I'm trying to analyse audio and visual features in tandem. My audio speech features are mel-frequency cepstrum co-efficients sampled at 100fps using the Hidden Markov Model Toolkit. My visual features come from a lip-tracking programme I built and are sampled at 29.97fps.
I know that I need to interpolate my visual features so that the sample rate is also 100fps, but I can't find a nice explanation or tutorial on how to do this online. Most of the help I have found comes from the speech recognition community which assumes a knowledge of interpolation on behalf of the reader, i.e. most cover the step with a simple "interpolate the visual features so that the sample rate equals 100fps".
Can anyone pooint me in the right direction?
Thanks a million
Since face movement is not low-pass filtered prior to video capture, most of the classic DSP interpolation methods may not apply. You might as well try linear interpolation of your features vectors to get from one set of time points to a set at a different set of time points. Just pick the 2 closest video frames and interpolate to get more data points in between. You could also try spline interpolation if your facial tracking algorithm measures accelerations in face motion.
Related
i am doing final year project as lane tracking using a camera. the most challenging task now is how i can measure distance between the camera (the car that carries it actually) and the lane.
While the lane is easily recognized (Hough line transform) but i found no way to measure distance to it.
given the fact that there is a way to measure distance to object in front of camera based on Pixel width of the object, but it does not work here be because the nearest point of the line, is blind in the camera.
What you want is to directly infer the depth map with a monocular camera.
You can refer my answer here
https://stackoverflow.com/a/64687551/11530294
Usually, we need a photometric measurement from a different position in the world to form a geometric understanding of the world(a.k.a depth map). For a single image, it is not possible to measure the geometric, but it is possible to infer depth from prior understanding.
One way for a single image to work is to use a deep learning-based method to direct infer depth. Usually, the deep learning-based approaches are all based on python, so if you only familiar with python, then this is the approach that you should go for. If the image is small enough, i think it is possible for realtime performance. There are many of this kind of work using CAFFE, TF, TORCH etc. you can search on git hub for more option. The one I posted here is what i used recently
reference:
Godard, Clément, et al. "Digging into self-supervised monocular depth estimation." Proceedings of the IEEE international conference on computer vision. 2019.
Source code: https://github.com/nianticlabs/monodepth2
The other way is to use a large FOV video for a single camera-based SLAM. This one has various constraints such as need good features, large FOV, slow motion, etc. You can find many of this work such as DTAM, LSDSLAM, DSO, etc. There are a couple of other packages from HKUST or ETH that does the mapping given the position(e.g if you have GPS/compass), some of the famous names are REMODE+SVO open_quadtree_mapping etc.
One typical example for a single camera-based SLAM would be LSDSLAM. It is a realtime SLAM.
This one is implemented based on ROS-C++, I remember they do publish the depth image. And you can write a python node to subscribe to the depth directly or the global optimized point cloud and project it into a depth map of any view angle.
reference: Engel, Jakob, Thomas Schöps, and Daniel Cremers. "LSD-SLAM: Large-scale direct monocular SLAM." European conference on computer vision. Springer, Cham, 2014.
source code: https://github.com/tum-vision/lsd_slam
I want to do pedestrian detection and tracking.
Input: Video Stream from CCTV camera.
Output:
#(no of) people going from left to right
# people going from right to left
# No. of people in the middle
What have i done so far:
For pedestrian detection I am using HOG and SVM. The detection is decent with high false positive rate. And its very slow as i am running in android platform.
Question:
After detection how to do I calculate the required values listed above. Can anyone tell me what is the tracking algorithm I have to use and any good algorithm for pedestrian detection.
Or should I use tracking algorithm? Is there a way to do without it?
Any references to codes/blogs/technical papers is appreciated.
Platform: C++ & OpenCV / android.
--Thanks
This is somehow close to a research problem.
You may want to have a look to this website which gathers a lot of references.
In particular, the work done by the group from Oxford present therein is pretty close to what you are doing, since their are using HOG for detection. (That work has been extremely illuminating for me).
EPFL and Julich have as well work done in the field.
You may also want to give a look to this review which describes several detection/tracking techniques, often involving variants of the HOG algorithm.
Along with #Acorbe response, I suggest the publications section of this (archived) website.
