TLV320AIC3104: Clock related issue

Hi,

I will get back to you on Tuesday about this, sorry about the delay. 

Hi Kokila,

Hi again. I believe it will be called at runtime, and if not, it would be when you set the clock in your DTS file. Can you provide your DTS and let's see where we need to define the MCLK/BCLK rate. I would think it is in the <clocks> section.

https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git/tree/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml

Best,
Mir

Hi Mir,

This is my DTS entry for the TLV. and for the clocks param what is the expected value we need to give?

tlv320aic320: tlv320aic320@18 {
                compatible = "ti,tlv320aic3104";
                reg = <0x18>;
                #sound-dai-cells = <0>;

                ai3x-micbias-vg = <3>;
                //reset-gpios = <&io_expander 11 GPIO_ACTIVE_LOW>;
                AVDD-supply = <&codec_3v3_reg>;
                IOVDD-supply = <&codec_3v3_reg>;
                DRVDD-supply = <&codec_1v8_reg>;
                DVDD-supply = <&codec_1v8_reg>;
        };

Hi,

I found this old e2e talking about the same problem:  https://e2e.ti.com/support/audio-group/audio/f/audio-forum/702805/rtos-tlv320aic3105-tlv320aic3105-codec-showing-system-clock-sysclk-as-0 

They suggest adding in the line aic3x->sysclk = <value> in the driver itself... but according to other forum posts, it may be expecting your SoC to change the MCLK rate. You can check out this forum for some suggestions for the DTS or changing the driver to a hard coded MCLK:

https://community.nxp.com/t5/i-MX-Processors/i-MX8QXP-SAI-MCLK-frequency/m-p/1097084#M160628

https://www.programmersought.com/article/40346906740/

Sorry I am not more help, I am not a Linux DTS expert.

Best,
Mir

Hello Mir,

Now my PLL error is solved, and Iam able to capture, but after capturing when i try to check the wav file, there is no valid data.

exynosauto9_easdk_android_vm:/ # i2cdump -y -f 29 0x18
     0  1  2  3  4  5  6  7  8  9  a  b  c  d  e  f    0123456789abcdef
00: 00 00 00 91 10 0c dc 0a 00 00 00 01 00 00 00 20    ...?????...?...
10: 20 ff ff 04 78 78 04 78 78 06 00 fe 00 00 fe 00     ???xx?xx?.?..?.
20: 00 00 00 00 cc 00 00 00 40 00 00 af af 00 2f af    ....?...@..??./?
30: 00 00 00 0c 00 2f af 00 00 00 0c 00 00 00 00 2f    ...?./?...?..../
40: af 0c 00 00 00 00 2f af 0c 00 00 00 00 00 00 00    ??..../??.......
50: 00 2f af 00 00 00 08 00 00 00 00 2f af 08 00 00    ./?...?..../??..
60: 00 00 00 00 00 00 a2 00 00 00 00 00 00 00 00 00    ......?.........
70: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00    ................
80: 00 00 00 91 10 0c dc 0a 00 00 00 01 00 00 00 20    ...?????...?...
90: 20 ff ff 04 78 78 04 78 78 06 00 fe 00 00 fe 00     ???xx?xx?.?..?.
a0: 00 00 00 00 cc 00 00 00 40 00 00 af af 00 2f af    ....?...@..??./?
b0: 00 00 00 0c 00 2f af 00 00 00 0c 00 00 00 00 2f    ...?./?...?..../
c0: af 0c 00 00 00 00 2f af 0c 00 00 00 00 00 00 00    ??..../??.......
d0: 00 2f af 00 00 00 08 00 00 00 00 2f af 08 00 00    ./?...?..../??..
e0: 00 00 00 00 00 00 a2 00 00 00 00 00 00 00 00 00    ......?.........
f0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00    ................
exynosauto9_easdk_android_vm:/ #

Also, Let me know for routing of the audio, should we set any value for mixer?

Please find the logs while try to capture the audio.

exynosauto9_easdk_android_vm:/ # tinycap /mnt/test.wav -d 16 -c 2 -r 48000 -b 16
Capturing sample: 2 ch, 48000 hz, 16 bit
Captured 417792 frames
exynosauto9_easdk_android_vm:/ #

Hi,

Can you share what you changed to fix your PLL/clocks? Just for future reference. 

And yes, you need to add dapm_routes to the DTS file. You can find the options by searching "snd_soc_dapm_route" in the driver aic3x.c file: https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git/tree/sound/soc/codecs/tlv320aic3x.c And the codec pin names are in the yaml help file for the driver: https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git/tree/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml

There are common values in the "intercon" section but the AIC3104 has its own few options as well. It depends which pins you are trying to use. 

