| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| rtc_event_log_source.cc | 35 const rtclog::RtpPacket* GetRtpPacket(const rtclog::Event& event) {
36 if (!event.has_type() || event.type() != rtclog::Event::RTP_EVENT)
40 const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
41 if (!rtp_packet.has_type() || rtp_packet.type() != rtclog::AUDIO ||
50 const rtclog::AudioPlayoutEvent* GetAudioPlayoutEvent(
51 const rtclog::Event& event) {
52 if (!event.has_type() || event.type() != rtclog::Event::AUDIO_PLAYOUT_EVENT)
56 const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
80 const rtclog::Event& event = event_log_->stream(rtp_packet_index_)
[all...] |
| rtc_event_log_source.h | 25 namespace rtclog { namespace in namespace:webrtc
27 } // namespace rtclog
61 rtc::scoped_ptr<rtclog::EventStream> event_log_;
|
| /external/webrtc/webrtc/call/ |
| rtc_event_log.cc | 98 void HandleEvent(rtclog::Event* event) EXCLUSIVE_LOCKS_REQUIRED(crit_);
101 void StoreToFile(rtclog::Event* event) EXCLUSIVE_LOCKS_REQUIRED(crit_);
104 void AddRecentEvent(const rtclog::Event& event)
112 rtclog::EventStream stream_ GUARDED_BY(crit_);
113 std::deque<rtclog::Event> recent_log_events_ GUARDED_BY(crit_);
114 std::vector<rtclog::Event> config_events_ GUARDED_BY(crit_);
133 rtclog::VideoReceiveConfig_RtcpMode ConvertRtcpMode(RtcpMode rtcp_mode) {
136 return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
138 return rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE;
141 return rtclog::VideoReceiveConfig::RTCP_COMPOUND
[all...] |
| rtc_event_log_unittest.cc | 58 MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
60 case rtclog::MediaType::ANY:
62 case rtclog::MediaType::AUDIO:
64 case rtclog::MediaType::VIDEO:
66 case rtclog::MediaType::DATA:
75 ::testing::AssertionResult IsValidBasicEvent(const rtclog::Event& event) {
80 rtclog::Event_EventType type = event.type();
81 if ((type == rtclog::Event::RTP_EVENT) != event.has_rtp_packet())
85 if ((type == rtclog::Event::RTCP_EVENT) != event.has_rtcp_packet())
89 if ((type == rtclog::Event::AUDIO_PLAYOUT_EVENT) !
[all...] |
| rtc_event_log2rtp_dump.cc | 97 webrtc::rtclog::EventStream event_stream;
120 const webrtc::rtclog::Event& event = event_stream.stream(i);
127 const webrtc::rtclog::RtpPacket& rtp_packet = event.rtp_packet();
128 if (FLAGS_noaudio && rtp_packet.type() == webrtc::rtclog::AUDIO)
130 if (FLAGS_novideo && rtp_packet.type() == webrtc::rtclog::VIDEO)
132 if (FLAGS_nodata && rtp_packet.type() == webrtc::rtclog::DATA)
167 const webrtc::rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
168 if (FLAGS_noaudio && rtcp_packet.type() == webrtc::rtclog::AUDIO)
170 if (FLAGS_novideo && rtcp_packet.type() == webrtc::rtclog::VIDEO)
172 if (FLAGS_nodata && rtcp_packet.type() == webrtc::rtclog::DATA
[all...] |
| rtc_event_log.h | 25 namespace rtclog { namespace in namespace:webrtc
27 } // namespace rtclog
88 rtclog::EventStream* result);
|
| rtc_event_log.proto | 3 package webrtc.rtclog;
|
| /external/webrtc/webrtc/examples/objc/AppRTCDemo/ |
| ARDWebSocketChannel.m | 46 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);
|
| ARDSDPUtils.m | 57 RTCLog(@"No m=video line, so can't prefer %@", codec);
61 RTCLog(@"No rtpmap for %@", codec);
|
| ARDAppEngineClient.m | 55 RTCLog(@"Joining room:%@ on room server.", roomId);
99 RTCLog(@"C->RS POST: %@", message);
145 RTCLog(@"C->RS: BYE");
157 RTCLog(@"Left room:%@ on room server.", roomId);
|
| ARDAppClient.m | 243 RTCLog(@"Joined room:%@ on room server.", roomId);
335 RTCLog(@"Signaling state changed: %d", stateChanged);
341 RTCLog(@"Received %lu video tracks and %lu audio tracks",
353 RTCLog(@"Stream was removed.");
358 RTCLog(@"WARNING: Renegotiation needed but unimplemented.");
363 RTCLog(@"ICE state changed: %d", newState);
371 RTCLog(@"ICE gathering state changed: %d", newState);
|
| /external/webrtc/webrtc/examples/objc/AppRTCDemo/ios/ |
| ARDVideoCallViewController.m | 60 RTCLog(@"Client connected.");
63 RTCLog(@"Client connecting.");
66 RTCLog(@"Client disconnected.");
74 RTCLog(@"ICE state changed: %d", state);
|
| /external/webrtc/webrtc/api/objc/ |
| RTCIceCandidate.mm | 62 RTCLog(@"Failed to create ICE candidate: %s\nline: %s",
|
| /external/webrtc/webrtc/base/objc/ |
| RTCLogging.h | 75 #define RTCLog(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
|
| /external/webrtc/talk/app/webrtc/objc/public/ |
| RTCLogging.h | 92 #define RTCLog(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
|
