[PATCH] fix channel ordering

Fix /boot/config.txt for bookworm installs

Makefile

@@ -1,3 +1,17 @@
+#
+# 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
@@ -14,14 +28,25 @@ endif
 endif
 
 
+
+else
+
+DEST := /lib/modules/$(uname_r)/kernel
+
 all:
-	make -C /lib/modules/$(shell uname -r)/build M=$(PWD) modules
+	make -C /lib/modules/$(uname_r)/build M=$(PWD) modules
 
 clean:
-	make -C /lib/modules/$(shell uname -r)/build M=$(PWD) clean
+	make -C /lib/modules/$(uname_r)/build M=$(PWD) clean
 
 install:
-	sudo cp 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 cp snd-soc-ac108.ko ${DEST}/sound/soc/codecs/
+	sudo cp snd-soc-wm8960.ko ${DEST}/sound/soc/codecs/
+	sudo cp snd-soc-seeed-voicecard.ko ${DEST}/sound/soc/bcm/
 	sudo depmod -a
+
+
+.PHONY: all clean install
+
+endif
+

README.md

@@ -1,231 +1,20 @@
 # seeed-voicecard
 
-[![Join the chat at https://gitter.im/seeed-voicecard/Lobby](https://badges.gitter.im/seeed-voicecard/Lobby.svg)](https://gitter.im/seeed-voicecard/Lobby?utm_source=badge&utm_medium=badge&utm_campaign=pr-badge&utm_content=badge)
-
-The drivers 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.
+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.
 
 ### 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/respeaker/seeed-voicecard
+git clone https://github.com/HinTak/seeed-voicecard
 cd seeed-voicecard
 sudo ./install.sh
 sudo reboot
 ```
+## ReSpeaker Documentation
 
-## ReSpeaker Mic Hat
+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)
 
-[![](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
 
@@ -255,7 +44,21 @@ 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). 
+
+

ac101.c

@@ -37,6 +37,7 @@
 #include <linux/gpio/consumer.h>
 #include <linux/regmap.h>
 #include <linux/input.h>
+#include <linux/delay.h>
 #include "ac101_regs.h"
 #include "ac10x.h"
 
@@ -77,6 +78,8 @@ 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;
@@ -84,16 +87,20 @@ 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;
 
-	return regmap_write(ac10x->regmap101, reg, val);
+	v = regmap_write(ac10x->regmap101, reg, val);
+	return v;
 }
 
 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;
 
-	return regmap_update_bits(ac10x->regmap101, reg, mask, value);
+	v = regmap_update_bits(ac10x->regmap101, reg, mask, value);
+	return v;
 }
 
 
@@ -112,7 +119,7 @@ static void switch_hw_config(struct snd_soc_codec *codec)
 {
 	int r;
 
-	AC101_DBG("%s,line:%d\n",__func__,__LINE__);
+	AC101_DBG();
 
 	/*HMIC/MMIC BIAS voltage level select:2.5v*/
 	ac101_update_bits(codec, OMIXER_BST1_CTRL, (0xf<<BIASVOLTAGE), (0xf<<BIASVOLTAGE));
@@ -155,7 +162,7 @@ static void switch_hw_config(struct snd_soc_codec *codec)
  */
 static void switch_status_update(struct ac10x_priv *ac10x)
 {
-	AC101_DBG("%s,line:%d,ac10x->state:%d\n", __func__, __LINE__, ac10x->state);
+	AC101_DBG("ac10x->state:%d\n", ac10x->state);
 
         input_report_switch(ac10x->inpdev, SW_HEADPHONE_INSERT, ac10x->state);
         input_sync(ac10x->inpdev);
@@ -176,7 +183,7 @@ 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;
@@ -210,17 +217,17 @@ static int __ac101_get_hmic_data(struct snd_soc_codec *codec) {
 	#ifdef AC101_DEBG
 	static long counter;
 	#endif
-	int r;
-	int d;
+	int r, d;
 
 	d = GET_HMIC_DATA(ac101_read(codec, HMIC_STS));
 
 	r = 0x1 << HMIC_DATA_PEND;
 	ac101_write(codec, HMIC_STS, r);
 
-	AC101_DBG("%s,line:%d HMIC_DATA(%3ld): %02X\n", __func__, __LINE__,
-			counter++, d
-		);
+	/* prevent i2c accessing too frequently */
+	usleep_range(1500, 3000);
+
+	AC101_DBG("HMIC_DATA(%3ld): %02X\n", counter++, d);
 	return d;
 }
 
@@ -255,7 +262,7 @@ static void work_cb_earphone_switch(struct work_struct *work)
 				input_report_key(ac10x->inpdev, KEY_HEADSETHOOK, 1);
 				input_sync(ac10x->inpdev);
 
-				AC101_DBG("%s,line:%d KEY_HEADSETHOOK1\n", __func__, __LINE__);
+				AC101_DBG("KEY_HEADSETHOOK1\n");
 
 				if (hook_flag1 != hook_flag2)
 					hook_flag1 = hook_flag2 = 0;
@@ -273,7 +280,7 @@ static void work_cb_earphone_switch(struct work_struct *work)
 				input_report_key(ac10x->inpdev, KEY_VOLUMEUP, 0);
 				input_sync(ac10x->inpdev);
 
-				AC101_DBG("%s,line:%d HMIC_DATA: %d KEY_VOLUMEUP\n", __func__, __LINE__, t);
+				AC101_DBG("HMIC_DATA: %d KEY_VOLUMEUP\n", t);
 			}
 			if (ac10x->pullout_cntr)
 				ac10x->pullout_cntr--;
@@ -286,7 +293,7 @@ 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("%s,line:%d KEY_VOLUMEDOWN\n", __func__, __LINE__);
+				AC101_DBG("KEY_VOLUMEDOWN\n");
 			}
 			if (ac10x->pullout_cntr)
 				ac10x->pullout_cntr--;
@@ -302,7 +309,7 @@ static void work_cb_earphone_switch(struct work_struct *work)
 				input_report_key(ac10x->inpdev, KEY_HEADSETHOOK, 0);
 				input_sync(ac10x->inpdev);
 
-				AC101_DBG("%s,line:%d KEY_HEADSETHOOK0\n", __func__, __LINE__);
+				AC101_DBG("KEY_HEADSETHOOK0\n");
 			}
 		}
 	} else {
@@ -356,7 +363,7 @@ 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;
-	int ret;
+	long ret;
 
 	ac10x->gpiod_irq = devm_gpiod_get_optional(&i2c->dev, "switch-irq", GPIOD_IN);
 	if (IS_ERR(ac10x->gpiod_irq)) {
@@ -369,7 +376,7 @@ static int ac101_switch_probe(struct ac10x_priv *ac10x) {
 
 	ac10x->irq = gpiod_to_irq(ac10x->gpiod_irq);
 	if (IS_ERR_VALUE(ac10x->irq)) {
-		pr_info("[ac101] map gpio to irq failed, errno = %d\n", ac10x->irq);
+		pr_warn("[ac101] map gpio to irq failed, errno = %ld\n", ac10x->irq);
 		ac10x->irq = 0;
 		goto _err_irq;
 	}
@@ -377,7 +384,7 @@ 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_info("[ac101] request virq %d failed, errno = %d\n", ac10x->irq, ret);
+		pr_warn("[ac101] request virq %ld failed, errno = %ld\n", ac10x->irq, ret);
 		goto _err_irq;
 	}
 
@@ -426,7 +433,7 @@ _err_input_register_device:
 _err_input_allocate_device:
 
 	if (ac10x->irq) {
-		devm_free_irq(&i2c->dev, ac10x->irq, NULL);
+		devm_free_irq(&i2c->dev, ac10x->irq, ac10x);
 		ac10x->irq = 0;
 	}
 _err_irq:
@@ -580,7 +587,7 @@ 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("%s,line:%d\n",__func__,__LINE__);
+		AC101_DBG();
 		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));
@@ -610,7 +617,7 @@ 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:%s,line:%d\n", __func__, __LINE__);
+		AC101_DBG("post:open\n");
 		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));
@@ -618,7 +625,7 @@ static int ac101_headphone_event(struct snd_soc_codec* codec, int event) {
 		break;
 	case SND_SOC_DAPM_PRE_PMD:
 		/*close*/
-		AC101_DBG("pre:close:%s,line:%d\n", __func__, __LINE__);
+		AC101_DBG("pre:close\n");
 		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));
@@ -637,9 +644,7 @@ 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;
-
-	/* spin_lock move to machine trigger */
+	int ret = 0;
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
@@ -653,9 +658,9 @@ 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("%s() L%d start sysclk failed\n", __func__, __LINE__);
+				AC101_DBG("start sysclk failed\n");
 			} else {
-				AC101_DBG("%s() L%d hw sysclk enable\n", __func__, __LINE__);
+				AC101_DBG("hw sysclk enable\n");
 				ac10x->aif1_clken++;
 			}
 		}
@@ -669,19 +674,18 @@ 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("%s() L%d stop sysclk failed\n", __func__, __LINE__);
+				AC101_DBG("stop sysclk failed\n");
 			} else {
-				AC101_DBG("%s() L%d hw sysclk disable\n", __func__, __LINE__);
+				AC101_DBG("hw sysclk disable\n");
 				ac10x->aif1_clken = 0;
 			}
 			break;
 		}
 	}
 
-	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);
+	AC101_DBG("event=%d pre_up/%d post_down/%d\n", event, SND_SOC_DAPM_PRE_PMU, SND_SOC_DAPM_POST_PMD);
 
-	return 0;
+	return ret;
 }
 
 /**
@@ -778,8 +782,8 @@ 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 volume", SPKOUT_CTRL, SPK_VOL, 0x1f, 0, speaker_vol_tlv),
-	SOC_SINGLE_TLV("headphone volume", HPOUT_CTRL, HP_VOL, 0x3f, 0, headphone_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),
 };
 
 /* PLL divisors */
@@ -907,7 +911,7 @@ 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("%s() L%d mute=%d\n", __func__, __LINE__, mute);
+	AC101_DBG("mute=%d\n",  mute);
 
 	ac101_write(codec, DAC_VOL_CTRL, mute? 0: 0xA0A0);
 
@@ -949,12 +953,12 @@ 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("%s,line:%d stream = %s, play: %d, capt: %d, active: %d\n", __func__, __LINE__,
+	AC101_DBG("stream = %s, play: %d, capt: %d, active: %d\n", 
 		snd_pcm_stream_str(substream),
-		codec_dai->playback_active, codec_dai->capture_active,
-		codec_dai->active);
+		codec_dai->stream[SNDRV_PCM_STREAM_PLAYBACK].active, codec_dai->stream[SNDRV_PCM_STREAM_CAPTURE].active,
+		snd_soc_dai_active(codec_dai));
 
-	if (!codec_dai->active) {
+	if (!snd_soc_dai_active(codec_dai)) {
 		ac10x->aif1_clken = 1;
 		ac101_aif1clk(codec, SND_SOC_DAPM_POST_PMD, 0);
 	} else {
@@ -968,7 +972,7 @@ 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("%s, line:%d, pll_id:%d\n", __func__, __LINE__, pll_id);
+	AC101_DBG("pll_id:%d\n",  pll_id);
 
 	/* clear volatile reserved bits*/
 	ac101_update_bits(codec, SYSCLK_CTRL, 0xFF & ~(0x1 << SYSCLK_ENA), 0x0);
@@ -1040,7 +1044,7 @@ int ac101_hw_params(struct snd_pcm_substream *substream,
 	int reg_val, freq_out;
 	unsigned channels;
 
-	AC101_DBG("%s() L%d +++\n", __func__, __LINE__);
+	AC101_DBG("+++\n");
 
 	if (_MASTER_MULTI_CODEC == _MASTER_AC101 && ac101_sysclk_started()) {
 		/* not configure hw_param twice if stream is playback, tell the caller it's started */
@@ -1076,7 +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->capture_active && dmic_used && codec_aif1_fs[i].samp_rate == 44100) {
+			if (codec_dai->stream[SNDRV_PCM_STREAM_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));
@@ -1126,7 +1130,7 @@ 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("%s() L%d ---\n", __func__, __LINE__);
+	AC101_DBG("---\n");
 	return 0;
 }
 
@@ -1136,7 +1140,7 @@ 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("%s() L%d\n", __func__, __LINE__);
+	AC101_DBG();
 
 	/*
 	 * 	master or slave selection
@@ -1148,14 +1152,14 @@ 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_warn("AC101 as Master\n");
+		pr_info("AC101 as Master\n");
 		reg_val |= (0x0<<AIF1_MSTR_MOD);
 		break;
 		#else
-		pr_warn("AC108 as Master\n");
+		pr_info("AC108 as Master\n");
 		#endif
 	case SND_SOC_DAIFMT_CBS_CFS:   /* codec clk & frm slave, ap is master*/
-		pr_warn("AC101 as Slave\n");
+		pr_info("AC101 as Slave\n");
 		reg_val |= (0x1<<AIF1_MSTR_MOD);
 		break;
 	default:
@@ -1225,7 +1229,7 @@ int ac101_audio_startup(struct snd_pcm_substream *substream,
 {
 	// struct snd_soc_codec *codec = codec_dai->codec;
 
-	AC101_DBG("\n\n%s,line:%d\n", __func__, __LINE__);
+	AC101_DBG("\n\n\n");
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 	}
@@ -1233,16 +1237,18 @@ 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) {
+static int ac101_set_clock(int y_start_n_stop, struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) {
+	int r;
+
 	if (y_start_n_stop) {
 		/* enable global clock */
-		ac101_aif1clk(static_ac10x->codec, SND_SOC_DAPM_PRE_PMU, 1);
+		r = ac101_aif1clk(static_ac10x->codec, SND_SOC_DAPM_PRE_PMU, 1);
 	} else {
 		/* disable global clock */
 		static_ac10x->aif1_clken = 1;
-		ac101_aif1clk(static_ac10x->codec, SND_SOC_DAPM_POST_PMD, 0);
+		r = ac101_aif1clk(static_ac10x->codec, SND_SOC_DAPM_POST_PMD, 0);
 	}
-	return 0;
+	return r;
 }
 #endif
 
@@ -1252,9 +1258,9 @@ 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("%s() stream=%s  cmd=%d\n",
-		__FUNCTION__,
+	AC101_DBG("stream=%s  cmd=%d\n",
 		snd_pcm_stream_str(substream),
 		cmd);
 
@@ -1263,6 +1269,7 @@ 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'
@@ -1272,15 +1279,21 @@ 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;
 }
 
@@ -1291,7 +1304,7 @@ 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("%s,line:%d, id=%d freq=%d, dir=%d\n", __func__, __LINE__,
+	AC101_DBG("id=%d freq=%d, dir=%d\n", 
 		clk_id, freq, dir);
 
 	ac10x->sysclk = freq;
@@ -1340,7 +1353,7 @@ 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("%s() L%d +++\n", __func__, __LINE__);
+	AC101_DBG("+++\n");
 
 	set_configuration(codec);
 	if (drc_used) {
@@ -1349,7 +1362,7 @@ 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("%s() L%d +++\n", __func__, __LINE__);
+	AC101_DBG("---\n");
 	return;
 }
 
@@ -1357,13 +1370,13 @@ 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("%s,line:%d, SND_SOC_BIAS_ON\n", __func__, __LINE__);
+		AC101_DBG("SND_SOC_BIAS_ON\n");
 		break;
 	case SND_SOC_BIAS_PREPARE:
-		AC101_DBG("%s,line:%d, SND_SOC_BIAS_PREPARE\n", __func__, __LINE__);
+		AC101_DBG("SND_SOC_BIAS_PREPARE\n");
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		AC101_DBG("%s,line:%d, SND_SOC_BIAS_STANDBY\n", __func__, __LINE__);
+		AC101_DBG("SND_SOC_BIAS_STANDBY\n");
 		#ifdef CONFIG_AC101_SWITCH_DETECT
 		switch_hw_config(codec);
 		#endif
@@ -1375,7 +1388,7 @@ 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("%s,line:%d, SND_SOC_BIAS_OFF\n", __func__, __LINE__);
+		AC101_DBG("SND_SOC_BIAS_OFF\n");
 		break;
 	}
 	snd_soc_codec_get_dapm(codec)->bias_level = level;
@@ -1389,7 +1402,7 @@ int ac101_codec_probe(struct snd_soc_codec *codec)
 
 	ac10x = dev_get_drvdata(codec->dev);
 	if (ac10x == NULL) {
-		AC101_DBG("not set client data %s() L%d\n", __func__, __LINE__);
+		AC101_DBG("not set client data!\n");
 		return -ENOMEM;
 	}
 	ac10x->codec = codec;
@@ -1440,12 +1453,20 @@ 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, NULL); */
+		devm_free_irq(codec->dev, ac10x->irq, ac10x);
 		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;
@@ -1490,32 +1511,32 @@ 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);
+		printk("read:start add:0x%x,count:0x%x\n", reg, num);
+
+		regcache_cache_bypass(ac10x->regmap101, true);
 		do {
-			value_r[i] = ac101_read(ac10x->codec, reg);
-			printk("0x%x: 0x%04x ",reg,value_r[i]);
-			reg+=1;
-			i++;
-			if(i == num)
+			value_r = ac101_read(ac10x->codec, reg);
+			printk("0x%x: 0x%04x ", reg++, value_r);
+			if (++i % 4 == 0 || i == num)
 				printk("\n");
-			if(i%4==0)
-				printk("\n");
-		} while(i<num);
+		} while (i < num);
+		regcache_cache_bypass(ac10x->regmap101, false);
 	}
 	return count;
 }
@@ -1591,7 +1612,7 @@ int ac101_probe(struct i2c_client *i2c, const struct i2c_device_id *id)
 	int ret = 0;
 	unsigned v = 0;
 
-	AC101_DBG("%s,line:%d\n", __func__, __LINE__);
+	AC101_DBG();
 
 	static_ac10x = ac10x;
 
