Audio application problem

Started by PROVENTEK MINDCRAFT AB in comp.dsp19 years ago 19 replies

Hi folks, I'm working with an dsp audio application and I desperately need an algorithm for tone control. The filter is described in my...

Hi folks, I'm working with an dsp audio application and I desperately need an algorithm for tone control. The filter is described in my spec as a Baxendall filter with the axial point of 1kHz and the characteristics is a maximum of +-6dB per octave. So far I have implemented low and high pass second order Butterworh IIR filters with coefficients calculated with Matlab and they work f...


SDR IQ samples of FM stereo to stereo audio help?

Started by Hannu Vuolasaho in comp.dsp10 years ago 9 replies

Hello everyone! I have been reading C algorithm for real-time DSP book for a while and the signal processing stuff feels so unrelated to real...

Hello everyone! I have been reading C algorithm for real-time DSP book for a while and the signal processing stuff feels so unrelated to real world problems. I invested DVB-T dongle and started playing with it. With all applications available the understanding how the system works is still little bit mystified to me. I even played with the gnuradio and got it to output audio. Howeve...


Regarding WMA10

Started by raja12345 in comp.dsp16 years ago

Hi All, I am a newcomer in the digital audio area & previously I worked in Video Decoders. I have embarked on my first project of...

Hi All, I am a newcomer in the digital audio area & previously I worked in Video Decoders. I have embarked on my first project of WMA10(Windows media audio 10). I had the reference codes & specifications for WMA9 & WMA10. Basically I need to find the difference between these two versions & then start the coding of difference on processor which I had the assembly language code for WMA9. I ...


Controlling satellite phone from Windows PC and routing audio between PC and sat phone?

Started by Mauritz Jameson in comp.dsp7 years ago 1 reply

Is there a way of connecting a Windows PC to a satellite phone so that the following can be done from software running on the PC: 1. Remotely...

Is there a way of connecting a Windows PC to a satellite phone so that the following can be done from software running on the PC: 1. Remotely control the satellite phone from software on the PC (i.e. answer call, hang up call, adjust speaker volume, adjust mic sensitivity etc) 2. Receive speaker audio stream as a digital 16-bit signed PCM mono audio stream. 3. Send 16-bit signed PCM mono audi...


Finding the main frequency of an input audio signal

Started by Anonymous in comp.dsp15 years ago 21 replies

What is the best/easiest method to determine the main frequency of the input audio signal? I am confused by some options (FFT, wavelets) and I...

What is the best/easiest method to determine the main frequency of the input audio signal? I am confused by some options (FFT, wavelets) and I am just looking for a simple one which doesn't have to be too much theory, but simply works. I still would like to learn something, not just reuse somebody's solution in part or in whole and not understand anything, but I wouldn't like to spend more th...


between BF561 and DVDP6437

Started by bharat pathak in comp.dsp15 years ago 5 replies

Hello All, I want to decide between BF561 from analog and DVDP6437 from TI for audio/video applications. My requirements are as...

Hello All, I want to decide between BF561 from analog and DVDP6437 from TI for audio/video applications. My requirements are as follows 1. do little bit of algorithmic work related to audio/video processing. eg. for video it could be video noise reduction, image scaling, deinterlacing, dithering, color enhancements contrast/brightness adjustements, mpeg NR and ...


audio loop with c6713

Started by laurasscurtu in comp.dsp8 years ago

Hello, I am trying to do a simple audio loop (playpack loop) using only the McBSP with TI C6713. I must mention, I am new to the DSP world and...

Hello, I am trying to do a simple audio loop (playpack loop) using only the McBSP with TI C6713. I must mention, I am new to the DSP world and I would very much appreciate any help. I am working with DSP/BIOS and with CSL functions. So this is what I have done so far: I have programed 2 hardware interrupt functions, one for receive and one for transmit. I have connected my laptop to t...


signal bandwidth for cellular networks

Started by Siddharth Mathur in comp.dsp19 years ago 1 reply

Hi When a speech or audio signal is sent through a GSM or CDMA cellular phone, what is the maximum audio frequency that is encoded by the...

Hi When a speech or audio signal is sent through a GSM or CDMA cellular phone, what is the maximum audio frequency that is encoded by the codec? For example, 4kHz is considered as the cutoff for a plain old telephone system. GSM uses a CELP based encoder (?) ,so the number may be related to the standard. Any approximate figures would be appreciated. TIA, Siddharth


Naive question? Does Log ( Log p1/p2) mean anything

Started by Richard Owlett in comp.dsp19 years ago 1 reply

I started playing with some human perceptible audio questions. If I understand correctly, humans perceive audio levels on a log power...

I started playing with some human perceptible audio questions. If I understand correctly, humans perceive audio levels on a log power scale. I'm starting to look a spectrum analysis problem ( intentionally vague as I'm not sure of what I want to look at ;) I wish to plot "data". Linear in intensity has obvious problems. Log (intensity squared) has much history. BUT I wish to lo...


dsPic adc for audio signals

Started by Anonymous in comp.dsp17 years ago 5 replies

Hi. With a dsPic (ie dsPic30F4013) I need to process two voice band channels (for telephony, fs=8kHz). Can there be problems to use the...

