| Index: content/renderer/media/webrtc/media_stream_remote_audio_track.h
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| diff --git a/content/renderer/media/webrtc/media_stream_remote_audio_track.h b/content/renderer/media/webrtc/media_stream_remote_audio_track.h
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| new file mode 100644
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| index 0000000000000000000000000000000000000000..9e48dfb40d7350b241d8a6db9466aed2272ec5f3
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| --- /dev/null
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| +++ b/content/renderer/media/webrtc/media_stream_remote_audio_track.h
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| @@ -0,0 +1,89 @@
|
| +// Copyright 2015 The Chromium Authors. All rights reserved.
|
| +// Use of this source code is governed by a BSD-style license that can be
|
| +// found in the LICENSE file.
|
| +
|
| +#ifndef CONTENT_RENDERER_MEDIA_WEBRTC_MEDIA_STREAM_REMOTE_AUDIO_TRACK_H_
|
| +#define CONTENT_RENDERER_MEDIA_WEBRTC_MEDIA_STREAM_REMOTE_AUDIO_TRACK_H_
|
| +
|
| +#include "base/memory/ref_counted.h"
|
| +#include "base/threading/thread_checker.h"
|
| +#include "content/renderer/media/media_stream_audio_track.h"
|
| +#include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
|
| +
|
| +namespace content {
|
| +
|
| +class MediaStreamRemoteAudioSource;
|
| +
|
| +// MediaStreamRemoteAudioTrack is a WebRTC specific implementation of an
|
| +// audio track received from a PeerConnection.
|
| +// TODO(tommi): Chrome shouldn't have to care about remote vs local so
|
| +// we should have a single track implementation that delegates to the
|
| +// sources that do different things depending on the type of source.
|
| +class MediaStreamRemoteAudioTrack : public MediaStreamAudioTrack {
|
| + public:
|
| + explicit MediaStreamRemoteAudioTrack(
|
| + const blink::WebMediaStreamSource& source, bool enabled);
|
| + ~MediaStreamRemoteAudioTrack() override;
|
| +
|
| + // MediaStreamTrack override.
|
| + void SetEnabled(bool enabled) override;
|
| +
|
| + // MediaStreamAudioTrack overrides.
|
| + void AddSink(MediaStreamAudioSink* sink) override;
|
| + void RemoveSink(MediaStreamAudioSink* sink) override;
|
| + media::AudioParameters GetOutputFormat() const override;
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| +
|
| + webrtc::AudioTrackInterface* GetAudioAdapter() override;
|
| +
|
| + private:
|
| + // MediaStreamAudioTrack override.
|
| + void OnStop() final;
|
| +
|
| + MediaStreamRemoteAudioSource* source() const;
|
| +
|
| + blink::WebMediaStreamSource source_;
|
| + bool enabled_;
|
| +};
|
| +
|
| +// Inheriting from ExtraData directly since MediaStreamAudioSource has
|
| +// too much unrelated bloat.
|
| +// TODO(tommi): MediaStreamAudioSource needs refactoring.
|
| +// TODO(miu): On it! ;-)
|
| +class MediaStreamRemoteAudioSource
|
| + : public blink::WebMediaStreamSource::ExtraData {
|
| + public:
|
| + explicit MediaStreamRemoteAudioSource(
|
| + const scoped_refptr<webrtc::AudioTrackInterface>& track);
|
| + ~MediaStreamRemoteAudioSource() override;
|
| +
|
| + // Controls whether or not the source is included in the main, mixed, audio
|
| + // output from WebRTC as rendered by WebRtcAudioRenderer (media players).
|
| + void SetEnabledForMixing(bool enabled);
|
| +
|
| + // Adds an audio sink for a track belonging to this source.
|
| + // |enabled| is the enabled state of the track and can be updated via
|
| + // a call to SetSinksEnabled.
|
| + void AddSink(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track,
|
| + bool enabled);
|
| +
|
| + // Removes an audio sink for a track belonging to this source.
|
| + void RemoveSink(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track);
|
| +
|
| + // Turns audio callbacks on/off for all sinks belonging to a track.
|
| + void SetSinksEnabled(MediaStreamAudioTrack* track, bool enabled);
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| +
|
| + // Removes all sinks belonging to a track.
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| + void RemoveAll(MediaStreamAudioTrack* track);
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| +
|
| + webrtc::AudioTrackInterface* GetAudioAdapter();
|
| +
|
| + private:
|
| + class AudioSink;
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| + std::unique_ptr<AudioSink> sink_;
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| + const scoped_refptr<webrtc::AudioTrackInterface> track_;
|
| + base::ThreadChecker thread_checker_;
|
| +};
|
| +
|
| +} // namespace content
|
| +
|
| +#endif // CONTENT_RENDERER_MEDIA_WEBRTC_MEDIA_STREAM_REMOTE_AUDIO_TRACK_H_
|
|
|