42 changed files with 1837 additions and 7118 deletions
--- a/44
+++ b/44
@ -1,52 +1,18 @@
-#
-# Peter Yang <turmary@126.com>
-# Copyright (c) 2019 Seeed Studio
-#
-# MIT License
-#
-
-uname_r=$(shell uname -r)
-
-# If KERNELRELEASE is defined, we've been invoked from the
-# kernel build system and can use its language
-ifneq ($(KERNELRELEASE),)
-# $(warning KERNELVERSION=$(KERNELVERSION))
-
 snd-soc-wm8960-objs := wm8960.o
-snd-soc-ac108-objs := ac108.o ac101.o
-snd-soc-seeed-voicecard-objs := seeed-voicecard.o
+snd-soc-ac108-objs := ac108.o


 obj-m += snd-soc-wm8960.o
 obj-m += snd-soc-ac108.o
-obj-m += snd-soc-seeed-voicecard.o

-ifdef DEBUG
-ifneq ($(DEBUG),0)
-	ccflags-y += -DDEBUG -DAC101_DEBG
-endif
-endif
-
-
-
-else
-
-DEST := /lib/modules/$(uname_r)/kernel

 all:
-	make -C /lib/modules/$(uname_r)/build M=$(PWD) modules
+	make -C /lib/modules/$(shell uname -r)/build M=$(PWD) modules

 clean:
-	make -C /lib/modules/$(uname_r)/build M=$(PWD) clean
+	make -C /lib/modules/$(shell uname -r)/build M=$(PWD) clean

 install:
-	sudo cp snd-soc-ac108.ko ${DEST}/sound/soc/codecs/
-	sudo cp snd-soc-wm8960.ko ${DEST}/sound/soc/codecs/
-	sudo cp snd-soc-seeed-voicecard.ko ${DEST}/sound/soc/bcm/
+	sudo cp snd-soc-ac108.ko /lib/modules/$(shell uname -r)/kernel/sound/soc/codecs/
+	sudo cp snd-soc-wm8960.ko /lib/modules/$(shell uname -r)/kernel/sound/soc/codecs/
 	sudo depmod -a
-
-
-.PHONY: all clean install
-
-endif
-
--- a/README.md
+++ b/README.md
@ -1,64 +1,149 @@
 # seeed-voicecard

-The drivers for [ReSpeaker Mic Hat](https://www.seeedstudio.com/ReSpeaker-2-Mics-Pi-HAT-p-2874.html), [ReSpeaker 4 Mic Array](https://www.seeedstudio.com/ReSpeaker-4-Mic-Array-for-Raspberry-Pi-p-2941.html), [6-Mics Circular Array Kit](), and [4-Mics Linear Array Kit]() for Raspberry Pi.
+[![Join the chat at https://gitter.im/seeed-voicecard/Lobby](https://badges.gitter.im/seeed-voicecard/Lobby.svg)](https://gitter.im/seeed-voicecard/Lobby?utm_source=badge&utm_medium=badge&utm_campaign=pr-badge&utm_content=badge)

-### Install seeed-voicecard
-Get the seeed voice card source code and install all linux kernel drivers
+The drivers of [ReSpeaker Mic Hat](https://www.seeedstudio.com/ReSpeaker-2-Mics-Pi-HAT-p-2874.html) and [ReSpeaker 4 Mic Array](https://www.seeedstudio.com/ReSpeaker-4-Mic-Array-for-Raspberry-Pi-p-2941.html) for Raspberry Pi.
+
+## ReSpeaker Mic Hat
+
+[![](https://github.com/SeeedDocument/MIC_HATv1.0_for_raspberrypi/blob/master/img/mic_hatv1.0.png?raw=true)](https://www.seeedstudio.com/ReSpeaker-2-Mics-Pi-HAT-p-2874.html)
+
+
+While the upstream wm8960 codec is not currently supported by current Pi kernel builds, upstream wm8960 has some bugs, we had fixed it. we must it build manually.
+
+Get the seeed voice card source code.
 ```bash
-git clone https://github.com/HinTak/seeed-voicecard
+git clone https://github.com/respeaker/seeed-voicecard
 cd seeed-voicecard
-sudo ./install.sh
+#for ReSpeaker 2-mic
+sudo ./install.sh 2mic
 sudo reboot
 ```
-## ReSpeaker Documentation

-Up to date documentation for reSpeaker products can be found in [Seeed Studio Wiki](https://wiki.seeedstudio.com/ReSpeaker/)!
-![](https://files.seeedstudio.com/wiki/ReSpeakerProductGuide/img/Raspberry_Pi_Mic_Array_Solutions.png)
-
-
-### Coherence
-
-Estimate the magnitude squared coherence using Welch’s method.
-![4-mics-linear-array-kit coherence](https://user-images.githubusercontent.com/3901856/37277486-beb1dd96-261f-11e8-898b-84405bfc7cea.png)  
-Note: 'CO 1-2' means the coherence between channel 1 and channel 2.
+Check that the sound card name matches the source code seeed-voicecard.

 ```bash
-# How to get the coherence of the captured audio(a.wav for example).
-sudo apt install python-numpy python-scipy python-matplotlib
-python tools/coherence.py a.wav
-
-# Requirement of the input audio file:
- format: WAV(Microsoft) signed 16-bit PCM
- channels: >=2
+#for ReSpeaker 2-mic
+pi@raspberrypi:~/seeed-voicecard $ aplay -l
+**** List of PLAYBACK Hardware Devices ****
+card 0: ALSA [bcm2835 ALSA], device 0: bcm2835 ALSA [bcm2835 ALSA]
+  Subdevices: 8/8
+  Subdevice #0: subdevice #0
+  Subdevice #1: subdevice #1
+  Subdevice #2: subdevice #2
+  Subdevice #3: subdevice #3
+  Subdevice #4: subdevice #4
+  Subdevice #5: subdevice #5
+  Subdevice #6: subdevice #6
+  Subdevice #7: subdevice #7
+card 0: ALSA [bcm2835 ALSA], device 1: bcm2835 ALSA [bcm2835 IEC958/HDMI]
+  Subdevices: 1/1
+  Subdevice #0: subdevice #0
+card 1: seeed2micvoicec [seeed-2mic-voicecard], device 0: bcm2835-i2s-wm8960-hifi wm8960-hifi-0 []
+  Subdevices: 1/1
+  Subdevice #0: subdevice #0
+pi@raspberrypi:~/seeed-voicecard $ arecord -l
+**** List of CAPTURE Hardware Devices ****
+card 1: seeed2micvoicec [seeed-2mic-voicecard], device 0: bcm2835-i2s-wm8960-hifi wm8960-hifi-0 []
+  Subdevices: 1/1
+  Subdevice #0: subdevice #0
+pi@raspberrypi:~/seeed-voicecard $ 
 ```

+#### Next step
+Go to https://github.com/respeaker/mic_hat to build voice enabled projects with Google Assistant SDK or Alexa Voice Service.
+
+## ReSpeaker 4 Mic Array
+
+[![](https://github.com/SeeedDocument/ReSpeaker-4-Mic-Array-for-Raspberry-Pi/blob/master/img/features.png?raw=true)](https://www.seeedstudio.com/ReSpeaker-4-Mic-Array-for-Raspberry-Pi-p-2941.html)
+
+The 4 Mic Array uses ac108 which includes 4 ADCs, we also write ac108 rapberry pi linux kernel driver.
+
+```bash
+git clone https://github.com/respeaker/seeed-voicecard
+cd seeed-voicecard
+sudo ./install.sh 4mic
+
+#reboot your Raspbian OS
+sudo reboot
+```
+
+Check that the sound card name matches the source code seeed-voicecard.
+
+```bash
+#for ReSpeaker 4-mic
+pi@raspberrypi:~ $ arecord -L
+null
+    Discard all samples (playback) or generate zero samples (capture)
+playback
+capture
+dmixed
+array
+ac108
+default:CARD=seeed4micvoicec
+    seeed-4mic-voicecard, 
+    Default Audio Device
+sysdefault:CARD=seeed4micvoicec
+    seeed-4mic-voicecard, 
+    Default Audio Device
+dmix:CARD=seeed4micvoicec,DEV=0
+    seeed-4mic-voicecard, 
+    Direct sample mixing device
+dsnoop:CARD=seeed4micvoicec,DEV=0
+    seeed-4mic-voicecard, 
+    Direct sample snooping device
+hw:CARD=seeed4micvoicec,DEV=0
+    seeed-4mic-voicecard, 
+    Direct hardware device without any conversions
+plughw:CARD=seeed4micvoicec,DEV=0
+    seeed-4mic-voicecard, 
+    Hardware device with all software conversions
+pi@raspberrypi:~ $ 
+```
+If you want to change the alsa settings, You can use `sudo alsactl --file=asound.state store` to save it.
+
+Test:
+```bash
+#for ReSpeaker 2-mic
+#It will capture sound an playback on hw:1
+arecord -f cd -Dhw:1 | aplay -Dhw:1
+```
+
+```bash
+#for ReSpeaker 4-mic
+#It will capture sound on AC108 and save as a.wav
+arecord -Dac108 -f S32_LE -r 16000 -c 4 a.wav
+```
 ### uninstall seeed-voicecard
 If you want to upgrade the driver , you need uninstall the driver first.

 ```
-pi@raspberrypi:~/seeed-voicecard $ sudo ./uninstall.sh 
-...
+pi@raspberrypi:~/seeed-voicecard $ sudo ./uninstall.sh 4mic
+delete dtoverlay=seeed-4mic-voicecard in /boot/config.txt
+delete snd-soc-ac108 in /etc/modules
+------------------------------------------------------
+Please reboot your raspberry pi to apply all settings
+Thank you!
+------------------------------------------------------
+pi@raspberrypi:~/seeed-voicecard $ sudo ./uninstall.sh 2mic
+delete dtoverlay=seeed-2mic-voicecard in /boot/config.txt
+remove seeed-2mic-voicecard.dtbo in /boot/overlays
+remove snd-soc-wm8960.ko
+delete snd-soc-wm8960 in /etc/modules
 ------------------------------------------------------
 Please reboot your raspberry pi to apply all settings
 Thank you!
 ------------------------------------------------------
 ```

