Dear Newsgroups, I gotta one of a kind interview on mini dv. When played back it both looks and sounds terrific!! Problem occurs when editing...

Dear Newsgroups, I gotta one of a kind interview on mini dv. When played back it both looks and sounds terrific!! Problem occurs when editing interview onto dvd. Audio becomes very muddy/unusable I could really use any suggestions on how I may cure this please?? Thanks in advance!! pittwindmill@see-below-no-spam hotmail.com

Hello, I would like to ask more experienced persons for help with the problem I'm working at... I have two audio signals: one outgoing voip...

Hello, I would like to ask more experienced persons for help with the problem I'm working at... I have two audio signals: one outgoing voip stream, and one incoming (from the other side) I would like to compare the streams, especially with regards to: - transport problems (RTT, jitter, packet loss) - acoustic distortion (transcoding between different codecs) First I need to find what...

Hi, new to this group...I am using Matlab to SIMULATE the de-mixing of two audio streams. What I did is as follows: 1. found two clean audio...

Hi, new to this group...I am using Matlab to SIMULATE the de-mixing of two audio streams. What I did is as follows: 1. found two clean audio sources on the internet S1, S2. (4 seconds at 11kHz, 16bit) 2. found four room/hall impulse responses H11,H12,H21,H22. (3 seconds at 11kHz, 16 bit) 3. Intention is to simulate pickup by two separate microphones the mixed streams X1, X2. 4. Hence Con...

I am trying to figure out interpolation (I have given up on decimation for now). My scenario is that I have 8 ksamples per second audio (from...

I am trying to figure out interpolation (I have given up on decimation for now). My scenario is that I have 8 ksamples per second audio (from a phone line), and I want to interpolate it to 96 ksamples per second. I know my audio is band limited within 0-4kHz, but other than that it could be anything -- human voice, modem tones, etc. I understand the most basic concepts. I could use lin...

Hi all Does anyone know of a DSP EVM (or product) with the following features: - 18-bit or greater DSP operations (fixed or floating...

Hi all Does anyone know of a DSP EVM (or product) with the following features: - 18-bit or greater DSP operations (fixed or floating point) - stereo audio ADC and DAC (> 80dB SFDR) - flash RAM - dirt cheap! Thanks! Ryan

Hi, We are going to develop Audio application using TMS320C6416. For input to AIC23 Codec, we use microphone. But AIC23 Codec is unable to detect...

Hi, We are going to develop Audio application using TMS320C6416. For input to AIC23 Codec, we use microphone. But AIC23 Codec is unable to detect the signal coming from MIC. To give microphone input, I had done all the required setting i.e INSEL bit of Analog audio path control register of AIC23 Codec is set to 1. Is any other setting is required ? Please suggest. Regards Vood00

Why do most software audio volume controls have linear tapers? I would have guesses that simulated log-taper controls would be...

Why do most software audio volume controls have linear tapers? I would have guesses that simulated log-taper controls would be standard. Jerry -- Engineering is the art of making what you want from things you can get.

Can anyone give a hit where to get any source code library in C or C++ to use audio fingerprinting technology?

Can anyone give a hit where to get any source code library in C or C++ to use audio fingerprinting technology?

I would like to use an audio codec based on WAVE PCM. It should be a little different though. The bit-resolution should be set to...

I would like to use an audio codec based on WAVE PCM. It should be a little different though. The bit-resolution should be set to equal 1/(sampling rate X # of channels). The bit-rate should be set to equal 1 bit per second. I would like to use this codec to transport audio files though the internet via email. I am looking for frequency response. In digital audio the sampling rate must be ...

Hi all, Many thanks for reading my post. I would like to design a custom DSP board for some quite strong audio processing. I would like to...

Hi all, Many thanks for reading my post. I would like to design a custom DSP board for some quite strong audio processing. I would like to put the complete project under GNU licence, so I would like to have a maximum of free (or not too expensive) tools. I think about using a TI C6722, it seems to fit my needs. I also found a GCC and BINUTILS port for C6000. My first question is : i...

Hello people, I'm kinda new to all this, but I think I'm in the right place for the issue I'm trying to solve: For an assignment I have to...

