///
///
///
///
///
/**
* Type Definitions for Gjs (https://gjs.guide/)
*
* These type definitions are automatically generated, do not edit them by hand.
* If you found a bug fix it in `ts-for-gir` or create a bug report on https://github.com/gjsify/ts-for-gir
*
* The based EJS template file is used for the generated .d.ts file of each GIR module like Gtk-4.0, GObject-2.0, ...
*/
declare module 'gi://GstRtp?version=1.0' {
// Module dependencies
import type GstBase from 'gi://GstBase?version=1.0';
import type Gst from 'gi://Gst?version=1.0';
import type GObject from 'gi://GObject?version=2.0';
import type GLib from 'gi://GLib?version=2.0';
import type GModule from 'gi://GModule?version=2.0';
export namespace GstRtp {
/**
* GstRtp-1.0
*/
/**
* Different types of feedback messages.
*/
/**
* Different types of feedback messages.
*/
export namespace RTCPFBType {
export const $gtype: GObject.GType;
}
enum RTCPFBType {
/**
* Invalid type
*/
FB_TYPE_INVALID,
/**
* Generic NACK
*/
RTPFB_TYPE_NACK,
/**
* Temporary Maximum Media Stream Bit Rate Request
*/
RTPFB_TYPE_TMMBR,
/**
* Temporary Maximum Media Stream Bit Rate
* Notification
*/
RTPFB_TYPE_TMMBN,
/**
* Request an SR packet for early
* synchronization
*/
RTPFB_TYPE_RTCP_SR_REQ,
RTPFB_TYPE_TWCC,
/**
* Picture Loss Indication
*/
PSFB_TYPE_PLI,
/**
* Slice Loss Indication
*/
PSFB_TYPE_SLI,
/**
* Reference Picture Selection Indication
*/
PSFB_TYPE_RPSI,
/**
* Application layer Feedback
*/
PSFB_TYPE_AFB,
/**
* Full Intra Request Command
*/
PSFB_TYPE_FIR,
/**
* Temporal-Spatial Trade-off Request
*/
PSFB_TYPE_TSTR,
/**
* Temporal-Spatial Trade-off Notification
*/
PSFB_TYPE_TSTN,
/**
* Video Back Channel Message
*/
PSFB_TYPE_VBCN,
}
/**
* Different types of SDES content.
*/
/**
* Different types of SDES content.
*/
export namespace RTCPSDESType {
export const $gtype: GObject.GType;
}
enum RTCPSDESType {
/**
* Invalid SDES entry
*/
INVALID,
/**
* End of SDES list
*/
END,
/**
* Canonical name
*/
CNAME,
/**
* User name
*/
NAME,
/**
* User's electronic mail address
*/
EMAIL,
/**
* User's phone number
*/
PHONE,
/**
* Geographic user location
*/
LOC,
/**
* Name of application or tool
*/
TOOL,
/**
* Notice about the source
*/
NOTE,
/**
* Private extensions
*/
PRIV,
/**
* H.323 callable address
*/
H323_CADDR,
/**
* Application Specific Identifier (RFC6776)
*/
APSI,
/**
* Reporting Group Identifier (RFC8861)
*/
RGRP,
/**
* RtpStreamId SDES item (RFC8852).
*/
RTP_STREAM_ID,
/**
* RepairedRtpStreamId SDES item (RFC8852).
*/
REPAIRED_RTP_STREAM_ID,
/**
* CLUE CaptId (RFC8849)
*/
CCID,
/**
* MID SDES item (RFC8843).
*/
MID,
}
/**
* Different RTCP packet types.
*/
/**
* Different RTCP packet types.
*/
export namespace RTCPType {
export const $gtype: GObject.GType;
}
enum RTCPType {
/**
* Invalid type
*/
INVALID,
/**
* Sender report
*/
SR,
/**
* Receiver report
*/
RR,
/**
* Source description
*/
SDES,
/**
* Goodbye
*/
BYE,
/**
* Application defined
*/
APP,
/**
* Transport layer feedback.
*/
RTPFB,
/**
* Payload-specific feedback.
*/
PSFB,
/**
* Extended report.
*/
XR,
}
/**
* Types of RTCP Extended Reports, those are defined in RFC 3611 and other RFCs
* according to the [IANA registry](https://www.iana.org/assignments/rtcp-xr-block-types/rtcp-xr-block-types.xhtml).
*/
/**
* Types of RTCP Extended Reports, those are defined in RFC 3611 and other RFCs
* according to the [IANA registry](https://www.iana.org/assignments/rtcp-xr-block-types/rtcp-xr-block-types.xhtml).
*/
export namespace RTCPXRType {
export const $gtype: GObject.GType;
}
enum RTCPXRType {
/**
* Invalid XR Report Block
*/
INVALID,
/**
* Loss RLE Report Block
*/
LRLE,
/**
* Duplicate RLE Report Block
*/
DRLE,
/**
* Packet Receipt Times Report Block
*/
PRT,
/**
* Receiver Reference Time Report Block
*/
RRT,
/**
* Delay since the last Receiver Report
*/
DLRR,
/**
* Statistics Summary Report Block
*/
SSUMM,
/**
* VoIP Metrics Report Block
*/
VOIP_METRICS,
}
/**
* Standard predefined fixed payload types.
*
* The official list is at:
* http://www.iana.org/assignments/rtp-parameters
*
* Audio:
* reserved: 19
* unassigned: 20-23,
*
* Video:
* unassigned: 24, 27, 29, 30, 35-71, 77-95
* Reserved for RTCP conflict avoidance: 72-76
*/
/**
* Standard predefined fixed payload types.
*
* The official list is at:
* http://www.iana.org/assignments/rtp-parameters
*
* Audio:
* reserved: 19
* unassigned: 20-23,
*
* Video:
* unassigned: 24, 27, 29, 30, 35-71, 77-95
* Reserved for RTCP conflict avoidance: 72-76
*/
class RTPPayload {
static $gtype: GObject.GType;
// Static fields
/**
* ITU-T G.711. mu-law audio (RFC 3551)
*/
static PCMU: number;
/**
* RFC 3551 says reserved
*/
static '1016': number;
/**
* RFC 3551 says reserved
*/
static G721: number;
/**
* GSM audio
*/
static GSM: number;
/**
* ITU G.723.1 audio
*/
static G723: number;
/**
* IMA ADPCM wave type (RFC 3551)
*/
static DVI4_8000: number;
/**
* IMA ADPCM wave type (RFC 3551)
*/
static DVI4_16000: number;
/**
* experimental linear predictive encoding
*/
static LPC: number;
/**
* ITU-T G.711 A-law audio (RFC 3551)
*/
static PCMA: number;
/**
* ITU-T G.722 (RFC 3551)
*/
static G722: number;
/**
* stereo PCM
*/
static L16_STEREO: number;
/**
* mono PCM
*/
static L16_MONO: number;
/**
* EIA & TIA standard IS-733
*/
static QCELP: number;
/**
* Comfort Noise (RFC 3389)
*/
static CN: number;
/**
* Audio MPEG 1-3.
*/
static MPA: number;
/**
* ITU-T G.728 Speech coder (RFC 3551)
*/
static G728: number;
/**
* IMA ADPCM wave type (RFC 3551)
*/
static DVI4_11025: number;
/**
* IMA ADPCM wave type (RFC 3551)
*/
static DVI4_22050: number;
/**
* ITU-T G.729 Speech coder (RFC 3551)
*/
static G729: number;
/**
* See RFC 2029
*/
static CELLB: number;
/**
* ISO Standards 10918-1 and 10918-2 (RFC 2435)
*/
static JPEG: number;
/**
* nv encoding by Ron Frederick
*/
static NV: number;
/**
* ITU-T Recommendation H.261 (RFC 2032)
*/
static H261: number;
/**
* Video MPEG 1 & 2 (RFC 2250)
*/
static MPV: number;
/**
* MPEG-2 transport stream (RFC 2250)
*/
static MP2T: number;
/**
* Video H263 (RFC 2190)
*/
static H263: number;
// Constructors
_init(...args: any[]): void;
}
/**
* The transfer profile to use.
*/
/**
* The transfer profile to use.
*/
export namespace RTPProfile {
export const $gtype: GObject.GType;
}
enum RTPProfile {
/**
* invalid profile
*/
UNKNOWN,
/**
* the Audio/Visual profile (RFC 3551)
*/
AVP,
/**
* the secure Audio/Visual profile (RFC 3711)
*/
SAVP,
/**
* the Audio/Visual profile with feedback (RFC 4585)
*/
AVPF,
/**
* the secure Audio/Visual profile with feedback (RFC 5124)
*/
SAVPF,
}
/**
* The maximum amount of SSRCs in a BYE packet.
*/
const RTCP_MAX_BYE_SSRC_COUNT: number;
/**
* The maximum amount of Receiver report blocks in RR and SR messages.
*/
const RTCP_MAX_RB_COUNT: number;
/**
* The maximum text length for an SDES item.
*/
const RTCP_MAX_SDES: number;
/**
* The maximum amount of SDES items.
*/
const RTCP_MAX_SDES_ITEM_COUNT: number;
/**
* Mask for version and packet type pair allowing reduced size
* packets, basically it accepts other types than RR and SR
*/
const RTCP_REDUCED_SIZE_VALID_MASK: number;
/**
* Mask for version, padding bit and packet type pair
*/
const RTCP_VALID_MASK: number;
/**
* Valid value for the first two bytes of an RTCP packet after applying
* #GST_RTCP_VALID_MASK to them.
*/
const RTCP_VALID_VALUE: number;
/**
* The supported RTCP version 2.
*/
const RTCP_VERSION: number;
const RTP_HDREXT_BASE: string;
/**
* Constant string used in element classification to signal that this element
* is a RTP header extension.
*/
const RTP_HDREXT_ELEMENT_CLASS: string;
const RTP_HDREXT_NTP_56: string;
const RTP_HDREXT_NTP_56_SIZE: number;
const RTP_HDREXT_NTP_64: string;
const RTP_HDREXT_NTP_64_SIZE: number;
const RTP_HEADER_EXTENSION_URI_METADATA_KEY: string;
const RTP_PAYLOAD_1016_STRING: string;
const RTP_PAYLOAD_CELLB_STRING: string;
const RTP_PAYLOAD_CN_STRING: string;
const RTP_PAYLOAD_DVI4_11025_STRING: string;
const RTP_PAYLOAD_DVI4_16000_STRING: string;
const RTP_PAYLOAD_DVI4_22050_STRING: string;
const RTP_PAYLOAD_DVI4_8000_STRING: string;
const RTP_PAYLOAD_DYNAMIC_STRING: string;
const RTP_PAYLOAD_G721_STRING: string;
const RTP_PAYLOAD_G722_STRING: string;
const RTP_PAYLOAD_G723_53: number;
const RTP_PAYLOAD_G723_53_STRING: string;
const RTP_PAYLOAD_G723_63: number;
const RTP_PAYLOAD_G723_63_STRING: string;
const RTP_PAYLOAD_G723_STRING: string;
const RTP_PAYLOAD_G728_STRING: string;
const RTP_PAYLOAD_G729_STRING: string;
const RTP_PAYLOAD_GSM_STRING: string;
const RTP_PAYLOAD_H261_STRING: string;
const RTP_PAYLOAD_H263_STRING: string;
const RTP_PAYLOAD_JPEG_STRING: string;
const RTP_PAYLOAD_L16_MONO_STRING: string;
const RTP_PAYLOAD_L16_STEREO_STRING: string;
const RTP_PAYLOAD_LPC_STRING: string;
const RTP_PAYLOAD_MP2T_STRING: string;
const RTP_PAYLOAD_MPA_STRING: string;
const RTP_PAYLOAD_MPV_STRING: string;
const RTP_PAYLOAD_NV_STRING: string;
const RTP_PAYLOAD_PCMA_STRING: string;
const RTP_PAYLOAD_PCMU_STRING: string;
const RTP_PAYLOAD_QCELP_STRING: string;
const RTP_PAYLOAD_TS41: number;
const RTP_PAYLOAD_TS41_STRING: string;
const RTP_PAYLOAD_TS48: number;
const RTP_PAYLOAD_TS48_STRING: string;
const RTP_SOURCE_META_MAX_CSRC_COUNT: number;
/**
* The supported RTP version 2.
