FFmpeg  4.3
Macros | Enumerations | Functions
rtp.h File Reference
#include "libavformat/avformat.h"
#include "libavcodec/avcodec.h"
#include "libavutil/mathematics.h"

Go to the source code of this file.

Macros

#define RTP_PT_PRIVATE   96
 
#define RTP_VERSION   2
 
#define RTP_MAX_SDES   256
 maximum text length for SDES More...
 
#define RTCP_TX_RATIO_NUM   5
 
#define RTCP_TX_RATIO_DEN   1000
 
#define RTP_XIPH_IDENT   0xfecdba
 
#define RTP_PT_IS_RTCP(x)
 
#define NTP_TO_RTP_FORMAT(x)   av_rescale((x), INT64_C(1) << 32, 1000000)
 

Enumerations

enum  RTCPType {
  RTCP_FIR = 192, RTCP_NACK, RTCP_SMPTETC, RTCP_IJ,
  RTCP_SR = 200, RTCP_RR, RTCP_SDES, RTCP_BYE,
  RTCP_APP, RTCP_RTPFB, RTCP_PSFB, RTCP_XR,
  RTCP_AVB, RTCP_RSI, RTCP_TOKEN
}
 

Functions

int ff_rtp_get_payload_type (AVFormatContext *fmt, AVCodecParameters *par, int idx)
 Return the payload type for a given stream used in the given format context. More...
 
int ff_rtp_get_codec_info (AVCodecParameters *par, int payload_type)
 Initialize a codec context based on the payload type. More...
 
const char * ff_rtp_enc_name (int payload_type)
 Return the encoding name (as defined in http://www.iana.org/assignments/rtp-parameters) for a given payload type. More...
 
enum AVCodecID ff_rtp_codec_id (const char *buf, enum AVMediaType codec_type)
 Return the codec id for the given encoding name and codec type. More...
 

Macro Definition Documentation

◆ RTP_PT_PRIVATE

#define RTP_PT_PRIVATE   96

Definition at line 77 of file rtp.h.

◆ RTP_VERSION

#define RTP_VERSION   2

Definition at line 78 of file rtp.h.

◆ RTP_MAX_SDES

#define RTP_MAX_SDES   256

maximum text length for SDES

Definition at line 79 of file rtp.h.

◆ RTCP_TX_RATIO_NUM

#define RTCP_TX_RATIO_NUM   5

Definition at line 82 of file rtp.h.

◆ RTCP_TX_RATIO_DEN

#define RTCP_TX_RATIO_DEN   1000

Definition at line 83 of file rtp.h.

◆ RTP_XIPH_IDENT

#define RTP_XIPH_IDENT   0xfecdba

Definition at line 89 of file rtp.h.

◆ RTP_PT_IS_RTCP

#define RTP_PT_IS_RTCP (   x)
Value:
(((x) >= RTCP_FIR && (x) <= RTCP_IJ) || \
((x) >= RTCP_SR && (x) <= RTCP_TOKEN))

Definition at line 110 of file rtp.h.

◆ NTP_TO_RTP_FORMAT

#define NTP_TO_RTP_FORMAT (   x)    av_rescale((x), INT64_C(1) << 32, 1000000)

Definition at line 113 of file rtp.h.

Enumeration Type Documentation

◆ RTCPType

enum RTCPType
Enumerator
RTCP_FIR 
RTCP_NACK 
RTCP_SMPTETC 
RTCP_IJ 
RTCP_SR 
RTCP_RR 
RTCP_SDES 
RTCP_BYE 
RTCP_APP 
RTCP_RTPFB 
RTCP_PSFB 
RTCP_XR 
RTCP_AVB 
RTCP_RSI 
RTCP_TOKEN 

Definition at line 92 of file rtp.h.

Function Documentation

◆ ff_rtp_get_payload_type()

int ff_rtp_get_payload_type ( AVFormatContext fmt,
AVCodecParameters par,
int  idx 
)

Return the payload type for a given stream used in the given format context.

Static payload types are derived from the codec. Dynamic payload type are derived from the id field in AVStream. The format context private option payload_type overrides both.

Parameters
fmtThe context of the format
parThe codec parameters
idxThe stream index
Returns
The payload type (the 'PT' field in the RTP header).

Definition at line 90 of file rtp.c.

Referenced by ff_rtp_chain_mux_open(), and rtp_write_header().

◆ ff_rtp_get_codec_info()

int ff_rtp_get_codec_info ( AVCodecParameters par,
int  payload_type 
)

Initialize a codec context based on the payload type.

Fill the codec_type and codec_id fields of a codec context with information depending on the payload type; for audio codecs, the channels and sample_rate fields are also filled.

Parameters
parThe codec parameters
payload_typeThe payload type (the 'PT' field in the RTP header)
Returns
In case of unknown payload type or dynamic payload type, a negative value is returned; otherwise, 0 is returned

Definition at line 71 of file rtp.c.

◆ ff_rtp_enc_name()

const char* ff_rtp_enc_name ( int  payload_type)

Return the encoding name (as defined in http://www.iana.org/assignments/rtp-parameters) for a given payload type.

Parameters
payload_typeThe payload type (the 'PT' field in the RTP header)
Returns
In case of unknown payload type or dynamic payload type, a pointer to an empty string is returned; otherwise, a pointer to a string containing the encoding name is returned

Definition at line 132 of file rtp.c.

◆ ff_rtp_codec_id()

enum AVCodecID ff_rtp_codec_id ( const char *  buf,
enum AVMediaType  codec_type 
)

Return the codec id for the given encoding name and codec type.

Parameters
bufA pointer to the string containing the encoding name
codec_typeThe codec type
Returns
In case of unknown encoding name, AV_CODEC_ID_NONE is returned; otherwise, the codec id is returned

Definition at line 143 of file rtp.c.

RTCP_IJ
@ RTCP_IJ
Definition: rtp.h:96
x
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new please make sure they are as small as space on each network bandwidth and so on benefit from smaller test cases Also keep in mind older checkouts use existing sample that means in practice generally do not remove or overwrite files as it likely would break older checkouts or releases Also all needed samples for a commit should be ideally before the push If you need an account for frequently uploading samples or you wish to help others by doing that send a mail to ffmpeg devel rsync vauL Duo x
Definition: fate.txt:150
RTCP_TOKEN
@ RTCP_TOKEN
Definition: rtp.h:107
RTCP_FIR
@ RTCP_FIR
Definition: rtp.h:93
RTCP_SR
@ RTCP_SR
Definition: rtp.h:97