GstRTPBuffer

The GstRTPBuffer helper functions makes it easy to parse and create regular GstBuffer objects that contain RTP payloads. These buffers are typically of 'application/x-rtp' GstCaps.

GstRTPBuffer

Data structure that points to an RTP packet. The size of the structure is made public to allow stack allocations.

Members

buffer (GstBuffer *) –

pointer to RTP buffer

state (guint) –

internal state

data (gpointer *) –

array of data

size (gsize *) –

array of size

map (GstMapInfo *) –

array of GstMapInfo


GstRTPBuffer

Data structure that points to an RTP packet. The size of the structure is made public to allow stack allocations.

Members

buffer (GstBuffer) –

pointer to RTP buffer

state (Number) –

internal state

data ([ Object ]) –

array of data

size ([ Number ]) –

array of size

map ([ GstMapInfo ]) –

array of GstMapInfo


GstRTPBuffer

Data structure that points to an RTP packet. The size of the structure is made public to allow stack allocations.

Members

buffer (GstBuffer) –

pointer to RTP buffer

state (int) –

internal state

data ([ object ]) –

array of data

size ([ int ]) –

array of size

map ([ GstMapInfo ]) –

array of GstMapInfo


Methods

gst_rtp_buffer_add_extension_onebyte_header

gboolean
gst_rtp_buffer_add_extension_onebyte_header (GstRTPBuffer * rtp,
                                             guint8 id,
                                             gconstpointer data,
                                             guint size)

Adds a RFC 5285 header extension with a one byte header to the end of the RTP header. If there is already a RFC 5285 header extension with a one byte header, the new extension will be appended. It will not work if there is already a header extension that does not follow the mechanism described in RFC 5285 or if there is a header extension with a two bytes header as described in RFC 5285. In that case, use gst_rtp_buffer_add_extension_twobytes_header

Parameters:

rtp

the RTP packet

id

The ID of the header extension (between 1 and 14).

data ( [array length=size] [element-type guint8] ) –

location for data

size

the size of the data in bytes

Returns

TRUE if header extension could be added


GstRtp.RTPBuffer.prototype.add_extension_onebyte_header

function GstRtp.RTPBuffer.prototype.add_extension_onebyte_header(id: Number, data: [ Number ], size: Number): {
    // javascript wrapper for 'gst_rtp_buffer_add_extension_onebyte_header'
}

Adds a RFC 5285 header extension with a one byte header to the end of the RTP header. If there is already a RFC 5285 header extension with a one byte header, the new extension will be appended. It will not work if there is already a header extension that does not follow the mechanism described in RFC 5285 or if there is a header extension with a two bytes header as described in RFC 5285. In that case, use GstRtp.RTPBuffer.prototype.add_extension_twobytes_header

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

id ( Number ) –

The ID of the header extension (between 1 and 14).

data ( [ Number ] ) –

location for data

size ( Number ) –

the size of the data in bytes

Returns ( Number ) –

true if header extension could be added


GstRtp.RTPBuffer.add_extension_onebyte_header

def GstRtp.RTPBuffer.add_extension_onebyte_header (self, id, data, size):
    #python wrapper for 'gst_rtp_buffer_add_extension_onebyte_header'

Adds a RFC 5285 header extension with a one byte header to the end of the RTP header. If there is already a RFC 5285 header extension with a one byte header, the new extension will be appended. It will not work if there is already a header extension that does not follow the mechanism described in RFC 5285 or if there is a header extension with a two bytes header as described in RFC 5285. In that case, use GstRtp.RTPBuffer.add_extension_twobytes_header

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

id ( int ) –

The ID of the header extension (between 1 and 14).

data ( [ int ] ) –

location for data

size ( int ) –

the size of the data in bytes

Returns ( bool ) –

True if header extension could be added


gst_rtp_buffer_add_extension_twobytes_header

gboolean
gst_rtp_buffer_add_extension_twobytes_header (GstRTPBuffer * rtp,
                                              guint8 appbits,
                                              guint8 id,
                                              gconstpointer data,
                                              guint size)

Adds a RFC 5285 header extension with a two bytes header to the end of the RTP header. If there is already a RFC 5285 header extension with a two bytes header, the new extension will be appended. It will not work if there is already a header extension that does not follow the mechanism described in RFC 5285 or if there is a header extension with a one byte header as described in RFC 5285. In that case, use gst_rtp_buffer_add_extension_onebyte_header

Parameters:

rtp

the RTP packet

appbits

Application specific bits

id

The ID of the header extension

data ( [array length=size] [element-type guint8] ) –

location for data

size

the size of the data in bytes

Returns

TRUE if header extension could be added


GstRtp.RTPBuffer.prototype.add_extension_twobytes_header

function GstRtp.RTPBuffer.prototype.add_extension_twobytes_header(appbits: Number, id: Number, data: [ Number ], size: Number): {
    // javascript wrapper for 'gst_rtp_buffer_add_extension_twobytes_header'
}

Adds a RFC 5285 header extension with a two bytes header to the end of the RTP header. If there is already a RFC 5285 header extension with a two bytes header, the new extension will be appended. It will not work if there is already a header extension that does not follow the mechanism described in RFC 5285 or if there is a header extension with a one byte header as described in RFC 5285. In that case, use GstRtp.RTPBuffer.prototype.add_extension_onebyte_header

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

appbits ( Number ) –

Application specific bits

id ( Number ) –

The ID of the header extension

data ( [ Number ] ) –

location for data

size ( Number ) –

the size of the data in bytes

Returns ( Number ) –

true if header extension could be added


GstRtp.RTPBuffer.add_extension_twobytes_header

def GstRtp.RTPBuffer.add_extension_twobytes_header (self, appbits, id, data, size):
    #python wrapper for 'gst_rtp_buffer_add_extension_twobytes_header'

Adds a RFC 5285 header extension with a two bytes header to the end of the RTP header. If there is already a RFC 5285 header extension with a two bytes header, the new extension will be appended. It will not work if there is already a header extension that does not follow the mechanism described in RFC 5285 or if there is a header extension with a one byte header as described in RFC 5285. In that case, use GstRtp.RTPBuffer.add_extension_onebyte_header

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

appbits ( int ) –

Application specific bits

id ( int ) –

The ID of the header extension

data ( [ int ] ) –

location for data

size ( int ) –

the size of the data in bytes

Returns ( bool ) –

True if header extension could be added


gst_rtp_buffer_get_csrc

guint32
gst_rtp_buffer_get_csrc (GstRTPBuffer * rtp,
                         guint8 idx)

Get the CSRC at index idx in buffer.

Parameters:

rtp

the RTP packet

idx

the index of the CSRC to get

Returns

the CSRC at index idx in host order.


GstRtp.RTPBuffer.prototype.get_csrc

function GstRtp.RTPBuffer.prototype.get_csrc(idx: Number): {
    // javascript wrapper for 'gst_rtp_buffer_get_csrc'
}

Get the CSRC at index idx in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

idx ( Number ) –

the index of the CSRC to get

Returns ( Number ) –

the CSRC at index idx in host order.


GstRtp.RTPBuffer.get_csrc

def GstRtp.RTPBuffer.get_csrc (self, idx):
    #python wrapper for 'gst_rtp_buffer_get_csrc'

Get the CSRC at index idx in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

idx ( int ) –

the index of the CSRC to get

Returns ( int ) –

the CSRC at index idx in host order.


gst_rtp_buffer_get_csrc_count

guint8
gst_rtp_buffer_get_csrc_count (GstRTPBuffer * rtp)

Get the CSRC count of the RTP packet in buffer.

Parameters:

rtp

the RTP packet

Returns

the CSRC count of buffer.


GstRtp.RTPBuffer.prototype.get_csrc_count

function GstRtp.RTPBuffer.prototype.get_csrc_count(): {
    // javascript wrapper for 'gst_rtp_buffer_get_csrc_count'
}

Get the CSRC count of the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( Number ) –

the CSRC count of buffer.


GstRtp.RTPBuffer.get_csrc_count

def GstRtp.RTPBuffer.get_csrc_count (self):
    #python wrapper for 'gst_rtp_buffer_get_csrc_count'

Get the CSRC count of the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( int ) –

the CSRC count of buffer.


gst_rtp_buffer_get_extension

gboolean
gst_rtp_buffer_get_extension (GstRTPBuffer * rtp)

Check if the extension bit is set on the RTP packet in buffer.

Parameters:

rtp

the RTP packet

Returns

TRUE if buffer has the extension bit set.


GstRtp.RTPBuffer.prototype.get_extension

function GstRtp.RTPBuffer.prototype.get_extension(): {
    // javascript wrapper for 'gst_rtp_buffer_get_extension'
}

Check if the extension bit is set on the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( Number ) –

TRUE if buffer has the extension bit set.


