Add Modal backend for audio mixdown

Co-authored-by: Igor Loskutov <igor.loskutoff@gmail.com>

CHANGELOG.md

@@ -1,5 +1,12 @@
 # Changelog
 
+## [0.28.0](https://github.com/Monadical-SAS/reflector/compare/v0.27.0...v0.28.0) (2026-01-20)
+
+
+### Features
+
+* worker affinity ([#819](https://github.com/Monadical-SAS/reflector/issues/819)) ([3b6540e](https://github.com/Monadical-SAS/reflector/commit/3b6540eae5b597449f98661bdf15483b77be3268))
+
 ## [0.27.0](https://github.com/Monadical-SAS/reflector/compare/v0.26.0...v0.27.0) (2025-12-26)
 
 

docker-compose.yml

@@ -34,7 +34,7 @@ services:
     environment:
       ENTRYPOINT: beat
 
-  hatchet-worker:
+  hatchet-worker-cpu:
     build:
       context: server
     volumes:
@@ -43,7 +43,20 @@ services:
     env_file:
       - ./server/.env
     environment:
-      ENTRYPOINT: hatchet-worker
+      ENTRYPOINT: hatchet-worker-cpu
+    depends_on:
+      hatchet:
+        condition: service_healthy
+  hatchet-worker-llm:
+    build:
+      context: server
+    volumes:
+      - ./server/:/app/
+      - /app/.venv
+    env_file:
+      - ./server/.env
+    environment:
+      ENTRYPOINT: hatchet-worker-llm
     depends_on:
       hatchet:
         condition: service_healthy

gpu/modal_deployments/deploy-all.sh

@@ -131,6 +131,15 @@ if [ -z "$DIARIZER_URL" ]; then
 fi
 echo "  -> $DIARIZER_URL"
 
+echo ""
+echo "Deploying mixdown (CPU audio processing)..."
+MIXDOWN_URL=$(modal deploy reflector_mixdown.py 2>&1 | grep -o 'https://[^ ]*web.modal.run' | head -1)
+if [ -z "$MIXDOWN_URL" ]; then
+    echo "Error: Failed to deploy mixdown. Check Modal dashboard for details."
+    exit 1
+fi
+echo "  -> $MIXDOWN_URL"
+
 # --- Output Configuration ---
 echo ""
 echo "=========================================="
@@ -147,4 +156,8 @@ echo ""
 echo "DIARIZATION_BACKEND=modal"
 echo "DIARIZATION_URL=$DIARIZER_URL"
 echo "DIARIZATION_MODAL_API_KEY=$API_KEY"
+echo ""
+echo "MIXDOWN_BACKEND=modal"
+echo "MIXDOWN_URL=$MIXDOWN_URL"
+echo "MIXDOWN_MODAL_API_KEY=$API_KEY"
 echo "# --- End Modal Configuration ---"

