import json import logging import os import random import re import string import time import traceback import torch import torch.nn as nn from funasr import AutoModel from funasr.metrics.compute_acc import compute_accuracy from funasr.register import tables from funasr.train_utils.device_funcs import force_gatherable, to_device from funasr.utils.datadir_writer import DatadirWriter from funasr.utils.load_utils import extract_fbank, load_audio_text_image_video from transformers import AutoConfig, AutoModelForCausalLM dtype_map = {"bf16": torch.bfloat16, "fp16": torch.float16, "fp32": torch.float32} @tables.register("model_classes", "FunASRNano") class FunASRNano(nn.Module): def __init__( self, audio_encoder: str = None, audio_encoder_conf: dict = None, audio_adaptor: str = None, audio_adaptor_conf: dict = None, llm: str = None, llm_conf: dict = None, input_size: int = 80, length_normalized_loss: bool = False, **kwargs, ): super().__init__() # audio encoder hub = audio_encoder_conf.get("hub", None) self.audio_encoder_activation_checkpoint = audio_encoder_conf.get( "activation_checkpoint", False ) if hub == "ms": model = AutoModel(model=audio_encoder, model_revision="master") audio_encoder_output_size = ( model.model.encoder_output_size if hasattr(model.model, "encoder_output_size") else -1 ) audio_encoder = ( model.model.model.encoder if hasattr(model.model, "model") else model.model.encoder ) else: encoder_class = tables.encoder_classes.get(audio_encoder) audio_encoder = encoder_class(input_size=input_size, **audio_encoder_conf) audio_encoder_output_size = audio_encoder.output_size() freeze = audio_encoder_conf.get("freeze", True) freeze_layer_num = int(audio_encoder_conf.get("freeze_layer_num", -1)) if freeze: for name, param in audio_encoder.named_parameters(): param.requires_grad = False audio_encoder.eval() self.audio_encoder = audio_encoder # llm self.llm = None init_param_path = llm_conf.get("init_param_path", None) llm_dim = None llm_load_kwargs = llm_conf.get("load_kwargs", {}) config = AutoConfig.from_pretrained(init_param_path) model = AutoModelForCausalLM.from_config(config, **llm_load_kwargs) freeze = llm_conf.get("freeze", True) if freeze: for name, param in model.named_parameters(): param.requires_grad = False model.eval() logging.info(f"use_lora: {llm_conf.get('use_lora', False)}") if llm_conf.get("use_lora", False): from omegaconf import DictConfig, OmegaConf lora_conf = llm_conf.get("lora_conf", {}) if isinstance(lora_conf, (OmegaConf, DictConfig)): lora_conf = OmegaConf.to_container(lora_conf, resolve=True) from peft import LoraConfig, PeftModel, get_peft_model lora_init_param_path = lora_conf.get("init_param_path", None) if lora_init_param_path is not None: logging.info(f"lora_init_param_path: {lora_init_param_path}") model = PeftModel.from_pretrained(model, lora_init_param_path) for name, param in model.named_parameters(): if not lora_conf.get("freeze_lora", False): if "lora_" in name: param.requires_grad = True else: peft_config = LoraConfig(**lora_conf) model = get_peft_model(model, peft_config) model.print_trainable_parameters() if llm_conf.get("activation_checkpoint", False): model.gradient_checkpointing_enable() self.llm_dtype = llm_conf.get("llm_dtype", "fp32") self.llm = model.to(dtype_map[self.llm_dtype]) llm_dim = model.get_input_embeddings().weight.shape[-1] # adaptor adaptor_class = tables.adaptor_classes.get(audio_adaptor) if audio_encoder_output_size > 0: audio_adaptor_conf["encoder_dim"] = audio_encoder_output_size audio_adaptor_conf["llm_dim"] = ( llm_dim if llm_dim is not None else audio_adaptor_conf["llm_dim"] ) audio_adaptor = adaptor_class(**audio_adaptor_conf) freeze = audio_adaptor_conf.get("freeze", False) if freeze: for name, param in audio_adaptor.named_parameters(): param.requires_grad = False audio_adaptor.eval() self.audio_adaptor = audio_adaptor self.length_normalized_loss = length_normalized_loss