GstRtp.RTPBuffer.get_extension

def GstRtp.RTPBuffer.get_extension (self):
    #python wrapper for 'gst_rtp_buffer_get_extension'

Check if the extension bit is set on the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( bool ) –

TRUE if buffer has the extension bit set.


gst_rtp_buffer_get_extension_bytes

GBytes *
gst_rtp_buffer_get_extension_bytes (GstRTPBuffer * rtp,
                                    guint16 * bits)

Similar to gst_rtp_buffer_get_extension_data, but more suitable for language bindings usage. bits will contain the extension 16 bits of custom data and the extension data (not including the extension header) is placed in a new GBytes structure.

If rtp did not contain an extension, this function will return NULL, with bits unchanged. If there is an extension header but no extension data then an empty GBytes will be returned.

Parameters:

rtp

the RTP packet

bits ( [out] ) –

location for header bits

Returns ( [transfer: full] ) –

A new GBytes if an extension header was present and NULL otherwise.

Since : 1.2


GstRtp.RTPBuffer.prototype.get_extension_bytes

function GstRtp.RTPBuffer.prototype.get_extension_bytes(): {
    // javascript wrapper for 'gst_rtp_buffer_get_extension_bytes'
}

Similar to gst_rtp_buffer_get_extension_data, but more suitable for language bindings usage. bits will contain the extension 16 bits of custom data and the extension data (not including the extension header) is placed in a new GLib.Bytes structure.

If rtp did not contain an extension, this function will return null, with bits unchanged. If there is an extension header but no extension data then an empty GLib.Bytes will be returned.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns a tuple made of:

( GLib.Bytes ) –

A new GLib.Bytes if an extension header was present and null otherwise.

bits ( Number ) –

A new GLib.Bytes if an extension header was present and null otherwise.

Since : 1.2


GstRtp.RTPBuffer.get_extension_bytes

def GstRtp.RTPBuffer.get_extension_bytes (self):
    #python wrapper for 'gst_rtp_buffer_get_extension_bytes'

Similar to gst_rtp_buffer_get_extension_data, but more suitable for language bindings usage. bits will contain the extension 16 bits of custom data and the extension data (not including the extension header) is placed in a new GLib.Bytes structure.

If rtp did not contain an extension, this function will return None, with bits unchanged. If there is an extension header but no extension data then an empty GLib.Bytes will be returned.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns a tuple made of:

( GLib.Bytes ) –

A new GLib.Bytes if an extension header was present and None otherwise.

bits ( int ) –

A new GLib.Bytes if an extension header was present and None otherwise.

Since : 1.2


gst_rtp_buffer_get_extension_data

gboolean
gst_rtp_buffer_get_extension_data (GstRTPBuffer * rtp,
                                   guint16 * bits,
                                   gpointer * data,
                                   guint * wordlen)

Get the extension data. bits will contain the extension 16 bits of custom data. data will point to the data in the extension and wordlen will contain the length of data in 32 bits words.

If buffer did not contain an extension, this function will return FALSE with bits, data and wordlen unchanged.

Parameters:

rtp

the RTP packet

bits ( [out] ) –

location for result bits

data ( [out] [array] [element-type guint8] [transfer: none] ) –

location for data

wordlen ( [out] ) –

location for length of data in 32 bits words

Returns

TRUE if buffer had the extension bit set.


gst_rtp_buffer_get_extension_onebyte_header

gboolean
gst_rtp_buffer_get_extension_onebyte_header (GstRTPBuffer * rtp,
                                             guint8 id,
                                             guint nth,
                                             gpointer * data,
                                             guint * size)

Parses RFC 5285 style header extensions with a one byte header. It will return the nth extension with the requested id.

Parameters:

rtp

the RTP packet

id

The ID of the header extension to be read (between 1 and 14).

nth

Read the nth extension packet with the requested ID

data

(out) (array length=size) (element-type guint8) (transfer none): location for data

size ( [out] ) –

the size of the data in bytes

Returns

TRUE if buffer had the requested header extension


GstRtp.RTPBuffer.prototype.get_extension_onebyte_header

function GstRtp.RTPBuffer.prototype.get_extension_onebyte_header(id: Number, nth: Number): {
    // javascript wrapper for 'gst_rtp_buffer_get_extension_onebyte_header'
}

Parses RFC 5285 style header extensions with a one byte header. It will return the nth extension with the requested id.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

id ( Number ) –

The ID of the header extension to be read (between 1 and 14).

nth ( Number ) –

Read the nth extension packet with the requested ID

Returns a tuple made of:

( Number ) –

TRUE if buffer had the requested header extension

data ( [ Number ] ) –

TRUE if buffer had the requested header extension

size ( Number ) –

TRUE if buffer had the requested header extension


GstRtp.RTPBuffer.get_extension_onebyte_header

def GstRtp.RTPBuffer.get_extension_onebyte_header (self, id, nth):
    #python wrapper for 'gst_rtp_buffer_get_extension_onebyte_header'

Parses RFC 5285 style header extensions with a one byte header. It will return the nth extension with the requested id.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

id ( int ) –

The ID of the header extension to be read (between 1 and 14).

nth ( int ) –

Read the nth extension packet with the requested ID

Returns a tuple made of:

( bool ) –

TRUE if buffer had the requested header extension

data ( [ int ] ) –

TRUE if buffer had the requested header extension

size ( int ) –

TRUE if buffer had the requested header extension


gst_rtp_buffer_get_extension_twobytes_header

gboolean
gst_rtp_buffer_get_extension_twobytes_header (GstRTPBuffer * rtp,
                                              guint8 * appbits,
                                              guint8 id,
                                              guint nth,
                                              gpointer * data,
                                              guint * size)

Parses RFC 5285 style header extensions with a two bytes header. It will return the nth extension with the requested id.

Parameters:

rtp

the RTP packet

appbits ( [out] ) –

Application specific bits

id

The ID of the header extension to be read (between 1 and 14).

nth

Read the nth extension packet with the requested ID

data

(out) (array length=size) (element-type guint8) (transfer none): location for data

size ( [out] ) –

the size of the data in bytes

Returns

TRUE if buffer had the requested header extension


GstRtp.RTPBuffer.prototype.get_extension_twobytes_header

function GstRtp.RTPBuffer.prototype.get_extension_twobytes_header(id: Number, nth: Number): {
    // javascript wrapper for 'gst_rtp_buffer_get_extension_twobytes_header'
}

Parses RFC 5285 style header extensions with a two bytes header. It will return the nth extension with the requested id.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

id ( Number ) –

The ID of the header extension to be read (between 1 and 14).

nth ( Number ) –

Read the nth extension packet with the requested ID

Returns a tuple made of:

( Number ) –

TRUE if buffer had the requested header extension

appbits ( Number ) –

TRUE if buffer had the requested header extension

data ( [ Number ] ) –

TRUE if buffer had the requested header extension

size ( Number ) –

TRUE if buffer had the requested header extension


GstRtp.RTPBuffer.get_extension_twobytes_header

def GstRtp.RTPBuffer.get_extension_twobytes_header (self, id, nth):
    #python wrapper for 'gst_rtp_buffer_get_extension_twobytes_header'

Parses RFC 5285 style header extensions with a two bytes header. It will return the nth extension with the requested id.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

id ( int ) –

The ID of the header extension to be read (between 1 and 14).

nth ( int ) –

Read the nth extension packet with the requested ID

Returns a tuple made of:

( bool ) –

TRUE if buffer had the requested header extension

appbits ( int ) –

TRUE if buffer had the requested header extension

data ( [ int ] ) –

TRUE if buffer had the requested header extension

size ( int ) –

TRUE if buffer had the requested header extension


gst_rtp_buffer_get_header_len

guint
gst_rtp_buffer_get_header_len (GstRTPBuffer * rtp)

Return the total length of the header in buffer. This include the length of the fixed header, the CSRC list and the extension header.

Parameters:

rtp

the RTP packet

Returns

The total length of the header in buffer.


GstRtp.RTPBuffer.prototype.get_header_len

function GstRtp.RTPBuffer.prototype.get_header_len(): {
    // javascript wrapper for 'gst_rtp_buffer_get_header_len'
}

Return the total length of the header in buffer. This include the length of the fixed header, the CSRC list and the extension header.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( Number ) –

The total length of the header in buffer.


GstRtp.RTPBuffer.get_header_len

def GstRtp.RTPBuffer.get_header_len (self):
    #python wrapper for 'gst_rtp_buffer_get_header_len'

Return the total length of the header in buffer. This include the length of the fixed header, the CSRC list and the extension header.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( int ) –

The total length of the header in buffer.


gst_rtp_buffer_get_marker

gboolean
gst_rtp_buffer_get_marker (GstRTPBuffer * rtp)

Check if the marker bit is set on the RTP packet in buffer.

