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Side by Side Diff: webrtc/video_engine/vie_sync_module.cc

Issue 1507903005: Revert of Merge webrtc/video_engine/ into webrtc/video/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Resolved merge conflict Created 5 years ago
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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/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/logging.h"
14 #include "webrtc/base/trace_event.h" 14 #include "webrtc/base/trace_event.h"
15 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" 15 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
16 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" 16 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
17 #include "webrtc/modules/video_coding/include/video_coding.h" 17 #include "webrtc/modules/video_coding/include/video_coding.h"
18 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" 18 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
19 #include "webrtc/video/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))
29 return -1; 29 return -1;
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165 165
166 if (voe_sync_interface_->SetMinimumPlayoutDelay( 166 if (voe_sync_interface_->SetMinimumPlayoutDelay(
167 voe_channel_id_, target_audio_delay_ms) == -1) { 167 voe_channel_id_, target_audio_delay_ms) == -1) {
168 LOG(LS_ERROR) << "Error setting voice delay."; 168 LOG(LS_ERROR) << "Error setting voice delay.";
169 } 169 }
170 vcm_->SetMinimumPlayoutDelay(target_video_delay_ms); 170 vcm_->SetMinimumPlayoutDelay(target_video_delay_ms);
171 return 0; 171 return 0;
172 } 172 }
173 173
174 } // namespace webrtc 174 } // namespace webrtc
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