chore(main): release 0.32.1 (#840)

* Add Modal backend for audio padding

- Create reflector_padding.py Modal deployment (CPU-based)
- Add PaddingWorkflow with conditional Modal/local backend
- Update deploy-all.sh to include padding deployment

---------

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

.gitignore

@@ -1,5 +1,6 @@
 .DS_Store
 server/.env
+server/.env.production
 .env
 Caddyfile
 server/exportdanswer

CHANGELOG.md

@@ -1,5 +1,27 @@
 # Changelog
 
+## [0.32.1](https://github.com/Monadical-SAS/reflector/compare/v0.32.0...v0.32.1) (2026-01-30)
+
+
+### Bug Fixes
+
+* daily multitrack pipeline finalze dependency fix ([23eb137](https://github.com/Monadical-SAS/reflector/commit/23eb1371cb9348c4b81eb12ad506b582f8a4799e))
+* match httpx pad with hatchet audio timeout ([c05d1f0](https://github.com/Monadical-SAS/reflector/commit/c05d1f03cd8369fc06efd455527e50246887efd0))
+
+## [0.32.0](https://github.com/Monadical-SAS/reflector/compare/v0.31.0...v0.32.0) (2026-01-30)
+
+
+### Features
+
+* modal padding ([#837](https://github.com/Monadical-SAS/reflector/issues/837)) ([7fde64e](https://github.com/Monadical-SAS/reflector/commit/7fde64e2529a1d37b0f7507c62d983a7bd0b5b89))
+
+## [0.31.0](https://github.com/Monadical-SAS/reflector/compare/v0.30.0...v0.31.0) (2026-01-23)
+
+
+### Features
+
+* mixdown optional ([#834](https://github.com/Monadical-SAS/reflector/issues/834)) ([fc3ef6c](https://github.com/Monadical-SAS/reflector/commit/fc3ef6c8933231c731fad84e7477a476a6220a5e))
+
 ## [0.30.0](https://github.com/Monadical-SAS/reflector/compare/v0.29.0...v0.30.0) (2026-01-23)
 
 

gpu/modal_deployments/deploy-all.sh

@@ -131,6 +131,15 @@ if [ -z "$DIARIZER_URL" ]; then
 fi
 echo "  -> $DIARIZER_URL"
 
+echo ""
+echo "Deploying padding (CPU audio processing via Modal SDK)..."
+modal deploy reflector_padding.py
+if [ $? -ne 0 ]; then
+    echo "Error: Failed to deploy padding. Check Modal dashboard for details."
+    exit 1
+fi
+echo "  -> reflector-padding.pad_track (Modal SDK function)"
+
 # --- Output Configuration ---
 echo ""
 echo "=========================================="
@@ -147,4 +156,6 @@ echo ""
 echo "DIARIZATION_BACKEND=modal"
 echo "DIARIZATION_URL=$DIARIZER_URL"
 echo "DIARIZATION_MODAL_API_KEY=$API_KEY"
+echo ""
+echo "# Padding uses Modal SDK (requires MODAL_TOKEN_ID/SECRET in worker containers)"
 echo "# --- End Modal Configuration ---"

