Audio Processing for musicqal application

Started by Anonymous in comp.dsp15 years ago 2 replies

Hi all, I wish to write a C++ program which generates sounds from FM instruments with additional processing (delays, reverbs etc.)....

Hi all, I wish to write a C++ program which generates sounds from FM instruments with additional processing (delays, reverbs etc.). The problem is I'm not sure what would be the best way to do this.Can anyonesuggest libraries etc that I might find useful? Would it be possible/useful to use VSTs within my program? Thanks, aine.


Detecting speech obscured by loud stationary noise

Started by Silash in comp.dsp13 years ago 2 replies

Hi, I would like to detect human speech in the presence of a very loud, very stationary external sound. It doesn't require a fine-grained...

Hi, I would like to detect human speech in the presence of a very loud, very stationary external sound. It doesn't require a fine-grained detection but that would be nice. It will ultimately be implemented in an FPGA. This isn't quite a negative SNR scenario because the "external sound" isn't Gaussian white noise, but a periodic waveform. So far I've learned that; 1) energy and power sta...


(Audio DSP)Comparing and masking audio

Started by Jeremy Smith in comp.dsp18 years ago 3 replies

Hi! I've been working on a program to take, say, a drum sound, then find it in another file (say, a song with that drum sound in it) by...

Hi! I've been working on a program to take, say, a drum sound, then find it in another file (say, a song with that drum sound in it) by comparing the frequency spectrum of the drum to various chunks of the song, so that it finds a close match of the spectrum charts. It compares everything in the frequency spectrum by using the Fast Fourier Transform. Once it knows where there is a du...


Moving coil exciter

Started by raymund hofmann in comp.dsp18 years ago 7 replies

I am looking for a "moving coil exciter" suitable for attaching to some surface like a window glass. I want to turn a window glass into...

I am looking for a "moving coil exciter" suitable for attaching to some surface like a window glass. I want to turn a window glass into a speaker. I don't want to achieve high sound pressures, but prefereably be able to go down to ~35 Hz (at sound pressures quite low). I guess signal processing and driving the exiter to it's limits is necessary to achieve linear response down to 35 Hz. I alr...


Higher upsampling with minimum phase downsampling produces more aliasing

Started by jungledmnc in comp.dsp7 years ago 20 replies

Hi, I'm programming a sound generator, based on wavetables. I have 8192 point wavetable. I create several band-limited "subwavetables" by...

Hi, I'm programming a sound generator, based on wavetables. I have 8192 point wavetable. I create several band-limited "subwavetables" by taking DFT, zeroing high octave(s) and IDFT. For generating particular pitch a choose a wavetable, which has all harmonics until 20k. Sound good so far, way better than just upsampling the original non-band-limited wavetable. The harmonics that exceed 22k ...


Sensors for monitoring motors and pumps?

Started by Rune Allnor in comp.dsp13 years ago 9 replies

Hi folks. Assume one wants to monitor the mechanical condition a pump or motor by means of acoustics, and this pump or motor is located in a...

Hi folks. Assume one wants to monitor the mechanical condition a pump or motor by means of acoustics, and this pump or motor is located in a room full of other pumps and motors. What sensors should one use to record the sound? A naive microphone would likely record the sound from all the pumps, meaning the acostic signature of the pump under investigation is corrupted by noise from al...


fixed point atan2

Started by Alex Hornung in comp.dsp11 years ago 19 replies

Hi, first off I'd like to apologize in case this question is/sounds extremely stupid, but I'm really stuck. I need an efficient atan2 that...

Hi, first off I'd like to apologize in case this question is/sounds extremely stupid, but I'm really stuck. I need an efficient atan2 that doesn't take up as much real estate on an FPGA as a CORDIC would. As such, I've been looking at the 'Trick' mentioned at dspguru[1]. So according to that trick, given: x = 0.1838 y = -0.1818 I would now do the following, since it is in the I...


Question regarding beamforming on data acquired through sound card

Started by Sylvia in comp.dsp15 years ago 2 replies

I have taken data from single microphone in MATLAB work space through MATLAB data acquisition tool box as left and right channels(inorder...

I have taken data from single microphone in MATLAB work space through MATLAB data acquisition tool box as left and right channels(inorder to simulate the data from 2 microphones).Here is the code I have used for sound data acquistion. AI = analoginput('winsound',0); chan = addchannel(AI,1); chan2=addchannel(AI,2); AI duration = 2; SampleRate = 44100; set(AI,'SampleRate',SampleRate); set...


Sick of sample-based "synthesis"!

Started by Radium in comp.dsp18 years ago 65 replies

Sample-based synths are stale and rigid. Any sound effect in action will noticeably quantize and alias the music. They are a hell an earsore for...

Sample-based synths are stale and rigid. Any sound effect in action will noticeably quantize and alias the music. They are a hell an earsore for life-wanting instruments such as synth pads and synth fx. The tone of synth pads are generated on FM synths! No wonder pads sound so crappy in samplers. A *real* digital (not analog) FM/modelling synth is a dream! It should be hard-coded and able ...


