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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 #if defined(WEBRTC_ANDROID) | 11 #if defined(WEBRTC_ANDROID) |
| 12 #include "webrtc/modules/audio_device/android/audio_device_template.h" | 12 #include "webrtc/modules/audio_device/android/audio_device_template.h" |
| 13 #include "webrtc/modules/audio_device/android/audio_record_jni.h" | 13 #include "webrtc/modules/audio_device/android/audio_record_jni.h" |
| 14 #include "webrtc/modules/audio_device/android/audio_track_jni.h" | 14 #include "webrtc/modules/audio_device/android/audio_track_jni.h" |
| 15 #include "webrtc/modules/utility/include/jvm_android.h" | 15 #include "webrtc/modules/utility/include/jvm_android.h" |
| 16 #endif | 16 #endif |
| 17 | 17 |
| 18 #include "webrtc/base/checks.h" | |
| 18 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" | 19 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" |
| 20 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" | |
| 19 #include "webrtc/system_wrappers/include/trace.h" | 21 #include "webrtc/system_wrappers/include/trace.h" |
| 22 #include "webrtc/voice_engine/channel_proxy.h" | |
| 20 #include "webrtc/voice_engine/voice_engine_impl.h" | 23 #include "webrtc/voice_engine/voice_engine_impl.h" |
| 21 | 24 |
| 22 namespace webrtc { | 25 namespace webrtc { |
| 23 | 26 |
| 24 // Counter to be ensure that we can add a correct ID in all static trace | 27 // Counter to be ensure that we can add a correct ID in all static trace |
| 25 // methods. It is not the nicest solution, especially not since we already | 28 // methods. It is not the nicest solution, especially not since we already |
| 26 // have a counter in VoEBaseImpl. In other words, there is room for | 29 // have a counter in VoEBaseImpl. In other words, there is room for |
| 27 // improvement here. | 30 // improvement here. |
| 28 static int32_t gVoiceEngineInstanceCounter = 0; | 31 static int32_t gVoiceEngineInstanceCounter = 0; |
| 29 | 32 |
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| 70 // Example: AudioDeviceBuffer::RequestPlayoutData() can access a | 73 // Example: AudioDeviceBuffer::RequestPlayoutData() can access a |
| 71 // partially deconstructed |_ptrCbAudioTransport| during destruction | 74 // partially deconstructed |_ptrCbAudioTransport| during destruction |
| 72 // if we don't call Terminate here. | 75 // if we don't call Terminate here. |
| 73 Terminate(); | 76 Terminate(); |
| 74 delete this; | 77 delete this; |
| 75 } | 78 } |
| 76 | 79 |
| 77 return new_ref; | 80 return new_ref; |
| 78 } | 81 } |
| 79 | 82 |
| 83 voe::ChannelProxy* VoiceEngineImpl::GetChannelProxy(int channel_id) { | |
| 84 RTC_DCHECK(channel_id >= 0); | |
| 85 CriticalSectionScoped cs(crit_sec()); | |
| 86 RTC_DCHECK(statistics().Initialized()); | |
| 87 return new voe::ChannelProxy(channel_manager().GetChannel(channel_id)); | |
| 88 } | |
|
kwiberg-webrtc
2015/11/25 10:44:59
I think this should return a scoped_ptr instead of
the sun
2015/11/25 12:28:01
Yes, that works great. Thanks.
| |
| 89 | |
| 80 VoiceEngine* VoiceEngine::Create() { | 90 VoiceEngine* VoiceEngine::Create() { |
| 81 Config* config = new Config(); | 91 Config* config = new Config(); |
| 82 return GetVoiceEngine(config, true); | 92 return GetVoiceEngine(config, true); |
| 83 } | 93 } |
| 84 | 94 |
| 85 VoiceEngine* VoiceEngine::Create(const Config& config) { | 95 VoiceEngine* VoiceEngine::Create(const Config& config) { |
| 86 return GetVoiceEngine(&config, false); | 96 return GetVoiceEngine(&config, false); |
| 87 } | 97 } |
| 88 | 98 |
| 89 int VoiceEngine::SetTraceFilter(unsigned int filter) { | 99 int VoiceEngine::SetTraceFilter(unsigned int filter) { |
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| 147 webrtc::JVM::Initialize(reinterpret_cast<JavaVM*>(javaVM), | 157 webrtc::JVM::Initialize(reinterpret_cast<JavaVM*>(javaVM), |
| 148 reinterpret_cast<jobject>(context)); | 158 reinterpret_cast<jobject>(context)); |
| 149 return 0; | 159 return 0; |
| 150 #else | 160 #else |
| 151 return -1; | 161 return -1; |
| 152 #endif | 162 #endif |
| 153 } | 163 } |
| 154 #endif | 164 #endif |
| 155 | 165 |
| 156 } // namespace webrtc | 166 } // namespace webrtc |
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