The personal website of Scott W Harden

Realtime Audio Visualization in Python

Python's "batteries included" nature makes it easy to interact with just about anything... except speakers and a microphone! As of this moment, there still are not standard libraries which which allow cross-platform interfacing with audio devices. There are some pretty convenient third-party modules, but I hope in the future a standard solution will be distributed with python. I appreciate the differences of Linux architectures such as ALSA and OSS, but toss in Windows and MacOS in the mix and it gets to be a huge mess. For Linux, would I even need anything fancy? I can run "cat file.wav > /dev/dsp" from a command prompt to play audio. There are some standard libraries for operating system specific sound (i.e., winsound), but I want something more versatile. The official audio wiki page on the subject lists a small collection of third-party platform-independent libraries. After excluding those which don't support microphone access (the ultimate goal of all my poking around in this subject), I dove a little deeper into sounddevice and PyAudio. Both of these I installed with pip (i.e., pip install pyaudio)

I really like the structure and documentation of sounddevice, but I decided to keep developing with PyAudio for now. Sounddevice seemed to take more system resources than PyAudio (in my limited test conditions: Windows 10 with very fast and modern hardware, Python 3), and would audibly "glitch" music as it was being played every time it attached or detached from the microphone stream. I tried streaming, but after about an hour I couldn't get clean live access to the microphone without glitching audio playback. Furthermore, every few times I ran this script it crashed my python kernel! I very rarely see this happening. iPython complained: "It seems the kernel died unexpectedly. Use 'Restart kernel' to continue using this console" and I eventually moved back to PyAudio. For a less "realtime" application, sounddevice might be a great solution. Here's the minimal case sounddevice script I tested with (that crashed sometimes). If you have a better one to do live high-speed audio capture, let me know!

import sounddevice #pip install sounddevice

for i in range(30): #30 updates in 1 second
    rec = sounddevice.rec(44100/30)
    sounddevice.wait()
    print(rec.shape)

Here's a simple demo to show how I get realtime microphone audio into numpy arrays using PyAudio. This isn't really that special. It's a good starting point though. Note that rather than have the user define a microphone source in the python script (I had a fancy menu system handling this for a while), I allow PyAudio to just look at the operating system's default input device. This seems like a realistic expectation, and saves time as long as you don't expect your user to be recording from two different devices at the same time. This script gets some audio from the microphone and shows the values in the console (ten times).

import pyaudio
import numpy as np

CHUNK = 4096 # number of data points to read at a time
RATE = 44100 # time resolution of the recording device (Hz)

p=pyaudio.PyAudio() # start the PyAudio class
stream=p.open(format=pyaudio.paInt16,channels=1,rate=RATE,input=True,
              frames_per_buffer=CHUNK) #uses default input device

# create a numpy array holding a single read of audio data
for i in range(10): #to it a few times just to see
    data = np.fromstring(stream.read(CHUNK),dtype=np.int16)
    print(data)

# close the stream gracefully
stream.stop_stream()
stream.close()
p.terminate()

I tried to push the limit a little bit and see how much useful data I could get from this console window. It turns out that it's pretty responsive! Here's a slight modification of the code, made to turn the console window into an impromptu VU meter.

import pyaudio
import numpy as np

CHUNK = 2**11
RATE = 44100

p=pyaudio.PyAudio()
stream=p.open(format=pyaudio.paInt16,channels=1,rate=RATE,input=True,
              frames_per_buffer=CHUNK)

for i in range(int(10*44100/1024)): #go for a few seconds
    data = np.fromstring(stream.read(CHUNK),dtype=np.int16)
    peak=np.average(np.abs(data))*2
    bars="#"*int(50*peak/2**16)
    print("%04d %05d %s"%(i,peak,bars))

stream.stop_stream()
stream.close()
p.terminate()

Result

The results are pretty good! The advantage here is that no libraries are required except PyAudio. For people interested in doing simple math (peak detection, frequency detection, etc.) this is a perfect starting point. Here's a quick cellphone video:

I've made realtime audio visualization (realtime FFT) scripts with Python before, but 80% of that code was creating a GUI. I want to see data in real time while I'm developing this code, but I really don't want to mess with GUI programming. I then had a crazy idea. Everyone has a web browser, which is a pretty good GUI... with a Python script to analyze audio and save graphs (a lot of them, quickly) and some JavaScript running in a browser to keep refreshing those graphs, I could get an idea of what the audio stream is doing in something kind of like real time. It was intended to be a hack, but I never expected it to work so well! Check this out...

Here's the python script to listen to the microphone and generate graphs:

import pyaudio
import numpy as np
import pylab
import time

RATE = 44100
CHUNK = int(RATE/20) # RATE / number of updates per second

def soundplot(stream):
    t1=time.time()
    data = np.fromstring(stream.read(CHUNK),dtype=np.int16)
    pylab.plot(data)
    pylab.title(i)
    pylab.grid()
    pylab.axis([0,len(data),-2**16/2,2**16/2])
    pylab.savefig("03.png",dpi=50)
    pylab.close('all')
    print("took %.02f ms"%((time.time()-t1)*1000))

if __name__=="__main__":
    p=pyaudio.PyAudio()
    stream=p.open(format=pyaudio.paInt16,channels=1,rate=RATE,input=True,
                  frames_per_buffer=CHUNK)
    for i in range(int(20*RATE/CHUNK)): #do this for 10 seconds
        soundplot(stream)
    stream.stop_stream()
    stream.close()
    p.terminate()

Here's the HTML file with JavaScript to keep reloading the image...

<html>
<script language="javascript">
function RefreshImage(){
document.pic0.src="03.png?a=" + String(Math.random()*99999999);
setTimeout('RefreshImage()',50);
}
</script>
<body onload="RefreshImage()">
<img name="pic0" src="03.png">
</body>
</html>

Operation

I couldn't believe my eyes. It's not elegant, but it's kind of functional!

Why stop there? I went ahead and wrote a microphone listening and processing class which makes this stuff easier. My ultimate goal hasn't been revealed yet, but I'm sure it'll be clear in a few weeks. Let's just say there's a lot of use in me visualizing streams of continuous data. Anyway, this class is the truly terrible attempt at a word pun by merging the words "SWH", "ear", and "Hear", into the official title "SWHear" which seems to be unique on Google. This class is minimal case, but can be easily modified to implement threaded recording (which won't cause the rest of the functions to hang) as well as mathematical manipulation of data, such as FFT. With the same HTML file as used above, here's the new python script and some video of the output:

import pyaudio
import time
import pylab
import numpy as np

class SWHear(object):
    """
    The SWHear class is made to provide access to continuously recorded
    (and mathematically processed) microphone data.
    """

    def __init__(self,device=None,startStreaming=True):
        """fire up the SWHear class."""
        print(" -- initializing SWHear")

        self.chunk = 4096 # number of data points to read at a time
        self.rate = 44100 # time resolution of the recording device (Hz)

        # for tape recording (continuous "tape" of recent audio)
        self.tapeLength=2 #seconds
        self.tape=np.empty(self.rate*self.tapeLength)*np.nan

        self.p=pyaudio.PyAudio() # start the PyAudio class
        if startStreaming:
            self.stream_start()

    ### LOWEST LEVEL AUDIO ACCESS
    # pure access to microphone and stream operations
    # keep math, plotting, FFT, etc out of here.

    def stream_read(self):
        """return values for a single chunk"""
        data = np.fromstring(self.stream.read(self.chunk),dtype=np.int16)
        #print(data)
        return data

    def stream_start(self):
        """connect to the audio device and start a stream"""
        print(" -- stream started")
        self.stream=self.p.open(format=pyaudio.paInt16,channels=1,
                                rate=self.rate,input=True,
                                frames_per_buffer=self.chunk)

    def stream_stop(self):
        """close the stream but keep the PyAudio instance alive."""
        if 'stream' in locals():
            self.stream.stop_stream()
            self.stream.close()
        print(" -- stream CLOSED")

    def close(self):
        """gently detach from things."""
        self.stream_stop()
        self.p.terminate()

