Files
webrtc/webrtc/voice_engine/shared_data.cc
tommi@webrtc.org 0c3e12b7bf Revamp the ProcessThreadImpl implementation.
* Add a new WakeUp method that gives a module a chance to be called back right away on the worker thread.
* Wrote unit tests for the class.
* Significantly reduce the amount of locking.
  - ProcessThreadImpl itself does a lot less locking.
  - Reimplemented the way we keep track of when to make calls to Process.
    This reduces the amount of calls to TimeUntilNextProcess and since most implementations of that function grab a lock, this means less locking.
* Renamed ProcessThread::CreateProcessThread to ProcessThread::Create.
* Added thread checks for Start/Stop.  Threading model of other functions is now documented.
* We now log an error if an implementation of TimeUntilNextProcess returns a negative value (some implementations do, but the method should only return a positive nr of ms).
* Removed the DestroyProcessThread method and instead force callers to use scoped_ptr<> to maintain object lifetime.

BUG=2822
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35999004

Cr-Commit-Position: refs/heads/master@{#8261}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8261 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-06 09:44:45 +00:00

121 lines
3.5 KiB
C++

/*
* 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.
*/
#include "webrtc/voice_engine/shared_data.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/voice_engine/channel.h"
#include "webrtc/voice_engine/output_mixer.h"
#include "webrtc/voice_engine/transmit_mixer.h"
namespace webrtc {
namespace voe {
static int32_t _gInstanceCounter = 0;
SharedData::SharedData(const Config& config) :
_instanceId(++_gInstanceCounter),
_apiCritPtr(CriticalSectionWrapper::CreateCriticalSection()),
_channelManager(_gInstanceCounter, config),
_engineStatistics(_gInstanceCounter),
_audioDevicePtr(NULL),
_moduleProcessThreadPtr(ProcessThread::Create()),
_externalRecording(false),
_externalPlayout(false)
{
Trace::CreateTrace();
if (OutputMixer::Create(_outputMixerPtr, _gInstanceCounter) == 0)
{
_outputMixerPtr->SetEngineInformation(_engineStatistics);
}
if (TransmitMixer::Create(_transmitMixerPtr, _gInstanceCounter) == 0)
{
_transmitMixerPtr->SetEngineInformation(*_moduleProcessThreadPtr,
_engineStatistics,
_channelManager);
}
_audioDeviceLayer = AudioDeviceModule::kPlatformDefaultAudio;
}
SharedData::~SharedData()
{
OutputMixer::Destroy(_outputMixerPtr);
TransmitMixer::Destroy(_transmitMixerPtr);
if (_audioDevicePtr) {
_audioDevicePtr->Release();
}
delete _apiCritPtr;
_moduleProcessThreadPtr->Stop();
Trace::ReturnTrace();
}
void SharedData::set_audio_device(AudioDeviceModule* audio_device)
{
// AddRef first in case the pointers are equal.
if (audio_device)
audio_device->AddRef();
if (_audioDevicePtr)
_audioDevicePtr->Release();
_audioDevicePtr = audio_device;
}
void SharedData::set_audio_processing(AudioProcessing* audioproc) {
audioproc_.reset(audioproc);
_transmitMixerPtr->SetAudioProcessingModule(audioproc);
_outputMixerPtr->SetAudioProcessingModule(audioproc);
}
int SharedData::NumOfSendingChannels() {
ChannelManager::Iterator it(&_channelManager);
int sending_channels = 0;
for (ChannelManager::Iterator it(&_channelManager); it.IsValid();
it.Increment()) {
if (it.GetChannel()->Sending())
++sending_channels;
}
return sending_channels;
}
int SharedData::NumOfPlayingChannels() {
ChannelManager::Iterator it(&_channelManager);
int playout_channels = 0;
for (ChannelManager::Iterator it(&_channelManager); it.IsValid();
it.Increment()) {
if (it.GetChannel()->Playing())
++playout_channels;
}
return playout_channels;
}
void SharedData::SetLastError(int32_t error) const {
_engineStatistics.SetLastError(error);
}
void SharedData::SetLastError(int32_t error,
TraceLevel level) const {
_engineStatistics.SetLastError(error, level);
}
void SharedData::SetLastError(int32_t error, TraceLevel level,
const char* msg) const {
_engineStatistics.SetLastError(error, level, msg);
}
} // namespace voe
} // namespace webrtc