feat: 3-mode selfhosted refactoring (--gpu, --cpu, --hosted) + audio token auth fallback (#896)

* fix: local processing instead of http server for cpu

* add fallback token if service worker doesnt work

* chore: rename processors to keep processor pattern up to date and allow other processors to be createed and used with env vars

server/.env.example

@@ -89,11 +89,11 @@ LLM_CONTEXT_WINDOW=16000
 ## =======================================================
 ## Audio Padding
 ##
-## backends: local (in-process PyAV), modal (HTTP API client)
-## Default is "local" — no external service needed.
+## backends: pyav (in-process PyAV), modal (HTTP API client)
+## Default is "pyav" — no external service needed.
 ## Set to "modal" when using Modal.com or self-hosted gpu/self_hosted/ container.
 ## =======================================================
-#PADDING_BACKEND=local
+#PADDING_BACKEND=pyav
 #PADDING_BACKEND=modal
 #PADDING_URL=https://xxxxx--reflector-padding-web.modal.run
 #PADDING_MODAL_API_KEY=xxxxx
@@ -101,8 +101,8 @@ LLM_CONTEXT_WINDOW=16000
 ## =======================================================
 ## Diarization
 ##
-## Only available on modal
-## To allow diarization, you need to expose expose the files to be dowloded by the pipeline
+## backends: modal (HTTP API), pyannote (in-process pyannote.audio)
+## To allow diarization, you need to expose expose the files to be downloaded by the pipeline
 ## =======================================================
 DIARIZATION_ENABLED=false
 DIARIZATION_BACKEND=modal

server/.env.selfhosted.example

@@ -32,26 +32,46 @@ AUTH_BACKEND=none
 
 # =======================================================
 # Specialized Models (Transcription, Diarization, Translation)
-# These run in the gpu/cpu container — NOT an LLM.
-# The "modal" backend means "HTTP API client" — it talks to
-# the self-hosted container, not Modal.com cloud.
+# These do NOT use an LLM. Configured per mode by the setup script:
+#
+# --gpu mode:    modal backends → GPU container (http://transcription:8000)
+# --cpu mode:    whisper/pyannote/marian/pyav → in-process ML on server/worker
+# --hosted mode: modal backends → user-provided remote GPU service URL
 # =======================================================
+
+# --- --gpu mode (default) ---
 TRANSCRIPT_BACKEND=modal
 TRANSCRIPT_URL=http://transcription:8000
 TRANSCRIPT_MODAL_API_KEY=selfhosted
-
 DIARIZATION_ENABLED=true
 DIARIZATION_BACKEND=modal
 DIARIZATION_URL=http://transcription:8000
-
 TRANSLATION_BACKEND=modal
 TRANSLATE_URL=http://transcription:8000
-
 PADDING_BACKEND=modal
 PADDING_URL=http://transcription:8000
 
-# HuggingFace token — optional, for gated models (e.g. pyannote).
-# Falls back to public S3 model bundle if not set.
+# --- --cpu mode (set by setup script) ---
+# TRANSCRIPT_BACKEND=whisper
+# DIARIZATION_BACKEND=pyannote
+# TRANSLATION_BACKEND=marian
+# PADDING_BACKEND=pyav
+
+# --- --hosted mode (set by setup script) ---
+# TRANSCRIPT_BACKEND=modal
+# TRANSCRIPT_URL=https://your-gpu-service.example.com
+# DIARIZATION_BACKEND=modal
+# DIARIZATION_URL=https://your-gpu-service.example.com
+# ... (all URLs point to one remote service)
+
+# Whisper model sizes for local transcription (--cpu mode)
+# Options: "tiny", "base", "small", "medium", "large-v2"
+# WHISPER_CHUNK_MODEL=tiny
+# WHISPER_FILE_MODEL=tiny
+
+# HuggingFace token — for gated models (e.g. pyannote diarization).
+# Required for --gpu and --cpu modes; falls back to public S3 bundle if not set.
+# Not needed for --hosted mode (remote service handles its own auth).
 # HF_TOKEN=hf_xxxxx
 
 # =======================================================

server/Dockerfile

@@ -6,7 +6,7 @@ ENV PYTHONUNBUFFERED=1 \
 
 # builder install base dependencies
 WORKDIR /tmp
-RUN apt-get update && apt-get install -y curl && apt-get clean
+RUN apt-get update && apt-get install -y curl ffmpeg && apt-get clean
 ADD https://astral.sh/uv/install.sh /uv-installer.sh
 RUN sh /uv-installer.sh && rm /uv-installer.sh
 ENV PATH="/root/.local/bin/:$PATH"

server/pyproject.toml

@@ -71,9 +71,12 @@ local = [
     "faster-whisper>=0.10.0",
 ]
 silero-vad = [
-    "silero-vad>=5.1.2",
+    "silero-vad==5.1.2",
     "torch>=2.8.0",
     "torchaudio>=2.8.0",
+    "pyannote.audio==3.4.0",
+    "pytorch-lightning<2.6",
+    "librosa==0.10.1",
 ]
 
 [tool.uv]

server/reflector/auth/__init__.py

@@ -14,6 +14,7 @@ current_user = auth_module.current_user
 current_user_optional = auth_module.current_user_optional
 parse_ws_bearer_token = auth_module.parse_ws_bearer_token
 current_user_ws_optional = auth_module.current_user_ws_optional
+verify_raw_token = auth_module.verify_raw_token
 
 # Optional router (e.g. for /auth/login in password backend)
 router = getattr(auth_module, "router", None)

server/reflector/auth/auth_jwt.py

@@ -144,3 +144,8 @@ async def current_user_ws_optional(websocket: "WebSocket") -> Optional[UserInfo]
     if not token:
         return None
     return await _authenticate_user(token, None, JWTAuth())
+
+
+async def verify_raw_token(token: str) -> Optional[UserInfo]:
+    """Verify a raw JWT token string (used for query-param auth fallback)."""
+    return await _authenticate_user(token, None, JWTAuth())

server/reflector/auth/auth_none.py

@@ -27,3 +27,8 @@ def parse_ws_bearer_token(websocket):
 
 async def current_user_ws_optional(websocket):
     return None
+
+
+async def verify_raw_token(token):
+    """Verify a raw JWT token string (used for query-param auth fallback)."""
+    return None

server/reflector/auth/auth_password.py

@@ -168,6 +168,11 @@ async def current_user_ws_optional(websocket: "WebSocket") -> Optional[UserInfo]
     return await _authenticate_user(token, None)
 
 
+async def verify_raw_token(token: str) -> Optional[UserInfo]:
+    """Verify a raw JWT token string (used for query-param auth fallback)."""
+    return await _authenticate_user(token, None)
+
+
 # --- Login router ---
 router = APIRouter(prefix="/auth", tags=["auth"])
 

server/reflector/processors/__init__.py

@@ -4,6 +4,8 @@ from .audio_diarization_auto import AudioDiarizationAutoProcessor  # noqa: F401
 from .audio_downscale import AudioDownscaleProcessor  # noqa: F401
 from .audio_file_writer import AudioFileWriterProcessor  # noqa: F401
 from .audio_merge import AudioMergeProcessor  # noqa: F401
+from .audio_padding import AudioPaddingProcessor  # noqa: F401
+from .audio_padding_auto import AudioPaddingAutoProcessor  # noqa: F401
 from .audio_transcript import AudioTranscriptProcessor  # noqa: F401
 from .audio_transcript_auto import AudioTranscriptAutoProcessor  # noqa: F401
 from .base import (  # noqa: F401

