GstRTPBasePayload

Provides a base class for RTP payloaders

GstRTPBasePayload

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──GstRTPBasePayload
                    ╰──GstRTPBaseAudioPayload

Members

element (GstElement) –
No description available

Class structure

GstRTPBasePayloadClass

Base class for audio RTP payloader.

Fields
parent_class (GstElementClass) –

the parent class


GstRtp.RTPBasePayloadClass

Base class for audio RTP payloader.

Attributes
parent_class (Gst.ElementClass) –

the parent class


GstRtp.RTPBasePayloadClass

Base class for audio RTP payloader.

Attributes
parent_class (Gst.ElementClass) –

the parent class


GstRTPBasePayload

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──GstRTPBasePayload
                    ╰──GstRTPBaseAudioPayload

Members

element (GstElement) –
No description available

GstRTPBasePayload

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──GstRTPBasePayload
                    ╰──GstRTPBaseAudioPayload

Members

element (GstElement) –
No description available

Methods

gst_rtp_base_payload_allocate_output_buffer

GstBuffer *
gst_rtp_base_payload_allocate_output_buffer (GstRTPBasePayload * payload,
                                             guint payload_len,
                                             guint8 pad_len,
                                             guint8 csrc_count)

Allocate a new GstBuffer with enough data to hold an RTP packet with minimum csrc_count CSRCs, a payload length of payload_len and padding of pad_len. If payload has source-info TRUE additional CSRCs may be allocated and filled with RTP source information.

Parameters:

payload

a GstRTPBasePayload

payload_len

the length of the payload

pad_len

the amount of padding

csrc_count

the minimum number of CSRC entries

Returns

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

Since : 1.16


GstRtp.RTPBasePayload.prototype.allocate_output_buffer

function GstRtp.RTPBasePayload.prototype.allocate_output_buffer(payload_len: Number, pad_len: Number, csrc_count: Number): {
    // javascript wrapper for 'gst_rtp_base_payload_allocate_output_buffer'
}

Allocate a new Gst.Buffer with enough data to hold an RTP packet with minimum csrc_count CSRCs, a payload length of payload_len and padding of pad_len. If payload has source-info true additional CSRCs may be allocated and filled with RTP source information.

Parameters:

payload_len ( Number ) –

the length of the payload

pad_len ( Number ) –

the amount of padding

csrc_count ( Number ) –

the minimum number of CSRC entries

Returns ( Gst.Buffer ) –

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

Since : 1.16


GstRtp.RTPBasePayload.allocate_output_buffer

def GstRtp.RTPBasePayload.allocate_output_buffer (self, payload_len, pad_len, csrc_count):
    #python wrapper for 'gst_rtp_base_payload_allocate_output_buffer'

Allocate a new Gst.Buffer with enough data to hold an RTP packet with minimum csrc_count CSRCs, a payload length of payload_len and padding of pad_len. If payload has source_info True additional CSRCs may be allocated and filled with RTP source information.

Parameters:

payload_len ( int ) –

the length of the payload

pad_len ( int ) –

the amount of padding

csrc_count ( int ) –

the minimum number of CSRC entries

Returns ( Gst.Buffer ) –

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

Since : 1.16


gst_rtp_base_payload_get_source_count

guint
gst_rtp_base_payload_get_source_count (GstRTPBasePayload * payload,
                                       GstBuffer * buffer)

Count the total number of RTP sources found in the meta of buffer, which will be automically added by gst_rtp_base_payload_allocate_output_buffer. If source-info is FALSE the count will be 0.

Parameters:

payload

a GstRTPBasePayload

buffer ( [transfer: none] ) –

a GstBuffer, typically the buffer to payload

Returns

The number of sources.

Since : 1.16


GstRtp.RTPBasePayload.prototype.get_source_count

function GstRtp.RTPBasePayload.prototype.get_source_count(buffer: Gst.Buffer): {
    // javascript wrapper for 'gst_rtp_base_payload_get_source_count'
}

Count the total number of RTP sources found in the meta of buffer, which will be automically added by GstRtp.RTPBasePayload.prototype.allocate_output_buffer. If source-info is false the count will be 0.

