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Reading Wave Files
On the Windows platform, the most common file extension for
audio files is "wav". MATLAB can
read such wave files via the command "wavread". The following
example reads the wave file
"sunday.wav" and display its waveform directly.
Example 1.
[y, fs]=wavread('sunday.wav');
sound(y, fs); % Playback of the sound data
(?????)time=(1:length(y))/fs; % Time vector on x-axis (??????)
plot(time, y)
In the above example, "fs" is the sample rate which is 16000 in
this case. This indicatesthat there are 44100 samples per second
when the clip was recorded. The vector "y" is a
column vector containing the samples of the speech signals. We
can use "sound(y, fs)"to play the audio signals read from the file.
"time" is a time vector in which each elementcorresponds to the
time of each sample. Therefore we can plot "y" against "t" to show
thewaveform directly.
Most audio signals are digitized to have a bit resolution of 8
or 16 bits. If we want to knowthe bit resolution of the input wave
file, we can use an extra output arguments to"wavread" to obtain
the information, such as
[y, fs, nbits]=wavread(sunday.wav');
Moreover, if we want to know the time duration of a stream of
audio signals, we can use"length(y)/fs" directly. The following
example can obtain most of the important information
of the wave file "sunday.wav".
Example 2
fileName='sunday.wav';[y, fs, nbits]=wavread(fileName);
sound(y, fs); % Playback of the sound data
(?????)time=(1:length(y))/fs; % Time vector on x-axis (??????)
plot(time, y)fprintf('Information of the sound file "%s":\n',
fileName);
fprintf('Duration = %g seconds\n', length(y)/fs);
fprintf('Sampling rate = %g samples/second\n', fs);fprintf('Bit
resolution = %g bits/sample\n', nbits);
From the above example, it can be observed that all the audio
signals are between -1and 1. However, each sample point is
represented by an 8-bit interger. How are theyrelated? First of
all, we need to know the following convention:
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1. If a wave file has a bit resolution of 8 bits, then each
sample point is stored as anunsigned integer between 0 and 255 (=
2^8-1).
2.
If a wave file has a bit resolution of 16 bits, then each sample
point is stored as anunsigned integer between -32768 (= 2^16/2) and
32767 (= 2^16/2-1).
Since almost all variables in MATLAB have the data type of
"double", therefore allsamples are converted into a floating-point
number between -1 and 1 for easymanipulation. Therefore to retrieve
the original integer values of the audio signals, we canproceed as
follows.
1. For 8-bit resolution, we can multiply "y" (the value obtained
by wavread) by 128and then plus 128.
2. For 16-bit resolution, we can multiply "y" (the value
obtained by wavread) by32768.
Here is an example.
Example 3
fileName='sunday.wav';
[y, fs, nbits]=wavread(fileName);
y0=y*(2^nbits/2)+(2^nbits/2); % y0 is the original values stored
in the wav
file
difference=sum(abs(y0-round(y0)))
In the above example, the difference is zero, indicating the
retrived y0 contains no fractional parts.
We can also use the command "wavread" to read a stereo wave
file. The returned variable willbe a matrix of 2 columns, each
containing the audio signals from a single channel. Example
follows.
Example 4
fileName='flanger.wav';
[y, fs]=wavread(fileName); % Read wave file
sound(y, fs); % Playback
left=y(:,1); % Left channel
right=y(:,2); % Right channelsubplot(2,1,1),
plot((1:length(left))/fs, left);
subplot(2,1,2), plot((1:length(right))/fs, right);
In the above example, MATLAB will read the wave file
"flanger.wav", play the stereo sound, and
plot two streams of audio signals in two subplots. Since the
intensities of these two channels are
more or less complemntary to each other, which gives an illusion
that the sound source is moving
back and forth between two speakers.
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If the wave file is too large to be read into memory directly,
we can also use "wavread" to read a
part of the audio signals directly. See the following
example.
