2
голосов
1ответов
3236 просмотров

How to write a downsampling function in Java

I'm trying to write a filtering function for an image, but I can't seem to wrap my head around (or remember) how to transfer all that math theory into code. Lets say I have the following function, where the ints inside the arrays are integers between 0 and 255 (pretty much grayscale pixels to ke...

3
голосов
5ответов
2806 просмотров

Is it possible to find stretches of silence in audio files with Javascript?

I've been working on a tool to transcribe recordings of speech with Javascript. Basically I'm hooking up key events to play, pause, and loop a file read in with the audio tag. There are a number of advanced existing desktop apps for doing this sort of thing (such as Transcriber -- here's a scree...

5
голосов
1ответов
8085 просмотров

Audio Processing - Tone Recognition

I've started developing a simple guitar tuner as a learning project for audio processing programs. Could someone recommend me an adequate library for this? Here's basically what I'm looking for: Microphone input. Real-time processing of the signal. I need to be able to perform note recognition...

1
голосов
1ответов
920 просмотров

Pitch recognition of musical notes on a smart phone, pt. 2

As a follow-up to my previous question, if I want my smartphone application to detect a certain musical note, and I only need to know whether the incoming sound is that musical note or not, with a certain amount of fuzziness, to allow the note to be off-key by x cents. Given that, is there a sup...

3
голосов
2ответов
7912 просмотров

How to generate a lower frequency version of a signal in Matlab?

With a sine input, I tried to modify it's frequency cutting some lower frequencies in the spectrum, shifting the main frequency towards zero. As the signal is not fftshifted I tried to do that by eliminating some samples at the begin and at the end of the fft vector: interval = 1; samplingFreque...

10
голосов
7ответов
19955 просмотров

How do I play a sound in Octave?

Octave appears to assume that a specific sound playing utility will be available on a system but doesn't seem to provide the ability to specify an alternate. In the error below, Octave is looking for ofsndplay, which is not a utility available on all systems. octave:38> sound(beamformed_20) ...

9
голосов
3ответов
22242 просмотров

Real-time pitch detection using FFT

I'm trying to do real-time pitch detection using C++. I'm testing some code from performous (http://performous.org/), because everything else hasn't worked for me. I know for sure that this works, but i just cant get it to work. I've been trying this for a few weeks now, and I haven't been able t...

23
голосов
5ответов
13253 просмотров

Pitch recognition of musical notes on a smart phone

With limited resources such as slower CPUs, code size and RAM, how best to detect the pitch of a musical note, similar to what an electronic or software tuner would do? Should I use: Kiss FFT FFTW Discrete Wavelet Transform autocorrelation zero crossing analysis octave-spaced filters other? ...

7
голосов
8ответов
4621 просмотров

TI DSP programming - is C fast enough or do I need an assembler?

I am going to write some image processing programs for Texas Instruments DaVinci platform. There are tools appropriate for programming in the C language, but I wonder if it is really possible to take full advantage of the DSP processor without resorting to an assembly language. Do you know about ...

3
голосов
4ответов
2507 просмотров

Avoiding floating point arithmetic

I wrote a small software synth for the iPhone. To further tune performance I measured my application with Shark and found that I am loosing a lot of time in float/SInt16 conversions. So I rewrote some parts to get around the conversions by pre-calculating lookup tables that return "ready-to-use"...

0
голосов
2ответов
748 просмотров

C++ api for understanding tone signals on a phone line

Is there any good c++ source codes or api for handling phone lines like understanding tone signals. For example i like to find out if the person enters 3 (it's likely that this is done using it's tone sound). Do i need a special modem for this purpose or it can be done using only standard modems.

8
голосов
5ответов
5616 просмотров

iPhone: CPU power to do DSP/Fourier transform/frequency domain?

I want to analyze MIC audio on an ongoing basis (not just a snipper or prerecorded sample), and display frequency graph and filter out certain aspects of the audio. Is the iPhone powerful enough for that? I suspect the answer is a yes, given the Google and iPhone voice recognition, Shazaam and ...

5
голосов
3ответов
4217 просмотров

OpenAL Real Time Audio Processing from Microphone

I would like to write a cross-platform application that can process and play back microphone data in real time. Imagine as a proof of concept a chat room where people can talk to each other and apply filters to their voices. Is OpenAL appropriate for this? If not, can someone provide an altern...

18
голосов
7ответов
15846 просмотров

Calculating vs. lookup tables for sine value performance?

Let's say you had to calculate the sine (cosine or tangent - whatever) where the domain is between 0.01 and 360.01. (using C#) What would be more performant? Using Math.Sin Using a lookup array with precalculated values I would anticpate that given the domain, option 2 would be much faster. ...

7
голосов
2ответов
4390 просмотров

How do I multiply the spectra of two images of different dimensions?

This is not a "programming" question. But I'm sure it's something that is widely known and understood in this community. I have an image, x, and a much smaller image, y, and I need to convolve the two by multiplying their FFTs. But since they are not the same size I don't know how to do the freq...