server/reflector/processors/_audio_download.py

@@ -0,0 +1,86 @@
+"""
+Shared audio download utility for local processors.
+
+Downloads audio from a URL to a temporary file for in-process ML inference.
+"""
+
+import asyncio
+import os
+import tempfile
+from pathlib import Path
+
+import requests
+
+from reflector.logger import logger
+
+S3_TIMEOUT = 60
+
+
+async def download_audio_to_temp(url: str) -> Path:
+    """Download audio from URL to a temporary file.
+
+    The caller is responsible for deleting the temp file after use.
+
+    Args:
+        url: Presigned URL or public URL to download audio from.
+
+    Returns:
+        Path to the downloaded temporary file.
+    """
+    loop = asyncio.get_event_loop()
+    return await loop.run_in_executor(None, _download_blocking, url)
+
+
+def _download_blocking(url: str) -> Path:
+    """Blocking download implementation."""
+    log = logger.bind(url=url[:80])
+    log.info("Downloading audio to temp file")
+
+    response = requests.get(url, stream=True, timeout=S3_TIMEOUT)
+    response.raise_for_status()
+
+    # Determine extension from content-type or URL
+    ext = _detect_extension(url, response.headers.get("content-type", ""))
+
+    fd, tmp_path = tempfile.mkstemp(suffix=ext)
+    try:
+        total_bytes = 0
+        with os.fdopen(fd, "wb") as f:
+            for chunk in response.iter_content(chunk_size=8192):
+                if chunk:
+                    f.write(chunk)
+                    total_bytes += len(chunk)
+        log.info("Audio downloaded", bytes=total_bytes, path=tmp_path)
+        return Path(tmp_path)
+    except Exception:
+        # Clean up on failure
+        try:
+            os.unlink(tmp_path)
+        except OSError:
+            pass
+        raise
+
+
+def _detect_extension(url: str, content_type: str) -> str:
+    """Detect audio file extension from URL or content-type."""
+    # Try URL path first
+    path = url.split("?")[0]  # Strip query params
+    for ext in (".wav", ".mp3", ".mp4", ".m4a", ".webm", ".ogg", ".flac"):
+        if path.lower().endswith(ext):
+            return ext
+
+    # Try content-type
+    ct_map = {
+        "audio/wav": ".wav",
+        "audio/x-wav": ".wav",
+        "audio/mpeg": ".mp3",
+        "audio/mp4": ".m4a",
+        "audio/webm": ".webm",
+        "audio/ogg": ".ogg",
+        "audio/flac": ".flac",
+    }
+    for ct, ext in ct_map.items():
+        if ct in content_type.lower():
+            return ext
+
+    return ".audio"

server/reflector/processors/_marian_translator_service.py

@@ -0,0 +1,76 @@
+"""
+MarianMT translation service.
+
+Singleton service that loads HuggingFace MarianMT translation models
+and reuses them across all MarianMT translator processor instances.
+
+Ported from gpu/self_hosted/app/services/translator.py for in-process use.
+"""
+
+import logging
+import threading
+
+from transformers import MarianMTModel, MarianTokenizer, pipeline
+
+logger = logging.getLogger(__name__)
+
+
+class MarianTranslatorService:
+    """MarianMT text translation service for in-process use."""
+
+    def __init__(self):
+        self._pipeline = None
+        self._current_pair = None
+        self._lock = threading.Lock()
+
+    def load(self, source_language: str = "en", target_language: str = "fr"):
+        """Load the translation model for a specific language pair."""
+        model_name = self._resolve_model_name(source_language, target_language)
+        logger.info(
+            "Loading MarianMT model: %s (%s -> %s)",
+            model_name,
+            source_language,
+            target_language,
+        )
+        tokenizer = MarianTokenizer.from_pretrained(model_name)
+        model = MarianMTModel.from_pretrained(model_name)
+        self._pipeline = pipeline("translation", model=model, tokenizer=tokenizer)
+        self._current_pair = (source_language.lower(), target_language.lower())
+
+    def _resolve_model_name(self, src: str, tgt: str) -> str:
+        """Resolve language pair to MarianMT model name."""
+        pair = (src.lower(), tgt.lower())
+        mapping = {
+            ("en", "fr"): "Helsinki-NLP/opus-mt-en-fr",
+            ("fr", "en"): "Helsinki-NLP/opus-mt-fr-en",
+            ("en", "es"): "Helsinki-NLP/opus-mt-en-es",
+            ("es", "en"): "Helsinki-NLP/opus-mt-es-en",
+            ("en", "de"): "Helsinki-NLP/opus-mt-en-de",
+            ("de", "en"): "Helsinki-NLP/opus-mt-de-en",
+        }
+        return mapping.get(pair, "Helsinki-NLP/opus-mt-en-fr")
+
+    def translate(self, text: str, source_language: str, target_language: str) -> dict:
+        """Translate text between languages.
+
+        Args:
+            text: Text to translate.
+            source_language: Source language code (e.g. "en").
+            target_language: Target language code (e.g. "fr").
+
+        Returns:
+            dict with "text" key containing {source_language: original, target_language: translated}.
+        """
+        pair = (source_language.lower(), target_language.lower())
+        if self._pipeline is None or self._current_pair != pair:
+            self.load(source_language, target_language)
+        with self._lock:
+            results = self._pipeline(
+                text, src_lang=source_language, tgt_lang=target_language
+            )
+        translated = results[0]["translation_text"] if results else ""
+        return {"text": {source_language: text, target_language: translated}}
+
+
+# Module-level singleton — shared across all MarianMT translator processors
+translator_service = MarianTranslatorService()

server/reflector/processors/_pyannote_diarization_service.py

@@ -0,0 +1,133 @@
+"""
+Pyannote diarization service using pyannote.audio.
+
+Singleton service that loads the pyannote speaker diarization model once
+and reuses it across all pyannote diarization processor instances.
+
+Ported from gpu/self_hosted/app/services/diarizer.py for in-process use.
+"""
+
+import logging
+import tarfile
+import threading
+from pathlib import Path
+from urllib.request import urlopen
+
+import torch
+import torchaudio
+import yaml
+from pyannote.audio import Pipeline
+
+from reflector.settings import settings
+
+logger = logging.getLogger(__name__)
+
+S3_BUNDLE_URL = "https://reflector-public.s3.us-east-1.amazonaws.com/pyannote-speaker-diarization-3.1.tar.gz"
+BUNDLE_CACHE_DIR = Path.home() / ".cache" / "pyannote-bundle"
+
+
+def _ensure_model(cache_dir: Path) -> str:
+    """Download and extract S3 model bundle if not cached."""
+    model_dir = cache_dir / "pyannote-speaker-diarization-3.1"
+    config_path = model_dir / "config.yaml"
+
+    if config_path.exists():
+        logger.info("Using cached model bundle at %s", model_dir)
+        return str(model_dir)
+
+    cache_dir.mkdir(parents=True, exist_ok=True)
+    tarball_path = cache_dir / "model.tar.gz"
+
+    logger.info("Downloading model bundle from %s", S3_BUNDLE_URL)
+    with urlopen(S3_BUNDLE_URL) as response, open(tarball_path, "wb") as f:
+        while chunk := response.read(8192):
+            f.write(chunk)
+
+    logger.info("Extracting model bundle")
+    with tarfile.open(tarball_path, "r:gz") as tar:
+        tar.extractall(path=cache_dir, filter="data")
+    tarball_path.unlink()
+
+    _patch_config(model_dir, cache_dir)
+    return str(model_dir)
+
+
+def _patch_config(model_dir: Path, cache_dir: Path) -> None:
+    """Rewrite config.yaml to reference local pytorch_model.bin paths."""
+    config_path = model_dir / "config.yaml"
+    with open(config_path) as f:
+        config = yaml.safe_load(f)
+
+    config["pipeline"]["params"]["segmentation"] = str(
+        cache_dir / "pyannote-segmentation-3.0" / "pytorch_model.bin"
+    )
+    config["pipeline"]["params"]["embedding"] = str(
+        cache_dir / "pyannote-wespeaker-voxceleb-resnet34-LM" / "pytorch_model.bin"
+    )
+
+    with open(config_path, "w") as f:
+        yaml.dump(config, f)
+
+    logger.info("Patched config.yaml with local model paths")
+
+
+class PyannoteDiarizationService:
+    """Pyannote speaker diarization service for in-process use."""
+
+    def __init__(self):
+        self._pipeline = None
+        self._device = "cpu"
+        self._lock = threading.Lock()
+
+    def load(self):
+        self._device = "cuda" if torch.cuda.is_available() else "cpu"
+        hf_token = settings.HF_TOKEN
+
+        if hf_token:
+            logger.info("Loading pyannote model from HuggingFace (HF_TOKEN set)")
+            self._pipeline = Pipeline.from_pretrained(
+                "pyannote/speaker-diarization-3.1",
+                use_auth_token=hf_token,
+            )
+        else:
+            logger.info("HF_TOKEN not set — loading model from S3 bundle")
+            model_path = _ensure_model(BUNDLE_CACHE_DIR)
+            config_path = Path(model_path) / "config.yaml"
+            self._pipeline = Pipeline.from_pretrained(str(config_path))
+
+        self._pipeline.to(torch.device(self._device))
+
+    def diarize_file(self, file_path: str, timestamp: float = 0.0) -> dict:
+        """Run speaker diarization on an audio file.
+
+        Args:
+            file_path: Path to the audio file.
+            timestamp: Offset to add to all segment timestamps.
+
+        Returns:
+            dict with "diarization" key containing list of
+            {"start": float, "end": float, "speaker": int} segments.
+        """
+        if self._pipeline is None:
+            self.load()
+        waveform, sample_rate = torchaudio.load(file_path)
+        with self._lock:
+            diarization = self._pipeline(
+                {"waveform": waveform, "sample_rate": sample_rate}
+            )
+        segments = []
+        for diarization_segment, _, speaker in diarization.itertracks(yield_label=True):
+            segments.append(
+                {
+                    "start": round(timestamp + diarization_segment.start, 3),
+                    "end": round(timestamp + diarization_segment.end, 3),
+                    "speaker": int(speaker[-2:])
+                    if speaker and speaker[-2:].isdigit()
+                    else 0,
+                }
+            )
+        return {"diarization": segments}
+
+
+# Module-level singleton — shared across all pyannote diarization processors
+diarization_service = PyannoteDiarizationService()

