2nd order decision directed PLL

Started by Momo Loulah in comp.dsp13 years ago 3 replies

Hi, I'm trying to implement a secod-order decision-directed phase-locked loop as part of acoustic algorithm. I know my loop constants but what...

Hi, I'm trying to implement a secod-order decision-directed phase-locked loop as part of acoustic algorithm. I know my loop constants but what I'm not sure about is how long do I have to integrate my phase error. It also occured to me that I might have to unwrap the phase also. Sorry if this sounds trivial - I just couldn't find any reference on this in the literature. Thanks, -Momo ...


Re: Opposite of Mu-law?

Started by Karl Uppiano in comp.dsp18 years ago

"Ben Bradley" wrote in message news:uu0ta055fd521oikgl9olo1mvg1tvr3e3e@4ax.com... > In rec.audio.tech,comp.dsp, Richard Owlett > ...

"Ben Bradley" wrote in message news:uu0ta055fd521oikgl9olo1mvg1tvr3e3e@4ax.com... > In rec.audio.tech,comp.dsp, Richard Owlett > wrote: > > > Ben Bradley wrote: > > [snip] > > > The music industry 'standards' for making a CD have changed in the > > > past decade or so. A song that sounds 'louder' is, at first listen, > > > deemed


Re: SERAF - Synaptic Energy Redistribution Audio Filter

Started by Stephen Norris in comp.dsp18 years ago

stephan.bernsee@web.de (Stephan M. Bernsee) wrote in message news: > > Sounds interesting, although I see no real connection to the >...

stephan.bernsee@web.de (Stephan M. Bernsee) wrote in message news: > > Sounds interesting, although I see no real connection to the > synthesizer from what I've read here. To be honest, I don't quite > understand how this SERAF is supposed to work anyway - do you have > more information? The bits I was able to dig out in this thread don'


I need Some help regarding speech to Text Conversion task

Started by Anonymous in comp.dsp17 years ago 2 replies

Hello all, I am implementing Speech to Text Conversion task using MATLAB. I implemented with LPC and Kohonnen's neural network...

Hello all, I am implementing Speech to Text Conversion task using MATLAB. I implemented with LPC and Kohonnen's neural network to do Speaker Identification task . Now i am able to idenify the speaker. With the same method i am able to identify the Vowel sounds also. Further i am not able to proceed to implement Consonent Conversion. I do not know about Phoneme also. ...


MELP / MELPe vocoder and IMBE vocoder

Started by CR in comp.dsp16 years ago 15 replies

Does anyone have any link or results of speech quality comparison between IMBE vocoder (or AMBE) and MELPe / MELP vocoder at rates 2400, 1200,...

Does anyone have any link or results of speech quality comparison between IMBE vocoder (or AMBE) and MELPe / MELP vocoder at rates 2400, 1200, 600 bps? I heard MELPe vocoder sounds much better, is that true? What about other aspects such as robustness to background noise & channel errors? Claudia R.


A question from Vaidyanathan's "Multirate Systems" book

Started by Rick Lyons in comp.dsp17 years ago 13 replies

Hello Earthlings, I recently ran across some material in Vaidyanathan's "Multirate Systems and Filter Banks" DSP book that discusses a...

Hello Earthlings, I recently ran across some material in Vaidyanathan's "Multirate Systems and Filter Banks" DSP book that discusses a way to improve the computational efficiency of polyphase filters used in non-integer decimation applications. Vaidyanathan's description (starting on page 128, and derived from a 1987 conference paper by C.-C Hsiao) sounds interesting except ...


Subject: Music & Acoustic Transcription (was: The Holy Grail of Transforms & Component Extraction)

Started by Alfred Einstead in comp.dsp9 years ago 2 replies

I will follow up on the note below with some You Tube demos, along with an outline of the analysis method used to recover the "Holy Grail", as...

I will follow up on the note below with some You Tube demos, along with an outline of the analysis method used to recover the "Holy Grail", as it were (i.e. clean separation of acoustic signals into their natural "chirp" components, in the same way that a person decomposes the sounds when hearing them). From 2012 Dec 15 http://groups.google.com/group/comp.dsp/msg/fb55797342ea084e > The hy


recognize material from sound !

Started by nino in comp.dsp13 years ago 8 replies

How ? correlation FFT ? thanks Is there a theory about ?

How ? correlation FFT ? thanks Is there a theory about ?


How to build a virtual acoustic studio hosted in a website?

Started by Piotr Mancini in comp.dsp3 years ago 1 reply

This is a long term question. We are in the process of building an Internet website that will allow the users to experiment, learn and ask...

This is a long term question. We are in the process of building an Internet website that will allow the users to experiment, learn and ask questions about distances, angles, trajectories, ballistics and sounds related to a very specific event. This is the website where we plan to host the simulations, data, etc. Free Open Source 3D Model of Dealey Plaza http://dealey-plaza.org/ That...


Power Chords

Started by Tim Wescott in comp.dsp7 years ago 45 replies

So, I'm trying to generate power chords in Scilab, and I'm just hearing ugly buzzing sounds. The closest I come to real power chords is if I...

So, I'm trying to generate power chords in Scilab, and I'm just hearing ugly buzzing sounds. The closest I come to real power chords is if I allow things to distort heavily. f = 500; // Hz t = (0:22050)/22050; y = 0.5 * (sin(2 * %pi * f * t) + sin(2 * %pi * 1.5 * f * t)); playsnd(y); Clues for the clueless? -- Tim Wescott Wescott Design Services http://www.wescottdesig...


downsampling -> FFT -> upsampling

Started by Fred T. Weiler in comp.dsp17 years ago 39 replies

Hi! I'm experimenting with FFT and inverse FFT, doing some filtering in real-time. It sounds good but it's a bit too CPU heavy so I decided to...

