AudioToText/audiotospeech.py

# -*- coding: utf-8 -*-

# IMPORTS
import os
import numpy as np
import requests
import tensorflow as tf
from tensorflow import keras
from tensorflow.keras import layers

# MODEL STUFF
# The set of characters accepted in the transcription.
characters = [x for x in "abcdefghijklmnopqrstuvwxyz'?! "]
# Mapping characters to integers
char_to_num = keras.layers.StringLookup(vocabulary=characters, oov_token="")
# Mapping integers back to original characters
num_to_char = keras.layers.StringLookup(
    vocabulary=char_to_num.get_vocabulary(), oov_token="", invert=True
)


# An integer scalar Tensor. The window length in samples.
frame_length = 256
# An integer scalar Tensor. The number of samples to step.
frame_step = 160
# An integer scalar Tensor. The size of the FFT to apply.
# If not provided, uses the smallest power of 2 enclosing frame_length.
fft_length = 384

# MODEL LOSS
def CTCLoss(y_true, y_pred):
    # Compute the training-time loss value
    batch_len = tf.cast(tf.shape(y_true)[0], dtype="int64")
    input_length = tf.cast(tf.shape(y_pred)[1], dtype="int64")
    label_length = tf.cast(tf.shape(y_true)[1], dtype="int64")

    input_length = input_length * tf.ones(shape=(batch_len, 1), dtype="int64")
    label_length = label_length * tf.ones(shape=(batch_len, 1), dtype="int64")

    loss = keras.backend.ctc_batch_cost(y_true, y_pred, input_length, label_length)
    return loss

# BUILD MODEL
def build_model(input_dim, output_dim, rnn_layers=5, rnn_units=128):
    """Model similar to DeepSpeech2."""
    # Model's input
    input_spectrogram = layers.Input((None, input_dim), name="input")
    # Expand the dimension to use 2D CNN.
    x = layers.Reshape((-1, input_dim, 1), name="expand_dim")(input_spectrogram)
    # Convolution layer 1
    x = layers.Conv2D(
        filters=32,
        kernel_size=[11, 41],
        strides=[2, 2],
        padding="same",
        use_bias=False,
        name="conv_1",
    )(x)
    x = layers.BatchNormalization(name="conv_1_bn")(x)
    x = layers.ReLU(name="conv_1_relu")(x)
    # Convolution layer 2
    x = layers.Conv2D(
        filters=32,
        kernel_size=[11, 21],
        strides=[1, 2],
        padding="same",
        use_bias=False,
        name="conv_2",
    )(x)
    x = layers.BatchNormalization(name="conv_2_bn")(x)
    x = layers.ReLU(name="conv_2_relu")(x)
    # Reshape the resulted volume to feed the RNNs layers
    x = layers.Reshape((-1, x.shape[-2] * x.shape[-1]))(x)
    # RNN layers
    for i in range(1, rnn_layers + 1):
        recurrent = layers.GRU(
            units=rnn_units,
            activation="tanh",
            recurrent_activation="sigmoid",
            use_bias=True,
            return_sequences=True,
            reset_after=True,
            name=f"gru_{i}",
        )
        x = layers.Bidirectional(
            recurrent, name=f"bidirectional_{i}", merge_mode="concat"
        )(x)
        if i < rnn_layers:
            x = layers.Dropout(rate=0.5)(x)
    # Dense layer
    x = layers.Dense(units=rnn_units * 2, name="dense_1")(x)
    x = layers.ReLU(name="dense_1_relu")(x)
    x = layers.Dropout(rate=0.5)(x)
    # Classification layer
    output = layers.Dense(units=output_dim + 1, activation="softmax")(x)
    # Model
    model = keras.Model(input_spectrogram, output, name="DeepSpeech_2")
    # Optimizer
    opt = keras.optimizers.Adam(learning_rate=1e-4)
    # Compile the model and return
    model.compile(optimizer=opt, loss=CTCLoss)
    return model

# GET AND INSTANTIATE MODEL
model = build_model(
    input_dim = fft_length // 2 + 1,
    output_dim = char_to_num.vocabulary_size(),
    rnn_units = 512,
)


# GET TEXT FROM MODEL PREDICTION
# A utility function to decode the output of the network
def decode_batch_predictions(pred):
    input_len = np.ones(pred.shape[0]) * pred.shape[1]
    # Use greedy search. For complex tasks, you can use beam search
    results = keras.backend.ctc_decode(pred, input_length=input_len, greedy=True)[0][0]
    # Iterate over the results and get back the text
    output_text = []
    for result in results:
        result = tf.strings.reduce_join(num_to_char(result)).numpy().decode("utf-8")
        output_text.append(result)
    return output_text


# PATH TO CKPT
# google share link
ckpt_link = 'https://drive.google.com/file/d/14mT_wJMuIqUEJSS12aAc6bnPCjYuLWGf/view?usp=sharing'

# Define the local filename to save data
local_file = 'AudioToTextCKPT.hdf5'

# Make http request for remote file data
data = requests.get(ckpt_link)

# Save file data to local copy
with open(local_file, 'wb')as file:
    file.write(data.content)

ckpt = local_file


# LOAD CKPT TO MODEL
model.load_weights(ckpt)

# CONVERT AUDIO TO TEXT
def AudioToText(wav_file):
    ###########################################
    ##  Process the Audio
    ##########################################
    # 1. Read wav file
    file = tf.io.read_file(wav_file)
    # 2. Decode the wav file
    audio, _ = tf.audio.decode_wav(file)
    audio = tf.squeeze(audio, axis=-1)
    # 3. Change type to float
    audio = tf.cast(audio, tf.float32)
    # 4. Get the spectrogram
    spectrogram = tf.signal.stft(
        audio, frame_length=frame_length, frame_step=frame_step, fft_length=fft_length
    )
    # 5. We only need the magnitude, which can be derived by applying tf.abs
    spectrogram = tf.abs(spectrogram)
    spectrogram = tf.math.pow(spectrogram, 0.5)
    # 6. normalisation
    means = tf.math.reduce_mean(spectrogram, 1, keepdims=True)
    stddevs = tf.math.reduce_std(spectrogram, 1, keepdims=True)
    spectrogram = (spectrogram - means) / (stddevs + 1e-10)

    pred = model.predict(spectrogram)
    
    output_text = decode_batch_predictions(pred)

    return output_text


# testing model
print(AudioToText('testWav.wav'))