DSPRelated.com

FFT and orthogonality....

Started by mash...@yahoo.com in Audio Signal Processing15 years ago 2 replies

Hi all! I am really new in DSP, finally i found this group. I hope members of DSP could help me honestly. My question is for what we use FFT...

Hi all! I am really new in DSP, finally i found this group. I hope members of DSP could help me honestly. My question is for what we use FFT (DFT, IFFT, IDFT) and why is needed? I read we use IFFT (or IDFT) in OFDM modulation scheme to transmit signals, what is the aim using it? (to change some properties ?) My second quesyion is about orthogonality. I can explain with mathemathical r...


Mixing of multiple audio files

Started by MyMo...@gmail.com in Audio Signal Processing15 years ago 2 replies

Hi everyone, I am new to the field of audio programming and gave myself some tasks to bring myself up to speed. The first task is programming a...

Hi everyone, I am new to the field of audio programming and gave myself some tasks to bring myself up to speed. The first task is programming a (simple) DAW, that can mix several channels of audio. I would like to get some opinions if my approach is somewhat ok. this is what I do: 1. defined audio buffer 2. for each WORD in the audio buffer I loop through all my channels and get the WOR...


How to deal with the overlap when constructing signal from frequency domain?

Started by hyee...@yahoo.com.cn in Audio Signal Processing15 years ago

All,here is 2 questions about constrcuting signal from frequency domain. Any comments are appreciated. 1. I transmit some speech signal,with...

All,here is 2 questions about constrcuting signal from frequency domain. Any comments are appreciated. 1. I transmit some speech signal,with no window, into frequency by fft, and do nothing and back to time domain by ifft at once. It sound good as the original siganl. But repeated above with hamming/hanning windowed signal,the constructed signal would induce some noise(Hiss or dithering). why?...


Help in choosing a DSP devlopment platform

Started by clif...@orangecactus.net in Audio Signal Processing15 years ago

Hello all- I am currently considering the purchase of a new DSP development platform which will be used for the creation of digital audio...

Hello all- I am currently considering the purchase of a new DSP development platform which will be used for the creation of digital audio processing units. At the moment both Analog Devices and Texas Instruments have some very good deals available on certain software and hardware pieces for this kind of thing, and I was wondering what some of you have found as far as "pluses" and "minuses" bet...


Pitch-Scaling with Phase Vocoder - Interpolating the Phase

Started by Sebastian Adolf in Audio Signal Processing15 years ago

Hello! I am currently writing my master-thesis on pitch-scaling and time-stretching and came across several problems... one of them i'll...

Hello! I am currently writing my master-thesis on pitch-scaling and time-stretching and came across several problems... one of them i'll describe here: In one of my pitch-scaling-implementations I am following the great bernsee-tutorial from http://www.dspdimension.com/admin/pitch-shifting-using-the-ft/ . My aim is now to extend his work with some interpolation-stuff for better signa...


Waterfall windowing problems

Started by elvi...@hotmail.com in Audio Signal Processing16 years ago 1 reply

Dears, I am writing a program to make the waterfall plot, but have some doubt. Can you help? 1. I am using square window, is it...

Dears, I am writing a program to make the waterfall plot, but have some doubt. Can you help? 1. I am using square window, is it correct? 2. Shall I keep the window size or shrink it in every step? Thanks for your help. Alvis


FFT output interpretation (FFTW v. Ooura)

Started by rela...@hotmail.co.uk in Audio Signal Processing16 years ago 14 replies

I'm new(ish) to FFT and to broaden my knowledge/experience I've been comparing FFTW and the Ooura FFT library. I'm getting a result with the...

I'm new(ish) to FFT and to broaden my knowledge/experience I've been comparing FFTW and the Ooura FFT library. I'm getting a result with the latter that I don't quite understand, however. I have been analysing 1024 samples from a WAV file (16-bit, sr 44100) using FFTW's real-to-complex transform and the equivalent rdft from the Ooura library. I get the same output apart from the DC bin. FFTW gives...


Basic Algorithm/formula to convert mic db output to db SPL

Started by jsnuff1 in Audio Signal Processing16 years ago 2 replies

I have a mic that outputs its level in decibels (the range this mic outputs seems to be between around -55db to 0db) and would like to find an...

I have a mic that outputs its level in decibels (the range this mic outputs seems to be between around -55db to 0db) and would like to find an algorithm that converts this value into decibel SPL. I have done some research and have used this formula that seems to do the basic job, but the values it outputs dont seem to be very accurate. Here is the formula im using now dbspl(db) = 2...


Is there such a thing as a simple DSP?

Started by George Bean in Audio Signal Processing16 years ago

I have an application to decode an FSK signal on a VHF carrier. In the past I would have used an EXAR XR2211A FSK demodulator chip feeding...

I have an application to decode an FSK signal on a VHF carrier. In the past I would have used an EXAR XR2211A FSK demodulator chip feeding a microcontroller but I assume with the proliferation of DSP's, the days of these chips and ones like them are numbered. Hoping to make the leap into the 21st century, I started investigating the incorporation of a DSP into my design. What I found was a be...


Mixing down from multiple input sources

Started by "d.abc51" in Audio Signal Processing16 years ago 4 replies

Hi all, I'm trying to code a program that can mix together multiple PCM wav files into a single output. It is essentially a sequencer, and works...

Hi all, I'm trying to code a program that can mix together multiple PCM wav files into a single output. It is essentially a sequencer, and works fine at points in the sequence where only one or two sounds are playing, however any more than that and the distortion can become unbearable. All I am doing right now is adding the sample values for each channel together and taking the sum of thes...


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