[PATCH] fix channel ordering

Fix /boot/config.txt for bookworm installs

LICENSE

@@ -0,0 +1,674 @@
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+                       Version 3, 29 June 2007
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+  IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING
+WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MODIFIES AND/OR CONVEYS
+THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES, INCLUDING ANY
+GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING OUT OF THE
+USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED TO LOSS OF
+DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD
+PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER PROGRAMS),
+EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF
+SUCH DAMAGES.
+
+  17. Interpretation of Sections 15 and 16.
+
+  If the disclaimer of warranty and limitation of liability provided
+above cannot be given local legal effect according to their terms,
+reviewing courts shall apply local law that most closely approximates
+an absolute waiver of all civil liability in connection with the
+Program, unless a warranty or assumption of liability accompanies a
+copy of the Program in return for a fee.
+
+                     END OF TERMS AND CONDITIONS
+
+            How to Apply These Terms to Your New Programs
+
+  If you develop a new program, and you want it to be of the greatest
+possible use to the public, the best way to achieve this is to make it
+free software which everyone can redistribute and change under these terms.
+
+  To do so, attach the following notices to the program.  It is safest
+to attach them to the start of each source file to most effectively
+state the exclusion of warranty; and each file should have at least
+the "copyright" line and a pointer to where the full notice is found.
+
+    {one line to give the program's name and a brief idea of what it does.}
+    Copyright (C) {year}  {name of author}
+
+    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 3 of the License, or
+    (at your option) any later version.
+
+    This program is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+    GNU General Public License for more details.
+
+    You should have received a copy of the GNU General Public License
+    along with this program.  If not, see <http://www.gnu.org/licenses/>.
+
+Also add information on how to contact you by electronic and paper mail.
+
+  If the program does terminal interaction, make it output a short
+notice like this when it starts in an interactive mode:
+
+    {project}  Copyright (C) {year}  {fullname}
+    This program comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
+    This is free software, and you are welcome to redistribute it
+    under certain conditions; type `show c' for details.
+
+The hypothetical commands `show w' and `show c' should show the appropriate
+parts of the General Public License.  Of course, your program's commands
+might be different; for a GUI interface, you would use an "about box".
+
+  You should also get your employer (if you work as a programmer) or school,
+if any, to sign a "copyright disclaimer" for the program, if necessary.
+For more information on this, and how to apply and follow the GNU GPL, see
+<http://www.gnu.org/licenses/>.
+
+  The GNU General Public License does not permit incorporating your program
+into proprietary programs.  If your program is a subroutine library, you
+may consider it more useful to permit linking proprietary applications with
+the library.  If this is what you want to do, use the GNU Lesser General
+Public License instead of this License.  But first, please read
+<http://www.gnu.org/philosophy/why-not-lgpl.html>.

Makefile

@@ -1,11 +1,52 @@
-obj-m := wm8960.o
+#
+# 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
+
+
+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/$(shell uname -r)/build M=$(PWD) modules
+	make -C /lib/modules/$(uname_r)/build M=$(PWD) modules
 
 clean:
-	make -C /lib/modules/$(shell uname -r)/build M=$(PWD) clean
+	make -C /lib/modules/$(uname_r)/build M=$(PWD) clean
 
 install:
-	sudo cp wm8960.ko /lib/modules/$(shell uname -r)
+	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 depmod -a
+
+
+.PHONY: all clean install
+
+endif
+

README.md

@@ -1,59 +1,64 @@
 # seeed-voicecard
 
-[![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)
+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.
 
-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.
-```
-git clone https://github.com/respeaker/seeed-voicecard
+### Install seeed-voicecard
+Get the seeed voice card source code and install all linux kernel drivers
+```bash
+git clone https://github.com/HinTak/seeed-voicecard
 cd seeed-voicecard
 sudo ./install.sh
-reboot
+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.
+
+```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
 ```
 
-Check that the sound card name matches the source code seeed-voicecard.
+### uninstall seeed-voicecard
+If you want to upgrade the driver , you need uninstall the driver first.
 
 ```
-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: seeedvoicecard [seeed-voicecard], device 0: bcm2835-i2s-wm8960-hifi wm8960-hifi-0 []
-  Subdevices: 1/1
-  Subdevice #0: subdevice #0
-pi@raspberrypi:~/seeed-voicecard$ 
+pi@raspberrypi:~/seeed-voicecard $ sudo ./uninstall.sh 
+...
+------------------------------------------------------
+Please reboot your raspberry pi to apply all settings
+Thank you!
+------------------------------------------------------
 ```
-Next apply the alsa controls setting
-```
-sudo alsactl --file=asound.state restore
-```
-If you want to change the alsa settings, You can use `sudo alsactl --file=asound.state store` to save it.
-
-Test:
-``` 
-arecord -f cd -Dhw:1 | aplay -Dhw:1
-```
-
-### with Google Assistant
-if you run the assistant but the playback is speed up considerably, try to configure alsa:
-
-```
-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). 
+
+

ac101_regs.h

@@ -0,0 +1,432 @@
+/*
+ * 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__

ac108.h

@@ -0,0 +1,774 @@
+/*
+ * ac108.h --  ac108 ALSA Soc Audio driver
+ *
+ * Author: panjunwen