gpu/modal_deployments/reflector_mixdown.py

@@ -0,0 +1,379 @@
+"""
+Reflector GPU backend - audio mixdown
+======================================
+
+CPU-intensive audio mixdown service for combining multiple audio tracks.
+Uses PyAV filter graph (amix) for high-quality audio mixing.
+"""
+
+import os
+import tempfile
+import time
+from fractions import Fraction
+
+import modal
+
+MIXDOWN_TIMEOUT = 900  # 15 minutes
+SCALEDOWN_WINDOW = 60  # 1 minute idle before shutdown
+
+app = modal.App("reflector-mixdown")
+
+# CPU-based image (no GPU needed for audio processing)
+image = (
+    modal.Image.debian_slim(python_version="3.12")
+    .apt_install("ffmpeg")  # Required by PyAV
+    .pip_install(
+        "av==13.1.0",  # PyAV for audio processing
+        "requests==2.32.3",  # HTTP for presigned URL downloads/uploads
+        "fastapi==0.115.12",  # API framework
+    )
+)
+
+
+@app.function(
+    cpu=4.0,  # 4 CPU cores for audio processing
+    timeout=MIXDOWN_TIMEOUT,
+    scaledown_window=SCALEDOWN_WINDOW,
+    secrets=[modal.Secret.from_name("reflector-gpu")],
+    image=image,
+)
+@modal.concurrent(max_inputs=10)
+@modal.asgi_app()
+def web():
+    import logging
+    import secrets
+    import shutil
+
+    import av
+    import requests
+    from av.audio.resampler import AudioResampler
+    from fastapi import Depends, FastAPI, HTTPException, status
+    from fastapi.security import OAuth2PasswordBearer
+    from pydantic import BaseModel
+
+    # Setup logging
+    logging.basicConfig(
+        level=logging.INFO, format="%(asctime)s - %(levelname)s - %(message)s"
+    )
+    logger = logging.getLogger(__name__)
+
+    app = FastAPI()
+    oauth2_scheme = OAuth2PasswordBearer(tokenUrl="token")
+
+    # Validate API key exists at startup
+    API_KEY = os.environ.get("REFLECTOR_GPU_APIKEY")
+    if not API_KEY:
+        raise RuntimeError("REFLECTOR_GPU_APIKEY not configured in Modal secrets")
+
+    def apikey_auth(apikey: str = Depends(oauth2_scheme)):
+        # Use constant-time comparison to prevent timing attacks
+        if secrets.compare_digest(apikey, API_KEY):
+            return
+        raise HTTPException(
+            status_code=status.HTTP_401_UNAUTHORIZED,
+            detail="Invalid API key",
+            headers={"WWW-Authenticate": "Bearer"},
+        )
+
+    class MixdownRequest(BaseModel):
+        track_urls: list[str]
+        output_url: str
+        target_sample_rate: int = 48000
+        expected_duration_sec: float | None = None
+
+    class MixdownResponse(BaseModel):
+        duration_ms: float
+        tracks_mixed: int
+        audio_uploaded: bool
+
+    def download_track(url: str, temp_dir: str, index: int) -> str:
+        """Download track from presigned URL to temp file using streaming."""
+        logger.info(f"Downloading track {index + 1}")
+        response = requests.get(url, stream=True, timeout=300)
+
+        if response.status_code == 404:
+            raise HTTPException(status_code=404, detail=f"Track {index} not found")
+        if response.status_code == 403:
+            raise HTTPException(
+                status_code=403, detail=f"Track {index} presigned URL expired"
+            )
+
+        response.raise_for_status()
+
+        temp_path = os.path.join(temp_dir, f"track_{index}.webm")
+        total_bytes = 0
+        with open(temp_path, "wb") as f:
+            for chunk in response.iter_content(chunk_size=8192):
+                if chunk:
+                    f.write(chunk)
+                    total_bytes += len(chunk)
+
+        logger.info(f"Track {index + 1} downloaded: {total_bytes} bytes")
+        return temp_path
+
+    def mixdown_tracks_modal(
+        track_paths: list[str],
+        output_path: str,
+        target_sample_rate: int,
+        expected_duration_sec: float | None,
+        logger,
+    ) -> float:
+        """Mix multiple audio tracks using PyAV filter graph.
+
+        Args:
+            track_paths: List of local file paths to audio tracks
+            output_path: Local path for output MP3 file
+            target_sample_rate: Sample rate for output (Hz)
+            expected_duration_sec: Optional fallback duration if container metadata unavailable
+            logger: Logger instance for progress tracking
+
+        Returns:
+            Duration in milliseconds
+        """
+        logger.info(f"Starting mixdown of {len(track_paths)} tracks")
+
+        # Build PyAV filter graph: N abuffer -> amix -> aformat -> sink
+        graph = av.filter.Graph()
+        inputs = []
+
+        for idx in range(len(track_paths)):
+            args = (
+                f"time_base=1/{target_sample_rate}:"
+                f"sample_rate={target_sample_rate}:"
+                f"sample_fmt=s32:"
+                f"channel_layout=stereo"
+            )
+            in_ctx = graph.add("abuffer", args=args, name=f"in{idx}")
+            inputs.append(in_ctx)
+
+        mixer = graph.add("amix", args=f"inputs={len(inputs)}:normalize=0", name="mix")
+        fmt = graph.add(
+            "aformat",
+            args=f"sample_fmts=s32:channel_layouts=stereo:sample_rates={target_sample_rate}",
+            name="fmt",
+        )
+        sink = graph.add("abuffersink", name="out")
+
+        # Connect inputs to mixer (no delays for Modal implementation)
+        for idx, in_ctx in enumerate(inputs):
+            in_ctx.link_to(mixer, 0, idx)
+
+        mixer.link_to(fmt)
+        fmt.link_to(sink)
+        graph.configure()
+
+        # Open all containers
+        containers = []
+        try:
+            for i, path in enumerate(track_paths):
+                try:
+                    c = av.open(path)
+                    containers.append(c)
+                except Exception as e:
+                    logger.warning(
+                        f"Failed to open container {i}: {e}",
+                    )
+
+            if not containers:
+                raise ValueError("Could not open any track containers")
+
+            # Calculate total duration for progress reporting
+            max_duration_sec = 0.0
+            for c in containers:
+                if c.duration is not None:
+                    dur_sec = c.duration / av.time_base
+                    max_duration_sec = max(max_duration_sec, dur_sec)
+            if max_duration_sec == 0.0 and expected_duration_sec:
+                max_duration_sec = expected_duration_sec
+
+            # Setup output container
+            out_container = av.open(output_path, "w", format="mp3")
+            out_stream = out_container.add_stream("libmp3lame", rate=target_sample_rate)
+
+            decoders = [c.decode(audio=0) for c in containers]
+            active = [True] * len(decoders)
+            resamplers = [
+                AudioResampler(format="s32", layout="stereo", rate=target_sample_rate)
+                for _ in decoders
+            ]
+
+            current_max_time = 0.0
+            last_log_time = time.monotonic()
+            start_time = time.monotonic()
+
+            total_duration = 0
+
+            while any(active):
+                for i, (dec, is_active) in enumerate(zip(decoders, active)):
+                    if not is_active:
+                        continue
+                    try:
+                        frame = next(dec)
+                    except StopIteration:
+                        active[i] = False
+                        inputs[i].push(None)  # Signal end of stream
+                        continue
+
+                    if frame.sample_rate != target_sample_rate:
+                        continue
+
+                    # Progress logging (every 5 seconds)
+                    if frame.time is not None:
+                        current_max_time = max(current_max_time, frame.time)
+                        now = time.monotonic()
+                        if now - last_log_time >= 5.0:
+                            elapsed = now - start_time
+                            if max_duration_sec > 0:
+                                progress_pct = min(
+                                    100.0, (current_max_time / max_duration_sec) * 100
+                                )
+                                logger.info(
+                                    f"Mixdown progress: {progress_pct:.1f}% @ {current_max_time:.1f}s (elapsed: {elapsed:.1f}s)"
+                                )
+                            else:
+                                logger.info(
+                                    f"Mixdown progress: @ {current_max_time:.1f}s (elapsed: {elapsed:.1f}s)"
+                                )
+                            last_log_time = now
+
+                    out_frames = resamplers[i].resample(frame) or []
+                    for rf in out_frames:
+                        rf.sample_rate = target_sample_rate
+                        rf.time_base = Fraction(1, target_sample_rate)
+                        inputs[i].push(rf)
+
+                    # Pull mixed frames from sink and encode
+                    while True:
+                        try:
+                            mixed = sink.pull()
+                        except Exception:
+                            break
+                        mixed.sample_rate = target_sample_rate
+                        mixed.time_base = Fraction(1, target_sample_rate)
+
+                        # Encode and mux
+                        for packet in out_stream.encode(mixed):
+                            out_container.mux(packet)
+                            total_duration += packet.duration
+
+            # Flush remaining frames from filter graph
+            while True:
+                try:
+                    mixed = sink.pull()
+                except Exception:
+                    break
+                mixed.sample_rate = target_sample_rate
+                mixed.time_base = Fraction(1, target_sample_rate)
+
+                for packet in out_stream.encode(mixed):
+                    out_container.mux(packet)
+                    total_duration += packet.duration
+
+            # Flush encoder
+            for packet in out_stream.encode():
+                out_container.mux(packet)
+                total_duration += packet.duration
+
+            # Calculate duration in milliseconds
+            if total_duration > 0:
+                # Use the same calculation as AudioFileWriterProcessor
+                duration_ms = round(
+                    float(total_duration * out_stream.time_base * 1000), 2
+                )
+            else:
+                duration_ms = 0.0
+
+            out_container.close()
+            logger.info(f"Mixdown complete: duration={duration_ms}ms")
+
+        finally:
+            # Cleanup all containers
+            for c in containers:
+                if c is not None:
+                    try:
+                        c.close()
+                    except Exception:
+                        pass
+
+        return duration_ms
+
+    @app.post("/v1/audio/mixdown", dependencies=[Depends(apikey_auth)])
+    def mixdown(request: MixdownRequest) -> MixdownResponse:
+        """Mix multiple audio tracks into a single MP3 file.
+
+        Tracks are downloaded from presigned S3 URLs, mixed using PyAV,
+        and uploaded to a presigned S3 PUT URL.
+        """
+        if not request.track_urls:
+            raise HTTPException(status_code=400, detail="No track URLs provided")
+
+        logger.info(f"Mixdown request: {len(request.track_urls)} tracks")
+
+        temp_dir = tempfile.mkdtemp()
+        temp_files = []
+        output_mp3_path = None
+
+        try:
+            # Download all tracks
+            for i, url in enumerate(request.track_urls):
+                temp_path = download_track(url, temp_dir, i)
+                temp_files.append(temp_path)
+
+            # Mix tracks
+            output_mp3_path = os.path.join(temp_dir, "mixed.mp3")
+            duration_ms = mixdown_tracks_modal(
+                temp_files,
+                output_mp3_path,
+                request.target_sample_rate,
+                request.expected_duration_sec,
+                logger,
+            )
+
+            # Upload result to S3
+            logger.info("Uploading result to S3")
+            file_size = os.path.getsize(output_mp3_path)
+            with open(output_mp3_path, "rb") as f:
+                upload_response = requests.put(
+                    request.output_url, data=f, timeout=300
+                )
+
+            if upload_response.status_code == 403:
+                raise HTTPException(
+                    status_code=403, detail="Output presigned URL expired"
+                )
+
+            upload_response.raise_for_status()
+            logger.info(f"Upload complete: {file_size} bytes")
+
+            return MixdownResponse(
+                duration_ms=duration_ms,
+                tracks_mixed=len(request.track_urls),
+                audio_uploaded=True,
+            )
+
+        except HTTPException:
+            raise
+        except Exception as e:
+            logger.error(f"Mixdown failed: {e}", exc_info=True)
+            raise HTTPException(status_code=500, detail=f"Mixdown failed: {str(e)}")
+
+        finally:
+            # Cleanup temp files
+            for temp_path in temp_files:
+                try:
+                    os.unlink(temp_path)
+                except Exception as e:
+                    logger.warning(f"Failed to cleanup temp file {temp_path}: {e}")
+
+            if output_mp3_path and os.path.exists(output_mp3_path):
+                try:
+                    os.unlink(output_mp3_path)
+                except Exception as e:
+                    logger.warning(f"Failed to cleanup output file {output_mp3_path}: {e}")
+
+            try:
+                shutil.rmtree(temp_dir)
+            except Exception as e:
+                logger.warning(f"Failed to cleanup temp directory {temp_dir}: {e}")
+
+    return app

