Audio pops in live audio streaming system.

Started by in comp.dsp11 years ago 10 replies

In a live audio streaming system, audio pops are caused by network packet delay. How to smooth the sharp decay, and what stuff should be filled...

In a live audio streaming system, audio pops are caused by network packet delay. How to smooth the sharp decay, and what stuff should be filled in the empty buffer?


Audio CD Error Correction

Started by Tim Wescott in comp.dsp6 years ago 20 replies

I vaguely remember reading a statement about audio CDs, to the effect that the error correction scheme is designed with some sort of magic so...

I vaguely remember reading a statement about audio CDs, to the effect that the error correction scheme is designed with some sort of magic so that it suffers a certain amount of soft degradation before the audio goes out entirely. This was a long, long time ago, and either I or the article I was reading could have it wrong. Is anyone up on CD audio recording formats? Is the above st...


Mathlab as experimental audio DSP workstation?

Started by Anonymous in comp.dsp11 years ago 11 replies

Could Mathlab make sense using it as a evaluation platform for audio DSP? This means: Streaming high quality audio using a firewire audio...

Could Mathlab make sense using it as a evaluation platform for audio DSP? This means: Streaming high quality audio using a firewire audio interface for example through a experimental adaptive filter coded in mathlab. And doing measurements and visualizing results in vaious ways (and fast). And doing simple GUI's for controlling that conveniently.


Undo frequency change by audio device before echo cancellation?

Started by roschler in comp.dsp9 years ago 3 replies

I am squarely in the camp of knowing barely enough DSP to be dangerous. I have successfully coded up some FIR filters to assist in sample rate...

I am squarely in the camp of knowing barely enough DSP to be dangerous. I have successfully coded up some FIR filters to assist in sample rate conversions between audio files but that's about it. My ambition could easily exceed my capacity here. I want to do echo cancellation of audio coming into a microphone that is positioned right next to an output speaker on an audio device I own. W...


Contemporary Audio DSP

Started by MarkAren in comp.dsp13 years ago 1 reply

Hi All, I am just about to start a reasonably simple DSP project processing audio (mainly filtering), I need 24 bit data, on board 8kw FLASH...

Hi All, I am just about to start a reasonably simple DSP project processing audio (mainly filtering), I need 24 bit data, on board 8kw FLASH and 4kw RAM, > 30MIPS and low cost development environment. I need a small parts count so I can interface an AES/EBU (S/PDIF) audio RX to a DAC -- I2S is the obvious interface. Are there any newer DSPs that support this audio interface directly as


USB Audio Dongles (looking for FD, Full bidirectional Stereo)

Started by Rich Osman in comp.dsp15 years ago 1 reply

I'm working on a hobby project that processes several communcations grade audio channels and I'm looking for a *cheap* way to get eight full...

I'm working on a hobby project that processes several communcations grade audio channels and I'm looking for a *cheap* way to get eight full duplex audio paths. The on-board interface today is usually full deplx stereo, so that's two. I've been looking at cheap USB audio interfaces and they all seem to have mic rather than line inputs, so a single channel input. Does anyone know of o...


Audio Processing - why Subband? why Octave banks? and why Uniform banks?

Started by Yan.L in comp.dsp14 years ago 1 reply

Hi, all I believe this is a very basic and old question. I know there are a number of advantages of subband processing. Here I'd like to ask...

Hi, all I believe this is a very basic and old question. I know there are a number of advantages of subband processing. Here I'd like to ask for your opinions of why subband is so widely used in audio signal processing applications, not only in audio compression, but on many other audio related topics, e.g BSS, surround effects... Researchers also talked about octave-band filter bank...


embedding messages (ID numbers) in audio

Started by Natalie in comp.dsp10 years ago 11 replies

Hi everyone, I want to embed ID numbers in audio so that I can recover them on a separate device. Some of my constraints: -The audio will be...

Hi everyone, I want to embed ID numbers in audio so that I can recover them on a separate device. Some of my constraints: -The audio will be transmitted from a standard TV/DVD set-up (so DVD encoding, TV speakers). -The audio will be received and processed on a cell phone (so cell phone microphone, processing capabilities (but can be a high-end cell phone)) -There will be significant back...


Game Audio Conference in Austin

Started by Anonymous in comp.dsp13 years ago 4 replies

The Game Audio Conference is coming up September 6-7-8, 2006 in Austin, Texas. The GAC is designed to offer advanced game audio sessions...

The Game Audio Conference is coming up September 6-7-8, 2006 in Austin, Texas. The GAC is designed to offer advanced game audio sessions aimed at working industry professionals and is geared toward illuminating and proposing solutions to some of game audio's toughest problems. The conference includes a keynote address from EA's Charles Keenen, as well as, talks from the following speakers:...


Looping ~50ms portion of an audio

Started by p_robi86 in comp.dsp3 years ago 5 replies

Hello, I am writing an application in which I need to be able to move trough an audio file based on where the user places the cursor, and if...

