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Searched refs:RTCLog (Results 1 – 16 of 16) sorted by relevance

/external/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/
DARDBroadcastSampleHandler.m31 os_log_t rtc_os_log = os_log_create("com.google.AppRTCMobile", "RTCLog");
55 RTCLog(@"Broadcast started.");
91 RTCLog(@"Client connected.");
94 RTCLog("Client connecting.");
97 RTCLog(@"Client disconnected.");
103 RTCLog(@"ICE state changed: %ld", (long)state);
127 RTCLog(@"Error: %@", error);
/external/webrtc/sdk/objc/components/audio/
DRTCAudioSession.mm107 RTCLog(@"RTC_OBJC_TYPE(RTCAudioSession) (%p): init.", self);
117 RTCLog(@"RTC_OBJC_TYPE(RTCAudioSession) (%p): dealloc.", self);
209 RTCLog(@"Adding delegate: (%p)", delegate);
220 RTCLog(@"Removing delegate: (%p)", delegate);
405 RTCLog(@"Number of current activations: %d", _activationCount);
503 RTCLog(@"Audio session interruption began.");
509 RTCLog(@"Audio session interruption ended.");
530 RTCLog(@"Audio route changed:");
533 RTCLog(@"Audio route changed: ReasonUnknown");
536 RTCLog(@"Audio route changed: NewDeviceAvailable");
[all …]
DRTCAudioSession+Configuration.mm65 RTCLog(@"Set category to: %@", configuration.category);
76 RTCLog(@"Set mode to: %@", configuration.mode);
90 RTCLog(@"Set category options to: %ld",
105 RTCLog(@"Set preferred sample rate to: %.2f",
120 RTCLog(@"Set preferred IO buffer duration to: %f",
151 RTCLog(@"Set input number of channels to: %ld",
166 RTCLog(@"Set output number of channels to: %ld",
DRTCNativeAudioSessionDelegateAdapter.mm62 RTCLog(@"Ignoring RouteConfigurationChange");
/external/webrtc/examples/objc/AppRTCMobile/
DARDWebSocketChannel.m46 RTCLog(@"Opening WebSocket.");
93 RTCLog(@"C->WSS: %@", messageString);
98 RTCLog(@"C->WSS POST: %@", dataString);
115 RTCLog(@"C->WSS DELETE rid:%@ cid:%@", _roomId, _clientId);
129 RTCLog(@"WebSocket connection opened.");
155 RTCLog(@"WSS->C: %@", payload);
168 RTCLog(@"WebSocket closed with code: %ld reason:%@ wasClean:%d",
193 RTCLog(@"Registering on WSS for rid:%@ cid:%@", _roomId, _clientId);
DARDAppEngineClient.m55 RTCLog(@"Joining room:%@ on room server.", roomId);
94 RTCLog(@"C->RS POST: %@", message);
135 RTCLog(@"C->RS: BYE");
157 RTCLog(@"Left room:%@ on room server.", roomId);
DARDAppClient.m273 RTCLog(@"Joined room:%@ on room server.", roomId);
377 RTCLog(@"Signaling state changed: %ld", (long)stateChanged);
382 RTCLog(@"Stream with %lu video tracks and %lu audio tracks was added.",
390 RTCLog(@"Now receiving %@ on track %@.", track.kind, track.trackId);
395 RTCLog(@"Stream was removed.");
399 RTCLog(@"WARNING: Renegotiation needed but unimplemented.");
404 RTCLog(@"ICE state changed: %ld", (long)newState);
412 RTCLog(@"ICE+DTLS state changed: %ld", (long)newState);
417 RTCLog(@"ICE gathering state changed: %ld", (long)newState);
444 RTCLog(@"ICE candidate pair changed because: %@", reason);
/external/webrtc/sdk/objc/native/src/audio/
Daudio_device_ios.mm270 RTCLog(@"Average number of playout callbacks between glitches: %d",
458 RTCLog(@"Glitch warning is ignored. Probably caused by device switch.");
503 RTCLog(@"Interruption begin. IsInterrupted changed from %d to 1.", is_interrupted_);
505 RTCLog(@"Stopping the audio unit due to interruption begin.");
517 RTCLog(@"Interruption ended. IsInterrupted changed from %d to 0. "
541 RTCLog(@"%@", session);
546 RTCLog(@"Handling CanPlayOrRecord change to: %d", can_play_or_record);
552 RTCLog(@"Handling sample rate change to %f.", sample_rate);
556 RTCLog(@"Ignoring sample rate change to %f due to interruption.", sample_rate);
575 RTCLog(@"Handling playout sample rate change to: %f\n"
[all …]
Dvoice_processing_audio_unit.mm26 RTCLog(@"AudioStreamBasicDescription: {\n"
71 RTCLog(@"VPIO unit AGC: %u", static_cast<unsigned int>(*enabled));
196 RTCLog(@"Initializing audio unit with sample rate: %f", sample_rate);
245 RTCLog(@"Pause 100ms and try audio unit initialization again...");
250 RTCLog(@"Voice Processing I/O unit is now initialized.");
301 RTCLog(@"WebRTC.Audio.BuiltInAGCWasEnabledByDefault: %d",
307 RTCLog(@"WebRTC.Audio.BuiltInAGCIsEnabled: %u",
316 RTCLog(@"Starting audio unit.");
323 RTCLog(@"Started audio unit");
331 RTCLog(@"Stopping audio unit.");
[all …]
/external/webrtc/sdk/objc/components/renderer/opengl/
DRTCDefaultShader.mm100 RTCLog(@"Failed to get uniform variable locations in I420 shader");
124 RTCLog(@"Failed to get uniform variable locations in NV12 shader");
139 RTCLog(@"Failed to setup vertex buffer");
164 RTCLog(@"Failed to setup I420 program");
192 RTCLog(@"Failed to setup NV12 shader");
/external/webrtc/examples/objc/AppRTCMobile/ios/
DARDVideoCallViewController.m79 RTCLog(@"Client connected.");
82 RTCLog(@"Client connecting.");
85 RTCLog(@"Client disconnected.");
93 RTCLog(@"ICE state changed: %ld", (long)state);
192 RTCLog(@"Audio session detected glitch, total: %lld", totalNumberOfGlitches);
DARDMainViewController.m156 RTCLog(@"Dismissing VC");
173 RTCLog(@"Stopping audio loop due to WebRTC start.");
176 RTCLog(@"Setting isAudioEnabled to YES.");
185 RTCLog(@"audioSessionDidStopPlayOrRecord");
243 RTCLog(@"Starting audio loop due to WebRTC end.");
/external/webrtc/sdk/objc/components/capturer/
DRTCCameraVideoCapturer.m338 RTCLog(@"Capture session interrupted: %@", reasonString);
342 RTCLog(@"Capture session interruption ended.");
364 RTCLog(@"Capture session started.");
375 RTCLog(@"Capture session stopped.");
395 RTCLog(@"Restarting capture session after error.");
411 RTCLog(@"Restarting capture session on active.");
DRTCFileVideoCapturer.m100 RTCLog(@"File capturer started reading");
105 RTCLog(@"File capturer stopped.");
/external/webrtc/sdk/objc/base/
DRTCLogging.h67 #define RTCLog(format, ...) RTCLogInfo(format, ##__VA_ARGS__) macro
/external/webrtc/sdk/objc/api/peerconnection/
DRTCIceCandidate.mm68 RTCLog(@"Failed to create ICE candidate: %s\nline: %s",