Can anyone provide sample pseudocode or share some existing link that has sample code.
Like for example I have a mix audio of 1kHz or 2kHz or 8kHz or so, and I want to boost certain frequencies like 1kHz only in real-time.
Reading some DSP books and resources confuses me.
You just need to design and implement a suitable digital filter. This is a large and complex subject area though, so you won't get a simple answer here. Probably the best thing as a first step would be to read a good introductory book on DSP, e.g. Understanding DSP by Rick Lyons, which is a very good for beginners as it's not too heavy on the math and has a more practical bent than most such introductory DSP books.
For this particular application though what you are trying to do is similar to implementing a graphic equalizer, and there are many pointers to how to implement this kind of thing if you use e.g. "graphic equalizer" as a search term.
There's a lot of math behind digital filtering. Sorry, I think it is important to at least understand basic filters (like those used in electronics). If you don't want to go through the basics: best to get an audio graphics equaliser where you can play with the (virtual) sliders. If you want to implement a very specific filter, please read on.
Real time: depends on your computing platform. If this is a small micro (like AVR, Microchip PIC,..) you'll need an efficient algorithm. This is likely a IIR band pass filter. The equivalent of a graphics equaliser consists of multiple band pass filters, all summed together. See http://en.wikipedia.org/wiki/Infinite_impulse_response
A more computing intensive algorithm uses FIR filters. In that case you can also control the phase of the filtered signal. http://en.wikipedia.org/wiki/Finite_impulse_response
If you find an algorithm (i.e. IIR), you'll need to calculate the coefficients. The algorithm is simple, calculating the coefficients is not.
I found a book matching your question: Audio digital signal processing in real time
I browsed through it; it seems to have the right answers.
Related
Lets say I have the audio file for Happy Birthday. I want to convert that audio file into an audio file that sounds like this : happy birthday.
First, I'd like to know if I have the ability to program this? Can a highschooler who's almost finished with APCS program this?
If I can:
How would I change the bpm of the song? I've searched through a bunch of websites, but they weren't very helpful.
I know that audio files can be represented in waveforms. How would I scan for each individual wave in an audio file (I need this to isolate the notes)?
This is a very ambitious project, actually. One reason is that it involves using digital signal processing tools like FFT (Fast fourier transforms) to analyze the sound to pick out the pitches. You might be able to find a library that can do this, but as far as coding such a tool, that would involve a steep learning curve.
If you would like to look further into this, there is a good online resource called "The Scientists and Engineers Guide to Digital Signal Processing". I was able to work through and understand the discrete fourier transform with only high school math (lots of trig) and a bit of calculus. It was a lift, though.
Trying to analyze rhythm is also no easy task. Even with advanced tools provided in professional notation system such as Finale, people have trouble playing rhythms in time well enough for the best transcription tools. Algorithms that "quantize" the beats help but also limit the amount of detail that can be included in the playback.
My guess is that as interesting and worthwhile as this project would be, to bring it to completion before the semester ends would require putting together prebuilt pieces. A lot of programming is done that way, these days.
If you scale the project back to something like just getting your code to analyze a short sample of a single note and give its pitch, that would be both impressive and doable with a lot of work. It could be done with a DFT algorithm instead of requiring FFT, reducing the amount of info you'd have to acquire first. That way, you'd only have to work your way up to understanding and implementing the material on this link which is about calculating the DFT. Notice that there is example code in BASIC. The code examples throughout this book are a big help.
I just saw a paper by Cornell reconstructing faces from sound. But I am more interested in the timbre. It might be attacked with AI, but is there an easier way? For example, is instrument a going to be on a different range than instrument b.
For the most part, instruments are going to have overlapping frequency content. IDK the specific algorithms for isolating instruments--I've heard they do exist. I would think that a big element is not just tracking all the harmonics and frequency content, but looking for correspondences in volume changes or frequency changes of the different frequencies, in order to determine which frequencies should be grouped together as a single instrument. Since instruments often play the same notes at the same time, this would be no mean feat. If you are a beginner with digital signal processing, can I recommend "The Scientists and Engineers Guide to DSP" by Steve Smith? (Free download, good book on the fundamental knowledge needed to tackle such a project.)