gpu/modal_deployments/reflector_padding.py

@@ -0,0 +1,277 @@
+"""
+Reflector GPU backend - audio padding
+======================================
+
+CPU-intensive audio padding service for adding silence to audio tracks.
+Uses PyAV filter graph (adelay) for precise track synchronization.
+
+IMPORTANT: This padding logic is duplicated from server/reflector/utils/audio_padding.py
+for Modal deployment isolation (Modal can't import from server/reflector/). If you modify
+the PyAV filter graph or padding algorithm, you MUST update both:
+  - gpu/modal_deployments/reflector_padding.py (this file)
+  - server/reflector/utils/audio_padding.py
+
+Constants duplicated from server/reflector/utils/audio_constants.py for same reason.
+"""
+
+import os
+import tempfile
+from fractions import Fraction
+import math
+import asyncio
+
+import modal
+
+S3_TIMEOUT = 60 # happens 2 times
+PADDING_TIMEOUT = 600 + (S3_TIMEOUT * 2)
+SCALEDOWN_WINDOW = 60  # The maximum duration (in seconds) that individual containers can remain idle when scaling down.
+DISCONNECT_CHECK_INTERVAL = 2  # Check for client disconnect
+
+
+app = modal.App("reflector-padding")
+
+# CPU-based image
+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
+    )
+)
+
+# ref B0F71CE8-FC59-4AA5-8414-DAFB836DB711
+OPUS_STANDARD_SAMPLE_RATE = 48000
+# ref B0F71CE8-FC59-4AA5-8414-DAFB836DB711
+OPUS_DEFAULT_BIT_RATE = 128000
+
+
+@app.function(
+    cpu=2.0,
+    timeout=PADDING_TIMEOUT,
+    scaledown_window=SCALEDOWN_WINDOW,
+    image=image,
+)
+@modal.asgi_app()
+def web():
+    from fastapi import FastAPI, Request, HTTPException
+    from pydantic import BaseModel
+
+    class PaddingRequest(BaseModel):
+        track_url: str
+        output_url: str
+        start_time_seconds: float
+        track_index: int
+
+    class PaddingResponse(BaseModel):
+        size: int
+        cancelled: bool = False
+
+    web_app = FastAPI()
+
+    @web_app.post("/pad")
+    async def pad_track_endpoint(request: Request, req: PaddingRequest) -> PaddingResponse:
+        """Modal web endpoint for padding audio tracks with disconnect detection.
+        """
+        import logging
+
+        logging.basicConfig(level=logging.INFO, format="%(asctime)s - %(levelname)s - %(message)s")
+        logger = logging.getLogger(__name__)
+
+        if not req.track_url:
+            raise HTTPException(status_code=400, detail="track_url cannot be empty")
+        if not req.output_url:
+            raise HTTPException(status_code=400, detail="output_url cannot be empty")
+        if req.start_time_seconds <= 0:
+            raise HTTPException(status_code=400, detail=f"start_time_seconds must be positive, got {req.start_time_seconds}")
+        if req.start_time_seconds > 18000:
+            raise HTTPException(status_code=400, detail=f"start_time_seconds exceeds maximum 18000s (5 hours)")
+
+        logger.info(f"Padding request: track {req.track_index}, delay={req.start_time_seconds}s")
+
+        # Thread-safe cancellation flag shared between async disconnect checker and blocking thread
+        import threading
+        cancelled = threading.Event()
+
+        async def check_disconnect():
+            """Background task to check for client disconnect every 2 seconds."""
+            while not cancelled.is_set():
+                await asyncio.sleep(DISCONNECT_CHECK_INTERVAL)
+                if await request.is_disconnected():
+                    logger.warning("Client disconnected, setting cancellation flag")
+                    cancelled.set()
+                    break
+
+        # Start disconnect checker in background
+        disconnect_task = asyncio.create_task(check_disconnect())
+
+        try:
+            result = await asyncio.get_event_loop().run_in_executor(
+                None, _pad_track_blocking, req, cancelled, logger
+            )
+            return PaddingResponse(**result)
+        finally:
+            cancelled.set()
+            disconnect_task.cancel()
+            try:
+                await disconnect_task
+            except asyncio.CancelledError:
+                pass
+
+    def _pad_track_blocking(req, cancelled, logger) -> dict:
+        """Blocking CPU-bound padding work with periodic cancellation checks.
+
+        Args:
+            cancelled: threading.Event for thread-safe cancellation signaling
+        """
+        import av
+        import requests
+        from av.audio.resampler import AudioResampler
+        import time
+
+        temp_dir = tempfile.mkdtemp()
+        input_path = None
+        output_path = None
+        last_check = time.time()
+
+        try:
+            logger.info("Downloading track for padding")
+            response = requests.get(req.track_url, stream=True, timeout=S3_TIMEOUT)
+            response.raise_for_status()
+
+            input_path = os.path.join(temp_dir, "track.webm")
+            total_bytes = 0
+            chunk_count = 0
+            with open(input_path, "wb") as f:
+                for chunk in response.iter_content(chunk_size=8192):
+                    if chunk:
+                        f.write(chunk)
+                        total_bytes += len(chunk)
+                        chunk_count += 1
+
+                        # Check for cancellation every arbitrary amount of chunks
+                        if chunk_count % 12 == 0:
+                            now = time.time()
+                            if now - last_check >= DISCONNECT_CHECK_INTERVAL:
+                                if cancelled.is_set():
+                                    logger.info("Cancelled during download, exiting early")
+                                    return {"size": 0, "cancelled": True}
+                                last_check = now
+            logger.info(f"Track downloaded: {total_bytes} bytes")
+
+            if cancelled.is_set():
+                logger.info("Cancelled after download, exiting early")
+                return {"size": 0, "cancelled": True}
+
+            # Apply padding using PyAV
+            output_path = os.path.join(temp_dir, "padded.webm")
+            delay_ms = math.floor(req.start_time_seconds * 1000)
+            logger.info(f"Padding track {req.track_index} with {delay_ms}ms delay using PyAV")
+
+            in_container = av.open(input_path)
+            in_stream = next((s for s in in_container.streams if s.type == "audio"), None)
+            if in_stream is None:
+                raise ValueError("No audio stream in input")
+
+            with av.open(output_path, "w", format="webm") as out_container:
+                out_stream = out_container.add_stream("libopus", rate=OPUS_STANDARD_SAMPLE_RATE)
+                out_stream.bit_rate = OPUS_DEFAULT_BIT_RATE
+                graph = av.filter.Graph()
+
+                abuf_args = (
+                    f"time_base=1/{OPUS_STANDARD_SAMPLE_RATE}:"
+                    f"sample_rate={OPUS_STANDARD_SAMPLE_RATE}:"
+                    f"sample_fmt=s16:"
+                    f"channel_layout=stereo"
+                )
+                src = graph.add("abuffer", args=abuf_args, name="src")
+                aresample_f = graph.add("aresample", args="async=1", name="ares")
+                delays_arg = f"{delay_ms}|{delay_ms}"
+                adelay_f = graph.add("adelay", args=f"delays={delays_arg}:all=1", name="delay")
+                sink = graph.add("abuffersink", name="sink")
+
+                src.link_to(aresample_f)
+                aresample_f.link_to(adelay_f)
+                adelay_f.link_to(sink)
+                graph.configure()
+
+                resampler = AudioResampler(
+                    format="s16", layout="stereo", rate=OPUS_STANDARD_SAMPLE_RATE
+                )
+
+                for frame in in_container.decode(in_stream):
+                    # Check for cancellation periodically
+                    now = time.time()
+                    if now - last_check >= DISCONNECT_CHECK_INTERVAL:
+                        if cancelled.is_set():
+                            logger.info("Cancelled during processing, exiting early")
+                            in_container.close()
+                            return {"size": 0, "cancelled": True}
+                        last_check = now
+
+                    out_frames = resampler.resample(frame) or []
+                    for rframe in out_frames:
+                        rframe.sample_rate = OPUS_STANDARD_SAMPLE_RATE
+                        rframe.time_base = Fraction(1, OPUS_STANDARD_SAMPLE_RATE)
+                        src.push(rframe)
+
+                        while True:
+                            try:
+                                f_out = sink.pull()
+                            except Exception:
+                                break
+                            f_out.sample_rate = OPUS_STANDARD_SAMPLE_RATE
+                            f_out.time_base = Fraction(1, OPUS_STANDARD_SAMPLE_RATE)
+                            for packet in out_stream.encode(f_out):
+                                out_container.mux(packet)
+
+                # Flush filter graph
+                src.push(None)
+                while True:
+                    try:
+                        f_out = sink.pull()
+                    except Exception:
+                        break
+                    f_out.sample_rate = OPUS_STANDARD_SAMPLE_RATE
+                    f_out.time_base = Fraction(1, OPUS_STANDARD_SAMPLE_RATE)
+                    for packet in out_stream.encode(f_out):
+                        out_container.mux(packet)
+
+                # Flush encoder
+                for packet in out_stream.encode(None):
+                    out_container.mux(packet)
+
+            in_container.close()
+
+            file_size = os.path.getsize(output_path)
+            logger.info(f"Padding complete: {file_size} bytes")
+
+            logger.info("Uploading padded track to S3")
+
+            with open(output_path, "rb") as f:
+                upload_response = requests.put(req.output_url, data=f, timeout=S3_TIMEOUT)
+
+            upload_response.raise_for_status()
+            logger.info(f"Upload complete: {file_size} bytes")
+
+            return {"size": file_size}
+
+        finally:
+            if input_path and os.path.exists(input_path):
+                try:
+                    os.unlink(input_path)
+                except Exception as e:
+                    logger.warning(f"Failed to cleanup input file: {e}")
+            if output_path and os.path.exists(output_path):
+                try:
+                    os.unlink(output_path)
+                except Exception as e:
+                    logger.warning(f"Failed to cleanup output file: {e}")
+            try:
+                os.rmdir(temp_dir)
+            except Exception as e:
+                logger.warning(f"Failed to cleanup temp directory: {e}")
+
+    return web_app
+

server/pyproject.toml

@@ -8,7 +8,7 @@ readme = "README.md"
 dependencies = [
     "aiohttp>=3.9.0",
     "aiohttp-cors>=0.7.0",
-    "av>=10.0.0",
+    "av>=15.0.0",
     "requests>=2.31.0",
     "aiortc>=1.5.0",
     "sortedcontainers>=2.4.0",

server/reflector/hatchet/constants.py

@@ -37,5 +37,5 @@ LLM_RATE_LIMIT_PER_SECOND = 10
 TIMEOUT_SHORT = 60  # Quick operations: API calls, DB updates
 TIMEOUT_MEDIUM = 120  # Single LLM calls, waveform generation
 TIMEOUT_LONG = 180  # Action items (larger context LLM)
-TIMEOUT_AUDIO = 300  # Audio processing: padding, mixdown
+TIMEOUT_AUDIO = 720  # Audio processing: padding, mixdown
 TIMEOUT_HEAVY = 600  # Transcription, fan-out LLM tasks

server/reflector/hatchet/workflows/daily_multitrack_pipeline.py

@@ -561,13 +561,27 @@ async def mixdown_tracks(input: PipelineInput, ctx: Context) -> MixdownResult:
 