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef _AC108_H
+#define _AC108_H
+
+
+/*** AC108 Codec Register Define***/
+
+//Chip Reset
+#define CHIP_RST			0x00
+#define CHIP_RST_VAL		0x12
+
+//Power Control
+#define PWR_CTRL1			0x01
+#define PWR_CTRL2			0x02
+#define PWR_CTRL3			0x03
+#define PWR_CTRL4			0x04
+#define PWR_CTRL5			0x05
+#define PWR_CTRL6			0x06
+#define PWR_CTRL7			0x07
+#define PWR_CTRL8			0x08
+#define PWR_CTRL9			0x09
+
+//PLL Configure Control
+#define PLL_CTRL1			0x10
+#define PLL_CTRL2			0x11
+#define PLL_CTRL3			0x12
+#define PLL_CTRL4			0x13
+#define PLL_CTRL5			0x14
+#define PLL_CTRL6			0x16
+#define PLL_CTRL7			0x17
+#define PLL_LOCK_CTRL		0x18
+
+//System Clock Control
+#define SYSCLK_CTRL			0x20
+#define MOD_CLK_EN			0x21
+#define MOD_RST_CTRL		0x22
+#define DSM_CLK_CTRL		0x25
+
+//I2S Common Control
+#define I2S_CTRL			0x30
+#define I2S_BCLK_CTRL		0x31
+#define I2S_LRCK_CTRL1		0x32
+#define I2S_LRCK_CTRL2		0x33
+#define I2S_FMT_CTRL1		0x34
+#define I2S_FMT_CTRL2		0x35
+#define I2S_FMT_CTRL3		0x36
+
+//I2S TX1 Control
+#define I2S_TX1_CTRL1		0x38
+#define I2S_TX1_CTRL2		0x39
+#define I2S_TX1_CTRL3		0x3A
+#define I2S_TX1_CHMP_CTRL1	0x3C
+#define I2S_TX1_CHMP_CTRL2	0x3D
+#define I2S_TX1_CHMP_CTRL3	0x3E
+#define I2S_TX1_CHMP_CTRL4	0x3F
+
+//I2S TX2 Control
+#define I2S_TX2_CTRL1		0x40
+#define I2S_TX2_CTRL2		0x41
+#define I2S_TX2_CTRL3		0x42
+#define I2S_TX2_CHMP_CTRL1	0x44
+#define I2S_TX2_CHMP_CTRL2	0x45
+#define I2S_TX2_CHMP_CTRL3	0x46
+#define I2S_TX2_CHMP_CTRL4	0x47
+
+//I2S RX1 Control
+#define I2S_RX1_CTRL1		0x50
+#define I2S_RX1_CHMP_CTRL1	0x54
+#define I2S_RX1_CHMP_CTRL2	0x55
+#define I2S_RX1_CHMP_CTRL3	0x56
+#define I2S_RX1_CHMP_CTRL4	0x57
+
+//I2S Loopback Debug
+#define I2S_LPB_DEBUG		0x58
+
+//ADC Common Control
+#define ADC_SPRC			0x60
+#define ADC_DIG_EN			0x61
+#define DMIC_EN				0x62
+#define ADC_DSR				0x63
+#define ADC_FIR				0x64
+#define ADC_DDT_CTRL		0x65
+
+//HPF Control
+#define HPF_EN				0x66
+#define HPF_COEF_REGH1		0x67
+#define HPF_COEF_REGH2		0x68
+#define HPF_COEF_REGL1		0x69
+#define HPF_COEF_REGL2		0x6A
+#define HPF_GAIN_REGH1		0x6B
+#define HPF_GAIN_REGH2		0x6C
+#define HPF_GAIN_REGL1		0x6D
+#define HPF_GAIN_REGL2		0x6E
+
+//ADC Digital Channel Volume Control
+#define ADC1_DVOL_CTRL		0x70
+#define ADC2_DVOL_CTRL		0x71
+#define ADC3_DVOL_CTRL		0x72
+#define ADC4_DVOL_CTRL		0x73
+
+//ADC Digital Mixer Source and Gain Control
+#define ADC1_DMIX_SRC		0x76
+#define ADC2_DMIX_SRC		0x77
+#define ADC3_DMIX_SRC		0x78
+#define ADC4_DMIX_SRC		0x79
+
+//ADC Digital Debug Control
+#define ADC_DIG_DEBUG		0x7F
+
+//I2S Pad Drive Control
+#define I2S_DAT_PADDRV_CTRL	0x80
+#define I2S_CLK_PADDRV_CTRL	0x81
+
+//Analog PGA Control
+#define ANA_PGA1_CTRL		0x90
+#define ANA_PGA2_CTRL		0x91
+#define ANA_PGA3_CTRL		0x92
+#define ANA_PGA4_CTRL		0x93
+
+//MIC Offset Control
+#define MIC_OFFSET_CTRL1	0x96
+#define MIC_OFFSET_CTRL2	0x97
+#define MIC1_OFFSET_STATU1	0x98
+#define MIC1_OFFSET_STATU2	0x99
+#define MIC2_OFFSET_STATU1	0x9A
+#define MIC2_OFFSET_STATU2	0x9B
+#define MIC3_OFFSET_STATU1	0x9C
+#define MIC3_OFFSET_STATU2	0x9D
+#define MIC4_OFFSET_STATU1	0x9E
+#define MIC4_OFFSET_STATU2	0x9F
+
+//ADC1 Analog Control
+#define ANA_ADC1_CTRL1		0xA0
+#define ANA_ADC1_CTRL2		0xA1
+#define ANA_ADC1_CTRL3		0xA2
+#define ANA_ADC1_CTRL4		0xA3
+#define ANA_ADC1_CTRL5		0xA4
+#define ANA_ADC1_CTRL6		0xA5
+#define ANA_ADC1_CTRL7		0xA6
+
+//ADC2 Analog Control
+#define ANA_ADC2_CTRL1		0xA7
+#define ANA_ADC2_CTRL2		0xA8
+#define ANA_ADC2_CTRL3		0xA9
+#define ANA_ADC2_CTRL4		0xAA
+#define ANA_ADC2_CTRL5		0xAB
+#define ANA_ADC2_CTRL6		0xAC
+#define ANA_ADC2_CTRL7		0xAD
+
+//ADC3 Analog Control
+#define ANA_ADC3_CTRL1		0xAE
+#define ANA_ADC3_CTRL2		0xAF
+#define ANA_ADC3_CTRL3		0xB0
+#define ANA_ADC3_CTRL4		0xB1
+#define ANA_ADC3_CTRL5		0xB2
+#define ANA_ADC3_CTRL6		0xB3
+#define ANA_ADC3_CTRL7		0xB4
+
+//ADC4 Analog Control
+#define ANA_ADC4_CTRL1		0xB5
+#define ANA_ADC4_CTRL2		0xB6
+#define ANA_ADC4_CTRL3		0xB7
+#define ANA_ADC4_CTRL4		0xB8
+#define ANA_ADC4_CTRL5		0xB9
+#define ANA_ADC4_CTRL6		0xBA
+#define ANA_ADC4_CTRL7		0xBB
+
+//GPIO Configure
+#define GPIO_CFG1			0xC0
+#define GPIO_CFG2			0xC1
+#define GPIO_DAT			0xC2
+#define GPIO_DRV			0xC3
+#define GPIO_PULL			0xC4
+#define GPIO_INT_CFG		0xC5
+#define GPIO_INT_EN			0xC6
+#define GPIO_INT_STATUS		0xC7
+
+//Misc
+#define BGTC_DAT			0xD1
+#define BGVC_DAT			0xD2
+#define PRNG_CLK_CTRL		0xDF
+
+
+
+/*** AC108 Codec Register Bit Define***/
+
+/*PWR_CTRL1*/
+#define CP12_CTRL				4
+#define CP12_SENSE_SELECT		3
+
+/*PWR_CTRL2*/
+#define CP12_SENSE_FILT			6
+#define CP12_COMP_FF_EN			3
+#define CP12_FORCE_ENABLE		2
+#define CP12_FORCE_RSTB			1
+
+/*PWR_CTRL3*/
+#define LDO33DIG_CTRL			0
+
+/*PWR_CTRL6*/
+#define LDO33ANA_2XHDRM			2
+#define LDO33ANA_ENABLE			0
+
+/*PWR_CTRL7*/
+#define VREF_SEL				3
+#define VREF_FASTSTART_ENABLE	1
+#define VREF_ENABLE				0
+
+/*PWR_CTRL9*/
+#define VREFP_FASTSTART_ENABLE	7
+#define VREFP_RESCTRL			5
+#define VREFP_LPMODE			4
+#define IGEN_TRIM				1
+#define VREFP_ENABLE			0
+
+
+/*PLL_CTRL1*/
+#define PLL_IBIAS				4
+#define PLL_NDET				3
+#define PLL_LOCKED_STATUS		2
+#define PLL_COM_EN				1
+#define PLL_EN					0
+
+/*PLL_CTRL2*/
+#define PLL_PREDIV2				5
+#define PLL_PREDIV1				0
+
+/*PLL_CTRL3*/
+#define PLL_LOOPDIV_MSB			0
+
+/*PLL_CTRL4*/
+#define PLL_LOOPDIV_LSB			0
+
+/*PLL_CTRL5*/
+#define PLL_POSTDIV2			5
+#define PLL_POSTDIV1			0
+
+/*PLL_CTRL6*/
+#define PLL_LDO					6
+#define PLL_CP					0
+
+/*PLL_CTRL7*/
+#define PLL_CAP					6
+#define PLL_RES					4
+#define PLL_TEST_EN				0
+
+/*PLL_LOCK_CTRL*/
+#define LOCK_LEVEL1				2
+#define LOCK_LEVEL2				1
+#define PLL_LOCK_EN				0
+
+
+/*SYSCLK_CTRL*/
+#define PLLCLK_EN				7
+#define PLLCLK_SRC				4
+#define SYSCLK_SRC				3
+#define SYSCLK_EN				0
+
+/*MOD_CLK_EN & MOD_RST_CTRL*/
+#define I2S						7
+#define ADC_DIGITAL				4
+#define MIC_OFFSET_CALIBRATION	1
+#define ADC_ANALOG				0
+
+/*DSM_CLK_CTRL*/
+#define MIC_OFFSET_DIV			4
+#define DSM_CLK_SEL				0
+
+
+/*I2S_CTRL*/
+#define BCLK_IOEN				7
+#define LRCK_IOEN				6
+#define SDO2_EN					5
+#define SDO1_EN					4
+#define TXEN					2
+#define RXEN					1
+#define GEN						0
+
+/*I2S_BCLK_CTRL*/
+#define EDGE_TRANSFER			5
+#define BCLK_POLARITY			4
