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Issue 1782053002: Relanding https://codereview.webrtc.org/1715883002/ in pieces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 9 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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32 void FakeAudioSendStream::SetStats( 32 void FakeAudioSendStream::SetStats(
33 const webrtc::AudioSendStream::Stats& stats) { 33 const webrtc::AudioSendStream::Stats& stats) {
34 stats_ = stats; 34 stats_ = stats;
35 } 35 }
36 36
37 FakeAudioSendStream::TelephoneEvent 37 FakeAudioSendStream::TelephoneEvent
38 FakeAudioSendStream::GetLatestTelephoneEvent() const { 38 FakeAudioSendStream::GetLatestTelephoneEvent() const {
39 return latest_telephone_event_; 39 return latest_telephone_event_;
40 } 40 }
41 41
42 bool FakeAudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event, 42 bool FakeAudioSendStream::SendTelephoneEvent(int payload_type, int event,
43 uint32_t duration_ms) { 43 int duration_ms) {
44 latest_telephone_event_.payload_type = payload_type; 44 latest_telephone_event_.payload_type = payload_type;
45 latest_telephone_event_.event_code = event; 45 latest_telephone_event_.event_code = event;
46 latest_telephone_event_.duration_ms = duration_ms; 46 latest_telephone_event_.duration_ms = duration_ms;
47 return true; 47 return true;
48 } 48 }
49 49
50 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const { 50 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const {
51 return stats_; 51 return stats_;
52 } 52 }
53 53
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416 } 416 }
417 417
418 void FakeCall::SignalNetworkState(webrtc::NetworkState state) { 418 void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
419 network_state_ = state; 419 network_state_ = state;
420 } 420 }
421 421
422 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { 422 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
423 last_sent_packet_ = sent_packet; 423 last_sent_packet_ = sent_packet;
424 } 424 }
425 } // namespace cricket 425 } // namespace cricket
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