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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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/audio/audio_state.h" | 11 #include "webrtc/audio/audio_state.h" |
12 | 12 |
13 #include "webrtc/base/atomicops.h" | 13 #include "webrtc/base/atomicops.h" |
14 #include "webrtc/base/checks.h" | 14 #include "webrtc/base/checks.h" |
15 #include "webrtc/base/logging.h" | 15 #include "webrtc/base/logging.h" |
16 #include "webrtc/modules/audio_device/include/audio_device.h" | 16 #include "webrtc/modules/audio_device/include/audio_device.h" |
17 #include "webrtc/voice_engine/include/voe_errors.h" | 17 #include "webrtc/voice_engine/include/voe_errors.h" |
18 | 18 |
19 namespace webrtc { | 19 namespace webrtc { |
20 namespace internal { | 20 namespace internal { |
21 | 21 |
| 22 // TODO(peah): Remove the conditional in the audio_transport_proxy_ constructor |
| 23 // call when upstream dependencies have properly been resolved. |
22 AudioState::AudioState(const AudioState::Config& config) | 24 AudioState::AudioState(const AudioState::Config& config) |
23 : config_(config), | 25 : config_(config), |
24 voe_base_(config.voice_engine), | 26 voe_base_(config.voice_engine), |
25 audio_transport_proxy_(voe_base_->audio_transport(), | 27 audio_transport_proxy_(voe_base_->audio_transport(), |
26 voe_base_->audio_processing(), | 28 config_.audio_processing |
| 29 ? config_.audio_processing.get() |
| 30 : voe_base_->audio_processing(), |
27 config_.audio_mixer) { | 31 config_.audio_mixer) { |
28 process_thread_checker_.DetachFromThread(); | 32 process_thread_checker_.DetachFromThread(); |
29 RTC_DCHECK(config_.audio_mixer); | 33 RTC_DCHECK(config_.audio_mixer); |
30 | 34 |
31 // Only one AudioState should be created per VoiceEngine. | 35 // Only one AudioState should be created per VoiceEngine. |
32 RTC_CHECK(voe_base_->RegisterVoiceEngineObserver(*this) != -1); | 36 RTC_CHECK(voe_base_->RegisterVoiceEngineObserver(*this) != -1); |
33 | 37 |
34 auto* const device = voe_base_->audio_device_module(); | 38 auto* const device = voe_base_->audio_device_module(); |
35 RTC_DCHECK(device); | 39 RTC_DCHECK(device); |
36 | 40 |
(...skipping 52 matching lines...) Loading... |
89 typing_noise_detected_ = false; | 93 typing_noise_detected_ = false; |
90 } | 94 } |
91 } | 95 } |
92 } // namespace internal | 96 } // namespace internal |
93 | 97 |
94 rtc::scoped_refptr<AudioState> AudioState::Create( | 98 rtc::scoped_refptr<AudioState> AudioState::Create( |
95 const AudioState::Config& config) { | 99 const AudioState::Config& config) { |
96 return rtc::scoped_refptr<AudioState>(new internal::AudioState(config)); | 100 return rtc::scoped_refptr<AudioState>(new internal::AudioState(config)); |
97 } | 101 } |
98 } // namespace webrtc | 102 } // namespace webrtc |
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