Parameters:

rtp

the RTP packet

Returns

TRUE if buffer has the marker bit set.


GstRtp.RTPBuffer.prototype.get_marker

function GstRtp.RTPBuffer.prototype.get_marker(): {
    // javascript wrapper for 'gst_rtp_buffer_get_marker'
}

Check if the marker bit is set on the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( Number ) –

TRUE if buffer has the marker bit set.


GstRtp.RTPBuffer.get_marker

def GstRtp.RTPBuffer.get_marker (self):
    #python wrapper for 'gst_rtp_buffer_get_marker'

Check if the marker bit is set on the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( bool ) –

TRUE if buffer has the marker bit set.


gst_rtp_buffer_get_packet_len

guint
gst_rtp_buffer_get_packet_len (GstRTPBuffer * rtp)

Return the total length of the packet in buffer.

Parameters:

rtp

the RTP packet

Returns

The total length of the packet in buffer.


GstRtp.RTPBuffer.prototype.get_packet_len

function GstRtp.RTPBuffer.prototype.get_packet_len(): {
    // javascript wrapper for 'gst_rtp_buffer_get_packet_len'
}

Return the total length of the packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( Number ) –

The total length of the packet in buffer.


GstRtp.RTPBuffer.get_packet_len

def GstRtp.RTPBuffer.get_packet_len (self):
    #python wrapper for 'gst_rtp_buffer_get_packet_len'

Return the total length of the packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( int ) –

The total length of the packet in buffer.


gst_rtp_buffer_get_padding

gboolean
gst_rtp_buffer_get_padding (GstRTPBuffer * rtp)

Check if the padding bit is set on the RTP packet in buffer.

Parameters:

rtp

the RTP packet

Returns

TRUE if buffer has the padding bit set.


GstRtp.RTPBuffer.prototype.get_padding

function GstRtp.RTPBuffer.prototype.get_padding(): {
    // javascript wrapper for 'gst_rtp_buffer_get_padding'
}

Check if the padding bit is set on the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( Number ) –

TRUE if buffer has the padding bit set.


GstRtp.RTPBuffer.get_padding

def GstRtp.RTPBuffer.get_padding (self):
    #python wrapper for 'gst_rtp_buffer_get_padding'

Check if the padding bit is set on the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( bool ) –

TRUE if buffer has the padding bit set.


gst_rtp_buffer_get_payload

gpointer
gst_rtp_buffer_get_payload (GstRTPBuffer * rtp)

Get a pointer to the payload data in buffer. This pointer is valid as long as a reference to buffer is held.

Parameters:

rtp

the RTP packet

Returns ( [array] [element-type guint8] [transfer: none] ) –

A pointer to the payload data in buffer.


gst_rtp_buffer_get_payload_buffer

GstBuffer *
gst_rtp_buffer_get_payload_buffer (GstRTPBuffer * rtp)

Create a buffer of the payload of the RTP packet in buffer. This function will internally create a subbuffer of buffer so that a memcpy can be avoided.

Parameters:

rtp

the RTP packet

Returns

A new buffer with the data of the payload.


GstRtp.RTPBuffer.prototype.get_payload_buffer

function GstRtp.RTPBuffer.prototype.get_payload_buffer(): {
    // javascript wrapper for 'gst_rtp_buffer_get_payload_buffer'
}

Create a buffer of the payload of the RTP packet in buffer. This function will internally create a subbuffer of buffer so that a memcpy can be avoided.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( Gst.Buffer ) –

A new buffer with the data of the payload.


GstRtp.RTPBuffer.get_payload_buffer

def GstRtp.RTPBuffer.get_payload_buffer (self):
    #python wrapper for 'gst_rtp_buffer_get_payload_buffer'

Create a buffer of the payload of the RTP packet in buffer. This function will internally create a subbuffer of buffer so that a memcpy can be avoided.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( Gst.Buffer ) –

A new buffer with the data of the payload.


gst_rtp_buffer_get_payload_bytes

GBytes *
gst_rtp_buffer_get_payload_bytes (GstRTPBuffer * rtp)

Similar to gst_rtp_buffer_get_payload, but more suitable for language bindings usage. The return value is a pointer to a GBytes structure containing the payload data in rtp.

Parameters:

rtp

the RTP packet

Returns ( [transfer: full] ) –

A new GBytes containing the payload data in rtp.

Since : 1.2


GstRtp.RTPBuffer.prototype.get_payload_bytes

function GstRtp.RTPBuffer.prototype.get_payload_bytes(): {
    // javascript wrapper for 'gst_rtp_buffer_get_payload_bytes'
}

Similar to gst_rtp_buffer_get_payload, but more suitable for language bindings usage. The return value is a pointer to a GLib.Bytes structure containing the payload data in rtp.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( GLib.Bytes ) –

A new GLib.Bytes containing the payload data in rtp.

Since : 1.2


GstRtp.RTPBuffer.get_payload_bytes

def GstRtp.RTPBuffer.get_payload_bytes (self):
    #python wrapper for 'gst_rtp_buffer_get_payload_bytes'

Similar to gst_rtp_buffer_get_payload, but more suitable for language bindings usage. The return value is a pointer to a GLib.Bytes structure containing the payload data in rtp.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( GLib.Bytes ) –

A new GLib.Bytes containing the payload data in rtp.

Since : 1.2


gst_rtp_buffer_get_payload_len

guint
gst_rtp_buffer_get_payload_len (GstRTPBuffer * rtp)

Get the length of the payload of the RTP packet in buffer.

Parameters:

rtp

the RTP packet

Returns

The length of the payload in buffer.


GstRtp.RTPBuffer.prototype.get_payload_len

function GstRtp.RTPBuffer.prototype.get_payload_len(): {
    // javascript wrapper for 'gst_rtp_buffer_get_payload_len'
}

Get the length of the payload of the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( Number ) –

The length of the payload in buffer.


GstRtp.RTPBuffer.get_payload_len

def GstRtp.RTPBuffer.get_payload_len (self):
    #python wrapper for 'gst_rtp_buffer_get_payload_len'

Get the length of the payload of the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( int ) –

The length of the payload in buffer.


gst_rtp_buffer_get_payload_subbuffer

GstBuffer *
gst_rtp_buffer_get_payload_subbuffer (GstRTPBuffer * rtp,
                                      guint offset,
                                      guint len)

Create a subbuffer of the payload of the RTP packet in buffer. offset bytes are skipped in the payload and the subbuffer will be of size len. If len is -1 the total payload starting from offset is subbuffered.

Parameters:

rtp

the RTP packet

offset

the offset in the payload

len

the length in the payload

Returns

A new buffer with the specified data of the payload.


GstRtp.RTPBuffer.prototype.get_payload_subbuffer

function GstRtp.RTPBuffer.prototype.get_payload_subbuffer(offset: Number, len: Number): {
    // javascript wrapper for 'gst_rtp_buffer_get_payload_subbuffer'
}

Create a subbuffer of the payload of the RTP packet in buffer. offset bytes are skipped in the payload and the subbuffer will be of size len. If len is -1 the total payload starting from offset is subbuffered.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

offset ( Number ) –

the offset in the payload

len ( Number ) –

the length in the payload

Returns ( Gst.Buffer ) –

A new buffer with the specified data of the payload.


GstRtp.RTPBuffer.get_payload_subbuffer

def GstRtp.RTPBuffer.get_payload_subbuffer (self, offset, len):
    #python wrapper for 'gst_rtp_buffer_get_payload_subbuffer'

Create a subbuffer of the payload of the RTP packet in buffer. offset bytes are skipped in the payload and the subbuffer will be of size len. If len is -1 the total payload starting from offset is subbuffered.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

offset ( int ) –

the offset in the payload

len ( int ) –

the length in the payload

Returns ( Gst.Buffer ) –

A new buffer with the specified data of the payload.


gst_rtp_buffer_get_payload_type

guint8
gst_rtp_buffer_get_payload_type (GstRTPBuffer * rtp)

Get the payload type of the RTP packet in buffer.

Parameters:

rtp

the RTP packet

Returns

The payload type.


GstRtp.RTPBuffer.prototype.get_payload_type

function GstRtp.RTPBuffer.prototype.get_payload_type(): {
    // javascript wrapper for 'gst_rtp_buffer_get_payload_type'
}

Get the payload type of the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( Number ) –

The payload type.


GstRtp.RTPBuffer.get_payload_type

def GstRtp.RTPBuffer.get_payload_type (self):
    #python wrapper for 'gst_rtp_buffer_get_payload_type'

Get the payload type of the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( int ) –

The payload type.


gst_rtp_buffer_get_seq

guint16
gst_rtp_buffer_get_seq (GstRTPBuffer * rtp)

Get the sequence number of the RTP packet in buffer.

