I am trying to get started with ROI encoding with the Nvidia Encoder NVENC.
As a first step I am trying to get the Nvidia demos to encode using ROI. I know that the switch -qpDeltaMapFile enables the flag enableExtQPDeltaMap. This allows me to send a file with a qp map that the encoder uses to tweak the values obtained by the rate control algorithm.
However there is absolutely no documentation on the format of this file. I tried to use one value per byte, and one byte per value assuming fixed size macroblocks of 16x16. It doesn't seem to work as I would expect.
Any guidance or references would help a lot.
There was a bug in my code. It actually works almost as I described.
Assume your screen is divided equally in 16x16 blocks, then each value will be added to the qp that the rate control algorithm chose. Each value passed is a signed integer, therefore a negative value will improve the quality while a positive value will decrease it. A value of 0 will stay with whatever the rate control algorithm decided.
Related
I would like to know how one could generate a wavetable out of a wav file for example.
I know a wavetable can be used in web audio api with setPerdiodic wave and I know how to use it.
But what do I need to do to create my own wavetables? I read about inverse FFT, but I did find nearly nothing. I don't need any code just an idea or a formula of how to get the wavetable from an wav file to a Buffer.
There are a few constraints here and I'm not sure how good the result will be.
Your wav file source can't be too long; the PeriodicWave object
only supports arrays up to size 8192 or so.
I'm going to assume your waveform is intended to be periodic. If the
last sample and the first aren't reasonably close to each other,
there will be a hard-to-reproduce jump.
The waveform must have zero mean, so if it doesn't you should remove
the mean.
With that taken care of, select a power of two greater than the length
of your wave file (not strictly needed, but most FFTs expect powers of
two). Zero-pad the wave file if the length is not a power of two.
Then compute the the FFT. You'll either get an array of complex
numbers or two arrays. Separate these out to real and imaginary
arrays and use them for contructing the PeriodicWave.
I'm trying to read variable length 1-D inputs into a Tensorflow CNN.
I have previously implemented reading fixed length inputs by first constructing a CSV file (where the first column is the label and the remaining columns are the input values - flattened spectrogram data all padded/truncated to the same length) using tf.TextLineReader().
This time I have a directory full of files each one containing a line of data I want to use as input (flattened spectrogram data again but I do not want to force them to the same dimensions), and the line lengths are not fixed. I'm getting an error trying to use the previous approach of compiling a CSV first. I looked into the documentation of tf.TextLineReader() and it specifies that all CSV rows must be the same shape, so I am stuck! Any help would be much appreciated, thanks :)
I'm assuming that the data isn't changing shape when you have a longer or shorter sample right? By that I mean that if you trained your network on arrays of 1000 pixels for example, with a kernel of say [5,1] size. That [5,1] kernel needs to see the same patterns in the variable length data as it did in the training data. If your data is stretched or shrunk, then the correct solution is to interpolate the data to the same size as the training data so the shapes/patterns match.
Assuming you just want variable length inputs, then in theory you should be able to do this by setting your batch size to 1 and varying the 1st dimension of the data.
So your input placeholder would look like:
X = tf.placeholder(dtype, shape=[1,None,1,1])
The 4 shape arguments are: 1=batch size; None=unknown first dimension size; 1=unused because it's a 1D dataset, 1=one channel images, again unused but necessary for tf.conv2d to receive the expected 4D image.
This is not very different from configuring tensorflow to support variable batch sizes. So you should review this link below and understand that process.
get the size of a variable batch dimension
Note that you can't use a batch size more than 1 here because you wouldn't be able to construct a matrix with missing values in the 2nd dimension. I expect the convolution operations to work with this variable dimension (though I haven't actually tried this).
Another option to deal with this problem would be to pad your inputs with 0's so they all have a common length, but that will need to have been trained into the model up front.
I want to fit a curve to data obtained from an FFT. While working on this, I remembered that an FFT gives binned data, and therefore I wondered if I should treat this differently with curve-fitting.
If the bins are narrow compared to the structure, I think it should not be necessary to treat the data differently, but for me that is not the case.
