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}
62 lines
2.2 KiB
C++
62 lines
2.2 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_SPATIAL_RESAMPLER_H
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#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_SPATIAL_RESAMPLER_H
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#include "webrtc/typedefs.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/common_video/libyuv/include/scaler.h"
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#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
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namespace webrtc {
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class VPMSpatialResampler {
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public:
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virtual ~VPMSpatialResampler() {};
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virtual int32_t SetTargetFrameSize(int32_t width, int32_t height) = 0;
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virtual void SetInputFrameResampleMode(VideoFrameResampling
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resampling_mode) = 0;
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virtual void Reset() = 0;
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virtual int32_t ResampleFrame(const VideoFrame& inFrame,
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VideoFrame* outFrame) = 0;
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virtual int32_t TargetWidth() = 0;
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virtual int32_t TargetHeight() = 0;
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virtual bool ApplyResample(int32_t width, int32_t height) = 0;
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};
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class VPMSimpleSpatialResampler : public VPMSpatialResampler {
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public:
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VPMSimpleSpatialResampler();
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~VPMSimpleSpatialResampler();
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virtual int32_t SetTargetFrameSize(int32_t width, int32_t height);
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virtual void SetInputFrameResampleMode(VideoFrameResampling resampling_mode);
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virtual void Reset();
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virtual int32_t ResampleFrame(const VideoFrame& inFrame,
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VideoFrame* outFrame);
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virtual int32_t TargetWidth();
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virtual int32_t TargetHeight();
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virtual bool ApplyResample(int32_t width, int32_t height);
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private:
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VideoFrameResampling resampling_mode_;
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int32_t target_width_;
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int32_t target_height_;
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Scaler scaler_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_SPATIAL_RESAMPLER_H
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