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requirements.txt

@@ -1,10 +1,9 @@
-torchaudio
-transformers
-librosa
-soundfile
-numpy
-phonemizer
-neucodec
-lmdeploy
-pyyaml
-torch==2.8.0
+gradio
+spaces
+torchaudio
+transformers
+librosa
+soundfile
+numpy
+phonemizer
+neucodec

vieneu_tts.py

@@ -0,0 +1,347 @@
+from pathlib import Path
+from typing import Generator
+import librosa
+import numpy as np
+import torch
+from neucodec import NeuCodec, DistillNeuCodec
+from transformers import AutoTokenizer, AutoModelForCausalLM
+from utils.phonemize_text import phonemize_text, phonemize_with_dict
+import re
+
+def _linear_overlap_add(frames: list[np.ndarray], stride: int) -> np.ndarray:
+    # original impl --> https://github.com/facebookresearch/encodec/blob/main/encodec/utils.py
+    assert len(frames)
+    dtype = frames[0].dtype
+    shape = frames[0].shape[:-1]
+
+    total_size = 0
+    for i, frame in enumerate(frames):
+        frame_end = stride * i + frame.shape[-1]
+        total_size = max(total_size, frame_end)
+
+    sum_weight = np.zeros(total_size, dtype=dtype)
+    out = np.zeros(*shape, total_size, dtype=dtype)
+
+    offset: int = 0
+    for frame in frames:
+        frame_length = frame.shape[-1]
+        t = np.linspace(0, 1, frame_length + 2, dtype=dtype)[1:-1]
+        weight = np.abs(0.5 - (t - 0.5))
+
+        out[..., offset : offset + frame_length] += weight * frame
+        sum_weight[offset : offset + frame_length] += weight
+        offset += stride
+    assert sum_weight.min() > 0
+    return out / sum_weight
+
+class VieNeuTTS:
+    def __init__(
+        self,
+        backbone_repo="pnnbao-ump/VieNeu-TTS",
+        backbone_device="cpu",
+        codec_repo="neuphonic/neucodec",
+        codec_device="cpu",
+    ):
+
+        # Constants
+        self.sample_rate = 24_000
+        self.max_context = 2048
+        self.hop_length = 480
+        self.streaming_overlap_frames = 1
+        self.streaming_frames_per_chunk = 25
+        self.streaming_lookforward = 5
+        self.streaming_lookback = 50
+        self.streaming_stride_samples = self.streaming_frames_per_chunk * self.hop_length
+
+        # ggml & onnx flags
+        self._is_quantized_model = False
+        self._is_onnx_codec = False
+
+        # HF tokenizer
+        self.tokenizer = None
+
+        # Load models
+        self._load_backbone(backbone_repo, backbone_device)
+        self._load_codec(codec_repo, codec_device)
+    
+    def _load_backbone(self, backbone_repo, backbone_device):
+        print(f"Loading backbone from: {backbone_repo} on {backbone_device} ...")
+
+        if backbone_repo.lower().endswith("gguf") or "gguf" in backbone_repo.lower():
+            try:
+                from llama_cpp import Llama
+            except ImportError as e:
+                raise ImportError(
+                    "Failed to import `llama_cpp`. "
+                    "Please install it with:\n"
+                    "    pip install llama-cpp-python"
+                ) from e
+            self.backbone = Llama.from_pretrained(
+                repo_id=backbone_repo,
+                filename="*.gguf",
+                verbose=False,
+                n_gpu_layers=-1 if backbone_device == "gpu" else 0,
+                n_ctx=self.max_context,
+                mlock=True,
+                flash_attn=True if backbone_device == "gpu" else False,
+            )
+            self._is_quantized_model = True
+            
+        else:
+            self.tokenizer = AutoTokenizer.from_pretrained(backbone_repo)
+            self.backbone = AutoModelForCausalLM.from_pretrained(backbone_repo).to(
+                torch.device(backbone_device)
+            )
+    
+    def _load_codec(self, codec_repo, codec_device):
+        print(f"Loading codec from: {codec_repo} on {codec_device} ...")
+        match codec_repo:
+            case "neuphonic/neucodec":
+                self.codec = NeuCodec.from_pretrained(codec_repo)
+                self.codec.eval().to(codec_device)
+            case "neuphonic/distill-neucodec":
+                self.codec = DistillNeuCodec.from_pretrained(codec_repo)
+                self.codec.eval().to(codec_device)
+            case "neuphonic/neucodec-onnx-decoder":
+                if codec_device != "cpu":
+                    raise ValueError("Onnx decoder only currently runs on CPU.")
+                try:
+                    from neucodec import NeuCodecOnnxDecoder
+                except ImportError as e:
+                    raise ImportError(
+                        "Failed to import the onnx decoder."
+                        " Ensure you have onnxruntime installed as well as neucodec >= 0.0.4."
