webkit  2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
Functions
rtp_analyze.cc File Reference
#include <assert.h>
#include <stdio.h>
#include <memory>
#include <vector>
#include "gflags/gflags.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"

Functions

 DEFINE_int32 (red, 117, "RTP payload type for RED")
 
 DEFINE_int32 (audio_level, 1, "Extension ID for audio level (RFC 6464)")
 
 DEFINE_int32 (abs_send_time, 3, "Extension ID for absolute sender time")
 
int main (int argc, char *argv[])
 

Function Documentation

◆ DEFINE_int32() [1/3]

DEFINE_int32 ( red  ,
117  ,
"RTP payload type for RED"   
)

◆ DEFINE_int32() [2/3]

DEFINE_int32 ( audio_level  ,
,
"Extension ID for audio level (RFC 6464)"   
)

◆ DEFINE_int32() [3/3]

DEFINE_int32 ( abs_send_time  ,
,
"Extension ID for absolute sender time  
)

◆ main()

int main ( int  argc,
char *  argv[] 
)