| Index: content/renderer/media/webrtc/peer_connection_remote_audio_source.h
|
| diff --git a/content/renderer/media/webrtc/peer_connection_remote_audio_source.h b/content/renderer/media/webrtc/peer_connection_remote_audio_source.h
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| deleted file mode 100644
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| index aa4f15d5b0e472d5d671b0352e7d50975a9794da..0000000000000000000000000000000000000000
|
| --- a/content/renderer/media/webrtc/peer_connection_remote_audio_source.h
|
| +++ /dev/null
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| @@ -1,102 +0,0 @@
|
| -// 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_PEER_CONNECTION_REMOTE_AUDIO_SOURCE_H_
|
| -#define CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_REMOTE_AUDIO_SOURCE_H_
|
| -
|
| -#include <memory>
|
| -
|
| -#include "base/memory/ref_counted.h"
|
| -#include "base/synchronization/lock.h"
|
| -#include "content/renderer/media/media_stream_audio_source.h"
|
| -#include "content/renderer/media/media_stream_audio_track.h"
|
| -#include "third_party/webrtc/api/mediastreaminterface.h"
|
| -
|
| -namespace media {
|
| -class AudioBus;
|
| -}
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| -
|
| -namespace content {
|
| -
|
| -// PeerConnectionRemoteAudioTrack is a WebRTC specific implementation of an
|
| -// audio track whose data is sourced from a PeerConnection.
|
| -class PeerConnectionRemoteAudioTrack final
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| - : NON_EXPORTED_BASE(public MediaStreamAudioTrack) {
|
| - public:
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| - explicit PeerConnectionRemoteAudioTrack(
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| - scoped_refptr<webrtc::AudioTrackInterface> track_interface);
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| - ~PeerConnectionRemoteAudioTrack() final;
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| -
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| - // If |track| is an instance of PeerConnectionRemoteAudioTrack, return a
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| - // type-casted pointer to it. Otherwise, return null.
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| - static PeerConnectionRemoteAudioTrack* From(MediaStreamAudioTrack* track);
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| -
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| - webrtc::AudioTrackInterface* track_interface() const {
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| - return track_interface_.get();
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| - }
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| -
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| - // MediaStreamAudioTrack override.
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| - void SetEnabled(bool enabled) override;
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| -
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| - private:
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| - // MediaStreamAudioTrack overrides.
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| - void* GetClassIdentifier() const final;
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| -
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| - const scoped_refptr<webrtc::AudioTrackInterface> track_interface_;
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| -
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| - // In debug builds, check that all methods that could cause object graph
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| - // or data flow changes are being called on the main thread.
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| - base::ThreadChecker thread_checker_;
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| -
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| - DISALLOW_COPY_AND_ASSIGN(PeerConnectionRemoteAudioTrack);
|
| -};
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| -
|
| -// Represents the audio provided by the receiving end of a PeerConnection.
|
| -class PeerConnectionRemoteAudioSource final
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| - : NON_EXPORTED_BASE(public MediaStreamAudioSource),
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| - NON_EXPORTED_BASE(protected webrtc::AudioTrackSinkInterface) {
|
| - public:
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| - explicit PeerConnectionRemoteAudioSource(
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| - scoped_refptr<webrtc::AudioTrackInterface> track_interface);
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| - ~PeerConnectionRemoteAudioSource() final;
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| -
|
| - protected:
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| - // MediaStreamAudioSource implementation.
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| - std::unique_ptr<MediaStreamAudioTrack> CreateMediaStreamAudioTrack(
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| - const std::string& id) final;
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| - bool EnsureSourceIsStarted() final;
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| - void EnsureSourceIsStopped() final;
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| -
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| - // webrtc::AudioTrackSinkInterface implementation.
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| - void OnData(const void* audio_data, int bits_per_sample, int sample_rate,
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| - size_t number_of_channels, size_t number_of_frames) final;
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| -
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| - private:
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| - // Interface to the implementation that calls OnData().
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| - const scoped_refptr<webrtc::AudioTrackInterface> track_interface_;
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| -
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| - // In debug builds, check that all methods that could cause object graph
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| - // or data flow changes are being called on the main thread.
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| - base::ThreadChecker thread_checker_;
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| -
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| - // True if |this| is receiving an audio flow as a sink of the remote
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| - // PeerConnection via |track_interface_|.
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| - bool is_sink_of_peer_connection_;
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| -
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| - // Buffer for converting from interleaved signed-integer PCM samples to the
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| - // planar float format. Only used on the thread that calls OnData().
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| - std::unique_ptr<media::AudioBus> audio_bus_;
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| -
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| - // In debug builds, use a "try lock" to sanity-check that there are no
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| - // concurrent calls to OnData(). See notes in OnData() implementation.
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| -#ifndef NDEBUG
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| - base::Lock single_audio_thread_guard_;
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| -#endif
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| -
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| - DISALLOW_COPY_AND_ASSIGN(PeerConnectionRemoteAudioSource);
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| -};
|
| -
|
| -} // namespace content
|
| -
|
| -#endif // CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_REMOTE_AUDIO_SOURCE_H_
|
|
|