Best,
Mir

Hi Mir,

For PLL issue, I tried to set the sysclk from the master, This automatically calls the set_sysclk function, which resolves PLL error.

 ret = snd_soc_dai_set_sysclk(codec_dai, 2, 24080000, SND_SOC_CLOCK_OUT);
        if (ret) {
            dev_err(card->dev, "failed to set tlv320aic31xx sysclk!\n");
            return ret;
        }

Now, Pls help me with the Mixer settings, what are the value we need to set for the Mic capture.?

I tried to capture the audio, The files is captured there is no valid audio data, I think iam missing some of the mixer routing. Please help me with the Mixer configuration

Hi Kokila,

You will want to include some routing in your dts file, like this:

         simple-audio-card,widgets =
                "Microphone", "Microphone Jack",
                "Microphone", "Internal Microphone",
                "Headphone", "Headphone Jack",
                "Line", "Line Out",
                "Speaker", "Internal Speaker";
            simple-audio-card,routing =
                "Line Out", "HPLCOM",
                "Line Out", "HPRCOM",
                "Headphone Jack", "HPLOUT",
                "Headphone Jack", "HPROUT",
                "Microphone Jack", "MIC2L",
                "Internal Microphone", "MIC1L",
                "Internal Speaker", "LLOUT",
                "Internal Speaker", "RLOUT";

-Mir

Hello Mir, 

Now we are able to Playback the audio, When we try to capture the audio is not getting captured. getting the junk data on the file.

These are the default value of the mixer's controls, Can you please comment on it? What are teh value we need to set to get the audio recorded properly?

522	ENUM	1	Left Line1L Mux                          single-ended
523	ENUM	1	Left Line1R Mux                          single-ended
524	ENUM	1	Right Line1L Mux                         single-ended
525	ENUM	1	Right Line1R Mux                         single-ended
526	BOOL	1	Left PGA Mixer Line1L Switch             On
527	BOOL	1	Left PGA Mixer Line1R Switch             Off
528	BOOL	1	Left PGA Mixer Mic2L Switch              Off
529	BOOL	1	Left PGA Mixer Mic2R Switch              Off
530	BOOL	1	Right PGA Mixer Line1R Switch            On
531	BOOL	1	Right PGA Mixer Line1L Switch            Off
532	BOOL	1	Right PGA Mixer Mic2L Switch             Off
533	BOOL	1	Right PGA Mixer Mic2R Switch             Off

Also, This is the Path Iam trying to Capture the audio via MIC.

What exactly the value we need to set for MICBIAS? Also Please find the I2cdump of the TLV Codec

exynosauto9_easdk_android_vm:/ # i2cdump -y -f 29 0x18

     0  1  2  3  4  5  6  7  8  9  a  b  c  d  e  f    0123456789abcdef

00: 00 00 00 91 10 00 00 0a 00 00 00 41 00 00 00 20    ...??..?...A... 

10: 20 00 00 04 78 78 04 78 78 06 00 fe 00 00 fe 00     ..?xx?xx?.?..?.

20: 00 00 00 00 44 00 00 00 40 00 00 af af 00 2f af    ....D...@..??./?

30: 00 00 00 0c 00 2f af 00 00 00 0c 00 00 00 00 2f    ...?./?...?..../

40: af 0c 00 00 00 00 2f af 0c 00 00 00 00 00 00 00    ??..../??.......

50: 00 2f af 00 00 00 08 00 00 00 00 2f af 08 00 00    ./?...?..../??..

60: 00 00 00 00 00 00 a2 00 00 00 00 00 00 00 00 00    ......?.........

70: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00    ................

80: 00 00 00 91 10 00 00 0a 00 00 00 01 00 00 00 20    ...??..?...?... 

90: 20 00 00 04 78 78 04 78 78 06 00 fe 00 00 fe 00     ..?xx?xx?.?..?.

a0: 00 00 00 00 44 00 00 00 40 00 00 af af 00 2f af    ....D...@..??./?

b0: 00 00 00 0c 00 2f af 00 00 00 0c 00 00 00 00 2f    ...?./?...?..../

c0: af 0c 00 00 00 00 2f af 0c 00 00 00 00 00 00 00    ??..../??.......

d0: 00 2f af 00 00 00 08 00 00 00 00 2f af 08 00 00    ./?...?..../??..

e0: 00 00 00 00 00 00 a2 00 00 00 00 00 00 00 00 00    ......?.........

f0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00    ................