     Path(output_path).unlink(missing_ok=True)
 
+    duration = duration_ms_callback_capture_container[0]
+
     async with fresh_db_connection():
-        from reflector.db.transcripts import transcripts_controller  # noqa: PLC0415
+        from reflector.db.transcripts import (  # noqa: PLC0415
+            TranscriptDuration,
+            transcripts_controller,
+        )
 
         transcript = await transcripts_controller.get_by_id(input.transcript_id)
         if transcript:
             await transcripts_controller.update(
-                transcript, {"audio_location": "storage"}
+                transcript, {"audio_location": "storage", "duration": duration}
+            )
+
+            duration_data = TranscriptDuration(duration=duration)
+            await append_event_and_broadcast(
+                input.transcript_id,
+                transcript,
+                "DURATION",
+                duration_data,
+                logger=logger,
             )
 
     ctx.log(f"mixdown_tracks complete: uploaded {file_size} bytes to {storage_path}")
@@ -1095,7 +1109,7 @@ async def identify_action_items(
 
 
 @daily_multitrack_pipeline.task(
-    parents=[generate_waveform, generate_title, generate_recap, identify_action_items],
+    parents=[process_tracks, generate_title, generate_recap, identify_action_items],
     execution_timeout=timedelta(seconds=TIMEOUT_SHORT),
     retries=3,
 )
@@ -1108,12 +1122,8 @@ async def finalize(input: PipelineInput, ctx: Context) -> FinalizeResult:
     """
     ctx.log("finalize: saving transcript and setting status to 'ended'")
 
-    mixdown_result = ctx.task_output(mixdown_tracks)
     track_result = ctx.task_output(process_tracks)
 
-    duration = mixdown_result.duration
-    all_words = track_result.all_words
-
     # Cleanup temporary padded S3 files (deferred until finalize for semantic parity with Celery)
     created_padded_files = track_result.created_padded_files
     if created_padded_files:
@@ -1133,7 +1143,6 @@ async def finalize(input: PipelineInput, ctx: Context) -> FinalizeResult:
 
     async with fresh_db_connection():
         from reflector.db.transcripts import (  # noqa: PLC0415
-            TranscriptDuration,
             TranscriptText,
             transcripts_controller,
         )
@@ -1142,8 +1151,6 @@ async def finalize(input: PipelineInput, ctx: Context) -> FinalizeResult:
         if transcript is None:
             raise ValueError(f"Transcript {input.transcript_id} not found in database")
 
-        merged_transcript = TranscriptType(words=all_words, translation=None)
-
         await append_event_and_broadcast(
             input.transcript_id,
             transcript,
@@ -1155,21 +1162,15 @@ async def finalize(input: PipelineInput, ctx: Context) -> FinalizeResult:
             logger=logger,
         )
 
-        # Save duration and clear workflow_run_id (workflow completed successfully)
+        # Clear workflow_run_id (workflow completed successfully)
-        # Note: title/long_summary/short_summary already saved by their callbacks
+        # Note: title/long_summary/short_summary/duration already saved by their callbacks
         await transcripts_controller.update(
             transcript,
             {
-                "duration": duration,
                 "workflow_run_id": None,  # Clear on success - no need to resume
             },
         )
 
-        duration_data = TranscriptDuration(duration=duration)
-        await append_event_and_broadcast(
-            input.transcript_id, transcript, "DURATION", duration_data, logger=logger
-        )
-
         await set_status_and_broadcast(input.transcript_id, "ended", logger=logger)
 
