Sound values dB rms conversion to dB 0-peak

Started by Donatas911 in comp.dsp7 years ago 4 replies

Good day for everyone, A question: How to convert a sound pressure level value in dB rms to the level in dB 0 to Peak value (having a values...

Good day for everyone, A question: How to convert a sound pressure level value in dB rms to the level in dB 0 to Peak value (having a values reference 1 micro Pascal)? As I recognized the relation for amplitude rms x 0.707 is valid for voltage values but that is different. D. _____________________________ Posted through www.DSPRelated.com


dynamic compression/noisegating/sound schemes

Started by Thomas Pototschnig in comp.dsp17 years ago 9 replies

hi NG I want to develop an small extern device with a small DSP for improving audio with dynamic compression, noise gating and sound...

hi NG I want to develop an small extern device with a small DSP for improving audio with dynamic compression, noise gating and sound schemes. But before I can start with the hardware and software of the DSP I want to test different algorithms on the PC. Unfortunately I didn't find anything which can help me for this project in the internet. Now my question: Does anyone know good sour...


Ultrasonic communication with consumer devices

Started by Piergiorgio Sartor in comp.dsp5 years ago 12 replies

Hi all, lately, there are more and more news about communication, using ultrasonic sound data transmission, between consumer devices, like...

Hi all, lately, there are more and more news about communication, using ultrasonic sound data transmission, between consumer devices, like smartphones. Here some links: https://www.newscientist.com/article/2110762-your-homes-online-gadgets-could-be-hacked-by-ultras ound/ http://arstechnica.com/tech-policy/2015/11/beware-of-ads-that-use-inaudible-sound-to-link-your-p hone-tv-tablet-and-pc


A Sound Mathematical Basis For Sampling - Lesson 3

Started by Airy R. Bean in comp.dsp17 years ago 7 replies

A Sound Mathematical Basis For Sampling - Lesson 3 -------------------------------------------------- Good Morning, once again, Boys and...

A Sound Mathematical Basis For Sampling - Lesson 3 -------------------------------------------------- Good Morning, once again, Boys and Girls! I'm sorry that I got called away yesterday; SWMBO, indeed, MBO! Today I'll derive for you the mathematics of sampling, based on an analysis of the circuits that we actually use, rather than on some dubious mathematics (I will show later why i...


Simple question regarding frequency detection

Started by Atri Mandal in comp.dsp18 years ago 13 replies

Hi, I am writing an application to detect a particular frequency(viz. 4.5 Khz) in an input audio signal. I have written the code to capture...

Hi, I am writing an application to detect a particular frequency(viz. 4.5 Khz) in an input audio signal. I have written the code to capture the sound using a microphone; but I have to properly design a digital filter which will return TRUE as soon as it detects sound of this frequency in the input signal. Also my project demands that I have to detect the signal by looking at a very small num...


DTMF detection through Sound Card

Started by dkurman in comp.dsp16 years ago 4 replies

Hi. I have posted two messages, but I can?t get this working. I mean, I'm getting the buffer from the sound card, and then I perform the...

Hi. I have posted two messages, but I can?t get this working. I mean, I'm getting the buffer from the sound card, and then I perform the goertzel algorithm with this, but the data I get is wrong almost all times. The steps are: Data: Buffer Size: 4096 Kb, N=4096, Sample Rate = 22050, Format: 16 bits 1) When I get the buffer completely filled, I read the buffer (two bytes a time) convertin...


Tracking one frequency

Started by Amanda Robin in comp.dsp16 years ago 4 replies

Please excuse this beginner question. I have about a minute's worth of sound data taken by a microphone and recorded into a file. The...

Please excuse this beginner question. I have about a minute's worth of sound data taken by a microphone and recorded into a file. The sound field has one frequency of interest and some random noise (at a lower magnitude than the random noise). My sample rate is about 6x the frequency of interest. I want to show the change in the frequency of interest alone. I am thinking to perfo...


Autocorrelation rocks for Engine RPM

Started by Robert Scott in comp.dsp8 years ago 22 replies

A while ago I posted a question about how to extract pitch information from the messy sound of an engine. It was for an iPhone app I...

A while ago I posted a question about how to extract pitch information from the messy sound of an engine. It was for an iPhone app I have developed which displays the RPM of an engine based on the sound it makes. Vladimir and other suggested autocorrelation as used in pitch detection in speech. Thanks, guys! After a long time struggling with various ad hoc ways to qualify peaks in the FFT...


Anybody Using S.F. Acoustic Mirror?

Started by Bob Cain in comp.dsp17 years ago 4 replies

Is anybody using Sound Forge (v5 and above) and Acoustic Mirror to calculate impulse responses from sweeps? I bought the old stand alone...

Is anybody using Sound Forge (v5 and above) and Acoustic Mirror to calculate impulse responses from sweeps? I bought the old stand alone Acoustic Modeler DX plugin from Sonic Foundry when they marketed it separately and it won't process recorded files or test sweeps or produce impulse responses over 16 bits. I'd like to know if the newer versions that are only bundled with Sound Fo...


spectral subtraction of signals other than noise

Started by adj in comp.dsp16 years ago 9 replies

i have been reading a lot of messages on spectral subtraction and how it is useful in reducing background noise. i have tried this in matlab and...

i have been reading a lot of messages on spectral subtraction and how it is useful in reducing background noise. i have tried this in matlab and i was highly successful in attenuating the noise from a sound file. but when i tried this with another sound file instead of noise the signal was not attenuated at all. in fact both of the signals could be heard perfectly well; and i can say pretty...


