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Issue 2234293002: Added new mixer to |check_targets| in .gn and fixed include/depend errors. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@mixer_dir_structure_comments
Patch Set: Depending on a system_wrappers implementation in audio_mixer broke chrome. Created 4 years, 4 months ago
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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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