This is for a personal project of mine, and I have no idea from where to start as it falls way beyond my comfort zone.
I know that there are a few language learning software out there that allows the user to record his or her voice and compare the pronounce with a native speaker of said language.
My question is, how to achieve this?
I mean, how one compares the pronunciation between the user and the native speaker?
If you're looking for something relatively simple, you could simply compute the MFCC (http://en.wikipedia.org/wiki/Mel-frequency_cepstrum) of the recording, and then look at something simple like the correlation between the recording and the average coefficients of that word being pronounced by a native speaker. The MFCC will transform the audio into a space where euclidean distance corresponds more closely with perceptual difference.
Of course, there are several possible problems:
Aligning the two recordings so the coefficients match up. To fix this, you could look at the maximum cross-correlation of the coefficients, rather than the simple correlation, so you will get an automatic "best alignment" for free. Also, you might have to clip off ends of the recording, so only the actual pronunciation of the word remains in the recording.
The MFCC maps to perceptual space, but might not correspond so well to accent inaccuracies. You could perhaps try to fix this by instead of comparing it to just the "ideal" pronunciation, comparing it to the average for several different types of mispronunciation, and looking at which model it is closest to.
Even good accented words will be on average some "distance" from the ideal. You'll have to take that into account, and compare the input's distance to the "relative" good distance.
Correlation might not be the best way to compare the relative similarity of two sounds. Experiment with lots of different metrics... try different L^p norms: (http://en.wikipedia.org/wiki/Lp_space), or try weighing the different MFCCs differently (if I recall, even after MFCC have been taken, although they are all supposed to have the same perceptual "weight", the ones in the middle are still more important for how we perceive a sound than the high or low ones.)
There might be certain parts of the sound where the pronunciation matters much more for the quality of the accent. Perhaps transient detection to find those positions and mark them as more important would be good. If you had a whole bunch of "good pronunciation" and "bad pronunciation" examples, you could probably automatically extract those locations.
Again, in the end the only way you're going to know which combination of these options works best is by testing.
I've read about adapting gaussian mixture models for the phonetic space of a general speaker to an individual. This might be useful for training for a non-canonical accent for private use.
If you just compare the speaker to a general pronunciation model, then the match might not be very good. So the idea is to adjust the models to fit the speaker better during individual training.
Speaker Verification using Adapted Gaussian Mixture Models
EDIT: looking over your question again, I think I answered a different question. But the technique uses similar models:
Model various language (Do you have lots of data for different languages? Collecting the data might be the hard part.) GMMs work well for this.
Compare the data point from the speaker to the various language models
Choose the model that is the best predictor for the speaker data as the winner.
Related
My friend Prasad Raghavendra and me, were trying to experiment with Machine Learning on audio.
We were doing it to learn and to explore interesting possibilities at any upcoming get-togethers.
I decided to see how deep learning or any machine learning can be fed with certain audios rated by humans (evaluation).
To our dismay, we found that the problem had to be split to accommodate for the dimensionality of input.
So, we decided to discard vocals and assess by accompaniments with an assumption that vocals and instruments are always correlated.
We tried to look for mp3/wav to MIDI converter. Unfortunately, they were only for single instruments on SourceForge and Github and other options are paid options. (Ableton Live, Fruity Loops etc.) We decided to take this as a sub-problem.
We thought of FFT, band-pass filters and moving window to accommodate for these.
But, we are not understanding as to how we can go about splitting instruments if chords are played and there are 5-6 instruments in file.
What are the algorithms that I can look for?
My friend knows to play Keyboard. So, I will be able to get MIDI data. But, are there any data-sets meant for this?
How many instruments can these algorithms detect?
How do we split the audio? We do not have multiple audios or the mixing matrix
We were also thinking about finding out the patterns of accompaniments and using those accompaniments in real-time while singing along. I guess we will be able to think about it once we get answers to 1,2,3 and 4. (We are thinking about both Chord progressions and Markovian dynamics)
Thanks for all help!
