Is Java a suitable alternative to C / C++ for realtime audio processing? I am considering an app with ~100 (at max) tracks of audio with delay lines (30s @ 48khz), filtering (512 point FIR?), and other DSP type operations occurring on each track simultaneously. The operations would be converted and performed in floating point. The sys...
There's this tech-festival in IIT-Bombay, India, where they're having an event called "Artbots" where we're supposed to design artbots with artistic abilities. I had an idea about a musical robot which takes a song as input, detects the notes in the song and plays it back on a piano. I need some method which will help me compute the pitc...
Update this question was previously titled as "Give me the name of a simple algorithm for signal(sound) pattern detection" My objective is to detect the presence of a given pattern in a noisy signal. I want to detect the presence of a species of insect recording the sounds with a microphone. I have previously recorded the sound of the ...
I am looking for a FFT library with a license that allow me to include it in a commercial C++ software. Free would be nice. (Intel IPPS is 199$ per head, which is a bit expensive for the number of developper that will be compiling it.) Anybody has good experience with such a FFT library? I found this page, which list a lot of librari...
Working on a tracking application using GPS. It is all fine, but sometimes because of the closed areas or bad weather I get inaccurate points. When you plot them, it just doesn't look right, with lots of hops/jumps. What algorithm should I run to filter out the bad signals It looks like an application of a blurring algorithm to me, but...
I have a range of dates and a measurement on each of those dates. I'd like to calculate an exponential moving average for each of the dates. Does anybody know how to do this? I'm new to python. It doesn't appear that averages are built into the standard python library, which strikes me as a little odd. Maybe I'm not looking in the r...
Not unlike a clap detector ("Clap on! clap clap Clap off! clap clap Clap on, clap off, the Clapper! clap clap ") I need to detect when a door closes. This is in a vehicle, which is easier than a room or household door: Listen: http://ubasics.com/so/van_driver_door_closing.wav Look: It's sampling at 16bits 4khz, and I'd like to av...
The TMS320C55x has a 17-bit MAC unit and a 40-bit accumulator. Why the non-power-of-2-width units? ...
Hi, My DAC is internally clocked at 48khz, I need to feed it from a stream that is also 48khz but is clocked from a different source. The two sources will drift over time and any buffering in between will inevitably overflow or underflow. Unfortunately, there is no rate control mechanism available for the source. How can this best be...
Hi, I may be doing an iPhone-based application doing near-real-time sound-processing (filtering, etc). I was wondering the best way to get started. Would I want to create an audio cue for recording and processing sound, as described here? Edit: I should be clear. I am not asking how to do signal processing, in general. I know some of...
I'm trying to build a video chat program using Adobe Flex but there is a giant problem with echos. If the participants arn't using headsets, everything they say echos. Worse, they can actually create positive feedback loop of echos that won't end until the mics are muted. Has anyone found a solution for this on the Flex/Flash platform...
I am trying to create a graphical spectrum analyzer in python. I am currently reading 1024 bytes of a 16 bit dual channel 44,100 Hz sample rate audio stream and averaging the amplitude of the 2 channels together. So now I have an array of 256 signed shorts. I now want to preform a fft on that array, using a module like numpy, and use ...
I am currently developing a piece of software using opencv and qt that plots data points. I need to be able fill in an image from incomplete data. I want to interpolate between the points I have. Can anyone recommend a library or function that could help me. I thought maybe the opencv reMap method but I can't seem to get that to work. T...
I'd like to compute power spectral density of time series; do some bandpass, lowpass, and highpass filtering; maybe some other basic stuff. Is there a nice open-source Java library to do this? I've hunted a bit without success (e.g., Googling "power spectral density java" or "signal processing java" and clicking through links, looking ...
In my application, i have to detect all the songs and BPM of those songs. Detect the songs are finished now I have to detect BPM of songs. Does anyone tell me that how can I detect BPM of each song by programming in Cocoa. Thanks, Haresh. ...
I need to run a CPU- and memory-heavy Python script (analyzing and altering a lengthy WAV file) as a background process on my web server (a VPS), between HTTP requests. The script takes up to 20 seconds to run and I am concerned about the performance on my server. Is there a good approach to either lower the priority of the process, per...
Folks, Matlab 2007b (7.5.0) has an avgpower function. See here: "The avgpower method uses a rectangle approximation to the integral to calculate the signal's average power using the PSD data stored in the object. "The avgpower method returns the average power of the signal which is the area under the PSD curve." Example invocation: ...
I'm wondering what is the recommended audio library to use? I'm attempting to make a small program that will aid in tuning instruments. (Piano, Guitar, etc.). I've read about ALSA & Marsyas audio libraries. I'm thinking the idea is to sample data from microphone, do analysis on chunks of 5-10ms (from what I've read). Then perform a FF...
When I use the (win32) waveInXXX functions in order to collect samples from a mic at a certain sampling frequency (say 8kHZ, even lower possibly), does the system/soundcard applies a lowpass filter on the input sample, or would i get aliasing? Should I sample at a higher frequency and do the filtering myself before lowering the sampling ...
I've actually posted this question before, but it hasn't been answered. Maybe I wasn't clear enough, so let me rephrase: As you know, when you're sampling a signal at a certain sampling rate, any frequency that's higher than half of the sampling rate gets aliased. In order to avoid it, you need to pass the signal (either in the analog f...