+                    ) from e
+                self.codec = NeuCodecOnnxDecoder.from_pretrained(codec_repo)
+                self._is_onnx_codec = True
+            case _:
+                raise ValueError(f"Unsupported codec repository: {codec_repo}")
+
+    def infer(self, text: str, ref_codes: np.ndarray | torch.Tensor, ref_text: str) -> np.ndarray:
+        """
+        Perform inference to generate speech from text using the TTS model and reference audio.
+
+        Args:
+            text (str): Input text to be converted to speech.
+            ref_codes (np.ndarray | torch.tensor): Encoded reference.
+            ref_text (str): Reference text for reference audio. Defaults to None.
+        Returns:
+            np.ndarray: Generated speech waveform.
+        """
+
+        # Generate tokens
+        if self._is_quantized_model:
+            output_str = self._infer_ggml(ref_codes, ref_text, text)
+        else:
+            prompt_ids = self._apply_chat_template(ref_codes, ref_text, text)
+            output_str = self._infer_torch(prompt_ids)
+
+        # Decode
+        wav = self._decode(output_str)
+
+        return wav
+
+    def infer_stream(self, text: str, ref_codes: np.ndarray | torch.Tensor, ref_text: str) -> Generator[np.ndarray, None, None]:
+        """
+        Perform streaming inference to generate speech from text using the TTS model and reference audio.
+
+        Args:
+            text (str): Input text to be converted to speech.
+            ref_codes (np.ndarray | torch.tensor): Encoded reference.
+            ref_text (str): Reference text for reference audio. Defaults to None.
+        Yields:
+            np.ndarray: Generated speech waveform.
+        """
+
+        if self._is_quantized_model:
+            return self._infer_stream_ggml(ref_codes, ref_text, text)
+        else:
+            raise NotImplementedError("Streaming is not implemented for the torch backend!")
+
+    def encode_reference(self, ref_audio_path: str | Path):
+        wav, _ = librosa.load(ref_audio_path, sr=16000, mono=True)
+        wav_tensor = torch.from_numpy(wav).float().unsqueeze(0).unsqueeze(0)  # [1, 1, T]
+        with torch.no_grad():
+            ref_codes = self.codec.encode_code(audio_or_path=wav_tensor).squeeze(0).squeeze(0)
+        return ref_codes
+
+    def _decode(self, codes: str):
+        """Decode speech tokens to audio waveform."""
+        # Extract speech token IDs using regex
+        speech_ids = [int(num) for num in re.findall(r"<\|speech_(\d+)\|>", codes)]
+        
+        if len(speech_ids) == 0:
+            raise ValueError(
+                "No valid speech tokens found in the output. "
+                "The model may not have generated proper speech tokens."
+            )
+        
+        # Onnx decode
+        if self._is_onnx_codec:
+            codes = np.array(speech_ids, dtype=np.int32)[np.newaxis, np.newaxis, :]
+            recon = self.codec.decode_code(codes)
+        # Torch decode
+        else:
+            with torch.no_grad():
+                codes = torch.tensor(speech_ids, dtype=torch.long)[None, None, :].to(
+                    self.codec.device
+                )
+                recon = self.codec.decode_code(codes).cpu().numpy()
+        
+        return recon[0, 0, :]
+    
+    def _apply_chat_template(self, ref_codes: list[int], ref_text: str, input_text: str) -> list[int]:
+        input_text = phonemize_with_dict(ref_text) + " " + phonemize_with_dict(input_text)
+
+        speech_replace = self.tokenizer.convert_tokens_to_ids("<|SPEECH_REPLACE|>")
+        speech_gen_start = self.tokenizer.convert_tokens_to_ids("<|SPEECH_GENERATION_START|>")
+        text_replace = self.tokenizer.convert_tokens_to_ids("<|TEXT_REPLACE|>")
+        text_prompt_start = self.tokenizer.convert_tokens_to_ids("<|TEXT_PROMPT_START|>")
+        text_prompt_end = self.tokenizer.convert_tokens_to_ids("<|TEXT_PROMPT_END|>")
+
+        input_ids = self.tokenizer.encode(input_text, add_special_tokens=False)
+        chat = """user: Convert the text to speech:<|TEXT_REPLACE|>\nassistant:<|SPEECH_REPLACE|>"""
+        ids = self.tokenizer.encode(chat)
+
+        text_replace_idx = ids.index(text_replace)
+        ids = (
+            ids[:text_replace_idx]
+            + [text_prompt_start]
+            + input_ids
+            + [text_prompt_end]
+            + ids[text_replace_idx + 1 :]  # noqa
+        )
+
+        speech_replace_idx = ids.index(speech_replace)
+        codes_str = "".join([f"<|speech_{i}|>" for i in ref_codes])
+        codes = self.tokenizer.encode(codes_str, add_special_tokens=False)
+        ids = ids[:speech_replace_idx] + [speech_gen_start] + list(codes)
+
+        return ids
+
+    def _infer_torch(self, prompt_ids: list[int]) -> str:
+        prompt_tensor = torch.tensor(prompt_ids).unsqueeze(0).to(self.backbone.device)
+        speech_end_id = self.tokenizer.convert_tokens_to_ids("<|SPEECH_GENERATION_END|>")
+        with torch.no_grad():
+            output_tokens = self.backbone.generate(
