I'm trying to partially differentiate the mass matrix with respect to the positions (including the free-floating base quaternions). However, I try to first substitute the unit quaternion constraint in the symbolic mass matrix i.e. and then differentiate with respect to all the positions (except q_0).
When I try to use plant.ToSymbolic() and then extract the symbolic mass matrix using plant.CalcMassMatrixViaInverseDynamics() (ManipulatorDynamics function taken from this example) it works for a single free-floating rigid body (3-DoF floating) and also for a free-floating box with a 3-DoF arm (6-DoF system) but the same process does not work when I change to a 7-DoF arm attached to the floating base (13-DoF system). The mass matrix is computed but the function to substitute the unit quaternion constraint takes too long (never finishes and returns). Portion of the applicable code:
q0 = Variable("q0")
q1 = Variable("q1")
q2 = Variable("q2")
q3 = Variable("q3")
unit_cons = Expression.sqrt(1 - q1**2 - q2**2 - q3**2)
mixed_q = np.array([q0, q1, q2, q3,
0, 0, 0,
0, 0, 0, 0, 0, 0, 0])
v = np.zeros(plant.num_velocities())
(M, Cv, tauG, B, tauExt) = ManipulatorDynamics(plant.ToSymbolic(), mixed_q, v)
M[0, 0].Substitute({q0: unit_cons})
This runs well for a single floating rigid body, floating body + 3-DoF arm. But the last line takes a long time and never returns for a floating base + 7-DoF arm.
Is there a better way to approach this with Drake's symbolic engine? Or would AutoDiff be a better choice for this? I only need the 1st partial derivative in this case.
As for some background, I am trying to create a custom linearization of the MultiBodyPlant which embeds the unit quaternion constraint. This requires the partial derivatives of the terms after substituting the unit quaternion constraint to only have the vector part of the quaternion in the equations of motion.
Or would AutoDiff be a better choice for this? I only need the 1st partial derivative in this case.
If you want to evaluate the partial differentiation, then I would suggest using autodiff. To handle your unit quaternion constraint, you can compute your q0 from the vector part of the quaternion.
Here is the pseudo code.
quaternion_vec_val = np.array([0.1, 0.2, 0.3])
q_non_quaternion_val = np.array([0.4, 0.5, 0.6])
q_variable = np.concantenate((quaternion_vec_val, q_non_quaternion_val))
# We want to take derivative w.r.t the vector part of the quaternion, and all non-quaternion variables. Hence we call InitializeAutoDiff on the variables we want to take derivative.
q_variable_ad = pydrake.math.Initialize(q_variable)
quaternion_scalar_ad = np.array([np.sqrt(1 - np.sum(q_variable_ad[:3]**2))])
q_ad = np.concantenate((quaternion_scalar_ad, q_variable_ad))
plant_ad = plant.ToAutoDiffXd()
# Initialize v_ad
v_ad = np.zeros(6, dtype=np.object)
for i in range(6):
v_ad[i] = pydrake.AutoDiffXd(0)
(M, C, g, B, tau_ext) = ManipulatorDynamics(plant_ad, q_ad, v_ad)
print(M(0, 0).derivatives())
(I haven't run the code, especially the part that initializes v_ad is fuzzy, but hopefully you get the idea).
Related
I am a student working with time-series data which we feed into a neural network for classification (my task is to build and train this NN).
We're told to use a band-pass filter of 10 Hz to 150 Hz since anything outside that is not interesting.
After applying the band-pass, I've also down-sampled the data to 300 samples per second (originally it was 768 Hz). My understanding of the Shannon Nyquist sampling theorem is that, after applying the band-pass, any information in the data will be perfectly preserved at this sample-rate.
However, I got into a discussion with my supervisor who claimed that 300 Hz might not be sufficient even if the signal was band-limited. She says that it is only the minimum sample rate, not necessarily the best sample rate.
My understanding of the sampling theorem makes me think the supervisor is obviously wrong, but I don't want to argue with my supervisor, especially in case I'm actually the one who has misunderstood.
Can anyone help to confirm my understanding or provide some clarification? And how should I take this up with my supervisor (if at all).
The Nyquist-Shannon theorem states that the sampling frequency should at-least be twice of bandwidth, i.e.,
fs > 2B
So, this is the minimal criteria. If the sampling frequency is less than 2B then there will be aliasing. There is no upper limit on sampling frequency, but more the sampling frequency, the better will be the reconstruction.
