GstRTPBaseDepayload

GstRtp.RTPBaseDepayload

GObject.Object
    ╰──GObject.InitiallyUnowned
        ╰──Gst.Object
            ╰──Gst.Element
                ╰──GstRtp.RTPBaseDepayload

GstRtp.RTPBaseDepayload.prototype.delayed

function GstRtp.RTPBaseDepayload.prototype.delayed(): {
    // javascript wrapper for 'gst_rtp_base_depayload_delayed'
}

Called from GstRTPBaseDepayload.process or GstRTPBaseDepayload.process_rtp_packet when the depayloader needs to keep the current input RTP header for use with the next output buffer.

The delayed buffer will remain until the end of processing the current output buffer and then enqueued for processing with the next output buffer.

A typical use-case is when the depayloader implementation will start a new output buffer for the current input RTP buffer but push the current output buffer first.

Must be called with the stream lock held.

Since : 1.24

GstRtp.RTPBaseDepayload.prototype.dropped

function GstRtp.RTPBaseDepayload.prototype.dropped(): {
    // javascript wrapper for 'gst_rtp_base_depayload_dropped'
}

Called from GstRTPBaseDepayload.process or GstRTPBaseDepayload.process_rtp_packet if the depayloader does not use the current buffer for the output buffer. This will either drop the delayed buffer or the last buffer from the header extension cache.

A typical use-case is when the depayloader implementation is dropping an input RTP buffer while waiting for the first keyframe.

Must be called with the stream lock held.

Since : 1.24

GstRtp.RTPBaseDepayload.prototype.flush

function GstRtp.RTPBaseDepayload.prototype.flush(keep_current: Number): {
    // javascript wrapper for 'gst_rtp_base_depayload_flush'
}

If GstRTPBaseDepayload.process or GstRTPBaseDepayload.process_rtp_packet drop an output buffer this function tells the base class to flush header extension cache as well.

This will not drop an input RTP header marked as delayed from GstRtp.RTPBaseDepayload.prototype.delayed.

If keep_current is true the current input RTP header will be kept and enqueued after flushing the previous input RTP headers.

A typical use-case for keep_current is when the depayloader implementation invalidates the current output buffer and starts a new one with the current RTP input buffer.

Must be called with the stream lock held.

Since : 1.24

GstRtp.RTPBaseDepayload.prototype.is_aggregate_hdrext_enabled

function GstRtp.RTPBaseDepayload.prototype.is_aggregate_hdrext_enabled(): {
    // javascript wrapper for 'gst_rtp_base_depayload_is_aggregate_hdrext_enabled'
}

Queries whether header extensions will be aggregated per depayloaded buffers.

Since : 1.24

GstRtp.RTPBaseDepayload.prototype.is_source_info_enabled

function GstRtp.RTPBaseDepayload.prototype.is_source_info_enabled(): {
    // javascript wrapper for 'gst_rtp_base_depayload_is_source_info_enabled'
}

Queries whether GstRtp.RTPSourceMeta will be added to depayloaded buffers.

Since : 1.16

GstRtp.RTPBaseDepayload.prototype.push

function GstRtp.RTPBaseDepayload.prototype.push(out_buf: Gst.Buffer): {
    // javascript wrapper for 'gst_rtp_base_depayload_push'
}

Push out_buf to the peer of filter. This function takes ownership of out_buf.

This function will by default apply the last incoming timestamp on the outgoing buffer when it didn't have a timestamp already.

GstRtp.RTPBaseDepayload.prototype.push_list

function GstRtp.RTPBaseDepayload.prototype.push_list(out_list: Gst.BufferList): {
    // javascript wrapper for 'gst_rtp_base_depayload_push_list'
}

Push out_list to the peer of filter. This function takes ownership of out_list.