Parameters:

buffer ( Gst.Buffer ) –

a Gst.Buffer, typically the buffer to payload

Returns ( Number ) –

The number of sources.

Since : 1.16


GstRtp.RTPBasePayload.get_source_count

def GstRtp.RTPBasePayload.get_source_count (self, buffer):
    #python wrapper for 'gst_rtp_base_payload_get_source_count'

Count the total number of RTP sources found in the meta of buffer, which will be automically added by GstRtp.RTPBasePayload.allocate_output_buffer. If source_info is False the count will be 0.

Parameters:

buffer ( Gst.Buffer ) –

a Gst.Buffer, typically the buffer to payload

Returns ( int ) –

The number of sources.

Since : 1.16


gst_rtp_base_payload_is_filled

gboolean
gst_rtp_base_payload_is_filled (GstRTPBasePayload * payload,
                                guint size,
                                GstClockTime duration)

Check if the packet with size and duration would exceed the configured maximum size.

Parameters:

payload

a GstRTPBasePayload

size

the size of the packet

duration

the duration of the packet

Returns

TRUE if the packet of size and duration would exceed the configured MTU or max_ptime.


GstRtp.RTPBasePayload.prototype.is_filled

function GstRtp.RTPBasePayload.prototype.is_filled(size: Number, duration: Number): {
    // javascript wrapper for 'gst_rtp_base_payload_is_filled'
}

Check if the packet with size and duration would exceed the configured maximum size.

Parameters:

size ( Number ) –

the size of the packet

duration ( Number ) –

the duration of the packet

Returns ( Number ) –

true if the packet of size and duration would exceed the configured MTU or max_ptime.


GstRtp.RTPBasePayload.is_filled

def GstRtp.RTPBasePayload.is_filled (self, size, duration):
    #python wrapper for 'gst_rtp_base_payload_is_filled'

Check if the packet with size and duration would exceed the configured maximum size.

Parameters:

size ( int ) –

the size of the packet

duration ( int ) –

the duration of the packet

Returns ( bool ) –

True if the packet of size and duration would exceed the configured MTU or max_ptime.


gst_rtp_base_payload_is_source_info_enabled

gboolean
gst_rtp_base_payload_is_source_info_enabled (GstRTPBasePayload * payload)

Queries whether the payloader will add contributing sources (CSRCs) to the RTP header from GstRTPSourceMeta.

Parameters:

payload

a GstRTPBasePayload

Returns

TRUE if source-info is enabled.

Since : 1.16


GstRtp.RTPBasePayload.prototype.is_source_info_enabled

function GstRtp.RTPBasePayload.prototype.is_source_info_enabled(): {
    // javascript wrapper for 'gst_rtp_base_payload_is_source_info_enabled'
}

Queries whether the payloader will add contributing sources (CSRCs) to the RTP header from GstRtp.RTPSourceMeta.

Parameters:

Returns ( Number ) –

true if source-info is enabled.

Since : 1.16


GstRtp.RTPBasePayload.is_source_info_enabled

def GstRtp.RTPBasePayload.is_source_info_enabled (self):
    #python wrapper for 'gst_rtp_base_payload_is_source_info_enabled'

Queries whether the payloader will add contributing sources (CSRCs) to the RTP header from GstRtp.RTPSourceMeta.

Parameters:

Returns ( bool ) –

True if source-info is enabled.

Since : 1.16


gst_rtp_base_payload_push

GstFlowReturn
gst_rtp_base_payload_push (GstRTPBasePayload * payload,
                           GstBuffer * buffer)

Push buffer to the peer element of the payloader. The SSRC, payload type, seqnum and timestamp of the RTP buffer will be updated first.

This function takes ownership of buffer.

Parameters:

payload

a GstRTPBasePayload

buffer

a GstBuffer

Returns

a GstFlowReturn.