*/
const RTP_VERSION: number;
/**
* Attaches RTP source information to `buffer`.
* @param buffer a #GstBuffer
* @param ssrc pointer to the SSRC
* @param csrc pointer to the CSRCs
* @returns the #GstRTPSourceMeta on @buffer.
*/
function buffer_add_rtp_source_meta(
buffer: Gst.Buffer,
ssrc?: number | null,
csrc?: number[] | null,
): RTPSourceMeta;
/**
* Find the #GstRTPSourceMeta on `buffer`.
* @param buffer a #GstBuffer
* @returns the #GstRTPSourceMeta or %NULL when there is no such metadata on @buffer.
*/
function buffer_get_rtp_source_meta(buffer: Gst.Buffer): RTPSourceMeta | null;
/**
* Open `buffer` for reading or writing, depending on `flags`. The resulting RTCP
* buffer state is stored in `rtcp`.
* @param buffer a buffer with an RTCP packet
* @param flags flags for the mapping
* @param rtcp resulting #GstRTCPBuffer
*/
function rtcp_buffer_map(buffer: Gst.Buffer, flags: Gst.MapFlags | null, rtcp: RTCPBuffer): boolean;
/**
* Create a new buffer for constructing RTCP packets. The packet will have a
* maximum size of `mtu`.
* @param mtu the maximum mtu size.
* @returns A newly allocated buffer.
*/
function rtcp_buffer_new(mtu: number): Gst.Buffer;
/**
* 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.
* @param data data for the new buffer
* @returns A newly allocated buffer with a copy of @data and of size @len.
*/
function rtcp_buffer_new_copy_data(data: Uint8Array | string): Gst.Buffer;
/**
* 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.
* @param data data for the new buffer
* @returns A newly allocated buffer with @data and of size @len.
*/
function rtcp_buffer_new_take_data(data: Uint8Array | string): Gst.Buffer;
/**
* Check if the data pointed to by `buffer` is a valid RTCP packet using
* gst_rtcp_buffer_validate_data().
* @param buffer the buffer to validate
* @returns TRUE if @buffer is a valid RTCP packet.
*/
function rtcp_buffer_validate(buffer: Gst.Buffer): boolean;
/**
* 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.
* @param data the data to validate
* @returns TRUE if the data points to a valid RTCP packet.
*/
function rtcp_buffer_validate_data(data: Uint8Array | string): boolean;
/**
* 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.
* @param data the data to validate
* @returns TRUE if the data points to a valid RTCP packet.
*/
function rtcp_buffer_validate_data_reduced(data: Uint8Array | string): boolean;
/**
* Check if the data pointed to by `buffer` is a valid RTCP packet using
* gst_rtcp_buffer_validate_reduced().
* @param buffer the buffer to validate
* @returns TRUE if @buffer is a valid RTCP packet.
*/
function rtcp_buffer_validate_reduced(buffer: Gst.Buffer): boolean;
/**
* Converts an NTP time to UNIX nanoseconds. `ntptime` can typically be
* the NTP time of an SR RTCP message and contains, in the upper 32 bits, the
* number of seconds since 1900 and, in the lower 32 bits, the fractional
* seconds. The resulting value will be the number of nanoseconds since 1970.
* @param ntptime an NTP timestamp
* @returns the UNIX time for @ntptime in nanoseconds.
*/
function rtcp_ntp_to_unix(ntptime: number): number;
/**
* Convert `name` into a `GstRTCPSDESType`. `name` is typically a key in a
* #GstStructure containing SDES items.
* @param name a SDES name
* @returns the #GstRTCPSDESType for @name or #GST_RTCP_SDES_PRIV when @name is a private sdes item.
*/
function rtcp_sdes_name_to_type(name: string): RTCPSDESType;
/**
* Converts `type` to the string equivalent. The string is typically used as a
* key in a #GstStructure containing SDES items.
* @param type a #GstRTCPSDESType
* @returns the string equivalent of @type
*/
function rtcp_sdes_type_to_name(type: RTCPSDESType | null): string;
/**
* Converts a UNIX timestamp in nanoseconds to an NTP time. The caller should
* pass a value with nanoseconds since 1970. The NTP time will, in the upper
* 32 bits, contain the number of seconds since 1900 and, in the lower 32
* bits, the fractional seconds. The resulting value can be used as an ntptime
* for constructing SR RTCP packets.
* @param unixtime an UNIX timestamp in nanoseconds
* @returns the NTP time for @unixtime.
*/
function rtcp_unix_to_ntp(unixtime: number): number;
/**
* Allocate enough data in `buffer` to hold an RTP packet with `csrc_count` CSRCs,
* a payload length of `payload_len` and padding of `pad_len`.
* `buffer` must be writable and all previous memory in `buffer` will be freed.
* If `pad_len` is >0, the padding bit will be set. All other RTP header fields
* will be set to 0/FALSE.
* @param buffer a #GstBuffer
* @param payload_len the length of the payload
* @param pad_len the amount of padding
* @param csrc_count the number of CSRC entries
*/
function rtp_buffer_allocate_data(
buffer: Gst.Buffer,
payload_len: number,
pad_len: number,
csrc_count: number,
): void;
/**
* Calculate the header length of an RTP packet with `csrc_count` CSRC entries.
* An RTP packet can have at most 15 CSRC entries.
* @param csrc_count the number of CSRC entries
* @returns The length of an RTP header with @csrc_count CSRC entries.
*/
function rtp_buffer_calc_header_len(csrc_count: number): number;
/**
* Calculate the total length of an RTP packet with a payload size of `payload_len,`
* a padding of `pad_len` and a `csrc_count` CSRC entries.
* @param payload_len the length of the payload
* @param pad_len the amount of padding
* @param csrc_count the number of CSRC entries
* @returns The total length of an RTP header with given parameters.
*/
function rtp_buffer_calc_packet_len(payload_len: number, pad_len: number, csrc_count: number): number;
/**
* Calculate the length of the payload of an RTP packet with size `packet_len,`
* a padding of `pad_len` and a `csrc_count` CSRC entries.
* @param packet_len the length of the total RTP packet
* @param pad_len the amount of padding
* @param csrc_count the number of CSRC entries
* @returns The length of the payload of an RTP packet with given parameters.
*/
function rtp_buffer_calc_payload_len(packet_len: number, pad_len: number, csrc_count: number): number;
/**
* Compare two sequence numbers, taking care of wraparounds. This function
* returns the difference between `seqnum1` and `seqnum2`.
* @param seqnum1 a sequence number
* @param seqnum2 a sequence number
* @returns a negative value if @seqnum1 is bigger than @seqnum2, 0 if they are equal or a positive value if @seqnum1 is smaller than @segnum2.
*/
function rtp_buffer_compare_seqnum(seqnum1: number, seqnum2: number): number;
/**
* Get the default clock-rate for the static payload type `payload_type`.
* @param payload_type the static payload type
* @returns the default clock rate or -1 if the payload type is not static or the clock-rate is undefined.
*/
function rtp_buffer_default_clock_rate(payload_type: number): number;
/**
* Update the `exttimestamp` field with the extended timestamp of `timestamp`
* For the first call of the method, `exttimestamp` should point to a location
* with a value of -1.
*
* This function is able to handle both forward and backward timestamps taking
* into account:
* - timestamp wraparound making sure that the returned value is properly increased.
* - timestamp unwraparound making sure that the returned value is properly decreased.
* @param exttimestamp a previous extended timestamp
* @param timestamp a new timestamp
* @returns The extended timestamp of @timestamp or 0 if the result can't go anywhere backwards.
*/
function rtp_buffer_ext_timestamp(exttimestamp: number, timestamp: number): [number, number];
/**
* Similar to gst_rtp_buffer_get_extension_onebyte_header, but working
* on the #GBytes you get from gst_rtp_buffer_get_extension_bytes.
* Parses RFC 5285 style header extensions with a one byte header. It will
* return the nth extension with the requested id.
* @param bytes #GBytes
* @param bit_pattern The bit-pattern. Anything but 0xBEDE is rejected.
* @param id The ID of the header extension to be read (between 1 and 14).
* @param nth Read the nth extension packet with the requested ID
* @returns TRUE if @bytes had the requested header extension
*/
function rtp_buffer_get_extension_onebyte_header_from_bytes(
bytes: GLib.Bytes | Uint8Array,
bit_pattern: number,
id: number,
nth: number,
): [boolean, Uint8Array];
/**
* Map the contents of `buffer` into `rtp`.
* @param buffer a #GstBuffer
* @param flags #GstMapFlags
* @returns %TRUE if @buffer could be mapped.
*/
function rtp_buffer_map(buffer: Gst.Buffer, flags: Gst.MapFlags | null): [boolean, RTPBuffer];
/**
* Allocate a new #GstBuffer with enough data to hold an RTP packet with
* `csrc_count` CSRCs, a payload length of `payload_len` and padding of `pad_len`.
* All other RTP header fields will be set to 0/FALSE.
* @param payload_len the length of the payload
* @param pad_len the amount of padding
* @param csrc_count the number of CSRC entries
* @returns A newly allocated buffer that can hold an RTP packet with given parameters.
*/
function rtp_buffer_new_allocate(payload_len: number, pad_len: number, csrc_count: number): Gst.Buffer;
/**
* Create a new #GstBuffer that can hold an RTP packet that is exactly
* `packet_len` long. The length of the payload depends on `pad_len` and
* `csrc_count` and can be calculated with gst_rtp_buffer_calc_payload_len().
* All RTP header fields will be set to 0/FALSE.
* @param packet_len the total length of the packet
* @param pad_len the amount of padding
* @param csrc_count the number of CSRC entries
* @returns A newly allocated buffer that can hold an RTP packet of @packet_len.
*/
function rtp_buffer_new_allocate_len(packet_len: number, pad_len: number, csrc_count: number): Gst.Buffer;
/**
* 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.
* @param data data for the new buffer
* @returns A newly allocated buffer with a copy of @data and of size @len.
*/
function rtp_buffer_new_copy_data(data: Uint8Array | string): Gst.Buffer;
/**
* 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.
* @param data data for the new buffer
* @returns A newly allocated buffer with @data and of size @len.
*/
function rtp_buffer_new_take_data(data: Uint8Array | string): Gst.Buffer;
/**
* Retrieve all the factories of the currently registered RTP header
* extensions. Call gst_element_factory_create() with each factory to create
* the associated #GstRTPHeaderExtension.
* @returns a #GList of #GstElementFactory's. Use gst_plugin_feature_list_free() after use
*/
function rtp_get_header_extension_list(): Gst.ElementFactory[];
/**
* Reads the NTP time from the `size` NTP-56 extension bytes in `data` and store the
* result in `ntptime`.