P.S.: We also tried FFT and we are able to see some harmonics. Is it due to Sinc() in fft when rectangular wave is input in time domain? Can that be used to determine timbre?
We were able to formulate the problem roughly. But, still, we are finding it difficult to formulate the problem. If we use frequency domain for certain frequency, then the instruments are indistinguishable. A trombone playing at 440 Hz or a Guitar playing at 440 Hz would have same frequency excepting timbre. We still do not know how we can determine timbre. We decided to go by time domain by considering notes. If a note exceeds a certain octave, we would use that as a separate dimension +1 for next octave, 0 for current octave and -1 for the previous octave.
If notes are represented by letters such as 'A', 'B', 'C' etc, then the problem reduces to mixing matrices.
O = MI during training.
M is the mixing matrix that will have to be found out using the known O output and I input of MIDI file.
During prediction though, M must be replaced by a probability matrix P which would be generated using previous M matrices.
The problem reduces to Ipredicted = P-1O. The error would then be reduced to LMSE of I. We can use DNN to adjust P using back-propagation.
But, in this approach, we assume that the notes 'A','B','C' etc are known. How do we detect them instantaneously or in small duration like 0.1 seconds? Because, template matching may not work due to harmonics. Any suggestions would be much appreciated.
Splitting out the different parts is a machine learning problem all to its own. Unfortunately, you can't look at this problem in audio land only. You must consider the music.
You need to train something to understand musical patterns and progressions in the context of the type of music you give it. It needs to understand what the different instruments sound like, both mixed and not mixed. It needs to understand how these instruments are often played together, if it's going to have any chance at all at separating what's going on.
This is a very, very difficult problem.
This is a very hard problem mainly because converting audio to pitch isnt very simple due to Nyquist folding harmonics that are 22Khz+ back down and also other harmonic introductions such as saturators/distortion and other analogue equipment that introduce harmonics.
The fundamental harmonic isnt always the loudest which is why your plan will not work.
The hardest thing to measure would be a distorted guitar. The harmonic some pedals/plugins can make is crazy.
I am working on a personal research project.
My objective is to be able to recognize a sound and identify if it belongs to the IPA or not by comparing it's waveform to a wave form in my data base. I have some skill with Mathematica, SciPy, and PyBrain.
For the first phase, I'm only using the English (US) phonetic alphabet.
I have a simple test bank of English phonetic alphabet sound files I found online. The trick here is:
I want to separate a sound file into wave forms that correspond to different syllables- this will take a learning algorithm. So, 'I like apples' would be cut up into the syllable waveforms that would make up the sentence.
Each waveform is then compared against the English PA's wave forms. I'm not certain how to do this part. I was thinking of using Praat to detect the waveforms, capture the image of the wave form and compare it to the one stored in the database with image analysis (which is kind of fun to do).
The damage here, is that I don't know how to make Praat generate a wave form file automatically then cut it up between syllables into waveform chunks. Logically, I would just prepare test cases for a learning algorithm and teach the comp to do it.
Instead of needing a wave form image- could I do this with fast Fourier transformation and compare two fft's- within x% margin of error consider it y syllable?
Frankly I don't really know about Praat, But I find your project super cool and interesting. I have experience with car motor's fault detection using it's sound, which might be connected to your project. I used Neural Networks and SVM to do the classification because multiple research papers proved it. Thus I didn't have any doubt about the way I chose. So my advice is maybe you should research and read some Papers about it. It really helps when you have questions like this (Will it work?, Can I use it instead or Am I using optimal solution? etc...). And good luck that's an awesome project :)
You could try Praat scripting.
Using just FFT will give you rather terrible results. Very long feature vector that will be really difficult to segment and run any training on it. That's thousands of points for a single syllable. Some deep neural networks are able to cope with it, but that's assuming you design them properly and provide huge training set. The advantage of using neural networks is that they can build features for you from the "raw data" (and I would consider fft also "raw"). However, when you work with sound, it's not that badly needed - you can manually engineer features. In case of sounds, science knows very well what sort of "features" sound have.