[Cont'd Tuesday Feb 21] Detect Sound of Breaking Glass with Labview

Started by Anonymous in comp.dsp16 years ago 2 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. Update: Hi guys. Thanks a lot for the exc...


High Gain, Low Volume, 'classic' guitar sound

Started by chuckles in comp.dsp15 years ago 8 replies

This is a bit tricky to explain, I've been playing around with clipping for a little while using a clean guitar, passed through a...

This is a bit tricky to explain, I've been playing around with clipping for a little while using a clean guitar, passed through a clipping algorithm which produces a square wave at its highest level (when passed a sinusoid) and a rounded version of the sinusoid for values 0-(max-1). The sound it produces becomes gradually 'fuzzier' as the level is set higher and higher. Its not great and prod...


volume change

Started by christos in comp.dsp13 years ago 4 replies

Does anyone knows how to change the volume by changing the amplitute taken from a 16bit audio stream.for now i have a byte array which is my...

Does anyone knows how to change the volume by changing the amplitute taken from a 16bit audio stream.for now i have a byte array which is my sample.i convert the byte array into an integer array.each element of the integer array is the amplitude of my sound.then i multuply each element of the integer array by 2 for example.the sound i get is full of noise.does enyone know what the problem is? ...


Example Digital Convolution in the Time Domain is Computationally Expensive

Started by johnnmonroe in comp.dsp6 years ago 4 replies

Looking for a succinct example showing convolution processing (in the time domain) is notoriously computationally intensive; e.g., the...

Looking for a succinct example showing convolution processing (in the time domain) is notoriously computationally intensive; e.g., the typical reverberation time of a room is approximately 0.3 seconds which corresponds to 2400 samples, i.e., taps (filter coefficients), for an 8 kHz sampled sound. Because the sound is sampled at 8 kHz, the "delay steps" are each of length 1/8000 (0.000125 seconds)...


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

Started by Bryan in comp.dsp11 years ago

I'm not sure exactly what you're asking, but if I interpret it correctly in that you're wondering why intonation is the way it is, and why middle...

I'm not sure exactly what you're asking, but if I interpret it correctly in that you're wondering why intonation is the way it is, and why middle C and A(440) tend to be anchor notes scales are built from, I recommend the following book(s): http://www.musimathics.com/ Very cool if you like math and music. The first volume covers what I just mentioned as well as the physics behind sounds and ...


Spectral Purity Measurement

Started by rickman in comp.dsp7 years ago 74 replies

I want to analyze the output of a DDS circuit and am wondering if an FFT is the best way to do this. I'm mainly concerned with the "close in"...

I want to analyze the output of a DDS circuit and am wondering if an FFT is the best way to do this. I'm mainly concerned with the "close in" spurs that are often generated by a DDS. My analysis of the errors involved in the sine generation is that they will be on the order of 1 ppm which I believe will be -240 dBc. Is that right? Sounds far too easy to get such good results. I gues...


Speech sound from the web: Sampling rate, mono vs stereo?

Started by Anonymous in comp.dsp7 years ago 1 reply

I am capturing human speech on the web. I find there are examples that people are sampling at 44.1 and 48kHz. All of them are also stereo. If...

I am capturing human speech on the web. I find there are examples that people are sampling at 44.1 and 48kHz. All of them are also stereo. If the sole purpose of capturing the sound is for extracting features, what might be the minimum or optimal sampling rate? Is there any value in Stereo signal? Am I correct that the left and right speech samples are identical in stereo - so just ignori...


MELPe vs. IMBE

Started by CR in comp.dsp16 years ago 1 reply

Does anyone have any results of quality comparison between IMBE (or AMBE) and MELPe at rates 2400, 1200, 600 bps? I heard MELPe sounds much...

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


"Maker board"

Started by Steve Pope in comp.dsp5 years ago 37 replies

I've recently seen in job listings, contract requirements etc. references to desiring engineers who are "experienced with Maker boards". What...

I've recently seen in job listings, contract requirements etc. references to desiring engineers who are "experienced with Maker boards". What exactly is a Maker board? I gather it's not a brand name, but some general concept ... it sounds a little new-age or something. Is there a specific meaning? Steve


Echo Cancellation on PC platform

Started by qfu72 in comp.dsp16 years ago 13 replies

we want to implement an acoustic echo canceller on a windows PC platform for chatting tools like MSN/skype. A serious problem we found in this...

we want to implement an acoustic echo canceller on a windows PC platform for chatting tools like MSN/skype. A serious problem we found in this kind of platform is the time delay (sound card play+speaker+air+mic.+ sound card record) between Ref. signal and Echo signal is not consist in every session 1)how could we esimate the time dealy? 2) why the AEC is so sensitive to time delay setting? ...


Using Re[X] or Im[X] from FFT result

Started by Jonas Rundberg in comp.dsp17 years ago 10 replies

Hello I am working on a software that will do frequency analysis of a sound signal. When doing the FFT (or DFT) the result is a compex...

Hello I am working on a software that will do frequency analysis of a sound signal. When doing the FFT (or DFT) the result is a compex number where the real part corresponds to the cosine component and the imaginary part corresponds to the sine component of the signal. If I analyze a sound consisting of a single frequency there is a slight difference in the DFT result, regarding the freque...