* @param data the data to read from
* @returns %TRUE on success.
*/
function rtp_hdrext_get_ntp_56(data: Uint8Array | string): [boolean, number];
/**
* Reads the NTP time from the `size` NTP-64 extension bytes in `data` and store the
* result in `ntptime`.
* @param data the data to read from
* @returns %TRUE on success.
*/
function rtp_hdrext_get_ntp_64(data: Uint8Array | string): [boolean, number];
/**
* Writes the NTP time in `ntptime` to the format required for the NTP-56 header
* extension. `data` must hold at least #GST_RTP_HDREXT_NTP_56_SIZE bytes.
* @param data the data to write to
* @param size the size of @data
* @param ntptime the NTP time
* @returns %TRUE on success.
*/
function rtp_hdrext_set_ntp_56(data: any | null, size: number, ntptime: number): boolean;
/**
* Writes the NTP time in `ntptime` to the format required for the NTP-64 header
* extension. `data` must hold at least #GST_RTP_HDREXT_NTP_64_SIZE bytes.
* @param data the data to write to
* @param size the size of @data
* @param ntptime the NTP time
* @returns %TRUE on success.
*/
function rtp_hdrext_set_ntp_64(data: any | null, size: number, ntptime: number): boolean;
/**
* Get the #GstRTPPayloadInfo for `media` and `encoding_name`. This function is
* mostly used to get the default clock-rate and bandwidth for dynamic payload
* types specified with `media` and `encoding` name.
*
* The search for `encoding_name` will be performed in a case insensitive way.
* @param media the media to find
* @param encoding_name the encoding name to find
* @returns a #GstRTPPayloadInfo or NULL when no info could be found.
*/
function rtp_payload_info_for_name(media: string, encoding_name: string): RTPPayloadInfo | null;
/**
* Get the #GstRTPPayloadInfo for `payload_type`. This function is
* mostly used to get the default clock-rate and bandwidth for static payload
* types specified with `payload_type`.
* @param payload_type the payload_type to find
* @returns a #GstRTPPayloadInfo or NULL when no info could be found.
*/
function rtp_payload_info_for_pt(payload_type: number): RTPPayloadInfo | null;
function rtp_source_meta_api_get_type(): GObject.GType;
function rtp_source_meta_get_info(): Gst.MetaInfo;
/**
* Additional RTP buffer flags. These flags can potentially be used on any
* buffers carrying RTP packets.
*
* Note that these are only valid for #GstCaps of type: application/x-rtp (x-rtcp).
* They can conflict with other extended buffer flags.
*/
/**
* Additional RTP buffer flags. These flags can potentially be used on any
* buffers carrying RTP packets.
*
* Note that these are only valid for #GstCaps of type: application/x-rtp (x-rtcp).
* They can conflict with other extended buffer flags.
*/
export namespace RTPBufferFlags {
export const $gtype: GObject.GType;
}
enum RTPBufferFlags {
/**
* The #GstBuffer was once wrapped
* in a retransmitted packet as specified by RFC 4588.
*/
RETRANSMISSION,
/**
* The packet represents redundant RTP packet.
* The flag is used in gstrtpstorage to be able to hold the packetback
* and use it only for recovery from packet loss.
* Since: 1.14
*/
REDUNDANT,
/**
* Offset to define more flags.
*/
LAST,
}
/**
* Additional mapping flags for gst_rtp_buffer_map().
*/
/**
* Additional mapping flags for gst_rtp_buffer_map().
*/
export namespace RTPBufferMapFlags {
export const $gtype: GObject.GType;
}
enum RTPBufferMapFlags {
/**
* Skip mapping and validation of RTP
* padding and RTP pad count when present. Useful for buffers where
* the padding may be encrypted.
*/
SKIP_PADDING,
/**
* Offset to define more flags
*/
LAST,
}
/**
* Direction to which to apply the RTP Header Extension
*/
/**
* Direction to which to apply the RTP Header Extension
*/
export namespace RTPHeaderExtensionDirection {
export const $gtype: GObject.GType;
}
enum RTPHeaderExtensionDirection {
/**
* Neither send nor
* receive RTP Header Extensions
*/
INACTIVE,
/**
* Only send RTP Header
* Extensions `GST_RTP_HEADER_EXTENSION_DIRECTION_RECVONLY:` Only
* receive RTP Header Extensions
*/
SENDONLY,
RECVONLY,
/**
* Send and receive RTP
* Header Extensions ext
*/
SENDRECV,
/**
* RTP header extension
* direction is inherited from the stream
*/
INHERITED,
}
/**
* Flags that apply to a RTP Audio/Video header extension.
*/
/**
* Flags that apply to a RTP Audio/Video header extension.
*/
export namespace RTPHeaderExtensionFlags {
export const $gtype: GObject.GType;
}
enum RTPHeaderExtensionFlags {
/**
* The one byte rtp extension header.
* 1-16 data bytes per extension with a maximum of
* 14 extension ids in total.
*/
ONE_BYTE,
/**
* The two byte rtp extension header.
* 256 data bytes per extension with a maximum of 255 (or 256
* including appbits) extensions in total.
*/
TWO_BYTE,
}
namespace RTPBaseAudioPayload {
// Constructor properties interface
interface ConstructorProps extends RTPBasePayload.ConstructorProps {
buffer_list: boolean;
bufferList: boolean;
}
}
/**
* Provides a base class for audio RTP payloaders for frame or sample based
* audio codecs (constant bitrate)
*
* This class derives from GstRTPBasePayload. It can be used for payloading
* audio codecs. It will only work with constant bitrate codecs. It supports
* both frame based and sample based codecs. It takes care of packing up the
* audio data into RTP packets and filling up the headers accordingly. The
* payloading is done based on the maximum MTU (mtu) and the maximum time per
* packet (max-ptime). The general idea is to divide large data buffers into
* smaller RTP packets. The RTP packet size is the minimum of either the MTU,
* max-ptime (if set) or available data. The RTP packet size is always larger or
* equal to min-ptime (if set). If min-ptime is not set, any residual data is
* sent in a last RTP packet. In the case of frame based codecs, the resulting
* RTP packets always contain full frames.
*
* ## Usage
*
* To use this base class, your child element needs to call either
* gst_rtp_base_audio_payload_set_frame_based() or
* gst_rtp_base_audio_payload_set_sample_based(). This is usually done in the
* element's `_init()` function. Then, the child element must call either
* gst_rtp_base_audio_payload_set_frame_options(),
* gst_rtp_base_audio_payload_set_sample_options() or
* gst_rtp_base_audio_payload_set_samplebits_options. Since
* GstRTPBaseAudioPayload derives from GstRTPBasePayload, the child element
* must set any variables or call/override any functions required by that base
* class. The child element does not need to override any other functions
* specific to GstRTPBaseAudioPayload.
*/
class RTPBaseAudioPayload extends RTPBasePayload {
static $gtype: GObject.GType;
// Properties
get buffer_list(): boolean;
set buffer_list(val: boolean);
get bufferList(): boolean;
set bufferList(val: boolean);
// Fields
payload: RTPBasePayload;
base_ts: Gst.ClockTime;
frame_size: number;
frame_duration: number;
sample_size: number;
// Constructors
constructor(properties?: Partial, ...args: any[]);
_init(...args: any[]): void;
// Methods
/**
* Create an RTP buffer and store `payload_len` bytes of the adapter as the
* payload. Set the timestamp on the new buffer to `timestamp` before pushing
* the buffer downstream.
*
* If `payload_len` is -1, all pending bytes will be flushed. If `timestamp` is
* -1, the timestamp will be calculated automatically.
* @param payload_len length of payload
* @param timestamp a #GstClockTime
* @returns a #GstFlowReturn
*/
flush(payload_len: number, timestamp: Gst.ClockTime): Gst.FlowReturn;
/**
* Gets the internal adapter used by the depayloader.
* @returns a #GstAdapter.
*/
get_adapter(): GstBase.Adapter;
/**
* Create an RTP buffer and store `payload_len` bytes of `data` as the
* payload. Set the timestamp on the new buffer to `timestamp` before pushing
* the buffer downstream.
* @param data data to set as payload
* @param timestamp a #GstClockTime
* @returns a #GstFlowReturn
*/
push(data: Uint8Array | string, timestamp: Gst.ClockTime): Gst.FlowReturn;
// Conflicted with GstRtp.RTPBasePayload.push
push(...args: never[]): any;
/**
* Tells #GstRTPBaseAudioPayload that the child element is for a frame based
* audio codec
*/
set_frame_based(): void;
/**
* Sets the options for frame based audio codecs.
* @param frame_duration The duraction of an audio frame in milliseconds.
* @param frame_size The size of an audio frame in bytes.
*/
set_frame_options(frame_duration: number, frame_size: number): void;
/**
* Tells #GstRTPBaseAudioPayload that the child element is for a sample based
* audio codec
*/
set_sample_based(): void;
/**
* Sets the options for sample based audio codecs.
* @param sample_size Size per sample in bytes.
*/
set_sample_options(sample_size: number): void;
/**
* Sets the options for sample based audio codecs.
* @param sample_size Size per sample in bits.
*/
set_samplebits_options(sample_size: number): void;
}
namespace RTPBaseDepayload {
// Signal callback interfaces
interface AddExtension {
(ext: RTPHeaderExtension): void;
}
interface ClearExtensions {
(): void;
}
interface RequestExtension {
(ext_id: number, ext_uri?: string | null): RTPHeaderExtension | null;
}
// Constructor properties interface
interface ConstructorProps extends Gst.Element.ConstructorProps {
auto_header_extension: boolean;
autoHeaderExtension: boolean;
extensions: Gst.ValueArray;
max_reorder: number;
maxReorder: number;
source_info: boolean;
sourceInfo: boolean;
stats: Gst.Structure;
}
}
/**
* Provides a base class for RTP depayloaders
*
* In order to handle RTP header extensions correctly if the
* depayloader aggregates multiple RTP packet payloads into one output
* buffer this class provides the function
* gst_rtp_base_depayload_set_aggregate_hdrext_enabled(). If the
* aggregation is enabled the virtual functions
* `GstRTPBaseDepayload`.process or
* `GstRTPBaseDepayload`.process_rtp_packet must tell the base class
* what happens to the current RTP packet. By default the base class
* assumes that the packet payload is used with the next output
* buffer.
*
* If the RTP packet will not be used with an output buffer
* gst_rtp_base_depayload_dropped() must be called. A typical
* situation would be if we are waiting for a keyframe.
*
* If the RTP packet will be used but not with the current output
* buffer but with the next one gst_rtp_base_depayload_delayed() must
* be called. This may happen if the current RTP packet signals the
* start of a new output buffer and the currently processed output
* buffer will be pushed first. The undelay happens implicitly once
* the current buffer has been pushed or
* gst_rtp_base_depayload_flush() has been called.
*
* If gst_rtp_base_depayload_flush() is called all RTP packets that
* have not been dropped since the last output buffer are dropped,
* e.g. if an output buffer is discarded due to malformed data. This
* may or may not include the current RTP packet depending on the 2nd
* parameter `keep_current`.
*
* Be aware that in case gst_rtp_base_depayload_push_list() is used
* each buffer will see the same list of RTP header extensions.
*/
abstract class RTPBaseDepayload extends Gst.Element {
static $gtype: GObject.GType;
// Properties
/**
* 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.
