#!/usr/bin/env python3 import sys import torch import logging import gradio as gr import speechbrain as sb from pathlib import Path import os import torchaudio from hyperpyyaml import load_hyperpyyaml from speechbrain.tokenizers.SentencePiece import SentencePiece from speechbrain.utils.data_utils import undo_padding from speechbrain.utils.distributed import run_on_main """Recipe for training a sequence-to-sequence ASR system with CommonVoice. The system employs a wav2vec2 encoder and a CTC decoder. Decoding is performed with greedy decoding (will be extended to beam search). To run this recipe, do the following: > python train_with_wav2vec2.py hparams/train_with_wav2vec2.yaml With the default hyperparameters, the system employs a pretrained wav2vec2 encoder. The wav2vec2 model is pretrained following the model given in the hprams file. It may be dependent on the language. The neural network is trained with CTC on sub-word units estimated with Byte Pairwise Encoding (BPE). The experiment file is flexible enough to support a large variety of different systems. By properly changing the parameter files, you can try different encoders, decoders, tokens (e.g, characters instead of BPE), training languages (all CommonVoice languages), and many other possible variations. Authors * Titouan Parcollet 2021 """ logger = logging.getLogger(__name__) # Define training procedure class ASR(sb.core.Brain): def compute_forward(self, batch, stage): """Forward computations from the waveform batches to the output probabilities.""" batch = batch.to(self.device) wavs, wav_lens = batch.sig wavs, wav_lens = wavs.to(self.device), wav_lens.to(self.device) if stage == sb.Stage.TRAIN: if hasattr(self.hparams, "augmentation"): wavs = self.hparams.augmentation(wavs, wav_lens) # Forward pass feats = self.modules.wav2vec2(wavs, wav_lens) x = self.modules.enc(feats) logits = self.modules.ctc_lin(x) p_ctc = self.hparams.log_softmax(logits) return p_ctc, wav_lens def treat_wav(self, sig): feats = self.modules.wav2vec2(sig.to("cpu"), torch.tensor([1]).to("cpu")) feats = self.modules.enc(feats) logits = self.modules.ctc_lin(feats) p_ctc = self.hparams.log_softmax(logits) predicted_words = [] for logs in p_ctc: text = decoder.decode(logs.detach().cpu().numpy()) predicted_words.append(text.split(" ")) return " ".join(predicted_words[0]) def compute_objectives(self, predictions, batch, stage): """Computes the loss (CTC) given predictions and targets.""" p_ctc, wav_lens = predictions ids = batch.id tokens, tokens_lens = batch.tokens loss = self.hparams.ctc_cost(p_ctc, tokens, wav_lens, tokens_lens) if stage != sb.Stage.TRAIN: predicted_tokens = sb.decoders.ctc_greedy_decode( p_ctc, wav_lens, blank_id=self.hparams.blank_index ) # Decode token terms to words if self.hparams.use_language_modelling: predicted_words = [] for logs in p_ctc: text = decoder.decode(logs.detach().cpu().numpy()) predicted_words.append(text.split(" ")) else: predicted_words = [ "".join(self.tokenizer.decode_ndim(utt_seq)).split(" ") for utt_seq in predicted_tokens ] # Convert indices to words target_words = [wrd.split(" ") for wrd in batch.wrd] self.wer_metric.append(ids, predicted_words, target_words) self.cer_metric.append(ids, predicted_words, target_words) return loss def fit_batch(self, batch): """Train the parameters given a single batch in input""" should_step = self.step % self.grad_accumulation_factor == 0 # Managing automatic mixed precision # TOFIX: CTC fine-tuning currently is unstable # This is certainly due to CTC being done in fp16 instead of fp32 if self.auto_mix_prec: with torch.cuda.amp.autocast(): with self.no_sync(): outputs = self.compute_forward(batch, sb.Stage.TRAIN) loss = self.compute_objectives(outputs, batch, sb.Stage.TRAIN) with self.no_sync(not should_step): self.scaler.scale( loss / self.grad_accumulation_factor ).backward() if should_step: if not self.hparams.wav2vec2.freeze: self.scaler.unscale_(self.wav2vec_optimizer) self.scaler.unscale_(self.model_optimizer) if self.check_gradients(loss): if not self.hparams.wav2vec2.freeze: if self.optimizer_step >= self.hparams.warmup_steps: self.scaler.step(self.wav2vec_optimizer) self.scaler.step(self.model_optimizer) self.scaler.update() self.zero_grad() self.optimizer_step += 1 else: # This is mandatory because HF models have a weird behavior with DDP # on the forward pass with self.no_sync(): outputs = self.compute_forward(batch, sb.Stage.TRAIN) loss = self.compute_objectives(outputs, batch, sb.Stage.TRAIN) with