modules/video_coding refactorings
The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc: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 --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
This commit is contained in:
86
webrtc/modules/video_coding/test/test_util.h
Normal file
86
webrtc/modules/video_coding/test/test_util.h
Normal file
@ -0,0 +1,86 @@
|
||||
/*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
|
||||
#define WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
|
||||
|
||||
/*
|
||||
* General declarations used through out VCM offline tests.
|
||||
*/
|
||||
|
||||
#include <string>
|
||||
|
||||
#include "webrtc/base/constructormagic.h"
|
||||
#include "webrtc/modules/include/module_common_types.h"
|
||||
#include "webrtc/modules/video_coding/include/video_coding.h"
|
||||
#include "webrtc/system_wrappers/include/event_wrapper.h"
|
||||
|
||||
enum { kMaxNackListSize = 250 };
|
||||
enum { kMaxPacketAgeToNack = 450 };
|
||||
|
||||
class NullEvent : public webrtc::EventWrapper {
|
||||
public:
|
||||
virtual ~NullEvent() {}
|
||||
|
||||
virtual bool Set() { return true; }
|
||||
|
||||
virtual bool Reset() { return true; }
|
||||
|
||||
virtual webrtc::EventTypeWrapper Wait(unsigned long max_time) {
|
||||
return webrtc::kEventTimeout;
|
||||
}
|
||||
|
||||
virtual bool StartTimer(bool periodic, unsigned long time) { return true; }
|
||||
|
||||
virtual bool StopTimer() { return true; }
|
||||
};
|
||||
|
||||
class NullEventFactory : public webrtc::EventFactory {
|
||||
public:
|
||||
virtual ~NullEventFactory() {}
|
||||
|
||||
virtual webrtc::EventWrapper* CreateEvent() {
|
||||
return new NullEvent;
|
||||
}
|
||||
};
|
||||
|
||||
class FileOutputFrameReceiver : public webrtc::VCMReceiveCallback {
|
||||
public:
|
||||
FileOutputFrameReceiver(const std::string& base_out_filename, uint32_t ssrc);
|
||||
virtual ~FileOutputFrameReceiver();
|
||||
|
||||
// VCMReceiveCallback
|
||||
virtual int32_t FrameToRender(webrtc::VideoFrame& video_frame);
|
||||
|
||||
private:
|
||||
std::string out_filename_;
|
||||
FILE* out_file_;
|
||||
FILE* timing_file_;
|
||||
int width_;
|
||||
int height_;
|
||||
int count_;
|
||||
|
||||
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FileOutputFrameReceiver);
|
||||
};
|
||||
|
||||
class CmdArgs {
|
||||
public:
|
||||
CmdArgs();
|
||||
|
||||
std::string codecName;
|
||||
webrtc::VideoCodecType codecType;
|
||||
int width;
|
||||
int height;
|
||||
int rtt;
|
||||
std::string inputFile;
|
||||
std::string outputFile;
|
||||
};
|
||||
|
||||
#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
|
||||
Reference in New Issue
Block a user