+### with Google Assistant
+if you run the assistant but the playback is speed up considerably, try to configure alsa:
+
+```bash
+sudo cp asound.conf /etc/asound.conf
+```
+
+If the alsa configuration doesn't solve the issue, try to use pulseaudio. See [#4](https://github.com/respeaker/seeed-voicecard/issues/4)
+
+
 Enjoy !
-
-### Technical support
-
-For hardware testing purposes we made a Rasperry Pi OS 5.10.17-v7l+ 32-bit image with reSpeaker drivers pre-installed, which you can download by clicking on [this link](https://files.seeedstudio.com/linux/Raspberry%20Pi%204%20reSpeaker/2021-05-07-raspios-buster-armhf-lite-respeaker.img.xz).
-
-We provide official support for using reSpeaker with the following OS:
- 32-bit Raspberry Pi OS
- 64-bit Raspberry Pi OS
-
-And following hardware platforms:
- Raspberry Pi 3 (all models), Raspberry Pi 4 (all models)
-
-Anything beyond the scope of official support is considered to be community supported. Support for other OS/hardware platforms can be added, provided MOQ requirements can be met. 
-
-If you have a technical problem when using reSpeaker with one of the officially supported platforms/OS, feel free to create an issue on Github. For general questions or suggestions, please use [Seeed forum](https://forum.seeedstudio.com/c/products/respeaker/15). 
-
-
--- a/ac101.c
+++ b/ac101.c
--- a/ac101_regs.h
+++ b/ac101_regs.h
@ -1,432 +0,0 @@
-/*
- * ac101_regs.h
- *
- * (C) Copyright 2017-2018
- * Seeed Technology Co., Ltd. <www.seeedstudio.com>
- *
- * PeterYang <linsheng.yang@seeed.cc>
- *
- * (C) Copyright 2010-2017
- * Reuuimlla Technology Co., Ltd. <www.reuuimllatech.com>
- * huangxin <huangxin@reuuimllatech.com>
- *
- * some simple description for this code
- *
- * 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 __AC101_REGS_H__
-#define __AC101_REGS_H__
-
-/*pll source*/
-#define AC101_MCLK1 1
-#define AC101_MCLK2 2
-#define AC101_BCLK1 3
-#define AC101_BCLK2 4
-
-#define AIF1_CLK 1
-#define AIF2_CLK 2
-
-#define	CHIP_AUDIO_RST		0x0
-#define PLL_CTRL1		0x1
-#define PLL_CTRL2		0x2
-#define SYSCLK_CTRL		0x3
-#define MOD_CLK_ENA		0x4
-#define MOD_RST_CTRL		0x5
-#define AIF_SR_CTRL		0x6
-
-#define AIF1_CLK_CTRL		0x10
-#define AIF1_ADCDAT_CTRL	0x11
-#define AIF1_DACDAT_CTRL	0x12
-#define AIF1_MXR_SRC		0x13
-#define AIF1_VOL_CTRL1		0x14
-#define AIF1_VOL_CTRL2		0x15
-#define AIF1_VOL_CTRL3		0x16
-#define AIF1_VOL_CTRL4		0x17
-#define AIF1_MXR_GAIN		0x18
-#define AIF1_RXD_CTRL		0x19
-#define ADC_DIG_CTRL		0x40
-#define ADC_VOL_CTRL		0x41
-#define ADC_DBG_CTRL		0x42
-
-#define HMIC_CTRL1			0x44
-#define HMIC_CTRL2			0x45
-#define HMIC_STS			0x46
-
-#define DAC_DIG_CTRL		0x48
-#define DAC_VOL_CTRL		0x49
-#define DAC_DBG_CTRL		0x4a
-#define DAC_MXR_SRC			0x4c
-#define DAC_MXR_GAIN		0x4d
-
-#define ADC_APC_CTRL		0x50
-#define ADC_SRC				0x51
-#define ADC_SRCBST_CTRL		0x52
-#define OMIXER_DACA_CTRL	0x53
-#define OMIXER_SR			0x54
-#define OMIXER_BST1_CTRL	0x55
-#define HPOUT_CTRL			0x56
-#define ESPKOUT_CTRL		0x57
-#define SPKOUT_CTRL			0x58
-#define LOUT_CTRL			0x59
-#define ADDA_TUNE1			0x5a
-#define ADDA_TUNE2			0x5b
-#define ADDA_TUNE3			0x5c
-#define HPOUT_STR			0x5d
-
-/*CHIP_AUDIO_RST*/
-#define AC101_CHIP_ID			0x0101
-
-/*PLL_CTRL1*/
-#define DPLL_DAC_BIAS		14
-#define PLL_POSTDIV_M		8
-#define CLOSE_LOOP		6
-#define INT			0
-
-/*PLL_CTRL2*/
-#define PLL_EN			15
-#define PLL_LOCK_STATUS		14
-#define PLL_PREDIV_NI		4
-#define PLL_POSTDIV_NF		0
-
-/*SYSCLK_CTRL*/
-#define PLLCLK_ENA			15
-#define PLLCLK_SRC			12
-#define AIF1CLK_ENA			11
-#define AIF1CLK_SRC			8
-#define AIF2CLK_ENA			7
-#define AIF2CLK_SRC			4
-#define SYSCLK_ENA			3
-#define SYSCLK_SRC			0
-
-/*MOD_CLK_ENA*/
-#define MOD_CLK_AIF1		15
-#define MOD_CLK_AIF2		14
-#define MOD_CLK_AIF3		13
-#define MOD_CLK_SRC1		11
-#define MOD_CLK_SRC2		10
-#define MOD_CLK_HPF_AGC		7
-#define MOD_CLK_HPF_DRC		6
-#define MOD_CLK_ADC_DIG		3
-#define MOD_CLK_DAC_DIG		2
-
-/*MOD_RST_CTRL*/
-#define MOD_RESET_CTL		0
-#define MOD_RESET_AIF1		15
-#define MOD_RESET_AIF2		14
-#define MOD_RESET_AIF3		13
-#define MOD_RESET_SRC1		11
-#define MOD_RESET_SRC2		10
-#define MOD_RESET_HPF_AGC	7
-#define MOD_RESET_HPF_DRC	6
-#define MOD_RESET_ADC_DIG	3
-#define MOD_RESET_DAC_DIG	2
-
-/*AIF_SR_CTRL*/
-#define AIF1_FS				12	//AIF1 Sample Rate
-#define AIF2_FS				8	//AIF2 Sample Rate
-#define SRC1_ENA			3
-#define SRC1_SRC			2
-#define SRC2_ENA			1
-#define SRC2_SRC			0
-
-/*AIF1LCK_CTRL*/
-#define AIF1_MSTR_MOD		15
-#define AIF1_BCLK_INV		14
-#define AIF1_LRCK_INV		13
-#define AIF1_BCLK_DIV		9
-#define AIF1_LRCK_DIV		6
-#define AIF1_WORK_SIZ		4
-#define AIF1_DATA_FMT		2
-#define DSP_MONO_PCM		1
-#define AIF1_TDMM_ENA		0
-
-/*AIF1_ADCDAT_CTRL*/
-#define AIF1_AD0L_ENA		15
-#define AIF1_AD0R_ENA		14
-#define AIF1_AD1L_ENA		13
-#define AIF1_AD1R_ENA		12
-#define AIF1_AD0L_SRC		10
-#define AIF1_AD0R_SRC		8
-#define AIF1_AD1L_SRC		6
-#define AIF1_AD1R_SRC		4
-#define AIF1_ADCP_ENA		3
-#define AIF1_ADUL_ENA		2
-#define AIF1_SLOT_SIZ		0
-
-/*AIF1_DACDAT_CTRL*/
-#define AIF1_DA0L_ENA		15
-#define AIF1_DA0R_ENA		14
-#define AIF1_DA1L_ENA		13
-#define AIF1_DA1R_ENA		12
-#define AIF1_DA0L_SRC		10
-#define AIF1_DA0R_SRC		8
-#define AIF1_DA1L_SRC		6
-#define AIF1_DA1R_SRC		4
-#define AIF1_DACP_ENA		3
-#define AIF1_DAUL_ENA		2
-#define AIF1_SLOT_SIZ		0
-
-/*AIF1_MXR_SRC*/
-#define AIF1_AD0L_AIF1_DA0L_MXR		15
-#define AIF1_AD0L_AIF2_DACL_MXR		14
-#define AIF1_AD0L_ADCL_MXR		13
-#define AIF1_AD0L_AIF2_DACR_MXR		12
-#define AIF1_AD0R_AIF1_DA0R_MXR		11
-#define AIF1_AD0R_AIF2_DACR_MXR		10
-#define AIF1_AD0R_ADCR_MXR		9
-#define AIF1_AD0R_AIF2_DACL_MXR		8
-#define AIF1_AD1L_AIF2_DACL_MXR		7
-#define AIF1_AD1L_ADCL_MXR		6
-#define AIF1_AD1L_MXR_SRC	6
-#define AIF1_AD1R_AIF2_DACR_MXR		3
-#define AIF1_AD1R_ADCR_MXR		2
-#define AIF1_AD1R_MXR_SRC	2
-
-/*AIF1_VOL_CTRL1*/
-#define AIF1_AD0L_VOL		8
-#define AIF1_AD0R_VOL		0
-
-/*AIF1_VOL_CTRL2*/
-#define AIF1_AD1L_VOL		8
-#define AIF1_AD1R_VOL		0
-
-/*AIF1_VOL_CTRL3*/
-#define AIF1_DA0L_VOL		8
-#define AIF1_DA0R_VOL		0
-
-/*AIF1_VOL_CTRL4*/
-#define AIF1_DA1L_VOL		8
-#define AIF1_DA1R_VOL 		0
-
-/*AIF1_MXR_GAIN*/
-#define AIF1_AD0L_MXR_GAIN	12
-#define AIF1_AD0R_MXR_GAIN	8
-#define AIF1_AD1L_MXR_GAIN	6
-#define AIF1_AD1R_MXR_GAIN	2
-
-/*AIF1_RXD_CTRL*/
-#define AIF1_N_DATA_DISCARD	8
-
-/*ADC_DIG_CTRL*/
-#define ENAD				15
-#define ENDM				14
-#define ADFIR32				13
-#define ADOUT_DTS			2
-#define ADOUT_DLY			1
-
-/*ADC_VOL_CTRL*/
-#define ADC_VOL_L			8
-#define ADC_VOL_R			0
-
-/*ADC_DBG_CTRL*/
-#define ADSW				15
-#define DMIC_CLK_PIN_CTRL 	12
-
-/*HMIC_CTRL1*/
-#define HMIC_M				12
-#define HMIC_N				8
-#define HMIC_DATA_IRQ_MODE	7
-#define HMIC_TH1_HYSTERESIS	5
-#define HMIC_PULLOUT_IRQ	4
-#define HMIC_PLUGIN_IRQ		3
-#define HMIC_KEYUP_IRQ		2
-#define HMIC_KEYDOWN_IRQ	1
-#define HMIC_DATA_IRQ_EN	0
-
-/*HMIC_CTRL2*/
-#define HMIC_SAMPLE_SELECT	14
-#define HMIC_TH2_HYSTERESIS	13
-#define HMIC_TH2			8
-#define HMIC_SF				6
-#define KEYUP_CLEAR			5
-#define HMIC_TH1			0
-
-/*HMIC_STS*/
-#define HMIC_DATA			8
-#define GET_HMIC_DATA(r)		(((r) >> HMIC_DATA) & 0x1F)
-#define HMIC_PULLOUT_PEND	4
-#define HMIC_PLUGIN_PEND	3
-#define HMIC_KEYUP_PEND		2
-#define HMKC_KEYDOWN_PEND	1
-#define HMIC_DATA_PEND		0
-#define HMIC_PEND_ALL		(0x1F)
-
-/*DAC_DIG_CTRL*/
-#define ENDA				15
-#define ENHPF				14
-#define DAFIR32				13
-#define MODQU				8
-
-/*DAC_VOL_CTRL*/
-#define DAC_VOL_L			8
-#define DAC_VOL_R			0
-
-/*DAC_DBG_CTRL*/
-#define DASW				15
-#define ENDWA_N				14
-#define DAC_MOD_DBG			13
-#define DAC_PTN_SEL			6
-#define DVC				0
-
-/*DAC_MXR_SRC*/
-#define DACL_MXR_AIF1_DA0L		15
-#define DACL_MXR_AIF1_DA1L		14
-#define DACL_MXR_AIF2_DACL		13
-#define DACL_MXR_ADCL			12
-#define DACL_MXR_SRC			12
-#define DACR_MXR_AIF1_DA0R		11
-#define DACR_MXR_AIF1_DA1R		10
-#define DACR_MXR_AIF2_DACR		9
-#define DACR_MXR_ADCR			8
-#define DACR_MXR_SRC		8
-
-/*DAC_MXR_GAIN*/
-#define DACL_MXR_GAIN		12
-#define DACR_MXR_GAIN		8
-
-/*ADC_APC_CTRL*/
-#define ADCREN				15
-#define ADCRG				12
-#define ADCLEN				11
-#define ADCLG				8
-#define MBIASEN				7
-#define MMIC_BIAS_CHOP_EN		6
-#define MMIC_BIAS_CHOP_CKS		4
-#define HBIASMOD			2
-#define HBIASEN				1
-#define HBIASADCEN			0
-
-/*ADC_SRC*/
-#define RADCMIXMUTEMIC1BOOST	  (13)
-#define RADCMIXMUTEMIC2BOOST	  (12)
-#define RADCMIXMUTELINEINLR		  (11)
-#define RADCMIXMUTELINEINR		  (10)
-#define RADCMIXMUTEAUXINR		  (9)
-#define RADCMIXMUTEROUTPUT		  (8)
-#define RADCMIXMUTELOUTPUT		  (7)
-#define LADCMIXMUTEMIC1BOOST	  (6)
-#define LADCMIXMUTEMIC2BOOST	  (5)
-#define LADCMIXMUTELINEINLR		  (4)
-#define LADCMIXMUTELINEINL		  (3)
-#define LADCMIXMUTEAUXINL		  (2)
-#define LADCMIXMUTELOUTPUT		  (1)
-#define LADCMIXMUTEROUTPUT		  (0)
-
-
-/*ADC_SRCBST_CTRL*/
-#define MIC1AMPEN			15
-#define ADC_MIC1G			12
-#define MIC2AMPEN			11
-#define ADC_MIC2G			8
-#define MIC2SLT				7
-#define LINEIN_PREG			4
-#define AUXI_PREG			0
-
-/*OMIXER_DACA_CTRL*/
-#define DACAREN				15
-#define DACALEN				14
-#define RMIXEN				13
-#define LMIXEN				12
-#define HPOUTPUTENABLE			8
-
-/*OMIXER_SR*/
-#define RMIXMUTEMIC1BOOST		  (13)
-#define RMIXMUTEMIC2BOOST		  (12)
-#define RMIXMUTELINEINLR		  (11)
-#define RMIXMUTELINEINR			  (10)
-#define RMIXMUTEAUXINR			  (9)
-#define RMIXMUTEDACR			  (8)
-#define RMIXMUTEDACL			  (7)
-#define LMIXMUTEMIC1BOOST		  (6)
-#define LMIXMUTEMIC2BOOST		  (5)
-#define LMIXMUTELINEINLR		  (4)
-#define LMIXMUTELINEINL			  (3)
-#define LMIXMUTEAUXINL			  (2)
-#define LMIXMUTEDACL			  (1)
-#define LMIXMUTEDACR			  (0)
-
-/*OMIXER_BST1_CTRL*/
-#define BIASVOLTAGE			12
-#define AXG				9
-#define OMIXER_MIC1G			6
-#define OMIXER_MIC2G			3
-#define LINEING				0
-
-/*HPOUT_CTRL*/
-#define RHPS				15
-#define LHPS				14
-#define RHPPA_MUTE			13
-#define LHPPA_MUTE			12
-#define HPPA_EN				11
-#define HP_VOL				4
-#define HPPA_DEL			2
-#define HPPA_IS				0
-
-/*ESPKOUT_CTRL*/
-#define EAR_RAMP_TIME		11
-#define	ESPA_OUT_CURRENT	9
-#define ESPSR				7
-#define ESPPA_MUTE			6
-#define ESPPA_EN			5
-#define ESP_VOL				0
-
-/*SPKOUT_CTRL*/
-#define HPCALICKS			13
-#define RSPKS				12
-#define RSPKINVEN			11
-#define RSPK_EN				9
-#define LSPKS				8
-#define LSPKINVEN			7
-#define LSPK_EN				5
-#define SPK_VOL				0
-
-/*LOUT_CTRL*/
-#define LINEOUTG			5
-#define LINEOUTEN			4
-#define LINEOUTS0			3
-#define LINEOUTS1			2
-#define LINEOUTS2			1
-#define LINEOUTS3			0
-
-/*ADDA_TUNE1*/
-#define CURRENT_TEST_SELECT	14
-#define BIHE_CTRL			12
-#define DITHER				11
-#define DITHER_CLK			9
-#define ZERO_CROSSOVER_EN	8
-#define ZERO_CROSSOVER_TIME 7
-#define EAR_SPEED_SELECT	6
-#define REF_CHOPPEN_CKS		4
-#define OPMIC_BIAS_CUR		0
-
-/*ADDA_TUNE2*/
-#define OPDAC_BIAS_CUR		14
-#define OPDRV_BIAS_CUR		12
-#define OPMIX_BIAS_CUR		10
-#define OPEAR_BIAS_CUR		8
-#define OPVR_BIAS_CUR		6
-#define OPAAF_BIAS_CUR		4
-#define OPADC1_BIAS_CUR		2
-#define OPADC2_BIAS_CUR		0
-
-/*ADDA_TUNE3*/
-#define LDOEN				15
-#define LDO_SEL				12
-#define BIASCALIVERIFY			11
-#define BIASMODE			10
-#define BIASCALIDATA			9
-#define OSCS				1
-#define OSCEN				0
-
-/*HPOUT_STR*/
-#define HPVL_SOFT_MOD		14
-#define	HPVL_STEP_CTRL		8
-#define  DACA_CHND_ENA		7
-#define HPPA_MXRD_ENA		6
-#define HPVL_CTRL_OUT		0
-
-#endif//__AC101_REGS_H__
--- a/ac108.c
+++ b/ac108.c
--- a/ac108_6mic.state
+++ b/ac108_6mic.state
@ -1,367 +0,0 @@
-state.ALSA {
-	control.1 {
-		iface MIXER
-		name 'PCM Playback Volume'
-		value 400
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '-10239 - 400'
-			dbmin -9999999
-			dbmax 400
-			dbvalue.0 400
-		}
-	}
-	control.2 {
-		iface MIXER
-		name 'PCM Playback Switch'
-		value true
-		comment {
-			access 'read write'
-			type BOOLEAN
-			count 1
-		}
-	}
-	control.3 {
-		iface MIXER
-		name 'PCM Playback Route'
-		value 1
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 2'
-		}
-	}
-	control.4 {
-		iface PCM
-		name 'IEC958 Playback Default'
-		value '0000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000'
-		comment {
-			access 'read write'
-			type IEC958
-			count 1
-		}
-	}
-	control.5 {
-		iface PCM
-		name 'IEC958 Playback Con Mask'
-		value '0200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000'
-		comment {
-			access read
-			type IEC958
-			count 1
-		}
-	}
-	control.6 {
-		iface PCM
-		name 'IEC958 Playback PCM Stream'
-		value '0000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000'
-		comment {
-			access 'read write inactive'
-			type IEC958
-			count 1
-		}
-	}
-}
-state.seeed8micvoicec {
-	control.1 {
-		iface MIXER
-		name 'CH1 digital volume'
-		value 208
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 255'
-			dbmin -11925
-			dbmax 7200
-			dbvalue.0 3675
-		}
-	}
-	control.2 {
-		iface MIXER
-		name 'CH2 digital volume'
-		value 208
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 255'
-			dbmin -11925
-			dbmax 7200
-			dbvalue.0 3675
-		}
-	}
-	control.3 {
-		iface MIXER
-		name 'CH3 digital volume'
-		value 208
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 255'
-			dbmin -11925
-			dbmax 7200
-			dbvalue.0 3675
-		}
-	}
-	control.4 {
-		iface MIXER
-		name 'CH4 digital volume'
-		value 208
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 255'
-			dbmin -11925
-			dbmax 7200
-			dbvalue.0 3675
-		}
-	}
-	control.5 {
-		iface MIXER
-		name 'ADC1 PGA gain'
-		value 0
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 31'
-			dbmin 0
-			dbmax 3100
-			dbvalue.0 0
-		}
-	}
-	control.6 {
-		iface MIXER
-		name 'ADC2 PGA gain'
-		value 0
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 31'
-			dbmin 0
-			dbmax 3100
-			dbvalue.0 0
-		}
-	}
-	control.7 {
-		iface MIXER
-		name 'ADC3 PGA gain'
-		value 0
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 31'
-			dbmin 0
-			dbmax 3100
-			dbvalue.0 0
-		}
-	}
-	control.8 {
-		iface MIXER
-		name 'ADC4 PGA gain'
-		value 0
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 31'
-			dbmin 0
-			dbmax 3100
-			dbvalue.0 0
-		}
-	}
-	control.9 {
-		iface MIXER
-		name 'CH5 digital volume'
-		value 208
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 255'
-			dbmin -11925
-			dbmax 7200
-			dbvalue.0 3675
-		}
-	}
-	control.10 {
-		iface MIXER
-		name 'CH6 digital volume'
-		value 208
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 255'
-			dbmin -11925
-			dbmax 7200
-			dbvalue.0 3675
-		}
-	}
-	control.11 {
-		iface MIXER
-		name 'CH7 digital volume'
-		value 198
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 255'
-			dbmin -11925
-			dbmax 7200
-			dbvalue.0 2925
-		}
-	}
-	control.12 {
-		iface MIXER
-		name 'CH8 digital volume'
-		value 198
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 255'
-			dbmin -11925
-			dbmax 7200
-			dbvalue.0 2925
-		}
-	}
-	control.13 {
-		iface MIXER
-		name 'ADC5 PGA gain'
-		value 0
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 31'
-			dbmin 0
-			dbmax 3100
-			dbvalue.0 0
-		}
-	}
-	control.14 {
-		iface MIXER
-		name 'ADC6 PGA gain'
-		value 0
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 31'
-			dbmin 0
-			dbmax 3100
-			dbvalue.0 0
-		}
-	}
-	control.15 {
-		iface MIXER
-		name 'ADC7 PGA gain'
-		value 0
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 31'
-			dbmin 0
-			dbmax 3100
-			dbvalue.0 0
-		}
-	}
-	control.16 {
-		iface MIXER
-		name 'ADC8 PGA gain'
-		value 0
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 31'
-			dbmin 0
-			dbmax 3100
-			dbvalue.0 0
-		}
-	}
-	control.17 {
-		iface MIXER
-		name 'DAC volume'
-		value.0 0
-		value.1 0
-		comment {
-			access 'read write'
-			type INTEGER
-			count 2
-			range '0 - 255'
-			dbmin -11925
-			dbmax 7200
-			dbvalue.0 -11925
-			dbvalue.1 -11925
-		}
-	}
-	control.18 {
-		iface MIXER
-		name 'DAC mixer gain'
-		value.0 0
-		value.1 0
-		comment {
-			access 'read write'
-			type INTEGER
-			count 2
-			range '0 - 15'
-			dbmin -600
-			dbmax 8400
-			dbvalue.0 -600
-			dbvalue.1 -600
-		}
-	}
-	control.19 {
-		iface MIXER
-		name 'digital volume'
-		value 51
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 63'
-			dbmin -7308
-			dbmax 0
-			dbvalue.0 -1392
-		}
-	}
-	control.20 {
-		iface MIXER
-		name 'Speaker Playback Volume'
-		value 25
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 31'
-			dbmin -4800
-			dbmax -150
-			dbvalue.0 -1050
-		}
-	}
-	control.21 {
-		iface MIXER
-		name 'Headphone Playback Volume'
-		value 52
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 63'
-			dbmin -6300
-			dbmax 0
-			dbvalue.0 -1100
-		}
-	}
-}
--- a/ac108_asound.state
+++ b/ac108_asound.state
@ -68,50 +68,104 @@ state.ALSA {
 state.seeed4micvoicec {
 	control.1 {
 		iface MIXER
-		name 'CH1 digital volume'
-		value 222
+		name 'OUT1 Mute'
+		value false
 		comment {
 			access 'read write'
-			type INTEGER
+			type BOOLEAN
 			count 1
-			range '0 - 255'
-			dbmin -11925
-			dbmax 7200
-			dbvalue.0 4725
 		}
 	}
 	control.2 {
 		iface MIXER
-		name 'CH2 digital volume'
-		value 222
+		name 'OUT2 Mute'
+		value true
 		comment {
 			access 'read write'
-			type INTEGER
+			type BOOLEAN
 			count 1
-			range '0 - 255'
-			dbmin -11925
-			dbmax 7200
-			dbvalue.0 4725
 		}
 	}
 	control.3 {
 		iface MIXER
-		name 'CH3 digital volume'
-		value 222
+		name 'TX1 Channel1~8 enable'
+		value '1-4 channels '
 		comment {
 			access 'read write'
-			type INTEGER
+			type ENUMERATED
 			count 1
-			range '0 - 255'
-			dbmin -11925
-			dbmax 7200
-			dbvalue.0 4725
+			item.0 'disable all'
+			item.1 '1-1 channels '
+			item.2 '1-2 channels '
+			item.3 '1-3 channels '
+			item.4 '1-4 channels '
+			item.5 '1-5 channels '
+			item.6 '1-6 channels '
+			item.7 '1-7 channels '
+			item.8 '1-8 channels '
 		}
 	}
 	control.4 {
 		iface MIXER
-		name 'CH4 digital volume'
-		value 222
+		name 'TX1 Channel9~16 enable'
+		value 'disable all'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 'disable all'
+			item.1 '8-9 channels '
+			item.2 '8-10 channels '
+			item.3 '8-11 channels '
+			item.4 '8-12 channels '
+			item.5 '8-13 channels '
+			item.6 '8-14 channels '
+			item.7 '8-15 channels '
+			item.8 '8-16 channels '
+		}
+	}
+	control.5 {
+		iface MIXER
+		name 'TX2 Channel1~8 enable'
+		value 'disable all'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 'disable all'
+			item.1 '1-1 channels '
+			item.2 '1-2 channels '
+			item.3 '1-3 channels '
+			item.4 '1-4 channels '
+			item.5 '1-5 channels '
+			item.6 '1-6 channels '
+			item.7 '1-7 channels '
+			item.8 '1-8 channels '
+		}
+	}
+	control.6 {
+		iface MIXER
+		name 'TX2 Channel9~16 enable'
+		value 'disable all'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 'disable all'
+			item.1 '8-9 channels '
+			item.2 '8-10 channels '
+			item.3 '8-11 channels '
+			item.4 '8-12 channels '
+			item.5 '8-13 channels '
+			item.6 '8-14 channels '
+			item.7 '8-15 channels '
+			item.8 '8-16 channels '
+		}
+	}
+	control.7 {
+		iface MIXER
+		name 'CH1 digital volume'
+		value 181
 		comment {
 			access 'read write'
 			type INTEGER
@ -119,55 +173,55 @@ state.seeed4micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 4725
-		}
-	}
-	control.5 {
-		iface MIXER
-		name 'ADC1 PGA gain'
-		value 0
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 31'
-			dbmin 0
-			dbmax 3100
-			dbvalue.0 0
-		}
-	}
-	control.6 {
-		iface MIXER
-		name 'ADC2 PGA gain'
-		value 0
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 31'
-			dbmin 0
-			dbmax 3100
-			dbvalue.0 0
-		}
-	}
-	control.7 {
-		iface MIXER
-		name 'ADC3 PGA gain'
-		value 0
-		comment {
-			access 'read write'
-			type INTEGER
-			count 1
-			range '0 - 31'
-			dbmin 0
-			dbmax 3100
-			dbvalue.0 0
+			dbvalue.0 1650
 		}
 	}
 	control.8 {
 		iface MIXER
-		name 'ADC4 PGA gain'
-		value 0
+		name 'CH2 digital volume'
+		value 181
+		comment {
+			access 'read write'
+			type INTEGER
+			count 1
+			range '0 - 255'
+			dbmin -11925
+			dbmax 7200
+			dbvalue.0 1650
+		}
+	}
+	control.9 {
+		iface MIXER
+		name 'CH3 digital volume'
+		value 181
+		comment {
+			access 'read write'
+			type INTEGER
+			count 1
+			range '0 - 255'
+			dbmin -11925
+			dbmax 7200
+			dbvalue.0 1650
+		}
+	}
+	control.10 {
+		iface MIXER
+		name 'CH4 digital volume'
+		value 181
+		comment {
+			access 'read write'
+			type INTEGER
+			count 1
+			range '0 - 255'
+			dbmin -11925
+			dbmax 7200
+			dbvalue.0 1650
+		}
+	}
+	control.11 {
+		iface MIXER
+		name 'ADC1 PGA gain'
+		value 27
 		comment {
 			access 'read write'
 			type INTEGER
@ -175,7 +229,725 @@ state.seeed4micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 0
+			dbvalue.0 2700
+		}
+	}
+	control.12 {
+		iface MIXER
+		name 'ADC2 PGA gain'
+		value 27
+		comment {
+			access 'read write'
+			type INTEGER
+			count 1
+			range '0 - 31'
+			dbmin 0
+			dbmax 3100
+			dbvalue.0 2700
+		}
+	}
+	control.13 {
+		iface MIXER
+		name 'ADC3 PGA gain'
+		value 27
+		comment {
+			access 'read write'
+			type INTEGER
+			count 1
+			range '0 - 31'
+			dbmin 0
+			dbmax 3100
+			dbvalue.0 2700
+		}
+	}
+	control.14 {
+		iface MIXER
+		name 'ADC4 PGA gain'
+		value 27
+		comment {
+			access 'read write'
+			type INTEGER
+			count 1
+			range '0 - 31'
+			dbmin 0
+			dbmax 3100
+			dbvalue.0 2700
+		}
+	}
+	control.15 {
+		iface MIXER
+		name 'Tx1 Channels'
+		value '4 channels '
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1 channels '
+			item.1 '2 channels '
+			item.2 '3 channels '
+			item.3 '4 channels '
+			item.4 '5 channels '
+			item.5 '6 channels '
+			item.6 '7 channels '
+			item.7 '8 channels '
+			item.8 '9 channels '
+			item.9 '10 channels '
+			item.10 '11 channels '
+			item.11 '12 channels '
+			item.12 '13 channels '
+			item.13 '14 channels '
+			item.14 '15 channels '
+			item.15 '16 channels '
+		}
+	}
+	control.16 {
+		iface MIXER
+		name 'Tx2 Channels'
+		value '1 channels '
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1 channels '
+			item.1 '2 channels '
+			item.2 '3 channels '
+			item.3 '4 channels '
+			item.4 '5 channels '
+			item.5 '6 channels '
+			item.6 '7 channels '
+			item.7 '8 channels '
+			item.8 '9 channels '
+			item.9 '10 channels '
+			item.10 '11 channels '
+			item.11 '12 channels '
+			item.12 '13 channels '
+			item.13 '14 channels '
+			item.14 '15 channels '
+			item.15 '16 channels '
+		}
+	}
+	control.17 {
+		iface MIXER
+		name 'Tx1 Channels 1 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.18 {
+		iface MIXER
+		name 'Tx1 Channels 2 MAP'
+		value '2st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.19 {
+		iface MIXER
+		name 'Tx1 Channels 3 MAP'
+		value '3st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.20 {
+		iface MIXER
+		name 'Tx1 Channels 4 MAP'
+		value '4st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.21 {
+		iface MIXER
+		name 'Tx1 Channels 5 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.22 {
+		iface MIXER
+		name 'Tx1 Channels 6 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.23 {
+		iface MIXER
+		name 'Tx1 Channels 7 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.24 {
+		iface MIXER
+		name 'Tx1 Channels 8 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.25 {
+		iface MIXER
+		name 'Tx1 Channels 9 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.26 {
+		iface MIXER
+		name 'Tx1 Channels 10 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.27 {
+		iface MIXER
+		name 'Tx1 Channels 11 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.28 {
+		iface MIXER
+		name 'Tx1 Channels 12 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.29 {
+		iface MIXER
+		name 'Tx1 Channels 13 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.30 {
+		iface MIXER
+		name 'Tx1 Channels 14 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.31 {
+		iface MIXER
+		name 'Tx1 Channels 15 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.32 {
+		iface MIXER
+		name 'Tx1 Channels 16 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.33 {
+		iface MIXER
+		name 'Tx2 Channels 1 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.34 {
+		iface MIXER
+		name 'Tx2 Channels 2 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.35 {
+		iface MIXER
+		name 'Tx2 Channels 3 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.36 {
+		iface MIXER
+		name 'Tx2 Channels 4 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.37 {
+		iface MIXER
+		name 'Tx2 Channels 5 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.38 {
+		iface MIXER
+		name 'Tx2 Channels 6 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.39 {
+		iface MIXER
+		name 'Tx2 Channels 7 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.40 {
+		iface MIXER
+		name 'Tx2 Channels 8 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.41 {
+		iface MIXER
+		name 'Tx2 Channels 9 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.42 {
+		iface MIXER
+		name 'Tx2 Channels 10 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.43 {
+		iface MIXER
+		name 'Tx2 Channels 11 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.44 {
+		iface MIXER
+		name 'Tx2 Channels 12 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.45 {
+		iface MIXER
+		name 'Tx2 Channels 13 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.46 {
+		iface MIXER
+		name 'Tx2 Channels 14 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.47 {
+		iface MIXER
+		name 'Tx2 Channels 15 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.48 {
+		iface MIXER
+		name 'Tx2 Channels 16 MAP'
+		value '1st adc sample'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 '1st adc sample'
+			item.1 '2st adc sample'
+			item.2 '3st adc sample'
+			item.3 '4st adc sample'
+		}
+	}
+	control.49 {
+		iface MIXER
+		name 'ADC4 Source'
+		value 'Analog ADC4'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 'Analog ADC1'
+			item.1 'Analog ADC2'
+			item.2 'Analog ADC3'
+			item.3 'Analog ADC4'
+		}
+	}
+	control.50 {
+		iface MIXER
+		name 'ADC3 Source'
+		value 'Analog ADC3'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 'Analog ADC1'
+			item.1 'Analog ADC2'
+			item.2 'Analog ADC3'
+			item.3 'Analog ADC4'
+		}
+	}
+	control.51 {
+		iface MIXER
+		name 'ADC2 Source'
+		value 'Analog ADC2'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 'Analog ADC1'
+			item.1 'Analog ADC2'
+			item.2 'Analog ADC3'
+			item.3 'Analog ADC4'
+		}
+	}
+	control.52 {
+		iface MIXER
+		name 'ADC1 Source'
+		value 'Analog ADC1'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 'Analog ADC1'
+			item.1 'Analog ADC2'
+			item.2 'Analog ADC3'
+			item.3 'Analog ADC4'
+		}
+	}
+	control.53 {
+		iface MIXER
+		name 'ADC1 Digital Mixer gc'
+		value 'disable all'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 'disable all'
+			item.1 'ADC1 data'
+			item.2 'ADC2 data'
+			item.3 'ADC3 data'
+			item.4 'ADC4 data'
+		}
+	}
+	control.54 {
+		iface MIXER
+		name 'ADC1 Digital Mixer src'
+		value 'ADC1 data'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 'disable all'
+			item.1 'ADC1 data'
+			item.2 'ADC2 data'
+			item.3 'ADC3 data'
+			item.4 'ADC4 data'
+		}
+	}
+	control.55 {
+		iface MIXER
+		name 'ADC2 Digital Mixer gc'
+		value 'disable all'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 'disable all'
+			item.1 'ADC1 data'
+			item.2 'ADC2 data'
+			item.3 'ADC3 data'
+			item.4 'ADC4 data'
+		}
+	}
+	control.56 {
+		iface MIXER
+		name 'ADC2 Digital Mixer src'
+		value 'ADC2 data'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 'disable all'
+			item.1 'ADC1 data'
+			item.2 'ADC2 data'
+			item.3 'ADC3 data'
+			item.4 'ADC4 data'
+		}
+	}
+	control.57 {
+		iface MIXER
+		name 'ADC3 Digital Mixer gc'
+		value 'disable all'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 'disable all'
+			item.1 'ADC1 data'
+			item.2 'ADC2 data'
+			item.3 'ADC3 data'
+			item.4 'ADC4 data'
+		}
+	}
+	control.58 {
+		iface MIXER
+		name 'ADC3 Digital Mixer src'
+		value 'ADC3 data'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 'disable all'
+			item.1 'ADC1 data'
+			item.2 'ADC2 data'
+			item.3 'ADC3 data'
+			item.4 'ADC4 data'
+		}
+	}
+	control.59 {
+		iface MIXER
+		name 'ADC4 Digital Mixer gc'
+		value 'disable all'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 'disable all'
+			item.1 'ADC1 data'
+			item.2 'ADC2 data'
+			item.3 'ADC3 data'
+			item.4 'ADC4 data'
+		}
+	}
+	control.60 {
+		iface MIXER
+		name 'ADC4 Digital Mixer src'
+		value 'ADC4 data'
+		comment {
+			access 'read write'
+			type ENUMERATED
+			count 1
+			item.0 'disable all'
+			item.1 'ADC1 data'
+			item.2 'ADC2 data'
+			item.3 'ADC3 data'
+			item.4 'ADC4 data'
 		}
 	}
 }
--- a/ac10x.h
+++ b/ac10x.h
@ -1,126 +0,0 @@
-/*
- * ac10x.h
- *
- * (C) Copyright 2017-2018
- * Seeed Technology Co., Ltd. <www.seeedstudio.com>
- *
- * PeterYang <linsheng.yang@seeed.cc>
- *
- * (C) Copyright 2010-2017
- * Reuuimlla Technology Co., Ltd. <www.reuuimllatech.com>
- * huangxin <huangxin@reuuimllatech.com>
- *
- * some simple description for this code
- *
- * 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 __AC10X_H__
-#define __AC10X_H__
-
-#define AC101_I2C_ID		4
-#define _MASTER_AC108		0
-#define _MASTER_AC101		1
-#define _MASTER_MULTI_CODEC	_MASTER_AC101
-
-/* enable headset detecting & headset button pressing */
-#define CONFIG_AC101_SWITCH_DETECT
-
-/* obsolete */
-#define CONFIG_AC10X_TRIG_LOCK	0
-
-
-#ifdef AC101_DEBG
-    #define AC101_DBG(format,args...)  printk("[AC101] %s() L%d " format, __func__, __LINE__, ##args)
-#else
-    #define AC101_DBG(...)
-#endif
-
-
-#include "sound-compatible-4.18.h"
-
-#ifdef CONFIG_AC101_SWITCH_DETECT
-enum headphone_mode_u {
-	HEADPHONE_IDLE,
-	FOUR_HEADPHONE_PLUGIN,
-	THREE_HEADPHONE_PLUGIN,
-};
-#endif
-
-struct ac10x_priv {
-	struct i2c_client *i2c[4];
-	struct regmap* i2cmap[4];
-	int codec_cnt;
-	unsigned sysclk;
-#define _FREQ_24_576K		24576000
-#define _FREQ_22_579K		22579200
-	unsigned mclk;	/* master clock or aif_clock/aclk */
-	int clk_id;
-	unsigned char i2s_mode;
-	unsigned char data_protocol;
-	struct delayed_work dlywork;
-	int tdm_chips_cnt;
-	int sysclk_en;
-
-	/* member for ac101 .begin */
-	struct snd_soc_codec *codec;
-	struct i2c_client *i2c101;
-	struct regmap* regmap101;
-
-	struct mutex dac_mutex;
-	u8 dac_enable;
-	spinlock_t lock;
-	u8 aif1_clken;
-
-	struct work_struct codec_resume;
-	struct gpio_desc* gpiod_spk_amp_gate;
-
-	#ifdef CONFIG_AC101_SWITCH_DETECT
-	struct gpio_desc* gpiod_irq;
-	long irq;
-	volatile int irq_cntr;
-	volatile int pullout_cntr;
-	volatile int state;
-
-	enum headphone_mode_u mode;
-	struct work_struct work_switch;
-	struct work_struct work_clear_irq;
-
-	struct input_dev* inpdev;
-	#endif
-	/* member for ac101 .end */
-};
-
-
-/* AC101 DAI operations */
-int ac101_audio_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai);
-void ac101_aif_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai);
-int ac101_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt);
-int ac101_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params,
-	struct snd_soc_dai *codec_dai);
-int ac101_trigger(struct snd_pcm_substream *substream, int cmd,
-	 	  struct snd_soc_dai *dai);
-int ac101_aif_mute(struct snd_soc_dai *codec_dai, int mute);
-
-/* codec driver specific */
-int ac101_codec_probe(struct snd_soc_codec *codec);
-int ac101_codec_remove(struct snd_soc_codec *codec);
-int ac101_codec_suspend(struct snd_soc_codec *codec);
-int ac101_codec_resume(struct snd_soc_codec *codec);
-int ac101_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level);
-
-/* i2c device specific */
-int ac101_probe(struct i2c_client *i2c, const struct i2c_device_id *id);
-void ac101_shutdown(struct i2c_client *i2c);
-int ac101_remove(struct i2c_client *i2c);
-
-/* seeed voice card export */
-int seeed_voice_card_register_set_clock(int stream, int (*set_clock)(int, struct snd_pcm_substream *, int, struct snd_soc_dai *));
-
-int ac10x_fill_regcache(struct device* dev, struct regmap* map);
-
-#endif//__AC10X_H__
--- a/asound_2mic.conf
+++ b/asound_2mic.conf
@ -1,15 +1,11 @@
 # The IPC key of dmix or dsnoop plugin must be unique
 # If 555555 or 666666 is used by other processes, use another one

