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bb547203bf
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
82 lines
2.9 KiB
C++
82 lines
2.9 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 <memory>
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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/include/process_thread.h"
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#include "webrtc/rtc_base/criticalsection.h"
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#include "webrtc/rtc_base/scoped_ref_ptr.h"
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#include "webrtc/rtc_base/task_queue.h"
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#include "webrtc/rtc_base/thread_annotations.h"
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#include "webrtc/rtc_base/thread_checker.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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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.get(); }
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void set_audio_device(
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const rtc::scoped_refptr<AudioDeviceModule>& audio_device);
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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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rtc::CriticalSection* crit_sec() { return &_apiCritPtr; }
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ProcessThread* process_thread() { return _moduleProcessThreadPtr.get(); }
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rtc::TaskQueue* encoder_queue();
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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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rtc::ThreadChecker construction_thread_;
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const uint32_t _instanceId;
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rtc::CriticalSection _apiCritPtr;
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ChannelManager _channelManager;
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Statistics _engineStatistics;
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rtc::scoped_refptr<AudioDeviceModule> _audioDevicePtr;
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OutputMixer* _outputMixerPtr;
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TransmitMixer* _transmitMixerPtr;
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std::unique_ptr<ProcessThread> _moduleProcessThreadPtr;
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// |encoder_queue| is defined last to ensure all pending tasks are cancelled
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// and deleted before any other members.
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rtc::TaskQueue encoder_queue_ RTC_ACCESS_ON(construction_thread_);
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SharedData();
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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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