         ctx.log(

server/reflector/hatchet/workflows/padding_workflow.py

@@ -0,0 +1,165 @@
+"""
+Hatchet child workflow: PaddingWorkflow
+Handles individual audio track padding via Modal.com backend.
+"""
+
+from datetime import timedelta
+
+import av
+from hatchet_sdk import Context
+from pydantic import BaseModel
+
+from reflector.hatchet.client import HatchetClientManager
+from reflector.hatchet.constants import TIMEOUT_AUDIO
+from reflector.hatchet.workflows.models import PadTrackResult
+from reflector.logger import logger
+from reflector.utils.audio_constants import PRESIGNED_URL_EXPIRATION_SECONDS
+from reflector.utils.audio_padding import extract_stream_start_time_from_container
+
+
+class PaddingInput(BaseModel):
+    """Input for individual track padding."""
+
+    track_index: int
+    s3_key: str
+    bucket_name: str
+    transcript_id: str
+
+
+hatchet = HatchetClientManager.get_client()
+
+padding_workflow = hatchet.workflow(
+    name="PaddingWorkflow", input_validator=PaddingInput
+)
+
+
+@padding_workflow.task(execution_timeout=timedelta(seconds=TIMEOUT_AUDIO), retries=3)
+async def pad_track(input: PaddingInput, ctx: Context) -> PadTrackResult:
+    """Pad audio track with silence based on WebM container start_time."""
+    ctx.log(f"pad_track: track {input.track_index}, s3_key={input.s3_key}")
+    logger.info(
+        "[Hatchet] pad_track",
+        track_index=input.track_index,
+        s3_key=input.s3_key,
+        transcript_id=input.transcript_id,
+    )
+
+    try:
+        # Create fresh storage instance to avoid aioboto3 fork issues
+        from reflector.settings import settings  # noqa: PLC0415
+        from reflector.storage.storage_aws import AwsStorage  # noqa: PLC0415
+
+        storage = AwsStorage(
+            aws_bucket_name=settings.TRANSCRIPT_STORAGE_AWS_BUCKET_NAME,
+            aws_region=settings.TRANSCRIPT_STORAGE_AWS_REGION,
+            aws_access_key_id=settings.TRANSCRIPT_STORAGE_AWS_ACCESS_KEY_ID,
+            aws_secret_access_key=settings.TRANSCRIPT_STORAGE_AWS_SECRET_ACCESS_KEY,
+        )
+
+        source_url = await storage.get_file_url(
+            input.s3_key,
+            operation="get_object",
+            expires_in=PRESIGNED_URL_EXPIRATION_SECONDS,
+            bucket=input.bucket_name,
+        )
+
+        # Extract start_time to determine if padding needed
+        with av.open(source_url) as in_container:
+            if in_container.duration:
+                try:
+                    duration = timedelta(seconds=in_container.duration // 1_000_000)
+                    ctx.log(
+                        f"pad_track: track {input.track_index}, duration={duration}"
+                    )
+                except (ValueError, TypeError, OverflowError) as e:
+                    ctx.log(
+                        f"pad_track: track {input.track_index}, duration error: {str(e)}"
+                    )
+
+            start_time_seconds = extract_stream_start_time_from_container(
+                in_container, input.track_index, logger=logger
+            )
+
+        if start_time_seconds <= 0:
+            logger.info(
+                f"Track {input.track_index} requires no padding",
+                track_index=input.track_index,
+            )
+            return PadTrackResult(
+                padded_key=input.s3_key,
+                bucket_name=input.bucket_name,
+                size=0,
+                track_index=input.track_index,
+            )
+
+        storage_path = f"file_pipeline_hatchet/{input.transcript_id}/tracks/padded_{input.track_index}.webm"
+
+        # Presign PUT URL for output (Modal will upload directly)
+        output_url = await storage.get_file_url(
+            storage_path,
+            operation="put_object",
+            expires_in=PRESIGNED_URL_EXPIRATION_SECONDS,
+        )
+
+        import httpx  # noqa: PLC0415
+
+        from reflector.processors.audio_padding_modal import (  # noqa: PLC0415
+            AudioPaddingModalProcessor,
+        )
+
+        try:
+            processor = AudioPaddingModalProcessor()
+            result = await processor.pad_track(
+                track_url=source_url,
+                output_url=output_url,
+                start_time_seconds=start_time_seconds,
+                track_index=input.track_index,
+            )
+            file_size = result.size
+
+            ctx.log(f"pad_track: Modal returned size={file_size}")
+        except httpx.HTTPStatusError as e:
+            error_detail = e.response.text if hasattr(e.response, "text") else str(e)
+            logger.error(
+                "[Hatchet] Modal padding HTTP error",
+                transcript_id=input.transcript_id,
+                track_index=input.track_index,
+                status_code=e.response.status_code if hasattr(e, "response") else None,
+                error=error_detail,
+                exc_info=True,
+            )
+            raise Exception(
+                f"Modal padding failed: HTTP {e.response.status_code}"
+            ) from e
+        except httpx.TimeoutException as e:
+            logger.error(
+                "[Hatchet] Modal padding timeout",
+                transcript_id=input.transcript_id,
+                track_index=input.track_index,
+                error=str(e),
+                exc_info=True,
+            )
+            raise Exception("Modal padding timeout") from e
+
+        logger.info(
+            "[Hatchet] pad_track complete",
+            track_index=input.track_index,
+            padded_key=storage_path,
+        )
+
+        return PadTrackResult(
+            padded_key=storage_path,
+            bucket_name=None,  # None = use default transcript storage bucket
+            size=file_size,
+            track_index=input.track_index,
+        )
+
+    except Exception as e:
+        logger.error(
+            "[Hatchet] pad_track failed",
+            transcript_id=input.transcript_id,
+            track_index=input.track_index,
+            error=str(e),
+            exc_info=True,
+        )
+        raise

server/reflector/hatchet/workflows/track_processing.py

@@ -14,9 +14,7 @@ Hatchet workers run in forked processes; fresh imports per task ensure
 storage/DB connections are not shared across forks.
 """
 
-import tempfile
 from datetime import timedelta
-from pathlib import Path
 
 import av
 from hatchet_sdk import Context
@@ -27,10 +25,7 @@ from reflector.hatchet.constants import TIMEOUT_AUDIO, TIMEOUT_HEAVY
 from reflector.hatchet.workflows.models import PadTrackResult, TranscribeTrackResult
 from reflector.logger import logger
 from reflector.utils.audio_constants import PRESIGNED_URL_EXPIRATION_SECONDS
-from reflector.utils.audio_padding import (
+from reflector.utils.audio_padding import extract_stream_start_time_from_container
-    apply_audio_padding_to_file,
-    extract_stream_start_time_from_container,
-)
 
 
 class TrackInput(BaseModel):
@@ -83,63 +78,44 @@ async def pad_track(input: TrackInput, ctx: Context) -> PadTrackResult:
         )
 
         with av.open(source_url) as in_container:
-            if in_container.duration:
-                try:
-                    duration = timedelta(seconds=in_container.duration // 1_000_000)
-                    ctx.log(
-                        f"pad_track: track {input.track_index}, duration={duration}"
-                    )
-                except Exception:
-                    ctx.log(f"pad_track: track {input.track_index}, duration=ERROR")
-
             start_time_seconds = extract_stream_start_time_from_container(
                 in_container, input.track_index, logger=logger
             )
 
-            # If no padding needed, return original S3 key
+        # If no padding needed, return original S3 key
-            if start_time_seconds <= 0:
+        if start_time_seconds <= 0:
-                logger.info(
+            logger.info(
-                    f"Track {input.track_index} requires no padding",
+                f"Track {input.track_index} requires no padding",
-                    track_index=input.track_index,
+                track_index=input.track_index,
-                )
+            )
-                return PadTrackResult(
+            return PadTrackResult(
-                    padded_key=input.s3_key,
+                padded_key=input.s3_key,
-                    bucket_name=input.bucket_name,
+                bucket_name=input.bucket_name,
-                    size=0,
+                size=0,
-                    track_index=input.track_index,
+                track_index=input.track_index,
-                )
+            )
 
-            with tempfile.NamedTemporaryFile(suffix=".webm", delete=False) as temp_file:
+        storage_path = f"file_pipeline_hatchet/{input.transcript_id}/tracks/padded_{input.track_index}.webm"
-                temp_path = temp_file.name
 
-            try:
+        # Presign PUT URL for output (Modal uploads directly)
-                apply_audio_padding_to_file(
+        output_url = await storage.get_file_url(
-                    in_container,
+            storage_path,
-                    temp_path,
+            operation="put_object",
-                    start_time_seconds,
+            expires_in=PRESIGNED_URL_EXPIRATION_SECONDS,
-                    input.track_index,
+        )
-                    logger=logger,
-                )
 
-                file_size = Path(temp_path).stat().st_size
+        from reflector.processors.audio_padding_modal import (  # noqa: PLC0415
-                storage_path = f"file_pipeline_hatchet/{input.transcript_id}/tracks/padded_{input.track_index}.webm"
+            AudioPaddingModalProcessor,
+        )
 