server/reflector/processors/audio_diarization_pyannote.py

@@ -0,0 +1,37 @@
+"""
+Pyannote audio diarization processor using pyannote.audio in-process.
+
+Downloads audio from URL, runs pyannote diarization locally,
+and returns speaker segments. No HTTP backend needed.
+"""
+
+import asyncio
+import os
+
+from reflector.processors._audio_download import download_audio_to_temp
+from reflector.processors._pyannote_diarization_service import diarization_service
+from reflector.processors.audio_diarization import AudioDiarizationProcessor
+from reflector.processors.audio_diarization_auto import AudioDiarizationAutoProcessor
+from reflector.processors.types import AudioDiarizationInput
+
+
+class AudioDiarizationPyannoteProcessor(AudioDiarizationProcessor):
+    INPUT_TYPE = AudioDiarizationInput
+
+    async def _diarize(self, data: AudioDiarizationInput):
+        """Run pyannote diarization on audio from URL."""
+        tmp_path = await download_audio_to_temp(data.audio_url)
+        try:
+            loop = asyncio.get_event_loop()
+            result = await loop.run_in_executor(
+                None, diarization_service.diarize_file, str(tmp_path)
+            )
+            return result["diarization"]
+        finally:
+            try:
+                os.unlink(tmp_path)
+            except OSError:
+                pass
+
+
+AudioDiarizationAutoProcessor.register("pyannote", AudioDiarizationPyannoteProcessor)

server/reflector/processors/audio_padding.py

@@ -0,0 +1,23 @@
+"""
+Base class for audio padding processors.
+"""
+
+from pydantic import BaseModel
+
+
+class PaddingResponse(BaseModel):
+    size: int
+    cancelled: bool = False
+
+
+class AudioPaddingProcessor:
+    """Base class for audio padding processors."""
+
+    async def pad_track(
+        self,
+        track_url: str,
+        output_url: str,
+        start_time_seconds: float,
+        track_index: int,
+    ) -> PaddingResponse:
+        raise NotImplementedError

server/reflector/processors/audio_padding_auto.py

@@ -1,9 +1,10 @@
 import importlib
 
+from reflector.processors.audio_padding import AudioPaddingProcessor
 from reflector.settings import settings
 
 
-class AudioPaddingAutoProcessor:
+class AudioPaddingAutoProcessor(AudioPaddingProcessor):
     _registry = {}
 
     @classmethod

server/reflector/processors/audio_padding_modal.py

@@ -6,19 +6,14 @@ import asyncio
 import os
 
 import httpx
-from pydantic import BaseModel
 
 from reflector.hatchet.constants import TIMEOUT_AUDIO
 from reflector.logger import logger
+from reflector.processors.audio_padding import AudioPaddingProcessor, PaddingResponse
 from reflector.processors.audio_padding_auto import AudioPaddingAutoProcessor
 
 
-class PaddingResponse(BaseModel):
-    size: int
-    cancelled: bool = False
-
-
-class AudioPaddingModalProcessor:
+class AudioPaddingModalProcessor(AudioPaddingProcessor):
     """Audio padding processor using Modal.com CPU backend via HTTP."""
 
     def __init__(

server/reflector/processors/audio_padding_pyav.py

@@ -1,5 +1,5 @@
 """
-Local audio padding processor using PyAV.
+PyAV audio padding processor.
 
 Pads audio tracks with silence directly in-process (no HTTP).
 Reuses the shared PyAV utilities from reflector.utils.audio_padding.
@@ -12,15 +12,15 @@ import tempfile
 import av
 
 from reflector.logger import logger
+from reflector.processors.audio_padding import AudioPaddingProcessor, PaddingResponse
 from reflector.processors.audio_padding_auto import AudioPaddingAutoProcessor
-from reflector.processors.audio_padding_modal import PaddingResponse
 from reflector.utils.audio_padding import apply_audio_padding_to_file
 
 S3_TIMEOUT = 60
 
 
-class AudioPaddingLocalProcessor:
-    """Audio padding processor using local PyAV (no HTTP backend)."""
+class AudioPaddingPyavProcessor(AudioPaddingProcessor):
+    """Audio padding processor using PyAV (no HTTP backend)."""
 
     async def pad_track(
         self,
@@ -29,7 +29,7 @@ class AudioPaddingLocalProcessor:
         start_time_seconds: float,
         track_index: int,
     ) -> PaddingResponse:
-        """Pad audio track with silence locally via PyAV.
+        """Pad audio track with silence via PyAV.
 
         Args:
             track_url: Presigned GET URL for source audio track
@@ -130,4 +130,4 @@ class AudioPaddingLocalProcessor:
                 log.warning("Failed to cleanup temp directory", error=str(e))
 
 
-AudioPaddingAutoProcessor.register("local", AudioPaddingLocalProcessor)
+AudioPaddingAutoProcessor.register("pyav", AudioPaddingPyavProcessor)

server/reflector/processors/audio_transcript_whisper.py

@@ -3,13 +3,17 @@ from faster_whisper import WhisperModel
 from reflector.processors.audio_transcript import AudioTranscriptProcessor
 from reflector.processors.audio_transcript_auto import AudioTranscriptAutoProcessor
 from reflector.processors.types import AudioFile, Transcript, Word
+from reflector.settings import settings
 
 
 class AudioTranscriptWhisperProcessor(AudioTranscriptProcessor):
     def __init__(self):
         super().__init__()
         self.model = WhisperModel(
-            "tiny", device="cpu", compute_type="float32", num_workers=12
+            settings.WHISPER_CHUNK_MODEL,
+            device="cpu",
+            compute_type="float32",
+            num_workers=12,
         )
 
     async def _transcript(self, data: AudioFile):

server/reflector/processors/file_diarization_pyannote.py

@@ -0,0 +1,39 @@
+"""
+Pyannote file diarization processor using pyannote.audio in-process.
+
+Downloads audio from URL, runs pyannote diarization locally,
+and returns speaker segments. No HTTP backend needed.
+"""
+
+import asyncio
+import os
+
+from reflector.processors._audio_download import download_audio_to_temp
+from reflector.processors._pyannote_diarization_service import diarization_service
+from reflector.processors.file_diarization import (
+    FileDiarizationInput,
+    FileDiarizationOutput,
+    FileDiarizationProcessor,
+)
+from reflector.processors.file_diarization_auto import FileDiarizationAutoProcessor
+
+
+class FileDiarizationPyannoteProcessor(FileDiarizationProcessor):
+    async def _diarize(self, data: FileDiarizationInput):
+        """Run pyannote diarization on file from URL."""
+        self.logger.info(f"Starting pyannote diarization from {data.audio_url}")
+        tmp_path = await download_audio_to_temp(data.audio_url)
+        try:
+            loop = asyncio.get_event_loop()
+            result = await loop.run_in_executor(
+                None, diarization_service.diarize_file, str(tmp_path)
+            )
+            return FileDiarizationOutput(diarization=result["diarization"])
+        finally:
+            try:
+                os.unlink(tmp_path)
+            except OSError:
+                pass
+
+
+FileDiarizationAutoProcessor.register("pyannote", FileDiarizationPyannoteProcessor)