I am using the command line tool aubiopitch to analyze voice recordings. My goal is to determine the fundamental frequency of the voice recorded. I know, of course, that the frequency varies – that's why I want to calculate an "average" in Hz over a 30-second recording.
My question: aubio uses different methods to determine the pitch of a recording: Schmitt trigger, harmonic comb, yin, yinfft etc. Which one of those would be my preferred choice when dealing with pure human voice recordings (no background music, atmo etc.).
I would recommend using yinfast or yinfft (default). For a discussion of the algorithms, their parameters, and their performance, see Chapter 3 of this document.
Note that the median is better suited than the average in this case.
CREPE is good and outperforms many others since it uses advanced neural-network machine learning for pitch prediction. It might be unstable in unseen conditions though and might not be very easy to plug since it requires tensorflow.
For more traditional and lightweight solution oyu can try REAPER.
I want to take two sounds that contain a dominant frequency and say 'this one is higher than this one'. I could do FFT, find the frequency with the greatest amplitude of each and compare them. I'm wondering if, as I have a specific task, there may be a simpler algorithm.
The sounds are quite dirty with many frequencies, but contain a clear dominant pitch. They aren't perfectly produced sine waves.
Given that the sounds are quite dirty, I would suggest starting to develop the algorithm with the output of an FFT as it'll be much simpler to diagnose any problems. Then when you're happy that it's working you can think about optimising/simplifying.
As a rule of thumb when developing this kind of numeric algorithm, I always try to operate first in the most relevant domain (in this case you're interested in frequencies, so analyse in frequency space) at the start, and once everything is behaving itself consider shortcuts/optimisations. That way you can test the latter solution against the best-performing former.
In the general case, decent pitch detection/estimation generally requires a more sophisticated algorithm than looking at FFT peaks, not a simpler algorithm.
There are a variety of pitch detection methods ranging in sophistication from counting zero-crossing (which obviously won't work in your case) to extremely complex algorithms.
While the frequency domain methods seems most appropriate, it's not as simple as "taking the FFT". If your data is very noisy, you may have spurious peaks that are higher than what you would consider to be the dominant frequency. One solution is use window overlapping segments of your signal, and do STFTs, and average the results. But this raises more questions: how big should the windows be? In this case, it depends on how far apart you expect those dominant peaks to be, how long your recordings are, etc. (Note: FFT methods can resolve to better than one-bin size by taking into account phase information. In this case, you would have to do something more complex than averaging all your FFT windows together).
Another approach would be a time-domain method, such as YIN:
http://recherche.ircam.fr/equipes/pcm/cheveign/pss/2002_JASA_YIN.pdf
Wikipedia discusses some more methods:
http://en.wikipedia.org/wiki/Pitch_detection_algorithm
You can also explore some more methods in chapter 9 of this book:
http://www.amazon.com/DAFX-Digital-Udo-ouml-lzer/dp/0471490784
You can get matlab sourcecode for yin from chapter 9 of that book here:
http://www2.hsu-hh.de/ant/dafx2002/DAFX_Book_Page_2nd_edition/matlab.html
I am looking for a way to compare a user submitted audio recording against a reference recording for comparison in order to give someone a grade or percentage for language learning.
I realize that this is a very un-scientific way of doing things and is more than a gimmick than anything.
My first thoughts are some sort of audio fingerprinting, or waveform comparison.
Any ideas where I should be looking?
This is by no means a trivial problem to solve, though there is an abundance of research on the topic. Presently the most successful forms of machine learning in the speech recognition domain apply Hidden Markov Model techniques.
You may also want to take a look at existing implementations of HMM algorithms. One such library in its early stages is ghmm.
Perhaps even better and more readily applicable to your problem is HTK.
In addition to chomp's great answer, one important keyword you probably need to look up is Dynamic Time Warping (DTW). This is the wikipedia article: http://en.wikipedia.org/wiki/Dynamic_time_warping