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import torch
from speechbrain.inference.interfaces import Pretrained
import librosa
import numpy as np
import torchaudio
import os
class ASR(Pretrained):
def __init__(self, *args, **kwargs):
super().__init__(*args, **kwargs)
def encode_batch_w2v2(self, device, wavs, wav_lens=None, normalize=False):
wavs = wavs.to(device)
wav_lens = wav_lens.to(device)
# Forward pass
encoded_outputs = self.mods.encoder_w2v2(wavs.detach())
# append
tokens_bos = torch.zeros((wavs.size(0), 1), dtype=torch.long).to(device)
embedded_tokens = self.mods.embedding(tokens_bos)
decoder_outputs, _ = self.mods.decoder(embedded_tokens, encoded_outputs, wav_lens)
# Output layer for seq2seq log-probabilities
predictions = self.hparams.test_search(encoded_outputs, wav_lens)[0]
# predicted_words = [self.hparams.tokenizer.decode_ids(prediction).split(" ") for prediction in predictions]
predicted_words = []
for prediction in predictions:
prediction = [token for token in prediction if token != 0]
predicted_words.append(self.hparams.tokenizer.decode_ids(prediction).split(" "))
prediction = []
for sent in predicted_words:
sent = self.filter_repetitions(sent, 3)
prediction.append(sent)
predicted_words = prediction
return predicted_words
def filter_repetitions(self, seq, max_repetition_length):
seq = list(seq)
output = []
max_n = len(seq) // 2
for n in range(max_n, 0, -1):
max_repetitions = max(max_repetition_length // n, 1)
# Don't need to iterate over impossible n values:
# len(seq) can change a lot during iteration
if (len(seq) <= n*2) or (len(seq) <= max_repetition_length):
continue
iterator = enumerate(seq)
# Fill first buffers:
buffers = [[next(iterator)[1]] for _ in range(n)]
for seq_index, token in iterator:
current_buffer = seq_index % n
if token != buffers[current_buffer][-1]:
# No repeat, we can flush some tokens
buf_len = sum(map(len, buffers))
flush_start = (current_buffer-buf_len) % n
# Keep n-1 tokens, but possibly mark some for removal
for flush_index in range(buf_len - buf_len%n):
if (buf_len - flush_index) > n-1:
to_flush = buffers[(flush_index + flush_start) % n].pop(0)
else:
to_flush = None
# Here, repetitions get removed:
if (flush_index // n < max_repetitions) and to_flush is not None:
output.append(to_flush)
elif (flush_index // n >= max_repetitions) and to_flush is None:
output.append(to_flush)
buffers[current_buffer].append(token)
# At the end, final flush
current_buffer += 1
buf_len = sum(map(len, buffers))
flush_start = (current_buffer-buf_len) % n
for flush_index in range(buf_len):
to_flush = buffers[(flush_index + flush_start) % n].pop(0)
# Here, repetitions just get removed:
if flush_index // n < max_repetitions:
output.append(to_flush)
seq = []
to_delete = 0
for token in output:
if token is None:
to_delete += 1
elif to_delete > 0:
to_delete -= 1
else:
seq.append(token)
output = []
return seq
def classify_file_w2v2(self, file, vad_model, device):
# Get audio length in seconds
sr = 16000
max_segment_length = 30
# waveform, sr = librosa.load(file, sr=sr)
waveform, file_sr = torchaudio.load(file)
# resample if not 16kHz
if file_sr != sr:
waveform = torchaudio.transforms.Resample(file_sr, sr)(waveform)
# limit to 1 min
waveform = waveform[:, :60*sr]
waveform = waveform.squeeze()
audio_length = len(waveform) / sr
print(f"Audio length: {audio_length:.2f} seconds")
if audio_length >= max_segment_length:
print(f"Audio is too long ({audio_length:.2f} seconds), splitting into segments")
# save waveform temporarily
torchaudio.save("temp.wav", waveform.unsqueeze(0), sr)
# get boundaries based on VAD
boundaries = vad_model.get_speech_segments("temp.wav",
large_chunk_size=30,
small_chunk_size=10,
apply_energy_VAD=True,
double_check=True)
# remove temp file
os.remove("temp.wav")
# Merge the segments to max max_segment_length
segments = []
current_start = boundaries[0][0].item()
current_end = boundaries[0][1].item()
for i in range(1, len(boundaries)):
next_start = boundaries[i][0].item()
next_end = boundaries[i][1].item()
# Check if the current segment can merge with the next segment
if (current_end - current_start) + (next_end - next_start) <= max_segment_length:
# Extend the current segment
current_end = next_end
else:
# Add the current segment to the result and start a new one
segments.append([current_start, current_end])
current_start = next_start
current_end = next_end
# Add the last segment
segments.append([current_start, current_end])
# Process each segment
outputs = []
for i, segment in enumerate(segments):
start, end = segment
start = int(start * sr)
end = int(end * sr)
segment = waveform[start:end]
print(f"Processing segment {i + 1}/{len(segments)}, length: {len(segment) / sr:.2f} seconds")
# import soundfile as sf
# sf.write(f"outputs/segment_{i}.wav", segment, sr)
segment_tensor = torch.tensor(segment).to(device)
# Fake a batch for the segment
batch = segment_tensor.unsqueeze(0).to(device)
rel_length = torch.tensor([1.0]).to(device) # Adjust if necessary
# Pass the segment through the ASR model
result = " ".join(self.encode_batch_w2v2(device, batch, rel_length)[0])
# outputs.append(result)
yield result
else:
waveform, file_sr = torchaudio.load(file)
# resample if not 16kHz
if file_sr != sr:
waveform = torchaudio.transforms.Resample(file_sr, sr)(waveform)
waveform = waveform.to(device)
rel_length = torch.tensor([1.0]).to(device)
outputs = " ".join(self.encode_batch_w2v2(device, waveform, rel_length)[0])
yield outputs
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