How to measure SNR in an music file?

Started by bhas...@soctronics.com in Audio Signal Processing9 years ago 1 reply

Hi everybody, I need a method or tool to measure the quality of an audio signal after doing some processing...

Hi everybody, I need a method or tool to measure the quality of an audio signal after doing some processing (interpolation,...etc). First of all how to find which is noise in a music file(through some method)? What are the parameters which represents the quality of the audio signal(I mean 20KHz bandwidth)? How to measure them? Thanks in advance. Regard...


Need a developer for a voice processing device

Started by Thomas David Kehoe in Audio Signal Processing9 years ago

Casa Futura Technologies is the leading manufacturer of electronic assistive devices for speech disorders. We need to contract a developer to...

Casa Futura Technologies is the leading manufacturer of electronic assistive devices for speech disorders. We need to contract a developer to make a speech aid for Parkinson's patients. For 15+ years our devices have been based on karaoke chips, but these chips are becoming obsolete and our devices are becoming more complex, so we need to move to a DSP-based platform. (We considered developin...


STFT or FFT Waterfall for loudspeaker?

Started by elvi...@hotmail.com in Audio Signal Processing9 years ago 2 replies

Dears, I wonder to know in those audio test equipment (e.g. CLIO, Audio Precision), they have a function called Waterfall or CSD. It is for...

Dears, I wonder to know in those audio test equipment (e.g. CLIO, Audio Precision), they have a function called Waterfall or CSD. It is for analyzing a speaker time-frequency relation from an impulse. Do they use FFT or STFT to generate the spectrum? Also, what windows do they use? I tried to FFT an impulse with shifting square window but I got a lot of low frequency noise after the first F...


help for increasing the accuracy for a content based audio classification syste

Started by Lita in Audio Signal Processing9 years ago

Hi; We are doing the above mentioned project for audio classification.We are coding in MATLAB.We have tried implementing the code by...

Hi; We are doing the above mentioned project for audio classification.We are coding in MATLAB.We have tried implementing the code by increasing the number of classes in the database from 4 to 7 and also the number of MFCC's from 13 to 30.We stuck to using left right HMM for now.In spite of all this the system accuracy remains more or less unaffected.By accuracy I mean the percentage of cli...


How make 3d sound in matlab??

Started by robs...@yahoo.com in Audio Signal Processing9 years ago

Hello all i trying make 3d sound in matlab (like virtual hair cut), and i think that is depend on amplitude(far,near) and shift(left, right)...

Hello all i trying make 3d sound in matlab (like virtual hair cut), and i think that is depend on amplitude(far,near) and shift(left, right) of orginal signal i success in part of amplitude, but not in shift the adea is one channel far become near and other near become far (like person who talk and move from left to right) this is the code i write it (one channel record), no error but...


Re: Calculation of filter coefficient in Sigma studio

Started by ramm...@ymail.com in Audio Signal Processing9 years ago

hi.... in my previous posy i found w0 is not clear so resending my queries again. i'm currently working on Analog Devices Dsp processor ADAU...

hi.... in my previous posy i found w0 is not clear so resending my queries again. i'm currently working on Analog Devices Dsp processor ADAU 1701.. Sigma studio is the software we are now on,in that i have some queries in the way they calculate filter coefficient. - main specification i needed to design a filter 1)Type[LPF,HPF, etc] 2)frequency 3)Q 4)Gain eg: I want a 2nd ord...


Filter coefficient calculation in Sigma studio

Started by ramm...@ymail.com in Audio Signal Processing9 years ago

hi.... i'm currently working on Analog Devices Dsp processor ADAU 1701.. Sigma studio is the software we are now on,in that i have some queries...

hi.... i'm currently working on Analog Devices Dsp processor ADAU 1701.. Sigma studio is the software we are now on,in that i have some queries in the way they calculate filter coefficient. - main specification i needed to design a filter 1)Type[LPF,HPF, etc] 2)frequency 3)Q 4)Gain eg: I want a 2nd order LPF with Frequency= 1000 Q=0.5 Gain=1 The given below derivation is th...


DSP for real-time audio effects processing

Started by joan...@gmail.com in Audio Signal Processing9 years ago 5 replies

Hi there, I am looking for an affordable DSP in price to manufacture a small device capable of real-time audio effects processing through a...

Hi there, I am looking for an affordable DSP in price to manufacture a small device capable of real-time audio effects processing through a microphone. Is there anybody who can recommend me which DSP could be the most interesting to program and use for this application? I have been checking DSPs from Analog Devices, Texas Instruments and Freescale, but I am a bit lost in the different models, a...


FIR in audio processing

Started by ramm...@ymail.com in Audio Signal Processing9 years ago 3 replies

hi... I'm new in dsp field.Just a fresher working for Audio product based company. i have some queries please help me. 1. why we choose IIR...

hi... I'm new in dsp field.Just a fresher working for Audio product based company. i have some queries please help me. 1. why we choose IIR filter design,for practical application,especially for processing audio signal. 2. Why not FIR? 3. Depending on the response there are many IIR filter(butterworth,bessel etc).Is there any sub division for FIR? 4. What are the disadvantage of FIR filte...


FIR in audio processing

Started by Rajmathi S in Audio Signal Processing9 years ago

Hi Rammya, ? 1. Why we choose IIR filter design for practical application, especially for processing audio signal? ? We can design an IIR...

Hi Rammya, ? 1. Why we choose IIR filter design for practical application, especially for processing audio signal? ? We can design an IIR Filter of far less order than an FIR filter giving us similar desired response. This greatly reduces the hardware resources required. On the flip side, IIR Filters have a non-linear phase response, which if acceptable, the filter is best designed as IIR. Als...


Ask a Question to the DSPRelated community

To significantly increase your chances of receiving answers, please make sure to:

  1. Use a meaningful title
  2. Express your question clearly and well
  3. Do not use this forum to promote your product, service or business
  4. Write in clear, grammatical, correctly-spelled language
  5. Do not post content that violates a copyright