update

funasr/models/llm_asr/model.py

@@ -1247,6 +1247,567 @@ class LLMASR4(nn.Module):
 
         return output
 
+    def inference_prepare(
+        self,
+        data_in,
+        data_lengths=None,
+        key: list = None,
+        tokenizer=None,
+        frontend=None,
+        **kwargs,
+    ):
+
+        meta_data = {}
+        prompt = kwargs.get("prompt", None)
+
+        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:
+            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)
+
+        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}"
+                        )
+                        # import pdb;
+                        # pdb.set_trace()
+                        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,
+    ):
+
+        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])
+
+            if not kwargs.get("tearchforing", False):
+
+                generated_ids = self.llm.generate(
+                    inputs_embeds=inputs_embeds, max_new_tokens=kwargs.get("max_length", 512)
+                )
+                # generated_ids = [
+                #     output_ids[len(input_id) :]
+                #     for input_id, output_ids in zip(input_ids, generated_ids)
+                # ]
+                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)
+                # attention_mask = attention_mask.to(dtype_map[llm_dtype])
+                model_outputs = self.llm(
+                    inputs_embeds=inputs_embeds, attention_mask=attention_mask, labels=labels_ids
+                )
+
+                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("[^\w\s\u3000\u4e00-\u9fff]+", "", response)
+        result_i = {"key": key[0], "text": response, "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
+
+
+@tables.register("model_classes", "LLMASR5")
+class LLMASR5(nn.Module):
+    """ """
+
+    def __init__(
+        self,
+        specaug: str = None,
+        specaug_conf: dict = None,
+        normalize: str = None,
+        normalize_conf: dict = None,
+        audio_encoder: str = None,
+        audio_encoder_conf: dict = None,
+        audio_adaptor: str = None,
+        audio_adaptor_conf: dict = None,
+        decoder: str = None,
+        decoder_conf: dict = None,
+        ctc: str = None,
+        ctc_conf: dict = None,
+        ctc_weight: float = 0.5,
+        llm: str = None,
+        llm_conf: dict = None,
+        input_size: int = 80,
+        vocab_size: int = -1,
+        ignore_id: int = -1,
+        blank_id: int = 0,
+        sos: int = 1,
+        eos: int = 2,
+        lsm_weight: float = 0.0,
+        length_normalized_loss: bool = False,
+        report_cer: bool = True,
+        report_wer: bool = True,
+        sym_space: str = "<space>",
+        sym_blank: str = "<blank>",
+        # extract_feats_in_collect_stats: bool = True,
+        share_embedding: bool = False,
+        # preencoder: Optional[AbsPreEncoder] = None,
+        # postencoder: Optional[AbsPostEncoder] = None,
+        **kwargs,
+    ):
+
+        super().__init__()
+
+        # audio encoder
+        hub = audio_encoder_conf.get("hub", None)
+        if hub == "ms":
+            from funasr import AutoModel
+
+            model = AutoModel(model=audio_encoder, model_revision="master")
+            # frontend = model.kwargs.get("frontend")
+            audio_encoder_output_size = model.model.encoder_output_size
+
+            audio_encoder = (
+                model.model.model.encoder if hasattr(model.model, "model") else model.model.encoder
+            )
+
+            # self.frontend = frontend
+
+        elif hub == "hf":
+            pass
+        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_layer_num > 0:
+        #     freeze_layer_num = range(freeze_layer_num)
+
+        if freeze:
+            for name, param in audio_encoder.named_parameters():
+                if freeze_layer_num > 0:
+                    idx = re.search(r"\.\d+\.", name)
+                    if idx is not None:
+                        beg, end = idx.regs[0]
+                        layer_id = int(name[beg + 1 : end - 1])
+                        if layer_id < freeze_layer_num:
+                            param.requires_grad = False
+                    elif "ln_post." not in name:
+                        param.requires_grad = False
+                else:
+                    param.requires_grad = False
+
+            audio_encoder.eval()
+
+        self.audio_encoder = audio_encoder
+
+        # llm
+        self.llm = None
+
+        from transformers import AutoModelForCausalLM, AutoTokenizer, AutoConfig
+
+        init_param_path = llm_conf.get("init_param_path", "vicuna-7b-v1.5")
+
