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TMessagesProj/jni/voip/webrtc/pc/remote_audio_source.h
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TMessagesProj/jni/voip/webrtc/pc/remote_audio_source.h
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/*
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* Copyright 2014 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 PC_REMOTE_AUDIO_SOURCE_H_
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#define PC_REMOTE_AUDIO_SOURCE_H_
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#include <stdint.h>
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#include <list>
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#include <string>
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#include "absl/types/optional.h"
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#include "api/call/audio_sink.h"
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#include "api/media_stream_interface.h"
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#include "api/notifier.h"
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#include "api/task_queue/task_queue_base.h"
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#include "media/base/media_channel.h"
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#include "rtc_base/synchronization/mutex.h"
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namespace webrtc {
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// This class implements the audio source used by the remote audio track.
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// This class works by configuring itself as a sink with the underlying media
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// engine, then when receiving data will fan out to all added sinks.
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class RemoteAudioSource : public Notifier<AudioSourceInterface> {
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public:
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// In Unified Plan, receivers map to m= sections and their tracks and sources
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// survive SSRCs being reconfigured. The life cycle of the remote audio source
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// is associated with the life cycle of the m= section, and thus even if an
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// audio channel is destroyed the RemoteAudioSource should kSurvive.
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//
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// In Plan B however, remote audio sources map 1:1 with an SSRCs and if an
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// audio channel is destroyed, the RemoteAudioSource should kEnd.
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enum class OnAudioChannelGoneAction {
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kSurvive,
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kEnd,
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};
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explicit RemoteAudioSource(
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TaskQueueBase* worker_thread,
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OnAudioChannelGoneAction on_audio_channel_gone_action);
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// Register and unregister remote audio source with the underlying media
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// engine.
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void Start(cricket::VoiceMediaReceiveChannelInterface* media_channel,
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absl::optional<uint32_t> ssrc);
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void Stop(cricket::VoiceMediaReceiveChannelInterface* media_channel,
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absl::optional<uint32_t> ssrc);
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void SetState(SourceState new_state);
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// MediaSourceInterface implementation.
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MediaSourceInterface::SourceState state() const override;
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bool remote() const override;
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// AudioSourceInterface implementation.
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void SetVolume(double volume) override;
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void RegisterAudioObserver(AudioObserver* observer) override;
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void UnregisterAudioObserver(AudioObserver* observer) override;
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void AddSink(AudioTrackSinkInterface* sink) override;
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void RemoveSink(AudioTrackSinkInterface* sink) override;
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protected:
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~RemoteAudioSource() override;
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private:
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// These are callbacks from the media engine.
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class AudioDataProxy;
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void OnData(const AudioSinkInterface::Data& audio);
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void OnAudioChannelGone();
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TaskQueueBase* const main_thread_;
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TaskQueueBase* const worker_thread_;
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const OnAudioChannelGoneAction on_audio_channel_gone_action_;
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std::list<AudioObserver*> audio_observers_;
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Mutex sink_lock_;
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std::list<AudioTrackSinkInterface*> sinks_;
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SourceState state_;
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};
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} // namespace webrtc
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#endif // PC_REMOTE_AUDIO_SOURCE_H_
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