server/reflector/processors/file_transcript_whisper.py

@@ -0,0 +1,275 @@
+"""
+Local file transcription processor using faster-whisper with Silero VAD pipeline.
+
+Downloads audio from URL, segments it using Silero VAD, transcribes each
+segment with faster-whisper, and merges results. No HTTP backend needed.
+
+VAD pipeline ported from gpu/self_hosted/app/services/transcriber.py.
+"""
+
+import asyncio
+import os
+import shutil
+import subprocess
+import threading
+from typing import Generator
+
+import numpy as np
+from silero_vad import VADIterator, load_silero_vad
+
+from reflector.processors._audio_download import download_audio_to_temp
+from reflector.processors.file_transcript import (
+    FileTranscriptInput,
+    FileTranscriptProcessor,
+)
+from reflector.processors.file_transcript_auto import FileTranscriptAutoProcessor
+from reflector.processors.types import Transcript, Word
+from reflector.settings import settings
+
+SAMPLE_RATE = 16000
+
+VAD_CONFIG = {
+    "batch_max_duration": 30.0,
+    "silence_padding": 0.5,
+    "window_size": 512,
+}
+
+
+class FileTranscriptWhisperProcessor(FileTranscriptProcessor):
+    """Transcribe complete audio files using local faster-whisper with VAD."""
+
+    def __init__(self, **kwargs):
+        super().__init__(**kwargs)
+        self._model = None
+        self._lock = threading.Lock()
+
+    def _ensure_model(self):
+        """Lazy-load the whisper model on first use."""
+        if self._model is not None:
+            return
+
+        import faster_whisper
+        import torch
+
+        device = "cuda" if torch.cuda.is_available() else "cpu"
+        compute_type = "float16" if device == "cuda" else "int8"
+        model_name = settings.WHISPER_FILE_MODEL
+
+        self.logger.info(
+            "Loading whisper model",
+            model=model_name,
+            device=device,
+            compute_type=compute_type,
+        )
+        self._model = faster_whisper.WhisperModel(
+            model_name,
+            device=device,
+            compute_type=compute_type,
+            num_workers=1,
+        )
+
+    async def _transcript(self, data: FileTranscriptInput):
+        """Download file, run VAD segmentation, transcribe each segment."""
+        tmp_path = await download_audio_to_temp(data.audio_url)
+        try:
+            loop = asyncio.get_event_loop()
+            result = await loop.run_in_executor(
+                None,
+                self._transcribe_file_blocking,
+                str(tmp_path),
+                data.language,
+            )
+            return result
+        finally:
+            try:
+                os.unlink(tmp_path)
+            except OSError:
+                pass
+
+    def _transcribe_file_blocking(self, file_path: str, language: str) -> Transcript:
+        """Blocking transcription with VAD pipeline."""
+        self._ensure_model()
+
+        audio_array = _load_audio_via_ffmpeg(file_path, SAMPLE_RATE)
+
+        # VAD segmentation → batch merging
+        merged_batches: list[tuple[float, float]] = []
+        batch_start = None
+        batch_end = None
+        max_duration = VAD_CONFIG["batch_max_duration"]
+
+        for seg_start, seg_end in _vad_segments(audio_array):
+            if batch_start is None:
+                batch_start, batch_end = seg_start, seg_end
+                continue
+            if seg_end - batch_start <= max_duration:
+                batch_end = seg_end
+            else:
+                merged_batches.append((batch_start, batch_end))
+                batch_start, batch_end = seg_start, seg_end
+
+        if batch_start is not None and batch_end is not None:
+            merged_batches.append((batch_start, batch_end))
+
+        # If no speech detected, try transcribing the whole file
+        if not merged_batches:
+            return self._transcribe_whole_file(file_path, language)
+
+        # Transcribe each batch
+        all_words = []
+        for start_time, end_time in merged_batches:
+            s_idx = int(start_time * SAMPLE_RATE)
+            e_idx = int(end_time * SAMPLE_RATE)
+            segment = audio_array[s_idx:e_idx]
+            segment = _pad_audio(segment, SAMPLE_RATE)
+
+            with self._lock:
+                segments, _ = self._model.transcribe(
+                    segment,
+                    language=language,
+                    beam_size=5,
+                    word_timestamps=True,
+                    vad_filter=True,
+                    vad_parameters={"min_silence_duration_ms": 500},
+                )
+                segments = list(segments)
+
+            for seg in segments:
+                for w in seg.words:
+                    all_words.append(
+                        {
+                            "word": w.word,
+                            "start": round(float(w.start) + start_time, 2),
+                            "end": round(float(w.end) + start_time, 2),
+                        }
+                    )
+
+        all_words = _enforce_word_timing_constraints(all_words)
+
+        words = [
+            Word(text=w["word"], start=w["start"], end=w["end"]) for w in all_words
+        ]
+        words.sort(key=lambda w: w.start)
+        return Transcript(words=words)
+
+    def _transcribe_whole_file(self, file_path: str, language: str) -> Transcript:
+        """Fallback: transcribe entire file without VAD segmentation."""
+        with self._lock:
+            segments, _ = self._model.transcribe(
+                file_path,
+                language=language,
+                beam_size=5,
+                word_timestamps=True,
+                vad_filter=True,
+                vad_parameters={"min_silence_duration_ms": 500},
+            )
+            segments = list(segments)
+
+        words = []
+        for seg in segments:
+            for w in seg.words:
+                words.append(
+                    Word(
+                        text=w.word,
+                        start=round(float(w.start), 2),
+                        end=round(float(w.end), 2),
+                    )
+                )
+        return Transcript(words=words)
+
+
+# --- VAD helpers (ported from gpu/self_hosted/app/services/transcriber.py) ---
+# IMPORTANT: This VAD segment logic is duplicated for deployment isolation.
+# If you modify this, consider updating the GPU service copy as well:
+#   - gpu/self_hosted/app/services/transcriber.py
+#   - gpu/modal_deployments/reflector_transcriber.py
+#   - gpu/modal_deployments/reflector_transcriber_parakeet.py
+
+
+def _load_audio_via_ffmpeg(
+    input_path: str, sample_rate: int = SAMPLE_RATE
+) -> np.ndarray:
+    """Load audio file via ffmpeg, converting to mono float32 at target sample rate."""
+    ffmpeg_bin = shutil.which("ffmpeg") or "ffmpeg"
+    cmd = [
+        ffmpeg_bin,
+        "-nostdin",
+        "-threads",
+        "1",
+        "-i",
+        input_path,
+        "-f",
+        "f32le",
+        "-acodec",
+        "pcm_f32le",
+        "-ac",
+        "1",
+        "-ar",
+        str(sample_rate),
+        "pipe:1",
+    ]
+    proc = subprocess.run(
+        cmd, stdout=subprocess.PIPE, stderr=subprocess.PIPE, check=True
+    )
+    return np.frombuffer(proc.stdout, dtype=np.float32)
+
+
+def _vad_segments(
+    audio_array: np.ndarray,
+    sample_rate: int = SAMPLE_RATE,
+    window_size: int = VAD_CONFIG["window_size"],
+) -> Generator[tuple[float, float], None, None]:
+    """Detect speech segments using Silero VAD."""
+    vad_model = load_silero_vad(onnx=False)
+    iterator = VADIterator(vad_model, sampling_rate=sample_rate)
+    start = None
+
+    for i in range(0, len(audio_array), window_size):
+        chunk = audio_array[i : i + window_size]
+        if len(chunk) < window_size:
+            chunk = np.pad(chunk, (0, window_size - len(chunk)), mode="constant")
+        speech = iterator(chunk)
+        if not speech:
+            continue
+        if "start" in speech:
+            start = speech["start"]
+            continue
+        if "end" in speech and start is not None:
+            end = speech["end"]
+            yield (start / float(SAMPLE_RATE), end / float(SAMPLE_RATE))
+            start = None
+
+    # Handle case where audio ends while speech is still active
+    if start is not None:
+        audio_duration = len(audio_array) / float(sample_rate)
+        yield (start / float(SAMPLE_RATE), audio_duration)
+
+    iterator.reset_states()
+
+
+def _pad_audio(audio_array: np.ndarray, sample_rate: int = SAMPLE_RATE) -> np.ndarray:
+    """Pad short audio with silence for VAD compatibility."""
+    audio_duration = len(audio_array) / sample_rate
+    if audio_duration < VAD_CONFIG["silence_padding"]:
+        silence_samples = int(sample_rate * VAD_CONFIG["silence_padding"])
+        silence = np.zeros(silence_samples, dtype=np.float32)
+        return np.concatenate([audio_array, silence])
+    return audio_array
+
+
+def _enforce_word_timing_constraints(words: list[dict]) -> list[dict]:
+    """Ensure no word end time exceeds the next word's start time."""
+    if len(words) <= 1:
+        return words
+    enforced: list[dict] = []
+    for i, word in enumerate(words):
+        current = dict(word)
+        if i < len(words) - 1:
+            next_start = words[i + 1]["start"]
+            if current["end"] > next_start:
+                current["end"] = next_start
+        enforced.append(current)
+    return enforced
+
+
+FileTranscriptAutoProcessor.register("whisper", FileTranscriptWhisperProcessor)

