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Time series dBFS plot output modification - current output plot not as expected (matplotlib)

I'm trying to plot the Amplitude (dBFS) vs. Time (s) plot of an audio (.wav) file using matplotlib. I managed to do that with the following code:
def convert_to_decibel(sample):
ref = 32768 # Using a signed 16-bit PCM format wav file. So, 2^16 is the max. value.
if sample!=0:
return 20 * np.log10(abs(sample) / ref)
else:
return 20 * np.log10(0.000001)
from scipy.io.wavfile import read as readWav
from scipy.fftpack import fft
import matplotlib.pyplot as gplot1
import matplotlib.pyplot as gplot2
import numpy as np
import struct
import gc
wavfile1 = '/home/user01/audio/speech.wav'
wavsamplerate1, wavdata1 = readWav(wavfile1)
wavdlen1 = wavdata1.size
wavdtype1 = wavdata1.dtype
gplot1.rcParams['figure.figsize'] = [15, 5]
pltaxis1 = gplot1.gca()
gplot1.axhline(y=0, c="black")
gplot1.xticks(np.arange(0, 10, 0.5))
gplot1.yticks(np.arange(-200, 200, 5))
gplot1.grid(linestyle = '--')
wavdata3 = np.array([convert_to_decibel(i) for i in wavdata1], dtype=np.int16)
yvals3 = wavdata3
t3 = wavdata3.size / wavsamplerate1
xvals3 = np.linspace(0, t3, wavdata3.size)
pltaxis1.set_xlim([0, t3 + 2])
pltaxis1.set_title('Amplitude (dBFS) vs Time(s)')
pltaxis1.plot(xvals3, yvals3, '-')
which gives the following output:
I had also plotted the Power Spectral Density (PSD, in dBm) using the code below:
from scipy.signal import welch as psd # Computes PSD using Welch's method.
fpsd, wPSD = psd(wavdata1, wavsamplerate1, nperseg=1024)
gplot2.rcParams['figure.figsize'] = [15, 5]
pltpsdm = gplot2.gca()
gplot2.axhline(y=0, c="black")
pltpsdm.plot(fpsd, 20*np.log10(wPSD))
gplot2.xticks(np.arange(0, 4000, 400))
gplot2.yticks(np.arange(-150, 160, 10))
pltpsdm.set_xlim([0, 4000])
pltpsdm.set_ylim([-150, 150])
gplot2.grid(linestyle = '--')
which gives the output as:
The second output above, using the Welch's method plots a more presentable output. The dBFS plot though informative is not very presentable IMO. Is this because of:
the difference in the domains (time in case of 1st output vs frequency in the 2nd output)?
the way plot function is implemented in pyplot?
Also, is there a way I can plot my dBFS output as a peak-to-peak style of plot just like in my PSD (dBm) plot rather than a dense stem plot?
Would be much helpful and would appreciate any pointers, answers or suggestions from experts here as I'm just a beginner with matplotlib and plots in python in general.