-                logger.info(
+        processor = AudioPaddingModalProcessor()
-                    f"About to upload padded track",
+        result = await processor.pad_track(
-                    key=storage_path,
+            track_url=source_url,
-                    size=file_size,
+            output_url=output_url,
-                )
+            start_time_seconds=start_time_seconds,
-
+            track_index=input.track_index,
-                with open(temp_path, "rb") as padded_file:
+        )
-                    await storage.put_file(storage_path, padded_file)
+        file_size = result.size
-
-                logger.info(
-                    f"Uploaded padded track to S3",
-                    key=storage_path,
-                    size=file_size,
-                )
-            finally:
-                Path(temp_path).unlink(missing_ok=True)
 
         ctx.log(f"pad_track complete: track {input.track_index} -> {storage_path}")
         logger.info(

server/reflector/processors/audio_padding_modal.py

@@ -0,0 +1,113 @@
+"""
+Modal.com backend for audio padding.
+"""
+
+import asyncio
+import os
+
+import httpx
+from pydantic import BaseModel
+
+from reflector.hatchet.constants import TIMEOUT_AUDIO
+from reflector.logger import logger
+
+
+class PaddingResponse(BaseModel):
+    size: int
+    cancelled: bool = False
+
+
+class AudioPaddingModalProcessor:
+    """Audio padding processor using Modal.com CPU backend via HTTP."""
+
+    def __init__(
+        self, padding_url: str | None = None, modal_api_key: str | None = None
+    ):
+        self.padding_url = padding_url or os.getenv("PADDING_URL")
+        if not self.padding_url:
+            raise ValueError(
+                "PADDING_URL required to use AudioPaddingModalProcessor. "
+                "Set PADDING_URL environment variable or pass padding_url parameter."
+            )
+
+        self.modal_api_key = modal_api_key or os.getenv("MODAL_API_KEY")
+
+    async def pad_track(
+        self,
+        track_url: str,
+        output_url: str,
+        start_time_seconds: float,
+        track_index: int,
+    ) -> PaddingResponse:
+        """Pad audio track with silence via Modal backend.
+
+        Args:
+            track_url: Presigned GET URL for source audio track
+            output_url: Presigned PUT URL for output WebM
+            start_time_seconds: Amount of silence to prepend
+            track_index: Track index for logging
+        """
+        if not track_url:
+            raise ValueError("track_url cannot be empty")
+        if start_time_seconds <= 0:
+            raise ValueError(
+                f"start_time_seconds must be positive, got {start_time_seconds}"
+            )
+
+        log = logger.bind(track_index=track_index, padding_seconds=start_time_seconds)
+        log.info("Sending Modal padding HTTP request")
+
+        url = f"{self.padding_url}/pad"
+
+        headers = {}
+        if self.modal_api_key:
+            headers["Authorization"] = f"Bearer {self.modal_api_key}"
+
+        try:
+            async with httpx.AsyncClient(timeout=TIMEOUT_AUDIO) as client:
+                response = await client.post(
+                    url,
+                    headers=headers,
+                    json={
+                        "track_url": track_url,
+                        "output_url": output_url,
+                        "start_time_seconds": start_time_seconds,
+                        "track_index": track_index,
+                    },
+                    follow_redirects=True,
+                )
+
+                if response.status_code != 200:
+                    error_body = response.text
+                    log.error(
+                        "Modal padding API error",
+                        status_code=response.status_code,
+                        error_body=error_body,
+                    )
+
+                response.raise_for_status()
+                result = response.json()
+
+            # Check if work was cancelled
+            if result.get("cancelled"):
+                log.warning("Modal padding was cancelled by disconnect detection")
+                raise asyncio.CancelledError(
+                    "Padding cancelled due to client disconnect"
+                )
+
+            log.info("Modal padding complete", size=result["size"])
+            return PaddingResponse(**result)
+        except asyncio.CancelledError:
+            log.warning(
+                "Modal padding cancelled (Hatchet timeout, disconnect detected on Modal side)"
+            )
+            raise
+        except httpx.TimeoutException as e:
+            log.error("Modal padding timeout", error=str(e), exc_info=True)
+            raise Exception(f"Modal padding timeout: {e}") from e
+        except httpx.HTTPStatusError as e:
+            log.error("Modal padding HTTP error", error=str(e), exc_info=True)
+            raise Exception(f"Modal padding HTTP error: {e}") from e
+        except Exception as e:
+            log.error("Modal padding unexpected error", error=str(e), exc_info=True)
+            raise

server/reflector/settings.py

@@ -98,6 +98,10 @@ class Settings(BaseSettings):
     # Diarization: local pyannote.audio
     DIARIZATION_PYANNOTE_AUTH_TOKEN: str | None = None
 
+    # Audio Padding (Modal.com backend)
+    PADDING_URL: str | None = None
+    PADDING_MODAL_API_KEY: str | None = None
+
     # Sentry
     SENTRY_DSN: str | None = None
 

server/reflector/utils/audio_constants.py

@@ -5,7 +5,9 @@ Used by both Hatchet workflows and Celery pipelines for consistent audio encodin
 """
 
 # Opus codec settings
+# ref B0F71CE8-FC59-4AA5-8414-DAFB836DB711
 OPUS_STANDARD_SAMPLE_RATE = 48000
+# ref B0F71CE8-FC59-4AA5-8414-DAFB836DB711
 OPUS_DEFAULT_BIT_RATE = 128000  # 128kbps for good speech quality
 
 # S3 presigned URL expiration

server/uv.lock

@@ -159,21 +159,20 @@ wheels = [
 
 [[package]]
 name = "aiortc"
-version = "1.13.0"
+version = "1.14.0"
 source = { registry = "https://pypi.org/simple" }
 dependencies = [
     { name = "aioice" },
     { name = "av" },
-    { name = "cffi" },
     { name = "cryptography" },
     { name = "google-crc32c" },
     { name = "pyee" },
     { name = "pylibsrtp" },
     { name = "pyopenssl" },
 ]
-sdist = { url = "https://files.pythonhosted.org/packages/62/03/bc947d74c548e0c17cf94e5d5bdacaed0ee9e5b2bb7b8b8cf1ac7a7c01ec/aiortc-1.13.0.tar.gz", hash = "sha256:5d209975c22d0910fb5a0f0e2caa828f2da966c53580f7c7170ac3a16a871620", size = 1179894 }
+sdist = { url = "https://files.pythonhosted.org/packages/51/9c/4e027bfe0195de0442da301e2389329496745d40ae44d2d7c4571c4290ce/aiortc-1.14.0.tar.gz", hash = "sha256:adc8a67ace10a085721e588e06a00358ed8eaf5f6b62f0a95358ff45628dd762", size = 1180864 }
 wheels = [
-    { url = "https://files.pythonhosted.org/packages/87/29/765633cab5f1888890f5f172d1d53009b9b14e079cdfa01a62d9896a9ea9/aiortc-1.13.0-py3-none-any.whl", hash = "sha256:9ccccec98796f6a96bd1c3dd437a06da7e0f57521c96bd56e4b965a91b03a0a0", size = 92910 },
+    { url = "https://files.pythonhosted.org/packages/57/ab/31646a49209568cde3b97eeade0d28bb78b400e6645c56422c101df68932/aiortc-1.14.0-py3-none-any.whl", hash = "sha256:4b244d7e482f4e1f67e685b3468269628eca1ec91fa5b329ab517738cfca086e", size = 93183 },
 ]
 