-
-# use samplerate to resample as speexdsp resample is bad
-defaults.pcm.rate_converter "samplerate"
-
-pcm.!default {
-    type asym
-    playback.pcm "playback"
-    capture.pcm "capture"
-}
+# pcm.!default {
+#     type asym
+#     playback.pcm "playback"
+#     capture.pcm "capture"
+# }

 pcm.playback {
    type plug
@ -23,16 +19,21 @@ pcm.capture {

 pcm.dmixed {
    type dmix
-    slave.pcm "hw:seeed2micvoicec"
+    slave.pcm "hw:0,0"
    ipc_key 555555 
 }

 pcm.array {
    type dsnoop
    slave {
-        pcm "hw:seeed2micvoicec"
+        pcm "hw:0,0"
        channels 2
    }
    ipc_key 666666
 }

+pcm.ac108 {
+        type ac108
+        slavepcm "hw:1,0"
+        channels 4
+}
--- a/asound_4mic.conf
+++ b/asound_4mic.conf
@ -1,33 +0,0 @@
-# The IPC key of dmix or dsnoop plugin must be unique
-# If 555555 or 666666 is used by other processes, use another one
-
-# use samplerate to resample as speexdsp resample is bad
-defaults.pcm.rate_converter "samplerate"
-
-pcm.!default {
-    type asym
-    playback.pcm "playback"
-    capture.pcm "ac108"
-}
-
-pcm.playback {
-    type plug
-    slave.pcm "hw:ALSA"
-}
-
-# pcm.dmixed {
-#     type dmix
-#     slave.pcm "hw:0,0"
-#     ipc_key 555555 
-# }
-
-pcm.ac108 {
-    type plug
-    slave.pcm "hw:seeed4micvoicec"
-}
-
-# pcm.multiapps {
-#     type dsnoop
-#     ac108-slavepcm "hw:1,0"
-#     ipc_key 666666
-# }
--- a/asound_6mic.conf
+++ b/asound_6mic.conf
@ -1,90 +0,0 @@
-# The IPC key of dmix or dsnoop plugin must be unique
-# If 555555 or 666666 is used by other processes, use another one
-
-# use samplerate to resample as speexdsp resample is broken
-defaults.pcm.rate_converter "samplerate"
-
-pcm.!default {
-    type asym
-    playback.pcm "dmixer"
-    capture.pcm "ac108"
-}
-
-
-pcm.ac108 {
-    type plug
-    slave {
-        rate 48000
-        format S32_LE
-        pcm "hw:seeed8micvoicec"
-    }
-}
-
-# pcm.multiapps {
-#     type plug
-#     slave.pcm {
-#         type dsnoop
-#         slave {
-#             rate 48000
-#             format S32_LE
-#             pcm "hw:seeed8micvoicec"
-#         }
-#         ipc_key 666666
-#     }
-# }
-
-pcm.dmixer {
-    type plug
-    slave {
-        pcm {
-            type dmix
-            ipc_key 555555
-            slave {	
-                pcm "hw:seeed8micvoicec"
-                format S32_LE
-                channels 8
-            }
-            bindings {
-                0 0
-                1 1
-                2 2
-                3 3
-                4 4
-                5 5
-                6 6
-                7 7
-            }
-        }
-        channels 8
-        format S32_LE
-        rate 48000
-    }
-    ttable.0.0 1
-    ttable.1.1 1
-    ttable.0.2 1
-    ttable.1.3 1
-    ttable.0.4 1
-    ttable.1.5 1
-    ttable.0.6 1
-    ttable.1.7 1
-}
-
-pcm.ac101 {
-    type plug
-    slave {
-        pcm "hw:seeed8micvoicec"
-        channels 8
-        format S32_LE
-        rate 48000
-    }
-    ttable.0.0 1
-    ttable.1.1 1
-    ttable.0.2 1
-    ttable.1.3 1
-    ttable.0.4 1
-    ttable.1.5 1
-    ttable.0.6 1
-    ttable.1.7 1
-}
-
-
--- a/builddtbo.sh
+++ b/builddtbo.sh
@ -1,10 +1,7 @@
-#!/bin/sh
 #dtoverlay -r seeed-2mic-voicecard
-DTC_FLAGS="-b 0 -Wno-unit_address_vs_reg -I dts -O dtb"

-dtc -@ $DTC_FLAGS -o seeed-2mic-voicecard.dtbo seeed-2mic-voicecard-overlay.dts
-dtc -@ $DTC_FLAGS -o seeed-4mic-voicecard.dtbo seeed-4mic-voicecard-overlay.dts
-dtc -@ $DTC_FLAGS -o seeed-8mic-voicecard.dtbo seeed-8mic-voicecard-overlay.dts
+dtc -@ -I dts -O dtb -o seeed-2mic-voicecard.dtbo seeed-2mic-voicecard-overlay.dts
+dtc -@ -I dts -O dtb -o seeed-4mic-voicecard.dtbo seeed-4mic-voicecard-overlay.dts

 # cp *.dtbo /boot/overlays
 # dtoverlay seeed-2mic-voicecard
--- a/dkms.conf
+++ b/dkms.conf
@ -1,15 +1,7 @@
 PACKAGE_NAME="seeed-voicecard"
-PACKAGE_VERSION="0.3"
+PACKAGE_VERSION="0.2"
 BUILT_MODULE_NAME[0]="snd-soc-wm8960"
 BUILT_MODULE_NAME[1]="snd-soc-ac108"
-BUILT_MODULE_NAME[2]="snd-soc-seeed-voicecard"
 DEST_MODULE_LOCATION[0]="/kernel/sound/soc/codecs"
 DEST_MODULE_LOCATION[1]="/kernel/sound/soc/codecs"
-DEST_MODULE_LOCATION[2]="/kernel/sound/soc/bcm"
-PATCH[0]="back-to-v4.19.diff"
-PATCH[1]="back-to-v5.4.diff"
-PATCH[2]="back-to-v5.8.diff"
-PATCH_MATCH[0]="^4\.19\.*"
-PATCH_MATCH[1]="^5\.4\.*"
-PATCH_MATCH[2]="^5\.8\.*"
 AUTOINSTALL="yes"
--- a/install.sh
+++ b/install.sh
@ -5,134 +5,34 @@ if [[ $EUID -ne 0 ]]; then
   exit 1
 fi

-# Check for enough space on /boot volume
-boot_line=$(df -h | grep /boot | head -n 1)
-if [ "x${boot_line}" = "x" ]; then
-  echo "Warning: /boot volume not found .."
-else
-  boot_space=$(echo $boot_line | awk '{print $4;}')
-  free_space=$(echo "${boot_space%?}")
-  unit="${boot_space: -1}"
-  if [[ "$unit" = "K" ]]; then
-    echo "Error: Not enough space left ($boot_space) on /boot"
-    exit 1
-  elif [[ "$unit" = "M" ]]; then
-    if [ "$free_space" -lt "25" ]; then
-      echo "Error: Not enough space left ($boot_space) on /boot"
-      exit 1
-    fi
-  fi
-fi
-
-#
-# make sure that we are on something ARM/Raspberry related
-# either a bare metal Raspberry or a qemu session with
-# Raspberry stuff available
-# - check for /boot/overlays
-# - dtparam and dtoverlay is available
-errorFound=0
-OVERLAYS=/boot/overlays
-[ -d /boot/firmware/overlays ] && OVERLAYS=/boot/firmware/overlays
-
-if [ ! -d $OVERLAYS ] ; then
-  echo "$OVERLAYS not found or not a directory" 1>&2
-  errorFound=1
-fi
-# should we also check for alsactl and amixer used in seeed-voicecard?
-PATH=$PATH:/opt/vc/bin
-for cmd in dtparam dtoverlay ; do
-  if ! which $cmd &>/dev/null ; then
-    echo "$cmd not found" 1>&2
-    echo "You may need to run ./ubuntu-prerequisite.sh"
-    errorFound=1
-  fi
-done
-if [ $errorFound = 1 ] ; then
-  echo "Errors found, exiting." 1>&2
+is_Raspberry=$(cat /proc/device-tree/model | awk  '{print $1}')
+if [ "x${is_Raspberry}" != "xRaspberry" ] ; then
+  echo "Sorry, this drivers only works on raspberry pi"
  exit 1
 fi

-ver="0.3"
-uname_r=$(uname -r)
+ver="0.2"
+card=$1
+
+if [ "x${card}" = "x" ] ; then
+  echo "Usage: ./install 2mic|4mic"
+  exit 1
+fi

 # we create a dir with this version to ensure that 'dkms remove' won't delete
 # the sources during kernel updates
 marker="0.0.0"

-_VER_RUN=
-function get_kernel_version() {
-  local ZIMAGE IMG_OFFSET
-
-  _VER_RUN=""
-  [ -z "$_VER_RUN" ] && {
-    ZIMAGE=/boot/kernel.img
-    [ -f /boot/firmware/vmlinuz ] && ZIMAGE=/boot/firmware/vmlinuz
-    # 64-bit-only kernel package
-    [ ! -f /boot/kernel.img ] && [ -f /boot/kernel8.img ] && ZIMAGE=/boot/kernel8.img
-    IMG_OFFSET=$(LC_ALL=C grep -abo $'\x1f\x8b\x08\x00' $ZIMAGE | head -n 1 | cut -d ':' -f 1)
-    _VER_RUN=$(dd if=$ZIMAGE obs=64K ibs=4 skip=$(( IMG_OFFSET / 4)) 2>/dev/null | zcat | grep -a -m1 "Linux version" | LC_ALL=C sed -e 's/^.*Linux/Linux/' | strings | awk '{ print $3; }')
-  }
-  echo "$_VER_RUN"
-  return 0
-}
-
-function check_kernel_headers() {
-  VER_RUN=$(get_kernel_version)
-  VER_HDR=$(dpkg -L raspberrypi-kernel-headers | egrep -m1 "/lib/modules/[^\/]+/build" | awk -F'/' '{ print $4; }')
-  [ "X$VER_RUN" == "X$VER_HDR" ] && {
-    return 0
-  }
-  VER_HDR=$(dpkg -L linux-headers-$VER_RUN | egrep -m1 "/lib/modules/[^\/]+/build" | awk -F'/' '{ print $4; }')
-  [ "X$VER_RUN" == "X$VER_HDR" ] && {
-    return 0
-  }
-
-  # echo RUN=$VER_RUN HDR=$VER_HDR
-  echo " !!! Your kernel version is $VER_RUN"
-  echo "     Not found *** corresponding *** kernel headers with apt-get."
-  echo "     This may occur if you have ran 'rpi-update'."
-  echo " Choose  *** y *** will revert the kernel to version $VER_HDR then continue."
-  echo " Choose  *** N *** will exit without this driver support, by default."
-  read -p "Would you like to proceed? (y/N)" -n 1 -r -s
-  echo
-  if ! [[ $REPLY =~ ^[Yy]$ ]]; then
-    exit 1;
-  fi
-
-  apt-get -y --reinstall install raspberrypi-kernel
-}
-
-# update and install required packages
-which apt &>/dev/null
-if [[ $? -eq 0 ]]; then
-  apt update -y
-  # Raspbian kernel packages
+apt update
 apt-get -y install raspberrypi-kernel-headers raspberrypi-kernel
-  # Recent Raspbian has 64-bit kernel on 32-bit userspace
-  apt-get -y install gcc-aarch64-linux-gnu
-  # Ubuntu kernel packages
-  apt-get -y install linux-raspi linux-headers-raspi linux-image-raspi
-  apt-get -y install dkms git i2c-tools libasound2-plugins
-  # rpi-update checker
-  check_kernel_headers
-fi
-
-# Arch Linux
-which pacman &>/dev/null
-if [[ $? -eq 0 ]]; then
-  pacman -Syu --needed git gcc automake make dkms linux-raspberrypi-headers i2c-tools
-fi
+apt-get -y install  dkms

 # locate currently installed kernels (may be different to running kernel if
 # it's just been updated)
-base_ver=$(get_kernel_version)
-base_ver=${base_ver%%[-+]*}
-#kernels="${base_ver}+ ${base_ver}-v7+ ${base_ver}-v7l+"
-kernels=$(uname -r)
+kernels=$(ls /lib/modules | sed "s/^/-k /")
+uname_r=$(uname -r)

 function install_module {
-  local _i
-
  src=$1
  mod=$2

@ -141,81 +41,67 @@ function install_module {
  fi

  if [[ -e /usr/src/$mod-$ver || -e /var/lib/dkms/$mod/$ver ]]; then
-    dkms remove --force -m $mod -v $ver --all
+    dkms remove -m $mod -v $ver --all
    rm -rf /usr/src/$mod-$ver
  fi
-
  mkdir -p /usr/src/$mod-$ver
  cp -a $src/* /usr/src/$mod-$ver/
-
  dkms add -m $mod -v $ver
-  for _i in $kernels; do
-    dkms build -k $_i -m $mod -v $ver && {
-      dkms install --force -k $_i -m $mod -v $ver
-    }
-  done
+  dkms build $kernels -m $mod -v $ver && dkms install $kernels -m $mod -v $ver

  mkdir -p /var/lib/dkms/$mod/$ver/$marker
 }
-
+if [ ! -f "/boot/overlays/seeed-4mic-voicecard.dtbo" ] && [ ! -f "/lib/modules/${uname_r}/kernel/sound/soc/codecs/snd-soc-ac108.ko" ] ; then
  install_module "./" "seeed-voicecard"
+  cp seeed-2mic-voicecard.dtbo /boot/overlays
+  cp seeed-4mic-voicecard.dtbo /boot/overlays
+  cp ac108_plugin/libasound_module_pcm_ac108.so   /usr/lib/arm-linux-gnueabihf/alsa-lib/  
+  cp asound.conf /etc/
+else
+  echo "card driver already installed"
+fi

-
-# install dtbos
-cp seeed-2mic-voicecard.dtbo $OVERLAYS
-cp seeed-4mic-voicecard.dtbo $OVERLAYS
-cp seeed-8mic-voicecard.dtbo $OVERLAYS
-
-#install alsa plugins
-# no need this plugin now
-# install -D ac108_plugin/libasound_module_pcm_ac108.so /usr/lib/arm-linux-gnueabihf/alsa-lib/
-rm -f /usr/lib/arm-linux-gnueabihf/alsa-lib/libasound_module_pcm_ac108.so
-
-#set kernel modules
-grep -q "^snd-soc-seeed-voicecard$" /etc/modules || \
-  echo "snd-soc-seeed-voicecard" >> /etc/modules
-grep -q "^snd-soc-ac108$" /etc/modules || \
+grep -q "snd-soc-ac108" /etc/modules || \
  echo "snd-soc-ac108" >> /etc/modules
-grep -q "^snd-soc-wm8960$" /etc/modules || \
+grep -q "snd-soc-wm8960" /etc/modules || \
  echo "snd-soc-wm8960" >> /etc/modules  

-#set dtoverlays
-CONFIG=/boot/config.txt
-[ -f /boot/firmware/config.txt ] && CONFIG=/boot/firmware/config.txt
-[ -f /boot/firmware/usercfg.txt ] && CONFIG=/boot/firmware/usercfg.txt

-sed -i -e 's:#dtparam=i2c_arm=on:dtparam=i2c_arm=on:g'  $CONFIG || true
-grep -q "^dtoverlay=i2s-mmap$" $CONFIG || \
-  echo "dtoverlay=i2s-mmap" >> $CONFIG
+grep -q "dtoverlay=i2s-mmap" /boot/config.txt || \
+  echo "dtoverlay=i2s-mmap" >> /boot/config.txt


-grep -q "^dtparam=i2s=on$" $CONFIG || \
-  echo "dtparam=i2s=on" >> $CONFIG
+grep -q "dtparam=i2s=on" /boot/config.txt || \
+  echo "dtparam=i2s=on" >> /boot/config.txt

-#install config files
-mkdir /etc/voicecard || true
-cp *.conf /etc/voicecard
-cp *.state /etc/voicecard
-
-#create git repo
-git_email=$(git config --global --get user.email)
-git_name=$(git config --global --get user.name)
-if [ "x${git_email}" == "x" ] || [ "x${git_name}" == "x" ] ; then
-    echo "setup git config"
-    git config --global user.email "respeaker@seeed.cc"
-    git config --global user.name "respeaker"
+has_2mic=$(grep seeed-2mic-voicecard /boot/config.txt)
+has_4mic=$(grep seeed-4mic-voicecard /boot/config.txt)
+case "${card}" in
+   "2mic") 
+    echo "cp wm8960_asound.state /var/lib/alsa/asound.state"
+    cp wm8960_asound.state /var/lib/alsa/asound.state
+    if [ "x${has_4mic}" != x ] ; then
+      echo "has 4mic before, now remove it"
+      sed -i "s/dtoverlay=seeed-4mic-voicecard//g" /boot/config.txt
    fi
-echo "git init"
-git --git-dir=/etc/voicecard/.git init
-echo "git add --all"
-git --git-dir=/etc/voicecard/.git --work-tree=/etc/voicecard/ add --all
-echo "git commit -m \"origin configures\""
-git --git-dir=/etc/voicecard/.git --work-tree=/etc/voicecard/ commit  -m "origin configures"
+    grep -q "dtoverlay=seeed-2mic-voicecard" /boot/config.txt || \
+      echo "dtoverlay=seeed-2mic-voicecard" >> /boot/config.txt
      
-cp seeed-voicecard /usr/bin/
-cp seeed-voicecard.service /lib/systemd/system/
-systemctl enable  seeed-voicecard.service
-systemctl start   seeed-voicecard
+   ;;
+   "4mic") 
+    echo "cp ac108_asound.state /var/lib/alsa/asound.state"
+    cp ac108_asound.state /var/lib/alsa/asound.state
+    if [ "x${has_2mic}" != x ] ; then
+      echo "has 2mic before, now remove it"
+      sed -i "s/dtoverlay=seeed-2mic-voicecard//g" /boot/config.txt
+    fi
+    grep -q "dtoverlay=seeed-4mic-voicecard" /boot/config.txt || \
+      echo "dtoverlay=seeed-4mic-voicecard" >> /boot/config.txt    
+   ;;
+   *) 
+    echo "Please use 2mic or 4mic"
+   ;;
+esac