server/reflector/processors/transcript_translator_marian.py

@@ -0,0 +1,50 @@
+"""
+MarianMT transcript translator processor using HuggingFace MarianMT in-process.
+
+Translates transcript text using HuggingFace MarianMT models
+locally. No HTTP backend needed.
+"""
+
+import asyncio
+
+from reflector.processors._marian_translator_service import translator_service
+from reflector.processors.transcript_translator import TranscriptTranslatorProcessor
+from reflector.processors.transcript_translator_auto import (
+    TranscriptTranslatorAutoProcessor,
+)
+from reflector.processors.types import TranslationLanguages
+
+
+class TranscriptTranslatorMarianProcessor(TranscriptTranslatorProcessor):
+    """Translate transcript text using MarianMT models."""
+
+    async def _translate(self, text: str) -> str | None:
+        source_language = self.get_pref("audio:source_language", "en")
+        target_language = self.get_pref("audio:target_language", "en")
+
+        languages = TranslationLanguages()
+        assert languages.is_supported(target_language)
+
+        self.logger.debug(f"MarianMT translate {text=}")
+
+        loop = asyncio.get_event_loop()
+        result = await loop.run_in_executor(
+            None,
+            translator_service.translate,
+            text,
+            source_language,
+            target_language,
+        )
+
+        if target_language in result["text"]:
+            translation = result["text"][target_language]
+        else:
+            translation = None
+
+        self.logger.debug(f"Translation result: {text=}, {translation=}")
+        return translation
+
+
+TranscriptTranslatorAutoProcessor.register(
+    "marian", TranscriptTranslatorMarianProcessor
+)

server/reflector/settings.py

@@ -40,11 +40,19 @@ class Settings(BaseSettings):
     # backends: silero, frames
     AUDIO_CHUNKER_BACKEND: str = "frames"
 
+    # HuggingFace token for gated models (pyannote diarization in --cpu mode)
+    HF_TOKEN: str | None = None
+
     # Audio Transcription
     # backends:
     #   - whisper: in-process model loading (no HTTP, runs in same process)
     #   - modal: HTTP API client (works with Modal.com OR self-hosted gpu/self_hosted/)
     TRANSCRIPT_BACKEND: str = "whisper"
+
+    # Whisper model sizes for local transcription
+    # Options: "tiny", "base", "small", "medium", "large-v2"
+    WHISPER_CHUNK_MODEL: str = "tiny"
+    WHISPER_FILE_MODEL: str = "tiny"
     TRANSCRIPT_URL: str | None = None
     TRANSCRIPT_TIMEOUT: int = 90
     TRANSCRIPT_FILE_TIMEOUT: int = 600
@@ -100,7 +108,7 @@ class Settings(BaseSettings):
     )
 
     # Diarization
-    # backend: modal — HTTP API client (works with Modal.com OR self-hosted gpu/self_hosted/)
+    # backends: modal — HTTP API client, pyannote — in-process pyannote.audio
     DIARIZATION_ENABLED: bool = True
     DIARIZATION_BACKEND: str = "modal"
     DIARIZATION_URL: str | None = None
@@ -111,9 +119,9 @@ class Settings(BaseSettings):
 
     # Audio Padding
     # backends:
-    #   - local: in-process PyAV padding (no HTTP, runs in same process)
+    #   - pyav: in-process PyAV padding (no HTTP, runs in same process)
     #   - modal: HTTP API client (works with Modal.com OR self-hosted gpu/self_hosted/)
-    PADDING_BACKEND: str = "local"
+    PADDING_BACKEND: str = "pyav"
     PADDING_URL: str | None = None
     PADDING_MODAL_API_KEY: str | None = None
 

server/reflector/views/transcripts_audio.py

@@ -13,7 +13,6 @@ from fastapi import APIRouter, Depends, HTTPException, Request, Response, status
 import reflector.auth as auth
 from reflector.db.transcripts import AudioWaveform, transcripts_controller
 from reflector.settings import settings
-from reflector.views.transcripts import ALGORITHM
 
 from ._range_requests_response import range_requests_response
 
@@ -36,16 +35,23 @@ async def transcript_get_audio_mp3(
 ):
     user_id = user["sub"] if user else None
     if not user_id and token:
-        unauthorized_exception = HTTPException(
-            status_code=status.HTTP_401_UNAUTHORIZED,
-            detail="Invalid or expired token",
-            headers={"WWW-Authenticate": "Bearer"},
-        )
         try:
-            payload = jwt.decode(token, settings.SECRET_KEY, algorithms=[ALGORITHM])
-            user_id: str = payload.get("sub")
-        except jwt.PyJWTError:
-            raise unauthorized_exception
+            token_user = await auth.verify_raw_token(token)
+        except Exception:
+            token_user = None
+        # Fallback: try as internal HS256 token (created by _generate_local_audio_link)
+        if not token_user:
+            try:
+                payload = jwt.decode(token, settings.SECRET_KEY, algorithms=["HS256"])
+                user_id = payload.get("sub")
+            except jwt.PyJWTError:
+                raise HTTPException(
+                    status_code=status.HTTP_401_UNAUTHORIZED,
+                    detail="Invalid or expired token",
+                    headers={"WWW-Authenticate": "Bearer"},
+                )
+        else:
+            user_id = token_user["sub"]
 
     transcript = await transcripts_controller.get_by_id_for_http(
         transcript_id, user_id=user_id