Parameters:

rtp

the RTP packet

Returns

The sequence number in host order.


GstRtp.RTPBuffer.prototype.get_seq

function GstRtp.RTPBuffer.prototype.get_seq(): {
    // javascript wrapper for 'gst_rtp_buffer_get_seq'
}

Get the sequence number of the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( Number ) –

The sequence number in host order.


GstRtp.RTPBuffer.get_seq

def GstRtp.RTPBuffer.get_seq (self):
    #python wrapper for 'gst_rtp_buffer_get_seq'

Get the sequence number of the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( int ) –

The sequence number in host order.


gst_rtp_buffer_get_ssrc

guint32
gst_rtp_buffer_get_ssrc (GstRTPBuffer * rtp)

Get the SSRC of the RTP packet in buffer.

Parameters:

rtp

the RTP packet

Returns

the SSRC of buffer in host order.


GstRtp.RTPBuffer.prototype.get_ssrc

function GstRtp.RTPBuffer.prototype.get_ssrc(): {
    // javascript wrapper for 'gst_rtp_buffer_get_ssrc'
}

Get the SSRC of the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( Number ) –

the SSRC of buffer in host order.


GstRtp.RTPBuffer.get_ssrc

def GstRtp.RTPBuffer.get_ssrc (self):
    #python wrapper for 'gst_rtp_buffer_get_ssrc'

Get the SSRC of the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( int ) –

the SSRC of buffer in host order.


gst_rtp_buffer_get_timestamp

guint32
gst_rtp_buffer_get_timestamp (GstRTPBuffer * rtp)

Get the timestamp of the RTP packet in buffer.

Parameters:

rtp

the RTP packet

Returns

The timestamp in host order.


GstRtp.RTPBuffer.prototype.get_timestamp

function GstRtp.RTPBuffer.prototype.get_timestamp(): {
    // javascript wrapper for 'gst_rtp_buffer_get_timestamp'
}

Get the timestamp of the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( Number ) –

The timestamp in host order.


GstRtp.RTPBuffer.get_timestamp

def GstRtp.RTPBuffer.get_timestamp (self):
    #python wrapper for 'gst_rtp_buffer_get_timestamp'

Get the timestamp of the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( int ) –

The timestamp in host order.


gst_rtp_buffer_get_version

guint8
gst_rtp_buffer_get_version (GstRTPBuffer * rtp)

Get the version number of the RTP packet in buffer.

Parameters:

rtp

the RTP packet

Returns

The version of buffer.


GstRtp.RTPBuffer.prototype.get_version

function GstRtp.RTPBuffer.prototype.get_version(): {
    // javascript wrapper for 'gst_rtp_buffer_get_version'
}

Get the version number of the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( Number ) –

The version of buffer.


GstRtp.RTPBuffer.get_version

def GstRtp.RTPBuffer.get_version (self):
    #python wrapper for 'gst_rtp_buffer_get_version'

Get the version number of the RTP packet in buffer.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

Returns ( int ) –

The version of buffer.


gst_rtp_buffer_pad_to

gst_rtp_buffer_pad_to (GstRTPBuffer * rtp,
                       guint len)

Set the amount of padding in the RTP packet in buffer to len. If len is 0, the padding is removed.

NOTE: This function does not work correctly.

Parameters:

rtp

the RTP packet

len

the new amount of padding


GstRtp.RTPBuffer.prototype.pad_to

function GstRtp.RTPBuffer.prototype.pad_to(len: Number): {
    // javascript wrapper for 'gst_rtp_buffer_pad_to'
}

Set the amount of padding in the RTP packet in buffer to len. If len is 0, the padding is removed.

NOTE: This function does not work correctly.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

len ( Number ) –

the new amount of padding


GstRtp.RTPBuffer.pad_to

def GstRtp.RTPBuffer.pad_to (self, len):
    #python wrapper for 'gst_rtp_buffer_pad_to'

Set the amount of padding in the RTP packet in buffer to len. If len is 0, the padding is removed.

NOTE: This function does not work correctly.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

len ( int ) –

the new amount of padding


gst_rtp_buffer_set_csrc

gst_rtp_buffer_set_csrc (GstRTPBuffer * rtp,
                         guint8 idx,
                         guint32 csrc)

Modify the CSRC at index idx in buffer to csrc.

Parameters:

rtp

the RTP packet

idx

the CSRC index to set

csrc

the CSRC in host order to set at idx


GstRtp.RTPBuffer.prototype.set_csrc

function GstRtp.RTPBuffer.prototype.set_csrc(idx: Number, csrc: Number): {
    // javascript wrapper for 'gst_rtp_buffer_set_csrc'
}

Modify the CSRC at index idx in buffer to csrc.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

idx ( Number ) –

the CSRC index to set

csrc ( Number ) –

the CSRC in host order to set at idx


GstRtp.RTPBuffer.set_csrc

def GstRtp.RTPBuffer.set_csrc (self, idx, csrc):
    #python wrapper for 'gst_rtp_buffer_set_csrc'

Modify the CSRC at index idx in buffer to csrc.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

idx ( int ) –

the CSRC index to set

csrc ( int ) –

the CSRC in host order to set at idx


gst_rtp_buffer_set_extension

gst_rtp_buffer_set_extension (GstRTPBuffer * rtp,
                              gboolean extension)

Set the extension bit on the RTP packet in buffer to extension.

Parameters:

rtp

the RTP packet

extension

the new extension


GstRtp.RTPBuffer.prototype.set_extension

function GstRtp.RTPBuffer.prototype.set_extension(extension: Number): {
    // javascript wrapper for 'gst_rtp_buffer_set_extension'
}

Set the extension bit on the RTP packet in buffer to extension.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

extension ( Number ) –

the new extension


GstRtp.RTPBuffer.set_extension

def GstRtp.RTPBuffer.set_extension (self, extension):
    #python wrapper for 'gst_rtp_buffer_set_extension'

Set the extension bit on the RTP packet in buffer to extension.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

extension ( bool ) –

the new extension


gst_rtp_buffer_set_extension_data

gboolean
gst_rtp_buffer_set_extension_data (GstRTPBuffer * rtp,
                                   guint16 bits,
                                   guint16 length)

Set the extension bit of the rtp buffer and fill in the bits and length of the extension header. If the existing extension data is not large enough, it will be made larger.

Parameters:

rtp

the RTP packet

bits

the bits specific for the extension

length

the length that counts the number of 32-bit words in the extension, excluding the extension header ( therefore zero is a valid length)

Returns

True if done.


GstRtp.RTPBuffer.prototype.set_extension_data

function GstRtp.RTPBuffer.prototype.set_extension_data(bits: Number, length: Number): {
    // javascript wrapper for 'gst_rtp_buffer_set_extension_data'
}

Set the extension bit of the rtp buffer and fill in the bits and length of the extension header. If the existing extension data is not large enough, it will be made larger.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

bits ( Number ) –

the bits specific for the extension

length ( Number ) –

the length that counts the number of 32-bit words in the extension, excluding the extension header ( therefore zero is a valid length)

Returns ( Number ) –

True if done.


GstRtp.RTPBuffer.set_extension_data

def GstRtp.RTPBuffer.set_extension_data (self, bits, length):
    #python wrapper for 'gst_rtp_buffer_set_extension_data'

Set the extension bit of the rtp buffer and fill in the bits and length of the extension header. If the existing extension data is not large enough, it will be made larger.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

bits ( int ) –

the bits specific for the extension

length ( int ) –

the length that counts the number of 32-bit words in the extension, excluding the extension header ( therefore zero is a valid length)

Returns ( bool ) –

True if done.


gst_rtp_buffer_set_marker

gst_rtp_buffer_set_marker (GstRTPBuffer * rtp,
                           gboolean marker)

Set the marker bit on the RTP packet in buffer to marker.

Parameters:

rtp

the RTP packet

marker

the new marker


GstRtp.RTPBuffer.prototype.set_marker

function GstRtp.RTPBuffer.prototype.set_marker(marker: Number): {
    // javascript wrapper for 'gst_rtp_buffer_set_marker'
}

Set the marker bit on the RTP packet in buffer to marker.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

marker ( Number ) –

the new marker


GstRtp.RTPBuffer.set_marker

def GstRtp.RTPBuffer.set_marker (self, marker):
    #python wrapper for 'gst_rtp_buffer_set_marker'

Set the marker bit on the RTP packet in buffer to marker.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

marker ( bool ) –

the new marker


gst_rtp_buffer_set_packet_len

gst_rtp_buffer_set_packet_len (GstRTPBuffer * rtp,
                               guint len)

Set the total rtp size to len. The data in the buffer will be made larger if needed. Any padding will be removed from the packet.