 echo "------------------------------------------------------"
 echo "Please reboot your raspberry pi to apply all settings"
--- a/patches/back-to-v4.19.diff
+++ b/patches/back-to-v4.19.diff
@ -1,454 +0,0 @@
-diff --git a/ac101.c b/ac101.c
-index 23837a7..41c15f3 100644
--- a/ac101.c
-+++ b/ac101.c
-@@ -955,10 +955,10 @@ void ac101_aif_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai
- 
- 	AC101_DBG("stream = %s, play: %d, capt: %d, active: %d\n", 
- 		snd_pcm_stream_str(substream),
-		codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE],
-		snd_soc_dai_active(codec_dai));
-+		codec_dai->playback_active, codec_dai->capture_active,
-+		codec_dai->active);
- 
-	if (!snd_soc_dai_active(codec_dai)) {
-+	if (!codec_dai->active) {
- 		ac10x->aif1_clken = 1;
- 		ac101_aif1clk(codec, SND_SOC_DAPM_POST_PMD, 0);
- 	} else {
-@@ -1080,7 +1080,7 @@ int ac101_hw_params(struct snd_pcm_substream *substream,
- 	freq_out = _FREQ_24_576K;
- 	for (i = 0; i < ARRAY_SIZE(codec_aif1_fs); i++) {
- 		if (codec_aif1_fs[i].samp_rate == params_rate(params)) {
-			if (codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] && dmic_used && codec_aif1_fs[i].samp_rate == 44100) {
-+			if (codec_dai->capture_active && dmic_used && codec_aif1_fs[i].samp_rate == 44100) {
- 				ac101_update_bits(codec, AIF_SR_CTRL, (0xf<<AIF1_FS), (0x4<<AIF1_FS));
- 			} else {
- 				ac101_update_bits(codec, AIF_SR_CTRL, (0xf<<AIF1_FS), ((codec_aif1_fs[i].srbit)<<AIF1_FS));
-diff --git a/ac108.c b/ac108.c
-index d5dd12d..0a8c462 100644
--- a/ac108.c
-+++ b/ac108.c
-@@ -653,7 +653,7 @@ static int ac108_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_h
- 
- 	dev_dbg(dai->dev, "%s() stream=%s play:%d capt:%d +++\n", __func__,
- 			snd_pcm_stream_str(substream),
-			dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]);
-+			dai->playback_active, dai->capture_active);
- 
- 	if (ac10x->i2c101) {
- 		ret = ac101_hw_params(substream, params, dai);
-@@ -664,8 +664,8 @@ static int ac108_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_h
- 		}
- 	}
- 
-	if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE && dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])
-	 || (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && dai->stream_active[SNDRV_PCM_STREAM_CAPTURE])) {
-+	if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE && dai->playback_active)
-+	 || (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && dai->capture_active)) {
- 		/* not configure hw_param twice */
- 		/* return 0; */
- 	}
-@@ -810,9 +810,6 @@ static int ac108_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int fr
- 
- 	struct ac10x_priv *ac10x = snd_soc_dai_get_drvdata(dai);
- 
-	if (freq != 24000000 || clk_id != SYSCLK_SRC_PLL)
-		dev_warn(dai->dev, "ac108_set_sysclk freq = %d clk = %d\n", freq, clk_id);
-
- 	freq = 24000000;
- 	clk_id = SYSCLK_SRC_PLL;
- 
-@@ -1124,7 +1121,7 @@ void ac108_aif_shutdown(struct snd_pcm_substream *substream,
- 	}
- }
- 
-int ac108_aif_mute(struct snd_soc_dai *dai, int mute, int direction) {
-+int ac108_aif_mute(struct snd_soc_dai *dai, int mute) {
- 	struct snd_soc_codec *codec = dai->codec;
- 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
- 
-@@ -1145,13 +1142,12 @@ static const struct snd_soc_dai_ops ac108_dai_ops = {
- 	.hw_params	= ac108_hw_params,
- 	.prepare	= ac108_prepare,
- 	.trigger	= ac108_trigger,
-	.mute_stream	= ac108_aif_mute,
-+	.digital_mute	= ac108_aif_mute,
- 
- 	/*DAI format configuration*/
- 	.set_fmt	= ac108_set_fmt,
- 
- 	// .hw_free = ac108_hw_free,
-	.no_capture_mute = 1,
- };
- 
- static  struct snd_soc_dai_driver ac108_dai0 = {
-diff --git a/seeed-voicecard.c b/seeed-voicecard.c
-index b90af93..af6db74 100644
--- a/seeed-voicecard.c
-+++ b/seeed-voicecard.c
-@@ -28,8 +28,6 @@
- #include <sound/simple_card_utils.h>
- #include "ac10x.h"
- 
-#define LINUX_VERSION_IS_GEQ(x1,x2,x3)	(LINUX_VERSION_CODE >= KERNEL_VERSION(x1,x2,x3))
-
- /*
-  * single codec:
-  *	0 - allow multi codec
-@@ -42,9 +40,6 @@ struct seeed_card_data {
- 	struct seeed_dai_props {
- 		struct asoc_simple_dai cpu_dai;
- 		struct asoc_simple_dai codec_dai;
-		struct snd_soc_dai_link_component cpus;   /* single cpu */
-		struct snd_soc_dai_link_component codecs; /* single codec */
-		struct snd_soc_dai_link_component platforms;
- 		unsigned int mclk_fs;
- 	} *dai_props;
- 	unsigned int mclk_fs;
-@@ -97,16 +92,16 @@ static int seeed_voice_card_startup(struct snd_pcm_substream *substream)
- 	if (ret)
- 		clk_disable_unprepare(dai_props->cpu_dai.clk);
- 
-	if (asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min) {
-		priv->channels_playback_default = asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min;
-+	if (rtd->cpu_dai->driver->playback.channels_min) {
-+		priv->channels_playback_default = rtd->cpu_dai->driver->playback.channels_min;
- 	}
-	if (asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min) {
-		priv->channels_capture_default = asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min;
-+	if (rtd->cpu_dai->driver->capture.channels_min) {
-+		priv->channels_capture_default = rtd->cpu_dai->driver->capture.channels_min;
- 	}
-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min = priv->channels_playback_override;
-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_max = priv->channels_playback_override;
-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min = priv->channels_capture_override;
-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_max = priv->channels_capture_override;
-+	rtd->cpu_dai->driver->playback.channels_min = priv->channels_playback_override;
-+	rtd->cpu_dai->driver->playback.channels_max = priv->channels_playback_override;
-+	rtd->cpu_dai->driver->capture.channels_min = priv->channels_capture_override;
-+	rtd->cpu_dai->driver->capture.channels_max = priv->channels_capture_override;
- 
- 	return ret;
- }
-@@ -118,10 +113,10 @@ static void seeed_voice_card_shutdown(struct snd_pcm_substream *substream)
- 	struct seeed_dai_props *dai_props =
- 		seeed_priv_to_props(priv, rtd->num);
- 
-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min = priv->channels_playback_default;
-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_max = priv->channels_playback_default;
-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min = priv->channels_capture_default;
-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_max = priv->channels_capture_default;
-+	rtd->cpu_dai->driver->playback.channels_min = priv->channels_playback_default;
-+	rtd->cpu_dai->driver->playback.channels_max = priv->channels_playback_default;
-+	rtd->cpu_dai->driver->capture.channels_min = priv->channels_capture_default;
-+	rtd->cpu_dai->driver->capture.channels_max = priv->channels_capture_default;
- 
- 	clk_disable_unprepare(dai_props->cpu_dai.clk);
- 
-@@ -132,8 +127,8 @@ static int seeed_voice_card_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 = asoc_rtd_to_codec(rtd, 0);
-	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
-+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- 	struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
- 	struct seeed_dai_props *dai_props =
- 		seeed_priv_to_props(priv, rtd->num);
-@@ -197,7 +192,7 @@ static void work_cb_codec_clk(struct work_struct *work)
- static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd)
- {
- 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
-+	struct snd_soc_dai *dai = rtd->codec_dai;
- 	struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
- 	#if CONFIG_AC10X_TRIG_LOCK
- 	unsigned long flags;
-@@ -206,7 +201,7 @@ static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd
- 
- 	dev_dbg(rtd->card->dev, "%s() stream=%s  cmd=%d play:%d, capt:%d\n",
- 		__FUNCTION__, snd_pcm_stream_str(substream), cmd,
-		dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]);
-+		dai->playback_active, dai->capture_active);
- 
- 	switch (cmd) {
- 	case SNDRV_PCM_TRIGGER_START:
-@@ -228,7 +223,7 @@ static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd
- 	case SNDRV_PCM_TRIGGER_SUSPEND:
- 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- 		/* capture channel resync, if overrun */
-		if (dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] && substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-+		if (dai->capture_active && substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- 			break;
- 		}
- 
-@@ -248,7 +243,7 @@ static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd
- 
- 	dev_dbg(rtd->card->dev, "%s() stream=%s  cmd=%d play:%d, capt:%d;finished %d\n",
- 		__FUNCTION__, snd_pcm_stream_str(substream), cmd,
-		dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], dai->stream_active[SNDRV_PCM_STREAM_CAPTURE], ret);
-+		dai->playback_active, dai->capture_active, ret);
- 
- 	return ret;
- }
-@@ -387,8 +382,8 @@ static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd)
- static int seeed_voice_card_dai_init(struct snd_soc_pcm_runtime *rtd)
- {
- 	struct seeed_card_data *priv =	snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *codec = asoc_rtd_to_codec(rtd, 0);
-	struct snd_soc_dai *cpu = asoc_rtd_to_cpu(rtd, 0);
-+	struct snd_soc_dai *codec = rtd->codec_dai;
-+	struct snd_soc_dai *cpu = rtd->cpu_dai;
- 	struct seeed_dai_props *dai_props =
- 		seeed_priv_to_props(priv, rtd->num);
- 	int ret;
-@@ -453,19 +448,20 @@ static int seeed_voice_card_dai_link_of(struct device_node *node,
- 		goto dai_link_of_err;
- 	}
- 
-	ret = asoc_simple_parse_daifmt(dev, node, codec,
-+	ret = asoc_simple_card_parse_daifmt(dev, node, codec,
- 					    prefix, &dai_link->dai_fmt);
- 	if (ret < 0)
- 		goto dai_link_of_err;
- 
- 	of_property_read_u32(node, "mclk-fs", &dai_props->mclk_fs);
- 
-	ret = asoc_simple_parse_cpu(cpu, dai_link, &single_cpu);
-+	ret = asoc_simple_card_parse_cpu(cpu, dai_link,
-+					 DAI, CELL, &single_cpu);
- 	if (ret < 0)
- 		goto dai_link_of_err;
- 
- 	#if _SINGLE_CODEC
-	ret = asoc_simple_parse_codec(codec, dai_link);
-+	ret = asoc_simple_card_parse_codec(codec, dai_link, DAI, CELL);
- 	if (ret < 0)
- 		goto dai_link_of_err;
- 	#else
-@@ -477,7 +473,7 @@ static int seeed_voice_card_dai_link_of(struct device_node *node,
- 	dev_dbg(dev, "dai_link num_codecs = %d\n", dai_link->num_codecs);
- 	#endif
- 
-	ret = asoc_simple_parse_platform(plat, dai_link);
-+	ret = asoc_simple_card_parse_platform(plat, dai_link, DAI, CELL);
- 	if (ret < 0)
- 		goto dai_link_of_err;
- 
-@@ -502,7 +498,7 @@ static int seeed_voice_card_dai_link_of(struct device_node *node,
- 	#if LINUX_VERSION_CODE <= KERNEL_VERSION(4,10,0)
- 	ret = asoc_simple_card_parse_clk_cpu(cpu, dai_link, cpu_dai);
- 	#else
-	ret = asoc_simple_parse_clk_cpu(dev, cpu, dai_link, cpu_dai);
-+	ret = asoc_simple_card_parse_clk_cpu(dev, cpu, dai_link, cpu_dai);
- 	#endif
- 	if (ret < 0)
- 		goto dai_link_of_err;
-@@ -510,16 +506,16 @@ static int seeed_voice_card_dai_link_of(struct device_node *node,
- 	#if LINUX_VERSION_CODE <= KERNEL_VERSION(4,10,0)
- 	ret = asoc_simple_card_parse_clk_codec(codec, dai_link, codec_dai);
- 	#else
-	ret = asoc_simple_parse_clk_codec(dev, codec, dai_link, codec_dai);
-+	ret = asoc_simple_card_parse_clk_codec(dev, codec, dai_link, codec_dai);
- 	#endif
- 	if (ret < 0)
- 		goto dai_link_of_err;
- 
-	ret = asoc_simple_set_dailink_name(dev, dai_link,
-+	ret = asoc_simple_card_set_dailink_name(dev, dai_link,
- 						"%s-%s",
-						dai_link->cpus->dai_name,
-+						dai_link->cpu_dai_name,
- 						#if _SINGLE_CODEC
-						dai_link->codecs->dai_name
-+						dai_link->codec_dai_name
- 						#else
- 						dai_link->codecs[0].dai_name
- 						#endif
-@@ -533,19 +529,21 @@ static int seeed_voice_card_dai_link_of(struct device_node *node,
- 	dev_dbg(dev, "\tname : %s\n", dai_link->stream_name);
- 	dev_dbg(dev, "\tformat : %04x\n", dai_link->dai_fmt);
- 	dev_dbg(dev, "\tcpu : %s / %d\n",
-		dai_link->cpus->dai_name,
-+		dai_link->cpu_dai_name,
- 		dai_props->cpu_dai.sysclk);
- 	dev_dbg(dev, "\tcodec : %s / %d\n",
- 		#if _SINGLE_CODEC
-		dai_link->codecs->dai_name,
-+		dai_link->codec_dai_name,
- 		#else
- 		dai_link->codecs[0].dai_name,
- 		#endif
- 		dai_props->codec_dai.sysclk);
- 
-	asoc_simple_canonicalize_cpu(dai_link, single_cpu);
-+	asoc_simple_card_canonicalize_cpu(dai_link, single_cpu);
- 	#if _SINGLE_CODEC
-	asoc_simple_canonicalize_platform(dai_link);
-+	ret = asoc_simple_card_canonicalize_dailink(dai_link);
-+	if (ret < 0)
-+		goto dai_link_of_err;
- 	#endif
- 
- dai_link_of_err:
-@@ -578,7 +576,7 @@ static int seeed_voice_card_parse_aux_devs(struct device_node *node,
- 		aux_node = of_parse_phandle(node, PREFIX "aux-devs", i);
- 		if (!aux_node)
- 			return -EINVAL;
-		priv->snd_card.aux_dev[i].dlc.of_node = aux_node;
-+		priv->snd_card.aux_dev[i].codec_of_node = aux_node;
- 	}
- 
- 	priv->snd_card.num_aux_devs = n;
-@@ -638,7 +636,7 @@ static int seeed_voice_card_parse_of(struct device_node *node,
- 			goto card_parse_end;
- 	}
- 
-	ret = asoc_simple_parse_card_name(&priv->snd_card, PREFIX);
-+	ret = asoc_simple_card_parse_card_name(&priv->snd_card, PREFIX);
- 	if (ret < 0)
- 		goto card_parse_end;
- 
-@@ -743,7 +741,7 @@ static int seeed_voice_card_probe(struct platform_device *pdev)
- 	struct seeed_dai_props *dai_props;
- 	struct device_node *np = pdev->dev.of_node;
- 	struct device *dev = &pdev->dev;
-	int num, ret, i;
-+	int num, ret;
- 
- 	/* Get the number of DAI links */
- 	if (np && of_get_child_by_name(np, PREFIX "dai-link"))
-@@ -761,25 +759,6 @@ static int seeed_voice_card_probe(struct platform_device *pdev)
- 	if (!dai_props || !dai_link)
- 		return -ENOMEM;
- 
-	/*
-	 * Use snd_soc_dai_link_component instead of legacy style
-	 * It is codec only. but cpu/platform will be supported in the future.
-	 * see
-	 *      soc-core.c :: snd_soc_init_multicodec()
-	 *
-	 * "platform" might be removed
-	 * see
-	 *      simple-card-utils.c :: asoc_simple_canonicalize_platform()
-	 */
-	for (i = 0; i < num; i++) {
-		dai_link[i].cpus		= &dai_props[i].cpus;
-		dai_link[i].num_cpus		= 1;
-		dai_link[i].codecs		= &dai_props[i].codecs;
-		dai_link[i].num_codecs		= 1;
-		dai_link[i].platforms		= &dai_props[i].platforms;
-		dai_link[i].num_platforms	= 1;
-	}
-
- 	priv->dai_props			= dai_props;
- 	priv->dai_link			= dai_link;
- 
-@@ -798,9 +777,6 @@ static int seeed_voice_card_probe(struct platform_device *pdev)
- 		}
- 	} else {
- 		struct seeed_card_info *cinfo;
-		struct snd_soc_dai_link_component *cpus;
-		struct snd_soc_dai_link_component *codecs;
-		struct snd_soc_dai_link_component *platform;
- 
- 		cinfo = dev->platform_data;
- 		if (!cinfo) {
-@@ -817,19 +793,13 @@ static int seeed_voice_card_probe(struct platform_device *pdev)
- 			return -EINVAL;
- 		}
- 
-		cpus			= dai_link->cpus;
-		cpus->dai_name		= cinfo->cpu_dai.name;
-
-		codecs			= dai_link->codecs;
-		codecs->name		= cinfo->codec;
-		codecs->dai_name	= cinfo->codec_dai.name;
-
-		platform		= dai_link->platforms;
-		platform->name		= cinfo->platform;
-
- 		priv->snd_card.name	= (cinfo->card) ? cinfo->card : cinfo->name;
- 		dai_link->name		= cinfo->name;
- 		dai_link->stream_name	= cinfo->name;
-+		dai_link->platform_name	= cinfo->platform;
-+		dai_link->codec_name	= cinfo->codec;
-+		dai_link->cpu_dai_name	= cinfo->cpu_dai.name;
-+		dai_link->codec_dai_name = cinfo->codec_dai.name;
- 		dai_link->dai_fmt	= cinfo->daifmt;
- 		dai_link->init		= seeed_voice_card_dai_init;
- 		memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai,
-@@ -853,7 +823,7 @@ static int seeed_voice_card_probe(struct platform_device *pdev)
- 		return ret;
- 
- err:
-	asoc_simple_clean_reference(&priv->snd_card);
-+	asoc_simple_card_clean_reference(&priv->snd_card);
- 
- 	return ret;
- }
-@@ -865,7 +835,7 @@ static int seeed_voice_card_remove(struct platform_device *pdev)
- 
- 	if (cancel_work_sync(&priv->work_codec_clk) != 0) {
- 	}
-	return asoc_simple_clean_reference(card);
-+	return asoc_simple_card_clean_reference(card);
- }
- 
- static const struct of_device_id seeed_voice_of_match[] = {
-diff --git a/sound-compatible-4.18.h b/sound-compatible-4.18.h
-index 550b3a7..6c1a014 100644
--- a/sound-compatible-4.18.h
-+++ b/sound-compatible-4.18.h
-@@ -16,6 +16,10 @@
- #endif
- 
- #if LINUX_VERSION_CODE < KERNEL_VERSION(5,4,0)
-+#ifndef __has_attribute
-+# define __has_attribute(x) __GCC4_has_attribute_##x
-+# define __GCC4_has_attribute___fallthrough__         0
-+#endif
- #if __has_attribute(__fallthrough__)
- # define fallthrough                    __attribute__((__fallthrough__))
- #else
-@@ -31,11 +35,7 @@
- #define snd_soc_codec_get_dapm     snd_soc_component_get_dapm
- #define snd_soc_codec_get_bias_level snd_soc_component_get_bias_level
- #define snd_soc_kcontrol_codec     snd_soc_kcontrol_component
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,9,0)
-#define snd_soc_read               snd_soc_component_read
-#else
- #define snd_soc_read               snd_soc_component_read32
-#endif
- #define snd_soc_register_codec     devm_snd_soc_register_component
- #define snd_soc_unregister_codec   snd_soc_unregister_component
- #define snd_soc_update_bits        snd_soc_component_update_bits
-diff --git a/wm8960.c b/wm8960.c
-index 465c6dc..34d4dad 100644
--- a/wm8960.c
-+++ b/wm8960.c
-@@ -796,7 +796,7 @@ static int wm8960_hw_free(struct snd_pcm_substream *substream,
- 	return 0;
- }
- 
-static int wm8960_mute(struct snd_soc_dai *dai, int mute, int direction)
-+static int wm8960_mute(struct snd_soc_dai *dai, int mute)
- {
- 	struct snd_soc_codec *codec = dai->codec;
- 
-@@ -1236,12 +1236,11 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
- static const struct snd_soc_dai_ops wm8960_dai_ops = {
- 	.hw_params = wm8960_hw_params,
- 	.hw_free = wm8960_hw_free,
-	.mute_stream = wm8960_mute,
-+	.digital_mute = wm8960_mute,
- 	.set_fmt = wm8960_set_dai_fmt,
- 	.set_clkdiv = wm8960_set_dai_clkdiv,
- 	.set_pll = wm8960_set_dai_pll,
- 	.set_sysclk = wm8960_set_dai_sysclk,
-	.no_capture_mute = 1,
- };
- 
- static struct snd_soc_dai_driver wm8960_dai = {
--- a/patches/back-to-v5.4.diff
+++ b/patches/back-to-v5.4.diff
@ -1,245 +0,0 @@
-diff --git a/ac101.c b/ac101.c
-index be9b1d8..343f030 100644
--- a/ac101.c
-+++ b/ac101.c
-@@ -955,10 +955,10 @@ void ac101_aif_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai
- 
- 	AC101_DBG("stream = %s, play: %d, capt: %d, active: %d\n", 
- 		snd_pcm_stream_str(substream),
-		codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE],
-		snd_soc_dai_active(codec_dai));
-+		codec_dai->playback_active, codec_dai->capture_active,
-+		codec_dai->active);
- 
-	if (!snd_soc_dai_active(codec_dai)) {
-+	if (!codec_dai->active) {
- 		ac10x->aif1_clken = 1;
- 		ac101_aif1clk(codec, SND_SOC_DAPM_POST_PMD, 0);
- 	} else {
-@@ -1080,7 +1080,7 @@ int ac101_hw_params(struct snd_pcm_substream *substream,
- 	freq_out = _FREQ_24_576K;
- 	for (i = 0; i < ARRAY_SIZE(codec_aif1_fs); i++) {
- 		if (codec_aif1_fs[i].samp_rate == params_rate(params)) {
-			if (codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] && dmic_used && codec_aif1_fs[i].samp_rate == 44100) {
-+			if (codec_dai->capture_active && dmic_used && codec_aif1_fs[i].samp_rate == 44100) {
- 				ac101_update_bits(codec, AIF_SR_CTRL, (0xf<<AIF1_FS), (0x4<<AIF1_FS));
- 			} else {
- 				ac101_update_bits(codec, AIF_SR_CTRL, (0xf<<AIF1_FS), ((codec_aif1_fs[i].srbit)<<AIF1_FS));
-diff --git a/ac108.c b/ac108.c
-index 4663df0..12ab27b 100644
--- a/ac108.c
-+++ b/ac108.c
-@@ -653,7 +653,7 @@ static int ac108_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_h
- 
- 	dev_dbg(dai->dev, "%s() stream=%s play:%d capt:%d +++\n", __func__,
- 			snd_pcm_stream_str(substream),
-			dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]);
-+			dai->playback_active, dai->capture_active);
- 
- 	if (ac10x->i2c101) {
- 		ret = ac101_hw_params(substream, params, dai);
-@@ -664,8 +664,8 @@ static int ac108_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_h
- 		}
- 	}
- 
-	if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE && dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])
-	 || (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && dai->stream_active[SNDRV_PCM_STREAM_CAPTURE])) {
-+	if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE && dai->playback_active)
-+	 || (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && dai->capture_active)) {
- 		/* not configure hw_param twice */
- 		/* return 0; */
- 	}
-@@ -1124,7 +1124,7 @@ void ac108_aif_shutdown(struct snd_pcm_substream *substream,
- 	}
- }
- 
-int ac108_aif_mute(struct snd_soc_dai *dai, int mute, int direction) {
-+int ac108_aif_mute(struct snd_soc_dai *dai, int mute) {
- 	struct snd_soc_codec *codec = dai->codec;
- 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
- 
-@@ -1145,13 +1145,12 @@ static const struct snd_soc_dai_ops ac108_dai_ops = {
- 	.hw_params	= ac108_hw_params,
- 	.prepare	= ac108_prepare,
- 	.trigger	= ac108_trigger,
-	.mute_stream	= ac108_aif_mute,
-+	.digital_mute	= ac108_aif_mute,
- 
- 	/*DAI format configuration*/
- 	.set_fmt	= ac108_set_fmt,
- 
- 	// .hw_free = ac108_hw_free,
-	.no_capture_mute = 1,
- };
- 
- static  struct snd_soc_dai_driver ac108_dai0 = {
-diff --git a/seeed-voicecard.c b/seeed-voicecard.c
-index c6d9048..43535aa 100644
--- a/seeed-voicecard.c
-+++ b/seeed-voicecard.c
-@@ -96,16 +96,16 @@ static int seeed_voice_card_startup(struct snd_pcm_substream *substream)
- 	if (ret)
- 		clk_disable_unprepare(dai_props->cpu_dai.clk);
- 
-	if (asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min) {
-		priv->channels_playback_default = asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min;
-+	if (rtd->cpu_dai->driver->playback.channels_min) {
-+		priv->channels_playback_default = rtd->cpu_dai->driver->playback.channels_min;
- 	}
-	if (asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min) {
-		priv->channels_capture_default = asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min;
-+	if (rtd->cpu_dai->driver->capture.channels_min) {
-+		priv->channels_capture_default = rtd->cpu_dai->driver->capture.channels_min;
- 	}
-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min = priv->channels_playback_override;
-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_max = priv->channels_playback_override;
-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min = priv->channels_capture_override;
-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_max = priv->channels_capture_override;
-+	rtd->cpu_dai->driver->playback.channels_min = priv->channels_playback_override;
-+	rtd->cpu_dai->driver->playback.channels_max = priv->channels_playback_override;
-+	rtd->cpu_dai->driver->capture.channels_min = priv->channels_capture_override;
-+	rtd->cpu_dai->driver->capture.channels_max = priv->channels_capture_override;
- 
- 	return ret;
- }
-@@ -117,10 +117,10 @@ static void seeed_voice_card_shutdown(struct snd_pcm_substream *substream)
- 	struct seeed_dai_props *dai_props =
- 		seeed_priv_to_props(priv, rtd->num);
- 
-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min = priv->channels_playback_default;
-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_max = priv->channels_playback_default;
-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min = priv->channels_capture_default;
-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_max = priv->channels_capture_default;
-+	rtd->cpu_dai->driver->playback.channels_min = priv->channels_playback_default;
-+	rtd->cpu_dai->driver->playback.channels_max = priv->channels_playback_default;
-+	rtd->cpu_dai->driver->capture.channels_min = priv->channels_capture_default;
-+	rtd->cpu_dai->driver->capture.channels_max = priv->channels_capture_default;
- 
- 	clk_disable_unprepare(dai_props->cpu_dai.clk);
- 
-@@ -131,8 +131,8 @@ static int seeed_voice_card_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 = asoc_rtd_to_codec(rtd, 0);
-	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
-+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- 	struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
- 	struct seeed_dai_props *dai_props =
- 		seeed_priv_to_props(priv, rtd->num);
-@@ -196,7 +196,7 @@ static void work_cb_codec_clk(struct work_struct *work)
- static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd)
- {
- 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
-+	struct snd_soc_dai *dai = rtd->codec_dai;
- 	struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
- 	#if CONFIG_AC10X_TRIG_LOCK
- 	unsigned long flags;
-@@ -205,7 +205,7 @@ static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd
- 
- 	dev_dbg(rtd->card->dev, "%s() stream=%s  cmd=%d play:%d, capt:%d\n",
- 		__FUNCTION__, snd_pcm_stream_str(substream), cmd,
-		dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]);
-+		dai->playback_active, dai->capture_active);
- 
- 	switch (cmd) {
- 	case SNDRV_PCM_TRIGGER_START:
-@@ -227,7 +227,7 @@ static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd
- 	case SNDRV_PCM_TRIGGER_SUSPEND:
- 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- 		/* capture channel resync, if overrun */
-		if (dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] && substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-+		if (dai->capture_active && substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- 			break;
- 		}
- 
-@@ -252,7 +247,7 @@ static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd
- 
- 	dev_dbg(rtd->card->dev, "%s() stream=%s  cmd=%d play:%d, capt:%d;finished %d\n",
- 		__FUNCTION__, snd_pcm_stream_str(substream), cmd,
-		dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], dai->stream_active[SNDRV_PCM_STREAM_CAPTURE], ret);
-+		dai->playback_active, dai->capture_active, ret);
- 
- 	return ret;
- }
-@@ -337,8 +337,8 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai,
- static int seeed_voice_card_dai_init(struct snd_soc_pcm_runtime *rtd)
- {
- 	struct seeed_card_data *priv =	snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *codec = asoc_rtd_to_codec(rtd, 0);
-	struct snd_soc_dai *cpu = asoc_rtd_to_cpu(rtd, 0);
-+	struct snd_soc_dai *codec = rtd->codec_dai;
-+	struct snd_soc_dai *cpu = rtd->cpu_dai;
- 	struct seeed_dai_props *dai_props =
- 		seeed_priv_to_props(priv, rtd->num);
- 	int ret;
-@@ -636,11 +636,11 @@ static int seeed_voice_card_probe(struct platform_device *pdev)
- 	 * Use snd_soc_dai_link_component instead of legacy style
- 	 * It is codec only. but cpu/platform will be supported in the future.
- 	 * see
-	 *      soc-core.c :: snd_soc_init_multicodec()
-+	 *	soc-core.c :: snd_soc_init_multicodec()
- 	 *
- 	 * "platform" might be removed
- 	 * see
-	 *      simple-card-utils.c :: asoc_simple_canonicalize_platform()
-+	 *	simple-card-utils.c :: asoc_simple_canonicalize_platform()
- 	 */
- 	for (i = 0; i < num; i++) {
- 		dai_link[i].cpus		= &dai_props[i].cpus;
-diff --git a/sound-compatible-4.18.h b/sound-compatible-4.18.h
-index 080325b..eefa7de 100644
--- a/sound-compatible-4.18.h
-+++ b/sound-compatible-4.18.h
-@@ -16,6 +16,10 @@
- #endif
- 
- #if LINUX_VERSION_CODE < KERNEL_VERSION(5,4,0)
-+#ifndef __has_attribute
-+# define __has_attribute(x) __GCC4_has_attribute_##x
-+# define __GCC4_has_attribute___fallthrough__         0
-+#endif
- #if __has_attribute(__fallthrough__)
- # define fallthrough                    __attribute__((__fallthrough__))
- #else
-@@ -31,11 +35,7 @@
- #define snd_soc_codec_get_dapm     snd_soc_component_get_dapm
- #define snd_soc_codec_get_bias_level snd_soc_component_get_bias_level
- #define snd_soc_kcontrol_codec     snd_soc_kcontrol_component
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,9,0)
-#define snd_soc_read               snd_soc_component_read
-#else
- #define snd_soc_read               snd_soc_component_read32
-#endif
- #define snd_soc_register_codec     devm_snd_soc_register_component
- #define snd_soc_unregister_codec   snd_soc_unregister_component
- #define snd_soc_update_bits        snd_soc_component_update_bits
-diff --git a/wm8960.c b/wm8960.c
-index 465c6dc..34d4dad 100644
--- a/wm8960.c
-+++ b/wm8960.c
-@@ -796,7 +796,7 @@ static int wm8960_hw_free(struct snd_pcm_substream *substream,
- 	return 0;
- }
- 
-static int wm8960_mute(struct snd_soc_dai *dai, int mute, int direction)
-+static int wm8960_mute(struct snd_soc_dai *dai, int mute)
- {
- 	struct snd_soc_codec *codec = dai->codec;
- 
-@@ -1236,12 +1236,11 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
- static const struct snd_soc_dai_ops wm8960_dai_ops = {
- 	.hw_params = wm8960_hw_params,
- 	.hw_free = wm8960_hw_free,
-	.mute_stream = wm8960_mute,
-+	.digital_mute = wm8960_mute,
- 	.set_fmt = wm8960_set_dai_fmt,
- 	.set_clkdiv = wm8960_set_dai_clkdiv,
- 	.set_pll = wm8960_set_dai_pll,
- 	.set_sysclk = wm8960_set_dai_sysclk,
-	.no_capture_mute = 1,
- };
- 
- static struct snd_soc_dai_driver wm8960_dai = {
--- a/patches/back-to-v5.8.diff
+++ b/patches/back-to-v5.8.diff
@ -1,71 +0,0 @@
-diff --git a/ac108.c b/ac108.c
-index 4663df0..67edeae 100644
--- a/ac108.c
-+++ b/ac108.c
-@@ -1124,7 +1124,7 @@ void ac108_aif_shutdown(struct snd_pcm_substream *substream,
- 	}
- }
- 
-int ac108_aif_mute(struct snd_soc_dai *dai, int mute, int direction) {
-+int ac108_aif_mute(struct snd_soc_dai *dai, int mute) {
- 	struct snd_soc_codec *codec = dai->codec;
- 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
- 
-@@ -1145,13 +1145,12 @@ static const struct snd_soc_dai_ops ac108_dai_ops = {
- 	.hw_params	= ac108_hw_params,
- 	.prepare	= ac108_prepare,
- 	.trigger	= ac108_trigger,
-	.mute_stream	= ac108_aif_mute,
-+	.digital_mute	= ac108_aif_mute,
- 
- 	/*DAI format configuration*/
- 	.set_fmt	= ac108_set_fmt,
- 
- 	// .hw_free = ac108_hw_free,
-	.no_capture_mute = 1,
- };
- 
- static  struct snd_soc_dai_driver ac108_dai0 = {
-diff --git a/sound-compatible-4.18.h b/sound-compatible-4.18.h
-index 080325b..faed848 100644
--- a/sound-compatible-4.18.h
-+++ b/sound-compatible-4.18.h
-@@ -31,11 +31,7 @@
- #define snd_soc_codec_get_dapm     snd_soc_component_get_dapm
- #define snd_soc_codec_get_bias_level snd_soc_component_get_bias_level
- #define snd_soc_kcontrol_codec     snd_soc_kcontrol_component
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,9,0)
-#define snd_soc_read               snd_soc_component_read
-#else
- #define snd_soc_read               snd_soc_component_read32
-#endif
- #define snd_soc_register_codec     devm_snd_soc_register_component
- #define snd_soc_unregister_codec   snd_soc_unregister_component
- #define snd_soc_update_bits        snd_soc_component_update_bits
-diff --git a/wm8960.c b/wm8960.c
-index 465c6dc..34d4dad 100644
--- a/wm8960.c
-+++ b/wm8960.c
-@@ -796,7 +796,7 @@ static int wm8960_hw_free(struct snd_pcm_substream *substream,
- 	return 0;
- }
- 
-static int wm8960_mute(struct snd_soc_dai *dai, int mute, int direction)
-+static int wm8960_mute(struct snd_soc_dai *dai, int mute)
- {
- 	struct snd_soc_codec *codec = dai->codec;
- 
-@@ -1236,12 +1236,11 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
- static const struct snd_soc_dai_ops wm8960_dai_ops = {
- 	.hw_params = wm8960_hw_params,
- 	.hw_free = wm8960_hw_free,
-	.mute_stream = wm8960_mute,
-+	.digital_mute = wm8960_mute,
- 	.set_fmt = wm8960_set_dai_fmt,
- 	.set_clkdiv = wm8960_set_dai_clkdiv,
- 	.set_pll = wm8960_set_dai_pll,
- 	.set_sysclk = wm8960_set_dai_sysclk,
-	.no_capture_mute = 1,
- };
- 
- static struct snd_soc_dai_driver wm8960_dai = {
--- a/pulseaudio/91-seeedvoicecard.rules
+++ b/pulseaudio/91-seeedvoicecard.rules
@ -1,8 +0,0 @@
-SUBSYSTEM!="sound", GOTO="seeedvoicecard_end"
-ACTION!="change", GOTO="seeedvoicecard_end"
-KERNEL!="card*", GOTO="seeedvoicecard_end"
-
-ATTR{id}=="seeed4micvoicec",ENV{PULSE_PROFILE_SET}="seeed-voicecard-4mic.conf"
-ATTR{id}=="seeed8micvoicec",ENV{PULSE_PROFILE_SET}="seeed-voicecard-8mic.conf"
-
-LABEL="seeedvoicecard_end"
--- a/pulseaudio/README.md
+++ b/pulseaudio/README.md
@ -1,251 +0,0 @@
-# PulseAudio Configuration for seeed-voicecard
-
-Follow this guide if you want to use your seeed-voicecard as a default source/sink of pulseaudio.
-
-### Prerequisites
-
-1. Download PulseAudio
-```
-sudo apt install -y pulseaudio
-```
-
-2. PulseAudio Profiles
-```
-cd seeed-voicecard/pulseaudio
-sudo cp pulse_config_4mic/seeed-voicecard.conf /usr/share/pulseaudio/alsa-mixer/profile-sets/seeed-voicecard-4mic.conf
-sudo cp pulse_config_6mic/seeed-voicecard.conf /usr/share/pulseaudio/alsa-mixer/profile-sets/seeed-voicecard-8mic.conf
-```
-
-3. Add `udev` Rules
-
-During the system start, when the card "seeed4micvoicec" is detected, the PULSE_PROFILE_SET variable will be set in the udev database, and PulseAudio will be forced to use `seeed-voicecard-4mic.conf`. Similarly, if the card "seeed8micvoicec" is detected, PulseAudio will be forced to use `seeed-voicecard-8mic.conf`.
-
-```
-sudo cp 91-seeedvoicecard.rules /etc/udev/rules.d/91-seeedvoicecard.rules
-```
-
-### ReSpeaker 4 Mic Array
-
-<!--
-1. Download pulseaudio
-```
-sudo apt install pulseaudio
-```
-
-2. First, you need to write [a profile for pulse](https://www.freedesktop.org/wiki/Software/PulseAudio/Backends/ALSA/Profiles/)
-```
-cd seeed-voicecard
-cd pulseaudio
-cd pulse_config_4mic
-sudo cp seeed-voicecard.conf /usr/share/pulseaudio/alsa-mixer/profile-sets/
-```
-
-3. Edit `udev rules`
-
-    During the system start, when the card "seeed4micvoicec" is detected, the PULSE_PROFILE_SET variable will be set in the udev database, and PulseAudio will be forced to use `seeed-voicecard.conf`.
-
-    ```
-    # have a look at /lib/udev/rules.d/90-pulseaudio.rules
-    sudo vim /lib/udev/rules.d/90-pulseaudio.rules
-    # add the following lines at about line 87(behind the setting for some laptops and before the line GOTO="pulseaudio_end")
-    # Seeed Voicecard
-    ATTR{id}=="seeed4micvoicec",ATTR{number}=="1",ENV{PULSE_PROFILE_SET}="seeed-voicecard.conf"
-    ```
-    ![](./udev_rules_4mic.png)
-
-    The value of `ATTR{number}` can be found with:
-
-    ```
-    udevadm info -a -p /sys/class/sound/card0/
-    # or udevadm info -a -p /sys/class/sound/card1/
-    ```
-
-    For example, in Raspberry Pi, we can find `ATTR{id}=="seeed4micvoicec"` and `ATTR{number}=="1"` with command `udevadm info -a -p /sys/class/sound/card1/`:
-
-    ```
-    pi@raspberrypi:~ $ udevadm info -a -p  /sys/class/sound/card1/
-
-    Udevadm info starts with the device specified by the devpath and then
-    walks up the chain of parent devices. It prints for every device
-    found, all possible attributes in the udev rules key format.
-    A rule to match, can be composed by the attributes of the device
-    and the attributes from one single parent device.
-
-    looking at device '/devices/platform/soc/soc:sound/sound/card1':
-        KERNEL=="card1"
-        SUBSYSTEM=="sound"
-        DRIVER==""
-        ATTR{id}=="seeed4micvoicec"
-        ATTR{number}=="1"
-
-    looking at parent device '/devices/platform/soc/soc:sound':
-        KERNELS=="soc:sound"
-        SUBSYSTEMS=="platform"
-        DRIVERS=="seeed-voicecard"
-        ATTRS{driver_override}=="(null)"
-
-    looking at parent device '/devices/platform/soc':
-        KERNELS=="soc"
-        SUBSYSTEMS=="platform"
-        DRIVERS==""
-        ATTRS{driver_override}=="(null)"
-
-    looking at parent device '/devices/platform':
-        KERNELS=="platform"
-        SUBSYSTEMS==""
-        DRIVERS==""
-    ``` -->
-
-1. config `default.pa` and `daemon.conf`
-```
-sudo cp pulse_config_4mic/default.pa /etc/pulse/
-sudo cp pulse_config_4mic/daemon.conf /etc/pulse/
-```
-
-2. reboot raspberry pi and check
-```
-sudo reboot
-pulseaudio --start  # start pulse at first
-pactl info  # check the setting
-
-Server String: /run/user/1000/pulse/native
-Library Protocol Version: 32
-Server Protocol Version: 32
-Is Local: yes
-Client Index: 18
-Tile Size: 65496
-User Name: pi
-Host Name: raspberrypi
-Server Name: pulseaudio
-Server Version: 10.0
-Default Sample Specification: s16le 4ch 96000Hz
-Default Channel Map: front-left,front-center,front-right,rear-center
-Default Sink: alsa_output.platform-soc_audio.analog-stereo
-Default Source: alsa_input.platform-soc_sound.seeed-source
-Cookie: 3b12:70b3
-```
-
-### 6-Mics Circular Array Kit and 4-Mics Linear Array
-
-<!--
-1. Download pulseaudio
-```
-sudo apt install pulseaudio
-```
-
-2. First, you need to write [a profile for pulse](https://www.freedesktop.org/wiki/Software/PulseAudio/Backends/ALSA/Profiles/)
-```
-cd seeed-voicecard
-cd pulseaudio
-cd pulse_config_6mic
-sudo cp seeed-voicecard.conf /usr/share/pulseaudio/alsa-mixer/profile-sets/
-```
-
-3. Edit `udev rules`
-
-    During the system start, when the card "seeed8micvoicec" is detected, the PULSE_PROFILE_SET variable will be set in the udev database, and PulseAudio will be forced to use `seeed-voicecard.conf`.
-
-    ```
-    # have a look at /lib/udev/rules.d/90-pulseaudio.rules
-    sudo vim /lib/udev/rules.d/90-pulseaudio.rules
-    # add the following lines at about line 87(behind the setting for some laptops and before the line GOTO="pulseaudio_end")
-    # Seeed Voicecard
-    ATTR{id}=="seeed8micvoicec",ATTR{number}=="1",ENV{PULSE_PROFILE_SET}="seeed-voicecard.conf"
-    ```
-    ![](./udev_rules_6mic.png)
-
-    The value of `ATTR{number}` can be found with:
-
-    ```
-    udevadm info -a -p /sys/class/sound/card0/
-    # or udevadm info -a -p /sys/class/sound/card1/
-    ```
-
-    For example, in Raspberry Pi, we can find `ATTR{id}=="seeed8micvoicec"` and `ATTR{number}=="1"` with command `udevadm info -a -p /sys/class/sound/card1/`:
-
-    ```
-    pi@raspberrypi:~ $ udevadm info -a -p  /sys/class/sound/card1/
-
-    Udevadm info starts with the device specified by the devpath and then
-    walks up the chain of parent devices. It prints for every device
-    found, all possible attributes in the udev rules key format.
-    A rule to match, can be composed by the attributes of the device
-    and the attributes from one single parent device.
-
-    looking at device '/devices/platform/soc/soc:sound/sound/card1':
-        KERNEL=="card1"
-        SUBSYSTEM=="sound"
-        DRIVER==""
-        ATTR{id}=="seeed8micvoicec"
-        ATTR{number}=="1"
-
-    looking at parent device '/devices/platform/soc/soc:sound':
-        KERNELS=="soc:sound"
-        SUBSYSTEMS=="platform"
-        DRIVERS=="seeed-voicecard"
-        ATTRS{driver_override}=="(null)"
-
-    looking at parent device '/devices/platform/soc':
-        KERNELS=="soc"
-        SUBSYSTEMS=="platform"
-        DRIVERS==""
-        ATTRS{driver_override}=="(null)"
-
-    looking at parent device '/devices/platform':
-        KERNELS=="platform"
-        SUBSYSTEMS==""
-        DRIVERS==""
-    ``` -->
-
-1. config `default.pa` and `daemon.conf`
-```
-sudo cp pulse_config_6mic/default.pa /etc/pulse/
-sudo cp pulse_config_6mic/daemon.conf /etc/pulse/
-```
-
-2. reboot raspberry pi and check
-```
-sudo reboot
-pulseaudio --start  # start pulse at first
-pactl info  # check the setting
-
-# The output should be like this
-# You could see the default sink is seeed-2ch and default source is seeed-8ch
-pi@raspberrypi:~ $ pactl info
-Server String: /run/user/1000/pulse/native
-Library Protocol Version: 32
-Server Protocol Version: 32
-Is Local: yes
-Client Index: 6
-Tile Size: 65496
-User Name: pi
-Host Name: raspberrypi
-Server Name: pulseaudio
-Server Version: 10.0
-Default Sample Specification: s32le 8ch 96000Hz
-Default Channel Map: front-left,front-left-of-center,front-center,front-right,front-right-of-center,rear-center,aux0,aux1
-Default Sink: alsa_output.platform-soc_sound.seeed-2ch
-Default Source: alsa_input.platform-soc_sound.seeed-8ch
-Cookie: 3523:e5af
-```
-
-### FAQ
-
-1. Default Sink/Source not right
-
-    Make sure there is no any other daemon or using the audio device. Then check profile and udev rules.
-
-    `pacmd list-sinks` and `pacmd list-sources` can be used to show the avaiable sinks/sources, after pulseaudio is started.
-
-2. Can't start PulseAudio
-
-    Check `default.pa` and `daemon.conf`
-
-3. How to get PulseAudio started automatically
-
-    Normally the PulseAudio server is started automatically. If you want to disable it, you can set `autospawn = no` in `~/.config/pulse/client.conf` or `/etc/pulse/client.conf`.
-    [Click this for more details](https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Running/).
-
-4. Why the default sample rate is 96000? What if my audio's sample rate is not the same as the default?
-
-    For the other sample rate audio, PulseAudio will resample it into 96K, which means that if your audio's sample rate is lower than 96K, it will get smoothing.
--- a/pulseaudio/pulse_config_4mic/daemon.conf
+++ b/pulseaudio/pulse_config_4mic/daemon.conf
@ -1,87 +0,0 @@
-# This file is part of PulseAudio.
-#
-# PulseAudio is free software; you can redistribute it and/or modify
-# it under the terms of the GNU Lesser General Public License as published by
-# the Free Software Foundation; either version 2 of the License, or
-# (at your option) any later version.
-#
-# PulseAudio 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.
-#
-# You should have received a copy of the GNU Lesser General Public License
-# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
-
-## Configuration file for the PulseAudio daemon. See pulse-daemon.conf(5) for
-## more information. Default values are commented out.  Use either ; or # for
-## commenting.
-
-; daemonize = no
-; fail = yes
-; allow-module-loading = yes
-; allow-exit = yes
-; use-pid-file = yes
-; system-instance = no
-; local-server-type = user
-; enable-shm = yes
-; enable-memfd = yes
-; shm-size-bytes = 0 # setting this 0 will use the system-default, usually 64 MiB
-; lock-memory = no
-; cpu-limit = no
-
-; high-priority = yes
-; nice-level = -11
-
-; realtime-scheduling = yes
-; realtime-priority = 5
-
-; exit-idle-time = 20
-; scache-idle-time = 20
-
-; dl-search-path = (depends on architecture)
-
-; load-default-script-file = yes
-; default-script-file = /etc/pulse/default.pa
-
-; log-target = auto
-; log-level = notice
-; log-meta = no
-; log-time = no
-; log-backtrace = 0
-
-; resample-method = speex-float-1
-; enable-remixing = yes
-; enable-lfe-remixing = no
-; lfe-crossover-freq = 0
-
-; flat-volumes = yes
-
-; rlimit-fsize = -1
-; rlimit-data = -1
-; rlimit-stack = -1
-; rlimit-core = -1
-; rlimit-as = -1
-; rlimit-rss = -1
-; rlimit-nproc = -1
-; rlimit-nofile = 256
-; rlimit-memlock = -1
-; rlimit-locks = -1
-; rlimit-sigpending = -1
-; rlimit-msgqueue = -1
-; rlimit-nice = 31
-; rlimit-rtprio = 9
-; rlimit-rttime = 200000
-
-default-sample-format = s16le
-default-sample-rate = 96000
-; alternate-sample-rate = 48000
-default-sample-channels = 4
-; default-channel-map = front-left,front-right
-
-; default-fragments = 4
-; default-fragment-size-msec = 25
-
-; enable-deferred-volume = yes
-; deferred-volume-safety-margin-usec = 8000
-; deferred-volume-extra-delay-usec = 0
--- a/pulseaudio/pulse_config_4mic/default.pa
+++ b/pulseaudio/pulse_config_4mic/default.pa
@ -1,144 +0,0 @@
-#!/usr/bin/pulseaudio -nF
-#
-# This file is part of PulseAudio.
-#
-# PulseAudio is free software; you can redistribute it and/or modify it
-# under the terms of the GNU Lesser General Public License as published by
-# the Free Software Foundation; either version 2 of the License, or
-# (at your option) any later version.
-#
-# PulseAudio 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.
-#
-# You should have received a copy of the GNU Lesser General Public License
-# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
-
-# This startup script is used only if PulseAudio is started per-user
-# (i.e. not in system mode)
-
-.fail
-
-### Automatically restore the volume of streams and devices
-load-module module-device-restore
-load-module module-stream-restore
-load-module module-card-restore
-
-### Automatically augment property information from .desktop files
-### stored in /usr/share/application
-load-module module-augment-properties
-
-### Should be after module-*-restore but before module-*-detect
-load-module module-switch-on-port-available
-
-### Load audio drivers statically
-### (it's probably better to not load these drivers manually, but instead
-### use module-udev-detect -- see below -- for doing this automatically)
-#load-module module-alsa-sink device="hw:1,0" channels=8 rate=48000 format=s32le
-#load-module module-alsa-source device="hw:1,0" channels=8 rate=48000 format=s32le
-#load-module module-oss device="/dev/dsp" sink_name=output source_name=input
-#load-module module-oss-mmap device="/dev/dsp" sink_name=output source_name=input
-#load-module module-null-sink
-#load-module module-pipe-sink
-
-### Automatically load driver modules depending on the hardware available
-.ifexists module-udev-detect.so
-load-module module-udev-detect 
-#channels=8 rate=48000 format=s32le
-.else
-### Use the static hardware detection module (for systems that lack udev support)
-load-module module-detect
-.endif
-
-### Automatically connect sink and source if JACK server is present
-.ifexists module-jackdbus-detect.so
-.nofail
-load-module module-jackdbus-detect channels=2
-.fail
-.endif
-
-### Automatically load driver modules for Bluetooth hardware
-.ifexists module-bluetooth-policy.so
-load-module module-bluetooth-policy
-.endif
-
-.ifexists module-bluetooth-discover.so
-load-module module-bluetooth-discover
-.endif
-
-### Load several protocols
-.ifexists module-esound-protocol-unix.so
-load-module module-esound-protocol-unix
-.endif
-load-module module-native-protocol-unix
-
-### Network access (may be configured with paprefs, so leave this commented
-### here if you plan to use paprefs)
-#load-module module-esound-protocol-tcp
-#load-module module-native-protocol-tcp
-#load-module module-zeroconf-publish
-
-### Load the RTP receiver module (also configured via paprefs, see above)
-#load-module module-rtp-recv
-
-### Load the RTP sender module (also configured via paprefs, see above)
-#load-module module-null-sink sink_name=rtp format=s16be channels=2 rate=44100 sink_properties="device.description='RTP Multicast Sink'"
-#load-module module-rtp-send source=rtp.monitor
-
-### Load additional modules from GConf settings. This can be configured with the paprefs tool.
-### Please keep in mind that the modules configured by paprefs might conflict with manually
-### loaded modules.
-.ifexists module-gconf.so
-.nofail
-load-module module-gconf
-.fail
-.endif
-
-### Automatically restore the default sink/source when changed by the user
-### during runtime
-### NOTE: This should be loaded as early as possible so that subsequent modules
-### that look up the default sink/source get the right value
-load-module module-default-device-restore
-
-### Automatically move streams to the default sink if the sink they are
-### connected to dies, similar for sources
-load-module module-rescue-streams
-
-### Make sure we always have a sink around, even if it is a null sink.
-load-module module-always-sink
-
-### Honour intended role device property
-load-module module-intended-roles
-
-### Automatically suspend sinks/sources that become idle for too long
-load-module module-suspend-on-idle
-
-### If autoexit on idle is enabled we want to make sure we only quit
-### when no local session needs us anymore.
-.ifexists module-console-kit.so
-load-module module-console-kit
-.endif
-.ifexists module-systemd-login.so
-load-module module-systemd-login
-.endif
-
-### Enable positioned event sounds
-load-module module-position-event-sounds
-
-### Cork music/video streams when a phone stream is active
-load-module module-role-cork
-
-### Modules to allow autoloading of filters (such as echo cancellation)
-### on demand. module-filter-heuristics tries to determine what filters
-### make sense, and module-filter-apply does the heavy-lifting of
-### loading modules and rerouting streams.
-load-module module-filter-heuristics
-load-module module-filter-apply
-
-### Make some devices default
-#set-default-sink output
-#set-default-source input
-set-default-source alsa_input.platform-soc_sound.seeed-source
-set-default-sink alsa_output.platform-soc_sound.seeed-sink
-
--- a/pulseaudio/pulse_config_4mic/seeed-voicecard.conf
+++ b/pulseaudio/pulse_config_4mic/seeed-voicecard.conf
@ -1,17 +0,0 @@
-# /usr/share/pulseaudio/alsa-mixer/profile-sets/seeed-voicecard.conf
-
-[General]
-auto-profiles = no
-[Mapping seeed-source]
-device-strings = hw:%f
-channel-map = front-left,front-right,rear-left,rear-right
-exact-channels = false
-fallback = yes
-paths-input = seeed-source
-priority = 3
-direction = input
-
-[Profile input:seeed-source]
-input-mappings = seeed-source
-priority = 5
-skip-probe = yes
--- a/pulseaudio/pulse_config_6mic/daemon.conf
+++ b/pulseaudio/pulse_config_6mic/daemon.conf
@ -1,87 +0,0 @@
-# This file is part of PulseAudio.
-#
-# PulseAudio is free software; you can redistribute it and/or modify
-# it under the terms of the GNU Lesser General Public License as published by
-# the Free Software Foundation; either version 2 of the License, or
-# (at your option) any later version.
-#
-# PulseAudio 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.
-#
-# You should have received a copy of the GNU Lesser General Public License
-# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
-
-## Configuration file for the PulseAudio daemon. See pulse-daemon.conf(5) for
-## more information. Default values are commented out.  Use either ; or # for
-## commenting.
-
-; daemonize = no
-; fail = yes
-; allow-module-loading = yes
-; allow-exit = yes
-; use-pid-file = yes
-; system-instance = no
-; local-server-type = user
-; enable-shm = yes
-; enable-memfd = yes
-; shm-size-bytes = 0 # setting this 0 will use the system-default, usually 64 MiB
-; lock-memory = no
-; cpu-limit = no
-
-; high-priority = yes
-; nice-level = -11
-
-; realtime-scheduling = yes
-; realtime-priority = 5
-
-; exit-idle-time = 20
-; scache-idle-time = 20
-
-; dl-search-path = (depends on architecture)
-
-; load-default-script-file = yes
-; default-script-file = /etc/pulse/default.pa
-
-; log-target = auto
-; log-level = notice
-; log-meta = no
-; log-time = no
-; log-backtrace = 0
-
-; resample-method = speex-float-1
-; enable-remixing = yes
-; enable-lfe-remixing = no
-; lfe-crossover-freq = 0
-
-; flat-volumes = yes
-
-; rlimit-fsize = -1
-; rlimit-data = -1
-; rlimit-stack = -1
-; rlimit-core = -1
-; rlimit-as = -1
-; rlimit-rss = -1
-; rlimit-nproc = -1
-; rlimit-nofile = 256
-; rlimit-memlock = -1
-; rlimit-locks = -1
-; rlimit-sigpending = -1
-; rlimit-msgqueue = -1
-; rlimit-nice = 31
-; rlimit-rtprio = 9
-; rlimit-rttime = 200000
-
-default-sample-format = s32le
-default-sample-rate = 96000
-; alternate-sample-rate = 48000
-default-sample-channels = 8
-; default-channel-map = front-left,front-right
-
-; default-fragments = 4
-; default-fragment-size-msec = 25
-
-; enable-deferred-volume = yes
-; deferred-volume-safety-margin-usec = 8000
-; deferred-volume-extra-delay-usec = 0
--- a/pulseaudio/pulse_config_6mic/default.pa
+++ b/pulseaudio/pulse_config_6mic/default.pa
@ -1,143 +0,0 @@
-#!/usr/bin/pulseaudio -nF
-#
-# This file is part of PulseAudio.
-#
-# PulseAudio is free software; you can redistribute it and/or modify it
-# under the terms of the GNU Lesser General Public License as published by
-# the Free Software Foundation; either version 2 of the License, or
-# (at your option) any later version.
-#
-# PulseAudio 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.
-#
-# You should have received a copy of the GNU Lesser General Public License
-# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
-
-# This startup script is used only if PulseAudio is started per-user
-# (i.e. not in system mode)
-
-.fail
-
-### Automatically restore the volume of streams and devices
-load-module module-device-restore
-load-module module-stream-restore
-load-module module-card-restore
-
-### Automatically augment property information from .desktop files
-### stored in /usr/share/application
-load-module module-augment-properties
-
-### Should be after module-*-restore but before module-*-detect
-load-module module-switch-on-port-available
-
-### Load audio drivers statically
-### (it's probably better to not load these drivers manually, but instead
-### use module-udev-detect -- see below -- for doing this automatically)
-#load-module module-alsa-sink device="hw:1,0" channels=8 rate=48000 format=s32le
-#load-module module-alsa-source device="hw:1,0" channels=8 rate=48000 format=s32le
-#load-module module-oss device="/dev/dsp" sink_name=output source_name=input
-#load-module module-oss-mmap device="/dev/dsp" sink_name=output source_name=input
-#load-module module-null-sink
-#load-module module-pipe-sink
-
-### Automatically load driver modules depending on the hardware available
-.ifexists module-udev-detect.so
-load-module module-udev-detect 
-#channels=8 rate=48000 format=s32le
-.else
-### Use the static hardware detection module (for systems that lack udev support)
-load-module module-detect
-.endif
-
-### Automatically connect sink and source if JACK server is present
-.ifexists module-jackdbus-detect.so
-.nofail
-load-module module-jackdbus-detect channels=2
-.fail
-.endif
-
-### Automatically load driver modules for Bluetooth hardware
-.ifexists module-bluetooth-policy.so
-load-module module-bluetooth-policy
-.endif
-
-.ifexists module-bluetooth-discover.so
-load-module module-bluetooth-discover
-.endif
-
-### Load several protocols
-.ifexists module-esound-protocol-unix.so
-load-module module-esound-protocol-unix
-.endif
-load-module module-native-protocol-unix
-
-### Network access (may be configured with paprefs, so leave this commented
-### here if you plan to use paprefs)
-#load-module module-esound-protocol-tcp
-#load-module module-native-protocol-tcp
-#load-module module-zeroconf-publish
-
-### Load the RTP receiver module (also configured via paprefs, see above)
-#load-module module-rtp-recv
-
-### Load the RTP sender module (also configured via paprefs, see above)
-#load-module module-null-sink sink_name=rtp format=s16be channels=2 rate=44100 sink_properties="device.description='RTP Multicast Sink'"
-#load-module module-rtp-send source=rtp.monitor
-
-### Load additional modules from GConf settings. This can be configured with the paprefs tool.
-### Please keep in mind that the modules configured by paprefs might conflict with manually
-### loaded modules.
-.ifexists module-gconf.so
-.nofail
-load-module module-gconf
-.fail
-.endif
-
-### Automatically restore the default sink/source when changed by the user
-### during runtime
-### NOTE: This should be loaded as early as possible so that subsequent modules
-### that look up the default sink/source get the right value
-load-module module-default-device-restore
-
-### Automatically move streams to the default sink if the sink they are
-### connected to dies, similar for sources
-load-module module-rescue-streams
-
-### Make sure we always have a sink around, even if it is a null sink.
-load-module module-always-sink
-
-### Honour intended role device property
-load-module module-intended-roles
-
-### Automatically suspend sinks/sources that become idle for too long
-load-module module-suspend-on-idle
-
-### If autoexit on idle is enabled we want to make sure we only quit
-### when no local session needs us anymore.
-.ifexists module-console-kit.so
-load-module module-console-kit
-.endif
-.ifexists module-systemd-login.so
-load-module module-systemd-login
-.endif
-
-### Enable positioned event sounds
-load-module module-position-event-sounds
-
-### Cork music/video streams when a phone stream is active
-load-module module-role-cork
-
-### Modules to allow autoloading of filters (such as echo cancellation)
-### on demand. module-filter-heuristics tries to determine what filters
-### make sense, and module-filter-apply does the heavy-lifting of
-### loading modules and rerouting streams.
-load-module module-filter-heuristics
-load-module module-filter-apply
-
-### Make some devices default
-#set-default-sink output
-#set-default-source input
-set-default-source alsa_input.platform-soc_sound.seeed-8ch
-set-default-sink alsa_output.platform-soc_sound.seeed-2ch
--- a/pulseaudio/pulse_config_6mic/seeed-voicecard.conf
+++ b/pulseaudio/pulse_config_6mic/seeed-voicecard.conf
@ -1,34 +0,0 @@
-# /usr/share/pulseaudio/alsa-mixer/profile-sets/seeed-voiced.conf
-
-[General]
-auto-profiles = no
-[Mapping seeed-8ch]
-device-strings = hw:%f
-channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right
-exact-channels = false
-fallback = yes
-paths-input = seeed-8ch
-priority = 3
-direction = input
-[Mapping seeed-2ch]
-device-strings = hw:%f
-channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right
-exact-channels = false
-exact-channels = false
-fallback = yes
-paths-output = seeed-2ch
-direction = output
-priority = 2
-[Profile output:seeed-2ch+input:seeed-8ch]
-output-mappings = seeed-2ch
-input-mappings = seeed-8ch
-priority = 100
-skip-probe = yes
-[Profile output:seeed-2ch]
-output-mappings = seeed-2ch
-priority = 4
-skip-probe = yes
-[Profile input:seeed-8ch]
-input-mappings = seeed-8ch
-priority = 5
-skip-probe = yes
--- a/pulseaudio/udev_rules_4mic.png
+++ b/pulseaudio/udev_rules_4mic.png
--- a/pulseaudio/udev_rules_6mic.png
+++ b/pulseaudio/udev_rules_6mic.png
--- a/seeed-2mic-voicecard-overlay.dts
+++ b/seeed-2mic-voicecard-overlay.dts
@ -38,8 +38,44 @@
 		};
    };

