Following the instructions I was able to make it work. The audio playback is working pretty well. However, I cannot test the audio input. I saw that it is a line level input (line in), so it cannot be connected to a regular microphone.
I want to test it through an echo scheme. I will produce some sound through q source (connected to the line in input of the cape) and check the response in the audio output, through an earphone.
I tried to connect the cape line in to the earphone output of an old cellphone through a male male jack cable (I saw in some forums that the line level is similar to the earphone output of some appliances, apart from the impedance difference.). Unfortunately, it didn’t work.
The interesting thing is that when the cable is connected to the cape audio input and the other connector is unplluged, I can hear the noise from static, similar to an unplugged to the amplifier electric guitar .
Can someone suggest me a simple form to test the audio input?
Following the instructions I was able to make it work. The audio playback is working pretty well. However, I cannot test the audio input. I saw that it is a line level input (line in), so it cannot be connected to a regular microphone.
I want to test it through an echo scheme. I will produce some sound through q source (connected to the line in input of the cape) and check the response in the audio output, through an earphone.
I tried to connect the cape line in to the earphone output of an old cellphone through a male male jack cable (I saw in some forums that the line level is similar to the earphone output of some appliances, apart from the impedance difference.). Unfortunately, it didn’t work.
The interesting thing is that when the cable is connected to the cape audio input and the other connector is unplluged, I can hear the noise from static, similar to an unplugged to the amplifier electric guitar .
Can someone suggest me a simple form to test the audio input?
I don’t have the Audio Cape, but I do have a TLV320AIC3106EVM connected to my BBB and it works fine. You may want to try the following command:
This will show you a bar graph in the terminal so you can see the input levels. I would just connect the output from you phone to the line input of your audio cape. In another terminal, run amixer to adjust the input levels.
It seems that I messed with the amixer parameters. Now I can not even
hear from the speaker test. Is there a way of reseting alsa
parameters, not only the general audio volume?
I saw something like
alsactl restore -P
and removing '/var/lib/alsa/asound.state
but it didn't work for parameters specifically to the DA830 EVM.
Why the audio cape is recognized as a DA830 EVM?
I suspect that the audio input is not working due to one of that parameters.
It seems that I messed with the amixer parameters. Now I can not even
hear from the speaker test. Is there a way of reseting alsa
parameters, not only the general audio volume?
I saw something like
alsactl restore -P
and removing '/var/lib/alsa/asound.state
After this, you need to restart alsa-utils. Also make sure you don¹t have
.asoundrc in your home folder.
but it didn't work for parameters specifically to the DA830 EVM.
Why the audio cape is recognized as a DA830 EVM?
The TLV320AIC3xxx codec is recognized as a DA830 EVM. If you look in the
audio cape DTS file, you will see ti,model = ³DA830 EVM².
It seems that I messed with the amixer parameters. Now I can not even
hear from the speaker test. Is there a way of reseting alsa
parameters, not only the general audio volume?
I saw something like
alsactl restore -P
and removing '/var/lib/alsa/asound.state
in my use of audio and this card from the console I found alasmixer much easier to use. I just went through and set all the levels to 100% and it works well. Try using alsamixer it should get things fixed up in an easy and intuitive way. F1 and the man page cover the controls.
in my use of audio and this card from the console I found alasmixer much easier to use. I just went through and set all the levels to 100% and it works well. Try using alsamixer it should get things fixed up in an easy and intuitive way. F1 and the man page cover the controls.
This is true, and if you run alsamixer in a SSH session, it looks much better than in a tty console like minicom.
Thank you all for the support so far. I did it. Now it is working.
It's a shame to admit but It was the capture sound level that was too
low. I used alsamixer and entered in the capture tab (F4) and put all
the inputs in the highest level.
I’ll probably try the same thing but inject the 1.8V from my bench supply.
Look at /arch/arm/boot/dts/am335x-bone-common-pinmux.dtsi line 220. These are the pins used to connect to the TLV320AIC3106EVM. The table below shows how I got this working.
I am actually trying to route to the mono line but having difficulties, have you attempted to do that? We have a custom board with an on-board TLV320AIC3106 but want to connect up the EVM to a stock BBB to isolate potential hardware issues (and we can’t easily access the mono line as it is connected directly to the microphone input of another component). Using alsamixer, we’ve turned on the mono line, upped the DAC mixing volume to max and turned on DAC L mixing. DAC L is set to output to the mixer (L1). I was able to get this to work using the Windows software using what look to be similar / identical settings.
Ultimately, we are trying to loopback USB microphone input on one of the digital channels through to the MONO line, while at the same time routing LINE1L to the LEFT channel. Hopefully this is possible. It’s an inherited board design so we’re hoping to avoid hardware changes.
One thing we’ve been struggling to understand is what the audio routing section does in the device tree overlay so any advice you can give on that would be appreciated.
I am actually trying to route to the mono line but having difficulties, have you attempted to do that? We have a custom board with an on-board TLV320AIC3106 but want to connect up the EVM to a stock BBB to isolate potential hardware issues (and we can’t easily access the mono line as it is connected directly to the microphone input of another component). Using alsamixer, we’ve turned on the mono line, upped the DAC mixing volume to max and turned on DAC L mixing. DAC L is set to output to the mixer (L1). I was able to get this to work using the Windows software using what look to be similar / identical settings.
Ultimately, we are trying to loopback USB microphone input on one of the digital channels through to the MONO line, while at the same time routing LINE1L to the LEFT channel. Hopefully this is possible. It’s an inherited board design so we’re hoping to avoid hardware changes.
One thing we’ve been struggling to understand is what the audio routing section does in the device tree overlay so any advice you can give on that would be appreciated.
Look at /Documentation/devicetree/bindings/sound/tlv320aic3x.txt
I think you need MONO_LOUT connected to LINE1L
See also /Documentation/devicetree/bindings/sound/davinci-evm-audio.txt