server/reflector/app.py

@@ -18,7 +18,6 @@ from reflector.views.rooms import router as rooms_router
 from reflector.views.rtc_offer import router as rtc_offer_router
 from reflector.views.transcripts import router as transcripts_router
 from reflector.views.transcripts_audio import router as transcripts_audio_router
-from reflector.views.transcripts_chat import router as transcripts_chat_router
 from reflector.views.transcripts_participants import (
     router as transcripts_participants_router,
 )
@@ -91,7 +90,6 @@ app.include_router(transcripts_participants_router, prefix="/v1")
 app.include_router(transcripts_speaker_router, prefix="/v1")
 app.include_router(transcripts_upload_router, prefix="/v1")
 app.include_router(transcripts_websocket_router, prefix="/v1")
-app.include_router(transcripts_chat_router, prefix="/v1")
 app.include_router(transcripts_webrtc_router, prefix="/v1")
 app.include_router(transcripts_process_router, prefix="/v1")
 app.include_router(user_router, prefix="/v1")

server/reflector/hatchet/run_workers.py

@@ -1,77 +0,0 @@
-"""
-Run Hatchet workers for the multitrack pipeline.
-Runs as a separate process, just like Celery workers.
-
-Usage:
-    uv run -m reflector.hatchet.run_workers
-
-    # Or via docker:
-    docker compose exec server uv run -m reflector.hatchet.run_workers
-"""
-
-import signal
-import sys
-
-from hatchet_sdk.rate_limit import RateLimitDuration
-
-from reflector.hatchet.constants import LLM_RATE_LIMIT_KEY, LLM_RATE_LIMIT_PER_SECOND
-from reflector.logger import logger
-from reflector.settings import settings
-
-
-def main() -> None:
-    """Start Hatchet worker polling."""
-    if not settings.HATCHET_ENABLED:
-        logger.error("HATCHET_ENABLED is False, not starting workers")
-        sys.exit(1)
-
-    if not settings.HATCHET_CLIENT_TOKEN:
-        logger.error("HATCHET_CLIENT_TOKEN is not set")
-        sys.exit(1)
-
-    logger.info(
-        "Starting Hatchet workers",
-        debug=settings.HATCHET_DEBUG,
-    )
-
-    # Import here (not top-level) - workflow modules call HatchetClientManager.get_client()
-    # at module level because Hatchet SDK decorators (@workflow.task) bind at import time.
-    # Can't use lazy init: decorators need the client object when function is defined.
-    from reflector.hatchet.client import HatchetClientManager  # noqa: PLC0415
-    from reflector.hatchet.workflows import (  # noqa: PLC0415
-        daily_multitrack_pipeline,
-        subject_workflow,
-        topic_chunk_workflow,
-        track_workflow,
-    )
-
-    hatchet = HatchetClientManager.get_client()
-
-    hatchet.rate_limits.put(
-        LLM_RATE_LIMIT_KEY, LLM_RATE_LIMIT_PER_SECOND, RateLimitDuration.SECOND
-    )
-
-    worker = hatchet.worker(
-        "reflector-pipeline-worker",
-        workflows=[
-            daily_multitrack_pipeline,
-            subject_workflow,
-            topic_chunk_workflow,
-            track_workflow,
-        ],
-    )
-
-    def shutdown_handler(signum: int, frame) -> None:
-        logger.info("Received shutdown signal, stopping workers...")
-        # Worker cleanup happens automatically on exit
-        sys.exit(0)
-
-    signal.signal(signal.SIGINT, shutdown_handler)
-    signal.signal(signal.SIGTERM, shutdown_handler)
-
-    logger.info("Starting Hatchet worker polling...")
-    worker.start()
-
-
-if __name__ == "__main__":
-    main()

server/reflector/hatchet/run_workers_cpu.py

@@ -0,0 +1,48 @@
+"""
+CPU-heavy worker pool for audio processing tasks.
+Handles ONLY: mixdown_tracks
+
+Configuration:
+- slots=1: Only mixdown (already serialized globally with max_runs=1)
+- Worker affinity: pool=cpu-heavy
+"""
+
+from reflector.hatchet.client import HatchetClientManager
+from reflector.hatchet.workflows.daily_multitrack_pipeline import (
+    daily_multitrack_pipeline,
+)
+from reflector.logger import logger
+from reflector.settings import settings
+
+
+def main():
+    if not settings.HATCHET_ENABLED:
+        logger.error("HATCHET_ENABLED is False, not starting CPU workers")
+        return
+
+    hatchet = HatchetClientManager.get_client()
+
+    logger.info(
+        "Starting Hatchet CPU worker pool (mixdown only)",
+        worker_name="cpu-worker-pool",
+        slots=1,
+        labels={"pool": "cpu-heavy"},
+    )
+
+    cpu_worker = hatchet.worker(
+        "cpu-worker-pool",
+        slots=1,  # Only 1 mixdown at a time (already serialized globally)
+        labels={
+            "pool": "cpu-heavy",
+        },
+        workflows=[daily_multitrack_pipeline],
+    )
+
+    try:
+        cpu_worker.start()
+    except KeyboardInterrupt:
+        logger.info("Received shutdown signal, stopping CPU workers...")
+
+
+if __name__ == "__main__":
+    main()

