37 changed files with 643 additions and 2875 deletions
--- a/35
+++ b/35
@ -1,17 +1,3 @@
-#
-# 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
@ -28,25 +14,14 @@ endif
 endif


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

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

 install:
-	sudo cp snd-soc-ac108.ko ${DEST}/sound/soc/codecs/
-	sudo cp snd-soc-wm8960.ko ${DEST}/sound/soc/codecs/
-	sudo cp snd-soc-seeed-voicecard.ko ${DEST}/sound/soc/bcm/
+	sudo cp snd-soc-ac108.ko /lib/modules/$(shell uname -r)/kernel/sound/soc/codecs/
+	sudo cp snd-soc-wm8960.ko /lib/modules/$(shell uname -r)/kernel/sound/soc/codecs/
+	sudo cp snd-soc-seeed-voicecard.ko /lib/modules/$(shell uname -r)/kernel/sound/soc/bcm/
 	sudo depmod -a
-
-
-.PHONY: all clean install
-
-endif
-
--- a/README.md
+++ b/README.md
@ -1,20 +1,231 @@
 # seeed-voicecard

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

 ### Install seeed-voicecard
-Get the seeed voice card source code and install all linux kernel drivers
+Get the seeed voice card source code. and install all linux kernel drivers
 ```bash
-git clone https://github.com/HinTak/seeed-voicecard
+git clone https://github.com/respeaker/seeed-voicecard
 cd seeed-voicecard
-sudo ./install.sh
+sudo ./install.sh 
 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)
+## ReSpeaker Mic Hat

+[![](https://github.com/SeeedDocument/MIC_HATv1.0_for_raspberrypi/blob/master/img/mic_hatv1.0.png?raw=true)](https://www.seeedstudio.com/ReSpeaker-2-Mics-Pi-HAT-p-2874.html)
+
+While the upstream wm8960 codec is not currently supported by current Pi kernel builds, upstream wm8960 has some bugs, we had fixed it. we must it build manually.
+
+Check that the sound card name matches the source code seeed-voicecard.
+
+```bash
+#for ReSpeaker 2-mic
+pi@raspberrypi:~/seeed-voicecard $ aplay -l
+**** List of PLAYBACK Hardware Devices ****
+card 0: ALSA [bcm2835 ALSA], device 0: bcm2835 ALSA [bcm2835 ALSA]
+  Subdevices: 8/8
+  Subdevice #0: subdevice #0
+  Subdevice #1: subdevice #1
+  Subdevice #2: subdevice #2
+  Subdevice #3: subdevice #3
+  Subdevice #4: subdevice #4
+  Subdevice #5: subdevice #5
+  Subdevice #6: subdevice #6
+  Subdevice #7: subdevice #7
+card 0: ALSA [bcm2835 ALSA], device 1: bcm2835 ALSA [bcm2835 IEC958/HDMI]
+  Subdevices: 1/1
+  Subdevice #0: subdevice #0
+card 1: seeed2micvoicec [seeed-2mic-voicecard], device 0: bcm2835-i2s-wm8960-hifi wm8960-hifi-0 []
+  Subdevices: 1/1
+  Subdevice #0: subdevice #0
+pi@raspberrypi:~/seeed-voicecard $ arecord -l
+**** List of CAPTURE Hardware Devices ****
+card 1: seeed2micvoicec [seeed-2mic-voicecard], device 0: bcm2835-i2s-wm8960-hifi wm8960-hifi-0 []
+  Subdevices: 1/1
+  Subdevice #0: subdevice #0
+pi@raspberrypi:~/seeed-voicecard $ 
+```
+If you want to change the alsa settings, You can use `sudo alsactl --file=/etc/voicecard/wm8960_asound.state  store` to save it.
+
+
+
+#### Next step
+Go to https://github.com/respeaker/mic_hat to build voice enabled projects with Google Assistant SDK or Alexa Voice Service.
+
+## ReSpeaker 4 Mic Array
+
+[![](https://github.com/SeeedDocument/ReSpeaker-4-Mic-Array-for-Raspberry-Pi/blob/master/img/features.png?raw=true)](https://www.seeedstudio.com/ReSpeaker-4-Mic-Array-for-Raspberry-Pi-p-2941.html)
+
+The 4 Mic Array uses ac108 which includes 4 ADCs, we also write ac108 rapberry pi linux kernel driver.
+
+Check that the sound card name matches the source code seeed-voicecard.
+
+```bash
+#for ReSpeaker 4 Mic Array
+pi@raspberrypi:~ $ arecord -L
+null
+    Discard all samples (playback) or generate zero samples (capture)
+playback
+capture
+dmixed
+array
+ac108
+default:CARD=seeed4micvoicec
+    seeed-4mic-voicecard, 
+    Default Audio Device
+sysdefault:CARD=seeed4micvoicec
+    seeed-4mic-voicecard, 
+    Default Audio Device
+dmix:CARD=seeed4micvoicec,DEV=0
+    seeed-4mic-voicecard, 
+    Direct sample mixing device
+dsnoop:CARD=seeed4micvoicec,DEV=0
+    seeed-4mic-voicecard, 
+    Direct sample snooping device
+hw:CARD=seeed4micvoicec,DEV=0
+    seeed-4mic-voicecard, 
+    Direct hardware device without any conversions
+plughw:CARD=seeed4micvoicec,DEV=0
+    seeed-4mic-voicecard, 
+    Hardware device with all software conversions
+pi@raspberrypi:~ $ 
+```
+If you want to change the alsa settings, You can use `sudo alsactl --file=/etc/voicecard/ac108_asound.state  store` to save it.
+
+## 6-Mics Circular Array Kit
+
+[![](https://user-images.githubusercontent.com/3901856/37268348-6adef768-2600-11e8-8861-588b1c3ea142.png)]()
+
+The 6 Mics Circular Array Kit uses ac108 x 2 / ac101 x 1 / micphones x 6, includes 8 ADCs and 2 DACs.
+
+The driver is implemented with 8 input channels & 8 output channels.
+>**The first 6 input channel are MIC recording data,  
+the rest 2 input channel are echo channel of playback  
+The first 2 output channel are playing data, the rest 6 output channel are dummy**
+
+
+Check that the sound card name matches the source code seeed-voicecard.
+```bash
+#for 6 Mic Circular Array
+pi@raspberrypi:~ $ arecord -L
+null
+    Discard all samples (playback) or generate zero samples (capture)
+default
+playback
+dmixed
+ac108
+multiapps
+ac101
+sysdefault:CARD=seeed8micvoicec
+    seeed-8mic-voicecard,
+    Default Audio Device
+dmix:CARD=seeed8micvoicec,DEV=0
+    seeed-8mic-voicecard,
+    Direct sample mixing device
+dsnoop:CARD=seeed8micvoicec,DEV=0
+    seeed-8mic-voicecard,
+    Direct sample snooping device
+hw:CARD=seeed8micvoicec,DEV=0
+    seeed-8mic-voicecard,
+    Direct hardware device without any conversions
+plughw:CARD=seeed8micvoicec,DEV=0
+    seeed-8mic-voicecard,
+    Hardware device with all software conversions
+    
+pi@raspberrypi:~ $ aplay -L
+null
+    Discard all samples (playback) or generate zero samples (capture)
+default
+playback
+dmixed
+ac108
+multiapps
+ac101
+sysdefault:CARD=ALSA
+    bcm2835 ALSA, bcm2835 ALSA
+    Default Audio Device
+dmix:CARD=ALSA,DEV=0
+    bcm2835 ALSA, bcm2835 ALSA
+    Direct sample mixing device
+dmix:CARD=ALSA,DEV=1
+    bcm2835 ALSA, bcm2835 IEC958/HDMI
+    Direct sample mixing device
+dsnoop:CARD=ALSA,DEV=0
+    bcm2835 ALSA, bcm2835 ALSA
+    Direct sample snooping device
+dsnoop:CARD=ALSA,DEV=1
+    bcm2835 ALSA, bcm2835 IEC958/HDMI
+    Direct sample snooping device
+hw:CARD=ALSA,DEV=0
+    bcm2835 ALSA, bcm2835 ALSA
+    Direct hardware device without any conversions
+hw:CARD=ALSA,DEV=1
+    bcm2835 ALSA, bcm2835 IEC958/HDMI
+    Direct hardware device without any conversions
+plughw:CARD=ALSA,DEV=0
+    bcm2835 ALSA, bcm2835 ALSA
+    Hardware device with all software conversions
+plughw:CARD=ALSA,DEV=1
+    bcm2835 ALSA, bcm2835 IEC958/HDMI
+    Hardware device with all software conversions
+sysdefault:CARD=seeed8micvoicec
+    seeed-8mic-voicecard,
+    Default Audio Device
+dmix:CARD=seeed8micvoicec,DEV=0
+    seeed-8mic-voicecard,
+    Direct sample mixing device
+dsnoop:CARD=seeed8micvoicec,DEV=0
+    seeed-8mic-voicecard,
+    Direct sample snooping device
+hw:CARD=seeed8micvoicec,DEV=0
+    seeed-8mic-voicecard,
+    Direct hardware device without any conversions
+plughw:CARD=seeed8micvoicec,DEV=0
+    seeed-8mic-voicecard,
+    Hardware device with all software conversions
+```
+
+## 4-Mics Linear Array Kit
+
+[![](https://user-images.githubusercontent.com/3901856/37194106-a0ccebce-23a7-11e8-88c5-ec611e44ec49.png)]()
+
+In contrast to 6-Mics Circular Array Kit for Raspberry Pi,
+the difference is only first 4 input channels are valid capture data.
+
+### Usage:
+```bash
+#for ReSpeaker 2-mic
+#It will capture sound an playback on hw:1
+arecord -f cd -Dhw:1 | aplay -Dhw:1
+```
+
+```bash
+#for ReSpeaker 4-mic
+#It will capture sound on AC108 and save as a.wav
+arecord -Dac108 -f S32_LE -r 16000 -c 4 a.wav
+```
+
+```bash
+#for 6-Mics Circular Array Kit and 4-Mics Linear Array Kit
+#It will capture sound on AC108 and save as a.wav
+arecord -Dac108 -f S32_LE -r 16000 -c 8 a.wav
+#Take care of that the captured mic audio is on the first 6 channels
+
+#It will play sound file a.wav on AC101
+aplay -D ac101 a.wav
+#Do not use -D plughw:1,0 directly except your wave file is single channel only.
+
+#Doing capture && playback the same time
+arecord -D hw:1,0 -f S32_LE -r 16000 -c 8 to_be_record.wav &
+#mono_to_play.wav is a mono channel wave file to play
+aplay -D plughw:1,0 -r 16000 mono_to_play.wav
+```
+**Note: Limit for developer using 6-Mics Circular Array Kit(or 4-Mics Linear Array Kit) doing capture & playback the same time:  
+1. capture must be start first, or else the capture channels will possibly be disorder.  
+2. playback output channels must fill with 8 same channels data or 4 same stereo channels data, or else the speaker or headphone will output nothing possibly.**

 ### Coherence

@ -44,21 +255,7 @@ Thank you!
 ------------------------------------------------------
 ```

