ec6fbd2776
First approach to remove parts of the heavy load done for encoding, and preparation for sending, from native audio thread to separate task queue. With this change we will give the native input audio thread more time to "relax" between successive audio captures. Separate profiling done on Android has verified that the change works well; the load is now redistributed and the load of the native AudioRecordThread is reduced. Similar conclusions should be valid for all other OS:es as well. BUG=NONE CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng Review-Url: https://codereview.webrtc.org/2665693002 Cr-Commit-Position: refs/heads/master@{#17488}
85 lines
3.0 KiB
C++
85 lines
3.0 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/base/criticalsection.h"
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#include "webrtc/base/scoped_ref_ptr.h"
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#include "webrtc/base/task_queue.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/base/thread_checker.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/include/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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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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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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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<AudioProcessing> audioproc_;
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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_ 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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