GstRTCPBuffer

GstRtp.RTCPBuffer

GstRtp.RTCPBuffer.prototype.add_packet

function GstRtp.RTCPBuffer.prototype.add_packet(type: GstRtp.RTCPType, packet: GstRtp.RTCPPacket): {
    // javascript wrapper for 'gst_rtcp_buffer_add_packet'
}

Add a new packet of type to rtcp. packet will point to the newly created packet.

GstRtp.RTCPBuffer.prototype.get_first_packet

function GstRtp.RTCPBuffer.prototype.get_first_packet(packet: GstRtp.RTCPPacket): {
    // javascript wrapper for 'gst_rtcp_buffer_get_first_packet'
}

Initialize a new GstRtp.RTCPPacket pointer that points to the first packet in rtcp.

GstRtp.RTCPBuffer.prototype.get_packet_count

function GstRtp.RTCPBuffer.prototype.get_packet_count(): {
    // javascript wrapper for 'gst_rtcp_buffer_get_packet_count'
}

Get the number of RTCP packets in rtcp.

GstRtp.RTCPBuffer.prototype.unmap

function GstRtp.RTCPBuffer.prototype.unmap(): {
    // javascript wrapper for 'gst_rtcp_buffer_unmap'
}

Finish rtcp after being constructed. This function is usually called after GstRtp.RTCPBuffer.prototype.map and after adding the RTCP items to the new buffer.

The function adjusts the size of rtcp with the total length of all the added packets.

GstRtp.RTCPBuffer.prototype.map

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

Open buffer for reading or writing, depending on flags. The resulting RTCP buffer state is stored in rtcp.

GstRtp.RTCPBuffer.prototype.new

function GstRtp.RTCPBuffer.prototype.new(mtu: Number): {
    // javascript wrapper for 'gst_rtcp_buffer_new'
}

Create a new buffer for constructing RTCP packets. The packet will have a maximum size of mtu.

GstRtp.RTCPBuffer.prototype.new_copy_data

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

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

GstRtp.RTCPBuffer.prototype.new_take_data

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

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

GstRtp.RTCPBuffer.prototype.validate

function GstRtp.RTCPBuffer.prototype.validate(buffer: Gst.Buffer): {
    // javascript wrapper for 'gst_rtcp_buffer_validate'
}

Check if the data pointed to by buffer is a valid RTCP packet using GstRtp.RTCPBuffer.prototype.validate_data.

GstRtp.RTCPBuffer.prototype.validate_data

function GstRtp.RTCPBuffer.prototype.validate_data(data: [ Number ], len: Number): {
    // javascript wrapper for 'gst_rtcp_buffer_validate_data'
}

Check if the data and size point to the data of a valid compound, non-reduced size RTCP packet. Use this function to validate a packet before using the other functions in this module.

GstRtp.RTCPBuffer.prototype.validate_data_reduced

function GstRtp.RTCPBuffer.prototype.validate_data_reduced(data: [ Number ], len: Number): {
    // javascript wrapper for 'gst_rtcp_buffer_validate_data_reduced'
}

Check if the data and size point to the data of a valid RTCP packet. Use this function to validate a packet before using the other functions in this module.

This function is updated to support reduced size rtcp packets according to RFC 5506 and will validate full compound RTCP packets as well as reduced size RTCP packets.

Since : 1.6

GstRtp.RTCPBuffer.prototype.validate_reduced

function GstRtp.RTCPBuffer.prototype.validate_reduced(buffer: Gst.Buffer): {
    // javascript wrapper for 'gst_rtcp_buffer_validate_reduced'
}

Check if the data pointed to by buffer is a valid RTCP packet using GstRtp.RTCPBuffer.prototype.validate_reduced.

Since : 1.6

GstRtp.RTCPPacket

GstRtp.RTCPPacket.prototype.add_profile_specific_ext

function GstRtp.RTCPPacket.prototype.add_profile_specific_ext(data: [ Number ], len: Number): {
    // javascript wrapper for 'gst_rtcp_packet_add_profile_specific_ext'
}

Add profile-specific extension data to packet. If packet already contains profile-specific extension data will be appended to the existing extension.

Since : 1.10

GstRtp.RTCPPacket.prototype.add_rb

function GstRtp.RTCPPacket.prototype.add_rb(ssrc: Number, fractionlost: Number, packetslost: Number, exthighestseq: Number, jitter: Number, lsr: Number, dlsr: Number): {
    // javascript wrapper for 'gst_rtcp_packet_add_rb'
}

Add a new report block to packet with the given values.