TLNR
This has nothing to do with pyplot.
The frequency domain is different from the time domain, but that's not why you didn't get what you want.
The calculation of dbFS in your code is wrong.
You should frame your data, calculate RMSs or peaks in every frame, and then convert that value to dbFS instead of applying this transformation to every sample point.
When we talk about the amplitude, we are talking about a periodic signal. And when we read in a series of data from a sound file, we read in a series of sample points of a signal(may be or be not periodic). The value of every sample point represents a, say, voltage value, or sound pressure value sampled at a specific time.
We assume that, within a very short time interval, maybe 10ms for example, the signal is stationary. Every such interval is called a frame.
Some specific function is applied to each frame usually, to reduce the sudden change at the edge of this frame, and these functions are called window functions. If you did nothing to every frame, you added rectangle windows to them.
An example: when the sampling frequency of your sound is 44100Hz, in a 10ms-long frame, there are 44100*0.01=441 sample points. That's what the nperseg argument means in your psd function but it has nothing to do with dbFS.
Given the knowledge above, now we can talk about the amplitude.
There are two methods a get the value of amplitude in every frame:
The most straightforward one is to get the maximum(peak) values in every frame.
Another one is to calculate the RMS(Root Mean Sqaure) of every frame.
After that, the peak values or RMS values can be converted to dbFS values.
Let's start coding:
import numpy as np
import matplotlib.pyplot as plt
from scipy.io import wavfile
# Determine full scall(maximum possible amplitude) by bit depth
bit_depth = 16
full_scale = 2 ** bit_depth
# dbFS function
to_dbFS = lambda x: 20 * np.log10(x / full_scale)
# Read in the wave file
fname = "01.wav"
fs,data = wavfile.read(fname)
# Determine frame length(number of sample points in a frame) and total frame numbers by window length(how long is a frame in seconds)
window_length = 0.01
signal_length = data.shape[0]
frame_length = int(window_length * fs)
nframes = signal_length // frame_length
# Get frames by broadcast. No overlaps are used.
idx = frame_length * np.arange(nframes)[:,None] + np.arange(frame_length)
frames = data[idx].astype("int64") # Convert to in 64 to avoid integer overflow
# Get RMS and peaks
rms = ((frames**2).sum(axis=1)/frame_length)**.5
peaks = np.abs(frames).max(axis=1)
# Convert them to dbfs
dbfs_rms = to_dbFS(rms)
dbfs_peak = to_dbFS(peaks)
# Let's start to plot
# Get time arrays of every sample point and ever frame
frame_time = np.arange(nframes) * window_length
data_time = np.linspace(0,signal_length/fs,signal_length)
# Plot
f,ax = plt.subplots()
ax.plot(data_time,data,color="k",alpha=.3)
# Plot the dbfs values on a twin x Axes since the y limits are not comparable between data values and dbfs
tax = ax.twinx()
tax.plot(frame_time,dbfs_rms,label="RMS")
tax.plot(frame_time,dbfs_peak,label="Peak")
tax.legend()
f.tight_layout()
# Save serval details
f.savefig("whole.png",dpi=300)
ax.set_xlim(1,2)
f.savefig("1-2sec.png",dpi=300)
ax.set_xlim(1.295,1.325)
f.savefig("1.2-1.3sec.png",dpi=300)
The whole time span looks like(the unit of the right axis is dbFS):
And the voiced part looks like:
You can see that the dbFS values become greater while the amplitudes become greater at the vowel start point:

Is there an easy way of finding frequency of envelopes in sound signals?

I have a sound signal of 5 secs length and it is from the sound of a propeller. I need to find rpm of the propeller by finding frequency of the envelopes.
import wave
import numpy as np
import matplotlib.pyplot as plt
raw = wave.open('/content/drive/MyDrive/Demon.wav','r')
signal = raw.readframes(-1)
signal = np.frombuffer(signal , dtype="int16")
frate = raw.getframerate()
time = np.linspace(0,len(signal) / frate,num = len(signal))
plt.figure(1)
plt.title("Sound Wave")
plt.xlabel("Time")
plt.plot(time, signal)
plt.show()
Here is the link to the sound file itself: https://sndup.net/5v3j
And since it is a 5 second-length signal and has 80.000 samples, I want to see it in details by looking 1 second part of the signal.
partial_signal = signal [1 : 16000]
partial_time = time[1 : 16000]
plt.plot(partial_time,partial_signal)
plt.show()
Output of the plot is shown below.
Edit: Looks like image will not show up here is the link to the image:
https://imgur.com/P5lnSM1
Now I need to find frequency of the envelopes thus rpm of the propeller by using only python.