 

Hi,

Give me a day to read through this, sorry about the delay. The junk data looks like it is offset differently for each channel, this leads me to believe that either your mic bias is incorrect or it is set to AC or DC coupled when you do not intend, something like this. 

-Mir

Can you share your schematic? I see that in register 25, 0x19, micbias is powered down during your recording. Also, there is an overflow flag in register 11 (0x0b) that says the right ADC has overflowed, and we see that your right channel on the Audacity screenshot is close to +1. So, I think that your bias on the microphone signals may be incorrect. We expect analog input signals to be between 0 and AVDD, and micbias may be somewhere between 1.8 and 2.5V. So, if the signals are not externally biased, then there will be an issue since your micbias is not enabled (and so the signal is not properly AC-coupled). Can you also measure the voltage coming out of the microphone to check that range and verify that you have analog signal coming out as you expect?

Best,
Mir

Hi Mir, 

As you pointed out, we found an issue in Mic power supply and its resolved. after that I have tried capturing the Audio, still the audio is not recorded. The new waveform looks like this now.

Also, this is how i2cdump looks now, that ADC overflow is not happening.

exynosauto9_easdk_android_vm:/ # i2cdump -y -f 29 0x18
     0  1  2  3  4  5  6  7  8  9  a  b  c  d  e  f    0123456789abcdef
00: 00 00 00 91 10 0c dc 0a 00 00 00 01 00 00 00 20    ...?????...?...
10: 20 ff ff 04 78 78 04 78 78 06 00 fe 00 00 fe 00     ???xx?xx?.?..?.
20: 00 00 00 00 cc 00 00 00 40 00 00 af af 00 2f af    ....?...@..??./?
30: 00 00 00 0c 00 2f af 00 00 00 0c 00 00 00 00 2f    ...?./?...?..../
40: af 0c 00 00 00 00 2f af 0c 00 00 00 00 00 00 00    ??..../??.......
50: 00 2f af 00 00 00 08 00 00 00 00 2f af 08 00 00    ./?...?..../??..
60: 00 00 00 00 00 00 a2 00 00 00 00 00 00 00 00 00    ......?.........
70: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00    ................
80: 00 00 00 91 10 0c dc 0a 00 00 00 01 00 00 00 20    ...?????...?...
90: 20 ff ff 04 78 78 04 78 78 06 00 fe 00 00 fe 00     ???xx?xx?.?..?.
a0: 00 00 00 00 cc 00 00 00 40 00 00 af af 00 2f af    ....?...@..??./?
b0: 00 00 00 0c 00 2f af 00 00 00 0c 00 00 00 00 2f    ...?./?...?..../
c0: af 0c 00 00 00 00 2f af 0c 00 00 00 00 00 00 00    ??..../??.......
d0: 00 2f af 00 00 00 08 00 00 00 00 2f af 08 00 00    ./?...?..../??..
e0: 00 00 00 00 00 00 a2 00 00 00 00 00 00 00 00 00    ......?.........
f0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00    ................

Can you help us here?  Did I miss any register setting with respect to Capture via Mic?

Thanks,

Kokila

Hi,

Can you share your schematic? This will help me understand what you intend to do. I see that there is not an overflow anymore, but micbias is still powered down. I also noticed that your J.D is different than it was in the previous screenshot, what clocks are you providing and what clocks do you expect out of the chip? Or are you providing all clocks? Here are my notes on the input registers:

03 91 #pll enabled, p=q=1
04 10 #j=4
05 0c #d msb = 1100 
06 dc #d lsb = 1101 11 -> 823 ??
07 0a #48k
0b 01 #no overflows
0f 20 #ladc pga is not muted, gain=32db
10 20 #radc pga not muted, gain=32db
11 ff #mic2l and mic2r not connected to ladc 
12 ff #mic2l and mic2r not connected to radc
13 04 #mic1lp single ended, connected to ladc pga, 0db, ladc powered up
15 78 #line1r not connected to ladc
16 04 #mic1rp single ended, connected to radc pga, 0db, radc powered up
18 78 #mic1lp not connected to radc
19 06 #micbias powered DOWN -> ??

Best,
Mir

Hi Mir, 

Please find the Schematic.

Audio codec schematic.pdf

Thanks 

Kokila K

Hi,

I think there is a chance that the audio you are getting is expected, can you try to increase the PGA gain? I am not sure about the swing of your microphone, but I notice in the schematic that you have ~10dB reduction in the mic signal from the op amp filter circuit. So, there is a chance that the voltage making it to the ADC is just low - if you can change the gain of that op amp you should see some improvement in the signal's swing. To remove the DC offset in your audio signal, you can select a preset ADC high pass filter in page 0 register 12 (0x0c).

Best,
Mir