Hi all, I have an audio algorithm on Apple (Altivec), which requires about 1400 MFlops (Float32) for Stereo, mostly multiply (700 MFLops) and...

Hi all, I have an audio algorithm on Apple (Altivec), which requires about 1400 MFlops (Float32) for Stereo, mostly multiply (700 MFLops) and add (700 MFlops). The demonstrator currently runs on a Powerbook G4, currently in mono. The next job would be to built a prototype device to convince potential customers that I can built it on hardware, i.e. without requiring a dedicated computer, an...

Greetings! I am doing a bit of research on current technigues used for speech enhancement / noise suppression and I came across the following...

Greetings! I am doing a bit of research on current technigues used for speech enhancement / noise suppression and I came across the following papers: N. Virag, "Single channel speech enhancement based on masking properties of the human auditory system," IEEE Trans. Speech and Audio Processing, vol. 7, no. 2, pp. 126-137, Mar. 1999. J. D. Johnston, "Transform coding of audio signal usi...

I'm working on a simple audio processing system using the dsPIC30F4013. It's mainly intended for amateur radio use, and I've formed a Yahoo...

I'm working on a simple audio processing system using the dsPIC30F4013. It's mainly intended for amateur radio use, and I've formed a Yahoo group for people interested in using dsPICs for radio applications: http://groups.yahoo.com/group/radio_dspic/ Leon -- Leon Heller, G1HSM http://www.geocities.com/leon_heller

Hi, I'm looking some pointers concerning the design of lossless audio (plus "silence") codecs. I want to deploy these on either end of a...

Hi, I'm looking some pointers concerning the design of lossless audio (plus "silence") codecs. I want to deploy these on either end of a packet switched network (coder at server, decoder at client). I.e., they are intended primarily for communication bandwidth reduction. Push content into coder, pass over network, extract content via decoder, *consume* (and discard). I.e., the system ...

Hello, I would like to know the effect of phase response in filters, in practical applications. For example, when dealing with audio...

Hello, I would like to know the effect of phase response in filters, in practical applications. For example, when dealing with audio signals, if audio signal is passed through a filter in order to filter out undesired frequency component(s) and if the filter has linear phase response, I would assume that output after filtering is equivalent to adding individual frequeny components back to c...

Hallo, I am using .WAV files as audio signal input. For implementing D-4 Wavelet filter coefficients of both HPF and LPF, its said that I have...

Hallo, I am using .WAV files as audio signal input. For implementing D-4 Wavelet filter coefficients of both HPF and LPF, its said that I have to multiply with input array. How will you consider the input from WAV files as arrays, also what will be in the case of stereo, where it has left and right channel values? Also, I am in new in the world of wavelet. What is the purpose of scaling functio...

I just started with DSP and think this is a simple question. I'm mostly interested in audio processing using FFT. My question is, what is...

I just started with DSP and think this is a simple question. I'm mostly interested in audio processing using FFT. My question is, what is the maximum resolution of the phase (for each frequency)? i.e. When transforming a 1024 samples signal (each 16bit integer for audio), the first frequency, f=0, can only be zero or PI (plus or minus DC). The last frequency, f=512, can also only be zero or P...

hi, I had certain querries about adding dither in audio signals. 1) I presume one of the reasons to add it is to compensate for quantisation...

hi, I had certain querries about adding dither in audio signals. 1) I presume one of the reasons to add it is to compensate for quantisation error while converting analog to digital. Is there any other reason for adding it? 2) Is it added only when the signal strength is zero, or would it be advantageous to add it always. If quantization error is the primary consideration, I pr...

Can anyone tell me the best way to introduce a delay between audio input and output in PCs? I explain better. I have an external audio...

Can anyone tell me the best way to introduce a delay between audio input and output in PCs? I explain better. I have an external audio source, connected to the MIC input of the laptop. I would like that the soundcard outputs the such input with a 3 seconds delay? Are there software to do this? Daniele

Could some DSP guru please help me a bit ? I am creating a FFT based application that uses an audio input. The sampling details are...

Could some DSP guru please help me a bit ? I am creating a FFT based application that uses an audio input. The sampling details are as follows: encoding : PCM sampling frequency : 16000 Hz resolution : 16 bits channel : mono signed : true endianness : little (as this runs on an Intel processor) The raw bytes get collected in a byte array, and I take two bytes at a time (resolution is 1...

Could some DSP guru please help me a bit ? I am creating a FFT based application that uses an audio input. The sampling details are...