-
    fragment@3 {
+        target = <&sound>;
+        master_overlay: __dormant__ {
+            compatible = "simple-audio-card";
+            simple-audio-card,format = "i2s";
+            simple-audio-card,name = "seeed-2mic-voicecard";
+            simple-audio-card,bitclock-master = <&dailink0_master>;
+            simple-audio-card,frame-master = <&dailink0_master>;
+            status = "okay";
+                simple-audio-card,widgets =
+                        "Microphone", "Mic Jack",
+                        "Line", "Line In",
+                        "Line", "Line Out",
+                        "Speaker", "Speaker",
+                        "Headphone", "Headphone Jack";
+                simple-audio-card,routing =
+                        "Headphone Jack", "HP_L",
+                        "Headphone Jack", "HP_R",
+                        "Speaker", "SPK_LP",
+                        "Speaker", "SPK_LN",
+                        "LINPUT1", "Mic Jack",
+                        "LINPUT3", "Mic Jack",
+                        "RINPUT1", "Mic Jack",
+                        "RINPUT2", "Mic Jack";
+
+
+            simple-audio-card,cpu {
+                sound-dai = <&i2s>;
+            };
+            dailink0_master: simple-audio-card,codec {
+                sound-dai = <&wm8960>;
+                        clocks = <&wm8960_mclk>;
+                        clock-names = "mclk";
+            };
+        };
+    };
+
+    fragment@4 {
        target = <&sound>;
        slave_overlay: __overlay__ {
                compatible = "simple-audio-card";
@ -78,7 +114,8 @@
    };