+        model = AutoModelForCausalLM.from_pretrained(
+            init_param_path,
+            load_in_8bit=None,
+            device_map=None,
+            use_cache=None,
+        )
+        freeze = llm_conf.get("freeze", True)
+        if freeze:
+            for name, param in model.named_parameters():
+                param.requires_grad = False
+            model.eval()
+        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)
+        audio_adaptor_conf["encoder_dim"] = audio_encoder_output_size
+        audio_adaptor_conf["llm_dim"] = llm_dim
+        audio_adaptor = adaptor_class(**audio_adaptor_conf)
+        init_param_path = audio_adaptor_conf.get("init_param_path", None)
+        if init_param_path is not None:
+            src_state = torch.load(init_param_path, map_location="cpu")
+            flag = audio_adaptor.load_state_dict(src_state, strict=False)
+            logging.info(f"Loading audio_adaptor ckpt: {init_param_path}, status: {flag}")
+
+        self.audio_adaptor = audio_adaptor
+
+        self.error_calculator = None
+
+        self.length_normalized_loss = length_normalized_loss
+        self.beam_search = None
+
+    def forward(
+        self,
+        speech: torch.Tensor,
+        speech_lengths: torch.Tensor,
+        input_ids: torch.Tensor,
+        attention_mask: torch.Tensor,
+        labels_ids: torch.Tensor,
+        fbank_beg: torch.Tensor,
+        fbank_mask: torch.Tensor,
+        **kwargs,
+    ) -> Tuple[torch.Tensor, Dict[str, torch.Tensor], torch.Tensor]:
+        """Encoder + Decoder + Calc loss
+        Args:
+                speech: (Batch, Length, ...)
+                speech_lengths: (Batch, )
+                text: (Batch, Length)
+                text_lengths: (Batch,)
+        """
+        import pdb
+
+        pdb.set_trace()
+        if len(speech_lengths.size()) > 1:
+            speech_lengths = speech_lengths[:, 0]
+
+        batch_size_speech, frames, _ = speech.shape
+        batch_size, token_num = input_ids.shape
+
+        with torch.cuda.amp.autocast(enabled=False):
+            # 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)
+
+        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 = 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}"
+                        )
+                        # import pdb;
+                        # pdb.set_trace()
+                        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
+
+        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
+
+        stats = {}
+        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_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"]
+        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 encode(self, speech, speech_lengths):
+        # audio encoder
+        encoder_out, encoder_out_lens = self.audio_encoder(speech.permute(0, 2, 1), 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":
+                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\|>)")
+
+        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 i == 0:
+                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"
+            else:
+                source_input = f"<|im_start|>user\n{user_prompt}<|im_end|>\n<|im_start|>assistant\n"
+
+            splits = pattern.split(source_input)
+            source_ids = []
+            fbank_i = []
+            fbank_mask_i = []
+            fake_token_len_i = 0
+            fbank_beg_i = -1
+            fbank_lens_i = []
+            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("!"):  # !!bytes
+                            sub_str = eval(sub_str[1:])
+                        try:
+                            time1 = time.perf_counter()
+                            data_src = load_audio_text_image_video(sub_str, fs=frontend.fs)
+                            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 kwargs.get("permute", True):
+                            speech = speech.permute(0, 2, 1)
+                        if speech_lengths > kwargs.get("max_source_length", 5500):
+                            # logging.info(
+                            #     f"speech_lengths > max_source_length: {speech_lengths}>{self.max_source_length}, {item}"
+                            # )
+                            badcase_flag = True
+
+                        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 = speech[0, :, :]
+        # fbank_lens = torch.tensor(fbank_lens, dtype=torch.int32)
+        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,
@@ -1328,7 +1889,6 @@ class LLMASR4(nn.Module):
 
                     speech_idx += 1
         return inputs_embeds, contents, batch, source_ids, meta_data
-    
 
     def inference(
         self,