zombie meetings ignore (no-mistakes)

2025-12-21 12:49:06 +00:00 · 2025-11-25 19:35:50 -05:00
27 changed files with 382 additions and 2285 deletions
--- a/CHANGELOG.md
+++ b/CHANGELOG.md
@@ -1,34 +1,5 @@
 # Changelog
 ## [0.22.2](https://github.com/Monadical-SAS/reflector/compare/v0.22.1...v0.22.2) (2025-12-02)
 ### Bug Fixes
 * daily auto refresh fix ([#755](https://github.com/Monadical-SAS/reflector/issues/755)) ([fe47c46](https://github.com/Monadical-SAS/reflector/commit/fe47c46489c5aa0cc538109f7559cc9accb35c01))
 * Skip mixdown for multitrack ([#760](https://github.com/Monadical-SAS/reflector/issues/760)) ([b51b7aa](https://github.com/Monadical-SAS/reflector/commit/b51b7aa9176c1a53ba57ad99f5e976c804a1e80c))
 ## [0.22.1](https://github.com/Monadical-SAS/reflector/compare/v0.22.0...v0.22.1) (2025-11-27)
 ### Bug Fixes
 * participants update from daily ([#749](https://github.com/Monadical-SAS/reflector/issues/749)) ([7f0b728](https://github.com/Monadical-SAS/reflector/commit/7f0b728991c1b9f9aae702c96297eae63b561ef5))
 ## [0.22.0](https://github.com/Monadical-SAS/reflector/compare/v0.21.0...v0.22.0) (2025-11-26)
 ### Features
 * Multitrack segmentation ([#747](https://github.com/Monadical-SAS/reflector/issues/747)) ([d63040e](https://github.com/Monadical-SAS/reflector/commit/d63040e2fdc07e7b272e85a39eb2411cd6a14798))
 ## [0.21.0](https://github.com/Monadical-SAS/reflector/compare/v0.20.0...v0.21.0) (2025-11-26)
 ### Features
 * add transcript format parameter to GET endpoint ([#709](https://github.com/Monadical-SAS/reflector/issues/709)) ([f6ca075](https://github.com/Monadical-SAS/reflector/commit/f6ca07505f34483b02270a2ef3bd809e9d2e1045))
 ## [0.20.0](https://github.com/Monadical-SAS/reflector/compare/v0.19.0...v0.20.0) (2025-11-25)
--- a/docs/transcript.md
+++ b/docs/transcript.md
@@ -1,241 +0,0 @@
 # Transcript Formats
 The Reflector API provides multiple output formats for transcript data through the `transcript_format` query parameter on the GET `/v1/transcripts/{id}` endpoint.
 ## Overview
 When retrieving a transcript, you can specify the desired format using the `transcript_format` query parameter. The API supports four formats optimized for different use cases:
 - **text** - Plain text with speaker names (default)
 - **text-timestamped** - Timestamped text with speaker names
 - **webvtt-named** - WebVTT subtitle format with participant names
 - **json** - Structured JSON segments with full metadata
 All formats include participant information when available, resolving speaker IDs to actual names.
 ## Query Parameter Usage
 ```
 GET /v1/transcripts/{id}?transcript_format={format}
 ```
 ### Parameters
 - `transcript_format` (optional): The desired output format
  - Type: `"text" | "text-timestamped" | "webvtt-named" | "json"`
  - Default: `"text"`
 ## Format Descriptions
 ### Text Format (`text`)
 **Use case:** Simple, human-readable transcript for display or export.
 **Format:** Speaker names followed by their dialogue, one line per segment.
 **Example:**
 ```
 John Smith: Hello everyone
 Jane Doe: Hi there
 John Smith: How are you today?
 ```
 **Request:**
 ```bash
 GET /v1/transcripts/{id}?transcript_format=text
 ```
 **Response:**
 ```json
 {
  "id": "transcript_123",
  "name": "Meeting Recording",
  "transcript_format": "text",
  "transcript": "John Smith: Hello everyone\nJane Doe: Hi there\nJohn Smith: How are you today?",
  "participants": [
    {"id": "p1", "speaker": 0, "name": "John Smith"},
    {"id": "p2", "speaker": 1, "name": "Jane Doe"}
  ],
  ...
 }
 ```
 ### Text Timestamped Format (`text-timestamped`)
 **Use case:** Transcript with timing information for navigation or reference.
 **Format:** `[MM:SS]` timestamp prefix before each speaker and dialogue.
 **Example:**
 ```
 [00:00] John Smith: Hello everyone
 [00:05] Jane Doe: Hi there
 [00:12] John Smith: How are you today?
 ```
 **Request:**
 ```bash
 GET /v1/transcripts/{id}?transcript_format=text-timestamped
 ```
 **Response:**
 ```json
 {
  "id": "transcript_123",
  "name": "Meeting Recording",
  "transcript_format": "text-timestamped",
  "transcript": "[00:00] John Smith: Hello everyone\n[00:05] Jane Doe: Hi there\n[00:12] John Smith: How are you today?",
  "participants": [
    {"id": "p1", "speaker": 0, "name": "John Smith"},
    {"id": "p2", "speaker": 1, "name": "Jane Doe"}
  ],
  ...
 }
 ```
 ### WebVTT Named Format (`webvtt-named`)
 **Use case:** Subtitle files for video players, accessibility tools, or video editing.
 **Format:** Standard WebVTT subtitle format with voice tags using participant names.
 **Example:**
 ```
 WEBVTT
 00:00:00.000 --> 00:00:05.000
 <v John Smith>Hello everyone
 00:00:05.000 --> 00:00:12.000
 <v Jane Doe>Hi there
 00:00:12.000 --> 00:00:18.000
 <v John Smith>How are you today?
 ```
 **Request:**
 ```bash
 GET /v1/transcripts/{id}?transcript_format=webvtt-named
 ```
 **Response:**
 ```json
 {
  "id": "transcript_123",
  "name": "Meeting Recording",
  "transcript_format": "webvtt-named",
  "transcript": "WEBVTT\n\n00:00:00.000 --> 00:00:05.000\n<v John Smith>Hello everyone\n\n...",
  "participants": [
    {"id": "p1", "speaker": 0, "name": "John Smith"},
    {"id": "p2", "speaker": 1, "name": "Jane Doe"}
  ],
  ...
 }
 ```
 ### JSON Format (`json`)
 **Use case:** Programmatic access with full timing and speaker metadata.
 **Format:** Array of segment objects with speaker information, text content, and precise timing.
 **Example:**
 ```json
 [
  {
    "speaker": 0,
    "speaker_name": "John Smith",
    "text": "Hello everyone",
    "start": 0.0,
    "end": 5.0
  },
  {
    "speaker": 1,
    "speaker_name": "Jane Doe",
    "text": "Hi there",
    "start": 5.0,
    "end": 12.0
  },
  {
    "speaker": 0,
    "speaker_name": "John Smith",
    "text": "How are you today?",
    "start": 12.0,
    "end": 18.0
  }
 ]
 ```
 **Request:**
 ```bash
 GET /v1/transcripts/{id}?transcript_format=json
 ```
 **Response:**
 ```json
 {
  "id": "transcript_123",
  "name": "Meeting Recording",
  "transcript_format": "json",
  "transcript": [
    {
      "speaker": 0,
      "speaker_name": "John Smith",
      "text": "Hello everyone",
      "start": 0.0,
      "end": 5.0
    },
    {
      "speaker": 1,
      "speaker_name": "Jane Doe",
      "text": "Hi there",
      "start": 5.0,
      "end": 12.0
    }
  ],
  "participants": [
    {"id": "p1", "speaker": 0, "name": "John Smith"},
    {"id": "p2", "speaker": 1, "name": "Jane Doe"}
  ],
  ...
 }
 ```
 ## Response Structure
 All formats return the same base transcript metadata with an additional `transcript_format` field and format-specific `transcript` field:
 ### Common Fields
 - `id`: Transcript identifier
 - `user_id`: Owner user ID (if authenticated)
 - `name`: Transcript name
 - `status`: Processing status
 - `locked`: Whether transcript is locked for editing
 - `duration`: Total duration in seconds
 - `title`: Auto-generated or custom title
 - `short_summary`: Brief summary
 - `long_summary`: Detailed summary
 - `created_at`: Creation timestamp
 - `share_mode`: Access control setting
 - `source_language`: Original audio language
 - `target_language`: Translation target language
 - `reviewed`: Whether transcript has been reviewed
 - `meeting_id`: Associated meeting ID (if applicable)
 - `source_kind`: Source type (live, file, room)
 - `room_id`: Associated room ID (if applicable)
 - `audio_deleted`: Whether audio has been deleted
 - `participants`: Array of participant objects with speaker mappings
 ### Format-Specific Fields
 - `transcript_format`: The format identifier (discriminator field)
 - `transcript`: The formatted transcript content (string for text/webvtt formats, array for json format)
 ## Speaker Name Resolution
 All formats resolve speaker IDs to participant names when available:
 - If a participant exists for the speaker ID, their name is used
 - If no participant exists, a default name like "Speaker 0" is generated
 - Speaker IDs are integers (0, 1, 2, etc.) assigned during diarization
--- a/server/reflector/dailyco_api/responses.py
+++ b/server/reflector/dailyco_api/responses.py
@@ -68,7 +68,7 @@ class MeetingParticipant(BaseModel):
    Reference: https://docs.daily.co/reference/rest-api/meetings/get-meeting-participants
    """
-    user_id: NonEmptyString | None = Field(None, description="User identifier")
+    user_id: NonEmptyString = Field(description="User identifier")
    participant_id: NonEmptyString = Field(description="Participant session identifier")
    user_name: NonEmptyString | None = Field(None, description="User display name")
    join_time: int = Field(description="Join timestamp (Unix epoch seconds)")
--- a/server/reflector/db/recordings.py
+++ b/server/reflector/db/recordings.py
@@ -35,15 +35,8 @@ class Recording(BaseModel):
    status: Literal["pending", "processing", "completed", "failed"] = "pending"
    meeting_id: str | None = None
    # for multitrack reprocessing
    # track_keys can be empty list [] if recording finished but no audio was captured (silence/muted)
    # None means not a multitrack recording, [] means multitrack with no tracks
    track_keys: list[str] | None = None
    @property
    def is_multitrack(self) -> bool:
        """True if recording has separate audio tracks (1+ tracks counts as multitrack)."""
        return self.track_keys is not None and len(self.track_keys) > 0
 class RecordingController:
    async def create(self, recording: Recording):
--- a/server/reflector/pipelines/main_multitrack_pipeline.py
+++ b/server/reflector/pipelines/main_multitrack_pipeline.py
@@ -9,10 +9,7 @@ from av.audio.resampler import AudioResampler
 from celery import chain, shared_task
 from reflector.asynctask import asynctask
 from reflector.dailyco_api import MeetingParticipantsResponse
 from reflector.db.transcripts import (
    Transcript,
    TranscriptParticipant,
    TranscriptStatus,
    TranscriptWaveform,
    transcripts_controller,
@@ -31,14 +28,8 @@ from reflector.processors import AudioFileWriterProcessor
 from reflector.processors.audio_waveform_processor import AudioWaveformProcessor
 from reflector.processors.types import TitleSummary
 from reflector.processors.types import Transcript as TranscriptType
 from reflector.settings import settings
 from reflector.storage import Storage, get_transcripts_storage
 from reflector.utils.daily import (
    filter_cam_audio_tracks,
    parse_daily_recording_filename,
 )
 from reflector.utils.string import NonEmptyString
 from reflector.video_platforms.factory import create_platform_client
 # Audio encoding constants
 OPUS_STANDARD_SAMPLE_RATE = 48000
@@ -503,90 +494,6 @@ class PipelineMainMultitrack(PipelineMainBase):
                transcript=transcript, event="WAVEFORM", data=waveform
            )
    async def update_participants_from_daily(
        self, transcript: Transcript, track_keys: list[str]
    ) -> None:
        """Update transcript participants with user_id and names from Daily.co API."""