+#define BCLKDIV					0
+
+/*I2S_LRCK_CTRL1*/
+#define LRCK_POLARITY			4
+#define LRCK_PERIODH			0
+
+/*I2S_LRCK_CTRL2*/
+#define LRCK_PERIODL			0
+
+/*I2S_FMT_CTRL1*/
+#define ENCD_SEL				6
+#define MODE_SEL				4
+#define TX2_OFFSET				3
+#define TX1_OFFSET				2
+#define TX_SLOT_HIZ				1
+#define TX_STATE				0
+
+/*I2S_FMT_CTRL2*/
+#define SLOT_WIDTH_SEL			4
+#define SAMPLE_RESOLUTION		0
+
+/*I2S_FMT_CTRL3*/
+#define TX_MLS					7
+#define SEXT					5
+#define OUT2_MUTE				4
+#define OUT1_MUTE				3
+#define LRCK_WIDTH				2
+#define TX_PDM					0
+
+
+/*I2S_TX1_CTRL1*/
+#define TX1_CHSEL				0
+
+/*I2S_TX1_CTRL2*/
+#define TX1_CH8_EN				7
+#define TX1_CH7_EN				6
+#define TX1_CH6_EN				5
+#define TX1_CH5_EN				4
+#define TX1_CH4_EN				3
+#define TX1_CH3_EN				2
+#define TX1_CH2_EN				1
+#define TX1_CH1_EN				0
+
+/*I2S_TX1_CTRL3*/
+#define TX1_CH16_EN				7
+#define TX1_CH15_EN				6
+#define TX1_CH14_EN				5
+#define TX1_CH13_EN				4
+#define TX1_CH12_EN				3
+#define TX1_CH11_EN				2
+#define TX1_CH10_EN				1
+#define TX1_CH9_EN				0
+
+/*I2S_TX1_CHMP_CTRL1*/
+#define TX1_CH4_MAP				6
+#define TX1_CH3_MAP				4
+#define TX1_CH2_MAP				2
+#define TX1_CH1_MAP				0
+
+/*I2S_TX1_CHMP_CTRL2*/
+#define TX1_CH8_MAP				6
+#define TX1_CH7_MAP				4
+#define TX1_CH6_MAP				2
+#define TX1_CH5_MAP				0
+
+/*I2S_TX1_CHMP_CTRL3*/
+#define TX1_CH12_MAP			6
+#define TX1_CH11_MAP			4
+#define TX1_CH10_MAP			2
+#define TX1_CH9_MAP				0
+
+/*I2S_TX1_CHMP_CTRL4*/
+#define TX1_CH16_MAP			6
+#define TX1_CH15_MAP			4
+#define TX1_CH14_MAP			2
+#define TX1_CH13_MAP			0
+
+
+/*I2S_TX2_CTRL1*/
+#define TX2_CHSEL				0
+
+/*I2S_TX2_CHMP_CTRL1*/
+#define TX2_CH4_MAP				6
+#define TX2_CH3_MAP				4
+#define TX2_CH2_MAP				2
+#define TX2_CH1_MAP				0
+
+/*I2S_TX2_CHMP_CTRL2*/
+#define TX2_CH8_MAP				6
+#define TX2_CH7_MAP				4
+#define TX2_CH6_MAP				2
+#define TX2_CH5_MAP				0
+
+/*I2S_TX2_CHMP_CTRL3*/
+#define TX2_CH12_MAP			6
+#define TX2_CH11_MAP			4
+#define TX2_CH10_MAP			2
+#define TX2_CH9_MAP				0
+
+/*I2S_TX2_CHMP_CTRL4*/
+#define TX2_CH16_MAP			6
+#define TX2_CH15_MAP			4
+#define TX2_CH14_MAP			2
+#define TX2_CH13_MAP			0
+
+
+/*I2S_RX1_CTRL1*/
+#define RX1_CHSEL				0
+
+/*I2S_RX1_CHMP_CTRL1*/
+#define RX1_CH4_MAP				6
+#define RX1_CH3_MAP				4
+#define RX1_CH2_MAP				2
+#define RX1_CH1_MAP				0
+
+/*I2S_RX1_CHMP_CTRL2*/
+#define RX1_CH8_MAP				6
+#define RX1_CH7_MAP				4
+#define RX1_CH6_MAP				2
+#define RX1_CH5_MAP				0
+
+/*I2S_RX1_CHMP_CTRL3*/
+#define RX1_CH12_MAP			6
+#define RX1_CH11_MAP			4
+#define RX1_CH10_MAP			2
+#define RX1_CH9_MAP				0
+
+/*I2S_RX1_CHMP_CTRL4*/
+#define RX1_CH16_MAP			6
+#define RX1_CH15_MAP			4
+#define RX1_CH14_MAP			2
+#define RX1_CH13_MAP			0
+
+
+/*I2S_LPB_DEBUG*/
+#define I2S_LPB_DEBUG_EN		0
+
+
+/*ADC_SPRC*/
+#define ADC_FS_I2S1				0
+
+/*ADC_DIG_EN*/
+#define DG_EN					4
+#define ENAD4					3
+#define ENAD3					2
+#define ENAD2					1
+#define ENAD1					0
+
+/*DMIC_EN*/
+#define DMIC2_EN				1
+#define DMIC1_EN				0
+
+/*ADC_DSR*/
+#define DIG_ADC4_SRS			6
+#define DIG_ADC3_SRS			4
+#define DIG_ADC2_SRS			2
+#define DIG_ADC1_SRS			0
+
+/*ADC_DDT_CTRL*/
+#define ADOUT_DLY_EN			2
+#define ADOUT_DTS				0
+
+
+/*HPF_EN*/
+#define DIG_ADC4_HPF_EN			3
+#define DIG_ADC3_HPF_EN			2
+#define DIG_ADC2_HPF_EN			1
+#define DIG_ADC1_HPF_EN			0
+
+
+/*ADC1_DMIX_SRC*/
+#define ADC1_ADC4_DMXL_GC		7
+#define ADC1_ADC3_DMXL_GC		6
+#define ADC1_ADC2_DMXL_GC		5
+#define ADC1_ADC1_DMXL_GC		4
+#define ADC1_ADC4_DMXL_SRC		3
+#define ADC1_ADC3_DMXL_SRC		2
+#define ADC1_ADC2_DMXL_SRC		1
+#define ADC1_ADC1_DMXL_SRC		0
+
+/*ADC2_DMIX_SRC*/
+#define ADC2_ADC4_DMXL_GC		7
+#define ADC2_ADC3_DMXL_GC		6
+#define ADC2_ADC2_DMXL_GC		5
+#define ADC2_ADC1_DMXL_GC		4
+#define ADC2_ADC4_DMXL_SRC		3
+#define ADC2_ADC3_DMXL_SRC		2
+#define ADC2_ADC2_DMXL_SRC		1
+#define ADC2_ADC1_DMXL_SRC		0
+
+/*ADC3_DMIX_SRC*/
+#define ADC3_ADC4_DMXL_GC		7
+#define ADC3_ADC3_DMXL_GC		6
+#define ADC3_ADC2_DMXL_GC		5
+#define ADC3_ADC1_DMXL_GC		4
+#define ADC3_ADC4_DMXL_SRC		3
+#define ADC3_ADC3_DMXL_SRC		2
+#define ADC3_ADC2_DMXL_SRC		1
+#define ADC3_ADC1_DMXL_SRC		0
+
+/*ADC4_DMIX_SRC*/
+#define ADC4_ADC4_DMXL_GC		7
+#define ADC4_ADC3_DMXL_GC		6
+#define ADC4_ADC2_DMXL_GC		5
+#define ADC4_ADC1_DMXL_GC		4
+#define ADC4_ADC4_DMXL_SRC		3
+#define ADC4_ADC3_DMXL_SRC		2
+#define ADC4_ADC2_DMXL_SRC		1
+#define ADC4_ADC1_DMXL_SRC		0
+
+
+/*ADC_DIG_DEBUG*/
+#define ADC_PTN_SEL				0
+
+
+/*I2S_DAT_PADDRV_CTRL*/
+#define TX2_DAT_DRV				4
+#define TX1_DAT_DRV				0
+
+/*I2S_CLK_PADDRV_CTRL*/
+#define LRCK_DRV				4
+#define BCLK_DRV				0
+
+
+/*ANA_PGA1_CTRL*/
+#define ADC1_ANALOG_PGA			1
+#define ADC1_ANALOG_PGA_STEP	0
+
+/*ANA_PGA2_CTRL*/
+#define ADC2_ANALOG_PGA			1
+#define ADC2_ANALOG_PGA_STEP	0
+
+/*ANA_PGA3_CTRL*/
+#define ADC3_ANALOG_PGA			1
+#define ADC3_ANALOG_PGA_STEP	0
+
+/*ANA_PGA4_CTRL*/
+#define ADC4_ANALOG_PGA			1
+#define ADC4_ANALOG_PGA_STEP	0
+
+
+/*MIC_OFFSET_CTRL1*/
+#define MIC_OFFSET_CAL_EN4		3
+#define MIC_OFFSET_CAL_EN3		2
+#define MIC_OFFSET_CAL_EN2		1
+#define MIC_OFFSET_CAL_EN1		0
+
+/*MIC_OFFSET_CTRL2*/
+#define MIC_OFFSET_CAL_GAIN		3
+#define MIC_OFFSET_CAL_CHANNEL	1
+#define MIC_OFFSET_CAL_EN_ONCE	0
+
+/*MIC1_OFFSET_STATU1*/
+#define MIC1_OFFSET_CAL_DONE	7
+#define MIC1_OFFSET_CAL_RUN_STA	6
+#define MIC1_OFFSET_MSB			0
+
+/*MIC1_OFFSET_STATU2*/
+#define MIC1_OFFSET_LSB			0
+
+/*MIC2_OFFSET_STATU1*/
+#define MIC2_OFFSET_CAL_DONE	7
+#define MIC2_OFFSET_CAL_RUN_STA	6
+#define MIC2_OFFSET_MSB			0
+
+/*MIC2_OFFSET_STATU2*/
+#define MIC2_OFFSET_LSB			0
+
+/*MIC3_OFFSET_STATU1*/
+#define MIC3_OFFSET_CAL_DONE	7
+#define MIC3_OFFSET_CAL_RUN_STA	6
+#define MIC3_OFFSET_MSB			0
+
+/*MIC3_OFFSET_STATU2*/
+#define MIC3_OFFSET_LSB			0
+
+/*MIC4_OFFSET_STATU1*/
+#define MIC4_OFFSET_CAL_DONE	7
+#define MIC4_OFFSET_CAL_RUN_STA	6
+#define MIC4_OFFSET_MSB			0
+
+/*MIC4_OFFSET_STATU2*/
+#define MIC4_OFFSET_LSB			0
+
+
+/*ANA_ADC1_CTRL1*/
+#define ADC1_PGA_BYPASS			7
+#define ADC1_PGA_BYP_RCM		6
+#define ADC1_PGA_CTRL_RCM		4
+#define ADC1_PGA_MUTE			3
+#define ADC1_DSM_ENABLE			2
+#define ADC1_PGA_ENABLE			1
+#define ADC1_MICBIAS_EN			0
+
+/*ANA_ADC1_CTRL3*/
+#define ADC1_ANA_CAL_EN			5
+#define ADC1_SEL_OUT_EDGE		3
+#define ADC1_DSM_DISABLE		2
+#define ADC1_VREFP_DISABLE		1
