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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" | |
|
stefan-webrtc
2015/11/13 08:29:27
Why is this needed in video_receive_stream.cc?
mflodman
2015/11/13 08:35:36
It is needed for the line of code you commented on
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| 21 #include "webrtc/system_wrappers/include/clock.h" | 22 #include "webrtc/system_wrappers/include/clock.h" |
| 22 #include "webrtc/video/receive_statistics_proxy.h" | 23 #include "webrtc/video/receive_statistics_proxy.h" |
| 23 #include "webrtc/video_engine/call_stats.h" | 24 #include "webrtc/video_engine/call_stats.h" |
| 24 #include "webrtc/video_receive_stream.h" | 25 #include "webrtc/video_receive_stream.h" |
| 25 | 26 |
| 26 namespace webrtc { | 27 namespace webrtc { |
| 27 | 28 |
| 28 static bool UseSendSideBwe(const std::vector<RtpExtension>& extensions) { | 29 static bool UseSendSideBwe(const std::vector<RtpExtension>& extensions) { |
| 29 for (const auto& extension : extensions) { | 30 for (const auto& extension : extensions) { |
| 30 if (extension.name == RtpExtension::kTransportSequenceNumber) | 31 if (extension.name == RtpExtension::kTransportSequenceNumber) |
| (...skipping 123 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 154 LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString(); | 155 LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString(); |
| 155 | 156 |
| 156 bool send_side_bwe = UseSendSideBwe(config_.rtp.extensions); | 157 bool send_side_bwe = UseSendSideBwe(config_.rtp.extensions); |
| 157 | 158 |
| 158 RemoteBitrateEstimator* bitrate_estimator = | 159 RemoteBitrateEstimator* bitrate_estimator = |
| 159 congestion_controller_->GetRemoteBitrateEstimator(send_side_bwe); | 160 congestion_controller_->GetRemoteBitrateEstimator(send_side_bwe); |
| 160 | 161 |
| 161 vie_channel_.reset(new ViEChannel( | 162 vie_channel_.reset(new ViEChannel( |
| 162 num_cpu_cores, &transport_adapter_, process_thread, nullptr, | 163 num_cpu_cores, &transport_adapter_, process_thread, nullptr, |
| 163 congestion_controller_->GetBitrateController()-> | 164 congestion_controller_->GetBitrateController()-> |
| 164 CreateRtcpBandwidthObserver(), | 165 CreateRtcpBandwidthObserver(), |
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stefan-webrtc
2015/11/13 08:29:27
Is it possible to follow up with removing or setti
mflodman
2015/11/13 08:35:36
Absolutely, I'll take a look at this and I totally
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| 165 nullptr, bitrate_estimator, call_stats_->rtcp_rtt_stats(), | 166 nullptr, bitrate_estimator, call_stats_->rtcp_rtt_stats(), |
| 166 congestion_controller_->pacer(), congestion_controller_->packet_router(), | 167 congestion_controller_->pacer(), congestion_controller_->packet_router(), |
| 167 1, false)); | 168 1, false)); |
| 168 | 169 |
| 169 RTC_CHECK(vie_channel_->Init() == 0); | 170 RTC_CHECK(vie_channel_->Init() == 0); |
| 170 | 171 |
| 171 // Register the channel to receive stats updates. | 172 // Register the channel to receive stats updates. |
| 172 call_stats_->RegisterStatsObserver(vie_channel_->GetStatsObserver()); | 173 call_stats_->RegisterStatsObserver(vie_channel_->GetStatsObserver()); |
| 173 | 174 |
| 174 // TODO(pbos): This is not fine grained enough... | 175 // TODO(pbos): This is not fine grained enough... |
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| 378 return 0; | 379 return 0; |
| 379 } | 380 } |
| 380 | 381 |
| 381 void VideoReceiveStream::SignalNetworkState(NetworkState state) { | 382 void VideoReceiveStream::SignalNetworkState(NetworkState state) { |
| 382 vie_channel_->SetRTCPMode(state == kNetworkUp ? config_.rtp.rtcp_mode | 383 vie_channel_->SetRTCPMode(state == kNetworkUp ? config_.rtp.rtcp_mode |
| 383 : RtcpMode::kOff); | 384 : RtcpMode::kOff); |
| 384 } | 385 } |
| 385 | 386 |
| 386 } // namespace internal | 387 } // namespace internal |
| 387 } // namespace webrtc | 388 } // namespace webrtc |
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