+
+
+
 Enjoy !
-
-### Technical support
-
-For hardware testing purposes we made a Rasperry Pi OS 5.10.17-v7l+ 32-bit image with reSpeaker drivers pre-installed, which you can download by clicking on [this link](https://files.seeedstudio.com/linux/Raspberry%20Pi%204%20reSpeaker/2021-05-07-raspios-buster-armhf-lite-respeaker.img.xz).
-
-We provide official support for using reSpeaker with the following OS:
- 32-bit Raspberry Pi OS
- 64-bit Raspberry Pi OS
-
-And following hardware platforms:
- Raspberry Pi 3 (all models), Raspberry Pi 4 (all models)
-
-Anything beyond the scope of official support is considered to be community supported. Support for other OS/hardware platforms can be added, provided MOQ requirements can be met. 
-
-If you have a technical problem when using reSpeaker with one of the officially supported platforms/OS, feel free to create an issue on Github. For general questions or suggestions, please use [Seeed forum](https://forum.seeedstudio.com/c/products/respeaker/15). 
-
-
--- a/ac101.c
+++ b/ac101.c
@ -37,7 +37,6 @@
 #include <linux/gpio/consumer.h>
 #include <linux/regmap.h>
 #include <linux/input.h>
-#include <linux/delay.h>
 #include "ac101_regs.h"
 #include "ac10x.h"

@ -78,8 +77,6 @@ int ac101_read(struct snd_soc_codec *codec, unsigned reg) {
 	int r, v = 0;

 	if ((r = regmap_read(ac10x->regmap101, reg, &v)) < 0) {
-		dev_err(codec->dev, "read reg %02X fail\n",
-			 reg);
 		return r;
 	}
 	return v;
@ -87,20 +84,16 @@ int ac101_read(struct snd_soc_codec *codec, unsigned reg) {

 int ac101_write(struct snd_soc_codec *codec, unsigned reg, unsigned val) {
 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
-	int v;

-	v = regmap_write(ac10x->regmap101, reg, val);
-	return v;
+	return regmap_write(ac10x->regmap101, reg, val);
 }

 int ac101_update_bits(struct snd_soc_codec *codec, unsigned reg,
 			unsigned mask, unsigned value
 ) {
 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
-	int v;

-	v = regmap_update_bits(ac10x->regmap101, reg, mask, value);
-	return v;
+	return regmap_update_bits(ac10x->regmap101, reg, mask, value);
 }


@ -119,7 +112,7 @@ static void switch_hw_config(struct snd_soc_codec *codec)
 {
 	int r;

-	AC101_DBG();
+	AC101_DBG("%s,line:%d\n",__func__,__LINE__);

 	/*HMIC/MMIC BIAS voltage level select:2.5v*/
 	ac101_update_bits(codec, OMIXER_BST1_CTRL, (0xf<<BIASVOLTAGE), (0xf<<BIASVOLTAGE));
@ -162,7 +155,7 @@ static void switch_hw_config(struct snd_soc_codec *codec)
 */
 static void switch_status_update(struct ac10x_priv *ac10x)
 {
-	AC101_DBG("ac10x->state:%d\n", ac10x->state);
+	AC101_DBG("%s,line:%d,ac10x->state:%d\n", __func__, __LINE__, ac10x->state);

        input_report_switch(ac10x->inpdev, SW_HEADPHONE_INSERT, ac10x->state);
        input_sync(ac10x->inpdev);
@ -183,7 +176,7 @@ static void work_cb_clear_irq(struct work_struct *work)
 	reg_val = ac101_read(codec, HMIC_STS);
 	if (BIT(HMIC_PULLOUT_PEND) & reg_val) {
 		ac10x->pullout_cntr++;
-		AC101_DBG("ac10x->pullout_cntr: %d\n", ac10x->pullout_cntr);
+		AC101_DBG("ac10x->pullout_cntr: %d\n",ac10x->pullout_cntr);
 	}

 	reg_val |= HMIC_PEND_ALL;
@ -217,17 +210,17 @@ static int __ac101_get_hmic_data(struct snd_soc_codec *codec) {
 	#ifdef AC101_DEBG
 	static long counter;
 	#endif
-	int r, d;
+	int r;
+	int d;

 	d = GET_HMIC_DATA(ac101_read(codec, HMIC_STS));

 	r = 0x1 << HMIC_DATA_PEND;
 	ac101_write(codec, HMIC_STS, r);

-	/* prevent i2c accessing too frequently */
-	usleep_range(1500, 3000);
-
-	AC101_DBG("HMIC_DATA(%3ld): %02X\n", counter++, d);
+	AC101_DBG("%s,line:%d HMIC_DATA(%3ld): %02X\n", __func__, __LINE__,
+			counter++, d
+		);
 	return d;
 }

@ -262,7 +255,7 @@ static void work_cb_earphone_switch(struct work_struct *work)
 				input_report_key(ac10x->inpdev, KEY_HEADSETHOOK, 1);
 				input_sync(ac10x->inpdev);

-				AC101_DBG("KEY_HEADSETHOOK1\n");
+				AC101_DBG("%s,line:%d KEY_HEADSETHOOK1\n", __func__, __LINE__);

 				if (hook_flag1 != hook_flag2)
 					hook_flag1 = hook_flag2 = 0;
@ -280,7 +273,7 @@ static void work_cb_earphone_switch(struct work_struct *work)
 				input_report_key(ac10x->inpdev, KEY_VOLUMEUP, 0);
 				input_sync(ac10x->inpdev);

-				AC101_DBG("HMIC_DATA: %d KEY_VOLUMEUP\n", t);
+				AC101_DBG("%s,line:%d HMIC_DATA: %d KEY_VOLUMEUP\n", __func__, __LINE__, t);
 			}
 			if (ac10x->pullout_cntr)
 				ac10x->pullout_cntr--;
@ -293,7 +286,7 @@ static void work_cb_earphone_switch(struct work_struct *work)
 				input_sync(ac10x->inpdev);
 				input_report_key(ac10x->inpdev, KEY_VOLUMEDOWN, 0);
 				input_sync(ac10x->inpdev);
-				AC101_DBG("KEY_VOLUMEDOWN\n");
+				AC101_DBG("%s,line:%d KEY_VOLUMEDOWN\n", __func__, __LINE__);
 			}
 			if (ac10x->pullout_cntr)
 				ac10x->pullout_cntr--;
@ -309,7 +302,7 @@ static void work_cb_earphone_switch(struct work_struct *work)
 				input_report_key(ac10x->inpdev, KEY_HEADSETHOOK, 0);
 				input_sync(ac10x->inpdev);

-				AC101_DBG("KEY_HEADSETHOOK0\n");
+				AC101_DBG("%s,line:%d KEY_HEADSETHOOK0\n", __func__, __LINE__);
 			}
 		}
 	} else {
@ -363,7 +356,7 @@ static irqreturn_t audio_hmic_irq(int irq, void *para)

 static int ac101_switch_probe(struct ac10x_priv *ac10x) {
 	struct i2c_client *i2c = ac10x->i2c101;
-	long ret;
+	int ret;

 	ac10x->gpiod_irq = devm_gpiod_get_optional(&i2c->dev, "switch-irq", GPIOD_IN);
 	if (IS_ERR(ac10x->gpiod_irq)) {
@ -376,7 +369,7 @@ static int ac101_switch_probe(struct ac10x_priv *ac10x) {

 	ac10x->irq = gpiod_to_irq(ac10x->gpiod_irq);
 	if (IS_ERR_VALUE(ac10x->irq)) {
-		pr_warn("[ac101] map gpio to irq failed, errno = %ld\n", ac10x->irq);
+		pr_info("[ac101] map gpio to irq failed, errno = %d\n", ac10x->irq);
 		ac10x->irq = 0;
 		goto _err_irq;
 	}
@ -384,7 +377,7 @@ static int ac101_switch_probe(struct ac10x_priv *ac10x) {
 	/* request irq, set irq type to falling edge trigger */
 	ret = devm_request_irq(ac10x->codec->dev, ac10x->irq, audio_hmic_irq, IRQF_TRIGGER_FALLING, "SWTICH_EINT", ac10x);
 	if (IS_ERR_VALUE(ret)) {
-		pr_warn("[ac101] request virq %ld failed, errno = %ld\n", ac10x->irq, ret);
+		pr_info("[ac101] request virq %d failed, errno = %d\n", ac10x->irq, ret);
 		goto _err_irq;
 	}

@ -433,7 +426,7 @@ _err_input_register_device:
 _err_input_allocate_device:

 	if (ac10x->irq) {
-		devm_free_irq(&i2c->dev, ac10x->irq, ac10x);
+		devm_free_irq(&i2c->dev, ac10x->irq, NULL);
 		ac10x->irq = 0;
 	}
 _err_irq:
@ -587,7 +580,7 @@ static int late_enable_dac(struct snd_soc_codec* codec, int event) {
 	mutex_lock(&ac10x->dac_mutex);
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
-		AC101_DBG();
+		AC101_DBG("%s,line:%d\n",__func__,__LINE__);
 		if (ac10x->dac_enable == 0){
 			/*enable dac module clk*/
 			ac101_update_bits(codec, MOD_CLK_ENA, (0x1<<MOD_CLK_DAC_DIG), (0x1<<MOD_CLK_DAC_DIG));
@ -617,7 +610,7 @@ static int ac101_headphone_event(struct snd_soc_codec* codec, int event) {
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
 		/*open*/
-		AC101_DBG("post:open\n");
+		AC101_DBG("post:open:%s,line:%d\n", __func__, __LINE__);
 		ac101_update_bits(codec, OMIXER_DACA_CTRL, (0xf<<HPOUTPUTENABLE), (0xf<<HPOUTPUTENABLE));
 		msleep(10);
 		ac101_update_bits(codec, HPOUT_CTRL, (0x1<<HPPA_EN), (0x1<<HPPA_EN));
@ -625,7 +618,7 @@ static int ac101_headphone_event(struct snd_soc_codec* codec, int event) {
 		break;
 	case SND_SOC_DAPM_PRE_PMD:
 		/*close*/
-		AC101_DBG("pre:close\n");
+		AC101_DBG("pre:close:%s,line:%d\n", __func__, __LINE__);
 		ac101_update_bits(codec, HPOUT_CTRL, (0x3<<LHPPA_MUTE), (0x0<<LHPPA_MUTE));
 		msleep(10);
 		ac101_update_bits(codec, OMIXER_DACA_CTRL, (0xf<<HPOUTPUTENABLE), (0x0<<HPOUTPUTENABLE));
@ -644,7 +637,9 @@ static int ac101_sysclk_started(void) {

 static int ac101_aif1clk(struct snd_soc_codec* codec, int event, int quick) {
 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
-	int ret = 0;
+	int ret;
+
+	/* spin_lock move to machine trigger */

