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Side by Side Diff: webrtc/video/video_receive_stream.cc

Issue 1442003002: Remove BitrateController dependency fromVideoReceiveStream. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: 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/modules/bitrate_controller/include/bitrate_controller.h"
22 #include "webrtc/system_wrappers/include/clock.h" 21 #include "webrtc/system_wrappers/include/clock.h"
23 #include "webrtc/video/receive_statistics_proxy.h" 22 #include "webrtc/video/receive_statistics_proxy.h"
24 #include "webrtc/video_engine/call_stats.h" 23 #include "webrtc/video_engine/call_stats.h"
25 #include "webrtc/video_receive_stream.h" 24 #include "webrtc/video_receive_stream.h"
26 25
27 namespace webrtc { 26 namespace webrtc {
28 27
29 static bool UseSendSideBwe(const std::vector<RtpExtension>& extensions) { 28 static bool UseSendSideBwe(const std::vector<RtpExtension>& extensions) {
30 for (const auto& extension : extensions) { 29 for (const auto& extension : extensions) {
31 if (extension.name == RtpExtension::kTransportSequenceNumber) 30 if (extension.name == RtpExtension::kTransportSequenceNumber)
(...skipping 122 matching lines...) Expand 10 before | Expand all | Expand 10 after
154 call_stats_(call_stats) { 153 call_stats_(call_stats) {
155 LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString(); 154 LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
156 155
157 bool send_side_bwe = UseSendSideBwe(config_.rtp.extensions); 156 bool send_side_bwe = UseSendSideBwe(config_.rtp.extensions);
158 157
159 RemoteBitrateEstimator* bitrate_estimator = 158 RemoteBitrateEstimator* bitrate_estimator =
160 congestion_controller_->GetRemoteBitrateEstimator(send_side_bwe); 159 congestion_controller_->GetRemoteBitrateEstimator(send_side_bwe);
161 160
162 vie_channel_.reset(new ViEChannel( 161 vie_channel_.reset(new ViEChannel(
163 num_cpu_cores, &transport_adapter_, process_thread, nullptr, 162 num_cpu_cores, &transport_adapter_, process_thread, nullptr,
164 congestion_controller_->GetBitrateController()-> 163 nullptr, nullptr, bitrate_estimator, call_stats_->rtcp_rtt_stats(),
165 CreateRtcpBandwidthObserver(),
166 nullptr, bitrate_estimator, call_stats_->rtcp_rtt_stats(),
167 congestion_controller_->pacer(), congestion_controller_->packet_router(), 164 congestion_controller_->pacer(), congestion_controller_->packet_router(),
168 1, false)); 165 1, false));
169 166
170 RTC_CHECK(vie_channel_->Init() == 0); 167 RTC_CHECK(vie_channel_->Init() == 0);
171 168
172 // Register the channel to receive stats updates. 169 // Register the channel to receive stats updates.
173 call_stats_->RegisterStatsObserver(vie_channel_->GetStatsObserver()); 170 call_stats_->RegisterStatsObserver(vie_channel_->GetStatsObserver());
174 171
175 // TODO(pbos): This is not fine grained enough... 172 // TODO(pbos): This is not fine grained enough...
176 vie_channel_->SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0, false, 173 vie_channel_->SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0, false,
(...skipping 202 matching lines...) Expand 10 before | Expand all | Expand 10 after
379 return 0; 376 return 0;
380 } 377 }
381 378
382 void VideoReceiveStream::SignalNetworkState(NetworkState state) { 379 void VideoReceiveStream::SignalNetworkState(NetworkState state) {
383 vie_channel_->SetRTCPMode(state == kNetworkUp ? config_.rtp.rtcp_mode 380 vie_channel_->SetRTCPMode(state == kNetworkUp ? config_.rtp.rtcp_mode
384 : RtcpMode::kOff); 381 : RtcpMode::kOff);
385 } 382 }
386 383
387 } // namespace internal 384 } // namespace internal
388 } // namespace webrtc 385 } // namespace webrtc
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