+#define ADC1_AAF_DISABLE		0
+
+/*ANA_ADC1_CTRL6*/
+#define PGA_CTRL_TC				6
+#define PGA_CTRL_RC				4
+#define PGA_CTRL_I_LIN			2
+#define PGA_CTRL_I_IN			0
+
+/*ANA_ADC1_CTRL7*/
+#define PGA_CTRL_HI_Z			7
+#define PGA_CTRL_SHORT_RF		6
+#define PGA_CTRL_VCM_VG			4
+#define PGA_CTRL_VCM_IN			0
+
+
+/*ANA_ADC2_CTRL1*/
+#define ADC2_PGA_BYPASS			7
+#define ADC2_PGA_BYP_RCM		6
+#define ADC2_PGA_CTRL_RCM		4
+#define ADC2_PGA_MUTE			3
+#define ADC2_DSM_ENABLE			2
+#define ADC2_PGA_ENABLE			1
+#define ADC2_MICBIAS_EN			0
+
+/*ANA_ADC2_CTRL3*/
+#define ADC2_ANA_CAL_EN			5
+#define ADC2_SEL_OUT_EDGE		3
+#define ADC2_DSM_DISABLE		2
+#define ADC2_VREFP_DISABLE		1
+#define ADC2_AAF_DISABLE		0
+
+/*ANA_ADC2_CTRL6*/
+#define PGA_CTRL_IBOOST			7
+#define PGA_CTRL_IQCTRL			6
+#define PGA_CTRL_OABIAS			4
+#define PGA_CTRL_CMLP_DIS		3
+#define PGA_CTRL_PDB_RIN		2
+#define PGA_CTRL_PEAKDET		0
+
+/*ANA_ADC2_CTRL7*/
+#define AAF_LPMODE_EN			7
+#define AAF_STG2_IB_SEL			4
+#define AAFDSM_IB_DIV2			3
+#define AAF_STG1_IB_SEL			0
+
+
+/*ANA_ADC3_CTRL1*/
+#define ADC3_PGA_BYPASS			7
+#define ADC3_PGA_BYP_RCM		6
+#define ADC3_PGA_CTRL_RCM		4
+#define ADC3_PGA_MUTE			3
+#define ADC3_DSM_ENABLE			2
+#define ADC3_PGA_ENABLE			1
+#define ADC3_MICBIAS_EN			0
+
+/*ANA_ADC3_CTRL3*/
+#define ADC3_ANA_CAL_EN			5
+#define ADC3_INVERT_CLK			4
+#define ADC3_SEL_OUT_EDGE		3
+#define ADC3_DSM_DISABLE		2
+#define ADC3_VREFP_DISABLE		1
+#define ADC3_AAF_DISABLE		0
+
+/*ANA_ADC3_CTRL7*/
+#define DSM_COMP_IB_SEL			6
+#define DSM_OTA_CTRL			4
+#define DSM_LPMODE				3
+#define DSM_OTA_IB_SEL			0
+
+
+/*ANA_ADC4_CTRL1*/
+#define ADC4_PGA_BYPASS			7
+#define ADC4_PGA_BYP_RCM		6
+#define ADC4_PGA_CTRL_RCM		4
+#define ADC4_PGA_MUTE			3
+#define ADC4_DSM_ENABLE			2
+#define ADC4_PGA_ENABLE			1
+#define ADC4_MICBIAS_EN			0
+
+/*ANA_ADC4_CTRL3*/
+#define ADC4_ANA_CAL_EN			5
+#define ADC4_SEL_OUT_EDGE		3
+#define ADC4_DSM_DISABLE		2
+#define ADC4_VREFP_DISABLE		1
+#define ADC4_AAF_DISABLE		0
+
+/*ANA_ADC4_CTRL6*/
+#define DSM_DEMOFF				5
+#define DSM_EN_DITHER			4
+#define DSM_VREFP_LPMODE		2
+#define DSM_VREFP_OUTCTRL		0
+
+/*ANA_ADC4_CTRL7*/
+#define CK8M_EN					5
+#define OSC_EN					4
+#define ADC4_CLK_GATING			3
+#define ADC3_CLK_GATING			2
+#define ADC2_CLK_GATING			1
+#define ADC1_CLK_GATING			0
+
+
+/*GPIO_CFG1*/
+#define GPIO2_SELECT			4
+#define GPIO1_SELECT			0
+
+/*GPIO_CFG2*/
+#define GPIO4_SELECT			4
+#define GPIO3_SELECT			0
+
+/*GPIO_DAT*///order???
+#define GPIO4_DAT				3
+#define GPIO3_DAT				2
+#define GPIO2_DAT				1
+#define GPIO1_DAT				0
+
+/*GPIO_DRV*/
+#define GPIO4_DRV				6
+#define GPIO3_DRV				4
+#define GPIO2_DRV				2
+#define GPIO1_DRV				0
+
+/*GPIO_PULL*/
+#define GPIO4_PULL				6
+#define GPIO3_PULL				4
+#define GPIO2_PULL				2
+#define GPIO1_PULL				0
+
+/*GPIO_INT_CFG*/
+#define GPIO4_EINT_CFG			6
+#define GPIO3_EINT_CFG			4
+#define GPIO2_EINT_CFG			2
+#define GPIO1_EINT_CFG			0
+
+/*GPIO_INT_EN*///order???
+#define GPIO4_EINT_EN			3
+#define GPIO3_EINT_EN			2
+#define GPIO2_EINT_EN			1
+#define GPIO1_EINT_EN			0
+
+/*GPIO_INT_STATUS*///order???
+#define GPIO4_EINT_STA			3
+#define GPIO3_EINT_STA			2
+#define GPIO2_EINT_STA			1
+#define GPIO1_EINT_STA			0
+
+
+/*PRNG_CLK_CTRL*/
+#define PRNG_CLK_EN				1
+#define PRNG_CLK_POS			0
+
+
+
+/*** Some Config Value ***/
+
+//[SYSCLK_CTRL]: PLLCLK_SRC
+#define PLLCLK_SRC_MCLK			0
+#define PLLCLK_SRC_BCLK			1
+#define PLLCLK_SRC_GPIO2		2
+#define PLLCLK_SRC_GPIO3		3
+
+//[SYSCLK_CTRL]: SYSCLK_SRC
+#define SYSCLK_SRC_MCLK			0
+#define SYSCLK_SRC_PLL			1
+
+//I2S BCLK POLARITY Control
+#define BCLK_NORMAL_DRIVE_N_SAMPLE_P	0
+#define BCLK_INVERT_DRIVE_P_SAMPLE_N	1
+
+//I2S LRCK POLARITY Control
+#define	LRCK_LEFT_LOW_RIGHT_HIGH		0
+#define LRCK_LEFT_HIGH_RIGHT_LOW		1
+
+//I2S Format Selection
+#define PCM_FORMAT						0
+#define LEFT_JUSTIFIED_FORMAT			1
+#define RIGHT_JUSTIFIED_FORMAT			2
+
+
+//I2S data protocol types
+
+#define IS_ENCODING_MODE		 0
+
+#endif
+

ac108_6mic.state

@@ -0,0 +1,367 @@
+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
+		}
+	}
+}

ac108_asound.state

@@ -0,0 +1,181 @@
+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 0
+		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.seeed4micvoicec {
+	control.1 {
+		iface MIXER
+		name 'CH1 digital volume'
+		value 222
+		comment {
+			access 'read write'
+			type INTEGER
+			count 1
+			range '0 - 255'
+			dbmin -11925
+			dbmax 7200
+			dbvalue.0 4725
+		}
+	}
+	control.2 {
+		iface MIXER
+		name 'CH2 digital volume'
+		value 222
+		comment {
+			access 'read write'
+			type INTEGER
+			count 1
+			range '0 - 255'
+			dbmin -11925
+			dbmax 7200
+			dbvalue.0 4725
+		}
+	}
+	control.3 {
+		iface MIXER
+		name 'CH3 digital volume'
+		value 222
+		comment {
+			access 'read write'
+			type INTEGER
+			count 1
+			range '0 - 255'
+			dbmin -11925
+			dbmax 7200
+			dbvalue.0 4725
+		}
+	}
+	control.4 {
+		iface MIXER
+		name 'CH4 digital volume'
+		value 222
+		comment {
+			access 'read write'
+			type INTEGER
+			count 1
+			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
+		}
+	}
+	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
+		}
+	}
+}

ac108_plugin/Makefile

@@ -0,0 +1,58 @@
+
+# Quiet (set to @ for a quite compile)
+Q	?= @
+#Q	?=
+
+# Build Tools
+CC 	:= gcc
+CFLAGS += -I. -Wall -funroll-loops -ffast-math -fPIC -DPIC -O0 -g
+LD := gcc
+LDFLAGS += -Wall -shared -lasound
+
+SND_PCM_OBJECTS = pcm_ac108.o ac108_help.o
+SND_PCM_LIBS =
+SND_PCM_BIN = libasound_module_pcm_ac108.so
+
+#SND_CTL_OBJECTS = ctl_ac108.o ladspa_utils.o
+#SND_CTL_LIBS =
+#SND_CTL_BIN = libasound_module_ctl_ac108.so
+
+MULTIARCH:=$(shell gcc --print-multiarch)
+LIBDIR = lib/$(MULTIARCH)
+
+.PHONY: all clean dep load_default
+
+all: Makefile $(SND_PCM_BIN) $(SND_CTL_BIN)
+
+dep:
+	@echo DEP $@
+	$(Q)for i in *.c; do $(CC) -MM $(CFLAGS) "$${i}" ; done > makefile.dep
+
+-include makefile.dep
+
+$(SND_PCM_BIN): $(SND_PCM_OBJECTS)
+	@echo LD $@
+	$(Q)$(LD) $(LDFLAGS) $(SND_PCM_LIBS) $(SND_PCM_OBJECTS) -o $(SND_PCM_BIN)
+
+#$(SND_CTL_BIN): $(SND_CTL_OBJECTS)
+#	@echo LD $@
+#	$(Q)$(LD) $(LDFLAGS) $(SND_CTL_LIBS) $(SND_CTL_OBJECTS) -o $(SND_CTL_BIN)
+
+%.o: %.c
+	@echo GCC $<
+	$(Q)$(CC) -c $(CFLAGS) $(CPPFLAGS) $<
+
+clean:
+	@echo Cleaning...