@@ -1602,13 +1623,13 @@ 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) {
@@ -1643,7 +1664,7 @@ void ac101_shutdown(struct i2c_client *i2c)
 	int reg_val;
 
 	if (codec == NULL) {
-		pr_err("%s() L%d: no sound card.\n", __func__, __LINE__);
+		pr_err(": no sound card.\n");
 		return;
 	}
 

ac108.c

@@ -173,8 +173,11 @@ 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;
 
-	r = regmap_read(i2cm, reg, &v);
+	if ((r = regmap_read(i2cm, reg, &v)) < 0) {
+		pr_err("ac10x_read error->[REG-0x%02x]\n", reg);
+	} else {
 		*rt_val = v;
+	}
 	return r;
 }
 
@@ -450,11 +453,13 @@ 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++) {
-		ac10x_update_bits(reg, mask, val, ac10x->i2cmap[i]);
+		r |= ac10x_update_bits(reg, mask, val, ac10x->i2cmap[i]);
 	}
-	return 0;
+	return r;
 }
 
 static unsigned int ac108_codec_read(struct snd_soc_codec *codec, unsigned int reg) {
@@ -648,7 +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->playback_active, dai->capture_active);
+			dai->stream[SNDRV_PCM_STREAM_PLAYBACK].active, dai->stream[SNDRV_PCM_STREAM_CAPTURE].active);
 
 	if (ac10x->i2c101) {
 		ret = ac101_hw_params(substream, params, dai);
@@ -659,8 +664,8 @@ static int ac108_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_h
 		}
 	}
 
-	if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE && dai->playback_active)
-	 || (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && dai->capture_active)) {
+	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)) {
 		/* not configure hw_param twice */
 		/* return 0; */
 	}
@@ -805,6 +810,9 @@ 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;
 
@@ -860,6 +868,7 @@ 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");
 		/**
@@ -982,47 +991,50 @@ 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) {
-	u8 r;
+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;
 
 	dev_dbg(ac10x->codec->dev, "%s() L%d cmd:%d\n", __func__, __LINE__, y_start_n_stop);
 
 	/* spin_lock move to machine trigger */
 
-	if (y_start_n_stop)  {
-		if (ac10x->sysclk_en == 0) {
-
+	if (ac10x->i2c101 && _MASTER_MULTI_CODEC == _MASTER_AC101) {
+		ac101_trigger(substream, cmd, dai);
+	}
+	if (y_start_n_stop && ac10x->sysclk_en == 0) {
 		/* enable lrck clock */
-		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]);
+		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]);
 		}
 
 		/*0x10: PLL Common voltage enable, PLL enable */
-		ac108_multi_update_bits(PLL_CTRL1, 0x01 << PLL_EN | 0x01 << PLL_COM_EN,
+		ret = ret || ac108_multi_update_bits(PLL_CTRL1, 0x01 << PLL_EN | 0x01 << PLL_COM_EN,
 						   0x01 << PLL_EN | 0x01 << PLL_COM_EN, ac10x);
 		/* enable global clock */
-		ac108_multi_update_bits(I2S_CTRL, 0x1 << TXEN | 0x1 << GEN, 0x1 << TXEN | 0x1 << GEN, ac10x);
+		ret = ret || ac108_multi_update_bits(I2S_CTRL, 0x1 << TXEN | 0x1 << GEN, 0x1 << TXEN | 0x1 << GEN, ac10x);
 
 		ac10x->sysclk_en = 1UL;
-		}
-	} else if (ac10x->sysclk_en != 0) {
+	} else if (!y_start_n_stop && ac10x->sysclk_en != 0) {
 		/* disable global clock */
-		ac108_multi_update_bits(I2S_CTRL, 0x1 << TXEN | 0x1 << GEN, 0x0 << TXEN | 0x0 << GEN, ac10x);
+		ret = ret || ac108_multi_update_bits(I2S_CTRL, 0x1 << TXEN | 0x1 << GEN, 0x0 << TXEN | 0x0 << GEN, ac10x);
 
 		/*0x10: PLL Common voltage disable, PLL disable */
-		ac108_multi_update_bits(PLL_CTRL1, 0x01 << PLL_EN | 0x01 << PLL_COM_EN,
+		ret = ret || 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, &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_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;
 		}
+	}
 
-	return 0;
+	return ret;
 }
 
 static int ac108_prepare(struct snd_pcm_substream *substream,
@@ -1049,39 +1061,33 @@ 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);
 		}
-
-		/* delayed clock starting, move to machine trigger() */
+		spin_unlock_irqrestore(&ac10x->lock, flags);
+		ac108_set_clock(1, substream, cmd, dai);
 		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;
 	}
 
-__ret:
-	spin_unlock_irqrestore(&ac10x->lock, flags);
-
+	dev_dbg(dai->dev, "%s() stream=%s  cmd=%d; finished %d\n",
+		__FUNCTION__,
+		snd_pcm_stream_str(substream),
+		cmd, ret);
 	return ret;
 }
 
@@ -1115,7 +1121,7 @@ void ac108_aif_shutdown(struct snd_pcm_substream *substream,
 	}
 }
 
-int ac108_aif_mute(struct snd_soc_dai *dai, int mute) {
+int ac108_aif_mute(struct snd_soc_dai *dai, int mute, int direction) {
 	struct snd_soc_codec *codec = dai->codec;
 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
 
@@ -1136,12 +1142,13 @@ static const struct snd_soc_dai_ops ac108_dai_ops = {
 	.hw_params	= ac108_hw_params,
 	.prepare	= ac108_prepare,
 	.trigger	= ac108_trigger,
-	.digital_mute	= ac108_aif_mute,
+	.mute_stream	= 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 = {
@@ -1211,7 +1218,6 @@ 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;
@@ -1267,6 +1273,12 @@ 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);
@@ -1310,6 +1322,11 @@ 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) {
@@ -1325,12 +1342,16 @@ 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);
 
@@ -1348,6 +1369,9 @@ 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;
@@ -1383,7 +1407,15 @@ static const struct regmap_config ac108_regmap = {
 	.max_register = 0xDF,
 	.cache_type = REGCACHE_FLAT,
 };
-static int ac108_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *i2c_id) {
+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) {
 	struct device_node *np = i2c->dev.of_node;
 	unsigned int val = 0;
 	int ret = 0, index;
@@ -1396,11 +1428,11 @@ static int ac108_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *i
 		}
 	}
 
-	index = (int)i2c_id->driver_data;
+	index = (int)i2c_match_id(ac108_i2c_id, i2c)->driver_data;
 	if (index == AC101_I2C_ID) {
 		ac10x->i2c101 = i2c;
 		i2c_set_clientdata(i2c, ac10x);
-		ret = ac101_probe(i2c, i2c_id);
+		ret = ac101_probe(i2c, i2c_match_id(ac108_i2c_id, i2c));
 		if (ret) {
 			ac10x->i2c101 = NULL;
 			return ret;
@@ -1418,8 +1450,8 @@ static int ac108_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *i
 	if (of_property_read_u32(np, "tdm-chips-count", &val)) val = 1;
 	ac10x->tdm_chips_cnt = val;
 
-	pr_err(" ac10x i2c_id number: %d\n", index);
-	pr_err(" ac10x data protocol: %d\n", ac10x->data_protocol);
+	pr_info(" ac10x i2c_id number: %d\n", index);
+	pr_info(" ac10x data protocol: %d\n", ac10x->data_protocol);
 
 	ac10x->i2c[index] = i2c;
 	ac10x->i2cmap[index] = devm_regmap_init_i2c(i2c, &ac108_regmap);
@@ -1429,17 +1461,19 @@ static int ac108_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *i
 		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_err(" ac10x codec count  : %d\n", ac10x->codec_cnt);
+	pr_info(" ac10x codec count  : %d\n", ac10x->codec_cnt);
 
 	ret = sysfs_create_group(&i2c->dev.kobj, &ac108_debug_attr_group);
 	if (ret) {
@@ -1464,7 +1498,7 @@ __ret:
 	return ret;
 }
 
-static int ac108_i2c_remove(struct i2c_client *i2c) {
+static void ac108_i2c_remove(struct i2c_client *i2c) {
 	if (ac10x->codec != NULL) {
 		snd_soc_unregister_codec(&ac10x->i2c[_MASTER_INDEX]->dev);
 		ac10x->codec = NULL;
@@ -1489,17 +1523,8 @@ __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[] = {

ac108_6mic.state

@@ -69,7 +69,7 @@ state.seeed8micvoicec {
 	control.1 {
 		iface MIXER
 		name 'CH1 digital volume'
-		value 166
+		value 208
 		comment {
 			access 'read write'
 			type INTEGER
@@ -77,13 +77,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 525
+			dbvalue.0 3675
 		}
 	}
 	control.2 {
 		iface MIXER
 		name 'CH2 digital volume'
-		value 166
+		value 208
 		comment {
 			access 'read write'
 			type INTEGER
@@ -91,13 +91,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 525
+			dbvalue.0 3675
 		}
 	}
 	control.3 {
 		iface MIXER
 		name 'CH3 digital volume'
-		value 166
+		value 208
 		comment {
 			access 'read write'
 			type INTEGER
@@ -105,13 +105,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 525
+			dbvalue.0 3675
 		}
 	}
 	control.4 {
 		iface MIXER
 		name 'CH4 digital volume'
-		value 166
+		value 208
 		comment {
 			access 'read write'
 			type INTEGER
@@ -119,13 +119,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 525
+			dbvalue.0 3675
 		}
 	}
 	control.5 {
 		iface MIXER
 		name 'ADC1 PGA gain'
-		value 25
+		value 0
 		comment {
 			access 'read write'
 			type INTEGER
@@ -133,13 +133,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 2500
+			dbvalue.0 0
 		}
 	}
 	control.6 {
 		iface MIXER
 		name 'ADC2 PGA gain'
-		value 25
+		value 0
 		comment {
 			access 'read write'
 			type INTEGER
@@ -147,13 +147,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 2500
+			dbvalue.0 0
 		}
 	}
 	control.7 {
 		iface MIXER
 		name 'ADC3 PGA gain'
-		value 25
+		value 0
 		comment {
 			access 'read write'
 			type INTEGER
@@ -161,13 +161,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 2500
+			dbvalue.0 0
 		}
 	}
 	control.8 {
 		iface MIXER
 		name 'ADC4 PGA gain'
-		value 25
+		value 0
 		comment {
 			access 'read write'
 			type INTEGER
@@ -175,13 +175,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 2500
+			dbvalue.0 0
 		}
 	}
 	control.9 {
 		iface MIXER
 		name 'CH5 digital volume'
-		value 165
+		value 208
 		comment {
 			access 'read write'
 			type INTEGER
@@ -189,13 +189,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 450
+			dbvalue.0 3675
 		}
 	}
 	control.10 {
 		iface MIXER
 		name 'CH6 digital volume'
-		value 165
+		value 208
 		comment {
 			access 'read write'
 			type INTEGER
@@ -203,13 +203,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 450
+			dbvalue.0 3675
 		}
 	}
 	control.11 {
 		iface MIXER
 		name 'CH7 digital volume'
-		value 165
+		value 198
 		comment {
 			access 'read write'
 			type INTEGER
@@ -217,13 +217,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 450
+			dbvalue.0 2925
 		}
 	}
 	control.12 {
 		iface MIXER
 		name 'CH8 digital volume'
-		value 165
+		value 198
 		comment {
 			access 'read write'
 			type INTEGER
@@ -231,13 +231,13 @@ state.seeed8micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 450
+			dbvalue.0 2925
 		}
 	}
 	control.13 {
 		iface MIXER
 		name 'ADC5 PGA gain'
-		value 25
+		value 0
 		comment {
 			access 'read write'
 			type INTEGER
@@ -245,13 +245,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 2500
+			dbvalue.0 0
 		}
 	}
 	control.14 {
 		iface MIXER
 		name 'ADC6 PGA gain'
-		value 25
+		value 0
 		comment {
 			access 'read write'
 			type INTEGER
@@ -259,13 +259,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 2500
+			dbvalue.0 0
 		}
 	}
 	control.15 {
 		iface MIXER
 		name 'ADC7 PGA gain'
-		value 25
+		value 0
 		comment {
 			access 'read write'
 			type INTEGER
@@ -273,13 +273,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 2500
+			dbvalue.0 0
 		}
 	}
 	control.16 {
 		iface MIXER
 		name 'ADC8 PGA gain'
-		value 25
+		value 0
 		comment {
 			access 'read write'
 			type INTEGER
@@ -287,7 +287,7 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 2500
+			dbvalue.0 0
 		}
 	}
 	control.17 {
@@ -325,7 +325,7 @@ state.seeed8micvoicec {
 	control.19 {
 		iface MIXER
 		name 'digital volume'
-		value 49
+		value 51
 		comment {
 			access 'read write'
 			type INTEGER
@@ -333,13 +333,13 @@ state.seeed8micvoicec {
 			range '0 - 63'
 			dbmin -7308
 			dbmax 0
-			dbvalue.0 -1624
+			dbvalue.0 -1392
 		}
 	}
 	control.20 {
 		iface MIXER
-		name 'speaker volume'
-		value 18
+		name 'Speaker Playback Volume'
+		value 25
 		comment {
 			access 'read write'
 			type INTEGER
@@ -347,13 +347,13 @@ state.seeed8micvoicec {
 			range '0 - 31'
 			dbmin -4800
 			dbmax -150
-			dbvalue.0 -2100
+			dbvalue.0 -1050
 		}
 	}
 	control.21 {
 		iface MIXER
-		name 'headphone volume'
-		value 48
+		name 'Headphone Playback Volume'
+		value 52
 		comment {
 			access 'read write'
 			type INTEGER
@@ -361,7 +361,7 @@ state.seeed8micvoicec {
 			range '0 - 63'
 			dbmin -6300
 			dbmax 0
-			dbvalue.0 -1500
+			dbvalue.0 -1100
 		}
 	}
 }

ac108_asound.state

@@ -69,7 +69,7 @@ state.seeed4micvoicec {
 	control.1 {
 		iface MIXER
 		name 'CH1 digital volume'
-		value 172
+		value 222
 		comment {
 			access 'read write'
 			type INTEGER
@@ -77,13 +77,13 @@ state.seeed4micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 975
+			dbvalue.0 4725
 		}
 	}
 	control.2 {
 		iface MIXER
 		name 'CH2 digital volume'
-		value 172
+		value 222
 		comment {
 			access 'read write'
 			type INTEGER
@@ -91,13 +91,13 @@ state.seeed4micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 975
+			dbvalue.0 4725
 		}
 	}
 	control.3 {
 		iface MIXER
 		name 'CH3 digital volume'
-		value 172
+		value 222
 		comment {
 			access 'read write'
 			type INTEGER
@@ -105,13 +105,13 @@ state.seeed4micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 975
+			dbvalue.0 4725
 		}
 	}
 	control.4 {
 		iface MIXER
 		name 'CH4 digital volume'
-		value 172
+		value 222
 		comment {
 			access 'read write'
 			type INTEGER
@@ -119,13 +119,13 @@ state.seeed4micvoicec {
 			range '0 - 255'
 			dbmin -11925
 			dbmax 7200
-			dbvalue.0 975
+			dbvalue.0 4725
 		}
 	}
 	control.5 {
 		iface MIXER
 		name 'ADC1 PGA gain'
-		value 31
+		value 0
 		comment {
 			access 'read write'
 			type INTEGER
@@ -133,13 +133,13 @@ state.seeed4micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 3100
+			dbvalue.0 0
 		}
 	}
 	control.6 {
 		iface MIXER
 		name 'ADC2 PGA gain'
-		value 31
+		value 0
 		comment {
 			access 'read write'
 			type INTEGER
@@ -147,13 +147,13 @@ state.seeed4micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 3100
+			dbvalue.0 0
 		}
 	}
 	control.7 {
 		iface MIXER
 		name 'ADC3 PGA gain'
-		value 31
+		value 0
 		comment {
 			access 'read write'
 			type INTEGER
@@ -161,13 +161,13 @@ state.seeed4micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 3100
+			dbvalue.0 0
 		}
 	}
 	control.8 {
 		iface MIXER
 		name 'ADC4 PGA gain'
-		value 31
+		value 0
 		comment {
 			access 'read write'
 			type INTEGER
@@ -175,7 +175,7 @@ state.seeed4micvoicec {
 			range '0 - 31'
 			dbmin 0
 			dbmax 3100
-			dbvalue.0 3100
+			dbvalue.0 0
 		}
 	}
 }

ac10x.h

@@ -29,14 +29,19 @@
 /* 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] "format,##args)
+    #define AC101_DBG(format,args...)  printk("[AC101] %s() L%d " format, __func__, __LINE__, ##args)
 #else
     #define AC101_DBG(...)
