++ sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdmi-codec.c | 107 +++++++++++++++++++++-- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 42 +++++++++ sound/soc/mediatek/mt8173/mt8173-rt5650.c | 42 +++++++++ 5 files changed, 191 insertions(+), 8 deletions(-) -- 2.10.2

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[alsa-devel] [PATCH v3 00/17] ASoC: add OF graph base simple-card
by Kuninori Morimoto 16 Nov '16

16 Nov '16

Hi Rob, Mark These are v3 of OF graph base simple-card patch-set. 1) - 5) : soc-core prepare for OF graph base simple-card 6) - 10) : OF graph new feature 11) - 17) : OF graph base simple-card (depends on above 2 patch-set) >> Mark I could success to apply 1) - 5) on your topic/core and topic/simple, both. Please let me know if you can't do it. *** BLURB HERE *** Kuninori Morimoto (17): 1) ASoC: soc-core: adjust for graph on snd_soc_of_parse_card_name 2) ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_simple_widgets 3) ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_routing 4) ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_prefix 5) ASoC: soc-core: snd_soc_get_dai_name() become non static 6) of_graph: add of_graph_get_remote_endpoint() 7) of_graph: add of_graph_get_port_parent() 8) of_graph: add of_graph_get_top_port() 9) of_graph: add for_each_of_port() / for_each_of_endpoint_in_port() 10) of_graph: add of_graph_get_port/endpoint_count() 11) ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() 12) ASoC: simple-card-utils: add asoc_simple_card_try_to_probe_graph_card() 13) ASoC: simple-card-utils: adjust for graph on asoc_simple_card_parse_card_name 14) ASoC: add simple-graph-card document 15) ASoC: add simple-graph-card support 16) ASoC: add simple-graph-scu-card document 17) ASoC: add simple-graph-scu-card support .../bindings/sound/simple-graph-card.txt | 65 +++ .../bindings/sound/simple-graph-scu-card.txt | 65 +++ drivers/of/base.c | 160 ++++++- include/linux/of_graph.h | 54 +++ include/sound/simple_card_utils.h | 19 + include/sound/soc.h | 32 +- sound/soc/generic/Kconfig | 15 + sound/soc/generic/Makefile | 4 + sound/soc/generic/simple-card-utils.c | 98 ++++- sound/soc/generic/simple-card.c | 2 +- sound/soc/generic/simple-graph-card.c | 462 +++++++++++++++++++++ sound/soc/generic/simple-graph-scu-card.c | 417 +++++++++++++++++++ sound/soc/generic/simple-scu-card.c | 2 +- sound/soc/soc-core.c | 41 +- 14 files changed, 1401 insertions(+), 35 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/simple-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-graph-scu-card.txt create mode 100644 sound/soc/generic/simple-graph-card.c create mode 100644 sound/soc/generic/simple-graph-scu-card.c -- 1.9.1

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[alsa-devel] [PATCH] ASoC: wm8580: tidyup Kconfig tristate title
by Kuninori Morimoto 16 Nov '16

16 Nov '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> It is WM8580, not WM8523 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c67667b..f1c48d2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -914,7 +914,7 @@ config SND_SOC_WM8523 depends on I2C config SND_SOC_WM8580 - tristate "Wolfson Microelectronics WM8523 CODEC" + tristate "Wolfson Microelectronics WM8580 CODEC" depends on I2C config SND_SOC_WM8711 -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rl6231: add 19.2M to 4.096M pll preset table
by Bard Liao 15 Nov '16

15 Nov '16

Add a pll mapping table for 19.2M in and 4.096M out. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rl6231.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 1dc68ab..7b447d0 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -102,6 +102,7 @@ struct pll_calc_map { }; static const struct pll_calc_map pll_preset_table[] = { + {19200000, 4096000, 23, 14, 1, false}, {19200000, 24576000, 3, 30, 3, false}, }; -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 0/2] ASoC: omap-twl4030: remove legacy code
by Nicolae Rosia 15 Nov '16

15 Nov '16

This patch series removes the omap-twl4030 platform data since all users are now device tree based. These patches are based on Tony's omap-for-v4.10/legacy branch [0] and also on [1]. [0] https://git.kernel.org/cgit/linux/kernel/git/tmlind/linux-omap.git/log/?h=o… [1] https://patchwork.kernel.org/patch/9424067/ Nicolae Rosia (2): ASoC: omap-twl4030: rework probing for device tree only mfd: twl4030-audio: make module DT only drivers/mfd/twl4030-audio.c | 46 ++-------- include/linux/i2c/twl.h | 23 ----- include/linux/platform_data/omap-twl4030.h | 58 ------------- sound/soc/codecs/twl4030.c | 8 ++ sound/soc/omap/omap-twl4030.c | 133 ++++++++++------------------- 5 files changed, 61 insertions(+), 207 deletions(-) delete mode 100644 include/linux/platform_data/omap-twl4030.h -- 2.7.4

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