Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(225)

Side by Side Diff: webrtc/modules/audio_coding/test/PacketLossTest.cc

Issue 2411613002: Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. (Closed)
Patch Set: Created 4 years, 2 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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
(...skipping 91 matching lines...) Expand 10 before | Expand all | Expand 10 after
102 return false; 102 return false;
103 } 103 }
104 104
105 bool SenderWithFEC::SetPacketLossRate(int expected_loss_rate) { 105 bool SenderWithFEC::SetPacketLossRate(int expected_loss_rate) {
106 bool success = false; 106 bool success = false;
107 _acm->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { 107 _acm->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
108 if (!*encoder) { 108 if (!*encoder) {
109 // There is no existing encoder. 109 // There is no existing encoder.
110 return; 110 return;
111 } 111 }
112 (*encoder)->SetProjectedPacketLossRate(expected_loss_rate / 100.0f); 112 (*encoder)->OnReceivedUplinkPacketLossFraction(expected_loss_rate / 100.0f);
113 success = true; 113 success = true;
114 }); 114 });
115 return success; 115 return success;
116 } 116 }
117 117
118 PacketLossTest::PacketLossTest(int channels, int expected_loss_rate, 118 PacketLossTest::PacketLossTest(int channels, int expected_loss_rate,
119 int actual_loss_rate, int burst_length) 119 int actual_loss_rate, int burst_length)
120 : channels_(channels), 120 : channels_(channels),
121 in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz" : 121 in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz" :
122 "audio_coding/teststereo32kHz"), 122 "audio_coding/teststereo32kHz"),
(...skipping 41 matching lines...) Expand 10 before | Expand all | Expand 10 after
164 164
165 receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_, 165 receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_,
166 actual_loss_rate_, burst_length_); 166 actual_loss_rate_, burst_length_);
167 receiver_->Run(); 167 receiver_->Run();
168 receiver_->Teardown(); 168 receiver_->Teardown();
169 rtpFile.Close(); 169 rtpFile.Close();
170 #endif 170 #endif
171 } 171 }
172 172
173 } // namespace webrtc 173 } // namespace webrtc
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698