*/
get auto_header_extension(): boolean;
set auto_header_extension(val: boolean);
/**
* 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.
*/
get autoHeaderExtension(): boolean;
set autoHeaderExtension(val: boolean);
/**
* 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".
*/
get extensions(): Gst.ValueArray;
/**
* 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.
*/
get max_reorder(): number;
set max_reorder(val: 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.
*/
get maxReorder(): number;
set maxReorder(val: number);
/**
* Add RTP source information found in RTP header as meta to output buffer.
*/
get source_info(): boolean;
set source_info(val: boolean);
/**
* Add RTP source information found in RTP header as meta to output buffer.
*/
get sourceInfo(): boolean;
set sourceInfo(val: boolean);
/**
* 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, clock-rate of the stream
* * `npt-start`: #G_TYPE_UINT64, time of playback start
* * `npt-stop`: #G_TYPE_UINT64, time of playback stop
* * `play-speed`: #G_TYPE_DOUBLE, the playback speed
* * `play-scale`: #G_TYPE_DOUBLE, the playback scale
* * `running-time-dts`: #G_TYPE_UINT64, the last running-time of the
* last DTS
* * `running-time-pts`: #G_TYPE_UINT64, the last running-time of the
* last PTS
* * `seqnum`: #G_TYPE_UINT, the last seen seqnum
* * `timestamp`: #G_TYPE_UINT, the last seen RTP timestamp
*/
get stats(): Gst.Structure;
// Fields
sinkpad: Gst.Pad;
srcpad: Gst.Pad;
clock_rate: number;
need_newsegment: boolean;
// Constructors
constructor(properties?: Partial, ...args: any[]);
_init(...args: any[]): void;
// Signals
connect(id: string, callback: (...args: any[]) => any): number;
connect_after(id: string, callback: (...args: any[]) => any): number;
emit(id: string, ...args: any[]): void;
connect(signal: 'add-extension', callback: (_source: this, ext: RTPHeaderExtension) => void): number;
connect_after(signal: 'add-extension', callback: (_source: this, ext: RTPHeaderExtension) => void): number;
emit(signal: 'add-extension', ext: RTPHeaderExtension): void;
connect(signal: 'clear-extensions', callback: (_source: this) => void): number;
connect_after(signal: 'clear-extensions', callback: (_source: this) => void): number;
emit(signal: 'clear-extensions'): void;
connect(
signal: 'request-extension',
callback: (_source: this, ext_id: number, ext_uri: string | null) => RTPHeaderExtension | null,
): number;
connect_after(
signal: 'request-extension',
callback: (_source: this, ext_id: number, ext_uri: string | null) => RTPHeaderExtension | null,
): number;
emit(signal: 'request-extension', ext_id: number, ext_uri?: string | null): void;
// Virtual methods
/**
* custom event handling
* @param event
*/
vfunc_handle_event(event: Gst.Event): boolean;
/**
* signal the depayloader about packet loss
* @param event
*/
vfunc_packet_lost(event: Gst.Event): boolean;
/**
* 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.
* @param _in
*/
vfunc_process(_in: Gst.Buffer): Gst.Buffer;
/**
* 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.
* @param rtp_buffer
*/
vfunc_process_rtp_packet(rtp_buffer: RTPBuffer): Gst.Buffer;
/**
* configure the depayloader
* @param caps
*/
vfunc_set_caps(caps: Gst.Caps): boolean;
// Methods
/**
* 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.
*/
delayed(): void;
/**
* 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.
*/
dropped(): void;
/**
* 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
* gst_rtp_base_depayload_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.
* @param keep_current if the current RTP buffer shall be kept
*/
flush(keep_current: boolean): void;
/**
* Queries whether header extensions will be aggregated per depayloaded buffers.
* @returns %TRUE if aggregate-header-extension is enabled.
*/
is_aggregate_hdrext_enabled(): boolean;
/**
* Queries whether #GstRTPSourceMeta will be added to depayloaded buffers.
* @returns %TRUE if source-info is enabled.
*/
is_source_info_enabled(): boolean;
/**
* 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.
* @param out_buf a #GstBuffer
* @returns a #GstFlowReturn.
*/
push(out_buf: Gst.Buffer): Gst.FlowReturn;
/**
* Push `out_list` to the peer of `filter`. This function takes ownership of
* `out_list`.
* @param out_list a #GstBufferList
* @returns a #GstFlowReturn.
*/
push_list(out_list: Gst.BufferList): Gst.FlowReturn;
/**
* Enable or disable aggregating header extensions.
* @param enable whether to aggregate header extensions per output buffer
*/
set_aggregate_hdrext_enabled(enable: boolean): void;
/**
* Enable or disable adding #GstRTPSourceMeta to depayloaded buffers.
* @param enable whether to add meta about RTP sources to buffer
*/
set_source_info_enabled(enable: boolean): void;
}
namespace RTPBasePayload {
// Signal callback interfaces
interface AddExtension {
(ext: RTPHeaderExtension): void;
}
interface ClearExtensions {
(): void;
}
interface RequestExtension {
(ext_id: number, ext_uri: string): RTPHeaderExtension | null;
}
// Constructor properties interface
interface ConstructorProps extends Gst.Element.ConstructorProps {
auto_header_extension: boolean;
autoHeaderExtension: boolean;
extensions: Gst.ValueArray;
max_ptime: number;
maxPtime: number;
min_ptime: number;
minPtime: number;
mtu: number;
onvif_no_rate_control: boolean;
onvifNoRateControl: boolean;
perfect_rtptime: boolean;
perfectRtptime: boolean;
pt: number;
ptime_multiple: number;
ptimeMultiple: number;
scale_rtptime: boolean;
scaleRtptime: boolean;
seqnum: number;
seqnum_offset: number;
seqnumOffset: number;
source_info: boolean;
sourceInfo: boolean;
ssrc: number;
stats: Gst.Structure;
timestamp: number;
timestamp_offset: number;
timestampOffset: number;
}
}
/**
* Provides a base class for RTP payloaders
*/
abstract class RTPBasePayload extends Gst.Element {
static $gtype: GObject.GType;
// Properties
/**
* 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.
*/
get auto_header_extension(): boolean;
set auto_header_extension(val: boolean);
/**
* 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.
*/
get autoHeaderExtension(): boolean;
set autoHeaderExtension(val: boolean);
/**
* 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".
*/
get extensions(): Gst.ValueArray;
get max_ptime(): number;
set max_ptime(val: number);
get maxPtime(): number;
set maxPtime(val: number);
/**
* Minimum duration of the packet data in ns (can't go above MTU)
*/
get min_ptime(): number;
set min_ptime(val: number);
/**
* Minimum duration of the packet data in ns (can't go above MTU)
*/
get minPtime(): number;
set minPtime(val: number);
get mtu(): number;
set mtu(val: number);
/**
* Make the payloader timestamp packets according to the Rate-Control=no
* behaviour specified in the ONVIF replay spec.
*/
get onvif_no_rate_control(): boolean;
set onvif_no_rate_control(val: boolean);
/**
* Make the payloader timestamp packets according to the Rate-Control=no
* behaviour specified in the ONVIF replay spec.
*/
get onvifNoRateControl(): boolean;
set onvifNoRateControl(val: boolean);
/**
* 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.
*/
get perfect_rtptime(): boolean;
set perfect_rtptime(val: boolean);
/**
* 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.
*/
get perfectRtptime(): boolean;
set perfectRtptime(val: boolean);
get pt(): number;
set pt(val: number);
/**
* Force buffers to be multiples of this duration in ns (0 disables)
*/
get ptime_multiple(): number;
set ptime_multiple(val: number);
/**
* Force buffers to be multiples of this duration in ns (0 disables)
*/
get ptimeMultiple(): number;
set ptimeMultiple(val: 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.
*/
get scale_rtptime(): boolean;
set scale_rtptime(val: boolean);
/**
* 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.
*/
get scaleRtptime(): boolean;
set scaleRtptime(val: boolean);
get seqnum(): number;
get seqnum_offset(): number;
set seqnum_offset(val: number);
get seqnumOffset(): number;
set seqnumOffset(val: number);
/**
* Enable writing the CSRC field in allocated RTP header based on RTP source
* information found in the input buffer's #GstRTPSourceMeta.
*/
get source_info(): boolean;
set source_info(val: boolean);
/**
* Enable writing the CSRC field in allocated RTP header based on RTP source
* information found in the input buffer's #GstRTPSourceMeta.
*/
get sourceInfo(): boolean;
set sourceInfo(val: boolean);
get ssrc(): number;
set ssrc(val: number);
/**
* 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 #GstRTPBasePayload:seqnum
* * `timestamp` :#G_TYPE_UINT, RTP timestamp, same as #GstRTPBasePayload:timestamp
* * `ssrc` :#G_TYPE_UINT, The SSRC in use
* * `pt` :#G_TYPE_UINT, The Payload type in use, same as #GstRTPBasePayload: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
*/
get stats(): Gst.Structure;
get timestamp(): number;
get timestamp_offset(): number;
set timestamp_offset(val: number);
get timestampOffset(): number;
set timestampOffset(val: number);
// Fields
element: Gst.Element;
// Constructors
constructor(properties?: Partial, ...args: any[]);
_init(...args: any[]): void;
// Signals
connect(id: string, callback: (...args: any[]) => any): number;
connect_after(id: string, callback: (...args: any[]) => any): number;
emit(id: string, ...args: any[]): void;
connect(signal: 'add-extension', callback: (_source: this, ext: RTPHeaderExtension) => void): number;
connect_after(signal: 'add-extension', callback: (_source: this, ext: RTPHeaderExtension) => void): number;
emit(signal: 'add-extension', ext: RTPHeaderExtension): void;
connect(signal: 'clear-extensions', callback: (_source: this) => void): number;
connect_after(signal: 'clear-extensions', callback: (_source: this) => void): number;
emit(signal: 'clear-extensions'): void;
connect(
signal: 'request-extension',
callback: (_source: this, ext_id: number, ext_uri: string) => RTPHeaderExtension | null,
): number;
connect_after(
signal: 'request-extension',
callback: (_source: this, ext_id: number, ext_uri: string) => RTPHeaderExtension | null,
): number;
emit(signal: 'request-extension', ext_id: number, ext_uri: string): void;
// Virtual methods
/**
* get desired caps
* @param pad
* @param filter
*/
vfunc_get_caps(pad: Gst.Pad, filter: Gst.Caps): Gst.Caps;
/**
* process data
* @param buffer
*/
vfunc_handle_buffer(buffer: Gst.Buffer): Gst.FlowReturn;
/**
* custom query handling
* @param pad
* @param query
*/
vfunc_query(pad: Gst.Pad, query: Gst.Query): boolean;
// Conflicted with Gst.Element.vfunc_query
vfunc_query(...args: never[]): any;
/**
* configure the payloader
* @param caps
*/
vfunc_set_caps(caps: Gst.Caps): boolean;
/**
* custom event handling on the sinkpad
* @param event
*/
vfunc_sink_event(event: Gst.Event): boolean;
/**
* custom event handling on the srcpad
* @param event
*/
vfunc_src_event(event: Gst.Event): boolean;
// Methods
/**
* 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 #GstRTPBasePayload:source-info %TRUE additional
* CSRCs may be allocated and filled with RTP source information.
* @param payload_len the length of the payload
* @param pad_len the amount of padding
* @param csrc_count the minimum number of CSRC entries
* @returns A newly allocated buffer that can hold an RTP packet with given parameters.