 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
@ -658,9 +653,9 @@ static int ac101_aif1clk(struct snd_soc_codec* codec, int event, int quick) {
 			ret = ret || ac101_update_bits(codec, SYSCLK_CTRL, (0x1<<SYSCLK_ENA), (0x1<<SYSCLK_ENA));

 			if (ret) {
-				AC101_DBG("start sysclk failed\n");
+				AC101_DBG("%s() L%d start sysclk failed\n", __func__, __LINE__);
 			} else {
-				AC101_DBG("hw sysclk enable\n");
+				AC101_DBG("%s() L%d hw sysclk enable\n", __func__, __LINE__);
 				ac10x->aif1_clken++;
 			}
 		}
@ -674,18 +669,19 @@ static int ac101_aif1clk(struct snd_soc_codec* codec, int event, int quick) {
 			ret = ret || ac101_update_bits(codec, SYSCLK_CTRL, (0x1<<SYSCLK_ENA), (0x0<<SYSCLK_ENA));

 			if (ret) {
-				AC101_DBG("stop sysclk failed\n");
+				AC101_DBG("%s() L%d stop sysclk failed\n", __func__, __LINE__);
 			} else {
-				AC101_DBG("hw sysclk disable\n");
+				AC101_DBG("%s() L%d hw sysclk disable\n", __func__, __LINE__);
 				ac10x->aif1_clken = 0;
 			}
 			break;
 		}
 	}

-	AC101_DBG("event=%d pre_up/%d post_down/%d\n", event, SND_SOC_DAPM_PRE_PMU, SND_SOC_DAPM_POST_PMD);
+	AC101_DBG("%s() L%d event=%d pre_up/%d post_down/%d\n", __func__, __LINE__,
+		event, SND_SOC_DAPM_PRE_PMU, SND_SOC_DAPM_POST_PMD);

-	return ret;
+	return 0;
 }

 /**
@ -782,8 +778,8 @@ static struct snd_kcontrol_new ac101_controls[] = {
 	SOC_DOUBLE_TLV("DAC volume", DAC_VOL_CTRL, DAC_VOL_L, DAC_VOL_R, 0xff, 0, dac_vol_tlv),
 	SOC_DOUBLE_TLV("DAC mixer gain", DAC_MXR_GAIN, DACL_MXR_GAIN, DACR_MXR_GAIN, 0xf, 0, dac_mix_vol_tlv),
 	SOC_SINGLE_TLV("digital volume", DAC_DBG_CTRL, DVC, 0x3f, 1, dig_vol_tlv),
-	SOC_SINGLE_TLV("Speaker Playback Volume", SPKOUT_CTRL, SPK_VOL, 0x1f, 0, speaker_vol_tlv),
-	SOC_SINGLE_TLV("Headphone Playback Volume", HPOUT_CTRL, HP_VOL, 0x3f, 0, headphone_vol_tlv),
+	SOC_SINGLE_TLV("speaker volume", SPKOUT_CTRL, SPK_VOL, 0x1f, 0, speaker_vol_tlv),
+	SOC_SINGLE_TLV("headphone volume", HPOUT_CTRL, HP_VOL, 0x3f, 0, headphone_vol_tlv),
 };

 /* PLL divisors */
@ -911,7 +907,7 @@ int ac101_aif_mute(struct snd_soc_dai *codec_dai, int mute)
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);

-	AC101_DBG("mute=%d\n",  mute);
+	AC101_DBG("%s() L%d mute=%d\n", __func__, __LINE__, mute);

 	ac101_write(codec, DAC_VOL_CTRL, mute? 0: 0xA0A0);

@ -953,12 +949,12 @@ void ac101_aif_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);

-	AC101_DBG("stream = %s, play: %d, capt: %d, active: %d\n", 
+	AC101_DBG("%s,line:%d stream = %s, play: %d, capt: %d, active: %d\n", __func__, __LINE__,
 		snd_pcm_stream_str(substream),
-		codec_dai->stream[SNDRV_PCM_STREAM_PLAYBACK].active, codec_dai->stream[SNDRV_PCM_STREAM_CAPTURE].active,
-		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 {
@ -972,7 +968,7 @@ static int ac101_set_pll(struct snd_soc_dai *codec_dai, int pll_id, int source,
 	struct snd_soc_codec *codec = codec_dai->codec;
 	int i, m, n_i, n_f;

-	AC101_DBG("pll_id:%d\n",  pll_id);
+	AC101_DBG("%s, line:%d, pll_id:%d\n", __func__, __LINE__, pll_id);

 	/* clear volatile reserved bits*/
 	ac101_update_bits(codec, SYSCLK_CTRL, 0xFF & ~(0x1 << SYSCLK_ENA), 0x0);
@ -1044,7 +1040,7 @@ int ac101_hw_params(struct snd_pcm_substream *substream,
 	int reg_val, freq_out;
 	unsigned channels;

-	AC101_DBG("+++\n");
+	AC101_DBG("%s() L%d +++\n", __func__, __LINE__);

 	if (_MASTER_MULTI_CODEC == _MASTER_AC101 && ac101_sysclk_started()) {
 		/* not configure hw_param twice if stream is playback, tell the caller it's started */
@ -1080,7 +1076,7 @@ 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[SNDRV_PCM_STREAM_CAPTURE].active && 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));
@ -1130,7 +1126,7 @@ int ac101_hw_params(struct snd_pcm_substream *substream,
 	AC101_DBG("rate: %d , channels: %d , samp_res: %d",
 		params_rate(params), channels, aif1_slot_size);

-	AC101_DBG("---\n");
+	AC101_DBG("%s() L%d ---\n", __func__, __LINE__);
 	return 0;
 }

@ -1140,7 +1136,7 @@ int ac101_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	int AIF_CLK_CTRL = AIF1_CLK_CTRL;
 	struct snd_soc_codec *codec = codec_dai->codec;

-	AC101_DBG();
+	AC101_DBG("%s() L%d\n", __func__, __LINE__);

 	/*
 	 * 	master or slave selection
@ -1152,14 +1148,14 @@ int ac101_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	switch(fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBM_CFM:   /* codec clk & frm master, ap is slave*/
 		#if _MASTER_MULTI_CODEC == _MASTER_AC101
-		pr_info("AC101 as Master\n");
+		pr_warn("AC101 as Master\n");
 		reg_val |= (0x0<<AIF1_MSTR_MOD);
 		break;
 		#else
-		pr_info("AC108 as Master\n");
+		pr_warn("AC108 as Master\n");
 		#endif
 	case SND_SOC_DAIFMT_CBS_CFS:   /* codec clk & frm slave, ap is master*/
-		pr_info("AC101 as Slave\n");
+		pr_warn("AC101 as Slave\n");
 		reg_val |= (0x1<<AIF1_MSTR_MOD);
 		break;
 	default:
@ -1229,7 +1225,7 @@ int ac101_audio_startup(struct snd_pcm_substream *substream,
 {
 	// struct snd_soc_codec *codec = codec_dai->codec;

-	AC101_DBG("\n\n\n");
+	AC101_DBG("\n\n%s,line:%d\n", __func__, __LINE__);

 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 	}
@ -1237,18 +1233,16 @@ int ac101_audio_startup(struct snd_pcm_substream *substream,
 }

 #if _MASTER_MULTI_CODEC == _MASTER_AC101
-static int ac101_set_clock(int y_start_n_stop, struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) {
-	int r;
-
+static int ac101_set_clock(int y_start_n_stop) {
 	if (y_start_n_stop) {
 		/* enable global clock */
-		r = ac101_aif1clk(static_ac10x->codec, SND_SOC_DAPM_PRE_PMU, 1);
+		ac101_aif1clk(static_ac10x->codec, SND_SOC_DAPM_PRE_PMU, 1);
 	} else {
 		/* disable global clock */
 		static_ac10x->aif1_clken = 1;
-		r = ac101_aif1clk(static_ac10x->codec, SND_SOC_DAPM_POST_PMD, 0);
+		ac101_aif1clk(static_ac10x->codec, SND_SOC_DAPM_POST_PMD, 0);
 	}
-	return r;
+	return 0;
 }
 #endif

@ -1258,9 +1252,9 @@ int ac101_trigger(struct snd_pcm_substream *substream, int cmd,
 	struct snd_soc_codec *codec = dai->codec;
 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
 	int ret = 0;
-	unsigned long flags;

-	AC101_DBG("stream=%s  cmd=%d\n",
+	AC101_DBG("%s() stream=%s  cmd=%d\n",
+		__FUNCTION__,
 		snd_pcm_stream_str(substream),
 		cmd);

@ -1269,7 +1263,6 @@ int ac101_trigger(struct snd_pcm_substream *substream, int cmd,
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		#if _MASTER_MULTI_CODEC == _MASTER_AC101
-		spin_lock_irqsave(&ac10x->lock, flags);
 		if (ac10x->aif1_clken == 0){
 			/*
 			 * enable aif1clk, it' here due to reduce time between 'AC108 Sysclk Enable' and 'AC101 Sysclk Enable'
@ -1279,21 +1272,15 @@ int ac101_trigger(struct snd_pcm_substream *substream, int cmd,
 			ret = ret || ac101_update_bits(codec, MOD_CLK_ENA, (0x1<<MOD_CLK_AIF1), (0x1<<MOD_CLK_AIF1));
 			ret = ret || ac101_update_bits(codec, MOD_RST_CTRL, (0x1<<MOD_RESET_AIF1), (0x1<<MOD_RESET_AIF1));
 		}
-		spin_unlock_irqrestore(&ac10x->lock, flags);
-		ac101_set_clock(1, substream, cmd, dai);
 		#endif
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		ac101_set_clock(0, NULL, 0, NULL);
 		break;
 	default:
 		ret = -EINVAL;
 	}
-	AC101_DBG("stream=%s  cmd=%d;finished %d\n",
-		snd_pcm_stream_str(substream),
-		cmd, ret);
 	return ret;
 }

@ -1304,7 +1291,7 @@ static int ac101_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);

-	AC101_DBG("id=%d freq=%d, dir=%d\n", 
+	AC101_DBG("%s,line:%d, id=%d freq=%d, dir=%d\n", __func__, __LINE__,
 		clk_id, freq, dir);

 	ac10x->sysclk = freq;
@ -1353,7 +1340,7 @@ static void codec_resume_work(struct work_struct *work)
 	struct ac10x_priv *ac10x = container_of(work, struct ac10x_priv, codec_resume);
 	struct snd_soc_codec *codec = ac10x->codec;

-	AC101_DBG("+++\n");
+	AC101_DBG("%s() L%d +++\n", __func__, __LINE__);

 	set_configuration(codec);
 	if (drc_used) {
@ -1362,7 +1349,7 @@ static void codec_resume_work(struct work_struct *work)
 	/*enable this bit to prevent leakage from ldoin*/
 	ac101_update_bits(codec, ADDA_TUNE3, (0x1<<OSCEN), (0x1<<OSCEN));

-	AC101_DBG("---\n");
+	AC101_DBG("%s() L%d +++\n", __func__, __LINE__);
 	return;
 }

