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Audio routes through call speaker instead of media speaker when joining as a viewer-only participant

Aperta
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Valutazione

Difficoltà
4/5
Tempo stimato
3-5 giorni
Idoneità per principianti
38/100
Tipo di issue
Funzionalità
Chiarezza
Abbastanza chiara
Stato di attività
Tranquilla
Stack tecnologico
flutter
Ambito
audio-video-rtc, mobile

Direzione di ricerca

Start by reading the audio-routing setup around RoomOptions and Hardware.instance, then trace how the SDK handles audio sessions on Android and iOS when a viewer joins without publishing tracks. The issue does not name files or tests, so first locate those entry points and relevant platform tests. Done means receive-only participants use media playback routing without breaking audio for participants who publish tracks.

Scritto dal modello di indicizzazione a partire dal testo della issue.

Descrizione

Is your feature request related to a problem? Please describe.
When joining a livestream session as a pure viewer (no microphone, no camera), the audio is incorrectly routed through the device's call/earpiece speaker instead of the media/loudspeaker — the same routing used for phone calls.

This occurs even though the viewer is not publishing any audio or video track. The expected behavior is that a participant in a receive-only role should use media playback audio routing, consistent with how any video player or media app behaves. Instead, LiveKit appears to initialize WebRTC in MODE_IN_COMMUNICATION (Android) / AVAudioSessionCategory.playAndRecord (iOS) regardless of whether the local participant is actually publishing.

Describe the solution you'd like
When a participant joins with no published tracks (viewer-only / subscriber-only role), LiveKit should automatically apply media playback audio routing:

• Android: Use AudioManager.MODE_NORMAL with loudspeaker, not MODE_IN_COMMUNICATION
• iOS: Use AVAudioSessionCategory.playback instead of playAndRecord

Ideally, expose a participant role hint (e.g. ParticipantKind.viewer or a audioMode option in RoomOptions) so developers can explicitly declare the intended audio routing behavior at connection time, before WebRTC initializes the audio session.

Describe alternatives you've considered
• Calling Hardware.instance.setSpeakerphoneOn(true) before room.connect() — overridden internally when WebRTC initializes the audio session on first track subscription.

• Manually reconfiguring AudioSession after the first remote audio track is subscribed — partially works but requires a delay and is fragile, as LiveKit re-acquires the audio session asynchronously.

• Using AudioOutputOptions(speakerOn: true) in RoomOptions — does not change the underlying audio mode from communication to media playback.

Additional context
• SDK: livekit_client (Flutter)
• Platforms affected: Android & iOS
• Use case: Livestreaming with a single host and many viewer-only participants
• Root cause: WebRTC initializes the audio stack in voice communication mode unconditionally, even for receive-only participants. This is by design in WebRTC but should be overridden by LiveKit when no local tracks are published.

Lingua principale
Dart
Stelle
421
Fork
260
Merge medio
2g 5h
PR unite (30g)
17

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