GstRtp.RTPBasePayload¶

Subclasses:: GstRtp.RTPBaseAudioPayload

Methods¶

Inherited:: Gst.Element (81), Gst.Object (27), GObject.Object (37)
Structs:: Gst.ElementClass (10), GObject.ObjectClass (5)

	`allocate_output_buffer` (payload_len, pad_len, csrc_count)
	`get_source_count` (buffer)
	`is_filled` (size, duration)
	`is_source_info_enabled` ()
	`push` (buffer)
	`push_list` (list)
	`set_options` (media, dynamic, encoding_name, clock_rate)
	`set_outcaps_structure` (s)
	`set_source_info_enabled` (enable)

Virtual Methods¶

Inherited:: Gst.Element (16), Gst.Object (1), GObject.Object (7)

	`do_get_caps` (pad, filter)
	`do_handle_buffer` (buffer)
	`do_query` (pad, query)
	`do_set_caps` (caps)
	`do_sink_event` (event)
	`do_src_event` (event)

Properties¶

Inherited:: Gst.Object (2)

Name	Type	Flags	Short Description
`auto-header-extension`	`bool`	r/w	Whether RTP header extensions should be automatically enabled, if an implementation is available
`max-ptime`	`int`	r/w	Maximum duration of the packet data in ns (-1 = unlimited up to MTU)
`min-ptime`	`int`	r/w	Minimum duration of the packet data in ns (can’t go above MTU)
`mtu`	`int`	r/w	Maximum size of one packet
`onvif-no-rate-control`	`bool`	r/w	Enable ONVIF Rate-Control=no timestamping mode
`perfect-rtptime`	`bool`	r/w	Generate perfect RTP timestamps when possible
`pt`	`int`	r/w	The payload type of the packets
`ptime-multiple`	`int`	r/w	Force buffers to be multiples of this duration in ns (0 disables)
`scale-rtptime`	`bool`	r/w	Whether the RTP timestamp should be scaled with the rate (speed)
`seqnum`	`int`	r	The RTP sequence number of the last processed packet
`seqnum-offset`	`int`	r/w	Offset to add to all outgoing seqnum (-1 = random)
`source-info`	`bool`	r/w	Write CSRC based on buffer meta RTP source information
`ssrc`	`int`	r/w	The SSRC of the packets (default == random)
`stats`	`Gst.Structure`	r	Various statistics
`timestamp`	`int`	r	The RTP timestamp of the last processed packet
`timestamp-offset`	`int`	r/w	Offset to add to all outgoing timestamps (default = random)

Signals¶

Inherited:: Gst.Element (3), Gst.Object (1), GObject.Object (1)

Name	Short Description
`add-extension`	Add ext as an extension for writing part of an RTP header extension onto outgoing RTP packets.
`clear-extensions`	Clear all RTP header extensions used by this payloader.
`request-extension`	The returned ext must be configured with the correct ext_id and with the necessary attributes as required by the extension implementation.

Fields¶

Inherited:: Gst.Element (3), Gst.Object (1), GObject.Object (1)

Name	Type	Access
clock_rate	`int`	r
current_ssrc	`int`	r
dynamic	`bool`	r
element	`Gst.Element`	r
encoding_name	`str`	r
max_ptime	`int`	r
media	`str`	r
min_ptime	`int`	r
mtu	`int`	r
pt	`int`	r
ptime	`int`	r
ptime_multiple	`int`	r
segment	`Gst.Segment`	r
seqnum	`int`	r
seqnum_base	`int`	r
seqnum_offset	`int`	r
sinkpad	`Gst.Pad`	r
srcpad	`Gst.Pad`	r
ssrc	`int`	r
timestamp	`int`	r
ts_base	`int`	r
ts_offset	`int`	r

Class Details¶

class GstRtp.RTPBasePayload(**kwargs)¶

Bases:: Gst.Element
Abstract:: Yes
Structure:: GstRtp.RTPBasePayloadClass

Provides a base class for RTP payloaders

allocate_output_buffer(payload_len, pad_len, csrc_count)¶

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:

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

Return type:

Gst.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 self has GstRtp.RTPBasePayload :source-info True additional CSRCs may be allocated and filled with RTP source information.