GstRtp.RTPBasePayload.prototype.push

function GstRtp.RTPBasePayload.prototype.push(buffer: Gst.Buffer): {
    // javascript wrapper for 'gst_rtp_base_payload_push'
}

Push buffer to the peer element of the payloader. The SSRC, payload type, seqnum and timestamp of the RTP buffer will be updated first.

This function takes ownership of buffer.

Parameters:

buffer ( Gst.Buffer ) –

a Gst.Buffer

Returns ( Gst.FlowReturn ) –

a Gst.FlowReturn.


GstRtp.RTPBasePayload.push

def GstRtp.RTPBasePayload.push (self, buffer):
    #python wrapper for 'gst_rtp_base_payload_push'

Push buffer to the peer element of the payloader. The SSRC, payload type, seqnum and timestamp of the RTP buffer will be updated first.

This function takes ownership of buffer.

Parameters:

buffer ( Gst.Buffer ) –

a Gst.Buffer

Returns ( Gst.FlowReturn ) –

a Gst.FlowReturn.


gst_rtp_base_payload_push_list

GstFlowReturn
gst_rtp_base_payload_push_list (GstRTPBasePayload * payload,
                                GstBufferList * list)

Push list to the peer element of the payloader. The SSRC, payload type, seqnum and timestamp of the RTP buffer will be updated first.

This function takes ownership of list.

Parameters:

payload

a GstRTPBasePayload

list

a GstBufferList

Returns

a GstFlowReturn.


GstRtp.RTPBasePayload.prototype.push_list

function GstRtp.RTPBasePayload.prototype.push_list(list: Gst.BufferList): {
    // javascript wrapper for 'gst_rtp_base_payload_push_list'
}

Push list to the peer element of the payloader. The SSRC, payload type, seqnum and timestamp of the RTP buffer will be updated first.

This function takes ownership of list.

Parameters:

Returns ( Gst.FlowReturn ) –

a Gst.FlowReturn.


GstRtp.RTPBasePayload.push_list

def GstRtp.RTPBasePayload.push_list (self, list):
    #python wrapper for 'gst_rtp_base_payload_push_list'

Push list to the peer element of the payloader. The SSRC, payload type, seqnum and timestamp of the RTP buffer will be updated first.

This function takes ownership of list.

Parameters:

Returns ( Gst.FlowReturn ) –

a Gst.FlowReturn.


gst_rtp_base_payload_set_options

gst_rtp_base_payload_set_options (GstRTPBasePayload * payload,
                                  const gchar * media,
                                  gboolean dynamic,
                                  const gchar * encoding_name,
                                  guint32 clock_rate)

Set the rtp options of the payloader. These options will be set in the caps of the payloader. Subclasses must call this method before calling gst_rtp_base_payload_push or gst_rtp_base_payload_set_outcaps.

Parameters:

payload

a GstRTPBasePayload

media

the media type (typically "audio" or "video")

dynamic

if the payload type is dynamic

encoding_name

the encoding name

clock_rate

the clock rate of the media


GstRtp.RTPBasePayload.prototype.set_options

function GstRtp.RTPBasePayload.prototype.set_options(media: String, dynamic: Number, encoding_name: String, clock_rate: Number): {
    // javascript wrapper for 'gst_rtp_base_payload_set_options'
}

Set the rtp options of the payloader. These options will be set in the caps of the payloader. Subclasses must call this method before calling GstRtp.RTPBasePayload.prototype.push or gst_rtp_base_payload_set_outcaps (not introspectable).

Parameters:

media ( String ) –

the media type (typically "audio" or "video")

dynamic ( Number ) –

if the payload type is dynamic

encoding_name ( String ) –

the encoding name

clock_rate ( Number ) –

the clock rate of the media


GstRtp.RTPBasePayload.set_options

def GstRtp.RTPBasePayload.set_options (self, media, dynamic, encoding_name, clock_rate):
    #python wrapper for 'gst_rtp_base_payload_set_options'

Set the rtp options of the payloader. These options will be set in the caps of the payloader. Subclasses must call this method before calling GstRtp.RTPBasePayload.push or gst_rtp_base_payload_set_outcaps (not introspectable).