You can do that quite easily with a fast Fourier transform (FFT) applied on the signal amplitude. Here is an example:
import wave
import numpy as np
import matplotlib.pyplot as plt
from scipy.fft import rfft, rfftfreq
from scipy.ndimage import gaussian_filter
raw = wave.open('Demon.wav','r')
signal = raw.readframes(-1)
signal = np.frombuffer(signal , dtype="int16")
frate = raw.getframerate()
time = np.linspace(0,len(signal) / frate,num = len(signal))
# Compute the amplitude of the sound signal
signalAmplitude = signal.astype(np.float64)**2
# Filter the signal to remove very short-timed amplitude modulations (<= 1 ms)
signalAmplitude = gaussian_filter(signalAmplitude, sigma=frate/1000)
# Compute the frequency amplitude of the FFT signal
tmpFreq = np.abs(rfft(signalAmplitude))
# Get the associated practical frequency for this signal
hzFreq = rfftfreq(signal.shape[0], d=1/frate)
finalFrequency = hzFreq[1+tmpFreq[1:].argmax()]
print(finalFrequency)
# Show sound frequency diagram
plt.xticks(np.arange(21))
plt.xlim([1, 20]) # Show only interesting low frequencies
plt.plot(hzFreq, tmpFreq)
plt.show()
The frequency diagram is the following:
The final detected frequency is 3.0 Hz which is very consistent with what we can hear.

Creating similar spectrogram in continues wavelet transform compared to discret wavelet transform

Using PyWavelets and Matplotbib.Specgram on a signal gives more detailed plots with pywt.dwt then pywt.cwt. How can I get a pywt.cwt specgram in a similar way?
With dwt:
import pywt
import pywt.data
import matplotlib.pyplot as plot
from scipy import signal
from scipy.io import wavfile
bA, bD = pywt.dwt(datamean, 'db2')
powerSpectrum, freqenciesFound, time, imageAxis = plot.specgram(bA, NFFT = 387, Fs=100)
plot.xlabel('Time')
plot.ylabel('Frequency')
plot.show()
with this spectrogram plot:
https://imgur.com/a/bYb8bBS
With cwt:
widths = np.arange(1,5)
coef, freqs = pywt.cwt(datamean, widths,'morl')
powerSpectrum, freqenciesFound, time, imageAxis = plot.specgram(coef, NFFT = 129, Fs=100)
plot.xlabel('Time')
plot.ylabel('Frequency')
plot.show()
with this spectrogram plot:
https://imgur.com/a/GIINzJp
and for better results:
sig = datamean
widths = np.arange(1, 31)
cwtmatr = signal.cwt(sig, signal.ricker, widths)
plt.imshow(cwtmatr, extent=[-1, 1, 1, 5], cmap='PRGn', aspect='auto',
vmax=abs(cwtmatr).max(), vmin=-abs(cwtmatr).max())
plt.show()
with this spectrogram plot:
https://imgur.com/a/TnXqgGR
How can I get for cwt (spectrogram plot 2 and 3) a similar spectogram plot and style like in the first one?
It seems like the 1st spectrogram plot compared to the 3rd has much more details.

This would be better as a comment, but since I lack the Karma to do that:
You don't want to make a spectrogram with wavelets, but a scalogram instead. What it looks like you're doing above is projecting your data in a scale subspace (that correlates to frequency), then taking those scales and finding the frequency content of them which is not what you probably want.
The detail and approximation coefficients are what you would want to use directly. Unfortunately, PyWavelets doesn't have a simple plotting function to do this for you, AFAIK. Matlab does, and their help page may be illuminating if I fail.
def scalogram(data):
wave='db4'
coeff=pywt.wavedec(data,wave)
levels=len(coeff)
lengths=[len(co) for co in coeff]
col=np.max(lengths)
im=np.ones([levels,col])
col=col.astype(float)
for level in range(levels):
#print [lengths[level],col]
y=coeff[level]
if lengths[1+level]<col:
x=col/(lengths[1+level]+1)*np.arange(1,len(y)+1)
xi=np.linspace(0,int(col),int(col))
yi=griddata(points=x,values=y,xi=xi,method='nearest')
else:
yi=y
im[level,:]=yi
im[im==0]=np.nan
tiles=sum(lengths)-lengths[0]
return im,tiles
Wxx,tiles=scalogram(data)
IM=plt.imshow(np.log10(abs(Wxx)),aspect='auto')
plt.show()
There are better ways of doing that, but it works. This produces a square matrix similar to spectrogram in "Wxx", and tiles is simply a counter of the number of time-frequency tilings to compare to the number used in a SFFT.
I've attached a picture of what these tilings look like

How to get and plot a signal envelope

I would like to know if there is a function envelope in Python to have the same result as this
I have already tried an envelope function in Python but there is this result and it doesn't correspond with what I want.