+	$(Q)rm -vf *.o *.so
+
+install: all
+	@echo Installing...
+	$(Q)mkdir -p ${DESTDIR}/usr/$(LIBDIR)/alsa-lib/
+	$(Q)install -m 644 $(SND_PCM_BIN) ${DESTDIR}/usr/$(LIBDIR)/alsa-lib/
+	#$(Q)install -m 644 $(SND_CTL_BIN) ${DESTDIR}/usr/$(LIBDIR)/alsa-lib/
+
+uninstall:
+	@echo Un-installing...
+	$(Q)rm ${DESTDIR}/usr/lib/alsa-lib/$(SND_PCM_BIN)
+	#$(Q)rm ${DESTDIR}/usr/lib/alsa-lib/$(SND_CTL_BIN)

ac108_plugin/README.md

@@ -0,0 +1,5 @@
+#seeed-4mic-voicecard alsa plugin
+```
+sudo apt install libasound2-dev
+make && sudo make install
+```

ac108_plugin/ac108_help.c

@@ -0,0 +1,77 @@
+#include "ac108_help.h"
+
+void generate_sine(const snd_pcm_channel_area_t *areas, 
+			  snd_pcm_uframes_t offset,
+			  int count, double *_phase)
+{
+	 snd_pcm_format_t format = SND_PCM_FORMAT_S32;	/* sample format */
+	 unsigned int rate = 16000;			/* stream rate */
+	 unsigned int channels = 4;			/* count of channels */
+	 unsigned int buffer_time = 500000;		/* ring buffer length in us */
+	 unsigned int period_time = 100000;		/* period time in us */
+	 double freq = 160;				/* sinusoidal wave frequency in Hz */
+	 int verbose = 0;					/* verbose flag */
+	 int resample = 1;				/* enable alsa-lib resampling */
+	 int period_event = 0;				/* produce poll event after each period */
+		
+	static double max_phase = 2. * M_PI;
+	double phase = *_phase;
+	double step = max_phase*freq/(double)rate;
+	unsigned char *samples[channels];
+	int steps[channels];
+	unsigned int chn;
+	int format_bits = snd_pcm_format_width(format);
+	unsigned int maxval = (1 << (format_bits - 1)) - 1;
+	int bps = format_bits / 8;  /* bytes per sample */
+	int phys_bps = snd_pcm_format_physical_width(format) / 8;
+	int big_endian = snd_pcm_format_big_endian(format) == 1;
+	int to_unsigned = snd_pcm_format_unsigned(format) == 1;
+	int is_float = (format == SND_PCM_FORMAT_FLOAT_LE ||
+			format == SND_PCM_FORMAT_FLOAT_BE);
+
+	/* verify and prepare the contents of areas */
+	for (chn = 0; chn < channels; chn++) {
+		if ((areas[chn].first % 8) != 0) {
+			printf("areas[%i].first == %i, aborting...\n", chn, areas[chn].first);
+			exit(EXIT_FAILURE);
+		}
+		samples[chn] = /*(signed short *)*/(((unsigned char *)areas[chn].addr) + (areas[chn].first / 8));
+		if ((areas[chn].step % 16) != 0) {
+			printf("areas[%i].step == %i, aborting...\n", chn, areas[chn].step);
+			exit(EXIT_FAILURE);
+		}
+		steps[chn] = areas[chn].step / 8;
+		samples[chn] += offset * steps[chn];
+	}
+	/* fill the channel areas */
+	while (count-- > 0) {
+		union {
+			float f;
+			int i;
+		} fval;
+		int res, i;
+		if (is_float) {
+			fval.f = sin(phase);
+			res = fval.i;
+		} else
+			res = sin(phase) * maxval;
+		if (to_unsigned)
+			res ^= 1U << (format_bits - 1);
+		for (chn = 0; chn < channels; chn++) {
+			/* Generate data in native endian format */
+			if (big_endian) {
+				for (i = 0; i < bps; i++)
+					*(samples[chn] + phys_bps - 1 - i) = (res >> i * 8) & 0xff;
+			} else {
+				for (i = 0; i < bps; i++)
+					*(samples[chn] + i) = (res >>  i * 8) & 0xff;
+			}
+			samples[chn] += steps[chn];
+		}
+		phase += step;
+		if (phase >= max_phase)
+			phase -= max_phase;
+	}
+	*_phase = phase;
+}
+

ac108_plugin/ac108_help.h

@@ -0,0 +1,11 @@
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <alsa/asoundlib.h>
+#include <alsa/pcm_external.h>
+#include <alsa/pcm_plugin.h>
+#include <math.h>
+
+void generate_sine(const snd_pcm_channel_area_t *areas,        
+                          snd_pcm_uframes_t offset,
+                          int count, double *_phase);

ac108_plugin/pcm_ac108.c

@@ -0,0 +1,536 @@
+//https://github.com/HazouPH/android_device_motorola_smi-plus/blob/48029b4afc307c73181b108a5b0155b9f20856ca/smi-modules/alsa-lib_module_voice/pcm_voice.c
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+#include <alsa/asoundlib.h>
+#include <alsa/pcm_external.h>
+#include <alsa/pcm_plugin.h>
+#include "ac108_help.h"
+#include <math.h>
+
+#define ARRAY_SIZE(ary)	(sizeof(ary)/sizeof(ary[0]))
+#define  AC108_FRAME_SIZE 40960
+struct ac108_t {
+	snd_pcm_ioplug_t io;
+	snd_pcm_t *pcm;
+	snd_pcm_hw_params_t *hw_params;
+	unsigned int last_size;
+	unsigned int ptr;
+	unsigned int        latency;         // Delay in usec
+	unsigned int        bufferSize;      // Size of sample buffer
+};
+static unsigned char capture_buf[AC108_FRAME_SIZE];
+/* set up the fixed parameters of pcm PCM hw_parmas */
+static int ac108_slave_hw_params_half(struct ac108_t *capture, unsigned int rate,snd_pcm_format_t format) {
+	int err;
+    snd_pcm_uframes_t bufferSize = capture->bufferSize;
+    unsigned int latency = capture->latency;
+
+    unsigned int buffer_time = 0;
+    unsigned int period_time = 0;
+	if ((err = snd_pcm_hw_params_malloc(&capture->hw_params)) < 0) return err;
+
+	if ((err = snd_pcm_hw_params_any(capture->pcm, capture->hw_params)) < 0) {
+		SNDERR("Cannot get pcm hw_params");
+		goto out;
+	}
+	if ((err = snd_pcm_hw_params_set_access(capture->pcm, capture->hw_params,
+											SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
+		SNDERR("Cannot set pcm access RW_INTERLEAVED");
+		goto out;
+	}
+	if ((err = snd_pcm_hw_params_set_channels(capture->pcm, capture->hw_params, 2)) < 0) {
+		SNDERR("Cannot set pcm channels 2");
+		goto out;
+	}
+	if ((err = snd_pcm_hw_params_set_format(capture->pcm, capture->hw_params,
+											format)) < 0) {
+		SNDERR("Cannot set pcm format");
+		goto out;
+	}
+	if ((err = snd_pcm_hw_params_set_rate(capture->pcm, capture->hw_params, rate, 0)) < 0) {
+		SNDERR("Cannot set pcm rate %d", rate);
+		goto out;
+	}
+
+    err = snd_pcm_hw_params_get_buffer_time_max(capture->hw_params,
+            &buffer_time, 0);
+    if (buffer_time > 80000)
+        buffer_time = 80000;
+    period_time = buffer_time / 4;
+
+    err = snd_pcm_hw_params_set_period_time_near(capture->pcm, capture->hw_params,
+            &period_time, 0);
+    if (err < 0) {
+        SNDERR("Unable to set_period_time_near");
+        goto out;
+    }
+    err = snd_pcm_hw_params_set_buffer_time_near(capture->pcm, capture->hw_params,
+            &buffer_time, 0);
+    if (err < 0) {
+        SNDERR("Unable to set_buffer_time_near");
+        goto out;
+    }
+
+    capture->bufferSize = bufferSize;
+    capture->latency = latency;
+
+	return 0;
+
+out:
+	free(capture->hw_params);
+	capture->hw_params = NULL;
+	return err;
+}
+
+/*
+ * start and stop callbacks - just trigger pcm PCM
+ */
+static int ac108_start(snd_pcm_ioplug_t *io) {
+	struct ac108_t *capture = io->private_data;
+    if(!capture->pcm) {
+		SNDERR( "pcm is lost\n");
+	}	
+
+	return snd_pcm_start(capture->pcm);
+}
+
+static int ac108_stop(snd_pcm_ioplug_t *io) {
+	struct ac108_t *capture = io->private_data;
+
+	return snd_pcm_drop(capture->pcm);
+}
+/*
+ * pointer callback
+ *
+ * Calculate the current position from the delay of pcm PCM
+ */
+static snd_pcm_sframes_t ac108_pointer(snd_pcm_ioplug_t *io) {
+
+	struct ac108_t *capture = io->private_data;
+	int  size;
+	
+	assert(capture);
+
+
+	size = snd_pcm_avail(capture->pcm);
+	if (size < 0) 
+		return size;
+
+	size = size/2;
+
+	if (size > capture->last_size) {
+			capture->ptr += size - capture->last_size;
+			capture->ptr %= io->buffer_size;
+	}
+
+	//fprintf(stderr, "%s :%d %d %d %d %d %d\n", __func__,capture->ptr ,capture->last_size,size,  io->buffer_size,io->appl_ptr, io->hw_ptr);
+	capture->last_size = size;
+	
+	return capture->ptr;
+}
+
+/*
+ * transfer callback
+ */
+static snd_pcm_sframes_t ac108_transfer(snd_pcm_ioplug_t *io,
+										const snd_pcm_channel_area_t *dst_areas,
+										snd_pcm_uframes_t dst_offset,
+										snd_pcm_uframes_t size) {