    ### TAPE METHODS
    # tape is like a circular magnetic ribbon of tape that's continously
    # recorded and recorded over in a loop. self.tape contains this data.
    # the newest data is always at the end. Don't modify data on the type,
    # but rather do math on it (like FFT) as you read from it.

    def tape_add(self):
        """add a single chunk to the tape."""
        self.tape[:-self.chunk]=self.tape[self.chunk:]
        self.tape[-self.chunk:]=self.stream_read()

    def tape_flush(self):
        """completely fill tape with new data."""
        readsInTape=int(self.rate*self.tapeLength/self.chunk)
        print(" -- flushing %d s tape with %dx%.2f ms reads"%\
                  (self.tapeLength,readsInTape,self.chunk/self.rate))
        for i in range(readsInTape):
            self.tape_add()

    def tape_forever(self,plotSec=.25):
        t1=0
        try:
            while True:
                self.tape_add()
                if (time.time()-t1)>plotSec:
                    t1=time.time()
                    self.tape_plot()
        except:
            print(" ~~ exception (keyboard?)")
            return

    def tape_plot(self,saveAs="03.png"):
        """plot what's in the tape."""
        pylab.plot(np.arange(len(self.tape))/self.rate,self.tape)
        pylab.axis([0,self.tapeLength,-2**16/2,2**16/2])
        if saveAs:
            t1=time.time()
            pylab.savefig(saveAs,dpi=50)
            print("plotting saving took %.02f ms"%((time.time()-t1)*1000))
        else:
            pylab.show()
            print() #good for IPython
        pylab.close('all')

if __name__=="__main__":
    ear=SWHear()
    ear.tape_forever()
    ear.close()
    print("DONE")

I don't really intend anyone to actually do this, but it's a cool alternative to recording a small portion of audio, plotting it in a pop-up matplotlib window, and waiting for the user to close it to record a new fraction. I had a lot more text in here demonstrating real-time FFT, but I'd rather consolidate everything FFT related into a single post. For now, I'm happy pursuing microphone-related python projects with PyAudio.

Display a single frequency

Use Numpy's FFT() and FFTFREQ() to turn the linear data into frequency. Set that target and grab the FFT value corresponding to that frequency. I haven't tested this to be sure it's working, but it should at least be close...

import pyaudio
import numpy as np
np.set_printoptions(suppress=True) # don't use scientific notation

CHUNK = 4096 # number of data points to read at a time
RATE = 44100 # time resolution of the recording device (Hz)
TARGET = 2100 # show only this one frequency

p=pyaudio.PyAudio() # start the PyAudio class
stream=p.open(format=pyaudio.paInt16,channels=1,rate=RATE,input=True,
              frames_per_buffer=CHUNK) #uses default input device

# create a numpy array holding a single read of audio data
for i in range(10): #to it a few times just to see
    data = np.fromstring(stream.read(CHUNK),dtype=np.int16)
    fft = abs(np.fft.fft(data).real)
    fft = fft[:int(len(fft)/2)] # keep only first half
    freq = np.fft.fftfreq(CHUNK,1.0/RATE)
    freq = freq[:int(len(freq)/2)] # keep only first half
    assert freq[-1]>TARGET, "ERROR: increase chunk size"
    val = fft[np.where(freq>TARGET)[0][0]]
    print(val)

# close the stream gracefully
stream.stop_stream()
stream.close()
p.terminate()

Display Peak Frequency

If your goal is to determine which frequency is producing the loudest tone, use this function. I also added a few lines to graph the output in case you want to observe how it operates. I recommend testing this script with a tone generator, or a YouTube video containing tones of a range of frequencies like this one.

import pyaudio
import numpy as np
import matplotlib.pyplot as plt

np.set_printoptions(suppress=True) # don't use scientific notation

CHUNK = 4096 # number of data points to read at a time
RATE = 44100 # time resolution of the recording device (Hz)

p=pyaudio.PyAudio() # start the PyAudio class
stream=p.open(format=pyaudio.paInt16,channels=1,rate=RATE,input=True,
              frames_per_buffer=CHUNK) #uses default input device

# create a numpy array holding a single read of audio data
for i in range(10): #to it a few times just to see
    data = np.fromstring(stream.read(CHUNK),dtype=np.int16)
    data = data * np.hanning(len(data)) # smooth the FFT by windowing data
    fft = abs(np.fft.fft(data).real)
    fft = fft[:int(len(fft)/2)] # keep only first half
    freq = np.fft.fftfreq(CHUNK,1.0/RATE)
    freq = freq[:int(len(freq)/2)] # keep only first half
    freqPeak = freq[np.where(fft==np.max(fft))[0][0]]+1
    print("peak frequency: %d Hz"%freqPeak)

    # uncomment this if you want to see what the freq vs FFT looks like
    #plt.plot(freq,fft)
    #plt.axis([0,4000,None,None])
    #plt.show()
    #plt.close()

# close the stream gracefully
stream.stop_stream()
stream.close()
p.terminate()

Display Left and Right Levels

import pyaudio
import numpy as np

maxValue = 2**16
p=pyaudio.PyAudio()
stream=p.open(format=pyaudio.paInt16,channels=2,rate=44100,
              input=True, frames_per_buffer=1024)
while True:
    data = np.fromstring(stream.read(1024),dtype=np.int16)
    dataL = data[0::2]
    dataR = data[1::2]
    peakL = np.abs(np.max(dataL)-np.min(dataL))/maxValue
    peakR = np.abs(np.max(dataR)-np.min(dataR))/maxValue
    print("L:%00.02f R:%00.02f"%(peakL*100, peakR*100))

Output

L:47.26 R:45.17
L:47.55 R:45.63
L:49.44 R:45.98
L:45.27 R:49.80
L:44.39 R:45.75
L:47.50 R:46.96
L:41.49 R:42.64
L:42.95 R:41.39
L:49.56 R:49.62
L:48.29 R:48.80
L:45.03 R:47.62
L:47.99 R:49.35
L:41.58 R:49.21

Or with a tweak...

import pyaudio
import numpy as np

maxValue = 2**16
bars = 35
p=pyaudio.PyAudio()
stream=p.open(format=pyaudio.paInt16,channels=2,rate=44100,
              input=True, frames_per_buffer=1024)
while True:
    data = np.fromstring(stream.read(1024),dtype=np.int16)
    dataL = data[0::2]
    dataR = data[1::2]
    peakL = np.abs(np.max(dataL)-np.min(dataL))/maxValue
    peakR = np.abs(np.max(dataR)-np.min(dataR))/maxValue
    lString = "#"*int(peakL*bars)+"-"*int(bars-peakL*bars)
    rString = "#"*int(peakR*bars)+"-"*int(bars-peakR*bars)
    print("L=[%s]\tR=[%s]"%(lString, rString))

Graphical Output

Markdown source code last modified on January 18th, 2021
---
title: Realtime Audio Visualization in Python
date: 2016-07-19 04:44:48
tags: python, old
---

# Realtime Audio Visualization in Python

__Python's "batteries included" nature makes it easy to interact with just about anything... except speakers and a microphone!__ As of this moment, there still are not standard libraries which which allow cross-platform interfacing with audio devices. There are some pretty convenient third-party modules, but I hope in the future a standard solution will be distributed with python. I appreciate the differences of Linux architectures such as [ALSA](http://www.alsa-project.org/) and [OSS](https://en.wikipedia.org/wiki/Open_Sound_System), but toss in Windows and MacOS in the mix and it gets to be a huge mess. For Linux, would I even need anything fancy? I can run "`` cat file.wav > /dev/dsp ``" from a command prompt to play audio. There are some standard libraries for operating system specific sound (i.e., [winsound](https://docs.python.org/2/library/winsound.html)), but I want something more versatile. The [official audio wiki page on the subject](https://wiki.python.org/moin/Audio/) lists a small collection of third-party platform-independent libraries. After excluding those which don't support microphone access (the ultimate goal of all my poking around in this subject), I dove a little deeper into [sounddevice](http://python-sounddevice.readthedocs.io/en/0.3.3/) and [PyAudio](http://people.csail.mit.edu/hubert/pyaudio/). Both of these I installed with pip (i.e., `` pip install pyaudio ``)

