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Side by Side Diff: remoting/protocol/webrtc_audio_source_adapter.cc

Issue 2425483002: Remove usage of FOR_EACH_OBSERVER macro in remoting (Closed)
Patch Set: braces Created 4 years, 2 months ago
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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 "base/bind.h" 7 #include "base/bind.h"
8 #include "base/logging.h" 8 #include "base/logging.h"
9 #include "base/synchronization/lock.h" 9 #include "base/synchronization/lock.h"
10 #include "base/threading/thread_checker.h" 10 #include "base/threading/thread_checker.h"
(...skipping 107 matching lines...) Expand 10 before | Expand all | Expand 10 after
118 partial_frame_.insert(partial_frame_.end(), data.data(), 118 partial_frame_.insert(partial_frame_.end(), data.data(),
119 data.data() + bytes_to_append); 119 data.data() + bytes_to_append);
120 if (partial_frame_.size() < bytes_per_frame) { 120 if (partial_frame_.size() < bytes_per_frame) {
121 // Still don't have full frame. 121 // Still don't have full frame.
122 return; 122 return;
123 } 123 }
124 124
125 // Here |partial_frame_| always contains a full frame. 125 // Here |partial_frame_| always contains a full frame.
126 DCHECK_EQ(partial_frame_.size(), bytes_per_frame); 126 DCHECK_EQ(partial_frame_.size(), bytes_per_frame);
127 127
128 FOR_EACH_OBSERVER(webrtc::AudioTrackSinkInterface, audio_sinks_, 128 for (auto& observer : audio_sinks_) {
129 OnData(&partial_frame_.front(), kBytesPerSample * 8, 129 observer.OnData(&partial_frame_.front(), kBytesPerSample * 8,
130 sampling_rate_, kChannels, samples_per_frame)); 130 sampling_rate_, kChannels, samples_per_frame);
131 }
131 } 132 }
132 133
133 while (position + bytes_per_frame <= data.size()) { 134 while (position + bytes_per_frame <= data.size()) {
134 FOR_EACH_OBSERVER(webrtc::AudioTrackSinkInterface, audio_sinks_, 135 for (auto& observer : audio_sinks_) {
135 OnData(data.data() + position, kBytesPerSample * 8, 136 observer.OnData(data.data() + position, kBytesPerSample * 8,
136 sampling_rate_, kChannels, samples_per_frame)); 137 sampling_rate_, kChannels, samples_per_frame);
138 }
137 position += bytes_per_frame; 139 position += bytes_per_frame;
138 } 140 }
139 141
140 partial_frame_.assign(data.data() + position, data.data() + data.size()); 142 partial_frame_.assign(data.data() + position, data.data() + data.size());
141 } 143 }
142 144
143 WebrtcAudioSourceAdapter::WebrtcAudioSourceAdapter( 145 WebrtcAudioSourceAdapter::WebrtcAudioSourceAdapter(
144 scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner) 146 scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner)
145 : audio_task_runner_(audio_task_runner), core_(new Core()) {} 147 : audio_task_runner_(audio_task_runner), core_(new Core()) {}
146 148
(...skipping 35 matching lines...) Expand 10 before | Expand all | Expand 10 after
182 core_->RemoveSink(sink); 184 core_->RemoveSink(sink);
183 } 185 }
184 186
185 void WebrtcAudioSourceAdapter::RegisterObserver( 187 void WebrtcAudioSourceAdapter::RegisterObserver(
186 webrtc::ObserverInterface* observer) {} 188 webrtc::ObserverInterface* observer) {}
187 void WebrtcAudioSourceAdapter::UnregisterObserver( 189 void WebrtcAudioSourceAdapter::UnregisterObserver(
188 webrtc::ObserverInterface* observer) {} 190 webrtc::ObserverInterface* observer) {}
189 191
190 } // namespace protocol 192 } // namespace protocol
191 } // namespace remoting 193 } // namespace remoting
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