 #endif
 
 
+#include "sound-compatible-4.18.h"
+
 #ifdef CONFIG_AC101_SWITCH_DETECT
 enum headphone_mode_u {
 	HEADPHONE_IDLE,
@@ -75,7 +80,7 @@ struct ac10x_priv {
 
 	#ifdef CONFIG_AC101_SWITCH_DETECT
 	struct gpio_desc* gpiod_irq;
-	int irq;
+	long irq;
 	volatile int irq_cntr;
 	volatile int pullout_cntr;
 	volatile int state;
@@ -114,7 +119,7 @@ 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));
+int seeed_voice_card_register_set_clock(int stream, int (*set_clock)(int, struct snd_pcm_substream *, int, struct snd_soc_dai *));
 
 int ac10x_fill_regcache(struct device* dev, struct regmap* map);
 

asound_2mic.conf

@@ -1,6 +1,10 @@
 # The IPC key of dmix or dsnoop plugin must be unique
 # If 555555 or 666666 is used by other processes, use another one
 
+
+# use samplerate to resample as speexdsp resample is bad
+defaults.pcm.rate_converter "samplerate"
+
 pcm.!default {
     type asym
     playback.pcm "playback"
@@ -19,14 +23,14 @@ pcm.capture {
 
 pcm.dmixed {
     type dmix
-    slave.pcm "hw:0,0"
+    slave.pcm "hw:seeed2micvoicec"
     ipc_key 555555 
 }
 
 pcm.array {
     type dsnoop
     slave {
-        pcm "hw:0,0"
+        pcm "hw:seeed2micvoicec"
         channels 2
     }
     ipc_key 666666

asound_4mic.conf

@@ -1,6 +1,9 @@
 # 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"
@@ -9,22 +12,22 @@ pcm.!default {
 
 pcm.playback {
     type plug
-    slave.pcm "hw:0,0"
+    slave.pcm "hw:ALSA"
 }
 
-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 "multiapps"
+    slave.pcm "hw:seeed4micvoicec"
 }
 
-pcm.multiapps {
-    type dsnoop
-    ac108-slavepcm "hw:1,0"
-    ipc_key 666666
-}
+# pcm.multiapps {
+#     type dsnoop
+#     ac108-slavepcm "hw:1,0"
+#     ipc_key 666666
+# }

asound_6mic.conf

@@ -1,61 +1,90 @@
 # The IPC key of dmix or dsnoop plugin must be unique
 # If 555555 or 666666 is used by other processes, use another one
 
+# use samplerate to resample as speexdsp resample is broken
+defaults.pcm.rate_converter "samplerate"
+
 pcm.!default {
     type asym
-    playback.pcm "ac101"
+    playback.pcm "dmixer"
     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.pcm "multiapps"
+    slave {
+        rate 48000
+        format S32_LE
+        pcm "hw:seeed8micvoicec"
+    }
 }
 
-pcm.multiapps {
-    type dsnoop
-    slave.pcm "hw:1,0"
-    ipc_key 666666
+# pcm.multiapps {
+#     type plug
+#     slave.pcm {
+#         type dsnoop
+#         slave {
+#             rate 48000
+#             format S32_LE
+#             pcm "hw:seeed8micvoicec"
+#         }
+#         ipc_key 666666
+#     }
+# }
+
+pcm.dmixer {
+    type plug
+    slave {
+        pcm {
+            type dmix
+            ipc_key 555555
+            slave {	
+                pcm "hw:seeed8micvoicec"
+                format S32_LE
+                channels 8
+            }
+            bindings {
+                0 0
+                1 1
+                2 2
+                3 3
+                4 4
+                5 5
+                6 6
+                7 7
+            }
+        }
+        channels 8
+        format S32_LE
+        rate 48000
+    }
+    ttable.0.0 1
+    ttable.1.1 1
+    ttable.0.2 1
+    ttable.1.3 1
+    ttable.0.4 1
+    ttable.1.5 1
+    ttable.0.6 1
+    ttable.1.7 1
 }
 
 pcm.ac101 {
     type plug
     slave {
-		pcm {
-			type dmix
-			ipc_key 1048576
-				slave {
-				pcm		"hw:1,0"
-				format		S32_LE
-				# rate		16000
+        pcm "hw:seeed8micvoicec"
         channels 8
+        format S32_LE
+        rate 48000
     }
-			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
-	}
+    ttable.0.0 1
+    ttable.1.1 1
+    ttable.0.2 1
+    ttable.1.3 1
+    ttable.0.4 1
+    ttable.1.5 1
+    ttable.0.6 1
+    ttable.1.7 1
 }
 
 

builddtbo.sh

@@ -1,8 +1,10 @@
+#!/bin/sh
 #dtoverlay -r seeed-2mic-voicecard
+DTC_FLAGS="-b 0 -Wno-unit_address_vs_reg -I dts -O dtb"
 
-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
+dtc -@ $DTC_FLAGS -o seeed-2mic-voicecard.dtbo seeed-2mic-voicecard-overlay.dts
+dtc -@ $DTC_FLAGS -o seeed-4mic-voicecard.dtbo seeed-4mic-voicecard-overlay.dts
+dtc -@ $DTC_FLAGS -o seeed-8mic-voicecard.dtbo seeed-8mic-voicecard-overlay.dts
 
 # cp *.dtbo /boot/overlays
 # dtoverlay seeed-2mic-voicecard

dkms.conf

@@ -6,4 +6,10 @@ BUILT_MODULE_NAME[2]="snd-soc-seeed-voicecard"
 DEST_MODULE_LOCATION[0]="/kernel/sound/soc/codecs"
 DEST_MODULE_LOCATION[1]="/kernel/sound/soc/codecs"
 DEST_MODULE_LOCATION[2]="/kernel/sound/soc/bcm"
+PATCH[0]="back-to-v4.19.diff"
+PATCH[1]="back-to-v5.4.diff"
+PATCH[2]="back-to-v5.8.diff"
+PATCH_MATCH[0]="^4\.19\.*"
+PATCH_MATCH[1]="^5\.4\.*"
+PATCH_MATCH[2]="^5\.8\.*"
 AUTOINSTALL="yes"

install.sh

@@ -5,29 +5,134 @@ if [[ $EUID -ne 0 ]]; then
    exit 1
 fi
 
-is_Raspberry=$(cat /proc/device-tree/model | awk  '{print $1}')
-if [ "x${is_Raspberry}" != "xRaspberry" ] ; then
-  echo "Sorry, this drivers only works on raspberry pi"
+# 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
   exit 1
 fi
 
 ver="0.3"
-
+uname_r=$(uname -r)
 
 # we create a dir with this version to ensure that 'dkms remove' won't delete
 # the sources during kernel updates
 marker="0.0.0"
 
-apt update
-apt-get -y install raspberrypi-kernel-headers raspberrypi-kernel 
-apt-get -y install  dkms git i2c-tools
+_VER_RUN=
+function get_kernel_version() {
+  local ZIMAGE IMG_OFFSET
+
+  _VER_RUN=""
+  [ -z "$_VER_RUN" ] && {
+    ZIMAGE=/boot/kernel.img
+    [ -f /boot/firmware/vmlinuz ] && ZIMAGE=/boot/firmware/vmlinuz
+    # 64-bit-only kernel package
+    [ ! -f /boot/kernel.img ] && [ -f /boot/kernel8.img ] && ZIMAGE=/boot/kernel8.img
+    IMG_OFFSET=$(LC_ALL=C grep -abo $'\x1f\x8b\x08\x00' $ZIMAGE | head -n 1 | cut -d ':' -f 1)
+    _VER_RUN=$(dd if=$ZIMAGE obs=64K ibs=4 skip=$(( IMG_OFFSET / 4)) 2>/dev/null | zcat | grep -a -m1 "Linux version" | LC_ALL=C sed -e 's/^.*Linux/Linux/' | strings | awk '{ print $3; }')
+  }
+  echo "$_VER_RUN"
+  return 0
+}
+
+function check_kernel_headers() {
+  VER_RUN=$(get_kernel_version)
+  VER_HDR=$(dpkg -L raspberrypi-kernel-headers | egrep -m1 "/lib/modules/[^\/]+/build" | awk -F'/' '{ print $4; }')
+  [ "X$VER_RUN" == "X$VER_HDR" ] && {
+    return 0
+  }
+  VER_HDR=$(dpkg -L linux-headers-$VER_RUN | egrep -m1 "/lib/modules/[^\/]+/build" | awk -F'/' '{ print $4; }')
+  [ "X$VER_RUN" == "X$VER_HDR" ] && {
+    return 0
+  }
+
+  # echo RUN=$VER_RUN HDR=$VER_HDR
+  echo " !!! Your kernel version is $VER_RUN"
+  echo "     Not found *** corresponding *** kernel headers with apt-get."
+  echo "     This may occur if you have ran 'rpi-update'."
+  echo " Choose  *** y *** will revert the kernel to version $VER_HDR then continue."
+  echo " Choose  *** N *** will exit without this driver support, by default."
+  read -p "Would you like to proceed? (y/N)" -n 1 -r -s
+  echo
+  if ! [[ $REPLY =~ ^[Yy]$ ]]; then
+    exit 1;
+  fi
+
+  apt-get -y --reinstall install raspberrypi-kernel
+}
+
+# update and install required packages
+which apt &>/dev/null
+if [[ $? -eq 0 ]]; then
+  apt update -y
+  # Raspbian kernel packages
+  apt-get -y install raspberrypi-kernel-headers raspberrypi-kernel
+  # Recent Raspbian has 64-bit kernel on 32-bit userspace
+  apt-get -y install gcc-aarch64-linux-gnu
+  # Ubuntu kernel packages
+  apt-get -y install linux-raspi linux-headers-raspi linux-image-raspi
+  apt-get -y install dkms git i2c-tools libasound2-plugins
+  # rpi-update checker
+  check_kernel_headers
+fi
+
+# Arch Linux
+which pacman &>/dev/null
+if [[ $? -eq 0 ]]; then
+  pacman -Syu --needed git gcc automake make dkms linux-raspberrypi-headers i2c-tools
+fi
 
 # locate currently installed kernels (may be different to running kernel if
 # it's just been updated)
-kernels=$(ls /lib/modules | sed "s/^/-k /")
-uname_r=$(uname -r)
+base_ver=$(get_kernel_version)
+base_ver=${base_ver%%[-+]*}
+#kernels="${base_ver}+ ${base_ver}-v7+ ${base_ver}-v7l+"
+kernels=$(uname -r)
 
 function install_module {
+  local _i
+
   src=$1
   mod=$2
 
@@ -39,10 +144,16 @@ 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
-  dkms build $kernels -m $mod -v $ver && dkms install --force $kernels -m $mod -v $ver
+  for _i in $kernels; do
+    dkms build -k $_i -m $mod -v $ver && {
+      dkms install --force -k $_i -m $mod -v $ver
+    }
+  done
 
   mkdir -p /var/lib/dkms/$mod/$ver/$marker
 }
@@ -51,40 +162,60 @@ install_module "./" "seeed-voicecard"
 
 
 # install dtbos
-cp seeed-2mic-voicecard.dtbo /boot/overlays
-cp seeed-4mic-voicecard.dtbo /boot/overlays
-cp seeed-8mic-voicecard.dtbo /boot/overlays
+cp seeed-2mic-voicecard.dtbo $OVERLAYS
+cp seeed-4mic-voicecard.dtbo $OVERLAYS
+cp seeed-8mic-voicecard.dtbo $OVERLAYS
 
 #install alsa plugins
 # no need this plugin now
-# install -D ac108_plugin/libasound_module_pcm_ac108.so   /usr/lib/arm-linux-gnueabihf/alsa-lib/libasound_module_pcm_ac108.so
+# install -D ac108_plugin/libasound_module_pcm_ac108.so /usr/lib/arm-linux-gnueabihf/alsa-lib/
+rm -f /usr/lib/arm-linux-gnueabihf/alsa-lib/libasound_module_pcm_ac108.so
 
-#set kernel moduels
-grep -q "snd-soc-seeed-voicecard" /etc/modules || \
+#set kernel modules
+grep -q "^snd-soc-seeed-voicecard$" /etc/modules || \
   echo "snd-soc-seeed-voicecard" >> /etc/modules
-grep -q "snd-soc-ac108" /etc/modules || \
+grep -q "^snd-soc-ac108$" /etc/modules || \
   echo "snd-soc-ac108" >> /etc/modules
-grep -q "snd-soc-wm8960" /etc/modules || \
+grep -q "^snd-soc-wm8960$" /etc/modules || \
   echo "snd-soc-wm8960" >> /etc/modules
 
 #set dtoverlays
-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
+CONFIG=/boot/config.txt
+[ -f /boot/firmware/config.txt ] && CONFIG=/boot/firmware/config.txt
+[ -f /boot/firmware/usercfg.txt ] && CONFIG=/boot/firmware/usercfg.txt
+
+sed -i -e 's:#dtparam=i2c_arm=on:dtparam=i2c_arm=on:g'  $CONFIG || true
+grep -q "^dtoverlay=i2s-mmap$" $CONFIG || \
+  echo "dtoverlay=i2s-mmap" >> $CONFIG
 
 
-grep -q "dtparam=i2s=on" /boot/config.txt || \
-  echo "dtparam=i2s=on" >> /boot/config.txt
+grep -q "^dtparam=i2s=on$" $CONFIG || \
+  echo "dtparam=i2s=on" >> $CONFIG
 
 #install config files
 mkdir /etc/voicecard || true
 cp *.conf /etc/voicecard
 cp *.state /etc/voicecard
 
+#create git repo
+git_email=$(git config --global --get user.email)
+git_name=$(git config --global --get user.name)
+if [ "x${git_email}" == "x" ] || [ "x${git_name}" == "x" ] ; then
+    echo "setup git config"
+    git config --global user.email "respeaker@seeed.cc"
+    git config --global user.name "respeaker"
+fi
+echo "git init"
+git --git-dir=/etc/voicecard/.git init
+echo "git add --all"
+git --git-dir=/etc/voicecard/.git --work-tree=/etc/voicecard/ add --all
+echo "git commit -m \"origin configures\""
+git --git-dir=/etc/voicecard/.git --work-tree=/etc/voicecard/ commit  -m "origin configures"
 
 cp seeed-voicecard /usr/bin/
 cp seeed-voicecard.service /lib/systemd/system/
 systemctl enable  seeed-voicecard.service
+systemctl start   seeed-voicecard
 
 echo "------------------------------------------------------"
 echo "Please reboot your raspberry pi to apply all settings"

patches/back-to-v4.19.diff

@@ -0,0 +1,454 @@
+diff --git a/ac101.c b/ac101.c
+index 23837a7..41c15f3 100644
+--- a/ac101.c
++++ b/ac101.c
+@@ -955,10 +955,10 @@ void ac101_aif_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai
+ 
+ 	AC101_DBG("stream = %s, play: %d, capt: %d, active: %d\n", 
+ 		snd_pcm_stream_str(substream),
+-		codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE],
+-		snd_soc_dai_active(codec_dai));
++		codec_dai->playback_active, codec_dai->capture_active,
++		codec_dai->active);
+ 
+-	if (!snd_soc_dai_active(codec_dai)) {
++	if (!codec_dai->active) {
+ 		ac10x->aif1_clken = 1;
+ 		ac101_aif1clk(codec, SND_SOC_DAPM_POST_PMD, 0);
+ 	} else {
+@@ -1080,7 +1080,7 @@ int ac101_hw_params(struct snd_pcm_substream *substream,
+ 	freq_out = _FREQ_24_576K;
+ 	for (i = 0; i < ARRAY_SIZE(codec_aif1_fs); i++) {
+ 		if (codec_aif1_fs[i].samp_rate == params_rate(params)) {
+-			if (codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] && dmic_used && codec_aif1_fs[i].samp_rate == 44100) {
++			if (codec_dai->capture_active && dmic_used && codec_aif1_fs[i].samp_rate == 44100) {
+ 				ac101_update_bits(codec, AIF_SR_CTRL, (0xf<<AIF1_FS), (0x4<<AIF1_FS));
+ 			} else {
+ 				ac101_update_bits(codec, AIF_SR_CTRL, (0xf<<AIF1_FS), ((codec_aif1_fs[i].srbit)<<AIF1_FS));
+diff --git a/ac108.c b/ac108.c
+index d5dd12d..0a8c462 100644
+--- a/ac108.c
++++ b/ac108.c
+@@ -653,7 +653,7 @@ static int ac108_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_h
+ 
+ 	dev_dbg(dai->dev, "%s() stream=%s play:%d capt:%d +++\n", __func__,
+ 			snd_pcm_stream_str(substream),
+-			dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]);
++			dai->playback_active, dai->capture_active);
+ 
+ 	if (ac10x->i2c101) {
+ 		ret = ac101_hw_params(substream, params, dai);
+@@ -664,8 +664,8 @@ static int ac108_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_h
+ 		}
+ 	}
+ 
+-	if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE && dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])
+-	 || (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && dai->stream_active[SNDRV_PCM_STREAM_CAPTURE])) {
++	if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE && dai->playback_active)
++	 || (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && dai->capture_active)) {
+ 		/* not configure hw_param twice */
+ 		/* return 0; */
+ 	}
+@@ -810,9 +810,6 @@ static int ac108_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int fr
+ 
+ 	struct ac10x_priv *ac10x = snd_soc_dai_get_drvdata(dai);
+ 
+-	if (freq != 24000000 || clk_id != SYSCLK_SRC_PLL)
+-		dev_warn(dai->dev, "ac108_set_sysclk freq = %d clk = %d\n", freq, clk_id);
+-
+ 	freq = 24000000;
+ 	clk_id = SYSCLK_SRC_PLL;
+ 
+@@ -1124,7 +1121,7 @@ void ac108_aif_shutdown(struct snd_pcm_substream *substream,
+ 	}
+ }
+ 
+-int ac108_aif_mute(struct snd_soc_dai *dai, int mute, int direction) {
++int ac108_aif_mute(struct snd_soc_dai *dai, int mute) {
+ 	struct snd_soc_codec *codec = dai->codec;
+ 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
+ 
+@@ -1145,13 +1142,12 @@ static const struct snd_soc_dai_ops ac108_dai_ops = {
+ 	.hw_params	= ac108_hw_params,
+ 	.prepare	= ac108_prepare,
+ 	.trigger	= ac108_trigger,
+-	.mute_stream	= ac108_aif_mute,
++	.digital_mute	= ac108_aif_mute,
+ 
+ 	/*DAI format configuration*/
+ 	.set_fmt	= ac108_set_fmt,
+ 
+ 	// .hw_free = ac108_hw_free,
+-	.no_capture_mute = 1,
+ };
+ 
+ static  struct snd_soc_dai_driver ac108_dai0 = {
+diff --git a/seeed-voicecard.c b/seeed-voicecard.c
+index b90af93..af6db74 100644
+--- a/seeed-voicecard.c
++++ b/seeed-voicecard.c
+@@ -28,8 +28,6 @@
+ #include <sound/simple_card_utils.h>
+ #include "ac10x.h"
+ 
+-#define LINUX_VERSION_IS_GEQ(x1,x2,x3)	(LINUX_VERSION_CODE >= KERNEL_VERSION(x1,x2,x3))
+-
+ /*
+  * single codec:
+  *	0 - allow multi codec
+@@ -42,9 +40,6 @@ struct seeed_card_data {
+ 	struct seeed_dai_props {
+ 		struct asoc_simple_dai cpu_dai;
+ 		struct asoc_simple_dai codec_dai;
+-		struct snd_soc_dai_link_component cpus;   /* single cpu */
+-		struct snd_soc_dai_link_component codecs; /* single codec */
+-		struct snd_soc_dai_link_component platforms;
+ 		unsigned int mclk_fs;
+ 	} *dai_props;
+ 	unsigned int mclk_fs;