        if not transcript.recording_id:
            return
        try:
            async with create_platform_client("daily") as daily_client:
                id_to_name = {}
                id_to_user_id = {}
                try:
                    rec_details = await daily_client.get_recording(
                        transcript.recording_id
                    )
                    mtg_session_id = rec_details.mtgSessionId
                    if mtg_session_id:
                        try:
                            payload: MeetingParticipantsResponse = (
                                await daily_client.get_meeting_participants(
                                    mtg_session_id
                                )
                            )
                            for p in payload.data:
                                pid = p.participant_id
                                name = p.user_name
                                user_id = p.user_id
                                if name:
                                    id_to_name[pid] = name
                                if user_id:
                                    id_to_user_id[pid] = user_id
                        except Exception as e:
                            self.logger.warning(
                                "Failed to fetch Daily meeting participants",
                                error=str(e),
                                mtg_session_id=mtg_session_id,
                                exc_info=True,
                            )
                    else:
                        self.logger.warning(
                            "No mtgSessionId found for recording; participant names may be generic",
                            recording_id=transcript.recording_id,
                        )
                except Exception as e:
                    self.logger.warning(
                        "Failed to fetch Daily recording details",
                        error=str(e),
                        recording_id=transcript.recording_id,
                        exc_info=True,
                    )
                    return
                cam_audio_keys = filter_cam_audio_tracks(track_keys)
                for idx, key in enumerate(cam_audio_keys):
                    try:
                        parsed = parse_daily_recording_filename(key)
                        participant_id = parsed.participant_id
                    except ValueError as e:
                        self.logger.error(
                            "Failed to parse Daily recording filename",
                            error=str(e),
                            key=key,
                            exc_info=True,
                        )
                        continue
                    default_name = f"Speaker {idx}"
                    name = id_to_name.get(participant_id, default_name)
                    user_id = id_to_user_id.get(participant_id)
                    participant = TranscriptParticipant(
                        id=participant_id, speaker=idx, name=name, user_id=user_id
                    )
                    await transcripts_controller.upsert_participant(
                        transcript, participant
                    )
        except Exception as e:
            self.logger.warning(
                "Failed to map participant names", error=str(e), exc_info=True
            )
    async def process(self, bucket_name: str, track_keys: list[str]):
        transcript = await self.get_transcript()
        async with self.transaction():
@@ -595,12 +502,9 @@ class PipelineMainMultitrack(PipelineMainBase):
                {
                    "events": [],
                    "topics": [],
                    "participants": [],
                },
            )
        await self.update_participants_from_daily(transcript, track_keys)
        source_storage = get_transcripts_storage()
        transcript_storage = source_storage
@@ -632,21 +536,11 @@ class PipelineMainMultitrack(PipelineMainBase):
        transcript.data_path.mkdir(parents=True, exist_ok=True)
        if settings.SKIP_MIXDOWN:
            self.logger.warning(
                "SKIP_MIXDOWN enabled: Skipping mixdown and waveform generation. "
                "UI will have no audio playback or waveform.",
                num_tracks=len(padded_track_urls),
                transcript_id=transcript.id,
            )
        else:
        mp3_writer = AudioFileWriterProcessor(
            path=str(transcript.audio_mp3_filename),
            on_duration=self.on_duration,
        )
-            await self.mixdown_tracks(
+        await self.mixdown_tracks(padded_track_urls, mp3_writer, offsets_seconds=None)
                padded_track_urls, mp3_writer, offsets_seconds=None
            )
        await mp3_writer.flush()
        if not transcript.audio_mp3_filename.exists():
@@ -661,9 +555,7 @@ class PipelineMainMultitrack(PipelineMainBase):
            await transcript_storage.put_file(storage_path, mp3_file)
        mp3_url = await transcript_storage.get_file_url(storage_path)
-            await transcripts_controller.update(
+        await transcripts_controller.update(transcript, {"audio_location": "storage"})
                transcript, {"audio_location": "storage"}
            )
        self.logger.info(
            f"Uploaded mixed audio to storage",
--- a/server/reflector/processors/types.py
+++ b/server/reflector/processors/types.py
@@ -1,7 +1,6 @@
 import io
 import re
 import tempfile
 from collections import defaultdict
 from pathlib import Path
 from typing import Annotated, TypedDict
@@ -17,17 +16,6 @@ class DiarizationSegment(TypedDict):
 PUNC_RE = re.compile(r"[.;:?!…]")
 SENTENCE_END_RE = re.compile(r"[.?!…]$")
 # Max segment length for words_to_segments() - breaks on any punctuation (. ; : ? ! …)
 # when segment exceeds this limit. Used for non-multitrack recordings.
 MAX_SEGMENT_CHARS = 120
 # Max segment length for words_to_segments_by_sentence() - only breaks on sentence-ending
 # punctuation (. ? ! …) when segment exceeds this limit. Higher threshold allows complete
 # sentences in multitrack recordings where speakers overlap.
 # similar number to server/reflector/processors/transcript_liner.py
 MAX_SENTENCE_SEGMENT_CHARS = 1000
 class AudioFile(BaseModel):
@@ -88,6 +76,7 @@ def words_to_segments(words: list[Word]) -> list[TranscriptSegment]:
    # but separate if the speaker changes, or if the punctuation is a . , ; : ? !
    segments = []
    current_segment = None
    MAX_SEGMENT_LENGTH = 120
    for word in words:
        if current_segment is None:
@@ -117,7 +106,7 @@ def words_to_segments(words: list[Word]) -> list[TranscriptSegment]:
        current_segment.end = word.end
        have_punc = PUNC_RE.search(word.text)
-        if have_punc and (len(current_segment.text) > MAX_SEGMENT_CHARS):
+        if have_punc and (len(current_segment.text) > MAX_SEGMENT_LENGTH):
            segments.append(current_segment)
            current_segment = None
@@ -127,70 +116,6 @@ def words_to_segments(words: list[Word]) -> list[TranscriptSegment]:
    return segments
 def words_to_segments_by_sentence(words: list[Word]) -> list[TranscriptSegment]:
    """Group words by speaker, then split into sentences.
    For multitrack recordings where words from different speakers are interleaved
    by timestamp, this function first groups all words by speaker, then creates
    segments based on sentence boundaries within each speaker's words.
    This produces cleaner output than words_to_segments() which breaks on every
    speaker change, resulting in many tiny segments when speakers overlap.
    """
    if not words:
        return []
    # Group words by speaker, preserving order within each speaker
    by_speaker: dict[int, list[Word]] = defaultdict(list)
    for w in words:
        by_speaker[w.speaker].append(w)
    segments: list[TranscriptSegment] = []
    for speaker, speaker_words in by_speaker.items():
        current_text = ""
        current_start: float | None = None
        current_end: float = 0.0
        for word in speaker_words:
            if current_start is None:
                current_start = word.start
            current_text += word.text
            current_end = word.end
            # Check for sentence end or max length
            is_sentence_end = SENTENCE_END_RE.search(word.text.strip())
            is_too_long = len(current_text) >= MAX_SENTENCE_SEGMENT_CHARS
            if is_sentence_end or is_too_long:
                segments.append(
                    TranscriptSegment(
                        text=current_text,
                        start=current_start,
                        end=current_end,
                        speaker=speaker,
                    )
                )
                current_text = ""
                current_start = None
        # Flush remaining words for this speaker
        if current_text and current_start is not None:
            segments.append(
                TranscriptSegment(
                    text=current_text,
                    start=current_start,
                    end=current_end,
                    speaker=speaker,
                )
            )
    # Sort segments by start time
    segments.sort(key=lambda s: s.start)
    return segments
 class Transcript(BaseModel):
    translation: str | None = None
    words: list[Word] = []
@@ -229,9 +154,7 @@ class Transcript(BaseModel):
            word.start += offset
            word.end += offset
-    def as_segments(self, is_multitrack: bool = False) -> list[TranscriptSegment]:
+    def as_segments(self) -> list[TranscriptSegment]:
        if is_multitrack:
            return words_to_segments_by_sentence(self.words)
        return words_to_segments(self.words)
--- a/server/reflector/schemas/transcript_formats.py
+++ b/server/reflector/schemas/transcript_formats.py
@@ -1,17 +0,0 @@
 """Schema definitions for transcript format types and segments."""
 from typing import Literal
 from pydantic import BaseModel
 TranscriptFormat = Literal["text", "text-timestamped", "webvtt-named", "json"]
 class TranscriptSegment(BaseModel):
    """A single transcript segment with speaker and timing information."""
    speaker: int
    speaker_name: str
    text: str
    start: float
    end: float
--- a/server/reflector/services/transcript_process.py
+++ b/server/reflector/services/transcript_process.py
@@ -7,7 +7,7 @@ This module provides result-based error handling that works in both contexts:
 """
 from dataclasses import dataclass
-from typing import Literal, Union, assert_never
+from typing import Literal, Union
 import celery
 from celery.result import AsyncResult
@@ -18,6 +18,7 @@ from reflector.pipelines.main_file_pipeline import task_pipeline_file_process
 from reflector.pipelines.main_multitrack_pipeline import (
    task_pipeline_multitrack_process,
 )
 from reflector.utils.match import absurd
 from reflector.utils.string import NonEmptyString
@@ -154,7 +155,7 @@ def dispatch_transcript_processing(config: ProcessingConfig) -> AsyncResult:
    elif isinstance(config, FileProcessingConfig):
        return task_pipeline_file_process.delay(transcript_id=config.transcript_id)
    else:
-        assert_never(config)
+        absurd(config)
 def task_is_scheduled_or_active(task_name: str, **kwargs):
--- a/server/reflector/settings.py
+++ b/server/reflector/settings.py
@@ -138,14 +138,6 @@ class Settings(BaseSettings):
    DAILY_WEBHOOK_UUID: str | None = (
        None  # Webhook UUID for this environment. Not used by production code
    )
    # Multitrack processing
    # SKIP_MIXDOWN: When True, skips audio mixdown and waveform generation.
    # Transcription still works using individual tracks. Useful for:
    # - Diagnosing OOM issues in mixdown
    # - Fast processing when audio playback is not needed
    # Note: UI will have no audio playback or waveform when enabled.
    SKIP_MIXDOWN: bool = True
    # Platform Configuration
    DEFAULT_VIDEO_PLATFORM: Platform = WHEREBY_PLATFORM
--- a/server/reflector/utils/daily.py
+++ b/server/reflector/utils/daily.py
@@ -64,11 +64,6 @@ def recording_lock_key(recording_id: NonEmptyString) -> NonEmptyString:
    return f"recording:{recording_id}"
 def filter_cam_audio_tracks(track_keys: list[str]) -> list[str]:
    """Filter track keys to cam-audio tracks only (skip screen-audio, etc.)."""
    return [k for k in track_keys if "cam-audio" in k]
 def extract_base_room_name(daily_room_name: DailyRoomName) -> NonEmptyString:
    """
    Extract base room name from Daily.co timestamped room name.
--- a/server/reflector/utils/match.py
+++ b/server/reflector/utils/match.py
@@ -0,0 +1,10 @@
 from typing import NoReturn
 def assert_exhaustiveness(x: NoReturn) -> NoReturn:
    """Provide an assertion at type-check time that this function is never called."""
    raise AssertionError(f"Invalid value: {x!r}")
 def absurd(x: NoReturn) -> NoReturn:
    return assert_exhaustiveness(x)
--- a/server/reflector/utils/transcript_formats.py
+++ b/server/reflector/utils/transcript_formats.py
@@ -1,133 +0,0 @@
 """Utilities for converting transcript data to various output formats."""
 import webvtt
 from reflector.db.transcripts import TranscriptParticipant, TranscriptTopic
 from reflector.processors.types import (
    Transcript as ProcessorTranscript,
 )
 from reflector.schemas.transcript_formats import TranscriptSegment
 from reflector.utils.webvtt import seconds_to_timestamp
 def get_speaker_name(
    speaker: int, participants: list[TranscriptParticipant] | None
 ) -> str:
    """Get participant name for speaker or default to 'Speaker N'."""
    if participants:
        for participant in participants:
            if participant.speaker == speaker:
                return participant.name
    return f"Speaker {speaker}"
 def format_timestamp_mmss(seconds: float | int) -> str:
    """Format seconds as MM:SS timestamp."""
    minutes = int(seconds // 60)
    secs = int(seconds % 60)
    return f"{minutes:02d}:{secs:02d}"
 def transcript_to_text(
    topics: list[TranscriptTopic],
    participants: list[TranscriptParticipant] | None,
    is_multitrack: bool = False,
 ) -> str:
    """Convert transcript topics to plain text with speaker names."""
    lines = []
    for topic in topics:
        if not topic.words:
            continue
        transcript = ProcessorTranscript(words=topic.words)
        segments = transcript.as_segments(is_multitrack)
        for segment in segments:
            speaker_name = get_speaker_name(segment.speaker, participants)
            text = segment.text.strip()
            lines.append(f"{speaker_name}: {text}")
    return "\n".join(lines)
 def transcript_to_text_timestamped(
    topics: list[TranscriptTopic],
    participants: list[TranscriptParticipant] | None,
    is_multitrack: bool = False,
 ) -> str:
    """Convert transcript topics to timestamped text with speaker names."""
    lines = []
    for topic in topics:
        if not topic.words:
            continue
        transcript = ProcessorTranscript(words=topic.words)
        segments = transcript.as_segments(is_multitrack)
        for segment in segments:
            speaker_name = get_speaker_name(segment.speaker, participants)
            timestamp = format_timestamp_mmss(segment.start)
            text = segment.text.strip()
            lines.append(f"[{timestamp}] {speaker_name}: {text}")
    return "\n".join(lines)
 def topics_to_webvtt_named(
    topics: list[TranscriptTopic],
    participants: list[TranscriptParticipant] | None,
    is_multitrack: bool = False,
 ) -> str:
    """Convert transcript topics to WebVTT format with participant names."""
