Hello I'm new with DSPs (2 weeks) and got to make some easy effects like delay/reverb/flanger and some simple IIR/FIR implementation Now I...

Hello I'm new with DSPs (2 weeks) and got to make some easy effects like delay/reverb/flanger and some simple IIR/FIR implementation Now I was reading about making a compressor and my first thought was to use RMS, but I heard about audio envelope. The book I'm reading explains the formula, but doesn't tell much. Anyone know where I can find more sources? I don't understand how a low...

My aim is to compare two audio files. One is a recorded copy of the other. But they have differences, like some change in amplitude etc and...

My aim is to compare two audio files. One is a recorded copy of the other. But they have differences, like some change in amplitude etc and silence periods in the beginning and end. Is there some kind of a program I can use to do this. I would prefer some kind of C or C++ program which could do the same and tell me if the files are almost the same. That is have a high amount of correlation...

Can Anybody be kind enough to tell how to save the audio file played in matlab !! I tried with wavout but cannot save !! please tell me !!...

Can Anybody be kind enough to tell how to save the audio file played in matlab !! I tried with wavout but cannot save !! please tell me !! thanks

Hi, How can one linearly change (i.e., with an approx. straight line in a magnitude plot) the spectral tilt (st) of an audio sample by...

Hi, How can one linearly change (i.e., with an approx. straight line in a magnitude plot) the spectral tilt (st) of an audio sample by smaller steps than -6dB/Oct? For example, I found that for an audio x the st is -1.6 dB/Oct, and I want it to have a st of -2.3 dB/Oct, how can I calculate the coeff. of a filter such as once applied to x yields the desired st? thanks for any hint.

Hello, all. It's been many moons since I last posted here. I am doing a feasibility study for a client that entails locating a...

Hello, all. It's been many moons since I last posted here. I am doing a feasibility study for a client that entails locating a moderately powerful (~100 MMAC/s at better than 24x24 bits) DSP that consumes very little power. Furthermore, it would be even more lovely if the DSP lived on the same chip as an audio CODEC that did at least 16 bits at 44.1/48 kHz. I've scoured the web and I've also ...

Roman Rumian wrote: > Hi, > > does anybody know such a card ? > I need a minimum 6 channel 16-bit/48ks/s one. > > Best regards > > ...

Roman Rumian wrote: > Hi, > > does anybody know such a card ? > I need a minimum 6 channel 16-bit/48ks/s one. > > Best regards > > Roman Rumian > We've done an 8x8 audio daughter card for the 55x and 6711x DSKs. We can update the drivers to work with the 6713DSK. Contact us if you'd like more information. -Shawn Steenhagen www.appliedsignalprocessing.com

i've only done really basic work with audio ... so i'm not even sure what to call this? what i want to do is be able to take audio input from...

i've only done really basic work with audio ... so i'm not even sure what to call this? what i want to do is be able to take audio input from a microphone and filter out background noise. e.g. if i'm in my car and have music playing but also want to do speech recognition with a mic. the mic would get my spoken command along with the music being played. then i would have a separate feed of ...

Hi I am not totally new to DSP, I did do a module in it at University as part of my Masters, but that was around 15 years ago... I am...

Hi I am not totally new to DSP, I did do a module in it at University as part of my Masters, but that was around 15 years ago... I am looking at implementing audio effects, such as reverberation and amplifier/speaker modelling, by the convolution of an audio signal with an impulse response recorded from a room or amplifier. Specifically I want to implement a bass guitar amp model to improve ...

New to forum, new to industry, but 30 years in audio. __________________________________________ The header is not very clear, I know... ...

New to forum, new to industry, but 30 years in audio. __________________________________________ The header is not very clear, I know... but I have a question that I'd love to have addressed by anyone with experience in SRC for telephone call center software using low-Khz voice files for automated prompting. Issue: (assumed on my part) Distortion within a compressed audio file ...

hi a short time ago, I posted I was doing computed modulation. I now have 2 nice sidelobes. The q is this: Where mathematically, and more...

hi a short time ago, I posted I was doing computed modulation. I now have 2 nice sidelobes. The q is this: Where mathematically, and more to the point, where should I account for the necessity in code of doing an DC offset to audio array samples in order to achieve the "Carrier" in AM modulation ? procedure: I simply add some close sine frequencies to get a 300-3000 hz waveform ( ...

