Audio routes through call speaker instead of media speaker when joining as a viewer-only participant
维护者通常 1 天内回复
还没有人认领这个 Issue。
评估
- 难度
- 4/5
- 预计耗时
- 3-5 天
- 新手友好度
- 38/100
- Issue 类型
- 功能
- 描述清晰度
- 基本清楚
- 活跃度
- 冷清
- 技术栈
- flutter
调研方向
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.
由索引模型根据 Issue 内容生成。
描述
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.
- 主要语言
- Dart
- 星标
- 421
- 派生
- 260
- 平均合并
- 3 天 9 小时
- 30 天内合并 PR
- 11
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livekit/client-sdk-flutter 的其他 Issue
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难度 4/5 3-5 天 新手友好度 48/100
livekit/client-sdk-flutter#1227 ·
维护者通常 1 天内回复
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难度 4/5 3-5 天 新手友好度 45/100
livekit/client-sdk-flutter#1221 · 1 条评论 ·
维护者通常 1 天内回复
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难度 4/5 3-5 天 新手友好度 30/100
livekit/client-sdk-flutter#1219 · 1 个 reaction ·
维护者通常 1 天内回复
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iOS: `audioDeviceModule` becomes permanently `nil`/unavailable after backgrounding, even though `AVAudioSession` is never interrupted可能已有人在做 @hiroshihorie 于 31 天前认领。 未关闭
livekit/client-sdk-flutter#1206 · 2 条评论 · 已指派 1 人 ·
维护者通常 1 天内回复
-
[bug]Early reconnect during the initial room connection可能重新可做 @hiroshihorie 于 42 天前认领，目前没有进行中的 PR。 未关闭
livekit/client-sdk-flutter#1189 · 已指派 1 人 ·
维护者通常 1 天内回复
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