Though you don't mention exactly what function you use, it seems like you are using two different kinds of envelopes.
The way you call envelope in matlab, the relevant description is:
[yupper,ylower] = envelope(x) returns the upper and lower envelopes of
the input sequence, x, as the magnitude of its analytic signal. The
analytic signal of x is found using the discrete Fourier transform as
implemented in hilbert. The function initially removes the mean of x
and adds it back after computing the envelopes. If x is a matrix, then
envelope operates independently over each column of x.
Based on this, I suppose you would be looking for a way to get the Hilber transform in python. An example of this can be found here:
import numpy as np
import matplotlib.pyplot as plt
from scipy.signal import hilbert, chirp
duration = 1.0
fs = 400.0
samples = int(fs*duration)
t = np.arange(samples) / fs
signal = chirp(t, 20.0, t[-1], 100.0)
signal *= (1.0 + 0.5 * np.sin(2.0*np.pi*3.0*t) )
analytic_signal = hilbert(signal)
amplitude_envelope = np.abs(analytic_signal)
instantaneous_phase = np.unwrap(np.angle(analytic_signal))
instantaneous_frequency = np.diff(instantaneous_phase) / (2.0*np.pi) * fs
fig = plt.figure()
ax0 = fig.add_subplot(211)
ax0.plot(t, signal, label='signal')
ax0.plot(t, amplitude_envelope, label='envelope')
ax0.set_xlabel("time in seconds")
ax0.legend()
ax1 = fig.add_subplot(212)
ax1.plot(t[1:], instantaneous_frequency)
ax1.set_xlabel("time in seconds")
ax1.set_ylim(0.0, 120.0)
Resulting in:

Sometimes I would use obspy.signal.filter.envelope(data_array); But you can only get the upper line in your given example.
Obspy is a very useful package dealing with seismogram.