You can calculate these features with libraries like Yaafe. I recommend checking it even if you are not doing it in C++ or Python - the link I provided also delivers formulas for calculating them. I used some of them in my kiwi classifier.
Another good approach comes from scikit-talkbox, which provides exactly the tooling you might need.
I plan to write a conversation analysis software, which will recognize the individual speakers, their pitch and intensity. Pitch and intensity are somewhat straightforward (pitch via autocorrelation).
How would I go about recognizing individual speakers, so I can record his/her features? Will storing some heuristics for each speaker's frequencies be enough? I can assume that only one person speaks at a time (strictly non-overlapping). I can also assume that for training, each speaker can record a minute's worth of data before actual analysis.
Pitch and intensity on their own tell you nothing. You really need to analyse how pitch varies. In order to identify different speakers you need to transform the speech audio into some kind of feature space, and then make comparisons against your database of speakers in this feature space. The general term that you might want to Google for is prosody - see e.g. http://en.wikipedia.org/wiki/Prosody_(linguistics). While you're Googling you might also want to read up on speaker identification aka speaker recognition, see e.g. http://en.wikipedia.org/wiki/Speaker_identification
If you are still working on this... are you using speech-recognition on the sound input? Because Microsoft SAPI for example provides the application with a rich API for digging into the speech sound wave, which could make the speaker-recognition problem more tractable. I think you can get phoneme positions within the waveform. That would let you do power-spectrum analysis of vowels, for example, which could be used to generate features to distinguish speakers. (Before anybody starts muttering about pitch and volume, keep in mind that the formant curves come from vocal-tract shape and are fairly independent of pitch, which is vocal-cord frequency, and the relative position and relative amplitude of formants are (relatively!) independent of overall volume.) Phoneme duration in-context might also be a useful feature. Energy distribution during 'n' sounds could provide a 'nasality' feature. And so on. Just a thought. I expect to be working in this area myself.
A developer I am working with is developing a program that analyzes images of pavement to find cracks in the pavement. For every crack his program finds, it produces an entry in a file that tells me which pixels make up that particular crack. There are two problems with his software though:
1) It produces several false positives
2) If he finds a crack, he only finds small sections of it and denotes those sections as being separate cracks.
My job is to write software that will read this data, analyze it, and tell the difference between false-positives and actual cracks. I also need to determine how to group together all the small sections of a crack as one.
I have tried various ways of filtering the data to eliminate false-positives, and have been using neural networks to a limited degree of success to group cracks together. I understand there will be error, but as of now, there is just too much error. Does anyone have any insight for a non-AI expert as to the best way to accomplish my task or learn more about it? What kinds of books should I read, or what kind of classes should I take?
EDIT My question is more about how to notice patterns in my coworker's data and identify those patterns as actual cracks. It's the higher-level logic that I'm concerned with, not so much the low-level logic.
EDIT In all actuality, it would take AT LEAST 20 sample images to give an accurate representation of the data I'm working with. It varies a lot. But I do have a sample here, here, and here. These images have already been processed by my coworker's process. The red, blue, and green data is what I have to classify (red stands for dark crack, blue stands for light crack, and green stands for a wide/sealed crack).
In addition to the useful comments about image processing, it also sounds like you're dealing with a clustering problem.
Clustering algorithms come from the machine learning literature, specifically unsupervised learning. As the name implies, the basic idea is to try to identify natural clusters of data points within some large set of data.
For example, the picture below shows how a clustering algorithm might group a bunch of points into 7 clusters (indicated by circles and color):
(source: natekohl.net)
In your case, a clustering algorithm would attempt to repeatedly merge small cracks to form larger cracks, until some stopping criteria is met. The end result would be a smaller set of joined cracks. Of course, cracks are a little different than two-dimensional points -- part of the trick in getting a clustering algorithm to work here will be defining a useful distance metric between two cracks.
Popular clustering algorithms include k-means clustering (demo) and hierarchical clustering. That second link also has a nice step-by-step explanation of how k-means works.
