Audio

This module allows you play sounds from a speaker attached to the micro:bit.

The audio module can be imported as import audio or accessed via the microbit module as microbit.audio.

In order to use the audio module you will need to provide a sound source.

A sound source is an iterable (sequence, like list or tuple, or a generator) of frames, each of 32 samples. The audio modules plays samples at the rate of 7812.5 samples per second, which means that it can reproduce frequencies up to 3.9kHz.

Functions

audio.play(source, wait=True, pin=pin0, return_pin=None)

Play the source to completion.

Parameters:
  • source – An iterable sound source, each element of which must be an AudioFrame.

  • wait – If wait is True, this function will block until the source is exhausted.

  • pin – Specifies which pin the speaker is connected to.

  • return_pin – Specifies a differential pin to connect to the speaker instead of ground.

Classes

class audio.AudioFrame

An AudioFrame object is a list of 32 samples each of which is an unsigned byte (whole number between 0 and 255).

It takes just over 4 ms to play a single frame.

copyfrom(other)

Overwrite the data in this AudioFrame with the data from another AudioFrame instance.

Parameters:

otherAudioFrame instance from which to copy the data.

Using audio

You will need a sound source, as input to the play function. You can generate your own, like in examples/waveforms.py.

Technical Details

Note

You don’t need to understand this section to use the audio module. It is just here in case you wanted to know how it works.

The audio module can consumes an iterable (sequence, like list or tuple, or generator) of AudioFrame instances, each 32 samples at 7812.5 Hz, and uses linear interpolation to output a PWM signal at 32.5 kHz, which gives tolerable sound quality.

The function play fully copies all data from each AudioFrame before it calls next() for the next frame, so a sound source can use the same AudioFrame repeatedly.

The audio module has an internal 64 sample buffer from which it reads samples. When reading reaches the start or the mid-point of the buffer, it triggers a callback to fetch the next AudioFrame which is then copied into the buffer. This means that a sound source has under 4ms to compute the next AudioFrame, and for reliable operation needs to take less 2ms (which is 32000 cycles, so should be plenty).

Example

from microbit import display, sleep, button_a
import audio
import math

def repeated_frame(frame, count):
    for i in range(count):
        yield frame

# Press button A to skip to next wave.
def show_wave(name, frame, duration=1500):
    display.scroll(name + " wave", wait=False,delay=100)
    audio.play(repeated_frame(frame, duration),wait=False)
    for i in range(75):
        sleep(100)
        if button_a.is_pressed():
            display.clear()
            audio.stop()
            break

frame = audio.AudioFrame()

for i in range(len(frame)):
    frame[i] = int(math.sin(math.pi*i/16)*124+128.5)
show_wave("Sine", frame)

triangle = audio.AudioFrame()

QUARTER = len(triangle)//4
for i in range(QUARTER):
    triangle[i] = i*15
    triangle[i+QUARTER] = 248-i*15
    triangle[i+QUARTER*2] = 128-i*15
    triangle[i+QUARTER*3] = i*15+8
show_wave("Triangle", triangle)

square = audio.AudioFrame()

HALF = len(square)//2
for i in range(HALF):
    square[i] = 8
    square[i+HALF] = 248
show_wave("Square", square)
sleep(1000)

for i in range(len(frame)):
    frame[i] = 252-i*8
show_wave("Sawtooth", frame)

del frame

#Generate a waveform that goes from triangle to square wave, reasonably smoothly.
frames = [ None ] * 32
for i in range(32):
    frames[i] = frame = audio.AudioFrame()
    for j in range(len(triangle)):
        frame[j] = (triangle[j]*(32-i) + square[j]*i)>>5

def repeated_frames(frames, count):
    for frame in frames:
        for i in range(count):
            yield frame


display.scroll("Ascending wave", wait=False)
audio.play(repeated_frames(frames, 60))