__I really like the structure and documentation of sounddevice, but I decided to keep developing with PyAudio for now.__ Sounddevice seemed to take more system resources than PyAudio (in my limited test conditions: Windows 10 with very fast and modern hardware, Python 3), and would audibly "glitch" music as it was being played every time it attached or detached from the microphone stream. I tried streaming, but after about an hour I couldn't get clean live access to the microphone without glitching audio playback. Furthermore, every few times I ran this script it crashed my python kernel! I very rarely see this happening. iPython complained: "_It seems the kernel died unexpectedly. Use 'Restart kernel' to continue using this console_" and I eventually moved back to PyAudio. For a less "realtime" application, sounddevice might be a great solution. Here's the minimal case sounddevice script I tested with (that crashed sometimes). If you have a better one to do live high-speed audio capture, let me know!

```python
import sounddevice #pip install sounddevice

for i in range(30): #30 updates in 1 second
    rec = sounddevice.rec(44100/30)
    sounddevice.wait()
    print(rec.shape)
```

__Here's a simple demo to show how I get realtime microphone audio into numpy arrays using PyAudio.__ This isn't really that special. It's a good starting point though. Note that rather than have the user define a microphone source in the python script (I had a fancy menu system handling this for a while), I allow PyAudio to just look at the operating system's default input device. This seems like a realistic expectation, and saves time as long as you don't expect your user to be recording from two different devices at the same time. This script gets some audio from the microphone and shows the values in the console (ten times).

```python
import pyaudio
import numpy as np

CHUNK = 4096 # number of data points to read at a time
RATE = 44100 # time resolution of the recording device (Hz)

p=pyaudio.PyAudio() # start the PyAudio class
stream=p.open(format=pyaudio.paInt16,channels=1,rate=RATE,input=True,
              frames_per_buffer=CHUNK) #uses default input device

# create a numpy array holding a single read of audio data
for i in range(10): #to it a few times just to see
    data = np.fromstring(stream.read(CHUNK),dtype=np.int16)
    print(data)

# close the stream gracefully
stream.stop_stream()
stream.close()
p.terminate()
```

<div class="text-center img-border">

[![](01_thumb.jpg)](01.png)

</div>

I tried to push the limit a little bit and see how much useful data I could get from this console window. It turns out that it's pretty responsive! Here's a slight modification of the code, made to turn the console window into an impromptu [VU meter](https://en.wikipedia.org/wiki/VU_meter).

```python
import pyaudio
import numpy as np

CHUNK = 2**11
RATE = 44100

p=pyaudio.PyAudio()
stream=p.open(format=pyaudio.paInt16,channels=1,rate=RATE,input=True,
              frames_per_buffer=CHUNK)

for i in range(int(10*44100/1024)): #go for a few seconds
    data = np.fromstring(stream.read(CHUNK),dtype=np.int16)
    peak=np.average(np.abs(data))*2
    bars="#"*int(50*peak/2**16)
    print("%04d %05d %s"%(i,peak,bars))

stream.stop_stream()
stream.close()
p.terminate()
```

### Result

The results are pretty good! The advantage here is that _no_ libraries are required except PyAudio. For people interested in doing simple math (peak detection, frequency detection, etc.) this is a perfect starting point. Here's a quick cellphone video:

![](https://www.youtube.com/embed/xUzyDPsesK8)

I've made [realtime audio visualization (realtime FFT) scripts with Python before](https://www.swharden.com/wp/2013-05-09-realtime-fft-audio-visualization-with-python/), but 80% of that code was creating a GUI. I want to see data in real time while I'm developing this code, but I _really_ don't want to mess with GUI programming. I then had a crazy idea. Everyone has a web browser, which is a pretty good GUI... with a Python script to analyze audio and save graphs (a lot of them, quickly) and some JavaScript running in a browser to keep refreshing those graphs, I could get an idea of what the audio stream is doing in something kind of like real time. It was intended to be a hack, but I never expected it to work so well! Check this out...

__Here's the python script to listen to the microphone and generate graphs:__

```python
import pyaudio
import numpy as np
import pylab
import time

RATE = 44100
CHUNK = int(RATE/20) # RATE / number of updates per second

def soundplot(stream):
    t1=time.time()
    data = np.fromstring(stream.read(CHUNK),dtype=np.int16)
    pylab.plot(data)
    pylab.title(i)
    pylab.grid()
    pylab.axis([0,len(data),-2**16/2,2**16/2])
    pylab.savefig("03.png",dpi=50)
    pylab.close('all')
    print("took %.02f ms"%((time.time()-t1)*1000))

if __name__=="__main__":
    p=pyaudio.PyAudio()
    stream=p.open(format=pyaudio.paInt16,channels=1,rate=RATE,input=True,
                  frames_per_buffer=CHUNK)
    for i in range(int(20*RATE/CHUNK)): #do this for 10 seconds
        soundplot(stream)
    stream.stop_stream()
    stream.close()
    p.terminate()
```

__Here's the HTML file with JavaScript to keep reloading the image...__

```html
<html>
<script language="javascript">
function RefreshImage(){
document.pic0.src="03.png?a=" + String(Math.random()*99999999);
setTimeout('RefreshImage()',50);
}
</script>
<body onload="RefreshImage()">
<img name="pic0" src="03.png">
</body>
</html>
```

### Operation

I couldn't believe my eyes. It's not elegant, but it's kind of functional!

![](https://www.youtube.com/embed/80lAehBMUbE)

__Why stop there?__ I went ahead and wrote a microphone listening and processing class which makes this stuff easier. My ultimate goal hasn't been revealed yet, but I'm sure it'll be clear in a few weeks. Let's just say there's a lot of use in me visualizing streams of continuous data. Anyway, this class is the truly _terrible_ attempt at a word pun by merging the words "SWH", "ear", and "Hear", into the official title "SWHear" which seems to be [unique on Google](https://www.google.com/search?q=%2Bpython+%2Bswhear). This class is minimal case, but can be easily modified to implement threaded recording (which won't cause the rest of the functions to hang) as well as mathematical manipulation of data, such as FFT. With the same HTML file as used above, here's the new python script and some video of the output:

```python
import pyaudio
import time
import pylab
import numpy as np

class SWHear(object):
    """
    The SWHear class is made to provide access to continuously recorded
    (and mathematically processed) microphone data.
    """

    def __init__(self,device=None,startStreaming=True):
        """fire up the SWHear class."""
        print(" -- initializing SWHear")

        self.chunk = 4096 # number of data points to read at a time
        self.rate = 44100 # time resolution of the recording device (Hz)

        # for tape recording (continuous "tape" of recent audio)
        self.tapeLength=2 #seconds
        self.tape=np.empty(self.rate*self.tapeLength)*np.nan

        self.p=pyaudio.PyAudio() # start the PyAudio class
        if startStreaming:
            self.stream_start()

    ### LOWEST LEVEL AUDIO ACCESS
    # pure access to microphone and stream operations
    # keep math, plotting, FFT, etc out of here.

    def stream_read(self):
        """return values for a single chunk"""
        data = np.fromstring(self.stream.read(self.chunk),dtype=np.int16)
        #print(data)
        return data

    def stream_start(self):
        """connect to the audio device and start a stream"""
        print(" -- stream started")
        self.stream=self.p.open(format=pyaudio.paInt16,channels=1,
                                rate=self.rate,input=True,
                                frames_per_buffer=self.chunk)

    def stream_stop(self):
        """close the stream but keep the PyAudio instance alive."""
        if 'stream' in locals():
            self.stream.stop_stream()
            self.stream.close()
        print(" -- stream CLOSED")

    def close(self):
        """gently detach from things."""
        self.stream_stop()
        self.p.terminate()

    ### TAPE METHODS
    # tape is like a circular magnetic ribbon of tape that's continously
    # recorded and recorded over in a loop. self.tape contains this data.
    # the newest data is always at the end. Don't modify data on the type,
    # but rather do math on it (like FFT) as you read from it.

    def tape_add(self):
        """add a single chunk to the tape."""
        self.tape[:-self.chunk]=self.tape[self.chunk:]
        self.tape[-self.chunk:]=self.stream_read()

    def tape_flush(self):
        """completely fill tape with new data."""
        readsInTape=int(self.rate*self.tapeLength/self.chunk)
        print(" -- flushing %d s tape with %dx%.2f ms reads"%\
                  (self.tapeLength,readsInTape,self.chunk/self.rate))
        for i in range(readsInTape):
            self.tape_add()

    def tape_forever(self,plotSec=.25):
        t1=0
        try:
            while True:
                self.tape_add()
                if (time.time()-t1)>plotSec:
                    t1=time.time()
                    self.tape_plot()
        except:
            print(" ~~ exception (keyboard?)")
            return

    def tape_plot(self,saveAs="03.png"):
        """plot what's in the tape."""
        pylab.plot(np.arange(len(self.tape))/self.rate,self.tape)
        pylab.axis([0,self.tapeLength,-2**16/2,2**16/2])
        if saveAs:
            t1=time.time()
            pylab.savefig(saveAs,dpi=50)
            print("plotting saving took %.02f ms"%((time.time()-t1)*1000))
        else:
            pylab.show()
            print() #good for IPython
        pylab.close('all')

if __name__=="__main__":
    ear=SWHear()
    ear.tape_forever()
    ear.close()
    print("DONE")
```

