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code.py
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code.py
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import librosa
import soundfile
import os, glob, pickle
import numpy as np
from sklearn.model_selection import train_test_split
from sklearn.neural_network import MLPClassifier
from sklearn.metrics import accuracy_score
# Extract features (mfcc, chroma, mel) from a sound file
def extract_feature(file_name, mfcc, chroma, mel):
with soundfile.SoundFile(file_name) as sound_file:
X = sound_file.read(dtype="float32")
sample_rate = sound_file.samplerate
if chroma:
stft = np.abs(librosa.stft(X))
result = np.array([])
if mfcc:
mfccs = np.mean(librosa.feature.mfcc(y=X, sr=sample_rate, n_mfcc=40).T, axis=0)
result = np.hstack((result, mfccs))
if chroma:
chroma = np.mean(librosa.feature.chroma_stft(S=stft, sr=sample_rate).T, axis=0)
result = np.hstack((result, chroma))
if mel:
mel = np.mean(librosa.feature.melspectrogram(X, sr=sample_rate).T, axis=0)
result = np.hstack((result, mel))
return result
emotions = {
'01': 'neutral',
'02': 'calm',
'03': 'happy',
'04': 'sad',
'05': 'angry',
'06': 'fearful',
'07': 'disgust',
'08': 'surprised'
}
# Emotions to observe
observed_emotions = ['happy', 'fearful', 'angry', 'sad']
# Load the data and extract features for each sound file
def load_data(test_size=0.2):
x, y = [], []
for file in glob.glob(r"C:\Users\Prashanth\Ravdess data\\Actor_*\\*.wav"):
file_name = os.path.basename(file)
emotion = emotions[file_name.split("-")[2]]
if emotion not in observed_emotions:
continue
feature = extract_feature(file, mfcc=True, chroma=True, mel=True)
x.append(feature)
y.append(emotion)
return train_test_split(np.array(x), y, test_size=test_size, random_state=9)
# Split the dataset
x_train, x_test, y_train, y_test = load_data(test_size=0.25)
# Get the shape of the training and testing datasets
print((x_train.shape[0], x_test.shape[0]))
# Get the number of features extracted
print(f'Features extracted: {x_train.shape[1]}')
# Initialize the Multi Layer Perceptron Classifier
model = MLPClassifier(alpha=0.01, batch_size=256, epsilon=1e-08, hidden_layer_sizes=(300,), learning_rate='adaptive',
max_iter=500)
# Train the model
model.fit(x_train, y_train)
# Train the model
model.fit(x_train, y_train)
# Predict for the test set
y_pred = model.predict(x_test)
# Calculate the accuracy of our model
accuracy = accuracy_score(y_true=y_test, y_pred=y_pred)
# DataFlair - Print the accuracy
print("Accuracy: {:.2f}%".format(accuracy * 100))
if not os.path.isdir("result"):
os.mkdir("result")
pickle.dump(model, open("result/mlp_classifier.model", "wb"))
import pyaudio
import os
import wave
import pickle
from sys import byteorder
from array import array
from struct import pack
from sklearn.neural_network import MLPClassifier
THRESHOLD = 500
CHUNK_SIZE = 1024
FORMAT = pyaudio.paInt16
RATE = 16000
SILENCE = 30
def is_silent(snd_data):
"Returns 'True' if below the 'silent' threshold"
return max(snd_data) < THRESHOLD
def normalize(snd_data):
"Average the volume out"
MAXIMUM = 16384
times = float(MAXIMUM) / max(abs(i) for i in snd_data)
r = array('h')
for i in snd_data:
r.append(int(i * times))
return r
def trim(snd_data):
"Trim the blank spots at the start and end"
def _trim(snd_data):
snd_started = False
r = array('h')
for i in snd_data:
if not snd_started and abs(i) > THRESHOLD:
snd_started = True
r.append(i)
elif snd_started:
r.append(i)
return r
# Trim to the left
snd_data = _trim(snd_data)
# Trim to the right
snd_data.reverse()
snd_data = _trim(snd_data)
snd_data.reverse()
return snd_data
def add_silence(snd_data, seconds):
"Add silence to the start and end of 'snd_data' of length 'seconds' (float)"
r = array('h', [0 for i in range(int(seconds * RATE))])
r.extend(snd_data)
r.extend([0 for i in range(int(seconds * RATE))])
return r
def record():
"""
Record a word or words from the microphone and
return the data as an array of signed shorts.
Normalizes the audio, trims silence from the
start and end, and pads with 0.5 seconds of
blank sound to make sure VLC et al can play
it without getting chopped off.
"""
p = pyaudio.PyAudio()
stream = p.open(format=FORMAT, channels=1, rate=RATE,
input=True, output=True,
frames_per_buffer=CHUNK_SIZE)
num_silent = 0
snd_started = False
r = array('h')
while 1:
# little endian, signed short
snd_data = array('h', stream.read(CHUNK_SIZE))
if byteorder == 'big':
snd_data.byteswap()
r.extend(snd_data)
silent = is_silent(snd_data)
if silent and snd_started:
num_silent += 1
elif not silent and not snd_started:
snd_started = True
if snd_started and num_silent > SILENCE:
break
sample_width = p.get_sample_size(FORMAT)
stream.stop_stream()
stream.close()
p.terminate()
r = normalize(r)
r = trim(r)
r = add_silence(r, 0.5)
return sample_width, r
def record_to_file(path):
"Records from the microphone and outputs the resulting data to 'path'"
sample_width, data = record()
data = pack('<' + ('h' * len(data)), *data)
wf = wave.open(path, 'wb')
wf.setnchannels(1)
wf.setsampwidth(sample_width)
wf.setframerate(RATE)
wf.writeframes(data)
wf.close()
if __name__ == "__main__":
# load the saved model (after training)
model = pickle.load(open("result/mlp_classifier.model", "rb"))
print("Please talk")
filename = "test.wav"
# record the file (start talking)
record_to_file(filename)
# extract features and reshape it
features = extract_feature(filename, mfcc=True, chroma=True, mel=True).reshape(1, -1)
# predict
result = model.predict(features)[0]
# show the result !
print("result:", result)