26
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11ответов
55678 просмотров

Real time pitch detection

I'm trying to do real time pitch detection of a users singing, but I'm running into alot of problems. I've tried lots of methods, including FFT (FFT Problem (Returns random results)) and autocorrelation (Autocorrelation pitch detection returns random results with mic input), but I can't seem to g...

2
голосов
4ответов
3437 просмотров

Autocorrelation returns random results with mic input (using a high pass filter)

Sorry to ask a similar question to the one i asked before (FFT Problem (Returns random results)), but i've looked up pitch detection and autocorrelation and have found some code for pitch detection using autocorrelation. Im trying to do pitch detection of a users singing. Problem is, it keeps re...

3
голосов
4ответов
16576 просмотров

Simple signal processing in C#

I'm sampling a real-world sensor, and I need to display its filtered value. The signal is sampled at a rate of 10 Hz and during that period it could rise as much as 80 per cent of the maximum range. Earlier I've used Root Mean Square as a filter and just applying it to the last five values I've ...

5
голосов
6ответов
8842 просмотров

What is a "good" R value when comparing 2 signals using cross correlation?

I apologize for being a bit verbose in advance: if you want to skip all the background mumbo jumbo you can see my question down below. This is pretty much a follow up to a question I previously posted on how to compare two 1D (time dependent) signals. One of the answers I got was to use the cros...

4
голосов
2ответов
2871 просмотров

Measure audio noise level

I'm trying to get a qualitative handle on the amount of static or noise present in a audio stream. The normal content of the stream is voice or music. I've been experiementing with taking the stddev of the samples, and that does give me some handle on the presence of voice vs. empty channel noi...

2
голосов
2ответов
2384 просмотров

Automated transcription software

I've noticed that the wiki transcriptions for some of the recent Stack Overflow Podcasts are kind of weak. Clearly, this task calls for a computer program. Is transcribing audio to text (ideally with speaker labels so we know who said what) something that could feasibly be accomplished in softw...

11
голосов
6ответов
6381 просмотров

Algorithm to Match Time Dependent (1D) Signals

I was wondering if someone could point me to an algorithm/technique that is used to compare time dependent signals. Ideally, this hypothetical algorithm would take in 2 signals as inputs and return a number that would be the percentage similarity between the signals (0 being that the 2 signals ar...

23
голосов
2ответов
10835 просмотров

Learning Digital Signal Processing

What are some good resources for learning about DSP (including the mathematics and algorithms necessary for actually understanding these resources)? Let's assume that my math skills are rusty from lack of use as well, so a roadmap along the lines of: Stats refresher Calculus refresher Solid ne...

8
голосов
2ответов
5208 просмотров

Reverse Spectrogram A La Aphex Twin in MATLAB

I'm trying to convert an image into an audio signal in MATLAB by treating it as a spectrogram as in Aphex Twin's song on Windowlicker. Unfortunately, I'm having trouble getting a result. Here it what I have at the moment: function signal = imagetosignal(path, format) % Read in the image a...

-1
голосов
5ответов
3950 просмотров

SDR kit with 2.4GHz RF frontend?

Do you know a SDR (Software Defined Radio) kit with a 2.4GHz ISM band (2400MHz - 2483.5MHz) transceiver? I need to perform some software defined radio including customised modulation. Also the price for one kit should be at maximum $1000. I know there are some extremely expensive solutions out t...

3
голосов
2ответов
950 просмотров

Development kit for Bluetooth which allows customisation of the modulation algorithms

We need to perform some experiments on the Bluetooth protocol, and for this we need a development kit which allows us to implement/modify different parts of the Bluetooth protocol stack. We have been looking at the TI MSP430 Wireless Development Tool (EZ430-RF2500). This kit contains a MSP430 MC...

6
голосов
4ответов
4827 просмотров

8 bit audio samples to 16 bit

This is my "weekend" hobby problem. I have some well-loved single-cycle waveforms from the ROMs of a classic synthesizer. These are 8-bit samples (256 possible values). Because they are only 8 bits, the noise floor is pretty high. This is due to quantization error. Quantization error is pretty...

0
голосов
1ответов
486 просмотров

Integer FM Demodulation

What are some software (or FPGA) techniques suitable for FM demodulation? I've been experimenting in MATLAB to try and get an algorthm right, but I've been basing it on a analog reference material with limited results. I can make out the audio, but there is horrible distortions that I can't fix...

2
голосов
6ответов
7730 просмотров

Sine Table Interpolation

I want to put together a SDR system that tunes initially AM, later FM etc. The system I am planning to use to do this will have a sine lookup table for Direct Digital Synthesis (DDS). In order to tune properly I expect to need to be able to precisely control the frequency of the sine wave fed to ...

2
голосов
5ответов
4733 просмотров

Voice Alteration Algorithm

Could somebody point me to a voice alteration algorithm? Preferably in Java or C? Something that I could use to change a stream of recorded vocals into something that sounds like Optimus Prime. (FYI- Optimus Prime is the lead Autobot from transformers with a very distinctive sounding voice... not...