server/tests/test_processors_cpu.py

@@ -0,0 +1,450 @@
+"""
+Tests for in-process processor backends (--cpu mode).
+
+All ML model calls are mocked — no actual model loading needed.
+Tests verify processor registration, wiring, error handling, and data flow.
+"""
+
+import os
+import tempfile
+from pathlib import Path
+from unittest.mock import AsyncMock, MagicMock, patch
+
+import pytest
+
+from reflector.processors.file_diarization import (
+    FileDiarizationInput,
+    FileDiarizationOutput,
+)
+from reflector.processors.types import (
+    AudioDiarizationInput,
+    TitleSummaryWithId,
+    Transcript,
+    Word,
+)
+
+# ── Registration Tests ──────────────────────────────────────────────────
+
+
+def test_audio_diarization_pyannote_registers():
+    """Verify AudioDiarizationPyannoteProcessor registers with 'pyannote' backend."""
+    # Importing the module triggers registration
+    import reflector.processors.audio_diarization_pyannote  # noqa: F401
+    from reflector.processors.audio_diarization_auto import (
+        AudioDiarizationAutoProcessor,
+    )
+
+    assert "pyannote" in AudioDiarizationAutoProcessor._registry
+
+
+def test_file_diarization_pyannote_registers():
+    """Verify FileDiarizationPyannoteProcessor registers with 'pyannote' backend."""
+    import reflector.processors.file_diarization_pyannote  # noqa: F401
+    from reflector.processors.file_diarization_auto import FileDiarizationAutoProcessor
+
+    assert "pyannote" in FileDiarizationAutoProcessor._registry
+
+
+def test_transcript_translator_marian_registers():
+    """Verify TranscriptTranslatorMarianProcessor registers with 'marian' backend."""
+    import reflector.processors.transcript_translator_marian  # noqa: F401
+    from reflector.processors.transcript_translator_auto import (
+        TranscriptTranslatorAutoProcessor,
+    )
+
+    assert "marian" in TranscriptTranslatorAutoProcessor._registry
+
+
+def test_file_transcript_whisper_registers():
+    """Verify FileTranscriptWhisperProcessor registers with 'whisper' backend."""
+    import reflector.processors.file_transcript_whisper  # noqa: F401
+    from reflector.processors.file_transcript_auto import FileTranscriptAutoProcessor
+
+    assert "whisper" in FileTranscriptAutoProcessor._registry
+
+
+# ── Audio Download Utility Tests ────────────────────────────────────────
+
+
+@pytest.mark.asyncio
+async def test_download_audio_to_temp_success():
+    """Verify download_audio_to_temp downloads to a temp file and returns path."""
+    from reflector.processors._audio_download import download_audio_to_temp
+
+    mock_response = MagicMock()
+    mock_response.status_code = 200
+    mock_response.headers = {"content-type": "audio/wav"}
+    mock_response.iter_content.return_value = [b"fake audio data"]
+    mock_response.raise_for_status = MagicMock()
+
+    with patch("reflector.processors._audio_download.requests.get") as mock_get:
+        mock_get.return_value = mock_response
+
+        result = await download_audio_to_temp("https://example.com/test.wav")
+
+        assert isinstance(result, Path)
+        assert result.exists()
+        assert result.read_bytes() == b"fake audio data"
+        assert result.suffix == ".wav"
+
+        # Cleanup
+        os.unlink(result)
+
+
+@pytest.mark.asyncio
+async def test_download_audio_to_temp_cleanup_on_error():
+    """Verify temp file is cleaned up when download fails mid-write."""
+    from reflector.processors._audio_download import download_audio_to_temp
+
+    mock_response = MagicMock()
+    mock_response.headers = {"content-type": "audio/wav"}
+    mock_response.raise_for_status = MagicMock()
+
+    def fail_iter(*args, **kwargs):
+        raise ConnectionError("Download interrupted")
+
+    mock_response.iter_content = fail_iter
+
+    with patch("reflector.processors._audio_download.requests.get") as mock_get:
+        mock_get.return_value = mock_response
+
+        with pytest.raises(ConnectionError, match="Download interrupted"):
+            await download_audio_to_temp("https://example.com/test.wav")
+
+
+def test_detect_extension_from_url():
+    """Verify extension detection from URL path."""
+    from reflector.processors._audio_download import _detect_extension
+
+    assert _detect_extension("https://example.com/test.wav", "") == ".wav"
+    assert _detect_extension("https://example.com/test.mp3?signed=1", "") == ".mp3"
+    assert _detect_extension("https://example.com/test.webm", "") == ".webm"
+
+
+def test_detect_extension_from_content_type():
+    """Verify extension detection from content-type header."""
+    from reflector.processors._audio_download import _detect_extension
+
+    assert _detect_extension("https://s3.aws/uuid", "audio/mpeg") == ".mp3"
+    assert _detect_extension("https://s3.aws/uuid", "audio/wav") == ".wav"
+    assert _detect_extension("https://s3.aws/uuid", "audio/webm") == ".webm"
+
+
+def test_detect_extension_fallback():
+    """Verify fallback extension when neither URL nor content-type is recognized."""
+    from reflector.processors._audio_download import _detect_extension
+
+    assert (
+        _detect_extension("https://s3.aws/uuid", "application/octet-stream") == ".audio"
+    )
+
+
+# ── Audio Diarization Pyannote Processor Tests ──────────────────────────
+
+
+@pytest.mark.asyncio
+async def test_audio_diarization_pyannote_diarize():
+    """Verify pyannote audio diarization downloads, diarizes, and cleans up."""
+    from reflector.processors.audio_diarization_pyannote import (
+        AudioDiarizationPyannoteProcessor,
+    )
+
+    mock_diarization_result = {
+        "diarization": [
+            {"start": 0.0, "end": 2.5, "speaker": 0},
+            {"start": 2.5, "end": 5.0, "speaker": 1},
+        ]
+    }
+
+    # Create a temp file to simulate download
+    tmp = tempfile.NamedTemporaryFile(suffix=".wav", delete=False)
+    tmp.write(b"fake audio")
+    tmp.close()
+    tmp_path = Path(tmp.name)
+
+    processor = AudioDiarizationPyannoteProcessor()
+
+    with (
+        patch(
+            "reflector.processors.audio_diarization_pyannote.download_audio_to_temp",
+            new_callable=AsyncMock,
+            return_value=tmp_path,
+        ),
+        patch(
+            "reflector.processors.audio_diarization_pyannote.diarization_service"
+        ) as mock_svc,
+    ):
+        mock_svc.diarize_file.return_value = mock_diarization_result
+
+        data = AudioDiarizationInput(
+            audio_url="https://example.com/test.wav",
+            topics=[
+                TitleSummaryWithId(
+                    id="topic-1",
+                    title="Test Topic",
+                    summary="A test topic",
+                    timestamp=0.0,
+                    duration=5.0,
+                    transcript=Transcript(
+                        words=[Word(text="hello", start=0.0, end=1.0)]
+                    ),
+                )
+            ],
+        )
+        result = await processor._diarize(data)
+
+        assert result == mock_diarization_result["diarization"]
+        mock_svc.diarize_file.assert_called_once()
+
+
+# ── File Diarization Pyannote Processor Tests ───────────────────────────
+
+
+@pytest.mark.asyncio
+async def test_file_diarization_pyannote_diarize():
+    """Verify pyannote file diarization returns FileDiarizationOutput."""
+    from reflector.processors.file_diarization_pyannote import (
+        FileDiarizationPyannoteProcessor,
+    )
+
+    mock_diarization_result = {
+        "diarization": [
+            {"start": 0.0, "end": 3.0, "speaker": 0},
+            {"start": 3.0, "end": 6.0, "speaker": 1},
+        ]
+    }
+
+    tmp = tempfile.NamedTemporaryFile(suffix=".wav", delete=False)
+    tmp.write(b"fake audio")
+    tmp.close()
+    tmp_path = Path(tmp.name)
+
+    processor = FileDiarizationPyannoteProcessor()
+