server/reflector/hatchet/run_workers_llm.py

@@ -0,0 +1,56 @@
+"""
+LLM/I/O worker pool for all non-CPU tasks.
+Handles: all tasks except mixdown_tracks (transcription, LLM inference, orchestration)
+"""
+
+from reflector.hatchet.client import HatchetClientManager
+from reflector.hatchet.workflows.daily_multitrack_pipeline import (
+    daily_multitrack_pipeline,
+)
+from reflector.hatchet.workflows.subject_processing import subject_workflow
+from reflector.hatchet.workflows.topic_chunk_processing import topic_chunk_workflow
+from reflector.hatchet.workflows.track_processing import track_workflow
+from reflector.logger import logger
+from reflector.settings import settings
+
+SLOTS = 10
+WORKER_NAME = "llm-worker-pool"
+POOL = "llm-io"
+
+
+def main():
+    if not settings.HATCHET_ENABLED:
+        logger.error("HATCHET_ENABLED is False, not starting LLM workers")
+        return
+
+    hatchet = HatchetClientManager.get_client()
+
+    logger.info(
+        "Starting Hatchet LLM worker pool (all tasks except mixdown)",
+        worker_name=WORKER_NAME,
+        slots=SLOTS,
+        labels={"pool": POOL},
+    )
+
+    llm_worker = hatchet.worker(
+        WORKER_NAME,
+        slots=SLOTS,  # not all slots are probably used
+        labels={
+            "pool": POOL,
+        },
+        workflows=[
+            daily_multitrack_pipeline,
+            topic_chunk_workflow,
+            subject_workflow,
+            track_workflow,
+        ],
+    )
+
+    try:
+        llm_worker.start()
+    except KeyboardInterrupt:
+        logger.info("Received shutdown signal, stopping LLM workers...")
+
+
+if __name__ == "__main__":
+    main()

server/reflector/hatchet/workflows/daily_multitrack_pipeline.py

@@ -23,7 +23,12 @@ from pathlib import Path
 from typing import Any, Callable, Coroutine, Protocol, TypeVar
 
 import httpx
-from hatchet_sdk import Context
+from hatchet_sdk import (
+    ConcurrencyExpression,
+    ConcurrencyLimitStrategy,
+    Context,
+)
+from hatchet_sdk.labels import DesiredWorkerLabel
 from pydantic import BaseModel
 
 from reflector.dailyco_api.client import DailyApiClient
@@ -467,10 +472,24 @@ async def process_tracks(input: PipelineInput, ctx: Context) -> ProcessTracksRes
     parents=[process_tracks],
     execution_timeout=timedelta(seconds=TIMEOUT_AUDIO),
     retries=3,
+    desired_worker_labels={
+        "pool": DesiredWorkerLabel(
+            value="cpu-heavy",
+            required=True,
+            weight=100,
+        ),
+    },
+    concurrency=[
+        ConcurrencyExpression(
+            expression="'mixdown-global'",
+            max_runs=1,  # serialize mixdown to prevent resource contention
+            limit_strategy=ConcurrencyLimitStrategy.GROUP_ROUND_ROBIN,  # Queue
+        )
+    ],
 )
 @with_error_handling(TaskName.MIXDOWN_TRACKS)
 async def mixdown_tracks(input: PipelineInput, ctx: Context) -> MixdownResult:
-    """Mix all padded tracks into single audio file using PyAV (same as Celery)."""
+    """Mix all padded tracks into single audio file using PyAV or Modal backend."""
     ctx.log("mixdown_tracks: mixing padded tracks into single audio file")
 
     track_result = ctx.task_output(process_tracks)
@@ -494,7 +513,7 @@ async def mixdown_tracks(input: PipelineInput, ctx: Context) -> MixdownResult:
 
     storage = _spawn_storage()
 
-    # Presign URLs on demand (avoids stale URLs on workflow replay)
+    # Presign URLs for padded tracks (same expiration for both backends)
     padded_urls = []
     for track_info in padded_tracks:
         if track_info.key:
@@ -515,33 +534,104 @@ async def mixdown_tracks(input: PipelineInput, ctx: Context) -> MixdownResult:
         logger.error("Mixdown failed - no decodable audio frames found")
         raise ValueError("No decodable audio frames in any track")
 
-    output_path = tempfile.mktemp(suffix=".mp3")
+    output_key = f"{input.transcript_id}/audio.mp3"
-    duration_ms_callback_capture_container = [0.0]
 
-    async def capture_duration(d):
+    # Conditional: Modal or local backend
-        duration_ms_callback_capture_container[0] = d
+    if settings.MIXDOWN_BACKEND == "modal":
+        ctx.log("mixdown_tracks: using Modal backend")
 
-    writer = AudioFileWriterProcessor(path=output_path, on_duration=capture_duration)
+        # Presign PUT URL for output (Modal will upload directly)
+        output_url = await storage.get_file_url(
+            output_key,
+            operation="put_object",
+            expires_in=PRESIGNED_URL_EXPIRATION_SECONDS,
+        )
 
-    await mixdown_tracks_pyav(
+        from reflector.processors.audio_mixdown_modal import (  # noqa: PLC0415
-        valid_urls,
+            AudioMixdownModalProcessor,
-        writer,
+        )
-        target_sample_rate,
-        offsets_seconds=None,
-        logger=logger,
-        progress_callback=make_audio_progress_logger(ctx, TaskName.MIXDOWN_TRACKS),
-        expected_duration_sec=recording_duration if recording_duration > 0 else None,
-    )
-    await writer.flush()
 
-    file_size = Path(output_path).stat().st_size
+        try:
-    storage_path = f"{input.transcript_id}/audio.mp3"
+            processor = AudioMixdownModalProcessor()
+            result = await processor.mixdown(
+                track_urls=valid_urls,
+                output_url=output_url,
+                target_sample_rate=target_sample_rate,
+                expected_duration_sec=recording_duration
+                if recording_duration > 0
+                else None,
+            )
+            duration_ms = result.duration_ms
+            tracks_mixed = result.tracks_mixed
 
-    with open(output_path, "rb") as mixed_file:
+            ctx.log(
-        await storage.put_file(storage_path, mixed_file)
+                f"mixdown_tracks: Modal returned duration={duration_ms}ms, tracks={tracks_mixed}"
+            )
+        except httpx.HTTPStatusError as e:
+            error_detail = e.response.text if hasattr(e.response, "text") else str(e)
+            logger.error(
+                "[Hatchet] Modal mixdown HTTP error",
+                transcript_id=input.transcript_id,
+                status_code=e.response.status_code if hasattr(e, "response") else None,
+                error=error_detail,
+            )
+            raise RuntimeError(
+                f"Modal mixdown failed with HTTP {e.response.status_code}: {error_detail}"
+            )
+        except httpx.TimeoutException:
+            logger.error(
+                "[Hatchet] Modal mixdown timeout",
+                transcript_id=input.transcript_id,
+                timeout=settings.MIXDOWN_TIMEOUT,
+            )
+            raise RuntimeError(
+                f"Modal mixdown timeout after {settings.MIXDOWN_TIMEOUT}s"
+            )
+        except ValueError as e:
+            logger.error(
+                "[Hatchet] Modal mixdown validation error",
+                transcript_id=input.transcript_id,
+                error=str(e),
+            )
+            raise
+    else:
+        ctx.log("mixdown_tracks: using local backend")
 
-    Path(output_path).unlink(missing_ok=True)
+        # Existing local implementation
+        output_path = tempfile.mktemp(suffix=".mp3")
+        duration_ms_callback_capture_container = [0.0]
 
