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Issue 1728503002: Replace scoped_ptr with unique_ptr in webrtc/media/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@up1
Patch Set: 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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70 void FakeAudioReceiveStream::IncrementReceivedPackets() { 70 void FakeAudioReceiveStream::IncrementReceivedPackets() {
71 received_packets_++; 71 received_packets_++;
72 } 72 }
73 73
74 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const { 74 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
75 return stats_; 75 return stats_;
76 } 76 }
77 77
78 void FakeAudioReceiveStream::SetSink( 78 void FakeAudioReceiveStream::SetSink(
79 std::unique_ptr<webrtc::AudioSinkInterface> sink) { 79 std::unique_ptr<webrtc::AudioSinkInterface> sink) {
80 sink_ = rtc::UniqueToScoped(std::move(sink)); 80 sink_ = std::move(sink);
81 } 81 }
82 82
83 FakeVideoSendStream::FakeVideoSendStream( 83 FakeVideoSendStream::FakeVideoSendStream(
84 const webrtc::VideoSendStream::Config& config, 84 const webrtc::VideoSendStream::Config& config,
85 const webrtc::VideoEncoderConfig& encoder_config) 85 const webrtc::VideoEncoderConfig& encoder_config)
86 : sending_(false), 86 : sending_(false),
87 config_(config), 87 config_(config),
88 codec_settings_set_(false), 88 codec_settings_set_(false),
89 num_swapped_frames_(0) { 89 num_swapped_frames_(0) {
90 RTC_DCHECK(config.encoder_settings.encoder != NULL); 90 RTC_DCHECK(config.encoder_settings.encoder != NULL);
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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 } // namespace cricket 426 } // namespace cricket
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