self.no_sync(not should_step): (loss / self.grad_accumulation_factor).backward() if should_step: if self.check_gradients(loss): if not self.hparams.wav2vec2.freeze: if self.optimizer_step >= self.hparams.warmup_steps: self.wav2vec_optimizer.step() self.model_optimizer.step() self.zero_grad() self.optimizer_step += 1 self.on_fit_batch_end(batch, outputs, loss, should_step) return loss.detach().cpu() def evaluate_batch(self, batch, stage): """Computations needed for validation/test batches""" predictions = self.compute_forward(batch, stage=stage) with torch.no_grad(): loss = self.compute_objectives(predictions, batch, stage=stage) return loss.detach() def on_stage_start(self, stage, epoch): """Gets called at the beginning of each epoch""" if stage != sb.Stage.TRAIN: self.cer_metric = self.hparams.cer_computer() self.wer_metric = self.hparams.error_rate_computer() def on_stage_end(self, stage, stage_loss, epoch): """Gets called at the end of an epoch.""" # Compute/store important stats stage_stats = {"loss": stage_loss} if stage == sb.Stage.TRAIN: self.train_stats = stage_stats else: stage_stats["CER"] = self.cer_metric.summarize("error_rate") stage_stats["WER"] = self.wer_metric.summarize("error_rate") # Perform end-of-iteration things, like annealing, logging, etc. if stage == sb.Stage.VALID: old_lr_model, new_lr_model = self.hparams.lr_annealing_model( stage_stats["loss"] ) old_lr_wav2vec, new_lr_wav2vec = self.hparams.lr_annealing_wav2vec( stage_stats["loss"] ) sb.nnet.schedulers.update_learning_rate( self.model_optimizer, new_lr_model ) if not self.hparams.wav2vec2.freeze: sb.nnet.schedulers.update_learning_rate( self.wav2vec_optimizer, new_lr_wav2vec ) self.hparams.train_logger.log_stats( stats_meta={ "epoch": epoch, "lr_model": old_lr_model, "lr_wav2vec": old_lr_wav2vec, }, train_stats=self.train_stats, valid_stats=stage_stats, ) self.checkpointer.save_and_keep_only( meta={"WER": stage_stats["WER"]}, min_keys=["WER"], ) elif stage == sb.Stage.TEST: self.hparams.train_logger.log_stats( stats_meta={"Epoch loaded": self.hparams.epoch_counter.current}, test_stats=stage_stats, ) with open(self.hparams.wer_file, "w") as w: self.wer_metric.write_stats(w) def init_optimizers(self): "Initializes the wav2vec2 optimizer and model optimizer" # If the wav2vec encoder is unfrozen, we create the optimizer if not self.hparams.wav2vec2.freeze: self.wav2vec_optimizer = self.hparams.wav2vec_opt_class( self.modules.wav2vec2.parameters() ) if self.checkpointer is not None: self.checkpointer.add_recoverable( "wav2vec_opt", self.wav2vec_optimizer ) self.model_optimizer = self.hparams.model_opt_class( self.hparams.model.parameters() ) if self.checkpointer is not None: self.checkpointer.add_recoverable("modelopt", self.model_optimizer) def zero_grad(self, set_to_none=False): if not self.hparams.wav2vec2.freeze: self.wav2vec_optimizer.zero_grad(set_to_none) self.model_optimizer.zero_grad(set_to_none) # Define custom data procedure def dataio_prepare(hparams): """This function prepares the datasets to be used in the brain class. It also defines the data processing pipeline through user-defined functions.""" # 1. Define datasets data_folder = hparams["data_folder"] train_data = sb.dataio.dataset.DynamicItemDataset.from_csv( csv_path=hparams["train_csv"], replacements={"data_root": data_folder}, ) if hparams["sorting"] == "ascending": # we sort training data to speed up training and get better results. train_data = train_data.filtered_sorted( sort_key="duration", key_max_value={"duration": hparams["avoid_if_longer_than"]}, ) # when sorting do not shuffle in dataloader ! otherwise is pointless hparams["dataloader_options"]["shuffle"] = False elif hparams["sorting"] == "descending": train_data = train_data.filtered_sorted( sort_key="duration", reverse=True, key_max_value={"duration": hparams["avoid_if_longer_than"]}, ) # when sorting do not shuffle in dataloader ! otherwise is pointless hparams["dataloader_options"]["shuffle"] = False elif hparams["sorting"] == "random": pass else: raise NotImplementedError( "sorting must be random, ascending or descending" ) valid_data = sb.dataio.dataset.DynamicItemDataset.from_csv( csv_path=hparams["valid_csv"], replacements={"data_root": data_folder}, ) # We also sort the validation data so it is faster to validate valid_data = valid_data.filtered_sorted(sort_key="duration") test_datasets = {} for csv_file in hparams["test_csv"]: name = Path(csv_file).stem