Hi! I'm experimenting with FFT and inverse FFT, doing some filtering in real-time. It sounds good but it's a bit too CPU heavy so I decided to downsample the signal and apply the FFT and inverse FFT on samplerate/2 and then interpolate the resulting upsampling the output again to the original samplerate. However, I get some strange overtones when I resample the signal. I've tried differe...


Re: OT Re: trying to generate a wave file of 440 Hz...

Started by Jerry Avins in comp.dsp11 years ago

A440 is _entirely_ arbitrary. "French Baroque" A392 is another "standard". In general, standard pitch has been rising for centuries, as orchestra...

A440 is _entirely_ arbitrary. "French Baroque" A392 is another "standard". In general, standard pitch has been rising for centuries, as orchestra conductors strive for ever brighter sounds. The A440 standard was adopted in 1939 (my daughter's piano is older than that) replacing the A435 standard established in 1859. In general, compositions of Bach, Haydn, Mozart, and Beethoven were


Multi-Sharc architecture

Started by Jerome in comp.dsp17 years ago 4 replies

Hi All We are ready to start a new audio design featuring 4 DSPs 21262 from ADI. I was wondering if anyone has experience in multiple...

Hi All We are ready to start a new audio design featuring 4 DSPs 21262 from ADI. I was wondering if anyone has experience in multiple dsps architectures, more precisely in inter-communication between the dsps. Our architecture is currently as follow : - 1 dsp to interface inputs / outputs (audio codecs, spdif, usb audio ..) - 3 dsp to generate sounds based on our algorithms. the dsp 1...


Sample Rate Conversion by non integer factor

Started by David Reid in comp.dsp18 years ago 15 replies

How can this be done? My application is converting the sample rate of 44.1kHz wave files to sample rate X (eg change SR by factor of 0.75, or...

How can this be done? My application is converting the sample rate of 44.1kHz wave files to sample rate X (eg change SR by factor of 0.75, or 1.3) and playing back at 44.1kHz, so that the speed sounds different. This is being done in C++. So far, what i've read seems to only apply to SRC by integer factors. One method is: Upsampling: Interpolate zeroes (in time domain?), low pass filt...


Bell 103 / v.21 modem emulation software (FSK modutaion with sound card or WAVE file output under DOS or Win32)

Started by Daktaklakpak in comp.dsp15 years ago 11 replies

I'm looking for some DOS/Win32 software that will emulate a Bell 103 and ITU-T v.21 modem that uses a WAVE format file or a computer's sound...

I'm looking for some DOS/Win32 software that will emulate a Bell 103 and ITU-T v.21 modem that uses a WAVE format file or a computer's sound card for the modulation/demodulation audio signal output/input. These modems use AFSK modulation at 300 baud and 300 bps, at the following frequencies (in Hertz): Bell 103 Originate 1170 = Carrier 1070 = Space (Carrier - 100Hz) 1270 = ...


Resampling in stages?

Started by Anonymous in comp.dsp9 years ago 7 replies

Is there a reason (founded in DSP theory) to resample in stages instead of going directly from one sample rate to another? If so what is the...

Is there a reason (founded in DSP theory) to resample in stages instead of going directly from one sample rate to another? If so what is the reason? And where can I read about it? For example: When I resample from 8kHz to 48kHz the resulting audio sounds worse compared to the audio I get if I resample in stages from 8kHz to 24kHz to 48kHz. Thanks


Using correctly the quarter sine-wave symmetry in a basic DDS

Started by gretzteam in comp.dsp12 years ago 11 replies

Hi, In a very basic DDS, the first method used to reduce the ROM size is to only store one quarter of the sine-wave. Although this sounds very...

Hi, In a very basic DDS, the first method used to reduce the ROM size is to only store one quarter of the sine-wave. Although this sounds very obvious, I cannot figure out how to do it without storing 1/4th + 1 location in the ROM. For example, say we start with a 10-bit address for the full table (1024 locations). Ideally, we want to use the top two bits to determine which quadrant we are int...


Sound Laser...

Started by HardySpicer in comp.dsp12 years ago

http://physicsworld.com/cws/article/news/41857

http://physicsworld.com/cws/article/news/41857


[Ambiguity Function Derivation] What Is professor's Words (Von Neumann Measure? / L2 Measure?)

Started by Anonymous in comp.dsp16 years ago 8 replies

Hello, I taped a Sensor Fusion Course (one year ago) and I was wondering does anybody actually know what are the words the instructor is...

Hello, I taped a Sensor Fusion Course (one year ago) and I was wondering does anybody actually know what are the words the instructor is using? On the tape, it sounds something like Von Neummann Measure and L2 Measure which can't be correct. What do you think? The displayed viewgraphs in class are titled "Ambiguity Function Scenario" and then "The Ambiguity Function Development." Fir...


Recommended name for ADC counts?

Started by Chris Bore in comp.dsp9 years ago 17 replies

When creating metadata for a signal I give names to the values - for instance 'volts' or 'Pascals'. Before calibration, a signal read from an ADC...

When creating metadata for a signal I give names to the values - for instance 'volts' or 'Pascals'. Before calibration, a signal read from an ADC has a value that I tend to refer to as measured in 'ADC counts'. That sounds a bit clumsy - does anyone know of a routinely accepted name for that type of unit? Thanks, Chris Bore BORES Signal processing chris@bores.com www.bores.com