    vtt = webvtt.WebVTT()
    for topic in topics:
        if not topic.words:
            continue
        transcript = ProcessorTranscript(words=topic.words)
        segments = transcript.as_segments(is_multitrack)
        for segment in segments:
            speaker_name = get_speaker_name(segment.speaker, participants)
            text = segment.text.strip()
            text = f"<v {speaker_name}>{text}"
            caption = webvtt.Caption(
                start=seconds_to_timestamp(segment.start),
                end=seconds_to_timestamp(segment.end),
                text=text,
            )
            vtt.captions.append(caption)
    return vtt.content
 def transcript_to_json_segments(
    topics: list[TranscriptTopic],
    participants: list[TranscriptParticipant] | None,
    is_multitrack: bool = False,
 ) -> list[TranscriptSegment]:
    """Convert transcript topics to a flat list of JSON segments."""
    result = []
    for topic in topics:
        if not topic.words:
            continue
        transcript = ProcessorTranscript(words=topic.words)
        segments = transcript.as_segments(is_multitrack)
        for segment in segments:
            speaker_name = get_speaker_name(segment.speaker, participants)
            result.append(
                TranscriptSegment(
                    speaker=segment.speaker,
                    speaker_name=speaker_name,
                    text=segment.text.strip(),
                    start=segment.start,
                    end=segment.end,
                )
            )
    return result
--- a/server/reflector/utils/webvtt.py
+++ b/server/reflector/utils/webvtt.py
@@ -13,7 +13,7 @@ VttTimestamp = Annotated[str, "vtt_timestamp"]
 WebVTTStr = Annotated[str, "webvtt_str"]
-def seconds_to_timestamp(seconds: Seconds) -> VttTimestamp:
+def _seconds_to_timestamp(seconds: Seconds) -> VttTimestamp:
    # lib doesn't do that
    hours = int(seconds // 3600)
    minutes = int((seconds % 3600) // 60)
@@ -37,8 +37,8 @@ def words_to_webvtt(words: list[Word]) -> WebVTTStr:
        text = f"<v Speaker{segment.speaker}>{text}"
        caption = webvtt.Caption(
-            start=seconds_to_timestamp(segment.start),
+            start=_seconds_to_timestamp(segment.start),
-            end=seconds_to_timestamp(segment.end),
+            end=_seconds_to_timestamp(segment.end),
            text=text,
        )
        vtt.captions.append(caption)
--- a/server/reflector/video_platforms/daily.py
+++ b/server/reflector/video_platforms/daily.py
@@ -173,16 +173,15 @@ class DailyClient(VideoPlatformClient):
        self,
        room_name: DailyRoomName,
        enable_recording: bool,
-        user_id: NonEmptyString | None = None,
+        user_id: str | None = None,
-        is_owner: bool = False,
+    ) -> str:
    ) -> NonEmptyString:
        properties = MeetingTokenProperties(
            room_name=room_name,
            user_id=user_id,
            start_cloud_recording=enable_recording,
-            enable_recording_ui=False,
+            enable_recording_ui=not enable_recording,
            is_owner=is_owner,
        )
        request = CreateMeetingTokenRequest(properties=properties)
        result = await self._api_client.create_meeting_token(request)
        return result.token
--- a/server/reflector/views/rooms.py
+++ b/server/reflector/views/rooms.py
@@ -248,7 +248,7 @@ async def rooms_create(
        ics_url=room.ics_url,
        ics_fetch_interval=room.ics_fetch_interval,
        ics_enabled=room.ics_enabled,
-        platform=room.platform or settings.DEFAULT_VIDEO_PLATFORM,
+        platform=room.platform,
    )
@@ -556,7 +556,6 @@ async def rooms_join_meeting(
            meeting.room_name,
            enable_recording=enable_recording,
            user_id=user_id,
            is_owner=user_id == room.user_id,
        )
        meeting = meeting.model_copy()
        meeting.room_url = add_query_param(meeting.room_url, "t", token)
--- a/server/reflector/views/transcripts.py
+++ b/server/reflector/views/transcripts.py
@@ -1,22 +1,14 @@
 from datetime import datetime, timedelta, timezone
-from typing import Annotated, Literal, Optional, assert_never
+from typing import Annotated, Literal, Optional
 from fastapi import APIRouter, Depends, HTTPException, Query
 from fastapi_pagination import Page
 from fastapi_pagination.ext.databases import apaginate
 from jose import jwt
-from pydantic import (
+from pydantic import AwareDatetime, BaseModel, Field, constr, field_serializer
    AwareDatetime,
    BaseModel,
    Discriminator,
    Field,
    constr,
    field_serializer,
 )
 import reflector.auth as auth
 from reflector.db import get_database
 from reflector.db.recordings import recordings_controller
 from reflector.db.search import (
    DEFAULT_SEARCH_LIMIT,
    SearchLimit,
@@ -39,14 +31,7 @@ from reflector.db.transcripts import (
 )
 from reflector.processors.types import Transcript as ProcessorTranscript
 from reflector.processors.types import Word
 from reflector.schemas.transcript_formats import TranscriptFormat, TranscriptSegment
 from reflector.settings import settings
 from reflector.utils.transcript_formats import (
    topics_to_webvtt_named,
    transcript_to_json_segments,
    transcript_to_text,
    transcript_to_text_timestamped,
 )
 from reflector.ws_manager import get_ws_manager
 from reflector.zulip import (
    InvalidMessageError,
@@ -61,14 +46,6 @@ ALGORITHM = "HS256"
 DOWNLOAD_EXPIRE_MINUTES = 60
 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
 def create_access_token(data: dict, expires_delta: timedelta):
    to_encode = data.copy()
    expire = datetime.now(timezone.utc) + expires_delta
@@ -111,84 +88,10 @@ class GetTranscriptMinimal(BaseModel):
    audio_deleted: bool | None = None
-class GetTranscriptWithParticipants(GetTranscriptMinimal):
+class GetTranscript(GetTranscriptMinimal):
    participants: list[TranscriptParticipant] | None
 class GetTranscriptWithText(GetTranscriptWithParticipants):
    """
    Transcript response with plain text format.
    Format: Speaker names followed by their dialogue, one line per segment.
    Example:
        John Smith: Hello everyone
        Jane Doe: Hi there
    """
    transcript_format: Literal["text"] = "text"
    transcript: str
 class GetTranscriptWithTextTimestamped(GetTranscriptWithParticipants):
    """
    Transcript response with timestamped text format.
    Format: [MM:SS] timestamp prefix before each speaker and dialogue.
    Example:
        [00:00] John Smith: Hello everyone
        [00:05] Jane Doe: Hi there
    """
    transcript_format: Literal["text-timestamped"] = "text-timestamped"
    transcript: str
 class GetTranscriptWithWebVTTNamed(GetTranscriptWithParticipants):
    """
    Transcript response in WebVTT subtitle format with participant names.
    Format: Standard WebVTT with voice tags using participant names.
    Example:
        WEBVTT
        00:00:00.000 --> 00:00:05.000
        <v John Smith>Hello everyone
    """
    transcript_format: Literal["webvtt-named"] = "webvtt-named"
    transcript: str
 class GetTranscriptWithJSON(GetTranscriptWithParticipants):
    """
    Transcript response as structured JSON segments.
    Format: Array of segment objects with speaker info, text, and timing.
    Example:
        [
            {
                "speaker": 0,
                "speaker_name": "John Smith",
                "text": "Hello everyone",
                "start": 0.0,
                "end": 5.0
            }
        ]
    """
    transcript_format: Literal["json"] = "json"
    transcript: list[TranscriptSegment]
 GetTranscript = Annotated[
    GetTranscriptWithText
    | GetTranscriptWithTextTimestamped
    | GetTranscriptWithWebVTTNamed
    | GetTranscriptWithJSON,
    Discriminator("transcript_format"),
 ]
 class CreateTranscript(BaseModel):
    name: str
    source_language: str = Field("en")
@@ -325,7 +228,7 @@ async def transcripts_search(
    )
-@router.post("/transcripts", response_model=GetTranscriptWithParticipants)
+@router.post("/transcripts", response_model=GetTranscript)
 async def transcripts_create(
    info: CreateTranscript,
    user: Annotated[Optional[auth.UserInfo], Depends(auth.current_user_optional)],
@@ -369,7 +272,7 @@ class GetTranscriptTopic(BaseModel):
    segments: list[GetTranscriptSegmentTopic] = []
    @classmethod
-    def from_transcript_topic(cls, topic: TranscriptTopic, is_multitrack: bool = False):
+    def from_transcript_topic(cls, topic: TranscriptTopic):
        if not topic.words:
            # In previous version, words were missing
            # Just output a segment with speaker 0
@@ -393,7 +296,7 @@ class GetTranscriptTopic(BaseModel):
                    start=segment.start,
                    speaker=segment.speaker,
                )
-                for segment in transcript.as_segments(is_multitrack)
+                for segment in transcript.as_segments()
            ]
        return cls(
            id=topic.id,
@@ -410,8 +313,8 @@ class GetTranscriptTopicWithWords(GetTranscriptTopic):
    words: list[Word] = []
    @classmethod
-    def from_transcript_topic(cls, topic: TranscriptTopic, is_multitrack: bool = False):
+    def from_transcript_topic(cls, topic: TranscriptTopic):
-        instance = super().from_transcript_topic(topic, is_multitrack)
+        instance = super().from_transcript_topic(topic)
        if topic.words:
            instance.words = topic.words
        return instance
@@ -426,8 +329,8 @@ class GetTranscriptTopicWithWordsPerSpeaker(GetTranscriptTopic):
    words_per_speaker: list[SpeakerWords] = []
    @classmethod
-    def from_transcript_topic(cls, topic: TranscriptTopic, is_multitrack: bool = False):
+    def from_transcript_topic(cls, topic: TranscriptTopic):
-        instance = super().from_transcript_topic(topic, is_multitrack)
+        instance = super().from_transcript_topic(topic)
        if topic.words:
            words_per_speakers = []
            # group words by speaker
@@ -459,76 +362,14 @@ class GetTranscriptTopicWithWordsPerSpeaker(GetTranscriptTopic):
 async def transcript_get(
    transcript_id: str,
    user: Annotated[Optional[auth.UserInfo], Depends(auth.current_user_optional)],
    transcript_format: TranscriptFormat = "text",
 ):
    user_id = user["sub"] if user else None
-    transcript = await transcripts_controller.get_by_id_for_http(
+    return await transcripts_controller.get_by_id_for_http(
        transcript_id, user_id=user_id
    )
    is_multitrack = await _get_is_multitrack(transcript)
-    base_data = {
+@router.patch("/transcripts/{transcript_id}", response_model=GetTranscript)
        "id": transcript.id,
        "user_id": transcript.user_id,
        "name": transcript.name,
        "status": transcript.status,
        "locked": transcript.locked,
        "duration": transcript.duration,
        "title": transcript.title,
        "short_summary": transcript.short_summary,
        "long_summary": transcript.long_summary,
        "created_at": transcript.created_at,
        "share_mode": transcript.share_mode,
        "source_language": transcript.source_language,
        "target_language": transcript.target_language,
        "reviewed": transcript.reviewed,
        "meeting_id": transcript.meeting_id,
        "source_kind": transcript.source_kind,
        "room_id": transcript.room_id,
        "audio_deleted": transcript.audio_deleted,
        "participants": transcript.participants,
    }
    if transcript_format == "text":
        return GetTranscriptWithText(
            **base_data,
            transcript_format="text",
            transcript=transcript_to_text(
                transcript.topics, transcript.participants, is_multitrack
            ),
        )
    elif transcript_format == "text-timestamped":
        return GetTranscriptWithTextTimestamped(
            **base_data,
            transcript_format="text-timestamped",
            transcript=transcript_to_text_timestamped(
                transcript.topics, transcript.participants, is_multitrack
            ),
        )
    elif transcript_format == "webvtt-named":
        return GetTranscriptWithWebVTTNamed(
            **base_data,
            transcript_format="webvtt-named",
            transcript=topics_to_webvtt_named(
                transcript.topics, transcript.participants, is_multitrack
            ),
        )
    elif transcript_format == "json":
        return GetTranscriptWithJSON(
            **base_data,
            transcript_format="json",
            transcript=transcript_to_json_segments(
                transcript.topics, transcript.participants, is_multitrack
            ),
        )
    else:
        assert_never(transcript_format)
@router.patch(
    "/transcripts/{transcript_id}", response_model=GetTranscriptWithParticipants
 )
 async def transcript_update(
    transcript_id: str,
    info: UpdateTranscript,
@@ -578,12 +419,9 @@ async def transcript_get_topics(
        transcript_id, user_id=user_id
    )
    is_multitrack = await _get_is_multitrack(transcript)
    # convert to GetTranscriptTopic
    return [
-        GetTranscriptTopic.from_transcript_topic(topic, is_multitrack)
+        GetTranscriptTopic.from_transcript_topic(topic) for topic in transcript.topics
        for topic in transcript.topics
    ]
