webkit  2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
Public Types | Public Member Functions | Static Public Member Functions | List of all members
webrtc::test::RtpStream Class Reference

#include <delay_based_bwe_unittest_helper.h>

Public Types

enum  { kSendSideOffsetUs = 1000000 }
 
enum  { kSendSideOffsetUs = 1000000 }
 

Public Member Functions

 RtpStream (int fps, int bitrate_bps)
 
int64_t GenerateFrame (int64_t time_now_us, std::vector< PacketInfo > *packets)
 
int64_t next_rtp_time () const
 
void set_bitrate_bps (int bitrate_bps)
 
int bitrate_bps () const
 
 RtpStream (int fps, int bitrate_bps)
 
int64_t GenerateFrame (int64_t time_now_us, std::vector< PacketInfo > *packets)
 
int64_t next_rtp_time () const
 
void set_bitrate_bps (int bitrate_bps)
 
int bitrate_bps () const
 

Static Public Member Functions

static bool Compare (const std::unique_ptr< RtpStream > &lhs, const std::unique_ptr< RtpStream > &rhs)
 
static bool Compare (const std::unique_ptr< RtpStream > &lhs, const std::unique_ptr< RtpStream > &rhs)
 

Member Enumeration Documentation

◆ anonymous enum

anonymous enum
Enumerator
kSendSideOffsetUs 

◆ anonymous enum

anonymous enum
Enumerator
kSendSideOffsetUs 

Constructor & Destructor Documentation

◆ RtpStream() [1/2]

webrtc::test::RtpStream::RtpStream ( int  fps,
int  bitrate_bps 
)

◆ RtpStream() [2/2]

webrtc::test::RtpStream::RtpStream ( int  fps,
int  bitrate_bps 
)

Member Function Documentation

◆ bitrate_bps() [1/2]

int webrtc::test::RtpStream::bitrate_bps ( ) const

◆ bitrate_bps() [2/2]

int webrtc::test::RtpStream::bitrate_bps ( ) const

◆ Compare() [1/2]

bool webrtc::test::RtpStream::Compare ( const std::unique_ptr< RtpStream > &  lhs,
const std::unique_ptr< RtpStream > &  rhs 
)
static

◆ Compare() [2/2]

static bool webrtc::test::RtpStream::Compare ( const std::unique_ptr< RtpStream > &  lhs,
const std::unique_ptr< RtpStream > &  rhs 
)
static

◆ GenerateFrame() [1/2]

int64_t webrtc::test::RtpStream::GenerateFrame ( int64_t  time_now_us,
std::vector< PacketInfo > *  packets 
)

◆ GenerateFrame() [2/2]

int64_t webrtc::test::RtpStream::GenerateFrame ( int64_t  time_now_us,
std::vector< PacketInfo > *  packets 
)

◆ next_rtp_time() [1/2]

int64_t webrtc::test::RtpStream::next_rtp_time ( ) const

◆ next_rtp_time() [2/2]

int64_t webrtc::test::RtpStream::next_rtp_time ( ) const

◆ set_bitrate_bps() [1/2]

void webrtc::test::RtpStream::set_bitrate_bps ( int  bitrate_bps)

◆ set_bitrate_bps() [2/2]

void webrtc::test::RtpStream::set_bitrate_bps ( int  bitrate_bps)

The documentation for this class was generated from the following files: