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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 |
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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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