chore(main): release 0.22.2 (#756)

* multitrack mixdown optimisation

* skip mixdown for multitrack

* skip mixdown for multitrack

---------

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

CHANGELOG.md

@@ -1,5 +1,27 @@
 # 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)
 
 

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 = Field(description="User identifier")
+    user_id: NonEmptyString | None = Field(None, 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)")

server/reflector/db/recordings.py

@@ -35,8 +35,15 @@ 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):

server/reflector/pipelines/main_multitrack_pipeline.py

@@ -9,7 +9,10 @@ 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,
@@ -28,8 +31,14 @@ 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
@@ -494,6 +503,90 @@ 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():
@@ -502,9 +595,12 @@ class PipelineMainMultitrack(PipelineMainBase):
                 {
                     "events": [],
                     "topics": [],
+                    "participants": [],
                 },
             )
 
+        await self.update_participants_from_daily(transcript, track_keys)
+
         source_storage = get_transcripts_storage()
         transcript_storage = source_storage
 
@@ -536,43 +632,55 @@ class PipelineMainMultitrack(PipelineMainBase):
 
         transcript.data_path.mkdir(parents=True, exist_ok=True)
 
-        mp3_writer = AudioFileWriterProcessor(
+        if settings.SKIP_MIXDOWN:
-            path=str(transcript.audio_mp3_filename),
+            self.logger.warning(
-            on_duration=self.on_duration,
+                "SKIP_MIXDOWN enabled: Skipping mixdown and waveform generation. "
-        )
+                "UI will have no audio playback or waveform.",
-        await self.mixdown_tracks(padded_track_urls, mp3_writer, offsets_seconds=None)
+                num_tracks=len(padded_track_urls),
-        await mp3_writer.flush()
+                transcript_id=transcript.id,
+            )
+        else:
+            mp3_writer = AudioFileWriterProcessor(
+                path=str(transcript.audio_mp3_filename),
+                on_duration=self.on_duration,
+            )
+            await self.mixdown_tracks(
+                padded_track_urls, mp3_writer, offsets_seconds=None
+            )
+            await mp3_writer.flush()
 
-        if not transcript.audio_mp3_filename.exists():
+            if not transcript.audio_mp3_filename.exists():
-            raise Exception(
+                raise Exception(
-                "Mixdown failed - no MP3 file generated. Cannot proceed without playable audio."
+                    "Mixdown failed - no MP3 file generated. Cannot proceed without playable audio."
+                )
+
+            storage_path = f"{transcript.id}/audio.mp3"
+            # Use file handle streaming to avoid loading entire MP3 into memory
+            mp3_size = transcript.audio_mp3_filename.stat().st_size
+            with open(transcript.audio_mp3_filename, "rb") as mp3_file:
+                await transcript_storage.put_file(storage_path, mp3_file)
+            mp3_url = await transcript_storage.get_file_url(storage_path)
+
+            await transcripts_controller.update(
+                transcript, {"audio_location": "storage"}
             )
 
-        storage_path = f"{transcript.id}/audio.mp3"
+            self.logger.info(
-        # Use file handle streaming to avoid loading entire MP3 into memory
+                f"Uploaded mixed audio to storage",
-        mp3_size = transcript.audio_mp3_filename.stat().st_size
+                storage_path=storage_path,
-        with open(transcript.audio_mp3_filename, "rb") as mp3_file:
+                size=mp3_size,
-            await transcript_storage.put_file(storage_path, mp3_file)
+                url=mp3_url,
-        mp3_url = await transcript_storage.get_file_url(storage_path)
+            )
 