Hello, I am writing an application in which I need to be able to move trough an audio file based on where the user places the cursor, and if the user holds the cursor on a given position, I need to "freeze" the sound. In order to achieve this, I am looping 50ms of the audio from the position of the cursor. However this generates some noise in the audio, because the beginning and the end of...


Soundbite Audio Performance

Started by tomcee in comp.dsp7 years ago 2 replies

Has anyone measured the audio performance of the Freescale Soundbite audio board using simple pass through code? I am interested primarily in...

Has anyone measured the audio performance of the Freescale Soundbite audio board using simple pass through code? I am interested primarily in THD, freq response, etc....at 16/24 bits, 44.1/etc fs. Thank you , TomC


MP3 audio stream

Started by yoshita nanda in comp.dsp14 years ago 1 reply

dear all, I am a computer enggr. doing masters. The final Project involves lot of DSP. The project is on audio classification. The initial step...

dear all, I am a computer enggr. doing masters. The final Project involves lot of DSP. The project is on audio classification. The initial step is to obtain mp3 audio stream. Is there any freely available library to obtain the audio data as floating point values. This would help to reduce work load. I have tried using the LibMpeg3 but to i am stuck trying to convert to library. It would be kind...


How much processing can I do and still stream audio data

Started by Zach R. in comp.dsp16 years ago 6 replies

I am a new DSPer who is using a blackfin 535 ez-kit. I am trying to implement a program which requires streaming audio and am...

I am a new DSPer who is using a blackfin 535 ez-kit. I am trying to implement a program which requires streaming audio and am troubleshooting. How much processing can this chip support before it will no longer be able to stream the audio? Thanks, Zach


compression speed

Started by hyperbob in comp.dsp15 years ago 5 replies

Hello, I am wondering how much computation it takes to compress audio. This information is a little hard to come by. The general situation...

Hello, I am wondering how much computation it takes to compress audio. This information is a little hard to come by. The general situation is this: I am working on a program that will record either from the sound cards of several computers or from some card that provides several raw audio channels to be recorded from. I don't know how to estimate if I can compress all the audio in real ...


Identify this audio filter?

Started by JP in comp.dsp15 years ago 6 replies

I'm a (very) novice C++ programmer who occasionally likes to play around with audio DSP concepts. Could someone please help me find the name...

I'm a (very) novice C++ programmer who occasionally likes to play around with audio DSP concepts. Could someone please help me find the name and/or an implementation of the filtering algorithm I have in mind? What I'm looking for is the equivalent of a graphics programs' "sharpen" or "unsharp mask" but performed on a time vs. frequency spectrogram of a digitally sampled audio signal. (I'm n...


About asynchronous sample rate conversion

Started by nagual.hsu in comp.dsp13 years ago 13 replies

My audio capture and playback devices reference different crystal clocks. I want to play the captured audio as "realtime" as...

My audio capture and playback devices reference different crystal clocks. I want to play the captured audio as "realtime" as possible. Therefore I use Secret Rabbit Code to do dynamic sample rate conversion as some sort of clock skew compensation. Question 1: If the sample rate of the captured audio and the playback audio are all 8kHZ, do I have to use any low pass filter in case of...


Audio processing of more than one mic input

Started by Mauritz Jameson in comp.dsp6 years ago 10 replies

What is a typical setup for processing multiple microphone signals? In the system I'm currently working with, the audio driver calls a callback...

What is a typical setup for processing multiple microphone signals? In the system I'm currently working with, the audio driver calls a callback function which pushes the microphone audio to a queue and it wakes up a task to process the audio in the queue. However, with several microphone inputs and the current setup, I would have to choose which mic is the master; and the callback f


Looking for help with AGC

Started by Richard M. Hartman in comp.dsp15 years ago 4 replies

I am a software developer, not really an audio specialist. I need to implement an AGC for 8Khz 16-bit digital audio (under Windows). I tried...

I am a software developer, not really an audio specialist. I need to implement an AGC for 8Khz 16-bit digital audio (under Windows). I tried searching for AGC algorithms using Google, but the few mentions I saw were discussed using audio techie language and I do not understand how to convert this into practical code. For ex. the most promising one seems to be this posting: http://groups....


Recommend Book on Audio/Speech

Started by Davy in comp.dsp13 years ago

Hi all, I am interested in Audio/Speech region. Are Audio/Speech use different algorithm? And can you recommend some book on Audio/Speech...

Hi all, I am interested in Audio/Speech region. Are Audio/Speech use different algorithm? And can you recommend some book on Audio/Speech (I prefer book with Matlab code). Thanks! Best regards, Davy


Please Recommend Book on Audio/Speech

Started by Davy in comp.dsp13 years ago 5 replies

Hi all, I am interested in Audio/Speech region. Are Audio/Speech use different algorithm? And can you recommend some book on Audio/Speech...

Hi all, I am interested in Audio/Speech region. Are Audio/Speech use different algorithm? And can you recommend some book on Audio/Speech (I prefer book with Matlab code). Thanks! Best regards, Davy