*/
allocate_output_buffer(payload_len: number, pad_len: number, csrc_count: number): Gst.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 #GstRTPBasePayload:source-info is %FALSE the count will be 0.
* @param buffer a #GstBuffer, typically the buffer to payload
* @returns The number of sources.
*/
get_source_count(buffer: Gst.Buffer): number;
/**
* Check if the packet with `size` and `duration` would exceed the configured
* maximum size.
* @param size the size of the packet
* @param duration the duration of the packet
* @returns %TRUE if the packet of @size and @duration would exceed the configured MTU or max_ptime.
*/
is_filled(size: number, duration: Gst.ClockTime): boolean;
/**
* Queries whether the payloader will add contributing sources (CSRCs) to the
* RTP header from #GstRTPSourceMeta.
* @returns %TRUE if source-info is enabled.
*/
is_source_info_enabled(): boolean;
/**
* 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`.
* @param buffer a #GstBuffer
* @returns a #GstFlowReturn.
*/
push(buffer: Gst.Buffer): 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`.
* @param list a #GstBufferList
* @returns a #GstFlowReturn.
*/
push_list(list: Gst.BufferList): Gst.FlowReturn;
/**
* 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().
* @param media the media type (typically "audio" or "video")
* @param dynamic if the payload type is dynamic
* @param encoding_name the encoding name
* @param clock_rate the clock rate of the media
*/
set_options(media: string, dynamic: boolean, encoding_name: string, clock_rate: number): void;
/**
* Configure the output caps with the optional fields.
* @param s a #GstStructure with the caps fields
* @returns %TRUE if the caps could be set.
*/
set_outcaps_structure(s?: Gst.Structure | null): boolean;
/**
* Enable or disable adding contributing sources to RTP packets from
* #GstRTPSourceMeta.
* @param enable whether to add contributing sources to RTP packets
*/
set_source_info_enabled(enable: boolean): void;
}
namespace RTPHeaderExtension {
// Constructor properties interface
interface ConstructorProps extends Gst.Element.ConstructorProps {}
}
/**
* Instance struct for a RTP Audio/Video header extension.
*/
abstract class RTPHeaderExtension extends Gst.Element {
static $gtype: GObject.GType;
// Constructors
constructor(properties?: Partial, ...args: any[]);
_init(...args: any[]): void;
// Static methods
static create_from_uri(uri: string): RTPHeaderExtension | null;
static set_uri(uri: string): void;
// Virtual methods
/**
* This is used to know how much data a certain header extension will need for
* both allocating the resulting data, and deciding how much payload data can
* be generated.
*
* Implementations should return as accurate a value as is possible using the
* information given in the input `buffer`.
* @param input_meta a #GstBuffer
*/
vfunc_get_max_size(input_meta: Gst.Buffer): number;
/**
* retrieve the supported flags
*/
vfunc_get_supported_flags(): RTPHeaderExtensionFlags;
/**
* Read the RTP header extension from `data`.
* @param read_flags #GstRTPHeaderExtensionFlags for how the extension should be written
* @param data location to read the rtp header extension from
* @param buffer a #GstBuffer to modify if necessary
*/
vfunc_read(read_flags: RTPHeaderExtensionFlags, data: Uint8Array | string, buffer: Gst.Buffer): boolean;
/**
* set the necessary attributes that may be signaled e.g. with
* an SDP.
* @param direction
* @param attributes
*/
vfunc_set_attributes(direction: RTPHeaderExtensionDirection, attributes: string): boolean;
/**
* gst_rtp_header_extension_set_id() must have been called with a valid
* extension id that is contained in these caps.
*
* The only current known caps format is based on the SDP standard as produced
* by gst_sdp_media_attributes_to_caps().
* @param caps writable #GstCaps to modify
*/
vfunc_set_caps_from_attributes(caps: Gst.Caps): boolean;
/**
* Passes RTP payloader's sink (i.e. not payloaded) `caps` to the header
* extension.
* @param caps sink #GstCaps
*/
vfunc_set_non_rtp_sink_caps(caps: Gst.Caps): boolean;
/**
* Updates depayloader src caps based on the information received in RTP header.
* `caps` must be writable as this function may modify them.
* @param caps src #GstCaps to modify
*/
vfunc_update_non_rtp_src_caps(caps: Gst.Caps): boolean;
/**
* Writes the RTP header extension to `data` using information available from
* the `input_meta`. `data` will be sized to be at least the value returned
* from gst_rtp_header_extension_get_max_size().
* @param input_meta the input #GstBuffer to read information from if necessary
* @param write_flags #GstRTPHeaderExtensionFlags for how the extension should be written
* @param output output RTP #GstBuffer
* @param data location to write the rtp header extension into
*/
vfunc_write(
input_meta: Gst.Buffer,
write_flags: RTPHeaderExtensionFlags,
output: Gst.Buffer,
data: Uint8Array | string,
): number;
// Methods
/**
* Retrieve the direction
* @returns The direction
*/
get_direction(): RTPHeaderExtensionDirection;
get_id(): number;
/**
* This is used to know how much data a certain header extension will need for
* both allocating the resulting data, and deciding how much payload data can
* be generated.
*
* Implementations should return as accurate a value as is possible using the
* information given in the input `buffer`.
* @param input_meta a #GstBuffer
* @returns the maximum size of the data written by this extension
*/
get_max_size(input_meta: Gst.Buffer): number;
get_sdp_caps_field_name(): string;
get_supported_flags(): RTPHeaderExtensionFlags;
get_uri(): string | null;
/**
* Read the RTP header extension from `data`.
* @param read_flags #GstRTPHeaderExtensionFlags for how the extension should be written
* @param data location to read the rtp header extension from
* @param buffer a #GstBuffer to modify if necessary
* @returns whether the extension could be read from @data
*/
read(read_flags: RTPHeaderExtensionFlags | null, data: Uint8Array | string, buffer: Gst.Buffer): boolean;
/**
* gst_rtp_header_extension_set_id() must have been called with a valid
* extension id that is contained in these caps.
*
* The only current known caps format is based on the SDP standard as produced
* by gst_sdp_media_attributes_to_caps().
* @param caps the #GstCaps to configure this extension with
* @returns whether the @caps could be successfully set on @ext.
*/
set_attributes_from_caps(caps: Gst.Caps): boolean;
/**
* gst_rtp_header_extension_set_id() must have been called with a valid
* extension id that is contained in these caps.
*
* The only current known caps format is based on the SDP standard as produced
* by gst_sdp_media_attributes_to_caps().
* @param caps writable #GstCaps to modify
* @returns whether the configured attributes on @ext can successfully be set on @caps
*/
set_caps_from_attributes(caps: Gst.Caps): boolean;
/**
* Helper implementation for GstRTPExtensionClass::set_caps_from_attributes
* that sets the `ext` uri on caps with the specified extension id as required
* for sdp #GstCaps.
*
* Requires that the extension does not have any attributes or direction
* advertised in `caps`.
* @param caps #GstCaps to write fields into
* @param attributes
* @returns whether the @ext attributes could be set on @caps.
*/
set_caps_from_attributes_helper(caps: Gst.Caps, attributes: string): boolean;
/**
* Set the direction that this header extension should be used in.
* If #GST_RTP_HEADER_EXTENSION_DIRECTION_INHERITED is included, the
* direction will not be included in the caps (as it shouldn't be in the
* extmap line in the SDP).
* @param direction The direction
*/
set_direction(direction: RTPHeaderExtensionDirection | null): void;
/**
* sets the RTP extension id on `ext`
* @param ext_id The id of this extension
*/
set_id(ext_id: number): void;
/**
* Passes RTP payloader's sink (i.e. not payloaded) `caps` to the header
* extension.
* @param caps sink #GstCaps
* @returns Whether @caps could be read successfully
*/
set_non_rtp_sink_caps(caps: Gst.Caps): boolean;
/**
* Call this function in a subclass from #GstRTPHeaderExtensionClass::read to
* tell the depayloader whether the data just parsed from RTP packet require
* updating its src (non-RTP) caps. If `state` is TRUE, #GstRTPBaseDepayload will
* eventually invoke gst_rtp_header_extension_update_non_rtp_src_caps() to
* have the caps update applied. Applying the update also flips the internal
* "wants update" flag back to FALSE.
* @param state TRUE if caps update is needed
*/
set_wants_update_non_rtp_src_caps(state: boolean): void;
/**
* Updates depayloader src caps based on the information received in RTP header.
* `caps` must be writable as this function may modify them.
* @param caps src #GstCaps to modify
* @returns whether @caps were modified successfully
*/
update_non_rtp_src_caps(caps: Gst.Caps): boolean;
/**
* Call this function after gst_rtp_header_extension_read() to check if
* the depayloader's src caps need updating with data received in the last RTP
* packet.
* @returns Whether @ext wants to update depayloader's src caps.
*/
wants_update_non_rtp_src_caps(): boolean;
/**
* Writes the RTP header extension to `data` using information available from
* the `input_meta`. `data` will be sized to be at least the value returned
* from gst_rtp_header_extension_get_max_size().
* @param input_meta the input #GstBuffer to read information from if necessary
* @param write_flags #GstRTPHeaderExtensionFlags for how the extension should be written
* @param output output RTP #GstBuffer
* @param data location to write the rtp header extension into
* @returns the size of the data written, < 0 on failure
*/
write(
input_meta: Gst.Buffer,
write_flags: RTPHeaderExtensionFlags | null,
output: Gst.Buffer,
data: Uint8Array | string,
): number;
}
/**
* Note: The API in this module is not yet declared stable.
*
* The GstRTPCBuffer helper functions makes it easy to parse and create regular
* #GstBuffer objects that contain compound RTCP packets. These buffers are typically
* of 'application/x-rtcp' #GstCaps.
*
* An RTCP buffer consists of 1 or more #GstRTCPPacket structures that you can
* retrieve with gst_rtcp_buffer_get_first_packet(). #GstRTCPPacket acts as a pointer
* into the RTCP buffer; you can move to the next packet with
* gst_rtcp_packet_move_to_next().
*/
class RTCPBuffer {
static $gtype: GObject.GType;
// Constructors
_init(...args: any[]): void;
// Static methods
/**
* Open `buffer` for reading or writing, depending on `flags`. The resulting RTCP
* buffer state is stored in `rtcp`.
* @param buffer a buffer with an RTCP packet
* @param flags flags for the mapping
* @param rtcp resulting #GstRTCPBuffer
*/
static map(buffer: Gst.Buffer, flags: Gst.MapFlags, rtcp: RTCPBuffer): boolean;
/**
* Create a new buffer for constructing RTCP packets. The packet will have a
* maximum size of `mtu`.
* @param mtu the maximum mtu size.
*/
static ['new'](mtu: number): Gst.Buffer;
/**
* 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.
* @param data data for the new buffer
*/
static new_copy_data(data: Uint8Array | string): Gst.Buffer;
/**
* 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.
* @param data data for the new buffer
*/
static new_take_data(data: Uint8Array | string): Gst.Buffer;
/**
* Check if the data pointed to by `buffer` is a valid RTCP packet using
* gst_rtcp_buffer_validate_data().
* @param buffer the buffer to validate
*/
static validate(buffer: Gst.Buffer): boolean;
/**
* 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.
* @param data the data to validate
*/
static validate_data(data: Uint8Array | string): boolean;
/**
* 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.