![](https://www.youtube.com/embed/E94MuHtdg6Y)

__I don't really intend anyone to actually do this,__ but it's a cool alternative to recording a small portion of audio, plotting it in a pop-up matplotlib window, and waiting for the user to close it to record a new fraction. I had a lot more text in here demonstrating real-time FFT, but I'd rather consolidate everything FFT related into a single post. For now, I'm happy pursuing microphone-related python projects with PyAudio.

### Display a single frequency

Use [Numpy's FFT() and FFTFREQ()](https://docs.scipy.org/doc/numpy/reference/routines.fft.html) to turn the linear data into frequency. Set that target and grab the FFT value corresponding to that frequency. I haven't tested this to be sure it's working, but it should at least be close...

```python
import pyaudio
import numpy as np
np.set_printoptions(suppress=True) # don't use scientific notation

CHUNK = 4096 # number of data points to read at a time
RATE = 44100 # time resolution of the recording device (Hz)
TARGET = 2100 # show only this one frequency

p=pyaudio.PyAudio() # start the PyAudio class
stream=p.open(format=pyaudio.paInt16,channels=1,rate=RATE,input=True,
              frames_per_buffer=CHUNK) #uses default input device

# create a numpy array holding a single read of audio data
for i in range(10): #to it a few times just to see
    data = np.fromstring(stream.read(CHUNK),dtype=np.int16)
    fft = abs(np.fft.fft(data).real)
    fft = fft[:int(len(fft)/2)] # keep only first half
    freq = np.fft.fftfreq(CHUNK,1.0/RATE)
    freq = freq[:int(len(freq)/2)] # keep only first half
    assert freq[-1]>TARGET, "ERROR: increase chunk size"
    val = fft[np.where(freq>TARGET)[0][0]]
    print(val)

# close the stream gracefully
stream.stop_stream()
stream.close()
p.terminate()

```

### Display Peak Frequency

If your goal is to determine which frequency is producing the loudest tone, use this function. I also added a few lines to graph the output in case you want to observe how it operates. I recommend testing this script with a tone generator, or a YouTube video containing tones of a range of frequencies [like this one](https://www.youtube.com/watch?v=WfbMNJj2C4I).

```python
import pyaudio
import numpy as np
import matplotlib.pyplot as plt

np.set_printoptions(suppress=True) # don't use scientific notation

CHUNK = 4096 # number of data points to read at a time
RATE = 44100 # time resolution of the recording device (Hz)

p=pyaudio.PyAudio() # start the PyAudio class
stream=p.open(format=pyaudio.paInt16,channels=1,rate=RATE,input=True,
              frames_per_buffer=CHUNK) #uses default input device

# create a numpy array holding a single read of audio data
for i in range(10): #to it a few times just to see
    data = np.fromstring(stream.read(CHUNK),dtype=np.int16)
    data = data * np.hanning(len(data)) # smooth the FFT by windowing data
    fft = abs(np.fft.fft(data).real)
    fft = fft[:int(len(fft)/2)] # keep only first half
    freq = np.fft.fftfreq(CHUNK,1.0/RATE)
    freq = freq[:int(len(freq)/2)] # keep only first half
    freqPeak = freq[np.where(fft==np.max(fft))[0][0]]+1
    print("peak frequency: %d Hz"%freqPeak)

    # uncomment this if you want to see what the freq vs FFT looks like
    #plt.plot(freq,fft)
    #plt.axis([0,4000,None,None])
    #plt.show()
    #plt.close()

# close the stream gracefully
stream.stop_stream()
stream.close()
p.terminate()
```

### Display Left and Right Levels

```python
import pyaudio
import numpy as np

maxValue = 2**16
p=pyaudio.PyAudio()
stream=p.open(format=pyaudio.paInt16,channels=2,rate=44100,
              input=True, frames_per_buffer=1024)
while True:
    data = np.fromstring(stream.read(1024),dtype=np.int16)
    dataL = data[0::2]
    dataR = data[1::2]
    peakL = np.abs(np.max(dataL)-np.min(dataL))/maxValue
    peakR = np.abs(np.max(dataR)-np.min(dataR))/maxValue
    print("L:%00.02f R:%00.02f"%(peakL*100, peakR*100))
```

__Output__

```
L:47.26 R:45.17
L:47.55 R:45.63
L:49.44 R:45.98
L:45.27 R:49.80
L:44.39 R:45.75
L:47.50 R:46.96
L:41.49 R:42.64
L:42.95 R:41.39
L:49.56 R:49.62
L:48.29 R:48.80
L:45.03 R:47.62
L:47.99 R:49.35
L:41.58 R:49.21

```

Or with a tweak...

```python
import pyaudio
import numpy as np

maxValue = 2**16
bars = 35
p=pyaudio.PyAudio()
stream=p.open(format=pyaudio.paInt16,channels=2,rate=44100,
              input=True, frames_per_buffer=1024)
while True:
    data = np.fromstring(stream.read(1024),dtype=np.int16)
    dataL = data[0::2]
    dataR = data[1::2]
    peakL = np.abs(np.max(dataL)-np.min(dataL))/maxValue
    peakR = np.abs(np.max(dataR)-np.min(dataR))/maxValue
    lString = "#"*int(peakL*bars)+"-"*int(bars-peakL*bars)
    rString = "#"*int(peakR*bars)+"-"*int(bars-peakR*bars)
    print("L=[%s]\tR=[%s]"%(lString, rString))
```

### Graphical Output

<div class="text-center img-border">

[![](audio-graphical_thumb.jpg)](audio-graphical.png)

</div>

Epoch Timestamp Hashing

I was recently presented with the need to rename a folder of images based on a timestamp. This way, I can keep saving new files in that folder with overlapping filenames (i.e., 01.jpg, 02.jpg, 03.jpg, etc.), and every time I run this script all images are prepended with a timestamp. I still want the files to be sorted alphabetically, which is why an alphabetical timestamp (rather than a random hash) is preferred.