@@ -600,11 +438,9 @@ async def transcript_get_topics_with_words(
        transcript_id, user_id=user_id
    )
    is_multitrack = await _get_is_multitrack(transcript)
    # convert to GetTranscriptTopicWithWords
    return [
-        GetTranscriptTopicWithWords.from_transcript_topic(topic, is_multitrack)
+        GetTranscriptTopicWithWords.from_transcript_topic(topic)
        for topic in transcript.topics
    ]
@@ -623,17 +459,13 @@ async def transcript_get_topics_with_words_per_speaker(
        transcript_id, user_id=user_id
    )
    is_multitrack = await _get_is_multitrack(transcript)
    # get the topic from the transcript
    topic = next((t for t in transcript.topics if t.id == topic_id), None)
    if not topic:
        raise HTTPException(status_code=404, detail="Topic not found")
    # convert to GetTranscriptTopicWithWordsPerSpeaker
-    return GetTranscriptTopicWithWordsPerSpeaker.from_transcript_topic(
+    return GetTranscriptTopicWithWordsPerSpeaker.from_transcript_topic(topic)
        topic, is_multitrack
    )
@router.post("/transcripts/{transcript_id}/zulip")
--- a/server/reflector/views/transcripts_process.py
+++ b/server/reflector/views/transcripts_process.py
@@ -1,4 +1,4 @@
-from typing import Annotated, Optional, assert_never
+from typing import Annotated, Optional
 from fastapi import APIRouter, Depends, HTTPException
 from pydantic import BaseModel
@@ -15,6 +15,7 @@ from reflector.services.transcript_process import (
    prepare_transcript_processing,
    validate_transcript_for_processing,
 )
 from reflector.utils.match import absurd
 router = APIRouter()
@@ -43,7 +44,7 @@ async def transcript_process(
    elif isinstance(validation, ValidationOk):
        pass
    else:
-        assert_never(validation)
+        absurd(validation)
    config = await prepare_transcript_processing(validation)
--- a/server/reflector/worker/process.py
+++ b/server/reflector/worker/process.py
@@ -2,7 +2,6 @@ import json
 import os
 import re
 from datetime import datetime, timezone
 from typing import List
 from urllib.parse import unquote
 import av
@@ -12,7 +11,7 @@ from celery import shared_task
 from celery.utils.log import get_task_logger
 from pydantic import ValidationError
-from reflector.dailyco_api import RecordingResponse
+from reflector.dailyco_api import MeetingParticipantsResponse
 from reflector.db.daily_participant_sessions import (
    DailyParticipantSession,
    daily_participant_sessions_controller,
@@ -22,6 +21,7 @@ from reflector.db.recordings import Recording, recordings_controller
 from reflector.db.rooms import rooms_controller
 from reflector.db.transcripts import (
    SourceKind,
    TranscriptParticipant,
    transcripts_controller,
 )
 from reflector.pipelines.main_file_pipeline import task_pipeline_file_process
@@ -38,7 +38,7 @@ from reflector.storage import get_transcripts_storage
 from reflector.utils.daily import (
    DailyRoomName,
    extract_base_room_name,
-    filter_cam_audio_tracks,
+    parse_daily_recording_filename,
    recording_lock_key,
 )
 from reflector.video_platforms.factory import create_platform_client
@@ -240,6 +240,8 @@ async def _process_multitrack_recording_inner(
        )
    meeting = await meetings_controller.get_by_room_name(daily_room_name)
    if not meeting:
        raise Exception(f"Meeting not found: {daily_room_name}")
    room_name_base = extract_base_room_name(daily_room_name)
@@ -247,9 +249,6 @@ async def _process_multitrack_recording_inner(
    if not room:
        raise Exception(f"Room not found: {room_name_base}")
    if not meeting:
        raise Exception(f"Meeting not found: {room_name_base}")
    logger.info(
        "Found existing Meeting for recording",
        meeting_id=meeting.id,
@@ -273,7 +272,15 @@ async def _process_multitrack_recording_inner(
    # else: Recording already exists; metadata set at creation time
    transcript = await transcripts_controller.get_by_recording_id(recording.id)
-    if not transcript:
+    if transcript:
        await transcripts_controller.update(
            transcript,
            {
                "topics": [],
                "participants": [],
            },
        )
    else:
        transcript = await transcripts_controller.add(
            "",
            source_kind=SourceKind.ROOM,
@@ -286,10 +293,79 @@ async def _process_multitrack_recording_inner(
            room_id=room.id,
        )
    try:
        async with create_platform_client("daily") as daily_client:
            id_to_name = {}
            id_to_user_id = {}
            try:
                rec_details = await daily_client.get_recording(recording_id)
                mtg_session_id = rec_details.mtgSessionId
                if mtg_session_id:
                    try:
                        payload: MeetingParticipantsResponse = (
                            await daily_client.get_meeting_participants(mtg_session_id)
                        )
                        for p in payload.data:
                            pid = p.participant_id
                            assert (
                                pid is not None
                            ), "panic! participant id cannot be None"
                            name = p.user_name
                            user_id = p.user_id
                            if name:
                                id_to_name[pid] = name
                            if user_id:
                                id_to_user_id[pid] = user_id
                    except Exception as e:
                        logger.warning(
                            "Failed to fetch Daily meeting participants",
                            error=str(e),
                            mtg_session_id=mtg_session_id,
                            exc_info=True,
                        )
                else:
                    logger.warning(
                        "No mtgSessionId found for recording; participant names may be generic",
                        recording_id=recording_id,
                    )
            except Exception as e:
                logger.warning(
                    "Failed to fetch Daily recording details",
                    error=str(e),
                    recording_id=recording_id,
                    exc_info=True,
                )
            for idx, key in enumerate(track_keys):
                try:
                    parsed = parse_daily_recording_filename(key)
                    participant_id = parsed.participant_id
                except ValueError as e:
                    logger.error(
                        "Failed to parse Daily recording filename",
                        error=str(e),
                        key=key,
                        exc_info=True,
                    )
                    continue
                default_name = f"Speaker {idx}"
                name = id_to_name.get(participant_id, default_name)
                user_id = id_to_user_id.get(participant_id)
                participant = TranscriptParticipant(
                    id=participant_id, speaker=idx, name=name, user_id=user_id
                )
                await transcripts_controller.upsert_participant(transcript, participant)
    except Exception as e:
        logger.warning("Failed to map participant names", error=str(e), exc_info=True)
    task_pipeline_multitrack_process.delay(
        transcript_id=transcript.id,
        bucket_name=bucket_name,
-        track_keys=filter_cam_audio_tracks(track_keys),
+        track_keys=track_keys,
    )
@@ -314,7 +390,7 @@ async def poll_daily_recordings():
    async with create_platform_client("daily") as daily_client:
        # latest 100. TODO cursor-based state
-        api_recordings: List[RecordingResponse] = await daily_client.list_recordings()
+        api_recordings = await daily_client.list_recordings()
    if not api_recordings:
        logger.debug(
@@ -345,13 +421,11 @@ async def poll_daily_recordings():
    for recording in missing_recordings:
        if not recording.tracks:
-            if recording.status == "finished":
+            assert recording.status != "finished", (
-                logger.warning(
+                f"Recording {recording.id} has status='finished' but no tracks. "
-                    "Finished recording has no tracks (no audio captured)",
+                f"Daily.co API guarantees finished recordings have tracks available. "
-                    recording_id=recording.id,
+                f"room_name={recording.room_name}"
                    room_name=recording.room_name,
            )
            else:
            logger.debug(
                "No tracks in recording yet",
                recording_id=recording.id,
@@ -371,6 +445,15 @@ async def poll_daily_recordings():
            )
            continue
        meeting = await meetings_controller.get_by_room_name(recording.room_name)
        if not meeting:
            logger.warning(
                "Skipping recording - no matching meeting",
                recording_id=recording.id,
                room_name=recording.room_name,
            )
            continue
        logger.info(
            "Queueing missing recording for processing",
            recording_id=recording.id,
--- a/server/tests/test_poll_daily_recordings.py
+++ b/server/tests/test_poll_daily_recordings.py
@@ -59,17 +59,37 @@ def mock_recording_response():
    ]
@pytest.fixture
 def mock_meeting():
    """Mock meeting object."""
    from reflector.db.meetings import Meeting
    return Meeting(
        id="meeting-123",
        room_name="test-room-20251118120000",
        room_url="https://daily.co/test-room",
        host_room_url="https://daily.co/test-room",
        start_date=datetime.now(timezone.utc),
        end_date=datetime.now(timezone.utc) + timedelta(hours=1),
        room_id="room-123",
        platform="daily",
    )
@pytest.mark.asyncio
@patch("reflector.worker.process.settings")
@patch("reflector.worker.process.create_platform_client")
@patch("reflector.worker.process.recordings_controller.get_by_ids")
@patch("reflector.worker.process.meetings_controller.get_by_room_name")
@patch("reflector.worker.process.process_multitrack_recording.delay")
 async def test_poll_daily_recordings_processes_missing_recordings(
    mock_process_delay,
    mock_get_meeting,
    mock_get_recordings,
    mock_create_client,
    mock_settings,
    mock_recording_response,
    mock_meeting,
 ):
    """Test that poll_daily_recordings queues processing for recordings not in DB."""
    mock_settings.DAILYCO_STORAGE_AWS_BUCKET_NAME = "test-bucket"
@@ -85,6 +105,9 @@ async def test_poll_daily_recordings_processes_missing_recordings(
    # Mock DB controller - no existing recordings
    mock_get_recordings.return_value = []
    # Mock meeting exists for all recordings
    mock_get_meeting.return_value = mock_meeting
    # Execute - call the unwrapped async function
    poll_fn = _get_poll_daily_recordings_fn()
    await poll_fn()
@@ -113,6 +136,48 @@ async def test_poll_daily_recordings_processes_missing_recordings(
    assert calls[1].kwargs["track_keys"] == ["track1.webm"]
@pytest.mark.asyncio
@patch("reflector.worker.process.settings")
@patch("reflector.worker.process.create_platform_client")
@patch("reflector.worker.process.recordings_controller.get_by_ids")
@patch("reflector.worker.process.meetings_controller.get_by_room_name")
@patch("reflector.worker.process.process_multitrack_recording.delay")
 async def test_poll_daily_recordings_skips_recordings_without_meeting(
    mock_process_delay,
    mock_get_meeting,
    mock_get_recordings,
    mock_create_client,
    mock_settings,
    mock_recording_response,
 ):
    """Test that poll_daily_recordings skips recordings without matching meeting."""
    mock_settings.DAILYCO_STORAGE_AWS_BUCKET_NAME = "test-bucket"
    # Mock Daily.co API client
    mock_daily_client = AsyncMock()
    mock_daily_client.list_recordings = AsyncMock(return_value=mock_recording_response)
    mock_create_client.return_value.__aenter__ = AsyncMock(
        return_value=mock_daily_client
    )
    mock_create_client.return_value.__aexit__ = AsyncMock()
    # Mock DB controller - no existing recordings
    mock_get_recordings.return_value = []
    # Mock no meeting found
    mock_get_meeting.return_value = None
    # Execute - call the unwrapped async function
    poll_fn = _get_poll_daily_recordings_fn()
    await poll_fn()
    # Verify Daily.co API was called
    assert mock_daily_client.list_recordings.call_count == 1
    # Verify NO processing was queued (no matching meetings)
    assert mock_process_delay.call_count == 0
@pytest.mark.asyncio
@patch("reflector.worker.process.settings")
@patch("reflector.worker.process.create_platform_client")
--- a/server/tests/test_processor_transcript_segment.py
+++ b/server/tests/test_processor_transcript_segment.py
@@ -159,78 +159,3 @@ def test_processor_transcript_segment():
    assert segments[3].start == 30.72
    assert segments[4].start == 31.56
    assert segments[5].start == 32.38
 def test_processor_transcript_segment_multitrack_interleaved():
    """Test as_segments(is_multitrack=True) with interleaved speakers.
    Multitrack recordings have words from different speakers sorted by start time,
    causing frequent speaker alternation. The multitrack mode should group by
    speaker first, then split into sentences.