-        await transcripts_controller.update(transcript, {"audio_location": "storage"})
+            self.logger.info("Generating waveform from mixed audio")
-
+            waveform_processor = AudioWaveformProcessor(
-        self.logger.info(
+                audio_path=transcript.audio_mp3_filename,
-            f"Uploaded mixed audio to storage",
+                waveform_path=transcript.audio_waveform_filename,
-            storage_path=storage_path,
+                on_waveform=self.on_waveform,
-            size=mp3_size,
+            )
-            url=mp3_url,
+            waveform_processor.set_pipeline(self.empty_pipeline)
-        )
+            await waveform_processor.flush()
-
+            self.logger.info("Waveform generated successfully")
-        self.logger.info("Generating waveform from mixed audio")
-        waveform_processor = AudioWaveformProcessor(
-            audio_path=transcript.audio_mp3_filename,
-            waveform_path=transcript.audio_waveform_filename,
-            on_waveform=self.on_waveform,
-        )
-        waveform_processor.set_pipeline(self.empty_pipeline)
-        await waveform_processor.flush()
-        self.logger.info("Waveform generated successfully")
 
         speaker_transcripts: list[TranscriptType] = []
         for idx, padded_url in enumerate(padded_track_urls):

server/reflector/processors/types.py

@@ -1,6 +1,7 @@
 import io
 import re
 import tempfile
+from collections import defaultdict
 from pathlib import Path
 from typing import Annotated, TypedDict
 
@@ -16,6 +17,17 @@ 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):
@@ -76,7 +88,6 @@ 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:
@@ -106,7 +117,7 @@ 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_LENGTH):
+        if have_punc and (len(current_segment.text) > MAX_SEGMENT_CHARS):
             segments.append(current_segment)
             current_segment = None
 
@@ -116,6 +127,70 @@ 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] = []
@@ -154,7 +229,9 @@ class Transcript(BaseModel):
             word.start += offset
             word.end += offset
 
-    def as_segments(self) -> list[TranscriptSegment]:
+    def as_segments(self, is_multitrack: bool = False) -> list[TranscriptSegment]:
+        if is_multitrack:
+            return words_to_segments_by_sentence(self.words)
         return words_to_segments(self.words)
 
 

server/reflector/settings.py

@@ -138,6 +138,14 @@ 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
 

server/reflector/utils/daily.py

@@ -64,6 +64,11 @@ 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.

server/reflector/utils/transcript_formats.py

@@ -6,9 +6,6 @@ from reflector.db.transcripts import TranscriptParticipant, TranscriptTopic
 from reflector.processors.types import (
     Transcript as ProcessorTranscript,
 )
-from reflector.processors.types import (
-    words_to_segments,
-)
 from reflector.schemas.transcript_formats import TranscriptSegment
 from reflector.utils.webvtt import seconds_to_timestamp
 
@@ -32,7 +29,9 @@ def format_timestamp_mmss(seconds: float | int) -> str:
 
 
 def transcript_to_text(
-    topics: list[TranscriptTopic], participants: list[TranscriptParticipant] | None
+    topics: list[TranscriptTopic],
+    participants: list[TranscriptParticipant] | None,
+    is_multitrack: bool = False,
 ) -> str:
     """Convert transcript topics to plain text with speaker names."""
     lines = []
@@ -41,7 +40,7 @@ def transcript_to_text(
             continue
 
         transcript = ProcessorTranscript(words=topic.words)
-        segments = transcript.as_segments()
+        segments = transcript.as_segments(is_multitrack)
 
         for segment in segments:
             speaker_name = get_speaker_name(segment.speaker, participants)
@@ -52,7 +51,9 @@ def transcript_to_text(
 
 
 def transcript_to_text_timestamped(
-    topics: list[TranscriptTopic], participants: list[TranscriptParticipant] | None
+    topics: list[TranscriptTopic],
+    participants: list[TranscriptParticipant] | None,
+    is_multitrack: bool = False,
 ) -> str:
     """Convert transcript topics to timestamped text with speaker names."""
     lines = []
@@ -61,7 +62,7 @@ def transcript_to_text_timestamped(
             continue
 
         transcript = ProcessorTranscript(words=topic.words)
-        segments = transcript.as_segments()
+        segments = transcript.as_segments(is_multitrack)
 
         for segment in segments:
             speaker_name = get_speaker_name(segment.speaker, participants)
@@ -73,7 +74,9 @@ def transcript_to_text_timestamped(
 
 
 def topics_to_webvtt_named(
-    topics: list[TranscriptTopic], participants: list[TranscriptParticipant] | None
+    topics: list[TranscriptTopic],
+    participants: list[TranscriptParticipant] | None,
+    is_multitrack: bool = False,
 ) -> str:
     """Convert transcript topics to WebVTT format with participant names."""
     vtt = webvtt.WebVTT()
@@ -82,7 +85,8 @@ def topics_to_webvtt_named(
         if not topic.words:
             continue
 