  • At first I considered a long date such as 2014-04-19-01.jpg, but that adds so much text!...also, it doesn't include time of day.
  • If I include time of day, it becomes 2014-04-19-09-16-23-01.jpg
  • If I eliminate dashes to shorten it, it becomes hard to read, but might work 140419091623-01.jpg
  • If I use Unix Epoch time, it becomes 1397912944-01.jpg

The result I came up with uses base conversion and a string table of numbers and letters (in alphabetical order) to create a second-respecting timestamp hash using an arbitrary number of characters. For simplicity, I used 36 characters: 0-9, and a-z. I then wrote two functions to perform arbitrary base conversion, pulling characters from the hash. Although I could have nearly doubled my available characters by including the full ASCII table, respecting capitalization, I decided to keep it simple. The scheme goes like this:

  • Determine the date / time: 19-Apr-2014 13:08:55
  • Create an integer of Unix Epoch time (seconds past Jan 1, 1970): 1397912935
  • Do a base conversion from a character list: n4a4iv
  • My file name now becomes n4a4iv-01.jpg - I can accept this!and when I sort the folder alphabetically, they're in order by the timestamp

I can now represent any modern time, down to the second, with 6 characters. Here's some example output:

19-Apr-2014 13:08:55 <-> 1397912935 <-> n4a4iv
19-Apr-2014 13:08:56 <-> 1397912936 <-> n4a4iw
19-Apr-2014 13:08:57 <-> 1397912937 <-> n4a4ix
19-Apr-2014 13:08:58 <-> 1397912938 <-> n4a4iy
19-Apr-2014 13:08:59 <-> 1397912939 <-> n4a4iz
19-Apr-2014 13:09:00 <-> 1397912940 <-> n4a4j0
19-Apr-2014 13:09:01 <-> 1397912941 <-> n4a4j1
19-Apr-2014 13:09:02 <-> 1397912942 <-> n4a4j2
19-Apr-2014 13:09:03 <-> 1397912943 <-> n4a4j3
19-Apr-2014 13:09:04 <-> 1397912944 <-> n4a4j4

Interestingly, if I change my hash characters away from the list of 36 alphanumerics and replace it with just 0 and 1, I can encode/decode the date in binary:

19-Apr-2014 13:27:28 <-> 1397914048 <-> 1010011010100100111100111000000
19-Apr-2014 13:27:29 <-> 1397914049 <-> 1010011010100100111100111000001
19-Apr-2014 13:27:30 <-> 1397914050 <-> 1010011010100100111100111000010
19-Apr-2014 13:27:31 <-> 1397914051 <-> 1010011010100100111100111000011
19-Apr-2014 13:27:32 <-> 1397914052 <-> 1010011010100100111100111000100
19-Apr-2014 13:27:33 <-> 1397914053 <-> 1010011010100100111100111000101
19-Apr-2014 13:27:34 <-> 1397914054 <-> 1010011010100100111100111000110
19-Apr-2014 13:27:35 <-> 1397914055 <-> 1010011010100100111100111000111
19-Apr-2014 13:27:36 <-> 1397914056 <-> 1010011010100100111100111001000
19-Apr-2014 13:27:37 <-> 1397914057 <-> 1010011010100100111100111001001

Here's the code to generate / decode Unix epoch timestamps in Python:

hashchars='0123456789abcdefghijklmnopqrstuvwxyz'
#hashchars='01' #for binary

def epochToHash(n):
  hash=''
  while n>0:
    hash = hashchars[int(n % len(hashchars))] + hash
    n = int(n / len(hashchars))
  return hash

def epochFromHash(s):
  s=s[::-1]
  epoch=0
  for pos in range(len(s)):
    epoch+=hashchars.find(s[pos])*(len(hashchars)**pos)
  return epoch

import time
t=int(time.time())
for i in range(10):
  t=t+1
  print(time.strftime("%d-%b-%Y %H:%M:%S", time.gmtime(t)),
              "<->", t,"<->",epochToHash(t))
Markdown source code last modified on January 18th, 2021
---
title: Epoch Timestamp Hashing
date: 2014-04-19 08:31:53
tags: python, old
---

# Epoch Timestamp Hashing

__I was recently presented with the need to rename a folder of images based on a timestamp.__ This way, I can keep saving new files in that folder with overlapping filenames (i.e., <span style="color: #339966;">01.jpg</span>, <span style="color: #339966;">02.jpg</span>, <span style="color: #339966;">03.jpg</span>, etc.), and every time I run this script all images are prepended with a timestamp. I still want the files to be sorted alphabetically, which is why an alphabetical timestamp (rather than a random hash) is preferred.

*   At first I considered a long date such as <span style="color: #339966;">2014-04-19-01.jpg</span>, but that adds so much text!...also, it doesn't include time of day.
*   If I include time of day, it becomes <span style="color: #339966;">2014-04-19-09-16-23-01.jpg</span>
*   If I eliminate dashes to shorten it, it becomes hard to read, but might work <span style="color: #339966;">140419091623-01.jpg</span>
*   If I use [Unix Epoch](http://en.wikipedia.org/wiki/Unix_time) time, it becomes <span style="color: #339966;">1397912944-01.jpg</span>

__The result I came up with uses base conversion and a string table of numbers and letters (in alphabetical order) to create a second-respecting timestamp hash using an arbitrary number of characters.__ For simplicity, I used 36 characters: 0-9, and a-z. I then wrote two functions to perform arbitrary base conversion, pulling characters from the hash. Although I could have nearly doubled my available characters by including the full ASCII table, respecting capitalization, I decided to keep it simple. The scheme goes like this:

*   Determine the date / time: <span style="color: #339966;">19-Apr-2014 13:08:55</span>
*   Create an integer of [Unix Epoch](http://en.wikipedia.org/wiki/Unix_time) time (seconds past Jan 1, 1970):  <span style="color: #339966;">1397912935</span>
*   Do a base conversion from a character list: <span style="color: #339966;">n4a4iv</span>
*   My file name now becomes <span style="color: #888888;">n4a4iv-01.jpg</span> - I can accept this!_and when I sort the folder alphabetically, they're in order by the timestamp_

__I can now represent any modern time, down to the second, with 6 characters.__ Here's some example output:

```python
19-Apr-2014 13:08:55 <-> 1397912935 <-> n4a4iv
19-Apr-2014 13:08:56 <-> 1397912936 <-> n4a4iw
19-Apr-2014 13:08:57 <-> 1397912937 <-> n4a4ix
19-Apr-2014 13:08:58 <-> 1397912938 <-> n4a4iy
19-Apr-2014 13:08:59 <-> 1397912939 <-> n4a4iz
19-Apr-2014 13:09:00 <-> 1397912940 <-> n4a4j0
19-Apr-2014 13:09:01 <-> 1397912941 <-> n4a4j1
19-Apr-2014 13:09:02 <-> 1397912942 <-> n4a4j2
19-Apr-2014 13:09:03 <-> 1397912943 <-> n4a4j3
19-Apr-2014 13:09:04 <-> 1397912944 <-> n4a4j4
```

__Interestingly, if I change my hash characters away from the list of 36 alphanumerics and replace it with just 0 and 1, I can encode/decode the date in binary:__

```python
19-Apr-2014 13:27:28 <-> 1397914048 <-> 1010011010100100111100111000000
19-Apr-2014 13:27:29 <-> 1397914049 <-> 1010011010100100111100111000001
19-Apr-2014 13:27:30 <-> 1397914050 <-> 1010011010100100111100111000010
19-Apr-2014 13:27:31 <-> 1397914051 <-> 1010011010100100111100111000011
19-Apr-2014 13:27:32 <-> 1397914052 <-> 1010011010100100111100111000100
19-Apr-2014 13:27:33 <-> 1397914053 <-> 1010011010100100111100111000101
19-Apr-2014 13:27:34 <-> 1397914054 <-> 1010011010100100111100111000110
19-Apr-2014 13:27:35 <-> 1397914055 <-> 1010011010100100111100111000111
19-Apr-2014 13:27:36 <-> 1397914056 <-> 1010011010100100111100111001000
19-Apr-2014 13:27:37 <-> 1397914057 <-> 1010011010100100111100111001001
```