+    with (
+        patch(
+            "reflector.processors.file_diarization_pyannote.download_audio_to_temp",
+            new_callable=AsyncMock,
+            return_value=tmp_path,
+        ),
+        patch(
+            "reflector.processors.file_diarization_pyannote.diarization_service"
+        ) as mock_svc,
+    ):
+        mock_svc.diarize_file.return_value = mock_diarization_result
+
+        data = FileDiarizationInput(audio_url="https://example.com/test.wav")
+        result = await processor._diarize(data)
+
+        assert isinstance(result, FileDiarizationOutput)
+        assert len(result.diarization) == 2
+        assert result.diarization[0]["start"] == 0.0
+        assert result.diarization[1]["speaker"] == 1
+
+
+# ── Transcript Translator Marian Processor Tests ───────────────────────
+
+
+@pytest.mark.asyncio
+async def test_transcript_translator_marian_translate():
+    """Verify MarianMT translator calls service and extracts translation."""
+    from reflector.processors.transcript_translator_marian import (
+        TranscriptTranslatorMarianProcessor,
+    )
+
+    mock_result = {"text": {"en": "Hello world", "fr": "Bonjour le monde"}}
+
+    processor = TranscriptTranslatorMarianProcessor()
+
+    def fake_get_pref(key, default=None):
+        prefs = {"audio:source_language": "en", "audio:target_language": "fr"}
+        return prefs.get(key, default)
+
+    with (
+        patch.object(processor, "get_pref", side_effect=fake_get_pref),
+        patch(
+            "reflector.processors.transcript_translator_marian.translator_service"
+        ) as mock_svc,
+    ):
+        mock_svc.translate.return_value = mock_result
+
+        result = await processor._translate("Hello world")
+
+        assert result == "Bonjour le monde"
+        mock_svc.translate.assert_called_once_with("Hello world", "en", "fr")
+
+
+@pytest.mark.asyncio
+async def test_transcript_translator_marian_no_translation():
+    """Verify translator returns None when target language not in result."""
+    from reflector.processors.transcript_translator_marian import (
+        TranscriptTranslatorMarianProcessor,
+    )
+
+    mock_result = {"text": {"en": "Hello world"}}
+
+    processor = TranscriptTranslatorMarianProcessor()
+
+    def fake_get_pref(key, default=None):
+        prefs = {"audio:source_language": "en", "audio:target_language": "fr"}
+        return prefs.get(key, default)
+
+    with (
+        patch.object(processor, "get_pref", side_effect=fake_get_pref),
+        patch(
+            "reflector.processors.transcript_translator_marian.translator_service"
+        ) as mock_svc,
+    ):
+        mock_svc.translate.return_value = mock_result
+
+        result = await processor._translate("Hello world")
+
+        assert result is None
+
+
+# ── File Transcript Whisper Processor Tests ─────────────────────────────
+
+
+@pytest.mark.asyncio
+async def test_file_transcript_whisper_transcript():
+    """Verify whisper file processor downloads, transcribes, and returns Transcript."""
+    from reflector.processors.file_transcript import FileTranscriptInput
+    from reflector.processors.file_transcript_whisper import (
+        FileTranscriptWhisperProcessor,
+    )
+
+    tmp = tempfile.NamedTemporaryFile(suffix=".wav", delete=False)
+    tmp.write(b"fake audio")
+    tmp.close()
+    tmp_path = Path(tmp.name)
+
+    processor = FileTranscriptWhisperProcessor()
+
+    # Mock the blocking transcription method
+    mock_transcript = Transcript(
+        words=[
+            Word(text="Hello", start=0.0, end=0.5),
+            Word(text=" world", start=0.5, end=1.0),
+        ]
+    )
+
+    with (
+        patch(
+            "reflector.processors.file_transcript_whisper.download_audio_to_temp",
+            new_callable=AsyncMock,
+            return_value=tmp_path,
+        ),
+        patch.object(
+            processor,
+            "_transcribe_file_blocking",
+            return_value=mock_transcript,
+        ),
+    ):
+        data = FileTranscriptInput(
+            audio_url="https://example.com/test.wav", language="en"
+        )
+        result = await processor._transcript(data)
+
+        assert isinstance(result, Transcript)
+        assert len(result.words) == 2
+        assert result.words[0].text == "Hello"
+
+
+# ── VAD Helper Tests ────────────────────────────────────────────────────
+
+
+def test_enforce_word_timing_constraints():
+    """Verify word timing enforcement prevents overlapping times."""
+    from reflector.processors.file_transcript_whisper import (
+        _enforce_word_timing_constraints,
+    )
+
+    words = [
+        {"word": "hello", "start": 0.0, "end": 1.5},
+        {"word": "world", "start": 1.0, "end": 2.0},  # overlaps with previous
+        {"word": "test", "start": 2.0, "end": 3.0},
+    ]
+
+    result = _enforce_word_timing_constraints(words)
+
+    assert result[0]["end"] == 1.0  # Clamped to next word's start
+    assert result[1]["end"] == 2.0  # Clamped to next word's start
+    assert result[2]["end"] == 3.0  # Last word unchanged
+
+
+def test_enforce_word_timing_constraints_empty():
+    """Verify timing enforcement handles empty and single-word lists."""
+    from reflector.processors.file_transcript_whisper import (
+        _enforce_word_timing_constraints,
+    )
+
+    assert _enforce_word_timing_constraints([]) == []
+    assert _enforce_word_timing_constraints([{"word": "a", "start": 0, "end": 1}]) == [
+        {"word": "a", "start": 0, "end": 1}
+    ]
+
+
+def test_pad_audio_short():
+    """Verify short audio gets padded with silence."""
+    import numpy as np
+
+    from reflector.processors.file_transcript_whisper import _pad_audio
+
+    short_audio = np.zeros(100, dtype=np.float32)  # Very short
+    result = _pad_audio(short_audio, sample_rate=16000)
+
+    # Should be padded to at least silence_padding duration
+    assert len(result) > len(short_audio)
+
+
+def test_pad_audio_long():
+    """Verify long audio is not padded."""
+    import numpy as np
+
+    from reflector.processors.file_transcript_whisper import _pad_audio
+
+    long_audio = np.zeros(32000, dtype=np.float32)  # 2 seconds
+    result = _pad_audio(long_audio, sample_rate=16000)
+
+    assert len(result) == len(long_audio)
+
+
+# ── Translator Service Tests ────────────────────────────────────────────
+
+
+def test_translator_service_resolve_model():
+    """Verify model resolution for known and unknown language pairs."""
+    from reflector.processors._marian_translator_service import MarianTranslatorService
+
+    svc = MarianTranslatorService()
+
+    assert svc._resolve_model_name("en", "fr") == "Helsinki-NLP/opus-mt-en-fr"
+    assert svc._resolve_model_name("es", "en") == "Helsinki-NLP/opus-mt-es-en"
+    assert svc._resolve_model_name("en", "de") == "Helsinki-NLP/opus-mt-en-de"
+    # Unknown pair falls back to en->fr
+    assert svc._resolve_model_name("ja", "ko") == "Helsinki-NLP/opus-mt-en-fr"
+
+
+# ── Diarization Service Tests ───────────────────────────────────────────
+
+
+def test_diarization_service_singleton():
+    """Verify diarization_service is a module-level singleton."""
+    from reflector.processors._pyannote_diarization_service import (
+        PyannoteDiarizationService,
+        diarization_service,
+    )
+
+    assert isinstance(diarization_service, PyannoteDiarizationService)
+    assert diarization_service._pipeline is None  # Not loaded until first use
+
+
+def test_translator_service_singleton():
+    """Verify translator_service is a module-level singleton."""
+    from reflector.processors._marian_translator_service import (
+        MarianTranslatorService,
+        translator_service,
+    )
+
+    assert isinstance(translator_service, MarianTranslatorService)
+    assert translator_service._pipeline is None  # Not loaded until first use