@ -1370,13 +1357,13 @@ int ac101_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level le
 {
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		AC101_DBG("SND_SOC_BIAS_ON\n");
+		AC101_DBG("%s,line:%d, SND_SOC_BIAS_ON\n", __func__, __LINE__);
 		break;
 	case SND_SOC_BIAS_PREPARE:
-		AC101_DBG("SND_SOC_BIAS_PREPARE\n");
+		AC101_DBG("%s,line:%d, SND_SOC_BIAS_PREPARE\n", __func__, __LINE__);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		AC101_DBG("SND_SOC_BIAS_STANDBY\n");
+		AC101_DBG("%s,line:%d, SND_SOC_BIAS_STANDBY\n", __func__, __LINE__);
 		#ifdef CONFIG_AC101_SWITCH_DETECT
 		switch_hw_config(codec);
 		#endif
@ -1388,7 +1375,7 @@ int ac101_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level le
 		#endif
 		ac101_update_bits(codec, OMIXER_DACA_CTRL, (0xf<<HPOUTPUTENABLE), (0<<HPOUTPUTENABLE));
 		ac101_update_bits(codec, ADDA_TUNE3, (0x1<<OSCEN), (0<<OSCEN));
-		AC101_DBG("SND_SOC_BIAS_OFF\n");
+		AC101_DBG("%s,line:%d, SND_SOC_BIAS_OFF\n", __func__, __LINE__);
 		break;
 	}
 	snd_soc_codec_get_dapm(codec)->bias_level = level;
@ -1402,7 +1389,7 @@ int ac101_codec_probe(struct snd_soc_codec *codec)

 	ac10x = dev_get_drvdata(codec->dev);
 	if (ac10x == NULL) {
-		AC101_DBG("not set client data!\n");
+		AC101_DBG("not set client data %s() L%d\n", __func__, __LINE__);
 		return -ENOMEM;
 	}
 	ac10x->codec = codec;
@ -1453,20 +1440,12 @@ int ac101_codec_remove(struct snd_soc_codec *codec)
 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);

 	if (ac10x->irq) {
-		devm_free_irq(codec->dev, ac10x->irq, ac10x);
+		/* devm_free_irq(codec->dev, ac10x->irq, NULL); */
 		ac10x->irq = 0;
 	}

 	if (cancel_work_sync(&ac10x->work_switch) != 0) {
 	}
-
-	if (cancel_work_sync(&ac10x->work_clear_irq) != 0) {
-	}
-
-	if (ac10x->inpdev) {
-		input_unregister_device(ac10x->inpdev);
-		ac10x->inpdev = NULL;
-	}
 	#endif

 	return 0;
@ -1511,32 +1490,32 @@ int ac101_codec_resume(struct snd_soc_codec *codec)
 static ssize_t ac101_debug_store(struct device *dev,
 	struct device_attribute *attr, const char *buf, size_t count)
 {
+	static int val = 0, flag = 0;
+	u8 reg,num,i=0;
+	u16 value_w,value_r[128];
 	struct ac10x_priv *ac10x = dev_get_drvdata(dev);
-	int val = 0, flag = 0;
-	u16 value_w, value_r;
-	u8 reg, num, i=0;
-
 	val = simple_strtol(buf, NULL, 16);
 	flag = (val >> 24) & 0xF;
-	if (flag) {
+	if(flag) {
 		reg = (val >> 16) & 0xFF;
 		value_w =  val & 0xFFFF;
 		ac101_write(ac10x->codec, reg, value_w);
-		printk("write 0x%x to reg:0x%x\n", value_w, reg);
+		printk("write 0x%x to reg:0x%x\n",value_w,reg);
 	} else {
-		reg = (val >> 8) & 0xFF;
-		num = val & 0xff;
+		reg =(val>>8)& 0xFF;
+		num=val&0xff;
 		printk("\n");
-		printk("read:start add:0x%x,count:0x%x\n", reg, num);
-
-		regcache_cache_bypass(ac10x->regmap101, true);
+		printk("read:start add:0x%x,count:0x%x\n",reg,num);
 		do {
-			value_r = ac101_read(ac10x->codec, reg);
-			printk("0x%x: 0x%04x ", reg++, value_r);
-			if (++i % 4 == 0 || i == num)
+			value_r[i] = ac101_read(ac10x->codec, reg);
+			printk("0x%x: 0x%04x ",reg,value_r[i]);
+			reg+=1;
+			i++;
+			if(i == num)
 				printk("\n");
-		} while (i < num);
-		regcache_cache_bypass(ac10x->regmap101, false);
+			if(i%4==0)
+				printk("\n");
+		} while(i<num);
 	}
 	return count;
 }
@ -1612,7 +1591,7 @@ int ac101_probe(struct i2c_client *i2c, const struct i2c_device_id *id)
 	int ret = 0;
 	unsigned v = 0;

-	AC101_DBG();
+	AC101_DBG("%s,line:%d\n", __func__, __LINE__);

 	static_ac10x = ac10x;

@ -1623,13 +1602,13 @@ int ac101_probe(struct i2c_client *i2c, const struct i2c_device_id *id)
 		return ret;
 	}

+	ac10x_fill_regcache(&i2c->dev, ac10x->regmap101);
+
 	/* Chip reset */
-	regcache_cache_only(ac10x->regmap101, false);
+	/*
 	ret = regmap_write(ac10x->regmap101, CHIP_AUDIO_RST, 0);
 	msleep(50);
-
-	/* sync regcache for FLAT type */
-	ac10x_fill_regcache(&i2c->dev, ac10x->regmap101);
+	*/

 	ret = regmap_read(ac10x->regmap101, CHIP_AUDIO_RST, &v);
 	if (ret < 0) {
@ -1664,7 +1643,7 @@ void ac101_shutdown(struct i2c_client *i2c)
 	int reg_val;

 	if (codec == NULL) {
-		pr_err(": no sound card.\n");
+		pr_err("%s() L%d: no sound card.\n", __func__, __LINE__);
 		return;
 	}

--- a/ac108.c
+++ b/ac108.c
@ -173,11 +173,8 @@ static const DECLARE_TLV_DB_SCALE(tlv_ch_digital_vol, -11925,75,0);
 int ac10x_read(u8 reg, u8* rt_val, struct regmap* i2cm) {
 	int r, v = 0;

-	if ((r = regmap_read(i2cm, reg, &v)) < 0) {
-		pr_err("ac10x_read error->[REG-0x%02x]\n", reg);
-	} else {
-		*rt_val = v;
-	}
+	r = regmap_read(i2cm, reg, &v);
+	*rt_val = v;
 	return r;
 }

@ -453,13 +450,11 @@ static int ac108_multi_write(u8 reg, u8 val, struct ac10x_priv *ac10x) {
 }

 static int ac108_multi_update_bits(u8 reg, u8 mask, u8 val, struct ac10x_priv *ac10x) {
-	int r = 0;
 	u8 i;
-
 	for (i = 0; i < ac10x->codec_cnt; i++) {
-		r |= ac10x_update_bits(reg, mask, val, ac10x->i2cmap[i]);
+		ac10x_update_bits(reg, mask, val, ac10x->i2cmap[i]);
 	}
-	return r;
+	return 0;
 }

 static unsigned int ac108_codec_read(struct snd_soc_codec *codec, unsigned int reg) {
@ -653,7 +648,7 @@ 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[SNDRV_PCM_STREAM_PLAYBACK].active, dai->stream[SNDRV_PCM_STREAM_CAPTURE].active);
+			dai->playback_active, dai->capture_active);

 	if (ac10x->i2c101) {
 		ret = ac101_hw_params(substream, params, dai);
@ -664,8 +659,8 @@ static int ac108_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_h
 		}
 	}

-	if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE && dai->stream[SNDRV_PCM_STREAM_PLAYBACK].active)
-	 || (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && dai->stream[SNDRV_PCM_STREAM_CAPTURE].active)) {
+	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 +805,6 @@ 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;

@ -868,7 +860,6 @@ static int ac108_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) {
 			/* TODO: Both cpu_dai and codec_dai(AC108) be set as slave in DTS */
 			dev_dbg(dai->dev, "used as slave when AC101 is master\n");
 		}
-		fallthrough;
 	case SND_SOC_DAIFMT_CBS_CFS:    /*AC108 Slave*/
 		dev_dbg(dai->dev, "AC108 set to work as Slave\n");
 		/**
@ -991,50 +982,47 @@ static int ac108_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) {
 /*
 * due to miss channels order in cpu_dai, we meed defer the clock starting.
 */
-static int ac108_set_clock(int y_start_n_stop, struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) {
-	u8 reg;
-	int ret = 0;
+static int ac108_set_clock(int y_start_n_stop) {
+	u8 r;

 	dev_dbg(ac10x->codec->dev, "%s() L%d cmd:%d\n", __func__, __LINE__, y_start_n_stop);

 	/* spin_lock move to machine trigger */

-	if (ac10x->i2c101 && _MASTER_MULTI_CODEC == _MASTER_AC101) {
-		ac101_trigger(substream, cmd, dai);
-	}
-	if (y_start_n_stop && ac10x->sysclk_en == 0) {
+	if (y_start_n_stop)  {
+		if (ac10x->sysclk_en == 0) {
+
 		/* enable lrck clock */
-		ac10x_read(I2S_CTRL, &reg, ac10x->i2cmap[_MASTER_INDEX]);
-		if (reg & (0x01 << BCLK_IOEN)) {
-			ret = ret || ac10x_update_bits(I2S_CTRL, 0x03 << LRCK_IOEN, 0x03 << LRCK_IOEN, ac10x->i2cmap[_MASTER_INDEX]);
+		ac10x_read(I2S_CTRL, &r, ac10x->i2cmap[_MASTER_INDEX]);
+		if (r & (0x01 << BCLK_IOEN)) {
+			ac10x_update_bits(I2S_CTRL, 0x03 << LRCK_IOEN, 0x03 << LRCK_IOEN, ac10x->i2cmap[_MASTER_INDEX]);
 		}

 		/*0x10: PLL Common voltage enable, PLL enable */
-		ret = ret || ac108_multi_update_bits(PLL_CTRL1, 0x01 << PLL_EN | 0x01 << PLL_COM_EN,
+		ac108_multi_update_bits(PLL_CTRL1, 0x01 << PLL_EN | 0x01 << PLL_COM_EN,
 						   0x01 << PLL_EN | 0x01 << PLL_COM_EN, ac10x);
 		/* enable global clock */
-		ret = ret || ac108_multi_update_bits(I2S_CTRL, 0x1 << TXEN | 0x1 << GEN, 0x1 << TXEN | 0x1 << GEN, ac10x);
+		ac108_multi_update_bits(I2S_CTRL, 0x1 << TXEN | 0x1 << GEN, 0x1 << TXEN | 0x1 << GEN, ac10x);

 		ac10x->sysclk_en = 1UL;
-	} else if (!y_start_n_stop && ac10x->sysclk_en != 0) {
+		}
+	} else if (ac10x->sysclk_en != 0) {
 		/* disable global clock */
-		ret = ret || ac108_multi_update_bits(I2S_CTRL, 0x1 << TXEN | 0x1 << GEN, 0x0 << TXEN | 0x0 << GEN, ac10x);
+		ac108_multi_update_bits(I2S_CTRL, 0x1 << TXEN | 0x1 << GEN, 0x0 << TXEN | 0x0 << GEN, ac10x);

