Below are links to pieces of music and recordings of several notes. You are required to "transcribe" the music.
For transcription, you will have to determine the note or set of notes being played at each time.
This tgz file contains a recording of a harmonica piece rendering the song "blowin' in the wind". Also given are a collection of notes, and an example musical scale. Transcribe both the musical scale and the main song in terms of the notes.
This is a recording of "Polyshka Polye", played on the harmonica. It has been downloaded from youtube (with permission from the artist).
Below are a set of notes from a harmonica.
Note | Wav File |
E | e.wav |
F | f.wav |
G | g.wav |
A | a.wav |
B | b.wav |
C | c.wav |
D | d.wav |
E2 | e2.wav |
F2 | f2.wav |
G2 | g2.wav |
A2 | a2.wav |
Download the following matlab files: stft.m
You can read a wav file into matlab as follows:
[s,fs] = wavread('filename');
s = resample(s,16000,fs);
The recordings of the notes can be computed to a spectrum as follows:
spectrum = mean(abs(stft(s',2048,256,0,hann(2048))),2);
“spectrum
” will be a 1025 x 1 vector.
The recordings of the complete music can be read just as you read the notes. To convert it to a spectrogram do the following:
sft = stft(s',2048,256,0,hann(2048));
sphase = sft./abs(sft);
smag = abs(sft);
“smag
” will be a 1025 x K matrix (K is the number of spectral vectors in the matrix. We will also need “sphase
” to reconstruct the signal later.
Compute the spectrum for each of the notes. Compute the spectrogram matrix “smag
” for the music signal. This matrix is composed of K spectral vectors. Each vector represents 16 milliseconds of the signal.
You may find, projections, pseudo inverses, and dot products useful. If you know of any other techniques, you can use those too. Tricks like thresholding (setting all values of some variable that fall below a threshold to 0) might also help.
The output should be of the form of a matrix :
1 | 1 | 0 | 0 | 0 | 0 | 0 | 1 | . | . | . |
0 | 0 | 0 | 1 | 1 | 0 | 1 | 1 | . | . | . |
0 | 1 | 1 | 1 | 0 | 1 | 1 | 1 | . | . | . |
. | . | . | . | . | . | . | . | . | . | . |
Each row of the matrix represents one note. Hence there will be as many rows as you have notes in table 1.
Each column represents one of the columns in the spectrogram for the music. So if there are K vectors in the spectrogram, there will be K vectors in your output.
Each entry will denote if a note was found in that vector or not. For instance, if matrix entry (4,25) = 0, then the fourth note (d) was not found in the 25th spectral vector of the signal.
In the previous problems, the harmonica (or piano) notes that produce music when combined in some manner can be thought of as basis vectors for the music. That is to say, a linear combination of those vectors produces the music. Later in the class, we will look at methods that try to learn such bases automatically for different types of audio, such as a person's speech, music from a certain instrument, background noise etc.
One of the problems with learning this representation automatically is that the audio used to learn the bases for a certain kind of sound must contain that sound only and no other. If it contains other sound, or even an absence of sound, then the model that is learnt will capture something other than the intended source. It is easy to see that gathering real data for a task like this will result in non-optimal data sets.
Consider the case where the system is attempting to learn a set of basis vectors for a particular speaker. Listen to the data here for an example of data for the speaker collected in a natural environment. You will notice that the speaker is not speaking in the initial and end portions of the audio file. Also, there is background noise from an idling automobile throughout the video. Even if the recording was controlled in order to ensure that the speaker was speaking for the duration of the audio, the background noise would still appear. However, at the very least, we would like to extract the segments of audio where the speaker is present. One simple heuristic that is used for this kind of task is called thresholding.
Comment briefly on your observations. What differences do you see between the 2 sets of plots, why do you think there is/should be a difference, how do these differences change with the level of background noise, etc.
Solutions may be emailed to me, Sourish or Sohail. The message must have the subject line "MLSP assignment 1". It should include a 1 page report of what you did (can be longer), and the resulting matrix, as well as the synthesized audoi.