+@@ -97,16 +92,16 @@ static int seeed_voice_card_startup(struct snd_pcm_substream *substream)
+ 	if (ret)
+ 		clk_disable_unprepare(dai_props->cpu_dai.clk);
+ 
+-	if (asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min) {
+-		priv->channels_playback_default = asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min;
++	if (rtd->cpu_dai->driver->playback.channels_min) {
++		priv->channels_playback_default = rtd->cpu_dai->driver->playback.channels_min;
+ 	}
+-	if (asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min) {
+-		priv->channels_capture_default = asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min;
++	if (rtd->cpu_dai->driver->capture.channels_min) {
++		priv->channels_capture_default = rtd->cpu_dai->driver->capture.channels_min;
+ 	}
+-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min = priv->channels_playback_override;
+-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_max = priv->channels_playback_override;
+-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min = priv->channels_capture_override;
+-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_max = priv->channels_capture_override;
++	rtd->cpu_dai->driver->playback.channels_min = priv->channels_playback_override;
++	rtd->cpu_dai->driver->playback.channels_max = priv->channels_playback_override;
++	rtd->cpu_dai->driver->capture.channels_min = priv->channels_capture_override;
++	rtd->cpu_dai->driver->capture.channels_max = priv->channels_capture_override;
+ 
+ 	return ret;
+ }
+@@ -118,10 +113,10 @@ static void seeed_voice_card_shutdown(struct snd_pcm_substream *substream)
+ 	struct seeed_dai_props *dai_props =
+ 		seeed_priv_to_props(priv, rtd->num);
+ 
+-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min = priv->channels_playback_default;
+-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_max = priv->channels_playback_default;
+-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min = priv->channels_capture_default;
+-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_max = priv->channels_capture_default;
++	rtd->cpu_dai->driver->playback.channels_min = priv->channels_playback_default;
++	rtd->cpu_dai->driver->playback.channels_max = priv->channels_playback_default;
++	rtd->cpu_dai->driver->capture.channels_min = priv->channels_capture_default;
++	rtd->cpu_dai->driver->capture.channels_max = priv->channels_capture_default;
+ 
+ 	clk_disable_unprepare(dai_props->cpu_dai.clk);
+ 
+@@ -132,8 +127,8 @@ static int seeed_voice_card_hw_params(struct snd_pcm_substream *substream,
+ 				      struct snd_pcm_hw_params *params)
+ {
+ 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+-	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+-	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
++	struct snd_soc_dai *codec_dai = rtd->codec_dai;
++	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ 	struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
+ 	struct seeed_dai_props *dai_props =
+ 		seeed_priv_to_props(priv, rtd->num);
+@@ -197,7 +192,7 @@ static void work_cb_codec_clk(struct work_struct *work)
+ static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd)
+ {
+ 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+-	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
++	struct snd_soc_dai *dai = rtd->codec_dai;
+ 	struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
+ 	#if CONFIG_AC10X_TRIG_LOCK
+ 	unsigned long flags;
+@@ -206,7 +201,7 @@ static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd
+ 
+ 	dev_dbg(rtd->card->dev, "%s() stream=%s  cmd=%d play:%d, capt:%d\n",
+ 		__FUNCTION__, snd_pcm_stream_str(substream), cmd,
+-		dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]);
++		dai->playback_active, dai->capture_active);
+ 
+ 	switch (cmd) {
+ 	case SNDRV_PCM_TRIGGER_START:
+@@ -228,7 +223,7 @@ static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd
+ 	case SNDRV_PCM_TRIGGER_SUSPEND:
+ 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ 		/* capture channel resync, if overrun */
+-		if (dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] && substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
++		if (dai->capture_active && substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ 			break;
+ 		}
+ 
+@@ -248,7 +243,7 @@ static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd
+ 
+ 	dev_dbg(rtd->card->dev, "%s() stream=%s  cmd=%d play:%d, capt:%d;finished %d\n",
+ 		__FUNCTION__, snd_pcm_stream_str(substream), cmd,
+-		dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], dai->stream_active[SNDRV_PCM_STREAM_CAPTURE], ret);
++		dai->playback_active, dai->capture_active, ret);
+ 
+ 	return ret;
+ }
+@@ -387,8 +382,8 @@ static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd)
+ static int seeed_voice_card_dai_init(struct snd_soc_pcm_runtime *rtd)
+ {
+ 	struct seeed_card_data *priv =	snd_soc_card_get_drvdata(rtd->card);
+-	struct snd_soc_dai *codec = asoc_rtd_to_codec(rtd, 0);
+-	struct snd_soc_dai *cpu = asoc_rtd_to_cpu(rtd, 0);
++	struct snd_soc_dai *codec = rtd->codec_dai;
++	struct snd_soc_dai *cpu = rtd->cpu_dai;
+ 	struct seeed_dai_props *dai_props =
+ 		seeed_priv_to_props(priv, rtd->num);
+ 	int ret;
+@@ -453,19 +448,20 @@ static int seeed_voice_card_dai_link_of(struct device_node *node,
+ 		goto dai_link_of_err;
+ 	}
+ 
+-	ret = asoc_simple_parse_daifmt(dev, node, codec,
++	ret = asoc_simple_card_parse_daifmt(dev, node, codec,
+ 					    prefix, &dai_link->dai_fmt);
+ 	if (ret < 0)
+ 		goto dai_link_of_err;
+ 
+ 	of_property_read_u32(node, "mclk-fs", &dai_props->mclk_fs);
+ 
+-	ret = asoc_simple_parse_cpu(cpu, dai_link, &single_cpu);
++	ret = asoc_simple_card_parse_cpu(cpu, dai_link,
++					 DAI, CELL, &single_cpu);
+ 	if (ret < 0)
+ 		goto dai_link_of_err;
+ 
+ 	#if _SINGLE_CODEC
+-	ret = asoc_simple_parse_codec(codec, dai_link);
++	ret = asoc_simple_card_parse_codec(codec, dai_link, DAI, CELL);
+ 	if (ret < 0)
+ 		goto dai_link_of_err;
+ 	#else
+@@ -477,7 +473,7 @@ static int seeed_voice_card_dai_link_of(struct device_node *node,
+ 	dev_dbg(dev, "dai_link num_codecs = %d\n", dai_link->num_codecs);
+ 	#endif
+ 
+-	ret = asoc_simple_parse_platform(plat, dai_link);
++	ret = asoc_simple_card_parse_platform(plat, dai_link, DAI, CELL);
+ 	if (ret < 0)
+ 		goto dai_link_of_err;
+ 
+@@ -502,7 +498,7 @@ static int seeed_voice_card_dai_link_of(struct device_node *node,
+ 	#if LINUX_VERSION_CODE <= KERNEL_VERSION(4,10,0)
+ 	ret = asoc_simple_card_parse_clk_cpu(cpu, dai_link, cpu_dai);
+ 	#else
+-	ret = asoc_simple_parse_clk_cpu(dev, cpu, dai_link, cpu_dai);
++	ret = asoc_simple_card_parse_clk_cpu(dev, cpu, dai_link, cpu_dai);
+ 	#endif
+ 	if (ret < 0)
+ 		goto dai_link_of_err;
+@@ -510,16 +506,16 @@ static int seeed_voice_card_dai_link_of(struct device_node *node,
+ 	#if LINUX_VERSION_CODE <= KERNEL_VERSION(4,10,0)
+ 	ret = asoc_simple_card_parse_clk_codec(codec, dai_link, codec_dai);
+ 	#else
+-	ret = asoc_simple_parse_clk_codec(dev, codec, dai_link, codec_dai);
++	ret = asoc_simple_card_parse_clk_codec(dev, codec, dai_link, codec_dai);
+ 	#endif
+ 	if (ret < 0)
+ 		goto dai_link_of_err;
+ 
+-	ret = asoc_simple_set_dailink_name(dev, dai_link,
++	ret = asoc_simple_card_set_dailink_name(dev, dai_link,
+ 						"%s-%s",
+-						dai_link->cpus->dai_name,
++						dai_link->cpu_dai_name,
+ 						#if _SINGLE_CODEC
+-						dai_link->codecs->dai_name
++						dai_link->codec_dai_name
+ 						#else
+ 						dai_link->codecs[0].dai_name
+ 						#endif
+@@ -533,19 +529,21 @@ static int seeed_voice_card_dai_link_of(struct device_node *node,
+ 	dev_dbg(dev, "\tname : %s\n", dai_link->stream_name);
+ 	dev_dbg(dev, "\tformat : %04x\n", dai_link->dai_fmt);
+ 	dev_dbg(dev, "\tcpu : %s / %d\n",
+-		dai_link->cpus->dai_name,
++		dai_link->cpu_dai_name,
+ 		dai_props->cpu_dai.sysclk);
+ 	dev_dbg(dev, "\tcodec : %s / %d\n",
+ 		#if _SINGLE_CODEC
+-		dai_link->codecs->dai_name,
++		dai_link->codec_dai_name,
+ 		#else
+ 		dai_link->codecs[0].dai_name,
+ 		#endif
+ 		dai_props->codec_dai.sysclk);
+ 
+-	asoc_simple_canonicalize_cpu(dai_link, single_cpu);
++	asoc_simple_card_canonicalize_cpu(dai_link, single_cpu);
+ 	#if _SINGLE_CODEC
+-	asoc_simple_canonicalize_platform(dai_link);
++	ret = asoc_simple_card_canonicalize_dailink(dai_link);
++	if (ret < 0)
++		goto dai_link_of_err;
+ 	#endif
+ 
+ dai_link_of_err:
+@@ -578,7 +576,7 @@ static int seeed_voice_card_parse_aux_devs(struct device_node *node,
+ 		aux_node = of_parse_phandle(node, PREFIX "aux-devs", i);
+ 		if (!aux_node)
+ 			return -EINVAL;
+-		priv->snd_card.aux_dev[i].dlc.of_node = aux_node;
++		priv->snd_card.aux_dev[i].codec_of_node = aux_node;
+ 	}
+ 
+ 	priv->snd_card.num_aux_devs = n;
+@@ -638,7 +636,7 @@ static int seeed_voice_card_parse_of(struct device_node *node,
+ 			goto card_parse_end;
+ 	}
+ 
+-	ret = asoc_simple_parse_card_name(&priv->snd_card, PREFIX);
++	ret = asoc_simple_card_parse_card_name(&priv->snd_card, PREFIX);
+ 	if (ret < 0)
+ 		goto card_parse_end;
+ 
+@@ -743,7 +741,7 @@ static int seeed_voice_card_probe(struct platform_device *pdev)
+ 	struct seeed_dai_props *dai_props;
+ 	struct device_node *np = pdev->dev.of_node;
+ 	struct device *dev = &pdev->dev;
+-	int num, ret, i;
++	int num, ret;
+ 
+ 	/* Get the number of DAI links */
+ 	if (np && of_get_child_by_name(np, PREFIX "dai-link"))
+@@ -761,25 +759,6 @@ static int seeed_voice_card_probe(struct platform_device *pdev)
+ 	if (!dai_props || !dai_link)
+ 		return -ENOMEM;
+ 
+-	/*
+-	 * Use snd_soc_dai_link_component instead of legacy style
+-	 * It is codec only. but cpu/platform will be supported in the future.
+-	 * see
+-	 *      soc-core.c :: snd_soc_init_multicodec()
+-	 *
+-	 * "platform" might be removed
+-	 * see
+-	 *      simple-card-utils.c :: asoc_simple_canonicalize_platform()
+-	 */
+-	for (i = 0; i < num; i++) {
+-		dai_link[i].cpus		= &dai_props[i].cpus;
+-		dai_link[i].num_cpus		= 1;
+-		dai_link[i].codecs		= &dai_props[i].codecs;
+-		dai_link[i].num_codecs		= 1;
+-		dai_link[i].platforms		= &dai_props[i].platforms;
+-		dai_link[i].num_platforms	= 1;
+-	}
+-
+ 	priv->dai_props			= dai_props;
+ 	priv->dai_link			= dai_link;
+ 
+@@ -798,9 +777,6 @@ static int seeed_voice_card_probe(struct platform_device *pdev)
+ 		}
+ 	} else {
+ 		struct seeed_card_info *cinfo;
+-		struct snd_soc_dai_link_component *cpus;
+-		struct snd_soc_dai_link_component *codecs;
+-		struct snd_soc_dai_link_component *platform;
+ 
+ 		cinfo = dev->platform_data;
+ 		if (!cinfo) {
+@@ -817,19 +793,13 @@ static int seeed_voice_card_probe(struct platform_device *pdev)
+ 			return -EINVAL;
+ 		}
+ 
+-		cpus			= dai_link->cpus;
+-		cpus->dai_name		= cinfo->cpu_dai.name;
+-
+-		codecs			= dai_link->codecs;
+-		codecs->name		= cinfo->codec;
+-		codecs->dai_name	= cinfo->codec_dai.name;
+-
+-		platform		= dai_link->platforms;
+-		platform->name		= cinfo->platform;
+-
+ 		priv->snd_card.name	= (cinfo->card) ? cinfo->card : cinfo->name;
+ 		dai_link->name		= cinfo->name;
+ 		dai_link->stream_name	= cinfo->name;
++		dai_link->platform_name	= cinfo->platform;
++		dai_link->codec_name	= cinfo->codec;
++		dai_link->cpu_dai_name	= cinfo->cpu_dai.name;
++		dai_link->codec_dai_name = cinfo->codec_dai.name;
+ 		dai_link->dai_fmt	= cinfo->daifmt;
+ 		dai_link->init		= seeed_voice_card_dai_init;
+ 		memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai,
+@@ -853,7 +823,7 @@ static int seeed_voice_card_probe(struct platform_device *pdev)
+ 		return ret;
+ 
+ err:
+-	asoc_simple_clean_reference(&priv->snd_card);
++	asoc_simple_card_clean_reference(&priv->snd_card);
+ 
+ 	return ret;
+ }
+@@ -865,7 +835,7 @@ static int seeed_voice_card_remove(struct platform_device *pdev)
+ 
+ 	if (cancel_work_sync(&priv->work_codec_clk) != 0) {
+ 	}
+-	return asoc_simple_clean_reference(card);
++	return asoc_simple_card_clean_reference(card);
+ }
+ 
+ static const struct of_device_id seeed_voice_of_match[] = {
+diff --git a/sound-compatible-4.18.h b/sound-compatible-4.18.h
+index 550b3a7..6c1a014 100644
+--- a/sound-compatible-4.18.h
++++ b/sound-compatible-4.18.h
+@@ -16,6 +16,10 @@
+ #endif
+ 
+ #if LINUX_VERSION_CODE < KERNEL_VERSION(5,4,0)
++#ifndef __has_attribute
++# define __has_attribute(x) __GCC4_has_attribute_##x
++# define __GCC4_has_attribute___fallthrough__         0
++#endif
+ #if __has_attribute(__fallthrough__)
+ # define fallthrough                    __attribute__((__fallthrough__))
+ #else
+@@ -31,11 +35,7 @@
+ #define snd_soc_codec_get_dapm     snd_soc_component_get_dapm
+ #define snd_soc_codec_get_bias_level snd_soc_component_get_bias_level
+ #define snd_soc_kcontrol_codec     snd_soc_kcontrol_component
+-#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,9,0)
+-#define snd_soc_read               snd_soc_component_read
+-#else
+ #define snd_soc_read               snd_soc_component_read32
+-#endif
+ #define snd_soc_register_codec     devm_snd_soc_register_component
+ #define snd_soc_unregister_codec   snd_soc_unregister_component
+ #define snd_soc_update_bits        snd_soc_component_update_bits
+diff --git a/wm8960.c b/wm8960.c
+index 465c6dc..34d4dad 100644
+--- a/wm8960.c
++++ b/wm8960.c
+@@ -796,7 +796,7 @@ static int wm8960_hw_free(struct snd_pcm_substream *substream,
+ 	return 0;
+ }
+ 
+-static int wm8960_mute(struct snd_soc_dai *dai, int mute, int direction)
++static int wm8960_mute(struct snd_soc_dai *dai, int mute)
+ {
+ 	struct snd_soc_codec *codec = dai->codec;
+ 
+@@ -1236,12 +1236,11 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+ static const struct snd_soc_dai_ops wm8960_dai_ops = {
+ 	.hw_params = wm8960_hw_params,
+ 	.hw_free = wm8960_hw_free,
+-	.mute_stream = wm8960_mute,
++	.digital_mute = wm8960_mute,
+ 	.set_fmt = wm8960_set_dai_fmt,
+ 	.set_clkdiv = wm8960_set_dai_clkdiv,
+ 	.set_pll = wm8960_set_dai_pll,
+ 	.set_sysclk = wm8960_set_dai_sysclk,
+-	.no_capture_mute = 1,
+ };
+ 
+ static struct snd_soc_dai_driver wm8960_dai = {

patches/back-to-v5.4.diff

@@ -0,0 +1,245 @@
+diff --git a/ac101.c b/ac101.c
+index be9b1d8..343f030 100644
+--- a/ac101.c
++++ b/ac101.c
+@@ -955,10 +955,10 @@ void ac101_aif_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai
+ 
+ 	AC101_DBG("stream = %s, play: %d, capt: %d, active: %d\n", 
+ 		snd_pcm_stream_str(substream),
+-		codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE],
+-		snd_soc_dai_active(codec_dai));
++		codec_dai->playback_active, codec_dai->capture_active,
++		codec_dai->active);
+ 
+-	if (!snd_soc_dai_active(codec_dai)) {
++	if (!codec_dai->active) {
+ 		ac10x->aif1_clken = 1;
+ 		ac101_aif1clk(codec, SND_SOC_DAPM_POST_PMD, 0);
+ 	} else {
+@@ -1080,7 +1080,7 @@ int ac101_hw_params(struct snd_pcm_substream *substream,
+ 	freq_out = _FREQ_24_576K;
+ 	for (i = 0; i < ARRAY_SIZE(codec_aif1_fs); i++) {
+ 		if (codec_aif1_fs[i].samp_rate == params_rate(params)) {
+-			if (codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] && dmic_used && codec_aif1_fs[i].samp_rate == 44100) {
++			if (codec_dai->capture_active && dmic_used && codec_aif1_fs[i].samp_rate == 44100) {
+ 				ac101_update_bits(codec, AIF_SR_CTRL, (0xf<<AIF1_FS), (0x4<<AIF1_FS));
+ 			} else {
+ 				ac101_update_bits(codec, AIF_SR_CTRL, (0xf<<AIF1_FS), ((codec_aif1_fs[i].srbit)<<AIF1_FS));
+diff --git a/ac108.c b/ac108.c
+index 4663df0..12ab27b 100644
+--- a/ac108.c
++++ b/ac108.c
+@@ -653,7 +653,7 @@ static int ac108_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_h
