ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
88 lines
2.9 KiB
C++
88 lines
2.9 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_CONTENT_ANALYSIS_H
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#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_CONTENT_ANALYSIS_H
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/modules/video_processing/main/interface/video_processing_defines.h"
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#include "webrtc/typedefs.h"
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#include "webrtc/video_frame.h"
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namespace webrtc {
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class VPMContentAnalysis {
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public:
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// When |runtime_cpu_detection| is true, runtime selection of an optimized
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// code path is allowed.
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explicit VPMContentAnalysis(bool runtime_cpu_detection);
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~VPMContentAnalysis();
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// Initialize ContentAnalysis - should be called prior to
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// extractContentFeature
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// Inputs: width, height
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// Return value: 0 if OK, negative value upon error
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int32_t Initialize(int width, int height);
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// Extract content Feature - main function of ContentAnalysis
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// Input: new frame
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// Return value: pointer to structure containing content Analysis
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// metrics or NULL value upon error
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VideoContentMetrics* ComputeContentMetrics(const VideoFrame& inputFrame);
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// Release all allocated memory
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// Output: 0 if OK, negative value upon error
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int32_t Release();
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private:
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// return motion metrics
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VideoContentMetrics* ContentMetrics();
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// Normalized temporal difference metric: for motion magnitude
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typedef int32_t (VPMContentAnalysis::*TemporalDiffMetricFunc)();
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TemporalDiffMetricFunc TemporalDiffMetric;
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int32_t TemporalDiffMetric_C();
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// Motion metric method: call 2 metrics (magnitude and size)
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int32_t ComputeMotionMetrics();
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// Spatial metric method: computes the 3 frame-average spatial
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// prediction errors (1x2,2x1,2x2)
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typedef int32_t (VPMContentAnalysis::*ComputeSpatialMetricsFunc)();
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ComputeSpatialMetricsFunc ComputeSpatialMetrics;
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int32_t ComputeSpatialMetrics_C();
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#if defined(WEBRTC_ARCH_X86_FAMILY)
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int32_t ComputeSpatialMetrics_SSE2();
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int32_t TemporalDiffMetric_SSE2();
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#endif
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const uint8_t* orig_frame_;
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uint8_t* prev_frame_;
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int width_;
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int height_;
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int skip_num_;
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int border_;
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// Content Metrics: Stores the local average of the metrics.
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float motion_magnitude_; // motion class
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float spatial_pred_err_; // spatial class
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float spatial_pred_err_h_; // spatial class
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float spatial_pred_err_v_; // spatial class
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bool first_frame_;
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bool ca_Init_;
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VideoContentMetrics* content_metrics_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_CONTENT_ANALYSIS_H
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