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Side by Side Diff: webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc

Issue 1363573002: Wire up transport sequence number / send time callbacks to webrtc via libjingle. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Cleanup Created 5 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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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42 } 42 }
43 43
44 void TransportFeedbackAdapter::SetBitrateEstimator( 44 void TransportFeedbackAdapter::SetBitrateEstimator(
45 RemoteBitrateEstimator* rbe) { 45 RemoteBitrateEstimator* rbe) {
46 if (bitrate_estimator_.get() != rbe) { 46 if (bitrate_estimator_.get() != rbe) {
47 bitrate_estimator_.reset(rbe); 47 bitrate_estimator_.reset(rbe);
48 process_thread_->RegisterModule(rbe); 48 process_thread_->RegisterModule(rbe);
49 } 49 }
50 } 50 }
51 51
52 void TransportFeedbackAdapter::OnPacketSent(const PacketInfo& info) { 52 void TransportFeedbackAdapter::OnSentPacket(const PacketInfo& info) {
53 rtc::CritScope cs(&lock_); 53 rtc::CritScope cs(&lock_);
54 send_time_history_.AddAndRemoveOld(info); 54 send_time_history_.AddAndRemoveOld(info);
55 } 55 }
56 56
57 void TransportFeedbackAdapter::UpdateSendTime(uint16_t sequence_number,
58 int64_t send_time_ms) {
59 rtc::CritScope cs(&lock_);
60 send_time_history_.UpdateSendTime(sequence_number, send_time_ms);
61 }
62
57 void TransportFeedbackAdapter::OnTransportFeedback( 63 void TransportFeedbackAdapter::OnTransportFeedback(
58 const rtcp::TransportFeedback& feedback) { 64 const rtcp::TransportFeedback& feedback) {
59 int64_t timestamp_us = feedback.GetBaseTimeUs(); 65 int64_t timestamp_us = feedback.GetBaseTimeUs();
60 // Add timestamp deltas to a local time base selected on first packet arrival. 66 // Add timestamp deltas to a local time base selected on first packet arrival.
61 // This won't be the true time base, but makes it easier to manually inspect 67 // This won't be the true time base, but makes it easier to manually inspect
62 // time stamps. 68 // time stamps.
63 if (last_timestamp_us_ == kNoTimestamp) { 69 if (last_timestamp_us_ == kNoTimestamp) {
64 current_offset_ms_ = clock_->TimeInMilliseconds(); 70 current_offset_ms_ = clock_->TimeInMilliseconds();
65 } else { 71 } else {
66 int64_t delta = timestamp_us - last_timestamp_us_; 72 int64_t delta = timestamp_us - last_timestamp_us_;
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118 rtcp_bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate); 124 rtcp_bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate);
119 } 125 }
120 126
121 void TransportFeedbackAdapter::OnRttUpdate(int64_t avg_rtt_ms, 127 void TransportFeedbackAdapter::OnRttUpdate(int64_t avg_rtt_ms,
122 int64_t max_rtt_ms) { 128 int64_t max_rtt_ms) {
123 RTC_DCHECK(bitrate_estimator_.get() != nullptr); 129 RTC_DCHECK(bitrate_estimator_.get() != nullptr);
124 bitrate_estimator_->OnRttUpdate(avg_rtt_ms, max_rtt_ms); 130 bitrate_estimator_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
125 } 131 }
126 132
127 } // namespace webrtc 133 } // namespace webrtc
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