-        segments = words_to_segments(topic.words)
+        transcript = ProcessorTranscript(words=topic.words)
+        segments = transcript.as_segments(is_multitrack)
 
         for segment in segments:
             speaker_name = get_speaker_name(segment.speaker, participants)
@@ -100,19 +104,23 @@ def topics_to_webvtt_named(
 
 
 def transcript_to_json_segments(
-    topics: list[TranscriptTopic], participants: list[TranscriptParticipant] | None
+    topics: list[TranscriptTopic],
+    participants: list[TranscriptParticipant] | None,
+    is_multitrack: bool = False,
 ) -> list[TranscriptSegment]:
     """Convert transcript topics to a flat list of JSON segments."""
-    segments = []
+    result = []
 
     for topic in topics:
         if not topic.words:
             continue
 
         transcript = ProcessorTranscript(words=topic.words)
-        for segment in transcript.as_segments():
+        segments = transcript.as_segments(is_multitrack)
+
+        for segment in segments:
             speaker_name = get_speaker_name(segment.speaker, participants)
-            segments.append(
+            result.append(
                 TranscriptSegment(
                     speaker=segment.speaker,
                     speaker_name=speaker_name,
@@ -122,4 +130,4 @@ def transcript_to_json_segments(
                 )
             )
 
-    return segments
+    return result

server/reflector/views/transcripts.py

@@ -16,6 +16,7 @@ from pydantic import (
 
 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,
@@ -60,6 +61,14 @@ 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
@@ -360,7 +369,7 @@ class GetTranscriptTopic(BaseModel):
     segments: list[GetTranscriptSegmentTopic] = []
 
     @classmethod
-    def from_transcript_topic(cls, topic: TranscriptTopic):
+    def from_transcript_topic(cls, topic: TranscriptTopic, is_multitrack: bool = False):
         if not topic.words:
             # In previous version, words were missing
             # Just output a segment with speaker 0
@@ -384,7 +393,7 @@ class GetTranscriptTopic(BaseModel):
                     start=segment.start,
                     speaker=segment.speaker,
                 )
-                for segment in transcript.as_segments()
+                for segment in transcript.as_segments(is_multitrack)
             ]
         return cls(
             id=topic.id,
@@ -401,8 +410,8 @@ class GetTranscriptTopicWithWords(GetTranscriptTopic):
     words: list[Word] = []
 
     @classmethod
-    def from_transcript_topic(cls, topic: TranscriptTopic):
+    def from_transcript_topic(cls, topic: TranscriptTopic, is_multitrack: bool = False):
-        instance = super().from_transcript_topic(topic)
+        instance = super().from_transcript_topic(topic, is_multitrack)
         if topic.words:
             instance.words = topic.words
         return instance
@@ -417,8 +426,8 @@ class GetTranscriptTopicWithWordsPerSpeaker(GetTranscriptTopic):
     words_per_speaker: list[SpeakerWords] = []
 
     @classmethod
-    def from_transcript_topic(cls, topic: TranscriptTopic):
+    def from_transcript_topic(cls, topic: TranscriptTopic, is_multitrack: bool = False):
-        instance = super().from_transcript_topic(topic)
+        instance = super().from_transcript_topic(topic, is_multitrack)
         if topic.words:
             words_per_speakers = []
             # group words by speaker
@@ -457,6 +466,8 @@ async def transcript_get(
         transcript_id, user_id=user_id
     )
 