GstRtp.RTCPPacket.prototype.app_get_data

function GstRtp.RTCPPacket.prototype.app_get_data(): {
    // javascript wrapper for 'gst_rtcp_packet_app_get_data'
}

Get the application-dependent data attached to a RTPFB or PSFB packet.

Since : 1.10

GstRtp.RTCPPacket.prototype.app_get_data_length

function GstRtp.RTCPPacket.prototype.app_get_data_length(): {
    // javascript wrapper for 'gst_rtcp_packet_app_get_data_length'
}

Get the length of the application-dependent data attached to an APP packet.

Since : 1.10

GstRtp.RTCPPacket.prototype.app_get_name

function GstRtp.RTCPPacket.prototype.app_get_name(): {
    // javascript wrapper for 'gst_rtcp_packet_app_get_name'
}

Get the name field of the APP packet.

Since : 1.10

GstRtp.RTCPPacket.prototype.app_get_ssrc

function GstRtp.RTCPPacket.prototype.app_get_ssrc(): {
    // javascript wrapper for 'gst_rtcp_packet_app_get_ssrc'
}

Get the SSRC/CSRC field of the APP packet.

Since : 1.10

GstRtp.RTCPPacket.prototype.app_get_subtype

function GstRtp.RTCPPacket.prototype.app_get_subtype(): {
    // javascript wrapper for 'gst_rtcp_packet_app_get_subtype'
}

Get the subtype field of the APP packet.

Since : 1.10

GstRtp.RTCPPacket.prototype.app_set_data_length

function GstRtp.RTCPPacket.prototype.app_set_data_length(wordlen: Number): {
    // javascript wrapper for 'gst_rtcp_packet_app_set_data_length'
}

Set the length of the application-dependent data attached to an APP packet.

Since : 1.10

GstRtp.RTCPPacket.prototype.app_set_name

function GstRtp.RTCPPacket.prototype.app_set_name(name: String): {
    // javascript wrapper for 'gst_rtcp_packet_app_set_name'
}

Set the name field of the APP packet.

Since : 1.10

GstRtp.RTCPPacket.prototype.app_set_ssrc

function GstRtp.RTCPPacket.prototype.app_set_ssrc(ssrc: Number): {
    // javascript wrapper for 'gst_rtcp_packet_app_set_ssrc'
}

Set the SSRC/CSRC field of the APP packet.

Since : 1.10

GstRtp.RTCPPacket.prototype.app_set_subtype

function GstRtp.RTCPPacket.prototype.app_set_subtype(subtype: Number): {
    // javascript wrapper for 'gst_rtcp_packet_app_set_subtype'
}

Set the subtype field of the APP packet.

Since : 1.10

GstRtp.RTCPPacket.prototype.bye_add_ssrc

function GstRtp.RTCPPacket.prototype.bye_add_ssrc(ssrc: Number): {
    // javascript wrapper for 'gst_rtcp_packet_bye_add_ssrc'
}

Add ssrc to the BYE packet.

GstRtp.RTCPPacket.prototype.bye_add_ssrcs

function GstRtp.RTCPPacket.prototype.bye_add_ssrcs(ssrc: [ Number ], len: Number): {
    // javascript wrapper for 'gst_rtcp_packet_bye_add_ssrcs'
}

Adds len SSRCs in ssrc to BYE packet.

GstRtp.RTCPPacket.prototype.bye_get_nth_ssrc

function GstRtp.RTCPPacket.prototype.bye_get_nth_ssrc(nth: Number): {
    // javascript wrapper for 'gst_rtcp_packet_bye_get_nth_ssrc'
}

Get the nth SSRC of the BYE packet.

GstRtp.RTCPPacket.prototype.bye_get_reason

function GstRtp.RTCPPacket.prototype.bye_get_reason(): {
    // javascript wrapper for 'gst_rtcp_packet_bye_get_reason'
}

Get the reason in packet.

GstRtp.RTCPPacket.prototype.bye_get_reason_len

function GstRtp.RTCPPacket.prototype.bye_get_reason_len(): {
    // javascript wrapper for 'gst_rtcp_packet_bye_get_reason_len'
}

Get the length of the reason string.