GstRtp.RTPBaseDepayload.prototype.set_aggregate_hdrext_enabled

function GstRtp.RTPBaseDepayload.prototype.set_aggregate_hdrext_enabled(enable: Number): {
    // javascript wrapper for 'gst_rtp_base_depayload_set_aggregate_hdrext_enabled'
}

Enable or disable aggregating header extensions.

Since : 1.24

GstRtp.RTPBaseDepayload.prototype.set_source_info_enabled

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

Enable or disable adding GstRtp.RTPSourceMeta to depayloaded buffers.

Since : 1.16

request-extension

function request_extension_callback(self: GstRtp.RTPBaseDepayload, 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.

Flags: Run Last

Since : 1.20

add-extension

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

Add ext as an extension for reading part of an RTP header extension from incoming RTP packets.

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 depayloader.

Flags: Run Last / Action

Since : 1.20

auto-header-extension

“auto-header-extension” Number

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

Flags : Read / Write

Since : 1.20

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

max-reorder

“max-reorder” Number

Max seqnum reorder before the sender is assumed to have restarted.

When max-reorder is set to 0 all reordered/duplicate packets are considered coming from a restarted sender.

Flags : Read / Write

Since : 1.18

source-info

“source-info” Number

Add RTP source information found in RTP header as meta to output buffer.

Flags : Read / Write

Since : 1.16

stats

“stats” Gst.Structure

Various depayloader statistics retrieved atomically (and are therefore synchroized with each other). This property return a GstStructure named application/x-rtp-depayload-stats containing the following fields relating to the last processed buffer and current state of the stream being depayloaded:

• clock-rate: G_TYPE_UINT (not introspectable), clock-rate of the stream
• npt-start: G_TYPE_UINT64 (not introspectable), time of playback start
• npt-stop: G_TYPE_UINT64 (not introspectable), time of playback stop
• play-speed: G_TYPE_DOUBLE (not introspectable), the playback speed
• play-scale: G_TYPE_DOUBLE (not introspectable), the playback scale
• running-time-dts: G_TYPE_UINT64 (not introspectable), the last running-time of the last DTS
• running-time-pts: G_TYPE_UINT64 (not introspectable), the last running-time of the last PTS
• seqnum: G_TYPE_UINT (not introspectable), the last seen seqnum
• timestamp: G_TYPE_UINT (not introspectable), the last seen RTP timestamp

Flags : Read

vfunc_handle_event

function vfunc_handle_event(filter: GstRtp.RTPBaseDepayload, event: Gst.Event): {
    // javascript implementation of the 'handle_event' virtual method
}

custom event handling

vfunc_packet_lost

function vfunc_packet_lost(filter: GstRtp.RTPBaseDepayload, event: Gst.Event): {
    // javascript implementation of the 'packet_lost' virtual method
}

signal the depayloader about packet loss

vfunc_process

function vfunc_process(base: GstRtp.RTPBaseDepayload, in: Gst.Buffer): {
    // javascript implementation of the 'process' virtual method
}

process incoming rtp packets. Subclass must implement either this method or process_rtp_packet to process incoming rtp packets. If the child returns a buffer without a valid timestamp, the timestamp of the provided buffer will be applied to the result buffer and the buffer will be pushed. If this function returns null, nothing is pushed.

vfunc_process_rtp_packet

function vfunc_process_rtp_packet(base: GstRtp.RTPBaseDepayload, rtp_buffer: GstRtp.RTPBuffer): {
    // javascript implementation of the 'process_rtp_packet' virtual method
}

Same as the process virtual function, but slightly more efficient, since it is passed the rtp buffer structure that has already been mapped (with GST_MAP_READ) by the base class and thus does not have to be mapped again by the subclass. Can be used by the subclass to process incoming rtp packets. If the subclass returns a buffer without a valid timestamp, the timestamp of the input buffer will be applied to the result buffer and the output buffer will be pushed out. If this function returns null, nothing is pushed out. Since: 1.6.

vfunc_set_caps

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

configure the depayloader