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

Issue 1409133006: Make SendStatisticsProxy outlive ViEChannel. (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
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110 110
111 namespace internal { 111 namespace internal {
112 VideoSendStream::VideoSendStream( 112 VideoSendStream::VideoSendStream(
113 int num_cpu_cores, 113 int num_cpu_cores,
114 ProcessThread* module_process_thread, 114 ProcessThread* module_process_thread,
115 CallStats* call_stats, 115 CallStats* call_stats,
116 CongestionController* congestion_controller, 116 CongestionController* congestion_controller,
117 const VideoSendStream::Config& config, 117 const VideoSendStream::Config& config,
118 const VideoEncoderConfig& encoder_config, 118 const VideoEncoderConfig& encoder_config,
119 const std::map<uint32_t, RtpState>& suspended_ssrcs) 119 const std::map<uint32_t, RtpState>& suspended_ssrcs)
120 : transport_adapter_(config.send_transport), 120 : stats_proxy_(Clock::GetRealTimeClock(), config),
121 transport_adapter_(config.send_transport),
121 encoded_frame_proxy_(config.post_encode_callback), 122 encoded_frame_proxy_(config.post_encode_callback),
122 config_(config), 123 config_(config),
123 suspended_ssrcs_(suspended_ssrcs), 124 suspended_ssrcs_(suspended_ssrcs),
124 module_process_thread_(module_process_thread), 125 module_process_thread_(module_process_thread),
125 call_stats_(call_stats), 126 call_stats_(call_stats),
126 congestion_controller_(congestion_controller), 127 congestion_controller_(congestion_controller),
127 encoder_feedback_(new EncoderStateFeedback()), 128 encoder_feedback_(new EncoderStateFeedback()),
128 use_config_bitrate_(true), 129 use_config_bitrate_(true) {
129 stats_proxy_(Clock::GetRealTimeClock(), config) {
130 LOG(LS_INFO) << "VideoSendStream: " << config_.ToString(); 130 LOG(LS_INFO) << "VideoSendStream: " << config_.ToString();
131 RTC_DCHECK(!config_.rtp.ssrcs.empty()); 131 RTC_DCHECK(!config_.rtp.ssrcs.empty());
132 132
133 // Set up Call-wide sequence numbers, if configured for this send stream. 133 // Set up Call-wide sequence numbers, if configured for this send stream.
134 TransportFeedbackObserver* transport_feedback_observer = nullptr; 134 TransportFeedbackObserver* transport_feedback_observer = nullptr;
135 for (const RtpExtension& extension : config.rtp.extensions) { 135 for (const RtpExtension& extension : config.rtp.extensions) {
136 if (extension.name == RtpExtension::kTransportSequenceNumber) { 136 if (extension.name == RtpExtension::kTransportSequenceNumber) {
137 transport_feedback_observer = 137 transport_feedback_observer =
138 congestion_controller_->GetTransportFeedbackObserver(); 138 congestion_controller_->GetTransportFeedbackObserver();
139 break; 139 break;
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566 vie_channel_->IsSendingFecEnabled()); 566 vie_channel_->IsSendingFecEnabled());
567 567
568 // Restart the media flow 568 // Restart the media flow
569 vie_encoder_->Restart(); 569 vie_encoder_->Restart();
570 570
571 return true; 571 return true;
572 } 572 }
573 573
574 } // namespace internal 574 } // namespace internal
575 } // namespace webrtc 575 } // namespace webrtc
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