I'm not sure why it's a problem so this may not help too much:

1) you could try cutting the first half a second off the record file.

2) Many audio ADCs have a large DC offset and use  a digital high pass filter to remove it - it looks as if the RPi setup you have is only starting the ADC when you start the recording. If you start  a recording very soon after stopping  a previous recording do you still get the problem ?

The proper solution (if my guess about the error is correct) is to start up the ADC and let it settle before actually recording data - but this would require close control of the ADC which you may not have (without changing the driver code). You might be able to turn the high pass filter off - this will give you a DC offset which you could remove with software.

Do you have to use the RPi/Cirrus card combo ?

MK

Hello Michael,

My system wants to analyze the transmission channel,for this,  it is compared  an original file with the recorded file. At the same instant, playback and recording begins. The recorded file cannot be manipulated (it not possible try cutting the first half second) .

To setup the LineIn  (in second raspberry), it is necessary  to execute the script  "Record_from_lineIn.sh"

File:  Record_from_lineIn.sh

#!/bin/bash

# $1 added to support 1st line argument. i.e. "./Record_from_linein.sh -q" will stop all the control information being displayed on screen

#Record from onboard Line Input to AP

# +8dB input PGA gain

amixer $1 -Dhw:sndrpiwsp cset name='IN3L Volume' 8

amixer $1 -Dhw:sndrpiwsp cset name='IN3R Volume' 8

# better THD in normal mode vs lower noise floor in high performance

amixer $1 -Dhw:sndrpiwsp cset name='IN3 High Performance Switch' on

# Configure the input path for 0dB Gain,  HPF with a low cut off for DC removal

amixer $1 -Dhw:sndrpiwsp cset name='IN3L Digital Volume' 128

amixer $1 -Dhw:sndrpiwsp cset name='IN3R Digital Volume' 128

amixer $1 -Dhw:sndrpiwsp cset name='LHPF1 Input 1' IN3L

amixer $1 -Dhw:sndrpiwsp cset name='LHPF2 Input 1' IN3R

amixer $1 -Dhw:sndrpiwsp cset name='LHPF1 Mode' High-pass

amixer $1 -Dhw:sndrpiwsp cset name='LHPF2 Mode' High-pass

amixer $1 -Dhw:sndrpiwsp cset name='LHPF1 Coefficients' 240,3

amixer $1 -Dhw:sndrpiwsp cset name='LHPF2 Coefficients' 240,3

# Configure the Audio Interface and volume 0dB

amixer $1 -Dhw:sndrpiwsp cset name='AIF1TX1 Input 1' LHPF1

amixer $1 -Dhw:sndrpiwsp cset name='AIF1TX1 Input 1 Volume' 32

amixer $1 -Dhw:sndrpiwsp cset name='AIF1TX2 Input 1' LHPF2

amixer $1 -Dhw:sndrpiwsp cset name='AIF1TX2 Input 1 Volume'

I think that a high pass it setup, but I don´t know if  it necessary to setup any more to eliminate the transient signal.

Regards .

Jose

Attachments:

It looks like you have a capacitance component on your analog to digital converter.

You can have these types of events if you have a large stray capacitance caused by the input wires.

Are your audio input lines terminated?

You night have too much signal driving the device, which allows the off nominal signal to be higher than your signal ground.

Do some poking around your input device and look at the states before and after you begin to record.

DAB

There is a link to an earlier discussion here which tells you how you might adjust the High Pass filter:

DC offset on microphones and workaround.

or you could try to remove the DC offset at the line input.

MK

Thank you DAB and Machael for your answers.

Regards.

Jose.