I expect the right way to fit binned data is by minimizing not the difference between values of the bin and fit, but between bin area and the area beneath the fitted curve, for each bin, such that the energy in each bin matches the energy in the range of the bin as signified by the curve.
So my question is: am I thinking correctly about this? If not, how should I go about it?
Also, when looking around for information about this subject, I encountered the "Maximum log likelihood" for example, but did not find enough information about it to understand if and how it applied to my situation.
PS: I have no clue if this is the right site for this question, please let me know if there is a better place.
For an unwindowed FFT, the correct interpolation between bins is by using a Sinc (sin(x)/x) or periodic Sinc (Dirichlet) interpolation kernel. For an FFT of samples of a band-limited signal, thus will reconstruct the continuous spectrum.
A very simple and effective way of interpolating the spectrum (from an FFT) is to use zero-padding. It works both with and without windowing prior to the FFT.
Take your input vector of length N and extend it to length M*N, where M is an integer
Set all values beyond the original N values to zeros
Perform an FFT of length (N*M)
Calculate the magnitude of the ouput bins
What you get is the interpolated spectrum.
Best regards,
Jens
This can be done by using maximum log likelihood estimation. This is a method that finds the set of parameters that is most likely to have yielded the measured data - the technique originates in statistics.
I have finally found an understandable source for how to apply this to binned data. Sadly I cannot enter formulas here, so I refer to that source for a full explanation: slide 4 of this slide show.
EDIT:
For noisier signals this method did not seem to work very well. A method that was a bit more robust is a least squares fit, where the difference between the area is minimized, as suggested in the question.
I have not found any literature to defend this method, but it is similar to what happens in the maximum log likelihood estimation, and yields very similar results for noiseless test cases.
The documentation for both of these methods are both very generic wherever I look. I would like to know what exactly I'm looking at with the returned arrays I'm getting from each method.
For getByteTimeDomainData, what time period is covered with each pass? I believe most oscopes cover a 32 millisecond span for each pass. Is that what is covered here as well? For the actual element values themselves, the range seems to be 0 - 255. Is this equivalent to -1 - +1 volts?
For getByteFrequencyData the frequencies covered is based on the sampling rate, so each index is an actual frequency, but what about the actual element values themselves? Is there a dB range that is equivalent to the values returned in the returned array?
getByteTimeDomainData (and the newer getFloatTimeDomainData) return an array of the size you requested - its frequencyBinCount, which is calculated as half of the requested fftSize. That array is, of course, at the current sampleRate exposed on the AudioContext, so if it's the default 2048 fftSize, frequencyBinCount will be 1024, and if your device is running at 44.1kHz, that will equate to around 23ms of data.
The byte values do range between 0-255, and yes, that maps to -1 to +1, so 128 is zero. (It's not volts, but full-range unitless values.)
If you use getFloatFrequencyData, the values returned are in dB; if you use the Byte version, the values are mapped based on minDecibels/maxDecibels (see the minDecibels/maxDecibels description).
Mozilla 's documentation describes the difference between getFloatTimeDomainData and getFloatFrequencyData, which I summarize below. Mozilla docs reference the Web Audio
experiment ; the voice-change-o-matic. The voice-change-o-matic illustrates the conceptual difference to me (it only works in my Firefox browser; it does not work in my Chrome browser).
TimeDomain/getFloatTimeDomainData
TimeDomain functions are over some span of time.
We often visualize TimeDomain data using oscilloscopes.
In other words:
we visualize TimeDomain data with a line chart,
where the x-axis (aka the "original domain") is time
and the y axis is a measure of a signal (aka the "amplitude").
Change the voice-change-o-matic "visualizer setting" to Sinewave to
see getFloatTimeDomainData(...)
Frequency/getFloatFrequencyData
Frequency functions (GetByteFrequencyData) are at a point in time; i.e. right now; "the current frequency data"
We sometimes see these in mp3 players/ "winamp bargraph style" music players (aka "equalizer" visualizations).
In other words:
we visualize Frequency data with a bar graph
where the x-axis (aka "domain") are frequencies or frequency bands
and the y-axis is the strength of each frequency band
Change the voice-change-o-matic "visualizer setting" to Frequency bars to see getFloatFrequencyData(...)