+        async def capture_duration(d):
+            duration_ms_callback_capture_container[0] = d
+
+        writer = AudioFileWriterProcessor(
+            path=output_path, on_duration=capture_duration
+        )
+
+        await mixdown_tracks_pyav(
+            valid_urls,
+            writer,
+            target_sample_rate,
+            offsets_seconds=None,
+            logger=logger,
+            progress_callback=make_audio_progress_logger(ctx, TaskName.MIXDOWN_TRACKS),
+            expected_duration_sec=recording_duration
+            if recording_duration > 0
+            else None,
+        )
+        await writer.flush()
+
+        file_size = Path(output_path).stat().st_size
+        with open(output_path, "rb") as mixed_file:
+            await storage.put_file(output_key, mixed_file)
+
+        Path(output_path).unlink(missing_ok=True)
+        duration_ms = duration_ms_callback_capture_container[0]
+        tracks_mixed = len(valid_urls)
+
+        ctx.log(f"mixdown_tracks: local mixdown uploaded {file_size} bytes")
+
+    # Update DB (same for both backends)
     async with fresh_db_connection():
         from reflector.db.transcripts import transcripts_controller  # noqa: PLC0415
 
@@ -551,12 +641,12 @@ async def mixdown_tracks(input: PipelineInput, ctx: Context) -> MixdownResult:
                 transcript, {"audio_location": "storage"}
             )
 
-    ctx.log(f"mixdown_tracks complete: uploaded {file_size} bytes to {storage_path}")
+    ctx.log(f"mixdown_tracks complete: uploaded to {output_key}")
 
     return MixdownResult(
-        audio_key=storage_path,
+        audio_key=output_key,
-        duration=duration_ms_callback_capture_container[0],
+        duration=duration_ms,
-        tracks_mixed=len(valid_urls),
+        tracks_mixed=tracks_mixed,
     )
 
 

server/reflector/hatchet/workflows/topic_chunk_processing.py

@@ -7,7 +7,11 @@ Spawned dynamically by detect_topics via aio_run_many() for parallel processing.
 
 from datetime import timedelta
 
-from hatchet_sdk import ConcurrencyExpression, ConcurrencyLimitStrategy, Context
+from hatchet_sdk import (
+    ConcurrencyExpression,
+    ConcurrencyLimitStrategy,
+    Context,
+)
 from hatchet_sdk.rate_limit import RateLimit
 from pydantic import BaseModel
 
@@ -34,11 +38,13 @@ hatchet = HatchetClientManager.get_client()
 topic_chunk_workflow = hatchet.workflow(
     name="TopicChunkProcessing",
     input_validator=TopicChunkInput,
-    concurrency=ConcurrencyExpression(
+    concurrency=[
-        expression="'global'",  # constant string = global limit across all runs
+        ConcurrencyExpression(
-        max_runs=20,
+            expression="'global'",  # constant string = global limit across all runs
-        limit_strategy=ConcurrencyLimitStrategy.GROUP_ROUND_ROBIN,
+            max_runs=20,
-    ),
+            limit_strategy=ConcurrencyLimitStrategy.GROUP_ROUND_ROBIN,
+        )
+    ],
 )
 
 

server/reflector/processors/audio_mixdown_modal.py

@@ -0,0 +1,89 @@
+"""
+Modal.com backend for audio mixdown.
+
+Uses Modal's CPU containers to offload audio mixing from Hatchet workers.
+Communicates via presigned S3 URLs for both input and output.
+"""
+
+import httpx
+from pydantic import BaseModel
+
+from reflector.settings import settings
+
+
+class MixdownResponse(BaseModel):
+    """Response from Modal mixdown endpoint."""
+
+    duration_ms: float
+    tracks_mixed: int
+    audio_uploaded: bool
+
+
+class AudioMixdownModalProcessor:
+    """Audio mixdown processor using Modal.com CPU backend.
+
+    Sends track URLs (presigned GET) and output URL (presigned PUT) to Modal.
+    Modal handles download, mixdown via PyAV, and upload.
+    """
+
+    def __init__(self, modal_api_key: str | None = None):
+        if not settings.MIXDOWN_URL:
+            raise ValueError("MIXDOWN_URL required to use AudioMixdownModalProcessor")
+
+        self.mixdown_url = settings.MIXDOWN_URL + "/v1"
+        self.timeout = settings.MIXDOWN_TIMEOUT
+        self.modal_api_key = modal_api_key or settings.MIXDOWN_MODAL_API_KEY
+
+        if not self.modal_api_key:
+            raise ValueError(
+                "MIXDOWN_MODAL_API_KEY required to use AudioMixdownModalProcessor"
+            )
+
+    async def mixdown(
+        self,
+        track_urls: list[str],
+        output_url: str,
+        target_sample_rate: int,
+        expected_duration_sec: float | None = None,
+    ) -> MixdownResponse:
+        """Mix multiple audio tracks via Modal backend.
+
+        Args:
+            track_urls: List of presigned GET URLs for audio tracks (non-empty)
+            output_url: Presigned PUT URL for output MP3
+            target_sample_rate: Sample rate for output (Hz, must be positive)
+            expected_duration_sec: Optional fallback duration if container metadata unavailable
+
+        Returns:
+            MixdownResponse with duration_ms, tracks_mixed, audio_uploaded
+
+        Raises:
+            ValueError: If track_urls is empty or target_sample_rate invalid
+            httpx.HTTPStatusError: On HTTP errors (404, 403, 500, etc.)
+            httpx.TimeoutException: On timeout
+        """
+        # Validate inputs
+        if not track_urls:
+            raise ValueError("track_urls cannot be empty")
+        if target_sample_rate <= 0:
+            raise ValueError(
+                f"target_sample_rate must be positive, got {target_sample_rate}"
+            )
+        if expected_duration_sec is not None and expected_duration_sec < 0:
+            raise ValueError(
+                f"expected_duration_sec cannot be negative, got {expected_duration_sec}"
+            )
+
+        async with httpx.AsyncClient(timeout=self.timeout) as client:
+            response = await client.post(
+                f"{self.mixdown_url}/audio/mixdown",
+                headers={"Authorization": f"Bearer {self.modal_api_key}"},
+                json={
+                    "track_urls": track_urls,
+                    "output_url": output_url,
+                    "target_sample_rate": target_sample_rate,
+                    "expected_duration_sec": expected_duration_sec,
+                },
+            )
+            response.raise_for_status()
+            return MixdownResponse(**response.json())

server/reflector/settings.py

@@ -98,6 +98,17 @@ class Settings(BaseSettings):
     # Diarization: local pyannote.audio
     DIARIZATION_PYANNOTE_AUTH_TOKEN: str | None = None
 