 		/*0x10: PLL Common voltage disable, PLL disable */
-		ret = ret || ac108_multi_update_bits(PLL_CTRL1, 0x01 << PLL_EN | 0x01 << PLL_COM_EN,
+		ac108_multi_update_bits(PLL_CTRL1, 0x01 << PLL_EN | 0x01 << PLL_COM_EN,
 						   0x00 << PLL_EN | 0x00 << PLL_COM_EN, ac10x);

 		/* disable lrck clock if it's enabled */
-		ac10x_read(I2S_CTRL, &reg, ac10x->i2cmap[_MASTER_INDEX]);
-		if (reg & (0x01 << LRCK_IOEN)) {
-			ret = ret || ac10x_update_bits(I2S_CTRL, 0x03 << LRCK_IOEN, 0x01 << BCLK_IOEN, ac10x->i2cmap[_MASTER_INDEX]);
-		}
-		if (!ret) {
-			ac10x->sysclk_en = 0UL;
+		ac10x_read(I2S_CTRL, &r, ac10x->i2cmap[_MASTER_INDEX]);
+		if (r & (0x01 << LRCK_IOEN)) {
+			ac10x_update_bits(I2S_CTRL, 0x03 << LRCK_IOEN, 0x01 << BCLK_IOEN, ac10x->i2cmap[_MASTER_INDEX]);
 		}
+		ac10x->sysclk_en = 0UL;
 	}

-	return ret;
+	return 0;
 }

 static int ac108_prepare(struct snd_pcm_substream *substream,
@ -1061,33 +1049,39 @@ static int ac108_trigger(struct snd_pcm_substream *substream, int cmd,
 		snd_pcm_stream_str(substream),
 		cmd);

+	spin_lock_irqsave(&ac10x->lock, flags);
+
+	if (ac10x->i2c101 && _MASTER_MULTI_CODEC == _MASTER_AC101) {
+		ac101_trigger(substream, cmd, dai);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		       goto __ret;
+		}
+	}
+
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		spin_lock_irqsave(&ac10x->lock, flags);
 		/* disable global clock if lrck disabled */
 		ac10x_read(I2S_CTRL, &r, ac10x->i2cmap[_MASTER_INDEX]);
 		if ((r & (0x01 << BCLK_IOEN)) && (r & (0x01 << LRCK_IOEN)) == 0) {
 			/* disable global clock */
 			ac108_multi_update_bits(I2S_CTRL, 0x1 << TXEN | 0x1 << GEN, 0x0 << TXEN | 0x0 << GEN, ac10x);
 		}
-		spin_unlock_irqrestore(&ac10x->lock, flags);
-		ac108_set_clock(1, substream, cmd, dai);
+
+		/* delayed clock starting, move to machine trigger() */
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		ac108_set_clock(0, substream, cmd, dai);
 		break;
 	default:
 		ret = -EINVAL;
 	}

-	dev_dbg(dai->dev, "%s() stream=%s  cmd=%d; finished %d\n",
-		__FUNCTION__,
-		snd_pcm_stream_str(substream),
-		cmd, ret);
+__ret:
+	spin_unlock_irqrestore(&ac10x->lock, flags);
+
 	return ret;
 }

@ -1121,7 +1115,7 @@ 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);

@ -1142,13 +1136,12 @@ 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 = {
@ -1218,6 +1211,7 @@ static int ac108_add_widgets(struct snd_soc_codec *codec) {
 }

 static int ac108_codec_probe(struct snd_soc_codec *codec) {
+
 	spin_lock_init(&ac10x->lock);

 	ac10x->codec = codec;
@ -1273,12 +1267,6 @@ int ac108_codec_remove(struct snd_soc_codec *codec) {
 	}
 	return ac101_codec_remove(codec);
 }
-#if __NO_SND_SOC_CODEC_DRV
-void ac108_codec_remove_void(struct snd_soc_codec *codec) {
-	ac108_codec_remove(codec);
-}
-#define ac108_codec_remove ac108_codec_remove_void
-#endif

 int ac108_codec_suspend(struct snd_soc_codec *codec) {
 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
@ -1322,11 +1310,6 @@ static struct snd_soc_codec_driver ac10x_soc_codec_driver = {
 	.set_bias_level = ac108_set_bias_level,
 	.read		= ac108_codec_read,
 	.write		= ac108_codec_write,
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(4,17,0)
-	.idle_bias_on 	= 1,
-	.use_pmdown_time 	= 1,
-	.endianness 	= 1,
-#endif
 };

 static ssize_t ac108_store(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) {
@ -1342,16 +1325,12 @@ static ssize_t ac108_store(struct device *dev, struct device_attribute *attr, co
 		ac108_multi_write(reg, value_w, ac10x);
 		printk("Write 0x%02x to REG:0x%02x\n", value_w, reg);
 	} else {
-		int k;
-
 		reg = (val >> 8) & 0xFF;
 		num = val & 0xff;
 		printk("\nRead: start REG:0x%02x,count:0x%02x\n", reg, num);

-		for (k = 0; k < ac10x->codec_cnt; k++) {
-			regcache_cache_bypass(ac10x->i2cmap[k], true);
-		}
 		do {
+			int k;

 			memset(value_r, 0, sizeof value_r);

@ -1369,9 +1348,6 @@ static ssize_t ac108_store(struct device *dev, struct device_attribute *attr, co
 				printk("\n");
 			}
 		} while (i < num);
-		for (k = 0; k < ac10x->codec_cnt; k++) {
-			regcache_cache_bypass(ac10x->i2cmap[k], false);
-		}
 	}

 	return count;
@ -1407,15 +1383,7 @@ static const struct regmap_config ac108_regmap = {
 	.max_register = 0xDF,
 	.cache_type = REGCACHE_FLAT,
 };
-static const struct i2c_device_id ac108_i2c_id[] = {
-	{ "ac108_0", 0 },
-	{ "ac108_1", 1 },
-	{ "ac108_2", 2 },
-	{ "ac108_3", 3 },
-	{ "ac101", AC101_I2C_ID },
-	{ }
-};
-static int ac108_i2c_probe(struct i2c_client *i2c) {
+static int ac108_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *i2c_id) {
 	struct device_node *np = i2c->dev.of_node;
 	unsigned int val = 0;
 	int ret = 0, index;
@ -1428,11 +1396,11 @@ static int ac108_i2c_probe(struct i2c_client *i2c) {
 		}
 	}

-	index = (int)i2c_match_id(ac108_i2c_id, i2c)->driver_data;
+	index = (int)i2c_id->driver_data;
 	if (index == AC101_I2C_ID) {
 		ac10x->i2c101 = i2c;
 		i2c_set_clientdata(i2c, ac10x);
-		ret = ac101_probe(i2c, i2c_match_id(ac108_i2c_id, i2c));
+		ret = ac101_probe(i2c, i2c_id);
 		if (ret) {
 			ac10x->i2c101 = NULL;
 			return ret;
@ -1450,8 +1418,8 @@ static int ac108_i2c_probe(struct i2c_client *i2c) {
 	if (of_property_read_u32(np, "tdm-chips-count", &val)) val = 1;
 	ac10x->tdm_chips_cnt = val;

-	pr_info(" ac10x i2c_id number: %d\n", index);
-	pr_info(" ac10x data protocol: %d\n", ac10x->data_protocol);
+	pr_err(" ac10x i2c_id number: %d\n", index);
+	pr_err(" ac10x data protocol: %d\n", ac10x->data_protocol);

 	ac10x->i2c[index] = i2c;
 	ac10x->i2cmap[index] = devm_regmap_init_i2c(i2c, &ac108_regmap);
@ -1461,19 +1429,17 @@ static int ac108_i2c_probe(struct i2c_client *i2c) {
 		return ret;
 	}

+	ac10x_fill_regcache(&i2c->dev, ac10x->i2cmap[index]);
+
 	/*
 	 * Writing this register with 0x12 
 	 * will resets all register to their default state.
 	 */
-	regcache_cache_only(ac10x->i2cmap[index], false);
 	ret = regmap_write(ac10x->i2cmap[index], CHIP_RST, CHIP_RST_VAL);
 	msleep(1);

-	/* sync regcache for FLAT type */
-	ac10x_fill_regcache(&i2c->dev, ac10x->i2cmap[index]);
-
 	ac10x->codec_cnt++;
-	pr_info(" ac10x codec count  : %d\n", ac10x->codec_cnt);
+	pr_err(" ac10x codec count  : %d\n", ac10x->codec_cnt);

 	ret = sysfs_create_group(&i2c->dev.kobj, &ac108_debug_attr_group);
 	if (ret) {
@ -1498,7 +1464,7 @@ __ret:
 	return ret;
 }

-static void ac108_i2c_remove(struct i2c_client *i2c) {
+static int ac108_i2c_remove(struct i2c_client *i2c) {
 	if (ac10x->codec != NULL) {
 		snd_soc_unregister_codec(&ac10x->i2c[_MASTER_INDEX]->dev);
 		ac10x->codec = NULL;
@ -1523,8 +1489,17 @@ __ret:
 		kfree(ac10x);
 		ac10x = NULL;
 	}
+	return 0;
 }

+static const struct i2c_device_id ac108_i2c_id[] = {
+	{ "ac108_0", 0 },
+	{ "ac108_1", 1 },
+	{ "ac108_2", 2 },
+	{ "ac108_3", 3 },
+	{ "ac101", AC101_I2C_ID },
+	{ }
+};
 MODULE_DEVICE_TABLE(i2c, ac108_i2c_id);

