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

Issue 1930463002: Replace scoped_ptr with unique_ptr in webrtc/api/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 7 months ago
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1 /* 1 /*
2 * Copyright 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2014 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
11 #include "webrtc/api/remoteaudiosource.h" 11 #include "webrtc/api/remoteaudiosource.h"
12 12
13 #include <algorithm> 13 #include <algorithm>
14 #include <functional> 14 #include <functional>
15 #include <memory>
15 #include <utility> 16 #include <utility>
16 17
17 #include "webrtc/api/mediastreamprovider.h" 18 #include "webrtc/api/mediastreamprovider.h"
18 #include "webrtc/base/checks.h" 19 #include "webrtc/base/checks.h"
19 #include "webrtc/base/constructormagic.h" 20 #include "webrtc/base/constructormagic.h"
20 #include "webrtc/base/logging.h" 21 #include "webrtc/base/logging.h"
21 #include "webrtc/base/thread.h" 22 #include "webrtc/base/thread.h"
22 23
23 namespace webrtc { 24 namespace webrtc {
24 25
(...skipping 49 matching lines...) Expand 10 before | Expand all | Expand 10 after
74 RTC_DCHECK(sinks_.empty()); 75 RTC_DCHECK(sinks_.empty());
75 } 76 }
76 77
77 void RemoteAudioSource::Initialize(uint32_t ssrc, 78 void RemoteAudioSource::Initialize(uint32_t ssrc,
78 AudioProviderInterface* provider) { 79 AudioProviderInterface* provider) {
79 RTC_DCHECK(main_thread_->IsCurrent()); 80 RTC_DCHECK(main_thread_->IsCurrent());
80 // To make sure we always get notified when the provider goes out of scope, 81 // To make sure we always get notified when the provider goes out of scope,
81 // we register for callbacks here and not on demand in AddSink. 82 // we register for callbacks here and not on demand in AddSink.
82 if (provider) { // May be null in tests. 83 if (provider) { // May be null in tests.
83 provider->SetRawAudioSink( 84 provider->SetRawAudioSink(
84 ssrc, rtc::scoped_ptr<AudioSinkInterface>(new Sink(this))); 85 ssrc, std::unique_ptr<AudioSinkInterface>(new Sink(this)));
85 } 86 }
86 } 87 }
87 88
88 MediaSourceInterface::SourceState RemoteAudioSource::state() const { 89 MediaSourceInterface::SourceState RemoteAudioSource::state() const {
89 RTC_DCHECK(main_thread_->IsCurrent()); 90 RTC_DCHECK(main_thread_->IsCurrent());
90 return state_; 91 return state_;
91 } 92 }
92 93
93 bool RemoteAudioSource::remote() const { 94 bool RemoteAudioSource::remote() const {
94 RTC_DCHECK(main_thread_->IsCurrent()); 95 RTC_DCHECK(main_thread_->IsCurrent());
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151 } 152 }
152 153
153 void RemoteAudioSource::OnMessage(rtc::Message* msg) { 154 void RemoteAudioSource::OnMessage(rtc::Message* msg) {
154 RTC_DCHECK(main_thread_->IsCurrent()); 155 RTC_DCHECK(main_thread_->IsCurrent());
155 sinks_.clear(); 156 sinks_.clear();
156 state_ = MediaSourceInterface::kEnded; 157 state_ = MediaSourceInterface::kEnded;
157 FireOnChanged(); 158 FireOnChanged();
158 } 159 }
159 160
160 } // namespace webrtc 161 } // namespace webrtc
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