* @param data the data to validate
*/
static validate_data_reduced(data: Uint8Array | string): boolean;
/**
* Check if the data pointed to by `buffer` is a valid RTCP packet using
* gst_rtcp_buffer_validate_reduced().
* @param buffer the buffer to validate
*/
static validate_reduced(buffer: Gst.Buffer): boolean;
// Methods
/**
* Add a new packet of `type` to `rtcp`. `packet` will point to the newly created
* packet.
* @param type the #GstRTCPType of the new packet
* @param packet pointer to new packet
* @returns %TRUE if the packet could be created. This function returns %FALSE if the max mtu is exceeded for the buffer.
*/
add_packet(type: RTCPType | null, packet: RTCPPacket): boolean;
/**
* Initialize a new #GstRTCPPacket pointer that points to the first packet in
* `rtcp`.
* @param packet a #GstRTCPPacket
* @returns TRUE if the packet existed in @rtcp.
*/
get_first_packet(packet: RTCPPacket): boolean;
/**
* Get the number of RTCP packets in `rtcp`.
* @returns the number of RTCP packets in @rtcp.
*/
get_packet_count(): number;
/**
* Finish `rtcp` after being constructed. This function is usually called
* after gst_rtcp_buffer_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.
*/
unmap(): boolean;
}
/**
* Data structure that points to a packet at `offset` in `buffer`.
* The size of the structure is made public to allow stack allocations.
*/
class RTCPPacket {
static $gtype: GObject.GType;
// Fields
offset: number;
// Constructors
_init(...args: any[]): void;
// Methods
/**
* Add profile-specific extension `data` to `packet`. If `packet` already
* contains profile-specific extension `data` will be appended to the existing
* extension.
* @param data profile-specific data
* @returns %TRUE if the profile specific extension data was added.
*/
add_profile_specific_ext(data: Uint8Array | string): boolean;
/**
* Add a new report block to `packet` with the given values.
* @param ssrc data source being reported
* @param fractionlost fraction lost since last SR/RR
* @param packetslost the cumululative number of packets lost
* @param exthighestseq the extended last sequence number received
* @param jitter the interarrival jitter
* @param lsr the last SR packet from this source
* @param dlsr the delay since last SR packet
* @returns %TRUE if the packet was created. This function can return %FALSE if the max MTU is exceeded or the number of report blocks is greater than #GST_RTCP_MAX_RB_COUNT.
*/
add_rb(
ssrc: number,
fractionlost: number,
packetslost: number,
exthighestseq: number,
jitter: number,
lsr: number,
dlsr: number,
): boolean;
/**
* Get the application-dependent data attached to a RTPFB or PSFB `packet`.
* @returns A pointer to the data
*/
app_get_data(): number;
/**
* Get the length of the application-dependent data attached to an APP
* `packet`.
* @returns The length of data in 32-bit words.
*/
app_get_data_length(): number;
/**
* Get the name field of the APP `packet`.
* @returns The 4-byte name field, not zero-terminated.
*/
app_get_name(): string;
/**
* Get the SSRC/CSRC field of the APP `packet`.
* @returns The SSRC/CSRC.
*/
app_get_ssrc(): number;
/**
* Get the subtype field of the APP `packet`.
* @returns The subtype.
*/
app_get_subtype(): number;
/**
* Set the length of the application-dependent data attached to an APP
* `packet`.
* @param wordlen Length of the data in 32-bit words
* @returns %TRUE if there was enough space in the packet to add this much data.
*/
app_set_data_length(wordlen: number): boolean;
/**
* Set the name field of the APP `packet`.
* @param name 4-byte ASCII name
*/
app_set_name(name: string): void;
/**
* Set the SSRC/CSRC field of the APP `packet`.
* @param ssrc SSRC/CSRC of the packet
*/
app_set_ssrc(ssrc: number): void;
/**
* Set the subtype field of the APP `packet`.
* @param subtype subtype of the packet
*/
app_set_subtype(subtype: number): void;
/**
* Add `ssrc` to the BYE `packet`.
* @param ssrc an SSRC to add
* @returns %TRUE if the ssrc was added. This function can return %FALSE if the max MTU is exceeded or the number of sources blocks is greater than #GST_RTCP_MAX_BYE_SSRC_COUNT.
*/
bye_add_ssrc(ssrc: number): boolean;
/**
* Adds `len` SSRCs in `ssrc` to BYE `packet`.
* @param ssrc an array of SSRCs to add
* @returns %TRUE if the all the SSRCs were added. This function can return %FALSE if the max MTU is exceeded or the number of sources blocks is greater than #GST_RTCP_MAX_BYE_SSRC_COUNT.
*/
bye_add_ssrcs(ssrc: number[]): boolean;
/**
* Get the `nth` SSRC of the BYE `packet`.
* @param nth the nth SSRC to get
* @returns The @nth SSRC of @packet.
*/
bye_get_nth_ssrc(nth: number): number;
/**
* Get the reason in `packet`.
* @returns The reason for the BYE @packet or NULL if the packet did not contain a reason string. The string must be freed with g_free() after usage.
*/
bye_get_reason(): string | null;
/**
* Get the length of the reason string.
* @returns The length of the reason string or 0 when there is no reason string present.
*/
bye_get_reason_len(): number;
/**
* Get the number of SSRC fields in `packet`.
* @returns The number of SSRC fields in @packet.
*/
bye_get_ssrc_count(): number;
/**
* Set the reason string to `reason` in `packet`.
* @param reason a reason string
* @returns TRUE if the string could be set.
*/
bye_set_reason(reason: string): boolean;
/**
* The profile-specific extension data is copied into a new allocated
* memory area `data`. This must be freed with g_free() after usage.
* @returns %TRUE if there was valid data.
*/
copy_profile_specific_ext(): [boolean, Uint8Array];
/**
* Get the Feedback Control Information attached to a RTPFB or PSFB `packet`.
* @returns a pointer to the FCI
*/
fb_get_fci(): number;
/**
* Get the length of the Feedback Control Information attached to a
* RTPFB or PSFB `packet`.
* @returns The length of the FCI in 32-bit words.
*/
fb_get_fci_length(): number;
/**
* Get the media SSRC field of the RTPFB or PSFB `packet`.
* @returns the media SSRC.
*/
fb_get_media_ssrc(): number;
/**
* Get the sender SSRC field of the RTPFB or PSFB `packet`.
* @returns the sender SSRC.
*/
fb_get_sender_ssrc(): number;
/**
* Get the feedback message type of the FB `packet`.
* @returns The feedback message type.
*/
fb_get_type(): RTCPFBType;
/**
* Set the length of the Feedback Control Information attached to a
* RTPFB or PSFB `packet`.
* @param wordlen Length of the FCI in 32-bit words
* @returns %TRUE if there was enough space in the packet to add this much FCI
*/
fb_set_fci_length(wordlen: number): boolean;
/**
* Set the media SSRC field of the RTPFB or PSFB `packet`.
* @param ssrc a media SSRC
*/
fb_set_media_ssrc(ssrc: number): void;
/**
* Set the sender SSRC field of the RTPFB or PSFB `packet`.
* @param ssrc a sender SSRC
*/
fb_set_sender_ssrc(ssrc: number): void;
/**
* Set the feedback message type of the FB `packet`.
* @param type the #GstRTCPFBType to set
*/
fb_set_type(type: RTCPFBType | null): void;
/**
* Get the count field in `packet`.
* @returns The count field in @packet or -1 if @packet does not point to a valid packet.
*/
get_count(): number;
/**
* Get the length field of `packet`. This is the length of the packet in
* 32-bit words minus one.
* @returns The length field of @packet.
*/
get_length(): number;
/**
* Get the packet padding of the packet pointed to by `packet`.
* @returns If the packet has the padding bit set.
*/
get_padding(): boolean;
get_profile_specific_ext(): [boolean, Uint8Array];
get_profile_specific_ext_length(): number;
/**
* Parse the values of the `nth` report block in `packet` and store the result in
* the values.
* @param nth the nth report block in @packet
*/
get_rb(nth: number): [number, number, number, number, number, number, number];
/**
* Get the number of report blocks in `packet`.
* @returns The number of report blocks in @packet.
*/
get_rb_count(): number;
/**
* Get the packet type of the packet pointed to by `packet`.
* @returns The packet type or GST_RTCP_TYPE_INVALID when @packet is not pointing to a valid packet.
*/
get_type(): RTCPType;
/**
* Move the packet pointer `packet` to the next packet in the payload.
* Use gst_rtcp_buffer_get_first_packet() to initialize `packet`.
* @returns TRUE if @packet is pointing to a valid packet after calling this function.
*/
move_to_next(): boolean;
/**
* Removes the packet pointed to by `packet` and moves pointer to the next one
* @returns TRUE if @packet is pointing to a valid packet after calling this function.
*/
remove(): boolean;
/**
* Get the ssrc field of the RR `packet`.
* @returns the ssrc.
*/
rr_get_ssrc(): number;
/**
* Set the ssrc field of the RR `packet`.
* @param ssrc the SSRC to set
*/
rr_set_ssrc(ssrc: number): void;
/**
* Add a new SDES entry to the current item in `packet`.
* @param type the #GstRTCPSDESType of the SDES entry
* @param data the data
* @returns %TRUE if the item could be added, %FALSE if the MTU has been reached.
*/
sdes_add_entry(type: RTCPSDESType | null, data: Uint8Array | string): boolean;
/**
* Add a new SDES item for `ssrc` to `packet`.
* @param ssrc the SSRC of the new item to add
* @returns %TRUE if the item could be added, %FALSE if the maximum amount of items has been exceeded for the SDES packet or the MTU has been reached.
*/
sdes_add_item(ssrc: number): boolean;
/**
* This function is like gst_rtcp_packet_sdes_get_entry() but it returns a
* null-terminated copy of the data instead. use g_free() after usage.
* @param type result of the entry type
* @returns %TRUE if there was valid data.
*/
sdes_copy_entry(type: RTCPSDESType | null): [boolean, Uint8Array];
/**
* Move to the first SDES entry in the current item.
* @returns %TRUE if there was a first entry.
*/
sdes_first_entry(): boolean;
/**
* Move to the first SDES item in `packet`.
* @returns TRUE if there was a first item.
*/
sdes_first_item(): boolean;
/**
* 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
* gst_rtcp_packet_sdes_copy_entry() to get a null-terminated copy of the entry.
* @param type result of the entry type
* @returns %TRUE if there was valid data.
*/
sdes_get_entry(type: RTCPSDESType | null): [boolean, Uint8Array];
/**
* Get the number of items in the SDES packet `packet`.
* @returns The number of items in @packet.
*/
sdes_get_item_count(): number;
/**
* Get the SSRC of the current SDES item.
* @returns the SSRC of the current item.
*/
sdes_get_ssrc(): number;
/**
* Move to the next SDES entry in the current item.
* @returns %TRUE if there was a next entry.
*/
sdes_next_entry(): boolean;
/**
* Move to the next SDES item in `packet`.
* @returns TRUE if there was a next item.
*/
sdes_next_item(): boolean;
/**
* Set the `nth` new report block in `packet` with the given values.
*
* Note: Not implemented.
* @param nth the nth report block to set
* @param ssrc data source being reported
* @param fractionlost fraction lost since last SR/RR
* @param packetslost the cumululative number of packets lost
* @param exthighestseq the extended last sequence number received
* @param jitter the interarrival jitter
* @param lsr the last SR packet from this source
* @param dlsr the delay since last SR packet
*/
set_rb(
nth: number,
ssrc: number,
fractionlost: number,
packetslost: number,
exthighestseq: number,
jitter: number,
lsr: number,
dlsr: number,
): void;
/**
* Parse the SR sender info and store the values.