EDIT: This paper by some engineers at Phillips looks relevant to what you're trying to do:
Chenn-Jung Huang, Chua-Chin Wang, Chi-Feng Wu, "Image Processing Techniques for Wafer Defect Cluster Identification," IEEE Design and Test of Computers, vol. 19, no. 2, pp. 44-48, March/April, 2002.
They're doing a visual inspection for defects on silicon wafers, and use a median filter to remove noise before using a nearest-neighbor clustering algorithm to detect the defects.
Here are some related papers/books that they cite that might be useful:
M. Taubenlatt and J. Batchelder, “Patterned Wafer Inspection Using Spatial Filtering for Cluster Environment,” Applied Optics, vol. 31, no. 17, June 1992, pp. 3354-3362.
F.-L. Chen and S.-F. Liu, “A Neural-Network Approach to Recognize Defect Spatial Pattern in Semiconductor Fabrication.” IEEE Trans. Semiconductor Manufacturing, vol. 13, no. 3, Aug. 2000, pp. 366-373.
G. Earl, R. Johnsonbaugh, and S. Jost, Pattern Recognition and Image Analysis, Prentice Hall, Upper Saddle River, N.J., 1996.
Your problem falls in the very broad field of image classification. These types of problems can be notoriously difficult, and at the end of the day, solving them is an art. You must exploit every piece of knowledge you have about the problem domain to make it tractable.
One fundamental issue is normalization. You want to have similarly classified objects to be as similar as possible in their data representation. For example, if you have an image of the cracks, do all images have the same orientation? If not, then rotating the image may help in your classification. Similarly, scaling and translation (refer to this)
You also want to remove as much irrelevant data as possible from your training sets. Rather than directly working on the image, perhaps you could use edge extraction (for example Canny edge detection). This will remove all the 'noise' from the image, leaving only the edges. The exercise is then reduced to identifying which edges are the cracks and which are the natural pavement.
If you want to fast track to a solution then I suggest you first try the your luck with a Convolutional Neural Net, which can perform pretty good image classification with a minimum of preprocessing and noramlization. Its pretty well known in handwriting recognition, and might be just right for what you're doing.
I'm a bit confused by the way you've chosen to break down the problem. If your coworker isn't identifying complete cracks, and that's the spec, then that makes it your problem. But if you manage to stitch all the cracks together, and avoid his false positives, then haven't you just done his job?
That aside, I think this is an edge detection problem rather than a classification problem. If the edge detector is good, then your issues go away.
If you are still set on classification, then you are going to need a training set with known answers, since you need a way to quantify what differentiates a false positive from a real crack. However I still think it is unlikely that your classifier will be able to connect the cracks, since these are specific to each individual paving slab.
I have to agree with ire_and_curses, once you dive into the realm of edge detection to patch your co-developers crack detection, and remove his false positives, it seems as if you would be doing his job. If you can patch what his software did not detect, and remove his false positives around what he has given you. It seems like you would be able to do this for the full image.
If the spec is for him to detect the cracks, and you classify them, then it's his job to do the edge detection and remove false positives. And your job to take what he has given you and classify what type of crack it is. If you have to do edge detection to do that, then it sounds like you are not far from putting your co-developer out of work.
There are some very good answers here. But if you are unable to solve the problem, you may consider Mechanical Turk. In some cases it can be very cost-effective for stubborn problems. I know people who use it for all kinds of things like this (verification that a human can do easily but proves hard to code).
https://www.mturk.com/mturk/welcome
I am no expert by any means, but try looking at Haar Cascades. You may also wish to experiment with the OpenCV toolkit. These two things together do face detection and other object-detection tasks.
You may have to do "training" to develop a Haar Cascade for cracks in pavement.
What’s the best approach to recognize patterns in data, and what’s the best way to learn more on the topic?
The best approach is to study pattern recognition and machine learning. I would start with Duda's Pattern Classification and use Bishop's Pattern Recognition and Machine Learning as reference. It would take a good while for the material to sink in, but getting basic sense of pattern recognition and major approaches of classification problem should give you the direction. I can sit here and make some assumptions about your data, but honestly you probably have the best idea about the data set since you've been dealing with it more than anyone. Some of the useful technique for instance could be support vector machine and boosting.
