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// Copyright 2018 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_VIDEO_TRACK_SOURCE_H_
#define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_VIDEO_TRACK_SOURCE_H_
#include "base/memory/scoped_refptr.h"
#include "base/threading/thread_checker.h"
#include "content/common/content_export.h"
#include "media/base/video_frame_pool.h"
#include "third_party/webrtc/media/base/adaptedvideotracksource.h"
#include "third_party/webrtc/rtc_base/timestampaligner.h"
namespace content {
class CONTENT_EXPORT WebRtcVideoTrackSource
: public rtc::AdaptedVideoTrackSource {
public:
WebRtcVideoTrackSource(bool is_screencast,
absl::optional<bool> needs_denoising);
~WebRtcVideoTrackSource() override;
SourceState state() const override;
bool remote() const override;
bool is_screencast() const override;
absl::optional<bool> needs_denoising() const override;
void OnFrameCaptured(const scoped_refptr<media::VideoFrame>& frame);
using webrtc::VideoTrackSourceInterface::AddOrUpdateSink;
using webrtc::VideoTrackSourceInterface::RemoveSink;
private:
// |thread_checker_| is bound to the libjingle worker thread.
THREAD_CHECKER(thread_checker_);
media::VideoFramePool scaled_frame_pool_;
// State for the timestamp translation.
rtc::TimestampAligner timestamp_aligner_;
const bool is_screencast_;
const absl::optional<bool> needs_denoising_;
DISALLOW_COPY_AND_ASSIGN(WebRtcVideoTrackSource);
};
} // namespace content
#endif // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_VIDEO_TRACK_SOURCE_H_