Fourier Transform (aka Fast Fourier Transform/FFT)
Another way to think about "time domain vs frequency" is shown the diagram below, from Fast Fourier Transform wikipedia
getFloatTimeDomainData gives you the chart on on the top (x-axis is Time)
getFloatFrequencyData gives you the chart on the bottom (x-axis is Frequency)
a Fast Fourier Transform (FFT) converts the Time Domain data into Frequency data, in other words, FFT converts the first chart to the second chart.
cwilso has it backwards.
the time data array is the longer one (fftSize), and the frequency data array is the shorter one (half that, frequencyBinCount).
fftSize of 2048 at the usual sample rate of 44.1kHz means each sample has 1/44100 duration, you have 2048 samples at hand, and thus are covering a duration of 2048/44100 seconds, which 46 milliseconds, not 23 milliseconds. The frequencyBinCount is indeed 1024, but that refers to the frequency domain (as the name suggests), not the time domain, and it the computation 1024/44100, in this context, is about as meaningful as adding your birth date to the fftSize.
A little math illustrating what's happening: Fourier transform is a 'vector space isomorphism', that is, a mapping going bijectively (i.e., reversible) between 2 vector spaces of the same dimension; the 'time domain' and the 'frequency domain.' The vector space dimension we have here (in both cases) is fftSize.
So where does the 'half' come from? The frequency domain coefficients 'count double'. Either because they 'actually are' complex numbers, or because you have the 'sin' and the 'cos' flavor. Or, because you have a 'magnitude' and a 'phase', which you'll understand if you know how complex numbers work. (Those are 3 ways to say the same in a different jargon, so to speak.)
I don't know why the API only gives us half of the relevant numbers when it comes to frequency - I can only guess. And my guess is that those are the 'magnitude' numbers, and the 'phase' numbers are thrown out. The reason that this is my guess is that in applications, magnitude is far more important than phase. Still, I'm quite surprised that the API throws out information, and I'd be glad if some expert who actually knows (and isn't guessing) can confirm that it's indeed the magnitude. Or - even better (I love to learn) - correct me.
I have a series of data and need to detect peak values in the series within a certain number of readings (window size) and excluding a certain level of background "noise." I also need to capture the starting and stopping points of the appreciable curves (ie, when it starts ticking up and then when it stops ticking down).
The data are high precision floats.
Here's a quick sketch that captures the most common scenarios that I'm up against visually:
One method I attempted was to pass a window of size X along the curve going backwards to detect the peaks. It started off working well, but I missed a lot of conditions initially not anticipated. Another method I started to work out was a growing window that would discover the longer duration curves. Yet another approach used a more calculus based approach that watches for some velocity / gradient aspects. None seemed to hit the sweet spot, probably due to my lack of experience in statistical analysis.
Perhaps I need to use some kind of a statistical analysis package to cover my bases vs writing my own algorithm? Or would there be an efficient method for tackling this directly with SQL with some kind of local max techniques? I'm simply not sure how to approach this efficiently. Each method I try it seems that I keep missing various thresholds, detecting too many peak values or not capturing entire events (reporting a peak datapoint too early in the reading process).
Ultimately this is implemented in Ruby and so if you could advise as to the most efficient and correct way to approach this problem with Ruby that would be appreciated, however I'm open to a language agnostic algorithmic approach as well. Or is there a certain library that would address the various issues I'm up against in this scenario of detecting the maximum peaks?
my idea is simple, after get your windows of interest you will need find all the peaks in this window, you can just compare the last value with the next , after this you will have where the peaks occur and you can decide where are the best peak.
I wrote one simple source in matlab to show my idea!