Could some DSP guru please help me a bit ? I am creating a FFT based application that uses an audio input. The sampling details are as follows: encoding : PCM sampling frequency : 16000 Hz resolution : 16 bits channel : mono signed : true endianness : little (as this runs on an Intel processor) The raw bytes get collected in a byte array, and I take two bytes at a time (resolution is 1...

Hi everyone, I'm writing a tutorial series on how to create audio plug-ins using C++ and the WDL-OL Framework. I've finished the first 18...

Hi everyone, I'm writing a tutorial series on how to create audio plug-ins using C++ and the WDL-OL Framework. I've finished the first 18 posts, covering topics such as creating envelopes, multimode filters, anti-aliased oscillators, LFOs as well as receiving and processing MIDI and GUI programming. Basically we're building a subtractive synthesizer from scratch. Here's the table of

Hi, For an experiment i have a simple application that takes a quadrature signal from a radio and then does the following: Complex FFT -> ...

Hi, For an experiment i have a simple application that takes a quadrature signal from a radio and then does the following: Complex FFT -> Inverse FFT -> Cartesian to Polar -> audio out If I tune the radio to an AM signal I can demodulate it perfectly well by this but it depends on how the FFT is processed. In the FFT routine, I apply a window and if it is anything other than a rectangular wind

Hi, I am trying to extract MFCC coefficients for variable sample rate audio feed 1.Input Sample rate is dynamic and is not known in front (...

Hi, I am trying to extract MFCC coefficients for variable sample rate audio feed 1.Input Sample rate is dynamic and is not known in front ( an arbitrary media file ) though it is fixed per session and doesn’t change over time. 2.All Input feeds are being down sampled to a predefined sampling-rate ( say 8Khz or 16Khz ) 3.Re-sampling is done in the following way: a.2nd order Lowpass

Hello all, i have searched the internet for information about which of the two overlap methods is better but couldnt find any info. i am...

Hello all, i have searched the internet for information about which of the two overlap methods is better but couldnt find any info. i am implementing Decimation In Time fft in matlab. the entire process is: 1. audio sampling using wavread 2. computing fft of SEGMENTS of input audio samples 3. Computing fft of filter 4. Multiplying the two above fft's and taking inverse fft to get t...

Hi all, Im currently doing my final year project based on an Audio frequency pitch detector. I'm using the Analog- to- digital technique to...

Hi all, Im currently doing my final year project based on an Audio frequency pitch detector. I'm using the Analog- to- digital technique to extract the fundamental frequency of any sound that the device detects. However, I would like o know how using DSP chip is better to detect pit in comparison to ADC? Is there any downsides for using DSP chip instead? Thanks in advance ____________...

Hi all, I am developping an real time audio compression codec system.In my system I adopted 16-bit 48kHz to 5 bit 48kHz IMA-ADPCM algorithm,but...

Hi all, I am developping an real time audio compression codec system.In my system I adopted 16-bit 48kHz to 5 bit 48kHz IMA-ADPCM algorithm,but could not get the satisfied results,audio result was not good. I know from someone,there is an tech call "interpolated" will do the job. I would like to hear from some one how to work with interpolated or get some useful web URL for it. Thanks...

Dear All, I hope to do the audio sample rate convertion(resampling) between the ISDN and my device. The ISDN audio sample rate is 8000,...

Dear All, I hope to do the audio sample rate convertion(resampling) between the ISDN and my device. The ISDN audio sample rate is 8000, 16bits/sample, Mono but my device use 48000, 16bits/sample, Mono. I can do the resampling use some software library, but it is heavy loading. I hope to handle 30 calls(E1) concurrently. But I can only handle 2 calls on a P4 3.0 PC. Could you give me s...

Hello, I am using C++ on a linux little endian machine, and am interfacing to OpenSound system to produce my audio sound. So my goal is to be...

Hello, I am using C++ on a linux little endian machine, and am interfacing to OpenSound system to produce my audio sound. So my goal is to be able to reprduce the correct raw sound data corresponding to spectral data I already have (for example, I know the frequency versus power information for at least 128 bins, and over about 30 time points per second) I am using the four1 function found...

I have never before worked with either DSP hardware or microcontrollers. What are the advantages of using DSP hardware over using...

I have never before worked with either DSP hardware or microcontrollers. What are the advantages of using DSP hardware over using a microcontroller to perform DSP functions? What I am trying to do is measure the audio noise level (one part of an ambitious project). One idea I have is to use DSP hardware to digitize the audio input, do a DFT on it, integrate over the DFT, and use the su...