So, I think your supervisor is right in saying that it is the minimal condition and not the best one.
Actually, you and your supervisor are both wrong. The minimum sampling rate required to faithfully represent a real-valued time series whose spectrum lies between 10 Hz and 150 Hz is 140 Hz, not 300 Hz. I'll explain this, and then I'll explain some of the context that shows why you might want to "oversample", as it is referred to (spoiler alert: Bailian-Low Theorem). The supervisor is mixing folklore into the discussion, and when folklore is not properly-contexted, it tends to telephone tag into fakelore. (That's a common failing even in the peer-reviewed literature, by the way). And there's a lot of fakelore, here, that needs to be defogged.
For the following, I will use the following conventions.
There's no math layout on Stack Overflow (except what we already have with UTF-8), so ...
a^b denotes a raised to the power b.
∫_I (⋯x⋯) dx denotes an integral of (⋯x⋯) taken over all x ∈ I, with the default I = ℝ.
The support supp φ (or supp_x φ(x) to make the "x" explicit) of a function φ(x) is the smallest closed set containing all the x-es for which φ(x) ≠ 0. For regularly-behaving (e.g. continuously differentiable) functions that means a union of closed intervals and/or half-rays or the whole real line, itself. This figures centrally in the Shannon-Nyquist sampling theorem, as its main condition is that a spectrum have bounded support; i.e. a "finite bandwidth".
For the Fourier transform I will use the version that has the 2π up in the exponent, and for added convenience, I will use the convention 1^x = e^{2πix} = cos(2πx) + i sin(2πx) (which I refer to as the Ramanujan Convention, as it is the convention I frequently used in my previous life oops I mean which Ramanujan secretly used in his life to make the math a whole lot simpler).
The set ℤ = {⋯, -2, -1, 0, +1, +2, ⋯ } is the integers, and 1^{x+z} = 1^x for all z∈ℤ - making 1^x the archetype of a periodic function whose period is 1.
Thus, the Fourier transform f̂(ν) of a function f(t) and its inverse are given by:
f̂(ν) = ∫ f(t) 1^{-νt} dt, f(t) = ∫ f̂(ν) 1^{+νt} dν.
The spectrum of the time series given by the function f(t) is the function f̂(ν) of the cyclic frequency ν, which is what is measured in Hertz (Hz.); t, itself, being measured in seconds. A common convention is to use the angular frequency ω = 2πν, instead, but that muddies the picture.
The most important example, with respect to the issue at hand, is the Fourier transform χ̂_Ω of the interval function given by χ_Ω(t) = 1 if t ∈ [-½Ω,+½Ω] and χ_Ω(t) = 0 else:
χ̂_Ω(t) = ∫_[-½Ω,+½Ω] 1^ν dν
= {1^{+½Ω} - 1^{-½Ω}}/{2πi}
= {2i sin πΩ}/{2πi}
= Ω sinc πΩ
which is where the function sinc x = (sin πx)/(πx) comes into play.
The cardinal form of the sampling theorem is that a function f(t) can be sampled over an equally-spaced sampled domain T ≡ { kΔt: k ∈ ℤ }, if its spectrum is bounded by supp f̂ ⊆ [-½Ω,+½Ω] ⊆ [-1/(2Δt),+1/(2Δt)], with the sampling given as
f(t) = ∑_{t'∈T} f(t') Ω sinc(Ω(t - t')) Δt.
So, this generally applies to [over-]sampling with redundancy factors 1/(ΩΔt) ≥ 1. In the special case where the sampling is tight with ΩΔt = 1, then it reduces to the form
f(t) = ∑_{t'∈T} f(t') sinc({t - t'}/Δt).
In our case, supp f̂ = [10 Hz., 150 Hz.] so the tightest fits are with 1/Δt = Ω = 300 Hz.
This generalizes to equally-spaced sampled domains of the form T ≡ { t₀ + kΔt: k ∈ ℤ } without any modification.
But it also generalizes to frequency intervals supp f̂ = [ν₋,ν₊] of width Ω = ν₊ - ν₋ and center ν₀ = ½ (ν₋ + ν₊) to the following form:
f(t) = ∑_{t'∈T} f(t') 1^{ν₀(t - t')} Ω sinc(Ω(t - t')) Δt.
