tyler_wilson7
New member
Hi All,
This is my first time posting to any forum so I apologize if I break any norms...
I am a mechanical engineer that is in way over his head by taking on a DSP project. I am trying to make a voice amplification device for my grandfather who has lost a lot of his volume due to a tracheostomy surgery. After looking online for voice amplifiers on the market, I was rather disappointed and felt I could design something that would look less "old-man", be easier to use and adjust, and provide better quality sound. I am not sure what the final package will look like (it might be something that goes around his neck, or if I can make it small enough possibly something he can hide in his breast pocket or something), but the electrical system is going to be a Teensy 4.0 w/ 2 audio shields controlling 2 cheap electret microphones, which will send the I2S output to a class D amplifier which will drive 2 speakers (my original thought is two but if I shrink the form factor down it might go to one). My goal is to basically mimic hearing aid designs where there will be two microphones close to each other, but one will be slightly closer to his mouth/throat to pick up more voice, and the other will be slightly further away to then cancel out the system output and any other noise.
I have been working on this for about a month using the audio library and am very comfortable adjusting the parameters of the microphones and using mixers to combine and invert the signals. My latest code attempt left me able to almost completely cancel out my voice (as a test to see if the inversion was working) however I was still getting an insane amount of audio howling. After doing a lot of research I came to the conclusion I need an adaptive algorithm to take on this issue, but I am not nearly skilled enough to implement anything I found in research papers into actual code. This led me to an old 2016 post on this site (that I read earlier on but it seemed too complicated at the time) which used NLMS Audio Echo Cancellation between two microphone-speaker setups a certain distance away, but used two Teensy's. I am guessing the reasoning for the two Teensy's was a long transmission distance, but where my project will only have transmission distance of a few inches I want to modify the code to accommodate 2 mics with 1 Teensy + 2 Audio Shields.
Here is the link to the 2016 Post: https://forum.pjrc.com/threads/32910-Acoustic-Echo-Cancelation(AEC)
Here is the link to the GitHub page: https://github.com/kfillerup/NLMS_AEC
After combing through the code I have a pretty good idea at what is going on, however I am completely clueless as to how to modify the I2S inputs so that I can add a microphone which will fill the [x] signal array to be cancelled out. Right now it just says it is pulling from the far end transmission, but in reading through the thread it doesn't seem like he ever implemented the second Teensy so I can't tell where it left off. Can anyone help me in figuring out how to add the second microphone input?
Thanks,
Tyler
This is my first time posting to any forum so I apologize if I break any norms...
I am a mechanical engineer that is in way over his head by taking on a DSP project. I am trying to make a voice amplification device for my grandfather who has lost a lot of his volume due to a tracheostomy surgery. After looking online for voice amplifiers on the market, I was rather disappointed and felt I could design something that would look less "old-man", be easier to use and adjust, and provide better quality sound. I am not sure what the final package will look like (it might be something that goes around his neck, or if I can make it small enough possibly something he can hide in his breast pocket or something), but the electrical system is going to be a Teensy 4.0 w/ 2 audio shields controlling 2 cheap electret microphones, which will send the I2S output to a class D amplifier which will drive 2 speakers (my original thought is two but if I shrink the form factor down it might go to one). My goal is to basically mimic hearing aid designs where there will be two microphones close to each other, but one will be slightly closer to his mouth/throat to pick up more voice, and the other will be slightly further away to then cancel out the system output and any other noise.
I have been working on this for about a month using the audio library and am very comfortable adjusting the parameters of the microphones and using mixers to combine and invert the signals. My latest code attempt left me able to almost completely cancel out my voice (as a test to see if the inversion was working) however I was still getting an insane amount of audio howling. After doing a lot of research I came to the conclusion I need an adaptive algorithm to take on this issue, but I am not nearly skilled enough to implement anything I found in research papers into actual code. This led me to an old 2016 post on this site (that I read earlier on but it seemed too complicated at the time) which used NLMS Audio Echo Cancellation between two microphone-speaker setups a certain distance away, but used two Teensy's. I am guessing the reasoning for the two Teensy's was a long transmission distance, but where my project will only have transmission distance of a few inches I want to modify the code to accommodate 2 mics with 1 Teensy + 2 Audio Shields.
Here is the link to the 2016 Post: https://forum.pjrc.com/threads/32910-Acoustic-Echo-Cancelation(AEC)
Here is the link to the GitHub page: https://github.com/kfillerup/NLMS_AEC
After combing through the code I have a pretty good idea at what is going on, however I am completely clueless as to how to modify the I2S inputs so that I can add a microphone which will fill the [x] signal array to be cancelled out. Right now it just says it is pulling from the far end transmission, but in reading through the thread it doesn't seem like he ever implemented the second Teensy so I can't tell where it left off. Can anyone help me in figuring out how to add the second microphone input?
Thanks,
Tyler