New in version 1.16.

get_source_count(buffer)¶

Parameters:: buffer (Gst.Buffer) – a Gst.Buffer, typically the buffer to payload
Returns:: The number of sources.
Return type:: int

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 GstRtp.RTPBasePayload :source-info is False the count will be 0.

New in version 1.16.

is_filled(size, duration)¶

Parameters:

size (int) – the size of the packet
duration (int) – the duration of the packet

Returns:

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

Return type:

bool

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

is_source_info_enabled()¶

Returns:: True if source-info is enabled.
Return type:: bool

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

New in version 1.16.

push(buffer)¶

Parameters:: buffer (Gst.Buffer) – a Gst.Buffer
Returns:: a Gst.FlowReturn.
Return type:: Gst.FlowReturn

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.

push_list(list)¶

Parameters:: list (Gst.BufferList) – a Gst.BufferList
Returns:: a Gst.FlowReturn.
Return type:: Gst.FlowReturn

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.

set_options(media, dynamic, encoding_name, clock_rate)¶

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

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().

set_outcaps_structure(s)¶

Parameters:: s (Gst.Structure or None) – a Gst.Structure with the caps fields
Returns:: True if the caps could be set.
Return type:: bool

Configure the output caps with the optional fields.

New in version 1.20.

set_source_info_enabled(enable)¶

Parameters:: enable (bool) – whether to add contributing sources to RTP packets

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

New in version 1.16.

do_get_caps(pad, filter) virtual¶

Parameters:

pad (Gst.Pad) –
filter (Gst.Caps) –

Return type:

Gst.Caps

do_handle_buffer(buffer) virtual¶

Parameters:: buffer (Gst.Buffer) –
Return type:: Gst.FlowReturn

do_query(pad, query) virtual¶

Parameters:

pad (Gst.Pad) –
query (Gst.Query) –

Return type:

bool

do_set_caps(caps) virtual¶

Parameters:: caps (Gst.Caps) –
Return type:: bool

do_sink_event(event) virtual¶

Parameters:: event (Gst.Event) –
Return type:: bool

do_src_event(event) virtual¶

Parameters:: event (Gst.Event) –
Return type:: bool

Signal Details¶

GstRtp.RTPBasePayload.signals.add_extension(r_t_p_base_payload, ext)¶

Signal Name:

add-extension

Flags:

RUN_LAST, ACTION

Parameters:

r_t_p_base_payload (GstRtp.RTPBasePayload) – The object which received the signal
ext (GstRtp.RTPHeaderExtension) – the GstRtp.RTPHeaderExtension

Add ext as an extension for writing part of an RTP header extension onto outgoing RTP packets.

New in version 1.20.

GstRtp.RTPBasePayload.signals.clear_extensions(r_t_p_base_payload)¶

Signal Name:: clear-extensions
Flags:: RUN_LAST, ACTION
Parameters:: r_t_p_base_payload (GstRtp.RTPBasePayload) – The object which received the signal

Clear all RTP header extensions used by this payloader.

New in version 1.20.

GstRtp.RTPBasePayload.signals.request_extension(r_t_p_base_payload, ext_id, ext_uri)¶

Signal Name:

request-extension

Flags:

RUN_LAST

Parameters:

r_t_p_base_payload (GstRtp.RTPBasePayload) – The object which received the signal
ext_id (int) – the extension id being requested
ext_uri (str) – the extension URI being requested

Returns:

the GstRtp.RTPHeaderExtension for ext_id, or None

Return type:

GstRtp.RTPHeaderExtension or None

The returned ext must be configured with the correct ext_id and with the necessary attributes as required by the extension implementation.

New in version 1.20.