+	struct ac108_t *capture = io->private_data;
+	int chn;
+	unsigned char *dst_samples[io->channels];
+	int dst_steps[io->channels];
+	int bps = snd_pcm_format_width(io->format) / 8;  /* bytes per sample */
+	int i;
+	int count = 0;	
+	int err = 0;
+	unsigned char *src_buf;
+	unsigned char src_data[4][4];
+
+	
+	memset(capture_buf,0,AC108_FRAME_SIZE);
+
+	if(snd_pcm_avail(capture->pcm) > size*2){
+		if ((err = snd_pcm_readi (capture->pcm, capture_buf, size*2)) != size*2) {
+			SNDERR("read from audio interface failed %ld %d  %s!\n",size,err,snd_strerror (err));
+			exit(EXIT_FAILURE);
+			size = 0 ;
+		}
+	}else{
+		size = 0;
+	}
+#if 1	
+	/* verify and prepare the contents of areas */
+	for (chn = 0; chn < io->channels; chn++) {
+		if ((dst_areas[chn].first % 8) != 0) {
+			SNDERR("dst_areas[%i].first == %i, aborting...\n", chn, dst_areas[chn].first);
+			exit(EXIT_FAILURE);
+		}
+		dst_samples[chn] = /*(signed short *)*/(((unsigned char *)dst_areas[chn].addr) + (dst_areas[chn].first / 8));
+		if ((dst_areas[chn].step % 16) != 0) {
+			SNDERR("dst_areas[%i].step == %i, aborting...\n", chn, dst_areas[chn].step);
+			exit(EXIT_FAILURE);
+		}
+		dst_steps[chn] = dst_areas[chn].step / 8;
+		dst_samples[chn] += dst_offset * dst_steps[chn];
+	}
+#endif	
+	//  for(i = 0; i < size*2*bps;i++){
+	// 	fprintf(stderr,"%x ",capture_buf[i]);
+	// 	if(i%4 == 0)
+	// 		fprintf(stderr,"\n");
+	// }
+
+	//generate_sine(dst_areas, dst_offset,size, &count);
+	src_buf = capture_buf;
+#if 1
+	while(count < size){
+		for(chn = 0; chn < 4; chn++){
+			for (i = 0; i < bps; i++){
+				src_data[chn][i] = src_buf[i];
+			}
+			src_buf += bps ;						
+		}
+
+		for(chn = 0; chn < io->channels; chn++){
+			for (i = 0; i < bps; i++){
+				*(dst_samples[chn] + i) = src_data[chn][i];
+				//fprintf(stderr,"%x ",*(dst_samples[chn] + i));
+			}
+			//fprintf(stderr,"\n");
+			dst_samples[chn] += dst_steps[chn];	
+		}
+		count++;
+	}
+
+#endif
+
+	capture->last_size -= size;
+	
+	return size;
+}
+
+/*
+ * poll-related callbacks - just pass to pcm PCM
+ */
+static int ac108_poll_descriptors_count(snd_pcm_ioplug_t *io) {
+	struct ac108_t *capture = io->private_data;
+
+	return snd_pcm_poll_descriptors_count(capture->pcm);
+}
+
+static int ac108_poll_descriptors(snd_pcm_ioplug_t *io, struct pollfd *pfd,
+								  unsigned int space) {
+	struct ac108_t *capture = io->private_data;
+
+	return snd_pcm_poll_descriptors(capture->pcm, pfd, space);
+}
+
+static int ac108_poll_revents(snd_pcm_ioplug_t *io, struct pollfd *pfd,
+							  unsigned int nfds, unsigned short *revents) {
+	struct ac108_t *capture = io->private_data;
+
+	return snd_pcm_poll_descriptors_revents(capture->pcm, pfd, nfds, revents);
+}
+
+/*
+ * close callback
+ */
+static int ac108_close(snd_pcm_ioplug_t *io) {
+	struct ac108_t *capture = io->private_data;
+	if (capture->pcm)  
+		snd_pcm_close(capture->pcm);
+
+	return 0;
+}
+
+static int setSoftwareParams(struct ac108_t *capture) {
+	snd_pcm_sw_params_t *softwareParams;
+	int err;
+
+	snd_pcm_uframes_t bufferSize = 0;
+	snd_pcm_uframes_t periodSize = 0;
+	snd_pcm_uframes_t startThreshold, stopThreshold;
+	snd_pcm_sw_params_alloca(&softwareParams);
+
+	// Get the current software parameters
+	err = snd_pcm_sw_params_current(capture->pcm, softwareParams);
+	if (err < 0) {
+		SNDERR("Unable to get software parameters: %s", snd_strerror(err));
+		goto done;
+	}
+
+	// Configure ALSA to start the transfer when the buffer is almost full.
+	snd_pcm_get_params(capture->pcm, &bufferSize, &periodSize);
+
+
+	startThreshold = 1;
+	stopThreshold = bufferSize;
+
+
+	err = snd_pcm_sw_params_set_start_threshold(capture->pcm, softwareParams,
+												startThreshold);
+	if (err < 0) {
+		SNDERR("Unable to set start threshold to %lu frames: %s",
+				startThreshold, snd_strerror(err));
+		goto done;
+	}
+
+	err = snd_pcm_sw_params_set_stop_threshold(capture->pcm, softwareParams,
+											   stopThreshold);
+	if (err < 0) {
+		SNDERR("Unable to set stop threshold to %lu frames: %s",
+				stopThreshold, snd_strerror(err));
+		goto done;
+	}
+	// Allow the transfer to start when at least periodSize samples can be
+	// processed.
+	err = snd_pcm_sw_params_set_avail_min(capture->pcm, softwareParams,
+										  periodSize);
+	if (err < 0) {
+		SNDERR("Unable to configure available minimum to %lu: %s",
+				periodSize, snd_strerror(err));
+		goto done;
+	}
+
+	// Commit the software parameters back to the device.
+	err = snd_pcm_sw_params(capture->pcm, softwareParams);
+	if (err < 0) 
+		SNDERR("Unable to configure software parameters: %s",snd_strerror(err));
+
+
+
+	return 0;
+done:
+	snd_pcm_sw_params_free(softwareParams);
+
+	return err;
+}
+/*
+ * hw_params callback
+ *
+ * Set up pcm PCM according to the current parameters
+ */
+static int ac108_hw_params(snd_pcm_ioplug_t *io, snd_pcm_hw_params_t *params) {
+	struct ac108_t *capture = io->private_data;
+	snd_pcm_uframes_t period_size;
+	snd_pcm_uframes_t buffer_size;
+	int err;
+	if (!capture->hw_params) {
+		err = ac108_slave_hw_params_half(capture, 2*io->rate,io->format);
+		if (err < 0) {
+			SNDERR("ac108_slave_hw_params_half error\n");
+			return err;
+		}
+	}
+	period_size = io->period_size;
+	if ((err = snd_pcm_hw_params_set_period_size_near(capture->pcm, capture->hw_params,
+													  &period_size, NULL)) < 0) {
+		SNDERR("Cannot set pcm period size %ld", period_size);
+		return err;
+	}
+	buffer_size = io->buffer_size;
+	if ((err = snd_pcm_hw_params_set_buffer_size_near(capture->pcm, capture->hw_params,
+													  &buffer_size)) < 0) {
+		SNDERR("Cannot set pcm buffer size %ld", buffer_size);
+		return err;
+	}
+	if ((err = snd_pcm_hw_params(capture->pcm, capture->hw_params)) < 0) {
+		SNDERR("Cannot set pcm hw_params");
+		return err;
+	}
+	setSoftwareParams(capture);
+	return 0;
+}
+/*
+ * hw_free callback
+ */
+static int ac108_hw_free(snd_pcm_ioplug_t *io) {
+	struct ac108_t *capture = io->private_data;
+	free(capture->hw_params);
+	capture->hw_params = NULL;
+	
+	return snd_pcm_hw_free(capture->pcm);
+
+}
+
+
+static int ac108_prepare(snd_pcm_ioplug_t *io) {
+	struct ac108_t *capture = io->private_data;
+	capture->ptr = 0;
+	capture->last_size =0;
+	return snd_pcm_prepare(capture->pcm);
+}
+static int ac108_drain(snd_pcm_ioplug_t *io) {
+	struct ac108_t *capture = io->private_data;
+	
+	return snd_pcm_drain(capture->pcm);
+}
+#if 0
+static int ac108_sw_params(snd_pcm_ioplug_t *io, snd_pcm_sw_params_t *params) {
+	return 0;
+}
+#endif 
+static int ac108_delay(snd_pcm_ioplug_t * io, snd_pcm_sframes_t * delayp){
+
+	return 0;
+}
+/*
+ * callback table
+ */
+static snd_pcm_ioplug_callback_t a108_ops = {
+	.start = ac108_start,
+	.stop = ac108_stop,
+	.pointer = ac108_pointer,
+	.transfer = ac108_transfer,
+	.poll_descriptors_count = ac108_poll_descriptors_count,
+	.poll_descriptors = ac108_poll_descriptors,
+	.poll_revents = ac108_poll_revents,
+	.close = ac108_close,
+	.hw_params = ac108_hw_params,
+	.hw_free = ac108_hw_free,
+//	.sw_params = ac108_sw_params,
+	.prepare = ac108_prepare,
+	.drain = ac108_drain,
+	.delay = ac108_delay,
+};
+
+
+static int ac108_set_hw_constraint(struct ac108_t  *capture) {
+	static unsigned int accesses[] = {
+		SND_PCM_ACCESS_RW_INTERLEAVED 
+	};
+	unsigned int formats[] = { SND_PCM_FORMAT_S32,
+							   SND_PCM_FORMAT_S16 };
+
+	unsigned int  rates[] = {
+		8000,
+		16000,
+		32000,
+		44100,
+		48000,
+		96000
+	};
+	int err;
+
+
+	err = snd_pcm_ioplug_set_param_list(&capture->io,
+										SND_PCM_IOPLUG_HW_ACCESS,
+										ARRAY_SIZE(accesses),
+										accesses);
+	if (err < 0){
+		SNDERR("ioplug cannot set ac108 hw access");
+		return err;
+	} 
+