+ 
+ 	dev_dbg(dai->dev, "%s() stream=%s play:%d capt:%d +++\n", __func__,
+ 			snd_pcm_stream_str(substream),
+-			dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]);
++			dai->playback_active, dai->capture_active);
+ 
+ 	if (ac10x->i2c101) {
+ 		ret = ac101_hw_params(substream, params, dai);
+@@ -664,8 +664,8 @@ static int ac108_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_h
+ 		}
+ 	}
+ 
+-	if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE && dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])
+-	 || (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && dai->stream_active[SNDRV_PCM_STREAM_CAPTURE])) {
++	if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE && dai->playback_active)
++	 || (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && dai->capture_active)) {
+ 		/* not configure hw_param twice */
+ 		/* return 0; */
+ 	}
+@@ -1124,7 +1124,7 @@ void ac108_aif_shutdown(struct snd_pcm_substream *substream,
+ 	}
+ }
+ 
+-int ac108_aif_mute(struct snd_soc_dai *dai, int mute, int direction) {
++int ac108_aif_mute(struct snd_soc_dai *dai, int mute) {
+ 	struct snd_soc_codec *codec = dai->codec;
+ 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
+ 
+@@ -1145,13 +1145,12 @@ static const struct snd_soc_dai_ops ac108_dai_ops = {
+ 	.hw_params	= ac108_hw_params,
+ 	.prepare	= ac108_prepare,
+ 	.trigger	= ac108_trigger,
+-	.mute_stream	= ac108_aif_mute,
++	.digital_mute	= ac108_aif_mute,
+ 
+ 	/*DAI format configuration*/
+ 	.set_fmt	= ac108_set_fmt,
+ 
+ 	// .hw_free = ac108_hw_free,
+-	.no_capture_mute = 1,
+ };
+ 
+ static  struct snd_soc_dai_driver ac108_dai0 = {
+diff --git a/seeed-voicecard.c b/seeed-voicecard.c
+index c6d9048..43535aa 100644
+--- a/seeed-voicecard.c
++++ b/seeed-voicecard.c
+@@ -96,16 +96,16 @@ static int seeed_voice_card_startup(struct snd_pcm_substream *substream)
+ 	if (ret)
+ 		clk_disable_unprepare(dai_props->cpu_dai.clk);
+ 
+-	if (asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min) {
+-		priv->channels_playback_default = asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min;
++	if (rtd->cpu_dai->driver->playback.channels_min) {
++		priv->channels_playback_default = rtd->cpu_dai->driver->playback.channels_min;
+ 	}
+-	if (asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min) {
+-		priv->channels_capture_default = asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min;
++	if (rtd->cpu_dai->driver->capture.channels_min) {
++		priv->channels_capture_default = rtd->cpu_dai->driver->capture.channels_min;
+ 	}
+-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min = priv->channels_playback_override;
+-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_max = priv->channels_playback_override;
+-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min = priv->channels_capture_override;
+-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_max = priv->channels_capture_override;
++	rtd->cpu_dai->driver->playback.channels_min = priv->channels_playback_override;
++	rtd->cpu_dai->driver->playback.channels_max = priv->channels_playback_override;
++	rtd->cpu_dai->driver->capture.channels_min = priv->channels_capture_override;
++	rtd->cpu_dai->driver->capture.channels_max = priv->channels_capture_override;
+ 
+ 	return ret;
+ }
+@@ -117,10 +117,10 @@ static void seeed_voice_card_shutdown(struct snd_pcm_substream *substream)
+ 	struct seeed_dai_props *dai_props =
+ 		seeed_priv_to_props(priv, rtd->num);
+ 
+-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min = priv->channels_playback_default;
+-	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_max = priv->channels_playback_default;
+-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min = priv->channels_capture_default;
+-	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_max = priv->channels_capture_default;
++	rtd->cpu_dai->driver->playback.channels_min = priv->channels_playback_default;
++	rtd->cpu_dai->driver->playback.channels_max = priv->channels_playback_default;
++	rtd->cpu_dai->driver->capture.channels_min = priv->channels_capture_default;
++	rtd->cpu_dai->driver->capture.channels_max = priv->channels_capture_default;
+ 
+ 	clk_disable_unprepare(dai_props->cpu_dai.clk);
+ 
+@@ -131,8 +131,8 @@ static int seeed_voice_card_hw_params(struct snd_pcm_substream *substream,
+ 				      struct snd_pcm_hw_params *params)
+ {
+ 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+-	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+-	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
++	struct snd_soc_dai *codec_dai = rtd->codec_dai;
++	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ 	struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
+ 	struct seeed_dai_props *dai_props =
+ 		seeed_priv_to_props(priv, rtd->num);
+@@ -196,7 +196,7 @@ static void work_cb_codec_clk(struct work_struct *work)
+ static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd)
+ {
+ 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+-	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
++	struct snd_soc_dai *dai = rtd->codec_dai;
+ 	struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
+ 	#if CONFIG_AC10X_TRIG_LOCK
+ 	unsigned long flags;
+@@ -205,7 +205,7 @@ static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd
+ 
+ 	dev_dbg(rtd->card->dev, "%s() stream=%s  cmd=%d play:%d, capt:%d\n",
+ 		__FUNCTION__, snd_pcm_stream_str(substream), cmd,
+-		dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]);
++		dai->playback_active, dai->capture_active);
+ 
+ 	switch (cmd) {
+ 	case SNDRV_PCM_TRIGGER_START:
+@@ -227,7 +227,7 @@ static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd
+ 	case SNDRV_PCM_TRIGGER_SUSPEND:
+ 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ 		/* capture channel resync, if overrun */
+-		if (dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] && substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
++		if (dai->capture_active && substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ 			break;
+ 		}
+ 
+@@ -252,7 +247,7 @@ static int seeed_voice_card_trigger(struct snd_pcm_substream *substream, int cmd
+ 
+ 	dev_dbg(rtd->card->dev, "%s() stream=%s  cmd=%d play:%d, capt:%d;finished %d\n",
+ 		__FUNCTION__, snd_pcm_stream_str(substream), cmd,
+-		dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], dai->stream_active[SNDRV_PCM_STREAM_CAPTURE], ret);
++		dai->playback_active, dai->capture_active, ret);
+ 
+ 	return ret;
+ }
+@@ -337,8 +337,8 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai,
+ static int seeed_voice_card_dai_init(struct snd_soc_pcm_runtime *rtd)
+ {
+ 	struct seeed_card_data *priv =	snd_soc_card_get_drvdata(rtd->card);
+-	struct snd_soc_dai *codec = asoc_rtd_to_codec(rtd, 0);
+-	struct snd_soc_dai *cpu = asoc_rtd_to_cpu(rtd, 0);
++	struct snd_soc_dai *codec = rtd->codec_dai;
++	struct snd_soc_dai *cpu = rtd->cpu_dai;
+ 	struct seeed_dai_props *dai_props =
+ 		seeed_priv_to_props(priv, rtd->num);
+ 	int ret;
+@@ -636,11 +636,11 @@ static int seeed_voice_card_probe(struct platform_device *pdev)
+ 	 * Use snd_soc_dai_link_component instead of legacy style
+ 	 * It is codec only. but cpu/platform will be supported in the future.
+ 	 * see
+-	 *      soc-core.c :: snd_soc_init_multicodec()
++	 *	soc-core.c :: snd_soc_init_multicodec()
+ 	 *
+ 	 * "platform" might be removed
+ 	 * see
+-	 *      simple-card-utils.c :: asoc_simple_canonicalize_platform()
++	 *	simple-card-utils.c :: asoc_simple_canonicalize_platform()
+ 	 */
+ 	for (i = 0; i < num; i++) {
+ 		dai_link[i].cpus		= &dai_props[i].cpus;
+diff --git a/sound-compatible-4.18.h b/sound-compatible-4.18.h
+index 080325b..eefa7de 100644
+--- a/sound-compatible-4.18.h
++++ b/sound-compatible-4.18.h
+@@ -16,6 +16,10 @@
+ #endif
+ 
+ #if LINUX_VERSION_CODE < KERNEL_VERSION(5,4,0)
++#ifndef __has_attribute
++# define __has_attribute(x) __GCC4_has_attribute_##x
++# define __GCC4_has_attribute___fallthrough__         0
++#endif
+ #if __has_attribute(__fallthrough__)
+ # define fallthrough                    __attribute__((__fallthrough__))
+ #else
+@@ -31,11 +35,7 @@
+ #define snd_soc_codec_get_dapm     snd_soc_component_get_dapm
+ #define snd_soc_codec_get_bias_level snd_soc_component_get_bias_level
+ #define snd_soc_kcontrol_codec     snd_soc_kcontrol_component
+-#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,9,0)
+-#define snd_soc_read               snd_soc_component_read
+-#else
+ #define snd_soc_read               snd_soc_component_read32
+-#endif
+ #define snd_soc_register_codec     devm_snd_soc_register_component
+ #define snd_soc_unregister_codec   snd_soc_unregister_component
+ #define snd_soc_update_bits        snd_soc_component_update_bits
+diff --git a/wm8960.c b/wm8960.c
+index 465c6dc..34d4dad 100644
+--- a/wm8960.c
++++ b/wm8960.c
+@@ -796,7 +796,7 @@ static int wm8960_hw_free(struct snd_pcm_substream *substream,
+ 	return 0;
+ }
+ 
+-static int wm8960_mute(struct snd_soc_dai *dai, int mute, int direction)
++static int wm8960_mute(struct snd_soc_dai *dai, int mute)
+ {
+ 	struct snd_soc_codec *codec = dai->codec;
+ 
+@@ -1236,12 +1236,11 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+ static const struct snd_soc_dai_ops wm8960_dai_ops = {
+ 	.hw_params = wm8960_hw_params,
+ 	.hw_free = wm8960_hw_free,
+-	.mute_stream = wm8960_mute,
++	.digital_mute = wm8960_mute,
+ 	.set_fmt = wm8960_set_dai_fmt,
+ 	.set_clkdiv = wm8960_set_dai_clkdiv,
+ 	.set_pll = wm8960_set_dai_pll,
+ 	.set_sysclk = wm8960_set_dai_sysclk,
+-	.no_capture_mute = 1,
+ };
+ 
+ static struct snd_soc_dai_driver wm8960_dai = {

patches/back-to-v5.8.diff

@@ -0,0 +1,71 @@
+diff --git a/ac108.c b/ac108.c
+index 4663df0..67edeae 100644
+--- a/ac108.c
++++ b/ac108.c
+@@ -1124,7 +1124,7 @@ void ac108_aif_shutdown(struct snd_pcm_substream *substream,
+ 	}
+ }
+ 
+-int ac108_aif_mute(struct snd_soc_dai *dai, int mute, int direction) {
++int ac108_aif_mute(struct snd_soc_dai *dai, int mute) {
+ 	struct snd_soc_codec *codec = dai->codec;
+ 	struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
+ 
+@@ -1145,13 +1145,12 @@ static const struct snd_soc_dai_ops ac108_dai_ops = {
+ 	.hw_params	= ac108_hw_params,
+ 	.prepare	= ac108_prepare,
+ 	.trigger	= ac108_trigger,
+-	.mute_stream	= ac108_aif_mute,
++	.digital_mute	= ac108_aif_mute,
+ 
+ 	/*DAI format configuration*/
+ 	.set_fmt	= ac108_set_fmt,
+ 
+ 	// .hw_free = ac108_hw_free,
+-	.no_capture_mute = 1,
+ };
+ 
+ static  struct snd_soc_dai_driver ac108_dai0 = {
+diff --git a/sound-compatible-4.18.h b/sound-compatible-4.18.h
+index 080325b..faed848 100644
+--- a/sound-compatible-4.18.h
++++ b/sound-compatible-4.18.h
+@@ -31,11 +31,7 @@
+ #define snd_soc_codec_get_dapm     snd_soc_component_get_dapm
+ #define snd_soc_codec_get_bias_level snd_soc_component_get_bias_level
+ #define snd_soc_kcontrol_codec     snd_soc_kcontrol_component
+-#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,9,0)
+-#define snd_soc_read               snd_soc_component_read
+-#else
+ #define snd_soc_read               snd_soc_component_read32
+-#endif
+ #define snd_soc_register_codec     devm_snd_soc_register_component
+ #define snd_soc_unregister_codec   snd_soc_unregister_component
+ #define snd_soc_update_bits        snd_soc_component_update_bits
+diff --git a/wm8960.c b/wm8960.c
+index 465c6dc..34d4dad 100644
+--- a/wm8960.c
++++ b/wm8960.c
+@@ -796,7 +796,7 @@ static int wm8960_hw_free(struct snd_pcm_substream *substream,
+ 	return 0;
+ }
+ 
+-static int wm8960_mute(struct snd_soc_dai *dai, int mute, int direction)
++static int wm8960_mute(struct snd_soc_dai *dai, int mute)
+ {
+ 	struct snd_soc_codec *codec = dai->codec;
+ 
+@@ -1236,12 +1236,11 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+ static const struct snd_soc_dai_ops wm8960_dai_ops = {
+ 	.hw_params = wm8960_hw_params,
+ 	.hw_free = wm8960_hw_free,
+-	.mute_stream = wm8960_mute,
++	.digital_mute = wm8960_mute,
+ 	.set_fmt = wm8960_set_dai_fmt,
+ 	.set_clkdiv = wm8960_set_dai_clkdiv,
+ 	.set_pll = wm8960_set_dai_pll,
+ 	.set_sysclk = wm8960_set_dai_sysclk,
+-	.no_capture_mute = 1,
+ };
+ 
+ static struct snd_soc_dai_driver wm8960_dai = {

pulseaudio/91-seeedvoicecard.rules

@@ -0,0 +1,8 @@
+SUBSYSTEM!="sound", GOTO="seeedvoicecard_end"
+ACTION!="change", GOTO="seeedvoicecard_end"
+KERNEL!="card*", GOTO="seeedvoicecard_end"
+
+ATTR{id}=="seeed4micvoicec",ENV{PULSE_PROFILE_SET}="seeed-voicecard-4mic.conf"
+ATTR{id}=="seeed8micvoicec",ENV{PULSE_PROFILE_SET}="seeed-voicecard-8mic.conf"
+
+LABEL="seeedvoicecard_end"

pulseaudio/README.md

@@ -0,0 +1,251 @@
+# PulseAudio Configuration for seeed-voicecard
+
+Follow this guide if you want to use your seeed-voicecard as a default source/sink of pulseaudio.
+
+### Prerequisites
+
+1. Download PulseAudio
+```
+sudo apt install -y pulseaudio
+```
+
+2. PulseAudio Profiles
+```
+cd seeed-voicecard/pulseaudio
+sudo cp pulse_config_4mic/seeed-voicecard.conf /usr/share/pulseaudio/alsa-mixer/profile-sets/seeed-voicecard-4mic.conf
+sudo cp pulse_config_6mic/seeed-voicecard.conf /usr/share/pulseaudio/alsa-mixer/profile-sets/seeed-voicecard-8mic.conf
+```
+
+3. Add `udev` Rules
+
+During the system start, when the card "seeed4micvoicec" is detected, the PULSE_PROFILE_SET variable will be set in the udev database, and PulseAudio will be forced to use `seeed-voicecard-4mic.conf`. Similarly, if the card "seeed8micvoicec" is detected, PulseAudio will be forced to use `seeed-voicecard-8mic.conf`.
+
+```
+sudo cp 91-seeedvoicecard.rules /etc/udev/rules.d/91-seeedvoicecard.rules
+```
+
+### ReSpeaker 4 Mic Array
+
+<!--
+1. Download pulseaudio
+```
+sudo apt install pulseaudio
+```
+
+2. First, you need to write [a profile for pulse](https://www.freedesktop.org/wiki/Software/PulseAudio/Backends/ALSA/Profiles/)
+```
+cd seeed-voicecard
+cd pulseaudio
+cd pulse_config_4mic
+sudo cp seeed-voicecard.conf /usr/share/pulseaudio/alsa-mixer/profile-sets/
+```
+
+3. Edit `udev rules`
+
+    During the system start, when the card "seeed4micvoicec" is detected, the PULSE_PROFILE_SET variable will be set in the udev database, and PulseAudio will be forced to use `seeed-voicecard.conf`.
+
+    ```
+    # have a look at /lib/udev/rules.d/90-pulseaudio.rules
+    sudo vim /lib/udev/rules.d/90-pulseaudio.rules
+    # add the following lines at about line 87(behind the setting for some laptops and before the line GOTO="pulseaudio_end")
+    # Seeed Voicecard
+    ATTR{id}=="seeed4micvoicec",ATTR{number}=="1",ENV{PULSE_PROFILE_SET}="seeed-voicecard.conf"
+    ```
+    ![](./udev_rules_4mic.png)
+
+    The value of `ATTR{number}` can be found with:
+
+    ```
+    udevadm info -a -p /sys/class/sound/card0/
+    # or udevadm info -a -p /sys/class/sound/card1/
+    ```
+
+    For example, in Raspberry Pi, we can find `ATTR{id}=="seeed4micvoicec"` and `ATTR{number}=="1"` with command `udevadm info -a -p /sys/class/sound/card1/`:
+
+    ```
+    pi@raspberrypi:~ $ udevadm info -a -p  /sys/class/sound/card1/
+
+    Udevadm info starts with the device specified by the devpath and then
+    walks up the chain of parent devices. It prints for every device
+    found, all possible attributes in the udev rules key format.
+    A rule to match, can be composed by the attributes of the device
+    and the attributes from one single parent device.
