ddfdfed3b5
- Change how the transmission offset is calculated, to incorporate the time since the frame was captured. - Break out RtpRtcpClock and move it to system_wrappers. - Use RtpRtcpClock to set the capture time in ms in the capture module. We must use the same clock as in the RTP module to be able to measure the time from capture until transmission. - Enables the RTP header extension for packet transmission time offsets. BUG= TEST=trybots Review URL: https://webrtc-codereview.appspot.com/666006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2489 4adac7df-926f-26a2-2b94-8c16560cd09d
111 lines
3.7 KiB
C++
111 lines
3.7 KiB
C++
/*
|
|
* 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_GENERIC_CODEC_TEST_H_
|
|
#define WEBRTC_MODULES_VIDEO_CODING_TEST_GENERIC_CODEC_TEST_H_
|
|
|
|
#include "video_coding.h"
|
|
|
|
#include <string.h>
|
|
#include <fstream>
|
|
|
|
#include "test_callbacks.h"
|
|
#include "test_util.h"
|
|
/*
|
|
Test consists of:
|
|
1. Sanity checks
|
|
2. Bit rate validation
|
|
3. Encoder control test / General API functionality
|
|
4. Decoder control test / General API functionality
|
|
|
|
*/
|
|
|
|
namespace webrtc {
|
|
|
|
int VCMGenericCodecTest(CmdArgs& args);
|
|
|
|
class FakeTickTime;
|
|
|
|
class GenericCodecTest
|
|
{
|
|
public:
|
|
GenericCodecTest(webrtc::VideoCodingModule* vcm,
|
|
webrtc::FakeTickTime* clock);
|
|
~GenericCodecTest();
|
|
static int RunTest(CmdArgs& args);
|
|
WebRtc_Word32 Perform(CmdArgs& args);
|
|
float WaitForEncodedFrame() const;
|
|
|
|
private:
|
|
void Setup(CmdArgs& args);
|
|
void Print();
|
|
WebRtc_Word32 TearDown();
|
|
void IncrementDebugClock(float frameRate);
|
|
|
|
webrtc::FakeTickTime* _clock;
|
|
webrtc::VideoCodingModule* _vcm;
|
|
webrtc::VideoCodec _sendCodec;
|
|
webrtc::VideoCodec _receiveCodec;
|
|
std::string _inname;
|
|
std::string _outname;
|
|
std::string _encodedName;
|
|
WebRtc_Word32 _sumEncBytes;
|
|
FILE* _sourceFile;
|
|
FILE* _decodedFile;
|
|
FILE* _encodedFile;
|
|
WebRtc_UWord16 _width;
|
|
WebRtc_UWord16 _height;
|
|
float _frameRate;
|
|
WebRtc_UWord32 _lengthSourceFrame;
|
|
WebRtc_UWord32 _timeStamp;
|
|
VCMDecodeCompleteCallback* _decodeCallback;
|
|
VCMEncodeCompleteCallback* _encodeCompleteCallback;
|
|
|
|
}; // end of GenericCodecTest class definition
|
|
|
|
class RTPSendCallback_SizeTest : public webrtc::Transport
|
|
{
|
|
public:
|
|
// constructor input: (receive side) rtp module to send encoded data to
|
|
RTPSendCallback_SizeTest() : _maxPayloadSize(0), _payloadSizeSum(0), _nPackets(0) {}
|
|
virtual int SendPacket(int channel, const void *data, int len);
|
|
virtual int SendRTCPPacket(int channel, const void *data, int len) {return 0;}
|
|
void SetMaxPayloadSize(WebRtc_UWord32 maxPayloadSize);
|
|
void Reset();
|
|
float AveragePayloadSize() const;
|
|
private:
|
|
WebRtc_UWord32 _maxPayloadSize;
|
|
WebRtc_UWord32 _payloadSizeSum;
|
|
WebRtc_UWord32 _nPackets;
|
|
};
|
|
|
|
class VCMEncComplete_KeyReqTest : public webrtc::VCMPacketizationCallback
|
|
{
|
|
public:
|
|
VCMEncComplete_KeyReqTest(webrtc::VideoCodingModule &vcm) : _vcm(vcm), _seqNo(0), _timeStamp(0) {}
|
|
WebRtc_Word32 SendData(
|
|
const webrtc::FrameType frameType,
|
|
const WebRtc_UWord8 payloadType,
|
|
WebRtc_UWord32 timeStamp,
|
|
int64_t capture_time_ms,
|
|
const WebRtc_UWord8* payloadData,
|
|
const WebRtc_UWord32 payloadSize,
|
|
const webrtc::RTPFragmentationHeader& fragmentationHeader,
|
|
const webrtc::RTPVideoHeader* videoHdr);
|
|
private:
|
|
webrtc::VideoCodingModule& _vcm;
|
|
WebRtc_UWord16 _seqNo;
|
|
WebRtc_UWord32 _timeStamp;
|
|
}; // end of VCMEncodeCompleteCallback
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_GENERIC_CODEC_TEST_H_
|