AGC in audio conference

Started by hech1 in comp.dsp16 years ago 3 replies

Hello, I am working on a VoIP audio conference. I want that all mixed sources have the same volume before being mixing so that when the speaker...

Hello, I am working on a VoIP audio conference. I want that all mixed sources have the same volume before being mixing so that when the speaker changes, participants will keep hearing the same level of volume. Do I need a AGC algorithm? All the AGC algorithms that I found on the Internet are about using AGC for a single source, could anybody explain to me how to use AGC for a conference. ...


Audio Engineer Interview Questions

Started by londonman13 in comp.dsp14 years ago 13 replies

All, I have an audio engineer interview coming up this Friday. Can anyone recommend any good questions they have come across? Hope this makes...

All, I have an audio engineer interview coming up this Friday. Can anyone recommend any good questions they have come across? Hope this makes for a good reference thread for future interviewees as well. Here are some questions I have come across from previous interviews (I'm a fresh out grad, so these might be easy for all you experienced folks out there): - What is the bit-depth of CD quali...


Extracting digital FM data from unwanted audio

Started by W Marsh in comp.dsp18 years ago 10 replies

Hi. I want to extract digital data (where 1022Hz represents a 0, and 2044Hz represents a 1) from an audio signal where it is mixed with a song....

Hi. I want to extract digital data (where 1022Hz represents a 0, and 2044Hz represents a 1) from an audio signal where it is mixed with a song. An example of the signal I want after separating from the noise is http://funge.8bit.co.uk/cleandata.gif My knowledge of DSP is purely investigative, i.e. I haven't really studied the subject, so I've no idea how to best approach this. I've tried s...


Best DSP board / chip for audio effects/processing

Started by in comp.dsp17 years ago 2 replies

Hi! I'm going to buy myself a DSP development kit in a few months time, for two reasons: 1. I want to learn more about DSP and how DSP...

Hi! I'm going to buy myself a DSP development kit in a few months time, for two reasons: 1. I want to learn more about DSP and how DSP processors work. 2. I'm going to make myself an audio mixer (studio mixer), and I want to be able to add effects, etc to the input-signals. Things like delays, noice gates, compressors, etc. Currently, I've been looking a bit at the Dev....


C6711 DSK

Started by Hamid in comp.dsp19 years ago 3 replies

Hi all I have recently bought a DSK C6711 and for reading the Audio input port and writing to it, I need to have some library...

Hi all I have recently bought a DSK C6711 and for reading the Audio input port and writing to it, I need to have some library files(Board support library(BSL)), anybody can help me how to get these lib files, also if anybody has some example file to read the Audio input with McBSP, I appreciate if you send it for me. Regards, Hamid


introduction to digital sound processing

Started by in comp.dsp19 years ago 12 replies

Hi! Could you recommend some literature on DSP for a beginner? I'm interested in how low/highpass filters for digital wave audio, audio encoders...

Hi! Could you recommend some literature on DSP for a beginner? I'm interested in how low/highpass filters for digital wave audio, audio encoders (mp3) and other stuff like that works. My initial knowledge is not much - I understand how wav files work (I've made a few tiny programs in C that make wav louder, convert stereo into mono etc.) and I have a basic understanding of programming.


Removing White Noise

Started by Duane in comp.dsp17 years ago 13 replies

Hey everyone, I have an audio application, and I'm trying to do some pre/post processing to improve the sound quality. I put in an adaptive...

Hey everyone, I have an audio application, and I'm trying to do some pre/post processing to improve the sound quality. I put in an adaptive filter to try and remove structured noise, but I'd like to remove white/pink noise from the signal somehow.. I've been looking at wavelet techniques, or whatever else I can find. Can anyone recommend a good way to reduce white noise from audio/ A...


How to perform filtering on a time window for an audio signal?

Started by dingke1980 in comp.dsp16 years ago 10 replies

For example, for a given 10s length audio signal. I want to perform LP filter from 3s to 5s time window. At the same time, the total sample number...

For example, for a given 10s length audio signal. I want to perform LP filter from 3s to 5s time window. At the same time, the total sample number should remain the same with the un-processed one. Who can tell me how to do it? Thx very much. Kevin _____________________________________ Do you know a company who employs DSP engineers? Is it already listed at http://dsprelated.com/empl...


Looking for graphical programming tool for doing audio dsp work.

Started by khung in comp.dsp10 years ago 7 replies

Hi everyone, I am looking for graphical programming tool for doing audio DSP work. I can think of these companies/tools: (1) Simulink (from...

Hi everyone, I am looking for graphical programming tool for doing audio DSP work. I can think of these companies/tools: (1) Simulink (from Mathworks) (2) LabView (from National Instruments) (3) Ptolemy Classic (from UC Berkeley EECS) (4) HP Vue Please let me know if you can think of any product that I did not mention. Thanks Kevin


Spectrogram Plot to Bitmap Output

Started by BobM in comp.dsp17 years ago 2 replies

Hi Everyone, I'm looking to make a small C/C++ command-line utility which takes a binary audio file as input and generates a spectrogram of a...

Hi Everyone, I'm looking to make a small C/C++ command-line utility which takes a binary audio file as input and generates a spectrogram of a given time range of the file. First off, I am already comfortable with reading/writing binary audio files, as well as the "back end" requirements for generating spectrogram information (FFT, windowing, etc.). What I am not familiar with is gener...


