From f4f545b7243435116f3cedc4f42cb39bfed3331e Mon Sep 17 00:00:00 2001
From: 游雁 <zhifu.gzf@alibaba-inc.com>
Date: 星期二, 30 四月 2024 00:06:43 +0800
Subject: [PATCH] batch

---
 funasr/models/sense_voice/model.py |  235 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
 1 files changed, 235 insertions(+), 0 deletions(-)

diff --git a/funasr/models/sense_voice/model.py b/funasr/models/sense_voice/model.py
index d06776b..c12107e 100644
--- a/funasr/models/sense_voice/model.py
+++ b/funasr/models/sense_voice/model.py
@@ -476,3 +476,238 @@
         results.append(result_i)
 
         return results, meta_data
+
+
+@tables.register("model_classes", "SenseVoiceFSMN")
+class SenseVoiceFSMN(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_conf = kwargs.get("decoder_conf", {})
+        decoder_class = tables.decoder_classes.get(decoder)
+        decoder = decoder_class(
+            vocab_size=dims.n_vocab,
+            encoder_output_size=dims.n_audio_state,
+            **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
+        _, text_tokens = text.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
+        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"] = text_tokens * batch_size
+        stats["batch_size_real_tokens"] = text_lengths.sum().item()
+        stats["padding_tokens"] = stats["batch_size_x_tokens"] - stats["batch_size_real_tokens"]
+        stats["batch_size_x_frames_plus_tokens"] = (text_tokens + 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
+        )
+        # decoder_out, _ = self.model.decoder(encoder_out, encoder_out_lens, ys_pad, 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

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