self.feat_permute = audio_encoder_conf.get("feat_permute", True) rank = int(os.environ.get("RANK", 0)) logging.info(f"rank: {rank}, model is builded.") def forward( self, speech: torch.Tensor = None, speech_lengths: torch.Tensor = None, input_ids: torch.Tensor = None, attention_mask: torch.Tensor = None, labels_ids: torch.Tensor = None, fbank_beg: torch.Tensor = None, fbank_mask: torch.Tensor = None, **kwargs, ): batch_size, token_num = input_ids.shape stats = {} input_ids[input_ids < 0] = 0 inputs_embeds = self.llm.model.get_input_embeddings()(input_ids) if speech is not None: if len(speech_lengths.size()) > 1: speech_lengths = speech_lengths[:, 0] batch_size_speech, frames, _ = speech.shape # audio encoder if self.audio_encoder_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) # audio_adaptor encoder_out, encoder_out_lens = self.audio_adaptor( encoder_out, encoder_out_lens ) batch_size, token_num, dims = inputs_embeds.shape fake_token_len = kwargs.get("fake_token_len") fake_token_len[fake_token_len < 0] = 0 fbank_beg[fbank_beg < 0] = 0 speech_idx = 0 for batch_idx in range(batch_size): for turn_id in range(fbank_beg.shape[1]): fbank_beg_idx = fbank_beg[batch_idx, turn_id].item() if fbank_beg_idx > 0: speech_token_len = fake_token_len[batch_idx, turn_id] speech_token = encoder_out[speech_idx, :speech_token_len, :] try: inputs_embeds[ batch_idx, fbank_beg_idx : fbank_beg_idx + speech_token_len, :, ] = speech_token except Exception as e: logging.error(f"{str(e)}, {traceback.format_exc()}") logging.info( f"batch_idx: {batch_idx}, inputs_embeds: {inputs_embeds.shape}, fbank_beg_idx: {fbank_beg_idx}, speech_token_len: {speech_token_len}, encoder_out: {encoder_out.shape}, encoder_out_lens: {encoder_out_lens}, fake_token_len: {fake_token_len}, speech_lengths: {speech_lengths}" ) speech_token_len = encoder_out_lens[speech_idx].item() speech_token = encoder_out[speech_idx, :speech_token_len, :] inputs_embeds[ batch_idx, fbank_beg_idx : fbank_beg_idx + speech_token_len, :, ] = speech_token speech_idx += 1 stats["batch_size_speech"] = batch_size_speech stats["batch_size_x_frames"] = frames * batch_size_speech stats["batch_size_real_frames"] = speech_lengths.sum().item() stats["padding_frames"] = ( stats["batch_size_x_frames"] - stats["batch_size_real_frames"] ) with torch.cuda.amp.autocast( enabled=True if self.llm_dtype != "fp32" else False, dtype=dtype_map[self.llm_dtype], ): labels_ids[labels_ids == -1] = -100 attention_mask[attention_mask < 0] = 0 model_outputs = self.llm( inputs_embeds=inputs_embeds.to(dtype_map[self.llm_dtype]), attention_mask=attention_mask, labels=labels_ids, ) loss = model_outputs.loss with torch.no_grad(): preds = torch.argmax(model_outputs.logits, -1) acc_att = compute_accuracy( preds[:, :-1], labels_ids[:, 1:], ignore_label=-100 ) stats["acc"] = acc_att stats["loss"] = torch.clone(loss.detach()) stats["batch_size"] = batch_size stats["batch_size_x_tokens"] = token_num * batch_size stats["batch_size_real_tokens"] = attention_mask.sum().item() stats["padding_tokens"] = ( stats["batch_size_x_tokens"] - stats["batch_size_real_tokens"] ) dialog_turns = (fbank_beg > 0).sum(-1) dialog_turns_max = torch.max(dialog_turns).int().item() dialog_turns_avg = dialog_turns.sum().item() / batch_size stats["dialog_turns_max"] = dialog_turns_max stats["dialog_turns_avg"] = dialog_turns_avg # force_gatherable: to-device and to-tensor if scalar for DataParallel if self.length_normalized_loss: batch_size = int((labels_ids > 0 + 1).sum()) loss, stats, weight = force_gatherable((loss, stats, batch_size), loss.device) return loss, stats, weight def forward_export(self, speech, speech_lengths, **kwargs): x, olens = self.audio_encoder(speech, speech_lengths) encoder_out, encoder_out_lens = self.audio_adaptor(x, olens) return encoder_out, encoder_out_lens def encode(self, speech, speech_lengths): # audio encoder if self.feat_permute: encoder_out, encoder_out_lens = self.audio_encoder( speech.permute(0, 2, 1), speech_lengths ) else: encoder_out, encoder_out_lens = self.audio_encoder(speech, speech_lengths) return encoder_out, encoder_out_lens def data_template(self, data): system, user, assistant = [], [], [] for i, item in enumerate(data): role = item["role"] content = item["content"] if role == "system": system.append(content) elif role == "user": if "audio" in item: audio = item["audio"] content = [content, audio] user.append(content) elif role == "assistant": assistant.append(content) system = system * len(user) contents = { "system": system, "user": user, "assistant": assistant, } return contents def data_load_speech( self, contents: dict, tokenizer, frontend, meta_data={}, **kwargs ): system = contents["system"] user = contents["user"] assistant = contents["assistant"] pattern = re.compile(r"(<\|startofspeech\|>.*?<\|endofspeech\|>)") do_think = True sys_prompt = True if "dataset_conf" in kwargs: do_think = kwargs["dataset_conf"].get("do_think", True) sys_prompt = kwargs["dataset_conf"].get("sys_prompt", True) input_ids, labels, fbank, fbank_lens, fbank_mask, fbank_beg, fake_token_len = ( [], [], [], [], [], [], [], ) input_source_ids = [] for i, (system_prompt, user_prompt, target_out) in enumerate( zip(system, user, assistant) ): if i >= kwargs.get("multiturn_num_max", 5): break if len(input_ids) > kwargs.get("max_token_length", 1500): break if isinstance(user_prompt, (list, tuple)): user_prompt, audio = user_prompt if i == 0: if kwargs.get("infer_with_assistant_input", False): source_input = f"<|im_start|>system\n{system_prompt}<|im_end|>\n<|im_start|>user\n{user_prompt}" if not sys_prompt: source_input = f"<|im_start|>user\n{user_prompt}" else: source_input = f"<|im_start|>system\n{system_prompt}<|im_end|>\n<|im_start|>user\n{user_prompt}<|im_end|>\n<|im_start|>assistant\n" if not sys_prompt: source_input = f"<|im_start|>user\n{user_prompt}<|im_end|>\n<|im_start|>assistant\n" else: if kwargs.get("infer_with_assistant_input", False): source_input = f"<|im_start|>user\n{user_prompt}" else: source_input = f"<|im_start|>user\n{user_prompt}<|im_end|>\n<|im_start|>assistant\n" if not do_think: source_input += "\n\n\n\n" splits = pattern.split(source_input) source_ids = [] fbank_mask_i = [] fake_token_len_i = 0 fbank_beg_i = -1 speech, speech_lengths = [], [] for k, sub_str in enumerate(splits): if not sub_str.startswith("<|startofspeech|>"): sub_token = tokenizer.encode(sub_str) source_ids += sub_token fbank_mask_i += [0] * len(sub_token) else: sub_str = sub_str.replace("<|startofspeech|>", "").replace( "<|endofspeech|>", "" ) if sub_str.startswith("!"): sub_str = sub_str[1:] if sub_str.startswith("!"): # !!: audio sample point sub_str = audio try: time1 = time.perf_counter() data_src = load_audio_text_image_video( sub_str, fs=frontend.fs, **kwargs ) time2 = time.perf_counter() meta_data["load_data"] = f"{time2 - time1:0.3f}" except Exception as e: logging.error( f"Loading wav failed! {str(e)}, {traceback.format_exc()}" ) speech, speech_lengths = extract_fbank( data_src, data_type=kwargs.get("data_type", "sound"), frontend=frontend, is_final=True, ) # speech: [b, T, d] time3 = time.perf_counter() meta_data["extract_feat"] = f"{time3 - time2:0.3f}" meta_data["batch_data_time"] = ( speech_lengths.sum().item() * frontend.frame_shift * frontend.lfr_n / 1000 ) if self.feat_permute: speech = speech.permute(0, 2, 1) olens = 1 + (speech_lengths[0].item() - 3 + 2 * 1) // 2 olens = 1 + (olens - 3 + 2 * 1) // 2 fake_token_len_i = (olens - 1) // 2 + 1 fake_token = [0] * fake_token_len_i fbank_beg_i = len(source_ids) source_ids += fake_token fbank_mask_i += [1] * len(fake_token) fbank_beg += [fbank_beg_i + len(input_ids)] fake_token_len += [fake_token_len_i] source_mask = [-100] * len(source_ids) target_out = f"{target_out}<|im_end|>" target_ids = tokenizer.encode(target_out) input_source_ids = input_ids + source_ids input_ids += source_ids + target_ids labels += source_mask + target_ids fbank_mask += fbank_mask_i if len(speech) > 0: fbank.append(speech[0, :, :]) fbank_lens.append(speech_lengths) input_ids = torch.tensor( input_ids, dtype=torch.int64 ) # [: self.max_token_length] attention_mask = torch.tensor([1] * len(input_ids), dtype=torch.int32) labels = torch.tensor(labels, dtype=torch.int64) # [: self.max_token_length] fbank_mask = torch.tensor(fbank_mask, dtype=torch.float32) fbank_beg = torch.tensor(fbank_beg, dtype=torch.int32) fake_token_len = torch.tensor(fake_token_len, dtype=torch.int32) source_ids = torch.tensor(input_source_ids, dtype=torch.int64) target_ids = torch.tensor(target_ids, dtype=torch.int64) if len(fbank) > 0: speech = torch.nn.utils.rnn.pad_sequence( fbank, batch_first=True, padding_value=0.0 ) speech_lengths = torch.nn.utils.rnn.pad_sequence( fbank_lens, batch_first=True, padding_value=-1 ) else: speech = [] speech_lengths = [] output = { "speech": speech, "speech_lengths": speech_lengths, "fbank_mask": fbank_mask[None, :], "fbank_beg": fbank_beg[None,], "fake_token_len": fake_token_len[None, :], "input_ids": input_ids[None,], "attention_mask": attention_mask[None,], "labels_ids": labels, "source_ids": source_ids[None, :], "target_ids": target_ids[None, :], } return output def inference_prepare( self, data_in, data_lengths=None, key: list = None, tokenizer=None, frontend=None, **kwargs, ): meta_data = {} if kwargs.get("batch_size", 1) > 1: raise NotImplementedError("batch decoding is not implemented") contents = self.data_template(data_in[0]) output = self.data_load_speech( contents, tokenizer, frontend, meta_data=meta_data, **kwargs ) batch = to_device(output, kwargs["device"]) # audio encoder speech = batch["speech"] if len(speech) > 0: if "audio_embedding" in kwargs and "audio_embedding_lens" in kwargs: encoder_out = kwargs["audio_embedding"] encoder_out_lens = kwargs["audio_embedding_lens"] else: speech_lengths = batch["speech_lengths"][:, 0] # fp16 if kwargs.get("fp16", False): speech = speech.to(torch.float16) elif kwargs.get("bf16", False): speech = speech.to(torch.bfloat16) # audio encoder encoder_out, encoder_out_lens = self.encode(speech, speech_lengths) # audio_adaptor encoder_out, encoder_out_lens = self.audio_adaptor( encoder_out, encoder_out_lens ) meta_data["audio_adaptor_out"] = encoder_out meta_data["audio_adaptor_out_lens"] = encoder_out_lens input_ids = batch["input_ids"] source_ids = batch["source_ids"] fbank_beg = batch["fbank_beg"] fake_token_len = batch["fake_token_len"] if not kwargs.get("tearchforing", False): input_ids = source_ids input_ids[input_ids < 0] = 0 inputs_embeds = self.llm.model.get_input_embeddings()(input_ids) batch_size, token_num, dims = inputs_embeds.shape fake_token_len[fake_token_len < 0] = 0 fbank_beg[fbank_beg < 0] = 0 speech_idx = 0 for batch_idx in range(batch_size): for turn_id in range(fbank_beg.shape[1]): fbank_beg_idx = fbank_beg[batch_idx, turn_id].item() if fbank_beg_idx > 0: speech_token_len = fake_token_len[batch_idx, turn_id] speech_token = encoder_out[speech_idx, :speech_token_len, :] try: inputs_embeds[ batch_idx, fbank_beg_idx : fbank_beg_idx + speech_token_len, :, ] = speech_token except Exception as e: # logging.error(f"{str(e)}, {traceback.format_exc()}") logging.info( f"batch_idx: {batch_idx}, inputs_embeds: {inputs_embeds.shape}, fbank_beg_idx: {fbank_beg_idx}, speech_token_len: {speech_token_len}, encoder_out: {encoder_out.shape}, encoder_out_lens: {encoder_out_lens}, fake_token_len: {fake_token_len}, speech_lengths: {speech_lengths}" ) speech_token_len = encoder_out_lens[speech_idx].item() speech_token = encoder_out[speech_idx, :speech_token_len, :] inputs_embeds[ batch_idx, fbank_beg_idx : fbank_beg_idx + speech_token_len, :, ] = speech_token speech_idx += 1 return inputs_embeds, contents, batch, source_ids, meta_data def inference( self, data_in, data_lengths=None, key: list = None, tokenizer=None, frontend=None, **kwargs, ): hotwords = kwargs.get("hotwords", []) if len(hotwords) > 0: hotwords = ", ".join(hotwords) prompt = f"请结合上下文信息,更加准确地完成语音转写任务。