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

Issue 1704983002: Simplify CongestionController. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Comments addressed. Created 4 years, 10 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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21 namespace webrtc { 21 namespace webrtc {
22 22
23 const int64_t kNoTimestamp = -1; 23 const int64_t kNoTimestamp = -1;
24 const int64_t kSendTimeHistoryWindowMs = 10000; 24 const int64_t kSendTimeHistoryWindowMs = 10000;
25 const int64_t kBaseTimestampScaleFactor = 25 const int64_t kBaseTimestampScaleFactor =
26 rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 8); 26 rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 8);
27 const int64_t kBaseTimestampRangeSizeUs = kBaseTimestampScaleFactor * (1 << 24); 27 const int64_t kBaseTimestampRangeSizeUs = kBaseTimestampScaleFactor * (1 << 24);
28 28
29 TransportFeedbackAdapter::TransportFeedbackAdapter( 29 TransportFeedbackAdapter::TransportFeedbackAdapter(
30 BitrateController* bitrate_controller, 30 BitrateController* bitrate_controller,
31 Clock* clock, 31 Clock* clock)
32 ProcessThread* process_thread)
33 : send_time_history_(clock, kSendTimeHistoryWindowMs), 32 : send_time_history_(clock, kSendTimeHistoryWindowMs),
34 bitrate_controller_(bitrate_controller), 33 bitrate_controller_(bitrate_controller),
35 process_thread_(process_thread),
36 clock_(clock), 34 clock_(clock),
37 current_offset_ms_(kNoTimestamp), 35 current_offset_ms_(kNoTimestamp),
38 last_timestamp_us_(kNoTimestamp) {} 36 last_timestamp_us_(kNoTimestamp) {}
39 37
40 TransportFeedbackAdapter::~TransportFeedbackAdapter() { 38 TransportFeedbackAdapter::~TransportFeedbackAdapter() {
41 if (bitrate_estimator_.get())
42 process_thread_->DeRegisterModule(bitrate_estimator_.get());
43 } 39 }
44 40
45 void TransportFeedbackAdapter::SetBitrateEstimator( 41 void TransportFeedbackAdapter::SetBitrateEstimator(
46 RemoteBitrateEstimator* rbe) { 42 RemoteBitrateEstimator* rbe) {
47 if (bitrate_estimator_.get() != rbe) { 43 if (bitrate_estimator_.get() != rbe) {
48 bitrate_estimator_.reset(rbe); 44 bitrate_estimator_.reset(rbe);
49 process_thread_->RegisterModule(rbe);
50 } 45 }
51 } 46 }
52 47
53 void TransportFeedbackAdapter::AddPacket(uint16_t sequence_number, 48 void TransportFeedbackAdapter::AddPacket(uint16_t sequence_number,
54 size_t length, 49 size_t length,
55 bool was_paced) { 50 bool was_paced) {
56 rtc::CritScope cs(&lock_); 51 rtc::CritScope cs(&lock_);
57 send_time_history_.AddAndRemoveOld(sequence_number, length, was_paced); 52 send_time_history_.AddAndRemoveOld(sequence_number, length, was_paced);
58 } 53 }
59 54
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127 bitrate_controller_->UpdateDelayBasedEstimate(bitrate); 122 bitrate_controller_->UpdateDelayBasedEstimate(bitrate);
128 } 123 }
129 124
130 void TransportFeedbackAdapter::OnRttUpdate(int64_t avg_rtt_ms, 125 void TransportFeedbackAdapter::OnRttUpdate(int64_t avg_rtt_ms,
131 int64_t max_rtt_ms) { 126 int64_t max_rtt_ms) {
132 RTC_DCHECK(bitrate_estimator_.get() != nullptr); 127 RTC_DCHECK(bitrate_estimator_.get() != nullptr);
133 bitrate_estimator_->OnRttUpdate(avg_rtt_ms, max_rtt_ms); 128 bitrate_estimator_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
134 } 129 }
135 130
136 } // namespace webrtc 131 } // namespace webrtc
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