Hi. With a dsPic (ie dsPic30F4013) I need to process two voice band channels (for telephony, fs=8kHz). Can there be problems to use the embedded adc, oversampling 8 times and then filtering? Do you think this may waste more than one half of dsp bandwidth? Why is not usual to use the embedded adc for audio signals? Thank you. pp


Decibel tutorial

Started by Anonymous in comp.dsp14 years ago

I have created a tutorial to explain decibels. It is located here: http://www.fourier-series.com/rf-concepts/decibels.html It consists of a...

I have created a tutorial to explain decibels. It is located here: http://www.fourier-series.com/rf-concepts/decibels.html It consists of a flash program that allows user interaction and also contains a fair amount of embedded audio explanations. The flash tutorial is about 7M , most of which is the embedded audio.


Sharp DSP notch filter

Started by Eeyore in comp.dsp14 years ago 83 replies

I am proposing to engage on a project regarding mains voltage 'purity' (and absence of ) with regard to audible clicks and pops in...

I am proposing to engage on a project regarding mains voltage 'purity' (and absence of ) with regard to audible clicks and pops in high-end professional and hi-fi audio equipment. Typical EMC filters operate in the RF band and are threfore no use to filter audio 'in band' noise that can travel through transformer interwinding capacitance etc. I have found some of the TI INA series that w...


measuring the quality of mixer

Started by srikk in comp.dsp16 years ago 1 reply

Hi, I have implemented the Audio Mixer, which will mix three Audio Signal. I have done implementation using MATLAB. I am getting the Mixed...

Hi, I have implemented the Audio Mixer, which will mix three Audio Signal. I have done implementation using MATLAB. I am getting the Mixed Signal. I performed listening test on it. It sounds good. I want some quality criteria to be defined to validate Mixed Output signal. Your Input will help me. Thanks in Advance, Regards, Sri Kant


FFTW3 fftw_plan_dft_r2c_1d and its inverse question

Started by dinsoft in comp.dsp17 years ago 1 reply

Hello, I am running FreeBSD. I installed FFTW3 from the ports. I am not a native english speaker so I apolozige for the mistakes. The goal...

Hello, I am running FreeBSD. I installed FFTW3 from the ports. I am not a native english speaker so I apolozige for the mistakes. The goal of my program is to perform the FFT in order to extract all the required informations from the audio signal (frequencies, amplitudes, phases) , same them in a file , and then rebuild the original audio signal from the file. A) I try the FFTW c2r to go...


Choosing a DSP for audio processing

Started by Jackson4 in comp.dsp15 years ago 12 replies

Hello all, My group is designing a small, inexpensive audio device (basically a PWM amp with variable/custom EQ processing), and we're looking...

Hello all, My group is designing a small, inexpensive audio device (basically a PWM amp with variable/custom EQ processing), and we're looking to select a DSP. I've perused datasheets from the majors, and could use some general advice. Our requirements - Inexpensive ($5-10 ballpark in quantity) 40 MIPS or so should be plenty if there are single-cycle MACs On-board program flash with co...


MP3/Ogg-Vorbis and SHARC

Started by Ian McBride in comp.dsp19 years ago 1 reply

I'm looking for experience with commercial/GPL MP3 or Ogg-Vorbis source libraries and SHARC architecture. I need to decode and play multiple...

I'm looking for experience with commercial/GPL MP3 or Ogg-Vorbis source libraries and SHARC architecture. I need to decode and play multiple audio streams for an industrial audio server. I've googled a lot and I see a few options, but I can't find any type of comparison or review data. Thanks a bunch for any input.


transferring audio data between SHARC bus and PCI bus on Audio PMC board via SHARCFIN ASIC

Started by salrome in comp.dsp17 years ago

hello i want to ask how can i transfer data on Audio PMC board between SHARC bus and PCI bus via SHARCFIN ASIC. thanks salrome

hello i want to ask how can i transfer data on Audio PMC board between SHARC bus and PCI bus via SHARCFIN ASIC. thanks salrome


How to decode SMPTE timecode from LTC audio track?

Started by stephenbuck in comp.dsp18 years ago 1 reply

Hi, I'm wondering if anyone has any code (C?) or suggestions for decoding SMPTE timecode from an LTC audio track? Thanks, Steve

Hi, I'm wondering if anyone has any code (C?) or suggestions for decoding SMPTE timecode from an LTC audio track? Thanks, Steve


Is a signal containing random numbers White Noise?

Started by Chris Barrett in comp.dsp16 years ago 28 replies

Let's say my audio is represented by a series of numbers and each number has a random value. Is my audio white noise? I think it is, but I'm...

Let's say my audio is represented by a series of numbers and each number has a random value. Is my audio white noise? I think it is, but I'm having trouble proving it to my self.


microphone array

Started by senece in comp.dsp15 years ago 10 replies

hello sir, I am doing project with microphone array. I want to feed multiple audio inputs to PC simultaneously. Is it possible to do that ?...

hello sir, I am doing project with microphone array. I want to feed multiple audio inputs to PC simultaneously. Is it possible to do that ? If it is possible how to do that ? And also I want to process these audio inputs in MATLAB. How can I get these into MATLAB ?