+
+    looking at device '/devices/platform/soc/soc:sound/sound/card1':
+        KERNEL=="card1"
+        SUBSYSTEM=="sound"
+        DRIVER==""
+        ATTR{id}=="seeed4micvoicec"
+        ATTR{number}=="1"
+
+    looking at parent device '/devices/platform/soc/soc:sound':
+        KERNELS=="soc:sound"
+        SUBSYSTEMS=="platform"
+        DRIVERS=="seeed-voicecard"
+        ATTRS{driver_override}=="(null)"
+
+    looking at parent device '/devices/platform/soc':
+        KERNELS=="soc"
+        SUBSYSTEMS=="platform"
+        DRIVERS==""
+        ATTRS{driver_override}=="(null)"
+
+    looking at parent device '/devices/platform':
+        KERNELS=="platform"
+        SUBSYSTEMS==""
+        DRIVERS==""
+    ``` -->
+
+1. config `default.pa` and `daemon.conf`
+```
+sudo cp pulse_config_4mic/default.pa /etc/pulse/
+sudo cp pulse_config_4mic/daemon.conf /etc/pulse/
+```
+
+2. reboot raspberry pi and check
+```
+sudo reboot
+pulseaudio --start  # start pulse at first
+pactl info  # check the setting
+
+Server String: /run/user/1000/pulse/native
+Library Protocol Version: 32
+Server Protocol Version: 32
+Is Local: yes
+Client Index: 18
+Tile Size: 65496
+User Name: pi
+Host Name: raspberrypi
+Server Name: pulseaudio
+Server Version: 10.0
+Default Sample Specification: s16le 4ch 96000Hz
+Default Channel Map: front-left,front-center,front-right,rear-center
+Default Sink: alsa_output.platform-soc_audio.analog-stereo
+Default Source: alsa_input.platform-soc_sound.seeed-source
+Cookie: 3b12:70b3
+```
+
+### 6-Mics Circular Array Kit and 4-Mics Linear Array
+
+<!--
+1. Download pulseaudio
+```
+sudo apt install pulseaudio
+```
+
+2. First, you need to write [a profile for pulse](https://www.freedesktop.org/wiki/Software/PulseAudio/Backends/ALSA/Profiles/)
+```
+cd seeed-voicecard
+cd pulseaudio
+cd pulse_config_6mic
+sudo cp seeed-voicecard.conf /usr/share/pulseaudio/alsa-mixer/profile-sets/
+```
+
+3. Edit `udev rules`
+
+    During the system start, when the card "seeed8micvoicec" is detected, the PULSE_PROFILE_SET variable will be set in the udev database, and PulseAudio will be forced to use `seeed-voicecard.conf`.
+
+    ```
+    # have a look at /lib/udev/rules.d/90-pulseaudio.rules
+    sudo vim /lib/udev/rules.d/90-pulseaudio.rules
+    # add the following lines at about line 87(behind the setting for some laptops and before the line GOTO="pulseaudio_end")
+    # Seeed Voicecard
+    ATTR{id}=="seeed8micvoicec",ATTR{number}=="1",ENV{PULSE_PROFILE_SET}="seeed-voicecard.conf"
+    ```
+    ![](./udev_rules_6mic.png)
+
+    The value of `ATTR{number}` can be found with:
+
+    ```
+    udevadm info -a -p /sys/class/sound/card0/
+    # or udevadm info -a -p /sys/class/sound/card1/
+    ```
+
+    For example, in Raspberry Pi, we can find `ATTR{id}=="seeed8micvoicec"` and `ATTR{number}=="1"` with command `udevadm info -a -p /sys/class/sound/card1/`:
+
+    ```
+    pi@raspberrypi:~ $ udevadm info -a -p  /sys/class/sound/card1/
+
+    Udevadm info starts with the device specified by the devpath and then
+    walks up the chain of parent devices. It prints for every device
+    found, all possible attributes in the udev rules key format.
+    A rule to match, can be composed by the attributes of the device
+    and the attributes from one single parent device.
+
+    looking at device '/devices/platform/soc/soc:sound/sound/card1':
+        KERNEL=="card1"
+        SUBSYSTEM=="sound"
+        DRIVER==""
+        ATTR{id}=="seeed8micvoicec"
+        ATTR{number}=="1"
+
+    looking at parent device '/devices/platform/soc/soc:sound':
+        KERNELS=="soc:sound"
+        SUBSYSTEMS=="platform"
+        DRIVERS=="seeed-voicecard"
+        ATTRS{driver_override}=="(null)"
+
+    looking at parent device '/devices/platform/soc':
+        KERNELS=="soc"
+        SUBSYSTEMS=="platform"
+        DRIVERS==""
+        ATTRS{driver_override}=="(null)"
+
+    looking at parent device '/devices/platform':
+        KERNELS=="platform"
+        SUBSYSTEMS==""
+        DRIVERS==""
+    ``` -->
+
+1. config `default.pa` and `daemon.conf`
+```
+sudo cp pulse_config_6mic/default.pa /etc/pulse/
+sudo cp pulse_config_6mic/daemon.conf /etc/pulse/
+```
+
+2. reboot raspberry pi and check
+```
+sudo reboot
+pulseaudio --start  # start pulse at first
+pactl info  # check the setting
+
+# The output should be like this
+# You could see the default sink is seeed-2ch and default source is seeed-8ch
+pi@raspberrypi:~ $ pactl info
+Server String: /run/user/1000/pulse/native
+Library Protocol Version: 32
+Server Protocol Version: 32
+Is Local: yes
+Client Index: 6
+Tile Size: 65496
+User Name: pi
+Host Name: raspberrypi
+Server Name: pulseaudio
+Server Version: 10.0
+Default Sample Specification: s32le 8ch 96000Hz
+Default Channel Map: front-left,front-left-of-center,front-center,front-right,front-right-of-center,rear-center,aux0,aux1
+Default Sink: alsa_output.platform-soc_sound.seeed-2ch
+Default Source: alsa_input.platform-soc_sound.seeed-8ch
+Cookie: 3523:e5af
+```
+
+### FAQ
+
+1. Default Sink/Source not right
+
+    Make sure there is no any other daemon or using the audio device. Then check profile and udev rules.
+
+    `pacmd list-sinks` and `pacmd list-sources` can be used to show the avaiable sinks/sources, after pulseaudio is started.
+
+2. Can't start PulseAudio
+
+    Check `default.pa` and `daemon.conf`
+
+3. How to get PulseAudio started automatically
+
+    Normally the PulseAudio server is started automatically. If you want to disable it, you can set `autospawn = no` in `~/.config/pulse/client.conf` or `/etc/pulse/client.conf`.
+    [Click this for more details](https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Running/).
+
+4. Why the default sample rate is 96000? What if my audio's sample rate is not the same as the default?
+
+    For the other sample rate audio, PulseAudio will resample it into 96K, which means that if your audio's sample rate is lower than 96K, it will get smoothing.

pulseaudio/pulse_config_4mic/daemon.conf

@@ -0,0 +1,87 @@
+# This file is part of PulseAudio.
+#
+# PulseAudio is free software; you can redistribute it and/or modify
+# it under the terms of the GNU Lesser General Public License as published by
+# the Free Software Foundation; either version 2 of the License, or
+# (at your option) any later version.
+#
+# PulseAudio is distributed in the hope that it will be useful, but
+# WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+# General Public License for more details.
+#
+# You should have received a copy of the GNU Lesser General Public License
+# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+
+## Configuration file for the PulseAudio daemon. See pulse-daemon.conf(5) for
+## more information. Default values are commented out.  Use either ; or # for
+## commenting.
+
+; daemonize = no
+; fail = yes
+; allow-module-loading = yes
+; allow-exit = yes
+; use-pid-file = yes
+; system-instance = no
+; local-server-type = user
+; enable-shm = yes
+; enable-memfd = yes
+; shm-size-bytes = 0 # setting this 0 will use the system-default, usually 64 MiB
+; lock-memory = no
+; cpu-limit = no
+
+; high-priority = yes
+; nice-level = -11
+
+; realtime-scheduling = yes
+; realtime-priority = 5
+
+; exit-idle-time = 20
+; scache-idle-time = 20
+
+; dl-search-path = (depends on architecture)
+
+; load-default-script-file = yes
+; default-script-file = /etc/pulse/default.pa
+
+; log-target = auto
+; log-level = notice
+; log-meta = no
+; log-time = no
+; log-backtrace = 0
+
+; resample-method = speex-float-1
+; enable-remixing = yes
+; enable-lfe-remixing = no
+; lfe-crossover-freq = 0
+
+; flat-volumes = yes
+
+; rlimit-fsize = -1
+; rlimit-data = -1
+; rlimit-stack = -1
+; rlimit-core = -1
+; rlimit-as = -1
+; rlimit-rss = -1
+; rlimit-nproc = -1
+; rlimit-nofile = 256
+; rlimit-memlock = -1
+; rlimit-locks = -1
+; rlimit-sigpending = -1
+; rlimit-msgqueue = -1
+; rlimit-nice = 31
+; rlimit-rtprio = 9
+; rlimit-rttime = 200000
+
+default-sample-format = s16le
+default-sample-rate = 96000
+; alternate-sample-rate = 48000
+default-sample-channels = 4
+; default-channel-map = front-left,front-right
+
+; default-fragments = 4
+; default-fragment-size-msec = 25
+
+; enable-deferred-volume = yes
+; deferred-volume-safety-margin-usec = 8000
+; deferred-volume-extra-delay-usec = 0

pulseaudio/pulse_config_4mic/default.pa

@@ -0,0 +1,144 @@
+#!/usr/bin/pulseaudio -nF
+#
+# This file is part of PulseAudio.
+#
+# PulseAudio is free software; you can redistribute it and/or modify it
+# under the terms of the GNU Lesser General Public License as published by
+# the Free Software Foundation; either version 2 of the License, or
+# (at your option) any later version.
+#
+# PulseAudio is distributed in the hope that it will be useful, but
+# WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+# General Public License for more details.
+#
+# You should have received a copy of the GNU Lesser General Public License
+# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+
+# This startup script is used only if PulseAudio is started per-user
+# (i.e. not in system mode)
+
+.fail
+
+### Automatically restore the volume of streams and devices
+load-module module-device-restore
+load-module module-stream-restore
+load-module module-card-restore
+
+### Automatically augment property information from .desktop files
+### stored in /usr/share/application
+load-module module-augment-properties
+
+### Should be after module-*-restore but before module-*-detect
+load-module module-switch-on-port-available
+
+### Load audio drivers statically
+### (it's probably better to not load these drivers manually, but instead
+### use module-udev-detect -- see below -- for doing this automatically)
+#load-module module-alsa-sink device="hw:1,0" channels=8 rate=48000 format=s32le
+#load-module module-alsa-source device="hw:1,0" channels=8 rate=48000 format=s32le
+#load-module module-oss device="/dev/dsp" sink_name=output source_name=input
+#load-module module-oss-mmap device="/dev/dsp" sink_name=output source_name=input
+#load-module module-null-sink
+#load-module module-pipe-sink
+
+### Automatically load driver modules depending on the hardware available
+.ifexists module-udev-detect.so
+load-module module-udev-detect 
+#channels=8 rate=48000 format=s32le
+.else
+### Use the static hardware detection module (for systems that lack udev support)
+load-module module-detect
+.endif
+
+### Automatically connect sink and source if JACK server is present
+.ifexists module-jackdbus-detect.so
+.nofail
+load-module module-jackdbus-detect channels=2
+.fail
+.endif
+
+### Automatically load driver modules for Bluetooth hardware
+.ifexists module-bluetooth-policy.so
+load-module module-bluetooth-policy
+.endif
+
+.ifexists module-bluetooth-discover.so
+load-module module-bluetooth-discover
+.endif
+
+### Load several protocols
+.ifexists module-esound-protocol-unix.so
+load-module module-esound-protocol-unix
+.endif
+load-module module-native-protocol-unix
+
+### Network access (may be configured with paprefs, so leave this commented
+### here if you plan to use paprefs)
+#load-module module-esound-protocol-tcp
+#load-module module-native-protocol-tcp
+#load-module module-zeroconf-publish
+
+### Load the RTP receiver module (also configured via paprefs, see above)
+#load-module module-rtp-recv
+
+### Load the RTP sender module (also configured via paprefs, see above)
+#load-module module-null-sink sink_name=rtp format=s16be channels=2 rate=44100 sink_properties="device.description='RTP Multicast Sink'"
+#load-module module-rtp-send source=rtp.monitor
+
+### Load additional modules from GConf settings. This can be configured with the paprefs tool.
+### Please keep in mind that the modules configured by paprefs might conflict with manually
+### loaded modules.
+.ifexists module-gconf.so
+.nofail
+load-module module-gconf
+.fail
+.endif
+
+### Automatically restore the default sink/source when changed by the user
+### during runtime
+### NOTE: This should be loaded as early as possible so that subsequent modules
+### that look up the default sink/source get the right value
+load-module module-default-device-restore
+
+### Automatically move streams to the default sink if the sink they are
+### connected to dies, similar for sources
+load-module module-rescue-streams
+
+### Make sure we always have a sink around, even if it is a null sink.
+load-module module-always-sink
+
+### Honour intended role device property
+load-module module-intended-roles
+
+### Automatically suspend sinks/sources that become idle for too long
+load-module module-suspend-on-idle
+
+### If autoexit on idle is enabled we want to make sure we only quit
+### when no local session needs us anymore.
+.ifexists module-console-kit.so
+load-module module-console-kit
+.endif
+.ifexists module-systemd-login.so
+load-module module-systemd-login
+.endif
+
+### Enable positioned event sounds
+load-module module-position-event-sounds
+
+### Cork music/video streams when a phone stream is active
+load-module module-role-cork
+
+### Modules to allow autoloading of filters (such as echo cancellation)
+### on demand. module-filter-heuristics tries to determine what filters
+### make sense, and module-filter-apply does the heavy-lifting of
+### loading modules and rerouting streams.
+load-module module-filter-heuristics
+load-module module-filter-apply
+
+### Make some devices default
+#set-default-sink output
+#set-default-source input
+set-default-source alsa_input.platform-soc_sound.seeed-source
+set-default-sink alsa_output.platform-soc_sound.seeed-sink
+

pulseaudio/pulse_config_4mic/seeed-voicecard.conf

@@ -0,0 +1,17 @@
+# /usr/share/pulseaudio/alsa-mixer/profile-sets/seeed-voicecard.conf
+
+[General]
+auto-profiles = no
+[Mapping seeed-source]
+device-strings = hw:%f
+channel-map = front-left,front-right,rear-left,rear-right
+exact-channels = false
+fallback = yes
+paths-input = seeed-source
+priority = 3
+direction = input
+
+[Profile input:seeed-source]
+input-mappings = seeed-source
+priority = 5
+skip-probe = yes

pulseaudio/pulse_config_6mic/daemon.conf

@@ -0,0 +1,87 @@
+# This file is part of PulseAudio.
+#
+# PulseAudio is free software; you can redistribute it and/or modify
+# it under the terms of the GNU Lesser General Public License as published by
+# the Free Software Foundation; either version 2 of the License, or
+# (at your option) any later version.
+#
+# PulseAudio is distributed in the hope that it will be useful, but
+# WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+# General Public License for more details.
+#
+# You should have received a copy of the GNU Lesser General Public License
+# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+
+## Configuration file for the PulseAudio daemon. See pulse-daemon.conf(5) for
+## more information. Default values are commented out.  Use either ; or # for
+## commenting.
+
+; daemonize = no
+; fail = yes
+; allow-module-loading = yes
+; allow-exit = yes
+; use-pid-file = yes
+; system-instance = no
+; local-server-type = user
+; enable-shm = yes
+; enable-memfd = yes
+; shm-size-bytes = 0 # setting this 0 will use the system-default, usually 64 MiB
+; lock-memory = no
+; cpu-limit = no
+
+; high-priority = yes
+; nice-level = -11
+
+; realtime-scheduling = yes
+; realtime-priority = 5
+
+; exit-idle-time = 20
+; scache-idle-time = 20
+
+; dl-search-path = (depends on architecture)
+
+; load-default-script-file = yes
+; default-script-file = /etc/pulse/default.pa
+
+; log-target = auto
+; log-level = notice
+; log-meta = no
+; log-time = no
+; log-backtrace = 0
+
+; resample-method = speex-float-1
+; enable-remixing = yes
+; enable-lfe-remixing = no
+; lfe-crossover-freq = 0
+
+; flat-volumes = yes
+
+; rlimit-fsize = -1
+; rlimit-data = -1
+; rlimit-stack = -1
+; rlimit-core = -1
+; rlimit-as = -1
+; rlimit-rss = -1
+; rlimit-nproc = -1
+; rlimit-nofile = 256
+; rlimit-memlock = -1
+; rlimit-locks = -1
+; rlimit-sigpending = -1
+; rlimit-msgqueue = -1
+; rlimit-nice = 31
+; rlimit-rtprio = 9
+; rlimit-rttime = 200000
+
+default-sample-format = s32le
+default-sample-rate = 96000
+; alternate-sample-rate = 48000
+default-sample-channels = 8
+; default-channel-map = front-left,front-right
+
+; default-fragments = 4
+; default-fragment-size-msec = 25
+
+; enable-deferred-volume = yes
+; deferred-volume-safety-margin-usec = 8000
+; deferred-volume-extra-delay-usec = 0

pulseaudio/pulse_config_6mic/default.pa

@@ -0,0 +1,143 @@
+#!/usr/bin/pulseaudio -nF
+#
+# This file is part of PulseAudio.
+#
+# PulseAudio is free software; you can redistribute it and/or modify it
+# under the terms of the GNU Lesser General Public License as published by
+# the Free Software Foundation; either version 2 of the License, or
+# (at your option) any later version.
+#
+# PulseAudio is distributed in the hope that it will be useful, but
+# WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+# General Public License for more details.
