Hi, I'm looking to buy a DSP evaluation board to play around with in my sparetime. I prefer fixed-point DSPs, but floating-point is also...

Hi, I'm looking to buy a DSP evaluation board to play around with in my sparetime. I prefer fixed-point DSPs, but floating-point is also ok. Audio sampling rate should be minimum 24kHz. The board is primarily going to be used for audio signal processing applications (hearing aids). Requirements: * Algebraic assembly language (like the algebraic assembly syntax of Analog Device...

Hi, I'm an engineering univeristy student and I'm doing some experiments with digital audio filters. I'm quite pratice with general purpose...

Hi, I'm an engineering univeristy student and I'm doing some experiments with digital audio filters. I'm quite pratice with general purpose micros but i'm absolutely new to the DSP world. The intent is to provide a 44.1 / 48 Khz 16 bit source to a DSP and make the DSP act as a lowpass/bandpass/higpass crossover filter. Searching on the internet seems that FIR filters are the best solution ...

Dear Members; We are looking for 1 or 2 engineering partners on a wireless microphone project. We're needing partners with experience in...

Dear Members; We are looking for 1 or 2 engineering partners on a wireless microphone project. We're needing partners with experience in speech recognition, audio codecs, networks/IP address issues and audio compression algorithims. The project includes assigning a wireless mike an IP address (as a standalone network device) transceiving encrypted digital signal and IC/hardware design...

Hi, I'm coding a synthesizer for musical purposes. DSP is not my major, so I'm hitting a kind of road block here. I'm modulating a filter...

Hi, I'm coding a synthesizer for musical purposes. DSP is not my major, so I'm hitting a kind of road block here. I'm modulating a filter cutoff at audio rate, (audio input signal is a saw tooth and the modulator is a sine wave). At high resonance, the modulation seams to produce side-bands around the frequency of the modulator. If I sweep the filter cutoff manually and approach those side b...

Hey, Im working with an ADSP-21364 SHARC processor and EZKIT. I have a problem when trying to integrate two programs that use the serial...

Hey, Im working with an ADSP-21364 SHARC processor and EZKIT. I have a problem when trying to integrate two programs that use the serial ports. My first program is based on the "Block based talkthrough" example, SPORT0 is used to take audio input from the ADC and SPORT2 is used to output the processed audio to a DAC. This stand-alone program functions correctly. My second program...

Good evening everyone, unfortunaatly I didn't find any answer in google. Designing a digital audio synthesizer, how do I handle the fast change...

Good evening everyone, unfortunaatly I didn't find any answer in google. Designing a digital audio synthesizer, how do I handle the fast change of filter coefficients? For now I implemented say 4 coefficients sets for a lowpass filter, with (say) cutoff freqs of 5000, 3000, 1000, 500 hz. I can't generate and keep in memory 128 (MIDI.... ) different coefficient sets, but even if my sets are all ...

Contact: dddial@pedley-richard.com Our client is a global leader of advanced semiconductor solutions to lead-edge communications companies...

Contact: dddial@pedley-richard.com Our client is a global leader of advanced semiconductor solutions to lead-edge communications companies that drive innovation and convergence in voice, data, and wireless networks. The position is based in Austin, TX and will require a combination of technical dsp audio architecture knowledge (ie- drivers and/or advanced applications) and leadership bac...

Contact: dddial@pedley-richard.com Our client is a global leader of advanced semiconductor solutions to lead-edge communications companies...

Contact: dddial@pedley-richard.com Our client is a global leader of advanced semiconductor solutions to lead-edge communications companies that drive innovation and convergence in voice, data, and wireless networks. The position is based in Austin, TX and will require a combination of technical dsp audio architecture knowledge (ie- drivers and/or advanced applications) and leadership bac...

I Need audio Anomaly file collection ( library) for EVRC family codecs on CDMA networks. Any ideas?

