73d65513f1
This CL modifies the ADM interface to ensure that an external ADM can't call Create and Destroy any longer. It also contains some minor style nits to conform better with the Chromium style guide. Review URL: http://webrtc-codereview.appspot.com/133014 git-svn-id: http://webrtc.googlecode.com/svn/trunk@552 4adac7df-926f-26a2-2b94-8c16560cd09d
97 lines
2.8 KiB
C++
97 lines
2.8 KiB
C++
/*
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* Copyright (c) 2011 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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#include "shared_data.h"
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#include "audio_processing.h"
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#include "critical_section_wrapper.h"
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#include "channel.h"
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#include "output_mixer.h"
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#include "trace.h"
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#include "transmit_mixer.h"
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namespace webrtc {
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namespace voe {
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static WebRtc_Word32 _gInstanceCounter = 0;
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SharedData::SharedData() :
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_instanceId(++_gInstanceCounter),
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_apiCritPtr(CriticalSectionWrapper::CreateCriticalSection()),
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_channelManager(_gInstanceCounter),
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_engineStatistics(_gInstanceCounter),
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_audioDevicePtr(NULL),
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_audioProcessingModulePtr(NULL),
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_moduleProcessThreadPtr(ProcessThread::CreateProcessThread()),
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_externalRecording(false),
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_externalPlayout(false)
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{
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Trace::CreateTrace();
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Trace::SetLevelFilter(WEBRTC_VOICE_ENGINE_DEFAULT_TRACE_FILTER);
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if (OutputMixer::Create(_outputMixerPtr, _gInstanceCounter) == 0)
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{
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_outputMixerPtr->SetEngineInformation(_engineStatistics);
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}
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if (TransmitMixer::Create(_transmitMixerPtr, _gInstanceCounter) == 0)
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{
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_transmitMixerPtr->SetEngineInformation(*_moduleProcessThreadPtr,
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_engineStatistics,
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_channelManager);
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}
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_audioDeviceLayer = AudioDeviceModule::kPlatformDefaultAudio;
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}
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SharedData::~SharedData()
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{
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OutputMixer::Destroy(_outputMixerPtr);
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TransmitMixer::Destroy(_transmitMixerPtr);
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if (_audioDevicePtr) {
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_audioDevicePtr->Release();
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}
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AudioProcessing::Destroy(_audioProcessingModulePtr);
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delete _apiCritPtr;
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ProcessThread::DestroyProcessThread(_moduleProcessThreadPtr);
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Trace::ReturnTrace();
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}
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WebRtc_UWord16
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SharedData::NumOfSendingChannels()
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{
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WebRtc_Word32 numOfChannels = _channelManager.NumOfChannels();
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if (numOfChannels <= 0)
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{
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return 0;
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}
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WebRtc_UWord16 nChannelsSending(0);
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WebRtc_Word32* channelsArray = new WebRtc_Word32[numOfChannels];
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_channelManager.GetChannelIds(channelsArray, numOfChannels);
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for (int i = 0; i < numOfChannels; i++)
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{
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voe::ScopedChannel sc(_channelManager, channelsArray[i]);
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Channel* chPtr = sc.ChannelPtr();
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if (chPtr)
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{
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if (chPtr->Sending())
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{
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nChannelsSending++;
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}
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}
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}
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delete [] channelsArray;
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return nChannelsSending;
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}
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} // namespace voe
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} // namespace webrtc
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