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Issue 1783263002: Replace scoped_ptr with unique_ptr in webrtc/pc/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 9 months ago
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1 /* 1 /*
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2012 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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1203 ASSERT(false); 1203 ASSERT(false);
1204 } 1204 }
1205 } 1205 }
1206 1206
1207 void WebRtcSession::SetRawAudioSink(uint32_t ssrc, 1207 void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
1208 rtc::scoped_ptr<AudioSinkInterface> sink) { 1208 rtc::scoped_ptr<AudioSinkInterface> sink) {
1209 ASSERT(signaling_thread()->IsCurrent()); 1209 ASSERT(signaling_thread()->IsCurrent());
1210 if (!voice_channel_) 1210 if (!voice_channel_)
1211 return; 1211 return;
1212 1212
1213 voice_channel_->SetRawAudioSink(ssrc, std::move(sink)); 1213 voice_channel_->SetRawAudioSink(ssrc, rtc::ScopedToUnique(std::move(sink)));
1214 } 1214 }
1215 1215
1216 bool WebRtcSession::SetCaptureDevice(uint32_t ssrc, 1216 bool WebRtcSession::SetCaptureDevice(uint32_t ssrc,
1217 cricket::VideoCapturer* camera) { 1217 cricket::VideoCapturer* camera) {
1218 ASSERT(signaling_thread()->IsCurrent()); 1218 ASSERT(signaling_thread()->IsCurrent());
1219 1219
1220 if (!video_channel_) { 1220 if (!video_channel_) {
1221 // |video_channel_| doesnt't exist. Probably because the remote end doesnt't 1221 // |video_channel_| doesnt't exist. Probably because the remote end doesnt't
1222 // support video. 1222 // support video.
1223 LOG(LS_WARNING) << "Video not used in this call."; 1223 LOG(LS_WARNING) << "Video not used in this call.";
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2077 } 2077 }
2078 } 2078 }
2079 2079
2080 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel, 2080 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel,
2081 const rtc::SentPacket& sent_packet) { 2081 const rtc::SentPacket& sent_packet) {
2082 RTC_DCHECK(worker_thread()->IsCurrent()); 2082 RTC_DCHECK(worker_thread()->IsCurrent());
2083 media_controller_->call_w()->OnSentPacket(sent_packet); 2083 media_controller_->call_w()->OnSentPacket(sent_packet);
2084 } 2084 }
2085 2085
2086 } // namespace webrtc 2086 } // namespace webrtc
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