Parameters:

media ( str ) –

the media type (typically "audio" or "video")

dynamic ( bool ) –

if the payload type is dynamic

encoding_name ( str ) –

the encoding name

clock_rate ( int ) –

the clock rate of the media


gst_rtp_base_payload_set_outcaps

gboolean
gst_rtp_base_payload_set_outcaps (GstRTPBasePayload * payload,
                                  const gchar * fieldname,
                                  ... ...)

Configure the output caps with the optional parameters.

Variable arguments should be in the form field name, field type (as a GType), value(s). The last variable argument should be NULL.

Parameters:

payload

a GstRTPBasePayload

fieldname

the first field name or NULL

...

field values

Returns

TRUE if the caps could be set.


gst_rtp_base_payload_set_source_info_enabled

gst_rtp_base_payload_set_source_info_enabled (GstRTPBasePayload * payload,
                                              gboolean enable)

Enable or disable adding contributing sources to RTP packets from GstRTPSourceMeta.

Parameters:

payload

a GstRTPBasePayload

enable

whether to add contributing sources to RTP packets

Since : 1.16


GstRtp.RTPBasePayload.prototype.set_source_info_enabled

function GstRtp.RTPBasePayload.prototype.set_source_info_enabled(enable: Number): {
    // javascript wrapper for 'gst_rtp_base_payload_set_source_info_enabled'
}

Enable or disable adding contributing sources to RTP packets from GstRtp.RTPSourceMeta.

Parameters:

enable ( Number ) –

whether to add contributing sources to RTP packets

Since : 1.16


GstRtp.RTPBasePayload.set_source_info_enabled

def GstRtp.RTPBasePayload.set_source_info_enabled (self, enable):
    #python wrapper for 'gst_rtp_base_payload_set_source_info_enabled'

Enable or disable adding contributing sources to RTP packets from GstRtp.RTPSourceMeta.

Parameters:

enable ( bool ) –

whether to add contributing sources to RTP packets

Since : 1.16


Properties

max-ptime

“max-ptime” gint64

Flags : Read / Write


max-ptime

“max-ptime” Number

Flags : Read / Write


max_ptime

“self.props.max_ptime” int

Flags : Read / Write


min-ptime

“min-ptime” gint64

Minimum duration of the packet data in ns (can't go above MTU)

Flags : Read / Write


min-ptime

“min-ptime” Number

Minimum duration of the packet data in ns (can't go above MTU)

Flags : Read / Write


min_ptime

“self.props.min_ptime” int

Minimum duration of the packet data in ns (can't go above MTU)

Flags : Read / Write


mtu

“mtu” guint

Flags : Read / Write


mtu

“mtu” Number

Flags : Read / Write


mtu

“self.props.mtu” int

Flags : Read / Write


onvif-no-rate-control

“onvif-no-rate-control” gboolean

Make the payloader timestamp packets according to the Rate-Control=no behaviour specified in the ONVIF replay spec.

Flags : Read / Write


onvif-no-rate-control

“onvif-no-rate-control” Number

Make the payloader timestamp packets according to the Rate-Control=no behaviour specified in the ONVIF replay spec.

Flags : Read / Write


onvif_no_rate_control

“self.props.onvif_no_rate_control” bool

Make the payloader timestamp packets according to the Rate-Control=no behaviour specified in the ONVIF replay spec.