+    is_multitrack = await _get_is_multitrack(transcript)
+
     base_data = {
         "id": transcript.id,
         "user_id": transcript.user_id,
@@ -483,14 +494,16 @@ async def transcript_get(
         return GetTranscriptWithText(
             **base_data,
             transcript_format="text",
-            transcript=transcript_to_text(transcript.topics, transcript.participants),
+            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
+                transcript.topics, transcript.participants, is_multitrack
             ),
         )
     elif transcript_format == "webvtt-named":
@@ -498,7 +511,7 @@ async def transcript_get(
             **base_data,
             transcript_format="webvtt-named",
             transcript=topics_to_webvtt_named(
-                transcript.topics, transcript.participants
+                transcript.topics, transcript.participants, is_multitrack
             ),
         )
     elif transcript_format == "json":
@@ -506,7 +519,7 @@ async def transcript_get(
             **base_data,
             transcript_format="json",
             transcript=transcript_to_json_segments(
-                transcript.topics, transcript.participants
+                transcript.topics, transcript.participants, is_multitrack
             ),
         )
     else:
@@ -565,9 +578,12 @@ 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) for topic in transcript.topics
+        GetTranscriptTopic.from_transcript_topic(topic, is_multitrack)
+        for topic in transcript.topics
     ]
 
 
@@ -584,9 +600,11 @@ 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)
+        GetTranscriptTopicWithWords.from_transcript_topic(topic, is_multitrack)
         for topic in transcript.topics
     ]
 
@@ -605,13 +623,17 @@ 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(topic)
+    return GetTranscriptTopicWithWordsPerSpeaker.from_transcript_topic(
+        topic, is_multitrack
+    )
 
 
 @router.post("/transcripts/{transcript_id}/zulip")

server/reflector/worker/process.py

@@ -2,6 +2,7 @@ import json
 import os
 import re
 from datetime import datetime, timezone
+from typing import List
 from urllib.parse import unquote
 
 import av
@@ -11,7 +12,7 @@ from celery import shared_task
 from celery.utils.log import get_task_logger
 from pydantic import ValidationError
 
-from reflector.dailyco_api import MeetingParticipantsResponse
+from reflector.dailyco_api import RecordingResponse
 from reflector.db.daily_participant_sessions import (
     DailyParticipantSession,
     daily_participant_sessions_controller,
@@ -21,7 +22,6 @@ 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,
-    parse_daily_recording_filename,
+    filter_cam_audio_tracks,
     recording_lock_key,
 )
 from reflector.video_platforms.factory import create_platform_client
@@ -273,15 +273,7 @@ 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 transcript:
+    if not transcript:
-        await transcripts_controller.update(
-            transcript,
-            {
-                "topics": [],
-                "participants": [],
-            },
-        )
-    else:
         transcript = await transcripts_controller.add(
             "",
             source_kind=SourceKind.ROOM,
@@ -294,79 +286,10 @@ 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=track_keys,
+        track_keys=filter_cam_audio_tracks(track_keys),
     )
 
 
@@ -391,7 +314,7 @@ async def poll_daily_recordings():
 
     async with create_platform_client("daily") as daily_client:
         # latest 100. TODO cursor-based state
-        api_recordings = await daily_client.list_recordings()
+        api_recordings: List[RecordingResponse] = await daily_client.list_recordings()
 
     if not api_recordings:
         logger.debug(
@@ -422,17 +345,19 @@ async def poll_daily_recordings():
 
     for recording in missing_recordings:
         if not recording.tracks:
-            assert recording.status != "finished", (
+            if recording.status == "finished":
-                f"Recording {recording.id} has status='finished' but no tracks. "
+                logger.warning(
-                f"Daily.co API guarantees finished recordings have tracks available. "
+                    "Finished recording has no tracks (no audio captured)",
-                f"room_name={recording.room_name}"
+                    recording_id=recording.id,
-            )
+                    room_name=recording.room_name,
-            logger.debug(
+                )
-                "No tracks in recording yet",
+            else:
-                recording_id=recording.id,
+                logger.debug(
-                room_name=recording.room_name,
+                    "No tracks in recording yet",
-                status=recording.status,
+                    recording_id=recording.id,
-            )
+                    room_name=recording.room_name,
+                    status=recording.status,
+                )
             continue
 
         track_keys = [t.s3Key for t in recording.tracks if t.type == "audio"]

server/tests/test_processor_transcript_segment.py

@@ -159,3 +159,78 @@ 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

server/tests/test_transcript_formats.py

@@ -273,8 +273,17 @@ async def test_transcript_formats_with_multiple_speakers():
 