GstRtp.RTCPPacket.prototype.bye_get_ssrc_count

function GstRtp.RTCPPacket.prototype.bye_get_ssrc_count(): {
    // javascript wrapper for 'gst_rtcp_packet_bye_get_ssrc_count'
}

Get the number of SSRC fields in packet.

GstRtp.RTCPPacket.prototype.bye_set_reason

function GstRtp.RTCPPacket.prototype.bye_set_reason(reason: String): {
    // javascript wrapper for 'gst_rtcp_packet_bye_set_reason'
}

Set the reason string to reason in packet.

GstRtp.RTCPPacket.prototype.copy_profile_specific_ext

function GstRtp.RTCPPacket.prototype.copy_profile_specific_ext(): {
    // javascript wrapper for 'gst_rtcp_packet_copy_profile_specific_ext'
}

The profile-specific extension data is copied into a new allocated memory area data. This must be freed with GLib.prototype.free after usage.

Since : 1.10

GstRtp.RTCPPacket.prototype.fb_get_fci

function GstRtp.RTCPPacket.prototype.fb_get_fci(): {
    // javascript wrapper for 'gst_rtcp_packet_fb_get_fci'
}

Get the Feedback Control Information attached to a RTPFB or PSFB packet.

GstRtp.RTCPPacket.prototype.fb_get_fci_length

function GstRtp.RTCPPacket.prototype.fb_get_fci_length(): {
    // javascript wrapper for 'gst_rtcp_packet_fb_get_fci_length'
}

Get the length of the Feedback Control Information attached to a RTPFB or PSFB packet.

GstRtp.RTCPPacket.prototype.fb_get_media_ssrc

function GstRtp.RTCPPacket.prototype.fb_get_media_ssrc(): {
    // javascript wrapper for 'gst_rtcp_packet_fb_get_media_ssrc'
}

Get the media SSRC field of the RTPFB or PSFB packet.

GstRtp.RTCPPacket.prototype.fb_get_sender_ssrc

function GstRtp.RTCPPacket.prototype.fb_get_sender_ssrc(): {
    // javascript wrapper for 'gst_rtcp_packet_fb_get_sender_ssrc'
}

Get the sender SSRC field of the RTPFB or PSFB packet.

GstRtp.RTCPPacket.prototype.fb_get_type

function GstRtp.RTCPPacket.prototype.fb_get_type(): {
    // javascript wrapper for 'gst_rtcp_packet_fb_get_type'
}

Get the feedback message type of the FB packet.

GstRtp.RTCPPacket.prototype.fb_set_fci_length

function GstRtp.RTCPPacket.prototype.fb_set_fci_length(wordlen: Number): {
    // javascript wrapper for 'gst_rtcp_packet_fb_set_fci_length'
}

Set the length of the Feedback Control Information attached to a RTPFB or PSFB packet.

GstRtp.RTCPPacket.prototype.fb_set_media_ssrc

function GstRtp.RTCPPacket.prototype.fb_set_media_ssrc(ssrc: Number): {
    // javascript wrapper for 'gst_rtcp_packet_fb_set_media_ssrc'
}

Set the media SSRC field of the RTPFB or PSFB packet.

GstRtp.RTCPPacket.prototype.fb_set_sender_ssrc

function GstRtp.RTCPPacket.prototype.fb_set_sender_ssrc(ssrc: Number): {
    // javascript wrapper for 'gst_rtcp_packet_fb_set_sender_ssrc'
}

Set the sender SSRC field of the RTPFB or PSFB packet.

GstRtp.RTCPPacket.prototype.fb_set_type

function GstRtp.RTCPPacket.prototype.fb_set_type(type: GstRtp.RTCPFBType): {
    // javascript wrapper for 'gst_rtcp_packet_fb_set_type'
}

Set the feedback message type of the FB packet.

GstRtp.RTCPPacket.prototype.get_count

function GstRtp.RTCPPacket.prototype.get_count(): {
    // javascript wrapper for 'gst_rtcp_packet_get_count'
}

Get the count field in packet.

GstRtp.RTCPPacket.prototype.get_length

function GstRtp.RTCPPacket.prototype.get_length(): {
    // javascript wrapper for 'gst_rtcp_packet_get_length'
}

Get the length field of packet. This is the length of the packet in 32-bit words minus one.