+                prompt_tensor,
+                max_length=self.max_context,
+                eos_token_id=speech_end_id,
+                do_sample=True,
+                temperature=1,
+                top_k=50,
+                use_cache=True,
+                min_new_tokens=50,
+            )
+        input_length = prompt_tensor.shape[-1]
+        output_str = self.tokenizer.decode(
+            output_tokens[0, input_length:].cpu().numpy().tolist(), add_special_tokens=False
+        )
+        return output_str
+
+    def _infer_ggml(self, ref_codes: list[int], ref_text: str, input_text: str) -> str:
+        ref_text = phonemize_with_dict(ref_text)
+        input_text = phonemize_with_dict(input_text)
+
+        codes_str = "".join([f"<|speech_{idx}|>" for idx in ref_codes])
+        prompt = (
+            f"user: Convert the text to speech:<|TEXT_PROMPT_START|>{ref_text} {input_text}"
+            f"<|TEXT_PROMPT_END|>\nassistant:<|SPEECH_GENERATION_START|>{codes_str}"
+        )
+        output = self.backbone(
+            prompt,
+            max_tokens=self.max_context,
+            temperature=1.0,
+            top_k=50,
+            stop=["<|SPEECH_GENERATION_END|>"],
+        )
+        output_str = output["choices"][0]["text"]
+        return output_str
+
+    def _infer_stream_ggml(self, ref_codes: torch.Tensor, ref_text: str, input_text: str) -> Generator[np.ndarray, None, None]:
+        ref_text = phonemize_with_dict(ref_text)
+        input_text = phonemize_with_dict(input_text)
+
+        codes_str = "".join([f"<|speech_{idx}|>" for idx in ref_codes])
+        prompt = (
+            f"user: Convert the text to speech:<|TEXT_PROMPT_START|>{ref_text} {input_text}"
+            f"<|TEXT_PROMPT_END|>\nassistant:<|SPEECH_GENERATION_START|>{codes_str}"
+        )
+
+        audio_cache: list[np.ndarray] = []
+        token_cache: list[str] = [f"<|speech_{idx}|>" for idx in ref_codes]
+        n_decoded_samples: int = 0
+        n_decoded_tokens: int = len(ref_codes)
+
+        for item in self.backbone(
+            prompt,
+            max_tokens=self.max_context,
+            temperature=0.2,
+            top_k=50,
+            stop=["<|SPEECH_GENERATION_END|>"],
+            stream=True
+        ):
+            output_str = item["choices"][0]["text"]
+            token_cache.append(output_str)
+
+            if len(token_cache[n_decoded_tokens:]) >= self.streaming_frames_per_chunk + self.streaming_lookforward:
+
+                # decode chunk
+                tokens_start = max(
+                    n_decoded_tokens
+                    - self.streaming_lookback
+                    - self.streaming_overlap_frames,
+                    0
+                )
+                tokens_end = (
+                    n_decoded_tokens
+                    + self.streaming_frames_per_chunk
+                    + self.streaming_lookforward
+                    + self.streaming_overlap_frames
+                )
+                sample_start = (
+                    n_decoded_tokens - tokens_start
+                ) * self.hop_length
+                sample_end = (
+                    sample_start
+                    + (self.streaming_frames_per_chunk + 2 * self.streaming_overlap_frames) * self.hop_length
+                )
+                curr_codes = token_cache[tokens_start:tokens_end]
+                recon = self._decode("".join(curr_codes))
+                recon = recon[sample_start:sample_end]
+                audio_cache.append(recon)
+
+                # postprocess
+                processed_recon = _linear_overlap_add(
+                    audio_cache, stride=self.streaming_stride_samples
+                )
+                new_samples_end = len(audio_cache) * self.streaming_stride_samples
+                processed_recon = processed_recon[
+                    n_decoded_samples:new_samples_end
+                ]
+                n_decoded_samples = new_samples_end
+                n_decoded_tokens += self.streaming_frames_per_chunk
+                yield processed_recon
+
+        # final decoding handled separately as non-constant chunk size
+        remaining_tokens = len(token_cache) - n_decoded_tokens
+        if len(token_cache) > n_decoded_tokens:
+            tokens_start = max(
+                len(token_cache)
+                - (self.streaming_lookback + self.streaming_overlap_frames + remaining_tokens), 
+                0
+            )
+            sample_start = (
+                len(token_cache) 
+                - tokens_start 
+                - remaining_tokens 
+                - self.streaming_overlap_frames
+            ) * self.hop_length
+            curr_codes = token_cache[tokens_start:]
+            recon = self._decode("".join(curr_codes))
+            recon = recon[sample_start:]
+            audio_cache.append(recon)
+
+            processed_recon = _linear_overlap_add(audio_cache, stride=self.streaming_stride_samples)
+            processed_recon = processed_recon[n_decoded_samples:]
+            yield processed_recon