Edit: An interesting application of boosting is real-time face detection. See Viola/Jones's Rapid Object Detection using a Boosted Cascade of Simple
Features (pdf). Also, looking at the sample images, I'd say you should try improving the edge detection a bit. Maybe smoothing the image with Gaussian and running more aggressive edge detection can increase detection of smaller cracks.
I suggest you pick up any image processing textbook and read on the subject.
Particularly, you might be interested in Morphological Operations like Dilation and Erosion, which complements the job of an edge detector. Plenty of materials on the net...
This is an image processing problem. There are lots of books written on the subject, and much of the material in these books will go beyond a line-detection problem like this. Here is the outline of one technique that would work for the problem.
When you find a crack, you find some pixels that make up the crack. Edge detection filters or other edge detection methods can be used for this.
Start with one (any) pixel in a crack, then "follow" it to make a multipoint line out of the crack -- save the points that make up the line. You can remove some intermediate points if they lie close to a straight line. Do this with all the crack pixels. If you have a star-shaped crack, don't worry about it. Just follow the pixels in one (or two) directions to make up a line, then remove these pixels from the set of crack pixels. The other legs of the star will recognized as separate lines (for now).
You might perform some thinning on the crack pixels before step 1. In other words, check the neighbors of the pixels, and if there are too many then ignore that pixel. (This is a simplification -- you can find several algorithms for this.) Another preprocessing step might be to remove all the lines that are too thin or two faint. This might help with the false positives.
Now you have a lot of short, multipoint lines. For the endpoints of each line, find the nearest line. If the lines are within a tolerance, then "connect" the lines -- link them or add them to the same structure or array. This way, you can connect the close cracks, which would likely be the same crack in the concrete.
It seems like no matter the algorithm, some parameter adjustment will be necessary for good performance. Write it so it's easy to make minor changes in things like intensity thresholds, minimum and maximum thickness, etc.
Depending on the usage environment, you might want to allow user judgement do determine the questionable cases, and/or allow a user to review the all the cracks and click to combine, split or remove detected cracks.
You got some very good answer, esp. #Nate's, and all the links and books suggested are worthwhile. However, I'm surprised nobody suggested the one book that would have been my top pick -- O'Reilly's Programming Collective Intelligence. The title may not seem germane to your question, but, believe me, the contents are: one of the most practical, programmer-oriented coverage of data mining and "machine learning" I've ever seen. Give it a spin!-)
It sounds a little like a problem there is in Rock Mechanics, where there are joints in a rock mass and these joints have to be grouped into 'sets' by orientation, length and other properties. In this instance one method that works well is clustering, although classical K-means does seem to have a few problems which I have addressed in the past using a genetic algorithm to run the interative solution.
In this instance I suspect it might not work quite the same way. In this case I suspect that you need to create your groups to start with i.e. longitudinal, transverse etc. and define exactly what the behviour of each group is i.e. can a single longitudinal crack branch part way along it's length, and if it does what does that do to it's classification.
Once you have that then for each crack, I would generate a random crack or pattern of cracks based on the classification you have created. You can then use something like a least squares approach to see how closely the crack you are checking fits against the random crack / cracks you have generated. You can repeat this analysis many times in the manner of a Monte-Carlo analysis to identify which of the randomly generated crack / cracks best fits the one you are checking.
To then deal with the false positives you will need to create a pattern for each of the different types of false positives i.e. the edge of a kerb is a straight line. You will then be able to run the analysis picking out which is the most likely group for each crack you analyse.
Finally, you will need to 'tweak' the definition of different crack types to try and get a better result. I guess this could either use an automated approach or a manual approach depending on how you define your different crack types.
One other modification that sometimes helps when I'm doing problems like this is to have a random group. By tweaking the sensitivity of a random group i.e. how more or less likely a crack is to be included in the random group, you can sometimes adjust the sensitivty of the model to complex patterns that don't really fit anywhere.
Good luck, looks to me like you have a real challenge.
You should read about data mining, specially pattern mining.