+	if ((err = snd_pcm_ioplug_set_param_list(&capture->io, SND_PCM_IOPLUG_HW_FORMAT,
+											 ARRAY_SIZE(formats), formats)) < 0 ||
+		(err = snd_pcm_ioplug_set_param_minmax(&capture->io, SND_PCM_IOPLUG_HW_CHANNELS,
+											   1, 4)) < 0 ||
+		(err = snd_pcm_ioplug_set_param_list(&capture->io, SND_PCM_IOPLUG_HW_RATE,
+												ARRAY_SIZE(rates), rates)) < 0) 
+												{
+		SNDERR("ioplug cannot set ac108 format channel rate!");											
+		return err;
+	}
+	err = snd_pcm_ioplug_set_param_minmax(&capture->io,SND_PCM_IOPLUG_HW_BUFFER_BYTES,
+										  1, 4 * 1024 * 1024);
+	if (err < 0){ 
+		SNDERR("ioplug cannot set ac108 hw buffer bytes");		
+		return err;
+	}
+
+	err = snd_pcm_ioplug_set_param_minmax(&capture->io,SND_PCM_IOPLUG_HW_PERIOD_BYTES,
+										  128, 2 * 1024 * 1024);
+	if (err < 0) {
+		SNDERR("ioplug cannot set ac108 hw period bytes");		
+		return err;
+	}
+
+	err = snd_pcm_ioplug_set_param_minmax(&capture->io, SND_PCM_IOPLUG_HW_PERIODS,3, 1024);
+	if (err < 0) {
+		SNDERR("ioplug cannot set ac108 hw periods");		
+		return err;
+	}									  
+	return 0;
+}
+
+/*
+ * Main entry point
+ */
+SND_PCM_PLUGIN_DEFINE_FUNC(ac108) {
+	snd_config_iterator_t i, next;
+	int err;
+	const char *pcm_string = NULL;
+	struct ac108_t *capture;
+	int channels;
+	if (stream != SND_PCM_STREAM_CAPTURE) {
+		SNDERR("a108 is only for capture");
+		return -EINVAL;
+	}
+
+	snd_config_for_each(i, next, conf) {
+		snd_config_t *n = snd_config_iterator_entry(i);
+		const char *id;
+		if (snd_config_get_id(n, &id) < 0) continue;
+		if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue;
+
+		if (strcmp(id, "slavepcm") == 0) {
+			if (snd_config_get_string(n, &pcm_string) < 0) {
+				SNDERR("ac108 slavepcm must be a string");
+				return -EINVAL;
+			}
+			continue;
+		}
+
+		if (strcmp(id, "channels") == 0) {
+			long val;
+			if (snd_config_get_integer(n, &val) < 0) {
+				SNDERR("Invalid type for %s", id);
+				return -EINVAL;
+			}
+			channels = val;
+			if (channels != 2 && channels != 4 && channels != 6) {
+				SNDERR("channels must be 2, 4 or 6");
+				return -EINVAL;
+			}
+			continue;
+		}
+	}
+
+
+	capture = calloc(1, sizeof(*capture));
+	if (!capture) {
+		SNDERR("cannot allocate");
+		return -ENOMEM;
+	}
+	err = snd_pcm_open(&capture->pcm, pcm_string, stream, mode);
+	if (err < 0) goto error;
+
+
+	
+	//SND_PCM_NONBLOCK
+	capture->io.version = SND_PCM_IOPLUG_VERSION;
+	capture->io.name = "AC108 decode Plugin";
+	capture->io.mmap_rw = 0;
+	capture->io.callback = &a108_ops;
+	capture->io.private_data = capture;
+
+	err = snd_pcm_ioplug_create(&capture->io, name, stream, mode);
+	if (err < 0) goto error;
+
+	if ((err = ac108_set_hw_constraint(capture)) < 0) {
+		snd_pcm_ioplug_delete(&capture->io);
+		return err;
+	}
+	*pcmp = capture->io.pcm;
+	return 0;
+	
+error:
+	if (capture->pcm) snd_pcm_close(capture->pcm);
+	free(capture);
+	return err;
+}
+
+SND_PCM_PLUGIN_SYMBOL(ac108);

ac10x.h

@@ -0,0 +1,126 @@
+/*
+ * 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__

asound_2mic.conf

@@ -1,6 +1,10 @@
 # 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"
@@ -19,14 +23,14 @@ pcm.capture {
 
 pcm.dmixed {
     type dmix
-    slave.pcm "hw:0,0"
+    slave.pcm "hw:seeed2micvoicec"
     ipc_key 555555 
 }
 
 pcm.array {
     type dsnoop
     slave {
-        pcm "hw:0,0"
+        pcm "hw:seeed2micvoicec"
         channels 2
     }
     ipc_key 666666

asound_4mic.conf

@@ -0,0 +1,33 @@
+# 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
+# }

asound_6mic.conf

@@ -0,0 +1,90 @@
+# 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
+}
+
+

builddtbo.sh

@@ -1,4 +1,10 @@
-dtoverlay -r seeed-voicecard
-dtc -@ -I dts -O dtb -o seeed-voicecard.dtbo seeed-voicecard-overlay.dts
-cp seeed-voicecard.dtbo /boot/overlays
-dtoverlay seeed-voicecard
+#!/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
+
+# cp *.dtbo /boot/overlays
+# dtoverlay seeed-2mic-voicecard

dkms.conf

@@ -1,5 +1,15 @@
 PACKAGE_NAME="seeed-voicecard"
-PACKAGE_VERSION="0.1"
-BUILT_MODULE_NAME[0]="wm8960"
+PACKAGE_VERSION="0.3"
+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"

install.sh

@@ -5,22 +5,134 @@ 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
 
-ver="0.1"
+#
+# 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 [ $errorFound = 1 ] ; then
+  echo "Errors found, exiting." 1>&2
+  exit 1
+fi
+
+ver="0.3"
+uname_r=$(uname -r)
 
 # we create a dir with this version to ensure that 'dkms remove' won't delete
 # the sources during kernel updates
 marker="0.0.0"
 
-apt update
-apt-get -y install raspberrypi-kernel-headers raspberrypi-kernel
-apt-get -y install  dkms
+_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-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
 
 # locate currently installed kernels (may be different to running kernel if
 # it's just been updated)
-kernels=$(ls /lib/modules | sed "s/^/-k /")
+base_ver=$(get_kernel_version)
+base_ver=${base_ver%%[-+]*}
+#kernels="${base_ver}+ ${base_ver}-v7+ ${base_ver}-v7l+"
+kernels=$(uname -r)
 
 function install_module {
+  local _i
+
   src=$1
   mod=$2
 
@@ -29,13 +141,19 @@ function install_module {
   fi
 
   if [[ -e /usr/src/$mod-$ver || -e /var/lib/dkms/$mod/$ver ]]; then
-    dkms remove -m $mod -v $ver --all
+    dkms remove --force -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
-  dkms build $kernels -m $mod -v $ver && dkms install $kernels -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
 
   mkdir -p /var/lib/dkms/$mod/$ver/$marker
 }
@@ -43,23 +161,61 @@ function install_module {
 install_module "./" "seeed-voicecard"
 
 
-(
-  cp seeed-voicecard.dtbo /boot/overlays
-  cp asound.state /var/lib/alsa/asound.state
-)
+# install dtbos
+cp seeed-2mic-voicecard.dtbo $OVERLAYS
+cp seeed-4mic-voicecard.dtbo $OVERLAYS
+cp seeed-8mic-voicecard.dtbo $OVERLAYS
 
-echo 'wm8960' | sudo tee --append /etc/modules > /dev/null
+#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
 
-sed -i \
-  -e "s/^dtparam=audio=on/#\0/" \
-  -e "s/^#\(dtparam=i2s=on\)/\1/" \
-  /boot/config.txt
-grep -q "dtoverlay=i2s-mmap" /boot/config.txt || \
-  echo "dtoverlay=i2s-mmap" >> /boot/config.txt
-grep -q "dtoverlay=seeed-voicecard" /boot/config.txt || \
-  echo "dtoverlay=seeed-voicecard" >> /boot/config.txt
-grep -q "dtparam=i2s=on" /boot/config.txt || \
-  echo "dtparam=i2s=on" >> /boot/config.txt
+#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 || \
+  echo "snd-soc-ac108" >> /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" >> $CONFIG
+
+
+grep -q "^dtparam=i2s=on$" $CONFIG || \
+  echo "dtparam=i2s=on" >> $CONFIG
+
+#install config files
+mkdir /etc/voicecard || true
+cp *.conf /etc/voicecard
+cp *.state /etc/voicecard
+
+#create git repo
+git_email=$(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"
+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"
+
+cp seeed-voicecard /usr/bin/
+cp seeed-voicecard.service /lib/systemd/system/
+systemctl enable  seeed-voicecard.service
+systemctl start   seeed-voicecard
 
 echo "------------------------------------------------------"
 echo "Please reboot your raspberry pi to apply all settings"

patches/back-to-v4.19.diff

@@ -0,0 +1,454 @@
+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 = {

patches/back-to-v5.4.diff

@@ -0,0 +1,245 @@
+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 = {

patches/back-to-v5.8.diff

@@ -0,0 +1,71 @@
+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 = {

pulseaudio/91-seeedvoicecard.rules

@@ -0,0 +1,8 @@
+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"

pulseaudio/README.md

@@ -0,0 +1,251 @@
+# 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.