test_datasets[name] = sb.dataio.dataset.DynamicItemDataset.from_csv( csv_path=csv_file, replacements={"data_root": data_folder} ) test_datasets[name] = test_datasets[name].filtered_sorted( sort_key="duration" ) datasets = [train_data, valid_data] + [i for k, i in test_datasets.items()] # 2. Define audio pipeline: @sb.utils.data_pipeline.takes("wav") @sb.utils.data_pipeline.provides("sig") def audio_pipeline(wav): info = torchaudio.info(wav) sig = sb.dataio.dataio.read_audio(wav) resampled = torchaudio.transforms.Resample( info.sample_rate, hparams["sample_rate"], )(sig) return resampled sb.dataio.dataset.add_dynamic_item(datasets, audio_pipeline) label_encoder = sb.dataio.encoder.CTCTextEncoder() # 3. Define text pipeline: @sb.utils.data_pipeline.takes("wrd") @sb.utils.data_pipeline.provides( "wrd", "char_list", "tokens_list", "tokens" ) def text_pipeline(wrd): yield wrd char_list = list(wrd) yield char_list tokens_list = label_encoder.encode_sequence(char_list) yield tokens_list tokens = torch.LongTensor(tokens_list) yield tokens sb.dataio.dataset.add_dynamic_item(datasets, text_pipeline) lab_enc_file = os.path.join(hparams["save_folder"], "label_encoder.txt") special_labels = { "blank_label": hparams["blank_index"], "unk_label": hparams["unk_index"] } label_encoder.load_or_create( path=lab_enc_file, from_didatasets=[train_data], output_key="char_list", special_labels=special_labels, sequence_input=True, ) # 4. Set output: sb.dataio.dataset.set_output_keys( datasets, ["id", "sig", "wrd", "char_list", "tokens"], ) return train_data, valid_data, test_datasets, label_encoder # Load hyperparameters file with command-line overrides hparams_file, run_opts, overrides = sb.parse_arguments(["train_semi.yaml"]) with open(hparams_file) as fin: hparams = load_hyperpyyaml(fin, overrides) # If --distributed_launch then # create ddp_group with the right communication protocol sb.utils.distributed.ddp_init_group(run_opts) # Create experiment directory sb.create_experiment_directory( experiment_directory=hparams["output_folder"], hyperparams_to_save=hparams_file, overrides=overrides, ) # Due to DDP, we do the preparation ONLY on the main python process # Defining tokenizer and loading it # Create the datasets objects as well as tokenization and encoding :-D label_encoder = sb.dataio.encoder.CTCTextEncoder() lab_enc_file = os.path.join(hparams["save_folder"], "label_encoder.txt") special_labels = { "blank_label": hparams["blank_index"], "unk_label": hparams["unk_index"] } label_encoder.load_or_create( path=lab_enc_file, from_didatasets=[[]], output_key="char_list", special_labels=special_labels, sequence_input=True, ) from pyctcdecode import build_ctcdecoder ind2lab = label_encoder.ind2lab print(ind2lab) labels = [ind2lab[x] for x in range(len(ind2lab))] labels = [""] + labels[1:-1] + ["1"] # Replace the token with a blank character, needed for PyCTCdecode print(labels) decoder = build_ctcdecoder( labels, kenlm_model_path=hparams["ngram_lm_path"], # .arpa or .bin alpha=0.5, # Default by KenLM beta=1.0, # Default by KenLM ) # Trainer initialization run_opts["device"] = "cpu" asr_brain = ASR( modules=hparams["modules"], hparams=hparams, run_opts=run_opts, checkpointer=hparams["checkpointer"], ) # Adding objects to trainer. asr_brain.tokenizer = label_encoder asr_brain.checkpointer.recover_if_possible(device="cpu") asr_brain.modules.eval() title = "Tunisian Speech Recognition" def treat_wav_file(file_mic, file_upload, asr=asr_brain, device="cpu"): if (file_mic is not None) and (file_upload is not None): warn_output = "WARNING: You've uploaded an audio file and used the microphone. The recorded file from the microphone will be used and the uploaded audio will be discarded.\n" wav = file_mic elif (file_mic is None) and (file_upload is None): return "ERROR: You have to either use the microphone or upload an audio file" elif file_mic is not None: wav = file_mic else: wav = file_upload info = torchaudio.info(wav) sr = info.sample_rate sig = sb.dataio.dataio.read_audio(wav) if len(sig.shape) > 1: sig = torch.mean(sig, dim=1) sig = torch.unsqueeze(sig, 0) tensor_wav = sig.to(device) resampled = torchaudio.functional.resample(tensor_wav, sr, 16000) sentence = asr.treat_wav(resampled) return sentence gr.Interface( title = title, description = description, fn=treat_wav_file, inputs=[gr.Audio(sources="microphone", type='filepath', label = "record"), gr.Audio(sources="upload", type='filepath', label="filein")] ,outputs="text").launch()