mirror of
https://github.com/jamiepine/voicebox.git
synced 2026-09-28 22:55:15 -07:00
Merge branch 'main' into fix/model-size-selection-ignored
This commit is contained in:
@@ -29,9 +29,23 @@ class PyTorchTTSBackend:
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"""Get the best available device."""
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if torch.cuda.is_available():
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return "cuda"
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elif hasattr(torch.backends, 'mps') and torch.backends.mps.is_available():
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# MPS can have issues, use CPU for stability
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return "cpu"
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# Intel Arc / Intel Xe GPU via intel-extension-for-pytorch (IPEX)
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try:
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import intel_extension_for_pytorch # noqa: F401
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if hasattr(torch, 'xpu') and torch.xpu.is_available():
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return "xpu"
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except ImportError:
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pass
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# Any GPU on Windows via DirectML (torch-directml)
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try:
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import torch_directml
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if torch_directml.device_count() > 0:
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return torch_directml.device(0)
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except ImportError:
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pass
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# MPS (Apple Silicon) — kept for completeness but MLX backend is preferred
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if hasattr(torch.backends, 'mps') and torch.backends.mps.is_available():
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return "cpu" # MPS disabled for stability; MLX backend handles Apple Silicon
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return "cpu"
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def is_loaded(self) -> bool:
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@@ -166,11 +180,21 @@ class PyTorchTTSBackend:
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# Load the model (tqdm is patched, but filters out non-download progress)
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try:
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self.model = Qwen3TTSModel.from_pretrained(
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model_path,
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device_map=self.device,
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torch_dtype=torch.float32 if self.device == "cpu" else torch.bfloat16,
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)
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# Don't pass device_map on CPU: accelerate's meta-tensor mechanism
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# causes "Cannot copy out of meta tensor" when moving to CPU.
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# Instead load directly then call .to(device) if needed.
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if self.device == "cpu":
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self.model = Qwen3TTSModel.from_pretrained(
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model_path,
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torch_dtype=torch.float32,
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low_cpu_mem_usage=False,
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)
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else:
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self.model = Qwen3TTSModel.from_pretrained(
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model_path,
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device_map=self.device,
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torch_dtype=torch.bfloat16,
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)
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finally:
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# Exit the patch context
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tracker_context.__exit__(None, None, None)
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@@ -358,9 +382,22 @@ class PyTorchSTTBackend:
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"""Get the best available device."""
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if torch.cuda.is_available():
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return "cuda"
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elif hasattr(torch.backends, 'mps') and torch.backends.mps.is_available():
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# MPS support for Whisper
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return "cpu" # Use CPU for stability
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# Intel Arc / Intel Xe GPU via intel-extension-for-pytorch (IPEX)
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try:
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import intel_extension_for_pytorch # noqa: F401
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if hasattr(torch, 'xpu') and torch.xpu.is_available():
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return "xpu"
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except ImportError:
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pass
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# Any GPU on Windows via DirectML (torch-directml)
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try:
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import torch_directml
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if torch_directml.device_count() > 0:
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return torch_directml.device(0)
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except ImportError:
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pass
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if hasattr(torch.backends, 'mps') and torch.backends.mps.is_available():
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return "cpu" # MPS disabled for stability
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return "cpu"
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def is_loaded(self) -> bool:
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@@ -4,8 +4,17 @@ Configuration module for voicebox backend.
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Handles data directory configuration for production bundling.
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"""
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import os
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from pathlib import Path
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# Allow users to override the HuggingFace model download directory.
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# Set VOICEBOX_MODELS_DIR to an absolute path before starting the server.
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# This sets HF_HUB_CACHE so all huggingface_hub downloads go to that path.
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_custom_models_dir = os.environ.get("VOICEBOX_MODELS_DIR")
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if _custom_models_dir:
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os.environ["HF_HUB_CACHE"] = _custom_models_dir
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print(f"[config] Model download path set to: {_custom_models_dir}")
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# Default data directory (used in development)
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_data_dir = Path("data")
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+147
-36
@@ -22,6 +22,24 @@ import uuid
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import asyncio
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import signal
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import os
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from urllib.parse import quote
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def _safe_content_disposition(disposition_type: str, filename: str) -> str:
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"""Build a Content-Disposition header that is safe for non-ASCII filenames.
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Uses RFC 5987 ``filename*`` parameter so that browsers can decode
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UTF-8 filenames while the ``filename`` fallback stays ASCII-only.
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"""
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ascii_name = "".join(
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c for c in filename if c.isascii() and (c.isalnum() or c in " -_.")