__Here's the code to generate / decode Unix epoch timestamps in Python:__

```python
hashchars='0123456789abcdefghijklmnopqrstuvwxyz'
#hashchars='01' #for binary

def epochToHash(n):
  hash=''
  while n>0:
    hash = hashchars[int(n % len(hashchars))] + hash
    n = int(n / len(hashchars))
  return hash

def epochFromHash(s):
  s=s[::-1]
  epoch=0
  for pos in range(len(s)):
    epoch+=hashchars.find(s[pos])*(len(hashchars)**pos)
  return epoch

import time
t=int(time.time())
for i in range(10):
  t=t+1
  print(time.strftime("%d-%b-%Y %H:%M:%S", time.gmtime(t)),
              "<->", t,"<->",epochToHash(t))
```

Calculate QRSS Transmission Time with Python

How long does a particular bit of Morse code take to transmit at a certain speed? This is a simple question, but when sitting down trying to design schemes for 10-minute-window QRSS, it doesn't always have a quick and simple answer. Yeah, you could sit down and draw the pattern on paper and add-up the dots and dashes, but why do on paper what you can do in code? The following speaks for itself. I made the top line say my call sign in Morse code (AJ4VD), and the program does the rest. I now see that it takes 570 seconds to transmit AJ4VD at QRSS 10 speed (ten second dots), giving me 30 seconds of free time to kill.

Output of the following script, displaying info about "AJ4VD" (my call sign).

Here's the Python code I whipped-up to generate the results:

xmit=" .- .--- ....- ...- -..  " #callsign
dot,dash,space,seq="_-","_---","_",""
for c in xmit:
    if c==" ": seq+=space
    elif c==".": seq+=dot
    elif c=="-": seq+=dash
print "QRSS sequence:n",seq,"n"
for sec in [1,3,5,10,20,30,60]:
    tot=len(seq)*sec
    print "QRSS %02d: %d sec (%.01f min)"%(sec,tot,tot/60.0)

How ready am I to implement this in the microchip? Pretty darn close. I've got a surprisingly stable software-based time keeping solution running continuously executing a "tick()" function thanks to hardware interrupts. It was made easy thanks to Frank Zhao's AVR Timer Calculator. I could get it more exact by using a /1 prescaler instead of a /64, but this well within the range of acceptability so I'm calling it quits!

Output frequency is 1.0000210 Hz. That'll drift 2.59 sec/day. I'm cool with that.

Markdown source code last modified on January 18th, 2021
---
title: Calculate QRSS Transmission Time with Python
date: 2013-06-23 21:19:59
tags: qrss, python, old
---

# Calculate QRSS Transmission Time with Python

__How long does a particular bit of Morse code take to transmit at a certain speed?__ This is a simple question, but when sitting down trying to design schemes for 10-minute-window QRSS, it doesn't always have a quick and simple answer. Yeah, you could sit down and draw the pattern on paper and add-up the dots and dashes, but why do on paper what you can do in code? The following speaks for itself. I made the top line say my call sign in Morse code (AJ4VD), and the program does the rest. I now see that it takes 570 seconds to transmit AJ4VD at QRSS 10 speed (ten second dots), giving me 30 seconds of free time to kill.


<div class="text-center">

[![](qrss-calclator_thumb.jpg)](qrss-calclator.png)

</div>

Output of the following script, displaying info about "AJ4VD" (my call sign).

__Here's the Python code I whipped-up to generate the results:__

```python
xmit=" .- .--- ....- ...- -..  " #callsign
dot,dash,space,seq="_-","_---","_",""
for c in xmit:
    if c==" ": seq+=space
    elif c==".": seq+=dot
    elif c=="-": seq+=dash
print "QRSS sequence:n",seq,"n"
for sec in [1,3,5,10,20,30,60]:
    tot=len(seq)*sec
    print "QRSS %02d: %d sec (%.01f min)"%(sec,tot,tot/60.0)
```

__How ready am I to implement this in the microchip? __Pretty darn close. I've got a surprisingly stable software-based time keeping solution running continuously executing a "tick()" function thanks to hardware interrupts. It was made easy thanks to [Frank Zhao's AVR Timer Calculator](http://www.frank-zhao.com/cache/avrtimercalc.php). I could get it more exact by using a /1 prescaler instead of a /64, but this well within the range of acceptability so I'm calling it quits!


<div class="text-center img-border">

[![](photo-11_thumb.jpg)](photo-11.jpg)

</div>

Output frequency is 1.0000210 Hz. That'll drift 2.59 sec/day. I'm cool with that.

Realtime image pixelmap from Numpy array data in Qt

Consider realtime spectrograph software like QRSS VD. It's primary function is to scroll a potentially huge data-rich image across the screen. In Python, this is often easier said than done.__ If you're not careful, you can tackle this problem inefficiently and get terrible frame rates (<5FPS) or eat a huge amount of system resources (I get complaints often that QRSS VD takes up a lot of processor resources, and 99% of it is drawing the images). In the past, I've done it at least 4 different ways (one, two, three, four, five). Note that "four" seems to be the absolute fastest option so far. I've been keeping an eye out for a while now contemplating the best way to rapidly draw color-mapped 8-bit data in a python program. Now that I'm doing a majority of my graphical development with PyQt and QtDesigner (packaged with PythonXY), I ended-up with a solution that looks like this (plotting random data with a colormap):

1.) in QtDesigner, create a form with a scrollAreaWidget

2.) in QtDesigner, add a label inside the scrollAreaWidget

3.) in code, resize label and also scrollAreaWidgetContents to fit data (disable "widgetResizable")

4.) in code, create a QImage from a 2D numpy array (dtype=uint8)

5.) in code, set label pixmap to QtGui.QPixmap.fromImage(QImage)

That's pretty much it! Here are some highlights of my program. Note that the code for the GUI is in a separate file, and must be downloaded from the ZIP provided at the bottom. Hope it helps someone else out there who might want to do something similar!

import ui_main
import sys
from PyQt4 import QtCore, QtGui

import sys
from PyQt4 import Qt
import PyQt4.Qwt5 as Qwt
from PIL import Image
import numpy
import time

spectroWidth=1000
spectroHeight=1000

a=numpy.random.random(spectroHeight*spectroWidth)*255
a=numpy.reshape(a,(spectroHeight,spectroWidth))
a=numpy.require(a, numpy.uint8, 'C')

COLORTABLE=[]
for i in range(256): COLORTABLE.append(QtGui.qRgb(i/4,i,i/2))

def updateData():
    global a
    a=numpy.roll(a,-5)
    QI=QtGui.QImage(a.data, spectroWidth, spectroHeight, QtGui.QImage.Format_Indexed8)
    QI.setColorTable(COLORTABLE)
    uimain.label.setPixmap(QtGui.QPixmap.fromImage(QI))

if __name__ == "__main__":
    app = QtGui.QApplication(sys.argv)
    win_main = ui_main.QtGui.QWidget()
    uimain = ui_main.Ui_win_main()
    uimain.setupUi(win_main)

    # SET UP IMAGE
    uimain.IM = QtGui.QImage(spectroWidth, spectroHeight, QtGui.QImage.Format_Indexed8)
    uimain.label.setGeometry(QtCore.QRect(0,0,spectroWidth,spectroHeight))
    uimain.scrollAreaWidgetContents.setGeometry(QtCore.QRect(0,0,spectroWidth,spectroHeight))

    # SET UP RECURRING EVENTS
    uimain.timer = QtCore.QTimer()
    uimain.timer.start(.1)
    win_main.connect(uimain.timer, QtCore.SIGNAL('timeout()'), updateData)

    ### DISPLAY WINDOWS
    win_main.show()
    sys.exit(app.exec_())
Markdown source code last modified on January 18th, 2021
---
title: Realtime image pixelmap from Numpy array data in Qt
date: 2013-06-03 22:40:56
tags: python, old
---