In your case, you have ν₋ = 10 Hz., ν₊ = 150 Hz., Ω = 140 Hz., ν₀ = 80 Hz. with the condition Δt ≤ 1/140 second, a sampling rate of at least 140 Hz. with
f(t) = (140 Δt) ∑_{t'∈T} f(t') 1^{80(t - t')} sinc(140(t - t')).
where t and Δt are in seconds.
There is a larger context to all of this. One of the main places where this can be used is for transforms devised from an overlapping set of windowed filters in the frequency domain - a typical case in point being transforms for the time-scale plane, like the S-transform or the continuous wavelet transform.
Since you want the filters to be smoothly-windowed functions, without sharp corners, then in order for them to provide a complete set that adds up to a finite non-zero value over all of the frequency spectrum (so that they can all be normalized, in tandem, by dividing out by this sum), then their respective supports have to overlap.
(Edit: Generalized this example to cover both equally-spaced and logarithmic-spaced intervals.)
One example of such a set would be filters that have end-point frequencies taken from the set
Π = { p₀ (α + 1)ⁿ + β {(α + 1)ⁿ - 1} / α: n ∈ {0,1,2,⋯} }
So, for interval n (counting from n = 0), you would have ν₋ = p_n and ν₊ = p_{n+1}, where the members of Π are enumerated
p_n = p₀ (α + 1)ⁿ + β {(α + 1)ⁿ - 1} / α,
Δp_n = p_{n+1} - p_n = α p_n + β = (α p₀ + β)(α + 1)ⁿ,
n ∈ {0,1,2,⋯}
The center frequency of interval n would then be ν₀ = p_n + ½ Δp₀ (α + 1)ⁿ and the width would be Ω = Δp₀ (α + 1)ⁿ, but the actual support for the filter would overlap into a good part of the neighboring intervals, so that when you add up the filters that cover a given frequency ν the sum doesn't drop down to 0 as ν approaches any of the boundary points. (In the limiting case α → 0, this produces an equally-spaced frequency domain, suitable for an equalizer, while in the case β → 0, it produces a logarithmic scale with base α + 1, where octaves are equally-spaced.)
The other main place where you may apply this is to time-frequency analysis and spectrograms. Here, the role of a function f and its Fourier transform f̂ are reversed and the role of the frequency bandwidth Ω is now played by the (reciprocal) time bandwidth 1/Ω. You want to break up a time series, given by a function f(t) into overlapping segments f̃(q,λ) = g(λ)* f(q + λ), with smooth windowing given by the functions g(λ) with bounded support supp g ⊆ [-½ 1/Ω, +½ 1/Ω], and with interval spacing Δq much larger than the time sampling Δt (the ratio Δq/Δt is called the "hop" factor). The analogous role of Δt is played, here, by the frequency interval in the spectrogram Δp = Ω, which is now constant.
Edit: (Fixed the numbers for the Audacity example)
The minimum sampling rate for both supp_λ g and supp_λ f(q,λ) is Δq = 1/Ω = 1/Δp, and the corresponding redundancy factor is 1/(ΔpΔq). Audacity, for instance, uses a redundancy factor of 2 for its spectrograms. A typical value for Δp might be 44100/2048 Hz., while the time-sampling rate is Δt = 1/(2×3×5×7)² second (corresponding to 1/Δt = 44100 Hz.). With a redundancy factor of 2, Δq would be 1024/44100 second and the hop factor would be Δq/Δt = 1024.
If you try to fit the sampling windows, in either case, to the actual support of the band-limited (or time-limited) function, then the windows won't overlap and the only way to keep their sum from dropping to 0 on the boundary points would be for the windowing functions to have sharp corners on the boundaries, which would wreak havoc on their corresponding Fourier transforms.
The Balian-Low Theorem makes the actual statement on the matter.
https://encyclopediaofmath.org/wiki/Balian-Low_theorem
And a shout-out to someone I've been talking with, recently, about DSP-related matters and his monograph, which provides an excellent introductory reference to a lot of the issues discussed here.
A Friendly Guide To Wavelets
Gerald Kaiser
Birkhauser 1994
He said it's part of a trilogy, another installment of which is forthcoming.