My example are in wave from audio file :-)
waveFile='Chick_eco.wav';
[y, fs, nbits]=wavread(waveFile);
subplot(2,2,1); plot(y); legend('Original signal');
startIndex=15000;
WindowSize=100;
endIndex=startIndex+WindowSize-1;
frame = y(startIndex:endIndex);
nframe=length(frame)
%find the peaks
peaks = zeros(nframe,1);
k=3;
while(k <= nframe - 1)
y1 = frame(k - 1);
y2 = frame(k);
y3 = frame(k + 1);
if (y2 > 0)
if (y2 > y1 && y2 >= y3)
peaks(k)=frame(k);
end
end
k=k+1;
end
peaks2=peaks;
peaks2(peaks2<=0)=nan;
subplot(2,2,2); plot(frame); legend('Get Window Length = 100');
subplot(2,2,3); plot(peaks); legend('Where are the PEAKS');
subplot(2,2,4); plot(frame); legend('Peaks in the Window');
hold on; plot(peaks2, '*');
for j = 1 : nframe
if (peaks(j) > 0)
fprintf('Local=%i\n', j);
fprintf('Value=%i\n', peaks(j));
end
end
%Where the Local Maxima occur
[maxivalue, maxi]=max(peaks)
you can see all the peaks and where it occurs
Local=37
Value=3.266296e-001
Local=51
Value=4.333496e-002
Local=65
Value=5.049438e-001
Local=80
Value=4.286804e-001
Local=84
Value=3.110046e-001
I'll propose a couple of different ideas. One is to use discrete wavelets, the other is to use the geographer's concept of prominence.
Wavelets: Apply some sort of wavelet decomposition to your data. There are multiple choices, with Daubechies wavelets being the most widely used. You want the low frequency peaks. Zero out the high frequency wavelet elements, reconstruct your data, and look for local extrema.
Prominence: Those noisy peaks and valleys are of key interest to geographers. They want to know exactly which of a mountain's multiple little peaks is tallest, the exact location of the lowest point in the valley. Find the local minima and maxima in your data set. You should have a sequence of min/max/min/max/.../min. (You might want to add an arbitrary end points that are lower than your global minimum.) Consider a min/max/min sequence. Classify each of these triples per the difference between the max and the larger of the two minima. Make a reduced sequence that replaces the smallest of these triples with the smaller of the two minima. Iterate until you get down to a single min/max/min triple. In your example, you want the next layer down, the min/max/min/max/min sequence.
Note: I'm going to describe the algorithmic steps as if each pass were distinct. Obviously, in a specific implementation, you can combine steps where it makes sense for your application. For the purposes of my explanation, it makes the text a little more clear.
I'm going to make some assumptions about your problem:
The windows of interest (the signals that you are looking for) cover a fraction of the entire data space (i.e., it's not one long signal).
The windows have significant scope (i.e., they aren't one pixel wide on your picture).
The windows have a minimum peak of interest (i.e., even if the signal exceeds the background noise, the peak must have an additional signal excess of the background).
The windows will never overlap (i.e., each can be examined as a distinct sub-problem out of context of the rest of the signal).
Given those, you can first look through your data stream for a set of windows of interest. You can do this by making a first pass through the data: moving from left to right, look for noise threshold crossing points. If the signal was below the noise floor and exceeds it on the next sample, that's a candidate starting point for a window (vice versa for the candidate end point).
Now make a pass through your candidate windows: compare the scope and contents of each window with the values defined above. To use your picture as an example, the small peaks on the left of the image barely exceed the noise floor and do so for too short a time. However, the window in the center of the screen clearly has a wide time extent and a significant max value. Keep the windows that meet your minimum criteria, discard those that are trivial.
Now to examine your remaining windows in detail (remember, they can be treated individually). The peak is easy to find: pass through the window and keep the local max. With respect to the leading and trailing edges of the signal, you can see n the picture that you have a window that's slightly larger than the actual point at which the signal exceeds the noise floor. In this case, you can use a finite difference approximation to calculate the first derivative of the signal. You know that the leading edge will be somewhat to the left of the window on the chart: look for a point at which the first derivative exceeds a positive noise floor of its own (the slope turns upwards sharply). Do the same for the trailing edge (which will always be to the right of the window).
Result: a set of time windows, the leading and trailing edges of the signals and the peak that occured in that window.
It looks like the definition of a window is the range of x over which y is above the threshold. So use that to determine the size of the window. Within that, locate the largest value, thus finding the peak.
If that fails, then what additional criteria do you have for defining a region of interest? You may need to nail down your implicit assumptions to more than 'that looks like a peak to me'.