# Realtime image pixelmap from Numpy array data in Qt

Consider realtime spectrograph software like [QRSS VD](http://www.swharden.com/blog/qrss_vd/#screenshots).  It's primary function is to scroll a potentially huge data-rich image across the screen. In Python, this is often easier said than done.__ If you're not careful, you can tackle this problem inefficiently and get terrible frame rates (<5FPS) or eat a huge amount of system resources (I get complaints often that QRSS VD takes up a lot of processor resources, and 99% of it is drawing the images).  In the past, I've done it at least 4 different ways ([one](http://www.swharden.com/blog/2010-03-05-animated-realtime-spectrograph-with-scrolling-waterfall-display-in-python/), [two](http://www.swharden.com/blog/2013-05-09-realtime-fft-audio-visualization-with-python/), [three](http://www.swharden.com/blog/qrss_vd/#screenshots), [four](http://www.swharden.com/blog/2010-06-24-fast-tk-pixelmap-generation-from-2d-numpy-arrays-in-python/), [five](http://www.swharden.com/blog/2010-03-05-realtime-fft-graph-of-audio-wav-file-or-microphone-input-with-python-scipy-and-wckgraph/)). Note that "four" seems to be the absolute fastest option so far. I've been keeping an eye out for a while now contemplating the best way to rapidly draw color-mapped 8-bit data in a python program. Now that I'm doing a majority of my graphical development with PyQt and QtDesigner (packaged with [PythonXY](https://code.google.com/p/pythonxy/)), I ended-up with a solution that looks like this (plotting random data with a colormap):


<div class="text-center img-border">

![](qt-scrolling-spectrograph.gif)

</div>

1.) in QtDesigner, create a form with a **scrollAreaWidget**

2.) in QtDesigner, add a **label** inside the **scrollAreaWidget**

3.) in code, resize **label** and also **scrollAreaWidgetContents **to fit data (disable "widgetResizable")

4.) in code, create a **QImage** from a 2D numpy array (dtype=uint8)

5.) in code, set **label** pixmap to QtGui.QPixmap.fromImage(**QImage**)

That's pretty much it! Here are some highlights of my program. Note that the code for the GUI is in a separate file, and must be downloaded from the ZIP provided at the bottom. Hope it helps someone else out there who might want to do something similar!

```python
import ui_main
import sys
from PyQt4 import QtCore, QtGui

import sys
from PyQt4 import Qt
import PyQt4.Qwt5 as Qwt
from PIL import Image
import numpy
import time

spectroWidth=1000
spectroHeight=1000

a=numpy.random.random(spectroHeight*spectroWidth)*255
a=numpy.reshape(a,(spectroHeight,spectroWidth))
a=numpy.require(a, numpy.uint8, 'C')

COLORTABLE=[]
for i in range(256): COLORTABLE.append(QtGui.qRgb(i/4,i,i/2))

def updateData():
    global a
    a=numpy.roll(a,-5)
    QI=QtGui.QImage(a.data, spectroWidth, spectroHeight, QtGui.QImage.Format_Indexed8)
    QI.setColorTable(COLORTABLE)
    uimain.label.setPixmap(QtGui.QPixmap.fromImage(QI))

if __name__ == "__main__":
    app = QtGui.QApplication(sys.argv)
    win_main = ui_main.QtGui.QWidget()
    uimain = ui_main.Ui_win_main()
    uimain.setupUi(win_main)

    # SET UP IMAGE
    uimain.IM = QtGui.QImage(spectroWidth, spectroHeight, QtGui.QImage.Format_Indexed8)
    uimain.label.setGeometry(QtCore.QRect(0,0,spectroWidth,spectroHeight))
    uimain.scrollAreaWidgetContents.setGeometry(QtCore.QRect(0,0,spectroWidth,spectroHeight))

    # SET UP RECURRING EVENTS
    uimain.timer = QtCore.QTimer()
    uimain.timer.start(.1)
    win_main.connect(uimain.timer, QtCore.SIGNAL('timeout()'), updateData)

    ### DISPLAY WINDOWS
    win_main.show()
    sys.exit(app.exec_())
```

Wireless Microcontroller / PC Interface for $3.21

Here I demonstrate a dirt-cheap method of transmitting data from any microchip to any PC using $3.21 in parts. I've had this idea for a while, but finally got it working tonight. On the transmit side, I'm having a an ATMEL AVR microcontroller (ATMega48) transmit data (every number from 0 to 200 over and over) wirelessly using 433mhz wireless modules. The PC receives the data through the microphone port of a sound card, and a cross-platform Python script I wrote decodes the data from the audio and graphs it on the screen. I did something similar back in 2011, but it wasn't wireless, and the software wasn't nearly as robust as it is now.

This is a proof-of-concept demonstration, and part of a larger project. I think there's a need for this type of thing though! It's unnecessarily hard to transfer data from a MCU to a PC as it is. There's USB (For AVR V-USB is a nightmare and requires a precise, specific clock speed, DIP chips don't have native USB, and some PIC DIP chips do but then you have to go through driver hell), USART RS-232 over serial port works (but who has serial ports these days?), or USART over USB RS-232 interface chips (like FTDI FT-232, but surface mount only), but both also require precise, specific clock speeds. Pretend I want to just measure temperature once a minute. Do I really want to etch circuit boards and solder SMT components? Well, kinda, but I don't like feeling forced to. Some times you just want a no-nonsense way to get some numbers from your microchip to your computer. This project is a funky out-of-the-box alternative to traditional methods, and one that I hope will raise a few eyebrows.

Ultimately, I designed this project to eventually allow multiple "bursting" data transmitters to transmit on the same frequency routinely, thanks to syncing and forced-sync-loss (read on). It's part of what I'm tongue-in-cheek calling the Scott Harden RF Protocol (SH-RFP). In my goal application, I wish to have about 5 wireless temperature sensors all transmitting data to my PC. The receive side has some error checking in that it makes sure pulse sizes are intelligent and symmetrical (unlike random noise), and since each number is sent twice (with the second time being in reverse), there's another layer of error-detection. This is *NOT* a robust and accurate method to send critical data. It's a cheap way to send data. It is very range limited, and only is intended to work over a distance of ten or twenty feet. First, let's see it in action!

The RF modules are pretty simple. At 1.56 on ebay (with free shipping), they're cheap too! I won't go into detail documenting the ins and out of these things (that's done well elsewhere). Briefly, you give them +5V (VCC), 0V (GND), and flip their data pin (ATAD) on and off on the transmitter module, and the receiver module's DATA pin reflects the same state. The receiver uses a gain circuit which continuously increases gain until signal is detected, so if you're not transmitting it WILL decode noise and start flipping its output pin. Note that persistent high or low states are prone to noise too, so any protocol you use these things for should have rapid state transitions. It's also suggested that you maintain an average 50% duty cycle. These modules utilize amplitude shift keying (ASK) to transmit data wirelessly. The graphic below shows what that looks like at the RF level. Transmit and receive is improved by adding a quarter-wavelength vertical antenna to the "ANT" solder pad. At 433MHz, that is about 17cm, so I'm using a 17cm copper wire as an antenna.

Transmitting from the microcontroller is easy as pie! It’s just a matter of copying-in a few lines of C. It doesn’t rely on USART, SPI, I2C, or any other protocol. Part of why I developed this method is because I often use ATTiny44A which doesn’t have USART for serial interfacing. The “SH-RFP” is easy to implement just by adding a few lines of code. I can handle that. How does it work? I can define it simply by a few rules:

  • Pulses can be one of 3 lengths: A (0), B (1), or C (break).
  • Each pulse represents high, then low of that length.

To send a packet:

  • prime synchronization by sending ten ABCs
  • indicate we’re starting data by sending C.
  • for each number you want to send:
  • send your number bit by bit (A=0, B=1)
  • send your number bit by bit (A=1, B=0)
  • indicate number end by sending C.
  • tell PC to release the signal by sending ten Cs.

Decoding is the same thing in reverse. I use an eBay sound card at $1.29 (with free shipping) to get the signal into the PC. Synchronization is required to allow the PC to know that real data (not noise) is starting. Sending the same number twice (once with reversed bit polarity) is a proof-checking mechanisms that lets us throw-out data that isn’t accurate.