I am trying to linearize a free-floating system with a free-floating base and 3 joints (j1, j2, j3). As I understand the positions part of the system state is given by the vector (this matches MultibodyPlant::num_positions()):
q (10x1) = [base_quaternion (4x1), base_lin_position (3x1), j1_pos, j2_pos, j3_pos]
Since angular velocity requires only 3 components, the velocity part of the system state is written as (this matches MultibodyPlant::num_velocities()):
q_dot (9x1) = [base_rot_vel (3x1), base_lin_vel (3x1), j1_vel, j2_vel, j3_vel]
Using this, the full system state is given as (this works when using MultibodyPlant::SetPositionsAndVelocities) :
X (19x1) = [q (10x1),q_dot (9x1)]
With this, the system acceleration resulting from its dynamics and control forces X_dot = f(X, U) would be written as:
X_dot (18x1)= [q_dot (9x1), q_ddot (9x1)]
Due to the difference in the representation of rotations and angular velocities, the number of terms needed to define X and X_dot is different.
This brings to the following questions while linearizing the system about a point using Linearize:
The A and B matrices after linearization of a continuous-time MultibodyPlant represent the equation X_dot = A*X + B*u. However, there seems to be a mismatch here in the sizes of the arrays/matrices involved as X_dot (18x1) is different from matrices given by Linearize: A (19x19) and B (19x3). I don't then understand what accelerations does the matrix X_dot from the linear system equation represents with its size 19x1?
The above question is only for a continuous-time case. For a discrete-time system,the following equations hold without any issues with the matrix sizes:X[n+1] = A_d * X[n] + B_d * u[n]. However, it is not clear how the quaternion properties are maintained during this linearized forward simulation?
I think there is misunderstanding in the notation since q_dot ≠ v.
Instead, q_dot is simply the ordinary time-derivative of q.
q (10x1) = [base_quaternion (4x1), base_lin_position (3x1), j1_pos, j2_pos, j3_pos]
q_dot (10x1) = d/dt[base_quaternion (4x1), base_lin_position (3x1), j1_pos, j2_pos, j3_pos]
Angular velocity only has 3 components, so v (the velocity part of the system state) and its time-derivative v_dot are:
v (9x1) = [base_rot_vel (3x1), base_lin_vel (3x1), j1_vel, j2_vel, j3_vel]
v_dot (9x1) = d/dt[base_rot_vel (3x1), base_lin_vel (3x1), j1_vel, j2_vel, j3_vel]
The full system state X and its time-derivative x_dot are shown below.
X (19x1) = [q (10x1), v ( 9x1)]
X_dot (19x1) = [q_dot (10x1), v_dot (9x1)]
Note: X ≠ [q, q_dot], instead X = [q, v].
Similarly, X_dot ≠ [q_dot, q_ddot], instead X = [q_dot, v_dot].
I'm trying to understand RNNs and I would like to find a simple example that actually shows the one hot vectors and the numerical operations. Preferably conceptual since actual code may make it even more confusing. Most examples I google just show boxes with loops coming out of them and its really difficult to understand what exactly is going on. In the rare case where they do show the vectors its still difficult to see how they are getting the values.
for example I don't know where the values are coming from in this picture https://i1.wp.com/karpathy.github.io/assets/rnn/charseq.jpeg
If the example could integrate LSTMs and other popular extensions that would be cool too.
In the simple RNN case, a network accepts an input sequence x and produces an output sequence y while a hidden sequence h stores the network's dynamic state, such that at timestep i: x(i) ∊ ℝM, h(i) ∊ ℝN, y(i) ∊ ℝP the real valued vectors of M/N/P dimensions corresponding to input, hidden and output values respectively. The RNN changes its state and omits output based on the state equations:
h(t) = tanh(Wxh ∗ [x(t); h(t-1)]), where Wxh a linear map: ℝM+N ↦ ℝN, * the matrix multiplication and ; the concatenation operation. Concretely, to obtain h(t) you concatenate x(t) with h(t-1), you apply matrix multiplication between Wxh (of shape (M+N, N)) and the concatenated vector (of shape M+N) , and you use a tanh non-linearity on each element of the resulting vector (of shape N).
y(t) = sigmoid(Why * h(t)), where Why a linear map: ℝN ↦ ℝP. Concretely, you apply matrix multiplication between Why (of shape (N, P)) and h(t) (of shape N) to obtain a P-dimensional output vector, on which the sigmoid function is applied.
