An AudioStream is a container that holds Buffers, it also stores a sample rate, providing information about time. It is meant to easily manipulate multiple channels of audio at the same time. Most Nsound functions that operate on Buffers also operate on AudioStreams.
There are 3 general ways to create an AudioStream:
Call the constructor:
from Nsound import * a = AudioStream()
The default constructor sets the sample rate to 44100.0 and a single channel. The new AudioStream a is empty. Calling the getLength() method will return 0. Calling the getDuration() method will return 0.0.
To specify the sample rate:
a = AudioStream(44100.0)
This will create a new AudioStream with a sample rate of 44.1 kHz and a single channel. To create a stereo AudioStream:
a = AudioStream(44100.0, 2)
The is no limit to the number of channels an AudioStream can have. In practice, the number of channels and their duration will be limited to the amount of memory your computer has.
The underlying data structure that is held by the AudioStream class is a std::vector of Buffer objects. One can preallocate memory when creating an AudioStream by specify the number of samples to preallocate:
a = AudioStream(44100.0, 2, 1024)
The new AudioStream a is empty, even though memory was preallocated. Calling the getLength() method will return 0, calling getDuration() will return 0.0.
In general, you don’t need to worry about preallocating memory. It is meant to be useful when implementing new features in Nsound when the size of AudioStreams are already known.
The Buffer class includes some convience functions for creating Buffers that are filled with oness, random numbers or zeros:
from Nsound import * a1 = AudioStream.ones(44100.0, 2, 1.0) a2 = AudioStream.rand(44100.0, 2, 1.0) a3 = AudioStream.zeros(44100.0, 2, 1.0)
In the example above, 44100 samples (1 second) were stored in 2 channels.
An AudioStream can be created from a wavefile:
a = AudioStream("california.wav")
The new AudioStream a will contain all the samples in “california.wav”. If the wavefile has more than one channel, all channels are read and stored in the new AudioStream. The wavefile’s sample rate is also read and stored in the AudioStream.
The wavefile’s data will be converted into float64 with a range of (-1.0, 1.0.).