server/tests/test_transcripts_audio_token_auth.py

@@ -0,0 +1,327 @@
+"""Tests for audio mp3 endpoint token query-param authentication.
+
+Covers both password (HS256) and JWT/Authentik (RS256) auth backends,
+verifying that private transcripts can be accessed via ?token= query param.
+"""
+
+import shutil
+from datetime import datetime, timedelta, timezone
+from pathlib import Path
+from unittest.mock import patch
+
+import jwt
+import pytest
+from cryptography.hazmat.primitives import serialization
+from cryptography.hazmat.primitives.asymmetric import rsa
+
+# ---------------------------------------------------------------------------
+# Helpers
+# ---------------------------------------------------------------------------
+
+OWNER_USER_ID = "test-owner-user-id"
+
+
+def _create_hs256_token(user_id: str, secret: str, expired: bool = False) -> str:
+    """Create an HS256 JWT like the password auth backend does."""
+    delta = timedelta(minutes=-5) if expired else timedelta(hours=24)
+    payload = {
+        "sub": user_id,
+        "email": "test@example.com",
+        "exp": datetime.now(timezone.utc) + delta,
+    }
+    return jwt.encode(payload, secret, algorithm="HS256")
+
+
+def _generate_rsa_keypair():
+    """Generate a fresh RSA keypair for tests."""
+    private_key = rsa.generate_private_key(public_exponent=65537, key_size=2048)
+    public_pem = private_key.public_key().public_bytes(
+        serialization.Encoding.PEM, serialization.PublicFormat.SubjectPublicKeyInfo
+    )
+    return private_key, public_pem.decode()
+
+
+def _create_rs256_token(
+    authentik_uid: str,
+    private_key,
+    audience: str,
+    expired: bool = False,
+) -> str:
+    """Create an RS256 JWT like Authentik would issue."""
+    delta = timedelta(minutes=-5) if expired else timedelta(hours=1)
+    payload = {
+        "sub": authentik_uid,
+        "email": "authentik-user@example.com",
+        "aud": audience,
+        "exp": datetime.now(timezone.utc) + delta,
+    }
+    return jwt.encode(payload, private_key, algorithm="RS256")
+
+
+# ---------------------------------------------------------------------------
+# Fixtures
+# ---------------------------------------------------------------------------
+
+
+@pytest.fixture
+async def private_transcript(tmpdir):
+    """Create a private transcript owned by OWNER_USER_ID with an mp3 file.
+
+    Created directly via the controller (not HTTP) so no auth override
+    leaks into the test scope.
+    """
+    from reflector.db.transcripts import SourceKind, transcripts_controller
+    from reflector.settings import settings
+
+    settings.DATA_DIR = Path(tmpdir)
+
+    transcript = await transcripts_controller.add(
+        "Private audio test",
+        source_kind=SourceKind.FILE,
+        user_id=OWNER_USER_ID,
+        share_mode="private",
+    )
+    await transcripts_controller.update(transcript, {"status": "ended"})
+
+    # Copy a real mp3 to the expected location
+    audio_filename = transcript.audio_mp3_filename
+    mp3_source = Path(__file__).parent / "records" / "test_mathieu_hello.mp3"
+    audio_filename.parent.mkdir(parents=True, exist_ok=True)
+    shutil.copy(mp3_source, audio_filename)
+
+    yield transcript
+
+
+# ---------------------------------------------------------------------------
+# Core access control tests
+# ---------------------------------------------------------------------------
+
+
+@pytest.mark.asyncio
+async def test_audio_mp3_private_no_auth_returns_403(private_transcript, client):
+    """Without auth, accessing a private transcript's audio returns 403."""
+    response = await client.get(f"/transcripts/{private_transcript.id}/audio/mp3")
+    assert response.status_code == 403
+
+
+@pytest.mark.asyncio
+async def test_audio_mp3_with_bearer_header(private_transcript, client):
+    """Owner accessing audio via Authorization header works."""
+    from reflector.app import app
+    from reflector.auth import current_user_optional
+
+    # Temporarily override to simulate the owner being authenticated
+    app.dependency_overrides[current_user_optional] = lambda: {
+        "sub": OWNER_USER_ID,
+        "email": "test@example.com",
+    }
+    try:
+        response = await client.get(f"/transcripts/{private_transcript.id}/audio/mp3")
+    finally:
+        del app.dependency_overrides[current_user_optional]
+
+    assert response.status_code == 200
+    assert response.headers["content-type"] == "audio/mpeg"
+
+
+@pytest.mark.asyncio
+async def test_audio_mp3_public_transcript_no_auth_ok(tmpdir, client):
+    """Public transcripts are accessible without any auth."""
+    from reflector.db.transcripts import SourceKind, transcripts_controller
+    from reflector.settings import settings
+
+    settings.DATA_DIR = Path(tmpdir)
+
+    transcript = await transcripts_controller.add(
+        "Public audio test",
+        source_kind=SourceKind.FILE,
+        user_id=OWNER_USER_ID,
+        share_mode="public",
+    )
+    await transcripts_controller.update(transcript, {"status": "ended"})
+
+    audio_filename = transcript.audio_mp3_filename
+    mp3_source = Path(__file__).parent / "records" / "test_mathieu_hello.mp3"
+    audio_filename.parent.mkdir(parents=True, exist_ok=True)
+    shutil.copy(mp3_source, audio_filename)
+
+    response = await client.get(f"/transcripts/{transcript.id}/audio/mp3")
+    assert response.status_code == 200
+    assert response.headers["content-type"] == "audio/mpeg"
+
+
+# ---------------------------------------------------------------------------
+# Password auth backend tests (?token= with HS256)
+# ---------------------------------------------------------------------------
+
+
+@pytest.mark.asyncio
+async def test_audio_mp3_password_token_query_param(private_transcript, client):
+    """Password backend: valid HS256 ?token= grants access to private audio."""
+    from reflector.auth.auth_password import UserInfo
+    from reflector.settings import settings
+
+    token = _create_hs256_token(OWNER_USER_ID, settings.SECRET_KEY)
+
+    with patch("reflector.auth.verify_raw_token") as mock_verify:
+        mock_verify.return_value = UserInfo(sub=OWNER_USER_ID, email="test@example.com")
+        response = await client.get(
+            f"/transcripts/{private_transcript.id}/audio/mp3?token={token}"
+        )
+
+    assert response.status_code == 200
+    assert response.headers["content-type"] == "audio/mpeg"
+
+
+@pytest.mark.asyncio
+async def test_audio_mp3_password_expired_token_returns_401(private_transcript, client):
+    """Password backend: expired HS256 ?token= returns 401."""
+    from reflector.settings import settings
+
+    expired_token = _create_hs256_token(
+        OWNER_USER_ID, settings.SECRET_KEY, expired=True
+    )
+
+    with patch("reflector.auth.verify_raw_token") as mock_verify:
+        mock_verify.side_effect = jwt.ExpiredSignatureError("token expired")
+        response = await client.get(
+            f"/transcripts/{private_transcript.id}/audio/mp3" f"?token={expired_token}"
+        )
+
+    assert response.status_code == 401
+
+
+@pytest.mark.asyncio
+async def test_audio_mp3_password_wrong_user_returns_403(private_transcript, client):
+    """Password backend: valid token for a different user returns 403."""
+    from reflector.auth.auth_password import UserInfo
+    from reflector.settings import settings
+
+    token = _create_hs256_token("other-user-id", settings.SECRET_KEY)
+
+    with patch("reflector.auth.verify_raw_token") as mock_verify:
+        mock_verify.return_value = UserInfo(
+            sub="other-user-id", email="other@example.com"
+        )
+        response = await client.get(
+            f"/transcripts/{private_transcript.id}/audio/mp3?token={token}"
+        )
+
+    assert response.status_code == 403
+
+
+@pytest.mark.asyncio
+async def test_audio_mp3_invalid_token_returns_401(private_transcript, client):
+    """Garbage token string returns 401."""
+    with patch("reflector.auth.verify_raw_token") as mock_verify:
+        mock_verify.return_value = None
+        response = await client.get(
+            f"/transcripts/{private_transcript.id}/audio/mp3" "?token=not-a-real-token"
+        )
+
+    assert response.status_code == 401
+
+
+# ---------------------------------------------------------------------------
+# JWT/Authentik auth backend tests (?token= with RS256)
+# ---------------------------------------------------------------------------
+
+
+@pytest.mark.asyncio
+async def test_audio_mp3_authentik_token_query_param(private_transcript, client):
+    """Authentik backend: valid RS256 ?token= grants access to private audio."""
+    from reflector.auth.auth_password import UserInfo
+
+    private_key, _ = _generate_rsa_keypair()
+    token = _create_rs256_token("authentik-abc123", private_key, "test-audience")
+
+    with patch("reflector.auth.verify_raw_token") as mock_verify:
+        # Authentik flow maps authentik_uid -> internal user id
+        mock_verify.return_value = UserInfo(
+            sub=OWNER_USER_ID, email="authentik-user@example.com"
+        )
+        response = await client.get(
+            f"/transcripts/{private_transcript.id}/audio/mp3?token={token}"
+        )
+
+    assert response.status_code == 200
+    assert response.headers["content-type"] == "audio/mpeg"
+
+
+@pytest.mark.asyncio
+async def test_audio_mp3_authentik_expired_token_returns_401(
+    private_transcript, client
+):
+    """Authentik backend: expired RS256 ?token= returns 401."""
+    private_key, _ = _generate_rsa_keypair()
+    expired_token = _create_rs256_token(
+        "authentik-abc123", private_key, "test-audience", expired=True
+    )
+
+    with patch("reflector.auth.verify_raw_token") as mock_verify:
+        mock_verify.side_effect = jwt.ExpiredSignatureError("token expired")
+        response = await client.get(
+            f"/transcripts/{private_transcript.id}/audio/mp3" f"?token={expired_token}"
+        )
+
+    assert response.status_code == 401
+
+
+@pytest.mark.asyncio
+async def test_audio_mp3_authentik_wrong_user_returns_403(private_transcript, client):
+    """Authentik backend: valid RS256 token for different user returns 403."""
+    from reflector.auth.auth_password import UserInfo
+
+    private_key, _ = _generate_rsa_keypair()
+    token = _create_rs256_token("authentik-other", private_key, "test-audience")
+
+    with patch("reflector.auth.verify_raw_token") as mock_verify:
+        mock_verify.return_value = UserInfo(
+            sub="different-user-id", email="other@example.com"
+        )
+        response = await client.get(
+            f"/transcripts/{private_transcript.id}/audio/mp3?token={token}"
+        )
+
+    assert response.status_code == 403
+
+
+# ---------------------------------------------------------------------------
+# _generate_local_audio_link produces HS256 tokens — must be verifiable
+# by any auth backend
+# ---------------------------------------------------------------------------
+
+
+@pytest.mark.asyncio
+async def test_local_audio_link_token_works_with_authentik_backend(
+    private_transcript, client
+):
+    """_generate_local_audio_link creates an HS256 token via create_access_token.
+
+    When the Authentik (RS256) auth backend is active, verify_raw_token uses
+    JWTAuth which expects RS256 + public key. The HS256 token created by
+    _generate_local_audio_link will fail verification, returning 401.
+
+    This test documents the bug: the internal audio URL generated for the
+    diarization pipeline is unusable under the JWT auth backend.
+    """
+    from urllib.parse import parse_qs, urlparse
+
+    # Generate the internal audio link (uses create_access_token → HS256)
+    url = private_transcript._generate_local_audio_link()
+    parsed = urlparse(url)
+    token = parse_qs(parsed.query)["token"][0]
+
+    # Simulate what happens when the JWT/Authentik backend tries to verify
+    # this HS256 token: JWTAuth.verify_token expects RS256, so it raises.
+    with patch("reflector.auth.verify_raw_token") as mock_verify:
+        mock_verify.side_effect = jwt.exceptions.InvalidAlgorithmError(
+            "the specified alg value is not allowed"
+        )
+        response = await client.get(
+            f"/transcripts/{private_transcript.id}/audio/mp3?token={token}"
+        )
+
+    # BUG: this should be 200 (the token was created by our own server),
+    # but the Authentik backend rejects it because it's HS256, not RS256.
+    assert response.status_code == 200