pulseaudio/pulse_config_4mic/daemon.conf

@@ -0,0 +1,87 @@
+# 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

pulseaudio/pulse_config_4mic/default.pa

@@ -0,0 +1,144 @@
+#!/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
+

pulseaudio/pulse_config_4mic/seeed-voicecard.conf

@@ -0,0 +1,17 @@
+# /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

pulseaudio/pulse_config_6mic/daemon.conf

@@ -0,0 +1,87 @@
+# 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

pulseaudio/pulse_config_6mic/default.pa

@@ -0,0 +1,143 @@
+#!/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

pulseaudio/pulse_config_6mic/seeed-voicecard.conf

@@ -0,0 +1,34 @@
+# /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

seeed-2mic-voicecard-overlay.dts

@@ -38,49 +38,13 @@
 		};
     };
 
+
     fragment@3 {
-        target = <&sound>;
-        master_overlay: __dormant__ {
-            compatible = "simple-audio-card";
-            simple-audio-card,format = "i2s";
-            simple-audio-card,name = "seeed-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";
                 simple-audio-card,format = "i2s";
-                simple-audio-card,name = "seeed-voicecard"; 
+                simple-audio-card,name = "seeed-2mic-voicecard"; 
                 status = "okay";
                 simple-audio-card,widgets =
                         "Microphone", "Mic Jack",
@@ -114,8 +78,7 @@
     };
 
     __overrides__ {
-        alsaname = <&master_overlay>,"simple-audio-card,name",
-                    <&slave_overlay>,"simple-audio-card,name";
+        alsaname = <&slave_overlay>,"simple-audio-card,name";
         compatible = <&wm8960>,"compatible";
         master = <0>,"=2!3";
     };

seeed-4mic-voicecard-overlay.dts

@@ -0,0 +1,75 @@
+/dts-v1/;
+/plugin/;
+
+/ {
+	compatible = "brcm,bcm2708";
+
+	fragment@0 {
+		target = <&i2s>;
+		__overlay__ {
+			#sound-dai-cells = <0>;
+			status = "okay";
+		};
+	};
+
+	fragment@1 {
+		target-path = "/";
+		__overlay__ {
+			ac108_mclk: codec-mclk {
+				compatible = "fixed-clock";
+				#clock-cells = <0>;
+				clock-frequency = <24000000>;
+			};  
+		};
+	};
+
+	fragment@2 {
+		target = <&i2c1>;
+		__overlay__ {
+			#address-cells = <1>;
+			#size-cells = <0>;
+			status = "okay";
+
+			ac108_a: ac108@3b{
+				compatible = "x-power,ac108_0";
+				reg = <0x3b>;
+				#sound-dai-cells = <0>;
+				data-protocol = <0>;
+			};
+		};
+	};
+
+
+	fragment@3 {
+		target = <&sound>;
+
+		sound_overlay: __overlay__ {
+			compatible = "seeed-voicecard";
+			seeed-voice-card,format = "dsp_a";
+			seeed-voice-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 {
+				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 {
+				sound-dai = <&ac108_a>;
+				clocks =  <&ac108_mclk>;
+			};
+		};
+	};
+
+	__overrides__ {
+		card-name = <&sound_overlay>,"seeed-voice-card,name";
+	};
+};
+

seeed-8mic-voicecard-overlay.dts

@@ -0,0 +1,118 @@
+/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";
+	};
+};
+

seeed-voicecard

@@ -0,0 +1,163 @@
+#!/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

seeed-voicecard.c

@@ -0,0 +1,926 @@
+/*
+ * 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>");

seeed-voicecard.service

@@ -0,0 +1,12 @@
+[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

sound-compatible-4.18.h

@@ -0,0 +1,47 @@
+/*
+ * (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__
+

tools/coherence.py

@@ -0,0 +1,44 @@
+"""
+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()
+

tools/phase_test.py

@@ -0,0 +1,83 @@
+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()

ubuntu-prerequisite.sh

@@ -0,0 +1,37 @@
+#!/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

uninstall.sh

@@ -0,0 +1,100 @@
+#!/bin/bash
+
+if [[ $EUID -ne 0 ]]; then
+   echo "This script must be run as root (use sudo)" 1>&2
+   exit 1
+fi
+
+is_Raspberry=$(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
+
+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
+    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 $RET
+    fi
+}
+
+RPI_HATS="seeed-2mic-voicecard seeed-4mic-voicecard seeed-8mic-voicecard"
+
+PATH=$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
+
+rm  ${OVERLAYS}/seeed-2mic-voicecard.dtbo || true
+rm  ${OVERLAYS}/seeed-4mic-voicecard.dtbo || true
+rm  ${OVERLAYS}/seeed-8mic-voicecard.dtbo || true
+
+echo "remove alsa configs"
+rm -rf  /etc/voicecard/ || true
+
+echo "disabled seeed-voicecard.service "
+systemctl stop seeed-voicecard.service
+systemctl disable seeed-voicecard.service
+
+echo "remove seeed-voicecard"
+rm  /usr/bin/seeed-voicecard || true
+rm  /lib/systemd/system/seeed-voicecard.service || true
+
+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
+
+echo "------------------------------------------------------"
+echo "Please reboot your raspberry pi to apply all settings"
+echo "Thank you!"
+echo "------------------------------------------------------"

wm8960.c

@@ -25,6 +25,7 @@
 #include <sound/initval.h>
 #include <sound/tlv.h>
 #include <sound/wm8960.h>
+#include "sound-compatible-4.18.h"
 
 #include "wm8960.h"
 
@@ -226,11 +227,12 @@ 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 SNDRV_CTL_TLVD_DECLARE_DB_RANGE(micboost_tlv,
+static const DECLARE_TLV_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),
@@ -506,7 +508,11 @@ 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)
@@ -747,6 +753,7 @@ 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));
@@ -789,7 +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)
+static int wm8960_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_codec *codec = dai->codec;
 
@@ -1229,11 +1236,12 @@ 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,
-	.digital_mute = wm8960_mute,
+	.mute_stream = 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 = {
@@ -1251,7 +1259,11 @@ 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)
@@ -1274,7 +1286,12 @@ static int wm8960_probe(struct snd_soc_codec *codec)
 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,
+	.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 = {
@@ -1301,8 +1318,7 @@ static void wm8960_set_pdata_from_of(struct i2c_client *i2c,
 		pdata->shared_lrclk = true;
 }
 
-static int wm8960_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
+static int wm8960_i2c_probe(struct i2c_client *i2c)
 {
 	struct wm8960_data *pdata = dev_get_platdata(&i2c->dev);
 	struct wm8960_priv *wm8960;
@@ -1367,10 +1383,9 @@ static int wm8960_i2c_probe(struct i2c_client *i2c,
 	return ret;
 }
 
-static int wm8960_i2c_remove(struct i2c_client *client)
+static void wm8960_i2c_remove(struct i2c_client *client)
 {
 	snd_soc_unregister_codec(&client->dev);
-	return 0;
 }
 
 static const struct i2c_device_id wm8960_i2c_id[] = {

wm8960_asound.state

@@ -2,7 +2,7 @@ state.ALSA {
 	control.1 {
 		iface MIXER
 		name 'PCM Playback Volume'
-		value -1995
+		value -1994
 		comment {
 			access 'read write'
 			type INTEGER
@@ -10,7 +10,7 @@ state.ALSA {
 			range '-10239 - 400'
 			dbmin -9999999
 			dbmax 400
-			dbvalue.0 -1995
+			dbvalue.0 -1994
 		}
 	}
 	control.2 {
@@ -65,12 +65,12 @@ state.ALSA {
 		}
 	}
 }
-state.seeedvoicecard {
+state.seeed2micvoicec {
 	control.1 {
 		iface MIXER
 		name 'Capture Volume'
-		value.0 63
-		value.1 63
+		value.0 39
+		value.1 39
 		comment {
 			access 'read write'
 			type INTEGER
@@ -78,8 +78,8 @@ state.seeedvoicecard {
 			range '0 - 63'
 			dbmin -1725
 			dbmax 3000
-			dbvalue.0 3000
-			dbvalue.1 3000
+			dbvalue.0 1200
+			dbvalue.1 1200
 		}
 	}
 	control.2 {