A recent work at the end of last year also released a code base here:
https://bitbucket.org/rodrigob/doppia
There have also been earlier pedestrian detector works that have released code as well:
https://sites.google.com/site/wujx2001/home/c4
http://www.vision.caltech.edu/Image_Datasets/CaltechPedestrians
The best accurate way is to use tracking algorithm instead of statistic appearance counting of incoming people and detection occurred left right and middle..
You can use extended statistical models.. That produce how many inputs producing one of the outputs and back validate from output detection the input.
My experience is that tracking leads to better results than approach above. But is also little bit complicated. We talk about multi target tracking when the critical is match detection with tracked model which should be update based on detection. If tracking is matched with wrong model. The problems are there.
Here on youtube I developed some multi target tracker by simple LBP people detector, but multi model and kalman filter for tracking. Both capabilities are available in opencv. You need to when something is detected create new kalman filter for each object and update in case you match same detection. Predict in case detection is not here in frame and also remove the Kalman i it is not necessary to track any more.
1 Detect
2 Match detections with kalmans, hungarian algorithm and l2 norm. (for example)
3 Lot of work. Decide if kalman shoudl be established, remove, update, or results is not detected and should be predicted. This is lot of work here.
Pure statistic approach is less accurate, second one is for experience people at least one moth of coding and 3 month of tuning.. If you need to be faster and your resources are quite limited. You can by smart statistic achieve your results by pure detection much faster and little bit less accurate. People are judge the image and video tracking even multi target tracking is capable to beat human. Try to count and register each person in video and count exits point. You are not able to do this in some number of people. It is really repents on, what you want, application, customer you have, and results you show to customers. If this is 4 numbers income, left, right, middle and your error is 20 percent is still much more than one bored small paid guard should achieved by all day long counting..
https://www.youtube.com/watch?v=d-RCKfVjFI4
You can find on my BLOG Some dataset for people detection and car detection on my blog same as script for learning ideas, tutorials and tracking examples..
Opencv blog tutorials code and ideas
You can use KLT for this purpose as this will tell you the flow of person traveling from left to right then you can compute that by computing line length which in given example is drawn using cv2.line you can use input parameters of this functions to compute your case, little math involved. if there is a flow of pixels from left to right this is case 1 or right to left then case 3 and for no flow case 2. Or you can use this basic tutorial to track object movement. LINK
for my final thesis i am trying to build up an 3d face recognition system by combining color and depth information. the first step i did, is to realign the data-head to an given model-head using the iterative closest point algorithm. for the detection step i was thinking about using the libsvm. but i dont understand how to combine the depth and the color information to one feature vector? they are dependent information (each point consist of color (RGB), depth information and also scan quality).. what do you suggest to do? something like weighting?
edit:
last night i read an article about SURF/SIFT features i would like to use them! could it work? the concept would be the following: extracting this features out of the color image and the depth image (range image), using each feature as a single feature vector for the svm?
Concatenation is indeed a possibility. However, as you are working on 3d face recognition you should have some strategy as to how you go about it. Rotation and translation of faces will be hard to recognize using a "straightforward" approach.
You should decide whether you attempt to perform a detection of the face as a whole, or of sub-features. You could attempt to detect rotation by finding some core features (eyes, nose, etc).
Also, remember that SVMs are inherently binary (i.e. they separate between two classes). Depending on your exact application you will very likely have to employ some multi-class strategy (One-against-all or One-against-many).
I would recommend doing some literature research to see how others have attacked the problem (a google search will be a good start).
It sounds simple, but you can simply concatenate the two vectors into one. Many researchers do this.
What you arrived at is an important open problem. Yes, there are some ways to handle it, as mentioned here by Eamorr. For example you can concatenate and do PCA (or some non linear dimensionality reduction method). But it is kind of hard to defend the practicality of doing so, considering that PCA takes O(n^3) time in the number of features. This alone might be unreasonable for data in vision that may have thousands of features.
As mentioned by others, the easiest approach is to simply combine the two sets of features into one.
SVM is characterized by the normal to the maximum-margin hyperplane, where its components specify the weights/importance of the features, such that higher absolute values have a larger impact on the decision function. Thus SVM assigns weights to each feature all on its own.
In order for this to work, obviously you would have to normalize all the attributes to have the same scale (say transform all features to be in the range [-1,1] or [0,1])
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.
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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