GstRtp.RTCPPacket.prototype.get_padding

function GstRtp.RTCPPacket.prototype.get_padding(): {
    // javascript wrapper for 'gst_rtcp_packet_get_padding'
}

Get the packet padding of the packet pointed to by packet.

GstRtp.RTCPPacket.prototype.get_profile_specific_ext

function GstRtp.RTCPPacket.prototype.get_profile_specific_ext(): {
    // javascript wrapper for 'gst_rtcp_packet_get_profile_specific_ext'
}

Since : 1.10

GstRtp.RTCPPacket.prototype.get_profile_specific_ext_length

function GstRtp.RTCPPacket.prototype.get_profile_specific_ext_length(): {
    // javascript wrapper for 'gst_rtcp_packet_get_profile_specific_ext_length'
}

Since : 1.10

GstRtp.RTCPPacket.prototype.get_rb

function GstRtp.RTCPPacket.prototype.get_rb(nth: Number): {
    // javascript wrapper for 'gst_rtcp_packet_get_rb'
}

Parse the values of the nth report block in packet and store the result in the values.

GstRtp.RTCPPacket.prototype.get_rb_count

function GstRtp.RTCPPacket.prototype.get_rb_count(): {
    // javascript wrapper for 'gst_rtcp_packet_get_rb_count'
}

Get the number of report blocks in packet.

GstRtp.RTCPPacket.prototype.get_type

function GstRtp.RTCPPacket.prototype.get_type(): {
    // javascript wrapper for 'gst_rtcp_packet_get_type'
}

Get the packet type of the packet pointed to by packet.

GstRtp.RTCPPacket.prototype.move_to_next

function GstRtp.RTCPPacket.prototype.move_to_next(): {
    // javascript wrapper for 'gst_rtcp_packet_move_to_next'
}

Move the packet pointer packet to the next packet in the payload. Use GstRtp.RTCPBuffer.prototype.get_first_packet to initialize packet.

GstRtp.RTCPPacket.prototype.remove

function GstRtp.RTCPPacket.prototype.remove(): {
    // javascript wrapper for 'gst_rtcp_packet_remove'
}

Removes the packet pointed to by packet and moves pointer to the next one

GstRtp.RTCPPacket.prototype.rr_get_ssrc

function GstRtp.RTCPPacket.prototype.rr_get_ssrc(): {
    // javascript wrapper for 'gst_rtcp_packet_rr_get_ssrc'
}

Get the ssrc field of the RR packet.

GstRtp.RTCPPacket.prototype.rr_set_ssrc

function GstRtp.RTCPPacket.prototype.rr_set_ssrc(ssrc: Number): {
    // javascript wrapper for 'gst_rtcp_packet_rr_set_ssrc'
}

Set the ssrc field of the RR packet.

GstRtp.RTCPPacket.prototype.sdes_add_entry

function GstRtp.RTCPPacket.prototype.sdes_add_entry(type: GstRtp.RTCPSDESType, len: Number, data: [ Number ]): {
    // javascript wrapper for 'gst_rtcp_packet_sdes_add_entry'
}

Add a new SDES entry to the current item in packet.

GstRtp.RTCPPacket.prototype.sdes_add_item

function GstRtp.RTCPPacket.prototype.sdes_add_item(ssrc: Number): {
    // javascript wrapper for 'gst_rtcp_packet_sdes_add_item'
}

Add a new SDES item for ssrc to packet.

GstRtp.RTCPPacket.prototype.sdes_copy_entry

function GstRtp.RTCPPacket.prototype.sdes_copy_entry(type: GstRtp.RTCPSDESType): {
    // javascript wrapper for 'gst_rtcp_packet_sdes_copy_entry'
}

This function is like GstRtp.RTCPPacket.prototype.sdes_get_entry but it returns a null-terminated copy of the data instead. use GLib.prototype.free after usage.

GstRtp.RTCPPacket.prototype.sdes_first_entry

function GstRtp.RTCPPacket.prototype.sdes_first_entry(): {
    // javascript wrapper for 'gst_rtcp_packet_sdes_first_entry'
}

Move to the first SDES entry in the current item.

GstRtp.RTCPPacket.prototype.sdes_first_item

function GstRtp.RTCPPacket.prototype.sdes_first_item(): {
    // javascript wrapper for 'gst_rtcp_packet_sdes_first_item'
}

Move to the first SDES item in packet.