    """
    from reflector.processors.types import Transcript, Word
    # Simulate real multitrack data: words sorted by start time, speakers interleave
    # Speaker 0 says: "Hello there."
    # Speaker 1 says: "I'm good."
    # When sorted by time, words interleave
    transcript = Transcript(
        words=[
            Word(text="Hello ", start=0.0, end=0.5, speaker=0),
            Word(text="I'm ", start=0.5, end=0.8, speaker=1),
            Word(text="there.", start=0.5, end=1.0, speaker=0),
            Word(text="good.", start=1.0, end=1.5, speaker=1),
        ]
    )
    # Default behavior (is_multitrack=False): breaks on every speaker change = 4 segments
    segments_default = transcript.as_segments(is_multitrack=False)
    assert len(segments_default) == 4
    # Multitrack behavior: groups by speaker, then sentences = 2 segments
    segments_multitrack = transcript.as_segments(is_multitrack=True)
    assert len(segments_multitrack) == 2
    # Check content - sorted by start time
    assert segments_multitrack[0].speaker == 0
    assert segments_multitrack[0].text == "Hello there."
    assert segments_multitrack[0].start == 0.0
    assert segments_multitrack[0].end == 1.0
    assert segments_multitrack[1].speaker == 1
    assert segments_multitrack[1].text == "I'm good."
    assert segments_multitrack[1].start == 0.5
    assert segments_multitrack[1].end == 1.5
 def test_processor_transcript_segment_multitrack_overlapping_timestamps():
    """Test multitrack with exactly overlapping timestamps (real Daily.co data pattern)."""
    from reflector.processors.types import Transcript, Word
    # Real pattern from transcript 38d84d57: words with identical timestamps
    transcript = Transcript(
        words=[
            Word(text="speaking ", start=6.71, end=7.11, speaker=0),
            Word(text="Speaking ", start=6.71, end=7.11, speaker=1),
            Word(text="at ", start=7.11, end=7.27, speaker=0),
            Word(text="at ", start=7.11, end=7.27, speaker=1),
            Word(text="the ", start=7.27, end=7.43, speaker=0),
            Word(text="the ", start=7.27, end=7.43, speaker=1),
            Word(text="same ", start=7.43, end=7.59, speaker=0),
            Word(text="same ", start=7.43, end=7.59, speaker=1),
            Word(text="time.", start=7.59, end=8.0, speaker=0),
            Word(text="time.", start=7.59, end=8.0, speaker=1),
        ]
    )
    # Default: 10 segments (one per speaker change)
    segments_default = transcript.as_segments(is_multitrack=False)
    assert len(segments_default) == 10
    # Multitrack: 2 segments (one per speaker sentence)
    segments_multitrack = transcript.as_segments(is_multitrack=True)
    assert len(segments_multitrack) == 2
    # Both should have complete sentences
    assert "speaking at the same time." in segments_multitrack[0].text
    assert "Speaking at the same time." in segments_multitrack[1].text
--- a/server/tests/test_transcript_formats.py
+++ b/server/tests/test_transcript_formats.py
@@ -1,779 +0,0 @@
 """Tests for transcript format conversion functionality."""
 import pytest
 from reflector.db.transcripts import TranscriptParticipant, TranscriptTopic
 from reflector.processors.types import Word
 from reflector.utils.transcript_formats import (
    format_timestamp_mmss,
    get_speaker_name,
    topics_to_webvtt_named,
    transcript_to_json_segments,
    transcript_to_text,
    transcript_to_text_timestamped,
 )
@pytest.mark.asyncio
 async def test_get_speaker_name_with_participants():
    """Test speaker name resolution with participants list."""
    participants = [
        TranscriptParticipant(id="1", speaker=0, name="John Smith"),
        TranscriptParticipant(id="2", speaker=1, name="Jane Doe"),
    ]
    assert get_speaker_name(0, participants) == "John Smith"
    assert get_speaker_name(1, participants) == "Jane Doe"
    assert get_speaker_name(2, participants) == "Speaker 2"
@pytest.mark.asyncio
 async def test_get_speaker_name_without_participants():
    """Test speaker name resolution without participants list."""
    assert get_speaker_name(0, None) == "Speaker 0"
    assert get_speaker_name(1, None) == "Speaker 1"
    assert get_speaker_name(5, []) == "Speaker 5"
@pytest.mark.asyncio
 async def test_format_timestamp_mmss():
    """Test timestamp formatting to MM:SS."""
    assert format_timestamp_mmss(0) == "00:00"
    assert format_timestamp_mmss(5) == "00:05"
    assert format_timestamp_mmss(65) == "01:05"
    assert format_timestamp_mmss(125.7) == "02:05"
    assert format_timestamp_mmss(3661) == "61:01"
@pytest.mark.asyncio
 async def test_transcript_to_text():
    """Test plain text format conversion."""
    topics = [
        TranscriptTopic(
            id="1",
            title="Topic 1",
            summary="Summary 1",
            timestamp=0.0,
            words=[
                Word(text="Hello", start=0.0, end=1.0, speaker=0),
                Word(text=" world.", start=1.0, end=2.0, speaker=0),
            ],
        ),
        TranscriptTopic(
            id="2",
            title="Topic 2",
            summary="Summary 2",
            timestamp=2.0,
            words=[
                Word(text="How", start=2.0, end=3.0, speaker=1),
                Word(text=" are", start=3.0, end=4.0, speaker=1),
                Word(text=" you?", start=4.0, end=5.0, speaker=1),
            ],
        ),
    ]
    participants = [
        TranscriptParticipant(id="1", speaker=0, name="John Smith"),
        TranscriptParticipant(id="2", speaker=1, name="Jane Doe"),
    ]
    result = transcript_to_text(topics, participants)
    lines = result.split("\n")
    assert len(lines) == 2
    assert lines[0] == "John Smith: Hello world."
    assert lines[1] == "Jane Doe: How are you?"
@pytest.mark.asyncio
 async def test_transcript_to_text_timestamped():
    """Test timestamped text format conversion."""
    topics = [
        TranscriptTopic(
            id="1",
            title="Topic 1",
            summary="Summary 1",
            timestamp=0.0,
            words=[
                Word(text="Hello", start=0.0, end=1.0, speaker=0),
                Word(text=" world.", start=1.0, end=2.0, speaker=0),
            ],
        ),
        TranscriptTopic(
            id="2",
            title="Topic 2",
            summary="Summary 2",
            timestamp=65.0,
            words=[
                Word(text="How", start=65.0, end=66.0, speaker=1),
                Word(text=" are", start=66.0, end=67.0, speaker=1),
                Word(text=" you?", start=67.0, end=68.0, speaker=1),
            ],
        ),
    ]
    participants = [
        TranscriptParticipant(id="1", speaker=0, name="John Smith"),
        TranscriptParticipant(id="2", speaker=1, name="Jane Doe"),
    ]
    result = transcript_to_text_timestamped(topics, participants)
    lines = result.split("\n")
    assert len(lines) == 2
    assert lines[0] == "[00:00] John Smith: Hello world."
    assert lines[1] == "[01:05] Jane Doe: How are you?"
@pytest.mark.asyncio
 async def test_topics_to_webvtt_named():
    """Test WebVTT format conversion with participant names."""
    topics = [
        TranscriptTopic(
            id="1",
            title="Topic 1",
            summary="Summary 1",
            timestamp=0.0,
            words=[
                Word(text="Hello", start=0.0, end=1.0, speaker=0),
                Word(text=" world.", start=1.0, end=2.0, speaker=0),
            ],
        ),
    ]
    participants = [
        TranscriptParticipant(id="1", speaker=0, name="John Smith"),
    ]
    result = topics_to_webvtt_named(topics, participants)
    assert result.startswith("WEBVTT")
    assert "<v John Smith>" in result
    assert "00:00:00.000 --> 00:00:02.000" in result
    assert "Hello world." in result
@pytest.mark.asyncio
 async def test_transcript_to_json_segments():
    """Test JSON segments format conversion."""
    topics = [
        TranscriptTopic(
            id="1",
            title="Topic 1",
            summary="Summary 1",
            timestamp=0.0,
            words=[
                Word(text="Hello", start=0.0, end=1.0, speaker=0),
                Word(text=" world.", start=1.0, end=2.0, speaker=0),
            ],
        ),
        TranscriptTopic(
            id="2",
            title="Topic 2",
            summary="Summary 2",
            timestamp=2.0,
            words=[
                Word(text="How", start=2.0, end=3.0, speaker=1),
                Word(text=" are", start=3.0, end=4.0, speaker=1),
                Word(text=" you?", start=4.0, end=5.0, speaker=1),
            ],
        ),
    ]
    participants = [
        TranscriptParticipant(id="1", speaker=0, name="John Smith"),
        TranscriptParticipant(id="2", speaker=1, name="Jane Doe"),
    ]
    result = transcript_to_json_segments(topics, participants)
    assert len(result) == 2
    assert result[0].speaker == 0
    assert result[0].speaker_name == "John Smith"
    assert result[0].text == "Hello world."
    assert result[0].start == 0.0
    assert result[0].end == 2.0
    assert result[1].speaker == 1
    assert result[1].speaker_name == "Jane Doe"
    assert result[1].text == "How are you?"
    assert result[1].start == 2.0
    assert result[1].end == 5.0
@pytest.mark.asyncio
 async def test_transcript_formats_with_empty_topics():
    """Test format conversion with empty topics list."""
    topics = []
    participants = []
    assert transcript_to_text(topics, participants) == ""
    assert transcript_to_text_timestamped(topics, participants) == ""
    assert "WEBVTT" in topics_to_webvtt_named(topics, participants)
    assert transcript_to_json_segments(topics, participants) == []
@pytest.mark.asyncio
 async def test_transcript_formats_with_empty_words():
    """Test format conversion with topics containing no words."""
    topics = [
        TranscriptTopic(
            id="1",
            title="Topic 1",
            summary="Summary 1",
            timestamp=0.0,
            words=[],
        ),
    ]
    participants = []
    assert transcript_to_text(topics, participants) == ""
    assert transcript_to_text_timestamped(topics, participants) == ""
    assert "WEBVTT" in topics_to_webvtt_named(topics, participants)
    assert transcript_to_json_segments(topics, participants) == []
@pytest.mark.asyncio
 async def test_transcript_formats_with_multiple_speakers():
    """Test format conversion with multiple speaker changes."""
    topics = [
        TranscriptTopic(
            id="1",
            title="Topic 1",
            summary="Summary 1",
            timestamp=0.0,
            words=[
                Word(text="Hello", start=0.0, end=1.0, speaker=0),
                Word(text=" there.", start=1.0, end=2.0, speaker=0),
                Word(text="Hi", start=2.0, end=3.0, speaker=1),
                Word(text=" back.", start=3.0, end=4.0, speaker=1),
                Word(text="Good", start=4.0, end=5.0, speaker=0),
                Word(text=" morning.", start=5.0, end=6.0, speaker=0),
            ],
        ),
    ]
    participants = [
        TranscriptParticipant(id="1", speaker=0, name="Alice"),
        TranscriptParticipant(id="2", speaker=1, name="Bob"),
    ]
    text_result = transcript_to_text(topics, participants)
    lines = text_result.split("\n")
    assert len(lines) == 3
    assert "Alice: Hello there." in lines[0]
    assert "Bob: Hi back." in lines[1]
    assert "Alice: Good morning." in lines[2]
    json_result = transcript_to_json_segments(topics, participants)
    assert len(json_result) == 3
    assert json_result[0].speaker_name == "Alice"
    assert json_result[1].speaker_name == "Bob"
    assert json_result[2].speaker_name == "Alice"
@pytest.mark.asyncio
 async def test_transcript_formats_with_overlapping_speakers_multitrack():
    """Test format conversion for multitrack recordings with truly interleaved words.
    Multitrack recordings have words from different speakers sorted by start time,
    causing frequent speaker alternation. This tests the sentence-based segmentation
    that groups each speaker's words into complete sentences.
    """
    # Real multitrack data: words sorted by start time, speakers interleave
    # Alice says: "Hello there." (0.0-1.0)
    # Bob says: "I'm good." (0.5-1.5)
    # When sorted by time, words interleave: Hello, I'm, there., good.
    topics = [
        TranscriptTopic(
            id="1",
            title="Topic 1",
            summary="Summary 1",
            timestamp=0.0,
            words=[
                Word(text="Hello ", start=0.0, end=0.5, speaker=0),
                Word(text="I'm ", start=0.5, end=0.8, speaker=1),
                Word(text="there.", start=0.5, end=1.0, speaker=0),
                Word(text="good.", start=1.0, end=1.5, speaker=1),
            ],
        ),
    ]
    participants = [
        TranscriptParticipant(id="1", speaker=0, name="Alice"),
        TranscriptParticipant(id="2", speaker=1, name="Bob"),
    ]
    # With is_multitrack=True, should produce 2 segments (one per speaker sentence)
    # not 4 segments (one per speaker change)
    webvtt_result = topics_to_webvtt_named(topics, participants, is_multitrack=True)
    expected_webvtt = """WEBVTT
 00:00:00.000 --> 00:00:01.000
 <v Alice>Hello there.
 00:00:00.500 --> 00:00:01.500
 <v Bob>I'm good.
 """
    assert webvtt_result == expected_webvtt
    text_result = transcript_to_text(topics, participants, is_multitrack=True)
    lines = text_result.split("\n")
    assert len(lines) == 2
    assert "Alice: Hello there." in lines[0]
    assert "Bob: I'm good." in lines[1]
    timestamped_result = transcript_to_text_timestamped(
        topics, participants, is_multitrack=True
    )
    timestamped_lines = timestamped_result.split("\n")
    assert len(timestamped_lines) == 2
    assert "[00:00] Alice: Hello there." in timestamped_lines[0]
    assert "[00:00] Bob: I'm good." in timestamped_lines[1]
    segments = transcript_to_json_segments(topics, participants, is_multitrack=True)
    assert len(segments) == 2
    assert segments[0].speaker_name == "Alice"
    assert segments[0].text == "Hello there."
    assert segments[1].speaker_name == "Bob"
    assert segments[1].text == "I'm good."
