Kakao i Connect Live 커뮤니티

eliot.en · 1 September 2020 03:33

안드로이드 앱에서 알림을 받고 통화 연결을 하면 마이크가 활성화 되지 않는것 같습니다.
이번 테스트는 아니지만 이전 테스트에는 웹에서 호출한 목소리만 드리고 안드로이드 앱의 소리는 전달되지 않습니다.

아래 코드를 보면 “AAudio support is currently disabled on all devices!” 이런 내용이 있습니다.
어떤 설정을 추가하면 되나요?

09-01 12:22:58.648 … I/org.webrtc.Logging: WebRtcAudioManager: ctor@[name=Thread-45439, id=45439]
09-01 12:22:58.648 … I/org.webrtc.Logging: WebRtcAudioManager: Sample rate is set to 48000 Hz
09-01 12:22:58.658 … I/org.webrtc.Logging: WebRtcAudioEffects: canUseAcousticEchoCanceler: true
09-01 12:22:58.658 … I/org.webrtc.Logging: WebRtcAudioEffects: canUseNoiseSuppressor: true
09-01 12:22:58.668 … W/org.webrtc.Logging: WebRtcAudioManager: AAudio support is currently disabled on all devices!
09-01 12:22:58.668 … I/org.webrtc.Logging: WebRtcAudioManager: Android SDK: 23, Release: 6.0.1, Brand: samsung, Device: lentisltektt, Id: MMB29M, Hardware: qcom, Manufacturer: samsung, Model: SM-G906K, Product: lentisltektt
09-01 12:22:58.668 … I/org.webrtc.Logging: WebRtcAudioManager: Audio State: audio mode: MODE_IN_COMMUNICATION, has mic: true, mic muted: false, music active: false, speakerphone: true, BT SCO: false
09-01 12:22:58.668 … I/org.webrtc.Logging: WebRtcAudioManager: Audio State:
09-01 12:22:58.668 … I/org.webrtc.Logging: WebRtcAudioManager: fixed volume=false
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioManager: STREAM_VOICE_CALL: volume=5, max=5, muted=false
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioManager: STREAM_MUSIC: volume=10, max=15, muted=false
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioManager: STREAM_RING: volume=14, max=15, muted=false
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioManager: STREAM_ALARM: volume=11, max=15, muted=false
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioManager: STREAM_NOTIFICATION: volume=11, max=15, muted=false
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioManager: STREAM_SYSTEM: volume=11, max=15, muted=false
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioManager: Audio Devices:
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioManager: TYPE_BUILTIN_EARPIECE(out): id=1
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioManager: TYPE_BUILTIN_SPEAKER(out): id=2
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioManager: TYPE_BUILTIN_MIC(in): channels=[1, 2], encodings=[2], sample rates=[8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000], id=5
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioManager: TYPE_TELEPHONY(in): channels=[1, 2], encodings=[2], sample rates=[8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000], id=6
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioManager: SM-G906K is blacklisted for OpenSL ES usage!
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioManager: SM-G906K is blacklisted for OpenSL ES usage!
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioTrack: ctor@[name=Thread-45439, id=45439]
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioRecord: ctor@[name=Thread-45439, id=45439]
09-01 12:22:58.678 … I/org.webrtc.Logging: WebRtcAudioEffects: ctor@[name=Thread-45439, id=45439]
09-01 12:22:58.688 … I/org.webrtc.Logging: WebRtcAudioManager: init@[name=Thread-45439, id=45439]
09-01 12:22:58.688 … I/org.webrtc.Logging: WebRtcAudioManager: audio mode is: MODE_IN_COMMUNICATION
09-01 12:22:58.688 … I/org.webrtc.Logging: WebRtcAudioRecord: enableBuiltInAEC(true)
09-01 12:22:58.688 … I/org.webrtc.Logging: WebRtcAudioEffects: setAEC(true)
09-01 12:22:58.688 … I/org.webrtc.Logging: WebRtcAudioEffects: canUseAcousticEchoCanceler: true
09-01 12:22:58.688 … I/org.webrtc.Logging: WebRtcAudioRecord: enableBuiltInNS(true)
09-01 12:22:58.688 … I/org.webrtc.Logging: WebRtcAudioEffects: setNS(true)
09-01 12:22:58.688 … I/org.webrtc.Logging: WebRtcAudioEffects: canUseNoiseSuppressor: true
09-01 12:22:58.688 … I/org.webrtc.Logging: PeerConnectionFactory: onWorkerThreadReady
09-01 12:22:58.688 … I/org.webrtc.Logging: PeerConnectionFactory: onNetworkThreadReady
09-01 12:22:58.688 … I/org.webrtc.Logging: PeerConnectionFactory: onSignalingThreadReady
09-01 12:22:58.688 … D/WebSocketClient: connectChannel() - chId=1598930566935_vglwu
09-01 12:22:58.688 … I/org.webrtc.Logging: WebRtcAudioManager: VOICE_CALL stream volume: 5 (max=5)
09-01 12:22:58.698 … D/AppRTCAudioManager: WiredHeadsetReceiver.onReceive@[name=main, id=1]: a=android.intent.action.HEADSET_PLUG, s=unplugged, m=mic, n=null, sb=true

eliot.en · 2 September 2020 05:09

안드로이드 문제가 아니라 웹에 video 태그 대신에 audio 태그를 넣어야 하는 문제 였습니다. ㅜㅜ

view: {
remote: '#remoteVideo’,
local: ‘#localVieo’,
},
media: {
audio: true,
video: false,
}

github.com

RemoteMonster/web-sdk/blob/master/examples/full/voiceCall.html

<!DOCTYPE html>
<html lang="en">
<head>
    <meta charset="utf-8">
    <meta http-equiv="X-UA-Compatible" content="IE=edge">
    <meta name="viewport" content="width=device-width, initial-scale=1, shrink-to-fit=no">
    <meta name="description" content="RemoteMonster live media service demo">
    <meta name="author" content="Lucas Choi">

    <title>Voice Call</title>

    <!-- Custom fonts for this template-->
    <link href="vendor/fontawesome-free/css/all.min.css" rel="stylesheet" type="text/css">
    <link href="https://fonts.googleapis.com/css?family=Nunito:200,200i,300,300i,400,400i,600,600i,700,700i,800,800i,900,900i"
          rel="stylesheet">

    <!-- Custom styles for this template-->
    <link href="css/sb-admin-2.min.css" rel="stylesheet">
</head>

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calmglow · 17 September 2020 02:21

감사합니다

안드로이드 마이크 활성화 되지 않습니다