    __overrides__ {
-        alsaname = <&slave_overlay>,"simple-audio-card,name";
+        alsaname = <&master_overlay>,"simple-audio-card,name",
+                    <&slave_overlay>,"simple-audio-card,name";
        compatible = <&wm8960>,"compatible";
        master = <0>,"=2!3";
    };
--- a/seeed-2mic-voicecard.dtbo
+++ b/seeed-2mic-voicecard.dtbo
--- a/seeed-4mic-voicecard-overlay.dts
+++ b/seeed-4mic-voicecard-overlay.dts
@ -3,7 +3,6 @@

 / {
    compatible = "brcm,bcm2708";
-
 	fragment@0 {
 		target = <&i2s>;
 		__overlay__ {
@ -11,9 +10,8 @@
 			status = "okay";
        	};
 	};
-
    fragment@1 {
-		target-path = "/";
+        target-path = "/clocks";
        __overlay__ {
            ac108_mclk: codec-mclk {
                compatible = "fixed-clock";
@ -22,7 +20,6 @@
            };  
        };
    };
-
    fragment@2 {
 		target = <&i2c1>;
 		__overlay__ {
@ -34,42 +31,36 @@
 				compatible = "x-power,ac108_0";
 				reg = <0x3b>;
 				#sound-dai-cells = <0>;
-				data-protocol = <0>;
+                data-protocol = <1>;
 			};
 		};
    };


+
    fragment@3 {
        target = <&sound>;
-
        sound_overlay: __overlay__ {
-			compatible = "seeed-voicecard";
-			seeed-voice-card,format = "dsp_a";
-			seeed-voice-card,name = "seeed-4mic-voicecard"; 
+                compatible = "simple-audio-card";
+                simple-audio-card,format = "i2s";
+                simple-audio-card,name = "seeed-4mic-voicecard"; 
                status = "okay";
                
-			seeed-voice-card,bitclock-master = <&codec_dai>;
-			seeed-voice-card,frame-master = <&codec_dai>;
-			seeed-voice-card,channels-playback-override = <4>;
-			seeed-voice-card,channels-capture-override  = <4>;
-
-			cpu_dai: seeed-voice-card,cpu {
+                simple-audio-card,bitclock-master = <&dailink0_slave>;
+                simple-audio-card,frame-slave = <&dailink0_slave>;               
+                dailink0_slave: simple-audio-card,cpu {
                    sound-dai = <&i2s>;
-				dai-tdm-slot-num     = <2>;
-				dai-tdm-slot-width   = <32>;
-				dai-tdm-slot-tx-mask = <1 1 0 0>;
-				dai-tdm-slot-rx-mask = <1 1 0 0>;
                };
-			codec_dai: seeed-voice-card,codec {
+                codec_dai: simple-audio-card,codec {
                    sound-dai = <&ac108_a>;
                    clocks =  <&ac108_mclk>;
                };
        };
    };
-
    __overrides__ {
-		card-name = <&sound_overlay>,"seeed-voice-card,name";
-	};
+        card-name = <&sound_overlay>,"seeed-voicecard,name";
+    };    
+
+    
 };