Parameters:

rtp

the RTP packet

len

the new packet length


GstRtp.RTPBuffer.prototype.set_packet_len

function GstRtp.RTPBuffer.prototype.set_packet_len(len: Number): {
    // javascript wrapper for 'gst_rtp_buffer_set_packet_len'
}

Set the total rtp size to len. The data in the buffer will be made larger if needed. Any padding will be removed from the packet.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

len ( Number ) –

the new packet length


GstRtp.RTPBuffer.set_packet_len

def GstRtp.RTPBuffer.set_packet_len (self, len):
    #python wrapper for 'gst_rtp_buffer_set_packet_len'

Set the total rtp size to len. The data in the buffer will be made larger if needed. Any padding will be removed from the packet.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

len ( int ) –

the new packet length


gst_rtp_buffer_set_padding

gst_rtp_buffer_set_padding (GstRTPBuffer * rtp,
                            gboolean padding)

Set the padding bit on the RTP packet in buffer to padding.

Parameters:

rtp

the buffer

padding

the new padding


GstRtp.RTPBuffer.prototype.set_padding

function GstRtp.RTPBuffer.prototype.set_padding(padding: Number): {
    // javascript wrapper for 'gst_rtp_buffer_set_padding'
}

Set the padding bit on the RTP packet in buffer to padding.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the buffer

padding ( Number ) –

the new padding


GstRtp.RTPBuffer.set_padding

def GstRtp.RTPBuffer.set_padding (self, padding):
    #python wrapper for 'gst_rtp_buffer_set_padding'

Set the padding bit on the RTP packet in buffer to padding.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the buffer

padding ( bool ) –

the new padding


gst_rtp_buffer_set_payload_type

gst_rtp_buffer_set_payload_type (GstRTPBuffer * rtp,
                                 guint8 payload_type)

Set the payload type of the RTP packet in buffer to payload_type.

Parameters:

rtp

the RTP packet

payload_type

the new type


GstRtp.RTPBuffer.prototype.set_payload_type

function GstRtp.RTPBuffer.prototype.set_payload_type(payload_type: Number): {
    // javascript wrapper for 'gst_rtp_buffer_set_payload_type'
}

Set the payload type of the RTP packet in buffer to payload_type.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

payload_type ( Number ) –

the new type


GstRtp.RTPBuffer.set_payload_type

def GstRtp.RTPBuffer.set_payload_type (self, payload_type):
    #python wrapper for 'gst_rtp_buffer_set_payload_type'

Set the payload type of the RTP packet in buffer to payload_type.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

payload_type ( int ) –

the new type


gst_rtp_buffer_set_seq

gst_rtp_buffer_set_seq (GstRTPBuffer * rtp,
                        guint16 seq)

Set the sequence number of the RTP packet in buffer to seq.

Parameters:

rtp

the RTP packet

seq

the new sequence number


GstRtp.RTPBuffer.prototype.set_seq

function GstRtp.RTPBuffer.prototype.set_seq(seq: Number): {
    // javascript wrapper for 'gst_rtp_buffer_set_seq'
}

Set the sequence number of the RTP packet in buffer to seq.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

seq ( Number ) –

the new sequence number


GstRtp.RTPBuffer.set_seq

def GstRtp.RTPBuffer.set_seq (self, seq):
    #python wrapper for 'gst_rtp_buffer_set_seq'

Set the sequence number of the RTP packet in buffer to seq.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

seq ( int ) –

the new sequence number


gst_rtp_buffer_set_ssrc

gst_rtp_buffer_set_ssrc (GstRTPBuffer * rtp,
                         guint32 ssrc)

Set the SSRC on the RTP packet in buffer to ssrc.

Parameters:

rtp

the RTP packet

ssrc

the new SSRC


GstRtp.RTPBuffer.prototype.set_ssrc

function GstRtp.RTPBuffer.prototype.set_ssrc(ssrc: Number): {
    // javascript wrapper for 'gst_rtp_buffer_set_ssrc'
}

Set the SSRC on the RTP packet in buffer to ssrc.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

ssrc ( Number ) –

the new SSRC


GstRtp.RTPBuffer.set_ssrc

def GstRtp.RTPBuffer.set_ssrc (self, ssrc):
    #python wrapper for 'gst_rtp_buffer_set_ssrc'

Set the SSRC on the RTP packet in buffer to ssrc.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

ssrc ( int ) –

the new SSRC


gst_rtp_buffer_set_timestamp

gst_rtp_buffer_set_timestamp (GstRTPBuffer * rtp,
                              guint32 timestamp)

Set the timestamp of the RTP packet in buffer to timestamp.

Parameters:

rtp

the RTP packet

timestamp

the new timestamp


GstRtp.RTPBuffer.prototype.set_timestamp

function GstRtp.RTPBuffer.prototype.set_timestamp(timestamp: Number): {
    // javascript wrapper for 'gst_rtp_buffer_set_timestamp'
}

Set the timestamp of the RTP packet in buffer to timestamp.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

timestamp ( Number ) –

the new timestamp


GstRtp.RTPBuffer.set_timestamp

def GstRtp.RTPBuffer.set_timestamp (self, timestamp):
    #python wrapper for 'gst_rtp_buffer_set_timestamp'

Set the timestamp of the RTP packet in buffer to timestamp.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

timestamp ( int ) –

the new timestamp


gst_rtp_buffer_set_version

gst_rtp_buffer_set_version (GstRTPBuffer * rtp,
                            guint8 version)

Set the version of the RTP packet in buffer to version.

Parameters:

rtp

the RTP packet

version

the new version


GstRtp.RTPBuffer.prototype.set_version

function GstRtp.RTPBuffer.prototype.set_version(version: Number): {
    // javascript wrapper for 'gst_rtp_buffer_set_version'
}

Set the version of the RTP packet in buffer to version.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

version ( Number ) –

the new version


GstRtp.RTPBuffer.set_version

def GstRtp.RTPBuffer.set_version (self, version):
    #python wrapper for 'gst_rtp_buffer_set_version'

Set the version of the RTP packet in buffer to version.

Parameters:

rtp ( GstRtp.RTPBuffer ) –

the RTP packet

version ( int ) –

the new version


gst_rtp_buffer_unmap

gst_rtp_buffer_unmap (GstRTPBuffer * rtp)

Unmap rtp previously mapped with gst_rtp_buffer_map.

Parameters:

rtp

a GstRTPBuffer


GstRtp.RTPBuffer.prototype.unmap

function GstRtp.RTPBuffer.prototype.unmap(): {
    // javascript wrapper for 'gst_rtp_buffer_unmap'
}

Unmap rtp previously mapped with GstRtp.RTPBuffer.prototype.map.

Parameters:


GstRtp.RTPBuffer.unmap

def GstRtp.RTPBuffer.unmap (self):
    #python wrapper for 'gst_rtp_buffer_unmap'

Unmap rtp previously mapped with GstRtp.RTPBuffer.map.

Parameters:


Functions

gst_rtp_buffer_allocate_data

gst_rtp_buffer_allocate_data (GstBuffer * buffer,
                              guint payload_len,
                              guint8 pad_len,
                              guint8 csrc_count)

Allocate enough data in buffer to hold an RTP packet with csrc_count CSRCs, a payload length of payload_len and padding of pad_len. buffer must be writable and all previous memory in buffer will be freed. If pad_len is >0, the padding bit will be set. All other RTP header fields will be set to 0/FALSE.

Parameters:

buffer

a GstBuffer

payload_len

the length of the payload

pad_len

the amount of padding

csrc_count

the number of CSRC entries


GstRtp.RTPBuffer.prototype.allocate_data

function GstRtp.RTPBuffer.prototype.allocate_data(buffer: Gst.Buffer, payload_len: Number, pad_len: Number, csrc_count: Number): {
    // javascript wrapper for 'gst_rtp_buffer_allocate_data'
}

Allocate enough data in buffer to hold an RTP packet with csrc_count CSRCs, a payload length of payload_len and padding of pad_len. buffer must be writable and all previous memory in buffer will be freed. If pad_len is >0, the padding bit will be set. All other RTP header fields will be set to 0/FALSE.

Parameters:

buffer ( Gst.Buffer ) –

a Gst.Buffer

payload_len ( Number ) –

the length of the payload

pad_len ( Number ) –

the amount of padding

csrc_count ( Number ) –

the number of CSRC entries


GstRtp.RTPBuffer.allocate_data

def GstRtp.RTPBuffer.allocate_data (buffer, payload_len, pad_len, csrc_count):
    #python wrapper for 'gst_rtp_buffer_allocate_data'

Allocate enough data in buffer to hold an RTP packet with csrc_count CSRCs, a payload length of payload_len and padding of pad_len. buffer must be writable and all previous memory in buffer will be freed. If pad_len is >0, the padding bit will be set. All other RTP header fields will be set to 0/FALSE.