Phase shift of the signal after using FIR filter firwin in Python

So after my two last questions I come to my actual problem. Maybe somebody finds the error in my theoretical procedure or I did something wrong in programming.
I am implementing a bandpass filter in Python using scipy.signal (using the firwin function). My original signal consists of two frequencies (w_1=600Hz, w_2=800Hz). There might be a lot more frequencies that's why I need a bandpass filter.
In this case I want to filter the frequency band around 600 Hz, so I took 600 +/- 20Hz as cutoff frequencies. When I implemented the filter and reproduce the signal in the time domain using lfilter the right frequency is filtered.
To get rid of the phase shift I plotted the frequency response by using scipy.signal.freqz with the return h of firwin as numerator and 1 as predefined denumerator.
As described in the documentation of freqz I plotted the phase (== angle in the doc) as well and was able to look at the frequency response plot to get the phase shift for the frequency 600 Hz of the filtered signal.
So the phase delay t_p is
t_p=-(Tetha(w))/(w)
Unfortunately when I add this phase delay to the time data of my filtered signal, it has not got the same phase as the original 600 Hz signal.
I added the code. It is weird, before eliminating some part of the code to keep the minimum, the filtered signal started at the correct amplitude - now it is even worse.
################################################################################
#
# Filtering test
#
################################################################################
#
from math import *
import numpy as np
from scipy import signal
from scipy.signal import firwin, lfilter, lti
from scipy.signal import freqz
import matplotlib.pyplot as plt
import matplotlib.colors as colors
################################################################################
# Nb of frequencies in the original signal
nfrq = 2
F = [60,80]
################################################################################
# Sampling:
nitper = 16
nper = 50.
fmin = np.min(F)
fmax = np.max(F)
T0 = 1./fmin
dt = 1./fmax/nitper
#sampling frequency
fs = 1./dt
nyq_rate= fs/2
nitpermin = nitper*fmax/fmin
Nit = int(nper*nitpermin+1)
tps = np.linspace(0.,nper*T0,Nit)
dtf = fs/Nit
################################################################################
# Build analytic signal
# s = completeSignal(F,Nit,tps)
scomplete = np.zeros((Nit))
omg1 = 2.*pi*F[0]
omg2 = 2.*pi*F[1]
scomplete=scomplete+np.sin(omg1*tps)+np.sin(omg2*tps)
#ssingle = singleSignals(nfrq,F,Nit,tps)
ssingle=np.zeros((nfrq,Nit))
ssingle[0,:]=ssingle[0,:]+np.sin(omg1*tps)
ssingle[1,:]=ssingle[0,:]+np.sin(omg2*tps)
################################################################################
## Construction of the desired bandpass filter
lowcut = (60-2) # desired cutoff frequencies
highcut = (60+2)
ntaps = 451 # the higher and closer the signal frequencies, the more taps for the filter are required
taps_hamming = firwin(ntaps,[lowcut/nyq_rate, highcut/nyq_rate], pass_zero=False)
# Use lfilter to get the filtered signal
filtered_signal = lfilter(taps_hamming, 1, scomplete)
# The phase delay of the filtered signal
delay = ((ntaps-1)/2)/fs
plt.figure(1, figsize=(12, 9))
# Plot the signals
plt.plot(tps, scomplete,label="Original signal with %s freq" % nfrq)
plt.plot(tps-delay, filtered_signal,label="Filtered signal %s freq " % F[0])
plt.plot(tps, ssingle[0,:],label="original signal %s Hz" % F[0])
plt.grid(True)
plt.legend()
plt.xlim(0,1)
plt.xlabel('Time (s)')
plt.ylabel('Amplitude')
# Plot the frequency responses of the filter.
plt.figure(2, figsize=(12, 9))
plt.clf()
# First plot the desired ideal response as a green(ish) rectangle.
rect = plt.Rectangle((lowcut, 0), highcut - lowcut, 5.0,facecolor="#60ff60", alpha=0.2,label="ideal filter")
plt.gca().add_patch(rect)
# actual filter
w, h = freqz(taps_hamming, 1, worN=1000)
plt.plot((fs * 0.5 / np.pi) * w, abs(h), label="designed rectangular window filter")
plt.xlim(0,2*F[1])
plt.ylim(0, 1)
plt.grid(True)
plt.legend()
plt.xlabel('Frequency (Hz)')
plt.ylabel('Gain')
plt.title('Frequency response of FIR filter, %d taps' % ntaps)
plt.show()'

The delay of your FIR filter is simply 0.5*(n - 1)/fs, where n is the number of filter coefficients (i.e. "taps") and fs is the sample rate. Your implementation of this delay is fine.
The problem is that your array of time values tps is not correct. Take a look
at 1.0/(tps[1] - tps[0]); you'll see that it does not equal fs.
Change this:
tps = np.linspace(0.,nper*T0,Nit)
to, for example, this:
T = Nit / fs
tps = np.linspace(0., T, Nit, endpoint=False)
and your plots of the original and filtered 60 Hz signals will line up beautifully.
For another example, see http://wiki.scipy.org/Cookbook/FIRFilter.
In the script there, the delay is calculated on line 86. Below this, the delay is used to plot the original signal aligned with the filtered signal.
Note: The cookbook example uses scipy.signal.lfilter to apply the filter. A more efficient approach is to use numpy.convolve.

Seems like you may have had this answered already, but I believe that this is what the filtfilt function is used for. Basically, it does both a forward sweep and a backward sweep through your data, thus reversing the phase shift introduced by the initial filtering. Might be worth looking into.