--- a/seeed-4mic-voicecard.dtbo
+++ b/seeed-4mic-voicecard.dtbo
--- a/seeed-8mic-voicecard-overlay.dts
+++ b/seeed-8mic-voicecard-overlay.dts
@ -1,118 +0,0 @@
-/dts-v1/;
-/plugin/;
-
-/ {
-	compatible = "brcm,bcm2708";
-
-	fragment@0 {
-		target = <&i2s>;
-		__overlay__ {
-			#sound-dai-cells = <0>;
-			status = "okay";
-		};
-	};
-
-	fragment@1 {
-		target-path = "/";
-		__overlay__ {
-			ac10x_mclk: codec-mclk {
-				compatible = "fixed-clock";
-				#clock-cells = <0>;
-				clock-frequency = <24000000>;
-			};  
-		};
-	};
-
-	fragment@2 {
-		target = <&gpio>;
-		__overlay__ {
-			spk_amp_pins: spk_pins {
-				brcm,pins = <17 22>;
-				brcm,function = <1 0>; /* out in */
-				brcm,pull = <0 0>;     /* -   - */
-			};
-			gpclk0_pins: gpclk0_pins {
-				brcm,pins = <4>;
-				brcm,function = <4>; /* alt func 0 */
-				brcm,pull = <0>;     /* - */
-			};
-		};
-	};
-
-	fragment@3 {
-		target = <&i2c1>;
-		__overlay__ {
-			#address-cells = <1>;
-			#size-cells = <0>;
-			status = "okay";
-
-			ac101: ac101@1a{
-				compatible = "x-power,ac101";
-				pinctrl-names = "default";
-				pinctrl-0 = <&spk_amp_pins &gpclk0_pins>;
-				spk-amp-switch-gpios = <&gpio 17 0>;
-				switch-irq-gpios = <&gpio 22 0>;
-				reg = <0x1a>;
-				#sound-dai-cells = <0>;
-			};
-
-			ac108_a: ac108@35{
-				compatible = "x-power,ac108_0";
-				reg = <0x35>;
-				#sound-dai-cells = <0>;
-				data-protocol = <0>;
-				tdm-chips-count = <2>;
-			};
-
-			ac108_b: ac108@3b{
-				compatible = "x-power,ac108_1";
-				reg = <0x3b>;
-				#sound-dai-cells = <0>;
-				data-protocol = <0>;
-				tdm-chips-count = <2>;
-			};
-		};
-	};
-
-	fragment@4 {
-		target = <&sound>;
-
-		sound_overlay: __overlay__ {
-			compatible = "seeed-voicecard";
-			seeed-voice-card,name = "seeed-8mic-voicecard"; 
-			seeed-voice-card,channels-playback-override = <8>;
-			seeed-voice-card,channels-capture-override  = <8>;
-			#address-cells = <1>;
-			#size-cells = <0>;
-			status = "okay";
-
-			seeed-voice-card,dai-link@0 {
-				format = "dsp_a";
-				bitclock-master = <&codec0_dai>;
-				frame-master = <&codec0_dai>;
-				/* bitclock-inversion; */
-				/* frame-inversion; */
-				reg = <0>;
-
-				cpu {
-					sound-dai = <&i2s>;
-					dai-tdm-slot-num     = <2>;
-					dai-tdm-slot-width   = <32>;
-					dai-tdm-slot-tx-mask = <1 1 0 0>;
-					dai-tdm-slot-rx-mask = <1 1 0 0>;
-				};
-
-				codec0_dai: codec {
-					sound-dai = <&ac108_a>;
-					clocks =  <&ac10x_mclk>;
-					system-clock-id = <1>;
-				};
-			};
-		};
-	};
-
-	__overrides__ {
-		card-name = <&sound_overlay>,"seeed-voice-card,name";
-	};
-};
-
--- a/seeed-8mic-voicecard.dtbo
+++ b/seeed-8mic-voicecard.dtbo
--- a/163
+++ b/163
@ -1,163 +0,0 @@
-#!/bin/bash
-#
-# Copyright (c) 2018 Baozhu Zuo <zuobaozhu@gmail.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 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.
-
-set -x
-#exec 1>/var/log/$(basename $0).log 2>&1
-
-export PATH=$PATH:/opt/vc/bin
-OVERLAYS=/boot/overlays
-[ -d /boot/firmware/overlays ] && OVERLAYS=/boot/firmware/overlays
-
-#enable i2c interface
-dtparam -d $OVERLAYS i2c_arm=on
-modprobe i2c-dev
-
-#enable spi interface
-dtparam -d $OVERLAYS spi=on
-
-_VER_RUN=
-function get_kernel_version() {
-    local ZIMAGE IMG_OFFSET
-
-    _VER_RUN=""
-    [ -z "$_VER_RUN" ] && {
-        ZIMAGE=/boot/kernel.img
-        IMG_OFFSET=$(LC_ALL=C grep -abo $'\x1f\x8b\x08\x00' $ZIMAGE | head -n 1 | cut -d ':' -f 1)
-        # 64-bit-only kernel package
-        [ ! -f /boot/kernel.img ] && [ -f /boot/kernel8.img ] && ZIMAGE=/boot/kernel8.img
-        _VER_RUN=$(dd if=$ZIMAGE obs=64K ibs=4 skip=$(( IMG_OFFSET / 4)) 2>/dev/null | zcat | grep -a -m1 "Linux version" | LC_ALL=C sed -e 's/^.*Linux/Linux/' | strings | awk '{ print $3; }')
-    }
-    echo "$_VER_RUN"
-    return 0
-}
-
-CONFIG=/boot/config.txt
-[ -f /boot/firmware/usercfg.txt ] && CONFIG=/boot/firmware/usercfg.txt
-
-get_overlay() {
-    ov=$1
-    if grep -q -E "^dtoverlay=$ov" $CONFIG; then
-      echo 0
-    else
-      echo 1
-    fi
-}
-
-do_overlay() {
-    ov=$1
-    RET=$2
-    DEFAULT=--defaultno
-    CURRENT=0
-    if [ $(get_overlay $ov) -eq 0 ]; then
-        DEFAULT=
-        CURRENT=1
-    fi
-    if [ $RET -eq $CURRENT ]; then
-        ASK_TO_REBOOT=1
-    fi
-    if [ $RET -eq 0 ]; then
-        sed $CONFIG -i -e "s/^#dtoverlay=$ov/dtoverlay=$ov/"
-        if ! grep -q -E "^dtoverlay=$ov" $CONFIG; then
-            printf "dtoverlay=$ov\n" >> $CONFIG
-        fi
-        STATUS=enabled
-    elif [ $RET -eq 1 ]; then
-        sed $CONFIG -i -e "s/^dtoverlay=$ov/#dtoverlay=$ov/"
-        STATUS=disabled
-    else
-        return $RET
-    fi
-}
-
-
-is_1a=$(i2cdetect -y  1 0x1a 0x1a | egrep "(1a|UU)" | awk '{print $2}')
-is_35=$(i2cdetect -y  1 0x35 0x35 | egrep "(35|UU)" | awk '{print $2}')
-is_3b=$(i2cdetect -y  1 0x3b 0x3b | egrep "(3b|UU)" | awk '{print $2}')
-
-RPI_HATS="seeed-2mic-voicecard seeed-4mic-voicecard seeed-8mic-voicecard"
-overlay=""
-
-if [ "x${is_1a}" != "x" ] && [ "x${is_35}" == "x" ] ; then
-    echo "install 2mic"
-    overlay=seeed-2mic-voicecard
-    asound_conf=/etc/voicecard/asound_2mic.conf
-    asound_state=/etc/voicecard/wm8960_asound.state
-fi
-
-if [ "x${is_3b}" != "x" ] && [ "x${is_35}" == "x" ] ; then
-    echo "install 4mic"
-    overlay=seeed-4mic-voicecard
-    asound_conf=/etc/voicecard/asound_4mic.conf
-    asound_state=/etc/voicecard/ac108_asound.state
-fi
-
-if [ "x${is_3b}" != "x" ] && [ "x${is_35}" != "x" ] ; then
-    echo "install 6mic"
-    overlay=seeed-8mic-voicecard
-    asound_conf=/etc/voicecard/asound_6mic.conf
-    asound_state=/etc/voicecard/ac108_6mic.state
-fi
-
-if [ "$overlay" ]; then
-    echo Install $overlay ...
-
-    # Remove old configuration
-    rm /etc/asound.conf
-    rm /var/lib/alsa/asound.state
-
-    kernel_ver=$(uname -r) # get_kernel_version)
-    # echo kernel_ver=$kernel_ver
-
-    # TODO: dynamic dtoverlay Bug of v4.19.x
-    #       no DT node phandle inserted.
-    if [[ "$kernel_ver" =~ ^4\.19.*$ || "$kernel_ver" =~ ^5\.*$ ]]; then
-        for i in $RPI_HATS; do
-            if [ "$i" == "$overlay" ]; then
-                /bin/true #do_overlay $overlay 0
-            else
-                echo Uninstall $i ...
-                /bin/true #do_overlay $i 1
-            fi
-        done
-    fi
-    #make sure the driver loads correctly
-    dtoverlay -d $OVERLAYS $overlay || true
-
-
-    echo "create $overlay asound configure file"
-    ln -s $asound_conf /etc/asound.conf
-    echo "create $overlay asound status file"
-    ln -s $asound_state /var/lib/alsa/asound.state
-fi
-
-alsactl restore
-
-#Force 3.5mm ('headphone') jack
-#   The Raspberry Pi 4, released on 24th Jun 2019, has two HDMI ports,
-#   and can drive two displays with audios for two users simultaneously,
-#   in a "multiseat" configuration. The earlier single virtual ALSA
-#   option for re-directing audio playback between headphone jack and HDMI
-#   via a 'Routing' mixer setting was turned off eventually to allow
-#   simultaneous usage of all 3 playback devices.
-if aplay -l | grep -q "bcm2835 ALSA"; then
-    amixer cset numid=3 1 || true
-fi
--- a/seeed-voicecard.c
+++ b/seeed-voicecard.c
@ -1,926 +0,0 @@
-/*
- * SEEED voice card
- *
- * (C) Copyright 2017-2018
- * Seeed Technology Co., Ltd. <www.seeedstudio.com>
- *
- * base on ASoC simple sound card support
- *
- * Copyright (C) 2012 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@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.
- */
-/* #undef DEBUG */
-#include <linux/version.h>
-#include <linux/clk.h>
-#include <linux/device.h>
-#include <linux/gpio.h>
-#include <linux/module.h>
-#include <linux/of.h>
-#include <linux/of_gpio.h>
-#include <linux/platform_device.h>
-#include <linux/string.h>
-#include <sound/soc.h>
-#include <sound/soc-dai.h>
-#include <sound/simple_card_utils.h>
-#include "ac10x.h"
-
-#define LINUX_VERSION_IS_GEQ(x1,x2,x3)	(LINUX_VERSION_CODE >= KERNEL_VERSION(x1,x2,x3))
-
-
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,13,0)
-#define asoc_simple_parse_clk_cpu(dev, node, dai_link, simple_dai)      \
-  asoc_simple_parse_clk(dev, node, simple_dai, dai_link->cpus)
-#define asoc_simple_parse_clk_codec(dev, node, dai_link, simple_dai)    \
-  asoc_simple_parse_clk(dev, node, simple_dai, dai_link->codecs)
-#define asoc_simple_parse_cpu(node, dai_link, is_single_link)           \
-  asoc_simple_parse_dai(node, dai_link->cpus, is_single_link)
-#define asoc_simple_parse_codec(node, dai_link)                         \
-  asoc_simple_parse_dai(node, dai_link->codecs, NULL)
-#define asoc_simple_parse_platform(node, dai_link)                      \
-  asoc_simple_parse_dai(node, dai_link->platforms, NULL)
-#endif
-
-/*
- * single codec:
- *	0 - allow multi codec
- *	1 - yes
- */
-#define _SINGLE_CODEC		1
-
-struct seeed_card_data {
-	struct snd_soc_card snd_card;
-	struct seeed_dai_props {
-		struct asoc_simple_dai cpu_dai;
-		struct asoc_simple_dai codec_dai;
-		struct snd_soc_dai_link_component cpus;   /* single cpu */
-		struct snd_soc_dai_link_component codecs; /* single codec */
-		struct snd_soc_dai_link_component platforms;
-		unsigned int mclk_fs;
-	} *dai_props;
-	unsigned int mclk_fs;
-	unsigned channels_playback_default;
-	unsigned channels_playback_override;
-	unsigned channels_capture_default;
-	unsigned channels_capture_override;
-	struct snd_soc_dai_link *dai_link;
-	#if CONFIG_AC10X_TRIG_LOCK
-	spinlock_t lock;
-	#endif
-	struct work_struct work_codec_clk;
-	#define TRY_STOP_MAX	3
-	int try_stop;
-};
-
-struct seeed_card_info {
-	const char *name;
-	const char *card;
-	const char *codec;
-	const char *platform;
-
-	unsigned int daifmt;
-	struct asoc_simple_dai cpu_dai;
-	struct asoc_simple_dai codec_dai;
-};
-
-#define seeed_priv_to_card(priv) (&(priv)->snd_card)
-#define seeed_priv_to_dev(priv) ((priv)->snd_card.dev)
-#define seeed_priv_to_link(priv, i) ((priv)->snd_card.dai_link + (i))
-#define seeed_priv_to_props(priv, i) ((priv)->dai_props + (i))
-
-#define DAI	"sound-dai"
-#define CELL	"#sound-dai-cells"
-#define PREFIX	"seeed-voice-card,"
-
-static int seeed_voice_card_startup(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct seeed_card_data *priv =	snd_soc_card_get_drvdata(rtd->card);
-	struct seeed_dai_props *dai_props =
-		seeed_priv_to_props(priv, rtd->num);
-	int ret;
-
-	ret = clk_prepare_enable(dai_props->cpu_dai.clk);
-	if (ret)
-		return ret;
-
-	ret = clk_prepare_enable(dai_props->codec_dai.clk);
-	if (ret)
-		clk_disable_unprepare(dai_props->cpu_dai.clk);
-
-	if (asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min) {
-		priv->channels_playback_default = asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min;
-	}
-	if (asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min) {
-		priv->channels_capture_default = asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min;
-	}
-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min = priv->channels_playback_override;
-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_max = priv->channels_playback_override;
-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min = priv->channels_capture_override;
-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_max = priv->channels_capture_override;
-
-	return ret;
-}
-
-static void seeed_voice_card_shutdown(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct seeed_card_data *priv =	snd_soc_card_get_drvdata(rtd->card);
-	struct seeed_dai_props *dai_props =
-		seeed_priv_to_props(priv, rtd->num);
-
-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min = priv->channels_playback_default;
-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_max = priv->channels_playback_default;
-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min = priv->channels_capture_default;
-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_max = priv->channels_capture_default;
-
-	clk_disable_unprepare(dai_props->cpu_dai.clk);
-
-	clk_disable_unprepare(dai_props->codec_dai.clk);
-}
-
-static int seeed_voice_card_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 = asoc_rtd_to_codec(rtd, 0);
-	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
-	struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
-	struct seeed_dai_props *dai_props =
-		seeed_priv_to_props(priv, rtd->num);
-	unsigned int mclk, mclk_fs = 0;
-	int ret = 0;
-
-	if (priv->mclk_fs)
-		mclk_fs = priv->mclk_fs;
-	else if (dai_props->mclk_fs)
-		mclk_fs = dai_props->mclk_fs;
-
-	if (mclk_fs) {
-		mclk = params_rate(params) * mclk_fs;
-		ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
-					     SND_SOC_CLOCK_IN);
-		if (ret && ret != -ENOTSUPP)
-			goto err;
-
-		ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk,
-					     SND_SOC_CLOCK_OUT);
-		if (ret && ret != -ENOTSUPP)
-			goto err;
-	}
-	return 0;
-err:
-	return ret;
-}
-
-#define _SET_CLOCK_CNT		2
-static int (* _set_clock[_SET_CLOCK_CNT])(int y_start_n_stop, struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai);
-
-int seeed_voice_card_register_set_clock(int stream, int (*set_clock)(int, struct snd_pcm_substream *, int, struct snd_soc_dai *)) {
-	if (! _set_clock[stream]) {
-		_set_clock[stream] = set_clock;
-	}
-	return 0;
-}
-EXPORT_SYMBOL(seeed_voice_card_register_set_clock);
-
-/*
- * work_cb_codec_clk: clear audio codec inner clock.
- */
-static void work_cb_codec_clk(struct work_struct *work)
-{
-	struct seeed_card_data *priv = container_of(work, struct seeed_card_data, work_codec_clk);
-	int r = 0;
-
-	if (_set_clock[SNDRV_PCM_STREAM_CAPTURE]) {
-		r = r || _set_clock[SNDRV_PCM_STREAM_CAPTURE](0, NULL, 0, NULL); /* not using 2nd to 4th arg if 1st == 0 */
-	}
-	if (_set_clock[SNDRV_PCM_STREAM_PLAYBACK]) {
-		r = r || _set_clock[SNDRV_PCM_STREAM_PLAYBACK](0, NULL, 0, NULL); /* not using 2nd to 4th arg if 1st == 0 */
-	}
-
-	if (r && priv->try_stop++ < TRY_STOP_MAX) {
-		if (0 != schedule_work(&priv->work_codec_clk)) {}
-	}
-	return;
-}
-
-static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
-	struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
-	#if CONFIG_AC10X_TRIG_LOCK
-	unsigned long flags;
-	#endif
-	int ret = 0;
-
-	dev_dbg(rtd->card->dev, "%s() stream=%s  cmd=%d play:%d, capt:%d\n",
-		__FUNCTION__, snd_pcm_stream_str(substream), cmd,
-		dai->stream[SNDRV_PCM_STREAM_PLAYBACK].active, dai->stream[SNDRV_PCM_STREAM_CAPTURE].active);
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_RESUME:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (cancel_work_sync(&priv->work_codec_clk) != 0) {}
-		#if CONFIG_AC10X_TRIG_LOCK
-		/* I know it will degrades performance, but I have no choice */
-		spin_lock_irqsave(&priv->lock, flags);
-		#endif
-		// if (_set_clock[SNDRV_PCM_STREAM_CAPTURE]) _set_clock[SNDRV_PCM_STREAM_CAPTURE](1, substream, cmd, dai);
-		// if (_set_clock[SNDRV_PCM_STREAM_PLAYBACK]) _set_clock[SNDRV_PCM_STREAM_PLAYBACK](1, substream, cmd, dai);
-		#if CONFIG_AC10X_TRIG_LOCK
-		spin_unlock_irqrestore(&priv->lock, flags);
-		#endif
-		break;
-
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		/* capture channel resync, if overrun */
-		if (dai->stream[SNDRV_PCM_STREAM_CAPTURE].active && substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			break;
-		}
-
-		/* interrupt environment */
-		if (in_irq() || in_nmi() || in_serving_softirq()) {
-			priv->try_stop = 0;
-			if (0 != schedule_work(&priv->work_codec_clk)) {
-			}
-		} else {
-			// if (_set_clock[SNDRV_PCM_STREAM_CAPTURE]) _set_clock[SNDRV_PCM_STREAM_CAPTURE](0, NULL, 0, NULL); /* not using 2nd to 4th arg if 1st == 0 */
-			// if (_set_clock[SNDRV_PCM_STREAM_PLAYBACK]) _set_clock[SNDRV_PCM_STREAM_PLAYBACK](0, NULL, 0, NULL); /* not using 2nd to 4th arg if 1st == 0 */
-		}
-		break;
-	default:
-		ret = -EINVAL;
-	}
-
-	dev_dbg(rtd->card->dev, "%s() stream=%s  cmd=%d play:%d, capt:%d;finished %d\n",
-		__FUNCTION__, snd_pcm_stream_str(substream), cmd,
-		dai->stream[SNDRV_PCM_STREAM_PLAYBACK].active, dai->stream[SNDRV_PCM_STREAM_CAPTURE].active, ret);
-
-	return ret;
-}
-
-static struct snd_soc_ops seeed_voice_card_ops = {
-	.startup = seeed_voice_card_startup,
-	.shutdown = seeed_voice_card_shutdown,
-	.hw_params = seeed_voice_card_hw_params,
-	.trigger = seeed_voice_card_trigger,
-};
-
-static int asoc_simple_parse_dai(struct device_node *node,
-				 struct snd_soc_dai_link_component *dlc,
-				 int *is_single_link)
-{
-	struct of_phandle_args args;
-	int ret;
-
-	if (!node)
-		return 0;
-
-	/*
-	 * Get node via "sound-dai = <&phandle port>"
-	 * it will be used as xxx_of_node on soc_bind_dai_link()
-	 */
-	ret = of_parse_phandle_with_args(node, DAI, CELL, 0, &args);
-	if (ret)
-		return ret;
-
-	/*
-	 * FIXME
-	 *
-	 * Here, dlc->dai_name is pointer to CPU/Codec DAI name.
-	 * If user unbinded CPU or Codec driver, but not for Sound Card,
-	 * dlc->dai_name is keeping unbinded CPU or Codec
-	 * driver's pointer.
-	 *
-	 * If user re-bind CPU or Codec driver again, ALSA SoC will try
-	 * to rebind Card via snd_soc_try_rebind_card(), but because of
-	 * above reason, it might can't bind Sound Card.
-	 * Because Sound Card is pointing to released dai_name pointer.
-	 *
-	 * To avoid this rebind Card issue,
-	 * 1) It needs to alloc memory to keep dai_name eventhough
-	 *    CPU or Codec driver was unbinded, or
-	 * 2) user need to rebind Sound Card everytime
-	 *    if he unbinded CPU or Codec.
-	 */
-	ret = snd_soc_of_get_dai_name(node, &dlc->dai_name, 0);
-	if (ret < 0)
-		return ret;
-
-	dlc->of_node = args.np;
-
-	if (is_single_link)
-		*is_single_link = !args.args_count;
-
-	return 0;
-}
-
-static int asoc_simple_init_dai(struct snd_soc_dai *dai,
-				     struct asoc_simple_dai *simple_dai)
-{
-	int ret;
-
-	if (!simple_dai)
-		return 0;
-
-	if (simple_dai->sysclk) {
-		ret = snd_soc_dai_set_sysclk(dai, 0, simple_dai->sysclk,
-					     simple_dai->clk_direction);
-		if (ret && ret != -ENOTSUPP) {
-			dev_err(dai->dev, "simple-card: set_sysclk error\n");
-			return ret;
-		}
-	}
-
-	if (simple_dai->slots) {
-		ret = snd_soc_dai_set_bclk_ratio(dai,
-					       simple_dai->slots *
-					       simple_dai->slot_width);
-		if (ret && ret != -ENOTSUPP) {
-			dev_err(dai->dev, "simple-card: set_tdm_slot error\n");
-			return ret;
-		}
-	}
-
-	return 0;
-}
-
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,7,0)
-static inline int asoc_simple_component_is_codec(struct snd_soc_component *component)
-{
-	return component->driver->endianness;
-}
-
-static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_dai_link *dai_link = rtd->dai_link;
-	struct snd_soc_component *component;
-	struct snd_soc_pcm_stream *params;
-	struct snd_pcm_hardware hw;
-	int i, ret, stream;
-
-	/* Only Codecs */
-	for_each_rtd_components(rtd, i, component) {
-		if (!asoc_simple_component_is_codec(component))
-			return 0;
-	}
-
-	/* Assumes the capabilities are the same for all supported streams */
-	for (stream = 0; stream < 2; stream++) {
-		ret = snd_soc_runtime_calc_hw(rtd, &hw, stream);
-		if (ret == 0)
-			break;
-	}
-
-	if (ret < 0) {
-		dev_err(rtd->dev, "simple-card: no valid dai_link params\n");
-		return ret;
-	}
-
-	params = devm_kzalloc(rtd->dev, sizeof(*params), GFP_KERNEL);
-	if (!params)
-		return -ENOMEM;
-
-	params->formats = hw.formats;
-	params->rates = hw.rates;
-	params->rate_min = hw.rate_min;
-	params->rate_max = hw.rate_max;
-	params->channels_min = hw.channels_min;
-	params->channels_max = hw.channels_max;
-
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(6,4,0)
-	dai_link->c2c_params = params;
-	dai_link->num_c2c_params = 1;
-#else
-	/* apparently this goes back to 5.6.x */
-	dai_link->params = params;
-	dai_link->num_params = 1;
-#endif
-
-	return 0;
-}
-#endif
-
-static int seeed_voice_card_dai_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct seeed_card_data *priv =	snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *codec = asoc_rtd_to_codec(rtd, 0);
-	struct snd_soc_dai *cpu = asoc_rtd_to_cpu(rtd, 0);
-	struct seeed_dai_props *dai_props =
-		seeed_priv_to_props(priv, rtd->num);
-	int ret;
-
-	ret = asoc_simple_init_dai(codec, &dai_props->codec_dai);
-	if (ret < 0)
-		return ret;
-
-	ret = asoc_simple_init_dai(cpu, &dai_props->cpu_dai);
-	if (ret < 0)
-		return ret;
-
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,7,0)
-	ret = asoc_simple_init_dai_link_params(rtd);
-	if (ret < 0)
-		return ret;
-#endif
-
-	dev_dbg(rtd->card->dev, "codec \"%s\" mapping to cpu \"%s\"\n", codec->name, cpu->name);
-	return 0;
-}
-
-static int seeed_voice_card_dai_link_of(struct device_node *node,
-					struct seeed_card_data *priv,
-					int idx,
-					bool is_top_level_node)
-{
-	struct device *dev = seeed_priv_to_dev(priv);
-	struct snd_soc_dai_link *dai_link = seeed_priv_to_link(priv, idx);
-	struct seeed_dai_props *dai_props = seeed_priv_to_props(priv, idx);
-	struct asoc_simple_dai *cpu_dai = &dai_props->cpu_dai;
-	struct asoc_simple_dai *codec_dai = &dai_props->codec_dai;
-	struct device_node *cpu = NULL;
-	struct device_node *plat = NULL;
-	struct device_node *codec = NULL;
-	char prop[128];
-	char *prefix = "";
-	int ret, single_cpu;
-
-	/* For single DAI link & old style of DT node */
-	if (is_top_level_node)
-		prefix = PREFIX;
-
-	snprintf(prop, sizeof(prop), "%scpu", prefix);
-	cpu = of_get_child_by_name(node, prop);
-
-	if (!cpu) {
-		ret = -EINVAL;
-		dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop);
-		goto dai_link_of_err;
-	}
-
-	snprintf(prop, sizeof(prop), "%splat", prefix);
-	plat = of_get_child_by_name(node, prop);
-
-	snprintf(prop, sizeof(prop), "%scodec", prefix);
-	codec = of_get_child_by_name(node, prop);
-
-	if (!codec) {
-		ret = -EINVAL;
-		dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop);
-		goto dai_link_of_err;
-	}
-
-	ret = asoc_simple_parse_daifmt(dev, node, codec,
-					    prefix, &dai_link->dai_fmt);
-	if (ret < 0)
-		goto dai_link_of_err;
-
-	of_property_read_u32(node, "mclk-fs", &dai_props->mclk_fs);
-
-	ret = asoc_simple_parse_cpu(cpu, dai_link, &single_cpu);
-	if (ret < 0)
-		goto dai_link_of_err;
-
-	#if _SINGLE_CODEC
-	ret = asoc_simple_parse_codec(codec, dai_link);
-	if (ret < 0)
-		goto dai_link_of_err;
-	#else
-	ret = snd_soc_of_get_dai_link_codecs(dev, codec, dai_link);
-	if (ret < 0) {
-		dev_err(dev, "parse codec info error %d\n", ret);
-		goto dai_link_of_err;
-	}
-	dev_dbg(dev, "dai_link num_codecs = %d\n", dai_link->num_codecs);
-	#endif
-
-	ret = asoc_simple_parse_platform(plat, dai_link);
-	if (ret < 0)
-		goto dai_link_of_err;
-
-	ret = snd_soc_of_parse_tdm_slot(cpu,	&cpu_dai->tx_slot_mask,
-						&cpu_dai->rx_slot_mask,
-						&cpu_dai->slots,
-						&cpu_dai->slot_width);
-	dev_dbg(dev, "cpu_dai : slot,width,tx,rx = %d,%d,%d,%d\n", 
-			cpu_dai->slots, cpu_dai->slot_width,
-			cpu_dai->tx_slot_mask, cpu_dai->rx_slot_mask
-			);
-	if (ret < 0)
-		goto dai_link_of_err;
-
-	ret = snd_soc_of_parse_tdm_slot(codec,	&codec_dai->tx_slot_mask,
-						&codec_dai->rx_slot_mask,
-						&codec_dai->slots,
-						&codec_dai->slot_width);
-	if (ret < 0)
-		goto dai_link_of_err;
-
-	#if LINUX_VERSION_CODE <= KERNEL_VERSION(4,10,0)
-	ret = asoc_simple_card_parse_clk_cpu(cpu, dai_link, cpu_dai);
-	#else
-	ret = asoc_simple_parse_clk_cpu(dev, cpu, dai_link, cpu_dai);
-	#endif
-	if (ret < 0)
-		goto dai_link_of_err;
-
-	#if LINUX_VERSION_CODE <= KERNEL_VERSION(4,10,0)
-	ret = asoc_simple_card_parse_clk_codec(codec, dai_link, codec_dai);
-	#else
-	ret = asoc_simple_parse_clk_codec(dev, codec, dai_link, codec_dai);
-	#endif
-	if (ret < 0)
-		goto dai_link_of_err;
-
-	ret = asoc_simple_set_dailink_name(dev, dai_link,
-						"%s-%s",
-						dai_link->cpus->dai_name,
-						#if _SINGLE_CODEC
-						dai_link->codecs->dai_name
-						#else
-						dai_link->codecs[0].dai_name
-						#endif
-	);
-	if (ret < 0)
-		goto dai_link_of_err;
-
-	dai_link->ops = &seeed_voice_card_ops;
-	dai_link->init = seeed_voice_card_dai_init;
-
-	dev_dbg(dev, "\tname : %s\n", dai_link->stream_name);
-	dev_dbg(dev, "\tformat : %04x\n", dai_link->dai_fmt);
-	dev_dbg(dev, "\tcpu : %s / %d\n",
-		dai_link->cpus->dai_name,
-		dai_props->cpu_dai.sysclk);
-	dev_dbg(dev, "\tcodec : %s / %d\n",
-		#if _SINGLE_CODEC
-		dai_link->codecs->dai_name,
-		#else
-		dai_link->codecs[0].dai_name,
-		#endif
-		dai_props->codec_dai.sysclk);
-
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,13,0)
-	asoc_simple_canonicalize_cpu(dai_link->cpus, single_cpu);
-	#if _SINGLE_CODEC
-	asoc_simple_canonicalize_platform(dai_link->platforms, dai_link->cpus);
-	#endif
-#else
-	asoc_simple_canonicalize_cpu(dai_link, single_cpu);
-	#if _SINGLE_CODEC
-	asoc_simple_canonicalize_platform(dai_link);
-	#endif
-#endif
-
-dai_link_of_err:
-	of_node_put(cpu);
-	of_node_put(codec);
-
-	return ret;
-}
-
-static int seeed_voice_card_parse_aux_devs(struct device_node *node,
-					   struct seeed_card_data *priv)
-{
-	struct device *dev = seeed_priv_to_dev(priv);
-	struct device_node *aux_node;
-	int i, n, len;
-
-	if (!of_find_property(node, PREFIX "aux-devs", &len))
-		return 0;		/* Ok to have no aux-devs */
-
-	n = len / sizeof(__be32);
-	if (n <= 0)
-		return -EINVAL;
-
-	priv->snd_card.aux_dev = devm_kzalloc(dev,
-			n * sizeof(*priv->snd_card.aux_dev), GFP_KERNEL);
-	if (!priv->snd_card.aux_dev)
-		return -ENOMEM;
-
-	for (i = 0; i < n; i++) {
-		aux_node = of_parse_phandle(node, PREFIX "aux-devs", i);
-		if (!aux_node)
-			return -EINVAL;
-		priv->snd_card.aux_dev[i].dlc.of_node = aux_node;
-	}
-
-	priv->snd_card.num_aux_devs = n;
-	return 0;
-}
-
-static int seeed_voice_card_parse_of(struct device_node *node,
-				     struct seeed_card_data *priv)
-{
-	struct device *dev = seeed_priv_to_dev(priv);
-	struct device_node *dai_link;
-	int ret;
-
-	if (!node)
-		return -EINVAL;
-
-	dai_link = of_get_child_by_name(node, PREFIX "dai-link");
-
-	/* The off-codec widgets */
-	if (of_property_read_bool(node, PREFIX "widgets")) {
-		ret = snd_soc_of_parse_audio_simple_widgets(&priv->snd_card,
-					PREFIX "widgets");
-		if (ret)
-			goto card_parse_end;
-	}
-
-	/* DAPM routes */
-	if (of_property_read_bool(node, PREFIX "routing")) {
-		ret = snd_soc_of_parse_audio_routing(&priv->snd_card,
-					PREFIX "routing");
-		if (ret)
-			goto card_parse_end;
-	}
-
-	/* Factor to mclk, used in hw_params() */
-	of_property_read_u32(node, PREFIX "mclk-fs", &priv->mclk_fs);
-
-	/* Single/Muti DAI link(s) & New style of DT node */
-	if (dai_link) {
-		struct device_node *np = NULL;
-		int i = 0;
-
-		for_each_child_of_node(node, np) {
-			dev_dbg(dev, "\tlink %d:\n", i);
-			ret = seeed_voice_card_dai_link_of(np, priv,
-							   i, false);
-			if (ret < 0) {
-				of_node_put(np);
-				goto card_parse_end;
-			}
-			i++;
-		}
-	} else {
-		/* For single DAI link & old style of DT node */
-		ret = seeed_voice_card_dai_link_of(node, priv, 0, true);
-		if (ret < 0)
-			goto card_parse_end;
-	}
-
-	ret = asoc_simple_parse_card_name(&priv->snd_card, PREFIX);
-	if (ret < 0)
-		goto card_parse_end;
-
-	ret = seeed_voice_card_parse_aux_devs(node, priv);
-
-	priv->channels_playback_default  = 0;
-	priv->channels_playback_override = 2;
-	priv->channels_capture_default   = 0;
-	priv->channels_capture_override  = 2;
-	of_property_read_u32(node, PREFIX "channels-playback-default",
-				    &priv->channels_playback_default);
-	of_property_read_u32(node, PREFIX "channels-playback-override",
-				    &priv->channels_playback_override);
-	of_property_read_u32(node, PREFIX "channels-capture-default",
-				    &priv->channels_capture_default);
-	of_property_read_u32(node, PREFIX "channels-capture-override",
-				    &priv->channels_capture_override);
-
-card_parse_end:
-	of_node_put(dai_link);
-
-	return ret;
-}
-
-#ifdef DEBUG
-inline void seeed_debug_dai(struct seeed_card_data *priv,
-				  char *name,
-				  struct asoc_simple_dai *dai)
-{
-	struct device *dev = seeed_priv_to_dev(priv);
-
-	if (dai->name)
-		dev_dbg(dev, "%s dai name = %s\n",
-			name, dai->name);
-	if (dai->sysclk)
-		dev_dbg(dev, "%s sysclk = %d\n",
-			name, dai->sysclk);
-
-	dev_dbg(dev, "%s direction = %s\n",
-		name, dai->clk_direction ? "OUT" : "IN");
-
-	if (dai->slots)
-		dev_dbg(dev, "%s slots = %d\n", name, dai->slots);
-	if (dai->slot_width)
-		dev_dbg(dev, "%s slot width = %d\n", name, dai->slot_width);
-	if (dai->tx_slot_mask)
-		dev_dbg(dev, "%s tx slot mask = %d\n", name, dai->tx_slot_mask);
-	if (dai->rx_slot_mask)
-		dev_dbg(dev, "%s rx slot mask = %d\n", name, dai->rx_slot_mask);
-	if (dai->clk)
-		dev_dbg(dev, "%s clk %luHz\n", name, clk_get_rate(dai->clk));
-}
-
-inline void seeed_debug_info(struct seeed_card_data *priv)
-{
-	struct snd_soc_card *card = seeed_priv_to_card(priv);
-	struct device *dev = seeed_priv_to_dev(priv);
-
-	int i;
-
-	if (card->name)
-		dev_dbg(dev, "Card Name: %s\n", card->name);
-
-	for (i = 0; i < card->num_links; i++) {
-		struct seeed_dai_props *props = seeed_priv_to_props(priv, i);
-		struct snd_soc_dai_link *link = seeed_priv_to_link(priv, i);
-
-		dev_dbg(dev, "DAI%d\n", i);
-
-		seeed_debug_dai(priv, "cpu", &props->cpu_dai);
-		seeed_debug_dai(priv, "codec", &props->codec_dai);
-
-		if (link->name)
-			dev_dbg(dev, "dai name = %s\n", link->name);
-
-		dev_dbg(dev, "dai format = %04x\n", link->dai_fmt);
-
-		/*
-		if (props->adata.convert_rate)
-			dev_dbg(dev, "convert_rate = %d\n",
-				props->adata.convert_rate);
-		if (props->adata.convert_channels)
-			dev_dbg(dev, "convert_channels = %d\n",
-				props->adata.convert_channels);
-		if (props->codec_conf && props->codec_conf->name_prefix)
-			dev_dbg(dev, "name prefix = %s\n",
-				props->codec_conf->name_prefix);
-		*/
-		if (props->mclk_fs)
-			dev_dbg(dev, "mclk-fs = %d\n",
-				props->mclk_fs);
-	}
-}
-#else
-#define  seeed_debug_info(priv)
-#endif /* DEBUG */
-
-static int seeed_voice_card_probe(struct platform_device *pdev)
-{
-	struct seeed_card_data *priv;
-	struct snd_soc_dai_link *dai_link;
-	struct seeed_dai_props *dai_props;
-	struct device_node *np = pdev->dev.of_node;
-	struct device *dev = &pdev->dev;
-	int num, ret, i;
-
-	/* Get the number of DAI links */
-	if (np && of_get_child_by_name(np, PREFIX "dai-link"))
-		num = of_get_child_count(np);
-	else
-		num = 1;
-
-	/* Allocate the private data and the DAI link array */
-	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
-	if (!priv)
-		return -ENOMEM;
-
-	dai_props = devm_kzalloc(dev, sizeof(*dai_props) * num, GFP_KERNEL);
-	dai_link  = devm_kzalloc(dev, sizeof(*dai_link)  * num, GFP_KERNEL);
-	if (!dai_props || !dai_link)
-		return -ENOMEM;
-
-	/*
-	 * Use snd_soc_dai_link_component instead of legacy style
-	 * It is codec only. but cpu/platform will be supported in the future.
-	 * see
-	 *      soc-core.c :: snd_soc_init_multicodec()
-	 *
-	 * "platform" might be removed
-	 * see
-	 *      simple-card-utils.c :: asoc_simple_canonicalize_platform()
-	 */
-	for (i = 0; i < num; i++) {
-		dai_link[i].cpus		= &dai_props[i].cpus;
-		dai_link[i].num_cpus		= 1;
-		dai_link[i].codecs		= &dai_props[i].codecs;
-		dai_link[i].num_codecs		= 1;
-		dai_link[i].platforms		= &dai_props[i].platforms;
-		dai_link[i].num_platforms	= 1;
-	}
-
-	priv->dai_props			= dai_props;
-	priv->dai_link			= dai_link;
-
-	/* Init snd_soc_card */
-	priv->snd_card.owner		= THIS_MODULE;
-	priv->snd_card.dev		= dev;
-	priv->snd_card.dai_link		= priv->dai_link;
-	priv->snd_card.num_links	= num;
-
-	if (np && of_device_is_available(np)) {
-		ret = seeed_voice_card_parse_of(np, priv);
-		if (ret < 0) {
-			if (ret != -EPROBE_DEFER)
-				dev_err(dev, "parse error %d\n", ret);
-			goto err;
-		}
-	} else {
-		struct seeed_card_info *cinfo;
-		struct snd_soc_dai_link_component *cpus;
-		struct snd_soc_dai_link_component *codecs;
-		struct snd_soc_dai_link_component *platform;
-
-		cinfo = dev->platform_data;
-		if (!cinfo) {
-			dev_err(dev, "no info for seeed-voice-card\n");
-			return -EINVAL;
-		}
-
-		if (!cinfo->name ||
-		    !cinfo->codec_dai.name ||
-		    !cinfo->codec ||
-		    !cinfo->platform ||
-		    !cinfo->cpu_dai.name) {
-			dev_err(dev, "insufficient seeed_voice_card_info settings\n");
-			return -EINVAL;
-		}
-
-		cpus			= dai_link->cpus;
-		cpus->dai_name		= cinfo->cpu_dai.name;
-
-		codecs			= dai_link->codecs;
-		codecs->name		= cinfo->codec;
-		codecs->dai_name	= cinfo->codec_dai.name;
-
-		platform		= dai_link->platforms;
-		platform->name		= cinfo->platform;
-
-		priv->snd_card.name	= (cinfo->card) ? cinfo->card : cinfo->name;
-		dai_link->name		= cinfo->name;
-		dai_link->stream_name	= cinfo->name;
-		dai_link->dai_fmt	= cinfo->daifmt;
-		dai_link->init		= seeed_voice_card_dai_init;
-		memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai,
-					sizeof(priv->dai_props->cpu_dai));
-		memcpy(&priv->dai_props->codec_dai, &cinfo->codec_dai,
-					sizeof(priv->dai_props->codec_dai));
-	}
-
-	snd_soc_card_set_drvdata(&priv->snd_card, priv);
-
-	#if CONFIG_AC10X_TRIG_LOCK
-	spin_lock_init(&priv->lock);
-	#endif
-
-	INIT_WORK(&priv->work_codec_clk, work_cb_codec_clk);
-
-	seeed_debug_info(priv);
-
-	ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card);
-	if (ret >= 0)
-		return ret;
-
-err:
-	asoc_simple_clean_reference(&priv->snd_card);
-
-	return ret;
-}
-
-static int seeed_voice_card_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-	struct seeed_card_data *priv = snd_soc_card_get_drvdata(card);
-
-	if (cancel_work_sync(&priv->work_codec_clk) != 0) {
-	}
-	asoc_simple_clean_reference(card);
-
-	return 0;
-}
-
-static const struct of_device_id seeed_voice_of_match[] = {
-	{ .compatible = "seeed-voicecard", },
-	{},
-};
-MODULE_DEVICE_TABLE(of, seeed_voice_of_match);
-
-static struct platform_driver seeed_voice_card = {
-	.driver = {
-		.name = "seeed-voicecard",
-		.pm = &snd_soc_pm_ops,
-		.of_match_table = seeed_voice_of_match,
-	},
-	.probe = seeed_voice_card_probe,
-	.remove = seeed_voice_card_remove,
-};
-
-module_platform_driver(seeed_voice_card);
-
-MODULE_ALIAS("platform:seeed-voice-card");
-MODULE_LICENSE("GPL v2");
-MODULE_DESCRIPTION("ASoC SEEED Voice Card");
-MODULE_AUTHOR("PeterYang<linsheng.yang@seeed.cc>");
--- a/seeed-voicecard.service
+++ b/seeed-voicecard.service
@ -1,12 +0,0 @@
-[Unit]
-Description=Seeed Voicecard service
-After=alsa-restore.service
- 
-[Service]
-Type=oneshot
-RemainAfterExit=yes
-ExecStart=/usr/bin/seeed-voicecard
-User=root
- 
-[Install]
-WantedBy=sysinit.target
--- a/sound-compatible-4.18.h
+++ b/sound-compatible-4.18.h
@ -1,47 +0,0 @@
-/*
- * (C) Copyright 2017-2018
- * Seeed Technology Co., Ltd. <www.seeedstudio.com>
- *
- * PeterYang <linsheng.yang@seeed.cc>
- */
-#ifndef __SOUND_COMPATIBLE_4_18_H__
-#define __SOUND_COMPATIBLE_4_18_H__
-
-#include <linux/version.h>
-
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(4,17,0)
-#define __NO_SND_SOC_CODEC_DRV     1
-#else
-#define __NO_SND_SOC_CODEC_DRV     0
-#endif
-
-#if LINUX_VERSION_CODE < KERNEL_VERSION(5,4,0)
-#if __has_attribute(__fallthrough__)
-# define fallthrough                    __attribute__((__fallthrough__))
-#else
-# define fallthrough                    do {} while (0)  /* fallthrough */
-#endif
-#endif
-
-#if __NO_SND_SOC_CODEC_DRV
-#define codec                      component
-#define snd_soc_codec              snd_soc_component
-#define snd_soc_codec_driver       snd_soc_component_driver
-#define snd_soc_codec_get_drvdata  snd_soc_component_get_drvdata
-#define snd_soc_codec_get_dapm     snd_soc_component_get_dapm
-#define snd_soc_codec_get_bias_level snd_soc_component_get_bias_level
-#define snd_soc_kcontrol_codec     snd_soc_kcontrol_component
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,9,0)
-#define snd_soc_read               snd_soc_component_read
-#else
-#define snd_soc_read               snd_soc_component_read32
-#endif
-#define snd_soc_register_codec     devm_snd_soc_register_component
-#define snd_soc_unregister_codec   snd_soc_unregister_component
-#define snd_soc_update_bits        snd_soc_component_update_bits
-#define snd_soc_write              snd_soc_component_write
-#define snd_soc_add_codec_controls snd_soc_add_component_controls
-#endif
-
-#endif//__SOUND_COMPATIBLE_4_18_H__
-
--- a/tools/coherence.py
+++ b/tools/coherence.py
@ -1,44 +0,0 @@
-"""
-Estimate the magnitude squared coherence estimate,
-
- requirements
-    sudo apt install python-numpy python-scipy python-matplotlib
-"""
-
-
-import sys
-import wave
-import numpy as np
-from scipy import signal
-import matplotlib.pyplot as plt
-
-if len(sys.argv) < 2:
-    print('Usage: python {} audio.wav'.format(sys.argv[0]))
-    sys.exit(1)
-
-wav = wave.open(sys.argv[1], 'rb')
-channels = wav.getnchannels()
-frames = wav.readframes(wav.getnframes())
-fs = wav.getframerate()
-wav.close()
-
-print("channels: %d" % channels)
-print("rate    : %d" % fs)
-print("frames  : %d" % wav.getnframes())
-
-array = np.fromstring(frames, dtype='int16')
-
-ch0 = array[0::channels]
-
-fig, ax = plt.subplots()
-
-for ch in range(1, channels):
-    f, c = signal.coherence(ch0, array[ch::channels], fs, nperseg=1024)
-    ax.semilogy(f, c, label="CO 1-%d" % (ch + 1))
-
-legend = ax.legend(loc='lower right', shadow=True, fontsize='small')
-
-plt.xlabel('frequency [Hz]')
-plt.ylabel('Coherence')
-plt.show()
-
--- a/tools/phase_test.py
+++ b/tools/phase_test.py
@ -1,83 +0,0 @@
-import sys
-import wave
-import numpy as np
-
-
-if len(sys.argv) != 2:
-    print('Usage: {} multi.wav'.format(sys.argv[0]))
-    sys.exit(1)
-
-
-multi = wave.open(sys.argv[1], 'rb')
-rate = multi.getframerate()
-channels = multi.getnchannels()
-
-if channels <= 1:
-    sys.exit(1)
-
-N = rate 
-
-window = np.hanning(N)
-
-interp = 4*8
-max_offset = int(rate * 0.1 / 340 * interp)
-
-def gcc_phat(sig, refsig, fs=1, max_tau=None, interp=16):
-    '''
-    This function computes the offset between the signal sig and the reference signal refsig
-    using the Generalized Cross Correlation - Phase Transform (GCC-PHAT)method.
-    '''
-    
-    # make sure the length for the FFT is larger or equal than len(sig) + len(refsig)
-    n = sig.shape[0] + refsig.shape[0]
-
-    # Generalized Cross Correlation Phase Transform
-    SIG = np.fft.rfft(sig, n=n)
-    REFSIG = np.fft.rfft(refsig, n=n)
-    R = SIG * np.conj(REFSIG)
-    #R /= np.abs(R)
-
-    cc = np.fft.irfft(R, n=(interp * n))
-
-    max_shift = int(interp * n / 2)
-    if max_tau:
-        max_shift = np.minimum(int(interp * fs * max_tau), max_shift)
-
-    cc = np.concatenate((cc[-max_shift:], cc[:max_shift+1]))
-
-    # find max cross correlation index
-    shift = np.argmax(np.abs(cc)) - max_shift
-
-    tau = shift / float(interp * fs)
-    
-    return tau, cc
-
-
-print(multi.getsampwidth())
-
-while True:
-    data = multi.readframes(N)
-
-    if len(data) != multi.getsampwidth() * N * channels:
-        print("done")
-        break
-
-    if multi.getsampwidth() == 2:
-        data = np.fromstring(data, dtype='int16')
-    else:
-        data = np.fromstring(data, dtype='int32')
-    ref_buf = data[0::channels]
-
-    offsets = []
-    for ch in range(1, channels):
-        sig_buf = data[ch::channels]
-        tau, _ = gcc_phat(sig_buf * window, ref_buf * window, fs=1, max_tau=max_offset, interp=interp)
-        # tau, _ = gcc_phat(sig_buf, ref_buf, fs=rate, max_tau=1)
-
-        offsets.append(tau)
-
-    print(offsets)
-
-print(multi.getframerate())
-
-multi.close()
--- a/ubuntu-prerequisite.sh
+++ b/ubuntu-prerequisite.sh
@ -1,37 +0,0 @@
-#!/bin/bash
-
-# Copyright (c) Hin-Tak Leung 2020
-#
-# Overview:
-#   This script compiles and install the Broadcom VideoCore tools,
-#   configure the dynamic loader for the non-standard library location,
-#   and update the loader cache.
-#
-#   A few steps explicitly requires root privilege, which are
-#   marked with "sudo". The rest is just checking for duplicate/previous
-#   action.
-#
-#   This derived from my command history on ubuntu 20.04.1 .YMMV
-
-sudo apt install -y git gcc g++ make alsa-utils cmake
-
-git clone git://github.com/raspberrypi/userland.git
-pushd userland/
-
-arch=$(uname -m)
-if [[ "$arch" =~ aarch64 ]]; then
-    ./buildme --aarch64
-else
-    ./buildme
-fi
-# ./buildme already includes "sudo make install" at the end
-
-popd
-
-# matches Raspbian's location:
-if [ ! -f /etc/ld.so.conf.d/00-vmcs.conf ] ; then
-    echo "/opt/vc/lib" | sudo tee -a /etc/ld.so.conf.d/00-vmcs.conf
-    sudo ldconfig -v
-else
-    echo "/etc/ld.so.conf.d/00-vmcs.conf exists - no need to update ld.cache!"
-fi
--- a/uninstall.sh
+++ b/uninstall.sh
@ -12,87 +12,71 @@ if [ "x${is_Raspberry}" != "xRaspberry" ] ; then
 fi

