| Index: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
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| diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
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| deleted file mode 100644
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| index 72b80194b08ed09a01673c98c4ed9816aa4e6d74..0000000000000000000000000000000000000000
|
| --- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
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| +++ /dev/null
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| @@ -1,107 +0,0 @@
|
| -// Copyright 2014 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_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
|
| -#define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
|
| -
|
| -#include <vector>
|
| -
|
| -#include "base/memory/ref_counted.h"
|
| -#include "base/memory/scoped_vector.h"
|
| -#include "base/single_thread_task_runner.h"
|
| -#include "base/synchronization/lock.h"
|
| -#include "content/common/content_export.h"
|
| -#include "content/renderer/media/media_stream_audio_level_calculator.h"
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| -#include "content/renderer/media/media_stream_audio_processor.h"
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| -#include "third_party/webrtc/api/mediastreamtrack.h"
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| -#include "third_party/webrtc/media/base/audiorenderer.h"
|
| -
|
| -namespace cricket {
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| -class AudioRenderer;
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| -}
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| -
|
| -namespace webrtc {
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| -class AudioSourceInterface;
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| -class AudioProcessorInterface;
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| -}
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| -
|
| -namespace content {
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| -
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| -class MediaStreamAudioProcessor;
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| -class WebRtcAudioSinkAdapter;
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| -class WebRtcLocalAudioTrack;
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| -
|
| -// Provides an implementation of the webrtc::AudioTrackInterface that can be
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| -// bound/unbound to/from a MediaStreamAudioTrack. In other words, this is an
|
| -// adapter that sits between the media stream object graph and WebRtc's object
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| -// graph and proxies between the two.
|
| -class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter
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| - : NON_EXPORTED_BASE(
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| - public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) {
|
| - public:
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| - static scoped_refptr<WebRtcLocalAudioTrackAdapter> Create(
|
| - const std::string& label,
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| - webrtc::AudioSourceInterface* track_source);
|
| -
|
| - WebRtcLocalAudioTrackAdapter(
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| - const std::string& label,
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| - webrtc::AudioSourceInterface* track_source,
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| - scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner);
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| -
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| - ~WebRtcLocalAudioTrackAdapter() override;
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| -
|
| - void Initialize(WebRtcLocalAudioTrack* owner);
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| -
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| - // Set the object that provides shared access to the current audio signal
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| - // level. This method may only be called once, before the audio data flow
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| - // starts, and before any calls to GetSignalLevel() might be made.
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| - void SetLevel(scoped_refptr<MediaStreamAudioLevelCalculator::Level> level);
|
| -
|
| - // Method called by the WebRtcLocalAudioTrack to set the processor that
|
| - // applies signal processing on the data of the track.
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| - // This class will keep a reference of the |processor|.
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| - // Called on the main render thread.
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| - // This method may only be called once, before the audio data flow starts, and
|
| - // before any calls to GetAudioProcessor() might be made.
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| - void SetAudioProcessor(scoped_refptr<MediaStreamAudioProcessor> processor);
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| -
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| - // webrtc::MediaStreamTrack implementation.
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| - std::string kind() const override;
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| - bool set_enabled(bool enable) override;
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| -
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| - private:
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| - // webrtc::AudioTrackInterface implementation.
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| - void AddSink(webrtc::AudioTrackSinkInterface* sink) override;
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| - void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override;
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| - bool GetSignalLevel(int* level) override;
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| - rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor()
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| - override;
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| - webrtc::AudioSourceInterface* GetSource() const override;
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| -
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| - // Weak reference.
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| - WebRtcLocalAudioTrack* owner_;
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| -
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| - // The source of the audio track which handles the audio constraints.
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| - // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack.
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| - rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
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| -
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| - // Libjingle's signaling thread.
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| - const scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner_;
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| -
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| - // The audio processsor that applies audio processing on the data of audio
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| - // track. This must be set before calls to GetAudioProcessor() are made.
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| - scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
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| -
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| - // A vector of the peer connection sink adapters which receive the audio data
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| - // from the audio track.
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| - ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_;
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| -
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| - // Thread-safe accessor to current audio signal level. This must be set
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| - // before calls to GetSignalLevel() are made.
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| - scoped_refptr<MediaStreamAudioLevelCalculator::Level> level_;
|
| -};
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| -
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| -} // namespace content
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| -
|
| -#endif // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
|
|
|