Property Details¶

GstRtp.RTPBasePayload.props.auto_header_extension¶

Name:: auto-header-extension
Type:: bool
Default Value:: True
Flags:: READABLE, WRITABLE

If enabled, the payloader will automatically try to enable all the RTP header extensions provided in the src caps, saving the application the need to handle these extensions manually using the GstRtp.RTPBasePayload ::request-extension: signal.

New in version 1.20.

GstRtp.RTPBasePayload.props.max_ptime¶

Name:: max-ptime
Type:: int
Default Value:: -1
Flags:: READABLE, WRITABLE

Maximum duration of the packet data in ns (-1 = unlimited up to MTU)

GstRtp.RTPBasePayload.props.min_ptime¶

Name:: min-ptime
Type:: int
Default Value:: 0
Flags:: READABLE, WRITABLE

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

GstRtp.RTPBasePayload.props.mtu¶

Name:: mtu
Type:: int
Default Value:: 1400
Flags:: READABLE, WRITABLE

Maximum size of one packet

GstRtp.RTPBasePayload.props.onvif_no_rate_control¶

Name:: onvif-no-rate-control
Type:: bool
Default Value:: False
Flags:: READABLE, WRITABLE

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

New in version 1.16.

GstRtp.RTPBasePayload.props.perfect_rtptime¶

Name:: perfect-rtptime
Type:: bool
Default Value:: True
Flags:: READABLE, WRITABLE

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 GstRtp.RTPBasePayload is limited to handling perfect RTP timestamps for audio streams.

GstRtp.RTPBasePayload.props.pt¶

Name:: pt
Type:: int
Default Value:: 96
Flags:: READABLE, WRITABLE

The payload type of the packets

GstRtp.RTPBasePayload.props.ptime_multiple¶

Name:: ptime-multiple
Type:: int
Default Value:: 0
Flags:: READABLE, WRITABLE

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

GstRtp.RTPBasePayload.props.scale_rtptime¶

Name:: scale-rtptime
Type:: bool
Default Value:: True
Flags:: READABLE, WRITABLE

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.

New in version 1.18.

GstRtp.RTPBasePayload.props.seqnum¶

Name:: seqnum
Type:: int
Default Value:: 0
Flags:: READABLE

The RTP sequence number of the last processed packet

GstRtp.RTPBasePayload.props.seqnum_offset¶

Name:: seqnum-offset
Type:: int
Default Value:: -1
Flags:: READABLE, WRITABLE

Offset to add to all outgoing seqnum (-1 = random)

GstRtp.RTPBasePayload.props.source_info¶

Name:: source-info
Type:: bool
Default Value:: False
Flags:: READABLE, WRITABLE

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

New in version 1.16.

GstRtp.RTPBasePayload.props.ssrc¶

Name:: ssrc
Type:: int
Default Value:: 4294967295
Flags:: READABLE, WRITABLE

The SSRC of the packets (default == random)

GstRtp.RTPBasePayload.props.stats¶

Name:: stats
Type:: Gst.Structure
Default Value:: None
Flags:: READABLE

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 Gst.Structure 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 :GObject.TYPE_UINT, clock-rate of the stream
running-time :GObject.TYPE_UINT64, running time
seqnum :GObject.TYPE_UINT, sequence number, same as GstRtp.RTPBasePayload :seqnum
timestamp :GObject.TYPE_UINT, RTP timestamp, same as GstRtp.RTPBasePayload :timestamp
ssrc :GObject.TYPE_UINT, The SSRC in use
pt :GObject.TYPE_UINT, The Payload type in use, same as GstRtp.RTPBasePayload :pt
seqnum-offset :GObject.TYPE_UINT, The current offset added to the seqnum
timestamp-offset :GObject.TYPE_UINT, The current offset added to the timestamp

GstRtp.RTPBasePayload.props.timestamp¶

Name:: timestamp
Type:: int
Default Value:: 0
Flags:: READABLE

The RTP timestamp of the last processed packet

GstRtp.RTPBasePayload.props.timestamp_offset¶

Name:: timestamp-offset
Type:: int
Default Value:: 4294967295
Flags:: READABLE, WRITABLE

Offset to add to all outgoing timestamps (default = random)