 static const struct of_device_id ac108_of_match[] = {
--- a/ac108_6mic.state
+++ b/ac108_6mic.state
@ -69,7 +69,7 @@ state.seeed8micvoicec {
 	control.1 {
 		iface MIXER
 		name 'CH1 digital volume'
-		value 208
+		value 166
 		comment {
 			access 'read write'
 			type INTEGER
@ -77,13 +77,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 3675
+			dbvalue.0 525
 		}
 	}
 	control.2 {
 		iface MIXER
 		name 'CH2 digital volume'
-		value 208
+		value 166
 		comment {
 			access 'read write'
 			type INTEGER
@ -91,13 +91,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 3675
+			dbvalue.0 525
 		}
 	}
 	control.3 {
 		iface MIXER
 		name 'CH3 digital volume'
-		value 208
+		value 166
 		comment {
 			access 'read write'
 			type INTEGER
@ -105,13 +105,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 3675
+			dbvalue.0 525
 		}
 	}
 	control.4 {
 		iface MIXER
 		name 'CH4 digital volume'
-		value 208
+		value 166
 		comment {
 			access 'read write'
 			type INTEGER
@ -119,13 +119,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 3675
+			dbvalue.0 525
 		}
 	}
 	control.5 {
 		iface MIXER
 		name 'ADC1 PGA gain'
-		value 0
+		value 25
 		comment {
 			access 'read write'
 			type INTEGER
@ -133,13 +133,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 0
+			dbvalue.0 2500
 		}
 	}
 	control.6 {
 		iface MIXER
 		name 'ADC2 PGA gain'
-		value 0
+		value 25
 		comment {
 			access 'read write'
 			type INTEGER
@ -147,13 +147,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 0
+			dbvalue.0 2500
 		}
 	}
 	control.7 {
 		iface MIXER
 		name 'ADC3 PGA gain'
-		value 0
+		value 25
 		comment {
 			access 'read write'
 			type INTEGER
@ -161,13 +161,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 0
+			dbvalue.0 2500
 		}
 	}
 	control.8 {
 		iface MIXER
 		name 'ADC4 PGA gain'
-		value 0
+		value 25
 		comment {
 			access 'read write'
 			type INTEGER
@ -175,13 +175,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 0
+			dbvalue.0 2500
 		}
 	}
 	control.9 {
 		iface MIXER
 		name 'CH5 digital volume'
-		value 208
+		value 165
 		comment {
 			access 'read write'
 			type INTEGER
@ -189,13 +189,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 3675
+			dbvalue.0 450
 		}
 	}
 	control.10 {
 		iface MIXER
 		name 'CH6 digital volume'
-		value 208
+		value 165
 		comment {
 			access 'read write'
 			type INTEGER
@ -203,13 +203,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 3675
+			dbvalue.0 450
 		}
 	}
 	control.11 {
 		iface MIXER
 		name 'CH7 digital volume'
-		value 198
+		value 165
 		comment {
 			access 'read write'
 			type INTEGER
@ -217,13 +217,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 2925
+			dbvalue.0 450
 		}
 	}
 	control.12 {
 		iface MIXER
 		name 'CH8 digital volume'
-		value 198
+		value 165
 		comment {
 			access 'read write'
 			type INTEGER
@ -231,13 +231,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 2925
+			dbvalue.0 450
 		}
 	}
 	control.13 {
 		iface MIXER
 		name 'ADC5 PGA gain'
-		value 0
+		value 25
 		comment {
 			access 'read write'
 			type INTEGER
@ -245,13 +245,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 0
+			dbvalue.0 2500
 		}
 	}
 	control.14 {
 		iface MIXER
 		name 'ADC6 PGA gain'
-		value 0
+		value 25
 		comment {
 			access 'read write'
 			type INTEGER
@ -259,13 +259,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 0
+			dbvalue.0 2500
 		}
 	}
 	control.15 {
 		iface MIXER
 		name 'ADC7 PGA gain'
-		value 0
+		value 25
 		comment {
 			access 'read write'
 			type INTEGER
@ -273,13 +273,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 0
+			dbvalue.0 2500
 		}
 	}
 	control.16 {
 		iface MIXER
 		name 'ADC8 PGA gain'
-		value 0
+		value 25
 		comment {
 			access 'read write'
 			type INTEGER
@ -287,7 +287,7 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 0
+			dbvalue.0 2500
 		}
 	}
 	control.17 {
@ -325,7 +325,7 @@ state.seeed8micvoicec {
 	control.19 {
 		iface MIXER
 		name 'digital volume'
-		value 51
+		value 49
 		comment {
 			access 'read write'
 			type INTEGER
@ -333,13 +333,13 @@ state.seeed8micvoicec {
 			range '0 - 63'
 			dbmin -7308
 			dbmax 0
-			dbvalue.0 -1392
+			dbvalue.0 -1624
 		}
 	}
 	control.20 {
 		iface MIXER
-		name 'Speaker Playback Volume'
-		value 25
+		name 'speaker volume'
+		value 18
 		comment {
 			access 'read write'
 			type INTEGER
@ -347,13 +347,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin -4800
 			dbmax -150
-			dbvalue.0 -1050
+			dbvalue.0 -2100
 		}
 	}
 	control.21 {
 		iface MIXER
-		name 'Headphone Playback Volume'
-		value 52
+		name 'headphone volume'
+		value 48
 		comment {
 			access 'read write'
 			type INTEGER
@ -361,7 +361,7 @@ state.seeed8micvoicec {
 			range '0 - 63'
 			dbmin -6300
 			dbmax 0
-			dbvalue.0 -1100
+			dbvalue.0 -1500
 		}
 	}
 }
--- a/ac108_asound.state
+++ b/ac108_asound.state
@ -69,7 +69,7 @@ state.seeed4micvoicec {
 	control.1 {
 		iface MIXER
 		name 'CH1 digital volume'
-		value 222
+		value 172
 		comment {
 			access 'read write'
 			type INTEGER
@ -77,13 +77,13 @@ state.seeed4micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 4725
+			dbvalue.0 975
 		}
 	}
 	control.2 {
 		iface MIXER
 		name 'CH2 digital volume'
-		value 222
+		value 172
 		comment {
 			access 'read write'
 			type INTEGER
@ -91,13 +91,13 @@ state.seeed4micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 4725
+			dbvalue.0 975
 		}
 	}
 	control.3 {
 		iface MIXER
 		name 'CH3 digital volume'
-		value 222
+		value 172
 		comment {
 			access 'read write'
 			type INTEGER
@ -105,13 +105,13 @@ state.seeed4micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 4725
+			dbvalue.0 975
 		}
 	}
 	control.4 {
 		iface MIXER
 		name 'CH4 digital volume'
-		value 222
+		value 172
 		comment {
 			access 'read write'
 			type INTEGER
@ -119,13 +119,13 @@ state.seeed4micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 4725
+			dbvalue.0 975
 		}
 	}
 	control.5 {
 		iface MIXER
 		name 'ADC1 PGA gain'
-		value 0
+		value 31
 		comment {
 			access 'read write'
 			type INTEGER
@ -133,13 +133,13 @@ state.seeed4micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 0
+			dbvalue.0 3100
 		}
 	}
 	control.6 {
 		iface MIXER
 		name 'ADC2 PGA gain'
-		value 0
+		value 31
 		comment {
 			access 'read write'
 			type INTEGER
@ -147,13 +147,13 @@ state.seeed4micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 0
+			dbvalue.0 3100
 		}
 	}
 	control.7 {
 		iface MIXER
 		name 'ADC3 PGA gain'
-		value 0
+		value 31
 		comment {
 			access 'read write'
 			type INTEGER
@ -161,13 +161,13 @@ state.seeed4micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 0
+			dbvalue.0 3100
 		}
 	}
 	control.8 {
 		iface MIXER
 		name 'ADC4 PGA gain'
-		value 0
+		value 31
 		comment {
 			access 'read write'
 			type INTEGER
@ -175,7 +175,7 @@ state.seeed4micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 0
+			dbvalue.0 3100
 		}
 	}
 }
--- a/ac10x.h
+++ b/ac10x.h
@ -29,19 +29,14 @@
 /* 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)
+    #define AC101_DBG(format,args...)  printk("[AC101] "format,##args)
 #else
    #define AC101_DBG(...)
 #endif


-#include "sound-compatible-4.18.h"
-
 #ifdef CONFIG_AC101_SWITCH_DETECT
 enum headphone_mode_u {
 	HEADPHONE_IDLE,
@ -80,7 +75,7 @@ struct ac10x_priv {

 	#ifdef CONFIG_AC101_SWITCH_DETECT
 	struct gpio_desc* gpiod_irq;
-	long irq;
+	int irq;
 	volatile int irq_cntr;
 	volatile int pullout_cntr;
 	volatile int state;
@ -119,7 +114,7 @@ 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 seeed_voice_card_register_set_clock(int stream, int (*set_clock)(int));

 int ac10x_fill_regcache(struct device* dev, struct regmap* map);

--- a/asound_2mic.conf
+++ b/asound_2mic.conf
@ -1,10 +1,6 @@
 # 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"
@ -23,14 +19,14 @@ pcm.capture {

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

 pcm.array {
    type dsnoop
    slave {
-        pcm "hw:seeed2micvoicec"
+        pcm "hw:0,0"
        channels 2
    }
    ipc_key 666666
--- a/asound_4mic.conf
+++ b/asound_4mic.conf
@ -1,9 +1,6 @@
 # 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"
@ -12,22 +9,22 @@ pcm.!default {

 pcm.playback {
    type plug
-    slave.pcm "hw:ALSA"
+    slave.pcm "hw:0,0"
 }

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

 pcm.ac108 {
    type plug
-    slave.pcm "hw:seeed4micvoicec"
+    slave.pcm "multiapps"
 }

-# pcm.multiapps {
-#     type dsnoop
-#     ac108-slavepcm "hw:1,0"
-#     ipc_key 666666
-# }
+pcm.multiapps {
+    type dsnoop
+    ac108-slavepcm "hw:1,0"
+    ipc_key 666666
+}
--- a/asound_6mic.conf
+++ b/asound_6mic.conf
@ -1,90 +1,61 @@
 # 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"
+    playback.pcm "ac101"
    capture.pcm "ac108"
 }

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

 pcm.ac108 {
    type plug
-    slave {
-        rate 48000
-        format S32_LE
-        pcm "hw:seeed8micvoicec"
-    }
+    slave.pcm "multiapps"
 }

-# 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.multiapps {
+    type dsnoop
+    slave.pcm "hw:1,0"
+    ipc_key 666666
 }

 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
+	type plug
+	slave {
+		pcm {
+			type dmix
+			ipc_key 1048576
+				slave {
+				pcm		"hw:1,0"
+				format		S32_LE
+				# rate		16000
+				channels	8
+			}
+			bindings {
+				# map 2 channels input to
+				# first 2 channels of 8 output
+				0	0
+				1	1
+				0	2
+				1	3
+				0	4
+				1	5
+				0	6
+				1	7
+			}
+		}
+		channels	2
+	}
 }


--- a/builddtbo.sh
+++ b/builddtbo.sh
@ -1,10 +1,8 @@
-#!/bin/sh
 #dtoverlay -r seeed-2mic-voicecard
-DTC_FLAGS="-b 0 -Wno-unit_address_vs_reg -I dts -O dtb"

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

 # cp *.dtbo /boot/overlays
 # dtoverlay seeed-2mic-voicecard
--- a/dkms.conf
+++ b/dkms.conf
@ -6,10 +6,4 @@ BUILT_MODULE_NAME[2]="snd-soc-seeed-voicecard"
 DEST_MODULE_LOCATION[0]="/kernel/sound/soc/codecs"
 DEST_MODULE_LOCATION[1]="/kernel/sound/soc/codecs"
 DEST_MODULE_LOCATION[2]="/kernel/sound/soc/bcm"
-PATCH[0]="back-to-v4.19.diff"
-PATCH[1]="back-to-v5.4.diff"
-PATCH[2]="back-to-v5.8.diff"
-PATCH_MATCH[0]="^4\.19\.*"
-PATCH_MATCH[1]="^5\.4\.*"
-PATCH_MATCH[2]="^5\.8\.*"
 AUTOINSTALL="yes"
--- a/install.sh
+++ b/install.sh
@ -5,134 +5,29 @@ if [[ $EUID -ne 0 ]]; then
   exit 1
 fi

