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45 Threads found on edaboard.com: Voice Filter
Hi , since I am a starter in this audio processing , I was studying a code I got from edaboard discussion , i need to know the way in which they implemented the gain equation. can any body help me to understand the code ? clc; clear all; close all; load handel.mat filename = 'handel.wav'; =audioread('handel.wav'); v=rand (
The ADSL filters are Low-pass filters-you want a highpass filter. But your problem is more complicated because you don't want to distort the DSL signal. The 'normal' filters will distort the voice-band signals, but it doesn't matter too much, but if you start messing up your digital signals, you could have (...)
Your tiny speakers might sound like tweeters but they have a voice coil that is about 3 ohms so the 470uF output capacitors should charge or discharge in about 0.07 seconds. Do the tiny speakers have other capacitors in series?
I have listen many radio stations live programs on my radio receiver & noticed that a great amount of hum in voice. Also i noticed that hum is inbuilt in their transmission so there is no question of fault in receiver. My QUESTION IS THAT IS THERE ANY SMART TECHNIQUE OR SIMULATION TECHNIQUE TO REMOVE HUM FROM voice (AUDIO SIGNAL) DURING LI
I don't understand how to store voice signal in sd card and how to write code to store voicein sd card Plz, help me I want to make a recorder project
Passive RF voice or data filters shall be specified by many parameters of your choice. Passband width, often @-3dB Passband ripple (dB) Passband Group delay ripple (affects data bit shift) Bandstop or Rejection Bandwidth @ -X dB Input Z Output Current or power or input? Then you decide on topology and values and
Hi, We made a voice call using GSM module. But while playing voice which is stoed in apr33a3 module using GSM module, we are receiving a call itself.. can anyone give suggestion?
My project is: "Reading out a voice file and filter the voice signal with 5.5kHz bandwidth, 4 kHz and 3.2 kHz. (Self design digital filters with sampling rates to accurately perform this work). Listening to a voice file after filtering and describe the influence of the low-pass (...)
This is just a thought - ignoring the technical issues. Listening to 'raw' SSB through an AM demodulator it is clear that there is a correlation between the original voice and the resulting RF output, even though it isn't intelligible. Tuning through an SSB signal changes the recovered amplitude as the IF filter bandwidth is imposed on the baseband
I think 0.05 is your threshold... Now in this case is this is the threshold is in analog voice recording side or in digital data processing side.... If it is on digital side the you can use simple if else for this threshold based approach ..... But way that noise filtering is done in digital side using designing proper digital filter for (...)
The clearest voice audio would be given by the filter with minimum Group Delay. Bessel type has lowest group delay but a good compromise between group delay and rejection is Butterworth type.
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How does your filter do if the background amplitude is equal or greater than the voice amplitude?
Hi, I connected PWM output (VOUT1 and VOUT2) pins of aP8942A (pre recorded voice) to SIM900 MIC pins via filter circuit. When I first power on the board and make a call to SIM900 I am not able to listen the voice. After that if I power OFF and ON the module, I am able to listen the voice. Reset by using (...)
You sample your voice WITHOUT the noise to determine its spectrum, then create a filter that passes that frequency band. If the noise is wideband you can obviously only reject the noise outside your band of interest. That will improve Signal-to-noise ratio, but won't totally eliminate noise.
actually my FYP project is about to investigate the performance of voice echo over wireless project is using adaptive filter combined with bessel i want to know more about bessel filter..please anybody help me...:-(
sir, I am using a voice signal and obtained the subband signals by passing it through a 8 channel QMF filter bank I require the acs values of the sub band signals When computed using the matlab command XCORR ,autocorrelation in some bands is zero Can anybody kindly explain me how to evaluate the autocorrelation values upto lag 'k'.Moreover I want
what is your apllication? for voice or for data? if data, what is your data rate?
I want to take voice signal from condensor microphone i used a inverting opamp amplifier with a low pass filter of 3khz how much i should keep the gain for LM324 operating at +5v
The mechanical resonators part of the filter can attain very high quality factor Q, which gives to mechanical filters excellent selectivity. But the most important thing for a voice or CW communication system, the mechanical filters have good group delay response, making radios to sound different than others that use (...)