blob: 47c17ec2e7792cb90267e69b6ff3801915601974 [file] [log] [blame]
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <memory>
#include "webrtc/modules/video_processing/include/video_processing.h"
#include "webrtc/modules/video_processing/spatial_resampler.h"
#include "webrtc/modules/video_processing/video_decimator.h"
#include "webrtc/typedefs.h"
#include "webrtc/video_frame.h"
namespace webrtc {
class VideoDenoiser;
// All pointers/members in this class are assumed to be protected by the class
// owner.
class VPMFramePreprocessor {
void Reset();
// Enable temporal decimation.
void EnableTemporalDecimation(bool enable);
void SetInputFrameResampleMode(VideoFrameResampling resampling_mode);
// Set target resolution: frame rate and dimension.
int32_t SetTargetResolution(uint32_t width,
uint32_t height,
uint32_t frame_rate);
// Update incoming frame rate/dimension.
void UpdateIncomingframe_rate();
int32_t updateIncomingFrameSize(uint32_t width, uint32_t height);
// Set decimated values: frame rate/dimension.
uint32_t GetDecimatedFrameRate();
uint32_t GetDecimatedWidth() const;
uint32_t GetDecimatedHeight() const;
// Preprocess output:
void EnableDenoising(bool enable);
const VideoFrame* PreprocessFrame(const VideoFrame& frame);
// The content does not change so much every frame, so to reduce complexity
// we can compute new content metrics every |kSkipFrameCA| frames.
enum { kSkipFrameCA = 2 };
rtc::scoped_refptr<I420Buffer> denoised_buffer_[2];
VideoFrame denoised_frame_;
VideoFrame resampled_frame_;
VPMSpatialResampler* spatial_resampler_;
VPMVideoDecimator* vd_;
std::unique_ptr<VideoDenoiser> denoiser_;
uint8_t denoised_frame_toggle_;
uint32_t frame_cnt_;
} // namespace webrtc