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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #include "webrtc/video_engine/vie_sync_module.h" | 11 #include "webrtc/video_engine/vie_sync_module.h" |
12 | 12 |
| 13 #include "webrtc/base/trace_event.h" |
13 #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" | 14 #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" |
14 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" | 15 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" |
15 #include "webrtc/modules/video_coding/main/interface/video_coding.h" | 16 #include "webrtc/modules/video_coding/main/interface/video_coding.h" |
16 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" | 17 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
17 #include "webrtc/system_wrappers/interface/logging.h" | 18 #include "webrtc/system_wrappers/interface/logging.h" |
18 #include "webrtc/system_wrappers/interface/trace_event.h" | |
19 #include "webrtc/video_engine/stream_synchronization.h" | 19 #include "webrtc/video_engine/stream_synchronization.h" |
20 #include "webrtc/voice_engine/include/voe_video_sync.h" | 20 #include "webrtc/voice_engine/include/voe_video_sync.h" |
21 | 21 |
22 namespace webrtc { | 22 namespace webrtc { |
23 | 23 |
24 int UpdateMeasurements(StreamSynchronization::Measurements* stream, | 24 int UpdateMeasurements(StreamSynchronization::Measurements* stream, |
25 const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) { | 25 const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) { |
26 if (!receiver.Timestamp(&stream->latest_timestamp)) | 26 if (!receiver.Timestamp(&stream->latest_timestamp)) |
27 return -1; | 27 return -1; |
28 if (!receiver.LastReceivedTimeMs(&stream->latest_receive_time_ms)) | 28 if (!receiver.LastReceivedTimeMs(&stream->latest_receive_time_ms)) |
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179 } | 179 } |
180 sync_->SetTargetBufferingDelay(target_delay_ms); | 180 sync_->SetTargetBufferingDelay(target_delay_ms); |
181 // Setting initial playout delay to voice engine (video engine is updated via | 181 // Setting initial playout delay to voice engine (video engine is updated via |
182 // the VCM interface). | 182 // the VCM interface). |
183 voe_sync_interface_->SetInitialPlayoutDelay(voe_channel_id_, | 183 voe_sync_interface_->SetInitialPlayoutDelay(voe_channel_id_, |
184 target_delay_ms); | 184 target_delay_ms); |
185 return 0; | 185 return 0; |
186 } | 186 } |
187 | 187 |
188 } // namespace webrtc | 188 } // namespace webrtc |
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