Java Audio question - PLEASE HELP

Started by cppt...@yahoo.com in comp.dsp15 years ago 35 replies

Could some DSP or Java guru please help me ? I have a simple Java application that records audio signals and does simple analysis on...

Could some DSP or Java guru please help me ? I have a simple Java application that records audio signals and does simple analysis on the collected data. I am using PCM encoding, with sampling frequency 16000 Hz, 16 bits resolution, single or mono channel, little endian and signed. The data is read into a ByteArrayOutputStream and then as soon as I stop recording, the stream contents are wr...


Which Dev Board?

Started by Mauritz Jameson in comp.dsp11 years ago 4 replies

Which development board would you recommend for a real-time audio application which involves 8 FIR filters? Each FIR filter has 3000 taps and it...

Which development board would you recommend for a real-time audio application which involves 8 FIR filters? Each FIR filter has 3000 taps and it must be able to process audio blocks at 96kHz. As far as I understand that is equivalent of approx. 2304 MMAC/sec if the core CPU is to perform the filtering. I'm not sure I will be able to find a development board that can handle this? Or am I wro...


Audio codec for G.729/G.723.1/G.726/G.728

Started by Josh in comp.dsp17 years ago 2 replies

I found a website(http://www.imtelephone.com) that provide audio codec(G.729, G.723.1,G.726,G.728). It has a very simple interface and you can...

I found a website(http://www.imtelephone.com) that provide audio codec(G.729, G.723.1,G.726,G.728). It has a very simple interface and you can easily intergrate into your software. .


Seeking a simple audio energy detection algorithm

Started by Ury in comp.dsp13 years ago 5 replies

Hi, I receive a stream of linear 16 bit audio and need to identify the possibly "interesting" parts vs. plain silence or noise. Just...

Hi, I receive a stream of linear 16 bit audio and need to identify the possibly "interesting" parts vs. plain silence or noise. Just computing the average energy and comparing to a threshold is not sifficient since there may be a background noise. Is there a lightweight adaptive energy detection algorithm? Thanks, Ury.


pure software RDS?

Started by kcchesnut in comp.dsp17 years ago 1 reply

i'm a software programmer that is new to RDS/RBDS for broadcasting data with FM. wondering if it would be possible to write a pure RDS...

i'm a software programmer that is new to RDS/RBDS for broadcasting data with FM. wondering if it would be possible to write a pure RDS decoder. e.g. to have a USB FM tuner hooked up to my computer and getting the audio stream into my own application. then having the application do audio processing to retrieve the RDS commands and decode them. or will i need special hardware? Thanks, ca...


Regarding Real Time Problems of AEC

Started by Aparna Ram in comp.dsp16 years ago

Dear All, I have implemented AEC with and with out subbanding using NLMS algorithm and that code is working fine for all recorded...

Dear All, I have implemented AEC with and with out subbanding using NLMS algorithm and that code is working fine for all recorded audio files. But for real time audio files this algorithm is not effecting. If any one know the reason behind this then please give me your valuable suggestions. Thanks in advance. Regards, Aparna Ram.K.


[OT] patent search process for a new product

Started by SYL in comp.dsp15 years ago 28 replies

Hi, All Sorry for the off topic. I don't really know where I should ask. We are building a new audio product. It uses a number of...

Hi, All Sorry for the off topic. I don't really know where I should ask. We are building a new audio product. It uses a number of audio processing algorithms. Some of these algorithms were derived from recent publications. We wanna make sure we don't infringe any patent. What should I do? Do a keyword search on USPTO? What if I miss some "relevant" patents? We are not interested in fi...


Finding certain frequencies in an audio signal

Started by Thomas Schachtner in comp.dsp17 years ago 10 replies

Hi there, I would like to write an app which can detect certain frequencies in an audio signal. There are 11 different frequencies between...

Hi there, I would like to write an app which can detect certain frequencies in an audio signal. There are 11 different frequencies between 1060 and 2600 Hz (freq. no. 1 to 11). 5 of these freqs are sent as some kind of trigger signal with each tone having 70 ms length. My program should detect these signals and the order they are transmitted (e. g. freq. 1,4,8,3,5). I heard sth. a...


Confusion about FFT resolution - PLEASE HELP

Started by cppt...@yahoo.com in comp.dsp15 years ago 1 reply

Could some DSP guru please clarify the following ? I have a simple audio application, in which at records time about 20000 bytes from...

Could some DSP guru please clarify the following ? I have a simple audio application, in which at records time about 20000 bytes from the received audio signal. I am using PCM encoding with sampling frequency 16000, 16 bits, mono channel, little endian and signed. I wish to do FFT with the collected data. Since I collect bytes, I iterate through the buffer, collecting two bytes at a time, and...


Syncing multiple related audio tracks

Started by in comp.dsp13 years ago 8 replies

Hi, I'm looking for a method to automatically synchronize various audio tracks, recorded at the same place, with different devices. This...

Hi, I'm looking for a method to automatically synchronize various audio tracks, recorded at the same place, with different devices. This is intended to work at post-processing time (not in realtime.) Basically, I'm taking two audios track: one recorded by a camcorder, with poor mic quality, and an extra one recorded at the same time with a dedicated sound recorder, recording the same thi...