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

GstRtp.RTPBasePayload 

GObject.Object
    ╰──GObject.InitiallyUnowned
        ╰──Gst.Object
            ╰──Gst.Element
                ╰──GstRtp.RTPBasePayload
                    ╰──GstRtp.RTPBaseAudioPayload

Members

element (Gst.Element) –
No description available

GstRtp.RTPBasePayload 

GObject.Object
    ╰──GObject.InitiallyUnowned
        ╰──Gst.Object
            ╰──Gst.Element
                ╰──GstRtp.RTPBasePayload
                    ╰──GstRtp.RTPBaseAudioPayload

Members

element (Gst.Element) –
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 (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

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 (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

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:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

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:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

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:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

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:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

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:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

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:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

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 ( [transfer: full]) –

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:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

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:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

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 ( [transfer: full]) –

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:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

list (Gst.BufferList) –

a Gst.BufferList

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:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

list (Gst.BufferList) –

a Gst.BufferList

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:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

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:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

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_outcaps_structure 

gboolean
gst_rtp_base_payload_set_outcaps_structure (GstRTPBasePayload * payload,
                                            GstStructure * s)

Configure the output caps with the optional fields.

Parameters:

payload –

a GstRTPBasePayload

s ( [nullable]) –

a GstStructure with the caps fields

Returns –

TRUE if the caps could be set.

Since : 1.20

GstRtp.RTPBasePayload.prototype.set_outcaps_structure 

function GstRtp.RTPBasePayload.prototype.set_outcaps_structure(s: Gst.Structure): {
    // javascript wrapper for 'gst_rtp_base_payload_set_outcaps_structure'
}

Configure the output caps with the optional fields.

Parameters:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

s (Gst.Structure) –

a Gst.Structure with the caps fields

Returns (Number) –

true if the caps could be set.

Since : 1.20

GstRtp.RTPBasePayload.set_outcaps_structure 

def GstRtp.RTPBasePayload.set_outcaps_structure (self, s):
    #python wrapper for 'gst_rtp_base_payload_set_outcaps_structure'

Configure the output caps with the optional fields.

Parameters:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

s (Gst.Structure) –

a Gst.Structure with the caps fields

Returns (bool) –

True if the caps could be set.

Since : 1.20

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:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

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:

payload (GstRtp.RTPBasePayload) –

a GstRtp.RTPBasePayload

enable (bool) –

whether to add contributing sources to RTP packets

Since : 1.16

Signals

request-extension 

GstRTPHeaderExtension *
request_extension_callback (GstRTPBasePayload * self,
                            guint ext_id,
                            gchar * ext_uri,
                            gpointer user_data)

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

Parameters:

self –
No description available
ext_id –

the extension id being requested

ext_uri –

the extension URI being requested

user_data –
No description available
Returns ( [transfer: full][nullable]) –

the GstRTPHeaderExtension for ext_id, or NULL

Flags: Run Last

Since : 1.20

request-extension 

function request_extension_callback(self: GstRtp.RTPBasePayload, ext_id: Number, ext_uri: String, user_data: Object): {
    // javascript callback for the 'request-extension' signal
}

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

Parameters:

No description available
ext_id (Number) –

the extension id being requested

ext_uri (String) –

the extension URI being requested

user_data (Object) –
No description available

the GstRtp.RTPHeaderExtension for ext_id, or null

Flags: Run Last

Since : 1.20

request-extension 

def request_extension_callback (self, ext_id, ext_uri, *user_data):
    #python callback for the 'request-extension' signal

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

Parameters:

No description available
ext_id (int) –

the extension id being requested

ext_uri (str) –

the extension URI being requested

user_data (variadic) –
No description available

the GstRtp.RTPHeaderExtension for ext_id, or None

Flags: Run Last

Since : 1.20

Action Signals

add-extension 

g_signal_emit_by_name (self, "add-extension", ext, user_data);

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

Parameters:

self (GstRTPBasePayload *) –
No description available
ext (GstRTPHeaderExtension *, [transfer: full]) –

the GstRTPHeaderExtension

user_data (gpointer) –
No description available

Flags: Run Last / Action

Since : 1.20

add-extension 

let ret = self.emit ("add-extension", ext, user_data);