Data mining is the process of extracting patterns from data. As more data are gathered, with the amount of data doubling every three years, data mining is becoming an increasingly important tool to transform these data into information. It is commonly used in a wide range of profiling practices, such as marketing, surveillance, fraud detection and scientific discovery.
A good book on the subject is Data Mining: Practical Machine Learning Tools and Techniques
(source: waikato.ac.nz) ](http://www.amazon.com/Data-Mining-Ian-H-Witten/dp/3446215336 "ISBN 0-12-088407-0")
Basically what you have to do is apply statistical tools and methodologies to your datasets. The most used comparison methodologies are Student's t-test and the Chi squared test, to see if two unrelated variables are related with some confidence.
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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 pitches of the notes of the song. Any idea/suggestion on how to go about it?
This is exactly what I'm doing here as my last year project :) except one thing that my project is about tracking the pitch of human singing voice (and I don't have the robot to play the tune)
The quickest way I can think of is to utilize BASS library. It contains ready-to-use function that can give you FFT data from default recording device. Take a look at "livespec" code example that comes with BASS.
By the way, raw FFT data will not enough to determine fundamental frequency. You need algorithm such as Harmonic Product Spectrum to get the F0.
Another consideration is the audio source. If you are going to do FFT and apply Harmonic Product Spectrum on it. You will need to make sure the input has only one audio source. If it contains multiple sources such as in modern songs there will be to many frequencies to consider.
Harmonic Product Spectrum Theory
If the input signal is a musical note,
then its spectrum should consist of a
series of peaks, corresponding to
fundamental frequency with harmonic
components at integer multiples of the
fundamental frequency. Hence when we
compress the spectrum a number of
times (downsampling), and compare it
with the original spectrum, we can see
that the strongest harmonic peaks line
up. The first peak in the original
spectrum coincides with the second
peak in the spectrum compressed by a
factor of two, which coincides with
the third peak in the spectrum
compressed by a factor of three.
Hence, when the various spectrums are
multiplied together, the result will
form clear peak at the fundamental
frequency.
Method
First, we divide the input signal into
segments by applying a Hanning window,
where the window size and hop size are
given as an input. For each window,
we utilize the Short-Time Fourier
Transform to convert the input signal
from the time domain to the frequency
domain. Once the input is in the
frequency domain, we apply the
Harmonic Product Spectrum technique to
each window.
The HPS involves two steps:
downsampling and multiplication. To
downsample, we compressed the spectrum
twice in each window by resampling:
the first time, we compress the
original spectrum by two and the
second time, by three. Once this is
completed, we multiply the three
spectra together and find the
frequency that corresponds to the peak
(maximum value). This particular
frequency represents the fundamental
frequency of that particular window.
Limitations of the HPS method
Some nice features of this method
include: it is computationally
inexpensive, reasonably resistant to
additive and multiplicative noise, and
adjustable to different kind of
inputs. For instance, we could change
the number of compressed spectra to
use, and we could replace the spectral
multiplication with a spectral
addition. However, since human pitch
perception is basically logarithmic,
this means that low pitches may be
tracked less accurately than high
pitches.
Another severe shortfall of the HPS
method is that it its resolution is
only as good as the length of the FFT
used to calculate the spectrum. If we
perform a short and fast FFT, we are
limited in the number of discrete
frequencies we can consider. In order
to gain a higher resolution in our
output (and therefore see less
graininess in our pitch output), we
need to take a longer FFT which
requires more time.
from: http://cnx.org/content/m11714/latest/
Just a comment: The fundamental harmonic may as well be missing from a (harmonic) sound, this doesn't change the perceived pitch. As a limit case, if you take a square wave (say, a C# note) and completely suppress the first harmonic, the perceived note is still C#, in the same octave. In a way, our brain is able to compensate the absence of some harmonics, even the first, when it guesses a note.
Hence, to detect a pitch with frequency-domain techniques you should take into account all the harmonics (local maxima in the magnitude of the Fourier transform), and extract some sort of "greatest common divisor" of their frequencies. Pitch detection is not a trivial problem at all...