Parameters:

buffer ( Gst.Buffer ) –

a Gst.Buffer

payload_len ( int ) –

the length of the payload

pad_len ( int ) –

the amount of padding

csrc_count ( int ) –

the number of CSRC entries


gst_rtp_buffer_calc_header_len

guint
gst_rtp_buffer_calc_header_len (guint8 csrc_count)

Calculate the header length of an RTP packet with csrc_count CSRC entries. An RTP packet can have at most 15 CSRC entries.

Parameters:

csrc_count

the number of CSRC entries

Returns

The length of an RTP header with csrc_count CSRC entries.


GstRtp.RTPBuffer.prototype.calc_header_len

function GstRtp.RTPBuffer.prototype.calc_header_len(csrc_count: Number): {
    // javascript wrapper for 'gst_rtp_buffer_calc_header_len'
}

Calculate the header length of an RTP packet with csrc_count CSRC entries. An RTP packet can have at most 15 CSRC entries.

Parameters:

csrc_count ( Number ) –

the number of CSRC entries

Returns ( Number ) –

The length of an RTP header with csrc_count CSRC entries.


GstRtp.RTPBuffer.calc_header_len

def GstRtp.RTPBuffer.calc_header_len (csrc_count):
    #python wrapper for 'gst_rtp_buffer_calc_header_len'

Calculate the header length of an RTP packet with csrc_count CSRC entries. An RTP packet can have at most 15 CSRC entries.

Parameters:

csrc_count ( int ) –

the number of CSRC entries

Returns ( int ) –

The length of an RTP header with csrc_count CSRC entries.


gst_rtp_buffer_calc_packet_len

guint
gst_rtp_buffer_calc_packet_len (guint payload_len,
                                guint8 pad_len,
                                guint8 csrc_count)

Calculate the total length of an RTP packet with a payload size of payload_len, a padding of pad_len and a csrc_count CSRC entries.

Parameters:

payload_len

the length of the payload

pad_len

the amount of padding

csrc_count

the number of CSRC entries

Returns

The total length of an RTP header with given parameters.


GstRtp.RTPBuffer.prototype.calc_packet_len

function GstRtp.RTPBuffer.prototype.calc_packet_len(payload_len: Number, pad_len: Number, csrc_count: Number): {
    // javascript wrapper for 'gst_rtp_buffer_calc_packet_len'
}

Calculate the total length of an RTP packet with a payload size of payload_len, a padding of pad_len and a csrc_count CSRC entries.

Parameters:

payload_len ( Number ) –

the length of the payload

pad_len ( Number ) –

the amount of padding

csrc_count ( Number ) –

the number of CSRC entries

Returns ( Number ) –

The total length of an RTP header with given parameters.


GstRtp.RTPBuffer.calc_packet_len

def GstRtp.RTPBuffer.calc_packet_len (payload_len, pad_len, csrc_count):
    #python wrapper for 'gst_rtp_buffer_calc_packet_len'

Calculate the total length of an RTP packet with a payload size of payload_len, a padding of pad_len and a csrc_count CSRC entries.

Parameters:

payload_len ( int ) –

the length of the payload

pad_len ( int ) –

the amount of padding

csrc_count ( int ) –

the number of CSRC entries

Returns ( int ) –

The total length of an RTP header with given parameters.


gst_rtp_buffer_calc_payload_len

guint
gst_rtp_buffer_calc_payload_len (guint packet_len,
                                 guint8 pad_len,
                                 guint8 csrc_count)

Calculate the length of the payload of an RTP packet with size packet_len, a padding of pad_len and a csrc_count CSRC entries.

Parameters:

packet_len

the length of the total RTP packet

pad_len

the amount of padding

csrc_count

the number of CSRC entries

Returns

The length of the payload of an RTP packet with given parameters.


GstRtp.RTPBuffer.prototype.calc_payload_len

function GstRtp.RTPBuffer.prototype.calc_payload_len(packet_len: Number, pad_len: Number, csrc_count: Number): {
    // javascript wrapper for 'gst_rtp_buffer_calc_payload_len'
}

Calculate the length of the payload of an RTP packet with size packet_len, a padding of pad_len and a csrc_count CSRC entries.

Parameters:

packet_len ( Number ) –

the length of the total RTP packet

pad_len ( Number ) –

the amount of padding

csrc_count ( Number ) –

the number of CSRC entries

Returns ( Number ) –

The length of the payload of an RTP packet with given parameters.


GstRtp.RTPBuffer.calc_payload_len

def GstRtp.RTPBuffer.calc_payload_len (packet_len, pad_len, csrc_count):
    #python wrapper for 'gst_rtp_buffer_calc_payload_len'

Calculate the length of the payload of an RTP packet with size packet_len, a padding of pad_len and a csrc_count CSRC entries.

Parameters:

packet_len ( int ) –

the length of the total RTP packet

pad_len ( int ) –

the amount of padding

csrc_count ( int ) –

the number of CSRC entries

Returns ( int ) –

The length of the payload of an RTP packet with given parameters.


gst_rtp_buffer_compare_seqnum

gint
gst_rtp_buffer_compare_seqnum (guint16 seqnum1,
                               guint16 seqnum2)

Compare two sequence numbers, taking care of wraparounds. This function returns the difference between seqnum1 and seqnum2.

Parameters:

seqnum1

a sequence number

seqnum2

a sequence number

Returns

a negative value if seqnum1 is bigger than seqnum2, 0 if they are equal or a positive value if seqnum1 is smaller than segnum2.


GstRtp.RTPBuffer.prototype.compare_seqnum

function GstRtp.RTPBuffer.prototype.compare_seqnum(seqnum1: Number, seqnum2: Number): {
    // javascript wrapper for 'gst_rtp_buffer_compare_seqnum'
}

Compare two sequence numbers, taking care of wraparounds. This function returns the difference between seqnum1 and seqnum2.

Parameters:

seqnum1 ( Number ) –

a sequence number

seqnum2 ( Number ) –

a sequence number

Returns ( Number ) –

a negative value if seqnum1 is bigger than seqnum2, 0 if they are equal or a positive value if seqnum1 is smaller than segnum2.


GstRtp.RTPBuffer.compare_seqnum

def GstRtp.RTPBuffer.compare_seqnum (seqnum1, seqnum2):
    #python wrapper for 'gst_rtp_buffer_compare_seqnum'

Compare two sequence numbers, taking care of wraparounds. This function returns the difference between seqnum1 and seqnum2.

Parameters:

seqnum1 ( int ) –

a sequence number

seqnum2 ( int ) –

a sequence number

Returns ( int ) –

a negative value if seqnum1 is bigger than seqnum2, 0 if they are equal or a positive value if seqnum1 is smaller than segnum2.


gst_rtp_buffer_default_clock_rate

guint32
gst_rtp_buffer_default_clock_rate (guint8 payload_type)

Get the default clock-rate for the static payload type payload_type.

Parameters:

payload_type

the static payload type

Returns

the default clock rate or -1 if the payload type is not static or the clock-rate is undefined.


GstRtp.RTPBuffer.prototype.default_clock_rate

function GstRtp.RTPBuffer.prototype.default_clock_rate(payload_type: Number): {
    // javascript wrapper for 'gst_rtp_buffer_default_clock_rate'
}

Get the default clock-rate for the static payload type payload_type.

Parameters:

payload_type ( Number ) –

the static payload type

Returns ( Number ) –

the default clock rate or -1 if the payload type is not static or the clock-rate is undefined.


GstRtp.RTPBuffer.default_clock_rate

def GstRtp.RTPBuffer.default_clock_rate (payload_type):
    #python wrapper for 'gst_rtp_buffer_default_clock_rate'

Get the default clock-rate for the static payload type payload_type.

Parameters:

payload_type ( int ) –

the static payload type

Returns ( int ) –

the default clock rate or -1 if the payload type is not static or the clock-rate is undefined.


gst_rtp_buffer_ext_timestamp

guint64
gst_rtp_buffer_ext_timestamp (guint64 * exttimestamp,
                              guint32 timestamp)

Update the exttimestamp field with the extended timestamp of timestamp For the first call of the method, exttimestamp should point to a location with a value of -1.

This function is able to handle both forward and backward timestamps taking into account:

  • timestamp wraparound making sure that the returned value is properly increased.
  • timestamp unwraparound making sure that the returned value is properly decreased.

Parameters:

exttimestamp ( [inout] ) –

a previous extended timestamp

timestamp

a new timestamp

Returns

The extended timestamp of timestamp or 0 if the result can't go anywhere backwards.


GstRtp.RTPBuffer.prototype.ext_timestamp

function GstRtp.RTPBuffer.prototype.ext_timestamp(exttimestamp: Number, timestamp: Number): {
    // javascript wrapper for 'gst_rtp_buffer_ext_timestamp'
}

Update the exttimestamp field with the extended timestamp of timestamp For the first call of the method, exttimestamp should point to a location with a value of -1.

