a9cc40b7d2
[This CL is a rebase of an original CL by solenberg@: https://codereview.webrtc.org/2948763002/ which in turn was a rebase of an original CL by peah@: https://chromium-review.googlesource.com/c/527032/] Allow an external audio processing module to be used in WebRTC This CL adds support for optionally using an externally created audio processing module in a peerconnection. The ownership is shared between the peerconnection and the external creator of the module. As part of this the internal ownership of the audio processing module is moved from VoiceEngine to WebRtcVoiceEngine. BUG=webrtc:7775 Review-Url: https://codereview.webrtc.org/2961723004 Cr-Commit-Position: refs/heads/master@{#18837}
83 lines
2.9 KiB
C++
83 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/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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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_ 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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