*/
sr_get_sender_info(): [number, number, number, number, number];
/**
* Set the given values in the SR packet `packet`.
* @param ssrc the SSRC
* @param ntptime the NTP time
* @param rtptime the RTP time
* @param packet_count the packet count
* @param octet_count the octet count
*/
sr_set_sender_info(
ssrc: number,
ntptime: number,
rtptime: number,
packet_count: number,
octet_count: number,
): void;
/**
* Move to the first extended report block in XR `packet`.
* @returns TRUE if there was a first extended report block.
*/
xr_first_rb(): boolean;
xr_get_block_length(): number;
/**
* Get the extended report block type of the XR `packet`.
* @returns The extended report block type.
*/
xr_get_block_type(): RTCPXRType;
/**
* Parse the extended report block for DLRR report block type.
* @param nth the index of sub-block to retrieve.
* @param ssrc the SSRC of the receiver.
* @param last_rr the last receiver reference timestamp of @ssrc.
* @param delay the delay since @last_rr.
* @returns %TRUE if the report block is correctly parsed.
*/
xr_get_dlrr_block(nth: number, ssrc: number, last_rr: number, delay: number): boolean;
/**
* Retrieve the packet receipt time of `seq` which ranges in [begin_seq, end_seq).
* @param seq the sequence to retrieve the time.
* @param receipt_time the packet receipt time of @seq.
* @returns %TRUE if the report block returns the receipt time correctly.
*/
xr_get_prt_by_seq(seq: number, receipt_time: number): boolean;
/**
* Parse the Packet Recept Times Report Block from a XR `packet`
* @param ssrc the SSRC of the RTP data packet source being reported upon by this report block.
* @param thinning the amount of thinning performed on the sequence number space.
* @param begin_seq the first sequence number that this block reports on.
* @param end_seq the last sequence number that this block reports on plus one.
* @returns %TRUE if the report block is correctly parsed.
*/
xr_get_prt_info(ssrc: number, thinning: number, begin_seq: number, end_seq: number): boolean;
/**
* Parse the extended report block for Loss RLE and Duplicated LRE block type.
* @param ssrc the SSRC of the RTP data packet source being reported upon by this report block.
* @param thinning the amount of thinning performed on the sequence number space.
* @param begin_seq the first sequence number that this block reports on.
* @param end_seq the last sequence number that this block reports on plus one.
* @param chunk_count the number of chunks calculated by block length.
* @returns %TRUE if the report block is correctly parsed.
*/
xr_get_rle_info(
ssrc: number,
thinning: number,
begin_seq: number,
end_seq: number,
chunk_count: number,
): boolean;
/**
* Retrieve actual chunk data.
* @param nth the index of chunk to retrieve.
* @param chunk the @nth chunk.
* @returns %TRUE if the report block returns chunk correctly.
*/
xr_get_rle_nth_chunk(nth: number, chunk: number): boolean;
xr_get_rrt(timestamp: number): boolean;
/**
* Get the ssrc field of the XR `packet`.
* @returns the ssrc.
*/
xr_get_ssrc(): number;
/**
* Extract a basic information from static summary report block of XR `packet`.
* @param ssrc the SSRC of the source.
* @param begin_seq the first sequence number that this block reports on.
* @param end_seq the last sequence number that this block reports on plus one.
* @returns %TRUE if the report block is correctly parsed.
*/
xr_get_summary_info(ssrc: number, begin_seq: number, end_seq: number): boolean;
/**
* Extract jitter information from the statistics summary. If the jitter flag in
* a block header is set as zero, all of jitters will be zero.
* @param min_jitter the minimum relative transit time between two sequences.
* @param max_jitter the maximum relative transit time between two sequences.
* @param mean_jitter the mean relative transit time between two sequences.
* @param dev_jitter the standard deviation of the relative transit time between two sequences.
* @returns %TRUE if the report block is correctly parsed.
*/
xr_get_summary_jitter(
min_jitter: number,
max_jitter: number,
mean_jitter: number,
dev_jitter: number,
): boolean;
/**
* Get the number of lost or duplicate packets. If the flag in a block header
* is set as zero, `lost_packets` or `dup_packets` will be zero.
* @param lost_packets the number of lost packets between begin_seq and end_seq.
* @param dup_packets the number of duplicate packets between begin_seq and end_seq.
* @returns %TRUE if the report block is correctly parsed.
*/
xr_get_summary_pkt(lost_packets: number, dup_packets: number): boolean;
/**
* Extract the value of ttl for ipv4, or hop limit for ipv6.
* @param is_ipv4 the flag to indicate that the return values are ipv4 ttl or ipv6 hop limits.
* @param min_ttl the minimum TTL or Hop Limit value of data packets between two sequences.
* @param max_ttl the maximum TTL or Hop Limit value of data packets between two sequences.
* @param mean_ttl the mean TTL or Hop Limit value of data packets between two sequences.
* @param dev_ttl the standard deviation of the TTL or Hop Limit value of data packets between two sequences.
* @returns %TRUE if the report block is correctly parsed.
*/
xr_get_summary_ttl(
is_ipv4: boolean,
min_ttl: number,
max_ttl: number,
mean_ttl: number,
dev_ttl: number,
): boolean;
xr_get_voip_burst_metrics(
burst_density: number,
gap_density: number,
burst_duration: number,
gap_duration: number,
): boolean;
xr_get_voip_configuration_params(gmin: number, rx_config: number): boolean;
xr_get_voip_delay_metrics(roundtrip_delay: number, end_system_delay: number): boolean;
xr_get_voip_jitter_buffer_params(jb_nominal: number, jb_maximum: number, jb_abs_max: number): boolean;
xr_get_voip_metrics_ssrc(ssrc: number): boolean;
xr_get_voip_packet_metrics(loss_rate: number, discard_rate: number): boolean;
xr_get_voip_quality_metrics(
r_factor: number,
ext_r_factor: number,
mos_lq: number,
mos_cq: number,
): boolean;
xr_get_voip_signal_metrics(signal_level: number, noise_level: number, rerl: number, gmin: number): boolean;
/**
* Move to the next extended report block in XR `packet`.
* @returns TRUE if there was a next extended report block.
*/
xr_next_rb(): boolean;
}
type RTPBaseAudioPayloadClass = typeof RTPBaseAudioPayload;
abstract class RTPBaseAudioPayloadPrivate {
static $gtype: GObject.GType;
// Constructors
_init(...args: any[]): void;
}
type RTPBaseDepayloadClass = typeof RTPBaseDepayload;
abstract class RTPBaseDepayloadPrivate {
static $gtype: GObject.GType;
// Constructors
_init(...args: any[]): void;
}
type RTPBasePayloadClass = typeof RTPBasePayload;
abstract class RTPBasePayloadPrivate {
static $gtype: GObject.GType;
// Constructors
_init(...args: any[]): void;
}
/**
* The GstRTPBuffer helper functions makes it easy to parse and create regular
* #GstBuffer objects that contain RTP payloads. These buffers are typically of
* 'application/x-rtp' #GstCaps.
*/
class RTPBuffer {
static $gtype: GObject.GType;
// Fields
state: number;
data: any[];
size: number[];
// Constructors
_init(...args: any[]): void;
// Static methods
/**
* Allocate enough data in `buffer` to hold an RTP packet with `csrc_count` CSRCs,
* a payload length of `payload_len` and padding of `pad_len`.
* `buffer` must be writable and all previous memory in `buffer` will be freed.
* If `pad_len` is >0, the padding bit will be set. All other RTP header fields
* will be set to 0/FALSE.
* @param buffer a #GstBuffer
* @param payload_len the length of the payload
* @param pad_len the amount of padding
* @param csrc_count the number of CSRC entries
*/
static allocate_data(buffer: Gst.Buffer, payload_len: number, pad_len: number, csrc_count: number): void;
/**
* Calculate the header length of an RTP packet with `csrc_count` CSRC entries.
* An RTP packet can have at most 15 CSRC entries.
* @param csrc_count the number of CSRC entries
*/
static calc_header_len(csrc_count: number): number;
/**
* Calculate the total length of an RTP packet with a payload size of `payload_len,`
* a padding of `pad_len` and a `csrc_count` CSRC entries.
* @param payload_len the length of the payload
* @param pad_len the amount of padding
* @param csrc_count the number of CSRC entries
*/
static calc_packet_len(payload_len: number, pad_len: number, csrc_count: number): number;
/**
* Calculate the length of the payload of an RTP packet with size `packet_len,`
* a padding of `pad_len` and a `csrc_count` CSRC entries.
* @param packet_len the length of the total RTP packet
* @param pad_len the amount of padding
* @param csrc_count the number of CSRC entries
*/
static calc_payload_len(packet_len: number, pad_len: number, csrc_count: number): number;
/**
* Compare two sequence numbers, taking care of wraparounds. This function
* returns the difference between `seqnum1` and `seqnum2`.
* @param seqnum1 a sequence number
* @param seqnum2 a sequence number
*/
static compare_seqnum(seqnum1: number, seqnum2: number): number;
/**
* Get the default clock-rate for the static payload type `payload_type`.
* @param payload_type the static payload type
*/
static default_clock_rate(payload_type: number): number;
/**
* Update the `exttimestamp` field with the extended timestamp of `timestamp`
* For the first call of the method, `exttimestamp` should point to a location
* with a value of -1.
*
* This function is able to handle both forward and backward timestamps taking
* into account:
* - timestamp wraparound making sure that the returned value is properly increased.
* - timestamp unwraparound making sure that the returned value is properly decreased.
* @param exttimestamp a previous extended timestamp
* @param timestamp a new timestamp
*/
static ext_timestamp(exttimestamp: number, timestamp: number): [number, number];
/**
* Similar to gst_rtp_buffer_get_extension_onebyte_header, but working
* on the #GBytes you get from gst_rtp_buffer_get_extension_bytes.
* Parses RFC 5285 style header extensions with a one byte header. It will
* return the nth extension with the requested id.
* @param bytes #GBytes
* @param bit_pattern The bit-pattern. Anything but 0xBEDE is rejected.
* @param id The ID of the header extension to be read (between 1 and 14).
* @param nth Read the nth extension packet with the requested ID
*/
static get_extension_onebyte_header_from_bytes(
bytes: GLib.Bytes | Uint8Array,
bit_pattern: number,
id: number,
nth: number,
): [boolean, Uint8Array];
/**
* Map the contents of `buffer` into `rtp`.
* @param buffer a #GstBuffer
* @param flags #GstMapFlags
*/
static map(buffer: Gst.Buffer, flags: Gst.MapFlags): [boolean, RTPBuffer];
/**
* Allocate a new #GstBuffer with enough data to hold an RTP packet with
* `csrc_count` CSRCs, a payload length of `payload_len` and padding of `pad_len`.
* All other RTP header fields will be set to 0/FALSE.
* @param payload_len the length of the payload
* @param pad_len the amount of padding
* @param csrc_count the number of CSRC entries
*/
static new_allocate(payload_len: number, pad_len: number, csrc_count: number): Gst.Buffer;
/**
* Create a new #GstBuffer that can hold an RTP packet that is exactly
* `packet_len` long. The length of the payload depends on `pad_len` and
* `csrc_count` and can be calculated with gst_rtp_buffer_calc_payload_len().