This function is able to handle both forward and backward timestamps taking into account:

  • timestamp wraparound making sure that the returned value is properly increased.
  • timestamp unwraparound making sure that the returned value is properly decreased.

Parameters:

exttimestamp ( Number ) –

a previous extended timestamp

timestamp ( Number ) –

a new timestamp

Returns a tuple made of:

( Number ) –

The extended timestamp of timestamp or 0 if the result can't go anywhere backwards.

exttimestamp ( Number ) –

The extended timestamp of timestamp or 0 if the result can't go anywhere backwards.


GstRtp.RTPBuffer.ext_timestamp

def GstRtp.RTPBuffer.ext_timestamp (exttimestamp, timestamp):
    #python wrapper for 'gst_rtp_buffer_ext_timestamp'

Update the exttimestamp field with the extended timestamp of timestamp For the first call of the method, exttimestamp should point to a location with a value of -1.

This function is able to handle both forward and backward timestamps taking into account:

  • timestamp wraparound making sure that the returned value is properly increased.
  • timestamp unwraparound making sure that the returned value is properly decreased.

Parameters:

exttimestamp ( int ) –

a previous extended timestamp

timestamp ( int ) –

a new timestamp

Returns a tuple made of:

( int ) –

The extended timestamp of timestamp or 0 if the result can't go anywhere backwards.

exttimestamp ( int ) –

The extended timestamp of timestamp or 0 if the result can't go anywhere backwards.


gst_rtp_buffer_get_extension_onebyte_header_from_bytes

gboolean
gst_rtp_buffer_get_extension_onebyte_header_from_bytes (GBytes * bytes,
                                                        guint16 bit_pattern,
                                                        guint8 id,
                                                        guint nth,
                                                        gpointer * data,
                                                        guint * size)

Similar to gst_rtp_buffer_get_extension_onebyte_header, but working on the GBytes you get from gst_rtp_buffer_get_extension_bytes. Parses RFC 5285 style header extensions with a one byte header. It will return the nth extension with the requested id.

Parameters:

bytes

GBytes

bit_pattern

The bit-pattern. Anything but 0xBEDE is rejected.

id

The ID of the header extension to be read (between 1 and 14).

nth

Read the nth extension packet with the requested ID

data

(out) (array length=size) (element-type guint8) (transfer none): location for data

size ( [out] ) –

the size of the data in bytes

Returns

TRUE if bytes had the requested header extension

Since : 1.18


GstRtp.RTPBuffer.prototype.get_extension_onebyte_header_from_bytes

function GstRtp.RTPBuffer.prototype.get_extension_onebyte_header_from_bytes(bytes: GLib.Bytes, bit_pattern: Number, id: Number, nth: Number): {
    // javascript wrapper for 'gst_rtp_buffer_get_extension_onebyte_header_from_bytes'
}

Similar to gst_rtp_buffer_get_extension_onebyte_header, but working on the GLib.Bytes you get from gst_rtp_buffer_get_extension_bytes. Parses RFC 5285 style header extensions with a one byte header. It will return the nth extension with the requested id.

Parameters:

bytes ( GLib.Bytes ) –

GLib.Bytes

bit_pattern ( Number ) –

The bit-pattern. Anything but 0xBEDE is rejected.

id ( Number ) –

The ID of the header extension to be read (between 1 and 14).

nth ( Number ) –

Read the nth extension packet with the requested ID

Returns a tuple made of:

( Number ) –

TRUE if bytes had the requested header extension

data ( [ Number ] ) –

TRUE if bytes had the requested header extension

size ( Number ) –

TRUE if bytes had the requested header extension

Since : 1.18


GstRtp.RTPBuffer.get_extension_onebyte_header_from_bytes

def GstRtp.RTPBuffer.get_extension_onebyte_header_from_bytes (bytes, bit_pattern, id, nth):
    #python wrapper for 'gst_rtp_buffer_get_extension_onebyte_header_from_bytes'

Similar to gst_rtp_buffer_get_extension_onebyte_header, but working on the GLib.Bytes you get from gst_rtp_buffer_get_extension_bytes. Parses RFC 5285 style header extensions with a one byte header. It will return the nth extension with the requested id.

Parameters:

bytes ( GLib.Bytes ) –

GLib.Bytes

bit_pattern ( int ) –

The bit-pattern. Anything but 0xBEDE is rejected.

id ( int ) –

The ID of the header extension to be read (between 1 and 14).

nth ( int ) –

Read the nth extension packet with the requested ID

Returns a tuple made of:

( bool ) –

TRUE if bytes had the requested header extension

data ( [ int ] ) –

TRUE if bytes had the requested header extension

size ( int ) –

TRUE if bytes had the requested header extension

Since : 1.18


gst_rtp_buffer_map

gboolean
gst_rtp_buffer_map (GstBuffer * buffer,
                    GstMapFlags flags,
                    GstRTPBuffer * rtp)

Map the contents of buffer into rtp.

Parameters:

buffer

a GstBuffer

flags

GstMapFlags

rtp ( [out] ) –

a GstRTPBuffer

Returns

TRUE if buffer could be mapped.


GstRtp.RTPBuffer.prototype.map

function GstRtp.RTPBuffer.prototype.map(buffer: Gst.Buffer, flags: Gst.MapFlags): {
    // javascript wrapper for 'gst_rtp_buffer_map'
}

Map the contents of buffer into rtp.

Parameters:

buffer ( Gst.Buffer ) –

a Gst.Buffer

flags ( Gst.MapFlags ) –

Gst.MapFlags

Returns a tuple made of:

( Number ) –

true if buffer could be mapped.

rtp ( GstRtp.RTPBuffer ) –

true if buffer could be mapped.


GstRtp.RTPBuffer.map

def GstRtp.RTPBuffer.map (buffer, flags):
    #python wrapper for 'gst_rtp_buffer_map'

Map the contents of buffer into rtp.

Parameters:

buffer ( Gst.Buffer ) –

a Gst.Buffer

flags ( Gst.MapFlags ) –

Gst.MapFlags

Returns a tuple made of:

( bool ) –

True if buffer could be mapped.

rtp ( GstRtp.RTPBuffer ) –

True if buffer could be mapped.


gst_rtp_buffer_new_allocate

GstBuffer *
gst_rtp_buffer_new_allocate (guint payload_len,
                             guint8 pad_len,
                             guint8 csrc_count)

Allocate a new GstBuffer with enough data to hold an RTP packet with csrc_count CSRCs, a payload length of payload_len and padding of pad_len. All other RTP header fields will be set to 0/FALSE.

Parameters:

payload_len

the length of the payload

pad_len

the amount of padding

csrc_count

the number of CSRC entries

Returns

A newly allocated buffer that can hold an RTP packet with given parameters.


GstRtp.RTPBuffer.prototype.new_allocate

function GstRtp.RTPBuffer.prototype.new_allocate(payload_len: Number, pad_len: Number, csrc_count: Number): {
    // javascript wrapper for 'gst_rtp_buffer_new_allocate'
}

Allocate a new Gst.Buffer with enough data to hold an RTP packet with csrc_count CSRCs, a payload length of payload_len and padding of pad_len. All other RTP header fields will be set to 0/FALSE.

Parameters:

payload_len ( Number ) –

the length of the payload

pad_len ( Number ) –

the amount of padding

csrc_count ( Number ) –

the number of CSRC entries

Returns ( Gst.Buffer ) –

A newly allocated buffer that can hold an RTP packet with given parameters.


GstRtp.RTPBuffer.new_allocate

def GstRtp.RTPBuffer.new_allocate (payload_len, pad_len, csrc_count):
    #python wrapper for 'gst_rtp_buffer_new_allocate'

Allocate a new Gst.Buffer with enough data to hold an RTP packet with csrc_count CSRCs, a payload length of payload_len and padding of pad_len. All other RTP header fields will be set to 0/FALSE.

Parameters:

payload_len ( int ) –

the length of the payload

pad_len ( int ) –

the amount of padding

csrc_count ( int ) –

the number of CSRC entries

Returns ( Gst.Buffer ) –

A newly allocated buffer that can hold an RTP packet with given parameters.


gst_rtp_buffer_new_allocate_len

GstBuffer *
gst_rtp_buffer_new_allocate_len (guint packet_len,
                                 guint8 pad_len,
                                 guint8 csrc_count)

Create a new GstBuffer that can hold an RTP packet that is exactly packet_len long. The length of the payload depends on pad_len and csrc_count and can be calculated with gst_rtp_buffer_calc_payload_len. All RTP header fields will be set to 0/FALSE.