 [[package]]
@@ -327,28 +326,24 @@ wheels = [
 
 [[package]]
 name = "av"
-version = "14.4.0"
+version = "16.1.0"
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+sdist = { url = "https://files.pythonhosted.org/packages/78/cd/3a83ffbc3cc25b39721d174487fb0d51a76582f4a1703f98e46170ce83d4/av-16.1.0.tar.gz", hash = "sha256:a094b4fd87a3721dacf02794d3d2c82b8d712c85b9534437e82a8a978c175ffd", size = 4285203 }
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-    { url = "https://files.pythonhosted.org/packages/a7/8a/957da1f581aa1faa9a5dfa8b47ca955edb47f2b76b949950933b457bfa1d/av-14.4.0-cp312-cp312-musllinux_1_2_x86_64.whl", hash = "sha256:1661efbe9d975f927b8512d654704223d936f39016fad2ddab00aee7c40f412c", size = 37521981 },
-    { url = "https://files.pythonhosted.org/packages/28/76/3f1cf0568592f100fd68eb40ed8c491ce95ca3c1378cc2d4c1f6d1bd295d/av-14.4.0-cp312-cp312-win_amd64.whl", hash = "sha256:fbbeef1f421a3461086853d6464ad5526b56ffe8ccb0ab3fd0a1f121dfbf26ad", size = 27925944 },
 ]
 
 [[package]]
@@ -3267,7 +3262,7 @@ requires-dist = [
     { name = "aiohttp-cors", specifier = ">=0.7.0" },
     { name = "aiortc", specifier = ">=1.5.0" },
     { name = "alembic", specifier = ">=1.11.3" },
-    { name = "av", specifier = ">=10.0.0" },
+    { name = "av", specifier = ">=15.0.0" },
     { name = "celery", specifier = ">=5.3.4" },
     { name = "databases", extras = ["aiosqlite", "asyncpg"], specifier = ">=0.7.0" },
     { name = "fastapi", extras = ["standard"], specifier = ">=0.100.1" },

www/app/(app)/rooms/page.tsx

@@ -302,10 +302,10 @@ export default function RoomsList() {
         return;
       }
 
-      const platform: "whereby" | "daily" | null =
+      const platform: "whereby" | "daily" =
         room.platform === "whereby" || room.platform === "daily"
           ? room.platform
-          : null;
+          : "daily";
 
       const roomData = {
         name: room.name,

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

@@ -16,6 +16,7 @@ import {
 import { useError } from "../../../../(errors)/errorContext";
 import { useRouter } from "next/navigation";
 import { Box, Grid } from "@chakra-ui/react";
+import { parseNonEmptyString } from "../../../../lib/utils";
 
 export type TranscriptCorrect = {
   params: Promise<{
@@ -25,8 +26,7 @@ export type TranscriptCorrect = {
 
 export default function TranscriptCorrect(props: TranscriptCorrect) {
   const params = use(props.params);
-
+  const transcriptId = parseNonEmptyString(params.transcriptId);
-  const { transcriptId } = params;
 
   const updateTranscriptMutation = useTranscriptUpdate();
   const transcript = useTranscriptGet(transcriptId);

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

@@ -9,7 +9,9 @@ import React, { useEffect, useState, use } from "react";
 import FinalSummary from "./finalSummary";
 import TranscriptTitle from "../transcriptTitle";
 import Player from "../player";
+import { useWebSockets } from "../useWebSockets";
 import { useRouter } from "next/navigation";
+import { parseNonEmptyString } from "../../../lib/utils";
 import {
   Box,
   Flex,
@@ -30,7 +32,7 @@ type TranscriptDetails = {
 
 export default function TranscriptDetails(details: TranscriptDetails) {
   const params = use(details.params);
-  const transcriptId = params.transcriptId;
+  const transcriptId = parseNonEmptyString(params.transcriptId);
   const router = useRouter();
   const statusToRedirect = [
     "idle",
@@ -49,6 +51,7 @@ export default function TranscriptDetails(details: TranscriptDetails) {
     transcriptId,
     waiting || mp3.audioDeleted === true,
   );
+  useWebSockets(transcriptId);
   const useActiveTopic = useState<Topic | null>(null);
   const [finalSummaryElement, setFinalSummaryElement] =
     useState<HTMLDivElement | null>(null);

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

@@ -10,6 +10,7 @@ import {
 } from "@chakra-ui/react";
 import { useRouter } from "next/navigation";
 import { useTranscriptGet } from "../../../../lib/apiHooks";
+import { parseNonEmptyString } from "../../../../lib/utils";
 
 type TranscriptProcessing = {
   params: Promise<{
@@ -19,7 +20,7 @@ type TranscriptProcessing = {
 
 export default function TranscriptProcessing(details: TranscriptProcessing) {
   const params = use(details.params);
-  const transcriptId = params.transcriptId;
+  const transcriptId = parseNonEmptyString(params.transcriptId);
   const router = useRouter();
 
   const transcript = useTranscriptGet(transcriptId);

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

@@ -12,6 +12,7 @@ import { Box, Text, Grid, Heading, VStack, Flex } from "@chakra-ui/react";
 import LiveTrancription from "../../liveTranscription";
 import { useTranscriptGet } from "../../../../lib/apiHooks";
 import { TranscriptStatus } from "../../../../lib/transcript";
+import { parseNonEmptyString } from "../../../../lib/utils";
 
 type TranscriptDetails = {
   params: Promise<{
@@ -21,13 +22,14 @@ type TranscriptDetails = {
 
 const TranscriptRecord = (details: TranscriptDetails) => {
   const params = use(details.params);
-  const transcript = useTranscriptGet(params.transcriptId);
+  const transcriptId = parseNonEmptyString(params.transcriptId);
+  const transcript = useTranscriptGet(transcriptId);
   const [transcriptStarted, setTranscriptStarted] = useState(false);
   const useActiveTopic = useState<Topic | null>(null);
 
-  const webSockets = useWebSockets(params.transcriptId);
+  const webSockets = useWebSockets(transcriptId);
 
-  const mp3 = useMp3(params.transcriptId, true);
+  const mp3 = useMp3(transcriptId, true);
 
   const router = useRouter();
 

