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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/base/checks.h" |
| 19 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 19 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
| 20 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" | |
| 21 #include "webrtc/system_wrappers/include/trace.h" | 20 #include "webrtc/system_wrappers/include/trace.h" |
| 22 #include "webrtc/voice_engine/channel_proxy.h" | 21 #include "webrtc/voice_engine/channel_proxy.h" |
| 23 #include "webrtc/voice_engine/voice_engine_impl.h" | 22 #include "webrtc/voice_engine/voice_engine_impl.h" |
| 24 | 23 |
| 25 namespace webrtc { | 24 namespace webrtc { |
| 26 | 25 |
| 27 // Counter to be ensure that we can add a correct ID in all static trace | 26 // Counter to be ensure that we can add a correct ID in all static trace |
| 28 // methods. It is not the nicest solution, especially not since we already | 27 // methods. It is not the nicest solution, especially not since we already |
| 29 // have a counter in VoEBaseImpl. In other words, there is room for | 28 // have a counter in VoEBaseImpl. In other words, there is room for |
| 30 // improvement here. | 29 // improvement here. |
| (...skipping 28 matching lines...) Expand all Loading... |
| 59 Terminate(); | 58 Terminate(); |
| 60 delete this; | 59 delete this; |
| 61 } | 60 } |
| 62 | 61 |
| 63 return new_ref; | 62 return new_ref; |
| 64 } | 63 } |
| 65 | 64 |
| 66 rtc::scoped_ptr<voe::ChannelProxy> VoiceEngineImpl::GetChannelProxy( | 65 rtc::scoped_ptr<voe::ChannelProxy> VoiceEngineImpl::GetChannelProxy( |
| 67 int channel_id) { | 66 int channel_id) { |
| 68 RTC_DCHECK(channel_id >= 0); | 67 RTC_DCHECK(channel_id >= 0); |
| 69 CriticalSectionScoped cs(crit_sec()); | 68 rtc::CritScope cs(crit_sec()); |
| 70 RTC_DCHECK(statistics().Initialized()); | 69 RTC_DCHECK(statistics().Initialized()); |
| 71 return rtc::scoped_ptr<voe::ChannelProxy>( | 70 return rtc::scoped_ptr<voe::ChannelProxy>( |
| 72 new voe::ChannelProxy(channel_manager().GetChannel(channel_id))); | 71 new voe::ChannelProxy(channel_manager().GetChannel(channel_id))); |
| 73 } | 72 } |
| 74 | 73 |
| 75 VoiceEngine* VoiceEngine::Create() { | 74 VoiceEngine* VoiceEngine::Create() { |
| 76 Config* config = new Config(); | 75 Config* config = new Config(); |
| 77 return GetVoiceEngine(config, true); | 76 return GetVoiceEngine(config, true); |
| 78 } | 77 } |
| 79 | 78 |
| (...skipping 70 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 150 | 149 |
| 151 std::string VoiceEngine::GetVersionString() { | 150 std::string VoiceEngine::GetVersionString() { |
| 152 std::string version = "VoiceEngine 4.1.0"; | 151 std::string version = "VoiceEngine 4.1.0"; |
| 153 #ifdef WEBRTC_EXTERNAL_TRANSPORT | 152 #ifdef WEBRTC_EXTERNAL_TRANSPORT |
| 154 version += " (External transport build)"; | 153 version += " (External transport build)"; |
| 155 #endif | 154 #endif |
| 156 return version; | 155 return version; |
| 157 } | 156 } |
| 158 | 157 |
| 159 } // namespace webrtc | 158 } // namespace webrtc |
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