Additionally: * Moving all implementation inside RemoteAudioTrack into AudioTrack and remove RemoteAudioTrack. * AddSink/RemoveSink are now on all audio sources (like they are for video sources). While doing this I found that some of our tests are broken :) and fixed them. They were broken because AudioTrack didn't previously do much such as updating its state. BUG=chromium:569526 Review URL: https://codereview.webrtc.org/1522903002 Cr-Commit-Position: refs/heads/master@{#11026}
74 lines
2.9 KiB
C++
74 lines
2.9 KiB
C++
/*
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* libjingle
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* Copyright 2012 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifndef TALK_APP_WEBRTC_LOCALAUDIOSOURCE_H_
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#define TALK_APP_WEBRTC_LOCALAUDIOSOURCE_H_
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#include "talk/app/webrtc/mediastreaminterface.h"
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#include "talk/app/webrtc/notifier.h"
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#include "talk/app/webrtc/peerconnectioninterface.h"
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#include "talk/media/base/mediachannel.h"
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#include "webrtc/base/scoped_ptr.h"
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// LocalAudioSource implements AudioSourceInterface.
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// This contains settings for switching audio processing on and off.
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namespace webrtc {
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class MediaConstraintsInterface;
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class LocalAudioSource : public Notifier<AudioSourceInterface> {
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public:
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// Creates an instance of LocalAudioSource.
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static rtc::scoped_refptr<LocalAudioSource> Create(
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const PeerConnectionFactoryInterface::Options& options,
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const MediaConstraintsInterface* constraints);
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SourceState state() const override { return source_state_; }
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bool remote() const override { return false; }
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virtual const cricket::AudioOptions& options() const { return options_; }
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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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LocalAudioSource() : source_state_(kInitializing) {}
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~LocalAudioSource() override {}
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private:
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void Initialize(const PeerConnectionFactoryInterface::Options& options,
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const MediaConstraintsInterface* constraints);
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cricket::AudioOptions options_;
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SourceState source_state_;
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};
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} // namespace webrtc
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#endif // TALK_APP_WEBRTC_LOCALAUDIOSOURCE_H_
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