In other words, obtaining the output at time t requires iterating through the above equations for i=0,1,...,t. Therefore, the hidden state acts as a finite memory for the system, allowing for context-dependent computation (i.e. h(t) fully depends on both the history of the computation and the current input, and so does y(t)).
In the case of gated RNNs (GRU or LSTM), the state equations get somewhat harder to follow, due to the gating mechanisms which essentially allow selection between the input and the memory, but the core concept remains the same.
Numeric Example
Let's follow your example; we have M = 4, N = 3, P = 4, so Wxh is of shape (7, 3) and Why of shape (3, 4). We of course do not know the values of either W matrix, so we cannot reproduce the same results; we can still follow the process though.
At timestep t<0, we have h(t) = [0, 0, 0].
At timestep t=0, we receive input x(0) = [1, 0, 0, 0]. Concatenating x(0) with h(0-), we get [x(t); h(t-1)] = [1, 0, 0 ..., 0] (let's call this vector u to ease notation). We apply u * Wxh (i.e. multiplying a 7-dimensional vector with a 7 by 3 matrix) and get a vector v = [v1, v2, v3], where vi = Σj uj Wji = u1 W1i + u2 W2i + ... + u7 W7i. Finally, we apply tanh on v, obtaining h(0) = [tanh(v1), tanh(v2), tanh(v3)] = [0.3, -0.1, 0.9]. From h(0) we can also get y(0) via the same process; multiply h(0) with Why (i.e. 3 dimensional vector with a 3 by 4 matrix), get a vector s = [s1, s2, s3, s4], apply sigmoid on s and get σ(s) = y(0).
At timestep t=1, we receive input x(1) = [0, 1, 0, 0]. We concatenate x(1) with h(0) to get a new u = [0, 1, 0, 0, 0.3, -0.1, 0.9]. u is again multiplied with Wxh, and tanh is again applied on the result, giving us h(1) = [1, 0.3, 1]. Similarly, h(1) is multiplied by Why, giving us a new s vector on which we apply the sigmoid to obtain σ(s) = y(1).
This process continues until the input sequence finishes, ending the computation.
Note: I have ignored bias terms in the above equations because they do not affect the core concept and they make notation impossible to follow
I am trying to find a simple algorithm to find the correspondence between two sets of 2D points (registration). One set contains the template of an object I'd like to find and the second set mostly contains points that belong to the object of interest, but it can be noisy (missing points as well as additional points that do not belong to the object). Both sets contain roughly 40 points in 2D. The second set is a homography of the first set (translation, rotation and perspective transform).
I am interested in finding an algorithm for registration in order to get the point-correspondence. I will be using this information to find the transform between the two sets (all of this in OpenCV).
Can anyone suggest an algorithm, library or small bit of code that could do the job? As I'm dealing with small sets, it does not have to be super optimized. Currently, my approach is a RANSAC-like algorithm:
Choose 4 random points from set 1 and from set 2.
Compute transform matrix H (using openCV getPerspective())
Warp 1st set of points using H and test how they aligned to the 2nd set of points
Repeat 1-3 N times and choose best transform according to some metric (e.g. sum of squares).
Any ideas? Thanks for your input.
With python you can use Open3D librarry, wich is very easy to install in Anaconda. To your purpose ICP should work fine, so we'll use the classical ICP, wich minimizes point-to-point distances between closest points in every iteration. Here is the code to register 2 clouds:
import numpy as np
import open3d as o3d
# Parameters:
initial_T = np.identity(4) # Initial transformation for ICP
distance = 0.1 # The threshold distance used for searching correspondences
(closest points between clouds). I'm setting it to 10 cm.
# Read your point clouds:
source = o3d.io.read_point_cloud("point_cloud_1.xyz")
target = o3d.io.read_point_cloud("point_cloud_0.xyz")
# Define the type of registration:
type = o3d.pipelines.registration.TransformationEstimationPointToPoint(False)
# "False" means rigid transformation, scale = 1
# Define the number of iterations (I'll use 100):
iterations = o3d.pipelines.registration.ICPConvergenceCriteria(max_iteration = 100)
# Do the registration:
result = o3d.pipelines.registration.registration_icp(source, target, distance, initial_T, type, iterations)
result is a class with 4 things: the transformation T(4x4), 2 metrict (rmse and fitness) and the set of correspondences.
