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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" |
22 | 23 |
23 namespace webrtc { | 24 namespace webrtc { |
24 namespace { | 25 namespace { |
25 | 26 |
26 class SourceFrame { | 27 class SourceFrame { |
27 public: | 28 public: |
28 SourceFrame(MixerAudioSource* p, AudioFrame* a, bool m, bool was_mixed_before) | 29 SourceFrame(MixerAudioSource* p, AudioFrame* a, bool m, bool was_mixed_before) |
29 : audio_source_(p), | 30 : audio_source_(p), |
30 audio_frame_(a), | 31 audio_frame_(a), |
31 muted_(m), | 32 muted_(m), |
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134 output_frequency_(kDefaultFrequency), | 135 output_frequency_(kDefaultFrequency), |
135 sample_size_(0), | 136 sample_size_(0), |
136 audio_source_list_(), | 137 audio_source_list_(), |
137 additional_audio_source_list_(), | 138 additional_audio_source_list_(), |
138 num_mixed_audio_sources_(0), | 139 num_mixed_audio_sources_(0), |
139 use_limiter_(true), | 140 use_limiter_(true), |
140 time_stamp_(0) { | 141 time_stamp_(0) { |
141 thread_checker_.DetachFromThread(); | 142 thread_checker_.DetachFromThread(); |
142 } | 143 } |
143 | 144 |
144 NewAudioConferenceMixerImpl::~NewAudioConferenceMixerImpl() {} | |
145 | |
146 bool NewAudioConferenceMixerImpl::Init() { | 145 bool NewAudioConferenceMixerImpl::Init() { |
147 crit_.reset(CriticalSectionWrapper::CreateCriticalSection()); | 146 crit_.reset(CriticalSectionWrapper::CreateCriticalSection()); |
148 if (crit_.get() == NULL) | 147 if (crit_.get() == NULL) |
149 return false; | 148 return false; |
150 | 149 |
151 cb_crit_.reset(CriticalSectionWrapper::CreateCriticalSection()); | 150 cb_crit_.reset(CriticalSectionWrapper::CreateCriticalSection()); |
152 if (cb_crit_.get() == NULL) | 151 if (cb_crit_.get() == NULL) |
153 return false; | 152 return false; |
154 | 153 |
155 Config config; | 154 Config config; |
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585 | 584 |
586 if (error != limiter_->kNoError) { | 585 if (error != limiter_->kNoError) { |
587 WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_, | 586 WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_, |
588 "Error from AudioProcessing: %d", error); | 587 "Error from AudioProcessing: %d", error); |
589 RTC_NOTREACHED(); | 588 RTC_NOTREACHED(); |
590 return false; | 589 return false; |
591 } | 590 } |
592 return true; | 591 return true; |
593 } | 592 } |
594 } // namespace webrtc | 593 } // namespace webrtc |
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