+    # Audio Mixdown
+    # backends:
+    #   - local: in-process PyAV mixdown (runs in same process as Hatchet worker)
+    #   - modal: HTTP API client to Modal.com CPU container
+    MIXDOWN_BACKEND: str = "local"
+    MIXDOWN_URL: str | None = None
+    MIXDOWN_TIMEOUT: int = 900  # 15 minutes
+
+    # Mixdown: modal backend
+    MIXDOWN_MODAL_API_KEY: str | None = None
+
     # Sentry
     SENTRY_DSN: str | None = None
 

server/reflector/views/transcripts_chat.py

@@ -1,133 +0,0 @@
-"""
-Transcripts chat API
-====================
-
-WebSocket endpoint for bidirectional chat with LLM about transcript content.
-"""
-
-from typing import Optional
-
-from fastapi import APIRouter, WebSocket, WebSocketDisconnect
-from llama_index.core import Settings
-from llama_index.core.base.llms.types import ChatMessage, MessageRole
-
-from reflector.auth.auth_jwt import JWTAuth
-from reflector.db.recordings import recordings_controller
-from reflector.db.transcripts import transcripts_controller
-from reflector.db.users import user_controller
-from reflector.llm import LLM
-from reflector.settings import settings
-from reflector.utils.transcript_formats import topics_to_webvtt_named
-
-router = APIRouter()
-
-
-async def _get_is_multitrack(transcript) -> bool:
-    """Detect if transcript is from multitrack recording."""
-    if not transcript.recording_id:
-        return False
-    recording = await recordings_controller.get_by_id(transcript.recording_id)
-    return recording is not None and recording.is_multitrack
-
-
-@router.websocket("/transcripts/{transcript_id}/chat")
-async def transcript_chat_websocket(
-    transcript_id: str,
-    websocket: WebSocket,
-):
-    """WebSocket endpoint for chatting with LLM about transcript content."""
-    # 1. Auth check (optional) - extract token from WebSocket subprotocol header
-    # Browser can't send Authorization header for WS; use subprotocol: ["bearer", token]
-    raw_subprotocol = websocket.headers.get("sec-websocket-protocol") or ""
-    parts = [p.strip() for p in raw_subprotocol.split(",") if p.strip()]
-    token: Optional[str] = None
-    negotiated_subprotocol: Optional[str] = None
-    if len(parts) >= 2 and parts[0].lower() == "bearer":
-        negotiated_subprotocol = "bearer"
-        token = parts[1]
-
-    user_id: Optional[str] = None
-    if token:
-        try:
-            payload = JWTAuth().verify_token(token)
-            authentik_uid = payload.get("sub")
-
-            if authentik_uid:
-                user = await user_controller.get_by_authentik_uid(authentik_uid)
-                if user:
-                    user_id = user.id
-        except Exception:
-            # Auth failed - continue as anonymous
-            pass
-
-    # Get transcript (respects user_id for private transcripts)
-    transcript = await transcripts_controller.get_by_id_for_http(
-        transcript_id, user_id=user_id
-    )
-    if not transcript:
-        await websocket.close(code=1008)  # Policy violation (not found/unauthorized)
-        return
-
-    # 2. Accept connection (with negotiated subprotocol if present)
-    await websocket.accept(subprotocol=negotiated_subprotocol)
-
-    # 3. Generate WebVTT context
-    is_multitrack = await _get_is_multitrack(transcript)
-    webvtt = topics_to_webvtt_named(
-        transcript.topics, transcript.participants, is_multitrack
-    )
-
-    # Truncate if needed (15k char limit for POC)
-    webvtt_truncated = webvtt[:15000] if len(webvtt) > 15000 else webvtt
-
-    # 4. Configure LLM
-    llm = LLM(settings=settings, temperature=0.7)
-
-    # 5. System message with transcript context
-    system_msg = f"""You are analyzing this meeting transcript (WebVTT):
-
-{webvtt_truncated}
-
-Answer questions about content, speakers, timeline. Include timestamps when relevant."""
-
-    # 6. Conversation history
-    conversation_history = [ChatMessage(role=MessageRole.SYSTEM, content=system_msg)]
-
-    try:
-        # 7. Message loop
-        while True:
-            data = await websocket.receive_json()
-
-            if data.get("type") == "get_context":
-                # Return WebVTT context (for debugging/testing)
-                await websocket.send_json({"type": "context", "webvtt": webvtt})
-                continue
-
-            if data.get("type") != "message":
-                # Echo unknown types for backward compatibility
-                await websocket.send_json({"type": "echo", "data": data})
-                continue
-
-            # Add user message to history
-            user_msg = ChatMessage(role=MessageRole.USER, content=data.get("text", ""))
-            conversation_history.append(user_msg)
-
-            # Stream LLM response
-            assistant_msg = ""
-            chat_stream = await Settings.llm.astream_chat(conversation_history)
-            async for chunk in chat_stream:
-                token = chunk.delta or ""
-                if token:
-                    await websocket.send_json({"type": "token", "text": token})
-                    assistant_msg += token
-
-            # Save assistant response to history
-            conversation_history.append(
-                ChatMessage(role=MessageRole.ASSISTANT, content=assistant_msg)
-            )
-            await websocket.send_json({"type": "done"})
-
-    except WebSocketDisconnect:
-        pass
-    except Exception as e:
-        await websocket.send_json({"type": "error", "message": str(e)})

server/runserver.sh

@@ -7,8 +7,10 @@ elif [ "${ENTRYPOINT}" = "worker" ]; then
     uv run celery -A reflector.worker.app worker --loglevel=info
 elif [ "${ENTRYPOINT}" = "beat" ]; then
     uv run celery -A reflector.worker.app beat --loglevel=info
-elif [ "${ENTRYPOINT}" = "hatchet-worker" ]; then
+elif [ "${ENTRYPOINT}" = "hatchet-worker-cpu" ]; then
-    uv run python -m reflector.hatchet.run_workers
+    uv run python -m reflector.hatchet.run_workers_cpu
+elif [ "${ENTRYPOINT}" = "hatchet-worker-llm" ]; then
+    uv run python -m reflector.hatchet.run_workers_llm
 else
     echo "Unknown command"
 fi

