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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/logging.h" |
| 13 #include "webrtc/base/trace_event.h" | 14 #include "webrtc/base/trace_event.h" |
| 14 #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" | 15 #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" |
| 15 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" | 16 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" |
| 16 #include "webrtc/modules/video_coding/main/interface/video_coding.h" | 17 #include "webrtc/modules/video_coding/main/interface/video_coding.h" |
| 17 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" | 18 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| 18 #include "webrtc/system_wrappers/interface/logging.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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