I Need audio Anomaly file collection ( library) for EVRC family codecs on CDMA networks. Any ideas?

hi hi i am new to dsp and would like to buy a starter kit but since they are quite pricey would like some advice before purchasing as i am...

hi hi i am new to dsp and would like to buy a starter kit but since they are quite pricey would like some advice before purchasing as i am interested in audio dsp. can a starter kit take a audio signal from a pc, filter the samples and then return to pc for listening all in real time? What are the general capabilties of starter kits, do you have to buy more kit to carry out the above pr...

hello everybody, I search in google a lot but I can not find code of using gibbs sampling in audio (signal or sound) processing. if you can please...

hello everybody, I search in google a lot but I can not find code of using gibbs sampling in audio (signal or sound) processing. if you can please help me. thanks in advance

hi, ive seen that transforms takes signals from time domain to frequency domain. this case is evident in audio and speech coding. but what...

hi, ive seen that transforms takes signals from time domain to frequency domain. this case is evident in audio and speech coding. but what does frequency mean in images and video? is there something similar to oscillations as in audio signals in freq domain. please do help TIA.. -- Jagadeesh

Hi, I'm an engineering univeristy student and I'm doing some experiments with digital audio filters. I'm quite pratice with general purpose...

Hi, I'm an engineering univeristy student and I'm doing some experiments with digital audio filters. I'm quite pratice with general purpose micros but i'm absolutely new to the DSP world. The intent is to provide a 44.1 / 48 Khz 16 bit source to a DSP and make the DSP act as a lowpass/bandpass/higpass crossover filter. Searching on the internet seems that FIR filters are the best solution (IIR...

It is well known that the class D audio amplifiers do have the severe nonlinear distortion problems. This happens due to the fundamental...

It is well known that the class D audio amplifiers do have the severe nonlinear distortion problems. This happens due to the fundamental properties of the PWM process, as well as due to the non-idealities of the real amplifier. The perception of those distortions is quite different from that of the distortions of an analog amp. Due to the sampled nature of the PWM system, the efficienc...

Perhaps I'm forgetting something, but why would the sample rate of the digital audio delivery medium such as CD have _anything_ to do with...

Perhaps I'm forgetting something, but why would the sample rate of the digital audio delivery medium such as CD have _anything_ to do with the ease of sampling the original signal? It is certainly true that it is better to oversample the original analog input so that the antialiasing filter requirements can be greatly relaxed. However, once we're in the digital domain we can run some very ...

I note with bemused disgust that Macromedia has decided to refrain from allowing people who use their software to save recorded audio in a...

I note with bemused disgust that Macromedia has decided to refrain from allowing people who use their software to save recorded audio in a portable format. Does anyone have a method to convert Nellymoser audio to pulse code modulation in better than real time? They have attempted to make it hard, but it's easy to do it in real time, and it is easy to prove that it always will be easy in ...

Hi there, I have an audio signal (speech) sampled at 44.1 kHz and I want to shift its frequency by a fixed amount in Hz. Ie., I want a...

Hi there, I have an audio signal (speech) sampled at 44.1 kHz and I want to shift its frequency by a fixed amount in Hz. Ie., I want a frequency at 100 Hz to be at 110 Hz, and a frequency at 235 Hz to be at 245 Hz. I tried multiplying the signal with a sine tone but that gives me not one shift by +10Hz but a shift by -10Hz as well. So how should I do that? I suspect it should be easy ...

hi all, I'm looking for a free,popular C/C++ library for digital signal processing. I will use to test some algorithm for audio and...

hi all, I'm looking for a free,popular C/C++ library for digital signal processing. I will use to test some algorithm for audio and speech processing. I usuall work with Maltab, but now I want to move to C, may be with MS Visual Studio 2005. Thks for any advice!

Hi, I have a "one off" project that has to take a standard stereo 24bit audio stream at 192KHz sample rate, that I need to convert into...

Hi, I have a "one off" project that has to take a standard stereo 24bit audio stream at 192KHz sample rate, that I need to convert into mono. I've browsed AD, TI etc, and the eval kits seem overkill. I've seen the alesis AL3101 4$ DSP's but they only go up to 48K sample rate http://www.wavefrontsemi.com/products.html Any suggestion of a simple evaluation kit that will suit my needs? ...

Hi, I am in a process to improve the speech quality out of our current speech codec. Basically, our speech codec is an ADPCM codec with...

Hi, I am in a process to improve the speech quality out of our current speech codec. Basically, our speech codec is an ADPCM codec with bunch of interpolation and decimation filters and Sigma-Delta ADDA, and we have already adjusted all the filters so now it works just fine. However, we are still not 100% satisfied with the audio quality, we want to make the sound more clear so we are thin...