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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/new_audio_conference_mixer_impl.h" | 11 #include "webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.h" |
12 | 12 |
13 #include <algorithm> | 13 #include <algorithm> |
14 #include <functional> | 14 #include <functional> |
15 | 15 |
16 #include "webrtc/modules/audio_mixer/audio_frame_manipulator.h" | 16 #include "webrtc/modules/audio_mixer/audio_frame_manipulator.h" |
17 #include "webrtc/modules/audio_mixer/audio_mixer_defines.h" | 17 #include "webrtc/modules/audio_mixer/audio_mixer_defines.h" |
18 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 18 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
19 #include "webrtc/modules/utility/include/audio_frame_operations.h" | 19 #include "webrtc/modules/utility/include/audio_frame_operations.h" |
20 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" | 20 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
21 #include "webrtc/system_wrappers/include/trace.h" | 21 #include "webrtc/system_wrappers/include/trace.h" |
22 #include "webrtc/voice_engine/utility.h" | |
23 | 22 |
24 namespace webrtc { | 23 namespace webrtc { |
25 namespace { | 24 namespace { |
26 | 25 |
27 class SourceFrame { | 26 class SourceFrame { |
28 public: | 27 public: |
29 SourceFrame(MixerAudioSource* p, AudioFrame* a, bool m, bool was_mixed_before) | 28 SourceFrame(MixerAudioSource* p, AudioFrame* a, bool m, bool was_mixed_before) |
30 : audio_source_(p), | 29 : audio_source_(p), |
31 audio_frame_(a), | 30 audio_frame_(a), |
32 muted_(m), | 31 muted_(m), |
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135 output_frequency_(kDefaultFrequency), | 134 output_frequency_(kDefaultFrequency), |
136 sample_size_(0), | 135 sample_size_(0), |
137 audio_source_list_(), | 136 audio_source_list_(), |
138 additional_audio_source_list_(), | 137 additional_audio_source_list_(), |
139 num_mixed_audio_sources_(0), | 138 num_mixed_audio_sources_(0), |
140 use_limiter_(true), | 139 use_limiter_(true), |
141 time_stamp_(0) { | 140 time_stamp_(0) { |
142 thread_checker_.DetachFromThread(); | 141 thread_checker_.DetachFromThread(); |
143 } | 142 } |
144 | 143 |
| 144 NewAudioConferenceMixerImpl::~NewAudioConferenceMixerImpl() {} |
| 145 |
145 bool NewAudioConferenceMixerImpl::Init() { | 146 bool NewAudioConferenceMixerImpl::Init() { |
146 crit_.reset(CriticalSectionWrapper::CreateCriticalSection()); | 147 crit_.reset(CriticalSectionWrapper::CreateCriticalSection()); |
147 if (crit_.get() == NULL) | 148 if (crit_.get() == NULL) |
148 return false; | 149 return false; |
149 | 150 |
150 cb_crit_.reset(CriticalSectionWrapper::CreateCriticalSection()); | 151 cb_crit_.reset(CriticalSectionWrapper::CreateCriticalSection()); |
151 if (cb_crit_.get() == NULL) | 152 if (cb_crit_.get() == NULL) |
152 return false; | 153 return false; |
153 | 154 |
154 Config config; | 155 Config config; |
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584 | 585 |
585 if (error != limiter_->kNoError) { | 586 if (error != limiter_->kNoError) { |
586 WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_, | 587 WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_, |
587 "Error from AudioProcessing: %d", error); | 588 "Error from AudioProcessing: %d", error); |
588 RTC_NOTREACHED(); | 589 RTC_NOTREACHED(); |
589 return false; | 590 return false; |
590 } | 591 } |
591 return true; | 592 return true; |
592 } | 593 } |
593 } // namespace webrtc | 594 } // namespace webrtc |
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