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| 1 /* | 1 /* |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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/modules/audio_mixer/audio_mixer_impl.h" | 11 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
| 12 | 12 |
| 13 #include <algorithm> | 13 #include <algorithm> |
| 14 #include <functional> | 14 #include <functional> |
| 15 #include <iterator> | 15 #include <iterator> |
| 16 #include <utility> | 16 #include <utility> |
| 17 | 17 |
| 18 #include "webrtc/base/logging.h" | |
| 19 #include "webrtc/modules/audio_mixer/audio_frame_manipulator.h" | 18 #include "webrtc/modules/audio_mixer/audio_frame_manipulator.h" |
| 20 #include "webrtc/modules/audio_mixer/default_output_rate_calculator.h" | 19 #include "webrtc/modules/audio_mixer/default_output_rate_calculator.h" |
| 20 #include "webrtc/rtc_base/logging.h" |
| 21 | 21 |
| 22 namespace webrtc { | 22 namespace webrtc { |
| 23 namespace { | 23 namespace { |
| 24 | 24 |
| 25 struct SourceFrame { | 25 struct SourceFrame { |
| 26 SourceFrame(AudioMixerImpl::SourceStatus* source_status, | 26 SourceFrame(AudioMixerImpl::SourceStatus* source_status, |
| 27 AudioFrame* audio_frame, | 27 AudioFrame* audio_frame, |
| 28 bool muted) | 28 bool muted) |
| 29 : source_status(source_status), audio_frame(audio_frame), muted(muted) { | 29 : source_status(source_status), audio_frame(audio_frame), muted(muted) { |
| 30 RTC_DCHECK(source_status); | 30 RTC_DCHECK(source_status); |
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| 239 | 239 |
| 240 const auto iter = FindSourceInList(audio_source, &audio_source_list_); | 240 const auto iter = FindSourceInList(audio_source, &audio_source_list_); |
| 241 if (iter != audio_source_list_.end()) { | 241 if (iter != audio_source_list_.end()) { |
| 242 return (*iter)->is_mixed; | 242 return (*iter)->is_mixed; |
| 243 } | 243 } |
| 244 | 244 |
| 245 LOG(LS_ERROR) << "Audio source unknown"; | 245 LOG(LS_ERROR) << "Audio source unknown"; |
| 246 return false; | 246 return false; |
| 247 } | 247 } |
| 248 } // namespace webrtc | 248 } // namespace webrtc |
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