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).strip() or "download"
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utf8_name = quote(filename, safe="")
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return (
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f'{disposition_type}; filename="{ascii_name}"; '
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f"filename*=UTF-8''{utf8_name}"
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)
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from . import database, models, profiles, history, tts, transcribe, config, export_import, channels, stories, __version__
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from .database import get_db, Generation as DBGeneration, VoiceProfile as DBVoiceProfile
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@@ -77,10 +95,39 @@ async def health():
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tts_model = tts.get_tts_model()
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backend_type = get_backend_type()
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# Check for GPU availability (CUDA or MPS)
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# Check for GPU availability (CUDA, MPS, Intel Arc XPU, or DirectML)
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has_cuda = torch.cuda.is_available()
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has_mps = hasattr(torch.backends, 'mps') and torch.backends.mps.is_available()
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gpu_available = has_cuda or has_mps
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# Intel Arc / Intel Xe via intel-extension-for-pytorch (IPEX)
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has_xpu = False
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xpu_name = None
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try:
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import intel_extension_for_pytorch as ipex # noqa: F401
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if hasattr(torch, 'xpu') and torch.xpu.is_available():
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has_xpu = True
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try:
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xpu_name = torch.xpu.get_device_name(0)
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except Exception:
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xpu_name = "Intel GPU"
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except ImportError:
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pass
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# DirectML backend (torch-directml) for any Windows GPU
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has_directml = False
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directml_name = None
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try:
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import torch_directml
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if torch_directml.device_count() > 0:
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has_directml = True
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try:
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directml_name = torch_directml.device_name(0)
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except Exception:
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directml_name = "DirectML GPU"
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except ImportError:
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pass
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gpu_available = has_cuda or has_mps or has_xpu or has_directml or backend_type == "mlx"
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gpu_type = None
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if has_cuda:
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@@ -89,6 +136,10 @@ async def health():
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gpu_type = "MPS (Apple Silicon)"
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elif backend_type == "mlx":
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gpu_type = "Metal (Apple Silicon via MLX)"
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elif has_xpu:
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gpu_type = f"XPU ({xpu_name})"
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elif has_directml:
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gpu_type = f"DirectML ({directml_name})"
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vram_used = None
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if has_cuda:
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@@ -252,12 +303,17 @@ async def add_profile_sample(
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db: Session = Depends(get_db),
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):
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"""Add a sample to a voice profile."""
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# Save uploaded file to temporary location
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with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as tmp:
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# Preserve the uploaded file's extension so librosa can detect format correctly.
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# Defaulting to .wav was causing soundfile to reject MP3/WebM content as invalid WAV.
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_allowed_audio_exts = {'.wav', '.mp3', '.m4a', '.ogg', '.flac', '.aac', '.webm', '.opus'}
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_uploaded_ext = Path(file.filename or '').suffix.lower()
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file_suffix = _uploaded_ext if _uploaded_ext in _allowed_audio_exts else '.wav'
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with tempfile.NamedTemporaryFile(suffix=file_suffix, delete=False) as tmp:
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content = await file.read()
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tmp.write(content)
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tmp_path = tmp.name
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try:
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sample = await profiles.add_profile_sample(
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profile_id,
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@@ -268,6 +324,8 @@ async def add_profile_sample(
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return sample
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except ValueError as e:
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raise HTTPException(status_code=400, detail=str(e))
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except Exception as e:
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raise HTTPException(status_code=500, detail=f"Failed to process audio file: {str(e)}")
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finally:
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# Clean up temp file
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Path(tmp_path).unlink(missing_ok=True)
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@@ -388,7 +446,7 @@ async def export_profile(
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io.BytesIO(zip_bytes),
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media_type="application/zip",
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headers={
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"Content-Disposition": f'attachment; filename="{filename}"'
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"Content-Disposition": _safe_content_disposition("attachment", filename)
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}
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)
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except ValueError as e:
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@@ -543,44 +601,44 @@ async def generate_speech(
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raise HTTPException(status_code=404, detail="Profile not found")
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# Generate audio
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# Resolve model size and load the correct model FIRST.
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# This must happen before create_voice_prompt_for_profile because that
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# function calls load_model_async(None), which falls back to self.model_size.
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# If the model is already loaded with the right size at that point, it
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# returns immediately and the voice prompt is created by the correct model.
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tts_model = tts.get_tts_model()
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# Load the requested model size if different from current (async to not block)
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model_size = data.model_size or "1.7B"
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# Check if model needs to be downloaded first
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model_path = tts_model._get_model_path(model_size)
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if model_path.startswith("Qwen/"):
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# Model not cached - check if it exists remotely or needs download
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from huggingface_hub import constants as hf_constants
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repo_cache = Path(hf_constants.HF_HUB_CACHE) / ("models--" + model_path.replace("/", "--"))
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if not repo_cache.exists():
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# Start download in background
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model_name = f"qwen-tts-{model_size}"
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if not tts_model._is_model_cached(model_size):
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# Model is not fully cached — kick off a background download and tell
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# the client to retry once it's ready.