+#
+# You should have received a copy of the GNU Lesser General Public License
+# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+
+# This startup script is used only if PulseAudio is started per-user
+# (i.e. not in system mode)
+
+.fail
+
+### Automatically restore the volume of streams and devices
+load-module module-device-restore
+load-module module-stream-restore
+load-module module-card-restore
+
+### Automatically augment property information from .desktop files
+### stored in /usr/share/application
+load-module module-augment-properties
+
+### Should be after module-*-restore but before module-*-detect
+load-module module-switch-on-port-available
+
+### Load audio drivers statically
+### (it's probably better to not load these drivers manually, but instead
+### use module-udev-detect -- see below -- for doing this automatically)
+#load-module module-alsa-sink device="hw:1,0" channels=8 rate=48000 format=s32le
+#load-module module-alsa-source device="hw:1,0" channels=8 rate=48000 format=s32le
+#load-module module-oss device="/dev/dsp" sink_name=output source_name=input
+#load-module module-oss-mmap device="/dev/dsp" sink_name=output source_name=input
+#load-module module-null-sink
+#load-module module-pipe-sink
+
+### Automatically load driver modules depending on the hardware available
+.ifexists module-udev-detect.so
+load-module module-udev-detect 
+#channels=8 rate=48000 format=s32le
+.else
+### Use the static hardware detection module (for systems that lack udev support)
+load-module module-detect
+.endif
+
+### Automatically connect sink and source if JACK server is present
+.ifexists module-jackdbus-detect.so
+.nofail
+load-module module-jackdbus-detect channels=2
+.fail
+.endif
+
+### Automatically load driver modules for Bluetooth hardware
+.ifexists module-bluetooth-policy.so
+load-module module-bluetooth-policy
+.endif
+
+.ifexists module-bluetooth-discover.so
+load-module module-bluetooth-discover
+.endif
+
+### Load several protocols
+.ifexists module-esound-protocol-unix.so
+load-module module-esound-protocol-unix
+.endif
+load-module module-native-protocol-unix
+
+### Network access (may be configured with paprefs, so leave this commented
+### here if you plan to use paprefs)
+#load-module module-esound-protocol-tcp
+#load-module module-native-protocol-tcp
+#load-module module-zeroconf-publish
+
+### Load the RTP receiver module (also configured via paprefs, see above)
+#load-module module-rtp-recv
+
+### Load the RTP sender module (also configured via paprefs, see above)
+#load-module module-null-sink sink_name=rtp format=s16be channels=2 rate=44100 sink_properties="device.description='RTP Multicast Sink'"
+#load-module module-rtp-send source=rtp.monitor
+
+### Load additional modules from GConf settings. This can be configured with the paprefs tool.
+### Please keep in mind that the modules configured by paprefs might conflict with manually
+### loaded modules.
+.ifexists module-gconf.so
+.nofail
+load-module module-gconf
+.fail
+.endif
+
+### Automatically restore the default sink/source when changed by the user
+### during runtime
+### NOTE: This should be loaded as early as possible so that subsequent modules
+### that look up the default sink/source get the right value
+load-module module-default-device-restore
+
+### Automatically move streams to the default sink if the sink they are
+### connected to dies, similar for sources
+load-module module-rescue-streams
+
+### Make sure we always have a sink around, even if it is a null sink.
+load-module module-always-sink
+
+### Honour intended role device property
+load-module module-intended-roles
+
+### Automatically suspend sinks/sources that become idle for too long
+load-module module-suspend-on-idle
+
+### If autoexit on idle is enabled we want to make sure we only quit
+### when no local session needs us anymore.
+.ifexists module-console-kit.so
+load-module module-console-kit
+.endif
+.ifexists module-systemd-login.so
+load-module module-systemd-login
+.endif
+
+### Enable positioned event sounds
+load-module module-position-event-sounds
+
+### Cork music/video streams when a phone stream is active
+load-module module-role-cork
+
+### Modules to allow autoloading of filters (such as echo cancellation)
+### on demand. module-filter-heuristics tries to determine what filters
+### make sense, and module-filter-apply does the heavy-lifting of
+### loading modules and rerouting streams.
+load-module module-filter-heuristics
+load-module module-filter-apply
+
+### Make some devices default
+#set-default-sink output
+#set-default-source input
+set-default-source alsa_input.platform-soc_sound.seeed-8ch
+set-default-sink alsa_output.platform-soc_sound.seeed-2ch

pulseaudio/pulse_config_6mic/seeed-voicecard.conf

@@ -0,0 +1,34 @@
+# /usr/share/pulseaudio/alsa-mixer/profile-sets/seeed-voiced.conf
+
+[General]
+auto-profiles = no
+[Mapping seeed-8ch]
+device-strings = hw:%f
+channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right
+exact-channels = false
+fallback = yes
+paths-input = seeed-8ch
+priority = 3
+direction = input
+[Mapping seeed-2ch]
+device-strings = hw:%f
+channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right
+exact-channels = false
+exact-channels = false
+fallback = yes
+paths-output = seeed-2ch
+direction = output
+priority = 2
+[Profile output:seeed-2ch+input:seeed-8ch]
+output-mappings = seeed-2ch
+input-mappings = seeed-8ch
+priority = 100
+skip-probe = yes
+[Profile output:seeed-2ch]
+output-mappings = seeed-2ch
+priority = 4
+skip-probe = yes
+[Profile input:seeed-8ch]
+input-mappings = seeed-8ch
+priority = 5
+skip-probe = yes

seeed-4mic-voicecard-overlay.dts

@@ -13,7 +13,7 @@
 	};
 
 	fragment@1 {
-		target-path = "/clocks";
+		target-path = "/";
 		__overlay__ {
 			ac108_mclk: codec-mclk {
 				compatible = "fixed-clock";

seeed-8mic-voicecard-overlay.dts

@@ -13,7 +13,7 @@
 	};
 
 	fragment@1 {
-		target-path = "/clocks";
+		target-path = "/";
 		__overlay__ {
 			ac10x_mclk: codec-mclk {
 				compatible = "fixed-clock";
@@ -26,13 +26,15 @@
 	fragment@2 {
 		target = <&gpio>;
 		__overlay__ {
-			spk_amp_and_irq_pins: speaker_amp_and_irq_pins {
+			spk_amp_pins: spk_pins {
 				brcm,pins = <17 22>;
 				brcm,function = <1 0>; /* out in */
+				brcm,pull = <0 0>;     /* -   - */
 			};
 			gpclk0_pins: gpclk0_pins {
 				brcm,pins = <4>;
 				brcm,function = <4>; /* alt func 0 */
+				brcm,pull = <0>;     /* - */
 			};
 		};
 	};
@@ -47,7 +49,7 @@
 			ac101: ac101@1a{
 				compatible = "x-power,ac101";
 				pinctrl-names = "default";
-				pinctrl-0 = <&spk_amp_and_irq_pins>,<&gpclk0_pins>;
+				pinctrl-0 = <&spk_amp_pins &gpclk0_pins>;
 				spk-amp-switch-gpios = <&gpio 17 0>;
 				switch-irq-gpios = <&gpio 22 0>;
 				reg = <0x1a>;
@@ -80,6 +82,8 @@
 			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 {
@@ -88,6 +92,7 @@
 				frame-master = <&codec0_dai>;
 				/* bitclock-inversion; */
 				/* frame-inversion; */
+				reg = <0>;
 
 				cpu {
 					sound-dai = <&i2s>;

seeed-voicecard

@@ -21,76 +21,143 @@
 # THE SOFTWARE.
 
 set -x
-exec 1>/var/log/$(basename $0).log 2>&1
+#exec 1>/var/log/$(basename $0).log 2>&1
+
+export PATH=$PATH:/opt/vc/bin
+OVERLAYS=/boot/overlays
+[ -d /boot/firmware/overlays ] && OVERLAYS=/boot/firmware/overlays
+
 #enable i2c interface
-dtparam i2c_arm=on
+dtparam -d $OVERLAYS i2c_arm=on
 modprobe i2c-dev
 
 #enable spi interface
-dtparam spi=on
-sleep 1
+dtparam -d $OVERLAYS spi=on
+
+_VER_RUN=
+function get_kernel_version() {
+    local ZIMAGE IMG_OFFSET
+
+    _VER_RUN=""
+    [ -z "$_VER_RUN" ] && {
+        ZIMAGE=/boot/kernel.img
+        IMG_OFFSET=$(LC_ALL=C grep -abo $'\x1f\x8b\x08\x00' $ZIMAGE | head -n 1 | cut -d ':' -f 1)
+        # 64-bit-only kernel package
+        [ ! -f /boot/kernel.img ] && [ -f /boot/kernel8.img ] && ZIMAGE=/boot/kernel8.img
+        _VER_RUN=$(dd if=$ZIMAGE obs=64K ibs=4 skip=$(( IMG_OFFSET / 4)) 2>/dev/null | zcat | grep -a -m1 "Linux version" | LC_ALL=C sed -e 's/^.*Linux/Linux/' | strings | awk '{ print $3; }')
+    }
+    echo "$_VER_RUN"
+    return 0
+}
+
+CONFIG=/boot/config.txt
+[ -f /boot/firmware/usercfg.txt ] && CONFIG=/boot/firmware/usercfg.txt
+
+get_overlay() {
+    ov=$1
+    if grep -q -E "^dtoverlay=$ov" $CONFIG; then
+      echo 0
+    else
+      echo 1
+    fi
+}
+
+do_overlay() {
+    ov=$1
+    RET=$2
+    DEFAULT=--defaultno
+    CURRENT=0
+    if [ $(get_overlay $ov) -eq 0 ]; then
+        DEFAULT=
+        CURRENT=1
+    fi
+    if [ $RET -eq $CURRENT ]; then
+        ASK_TO_REBOOT=1
+    fi
+    if [ $RET -eq 0 ]; then
+        sed $CONFIG -i -e "s/^#dtoverlay=$ov/dtoverlay=$ov/"
+        if ! grep -q -E "^dtoverlay=$ov" $CONFIG; then
+            printf "dtoverlay=$ov\n" >> $CONFIG
+        fi
+        STATUS=enabled
+    elif [ $RET -eq 1 ]; then
+        sed $CONFIG -i -e "s/^dtoverlay=$ov/#dtoverlay=$ov/"
+        STATUS=disabled
+    else
+        return $RET
+    fi
+}
 
 
+is_1a=$(i2cdetect -y  1 0x1a 0x1a | egrep "(1a|UU)" | awk '{print $2}')
+is_35=$(i2cdetect -y  1 0x35 0x35 | egrep "(35|UU)" | awk '{print $2}')
+is_3b=$(i2cdetect -y  1 0x3b 0x3b | egrep "(3b|UU)" | awk '{print $2}')
 
-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}')
+RPI_HATS="seeed-2mic-voicecard seeed-4mic-voicecard seeed-8mic-voicecard"
+overlay=""
 
-if [ "x${is_1a}" == "x1a" ] && [ "x${is_35}" == "x" ] ; then
+if [ "x${is_1a}" != "x" ] && [ "x${is_35}" == "x" ] ; then
     echo "install 2mic"
-    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
+    overlay=seeed-2mic-voicecard
+    asound_conf=/etc/voicecard/asound_2mic.conf
+    asound_state=/etc/voicecard/wm8960_asound.state
 fi
 
-if [ "x${is_3b}" == "x3b" ] && [ "x${is_35}" == "x" ] ; then
+if [ "x${is_3b}" != "x" ] && [ "x${is_35}" == "x" ] ; then
     echo "install 4mic"
-    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
+    overlay=seeed-4mic-voicecard
+    asound_conf=/etc/voicecard/asound_4mic.conf
+    asound_state=/etc/voicecard/ac108_asound.state
 fi
 
-if [ "x${is_3b}" == "x3b" ] && [ "x${is_35}" == "x35" ] ; then
+if [ "x${is_3b}" != "x" ] && [ "x${is_35}" != "x" ] ; then
     echo "install 6mic"
-    dtoverlay seeed-8mic-voicecard
-    sleep 1
-    hw=$(aplay -l | grep seeed8micvoicec | awk '{print $2}' | sed 's/://')
+    overlay=seeed-8mic-voicecard
+    asound_conf=/etc/voicecard/asound_6mic.conf
+    asound_state=/etc/voicecard/ac108_6mic.state
+fi
 
-    cp /etc/voicecard/asound_6mic.conf /etc/asound.conf
+if [ "$overlay" ]; then
+    echo Install $overlay ...
 
-    old=$(cat /etc/asound.conf | grep hw: | awk 'NR==1 {print $2}' | sed 's/\"//g')
+    # Remove old configuration
+    rm /etc/asound.conf
+    rm /var/lib/alsa/asound.state
 
-    sed -i -e "s/${old}/hw:${hw},0/g" /etc/asound.conf  
+    kernel_ver=$(uname -r) # get_kernel_version)
+    # echo kernel_ver=$kernel_ver
 
-    cp /etc/voicecard/ac108_6mic.state /var/lib/alsa/asound.state
+    # TODO: dynamic dtoverlay Bug of v4.19.x
+    #       no DT node phandle inserted.
+    if [[ "$kernel_ver" =~ ^4\.19.*$ || "$kernel_ver" =~ ^5\.*$ ]]; then
+        for i in $RPI_HATS; do
+            if [ "$i" == "$overlay" ]; then
+                /bin/true #do_overlay $overlay 0
+            else
+                echo Uninstall $i ...
+                /bin/true #do_overlay $i 1
+            fi
+        done
+    fi
+    #make sure the driver loads correctly
+    dtoverlay -d $OVERLAYS $overlay || true
+
+
+    echo "create $overlay asound configure file"
+    ln -s $asound_conf /etc/asound.conf
+    echo "create $overlay asound status file"
+    ln -s $asound_state /var/lib/alsa/asound.state
 fi
 
 alsactl restore
 
-#Fore 3.5mm ('headphone') jack
-amixer cset numid=3 1 
-
-
+#Force 3.5mm ('headphone') jack
+#   The Raspberry Pi 4, released on 24th Jun 2019, has two HDMI ports,
+#   and can drive two displays with audios for two users simultaneously,
+#   in a "multiseat" configuration. The earlier single virtual ALSA
+#   option for re-directing audio playback between headphone jack and HDMI
+#   via a 'Routing' mixer setting was turned off eventually to allow
+#   simultaneous usage of all 3 playback devices.