如果没有相关信息,我们会留空。\n\n\n**上下文信息:**\n\n\n" prompt += f"热词列表:[{hotwords}]\n" else: prompt = "" language = kwargs.get("language", "auto") if language not in ("auto", "zh", "en", "ja"): language = "auto" if language == "auto": prompt += "语音转写" else: LANGUAGE_MAP = {"zh": "中文", "en": "英文", "ja": "日文"} prompt += f"语音转写成{LANGUAGE_MAP[language]}" itn = kwargs.get("itn", True) if not itn: prompt += ",不进行文本规整" prompt += ":" new_data_in = [] for data in data_in: if isinstance(data, str): new_data_in.append( [ {"role": "system", "content": "You are a helpful assistant."}, { "role": "user", "content": f"{prompt}<|startofspeech|>!{data}<|endofspeech|>", }, {"role": "assistant", "content": "null"}, ] ) elif isinstance(data, torch.Tensor): new_data_in.append( [ {"role": "system", "content": "You are a helpful assistant."}, { "role": "user", "content": f"{prompt}<|startofspeech|>!!<|endofspeech|>", "audio": data, }, {"role": "assistant", "content": "null"}, ] ) data_in = new_data_in if key is None: key = [] for _ in data_in: chars = string.ascii_letters + string.digits key.append( "rand_key_" + "".join(random.choice(chars) for _ in range(13)) ) return self.inference_llm( data_in, data_lengths=data_lengths, key=key, tokenizer=tokenizer, frontend=frontend, **kwargs, ) def inference_llm( self, data_in, data_lengths=None, key: list = None, tokenizer=None, frontend=None, **kwargs, ): inputs_embeds, contents, batch, source_ids, meta_data = self.inference_prepare( data_in, data_lengths, key, tokenizer, frontend, **kwargs ) llm_dtype = kwargs.get("llm_dtype", "fp32") if llm_dtype == "fp32": llm_dtype = "fp16" if kwargs.get("fp16", False) else llm_dtype llm_dtype = "bf16" if kwargs.get("bf16", False) else llm_dtype with torch.cuda.amp.autocast( enabled=True if llm_dtype != "fp32" else False, dtype=dtype_map[llm_dtype] ): label = contents["assistant"][-1] self.llm = self.llm.to(dtype_map[llm_dtype]) inputs_embeds = inputs_embeds.to(dtype_map[llm_dtype]) llm_kwargs = kwargs.get("llm_kwargs", {}) if not kwargs.get("teachforing", False): generated_ids = self.llm.generate( inputs_embeds=inputs_embeds, max_new_tokens=kwargs.get("max_length", 512), **llm_kwargs, ) response = tokenizer.batch_decode( generated_ids, skip_special_tokens=kwargs.get("skip_special_tokens", True), )[0] loss = None else: labels_ids = batch["labels_ids"] labels_ids[labels_ids == -1] = -100 attention_mask = batch.get("attention_mask", None) model_outputs = self.llm( inputs_embeds=inputs_embeds, attention_mask=attention_mask, labels=labels_ids, **llm_kwargs, ) preds = torch.argmax(model_outputs.logits, -1)[:, source_ids.shape[1] :] response = tokenizer.batch_decode( preds, add_special_tokens=False, skip_special_tokens=kwargs.get("skip_special_tokens", True), )[0] loss = model_outputs.loss.item() ibest_writer = None if kwargs.get("output_dir") is not None: if not hasattr(self, "writer"): self.writer = DatadirWriter(kwargs.get("output_dir")) ibest_writer = self.writer[f"{0 + 1}best_recog"] results = [] response_clean = re.sub(r"[^\w\s\u3000\u4e00-\u9fff]+", "", response) result_i = { "key": key[0], "text": re.sub(r'\s+', ' ', response.replace("/sil", " ")), "text_tn": response_clean, "label": label, } if loss is not None: result_i["loss"] = loss results.append(result_i) if ibest_writer is not None: ibest_writer["text"][key[0]] = response.replace("\n", " ") ibest_writer["label"][key[0]] = label.replace("\n", " ") ibest_writer["text_tn"][key[0]] = response_clean return results, meta_data @staticmethod def from_pretrained(model: str = None, **kwargs): from funasr import AutoModel model, kwargs = AutoModel.build_model( model=model, trust_remote_code=True, **kwargs ) return model, kwargs