How to implement a data acquisition system with pretrigger function on sound card

Started by Ze Ji in comp.dsp15 years ago 1 reply

As the title, I am willing to program in C language to use a sound card as a data acquisition system with the function of pretrigger, as used in...

As the title, I am willing to program in C language to use a sound card as a data acquisition system with the function of pretrigger, as used in Matlab. I have been searching on the internet for long time, but no useful information. There are some shareware of soft ocsiscope. Any idea? Thank you very much.


simple sound algorithms

Started by Anonymous in comp.dsp16 years ago 4 replies

Hello, I have written a simple sound editor for my university project that can read a .Wav file and saves its format and wave form data in...

Hello, I have written a simple sound editor for my university project that can read a .Wav file and saves its format and wave form data in arrays. Now I want to add capability to perform some effect on waveform data .I have written a Fade-In/Fadeout effect myself. but when I tried to find algorithms for other effects in internet I got confused! I almost know nothing about DSP or advanced sou...


Detect sound of breaking glass

Started by Anonymous in comp.dsp16 years ago 14 replies

Hello. I'm an undergraduate working on a project to build a home security system using DSP and NI's Speedy-33 board. My question is : 1) how can...

Hello. I'm an undergraduate working on a project to build a home security system using DSP and NI's Speedy-33 board. My question is : 1) how can I detect breaking glass sound using LabView? 2) how to connect external hardware(e.g siren, lights) to the Speedy-33 board that will be activated if breaking glass is detected. Thank you.


digital Sound IF filter must be linear phase?

Started by dtsao in comp.dsp14 years ago 5 replies

Hi, I need to build a digital filter for a sound signal. The SIF filter must remove the video portion of the channel and be compatible with...

Hi, I need to build a digital filter for a sound signal. The SIF filter must remove the video portion of the channel and be compatible with worldwide standards and have good rejection. From Matlab, it looks like this requires a tremendous amount of coefficients if built using FIR. So I wonder, does anyone know if a SIF filter MUST be linear phase? Or can it be done using an IIR? And what wou...


2 one-pole filters in series

Started by Emile in comp.dsp13 years ago 3 replies

Hi, im working on a schoolproject to render sound in a virtual environment. To do that i calculate sound propagation paths. At the moment i...

Hi, im working on a schoolproject to render sound in a virtual environment. To do that i calculate sound propagation paths. At the moment i model wall soundabsorption with onepole lowpass filters. Some paths bounce of several walls, causing several onepole filters working in series. I update my paths at about 25Hz and filtering i do in timedomain at 44100Hz. Now i would like to replace t...


simple code for audio (or signal or sound ) sampling using gibbs sampling

Started by Anonymous in comp.dsp6 years ago 2 replies

hello everybody, I search in google a lot but I can not find code of using gibbs sampling in audio (signal or sound) processing. if you can please...

hello everybody, I search in google a lot but I can not find code of using gibbs sampling in audio (signal or sound) processing. if you can please help me. thanks in advance


time variable delay

Started by Ralph Harry in comp.dsp17 years ago 9 replies

Hi to all, I need to implement a time variable delay on dsp in order to simulate the doppler effect of a sound source coming nearer and moving...

Hi to all, I need to implement a time variable delay on dsp in order to simulate the doppler effect of a sound source coming nearer and moving away. If I just pick the from the buffer the sample with delay corresponding to (distance of source)/(speed of sound), I get clicks when moving from one position to an other. What's the proper way (preferably not too expensive) to implement such a ...


sound recording in spectrogram

Started by nishant_jain_nj_ in comp.dsp11 years ago 1 reply

hello, i recorded my voice in .wav format in specrogram. I want to use that recorded file in matlab by using wavread command. but i m not able to...

hello, i recorded my voice in .wav format in specrogram. I want to use that recorded file in matlab by using wavread command. but i m not able to do so.it gives a error -------> Error using ==> wavread at 166 Data compression format (Format #65534) is not supported. what should i do? my aim is to record sound using spectrogram. identify the formants and extract them. i m recording " vowe


Re: PCM

Started by Ben Bradley in comp.dsp16 years ago

In comp.dsp,rec.audio.tech,sci.optics,rec.music.makers.synth,alt.os, On Mon, 03 Oct 2005 17:30:43 -0400, Jerry Avins wrote: > Arny Krueger...

In comp.dsp,rec.audio.tech,sci.optics,rec.music.makers.synth,alt.os, On Mon, 03 Oct 2005 17:30:43 -0400, Jerry Avins wrote: > Arny Krueger wrote: > > "Don Pearce" wrote in message > > ... > > > > > It would sound like noise. > > > > > > Indeed if it didn't sound like noise, I would be sending > > > the codec back to the writer for a better try. > > > > > >


Real To Complex Stream Conversion

Started by Isaac Gerg in comp.dsp17 years ago 8 replies

Hi all, I have a need to sample audio from my sound card and process it. However, the hardware I am processing it on works faster in...

Hi all, I have a need to sample audio from my sound card and process it. However, the hardware I am processing it on works faster in complex domain than real domain. I know this may sound funny, but it is true. Read on... In essence, I need to convert a real stream sampled at N to a complex stream sampled at N/2. Not sure what the best way to do this is or if it can even be don...