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Issue 1413333002: system_wrappers: rename interface -> include (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased again! Created 5 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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/video_receive_stream.h" 11 #include "webrtc/video/video_receive_stream.h"
12 12
13 #include <stdlib.h> 13 #include <stdlib.h>
14 14
15 #include <string> 15 #include <string>
16 16
17 #include "webrtc/base/checks.h" 17 #include "webrtc/base/checks.h"
18 #include "webrtc/base/logging.h" 18 #include "webrtc/base/logging.h"
19 #include "webrtc/call/congestion_controller.h" 19 #include "webrtc/call/congestion_controller.h"
20 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" 20 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
21 #include "webrtc/system_wrappers/interface/clock.h" 21 #include "webrtc/system_wrappers/include/clock.h"
22 #include "webrtc/video/receive_statistics_proxy.h" 22 #include "webrtc/video/receive_statistics_proxy.h"
23 #include "webrtc/video_engine/call_stats.h" 23 #include "webrtc/video_engine/call_stats.h"
24 #include "webrtc/video_receive_stream.h" 24 #include "webrtc/video_receive_stream.h"
25 25
26 namespace webrtc { 26 namespace webrtc {
27 27
28 static bool UseSendSideBwe(const std::vector<RtpExtension>& extensions) { 28 static bool UseSendSideBwe(const std::vector<RtpExtension>& extensions) {
29 for (const auto& extension : extensions) { 29 for (const auto& extension : extensions) {
30 if (extension.name == RtpExtension::kTransportSequenceNumber) 30 if (extension.name == RtpExtension::kTransportSequenceNumber)
31 return true; 31 return true;
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378 return 0; 378 return 0;
379 } 379 }
380 380
381 void VideoReceiveStream::SignalNetworkState(NetworkState state) { 381 void VideoReceiveStream::SignalNetworkState(NetworkState state) {
382 vie_channel_->SetRTCPMode(state == kNetworkUp ? config_.rtp.rtcp_mode 382 vie_channel_->SetRTCPMode(state == kNetworkUp ? config_.rtp.rtcp_mode
383 : RtcpMode::kOff); 383 : RtcpMode::kOff);
384 } 384 }
385 385
386 } // namespace internal 386 } // namespace internal
387 } // namespace webrtc 387 } // namespace webrtc
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