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Side by Side Diff: webrtc/api/webrtcsession.cc

Issue 1914153002: Remove calls to ScopedToUnique and UniqueToScoped (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@base-fix
Patch Set: Created 4 years, 8 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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1234 ASSERT(false); 1234 ASSERT(false);
1235 } 1235 }
1236 } 1236 }
1237 1237
1238 void WebRtcSession::SetRawAudioSink(uint32_t ssrc, 1238 void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
1239 rtc::scoped_ptr<AudioSinkInterface> sink) { 1239 rtc::scoped_ptr<AudioSinkInterface> sink) {
1240 ASSERT(signaling_thread()->IsCurrent()); 1240 ASSERT(signaling_thread()->IsCurrent());
1241 if (!voice_channel_) 1241 if (!voice_channel_)
1242 return; 1242 return;
1243 1243
1244 voice_channel_->SetRawAudioSink(ssrc, rtc::ScopedToUnique(std::move(sink))); 1244 voice_channel_->SetRawAudioSink(ssrc, std::move(sink));
1245 } 1245 }
1246 1246
1247 RtpParameters WebRtcSession::GetAudioRtpParameters(uint32_t ssrc) const { 1247 RtpParameters WebRtcSession::GetAudioRtpParameters(uint32_t ssrc) const {
1248 ASSERT(signaling_thread()->IsCurrent()); 1248 ASSERT(signaling_thread()->IsCurrent());
1249 if (voice_channel_) { 1249 if (voice_channel_) {
1250 return voice_channel_->GetRtpParameters(ssrc); 1250 return voice_channel_->GetRtpParameters(ssrc);
1251 } 1251 }
1252 return RtpParameters(); 1252 return RtpParameters();
1253 } 1253 }
1254 1254
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2155 } 2155 }
2156 } 2156 }
2157 2157
2158 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel, 2158 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel,
2159 const rtc::SentPacket& sent_packet) { 2159 const rtc::SentPacket& sent_packet) {
2160 RTC_DCHECK(worker_thread()->IsCurrent()); 2160 RTC_DCHECK(worker_thread()->IsCurrent());
2161 media_controller_->call_w()->OnSentPacket(sent_packet); 2161 media_controller_->call_w()->OnSentPacket(sent_packet);
2162 } 2162 }
2163 2163
2164 } // namespace webrtc 2164 } // namespace webrtc
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