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

 ver="0.3"
-uname_r=$(uname -r)
+

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

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

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

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

@ -144,16 +39,10 @@ function install_module {
    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
-  for _i in $kernels; do
-    dkms build -k $_i -m $mod -v $ver && {
-      dkms install --force -k $_i -m $mod -v $ver
-    }
-  done
+  dkms build $kernels -m $mod -v $ver && dkms install --force $kernels -m $mod -v $ver

  mkdir -p /var/lib/dkms/$mod/$ver/$marker
 }
@ -162,60 +51,40 @@ install_module "./" "seeed-voicecard"


 # install dtbos
-cp seeed-2mic-voicecard.dtbo $OVERLAYS
-cp seeed-4mic-voicecard.dtbo $OVERLAYS
-cp seeed-8mic-voicecard.dtbo $OVERLAYS
+cp seeed-2mic-voicecard.dtbo /boot/overlays
+cp seeed-4mic-voicecard.dtbo /boot/overlays
+cp seeed-8mic-voicecard.dtbo /boot/overlays

 #install alsa plugins
 # no need this plugin now
-# install -D ac108_plugin/libasound_module_pcm_ac108.so /usr/lib/arm-linux-gnueabihf/alsa-lib/
-rm -f /usr/lib/arm-linux-gnueabihf/alsa-lib/libasound_module_pcm_ac108.so
+# install -D ac108_plugin/libasound_module_pcm_ac108.so   /usr/lib/arm-linux-gnueabihf/alsa-lib/libasound_module_pcm_ac108.so

-#set kernel modules
-grep -q "^snd-soc-seeed-voicecard$" /etc/modules || \
+#set kernel moduels
+grep -q "snd-soc-seeed-voicecard" /etc/modules || \
  echo "snd-soc-seeed-voicecard" >> /etc/modules
-grep -q "^snd-soc-ac108$" /etc/modules || \
+grep -q "snd-soc-ac108" /etc/modules || \
  echo "snd-soc-ac108" >> /etc/modules
-grep -q "^snd-soc-wm8960$" /etc/modules || \
-  echo "snd-soc-wm8960" >> /etc/modules
+grep -q "snd-soc-wm8960" /etc/modules || \
+  echo "snd-soc-wm8960" >> /etc/modules  

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


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

 #install config files
 mkdir /etc/voicecard || true
 cp *.conf /etc/voicecard
 cp *.state /etc/voicecard

-#create git repo
-git_email=$(git config --global --get user.email)
-git_name=$(git config --global --get user.name)
-if [ "x${git_email}" == "x" ] || [ "x${git_name}" == "x" ] ; then
-    echo "setup git config"
-    git config --global user.email "respeaker@seeed.cc"
-    git config --global user.name "respeaker"
-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
+systemctl enable  seeed-voicecard.service 

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

 	fragment@1 {
-		target-path = "/";
+		target-path = "/clocks";
 		__overlay__ {
 			ac108_mclk: codec-mclk {
 				compatible = "fixed-clock";
--- a/seeed-4mic-voicecard.dtbo
+++ b/seeed-4mic-voicecard.dtbo
--- a/seeed-8mic-voicecard-overlay.dts
+++ b/seeed-8mic-voicecard-overlay.dts
@ -13,7 +13,7 @@
 	};

 	fragment@1 {
-		target-path = "/";
+		target-path = "/clocks";
 		__overlay__ {
 			ac10x_mclk: codec-mclk {
 				compatible = "fixed-clock";
@ -26,15 +26,13 @@
 	fragment@2 {
 		target = <&gpio>;
 		__overlay__ {
-			spk_amp_pins: spk_pins {
+			spk_amp_and_irq_pins: speaker_amp_and_irq_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>;     /* - */
 			};
 		};
 	};
@ -49,7 +47,7 @@
 			ac101: ac101@1a{
 				compatible = "x-power,ac101";
 				pinctrl-names = "default";
-				pinctrl-0 = <&spk_amp_pins &gpclk0_pins>;
+				pinctrl-0 = <&spk_amp_and_irq_pins>,<&gpclk0_pins>;
 				spk-amp-switch-gpios = <&gpio 17 0>;
 				switch-irq-gpios = <&gpio 22 0>;
 				reg = <0x1a>;
@ -82,8 +80,6 @@
 			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 {
@ -92,7 +88,6 @@
 				frame-master = <&codec0_dai>;
 				/* bitclock-inversion; */
 				/* frame-inversion; */
-				reg = <0>;

 				cpu {
 					sound-dai = <&i2s>;
--- a/seeed-8mic-voicecard.dtbo
+++ b/seeed-8mic-voicecard.dtbo
--- a/167
+++ b/167
@ -21,143 +21,76 @@
 # 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
-
+exec 1>/var/log/$(basename $0).log 2>&1
 #enable i2c interface
-dtparam -d $OVERLAYS i2c_arm=on
+dtparam 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
-}
+dtparam spi=on
+sleep 1


-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=""
+is_1a=$(i2cdetect -y  1 0x1a 0x1a | grep 1a | awk '{print $2}')
+is_35=$(i2cdetect -y  1 0x35 0x35 | grep 35 | awk '{print $2}')
+is_3b=$(i2cdetect -y  1 0x3b 0x3b | grep 3b | awk '{print $2}')

-if [ "x${is_1a}" != "x" ] && [ "x${is_35}" == "x" ] ; then
+if [ "x${is_1a}" == "x1a" ] && [ "x${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
+    dtoverlay seeed-2mic-voicecard
+    sleep 1
+    hw=$(aplay -l | grep seeed2micvoicec | awk '{print $2}' | sed 's/://')
+
+    cp /etc/voicecard/asound_2mic.conf /etc/asound.conf
+
+    echo "get old hw number"
+    old=$(cat /etc/asound.conf | grep hw: | awk 'NR==1 {print $2}' | sed 's/\"//g')
+
+    echo "replace new hw:${hw},0"
+    sed -i -e "s/${old}/hw:${hw},0/g" /etc/asound.conf
+
+    cp /etc/voicecard/wm8960_asound.state /var/lib/alsa/asound.state
 fi

-if [ "x${is_3b}" != "x" ] && [ "x${is_35}" == "x" ] ; then
+if [ "x${is_3b}" == "x3b" ] && [ "x${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
+    dtoverlay seeed-4mic-voicecard
+    sleep 1
+    hw=$(arecord -l | grep seeed4micvoicec | awk '{print $2}' | sed 's/://')
+
+    cp /etc/voicecard/asound_4mic.conf /etc/asound.conf
+
+    echo "get slavepcm line number"
+    line=$(grep -n ac108-slavepcm /etc/asound.conf | awk '{print $1}' | sed 's/://')
+
+    echo "delete ${line} slavepcm hw number"
+    sed -i -e "${line}d"  /etc/asound.conf
+
+    echo "insert slavepcm hw:${hw},0"
+    sed -i "${line}i slave.pcm \"hw:${hw},0\""  /etc/asound.conf
+
+    cp /etc/voicecard/ac108_asound.state /var/lib/alsa/asound.state
 fi

-if [ "x${is_3b}" != "x" ] && [ "x${is_35}" != "x" ] ; then
+if [ "x${is_3b}" == "x3b" ] && [ "x${is_35}" == "x35" ] ; then
    echo "install 6mic"
-    overlay=seeed-8mic-voicecard
-    asound_conf=/etc/voicecard/asound_6mic.conf
-    asound_state=/etc/voicecard/ac108_6mic.state
-fi
+    dtoverlay seeed-8mic-voicecard
+    sleep 1
+    hw=$(aplay -l | grep seeed8micvoicec | awk '{print $2}' | sed 's/://')

-if [ "$overlay" ]; then
-    echo Install $overlay ...
+    cp /etc/voicecard/asound_6mic.conf /etc/asound.conf

-    # Remove old configuration
-    rm /etc/asound.conf
-    rm /var/lib/alsa/asound.state
+    old=$(cat /etc/asound.conf | grep hw: | awk 'NR==1 {print $2}' | sed 's/\"//g')

-    kernel_ver=$(uname -r) # get_kernel_version)
-    # echo kernel_ver=$kernel_ver
+    sed -i -e "s/${old}/hw:${hw},0/g" /etc/asound.conf  

-    # 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
+    cp /etc/voicecard/ac108_6mic.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
+#Fore 3.5mm ('headphone') jack
+amixer cset numid=3 1 
+
+
--- a/seeed-voicecard.c
+++ b/seeed-voicecard.c
@ -26,23 +26,6 @@
 #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:
@ -56,9 +39,6 @@ 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;
@ -67,12 +47,7 @@ struct seeed_card_data {
 	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 {
@ -86,7 +61,6 @@ struct seeed_card_info {
 	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))
@ -111,16 +85,16 @@ 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;
 }
@ -132,10 +106,10 @@ 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);

@ -146,8 +120,8 @@ 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);
@ -177,9 +151,9 @@ err:
 }

 #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);
+static int (* _set_clock[_SET_CLOCK_CNT])(int y_start_n_stop);

-int seeed_voice_card_register_set_clock(int stream, int (*set_clock)(int, struct snd_pcm_substream *, int, struct snd_soc_dai *)) {
+int seeed_voice_card_register_set_clock(int stream, int (*set_clock)(int)) {
 	if (! _set_clock[stream]) {
 		_set_clock[stream] = set_clock;
 	}
@ -187,82 +161,44 @@ int seeed_voice_card_register_set_clock(int stream, int (*set_clock)(int, struct
 }
 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 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;
-	#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);
+		dai->playback_active, dai->capture_active);
+
+	/* I know it will degrades performance, but I have no choice */
+	spin_lock_irqsave(&priv->lock, flags);

 	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
+		if (_set_clock[SNDRV_PCM_STREAM_CAPTURE]) _set_clock[SNDRV_PCM_STREAM_CAPTURE](1);
+		if (_set_clock[SNDRV_PCM_STREAM_PLAYBACK]) _set_clock[SNDRV_PCM_STREAM_PLAYBACK](1);
 		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) {
+		if (dai->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 */
-		}
+		if (_set_clock[SNDRV_PCM_STREAM_CAPTURE]) _set_clock[SNDRV_PCM_STREAM_CAPTURE](0);
+		if (_set_clock[SNDRV_PCM_STREAM_PLAYBACK]) _set_clock[SNDRV_PCM_STREAM_PLAYBACK](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);
+	spin_unlock_irqrestore(&priv->lock, flags);

 	return ret;
 }
@ -274,165 +210,23 @@ static struct snd_soc_ops seeed_voice_card_ops = {
 	.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 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;

-	ret = asoc_simple_init_dai(codec, &dai_props->codec_dai);
+	ret = asoc_simple_card_init_dai(codec, &dai_props->codec_dai);
 	if (ret < 0)
 		return ret;

-	ret = asoc_simple_init_dai(cpu, &dai_props->cpu_dai);
+	ret = asoc_simple_card_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;
 }

