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Side by Side Diff: webrtc/media/engine/fakewebrtccall.cc

Issue 1748403002: Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Undid unneccessary changes to rtp_rtcp module. 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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416 config_.bitrate_config = bitrate_config; 416 config_.bitrate_config = bitrate_config;
417 } 417 }
418 418
419 void FakeCall::SignalNetworkState(webrtc::NetworkState state) { 419 void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
420 network_state_ = state; 420 network_state_ = state;
421 } 421 }
422 422
423 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) { 423 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
424 last_sent_packet_ = sent_packet; 424 last_sent_packet_ = sent_packet;
425 } 425 }
426
427 webrtc::RtcEventLog* FakeCall::RtcEventLog() {
428 return nullptr;
429 }
430
426 } // namespace cricket 431 } // namespace cricket
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