iMX6ULL in i2s slave mode, no input audio

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iMX6ULL in i2s slave mode, no input audio

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Contributor I

We have a board which has an iMX6ULL SOM module with a CS4270 codec attached via sai1. It is set up so that the codec is the i2s master. In this setup, we have found that audio output (playing back a file) is working fine, however we are not getting any input from the codec. The result of running `arecord` is a file of all 0's as seen below:

 

 

arecord --format=S24_LE --duration=5 --rate=48000 --file-type=raw -c 2 out.raw -vvv
Recording raw data 'out.raw' : Signed 24 bit Little Endian, Rate 48000 Hz, Stereo
Plug PCM: Hardware PCM card 0 'r0-audio' device 0 subdevice 0
Its setup is:
  stream       : CAPTURE
  access       : RW_INTERLEAVED
  format       : S24_LE
  subformat    : STD
  channels     : 2
  rate         : 48000
  exact rate   : 48000 (48000/1)
  msbits       : 32
  buffer_size  : 24000
  period_size  : 6000
  period_time  : 125000
  tstamp_mode  : NONE
  tstamp_type  : MONOTONIC
  period_step  : 1
  avail_min    : 6000
  period_event : 0
  start_threshold  : 1
  stop_threshold   : 24000
  silence_threshold: 0
  silence_size : 0
  boundary     : 1572864000
  appl_ptr     : 0
  hw_ptr       : 0
Max peak (12000 samples): 0x00000000 #                    0%
Max peak (12000 samples): 0x00000000 #                    0%
Max peak (12000 samples): 0x00000000 #                    0%
Max peak (12000 samples): 0x00000000 #                    0%
Max peak (12000 samples): 0x00000000 #                    0%
Max peak (12000 samples): 0x00000000 #                    0%
Max peak (12000 samples): 0x00000000 #                    0%
Max peak (12000 samples): 0x00000000 #                    0%
Max peak (12000 samples): 0x00000000 #                    0%
Max peak (12000 samples): 0x00000000 #                    0%
Max peak (12000 samples): 0x00000000 #                    0%
Max peak (12000 samples): 0x00000000 #                    0%
Max peak (12000 samples): 0x00000000 #                    0%
Max peak (12000 samples): 0x00000000 #                    0%
^CAborted by signal Interrupt...
Max peak (12000 samples): 0x00000000 #                    0%
root@r0:~# hexdump out.raw 
0000000 0000 0000 0000 0000 0000 0000 0000 0000
*
00afc80

 

 

 Here is what we have in our device tree:

 

 

#include "imx6ul.dtsi"

/{
...
	codec_clk: aud_clk {
		compatible = "fixed-clock";
		#clock-cells = <0>;
		clock-frequency = <12288000>;
	};

	sound {
		/* Simple-card driver definition */
		status = "okay";
		compatible = "simple-audio-card";
		simple-audio-card,name = "r0-audio";
		/* Define i2s format, see fig. 10 in cs4270 data sheet */
		simple-audio-card,format = "i2s";
		/* Set codec as clock master */
		simple-audio-card,bitclock-master = <&codec_dai>;
		simple-audio-card,frame-master = <&codec_dai>;

		/* Define the cpu audio block */
		cpu_dai: simple-audio-card,cpu {
			/* i2s from codec is connected to sai1 */
			sound-dai = <&sai1>;
			/* 2 data frames (stereo)
			 * 32-bit width (even though we only fill 24)
			 */
			dai-tdm-slot-num = <2>;
			dai-tdm-slot-width = <32>;
		};

		/* Define the codec audio block */
		codec_dai: simple-audio-card,codec {
			/* Point to the codec definition at i2c2 */
			sound-dai = <&codec>;
			/* Provide the clock definition since the codec is master */
			clocks = <&codec_clk 0>;
		};
	};

/* CS4270 codec is attached to sai1 on the CPU, this block enables it */
&sai1 {
	pinctrl-names = "default";
	pinctrl-0 = <&pinctrl_sai1>;
	status = "okay";
};

/* Audio I2C Bus */
&i2c2 {
	clock_frequency = <100000>;
	pinctrl-names = "default";
	pinctrl-0 = <&pinctrl_i2c2>;
	status = "okay";

	/* CS4270 codec is at address 0x48 */
	codec: cs4270@48 {
		#sound-dai-cells = <0>;
		status = "okay";
		compatible = "cirrus,cs4270";
		reg = <0x48>;
		reset-gpios = <&gpio2 11 0>;
	};
};

&iomuxc {
	pinctrl-names = "default";
	pinctrl-0 = <&pinctrl_hog>;

	imx6ul-evk {
		/* Pins required for sai1 (i2s to codec) */
		pinctrl_sai1: sai1grp {
			fsl,pins = <
				/* cs4270 codec wired to SAI1 */
				/* Note that the RX_BCLK and RX_SYNC clock follow the TX clocks */
				MX6UL_PAD_CSI_DATA04__SAI1_TX_SYNC	0x1b080
				MX6UL_PAD_CSI_DATA05__SAI1_TX_BCLK	0x1b080
				MX6UL_PAD_CSI_DATA06__SAI1_RX_DATA	0x11080
				MX6UL_PAD_CSI_DATA07__SAI1_TX_DATA	0x1f0b8
			>;
		};
    };
};

 

 

 Here is what we have in `asound.conf`

 

 

pcm.device{
        format S24_LE
        rate 48000
        type hw 
        card 0
        device 0
}

pcm.!default{
        type plug
        slave.pcm "device"
}

 

 

This is all working fine for audio playback. I have checked the iMX registers using devmem2 and it appears that everything is set up correctly for the sai module, however I am not able to get any audio data. We have verified on a scope that the SCLK, LRCLK, and i2s data is coming in correctly, however it seems that the data is never being received by ALSA applications. Any help would be appreciated! 

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NXP TechSupport
NXP TechSupport

Hi Willard

 

one can try amixer described in sect.7.5 Audio Mixer (AUDMIX)  i.MX Linux Reference Manual​

https://source.codeaurora.org/external/imx/imx-test/tree/test/mxc_amix_test?h=imx_5.4.24_2.1.0

 

Best regards
igor