@ -460,37 +254,32 @@ static int seeed_voice_card_dai_link_of(struct device_node *node,
 	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) {
+	if (!cpu || !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,
+	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
@ -502,7 +291,7 @@ 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;

@ -527,7 +316,7 @@ 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;
@ -535,16 +324,22 @@ 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,
+	#if _SINGLE_CODEC
+	ret = asoc_simple_card_canonicalize_dailink(dai_link);
+	if (ret < 0)
+		goto dai_link_of_err;
+	#endif
+
+	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
@ -558,27 +353,17 @@ 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);

-#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
+	asoc_simple_card_canonicalize_cpu(dai_link, single_cpu);

 dai_link_of_err:
 	of_node_put(cpu);
@ -610,7 +395,7 @@ 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;
@ -670,7 +455,7 @@ 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;

@ -695,79 +480,6 @@ card_parse_end:
 	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;
@ -775,7 +487,7 @@ 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"))
@ -793,25 +505,6 @@ 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;

@ -830,9 +523,6 @@ 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) {
@ -849,19 +539,13 @@ 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,
@ -872,20 +556,14 @@ static int seeed_voice_card_probe(struct platform_device *pdev)

 	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);
+	asoc_simple_card_clean_reference(&priv->snd_card);

 	return ret;
 }
@ -893,13 +571,8 @@ err:
 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;
+	return asoc_simple_card_clean_reference(card);
 }

 static const struct of_device_id seeed_voice_of_match[] = {
--- a/sound-compatible-4.18.h
+++ b/sound-compatible-4.18.h
@ -1,47 +0,0 @@
-/*
- * (C) Copyright 2017-2018
- * Seeed Technology Co., Ltd. <www.seeedstudio.com>
- *
- * PeterYang <linsheng.yang@seeed.cc>
- */
-#ifndef __SOUND_COMPATIBLE_4_18_H__
-#define __SOUND_COMPATIBLE_4_18_H__
-
-#include <linux/version.h>
-
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(4,17,0)
-#define __NO_SND_SOC_CODEC_DRV     1
-#else
-#define __NO_SND_SOC_CODEC_DRV     0
-#endif
-
-#if LINUX_VERSION_CODE < KERNEL_VERSION(5,4,0)
-#if __has_attribute(__fallthrough__)
-# define fallthrough                    __attribute__((__fallthrough__))
-#else
-# define fallthrough                    do {} while (0)  /* fallthrough */
-#endif
-#endif
-
-#if __NO_SND_SOC_CODEC_DRV
-#define codec                      component
-#define snd_soc_codec              snd_soc_component
-#define snd_soc_codec_driver       snd_soc_component_driver
-#define snd_soc_codec_get_drvdata  snd_soc_component_get_drvdata
-#define snd_soc_codec_get_dapm     snd_soc_component_get_dapm
-#define snd_soc_codec_get_bias_level snd_soc_component_get_bias_level
-#define snd_soc_kcontrol_codec     snd_soc_kcontrol_component
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,9,0)
-#define snd_soc_read               snd_soc_component_read
-#else
-#define snd_soc_read               snd_soc_component_read32
-#endif
-#define snd_soc_register_codec     devm_snd_soc_register_component
-#define snd_soc_unregister_codec   snd_soc_unregister_component
-#define snd_soc_update_bits        snd_soc_component_update_bits
-#define snd_soc_write              snd_soc_component_write
-#define snd_soc_add_codec_controls snd_soc_add_component_controls
-#endif
-
-#endif//__SOUND_COMPATIBLE_4_18_H__
-
--- a/tools/phase_test.py
+++ b/tools/phase_test.py
@ -1,83 +0,0 @@
-import sys
-import wave
-import numpy as np
-
-
-if len(sys.argv) != 2:
-    print('Usage: {} multi.wav'.format(sys.argv[0]))
-    sys.exit(1)
-
-
-multi = wave.open(sys.argv[1], 'rb')
-rate = multi.getframerate()
-channels = multi.getnchannels()
-
-if channels <= 1:
-    sys.exit(1)
-
-N = rate 
-
-window = np.hanning(N)
-
-interp = 4*8
-max_offset = int(rate * 0.1 / 340 * interp)
-
-def gcc_phat(sig, refsig, fs=1, max_tau=None, interp=16):
-    '''
-    This function computes the offset between the signal sig and the reference signal refsig
-    using the Generalized Cross Correlation - Phase Transform (GCC-PHAT)method.
-    '''
-    
-    # make sure the length for the FFT is larger or equal than len(sig) + len(refsig)
-    n = sig.shape[0] + refsig.shape[0]
-
-    # Generalized Cross Correlation Phase Transform
-    SIG = np.fft.rfft(sig, n=n)
-    REFSIG = np.fft.rfft(refsig, n=n)
-    R = SIG * np.conj(REFSIG)
-    #R /= np.abs(R)
-
-    cc = np.fft.irfft(R, n=(interp * n))
-
-    max_shift = int(interp * n / 2)
-    if max_tau:
-        max_shift = np.minimum(int(interp * fs * max_tau), max_shift)
-
-    cc = np.concatenate((cc[-max_shift:], cc[:max_shift+1]))
-
-    # find max cross correlation index
-    shift = np.argmax(np.abs(cc)) - max_shift
-
-    tau = shift / float(interp * fs)
-    
-    return tau, cc
-
-
-print(multi.getsampwidth())
-
-while True:
-    data = multi.readframes(N)
-
-    if len(data) != multi.getsampwidth() * N * channels:
-        print("done")
-        break
-
-    if multi.getsampwidth() == 2:
-        data = np.fromstring(data, dtype='int16')
-    else:
-        data = np.fromstring(data, dtype='int32')
-    ref_buf = data[0::channels]
-
-    offsets = []
-    for ch in range(1, channels):
-        sig_buf = data[ch::channels]
-        tau, _ = gcc_phat(sig_buf * window, ref_buf * window, fs=1, max_tau=max_offset, interp=interp)
-        # tau, _ = gcc_phat(sig_buf, ref_buf, fs=rate, max_tau=1)
-
-        offsets.append(tau)
-
-    print(offsets)
-
-print(multi.getframerate())
-
-multi.close()
--- a/ubuntu-prerequisite.sh
+++ b/ubuntu-prerequisite.sh
@ -1,37 +0,0 @@
-#!/bin/bash
-
-# Copyright (c) Hin-Tak Leung 2020
-#
-# Overview:
-#   This script compiles and install the Broadcom VideoCore tools,
-#   configure the dynamic loader for the non-standard library location,
-#   and update the loader cache.
-#
-#   A few steps explicitly requires root privilege, which are
-#   marked with "sudo". The rest is just checking for duplicate/previous
-#   action.
-#
-#   This derived from my command history on ubuntu 20.04.1 .YMMV
-
-sudo apt install -y git gcc g++ make alsa-utils cmake
-
-git clone git://github.com/raspberrypi/userland.git
-pushd userland/
-
-arch=$(uname -m)
-if [[ "$arch" =~ aarch64 ]]; then
-    ./buildme --aarch64
-else
-    ./buildme
-fi
-# ./buildme already includes "sudo make install" at the end
-
-popd
-
-# matches Raspbian's location:
-if [ ! -f /etc/ld.so.conf.d/00-vmcs.conf ] ; then
-    echo "/opt/vc/lib" | sudo tee -a /etc/ld.so.conf.d/00-vmcs.conf
-    sudo ldconfig -v
-else
-    echo "/etc/ld.so.conf.d/00-vmcs.conf exists - no need to update ld.cache!"
-fi
--- a/uninstall.sh
+++ b/uninstall.sh
@ -13,65 +13,17 @@ 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
+rm  /boot/overlays/seeed-2mic-voicecard.dtbo || true
+rm  /boot/overlays/seeed-4mic-voicecard.dtbo || true
+rm  /boot/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
+systemctl disable seeed-voicecard.service 

 echo "remove seeed-voicecard"
 rm  /usr/bin/seeed-voicecard || true
@ -81,18 +33,9 @@ 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 $i 1
-done
+rm  /lib/modules/${uname_r}/kernel/sound/soc/codecs/snd-soc-wm8960.ko || true
+rm  /lib/modules/${uname_r}/kernel/sound/soc/codecs/snd-soc-ac108.ko || true
+rm  /lib/modules/${uname_r}/kernel/sound/soc/bcm/snd-soc-seeed-voicecard.ko || true

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

 #include "wm8960.h"

@ -508,11 +507,7 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec)
 	 * list each time to find the desired power state do so now
 	 * and save the result.
 	 */
-	#if __NO_SND_SOC_CODEC_DRV
-	list_for_each_entry(w, &codec->card->widgets, list) {
-	#else
 	list_for_each_entry(w, &codec->component.card->widgets, list) {
-	#endif
 		if (w->dapm != dapm)
 			continue;
 		if (strcmp(w->name, "LOUT1 PGA") == 0)
@ -753,7 +748,6 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
 			iface |= 0x000c;
 			break;
 		}
-		fallthrough;
 	default:
 		dev_err(codec->dev, "unsupported width %d\n",
 			params_width(params));
@ -796,7 +790,7 @@ 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 +1230,11 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
 static const struct snd_soc_dai_ops wm8960_dai_ops = {
 	.hw_params = wm8960_hw_params,
 	.hw_free = wm8960_hw_free,
-	.mute_stream = wm8960_mute,
+	.digital_mute = wm8960_mute,
 	.set_fmt = wm8960_set_dai_fmt,
 	.set_clkdiv = wm8960_set_dai_clkdiv,
 	.set_pll = wm8960_set_dai_pll,
 	.set_sysclk = wm8960_set_dai_sysclk,
-	.no_capture_mute = 1,
 };

 static struct snd_soc_dai_driver wm8960_dai = {
@ -1259,11 +1252,7 @@ static struct snd_soc_dai_driver wm8960_dai = {
 		.rates = WM8960_RATES,
 		.formats = WM8960_FORMATS,},
 	.ops = &wm8960_dai_ops,
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,12,0)
-	.symmetric_rate = 1,
-#else
 	.symmetric_rates = 1,
-#endif
 };

 static int wm8960_probe(struct snd_soc_codec *codec)
@ -1286,12 +1275,7 @@ 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,
-	#if __NO_SND_SOC_CODEC_DRV
-	.idle_bias_on		= 1,
-	.use_pmdown_time	= 1,
-	.endianness		= 1,
-	#endif
+	.suspend_bias_off = true,
 };

 static const struct regmap_config wm8960_regmap = {
@ -1318,7 +1302,8 @@ static void wm8960_set_pdata_from_of(struct i2c_client *i2c,
 		pdata->shared_lrclk = true;
 }

-static int wm8960_i2c_probe(struct i2c_client *i2c)
+static int wm8960_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
 {
 	struct wm8960_data *pdata = dev_get_platdata(&i2c->dev);
 	struct wm8960_priv *wm8960;
@ -1383,9 +1368,10 @@ static int wm8960_i2c_probe(struct i2c_client *i2c)
 	return ret;
 }

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

 static const struct i2c_device_id wm8960_i2c_id[] = {