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model_name = f"qwen-tts-{model_size}"
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async def download_model_background():
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try:
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await tts_model.load_model_async(model_size)
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except Exception as e:
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task_manager.error_download(model_name, str(e))
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async def download_model_background():
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try:
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await tts_model.load_model_async(model_size)
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except Exception as e:
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task_manager.error_download(model_name, str(e))
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task_manager.start_download(model_name)
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asyncio.create_task(download_model_background())
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task_manager.start_download(model_name)
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asyncio.create_task(download_model_background())
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# Return 202 Accepted with download info
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raise HTTPException(
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status_code=202,
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detail={
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"message": f"Model {model_size} is being downloaded. Please wait and try again.",
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"model_name": model_name,
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"downloading": True
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}
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)
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raise HTTPException(
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status_code=202,
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detail={
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"message": f"Model {model_size} is being downloaded. Please wait and try again.",
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"model_name": model_name,
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"downloading": True,
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},
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)
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# Load the requested model BEFORE creating voice prompt,
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# so create_voice_prompt uses the correct model size
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# Load (or switch to) the requested model before building the voice prompt
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await tts_model.load_model_async(model_size)
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# Create voice prompt from profile
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# Create voice prompt from profile (model is already loaded with correct size)
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voice_prompt = await profiles.create_voice_prompt_for_profile(
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data.profile_id,
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db,
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@@ -628,6 +686,59 @@ async def generate_speech(
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raise HTTPException(status_code=500, detail=str(e))
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@app.post("/generate/stream")
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async def stream_speech(
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data: models.GenerationRequest,
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db: Session = Depends(get_db),
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):
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"""
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Generate speech and stream the WAV audio directly without saving to disk.
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Returns raw WAV bytes via a StreamingResponse so the client can start
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playing audio before the entire file has been received. This endpoint
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does NOT create a history entry — use /generate for that.
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"""
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profile = await profiles.get_profile(data.profile_id, db)
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if not profile:
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raise HTTPException(status_code=404, detail="Profile not found")
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tts_model = tts.get_tts_model()
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model_size = data.model_size or "1.7B"
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if not tts_model._is_model_cached(model_size):
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raise HTTPException(
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status_code=400,
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detail=f"Model {model_size} is not downloaded yet. Use /generate to trigger a download.",
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)
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# Load the correct model before building the voice prompt (fixes issue #96)
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await tts_model.load_model_async(model_size)
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voice_prompt = await profiles.create_voice_prompt_for_profile(data.profile_id, db)
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audio, sample_rate = await tts_model.generate(
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data.text,
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voice_prompt,
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data.language,
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data.seed,
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data.instruct,
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)
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wav_bytes = tts.audio_to_wav_bytes(audio, sample_rate)
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async def _wav_stream():
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# Yield in chunks so large responses don't block the event loop
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chunk_size = 64 * 1024 # 64 KB
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for i in range(0, len(wav_bytes), chunk_size):
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yield wav_bytes[i : i + chunk_size]
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return StreamingResponse(
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_wav_stream(),
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media_type="audio/wav",
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headers={"Content-Disposition": 'attachment; filename="speech.wav"'},
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)
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# ============================================
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# HISTORY ENDPOINTS
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# ============================================
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@@ -756,7 +867,7 @@ async def export_generation(
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io.BytesIO(zip_bytes),
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media_type="application/zip",
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headers={
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"Content-Disposition": f'attachment; filename="{filename}"'
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"Content-Disposition": _safe_content_disposition("attachment", filename)
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}
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)
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except ValueError as e:
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@@ -789,7 +900,7 @@ async def export_generation_audio(
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audio_path,
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media_type="audio/wav",
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headers={
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"Content-Disposition": f'attachment; filename="{filename}"'
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"Content-Disposition": _safe_content_disposition("attachment", filename)
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}
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)
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@@ -1057,7 +1168,7 @@ async def export_story_audio(
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io.BytesIO(audio_bytes),
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media_type="audio/wav",
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headers={
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"Content-Disposition": f'attachment; filename="{filename}"'
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"Content-Disposition": _safe_content_disposition("attachment", filename)
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}
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)
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except HTTPException:
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@@ -32,11 +32,3 @@ def audio_to_wav_bytes(audio: np.ndarray, sample_rate: int) -> bytes:
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sf.write(buffer, audio, sample_rate, format="WAV")
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buffer.seek(0)
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return buffer.read()
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def audio_to_wav_bytes(audio: np.ndarray, sample_rate: int) -> bytes:
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"""Convert audio array to WAV bytes."""
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buffer = io.BytesIO()
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sf.write(buffer, audio, sample_rate, format="WAV")
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buffer.seek(0)
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return buffer.read()
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Reference in New Issue
Block a user