webkit  2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
Classes | Namespaces | Macros | Enumerations | Functions
rtp_player.cc File Reference
#include "webrtc/modules/video_coding/test/rtp_player.h"
#include <stdio.h>
#include <cstdlib>
#include <map>
#include <memory>
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/video_coding/internal_defines.h"
#include "webrtc/modules/video_coding/test/test_util.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/test/rtp_file_reader.h"

Classes

class  webrtc::rtpplayer::RawRtpPacket
 
class  webrtc::rtpplayer::LostPackets
 
class  webrtc::rtpplayer::SsrcHandlers
 
class  webrtc::rtpplayer::RtpPlayerImpl
 

Namespaces

 webrtc
 
 webrtc::rtpplayer
 

Macros

#define DEBUG_LOG1(text, arg)
 

Enumerations

enum  { webrtc::rtpplayer::kMaxPacketBufferSize = 4096, webrtc::rtpplayer::kDefaultTransmissionTimeOffsetExtensionId = 2 }
 

Functions

RtpPlayerInterface * webrtc::rtpplayer::Create (const std::string &inputFilename, PayloadSinkFactoryInterface *payloadSinkFactory, Clock *clock, const PayloadTypes &payload_types, float lossRate, int64_t rttMs, bool reordering)
 

Macro Definition Documentation

◆ DEBUG_LOG1

#define DEBUG_LOG1 (   text,
  arg 
)