+if aplay -l | grep -q "bcm2835 ALSA"; then
+    amixer cset numid=3 1 || true
+fi

seeed-voicecard.c

@@ -26,6 +26,23 @@
 #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:
@@ -39,6 +56,9 @@ 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;
@@ -47,7 +67,12 @@ 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 {
@@ -61,6 +86,7 @@ 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))
@@ -85,16 +111,16 @@ static int seeed_voice_card_startup(struct snd_pcm_substream *substream)
 	if (ret)
 		clk_disable_unprepare(dai_props->cpu_dai.clk);
 
-	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->playback.channels_min) {
+		priv->channels_playback_default = asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min;
 	}
-	if (rtd->cpu_dai->driver->capture.channels_min) {
-		priv->channels_capture_default = rtd->cpu_dai->driver->capture.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;
 	}
-	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;
+	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min = priv->channels_playback_override;
+	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_max = priv->channels_playback_override;
+	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min = priv->channels_capture_override;
+	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_max = priv->channels_capture_override;
 
 	return ret;
 }
@@ -106,10 +132,10 @@ static void seeed_voice_card_shutdown(struct snd_pcm_substream *substream)
 	struct seeed_dai_props *dai_props =
 		seeed_priv_to_props(priv, rtd->num);
 
-	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;
+	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_min = priv->channels_playback_default;
+	asoc_rtd_to_cpu(rtd, 0)->driver->playback.channels_max = priv->channels_playback_default;
+	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_min = priv->channels_capture_default;
+	asoc_rtd_to_cpu(rtd, 0)->driver->capture.channels_max = priv->channels_capture_default;
 
 	clk_disable_unprepare(dai_props->cpu_dai.clk);
 
@@ -120,8 +146,8 @@ 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 = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct seeed_dai_props *dai_props =
 		seeed_priv_to_props(priv, rtd->num);
@@ -151,9 +177,9 @@ err:
 }
 
 #define _SET_CLOCK_CNT		2
-static int (* _set_clock[_SET_CLOCK_CNT])(int y_start_n_stop);
+static int (* _set_clock[_SET_CLOCK_CNT])(int y_start_n_stop, struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai);
 
-int seeed_voice_card_register_set_clock(int stream, int (*set_clock)(int)) {
+int seeed_voice_card_register_set_clock(int stream, int (*set_clock)(int, struct snd_pcm_substream *, int, struct snd_soc_dai *)) {
 	if (! _set_clock[stream]) {
 		_set_clock[stream] = set_clock;
 	}
@@ -161,44 +187,82 @@ int seeed_voice_card_register_set_clock(int stream, int (*set_clock)(int)) {
 }
 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 = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct seeed_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
+	#if CONFIG_AC10X_TRIG_LOCK
 	unsigned long flags;
+	#endif
 	int ret = 0;
 
 	dev_dbg(rtd->card->dev, "%s() stream=%s  cmd=%d play:%d, capt:%d\n",
 		__FUNCTION__, snd_pcm_stream_str(substream), cmd,
-		dai->playback_active, dai->capture_active);
-
-	/* I know it will degrades performance, but I have no choice */
-	spin_lock_irqsave(&priv->lock, flags);
+		dai->stream[SNDRV_PCM_STREAM_PLAYBACK].active, dai->stream[SNDRV_PCM_STREAM_CAPTURE].active);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (_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);
+		if (cancel_work_sync(&priv->work_codec_clk) != 0) {}
+		#if CONFIG_AC10X_TRIG_LOCK
+		/* I know it will degrades performance, but I have no choice */
+		spin_lock_irqsave(&priv->lock, flags);
+		#endif
+		// if (_set_clock[SNDRV_PCM_STREAM_CAPTURE]) _set_clock[SNDRV_PCM_STREAM_CAPTURE](1, substream, cmd, dai);
+		// if (_set_clock[SNDRV_PCM_STREAM_PLAYBACK]) _set_clock[SNDRV_PCM_STREAM_PLAYBACK](1, substream, cmd, dai);
+		#if CONFIG_AC10X_TRIG_LOCK
+		spin_unlock_irqrestore(&priv->lock, flags);
+		#endif
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		/* capture channel resync, if overrun */
-		if (dai->capture_active && substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		if (dai->stream[SNDRV_PCM_STREAM_CAPTURE].active && substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 			break;
 		}
-		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);
+
+		/* interrupt environment */
+		if (in_irq() || in_nmi() || in_serving_softirq()) {
+			priv->try_stop = 0;
+			if (0 != schedule_work(&priv->work_codec_clk)) {
+			}
+		} else {
+			// if (_set_clock[SNDRV_PCM_STREAM_CAPTURE]) _set_clock[SNDRV_PCM_STREAM_CAPTURE](0, NULL, 0, NULL); /* not using 2nd to 4th arg if 1st == 0 */
+			// if (_set_clock[SNDRV_PCM_STREAM_PLAYBACK]) _set_clock[SNDRV_PCM_STREAM_PLAYBACK](0, NULL, 0, NULL); /* not using 2nd to 4th arg if 1st == 0 */
+		}
 		break;
 	default:
 		ret = -EINVAL;
 	}
 
-	spin_unlock_irqrestore(&priv->lock, flags);
+	dev_dbg(rtd->card->dev, "%s() stream=%s  cmd=%d play:%d, capt:%d;finished %d\n",
+		__FUNCTION__, snd_pcm_stream_str(substream), cmd,
+		dai->stream[SNDRV_PCM_STREAM_PLAYBACK].active, dai->stream[SNDRV_PCM_STREAM_CAPTURE].active, ret);
 
 	return ret;
 }
@@ -210,23 +274,165 @@ 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 = rtd->codec_dai;
-	struct snd_soc_dai *cpu = rtd->cpu_dai;
+	struct snd_soc_dai *codec = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu = asoc_rtd_to_cpu(rtd, 0);
 	struct seeed_dai_props *dai_props =
 		seeed_priv_to_props(priv, rtd->num);
 	int ret;
 
-	ret = asoc_simple_card_init_dai(codec, &dai_props->codec_dai);
+	ret = asoc_simple_init_dai(codec, &dai_props->codec_dai);
 	if (ret < 0)
 		return ret;
 
-	ret = asoc_simple_card_init_dai(cpu, &dai_props->cpu_dai);
+	ret = asoc_simple_init_dai(cpu, &dai_props->cpu_dai);
 	if (ret < 0)
 		return ret;
 
+#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,7,0)
+	ret = asoc_simple_init_dai_link_params(rtd);
+	if (ret < 0)
+		return ret;
+#endif
+
+	dev_dbg(rtd->card->dev, "codec \"%s\" mapping to cpu \"%s\"\n", codec->name, cpu->name);
 	return 0;
 }
 
@@ -254,32 +460,37 @@ 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 (!cpu || !codec) {
+	if (!codec) {
 		ret = -EINVAL;
 		dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop);
 		goto dai_link_of_err;
 	}
 
-	ret = asoc_simple_card_parse_daifmt(dev, node, codec,
+	ret = asoc_simple_parse_daifmt(dev, node, codec,
 					    prefix, &dai_link->dai_fmt);
 	if (ret < 0)
 		goto dai_link_of_err;
 
 	of_property_read_u32(node, "mclk-fs", &dai_props->mclk_fs);
 
-	ret = asoc_simple_card_parse_cpu(cpu, dai_link,
-					 DAI, CELL, &single_cpu);
+	ret = asoc_simple_parse_cpu(cpu, dai_link, &single_cpu);
 	if (ret < 0)
 		goto dai_link_of_err;
 
 	#if _SINGLE_CODEC
-	ret = asoc_simple_card_parse_codec(codec, dai_link, DAI, CELL);
+	ret = asoc_simple_parse_codec(codec, dai_link);
 	if (ret < 0)
 		goto dai_link_of_err;
 	#else
@@ -291,7 +502,7 @@ 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_card_parse_platform(plat, dai_link, DAI, CELL);
+	ret = asoc_simple_parse_platform(plat, dai_link);
 	if (ret < 0)
 		goto dai_link_of_err;
 
@@ -316,7 +527,7 @@ 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_card_parse_clk_cpu(dev, cpu, dai_link, cpu_dai);
+	ret = asoc_simple_parse_clk_cpu(dev, cpu, dai_link, cpu_dai);
 	#endif
 	if (ret < 0)
 		goto dai_link_of_err;
@@ -324,22 +535,16 @@ 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_card_parse_clk_codec(dev, codec, dai_link, codec_dai);
+	ret = asoc_simple_parse_clk_codec(dev, codec, dai_link, codec_dai);
 	#endif
 	if (ret < 0)
 		goto dai_link_of_err;
 
-	#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,
+	ret = asoc_simple_set_dailink_name(dev, dai_link,
 						"%s-%s",
-						dai_link->cpu_dai_name,
+						dai_link->cpus->dai_name,
 						#if _SINGLE_CODEC
-						dai_link->codec_dai_name
+						dai_link->codecs->dai_name
 						#else
 						dai_link->codecs[0].dai_name
 						#endif
@@ -353,17 +558,27 @@ 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->cpu_dai_name,
+		dai_link->cpus->dai_name,
 		dai_props->cpu_dai.sysclk);
 	dev_dbg(dev, "\tcodec : %s / %d\n",
 		#if _SINGLE_CODEC
-		dai_link->codec_dai_name,
+		dai_link->codecs->dai_name,
 		#else
 		dai_link->codecs[0].dai_name,
 		#endif
 		dai_props->codec_dai.sysclk);
 
-	asoc_simple_card_canonicalize_cpu(dai_link, single_cpu);
+#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,13,0)
+	asoc_simple_canonicalize_cpu(dai_link->cpus, single_cpu);
+	#if _SINGLE_CODEC
+	asoc_simple_canonicalize_platform(dai_link->platforms, dai_link->cpus);
+	#endif
+#else
+	asoc_simple_canonicalize_cpu(dai_link, single_cpu);
+	#if _SINGLE_CODEC
+	asoc_simple_canonicalize_platform(dai_link);
+	#endif
+#endif
 
 dai_link_of_err:
 	of_node_put(cpu);
@@ -395,7 +610,7 @@ 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].codec_of_node = aux_node;
+		priv->snd_card.aux_dev[i].dlc.of_node = aux_node;
 	}
 
 	priv->snd_card.num_aux_devs = n;
@@ -455,7 +670,7 @@ static int seeed_voice_card_parse_of(struct device_node *node,
 			goto card_parse_end;
 	}
 
-	ret = asoc_simple_card_parse_card_name(&priv->snd_card, PREFIX);
+	ret = asoc_simple_parse_card_name(&priv->snd_card, PREFIX);
 	if (ret < 0)
 		goto card_parse_end;
 
@@ -480,6 +695,79 @@ 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;
@@ -487,7 +775,7 @@ 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;
+	int num, ret, i;
 
 	/* Get the number of DAI links */
 	if (np && of_get_child_by_name(np, PREFIX "dai-link"))
@@ -505,6 +793,25 @@ 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;
 
@@ -523,6 +830,9 @@ 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) {
@@ -539,13 +849,19 @@ 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,
@@ -556,14 +872,20 @@ 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_card_clean_reference(&priv->snd_card);
+	asoc_simple_clean_reference(&priv->snd_card);
 
 	return ret;
 }
@@ -571,8 +893,13 @@ 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);
 
-	return asoc_simple_card_clean_reference(card);
+	if (cancel_work_sync(&priv->work_codec_clk) != 0) {
+	}
+	asoc_simple_clean_reference(card);
+
+	return 0;
 }
 
 static const struct of_device_id seeed_voice_of_match[] = {

sound-compatible-4.18.h

@@ -0,0 +1,47 @@
+/*
+ * (C) Copyright 2017-2018
+ * Seeed Technology Co., Ltd. <www.seeedstudio.com>
+ *
+ * PeterYang <linsheng.yang@seeed.cc>
+ */
+#ifndef __SOUND_COMPATIBLE_4_18_H__
+#define __SOUND_COMPATIBLE_4_18_H__
+
+#include <linux/version.h>
+
+#if LINUX_VERSION_CODE >= KERNEL_VERSION(4,17,0)
+#define __NO_SND_SOC_CODEC_DRV     1
+#else
+#define __NO_SND_SOC_CODEC_DRV     0
+#endif
+
+#if LINUX_VERSION_CODE < KERNEL_VERSION(5,4,0)
+#if __has_attribute(__fallthrough__)
+# define fallthrough                    __attribute__((__fallthrough__))
+#else
+# define fallthrough                    do {} while (0)  /* fallthrough */
+#endif
+#endif
+
+#if __NO_SND_SOC_CODEC_DRV
+#define codec                      component
+#define snd_soc_codec              snd_soc_component
+#define snd_soc_codec_driver       snd_soc_component_driver
+#define snd_soc_codec_get_drvdata  snd_soc_component_get_drvdata
+#define snd_soc_codec_get_dapm     snd_soc_component_get_dapm
+#define snd_soc_codec_get_bias_level snd_soc_component_get_bias_level
+#define snd_soc_kcontrol_codec     snd_soc_kcontrol_component
+#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,9,0)
+#define snd_soc_read               snd_soc_component_read
+#else
+#define snd_soc_read               snd_soc_component_read32
+#endif
+#define snd_soc_register_codec     devm_snd_soc_register_component
+#define snd_soc_unregister_codec   snd_soc_unregister_component
+#define snd_soc_update_bits        snd_soc_component_update_bits
+#define snd_soc_write              snd_soc_component_write
+#define snd_soc_add_codec_controls snd_soc_add_component_controls
+#endif
+
+#endif//__SOUND_COMPATIBLE_4_18_H__
+

tools/phase_test.py

@@ -0,0 +1,83 @@
+import sys
+import wave
+import numpy as np
+
+
+if len(sys.argv) != 2:
+    print('Usage: {} multi.wav'.format(sys.argv[0]))
+    sys.exit(1)
+
+
+multi = wave.open(sys.argv[1], 'rb')
+rate = multi.getframerate()
+channels = multi.getnchannels()
+
+if channels <= 1:
+    sys.exit(1)
+
+N = rate 
+
+window = np.hanning(N)
+
+interp = 4*8
+max_offset = int(rate * 0.1 / 340 * interp)
+
+def gcc_phat(sig, refsig, fs=1, max_tau=None, interp=16):
+    '''
+    This function computes the offset between the signal sig and the reference signal refsig
+    using the Generalized Cross Correlation - Phase Transform (GCC-PHAT)method.
+    '''
+    
+    # make sure the length for the FFT is larger or equal than len(sig) + len(refsig)
+    n = sig.shape[0] + refsig.shape[0]
+
+    # Generalized Cross Correlation Phase Transform
+    SIG = np.fft.rfft(sig, n=n)
+    REFSIG = np.fft.rfft(refsig, n=n)
+    R = SIG * np.conj(REFSIG)
+    #R /= np.abs(R)
+
+    cc = np.fft.irfft(R, n=(interp * n))
+
+    max_shift = int(interp * n / 2)
+    if max_tau:
+        max_shift = np.minimum(int(interp * fs * max_tau), max_shift)
+
+    cc = np.concatenate((cc[-max_shift:], cc[:max_shift+1]))
+
+    # find max cross correlation index
+    shift = np.argmax(np.abs(cc)) - max_shift
+
+    tau = shift / float(interp * fs)
+    
+    return tau, cc
+
+
+print(multi.getsampwidth())
+
+while True:
+    data = multi.readframes(N)
+
+    if len(data) != multi.getsampwidth() * N * channels:
+        print("done")
+        break
+
+    if multi.getsampwidth() == 2:
+        data = np.fromstring(data, dtype='int16')
+    else:
+        data = np.fromstring(data, dtype='int32')
+    ref_buf = data[0::channels]
+
+    offsets = []
+    for ch in range(1, channels):
+        sig_buf = data[ch::channels]
+        tau, _ = gcc_phat(sig_buf * window, ref_buf * window, fs=1, max_tau=max_offset, interp=interp)
+        # tau, _ = gcc_phat(sig_buf, ref_buf, fs=rate, max_tau=1)
+
+        offsets.append(tau)
+
+    print(offsets)
+
+print(multi.getframerate())
+
+multi.close()

ubuntu-prerequisite.sh

@@ -0,0 +1,37 @@
+#!/bin/bash
+
+# Copyright (c) Hin-Tak Leung 2020
+#
+# Overview:
+#   This script compiles and install the Broadcom VideoCore tools,
+#   configure the dynamic loader for the non-standard library location,
+#   and update the loader cache.
+#
+#   A few steps explicitly requires root privilege, which are
+#   marked with "sudo". The rest is just checking for duplicate/previous
+#   action.
+#
+#   This derived from my command history on ubuntu 20.04.1 .YMMV
+
+sudo apt install -y git gcc g++ make alsa-utils cmake
+
+git clone git://github.com/raspberrypi/userland.git
+pushd userland/
+
+arch=$(uname -m)
+if [[ "$arch" =~ aarch64 ]]; then
+    ./buildme --aarch64
+else
+    ./buildme
+fi
+# ./buildme already includes "sudo make install" at the end
+
+popd
+
+# matches Raspbian's location:
+if [ ! -f /etc/ld.so.conf.d/00-vmcs.conf ] ; then
+    echo "/opt/vc/lib" | sudo tee -a /etc/ld.so.conf.d/00-vmcs.conf
+    sudo ldconfig -v
+else
+    echo "/etc/ld.so.conf.d/00-vmcs.conf exists - no need to update ld.cache!"
+fi

uninstall.sh

@@ -13,16 +13,64 @@ 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"
-rm  /boot/overlays/seeed-2mic-voicecard.dtbo || true
-rm  /boot/overlays/seeed-4mic-voicecard.dtbo || true
-rm  /boot/overlays/seeed-8mic-voicecard.dtbo || true
+for i in $RPI_HATS; do
+    dtoverlay -r $i
+done
+OVERLAYS=/boot/overlays
+[ -d /boot/firmware/overlays ] && OVERLAYS=/boot/firmware/overlays
+
+rm  ${OVERLAYS}/seeed-2mic-voicecard.dtbo || true
+rm  ${OVERLAYS}/seeed-4mic-voicecard.dtbo || true
+rm  ${OVERLAYS}/seeed-8mic-voicecard.dtbo || true
 
 echo "remove alsa configs"
 rm -rf  /etc/voicecard/ || true
 
 echo "disabled seeed-voicecard.service "
+systemctl stop seeed-voicecard.service
 systemctl disable seeed-voicecard.service
 
 echo "remove seeed-voicecard"
@@ -33,9 +81,18 @@ echo "remove dkms"
 rm  -rf /var/lib/dkms/seeed-voicecard || true
 
 echo "remove kernel modules"
-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
+rm  /lib/modules/*/kernel/sound/soc/codecs/snd-soc-wm8960.ko || true
+rm  /lib/modules/*/kernel/sound/soc/codecs/snd-soc-ac108.ko || true
+rm  /lib/modules/*/kernel/sound/soc/bcm/snd-soc-seeed-voicecard.ko || true
+rm  /lib/modules/*/updates/dkms/snd-soc-wm8960.ko || true
+rm  /lib/modules/*/updates/dkms/snd-soc-ac108.ko || true
+rm  /lib/modules/*/updates/dkms/snd-soc-seeed-voicecard.ko || true
+
+echo "remove $CONFIG configuration"
+for i in $RPI_HATS; do
+    echo Uninstall $i ...
+    do_overlay $i 1
+done
 
 echo "------------------------------------------------------"
 echo "Please reboot your raspberry pi to apply all settings"

wm8960.c

@@ -25,6 +25,7 @@
 #include <sound/initval.h>
 #include <sound/tlv.h>
 #include <sound/wm8960.h>
+#include "sound-compatible-4.18.h"
 
 #include "wm8960.h"
 
@@ -507,7 +508,11 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec)
 	 * list each time to find the desired power state do so now
 	 * and save the result.
 	 */
+	#if __NO_SND_SOC_CODEC_DRV
+	list_for_each_entry(w, &codec->card->widgets, list) {
+	#else
 	list_for_each_entry(w, &codec->component.card->widgets, list) {
+	#endif
 		if (w->dapm != dapm)
 			continue;
 		if (strcmp(w->name, "LOUT1 PGA") == 0)
@@ -748,6 +753,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
 			iface |= 0x000c;
 			break;
 		}
+		fallthrough;
 	default:
 		dev_err(codec->dev, "unsupported width %d\n",
 			params_width(params));
@@ -790,7 +796,7 @@ static int wm8960_hw_free(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8960_mute(struct snd_soc_dai *dai, int mute)
+static int wm8960_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_codec *codec = dai->codec;
 
@@ -1230,11 +1236,12 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
 static const struct snd_soc_dai_ops wm8960_dai_ops = {
 	.hw_params = wm8960_hw_params,
 	.hw_free = wm8960_hw_free,
-	.digital_mute = wm8960_mute,
+	.mute_stream = wm8960_mute,
 	.set_fmt = wm8960_set_dai_fmt,
 	.set_clkdiv = wm8960_set_dai_clkdiv,
 	.set_pll = wm8960_set_dai_pll,
 	.set_sysclk = wm8960_set_dai_sysclk,
+	.no_capture_mute = 1,
 };
 
 static struct snd_soc_dai_driver wm8960_dai = {
@@ -1252,7 +1259,11 @@ static struct snd_soc_dai_driver wm8960_dai = {
 		.rates = WM8960_RATES,
 		.formats = WM8960_FORMATS,},
 	.ops = &wm8960_dai_ops,
+#if LINUX_VERSION_CODE >= KERNEL_VERSION(5,12,0)
+	.symmetric_rate = 1,
+#else
 	.symmetric_rates = 1,
+#endif
 };
 
 static int wm8960_probe(struct snd_soc_codec *codec)
@@ -1276,6 +1287,11 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm8960 = {
 	.probe =	wm8960_probe,
 	.set_bias_level = wm8960_set_bias_level,
 	.suspend_bias_off	= true,
+	#if __NO_SND_SOC_CODEC_DRV
+	.idle_bias_on		= 1,
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
+	#endif
 };
 
 static const struct regmap_config wm8960_regmap = {
@@ -1302,8 +1318,7 @@ static void wm8960_set_pdata_from_of(struct i2c_client *i2c,
 		pdata->shared_lrclk = true;
 }
 
-static int wm8960_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
+static int wm8960_i2c_probe(struct i2c_client *i2c)
 {
 	struct wm8960_data *pdata = dev_get_platdata(&i2c->dev);
 	struct wm8960_priv *wm8960;
@@ -1368,10 +1383,9 @@ static int wm8960_i2c_probe(struct i2c_client *i2c,
 	return ret;
 }
 
-static int wm8960_i2c_remove(struct i2c_client *client)
+static void wm8960_i2c_remove(struct i2c_client *client)
 {
 	snd_soc_unregister_codec(&client->dev);
-	return 0;
 }
 
 static const struct i2c_device_id wm8960_i2c_id[] = {