#include <rtp_generator.h>
|
| RtpGenerator (int samples_per_ms, uint16_t start_seq_number=0, uint32_t start_timestamp=0, uint32_t start_send_time_ms=0, uint32_t ssrc=0x12345678) |
|
virtual | ~RtpGenerator () |
|
virtual uint32_t | GetRtpHeader (uint8_t payload_type, size_t payload_length_samples, WebRtcRTPHeader *rtp_header) |
|
void | set_drift_factor (double factor) |
|
| RtpGenerator (int samples_per_ms, uint16_t start_seq_number=0, uint32_t start_timestamp=0, uint32_t start_send_time_ms=0, uint32_t ssrc=0x12345678) |
|
virtual | ~RtpGenerator () |
|
virtual uint32_t | GetRtpHeader (uint8_t payload_type, size_t payload_length_samples, WebRtcRTPHeader *rtp_header) |
|
void | set_drift_factor (double factor) |
|
◆ RtpGenerator() [1/2]
webrtc::test::RtpGenerator::RtpGenerator |
( |
int |
samples_per_ms, |
|
|
uint16_t |
start_seq_number = 0 , |
|
|
uint32_t |
start_timestamp = 0 , |
|
|
uint32_t |
start_send_time_ms = 0 , |
|
|
uint32_t |
ssrc = 0x12345678 |
|
) |
| |
|
inline |
◆ ~RtpGenerator() [1/2]
virtual webrtc::test::RtpGenerator::~RtpGenerator |
( |
| ) |
|
|
inlinevirtual |
◆ RtpGenerator() [2/2]
webrtc::test::RtpGenerator::RtpGenerator |
( |
int |
samples_per_ms, |
|
|
uint16_t |
start_seq_number = 0 , |
|
|
uint32_t |
start_timestamp = 0 , |
|
|
uint32_t |
start_send_time_ms = 0 , |
|
|
uint32_t |
ssrc = 0x12345678 |
|
) |
| |
|
inline |
◆ ~RtpGenerator() [2/2]
virtual webrtc::test::RtpGenerator::~RtpGenerator |
( |
| ) |
|
|
inlinevirtual |
◆ GetRtpHeader() [1/2]
◆ GetRtpHeader() [2/2]
◆ set_drift_factor() [1/2]
void webrtc::test::RtpGenerator::set_drift_factor |
( |
double |
factor | ) |
|
◆ set_drift_factor() [2/2]
void webrtc::test::RtpGenerator::set_drift_factor |
( |
double |
factor | ) |
|
◆ drift_factor_
double webrtc::test::RtpGenerator::drift_factor_ |
|
protected |
◆ next_send_time_ms_
uint32_t webrtc::test::RtpGenerator::next_send_time_ms_ |
|
protected |
◆ samples_per_ms_
const int webrtc::test::RtpGenerator::samples_per_ms_ |
|
protected |
◆ seq_number_
uint16_t webrtc::test::RtpGenerator::seq_number_ |
|
protected |
◆ ssrc_
const uint32_t webrtc::test::RtpGenerator::ssrc_ |
|
protected |
◆ timestamp_
uint32_t webrtc::test::RtpGenerator::timestamp_ |
|
protected |
The documentation for this class was generated from the following files:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_generator.cc