server/tests/test_transcripts_chat.py

@@ -1,234 +0,0 @@
-"""Tests for transcript chat WebSocket endpoint."""
-
-import asyncio
-import threading
-import time
-from pathlib import Path
-
-import pytest
-from httpx_ws import aconnect_ws
-from uvicorn import Config, Server
-
-from reflector.db.transcripts import (
-    SourceKind,
-    TranscriptParticipant,
-    TranscriptTopic,
-    transcripts_controller,
-)
-from reflector.processors.types import Word
-
-
-@pytest.fixture
-def chat_appserver(tmpdir, setup_database):
-    """Start a real HTTP server for WebSocket testing."""
-    from reflector.app import app
-    from reflector.db import get_database
-    from reflector.settings import settings
-
-    DATA_DIR = settings.DATA_DIR
-    settings.DATA_DIR = Path(tmpdir)
-
-    # Start server in separate thread with its own event loop
-    host = "127.0.0.1"
-    port = 1256  # Different port from rtc tests
-    server_started = threading.Event()
-    server_exception = None
-    server_instance = None
-
-    def run_server():
-        nonlocal server_exception, server_instance
-        try:
-            # Create new event loop for this thread
-            loop = asyncio.new_event_loop()
-            asyncio.set_event_loop(loop)
-
-            config = Config(app=app, host=host, port=port, loop=loop)
-            server_instance = Server(config)
-
-            async def start_server():
-                # Initialize database connection in this event loop
-                database = get_database()
-                await database.connect()
-                try:
-                    await server_instance.serve()
-                finally:
-                    await database.disconnect()
-
-            # Signal that server is starting
-            server_started.set()
-            loop.run_until_complete(start_server())
-        except Exception as e:
-            server_exception = e
-            server_started.set()
-        finally:
-            loop.close()
-
-    server_thread = threading.Thread(target=run_server, daemon=True)
-    server_thread.start()
-
-    # Wait for server to start
-    server_started.wait(timeout=30)
-    if server_exception:
-        raise server_exception
-
-    # Wait for server to be fully ready
-    time.sleep(1)
-
-    yield server_instance, host, port
-
-    # Stop server
-    if server_instance:
-        server_instance.should_exit = True
-        server_thread.join(timeout=30)
-
-    settings.DATA_DIR = DATA_DIR
-
-
-@pytest.fixture
-async def test_transcript(setup_database):
-    """Create a test transcript for WebSocket tests."""
-    transcript = await transcripts_controller.add(
-        name="Test Transcript for Chat", source_kind=SourceKind.FILE
-    )
-    return transcript
-
-
-@pytest.fixture
-async def test_transcript_with_content(setup_database):
-    """Create a test transcript with actual content for WebVTT generation."""
-    transcript = await transcripts_controller.add(
-        name="Test Transcript with Content", source_kind=SourceKind.FILE
-    )
-
-    # Add participants
-    await transcripts_controller.update(
-        transcript,
-        {
-            "participants": [
-                TranscriptParticipant(id="1", speaker=0, name="Alice").model_dump(),
-                TranscriptParticipant(id="2", speaker=1, name="Bob").model_dump(),
-            ]
-        },
-    )
-
-    # Add topic with words
-    await transcripts_controller.upsert_topic(
-        transcript,
-        TranscriptTopic(
-            title="Introduction",
-            summary="Opening remarks",
-            timestamp=0.0,
-            words=[
-                Word(text="Hello ", start=0.0, end=1.0, speaker=0),
-                Word(text="everyone.", start=1.0, end=2.0, speaker=0),
-                Word(text="Hi ", start=2.0, end=3.0, speaker=1),
-                Word(text="there!", start=3.0, end=4.0, speaker=1),
-            ],
-        ),
-    )
-
-    return transcript
-
-
-@pytest.mark.asyncio
-async def test_chat_websocket_connection_success(test_transcript, chat_appserver):
-    """Test successful WebSocket connection to chat endpoint."""
-    server, host, port = chat_appserver
-    base_url = f"ws://{host}:{port}/v1"
-
-    async with aconnect_ws(f"{base_url}/transcripts/{test_transcript.id}/chat") as ws:
-        # Send unknown message type to test echo behavior
-        await ws.send_json({"type": "test", "text": "Hello"})
-
-        # Should receive echo for unknown types
-        response = await ws.receive_json()
-        assert response["type"] == "echo"
-        assert response["data"]["type"] == "test"
-
-
-@pytest.mark.asyncio
-async def test_chat_websocket_nonexistent_transcript(chat_appserver):
-    """Test WebSocket connection fails for nonexistent transcript."""
-    server, host, port = chat_appserver
-    base_url = f"ws://{host}:{port}/v1"
-
-    # Connection should fail or disconnect immediately for non-existent transcript
-    # Different behavior from successful connection
-    with pytest.raises(Exception):  # Will raise on connection or first operation
-        async with aconnect_ws(f"{base_url}/transcripts/nonexistent-id/chat") as ws:
-            await ws.send_json({"type": "message", "text": "Hello"})
-            await ws.receive_json()
-
-
-@pytest.mark.asyncio
-async def test_chat_websocket_multiple_messages(test_transcript, chat_appserver):
-    """Test sending multiple messages through WebSocket."""
-    server, host, port = chat_appserver
-    base_url = f"ws://{host}:{port}/v1"
-
-    async with aconnect_ws(f"{base_url}/transcripts/{test_transcript.id}/chat") as ws:
-        # Send multiple unknown message types (testing echo behavior)
-        messages = ["First message", "Second message", "Third message"]
-
-        for i, msg in enumerate(messages):
-            await ws.send_json({"type": f"test{i}", "text": msg})
-            response = await ws.receive_json()
-            assert response["type"] == "echo"
-            assert response["data"]["type"] == f"test{i}"
-            assert response["data"]["text"] == msg
-
-
-@pytest.mark.asyncio
-async def test_chat_websocket_disconnect_graceful(test_transcript, chat_appserver):
-    """Test WebSocket disconnects gracefully."""
-    server, host, port = chat_appserver
-    base_url = f"ws://{host}:{port}/v1"
-
-    async with aconnect_ws(f"{base_url}/transcripts/{test_transcript.id}/chat") as ws:
-        await ws.send_json({"type": "message", "text": "Hello"})
-        await ws.receive_json()
-        # Close handled by context manager - should not raise
-
-
-@pytest.mark.asyncio
-async def test_chat_websocket_context_generation(
-    test_transcript_with_content, chat_appserver
-):
-    """Test WebVTT context is generated on connection."""
-    server, host, port = chat_appserver
-    base_url = f"ws://{host}:{port}/v1"
-
-    async with aconnect_ws(
-        f"{base_url}/transcripts/{test_transcript_with_content.id}/chat"
-    ) as ws:
-        # Request context
-        await ws.send_json({"type": "get_context"})
-
-        # Receive context response
-        response = await ws.receive_json()
-        assert response["type"] == "context"
-        assert "webvtt" in response
-
-        # Verify WebVTT format
-        webvtt = response["webvtt"]
-        assert webvtt.startswith("WEBVTT")
-        assert "<v Alice>" in webvtt
-        assert "<v Bob>" in webvtt
-        assert "Hello everyone." in webvtt
-        assert "Hi there!" in webvtt
-
-
-@pytest.mark.asyncio
-async def test_chat_websocket_unknown_message_type(test_transcript, chat_appserver):
-    """Test unknown message types are echoed back."""
-    server, host, port = chat_appserver
-    base_url = f"ws://{host}:{port}/v1"
-
-    async with aconnect_ws(f"{base_url}/transcripts/{test_transcript.id}/chat") as ws:
-        # Send unknown message type
-        await ws.send_json({"type": "unknown", "data": "test"})
-
-        # Should receive echo
-        response = await ws.receive_json()
-        assert response["type"] == "echo"
-        assert response["data"]["type"] == "unknown"

