9080518a39
I mistakenly ommitted the checks when logging.h was ported from libjingle to webrtc. This caused a significant CPU cost for logs which were later filtered out anyway. Verified with LS_VERBOSE logging in neteq4, running: $ out/Release/modules_unittests \ --gtest_filter=NetEqDecodingTest.TestBitExactness \ --gtest_repeat=50 > time.txt $ grep "case ran" time.txt | grep "[0-9]* ms" -o | sort Results on a MacBook Retina, averaged over 5 runs: Verbose logs disabled: 666 ms Exisiting implementation, verbose logs enabled: 944 ms (1.42x) New implementation, verbose logs enabled: 673 ms (1.01x) BUG=2314 R=henrik.lundin@webrtc.org, henrike@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2160005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4682 4adac7df-926f-26a2-2b94-8c16560cd09d
156 lines
4.2 KiB
C++
156 lines
4.2 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#if defined(WEBRTC_ANDROID) && !defined(WEBRTC_ANDROID_OPENSLES)
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#include "webrtc/modules/audio_device/android/audio_device_jni_android.h"
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#endif
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#include "webrtc/system_wrappers/interface/trace.h"
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#include "webrtc/voice_engine/voice_engine_impl.h"
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namespace webrtc
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{
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// Counter to be ensure that we can add a correct ID in all static trace
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// methods. It is not the nicest solution, especially not since we already
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// have a counter in VoEBaseImpl. In other words, there is room for
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// improvement here.
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static int32_t gVoiceEngineInstanceCounter = 0;
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extern "C"
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{
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WEBRTC_DLLEXPORT VoiceEngine* GetVoiceEngine();
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VoiceEngine* GetVoiceEngine()
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{
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VoiceEngineImpl* self = new VoiceEngineImpl();
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if (self != NULL)
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{
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self->AddRef(); // First reference. Released in VoiceEngine::Delete.
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gVoiceEngineInstanceCounter++;
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}
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return self;
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}
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} // extern "C"
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int VoiceEngineImpl::AddRef() {
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return ++_ref_count;
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}
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// This implements the Release() method for all the inherited interfaces.
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int VoiceEngineImpl::Release() {
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int new_ref = --_ref_count;
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assert(new_ref >= 0);
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if (new_ref == 0) {
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WEBRTC_TRACE(kTraceApiCall, kTraceVoice, -1,
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"VoiceEngineImpl self deleting (voiceEngine=0x%p)",
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this);
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delete this;
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}
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return new_ref;
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}
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VoiceEngine* VoiceEngine::Create()
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{
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#if (defined _WIN32)
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HMODULE hmod_ = LoadLibrary(TEXT("VoiceEngineTestingDynamic.dll"));
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if (hmod_)
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{
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typedef VoiceEngine* (*PfnGetVoiceEngine)(void);
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PfnGetVoiceEngine pfn = (PfnGetVoiceEngine)GetProcAddress(
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hmod_,"GetVoiceEngine");
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if (pfn)
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{
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VoiceEngine* self = pfn();
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return (self);
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}
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}
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#endif
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return GetVoiceEngine();
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}
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int VoiceEngine::SetTraceFilter(unsigned int filter)
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{
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WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
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VoEId(gVoiceEngineInstanceCounter, -1),
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"SetTraceFilter(filter=0x%x)", filter);
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// Remember old filter
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uint32_t oldFilter = Trace::level_filter();
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Trace::set_level_filter(filter);
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// If previous log was ignored, log again after changing filter
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if (kTraceNone == oldFilter)
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{
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WEBRTC_TRACE(kTraceApiCall, kTraceVoice, -1,
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"SetTraceFilter(filter=0x%x)", filter);
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}
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return 0;
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}
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int VoiceEngine::SetTraceFile(const char* fileNameUTF8,
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bool addFileCounter)
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{
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int ret = Trace::SetTraceFile(fileNameUTF8, addFileCounter);
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WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
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VoEId(gVoiceEngineInstanceCounter, -1),
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"SetTraceFile(fileNameUTF8=%s, addFileCounter=%d)",
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fileNameUTF8, addFileCounter);
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return (ret);
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}
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int VoiceEngine::SetTraceCallback(TraceCallback* callback)
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{
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WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
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VoEId(gVoiceEngineInstanceCounter, -1),
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"SetTraceCallback(callback=0x%x)", callback);
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return (Trace::SetTraceCallback(callback));
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}
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bool VoiceEngine::Delete(VoiceEngine*& voiceEngine)
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{
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if (voiceEngine == NULL)
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return false;
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VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
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// Release the reference that was added in GetVoiceEngine.
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int ref = s->Release();
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voiceEngine = NULL;
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if (ref != 0) {
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WEBRTC_TRACE(kTraceWarning, kTraceVoice, -1,
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"VoiceEngine::Delete did not release the very last reference. "
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"%d references remain.", ref);
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}
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return true;
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}
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int VoiceEngine::SetAndroidObjects(void* javaVM, void* env, void* context)
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{
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#ifdef WEBRTC_ANDROID
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#ifdef WEBRTC_ANDROID_OPENSLES
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return 0;
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#else
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return AudioDeviceAndroidJni::SetAndroidAudioDeviceObjects(
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javaVM, env, context);
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#endif
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#else
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return -1;
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#endif
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}
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} // namespace webrtc
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