GstRtp.RTCPPacket.prototype.sdes_get_entry

function GstRtp.RTCPPacket.prototype.sdes_get_entry(type: GstRtp.RTCPSDESType): {
    // javascript wrapper for 'gst_rtcp_packet_sdes_get_entry'
}

Get the data of the current SDES item entry. type (when not NULL) will contain the type of the entry. data (when not NULL) will point to len bytes.

When type refers to a text item, data will point to a UTF8 string. Note that this UTF8 string is NOT null-terminated. Use GstRtp.RTCPPacket.prototype.sdes_copy_entry to get a null-terminated copy of the entry.

GstRtp.RTCPPacket.prototype.sdes_get_item_count

function GstRtp.RTCPPacket.prototype.sdes_get_item_count(): {
    // javascript wrapper for 'gst_rtcp_packet_sdes_get_item_count'
}

Get the number of items in the SDES packet packet.

GstRtp.RTCPPacket.prototype.sdes_get_ssrc

function GstRtp.RTCPPacket.prototype.sdes_get_ssrc(): {
    // javascript wrapper for 'gst_rtcp_packet_sdes_get_ssrc'
}

Get the SSRC of the current SDES item.

GstRtp.RTCPPacket.prototype.sdes_next_entry

function GstRtp.RTCPPacket.prototype.sdes_next_entry(): {
    // javascript wrapper for 'gst_rtcp_packet_sdes_next_entry'
}

Move to the next SDES entry in the current item.

GstRtp.RTCPPacket.prototype.sdes_next_item

function GstRtp.RTCPPacket.prototype.sdes_next_item(): {
    // javascript wrapper for 'gst_rtcp_packet_sdes_next_item'
}

Move to the next SDES item in packet.

GstRtp.RTCPPacket.prototype.set_rb

function GstRtp.RTCPPacket.prototype.set_rb(nth: Number, ssrc: Number, fractionlost: Number, packetslost: Number, exthighestseq: Number, jitter: Number, lsr: Number, dlsr: Number): {
    // javascript wrapper for 'gst_rtcp_packet_set_rb'
}

Set the nth new report block in packet with the given values.

Note: Not implemented.

GstRtp.RTCPPacket.prototype.sr_get_sender_info

function GstRtp.RTCPPacket.prototype.sr_get_sender_info(): {
    // javascript wrapper for 'gst_rtcp_packet_sr_get_sender_info'
}

Parse the SR sender info and store the values.

GstRtp.RTCPPacket.prototype.sr_set_sender_info

function GstRtp.RTCPPacket.prototype.sr_set_sender_info(ssrc: Number, ntptime: Number, rtptime: Number, packet_count: Number, octet_count: Number): {
    // javascript wrapper for 'gst_rtcp_packet_sr_set_sender_info'
}

Set the given values in the SR packet packet.

GstRtp.RTCPPacket.prototype.xr_first_rb

function GstRtp.RTCPPacket.prototype.xr_first_rb(): {
    // javascript wrapper for 'gst_rtcp_packet_xr_first_rb'
}

Move to the first extended report block in XR packet.

Since : 1.16

GstRtp.RTCPPacket.prototype.xr_get_block_length

function GstRtp.RTCPPacket.prototype.xr_get_block_length(): {
    // javascript wrapper for 'gst_rtcp_packet_xr_get_block_length'
}

Since : 1.16

GstRtp.RTCPPacket.prototype.xr_get_block_type

function GstRtp.RTCPPacket.prototype.xr_get_block_type(): {
    // javascript wrapper for 'gst_rtcp_packet_xr_get_block_type'
}

Get the extended report block type of the XR packet.

Since : 1.16

GstRtp.RTCPPacket.prototype.xr_get_dlrr_block

function GstRtp.RTCPPacket.prototype.xr_get_dlrr_block(nth: Number, ssrc: Number, last_rr: Number, delay: Number): {
    // javascript wrapper for 'gst_rtcp_packet_xr_get_dlrr_block'
}

Parse the extended report block for DLRR report block type.

Since : 1.16

GstRtp.RTCPPacket.prototype.xr_get_prt_by_seq

function GstRtp.RTCPPacket.prototype.xr_get_prt_by_seq(seq: Number, receipt_time: Number): {
    // javascript wrapper for 'gst_rtcp_packet_xr_get_prt_by_seq'
}

Retrieve the packet receipt time of seq which ranges in [begin_seq, end_seq).