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

@@ -7,6 +7,7 @@ import useMp3 from "../../useMp3";
 import { Center, VStack, Text, Heading } from "@chakra-ui/react";
 import FileUploadButton from "../../fileUploadButton";
 import { useTranscriptGet } from "../../../../lib/apiHooks";
+import { parseNonEmptyString } from "../../../../lib/utils";
 
 type TranscriptUpload = {
   params: Promise<{
@@ -16,12 +17,13 @@ type TranscriptUpload = {
 
 const TranscriptUpload = (details: TranscriptUpload) => {
   const params = use(details.params);
-  const transcript = useTranscriptGet(params.transcriptId);
+  const transcriptId = parseNonEmptyString(params.transcriptId);
+  const transcript = useTranscriptGet(transcriptId);
   const [transcriptStarted, setTranscriptStarted] = useState(false);
 
-  const webSockets = useWebSockets(params.transcriptId);
+  const webSockets = useWebSockets(transcriptId);
 
-  const mp3 = useMp3(params.transcriptId, true);
+  const mp3 = useMp3(transcriptId, true);
 
   const router = useRouter();
 

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

@@ -1,5 +1,6 @@
 import { useState } from "react";
 import type { components } from "../../reflector-api";
+import { parseMaybeNonEmptyString } from "../../lib/utils";
 
 type UpdateTranscript = components["schemas"]["UpdateTranscript"];
 type GetTranscriptWithParticipants =
@@ -32,7 +33,7 @@ const TranscriptTitle = (props: TranscriptTitle) => {
   const [isEditing, setIsEditing] = useState(false);
   const updateTranscriptMutation = useTranscriptUpdate();
   const participantsQuery = useTranscriptParticipants(
-    props.transcript?.id || null,
+    props.transcript?.id ? parseMaybeNonEmptyString(props.transcript.id) : null,
   );
 
   const updateTitle = async (newTitle: string, transcriptId: string) => {

www/app/(app)/transcripts/useMp3.ts

@@ -1,5 +1,6 @@
 import { useEffect, useState } from "react";
 import { useTranscriptGet } from "../../lib/apiHooks";
+import { parseMaybeNonEmptyString } from "../../lib/utils";
 import { useAuth } from "../../lib/AuthProvider";
 import { API_URL } from "../../lib/apiClient";
 
@@ -27,7 +28,7 @@ const useMp3 = (transcriptId: string, waiting?: boolean): Mp3Response => {
     data: transcript,
     isLoading: transcriptMetadataLoading,
     error: transcriptError,
-  } = useTranscriptGet(later ? null : transcriptId);
+  } = useTranscriptGet(later ? null : parseMaybeNonEmptyString(transcriptId));
 
   const [serviceWorker, setServiceWorker] =
     useState<ServiceWorkerRegistration | null>(null);

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

@@ -1,6 +1,7 @@
 import type { components } from "../../reflector-api";
 type Participant = components["schemas"]["Participant"];
 import { useTranscriptParticipants } from "../../lib/apiHooks";
+import { parseMaybeNonEmptyString } from "../../lib/utils";
 
 type ErrorParticipants = {
   error: Error;
@@ -32,7 +33,7 @@ const useParticipants = (transcriptId: string): UseParticipants => {
     isLoading: loading,
     error,
     refetch,
-  } = useTranscriptParticipants(transcriptId || null);
+  } = useTranscriptParticipants(parseMaybeNonEmptyString(transcriptId));
 
   // Type-safe return based on state
   if (error) {

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

@@ -1,5 +1,6 @@
 import type { components } from "../../reflector-api";
 import { useTranscriptTopicsWithWordsPerSpeaker } from "../../lib/apiHooks";
+import { parseMaybeNonEmptyString } from "../../lib/utils";
 
 type GetTranscriptTopicWithWordsPerSpeaker =
   components["schemas"]["GetTranscriptTopicWithWordsPerSpeaker"];
@@ -38,7 +39,7 @@ const useTopicWithWords = (
     error,
     refetch,
   } = useTranscriptTopicsWithWordsPerSpeaker(
-    transcriptId || null,
+    parseMaybeNonEmptyString(transcriptId),
     topicId || null,
   );
 

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

@@ -1,5 +1,6 @@
 import { useTranscriptTopics } from "../../lib/apiHooks";
 import type { components } from "../../reflector-api";
+import { parseMaybeNonEmptyString } from "../../lib/utils";
 
 type GetTranscriptTopic = components["schemas"]["GetTranscriptTopic"];
 
@@ -10,7 +11,11 @@ type TranscriptTopics = {
 };
 
 const useTopics = (id: string): TranscriptTopics => {
-  const { data: topics, isLoading: loading, error } = useTranscriptTopics(id);
+  const {
+    data: topics,
+    isLoading: loading,
+    error,
+  } = useTranscriptTopics(parseMaybeNonEmptyString(id));
 
   return {
     topics: topics || null,

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

@@ -1,5 +1,6 @@
 import type { components } from "../../reflector-api";
 import { useTranscriptWaveform } from "../../lib/apiHooks";
+import { parseMaybeNonEmptyString } from "../../lib/utils";
 
 type AudioWaveform = components["schemas"]["AudioWaveform"];
 
@@ -14,7 +15,7 @@ const useWaveform = (id: string, skip: boolean): AudioWaveFormResponse => {
     data: waveform,
     isLoading: loading,
     error,
-  } = useTranscriptWaveform(skip ? null : id);
+  } = useTranscriptWaveform(skip ? null : parseMaybeNonEmptyString(id));
 
   return {
     waveform: waveform || null,

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

@@ -7,6 +7,12 @@ type GetTranscriptSegmentTopic =
   components["schemas"]["GetTranscriptSegmentTopic"];
 import { useQueryClient } from "@tanstack/react-query";
 import { $api, WEBSOCKET_URL } from "../../lib/apiClient";
+import {
+  invalidateTranscript,
+  invalidateTranscriptTopics,
+  invalidateTranscriptWaveform,
+} from "../../lib/apiHooks";
+import { NonEmptyString } from "../../lib/utils";
 
 export type UseWebSockets = {
   transcriptTextLive: string;
@@ -369,15 +375,10 @@ export const useWebSockets = (transcriptId: string | null): UseWebSockets => {
             });
             console.debug("TOPIC event:", message.data);
             // Invalidate topics query to sync with WebSocket data
-            queryClient.invalidateQueries({
+            invalidateTranscriptTopics(
-              queryKey: $api.queryOptions(
+              queryClient,
-                "get",
+              transcriptId as NonEmptyString,
-                "/v1/transcripts/{transcript_id}/topics",
+            );
-                {
-                  params: { path: { transcript_id: transcriptId } },
-                },
-              ).queryKey,
-            });
             break;
 
           case "FINAL_SHORT_SUMMARY":
@@ -388,15 +389,7 @@ export const useWebSockets = (transcriptId: string | null): UseWebSockets => {
             if (message.data) {
               setFinalSummary(message.data);
               // Invalidate transcript query to sync summary
-              queryClient.invalidateQueries({
+              invalidateTranscript(queryClient, transcriptId as NonEmptyString);
-                queryKey: $api.queryOptions(
-                  "get",
-                  "/v1/transcripts/{transcript_id}",
-                  {
-                    params: { path: { transcript_id: transcriptId } },
-                  },
-                ).queryKey,
-              });
             }
             break;
 