 
 @pytest.mark.asyncio
-async def test_transcript_formats_with_overlapping_speakers():
+async def test_transcript_formats_with_overlapping_speakers_multitrack():
-    """Test format conversion when multiple speakers speak at the same time (overlapping timestamps)."""
+    """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",
@@ -282,11 +291,10 @@ async def test_transcript_formats_with_overlapping_speakers():
             summary="Summary 1",
             timestamp=0.0,
             words=[
-                Word(text="Hello", start=0.0, end=0.5, speaker=0),
+                Word(text="Hello ", start=0.0, end=0.5, speaker=0),
-                Word(text=" there.", start=0.5, end=1.0, speaker=0),
+                Word(text="I'm ", start=0.5, end=0.8, speaker=1),
-                # Speaker 1 overlaps with speaker 0 at 0.5-1.0
+                Word(text="there.", start=0.5, end=1.0, speaker=0),
-                Word(text="I'm", start=0.5, end=1.0, speaker=1),
+                Word(text="good.", start=1.0, end=1.5, speaker=1),
-                Word(text=" good.", start=1.0, end=1.5, speaker=1),
             ],
         ),
     ]
@@ -296,20 +304,9 @@ async def test_transcript_formats_with_overlapping_speakers():
         TranscriptParticipant(id="2", speaker=1, name="Bob"),
     ]
 
-    text_result = transcript_to_text(topics, participants)
+    # With is_multitrack=True, should produce 2 segments (one per speaker sentence)
-    lines = text_result.split("\n")
+    # not 4 segments (one per speaker change)
-    assert len(lines) >= 2
+    webvtt_result = topics_to_webvtt_named(topics, participants, is_multitrack=True)
-    assert any("Alice:" in line for line in lines)
-    assert any("Bob:" in line for line in lines)
-
-    timestamped_result = transcript_to_text_timestamped(topics, participants)
-    timestamped_lines = timestamped_result.split("\n")
-    assert len(timestamped_lines) >= 2
-    assert any("Alice:" in line for line in timestamped_lines)
-    assert any("Bob:" in line for line in timestamped_lines)
-    assert any("[00:00]" in line for line in timestamped_lines)
-
-    webvtt_result = topics_to_webvtt_named(topics, participants)
     expected_webvtt = """WEBVTT
 
 00:00:00.000 --> 00:00:01.000
@@ -320,23 +317,26 @@ async def test_transcript_formats_with_overlapping_speakers():
 """
     assert webvtt_result == expected_webvtt
 
-    segments = transcript_to_json_segments(topics, participants)
+    text_result = transcript_to_text(topics, participants, is_multitrack=True)
-    assert len(segments) >= 2
+    lines = text_result.split("\n")
-    speakers = {seg.speaker for seg in segments}
+    assert len(lines) == 2
-    assert 0 in speakers and 1 in speakers
+    assert "Alice: Hello there." in lines[0]
+    assert "Bob: I'm good." in lines[1]
 
-    alice_seg = next(seg for seg in segments if seg.speaker == 0)
+    timestamped_result = transcript_to_text_timestamped(
-    bob_seg = next(seg for seg in segments if seg.speaker == 1)
+        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]
 
-    # Verify timestamps overlap: Alice (0.0-1.0) and Bob (0.5-1.5) overlap at 0.5-1.0
+    segments = transcript_to_json_segments(topics, participants, is_multitrack=True)
-    assert alice_seg.start < bob_seg.end, "Alice segment should start before Bob ends"
+    assert len(segments) == 2
-    assert bob_seg.start < alice_seg.end, "Bob segment should start before Alice ends"
+    assert segments[0].speaker_name == "Alice"
-
+    assert segments[0].text == "Hello there."
-    overlap_start = max(alice_seg.start, bob_seg.start)
+    assert segments[1].speaker_name == "Bob"
-    overlap_end = min(alice_seg.end, bob_seg.end)
+    assert segments[1].text == "I'm good."
-    assert (
-        overlap_start < overlap_end
-    ), f"Segments should overlap between {overlap_start} and {overlap_end}"
 
 
 @pytest.mark.asyncio
@@ -573,3 +573,207 @@ async def test_api_transcript_format_default_is_text(client):
 
