blob: 703b6a4470e686c13505f506f728d3b8470625ee [file] [log] [blame]
/*
* 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)
#if defined(WEBRTC_ANDROID_OPENSLES)
#include "webrtc/modules/audio_device/android/audio_manager_jni.h"
#else
#include "webrtc/modules/audio_device/android/audio_device_jni_android.h"
#endif
#endif
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/system_wrappers/interface/trace.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(const Config* config, bool owns_config)
{
#if (defined _WIN32)
HMODULE hmod = LoadLibrary(TEXT("VoiceEngineTestingDynamic.dll"));
if (hmod) {
typedef VoiceEngine* (*PfnGetVoiceEngine)(void);
PfnGetVoiceEngine pfn = (PfnGetVoiceEngine)GetProcAddress(
hmod,"GetVoiceEngine");
if (pfn) {
VoiceEngine* self = pfn();
if (owns_config) {
delete config;
}
return (self);
}
}
#endif
VoiceEngineImpl* self = new VoiceEngineImpl(config, owns_config);
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);
delete this;
}
return new_ref;
}
VoiceEngine* VoiceEngine::Create() {
Config* config = new Config();
config->Set<AudioCodingModuleFactory>(new AudioCodingModuleFactory());
return GetVoiceEngine(config, true);
}
VoiceEngine* VoiceEngine::Create(const Config& config) {
return GetVoiceEngine(&config, false);
}
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;
}
int VoiceEngine::SetAndroidObjects(void* javaVM, void* env, void* context)
{
#ifdef WEBRTC_ANDROID
#ifdef WEBRTC_ANDROID_OPENSLES
AudioManagerJni::SetAndroidAudioDeviceObjects(javaVM, env, context);
return 0;
#else
return AudioDeviceAndroidJni::SetAndroidAudioDeviceObjects(
javaVM, env, context);
#endif
#else
return -1;
#endif
}
} // namespace webrtc