Flags : Read / Write


perfect-rtptime

“perfect-rtptime” gboolean

Try to use the offset fields to generate perfect RTP timestamps. When this option is disabled, RTP timestamps are generated from GST_BUFFER_PTS of each payloaded buffer. The PTSes of buffers may not necessarily increment with the amount of data in each input buffer, consider e.g. the case where the buffer arrives from a network which means that the PTS is unrelated to the amount of data. Because the RTP timestamps are generated from GST_BUFFER_PTS this can result in RTP timestamps that also don't increment with the amount of data in the payloaded packet. To circumvent this it is possible to set the perfect rtptime option enabled. When this option is enabled the payloader will increment the RTP timestamps based on GST_BUFFER_OFFSET which relates to the amount of data in each packet rather than the GST_BUFFER_PTS of each buffer and therefore the RTP timestamps will more closely correlate with the amount of data in each buffer. Currently GstRTPBasePayload is limited to handling perfect RTP timestamps for audio streams.

Flags : Read / Write


perfect-rtptime

“perfect-rtptime” Number

Try to use the offset fields to generate perfect RTP timestamps. When this option is disabled, RTP timestamps are generated from GST_BUFFER_PTS of each payloaded buffer. The PTSes of buffers may not necessarily increment with the amount of data in each input buffer, consider e.g. the case where the buffer arrives from a network which means that the PTS is unrelated to the amount of data. Because the RTP timestamps are generated from GST_BUFFER_PTS this can result in RTP timestamps that also don't increment with the amount of data in the payloaded packet. To circumvent this it is possible to set the perfect rtptime option enabled. When this option is enabled the payloader will increment the RTP timestamps based on GST_BUFFER_OFFSET which relates to the amount of data in each packet rather than the GST_BUFFER_PTS of each buffer and therefore the RTP timestamps will more closely correlate with the amount of data in each buffer. Currently GstRTPBasePayload is limited to handling perfect RTP timestamps for audio streams.

Flags : Read / Write


perfect_rtptime

“self.props.perfect_rtptime” bool

Try to use the offset fields to generate perfect RTP timestamps. When this option is disabled, RTP timestamps are generated from GST_BUFFER_PTS of each payloaded buffer. The PTSes of buffers may not necessarily increment with the amount of data in each input buffer, consider e.g. the case where the buffer arrives from a network which means that the PTS is unrelated to the amount of data. Because the RTP timestamps are generated from GST_BUFFER_PTS this can result in RTP timestamps that also don't increment with the amount of data in the payloaded packet. To circumvent this it is possible to set the perfect rtptime option enabled. When this option is enabled the payloader will increment the RTP timestamps based on GST_BUFFER_OFFSET which relates to the amount of data in each packet rather than the GST_BUFFER_PTS of each buffer and therefore the RTP timestamps will more closely correlate with the amount of data in each buffer. Currently GstRTPBasePayload is limited to handling perfect RTP timestamps for audio streams.

Flags : Read / Write


pt

“pt” guint

Flags : Read / Write


pt

“pt” Number

Flags : Read / Write


pt

“self.props.pt” int

Flags : Read / Write


ptime-multiple

“ptime-multiple” gint64

Force buffers to be multiples of this duration in ns (0 disables)

Flags : Read / Write


ptime-multiple

“ptime-multiple” Number

Force buffers to be multiples of this duration in ns (0 disables)

Flags : Read / Write


ptime_multiple

“self.props.ptime_multiple” int

Force buffers to be multiples of this duration in ns (0 disables)

Flags : Read / Write


scale-rtptime

“scale-rtptime” gboolean

Make the RTP packets' timestamps be scaled with the segment's rate (corresponding to RTSP speed parameter). Disabling this property means the timestamps will not be affected by the set delivery speed (RTSP speed).

Example: A server wants to allow streaming a recorded video in double speed but still have the timestamps correspond to the position in the video. This is achieved by the client setting RTSP Speed to 2 while the server has this property disabled.

Flags : Read / Write


scale-rtptime

“scale-rtptime” Number

Make the RTP packets' timestamps be scaled with the segment's rate (corresponding to RTSP speed parameter). Disabling this property means the timestamps will not be affected by the set delivery speed (RTSP speed).