* All RTP header fields will be set to 0/FALSE.
* @param packet_len the total length of the packet
* @param pad_len the amount of padding
* @param csrc_count the number of CSRC entries
*/
static new_allocate_len(packet_len: number, pad_len: number, csrc_count: number): Gst.Buffer;
/**
* 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.
* @param data data for the new buffer
*/
static new_copy_data(data: Uint8Array | string): Gst.Buffer;
/**
* 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.
* @param data data for the new buffer
*/
static new_take_data(data: Uint8Array | string): Gst.Buffer;
// Methods
/**
* Adds a RFC 5285 header extension with a one byte header to the end of the
* RTP header. If there is already a RFC 5285 header extension with a one byte
* header, the new extension will be appended.
* It will not work if there is already a header extension that does not follow
* the mechanism described in RFC 5285 or if there is a header extension with
* a two bytes header as described in RFC 5285. In that case, use
* gst_rtp_buffer_add_extension_twobytes_header()
* @param id The ID of the header extension (between 1 and 14).
* @param data location for data
* @returns %TRUE if header extension could be added
*/
add_extension_onebyte_header(id: number, data: Uint8Array | string): boolean;
/**
* Adds a RFC 5285 header extension with a two bytes header to the end of the
* RTP header. If there is already a RFC 5285 header extension with a two bytes
* header, the new extension will be appended.
* It will not work if there is already a header extension that does not follow
* the mechanism described in RFC 5285 or if there is a header extension with
* a one byte header as described in RFC 5285. In that case, use
* gst_rtp_buffer_add_extension_onebyte_header()
* @param appbits Application specific bits
* @param id The ID of the header extension
* @param data location for data
* @returns %TRUE if header extension could be added
*/
add_extension_twobytes_header(appbits: number, id: number, data: Uint8Array | string): boolean;
/**
* Get the CSRC at index `idx` in `buffer`.
* @param idx the index of the CSRC to get
* @returns the CSRC at index @idx in host order.
*/
get_csrc(idx: number): number;
/**
* Get the CSRC count of the RTP packet in `buffer`.
* @returns the CSRC count of @buffer.
*/
get_csrc_count(): number;
/**
* Check if the extension bit is set on the RTP packet in `buffer`.
* @returns TRUE if @buffer has the extension bit set.
*/
get_extension(): boolean;
/**
* Similar to gst_rtp_buffer_get_extension_data, but more suitable for language
* bindings usage. `bits` will contain the extension 16 bits of custom data and
* the extension data (not including the extension header) is placed in a new
* #GBytes structure.
*
* If `rtp` did not contain an extension, this function will return %NULL, with
* `bits` unchanged. If there is an extension header but no extension data then
* an empty #GBytes will be returned.
* @returns A new #GBytes if an extension header was present and %NULL otherwise.
*/
get_extension_data(): [GLib.Bytes | null, number];
/**
* Parses RFC 5285 style header extensions with a one byte header. It will
* return the nth extension with the requested id.
* @param id The ID of the header extension to be read (between 1 and 14).
* @param nth Read the nth extension packet with the requested ID
* @returns TRUE if @buffer had the requested header extension
*/
get_extension_onebyte_header(id: number, nth: number): [boolean, Uint8Array | null];
/**
* Parses RFC 5285 style header extensions with a two bytes header. It will
* return the nth extension with the requested id.
* @param id The ID of the header extension to be read (between 1 and 14).
* @param nth Read the nth extension packet with the requested ID
* @returns TRUE if @buffer had the requested header extension
*/
get_extension_twobytes_header(id: number, nth: number): [boolean, number, Uint8Array | null];
/**
* Return the total length of the header in `buffer`. This include the length of
* the fixed header, the CSRC list and the extension header.
* @returns The total length of the header in @buffer.
*/
get_header_len(): number;
/**
* Check if the marker bit is set on the RTP packet in `buffer`.
* @returns TRUE if @buffer has the marker bit set.
*/
get_marker(): boolean;
/**
* Return the total length of the packet in `buffer`.
* @returns The total length of the packet in @buffer.
*/
get_packet_len(): number;
/**
* Check if the padding bit is set on the RTP packet in `buffer`.
* @returns TRUE if @buffer has the padding bit set.
*/
get_padding(): boolean;
/**
* Create a buffer of the payload of the RTP packet in `buffer`. This function
* will internally create a subbuffer of `buffer` so that a memcpy can be
* avoided.
* @returns A new buffer with the data of the payload.
*/
get_payload_buffer(): Gst.Buffer;
/**
* Similar to gst_rtp_buffer_get_payload, but more suitable for language
* bindings usage. The return value is a pointer to a #GBytes structure
* containing the payload data in `rtp`.
* @returns A new #GBytes containing the payload data in @rtp.
*/
get_payload(): GLib.Bytes | null;
/**
* Get the length of the payload of the RTP packet in `buffer`.
* @returns The length of the payload in @buffer.
*/
get_payload_len(): number;
/**
* Create a subbuffer of the payload of the RTP packet in `buffer`. `offset` bytes
* are skipped in the payload and the subbuffer will be of size `len`.
* If `len` is -1 the total payload starting from `offset` is subbuffered.
* @param offset the offset in the payload
* @param len the length in the payload
* @returns A new buffer with the specified data of the payload.
*/
get_payload_subbuffer(offset: number, len: number): Gst.Buffer;
/**
* Get the payload type of the RTP packet in `buffer`.
* @returns The payload type.
*/
get_payload_type(): number;
/**
* Get the sequence number of the RTP packet in `buffer`.
* @returns The sequence number in host order.
*/
get_seq(): number;
/**
* Get the SSRC of the RTP packet in `buffer`.
* @returns the SSRC of @buffer in host order.
*/
get_ssrc(): number;
/**
* Get the timestamp of the RTP packet in `buffer`.
* @returns The timestamp in host order.
*/
get_timestamp(): number;
/**
* Get the version number of the RTP packet in `buffer`.
* @returns The version of @buffer.
*/
get_version(): number;
/**
* Set the amount of padding in the RTP packet in `buffer` to
* `len`. If `len` is 0, the padding is removed.
*
* NOTE: This function does not work correctly.
* @param len the new amount of padding
*/
pad_to(len: number): void;
/**
* Unsets the extension bit of the RTP buffer and removes the extension header
* and data.
*
* If the RTP buffer has no header extension data, the action has no effect.
* The RTP buffer must be mapped READWRITE only once and the underlying
* GstBuffer must be writable.
*/
remove_extension_data(): void;
/**
* Modify the CSRC at index `idx` in `buffer` to `csrc`.
* @param idx the CSRC index to set
* @param csrc the CSRC in host order to set at @idx
*/
set_csrc(idx: number, csrc: number): void;
/**
* Set the extension bit on the RTP packet in `buffer` to `extension`.
* @param extension the new extension
*/
set_extension(extension: boolean): void;
/**
* Set the extension bit of the rtp buffer and fill in the `bits` and `length` of the
* extension header. If the existing extension data is not large enough, it will
* be made larger.
*
* Will also shorten the extension data from 1.20.
* @param bits the bits specific for the extension
* @param length the length that counts the number of 32-bit words in the extension, excluding the extension header ( therefore zero is a valid length)
* @returns True if done.
*/
set_extension_data(bits: number, length: number): boolean;
/**
* Set the marker bit on the RTP packet in `buffer` to `marker`.
* @param marker the new marker
*/
set_marker(marker: boolean): void;
/**
* Set the total `rtp` size to `len`. The data in the buffer will be made
* larger if needed. Any padding will be removed from the packet.
* @param len the new packet length
*/
set_packet_len(len: number): void;
/**
* Set the padding bit on the RTP packet in `buffer` to `padding`.
* @param padding the new padding
*/
set_padding(padding: boolean): void;
/**
* Set the payload type of the RTP packet in `buffer` to `payload_type`.
* @param payload_type the new type
*/
set_payload_type(payload_type: number): void;
/**
* Set the sequence number of the RTP packet in `buffer` to `seq`.
* @param seq the new sequence number
*/
set_seq(seq: number): void;
/**
* Set the SSRC on the RTP packet in `buffer` to `ssrc`.
* @param ssrc the new SSRC
*/
set_ssrc(ssrc: number): void;
/**
* Set the timestamp of the RTP packet in `buffer` to `timestamp`.
* @param timestamp the new timestamp
*/
set_timestamp(timestamp: number): void;
/**
* Set the version of the RTP packet in `buffer` to `version`.
* @param version the new version
*/
set_version(version: number): void;
/**
* Unmap `rtp` previously mapped with gst_rtp_buffer_map().
*/
unmap(): void;
}
type RTPHeaderExtensionClass = typeof RTPHeaderExtension;
/**
* Structure holding default payload type information.
*/
class RTPPayloadInfo {
static $gtype: GObject.GType;
// Fields
payload_type: number;
media: string;
encoding_name: string;
clock_rate: number;
encoding_parameters: string;
bitrate: number;
// Constructors
constructor(
properties?: Partial<{
payload_type: number;
media: string;
encoding_name: string;
clock_rate: number;
encoding_parameters: string;
bitrate: number;
}>,
);
_init(...args: any[]): void;
// Static methods
/**
* Get the #GstRTPPayloadInfo for `media` and `encoding_name`. This function is
* mostly used to get the default clock-rate and bandwidth for dynamic payload
* types specified with `media` and `encoding` name.
*
* The search for `encoding_name` will be performed in a case insensitive way.
* @param media the media to find
* @param encoding_name the encoding name to find
*/
static for_name(media: string, encoding_name: string): RTPPayloadInfo | null;
/**
* Get the #GstRTPPayloadInfo for `payload_type`. This function is
* mostly used to get the default clock-rate and bandwidth for static payload
* types specified with `payload_type`.
* @param payload_type the payload_type to find
*/
static for_pt(payload_type: number): RTPPayloadInfo | null;
}
/**
* Meta describing the source(s) of the buffer.
*/
class RTPSourceMeta {
static $gtype: GObject.GType;
// Fields
ssrc: number;
ssrc_valid: boolean;
csrc: number[];
csrc_count: number;
// Constructors
_init(...args: any[]): void;
// Static methods
static get_info(): Gst.MetaInfo;
// Methods
/**
* Appends `csrc` to the list of contributing sources in `meta`.
* @param csrc the csrcs to append
* @returns %TRUE if all elements in @csrc was added, %FALSE otherwise.
*/
append_csrc(csrc: number[]): boolean;
/**
* Count the total number of RTP sources found in `meta,` both SSRC and CSRC.
* @returns The number of RTP sources
*/
get_source_count(): number;
/**
* Sets `ssrc` in `meta`. If `ssrc` is %NULL the ssrc of `meta` will be unset.
* @param ssrc pointer to the SSRC
* @returns %TRUE on success, %FALSE otherwise.
*/
set_ssrc(ssrc?: number | null): boolean;
}
/**
* Name of the imported GIR library
* `see` https://gitlab.gnome.org/GNOME/gjs/-/blob/master/gi/ns.cpp#L188
*/
const __name__: string;
/**
* Version of the imported GIR library
* `see` https://gitlab.gnome.org/GNOME/gjs/-/blob/master/gi/ns.cpp#L189
*/
const __version__: string;
}
export default GstRtp;
}
declare module 'gi://GstRtp' {
import GstRtp10 from 'gi://GstRtp?version=1.0';
export default GstRtp10;
}
// END