|  | /* | 
|  | *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 
|  | * | 
|  | *  Use of this source code is governed by a BSD-style license | 
|  | *  that can be found in the LICENSE file in the root of the source | 
|  | *  tree. An additional intellectual property rights grant can be found | 
|  | *  in the file PATENTS.  All contributing project authors may | 
|  | *  be found in the AUTHORS file in the root of the source tree. | 
|  | */ | 
|  |  | 
|  | #if defined(WEBRTC_ANDROID) | 
|  | #include "webrtc/modules/audio_device/android/audio_device_template.h" | 
|  | #include "webrtc/modules/audio_device/android/audio_record_jni.h" | 
|  | #include "webrtc/modules/audio_device/android/audio_track_jni.h" | 
|  | #include "webrtc/modules/utility/include/jvm_android.h" | 
|  | #endif | 
|  |  | 
|  | #include "webrtc/base/checks.h" | 
|  | #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 
|  | #include "webrtc/system_wrappers/include/trace.h" | 
|  | #include "webrtc/voice_engine/channel_proxy.h" | 
|  | #include "webrtc/voice_engine/voice_engine_impl.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | // Counter to be ensure that we can add a correct ID in all static trace | 
|  | // methods. It is not the nicest solution, especially not since we already | 
|  | // have a counter in VoEBaseImpl. In other words, there is room for | 
|  | // improvement here. | 
|  | static int32_t gVoiceEngineInstanceCounter = 0; | 
|  |  | 
|  | VoiceEngine* GetVoiceEngine() { | 
|  | VoiceEngineImpl* self = new VoiceEngineImpl(); | 
|  | if (self != NULL) { | 
|  | self->AddRef();  // First reference.  Released in VoiceEngine::Delete. | 
|  | gVoiceEngineInstanceCounter++; | 
|  | } | 
|  | return self; | 
|  | } | 
|  |  | 
|  | int VoiceEngineImpl::AddRef() { | 
|  | return ++_ref_count; | 
|  | } | 
|  |  | 
|  | // This implements the Release() method for all the inherited interfaces. | 
|  | int VoiceEngineImpl::Release() { | 
|  | int new_ref = --_ref_count; | 
|  | assert(new_ref >= 0); | 
|  | if (new_ref == 0) { | 
|  | WEBRTC_TRACE(kTraceApiCall, kTraceVoice, -1, | 
|  | "VoiceEngineImpl self deleting (voiceEngine=0x%p)", this); | 
|  |  | 
|  | // Clear any pointers before starting destruction. Otherwise worker- | 
|  | // threads will still have pointers to a partially destructed object. | 
|  | // Example: AudioDeviceBuffer::RequestPlayoutData() can access a | 
|  | // partially deconstructed |_ptrCbAudioTransport| during destruction | 
|  | // if we don't call Terminate here. | 
|  | Terminate(); | 
|  | delete this; | 
|  | } | 
|  |  | 
|  | return new_ref; | 
|  | } | 
|  |  | 
|  | std::unique_ptr<voe::ChannelProxy> VoiceEngineImpl::GetChannelProxy( | 
|  | int channel_id) { | 
|  | RTC_DCHECK(channel_id >= 0); | 
|  | rtc::CritScope cs(crit_sec()); | 
|  | RTC_DCHECK(statistics().Initialized()); | 
|  | return std::unique_ptr<voe::ChannelProxy>( | 
|  | new voe::ChannelProxy(channel_manager().GetChannel(channel_id))); | 
|  | } | 
|  |  | 
|  | VoiceEngine* VoiceEngine::Create() { | 
|  | return GetVoiceEngine(); | 
|  | } | 
|  |  | 
|  | int VoiceEngine::SetTraceFilter(unsigned int filter) { | 
|  | WEBRTC_TRACE(kTraceApiCall, kTraceVoice, | 
|  | VoEId(gVoiceEngineInstanceCounter, -1), | 
|  | "SetTraceFilter(filter=0x%x)", filter); | 
|  |  | 
|  | // Remember old filter | 
|  | uint32_t oldFilter = Trace::level_filter(); | 
|  | Trace::set_level_filter(filter); | 
|  |  | 
|  | // If previous log was ignored, log again after changing filter | 
|  | if (kTraceNone == oldFilter) { | 
|  | WEBRTC_TRACE(kTraceApiCall, kTraceVoice, -1, "SetTraceFilter(filter=0x%x)", | 
|  | filter); | 
|  | } | 
|  |  | 
|  | return 0; | 
|  | } | 
|  |  | 
|  | int VoiceEngine::SetTraceFile(const char* fileNameUTF8, bool addFileCounter) { | 
|  | int ret = Trace::SetTraceFile(fileNameUTF8, addFileCounter); | 
|  | WEBRTC_TRACE(kTraceApiCall, kTraceVoice, | 
|  | VoEId(gVoiceEngineInstanceCounter, -1), | 
|  | "SetTraceFile(fileNameUTF8=%s, addFileCounter=%d)", fileNameUTF8, | 
|  | addFileCounter); | 
|  | return (ret); | 
|  | } | 
|  |  | 
|  | int VoiceEngine::SetTraceCallback(TraceCallback* callback) { | 
|  | WEBRTC_TRACE(kTraceApiCall, kTraceVoice, | 
|  | VoEId(gVoiceEngineInstanceCounter, -1), | 
|  | "SetTraceCallback(callback=0x%x)", callback); | 
|  | return (Trace::SetTraceCallback(callback)); | 
|  | } | 
|  |  | 
|  | bool VoiceEngine::Delete(VoiceEngine*& voiceEngine) { | 
|  | if (voiceEngine == NULL) | 
|  | return false; | 
|  |  | 
|  | VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine); | 
|  | // Release the reference that was added in GetVoiceEngine. | 
|  | int ref = s->Release(); | 
|  | voiceEngine = NULL; | 
|  |  | 
|  | if (ref != 0) { | 
|  | WEBRTC_TRACE( | 
|  | kTraceWarning, kTraceVoice, -1, | 
|  | "VoiceEngine::Delete did not release the very last reference.  " | 
|  | "%d references remain.", | 
|  | ref); | 
|  | } | 
|  |  | 
|  | return true; | 
|  | } | 
|  |  | 
|  | #if !defined(WEBRTC_CHROMIUM_BUILD) | 
|  | // TODO(henrika): change types to JavaVM* and jobject instead of void*. | 
|  | int VoiceEngine::SetAndroidObjects(void* javaVM, void* context) { | 
|  | #ifdef WEBRTC_ANDROID | 
|  | webrtc::JVM::Initialize(reinterpret_cast<JavaVM*>(javaVM), | 
|  | reinterpret_cast<jobject>(context)); | 
|  | return 0; | 
|  | #else | 
|  | return -1; | 
|  | #endif | 
|  | } | 
|  | #endif | 
|  |  | 
|  | std::string VoiceEngine::GetVersionString() { | 
|  | std::string version = "VoiceEngine 4.1.0"; | 
|  | #ifdef WEBRTC_EXTERNAL_TRANSPORT | 
|  | version += " (External transport build)"; | 
|  | #endif | 
|  | return version; | 
|  | } | 
|  |  | 
|  | }  // namespace webrtc |