 uname_r=$(uname -r)
-
-CONFIG=/boot/config.txt
-[ -f /boot/firmware/config.txt ] && CONFIG=/boot/firmware/config.txt
-[ -f /boot/firmware/usercfg.txt ] && CONFIG=/boot/firmware/usercfg.txt
-
-get_overlay() {
-    ov=$1
-    if grep -q -E "^dtoverlay=$ov" $CONFIG; then
-      echo 0
-    else
-      echo 1
+card=$1
+if [ x${card} = "x" ] ; then
+	echo "Usage: ./uninstall 2mic|4mic"
+	exit 1
 fi
-}

-do_overlay() {
-    ov=$1
-    RET=$2
-    DEFAULT=--defaultno
-    CURRENT=0
-    if [ $(get_overlay $ov) -eq 0 ]; then
-        DEFAULT=
-        CURRENT=1
+if [ x${card} = "x2mic" ] ; then
+	echo "delete dtoverlay=seeed-2mic-voicecard in /boot/config.txt"
+	sed -i "s/dtoverlay=seeed-2mic-voicecard//g" /boot/config.txt
+	if [ -f /boot/overlays/seeed-2mic-voicecard.dtbo ] ; then
+		echo "remove seeed-2mic-voicecard.dtbo in /boot/overlays"
+		rm /boot/overlays/seeed-2mic-voicecard.dtbo
 	fi 
-    if [ $RET -eq $CURRENT ]; then
-        ASK_TO_REBOOT=1
+
+	if [ -f /lib/modules/${uname_r}/kernel/sound/soc/codecs/snd-soc-wm8960.ko ] ; then
+		echo "remove snd-soc-wm8960.ko"
+		rm  /lib/modules/${uname_r}/kernel/sound/soc/codecs/snd-soc-wm8960.ko
 	fi
-    if [ $RET -eq 0 ]; then
-        sed $CONFIG -i -e "s/^#dtoverlay=$ov/dtoverlay=$ov/"
-        if ! grep -q -E "^dtoverlay=$ov" $CONFIG; then
-            printf "dtoverlay=$ov\n" >> $CONFIG
+
+	if [ -d /var/lib/dkms/seeed-voicecard ] ; then
+		echo "remove seeed-voicecard dkms"
+		rm  -rf  /var/lib/dkms/seeed-voicecard
+	fi
+
+	echo "delete snd-soc-wm8960 in /etc/modules"
+	sed -i "s/snd-soc-wm8960//g" /etc/modules
+
+	if [ -f /var/lib/alsa/asound.state ] ; then 
+		echo  "remove wm8960_asound.state"
+		rm /var/lib/alsa/asound.state
 	fi
-        STATUS=enabled
-    elif [ $RET -eq 1 ]; then
-        sed $CONFIG -i -e "s/^dtoverlay=$ov/#dtoverlay=$ov/"
-        STATUS=disabled
-    else
-        return $RET
 fi
-}

-RPI_HATS="seeed-2mic-voicecard seeed-4mic-voicecard seeed-8mic-voicecard"
+if [ x${card} = "x4mic" ] ; then
+        echo "delete dtoverlay=seeed-4mic-voicecard in /boot/config.txt"
+        sed -i "s/dtoverlay=seeed-4mic-voicecard//g" /boot/config.txt
 	
-PATH=$PATH:/opt/vc/bin
-echo "remove dtbos"
-for i in $RPI_HATS; do
-    dtoverlay -r $i
-done
-OVERLAYS=/boot/overlays
-[ -d /boot/firmware/overlays ] && OVERLAYS=/boot/firmware/overlays
+	if [ -f /boot/overlays/seeed-4mic-voicecard.dtbo ] ; then
+        	echo "remove seeed-4mic-voicecard.dtbo in /boot/overlays"
+        	rm /boot/overlays/seeed-4mic-voicecard.dtbo
+	fi

-rm  ${OVERLAYS}/seeed-2mic-voicecard.dtbo || true
-rm  ${OVERLAYS}/seeed-4mic-voicecard.dtbo || true
-rm  ${OVERLAYS}/seeed-8mic-voicecard.dtbo || true
+	if [ -f /lib/modules/${uname_r}/kernel/sound/soc/codecs/snd-soc-ac108.ko ] ; then
+        	echo "remove snd-soc-ac108.ko"
+        	rm  /lib/modules/${uname_r}/kernel/sound/soc/codecs/snd-soc-ac108.ko
+	fi

-echo "remove alsa configs"
-rm -rf  /etc/voicecard/ || true
+	if [ -d /var/lib/dkms/seeed-voicecard ] ; then
+        echo "remove seeed-voicecard dkms"
+        rm  -rf  /var/lib/dkms/seeed-voicecard
+	fi 

-echo "disabled seeed-voicecard.service "
-systemctl stop seeed-voicecard.service
-systemctl disable seeed-voicecard.service
+        echo "delete snd-soc-ac108 in /etc/modules"
+        sed -i "s/snd-soc-ac108//g" /etc/modules                

-echo "remove seeed-voicecard"
-rm  /usr/bin/seeed-voicecard || true
-rm  /lib/systemd/system/seeed-voicecard.service || true
+	if [ -f /var/lib/alsa/asound.state ] ; then
+        	echo  "remove ac108_asound.state"
+        	rm /var/lib/alsa/asound.state
+	fi
 	
-echo "remove dkms"
-rm  -rf /var/lib/dkms/seeed-voicecard || true
-
-echo "remove kernel modules"
-rm  /lib/modules/*/kernel/sound/soc/codecs/snd-soc-wm8960.ko || true
-rm  /lib/modules/*/kernel/sound/soc/codecs/snd-soc-ac108.ko || true
-rm  /lib/modules/*/kernel/sound/soc/bcm/snd-soc-seeed-voicecard.ko || true
-rm  /lib/modules/*/updates/dkms/snd-soc-wm8960.ko || true
-rm  /lib/modules/*/updates/dkms/snd-soc-ac108.ko || true
-rm  /lib/modules/*/updates/dkms/snd-soc-seeed-voicecard.ko || true
-
-echo "remove $CONFIG configuration"
-for i in $RPI_HATS; do
-    echo Uninstall $i ...
-    do_overlay $i 1
-done
+	if [ -f /usr/lib/arm-linux-gnueabihf/alsa-lib/libasound_module_pcm_ac108.so ] ; then
+		echo "remove libasound_module_pcm_ac108.so in /usr/lib/arm-linux-gnueabihf/alsa-lib/ "
+		rm  /usr/lib/arm-linux-gnueabihf/alsa-lib/libasound_module_pcm_ac108.so
+	fi 
+fi

 echo "------------------------------------------------------"
 echo "Please reboot your raspberry pi to apply all settings"
--- a/wm8960.c
+++ b/wm8960.c
@ -25,7 +25,6 @@
 #include <sound/initval.h>
 #include <sound/tlv.h>
 #include <sound/wm8960.h>
-#include "sound-compatible-4.18.h"

 #include "wm8960.h"

@ -227,12 +226,11 @@ static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
 static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
 static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
 static const DECLARE_TLV_DB_SCALE(lineinboost_tlv, -1500, 300, 1);
-static const DECLARE_TLV_DB_RANGE(micboost_tlv,
+static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(micboost_tlv,
 	0, 1, TLV_DB_SCALE_ITEM(0, 1300, 0),
 	2, 3, TLV_DB_SCALE_ITEM(2000, 900, 0),
 );

-
 static const struct snd_kcontrol_new wm8960_snd_controls[] = {
 SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
 		 0, 63, 0, inpga_tlv),
@ -508,11 +506,7 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec)
 	 * list each time to find the desired power state do so now
 	 * and save the result.
 	 */
-	#if __NO_SND_SOC_CODEC_DRV
-	list_for_each_entry(w, &codec->card->widgets, list) {
-	#else
 	list_for_each_entry(w, &codec->component.card->widgets, list) {
-	#endif
 		if (w->dapm != dapm)
 			continue;
 		if (strcmp(w->name, "LOUT1 PGA") == 0)
@ -753,7 +747,6 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
 			iface |= 0x000c;
 			break;
 		}
-		fallthrough;
 	default:
 		dev_err(codec->dev, "unsupported width %d\n",
 			params_width(params));
@ -796,7 +789,7 @@ static int wm8960_hw_free(struct snd_pcm_substream *substream,
 	return 0;
 }

-static int wm8960_mute(struct snd_soc_dai *dai, int mute, int direction)
+static int wm8960_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;

@ -1236,12 +1229,11 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
 static const struct snd_soc_dai_ops wm8960_dai_ops = {
 	.hw_params = wm8960_hw_params,
 	.hw_free = wm8960_hw_free,
-	.mute_stream = wm8960_mute,
+	.digital_mute = wm8960_mute,
 	.set_fmt = wm8960_set_dai_fmt,
 	.set_clkdiv = wm8960_set_dai_clkdiv,
 	.set_pll = wm8960_set_dai_pll,
 	.set_sysclk = wm8960_set_dai_sysclk,
-	.no_capture_mute = 1,
 };

 static struct snd_soc_dai_driver wm8960_dai = {
@ -1259,11 +1251,7 @@ static struct snd_soc_dai_driver wm8960_dai = {
 		.rates = WM8960_RATES,
 		.formats = WM8960_FORMATS,},
 	.ops = &wm8960_dai_ops,
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,12,0)
-	.symmetric_rate = 1,
-#else
 	.symmetric_rates = 1,
-#endif
 };

 static int wm8960_probe(struct snd_soc_codec *codec)
@ -1287,11 +1275,6 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm8960 = {
 	.probe =	wm8960_probe,
 	.set_bias_level = wm8960_set_bias_level,
 	.suspend_bias_off = true,
-	#if __NO_SND_SOC_CODEC_DRV
-	.idle_bias_on		= 1,
-	.use_pmdown_time	= 1,
-	.endianness		= 1,
-	#endif
 };

 static const struct regmap_config wm8960_regmap = {
@ -1318,7 +1301,8 @@ static void wm8960_set_pdata_from_of(struct i2c_client *i2c,
 		pdata->shared_lrclk = true;
 }

-static int wm8960_i2c_probe(struct i2c_client *i2c)
+static int wm8960_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
 {
 	struct wm8960_data *pdata = dev_get_platdata(&i2c->dev);
 	struct wm8960_priv *wm8960;
@ -1383,9 +1367,10 @@ static int wm8960_i2c_probe(struct i2c_client *i2c)
 	return ret;
 }

-static void wm8960_i2c_remove(struct i2c_client *client)
+static int wm8960_i2c_remove(struct i2c_client *client)
 {
 	snd_soc_unregister_codec(&client->dev);
+	return 0;
 }

 static const struct i2c_device_id wm8960_i2c_id[] = {