@pytest.mark.asyncio
 async def test_api_transcript_format_text(client):
    """Test GET /transcripts/{id} with transcript_format=text."""
    response = await client.post("/transcripts", json={"name": "Test transcript"})
    assert response.status_code == 200
    tid = response.json()["id"]
    from reflector.db.transcripts import (
        TranscriptParticipant,
        TranscriptTopic,
        transcripts_controller,
    )
    from reflector.processors.types import Word
    transcript = await transcripts_controller.get_by_id(tid)
    await transcripts_controller.update(
        transcript,
        {
            "participants": [
                TranscriptParticipant(
                    id="1", speaker=0, name="John Smith"
                ).model_dump(),
                TranscriptParticipant(id="2", speaker=1, name="Jane Doe").model_dump(),
            ]
        },
    )
    await transcripts_controller.upsert_topic(
        transcript,
        TranscriptTopic(
            title="Topic 1",
            summary="Summary 1",
            timestamp=0,
            words=[
                Word(text="Hello", start=0, end=1, speaker=0),
                Word(text=" world.", start=1, end=2, speaker=0),
            ],
        ),
    )
    response = await client.get(f"/transcripts/{tid}?transcript_format=text")
    assert response.status_code == 200
    data = response.json()
    assert data["transcript_format"] == "text"
    assert "transcript" in data
    assert "John Smith: Hello world." in data["transcript"]
@pytest.mark.asyncio
 async def test_api_transcript_format_text_timestamped(client):
    """Test GET /transcripts/{id} with transcript_format=text-timestamped."""
    response = await client.post("/transcripts", json={"name": "Test transcript"})
    assert response.status_code == 200
    tid = response.json()["id"]
    from reflector.db.transcripts import (
        TranscriptParticipant,
        TranscriptTopic,
        transcripts_controller,
    )
    from reflector.processors.types import Word
    transcript = await transcripts_controller.get_by_id(tid)
    await transcripts_controller.update(
        transcript,
        {
            "participants": [
                TranscriptParticipant(
                    id="1", speaker=0, name="John Smith"
                ).model_dump(),
            ]
        },
    )
    await transcripts_controller.upsert_topic(
        transcript,
        TranscriptTopic(
            title="Topic 1",
            summary="Summary 1",
            timestamp=0,
            words=[
                Word(text="Hello", start=65, end=66, speaker=0),
                Word(text=" world.", start=66, end=67, speaker=0),
            ],
        ),
    )
    response = await client.get(
        f"/transcripts/{tid}?transcript_format=text-timestamped"
    )
    assert response.status_code == 200
    data = response.json()
    assert data["transcript_format"] == "text-timestamped"
    assert "transcript" in data
    assert "[01:05] John Smith: Hello world." in data["transcript"]
@pytest.mark.asyncio
 async def test_api_transcript_format_webvtt_named(client):
    """Test GET /transcripts/{id} with transcript_format=webvtt-named."""
    response = await client.post("/transcripts", json={"name": "Test transcript"})
    assert response.status_code == 200
    tid = response.json()["id"]
    from reflector.db.transcripts import (
        TranscriptParticipant,
        TranscriptTopic,
        transcripts_controller,
    )
    from reflector.processors.types import Word
    transcript = await transcripts_controller.get_by_id(tid)
    await transcripts_controller.update(
        transcript,
        {
            "participants": [
                TranscriptParticipant(
                    id="1", speaker=0, name="John Smith"
                ).model_dump(),
            ]
        },
    )
    await transcripts_controller.upsert_topic(
        transcript,
        TranscriptTopic(
            title="Topic 1",
            summary="Summary 1",
            timestamp=0,
            words=[
                Word(text="Hello", start=0, end=1, speaker=0),
                Word(text=" world.", start=1, end=2, speaker=0),
            ],
        ),
    )
    response = await client.get(f"/transcripts/{tid}?transcript_format=webvtt-named")
    assert response.status_code == 200
    data = response.json()
    assert data["transcript_format"] == "webvtt-named"
    assert "transcript" in data
    assert "WEBVTT" in data["transcript"]
    assert "<v John Smith>" in data["transcript"]
@pytest.mark.asyncio
 async def test_api_transcript_format_json(client):
    """Test GET /transcripts/{id} with transcript_format=json."""
    response = await client.post("/transcripts", json={"name": "Test transcript"})
    assert response.status_code == 200
    tid = response.json()["id"]
    from reflector.db.transcripts import (
        TranscriptParticipant,
        TranscriptTopic,
        transcripts_controller,
    )
    from reflector.processors.types import Word
    transcript = await transcripts_controller.get_by_id(tid)
    await transcripts_controller.update(
        transcript,
        {
            "participants": [
                TranscriptParticipant(
                    id="1", speaker=0, name="John Smith"
                ).model_dump(),
            ]
        },
    )
    await transcripts_controller.upsert_topic(
        transcript,
        TranscriptTopic(
            title="Topic 1",
            summary="Summary 1",
            timestamp=0,
            words=[
                Word(text="Hello", start=0, end=1, speaker=0),
                Word(text=" world.", start=1, end=2, speaker=0),
            ],
        ),
    )
    response = await client.get(f"/transcripts/{tid}?transcript_format=json")
    assert response.status_code == 200
    data = response.json()
    assert data["transcript_format"] == "json"
    assert "transcript" in data
    assert isinstance(data["transcript"], list)
    assert len(data["transcript"]) == 1
    assert data["transcript"][0]["speaker"] == 0
    assert data["transcript"][0]["speaker_name"] == "John Smith"
    assert data["transcript"][0]["text"] == "Hello world."
@pytest.mark.asyncio
 async def test_api_transcript_format_default_is_text(client):
    """Test GET /transcripts/{id} defaults to text format."""
    response = await client.post("/transcripts", json={"name": "Test transcript"})
    assert response.status_code == 200
    tid = response.json()["id"]
    from reflector.db.transcripts import TranscriptTopic, transcripts_controller
    from reflector.processors.types import Word
    transcript = await transcripts_controller.get_by_id(tid)
    await transcripts_controller.upsert_topic(
        transcript,
        TranscriptTopic(
            title="Topic 1",
            summary="Summary 1",
            timestamp=0,
            words=[
                Word(text="Hello", start=0, end=1, speaker=0),
            ],
        ),
    )
    response = await client.get(f"/transcripts/{tid}")
    assert response.status_code == 200
    data = response.json()
    assert data["transcript_format"] == "text"
    assert "transcript" in data
@pytest.mark.asyncio
 async def test_api_topics_endpoint_multitrack_segmentation(client):
    """Test GET /transcripts/{id}/topics uses sentence-based segmentation for multitrack.
    This tests the fix for TASKS2.md - ensuring /topics endpoints correctly detect
    multitrack recordings and use sentence-based segmentation instead of fragmenting
    on every speaker change.
    """
    from datetime import datetime, timezone
    from reflector.db.recordings import Recording, recordings_controller
    from reflector.db.transcripts import (
        TranscriptParticipant,
        TranscriptTopic,
        transcripts_controller,
    )
    from reflector.processors.types import Word
    # Create a multitrack recording (has track_keys)
    recording = Recording(
        bucket_name="test-bucket",
        object_key="test-key",
        recorded_at=datetime.now(timezone.utc),
        track_keys=["track1.webm", "track2.webm"],  # This makes it multitrack
    )
    await recordings_controller.create(recording)
    # Create transcript linked to the recording
    transcript = await transcripts_controller.add(
        name="Multitrack Test",
        source_kind="file",
        recording_id=recording.id,
    )
    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 interleaved words (as they appear in real multitrack data)
    await transcripts_controller.upsert_topic(
        transcript,
        TranscriptTopic(
            title="Topic 1",
            summary="Summary 1",
            timestamp=0,
            words=[
                Word(text="Hello ", start=0.0, end=0.5, speaker=0),
                Word(text="I'm ", start=0.5, end=0.8, speaker=1),
                Word(text="there.", start=0.5, end=1.0, speaker=0),
                Word(text="good.", start=1.0, end=1.5, speaker=1),
            ],
        ),
    )
    # Test /topics endpoint
    response = await client.get(f"/transcripts/{transcript.id}/topics")
    assert response.status_code == 200
    data = response.json()
    assert len(data) == 1
    topic = data[0]
    # Key assertion: multitrack should produce 2 segments (one per speaker sentence)
    # Not 4 segments (one per speaker change)
    assert len(topic["segments"]) == 2
    # Check content
    segment_texts = [s["text"] for s in topic["segments"]]
    assert "Hello there." in segment_texts
    assert "I'm good." in segment_texts
@pytest.mark.asyncio
 async def test_api_topics_endpoint_non_multitrack_segmentation(client):
    """Test GET /transcripts/{id}/topics uses default segmentation for non-multitrack.
    Ensures backward compatibility - transcripts without multitrack recordings
    should continue using the default speaker-change-based segmentation.
    """
    from reflector.db.transcripts import (
        TranscriptParticipant,
        TranscriptTopic,
        transcripts_controller,
    )
    from reflector.processors.types import Word
    # Create transcript WITHOUT recording (defaulted as not multitrack) TODO better heuristic
    response = await client.post("/transcripts", json={"name": "Test transcript"})
    assert response.status_code == 200
    tid = response.json()["id"]
    transcript = await transcripts_controller.get_by_id(tid)
    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 interleaved words
    await transcripts_controller.upsert_topic(
        transcript,
        TranscriptTopic(
            title="Topic 1",
            summary="Summary 1",
            timestamp=0,
            words=[
                Word(text="Hello ", start=0.0, end=0.5, speaker=0),
                Word(text="I'm ", start=0.5, end=0.8, speaker=1),
                Word(text="there.", start=0.5, end=1.0, speaker=0),
                Word(text="good.", start=1.0, end=1.5, speaker=1),
            ],
        ),
    )
    # Test /topics endpoint
    response = await client.get(f"/transcripts/{tid}/topics")
    assert response.status_code == 200
    data = response.json()
    assert len(data) == 1
    topic = data[0]
    # Non-multitrack: should produce 4 segments (one per speaker change)
    assert len(topic["segments"]) == 4
@pytest.mark.asyncio
 async def test_api_topics_with_words_endpoint_multitrack(client):
    """Test GET /transcripts/{id}/topics/with-words uses multitrack segmentation."""
