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

Issue 1706183002: Replace scoped_ptr with unique_ptr in webrtc/audio/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 months ago
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1 /* 1 /*
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2004 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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1331 return stream_->GetStats(); 1331 return stream_->GetStats();
1332 } 1332 }
1333 1333
1334 int channel() const { 1334 int channel() const {
1335 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1335 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1336 return config_.voe_channel_id; 1336 return config_.voe_channel_id;
1337 } 1337 }
1338 1338
1339 void SetRawAudioSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) { 1339 void SetRawAudioSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
1340 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1340 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1341 stream_->SetSink(std::move(sink)); 1341 stream_->SetSink(rtc::ScopedToUnique(std::move(sink)));
1342 } 1342 }
1343 1343
1344 private: 1344 private:
1345 void RecreateAudioReceiveStream( 1345 void RecreateAudioReceiveStream(
1346 bool use_transport_cc, 1346 bool use_transport_cc,
1347 const std::vector<webrtc::RtpExtension>& extensions) { 1347 const std::vector<webrtc::RtpExtension>& extensions) {
1348 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1348 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1349 if (stream_) { 1349 if (stream_) {
1350 call_->DestroyAudioReceiveStream(stream_); 1350 call_->DestroyAudioReceiveStream(stream_);
1351 stream_ = nullptr; 1351 stream_ = nullptr;
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2546 } 2546 }
2547 } else { 2547 } else {
2548 LOG(LS_INFO) << "Stopping playout for channel #" << channel; 2548 LOG(LS_INFO) << "Stopping playout for channel #" << channel;
2549 engine()->voe()->base()->StopPlayout(channel); 2549 engine()->voe()->base()->StopPlayout(channel);
2550 } 2550 }
2551 return true; 2551 return true;
2552 } 2552 }
2553 } // namespace cricket 2553 } // namespace cricket
2554 2554
2555 #endif // HAVE_WEBRTC_VOICE 2555 #endif // HAVE_WEBRTC_VOICE
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