Open gavrilikhin-d opened 1 year ago
When InCallManager is started in WebRTC call, it somehow creates lag on unmuting microphone on iOS. Without it everything works fine.
InCallManager
Came from: https://github.com/react-native-webrtc/react-native-webrtc/issues/1424
This problem appears on latest version (4.1.0) too
4.1.0
Any updates on this issue, guys? Is there anything that can be done to work around it?
When
InCallManager
is started in WebRTC call, it somehow creates lag on unmuting microphone on iOS. Without it everything works fine.Unmuting with native logs
``` [javascript] setting muted to false (webrtc_voice_engine.cc:1486): Setting voice channel options: AudioOptions {} (webrtc_voice_engine.cc:424): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, } (webrtc_voice_engine.cc:443): Always disable AEC on iOS. Use built-in instead. (webrtc_voice_engine.cc:453): Always disable AGC on iOS. Use built-in instead. (audio_device_generic.cc:18): BuiltInAECIsAvailable: Not supported on this platform (audio_device_generic.cc:28): BuiltInAGCIsAvailable: Not supported on this platform (audio_processing_impl.cc:911): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller: { enabled: 0, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 0, level: High }, transient_suppression: { enabled: 0 }, gain_controller1: { enabled: 0, mode: FixedDigital, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_gain_controller { enabled: 1, startup_min_volume: 0, clipped_level_min: 70, enable_digital_adaptive: 1, clipped_level_step: 15, clipped_ratio_threshold: 0.1, clipped_wait_frames: 300, clipping_predictor: { enabled: 0, mode: 0, window_length: 5, reference_window_length: 5, reference_window_delay: 5, clipping_threshold: -1, crest_factor_margin: 3, use_predicted_step: 1 }}}, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, headroom_db: 6, max_gain_db: 30, initial_gain_db: 8, max_gain_change_db_per_second: 3, max_output_noise_level_dbfs: -50 }, input_volume_control : { enabled 0}} (webrtc_voice_engine.cc:1500): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, } Start Monitoring (RTCLogging.mm:33): (audio_device_ios.mm:444 OnGetPlayoutData): Possible playout audio glitch detected. Time since last OnGetPlayoutData was 123 ms. (RTCLogging.mm:33): (audio_device_ios.mm:453 OnGetPlayoutData): Glitch warning is ignored. Probably caused by device switch. (RTCLogging.mm:33): (RTCAudioSession.mm:855 -[RTCAudioSession observeValueForKeyPath:ofObject:change:context:]): OutputVolumeDidChange to 0.500000 (RTCLogging.mm:33): (RTCAudioSession.mm:531 -[RTCAudioSession handleRouteChangeNotification:]): Audio route changed: (RTCLogging.mm:33): (RTCAudioSession.mm:544 -[RTCAudioSession handleRouteChangeNotification:]): Audio route changed: CategoryChange to :AVAudioSessionCategoryMultiRoute (RTCLogging.mm:33): (RTCAudioSession.mm:563 -[RTCAudioSession handleRouteChangeNotification:]): Previous route:Came from: https://github.com/react-native-webrtc/react-native-webrtc/issues/1424