    from datetime import datetime, timezone
    from reflector.db.recordings import Recording, recordings_controller
    from reflector.db.transcripts import (
        TranscriptParticipant,
        TranscriptTopic,
        transcripts_controller,
    )
    from reflector.processors.types import Word
    # Create multitrack recording
    recording = Recording(
        bucket_name="test-bucket",
        object_key="test-key-2",
        recorded_at=datetime.now(timezone.utc),
        track_keys=["track1.webm", "track2.webm"],
    )
    await recordings_controller.create(recording)
    transcript = await transcripts_controller.add(
        name="Multitrack Test 2",
        source_kind="file",
        recording_id=recording.id,
    )
    await transcripts_controller.update(
        transcript,
        {
            "participants": [
                TranscriptParticipant(id="1", speaker=0, name="Alice").model_dump(),
                TranscriptParticipant(id="2", speaker=1, name="Bob").model_dump(),
            ]
        },
    )
    await transcripts_controller.upsert_topic(
        transcript,
        TranscriptTopic(
            title="Topic 1",
            summary="Summary 1",
            timestamp=0,
            words=[
                Word(text="Hello ", start=0.0, end=0.5, speaker=0),
                Word(text="I'm ", start=0.5, end=0.8, speaker=1),
                Word(text="there.", start=0.5, end=1.0, speaker=0),
                Word(text="good.", start=1.0, end=1.5, speaker=1),
            ],
        ),
    )
    response = await client.get(f"/transcripts/{transcript.id}/topics/with-words")
    assert response.status_code == 200
    data = response.json()
    assert len(data) == 1
    topic = data[0]
    # Should have 2 segments (multitrack sentence-based)
    assert len(topic["segments"]) == 2
    # Should also have words field
    assert "words" in topic
    assert len(topic["words"]) == 4
--- a/www/app/(app)/transcripts/[transcriptId]/page.tsx
+++ b/www/app/(app)/transcripts/[transcriptId]/page.tsx
@@ -117,6 +117,15 @@ export default function TranscriptDetails(details: TranscriptDetails) {
    return <Modal title="Loading" text={"Loading transcript..."} />;
  }
  if (mp3.error) {
    return (
      <Modal
        title="Transcription error"
        text={`There was an error loading the recording. Error: ${mp3.error}`}
      />
    );
  }
  return (
    <>
      <Grid
@@ -138,12 +147,7 @@ export default function TranscriptDetails(details: TranscriptDetails) {
              />
            ) : !mp3.loading && (waveform.error || mp3.error) ? (
              <Box p={4} bg="red.100" borderRadius="md">
-                <Text>
+                <Text>Error loading this recording</Text>
                  Error loading{" "}
                  {[waveform.error && "waveform", mp3.error && "mp3"]
                    .filter(Boolean)
                    .join(" and ")}
                </Text>
              </Box>
            ) : (
              <Skeleton h={14} />
--- a/www/app/(app)/transcripts/shareCopy.tsx
+++ b/www/app/(app)/transcripts/shareCopy.tsx
@@ -1,16 +1,14 @@
 import { useState } from "react";
-import type { components, operations } from "../../reflector-api";
+import type { components } from "../../reflector-api";
-type GetTranscriptWithParticipants =
+type GetTranscript = components["schemas"]["GetTranscript"];
  components["schemas"]["GetTranscriptWithParticipants"];
 type GetTranscriptTopic = components["schemas"]["GetTranscriptTopic"];
-import { Button, BoxProps, Box, Menu, Text } from "@chakra-ui/react";
+import { Button, BoxProps, Box } from "@chakra-ui/react";
-import { LuChevronDown } from "react-icons/lu";
+import { buildTranscriptWithTopics } from "./buildTranscriptWithTopics";
-import { client } from "../../lib/apiClient";
+import { useTranscriptParticipants } from "../../lib/apiHooks";
 import { toaster } from "../../components/ui/toaster";
 type ShareCopyProps = {
  finalSummaryElement: HTMLDivElement | null;
-  transcript: GetTranscriptWithParticipants;
+  transcript: GetTranscript;
  topics: GetTranscriptTopic[];
 };
@@ -22,33 +20,11 @@ export default function ShareCopy({
 }: ShareCopyProps & BoxProps) {
  const [isCopiedSummary, setIsCopiedSummary] = useState(false);
  const [isCopiedTranscript, setIsCopiedTranscript] = useState(false);
-  const [isCopying, setIsCopying] = useState(false);
+  const participantsQuery = useTranscriptParticipants(transcript?.id || null);
  type ApiTranscriptFormat = NonNullable<
    operations["v1_transcript_get"]["parameters"]["query"]
  >["transcript_format"];
  const TRANSCRIPT_FORMATS = [
    "text",
    "text-timestamped",
    "webvtt-named",
    "json",
  ] as const satisfies ApiTranscriptFormat[];
  type TranscriptFormat = (typeof TRANSCRIPT_FORMATS)[number];
  const TRANSCRIPT_FORMAT_LABELS: { [k in TranscriptFormat]: string } = {
    text: "Plain text",
    "text-timestamped": "Text + timestamps",
    "webvtt-named": "WebVTT (named)",
    json: "JSON",
  };
  const formatOptions = TRANSCRIPT_FORMATS.map((f) => ({
    value: f,
    label: TRANSCRIPT_FORMAT_LABELS[f],
  }));
  const onCopySummaryClick = () => {
    const text_to_copy = finalSummaryElement?.innerText;
    if (text_to_copy) {
      navigator.clipboard.writeText(text_to_copy).then(() => {
        setIsCopiedSummary(true);
@@ -58,91 +34,27 @@ export default function ShareCopy({
    }
  };
-  const onCopyTranscriptFormatClick = async (format: TranscriptFormat) => {
+  const onCopyTranscriptClick = () => {
-    try {
+    const text_to_copy =
-      setIsCopying(true);
+      buildTranscriptWithTopics(
-      const { data, error } = await client.GET(
+        topics || [],
-        "/v1/transcripts/{transcript_id}",
+        participantsQuery?.data || null,
-        {
+        transcript?.title || null,
-          params: {
+      ) || "";
            path: { transcript_id: transcript.id },
            query: { transcript_format: format },
          },
        },
      );
      if (error) {
        console.error("Failed to copy transcript:", error);
        toaster.create({
          duration: 3000,
          render: () => (
            <Box bg="red.500" color="white" px={4} py={3} borderRadius="md">
              <Text fontWeight="bold">Error</Text>
              <Text fontSize="sm">Failed to fetch transcript</Text>
            </Box>
          ),
        });
        return;
      }
-      const copiedText =
+    text_to_copy &&
-        format === "json"
+      navigator.clipboard.writeText(text_to_copy).then(() => {
          ? JSON.stringify(data?.transcript ?? {}, null, 2)
          : String(data?.transcript ?? "");
      if (copiedText) {
        await navigator.clipboard.writeText(copiedText);
        setIsCopiedTranscript(true);
        // Reset the copied state after 2 seconds
        setTimeout(() => setIsCopiedTranscript(false), 2000);
      }
    } catch (e) {
      console.error("Failed to copy transcript:", e);
      toaster.create({
        duration: 3000,
        render: () => (
          <Box bg="red.500" color="white" px={4} py={3} borderRadius="md">
            <Text fontWeight="bold">Error</Text>
            <Text fontSize="sm">Failed to copy transcript</Text>
          </Box>
        ),
      });
    } finally {
      setIsCopying(false);
    }
  };
  return (
    <Box {...boxProps}>
-      <Menu.Root
+      <Button onClick={onCopyTranscriptClick} mr={2} variant="subtle">
        closeOnSelect={true}
        lazyMount={true}
        positioning={{ gutter: 4 }}
      >
        <Menu.Trigger asChild>
          <Button
            mr={2}
            variant="subtle"
            loading={isCopying}
            loadingText="Copying..."
          >
        {isCopiedTranscript ? "Copied!" : "Copy Transcript"}
            <LuChevronDown style={{ marginLeft: 6 }} />
      </Button>
        </Menu.Trigger>
        <Menu.Positioner>
          <Menu.Content>
            {formatOptions.map((opt) => (
              <Menu.Item
                key={opt.value}
                value={opt.value}
                _hover={{ backgroundColor: "gray.100" }}
                onClick={() => onCopyTranscriptFormatClick(opt.value)}
              >
                {opt.label}
              </Menu.Item>
            ))}
          </Menu.Content>
        </Menu.Positioner>
      </Menu.Root>
      <Button onClick={onCopySummaryClick} variant="subtle">
        {isCopiedSummary ? "Copied!" : "Copy Summary"}
      </Button>
--- a/www/app/[roomName]/components/DailyRoom.tsx
+++ b/www/app/[roomName]/components/DailyRoom.tsx
@@ -11,7 +11,6 @@ import {
  recordingTypeRequiresConsent,
 } from "../../lib/consent";
 import { useRoomJoinMeeting } from "../../lib/apiHooks";
 import { assertExists } from "../../lib/utils";
 type Meeting = components["schemas"]["Meeting"];
@@ -23,15 +22,16 @@ export default function DailyRoom({ meeting }: DailyRoomProps) {
  const router = useRouter();
  const params = useParams();
  const auth = useAuth();
-  const authLastUserId = auth.lastUserId;
+  const status = auth.status;
  const containerRef = useRef<HTMLDivElement>(null);
  const joinMutation = useRoomJoinMeeting();
  const [joinedMeeting, setJoinedMeeting] = useState<Meeting | null>(null);
  const roomName = params?.roomName as string;
  // Always call /join to get a fresh token with user_id
  useEffect(() => {
-    if (authLastUserId === undefined || !meeting?.id || !roomName) return;
+    if (status === "loading" || !meeting?.id || !roomName) return;
    const join = async () => {
      try {
@@ -50,17 +50,18 @@ export default function DailyRoom({ meeting }: DailyRoomProps) {
    };
    join();
-  }, [meeting?.id, roomName, authLastUserId]);
+  }, [meeting?.id, roomName, status]);
  const roomUrl = joinedMeeting?.host_room_url || joinedMeeting?.room_url;
  const isLoading =
    status === "loading" || joinMutation.isPending || !joinedMeeting;
  const handleLeave = useCallback(() => {
    router.push("/browse");
  }, [router]);
  useEffect(() => {
-    if (authLastUserId === undefined || !roomUrl || !containerRef.current)
+    if (isLoading || !roomUrl || !containerRef.current) return;
      return;
    let frame: DailyCall | null = null;
    let destroyed = false;
@@ -89,14 +90,9 @@ export default function DailyRoom({ meeting }: DailyRoomProps) {
        frame.on("left-meeting", handleLeave);
        // TODO this method must not ignore no-recording option
        // TODO this method is here to make dev environments work in some cases (not examined which)
        frame.on("joined-meeting", async () => {
          try {
-            assertExists(
+            await frame.startRecording({ type: "raw-tracks" });
              frame,
              "frame object got lost somewhere after frame.on was called",
            ).startRecording({ type: "raw-tracks" });
          } catch (error) {
            console.error("Failed to start recording:", error);
          }
@@ -108,9 +104,7 @@ export default function DailyRoom({ meeting }: DailyRoomProps) {
      }
    };
-    createAndJoin().catch((error) => {
+    createAndJoin();
      console.error("Failed to create and join meeting:", error);
    });
    return () => {
      destroyed = true;
@@ -120,9 +114,9 @@ export default function DailyRoom({ meeting }: DailyRoomProps) {
        });
      }
    };
-  }, [roomUrl, authLastUserId, handleLeave]);
+  }, [roomUrl, isLoading, handleLeave]);
-  if (authLastUserId === undefined) {
+  if (isLoading) {
    return (
      <Center width="100vw" height="100vh">
        <Spinner size="xl" />
--- a/www/app/lib/AuthProvider.tsx
+++ b/www/app/lib/AuthProvider.tsx
@@ -1,6 +1,6 @@
 "use client";
-import { createContext, useContext, useRef } from "react";
+import { createContext, useContext } from "react";
 import { useSession as useNextAuthSession } from "next-auth/react";
 import { signOut, signIn } from "next-auth/react";
 import { configureApiAuth } from "./apiClient";
@@ -25,9 +25,6 @@ type AuthContextType = (
  update: () => Promise<Session | null>;
  signIn: typeof signIn;
  signOut: typeof signOut;
  // TODO probably rename isLoading to isReloading and make THIS field "isLoading"
  // undefined is "not known", null is "is certainly logged out"
  lastUserId: CustomSession["user"]["id"] | null | undefined;
 };
 const AuthContext = createContext<AuthContextType | undefined>(undefined);
@@ -44,15 +41,10 @@ const noopAuthContext: AuthContextType = {
  signOut: async () => {
    throw new Error("signOut not supposed to be called");
  },
  lastUserId: undefined,
 };
 export function AuthProvider({ children }: { children: React.ReactNode }) {
  const { data: session, status, update } = useNextAuthSession();
  // referential comparison done in component, must be primitive /or cached
  const lastUserId = useRef<CustomSession["user"]["id"] | null | undefined>(
    null,
  );
  const contextValue: AuthContextType = isAuthEnabled
    ? {
@@ -81,16 +73,11 @@ export function AuthProvider({ children }: { children: React.ReactNode }) {
            case "authenticated": {
              const customSession = assertCustomSession(session);
              if (customSession?.error === REFRESH_ACCESS_TOKEN_ERROR) {
                // warning: call order-dependent
                lastUserId.current = null;
                // token had expired but next auth still returns "authenticated" so show user unauthenticated state
                return {
                  status: "unauthenticated" as const,
                };
              } else if (customSession?.accessToken) {
                // updates anyways with updated properties below
                // warning! execution order conscience, must be ran before reading lastUserId.current below
                lastUserId.current = customSession.user.id;
                return {
                  status,
                  accessToken: customSession.accessToken,
@@ -105,8 +92,6 @@ export function AuthProvider({ children }: { children: React.ReactNode }) {
              }
            }
            case "unauthenticated": {
              // warning: call order-dependent
              lastUserId.current = null;
              return { status: "unauthenticated" as const };
            }
            default: {
@@ -118,8 +103,6 @@ export function AuthProvider({ children }: { children: React.ReactNode }) {
        update,
        signIn,
        signOut,
        // for optimistic cases when we assume "loading" doesn't immediately invalidate the user
        lastUserId: lastUserId.current,
      }
    : noopAuthContext;
--- a/www/app/lib/authBackend.ts
+++ b/www/app/lib/authBackend.ts
@@ -32,11 +32,6 @@ async function getUserId(accessToken: string): Promise<string | null> {
    });
    if (!response.ok) {
      try {
        console.error(await response.text());
      } catch (e) {
        console.error("Failed to parse error response", e);
      }
      return null;
    }
@@ -148,7 +143,7 @@ export const authOptions = (): AuthOptions =>
          },
          async session({ session, token }) {
            const extendedToken = token as JWTWithAccessToken;
-            console.log("extendedToken", extendedToken);
+
            const userId = await getUserId(extendedToken.accessToken);
            return {
--- a/www/app/reflector-api.d.ts
+++ b/www/app/reflector-api.d.ts
@@ -696,7 +696,7 @@ export interface paths {
    patch?: never;
    trace?: never;
  };
-  "/v1/daily/webhook": {
+  "/v1/webhook": {
    parameters: {
      query?: never;
      header?: never;
@@ -708,27 +708,6 @@ export interface paths {
    /**
     * Webhook
     * @description Handle Daily webhook events.
     *
     *     Example webhook payload:
     *     {
     *       "version": "1.0.0",
     *       "type": "recording.ready-to-download",
     *       "id": "rec-rtd-c3df927c-f738-4471-a2b7-066fa7e95a6b-1692124192",
     *       "payload": {
     *         "recording_id": "08fa0b24-9220-44c5-846c-3f116cf8e738",
     *         "room_name": "Xcm97xRZ08b2dePKb78g",
     *         "start_ts": 1692124183,
     *         "status": "finished",
     *         "max_participants": 1,
     *         "duration": 9,
     *         "share_token": "ntDCL5k98Ulq", #gitleaks:allow
     *         "s3_key": "api-test-1j8fizhzd30c/Xcm97xRZ08b2dePKb78g/1692124183028"
     *       },
     *       "event_ts": 1692124192
     *     }
     *
     *     Daily.co circuit-breaker: After 3+ failed responses (4xx/5xx), webhook
     *     state→FAILED, stops sending events. Reset: scripts/recreate_daily_webhook.py
     */
    post: operations["v1_webhook"];
    delete?: never;
@@ -920,11 +899,81 @@ export interface components {
      target_language: string;
      source_kind?: components["schemas"]["SourceKind"] | null;
    };
    /**
     * DailyWebhookEvent
     * @description Daily webhook event structure.