I am recording electric guitar with several microphones. I want to ensure that each mic's audio is phase aligned with each other. So I was...

I am recording electric guitar with several microphones. I want to ensure that each mic's audio is phase aligned with each other. So I was thinking I could set one of the audio files as a reference file and then phase align everything to that reference signal. Anyone have any idea how on earth I would actually perform the phase alignment?? thanks, brian

I'm a musician and engineer and would like to start on DSP programming on audio on the Win platform. Can you recommend good books to start with....

I'm a musician and engineer and would like to start on DSP programming on audio on the Win platform. Can you recommend good books to start with. I have a grasp of audio file formats and some basics of the Fourier transforms. Despite that I would like to start studying the DSP from the scratch with the math described down to the detail as it's been a long [nope, longer than that;-)] time s...

Hi, I have a calculus tutorial that consists of 4 flash programs with embedded audio that goes over the basics of calculus. It is located...

Hi, I have a calculus tutorial that consists of 4 flash programs with embedded audio that goes over the basics of calculus. It is located here: http://www.fourier-series.com/Math/Calc/index.html I know this is pretty basic for this group, but I have posted all my other tutorials here, so what the heck. (Note - I have not completed the audio for the 4th tutorial program yet - soon...

Hi all, How to deal with audio files on CCS? To load data from audio file on to some buffer array,I have tried converting it into ASCII text...

Hi all, How to deal with audio files on CCS? To load data from audio file on to some buffer array,I have tried converting it into ASCII text file, but it makes the resulting file quite bulky(150MB) and its taking long time to add the header at the begining. any ideas? -aamer

Hi all, I've heard a lot of aliasing. What do they look like in image and audio/speech/music with aliasing? I did not see/hear any real...

Hi all, I've heard a lot of aliasing. What do they look like in image and audio/speech/music with aliasing? I did not see/hear any real aliasing stuff, so the concept of aliasing looks abstract to me, although I know how it occurs in terms of mathematics... I am also wondering about the anti-aliasing filter before downsampling... By filtering, you lose information, right? It is har...

Hi, I am new to DSP. Can anybody tell me how to determine the encoding of any .raw audio file? I mean I want to convert raw audio data to...

Hi, I am new to DSP. Can anybody tell me how to determine the encoding of any .raw audio file? I mean I want to convert raw audio data to PCMU data to be sent over RTP. Can you tell me how to do that? Thanks Sonika

I'm doing some audio programming to be used in a flight trainer. Previously, directsound was being used to perform all audio playback. ...

I'm doing some audio programming to be used in a flight trainer. Previously, directsound was being used to perform all audio playback. Now though, due to the need to do realtime capture and playback, we've switched to using ASIO because of the low latency associated with it. Anyway, Directsound had methods for doing things like volume, pan, and frequency control, but now with ASIO, this ha...

Sirs, Can anyone help me with a pointer to where I can purchase a good book on Advanced Audio Coding, AAC? -- Best regards, Anders...

Sirs, Can anyone help me with a pointer to where I can purchase a good book on Advanced Audio Coding, AAC? -- Best regards, Anders Buvarp anders@lsil.com

Hi there, is there any example code to build an Audio Limiter with the EZKIT LITE 2189M Evaluation Board ? Any suggestions ? Thanks in...

Hi there, is there any example code to build an Audio Limiter with the EZKIT LITE 2189M Evaluation Board ? Any suggestions ? Thanks in advance Roland

Hi Gentlemen, Please advise if you have thoughts on my questions. I'm developing a program which will be used as audio cross-over,...

Hi Gentlemen, Please advise if you have thoughts on my questions. I'm developing a program which will be used as audio cross-over, meaning that I need to filter the audio signal (HP, LP). I'm using SPUC library and FIR class defined there. FIR coeffs are calculated using Remez algorithm (remez_fir class) and Blackman window applied to them as well. Everything seems to be good - impulse respo...