how capture audio from a headphone with microphone using dsP

[ransiluj]

hey friends,
i'm doing noise suppression and communication project. what i basically doing is I'm going take the speech from the microphone that in headphone. and i have another mic and i'm taking background noise from that. Then i'm giving that noise and voice to dsPIC30f4011. i want know how i detect sound coming from the microphone using dsPIC. I have no idea about it. Please help me.
thank you

 

[audioguru]

The phases of the noise will be different between the two microphones due to different distances from the noises. You need to adjust the phases to be the same or you will not have cancellation.

 

[JimB]

Such a lot of complication.

Why don't you just do what aircraft headsets have done for the last 50 years?
There are two microphone capsules in the mic, wired in series, one capsule is facing the pilots mouth, the other capsule is facing away.
The background noise is the same in both mic capsules and cancels, the voice is greater in one capsule and so the result is voice with minimum noise.

JimB

 

[ransiluj]

Such a lot of complication.

Why don't you just do what aircraft headsets have done for the last 50 years?
There are two microphone capsules in the mic, wired in series, one capsule is facing the pilots mouth, the other capsule is facing away.
The background noise is the same in both mic capsules and cancels, the voice is greater in one capsule and so the result is voice with minimum noise.

JimB

this is my final year project... um making this using adaptive filters.. and then um communicating with another person using bluetooth.. this is good for noisy industry environment to communicate.. can you please help me

 

[ransiluj]

The phases of the noise will be different between the two microphones due to different distances from the noises. You need to adjust the phases to be the same or you will not have cancellation.

hello... um going to use adaptive filters.. i think that method is good right ?

 

[audioguru]

hello... um going to use adaptive filters.. i think that method is good right ?

I worked with boardroom tele-conferencing systems that used adaptive filters to reduce long distance feedback and echoes. They worked a little but usually some voice-switching attenuation was also needed.
They were "trained" by playing pink noise through the speakers which was picked up by the mic then the DSP made a model of the acoustics. The system cancelled voice sounds from the speakers from being transmitted back to the other end. It worked poorly when both ends talked at the same time.
If a mic or speaker was moved or if a door was opened then the model of the acoustics was ruined and the DSP failed unless it slowly re-trained from the voices.

Maybe your DSP can be trained by noise.

 

[magvitron]

if you are doing a project ie an academic project, if simulation is enough , do it via lab-view it has got an inbuilt adaptive block that you can use for capturing and cancelling noise, echo.. etc. if you know matlab, you can do adaptive filtering by that too. take a look at its help menu and look for adaptive filters, or you can use simulink.

 

[ransiluj]

if you are doing a project ie an academic project, if simulation is enough , do it via lab-view it has got an inbuilt adaptive block that you can use for capturing and cancelling noise, echo.. etc. if you know matlab, you can do adaptive filtering by that too. take a look at its help menu and look for adaptive filters, or you can use simulink.

i'm want to build hardware.. i want to build an adaptive filter inside the dsPIC.. i want to know how to capture the sound first using dsPIC. Please help

 

[ransiluj]

I worked with boardroom tele-conferencing systems that used adaptive filters to reduce long distance feedback and echoes. They worked a little but usually some voice-switching attenuation was also needed.
They were "trained" by playing pink noise through the speakers which was picked up by the mic then the DSP made a model of the acoustics. The system cancelled voice sounds from the speakers from being transmitted back to the other end. It worked poorly when both ends talked at the same time.
If a mic or speaker was moved or if a door was opened then the model of the acoustics was ruined and the DSP failed unless it slowly re-trained from the voices.

Maybe your DSP can be trained by noise.

i don't think there is echo for me. (coz of background noise.) i will give you example for wot i'm going to made. click this video.. this is the device
https://www.youtube.com/watch?v=j_SGdT7EmBQ
so u think that problem gonna cause me too ?

 

[magvitron]

taking inputs by ADC is the thing for you. You can do like this. mic>> preamp>>adc of the pic

 

[audioguru]

Did you notice that the voice level of the guy with the snow blower was jumping up and down? The system is cancelling the low frequency snow blower sound plus his voice!

Did you notice that the system cancels only low frequencies because their wavelength is long? Because the phases of the two mics are the same?

Some expensive cars also have a system that has a mic and powerful speakers that cancel some low frequency engine sound.