     */
    DailyWebhookEvent: {
      /** Type */
      type: string;
      /** Id */
      id: string;
      /** Ts */
      ts: number;
      /** Data */
      data: {
        [key: string]: unknown;
      };
    };
    /** DeletionStatus */
    DeletionStatus: {
      /** Status */
      status: string;
    };
    /** GetTranscript */
    GetTranscript: {
      /** Id */
      id: string;
      /** User Id */
      user_id: string | null;
      /** Name */
      name: string;
      /**
       * Status
       * @enum {string}
       */
      status:
        | "idle"
        | "uploaded"
        | "recording"
        | "processing"
        | "error"
        | "ended";
      /** Locked */
      locked: boolean;
      /** Duration */
      duration: number;
      /** Title */
      title: string | null;
      /** Short Summary */
      short_summary: string | null;
      /** Long Summary */
      long_summary: string | null;
      /** Created At */
      created_at: string;
      /**
       * Share Mode
       * @default private
       */
      share_mode: string;
      /** Source Language */
      source_language: string | null;
      /** Target Language */
      target_language: string | null;
      /** Reviewed */
      reviewed: boolean;
      /** Meeting Id */
      meeting_id: string | null;
      source_kind: components["schemas"]["SourceKind"];
      /** Room Id */
      room_id?: string | null;
      /** Room Name */
      room_name?: string | null;
      /** Audio Deleted */
      audio_deleted?: boolean | null;
      /** Participants */
      participants: components["schemas"]["TranscriptParticipant"][] | null;
    };
    /** GetTranscriptMinimal */
    GetTranscriptMinimal: {
      /** Id */
@@ -1056,345 +1105,6 @@ export interface components {
       */
      words_per_speaker: components["schemas"]["SpeakerWords"][];
    };
    /**
     * GetTranscriptWithJSON
     * @description Transcript response as structured JSON segments.
     *
     *     Format: Array of segment objects with speaker info, text, and timing.
     *     Example:
     *         [
     *             {
     *                 "speaker": 0,
     *                 "speaker_name": "John Smith",
     *                 "text": "Hello everyone",
     *                 "start": 0.0,
     *                 "end": 5.0
     *             }
     *         ]
     */
    GetTranscriptWithJSON: {
      /** Id */
      id: string;
      /** User Id */
      user_id: string | null;
      /** Name */
      name: string;
      /**
       * Status
       * @enum {string}
       */
      status:
        | "idle"
        | "uploaded"
        | "recording"
        | "processing"
        | "error"
        | "ended";
      /** Locked */
      locked: boolean;
      /** Duration */
      duration: number;
      /** Title */
      title: string | null;
      /** Short Summary */
      short_summary: string | null;
      /** Long Summary */
      long_summary: string | null;
      /** Created At */
      created_at: string;
      /**
       * Share Mode
       * @default private
       */
      share_mode: string;
      /** Source Language */
      source_language: string | null;
      /** Target Language */
      target_language: string | null;
      /** Reviewed */
      reviewed: boolean;
      /** Meeting Id */
      meeting_id: string | null;
      source_kind: components["schemas"]["SourceKind"];
      /** Room Id */
      room_id?: string | null;
      /** Room Name */
      room_name?: string | null;
      /** Audio Deleted */
      audio_deleted?: boolean | null;
      /** Participants */
      participants: components["schemas"]["TranscriptParticipant"][] | null;
      /**
       * @description discriminator enum property added by openapi-typescript
       * @enum {string}
       */
      transcript_format: "json";
      /** Transcript */
      transcript: components["schemas"]["TranscriptSegment"][];
    };
    /** GetTranscriptWithParticipants */
    GetTranscriptWithParticipants: {
      /** Id */
      id: string;
      /** User Id */
      user_id: string | null;
      /** Name */
      name: string;
      /**
       * Status
       * @enum {string}
       */
      status:
        | "idle"
        | "uploaded"
        | "recording"
        | "processing"
        | "error"
        | "ended";
      /** Locked */
      locked: boolean;
      /** Duration */
      duration: number;
      /** Title */
      title: string | null;
      /** Short Summary */
      short_summary: string | null;
      /** Long Summary */
      long_summary: string | null;
      /** Created At */
      created_at: string;
      /**
       * Share Mode
       * @default private
       */
      share_mode: string;
      /** Source Language */
      source_language: string | null;
      /** Target Language */
      target_language: string | null;
      /** Reviewed */
      reviewed: boolean;
      /** Meeting Id */
      meeting_id: string | null;
      source_kind: components["schemas"]["SourceKind"];
      /** Room Id */
      room_id?: string | null;
      /** Room Name */
      room_name?: string | null;
      /** Audio Deleted */
      audio_deleted?: boolean | null;
      /** Participants */
      participants: components["schemas"]["TranscriptParticipant"][] | null;
    };
    /**
     * GetTranscriptWithText
     * @description Transcript response with plain text format.
     *
     *     Format: Speaker names followed by their dialogue, one line per segment.
     *     Example:
     *         John Smith: Hello everyone
     *         Jane Doe: Hi there
     */
    GetTranscriptWithText: {
      /** Id */
      id: string;
      /** User Id */
      user_id: string | null;
      /** Name */
      name: string;
      /**
       * Status
       * @enum {string}
       */
      status:
        | "idle"
        | "uploaded"
        | "recording"
        | "processing"
        | "error"
        | "ended";
      /** Locked */
      locked: boolean;
      /** Duration */
      duration: number;
      /** Title */
      title: string | null;
      /** Short Summary */
      short_summary: string | null;
      /** Long Summary */
      long_summary: string | null;
      /** Created At */
      created_at: string;
      /**
       * Share Mode
       * @default private
       */
      share_mode: string;
      /** Source Language */
      source_language: string | null;
      /** Target Language */
      target_language: string | null;
      /** Reviewed */
      reviewed: boolean;
      /** Meeting Id */
      meeting_id: string | null;
      source_kind: components["schemas"]["SourceKind"];
      /** Room Id */
      room_id?: string | null;
      /** Room Name */
      room_name?: string | null;
      /** Audio Deleted */
      audio_deleted?: boolean | null;
      /** Participants */
      participants: components["schemas"]["TranscriptParticipant"][] | null;
      /**
       * @description discriminator enum property added by openapi-typescript
       * @enum {string}
       */
      transcript_format: "text";
      /** Transcript */
      transcript: string;
    };
    /**
     * GetTranscriptWithTextTimestamped
     * @description Transcript response with timestamped text format.
     *
     *     Format: [MM:SS] timestamp prefix before each speaker and dialogue.
     *     Example:
     *         [00:00] John Smith: Hello everyone
     *         [00:05] Jane Doe: Hi there
     */
    GetTranscriptWithTextTimestamped: {
      /** Id */
      id: string;
      /** User Id */
      user_id: string | null;
      /** Name */
      name: string;
      /**
       * Status
       * @enum {string}
       */
      status:
        | "idle"
        | "uploaded"
        | "recording"
        | "processing"
        | "error"
        | "ended";
      /** Locked */
      locked: boolean;
      /** Duration */
      duration: number;
      /** Title */
      title: string | null;
      /** Short Summary */
      short_summary: string | null;
      /** Long Summary */
      long_summary: string | null;
      /** Created At */
      created_at: string;
      /**
       * Share Mode
       * @default private
       */
      share_mode: string;
      /** Source Language */
      source_language: string | null;
      /** Target Language */
      target_language: string | null;
      /** Reviewed */
      reviewed: boolean;
      /** Meeting Id */
      meeting_id: string | null;
      source_kind: components["schemas"]["SourceKind"];
      /** Room Id */
      room_id?: string | null;
      /** Room Name */
      room_name?: string | null;
      /** Audio Deleted */
      audio_deleted?: boolean | null;
      /** Participants */
      participants: components["schemas"]["TranscriptParticipant"][] | null;
      /**
       * @description discriminator enum property added by openapi-typescript
       * @enum {string}
       */
      transcript_format: "text-timestamped";
      /** Transcript */
      transcript: string;
    };
    /**
     * GetTranscriptWithWebVTTNamed
     * @description Transcript response in WebVTT subtitle format with participant names.
     *
     *     Format: Standard WebVTT with voice tags using participant names.
     *     Example:
     *         WEBVTT
     *
     *         00:00:00.000 --> 00:00:05.000
     *         <v John Smith>Hello everyone
     */
    GetTranscriptWithWebVTTNamed: {
      /** Id */
      id: string;
      /** User Id */
      user_id: string | null;
      /** Name */
      name: string;
      /**
       * Status
       * @enum {string}
       */
      status:
        | "idle"
        | "uploaded"
        | "recording"
        | "processing"
        | "error"
        | "ended";
      /** Locked */
      locked: boolean;
      /** Duration */
      duration: number;
      /** Title */
      title: string | null;
      /** Short Summary */
      short_summary: string | null;
      /** Long Summary */
      long_summary: string | null;
      /** Created At */
      created_at: string;
      /**
       * Share Mode
       * @default private
       */
      share_mode: string;
      /** Source Language */
      source_language: string | null;
      /** Target Language */
      target_language: string | null;
      /** Reviewed */
      reviewed: boolean;
      /** Meeting Id */
      meeting_id: string | null;
      source_kind: components["schemas"]["SourceKind"];
      /** Room Id */
      room_id?: string | null;
      /** Room Name */
      room_name?: string | null;
      /** Audio Deleted */
      audio_deleted?: boolean | null;
      /** Participants */
      participants: components["schemas"]["TranscriptParticipant"][] | null;
      /**
       * @description discriminator enum property added by openapi-typescript
       * @enum {string}
       */
      transcript_format: "webvtt-named";
      /** Transcript */
      transcript: string;
    };
    /** HTTPValidationError */
    HTTPValidationError: {
      /** Detail */
@@ -1523,6 +1233,7 @@ export interface components {
      } | null;
      /**
       * Platform
       * @default whereby
       * @enum {string}
       */
      platform: "whereby" | "daily";
@@ -1614,6 +1325,7 @@ export interface components {
      ics_last_etag?: string | null;
      /**
       * Platform
       * @default whereby
       * @enum {string}
       */
      platform: "whereby" | "daily";
@@ -1665,6 +1377,7 @@ export interface components {
      ics_last_etag?: string | null;
      /**
       * Platform
       * @default whereby
       * @enum {string}
       */
      platform: "whereby" | "daily";
@@ -1810,24 +1523,6 @@ export interface components {
      speaker: number | null;
      /** Name */
      name: string;
      /** User Id */
      user_id?: string | null;
    };
    /**
     * TranscriptSegment
     * @description A single transcript segment with speaker and timing information.
     */
    TranscriptSegment: {
      /** Speaker */
      speaker: number;
      /** Speaker Name */
      speaker_name: string;
      /** Text */
      text: string;
      /** Start */
      start: number;
      /** End */
      end: number;
    };
    /** UpdateParticipant */
    UpdateParticipant: {
@@ -2616,7 +2311,7 @@ export interface operations {
          [name: string]: unknown;
        };
        content: {
-          "application/json": components["schemas"]["GetTranscriptWithParticipants"];
+          "application/json": components["schemas"]["GetTranscript"];
        };
      };
      /** @description Validation Error */
@@ -2674,13 +2369,7 @@ export interface operations {
  };
  v1_transcript_get: {
    parameters: {
-      query?: {
+      query?: never;
        transcript_format?:
          | "text"
          | "text-timestamped"
          | "webvtt-named"
          | "json";
      };
      header?: never;
      path: {
        transcript_id: string;
@@ -2695,11 +2384,7 @@ export interface operations {
          [name: string]: unknown;
        };
        content: {
-          "application/json":
+          "application/json": components["schemas"]["GetTranscript"];
            | components["schemas"]["GetTranscriptWithText"]
            | components["schemas"]["GetTranscriptWithTextTimestamped"]
            | components["schemas"]["GetTranscriptWithWebVTTNamed"]
            | components["schemas"]["GetTranscriptWithJSON"];
        };
      };
      /** @description Validation Error */
@@ -2765,7 +2450,7 @@ export interface operations {
          [name: string]: unknown;
        };
        content: {
-          "application/json": components["schemas"]["GetTranscriptWithParticipants"];
+          "application/json": components["schemas"]["GetTranscript"];
        };
      };
      /** @description Validation Error */
@@ -3571,7 +3256,11 @@ export interface operations {
      path?: never;
      cookie?: never;
    };
-    requestBody?: never;
+    requestBody: {
      content: {
        "application/json": components["schemas"]["DailyWebhookEvent"];
      };
    };
    responses: {
      /** @description Successful Response */
      200: {
@@ -3582,6 +3271,15 @@ export interface operations {
          "application/json": unknown;
        };
      };
      /** @description Validation Error */
      422: {
        headers: {
          [name: string]: unknown;
        };
        content: {
          "application/json": components["schemas"]["HTTPValidationError"];
        };
      };
    };
  };
 }