     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

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

@@ -117,15 +117,6 @@ 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
@@ -147,7 +138,12 @@ export default function TranscriptDetails(details: TranscriptDetails) {
               />
             ) : !mp3.loading && (waveform.error || mp3.error) ? (
               <Box p={4} bg="red.100" borderRadius="md">
-                <Text>Error loading this recording</Text>
+                <Text>
+                  Error loading{" "}
+                  {[waveform.error && "waveform", mp3.error && "mp3"]
+                    .filter(Boolean)
+                    .join(" and ")}
+                </Text>
               </Box>
             ) : (
               <Skeleton h={14} />

www/app/[roomName]/components/DailyRoom.tsx

@@ -11,6 +11,7 @@ import {
   recordingTypeRequiresConsent,
 } from "../../lib/consent";
 import { useRoomJoinMeeting } from "../../lib/apiHooks";
+import { assertExists } from "../../lib/utils";
 
 type Meeting = components["schemas"]["Meeting"];
 
@@ -22,16 +23,15 @@ export default function DailyRoom({ meeting }: DailyRoomProps) {
   const router = useRouter();
   const params = useParams();
   const auth = useAuth();
-  const status = auth.status;
+  const authLastUserId = auth.lastUserId;
   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 (status === "loading" || !meeting?.id || !roomName) return;
+    if (authLastUserId === undefined || !meeting?.id || !roomName) return;
 
     const join = async () => {
       try {
@@ -50,18 +50,17 @@ export default function DailyRoom({ meeting }: DailyRoomProps) {
     };
 
     join();
-  }, [meeting?.id, roomName, status]);
+  }, [meeting?.id, roomName, authLastUserId]);
 
   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 (isLoading || !roomUrl || !containerRef.current) return;
+    if (authLastUserId === undefined || !roomUrl || !containerRef.current)
+      return;
 
     let frame: DailyCall | null = null;
     let destroyed = false;
@@ -90,9 +89,14 @@ 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 {
-            await frame.startRecording({ type: "raw-tracks" });
+            assertExists(
+              frame,
+              "frame object got lost somewhere after frame.on was called",
+            ).startRecording({ type: "raw-tracks" });
           } catch (error) {
             console.error("Failed to start recording:", error);
           }
@@ -104,7 +108,9 @@ export default function DailyRoom({ meeting }: DailyRoomProps) {
       }
     };
 
-    createAndJoin();
+    createAndJoin().catch((error) => {
+      console.error("Failed to create and join meeting:", error);
+    });
 
     return () => {
       destroyed = true;
@@ -114,9 +120,9 @@ export default function DailyRoom({ meeting }: DailyRoomProps) {
         });
       }
     };
-  }, [roomUrl, isLoading, handleLeave]);
+  }, [roomUrl, authLastUserId, handleLeave]);
 
-  if (isLoading) {
+  if (authLastUserId === undefined) {
     return (
       <Center width="100vw" height="100vh">
         <Spinner size="xl" />

www/app/lib/AuthProvider.tsx

@@ -1,6 +1,6 @@
 "use client";
 
-import { createContext, useContext } from "react";
+import { createContext, useContext, useRef } from "react";
 import { useSession as useNextAuthSession } from "next-auth/react";
 import { signOut, signIn } from "next-auth/react";
 import { configureApiAuth } from "./apiClient";
@@ -25,6 +25,9 @@ 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);
@@ -41,10 +44,15 @@ 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
     ? {
@@ -73,11 +81,16 @@ 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,
@@ -92,6 +105,8 @@ export function AuthProvider({ children }: { children: React.ReactNode }) {
               }
             }
             case "unauthenticated": {
+              // warning: call order-dependent
+              lastUserId.current = null;
               return { status: "unauthenticated" as const };
             }
             default: {
@@ -103,6 +118,8 @@ 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;
 

www/app/lib/authBackend.ts

@@ -148,7 +148,7 @@ export const authOptions = (): AuthOptions =>
           },
           async session({ session, token }) {
             const extendedToken = token as JWTWithAccessToken;
-
+            console.log("extendedToken", extendedToken);
             const userId = await getUserId(extendedToken.accessToken);
 
             return {