www/app/(app)/transcripts/TranscriptChatModal.tsx

@@ -1,103 +0,0 @@
-"use client";
-
-import { useState } from "react";
-import { Box, Dialog, Input, IconButton } from "@chakra-ui/react";
-import { MessageCircle } from "lucide-react";
-import Markdown from "react-markdown";
-import "../../styles/markdown.css";
-import type { Message } from "./useTranscriptChat";
-
-interface TranscriptChatModalProps {
-  open: boolean;
-  onClose: () => void;
-  messages: Message[];
-  sendMessage: (text: string) => void;
-  isStreaming: boolean;
-  currentStreamingText: string;
-}
-
-export function TranscriptChatModal({
-  open,
-  onClose,
-  messages,
-  sendMessage,
-  isStreaming,
-  currentStreamingText,
-}: TranscriptChatModalProps) {
-  const [input, setInput] = useState("");
-
-  const handleSend = () => {
-    if (!input.trim()) return;
-    sendMessage(input);
-    setInput("");
-  };
-
-  return (
-    <Dialog.Root open={open} onOpenChange={(e) => !e.open && onClose()}>
-      <Dialog.Backdrop />
-      <Dialog.Positioner>
-        <Dialog.Content maxW="500px" h="600px">
-          <Dialog.Header>Transcript Chat</Dialog.Header>
-
-          <Dialog.Body overflowY="auto">
-            {messages.map((msg) => (
-              <Box
-                key={msg.id}
-                p={3}
-                mb={2}
-                bg={msg.role === "user" ? "blue.50" : "gray.50"}
-                borderRadius="md"
-              >
-                {msg.role === "user" ? (
-                  msg.text
-                ) : (
-                  <div className="markdown">
-                    <Markdown>{msg.text}</Markdown>
-                  </div>
-                )}
-              </Box>
-            ))}
-
-            {isStreaming && (
-              <Box p={3} bg="gray.50" borderRadius="md">
-                <div className="markdown">
-                  <Markdown>{currentStreamingText}</Markdown>
-                </div>
-                <Box as="span" className="animate-pulse">
-                  ▊
-                </Box>
-              </Box>
-            )}
-          </Dialog.Body>
-
-          <Dialog.Footer>
-            <Input
-              value={input}
-              onChange={(e) => setInput(e.target.value)}
-              onKeyDown={(e) => e.key === "Enter" && handleSend()}
-              placeholder="Ask about transcript..."
-              disabled={isStreaming}
-            />
-          </Dialog.Footer>
-        </Dialog.Content>
-      </Dialog.Positioner>
-    </Dialog.Root>
-  );
-}
-
-export function TranscriptChatButton({ onClick }: { onClick: () => void }) {
-  return (
-    <IconButton
-      position="fixed"
-      bottom="24px"
-      right="24px"
-      onClick={onClick}
-      size="lg"
-      colorPalette="blue"
-      borderRadius="full"
-      aria-label="Open chat"
-    >
-      <MessageCircle />
-    </IconButton>
-  );
-}

www/app/(app)/transcripts/[transcriptId]/page.tsx

@@ -18,15 +18,9 @@ import {
   Skeleton,
   Text,
   Spinner,
-  useDisclosure,
 } from "@chakra-ui/react";
 import { useTranscriptGet } from "../../../lib/apiHooks";
 import { TranscriptStatus } from "../../../lib/transcript";
-import {
-  TranscriptChatModal,
-  TranscriptChatButton,
-} from "../TranscriptChatModal";
-import { useTranscriptChat } from "../useTranscriptChat";
 
 type TranscriptDetails = {
   params: Promise<{
@@ -59,9 +53,6 @@ export default function TranscriptDetails(details: TranscriptDetails) {
   const [finalSummaryElement, setFinalSummaryElement] =
     useState<HTMLDivElement | null>(null);
 
-  const { open, onOpen, onClose } = useDisclosure();
-  const chat = useTranscriptChat(transcriptId);
-
   useEffect(() => {
     if (!waiting || !transcript.data) return;
 
@@ -128,15 +119,6 @@ export default function TranscriptDetails(details: TranscriptDetails) {
 
   return (
     <>
-      <TranscriptChatModal
-        open={open}
-        onClose={onClose}
-        messages={chat.messages}
-        sendMessage={chat.sendMessage}
-        isStreaming={chat.isStreaming}
-        currentStreamingText={chat.currentStreamingText}
-      />
-      <TranscriptChatButton onClick={onOpen} />
       <Grid
         templateColumns="1fr"
         templateRows="auto minmax(0, 1fr)"

www/app/(app)/transcripts/useTranscriptChat.ts

@@ -1,130 +0,0 @@
-"use client";
-
-import { useEffect, useState, useRef } from "react";
-import { getSession } from "next-auth/react";
-import { WEBSOCKET_URL } from "../../lib/apiClient";
-import { assertExtendedToken } from "../../lib/types";
-
-export type Message = {
-  id: string;
-  role: "user" | "assistant";
-  text: string;
-  timestamp: Date;
-};
-
-export type UseTranscriptChat = {
-  messages: Message[];
-  sendMessage: (text: string) => void;
-  isStreaming: boolean;
-  currentStreamingText: string;
-};
-
-export const useTranscriptChat = (transcriptId: string): UseTranscriptChat => {
-  const [messages, setMessages] = useState<Message[]>([]);
-  const [isStreaming, setIsStreaming] = useState(false);
-  const [currentStreamingText, setCurrentStreamingText] = useState("");
-  const wsRef = useRef<WebSocket | null>(null);
-  const streamingTextRef = useRef<string>("");
-  const isMountedRef = useRef<boolean>(true);
-
-  useEffect(() => {
-    isMountedRef.current = true;
-
-    const connectWebSocket = async () => {
-      const url = `${WEBSOCKET_URL}/v1/transcripts/${transcriptId}/chat`;
-
-      // Get auth token for WebSocket subprotocol
-      let protocols: string[] | undefined;
-      try {
-        const session = await getSession();
-        if (session) {
-          const token = assertExtendedToken(session).accessToken;
-          // Pass token via subprotocol: ["bearer", token]
-          protocols = ["bearer", token];
-        }
-      } catch (error) {
-        console.warn("Failed to get auth token for WebSocket:", error);
-      }
-
-      const ws = new WebSocket(url, protocols);
-      wsRef.current = ws;
-
-      ws.onopen = () => {
-        console.log("Chat WebSocket connected");
-      };
-
-      ws.onmessage = (event) => {
-        if (!isMountedRef.current) return;
-
-        const msg = JSON.parse(event.data);
-
-        switch (msg.type) {
-          case "token":
-            setIsStreaming(true);
-            streamingTextRef.current += msg.text;
-            setCurrentStreamingText(streamingTextRef.current);
-            break;
-
-          case "done":
-            // CRITICAL: Save the text BEFORE resetting the ref
-            // The setMessages callback may execute later, after ref is reset
-            const finalText = streamingTextRef.current;
-
-            setMessages((prev) => [
-              ...prev,
-              {
-                id: Date.now().toString(),
-                role: "assistant",
-                text: finalText,
-                timestamp: new Date(),
-              },
-            ]);
-            streamingTextRef.current = "";
-            setCurrentStreamingText("");
-            setIsStreaming(false);
-            break;
-
-          case "error":
-            console.error("Chat error:", msg.message);
-            setIsStreaming(false);
-            break;
-        }
-      };
-
-      ws.onerror = (error) => {
-        console.error("WebSocket error:", error);
-      };
-
-      ws.onclose = () => {
-        console.log("Chat WebSocket closed");
-      };
-    };
-
-    connectWebSocket();
-
-    return () => {
-      isMountedRef.current = false;
-      if (wsRef.current) {
-        wsRef.current.close();
-      }
-    };
-  }, [transcriptId]);
-
-  const sendMessage = (text: string) => {
-    if (!wsRef.current) return;
-
-    setMessages((prev) => [
-      ...prev,
-      {
-        id: Date.now().toString(),
-        role: "user",
-        text,
-        timestamp: new Date(),
-      },
-    ]);
-
-    wsRef.current.send(JSON.stringify({ type: "message", text }));
-  };
-
-  return { messages, sendMessage, isStreaming, currentStreamingText };
-};