@@ -405,15 +398,7 @@ export const useWebSockets = (transcriptId: string | null): UseWebSockets => {
             if (message.data) {
               setTitle(message.data.title);
               // Invalidate transcript query to sync title
-              queryClient.invalidateQueries({
+              invalidateTranscript(queryClient, transcriptId as NonEmptyString);
-                queryKey: $api.queryOptions(
-                  "get",
-                  "/v1/transcripts/{transcript_id}",
-                  {
-                    params: { path: { transcript_id: transcriptId } },
-                  },
-                ).queryKey,
-              });
             }
             break;
 
@@ -424,6 +409,10 @@ export const useWebSockets = (transcriptId: string | null): UseWebSockets => {
             );
             if (message.data) {
               setWaveForm(message.data.waveform);
+              invalidateTranscriptWaveform(
+                queryClient,
+                transcriptId as NonEmptyString,
+              );
             }
             break;
           case "DURATION":

www/app/[roomName]/MeetingSelection.tsx

@@ -26,7 +26,7 @@ import { useRouter } from "next/navigation";
 import { formatDateTime, formatStartedAgo } from "../lib/timeUtils";
 import MeetingMinimalHeader from "../components/MeetingMinimalHeader";
 import { NonEmptyString } from "../lib/utils";
-import { MeetingId } from "../lib/types";
+import { MeetingId, assertMeetingId } from "../lib/types";
 
 type Meeting = components["schemas"]["Meeting"];
 
@@ -315,7 +315,9 @@ export default function MeetingSelection({
                         variant="outline"
                         colorScheme="red"
                         size="md"
-                        onClick={() => handleEndMeeting(meeting.id)}
+                        onClick={() =>
+                          handleEndMeeting(assertMeetingId(meeting.id))
+                        }
                         loading={deactivateMeetingMutation.isPending}
                       >
                         <Icon as={LuX} me={2} />
@@ -460,7 +462,9 @@ export default function MeetingSelection({
                           variant="outline"
                           colorScheme="red"
                           size="md"
-                          onClick={() => handleEndMeeting(meeting.id)}
+                          onClick={() =>
+                            handleEndMeeting(assertMeetingId(meeting.id))
+                          }
                           loading={deactivateMeetingMutation.isPending}
                         >
                           <Icon as={LuX} me={2} />

www/app/lib/apiHooks.ts

@@ -6,6 +6,7 @@ import { QueryClient, useQueryClient } from "@tanstack/react-query";
 import type { components } from "../reflector-api";
 import { useAuth } from "./AuthProvider";
 import { MeetingId } from "./types";
+import { NonEmptyString } from "./utils";
 
 /*
  * XXX error types returned from the hooks are not always correct; declared types are ValidationError but real type could be string or any other
@@ -103,7 +104,7 @@ export function useTranscriptProcess() {
   });
 }
 
-export function useTranscriptGet(transcriptId: string | null) {
+export function useTranscriptGet(transcriptId: NonEmptyString | null) {
   return $api.useQuery(
     "get",
     "/v1/transcripts/{transcript_id}",
@@ -120,6 +121,16 @@ export function useTranscriptGet(transcriptId: string | null) {
   );
 }
 
+export const invalidateTranscript = (
+  queryClient: QueryClient,
+  transcriptId: NonEmptyString,
+) =>
+  queryClient.invalidateQueries({
+    queryKey: $api.queryOptions("get", "/v1/transcripts/{transcript_id}", {
+      params: { path: { transcript_id: transcriptId } },
+    }).queryKey,
+  });
+
 export function useRoomGet(roomId: string | null) {
   const { isAuthenticated } = useAuthReady();
 
@@ -297,7 +308,7 @@ export function useTranscriptUploadAudio() {
   );
 }
 
-export function useTranscriptWaveform(transcriptId: string | null) {
+export function useTranscriptWaveform(transcriptId: NonEmptyString | null) {
   return $api.useQuery(
     "get",
     "/v1/transcripts/{transcript_id}/audio/waveform",
@@ -312,7 +323,21 @@ export function useTranscriptWaveform(transcriptId: string | null) {
   );
 }
 
-export function useTranscriptMP3(transcriptId: string | null) {
+export const invalidateTranscriptWaveform = (
+  queryClient: QueryClient,
+  transcriptId: NonEmptyString,
+) =>
+  queryClient.invalidateQueries({
+    queryKey: $api.queryOptions(
+      "get",
+      "/v1/transcripts/{transcript_id}/audio/waveform",
+      {
+        params: { path: { transcript_id: transcriptId } },
+      },
+    ).queryKey,
+  });
+
+export function useTranscriptMP3(transcriptId: NonEmptyString | null) {
   const { isAuthenticated } = useAuthReady();
 
   return $api.useQuery(
@@ -329,7 +354,7 @@ export function useTranscriptMP3(transcriptId: string | null) {
   );
 }
 
-export function useTranscriptTopics(transcriptId: string | null) {
+export function useTranscriptTopics(transcriptId: NonEmptyString | null) {
   return $api.useQuery(
     "get",
     "/v1/transcripts/{transcript_id}/topics",
@@ -344,7 +369,23 @@ export function useTranscriptTopics(transcriptId: string | null) {
   );
 }
 
-export function useTranscriptTopicsWithWords(transcriptId: string | null) {
+export const invalidateTranscriptTopics = (
+  queryClient: QueryClient,
+  transcriptId: NonEmptyString,
+) =>
+  queryClient.invalidateQueries({
+    queryKey: $api.queryOptions(
+      "get",
+      "/v1/transcripts/{transcript_id}/topics",
+      {
+        params: { path: { transcript_id: transcriptId } },
+      },
+    ).queryKey,
+  });
+
+export function useTranscriptTopicsWithWords(
+  transcriptId: NonEmptyString | null,
+) {
   const { isAuthenticated } = useAuthReady();
 
   return $api.useQuery(
@@ -362,7 +403,7 @@ export function useTranscriptTopicsWithWords(transcriptId: string | null) {
 }
 
 export function useTranscriptTopicsWithWordsPerSpeaker(
-  transcriptId: string | null,
+  transcriptId: NonEmptyString | null,
   topicId: string | null,
 ) {
   const { isAuthenticated } = useAuthReady();
@@ -384,7 +425,7 @@ export function useTranscriptTopicsWithWordsPerSpeaker(
   );
 }
 
-export function useTranscriptParticipants(transcriptId: string | null) {
+export function useTranscriptParticipants(transcriptId: NonEmptyString | null) {
   const { isAuthenticated } = useAuthReady();
 
   return $api.useQuery(