From a software side, I’m using PyAudio to collect data from the sound card, and the PythonXY distribution to handle analysis with numpy, scipy, and plotting with QwtPlot, and general GUI functionality with PyQt. I think that’s about everything.

The demonstration interface is pretty self-explanatory. The top-right shows a sample piece of data. The top left is a histogram of the number of samples of each pulse width. A clean signal should have 3 pulses (A=0, B=1, C=break). Note that you’re supposed to look at the peaks to determine the best lengths to tell the software to use to distinguish A, B, and C. This was intentionally not hard-coded because I want to rapidly switch from one microcontroller platform to another which may be operating at a different clock speed, and if all the sudden it’s running 3 times slower it will be no problem to decide on the PC side. Slick, huh? The bottom-left shows data values coming in. The bottom-right graphs those values. Rate reporting lets us know that I'm receiving over 700 good data points a second. That's pretty cool, especially considering I'm recording at 44,100 Hz.

All source code (C files for an ATMega48 and Python scripts for the GUI) can be viewed here: SHRFP project on GitHub

If you use these concepts, hardware, or ideas in your project, let me know about it! Send me an email showing me your project – I’d love to see it. Good luck!

Markdown source code last modified on January 18th, 2021
---
title: Wireless Microcontroller / PC Interface for $3.21
date: 2013-05-19 01:32:46
tags: microcontroller, old, python
---

# Wireless Microcontroller / PC Interface for $3.21

__Here I demonstrate a dirt-cheap method of transmitting data from any microchip to any PC using $3.21 in parts.  __I've had this idea for a while, but finally got it working tonight. On the transmit side, I'm having a an ATMEL AVR microcontroller (ATMega48) transmit data (every number from 0 to 200 over and over) wirelessly using 433mhz wireless modules. The PC receives the data through the microphone port of a sound card, and a cross-platform Python script I wrote decodes the data from the audio and graphs it on the screen. I [did something similar back in 2011](http://www.swharden.com/blog/2011-07-09-sound-card-microcontrollerpc-communication/), but it wasn't wireless, and the software wasn't nearly as robust as it is now.

__This is a proof-of-concept demonstration, and part of a larger project.__ I think there's a need for this type of thing though! It's unnecessarily hard to transfer data from a MCU to a PC as it is. There's USB (For AVR [V-USB](http://www.obdev.at/products/vusb/index.html) is a nightmare and requires a precise, specific clock speed, DIP chips don't have native USB, and some PIC DIP chips do but then you have to go through driver hell), [USART RS-232 over serial port](http://www.swharden.com/blog/2009-05-14-simple-case-avrpc-serial-communication-via-max232/) works (but who has serial ports these days?), or USART over USB RS-232 interface chips (like [FTDI FT-232](http://www.ftdichip.com/Products/ICs/FT232R.htm), but surface mount only), but both also require precise, specific clock speeds. Pretend I want to just measure temperature once a minute. Do I _really_ want to etch circuit boards and solder SMT components? Well, kinda, but I don't like feeling forced to. Some times you just want a no-nonsense way to get some numbers from your microchip to your computer. This project is a funky out-of-the-box alternative to traditional methods, and one that I hope will raise a few eyebrows.

<div class="text-center img-border">

[![](c31_thumb.jpg)](c31.jpg)

</div>

__Ultimately, I designed this project to eventually allow multiple "bursting" data transmitters to transmit on the same frequency__ __routinely__, thanks to syncing and forced-sync-loss (read on). It's part of what I'm tongue-in-cheek calling the _Scott Harden RF Protocol_ (SH-RFP). In my goal application, I wish to have about 5 wireless temperature sensors all transmitting data to my PC.  The receive side has some error checking in that it makes sure pulse sizes are intelligent and symmetrical (unlike random noise), and since each number is sent twice (with the second time being in reverse), there's another layer of error-detection.  This is \*NOT\* a robust and accurate method to send critical data. It's a cheap way to send data. It is very range limited, and only is intended to work over a distance of ten or twenty feet. First, let's see it in action!

![](https://www.youtube.com/embed/GJHFldPwZvM)

__The RF modules are pretty simple. [At 1.56 on ebay](http://www.ebay.com/itm/KDQ11-NEW-1PCS-433MHZ-RF-TRANSMITTER-AND-RECEIVER-LINK-KIT-FOR-ARDUINO-SCA-1710-/350797631746?pt=LH_DefaultDomain_0&hash=item51ad2b1102) (with free shipping), they're cheap too!__ I won't go into detail documenting the ins and out of these things (that's done well elsewhere). Briefly, you give them +5V (VCC), 0V (GND), and flip their data pin (ATAD) on and off on the transmitter module, and the receiver module's DATA pin reflects the same state. The receiver uses a gain circuit which continuously increases gain until signal is detected, so if you're not transmitting it WILL decode noise and start flipping its output pin. Note that persistent high or low states are prone to noise too, so any protocol you use these things for should have rapid state transitions. It's also suggested that you maintain an average 50% duty cycle. These modules utilize [amplitude shift keying](http://en.wikipedia.org/wiki/Amplitude-shift_keying) (ASK) to transmit data wirelessly. The graphic below shows what that looks like at the RF level. Transmit and receive is improved by adding a quarter-wavelength vertical antenna to the "ANT" solder pad. At 433MHz, that is about 17cm, so I'm using a 17cm copper wire as an antenna.

__Transmitting from the microcontroller is easy as pie!__ It’s just a matter of copying-in a few lines of C.  It doesn’t rely on USART, SPI, I2C, or any other protocol. Part of why I developed this method is because I often use ATTiny44A which doesn’t have USART for serial interfacing. The “SH-RFP” is easy to implement just by adding a few lines of code. I can handle that.  How does it work? I can define it simply by a few rules:

*   Pulses can be one of 3 lengths: A (0), B (1), or C (break).
*   Each pulse represents high, then low of that length.

To send a packet:

*   prime synchronization by sending ten ABCs
*   indicate we’re starting data by sending C.
*   for each number you want to send:
  *   send your number bit by bit (A=0, B=1)
  *   send your number bit by bit (A=1, B=0)
  *   indicate number end by sending C.
*   tell PC to release the signal by sending ten Cs.

Decoding is the same thing in reverse. I use an [eBay sound card at $1.29](search.ebay.com/usb-sound-card) (with free shipping) to get the signal into the PC. </span> Synchronization is required to allow the PC to know that real data (not noise) is starting. Sending the same number twice (once with reversed bit polarity) is a proof-checking mechanisms that lets us throw-out data that isn’t accurate.

__From a software side,__ I’m using PyAudio to collect data from the sound card, and the PythonXY distribution to handle analysis with numpy, scipy, and plotting with QwtPlot, and general GUI functionality with PyQt. I think that’s about everything.

<div class="text-center img-border">

[![](SHRFP_thumb.jpg)](SHRFP.png)

</div>

__The demonstration interface is pretty self-explanatory.__ The top-right shows a sample piece of data. The top left is a histogram of the number of samples of each pulse width. A clean signal should have 3 pulses (A=0, B=1, C=break). Note that you’re supposed to look at the peaks to determine the best lengths to tell the software to use to distinguish A, B, and C. This was intentionally not hard-coded because I want to rapidly switch from one microcontroller platform to another which may be operating at a different clock speed, and if all the sudden it’s running 3 times slower it will be no problem to decide on the PC side. Slick, huh? The bottom-left shows data values coming in. The bottom-right graphs those values. Rate reporting lets us know that I'm receiving over 700 good data points a second. That's pretty cool, especially considering I'm recording at 44,100 Hz.

All source code (C files for an ATMega48 and Python scripts for the GUI) can be viewed here: [SHRFP project on GitHub](https://github.com/swharden/AVR-projects/tree/master/ATMega48%202013-05-14%20SHRFP%20monitor)

If you use these concepts, hardware, or ideas in your project, let me know about it! Send me an email showing me your project – I’d love to see it. Good luck!

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