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

Parameters:

No description available

the GstRtp.RTPHeaderExtension

user_data (Object) –
No description available

Flags: Run Last / Action

Since : 1.20

add-extension 

ret = self.emit ("add-extension", ext, user_data)

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

Parameters:

No description available

the GstRtp.RTPHeaderExtension

user_data (variadic) –
No description available

Flags: Run Last / Action

Since : 1.20

clear-extensions 

g_signal_emit_by_name (self, "clear-extensions", user_data);

Clear all RTP header extensions used by this payloader.

Parameters:

self (GstRTPBasePayload *) –
No description available
user_data (gpointer) –
No description available

Flags: Run Last / Action

Since : 1.20

clear-extensions 

let ret = self.emit ("clear-extensions", user_data);

Clear all RTP header extensions used by this payloader.

Parameters:

No description available
user_data (Object) –
No description available

Flags: Run Last / Action

Since : 1.20

clear-extensions 

ret = self.emit ("clear-extensions", user_data)

Clear all RTP header extensions used by this payloader.

Parameters:

No description available
user_data (variadic) –
No description available

Flags: Run Last / Action

Since : 1.20

Properties

auto-header-extension 

“auto-header-extension” gboolean

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 GstRTPBasePayload::request-extension: signal.

Flags : Read / Write

Since : 1.20

auto-header-extension 

“auto-header-extension” Number

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 GstRTPBasePayload::request-extension: signal.

Flags : Read / Write

Since : 1.20

auto_header_extension 

“self.props.auto_header_extension” bool

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 GstRTPBasePayload::request-extension: signal.

Flags : Read / Write

Since : 1.20

extensions 

“extensions” GstValueArray *

A list of already enabled RTP header extensions. This may be useful for finding out which extensions are already enabled (with add-extension signal) and picking a non-conflicting ID for a new extension that needs to be added on top of the existing ones.

Note that the value returned by reading this property is not dynamically updated when the set of enabled extensions changes by any of existing action signals. Rather, it represents the current state at the time the property is read.

Dynamic updates of this property can be received by subscribing to its corresponding "notify" signal, i.e. "notify::extensions".

Flags : Read

Since : 1.24

extensions 

“extensions” Gst.ValueArray

A list of already enabled RTP header extensions. This may be useful for finding out which extensions are already enabled (with add-extension signal) and picking a non-conflicting ID for a new extension that needs to be added on top of the existing ones.

Note that the value returned by reading this property is not dynamically updated when the set of enabled extensions changes by any of existing action signals. Rather, it represents the current state at the time the property is read.

Dynamic updates of this property can be received by subscribing to its corresponding "notify" signal, i.e. "notify::extensions".

Flags : Read

Since : 1.24

extensions 

“self.props.extensions” Gst.ValueArray

A list of already enabled RTP header extensions. This may be useful for finding out which extensions are already enabled (with add-extension signal) and picking a non-conflicting ID for a new extension that needs to be added on top of the existing ones.

Note that the value returned by reading this property is not dynamically updated when the set of enabled extensions changes by any of existing action signals. Rather, it represents the current state at the time the property is read.

Dynamic updates of this property can be received by subscribing to its corresponding "notify" signal, i.e. "notify::extensions".

Flags : Read

Since : 1.24

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

Since : 1.16

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

Since : 1.16

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

Since : 1.16

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

Since : 1.18

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

Since : 1.18

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

Since : 1.18

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

Since : 1.16

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

Since : 1.16

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

Since : 1.16

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

vfunc_get_caps 

function vfunc_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

do_get_caps 

def do_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

vfunc_handle_buffer 

function vfunc_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

do_handle_buffer 

def do_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

vfunc_query 

function vfunc_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

do_query 

def do_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

vfunc_set_caps 

function vfunc_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

do_set_caps 

def do_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

vfunc_sink_event 

function vfunc_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

do_sink_event 

def do_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

vfunc_src_event 

function vfunc_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

do_src_event 

def do_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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