Hello people, I'm kinda new to all this, but I think I'm in the right place for the issue I'm trying to solve: For an assignment I have to do in my study, I have to implement a simulation of a large sensor network, consisting of nodes equiped with microphones. The idea is to let the nodes record the audio from their surroundings, probably transform (simplify) it in some way and commu...

Hi admin, The code below to process the audio signal on red5.Audio signal on red5 is the type of data bytes.I want to change the data type to a...

Hi admin, The code below to process the audio signal on red5.Audio signal on red5 is the type of data bytes.I want to change the data type to a sample (float). public void start(IBroadcastStream broadcastStream, IScope scope) throws StreamException { if (log.isDebugEnabled()) log.debug("startTranscodingStream({},{})", broadcastStream.getPublishedName(), scope.getName()); recordStream(broa...

Hi all, I really hope this is the right place to post my question because I really need some guidance from some experts. I was told to find...

Hi all, I really hope this is the right place to post my question because I really need some guidance from some experts. I was told to find out a library, preferably LGPL, but in a worst case scenario also a commercial one would be ok, that would do resampling. Some scenarios where I will have to use this would be: 1) I get a file with some audio with the following attributes: Frequ...

Hello, I know this isn't very DSP related ;) but I think you can help me here. I've got two different approaches for programming a balance...

Hello, I know this isn't very DSP related ;) but I think you can help me here. I've got two different approaches for programming a balance control, and I'm not sure which one to take. I tried examining other audio programs but there were also differences. Let's say I load a WAV file and treat the left and right channel as two entirely DIFFERENT audio tracks, but when played back, it shoul...

Hi Folks, It's been a while since I looked at this NG but now I think I can use some fellow thinkers :) I've been asked to think about...

Hi Folks, It's been a while since I looked at this NG but now I think I can use some fellow thinkers :) I've been asked to think about comparing 2 audio signals to see if the audio is roughly (but with high probability) the same. One input is a broadcasted signal (over air or internet), the other one is its reference (the original signal). So the quality of both signals can differ, a...

Hi everybody, I got stuck with a simple DSP problem, and I'm not able to find a any resource that might help me. I was reading a well known...

Hi everybody, I got stuck with a simple DSP problem, and I'm not able to find a any resource that might help me. I was reading a well known article (Content Fingerprinting Using Wavelets, M.Covell, S.Baluja), in which there is a small description of audio processing: "We start our processing by converting the audio input (sampled at 5512 Hz) into a spectrogram. We create our spectrograms using ...

OMG. So many acronyms. DIO, SIO, IOM, TSK, SWI, PIP, PIO Would anybody be able to tell me what these acronyms might refer to? I'm trying...

OMG. So many acronyms. DIO, SIO, IOM, TSK, SWI, PIP, PIO Would anybody be able to tell me what these acronyms might refer to? I'm trying to get some example code running on a DSK TI6713 from Spectrum Digital. In particular, I'm trying to find a audio talk through project. There are three audio related example projects available with the board: tsk_audio, swi_audio, and pip_audio. --...

Can anyone recommend a dedicated audio processing chip with really low power requirements? Im looking at a performance of about 50-60 MIPS. The...

Can anyone recommend a dedicated audio processing chip with really low power requirements? Im looking at a performance of about 50-60 MIPS. The application is SISO (single-in, single-out). Any suggestions from the Pros? Thanks, --Shafik

Hello! I used dbPowerAmp converter to convert Real Audio .rm files to mp3, but now it asks to buy a license for mp3. How to convert Real Audio...

Hello! I used dbPowerAmp converter to convert Real Audio .rm files to mp3, but now it asks to buy a license for mp3. How to convert Real Audio .rm files to mp3 for free? Best regards, Dima +7 9163876746

Hi,guys! I have a question about digital audio system. As we know, a lot of digital audio system has a Pre-Emphasis circuit before the A/D...

Hi,guys! I have a question about digital audio system. As we know, a lot of digital audio system has a Pre-Emphasis circuit before the A/D circuit. The audio signal must pass a treble-boost filter before they go to the A/D. And my question is, what's the effect of this Pre-Emphasis circuit? To reduce the noise? To enhance the A/D's performance? Or other effect? Thanks!