Example: A server wants to allow streaming a recorded video in double speed but still have the timestamps correspond to the position in the video. This is achieved by the client setting RTSP Speed to 2 while the server has this property disabled.

Flags : Read / Write


scale_rtptime

“self.props.scale_rtptime” bool

Make the RTP packets' timestamps be scaled with the segment's rate (corresponding to RTSP speed parameter). Disabling this property means the timestamps will not be affected by the set delivery speed (RTSP speed).

Example: A server wants to allow streaming a recorded video in double speed but still have the timestamps correspond to the position in the video. This is achieved by the client setting RTSP Speed to 2 while the server has this property disabled.

Flags : Read / Write


seqnum

“seqnum” guint

Flags : Read


seqnum

“seqnum” Number

Flags : Read


seqnum

“self.props.seqnum” int

Flags : Read


seqnum-offset

“seqnum-offset” gint

Flags : Read / Write


seqnum-offset

“seqnum-offset” Number

Flags : Read / Write


seqnum_offset

“self.props.seqnum_offset” int

Flags : Read / Write


source-info

“source-info” gboolean

Enable writing the CSRC field in allocated RTP header based on RTP source information found in the input buffer's GstRTPSourceMeta.

Flags : Read / Write


source-info

“source-info” Number

Enable writing the CSRC field in allocated RTP header based on RTP source information found in the input buffer's GstRtp.RTPSourceMeta.

Flags : Read / Write


source_info

“self.props.source_info” bool

Enable writing the CSRC field in allocated RTP header based on RTP source information found in the input buffer's GstRtp.RTPSourceMeta.

Flags : Read / Write


ssrc

“ssrc” guint

Flags : Read / Write


ssrc

“ssrc” Number

Flags : Read / Write


ssrc

“self.props.ssrc” int

Flags : Read / Write


stats

“stats” GstStructure *

Various payloader statistics retrieved atomically (and are therefore synchroized with each other), these can be used e.g. to generate an RTP-Info header. This property return a GstStructure named application/x-rtp-payload-stats containing the following fields relating to the last processed buffer and current state of the stream being payloaded:

  • clock-rate :#G_TYPE_UINT, clock-rate of the stream
  • running-time :#G_TYPE_UINT64, running time
  • seqnum :#G_TYPE_UINT, sequence number, same as seqnum
  • timestamp :#G_TYPE_UINT, RTP timestamp, same as timestamp
  • ssrc :#G_TYPE_UINT, The SSRC in use
  • pt :#G_TYPE_UINT, The Payload type in use, same as pt
  • seqnum-offset :#G_TYPE_UINT, The current offset added to the seqnum
  • timestamp-offset :#G_TYPE_UINT, The current offset added to the timestamp

Flags : Read


stats

“stats” Gst.Structure

Various payloader statistics retrieved atomically (and are therefore synchroized with each other), these can be used e.g. to generate an RTP-Info header. This property return a GstStructure named application/x-rtp-payload-stats containing the following fields relating to the last processed buffer and current state of the stream being payloaded:

  • clock-rate :#G_TYPE_UINT, clock-rate of the stream
  • running-time :#G_TYPE_UINT64, running time
  • seqnum :#G_TYPE_UINT, sequence number, same as seqnum
  • timestamp :#G_TYPE_UINT, RTP timestamp, same as timestamp
  • ssrc :#G_TYPE_UINT, The SSRC in use
  • pt :#G_TYPE_UINT, The Payload type in use, same as pt
  • seqnum-offset :#G_TYPE_UINT, The current offset added to the seqnum
  • timestamp-offset :#G_TYPE_UINT, The current offset added to the timestamp

Flags : Read


stats

“self.props.stats” Gst.Structure

Various payloader statistics retrieved atomically (and are therefore synchroized with each other), these can be used e.g. to generate an RTP-Info header. This property return a GstStructure named application/x-rtp-payload-stats containing the following fields relating to the last processed buffer and current state of the stream being payloaded:

