webrtc/voice_engine/voice_engine_impl.cc

/*
 *  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/interface/jvm_android.h"
#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);

    // 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;
}

VoiceEngine* VoiceEngine::Create() {
  Config* config = new Config();
  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;
}

#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));
  // The Android ADM implementation supports dynamic selection of the audio
  // layer in both directions if a default audio layer is selected. Both
  // Java-based audio backends are initialized here to ensure that the user
  // can switch backend dynamically as well.
  typedef AudioDeviceTemplate<AudioRecordJni, AudioTrackJni> AudioDevice;
  if (javaVM && context) {
    AudioDevice::SetAndroidAudioDeviceObjects(javaVM, context);
  } else {
    AudioDevice::ClearAndroidAudioDeviceObjects();
  }
  return 0;
#else
  return -1;
#endif
}
#endif

}  // namespace webrtc