Parameters:

packet_len

the total length of the packet

pad_len

the amount of padding

csrc_count

the number of CSRC entries

Returns

A newly allocated buffer that can hold an RTP packet of packet_len.


GstRtp.RTPBuffer.prototype.new_allocate_len

function GstRtp.RTPBuffer.prototype.new_allocate_len(packet_len: Number, pad_len: Number, csrc_count: Number): {
    // javascript wrapper for 'gst_rtp_buffer_new_allocate_len'
}

Create a new Gst.Buffer that can hold an RTP packet that is exactly packet_len long. The length of the payload depends on pad_len and csrc_count and can be calculated with GstRtp.RTPBuffer.prototype.calc_payload_len. All RTP header fields will be set to 0/FALSE.

Parameters:

packet_len ( Number ) –

the total length of the packet

pad_len ( Number ) –

the amount of padding

csrc_count ( Number ) –

the number of CSRC entries

Returns ( Gst.Buffer ) –

A newly allocated buffer that can hold an RTP packet of packet_len.


GstRtp.RTPBuffer.new_allocate_len

def GstRtp.RTPBuffer.new_allocate_len (packet_len, pad_len, csrc_count):
    #python wrapper for 'gst_rtp_buffer_new_allocate_len'

Create a new Gst.Buffer that can hold an RTP packet that is exactly packet_len long. The length of the payload depends on pad_len and csrc_count and can be calculated with GstRtp.RTPBuffer.calc_payload_len. All RTP header fields will be set to 0/FALSE.

Parameters:

packet_len ( int ) –

the total length of the packet

pad_len ( int ) –

the amount of padding

csrc_count ( int ) –

the number of CSRC entries

Returns ( Gst.Buffer ) –

A newly allocated buffer that can hold an RTP packet of packet_len.


gst_rtp_buffer_new_copy_data

GstBuffer *
gst_rtp_buffer_new_copy_data (gconstpointer data,
                              gsize len)

Create a new buffer and set the data to a copy of len bytes of data and the size to len. The data will be freed when the buffer is freed.

Parameters:

data ( [array length=len] [element-type guint8] ) –

data for the new buffer

len

the length of data

Returns

A newly allocated buffer with a copy of data and of size len.


GstRtp.RTPBuffer.prototype.new_copy_data

function GstRtp.RTPBuffer.prototype.new_copy_data(data: [ Number ], len: Number): {
    // javascript wrapper for 'gst_rtp_buffer_new_copy_data'
}

Create a new buffer and set the data to a copy of len bytes of data and the size to len. The data will be freed when the buffer is freed.

Parameters:

data ( [ Number ] ) –

data for the new buffer

len ( Number ) –

the length of data

Returns ( Gst.Buffer ) –

A newly allocated buffer with a copy of data and of size len.


GstRtp.RTPBuffer.new_copy_data

def GstRtp.RTPBuffer.new_copy_data (data, len):
    #python wrapper for 'gst_rtp_buffer_new_copy_data'

Create a new buffer and set the data to a copy of len bytes of data and the size to len. The data will be freed when the buffer is freed.

Parameters:

data ( [ int ] ) –

data for the new buffer

len ( int ) –

the length of data

Returns ( Gst.Buffer ) –

A newly allocated buffer with a copy of data and of size len.


gst_rtp_buffer_new_take_data

GstBuffer *
gst_rtp_buffer_new_take_data (gpointer data,
                              gsize len)

Create a new buffer and set the data and size of the buffer to data and len respectively. data will be freed when the buffer is unreffed, so this function transfers ownership of data to the new buffer.

Parameters:

data

(array length=len) (transfer full) (element-type guint8): data for the new buffer

len

the length of data

Returns

A newly allocated buffer with data and of size len.


GstRtp.RTPBuffer.prototype.new_take_data

function GstRtp.RTPBuffer.prototype.new_take_data(data: [ Number ], len: Number): {
    // javascript wrapper for 'gst_rtp_buffer_new_take_data'
}

Create a new buffer and set the data and size of the buffer to data and len respectively. data will be freed when the buffer is unreffed, so this function transfers ownership of data to the new buffer.

Parameters:

data ( [ Number ] ) –

(array length=len) (transfer full) (element-type guint8): data for the new buffer

len ( Number ) –

the length of data

Returns ( Gst.Buffer ) –

A newly allocated buffer with data and of size len.


GstRtp.RTPBuffer.new_take_data

def GstRtp.RTPBuffer.new_take_data (data, len):
    #python wrapper for 'gst_rtp_buffer_new_take_data'

Create a new buffer and set the data and size of the buffer to data and len respectively. data will be freed when the buffer is unreffed, so this function transfers ownership of data to the new buffer.

Parameters:

data ( [ int ] ) –

(array length=len) (transfer full) (element-type guint8): data for the new buffer

len ( int ) –

the length of data

Returns ( Gst.Buffer ) –

A newly allocated buffer with data and of size len.


Enumerations

GstRTPBufferFlags

Additional RTP buffer flags. These flags can potentially be used on any buffers carrying RTP packets.

Note that these are only valid for GstCaps of type: application/x-rtp (x-rtcp). They can conflict with other extended buffer flags.

Members
GST_RTP_BUFFER_FLAG_RETRANSMISSION (1048576) –

The GstBuffer was once wrapped in a retransmitted packet as specified by RFC 4588.

GST_RTP_BUFFER_FLAG_REDUNDANT (2097152) –

The packet represents redundant RTP packet. The flag is used in gstrtpstorage to be able to hold the packetback and use it only for recovery from packet loss. Since: 1.14

GST_RTP_BUFFER_FLAG_LAST (268435456) –

Offset to define more flags.


GstRtp.RTPBufferFlags

Additional RTP buffer flags. These flags can potentially be used on any buffers carrying RTP packets.

Note that these are only valid for Gst.Caps of type: application/x-rtp (x-rtcp). They can conflict with other extended buffer flags.

Members
GstRtp.RTPBufferFlags.RETRANSMISSION (1048576) –

The Gst.Buffer was once wrapped in a retransmitted packet as specified by RFC 4588.

GstRtp.RTPBufferFlags.REDUNDANT (2097152) –

The packet represents redundant RTP packet. The flag is used in gstrtpstorage to be able to hold the packetback and use it only for recovery from packet loss. Since: 1.14

GstRtp.RTPBufferFlags.LAST (268435456) –

Offset to define more flags.


GstRtp.RTPBufferFlags

Additional RTP buffer flags. These flags can potentially be used on any buffers carrying RTP packets.

Note that these are only valid for Gst.Caps of type: application/x-rtp (x-rtcp). They can conflict with other extended buffer flags.

Members
GstRtp.RTPBufferFlags.RETRANSMISSION (1048576) –

The Gst.Buffer was once wrapped in a retransmitted packet as specified by RFC 4588.

GstRtp.RTPBufferFlags.REDUNDANT (2097152) –

The packet represents redundant RTP packet. The flag is used in gstrtpstorage to be able to hold the packetback and use it only for recovery from packet loss. Since: 1.14

GstRtp.RTPBufferFlags.LAST (268435456) –

Offset to define more flags.


GstRTPBufferMapFlags

Additional mapping flags for gst_rtp_buffer_map.

Members
GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING (65536) –

Skip mapping and validation of RTP padding and RTP pad count when present. Useful for buffers where the padding may be encrypted.

GST_RTP_BUFFER_MAP_FLAG_LAST (16777216) –

Offset to define more flags


GstRtp.RTPBufferMapFlags

Additional mapping flags for GstRtp.RTPBuffer.prototype.map.

Members
GstRtp.RTPBufferMapFlags.SKIP_PADDING (65536) –

Skip mapping and validation of RTP padding and RTP pad count when present. Useful for buffers where the padding may be encrypted.

GstRtp.RTPBufferMapFlags.LAST (16777216) –

Offset to define more flags


GstRtp.RTPBufferMapFlags

Additional mapping flags for GstRtp.RTPBuffer.map.

Members
GstRtp.RTPBufferMapFlags.SKIP_PADDING (65536) –

Skip mapping and validation of RTP padding and RTP pad count when present. Useful for buffers where the padding may be encrypted.

GstRtp.RTPBufferMapFlags.LAST (16777216) –

Offset to define more flags


Constants

GST_RTP_BUFFER_INIT

#define GST_RTP_BUFFER_INIT { NULL, 0, { NULL, NULL, NULL, NULL}, { 0, 0, 0, 0 }, \
  { GST_MAP_INFO_INIT, GST_MAP_INFO_INIT, GST_MAP_INFO_INIT, GST_MAP_INFO_INIT} }

GST_RTP_VERSION

#define GST_RTP_VERSION 2

The supported RTP version 2.


GstRtp.RTP_VERSION

The supported RTP version 2.


GstRtp.RTP_VERSION

The supported RTP version 2.


The results of the search are