To acess the transformation:
I used it a lot with 3D clouds obteined from Terrestrial Laser Scanners (TLS) and from robots (Velodiny LIDAR).
With MATLAB:
We'll use the point-to-point ICP again, because your data is 2D. Here is a minimum example with two point clouds random generated inside a triangle shape:
% Triangle vértices:
V1 = [-20, 0; -10, 10; 0, 0];
V2 = [-10, 0; 0, 10; 10, 0];
% Create clouds and show pair:
points = 5000
N1 = criar_nuvem_triangulo(V1,points);
N2 = criar_nuvem_triangulo(V2,points);
pcshowpair(N1,N2)
% Registrate pair N1->N2 and show:
[T,N1_tranformed,RMSE]=pcregistericp(N1,N2,'Metric','pointToPoint','MaxIterations',100);
pcshowpair(N1_tranformed,N2)
"criar_nuvem_triangulo" is a function to generate random point clouds inside a triangle:
function [cloud] = criar_nuvem_triangulo(V,N)
% Function wich creates 2D point clouds in triangle format using random
% points
% Parameters: V = Triangle vertices (3x2 Matrix)| N = Number of points
t = sqrt(rand(N, 1));
s = rand(N, 1);
P = (1 - t) * V(1, :) + bsxfun(#times, ((1 - s) * V(2, :) + s * V(3, :)), t);
points = [P,zeros(N,1)];
cloud = pointCloud(points)
end
results:
You may just use cv::findHomography. It is a RANSAC-based approach around cv::getPerspectiveTransform.
auto H = cv::findHomography(srcPoints, dstPoints, CV_RANSAC,3);
Where 3 is the reprojection threshold.
One traditional approach to solve your problem is by using point-set registration method when you don't have matching pair information. Point set registration is similar to method you are talking about.You can find matlab implementation here.
Thanks
I am trying to extract Rotation matrix and Translation vector from the essential matrix.
<pre><code>
SVD svd(E,SVD::MODIFY_A);
Mat svd_u = svd.u;
Mat svd_vt = svd.vt;
Mat svd_w = svd.w;
Matx33d W(0,-1,0,
1,0,0,
0,0,1);
Mat_<double> R = svd_u * Mat(W).t() * svd_vt; //or svd_u * Mat(W) * svd_vt;
Mat_<double> t = svd_u.col(2); //or -svd_u.col(2)
</code></pre>
However, when I am using R and T (e.g. to obtain rectified images), the result does not seem to be right(black images or some obviously wrong outputs), even so I used different combination of possible R and T.
I suspected to E. According to the text books, my calculation is right if we have:
E = U*diag(1, 1, 0)*Vt
In my case svd.w which is supposed to be diag(1, 1, 0) [at least in term of a scale], is not so. Here is an example of my output:
svd.w = [21.47903827647813; 20.28555196246256; 5.167099204708699e-010]
Also, two of the eigenvalues of E should be equal and the third one should be zero. In the same case the result is:
eigenvalues of E = 0.0000 + 0.0000i, 0.3143 +20.8610i, 0.3143 -20.8610i
As you see, two of them are complex conjugates.
Now, the questions are:
Is the decomposition of E and calculation of R and T done in a right way?
If the calculation is right, why the internal rules of essential matrix are not satisfied by the results?
If everything about E, R, and T is fine, why the rectified images obtained by them are not correct?
I get E from fundamental matrix, which I suppose to be right. I draw epipolar lines on both the left and right images and they all pass through the related points (for all the 16 points used to calculate the fundamental matrix).
Any help would be appreciated.
Thanks!
I see two issues.
First, discounting the negligible value of the third diagonal term, your E is about 6% off the ideal one: err_percent = (21.48 - 20.29) / 20.29 * 100 . Sounds small, but translated in terms of pixel error it may be an altogether larger amount.
So I'd start by replacing E with the ideal one after SVD decomposition: Er = U * diag(1,1,0) * Vt.
Second, the textbook decomposition admits 4 solutions, only one of which is physically plausible (i.e. with 3D points in front of the camera). You may be hitting one of non-physical ones. See http://en.wikipedia.org/wiki/Essential_matrix#Determining_R_and_t_from_E .