DAFX has about 30 pages dedicated to pitch detection, with examples and Matlab code.
Autocorrelation - http://en.wikipedia.org/wiki/Autocorrelation
Zero-crossing - http://en.wikipedia.org/wiki/Zero_crossing (this method is used in cheap guitar tuners)
Try YAAPT pitch tracking, which detects fundamental frequency in both time and frequency domains. You can download Matlab source code from the link and look for peaks in the FFT output using the spectral process part.
Python package http://bjbschmitt.github.io/AMFM_decompy/pYAAPT.html#
Did you try Wikipedia's article on pitch detection? It contains a few references that can be interesting to you.
In addition, here's a list of DSP applications and libraries, where you can poke around. The list only mentions Linux software packages, but many of them are cross-platform, and there's a lot of source code you can look at.
Just FYI, detecting the pitch of the notes in a monophonic recording is within reach of most DSP-savvy people. Detecting the pitches of all notes, including chords and stuff, is a lot harder.
Just a thought - but do you need to process a digital audio stream as input?
If not, consider using a symbolic representation of music (such as MIDI). The pitches of the notes will then be stated explicitly, and you can synthesize sounds (and movements) corresponding to the pitch, rhythm and many other musical parameters extremely easily.
If you need to analyse a digital audio stream (mp3, wav, live input, etc) bear in mind that while pitch detection of simple monophonic sounds is quite advanced, polyphonic pitch detection is an unsolved problem. In this case, you may find my answer to this question helpful.
For extracting the fundamental frequency of the melody from polyphonic music you could try the MELODIA plug-in: http://mtg.upf.edu/technologies/melodia
Extracting the F0's of all the instruments in a song (multi-F0 tracking) or transcribing them into notes is an even harder task. Both melody extraction and music transcription are still open research problems, so regardless of the algorithm/tool you use don't expect to obtain perfect results for either.
If you're trying to detect the notes of a polyphonic recording (multiple notes at the same time) good luck. That's a very tricky problem. I don't know of any way to listen to, say, a recording of a string quartet and have an algorithm separate the four voices. (Wavelets maybe?) If it's just one note at a time, there are several pitch tracking algorithms out there, many of them mentioned in other comments.
The algorithm you want to use will depend on the type of music you are listening to. If you want it to pick up people singing there are a lot of good algorithms out there designed specifically for voice. (That's where most of the research is.) If you are trying to pick up specific instruments you'll have to be a bit more creative. Voice algorithms can be simple because the range of the human singing voice is generally limited to about 100-2000 Hz. (Speaking range is much more narrow). The fundamental frequencies on a piano, however, go from about 27 Hz. to 4200 Hz., so you're dealing with a wider range usually ignored by voice pitch detection algorithms.
The waveform of most instruments is going to be fairly complex, with lots of harmonics, so a simple approach like counting zeros or just taking the autocorrelation won't work. If you knew roughly what frequency range you were looking in you could low-pass filter and then zero count. I'd think you'd be better off though with a more complex algorithm such as the Harmonic Product Spectrum mentioned by another user, or YAAPT ("Yet Another Algorithm for Pitch Tracking"), or something similar.
One last problem: some instruments, the piano in particular, will have the problem of missing fundamentals and inharmonicity. Missing fundamentals can be dealt with by the pitch tracking algorithms...in fact they have to be since fundamentals are often cut out in electronic transmission...though you'll probably still get some octave errors. Inharmonicity however, will give you problems if somebody plays a note in the bottom octaves of the piano. Normal pitch tracking algorithms aren't designed to deal with inharmonicity because the human voice is not significantly inharmonic.
You basically need a spectrum analyzer. You might be able to to a FFT on a recording of an analog input, but much depends on the resolution of the recording.
what immediately comes to my mind:
filter out very low frequencies (drums, bass-line),
filter out high frequencies (harmonics)
FFT,
look for peaks in the FFT output for the melody
I am not sure, if that works for very polyphonic sounds - maybe googling for "FFT, analysis, melody etc." will return more info on possible problems.
regards