import logging from dataclasses import dataclass from typing import Dict from typing import Iterable, Optional import types import time import numpy as np import torch import torch.nn.functional as F from torch import Tensor from torch import nn from torch.cuda.amp import autocast from funasr.metrics.compute_acc import compute_accuracy from funasr.losses.label_smoothing_loss import LabelSmoothingLoss from funasr.train_utils.device_funcs import force_gatherable from . import whisper_lib as whisper from funasr.utils.load_utils import load_audio_text_image_video, extract_fbank from funasr.register import tables @tables.register("model_classes", "SenseVoice") class SenseVoice(nn.Module): def __init__(self, *args, **kwargs): super().__init__() dims = kwargs.get("dims", {}) dims = whisper.model.ModelDimensions(**dims) model = whisper.model.Whisper(dims=dims) # encoder model.encoder.downsample_rate = kwargs.get("downsample_rate", 4) model.encoder.use_padmask = kwargs.get("use_padmask", True) from .encoder import sense_voice_encode_forward model.encoder.forward = types.MethodType(sense_voice_encode_forward, model.encoder) # decoder model.decoder.use_padmask = kwargs.get("use_padmask", True) from .decoder import sense_voice_decode_forward model.decoder.forward = types.MethodType(sense_voice_decode_forward, model.decoder) self.model = model self.encoder_output_size = self.model.dims.n_audio_state self.activation_checkpoint = kwargs.get("activation_checkpoint", False) self.ignore_id = kwargs.get("ignore_id", -1) self.vocab_size = kwargs.get("vocab_size", -1) self.length_normalized_loss = kwargs.get("length_normalized_loss", True) self.criterion_att = LabelSmoothingLoss( size=self.vocab_size, padding_idx=self.ignore_id, smoothing=kwargs.get("lsm_weight", 0.0), normalize_length=self.length_normalized_loss, ) specaug = kwargs.get("specaug", None) if specaug is not None: specaug_class = tables.specaug_classes.get(specaug) specaug = specaug_class(**kwargs.get("specaug_conf", {})) self.specaug = specaug def forward( self, speech: torch.Tensor, speech_lengths: torch.Tensor, text: torch.Tensor, text_lengths: torch.Tensor, **kwargs, ): target_mask = kwargs.get("target_mask", None) # import pdb; # pdb.set_trace() if len(text_lengths.size()) > 1: text_lengths = text_lengths[:, 0] if len(speech_lengths.size()) > 1: speech_lengths = speech_lengths[:, 0] batch_size = speech.shape[0] if self.activation_checkpoint: from torch.utils.checkpoint import checkpoint encoder_out, encoder_out_lens = checkpoint( self.encode, speech, speech_lengths, use_reentrant=False ) else: encoder_out, encoder_out_lens = self.encode(speech, speech_lengths) loss_att, acc_att, cer_att, wer_att = self._calc_att_loss( encoder_out, encoder_out_lens, text, text_lengths, target_mask=target_mask ) loss = loss_att stats = {} stats["acc"] = acc_att stats["loss"] = torch.clone(loss.detach()) stats["batch_size"] = batch_size # force_gatherable: to-device and to-tensor if scalar for DataParallel if self.length_normalized_loss: batch_size = int((text_lengths + 1).sum()) loss, stats, weight = force_gatherable((loss, stats, batch_size), loss.device) return loss, stats, weight def encode( self, speech: torch.Tensor, speech_lengths: torch.Tensor, **kwargs, ): """Encoder. Note that this method is used by asr_inference.py Args: speech: (Batch, Length, ...) speech_lengths: (Batch, ) ind: int """ with autocast(False): # Data augmentation if self.specaug is not None and self.training: speech, speech_lengths = self.specaug(speech, speech_lengths) # Forward encoder encoder_out, encoder_out_lens = self.model.encoder(speech.permute(0, 2, 1), speech_lengths) return encoder_out, encoder_out_lens def _calc_att_loss( self, encoder_out: torch.Tensor, encoder_out_lens: torch.Tensor, ys_pad: torch.Tensor, ys_pad_lens: torch.Tensor, **kwargs, ): target_mask = kwargs.get("target_mask", None) stats = {} # 1. Forward decoder decoder_out = self.model.decoder( x=ys_pad, xa=encoder_out, hlens=encoder_out_lens, ys_in_lens=ys_pad_lens ) # 2. Compute attention loss mask = torch.ones_like(ys_pad) * (-1) ys_pad_mask = (ys_pad * target_mask + mask * (1 - target_mask)).to(torch.int64) ys_pad_mask[ys_pad_mask == 0] = -1 loss_att = self.criterion_att(decoder_out[:, :-1, :], ys_pad_mask[:, 1:]) with torch.no_grad(): preds = torch.argmax(decoder_out, -1) acc_att = compute_accuracy( preds[:, :-1], ys_pad_mask[:, 1:], ignore_label=self.ignore_id ) return loss_att, acc_att, None, None def inference( self, data_in, data_lengths=None, key: list = None, tokenizer=None, frontend=None, **kwargs, ): if kwargs.get("batch_size", 1) > 1: raise NotImplementedError("batch decoding is not implemented") if frontend is None and not hasattr(self, "frontend"): frontend_class = tables.frontend_classes.get("WhisperFrontend") frontend = frontend_class( n_mels=self.model.dims.n_mels, do_pad_trim=kwargs.get("do_pad_trim", True) ) self.frontend = frontend else: frontend = frontend if frontend is not None else self.frontend meta_data = {} if ( isinstance(data_in, torch.Tensor) and kwargs.get("data_type", "sound") == "fbank" ): # fbank speech, speech_lengths = data_in, data_lengths if len(speech.shape) < 3: speech = speech[None, :, :] if speech_lengths is None: speech_lengths = speech.shape[1] else: # extract fbank feats time1 = time.perf_counter() audio_sample_list = load_audio_text_image_video( data_in, fs=frontend.fs if hasattr(frontend, "fs") else 16000, audio_fs=kwargs.get("fs", 16000), data_type=kwargs.get("data_type", "sound"), tokenizer=tokenizer, ) time2 = time.perf_counter() meta_data["load_data"] = f"{time2 - time1:0.3f}" speech, speech_lengths = extract_fbank( audio_sample_list, data_type=kwargs.get("data_type", "sound"), frontend=frontend ) time3 = time.perf_counter() meta_data["extract_feat"] = f"{time3 - time2:0.3f}" frame_shift = frontend.frame_shift if hasattr(frontend, "frame_shift") else 10 lfr_n = frontend.lfr_n if hasattr(frontend, "lfr_n") else 1 meta_data["batch_data_time"] = speech_lengths.sum().item() * frame_shift * lfr_n / 1000 speech = speech.to(device=kwargs["device"])[0, :, :] speech_lengths = speech_lengths.to(device=kwargs["device"]) DecodingOptions = kwargs.get("DecodingOptions", {}) task = DecodingOptions.get("task", "ASR") if isinstance(task, str): task = [task] task = "".join([f"<|{x}|>" for x in task]) initial_prompt = kwargs.get("initial_prompt", f"<|startoftranscript|>{task}") DecodingOptions["initial_prompt"] = initial_prompt language = DecodingOptions.get("language", None) language = None if language == "auto" else language DecodingOptions["language"] = language DecodingOptions["vocab_path"] = kwargs["tokenizer_conf"].get("vocab_path", None) if "without_timestamps" not in DecodingOptions: DecodingOptions["without_timestamps"] = True options = whisper.DecodingOptions(**DecodingOptions) result = whisper.decode(self.model, speech, options) text = f"{result.text}" results = [] result_i = {"key": key[0], "text": text} results.append(result_i) return results, meta_data @tables.register("model_classes", "SenseVoiceRWKV") class SenseVoiceRWKV(nn.Module): def __init__(self, *args, **kwargs): super().__init__() dims = kwargs.get("dims", {}) dims = whisper.model.ModelDimensions(**dims) model = whisper.model.Whisper(dims=dims) # encoder model.encoder.downsample_rate = kwargs.get("downsample_rate", 4) model.encoder.use_padmask = kwargs.get("use_padmask", True) from .encoder import sense_voice_encode_forward model.encoder.forward = types.MethodType(sense_voice_encode_forward, model.encoder) # decoder del model.decoder decoder = kwargs.get("decoder", "SenseVoiceDecoder") decoder_class = tables.decoder_classes.get(decoder) decoder = decoder_class( n_vocab=dims.n_vocab, n_ctx=dims.n_text_ctx, n_state=dims.n_text_state, n_head=dims.n_text_head, n_layer=dims.n_text_layer, **kwargs.get("decoder_conf"), ) model.decoder = decoder self.model = model self.encoder_output_size = self.model.dims.n_audio_state self.activation_checkpoint = kwargs.get("activation_checkpoint", False) self.ignore_id = kwargs.get("ignore_id", -1) self.vocab_size = kwargs.get("vocab_size", -1) self.length_normalized_loss = kwargs.get("length_normalized_loss", True) self.criterion_att = LabelSmoothingLoss( size=self.vocab_size, padding_idx=self.ignore_id, smoothing=kwargs.get("lsm_weight", 0.0), normalize_length=self.length_normalized_loss, ) specaug = kwargs.get("specaug", None) if specaug is not None: specaug_class = tables.specaug_classes.get(specaug) specaug = specaug_class(**kwargs.get("specaug_conf", {})) self.specaug = specaug def forward( self, speech: torch.Tensor, speech_lengths: torch.Tensor, text: torch.Tensor, text_lengths: torch.Tensor, **kwargs, ): target_mask = kwargs.get("target_mask", None) # import pdb; # pdb.set_trace() if len(text_lengths.size()) > 1: text_lengths = text_lengths[:, 0] if len(speech_lengths.size()) > 1: speech_lengths = speech_lengths[:, 0] batch_size, frames, _ = speech.shape if self.activation_checkpoint: from torch.utils.checkpoint import checkpoint encoder_out, encoder_out_lens = checkpoint( self.encode, speech, speech_lengths, use_reentrant=False ) else: encoder_out, encoder_out_lens = self.encode(speech, speech_lengths) loss_att, acc_att, cer_att, wer_att = self._calc_att_loss( encoder_out, encoder_out_lens, text, text_lengths, target_mask=target_mask ) loss = loss_att stats = {} stats["acc"] = acc_att stats["loss"] = torch.clone(loss.detach()) stats["batch_size"] = batch_size stats["batch_size_x_frames"] = frames * batch_size # force_gatherable: to-device and to-tensor if scalar for DataParallel if self.length_normalized_loss: batch_size = int((text_lengths + 1).sum()) loss, stats, weight = force_gatherable((loss, stats, batch_size), loss.device) return loss, stats, weight def encode( self, speech: torch.Tensor, speech_lengths: torch.Tensor, **kwargs, ): """Encoder. Note that this method is used by asr_inference.py Args: speech: (Batch, Length, ...) speech_lengths: (Batch, ) ind: int """ with autocast(False): # Data augmentation if self.specaug is not None and self.training: speech, speech_lengths = self.specaug(speech, speech_lengths) # Forward encoder encoder_out, encoder_out_lens = self.model.encoder(speech.permute(0, 2, 1), speech_lengths) return encoder_out, encoder_out_lens def _calc_att_loss( self, encoder_out: torch.Tensor, encoder_out_lens: torch.Tensor, ys_pad: torch.Tensor, ys_pad_lens: torch.Tensor, **kwargs, ): target_mask = kwargs.get("target_mask", None) stats = {} # 1. Forward decoder decoder_out = self.model.decoder( x=ys_pad, xa=encoder_out, hlens=encoder_out_lens, ys_in_lens=ys_pad_lens ) # 2. Compute attention loss mask = torch.ones_like(ys_pad) * (-1) ys_pad_mask = (ys_pad * target_mask + mask * (1 - target_mask)).to(torch.int64) ys_pad_mask[ys_pad_mask == 0] = -1 loss_att = self.criterion_att(decoder_out[:, :-1, :], ys_pad_mask[:, 1:]) with torch.no_grad(): preds = torch.argmax(decoder_out, -1) acc_att = compute_accuracy( preds[:, :-1], ys_pad_mask[:, 1:], ignore_label=self.ignore_id ) return loss_att, acc_att, None, None def inference( self, data_in, data_lengths=None, key: list = None, tokenizer=None, frontend=None, **kwargs, ): if kwargs.get("batch_size", 1) > 1: raise NotImplementedError("batch decoding is not implemented") if frontend is None and not hasattr(self, "frontend"): frontend_class = tables.frontend_classes.get("WhisperFrontend") frontend = frontend_class( n_mels=self.model.dims.n_mels, do_pad_trim=kwargs.get("do_pad_trim", True) ) self.frontend = frontend else: frontend = frontend if frontend is not None else self.frontend meta_data = {} if ( isinstance(data_in, torch.Tensor) and kwargs.get("data_type", "sound") == "fbank" ): # fbank speech, speech_lengths = data_in, data_lengths if len(speech.shape) < 3: speech = speech[None, :, :] if speech_lengths is None: speech_lengths = speech.shape[1] else: # extract fbank feats time1 = time.perf_counter() audio_sample_list = load_audio_text_image_video( data_in, fs=frontend.fs if hasattr(frontend, "fs") else 16000, audio_fs=kwargs.get("fs", 16000), data_type=kwargs.get("data_type", "sound"), tokenizer=tokenizer, ) time2 = time.perf_counter() meta_data["load_data"] = f"{time2 - time1:0.3f}" speech, speech_lengths = extract_fbank( audio_sample_list, data_type=kwargs.get("data_type", "sound"), frontend=frontend ) time3 = time.perf_counter() meta_data["extract_feat"] = f"{time3 - time2:0.3f}" frame_shift = frontend.frame_shift if hasattr(frontend, "frame_shift") else 10 lfr_n = frontend.lfr_n if hasattr(frontend, "lfr_n") else 1 meta_data["batch_data_time"] = speech_lengths.sum().item() * frame_shift * lfr_n / 1000 speech = speech.to(device=kwargs["device"])[0, :, :] speech_lengths = speech_lengths.to(device=kwargs["device"]) DecodingOptions = kwargs.get("DecodingOptions", {}) task = DecodingOptions.get("task", "ASR") if isinstance(task, str): task = [task] task = "".join([f"<|{x}|>" for x in task]) initial_prompt = kwargs.get("initial_prompt", f"<|startoftranscript|>{task}") DecodingOptions["initial_prompt"] = initial_prompt language = DecodingOptions.get("language", None) language = None if language == "auto" else language DecodingOptions["language"] = language DecodingOptions["vocab_path"] = kwargs["tokenizer_conf"].get("vocab_path", None) if "without_timestamps" not in DecodingOptions: DecodingOptions["without_timestamps"] = True options = whisper.DecodingOptions(**DecodingOptions) result = whisper.decode(self.model, speech, options) text = f"{result.text}" results = [] result_i = {"key": key[0], "text": text} results.append(result_i) return results, meta_data