0c3e12b7bf
* 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
92 lines
3.2 KiB
C++
92 lines
3.2 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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#ifndef WEBRTC_VOICE_ENGINE_SHARED_DATA_H
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#define WEBRTC_VOICE_ENGINE_SHARED_DATA_H
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/audio_device/include/audio_device.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/utility/interface/process_thread.h"
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#include "webrtc/voice_engine/channel_manager.h"
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#include "webrtc/voice_engine/statistics.h"
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#include "webrtc/voice_engine/voice_engine_defines.h"
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class ProcessThread;
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namespace webrtc {
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class Config;
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class CriticalSectionWrapper;
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namespace voe {
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class TransmitMixer;
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class OutputMixer;
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class SharedData
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{
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public:
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// Public accessors.
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uint32_t instance_id() const { return _instanceId; }
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Statistics& statistics() { return _engineStatistics; }
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ChannelManager& channel_manager() { return _channelManager; }
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AudioDeviceModule* audio_device() { return _audioDevicePtr; }
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void set_audio_device(AudioDeviceModule* audio_device);
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AudioProcessing* audio_processing() { return audioproc_.get(); }
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void set_audio_processing(AudioProcessing* audio_processing);
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TransmitMixer* transmit_mixer() { return _transmitMixerPtr; }
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OutputMixer* output_mixer() { return _outputMixerPtr; }
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CriticalSectionWrapper* crit_sec() { return _apiCritPtr; }
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bool ext_recording() const { return _externalRecording; }
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void set_ext_recording(bool value) { _externalRecording = value; }
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bool ext_playout() const { return _externalPlayout; }
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void set_ext_playout(bool value) { _externalPlayout = value; }
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ProcessThread* process_thread() { return _moduleProcessThreadPtr.get(); }
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AudioDeviceModule::AudioLayer audio_device_layer() const {
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return _audioDeviceLayer;
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}
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void set_audio_device_layer(AudioDeviceModule::AudioLayer layer) {
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_audioDeviceLayer = layer;
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}
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int NumOfSendingChannels();
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int NumOfPlayingChannels();
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// Convenience methods for calling statistics().SetLastError().
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void SetLastError(int32_t error) const;
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void SetLastError(int32_t error, TraceLevel level) const;
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void SetLastError(int32_t error, TraceLevel level,
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const char* msg) const;
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protected:
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const uint32_t _instanceId;
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CriticalSectionWrapper* _apiCritPtr;
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ChannelManager _channelManager;
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Statistics _engineStatistics;
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AudioDeviceModule* _audioDevicePtr;
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OutputMixer* _outputMixerPtr;
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TransmitMixer* _transmitMixerPtr;
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rtc::scoped_ptr<AudioProcessing> audioproc_;
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rtc::scoped_ptr<ProcessThread> _moduleProcessThreadPtr;
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bool _externalRecording;
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bool _externalPlayout;
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AudioDeviceModule::AudioLayer _audioDeviceLayer;
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SharedData(const Config& config);
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virtual ~SharedData();
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};
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} // namespace voe
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} // namespace webrtc
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#endif // WEBRTC_VOICE_ENGINE_SHARED_DATA_H
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