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https://github.com/mollyim/webrtc.git
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d367921eb1
This also changes RtpReceiver and RemoteAudioSource to have two-step initialization, since in Unified Plan RtpReceivers are created much earlier than in Plan B. Bug: webrtc:7600 Change-Id: Ia135d25eb8bcab22969007b3a825a5a43ce62bf4 Reviewed-on: https://webrtc-review.googlesource.com/39382 Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21681}
76 lines
2.3 KiB
C++
76 lines
2.3 KiB
C++
/*
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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_REMOTEAUDIOSOURCE_H_
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#define PC_REMOTEAUDIOSOURCE_H_
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#include <list>
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#include <string>
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#include "api/call/audio_sink.h"
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#include "api/notifier.h"
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#include "pc/channel.h"
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/messagehandler.h"
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namespace rtc {
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struct Message;
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class Thread;
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} // namespace rtc
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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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rtc::MessageHandler {
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public:
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explicit RemoteAudioSource(rtc::Thread* worker_thread);
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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::VoiceMediaChannel* media_channel, uint32_t ssrc);
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void Stop(cricket::VoiceMediaChannel* media_channel, uint32_t ssrc);
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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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void OnMessage(rtc::Message* msg) override;
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rtc::Thread* const main_thread_;
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rtc::Thread* const worker_thread_;
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std::list<AudioObserver*> audio_observers_;
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rtc::CriticalSection 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_REMOTEAUDIOSOURCE_H_
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