  • clock-rate :#G_TYPE_UINT, clock-rate of the stream
  • running-time :#G_TYPE_UINT64, running time
  • seqnum :#G_TYPE_UINT, sequence number, same as seqnum
  • timestamp :#G_TYPE_UINT, RTP timestamp, same as timestamp
  • ssrc :#G_TYPE_UINT, The SSRC in use
  • pt :#G_TYPE_UINT, The Payload type in use, same as pt
  • seqnum-offset :#G_TYPE_UINT, The current offset added to the seqnum
  • timestamp-offset :#G_TYPE_UINT, The current offset added to the timestamp

Flags : Read


timestamp

“timestamp” guint

Flags : Read


timestamp

“timestamp” Number

Flags : Read


timestamp

“self.props.timestamp” int

Flags : Read


timestamp-offset

“timestamp-offset” guint

Flags : Read / Write


timestamp-offset

“timestamp-offset” Number

Flags : Read / Write


timestamp_offset

“self.props.timestamp_offset” int

Flags : Read / Write


Virtual Methods

get_caps

GstCaps *
get_caps (GstRTPBasePayload * payload,
          GstPad * pad,
          GstCaps * filter)

get desired caps

Parameters:

payload
No description available
pad
No description available
filter
No description available
Returns
No description available

get_caps

function get_caps(payload: GstRtp.RTPBasePayload, pad: Gst.Pad, filter: Gst.Caps): {
    // javascript implementation of the 'get_caps' virtual method
}

get desired caps

Parameters:

payload ( GstRtp.RTPBasePayload ) –
No description available
pad ( Gst.Pad ) –
No description available
filter ( Gst.Caps ) –
No description available
Returns ( Gst.Caps ) –
No description available

get_caps

def get_caps (payload, pad, filter):
    #python implementation of the 'get_caps' virtual method

get desired caps

Parameters:

payload ( GstRtp.RTPBasePayload ) –
No description available
pad ( Gst.Pad ) –
No description available
filter ( Gst.Caps ) –
No description available
Returns ( Gst.Caps ) –
No description available

handle_buffer

GstFlowReturn
handle_buffer (GstRTPBasePayload * payload,
               GstBuffer * buffer)

process data

Parameters:

payload
No description available
buffer
No description available
Returns
No description available

handle_buffer

function handle_buffer(payload: GstRtp.RTPBasePayload, buffer: Gst.Buffer): {
    // javascript implementation of the 'handle_buffer' virtual method
}

process data

Parameters:

payload ( GstRtp.RTPBasePayload ) –
No description available
buffer ( Gst.Buffer ) –
No description available
Returns ( Gst.FlowReturn ) –
No description available

handle_buffer

def handle_buffer (payload, buffer):
    #python implementation of the 'handle_buffer' virtual method

process data

Parameters:

payload ( GstRtp.RTPBasePayload ) –
No description available
buffer ( Gst.Buffer ) –
No description available
Returns ( Gst.FlowReturn ) –
No description available

query

gboolean
query (GstRTPBasePayload * payload,
       GstPad * pad,
       GstQuery * query)

custom query handling

Parameters:

payload
No description available
pad
No description available
query
No description available
Returns
No description available

query

function query(payload: GstRtp.RTPBasePayload, pad: Gst.Pad, query: Gst.Query): {
    // javascript implementation of the 'query' virtual method
}

custom query handling

Parameters:

payload ( GstRtp.RTPBasePayload ) –
No description available
pad ( Gst.Pad ) –
No description available
query ( Gst.Query ) –
No description available
Returns ( Number ) –
No description available

query

def query (payload, pad, query):
    #python implementation of the 'query' virtual method

custom query handling

Parameters:

payload ( GstRtp.RTPBasePayload ) –
No description available
pad ( Gst.Pad ) –
No description available
query ( Gst.Query ) –
No description available
Returns ( bool ) –
No description available

set_caps

gboolean
set_caps (GstRTPBasePayload * payload,
          GstCaps * caps)

configure the payloader

Parameters:

