| Index: content/renderer/media/webrtc_local_audio_track.h
|
| diff --git a/content/renderer/media/webrtc_local_audio_track.h b/content/renderer/media/webrtc_local_audio_track.h
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| new file mode 100644
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| index 0000000000000000000000000000000000000000..1c88465d51eee729079c5019e3fc8cbdca322357
|
| --- /dev/null
|
| +++ b/content/renderer/media/webrtc_local_audio_track.h
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| @@ -0,0 +1,106 @@
|
| +// Copyright 2013 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_LOCAL_AUDIO_TRACK_H_
|
| +#define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
|
| +
|
| +#include <list>
|
| +#include <string>
|
| +
|
| +#include "base/macros.h"
|
| +#include "base/memory/ref_counted.h"
|
| +#include "base/synchronization/lock.h"
|
| +#include "base/threading/thread_checker.h"
|
| +#include "content/renderer/media/media_stream_audio_track.h"
|
| +#include "content/renderer/media/tagged_list.h"
|
| +#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
|
| +#include "media/base/audio_parameters.h"
|
| +
|
| +namespace media {
|
| +class AudioBus;
|
| +}
|
| +
|
| +namespace content {
|
| +
|
| +class MediaStreamAudioLevelCalculator;
|
| +class MediaStreamAudioProcessor;
|
| +class MediaStreamAudioSink;
|
| +class MediaStreamAudioSinkOwner;
|
| +class MediaStreamAudioTrackSink;
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| +
|
| +// A WebRtcLocalAudioTrack manages thread-safe connects/disconnects to sinks,
|
| +// and the delivery of audio data from the source to the sinks.
|
| +class CONTENT_EXPORT WebRtcLocalAudioTrack
|
| + : NON_EXPORTED_BASE(public MediaStreamAudioTrack) {
|
| + public:
|
| + explicit WebRtcLocalAudioTrack(
|
| + scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter);
|
| +
|
| + ~WebRtcLocalAudioTrack() override;
|
| +
|
| + // Add a sink to the track. This function will trigger a OnSetFormat()
|
| + // call on the |sink|.
|
| + // Called on the main render thread.
|
| + void AddSink(MediaStreamAudioSink* sink) override;
|
| +
|
| + // Remove a sink from the track.
|
| + // Called on the main render thread.
|
| + void RemoveSink(MediaStreamAudioSink* sink) override;
|
| +
|
| + // Overrides for MediaStreamTrack.
|
| + void SetEnabled(bool enabled) override;
|
| + webrtc::AudioTrackInterface* GetAudioAdapter() override;
|
| + media::AudioParameters GetOutputFormat() const override;
|
| +
|
| + // Method called by the capturer to deliver the capture data.
|
| + // Called on the capture audio thread.
|
| + void Capture(const media::AudioBus& audio_bus,
|
| + base::TimeTicks estimated_capture_time);
|
| +
|
| + // Method called by the capturer to set the audio parameters used by source
|
| + // of the capture data..
|
| + // Called on the capture audio thread.
|
| + void OnSetFormat(const media::AudioParameters& params);
|
| +
|
| + // Called by the capturer before the audio data flow begins to set the object
|
| + // that provides shared access to the current audio signal level.
|
| + void SetLevel(scoped_refptr<MediaStreamAudioLevelCalculator::Level> level);
|
| +
|
| + // Called by the capturer before the audio data flow begins to provide a
|
| + // reference to the audio processor so that the track can query stats from it.
|
| + void SetAudioProcessor(scoped_refptr<MediaStreamAudioProcessor> processor);
|
| +
|
| + private:
|
| + typedef TaggedList<MediaStreamAudioTrackSink> SinkList;
|
| +
|
| + // MediaStreamAudioTrack override.
|
| + void OnStop() final;
|
| +
|
| + // All usage of libjingle is through this adapter. The adapter holds
|
| + // a pointer to this object, but no reference.
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| + const scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_;
|
| +
|
| + // A tagged list of sinks that the audio data is fed to. Tags
|
| + // indicate tracks that need to be notified that the audio format
|
| + // has changed.
|
| + SinkList sinks_;
|
| +
|
| + // Tests that methods are called on libjingle's signaling thread.
|
| + base::ThreadChecker signal_thread_checker_;
|
| +
|
| + // Used to DCHECK that some methods are called on the capture audio thread.
|
| + base::ThreadChecker capture_thread_checker_;
|
| +
|
| + // Protects |params_| and |sinks_|.
|
| + mutable base::Lock lock_;
|
| +
|
| + // Audio parameters of the audio capture stream.
|
| + media::AudioParameters audio_parameters_;
|
| +
|
| + DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
|
| +};
|
| +
|
| +} // namespace content
|
| +
|
| +#endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
|
|
|