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| 1 // Copyright 2016 The Chromium Authors. All rights reserved. | 1 // Copyright 2016 The Chromium Authors. All rights reserved. |
| 2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
| 3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
| 4 | 4 |
| 5 #include "remoting/protocol/webrtc_audio_source_adapter.h" | 5 #include "remoting/protocol/webrtc_audio_source_adapter.h" |
| 6 | 6 |
| 7 #include <utility> |
| 8 |
| 7 #include "base/bind.h" | 9 #include "base/bind.h" |
| 8 #include "base/logging.h" | 10 #include "base/logging.h" |
| 9 #include "base/synchronization/lock.h" | 11 #include "base/synchronization/lock.h" |
| 10 #include "base/threading/thread_checker.h" | 12 #include "base/threading/thread_checker.h" |
| 11 #include "remoting/proto/audio.pb.h" | 13 #include "remoting/proto/audio.pb.h" |
| 12 #include "remoting/protocol/audio_source.h" | 14 #include "remoting/protocol/audio_source.h" |
| 13 | 15 |
| 14 namespace remoting { | 16 namespace remoting { |
| 15 namespace protocol { | 17 namespace protocol { |
| 16 | 18 |
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| 137 sampling_rate_, kChannels, samples_per_frame); | 139 sampling_rate_, kChannels, samples_per_frame); |
| 138 } | 140 } |
| 139 position += bytes_per_frame; | 141 position += bytes_per_frame; |
| 140 } | 142 } |
| 141 | 143 |
| 142 partial_frame_.assign(data.data() + position, data.data() + data.size()); | 144 partial_frame_.assign(data.data() + position, data.data() + data.size()); |
| 143 } | 145 } |
| 144 | 146 |
| 145 WebrtcAudioSourceAdapter::WebrtcAudioSourceAdapter( | 147 WebrtcAudioSourceAdapter::WebrtcAudioSourceAdapter( |
| 146 scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner) | 148 scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner) |
| 147 : audio_task_runner_(audio_task_runner), core_(new Core()) {} | 149 : audio_task_runner_(std::move(audio_task_runner)), core_(new Core()) {} |
| 148 | 150 |
| 149 WebrtcAudioSourceAdapter::~WebrtcAudioSourceAdapter() { | 151 WebrtcAudioSourceAdapter::~WebrtcAudioSourceAdapter() { |
| 150 audio_task_runner_->DeleteSoon(FROM_HERE, core_.release()); | 152 audio_task_runner_->DeleteSoon(FROM_HERE, core_.release()); |
| 151 } | 153 } |
| 152 | 154 |
| 153 void WebrtcAudioSourceAdapter::Start( | 155 void WebrtcAudioSourceAdapter::Start( |
| 154 std::unique_ptr<AudioSource> audio_source) { | 156 std::unique_ptr<AudioSource> audio_source) { |
| 155 audio_task_runner_->PostTask( | 157 audio_task_runner_->PostTask( |
| 156 FROM_HERE, base::Bind(&Core::Start, base::Unretained(core_.get()), | 158 FROM_HERE, base::Bind(&Core::Start, base::Unretained(core_.get()), |
| 157 base::Passed(&audio_source))); | 159 base::Passed(&audio_source))); |
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| 184 core_->RemoveSink(sink); | 186 core_->RemoveSink(sink); |
| 185 } | 187 } |
| 186 | 188 |
| 187 void WebrtcAudioSourceAdapter::RegisterObserver( | 189 void WebrtcAudioSourceAdapter::RegisterObserver( |
| 188 webrtc::ObserverInterface* observer) {} | 190 webrtc::ObserverInterface* observer) {} |
| 189 void WebrtcAudioSourceAdapter::UnregisterObserver( | 191 void WebrtcAudioSourceAdapter::UnregisterObserver( |
| 190 webrtc::ObserverInterface* observer) {} | 192 webrtc::ObserverInterface* observer) {} |
| 191 | 193 |
| 192 } // namespace protocol | 194 } // namespace protocol |
| 193 } // namespace remoting | 195 } // namespace remoting |
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