payload
No description available
caps
No description available
Returns
No description available

set_caps

function set_caps(payload: GstRtp.RTPBasePayload, caps: Gst.Caps): {
    // javascript implementation of the 'set_caps' virtual method
}

configure the payloader

Parameters:

payload ( GstRtp.RTPBasePayload ) –
No description available
caps ( Gst.Caps ) –
No description available
Returns ( Number ) –
No description available

set_caps

def set_caps (payload, caps):
    #python implementation of the 'set_caps' virtual method

configure the payloader

Parameters:

payload ( GstRtp.RTPBasePayload ) –
No description available
caps ( Gst.Caps ) –
No description available
Returns ( bool ) –
No description available

sink_event

gboolean
sink_event (GstRTPBasePayload * payload,
            GstEvent * event)

custom event handling on the sinkpad

Parameters:

payload
No description available
event
No description available
Returns
No description available

sink_event

function sink_event(payload: GstRtp.RTPBasePayload, event: Gst.Event): {
    // javascript implementation of the 'sink_event' virtual method
}

custom event handling on the sinkpad

Parameters:

payload ( GstRtp.RTPBasePayload ) –
No description available
event ( Gst.Event ) –
No description available
Returns ( Number ) –
No description available

sink_event

def sink_event (payload, event):
    #python implementation of the 'sink_event' virtual method

custom event handling on the sinkpad

Parameters:

payload ( GstRtp.RTPBasePayload ) –
No description available
event ( Gst.Event ) –
No description available
Returns ( bool ) –
No description available

src_event

gboolean
src_event (GstRTPBasePayload * payload,
           GstEvent * event)

custom event handling on the srcpad

Parameters:

payload
No description available
event
No description available
Returns
No description available

src_event

function src_event(payload: GstRtp.RTPBasePayload, event: Gst.Event): {
    // javascript implementation of the 'src_event' virtual method
}

custom event handling on the srcpad

Parameters:

payload ( GstRtp.RTPBasePayload ) –
No description available
event ( Gst.Event ) –
No description available
Returns ( Number ) –
No description available

src_event

def src_event (payload, event):
    #python implementation of the 'src_event' virtual method

custom event handling on the srcpad

Parameters:

payload ( GstRtp.RTPBasePayload ) –
No description available
event ( Gst.Event ) –
No description available
Returns ( bool ) –
No description available

Function Macros

GST_RTP_BASE_PAYLOAD_CAST

#define GST_RTP_BASE_PAYLOAD_CAST(obj) \
        ((GstRTPBasePayload*)(obj))

GST_RTP_BASE_PAYLOAD_MTU

#define GST_RTP_BASE_PAYLOAD_MTU(payload) (GST_RTP_BASE_PAYLOAD (payload)->mtu)

Get access to the configured MTU of payload.

Parameters:

payload

a GstRTPBasePayload


GST_RTP_BASE_PAYLOAD_PT

#define GST_RTP_BASE_PAYLOAD_PT(payload)  (GST_RTP_BASE_PAYLOAD (payload)->pt)

Get access to the configured payload type of payload.

Parameters:

payload

a GstRTPBasePayload


GST_RTP_BASE_PAYLOAD_SINKPAD

#define GST_RTP_BASE_PAYLOAD_SINKPAD(payload) (GST_RTP_BASE_PAYLOAD (payload)->sinkpad)

Get access to the sinkpad of payload.

Parameters:

payload

a GstRTPBasePayload


GST_RTP_BASE_PAYLOAD_SRCPAD

#define GST_RTP_BASE_PAYLOAD_SRCPAD(payload)  (GST_RTP_BASE_PAYLOAD (payload)->srcpad)

Get access to the srcpad of payload.

Parameters:

payload

a GstRTPBasePayload


Constants

GST_TYPE_RTP_BASE_PAYLOAD

#define GST_TYPE_RTP_BASE_PAYLOAD \
        (gst_rtp_base_payload_get_type())

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