rtpjitterbuffer: GStreamer Good Plugins 1.0 Plugins Reference Manual

## rtpjitterbuffer rtpjitterbuffer |

## Properties

gboolean | do-lost | Read / Write
gboolean | drop-on-latency | Read / Write
guint | latency | Read / Write
gint64 | ts-offset | Read / Write
RTPJitterBufferMode | mode | Read / Write
gint | percent | Read
gboolean | do-retransmission | Read / Write
gint | rtx-delay | Read / Write
gint | rtx-delay-reorder | Read / Write
gint | rtx-retry-period | Read / Write
gint | rtx-retry-timeout | Read / Write
GstStructure * | stats | Read
guint | rtx-min-delay | Read / Write
gint | rtx-min-retry-timeout | Read / Write
gint | rtx-max-retries | Read / Write
gboolean | rtx-next-seqnum | Read / Write
guint | max-dropout-time | Read / Write
guint | max-misorder-time | Read / Write
gint | max-rtcp-rtp-time-diff | Read / Write

## Signals

void | clear-pt-map | Action
void | handle-sync | Run Last
void | on-npt-stop | Run Last
GstCaps* | request-pt-map | Run Last
guint64 | set-active | Action

## Types and Values

struct | GstRtpJitterBuffer

## Object Hierarchy

        GObject
        ╰── GInitiallyUnowned
            ╰── GstObject
                ╰── GstElement
                    ╰── GstRtpJitterBuffer

## Description

This element reorders and removes duplicate RTP packets as they are received from a network source.

The element needs the clock-rate of the RTP payload in order to estimate the delay. This information is obtained either from the caps on the sink pad or, when no caps are present, from the “request-pt-map” signal. To clear the previous pt-map use the “clear-pt-map” signal.

The rtpjitterbuffer will wait for missing packets up to a configurable time limit using the “latency” property. Packets arriving too late are considered to be lost packets. If the “do-lost” property is set, lost packets will result in a custom serialized downstream event of name GstRTPPacketLost. The lost packet events are usually used by a depayloader or other element to create concealment data or some other logic to gracefully handle the missing packets.

The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incomming buffer and the rtptime inside the RTP packet to create a PTS on the outgoing buffer.

The jitterbuffer can also be configured to send early retransmission events upstream by setting the “do-retransmission” property. In this mode, the jitterbuffer tries to estimate when a packet should arrive and sends a custom upstream event named GstRTPRetransmissionRequest when the packet is considered late. The initial expected packet arrival time is calculated as follows:

- If seqnum N arrived at time T, seqnum N+1 is expected to arrive at T + packet-spacing + “rtx-delay”. The packet spacing is calculated from the DTS (or PTS is no DTS) of two consecutive RTP packets with different rtptime.

- If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm, seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any previously scheduled timeout is overwritten.

- If seqnum N arrived, all seqnum older than N - “rtx-delay-reorder” are considered late immediately. This is to request fast feedback for abonormally reorder packets before any of the previous timeouts is triggered.

A late packet triggers the GstRTPRetransmissionRequest custom upstream event. After the initial timeout expires and the retransmission event is sent, the timeout is scheduled for T + “rtx-retry-timeout”. If the missing packet did not arrive after “rtx-retry-timeout”, a new GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled again for T + “rtx-retry-timeout”. This repeats until “rtx-retry-period” elapsed, at which point no further retransmission requests are sent and the regular logic is performed to schedule a lost packet as discussed above.

This element acts as a live element and so adds “latency” to the pipeline.

This element will automatically be used inside rtpbin.

### Example pipelines

    1
    gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink

Connect to a streaming server and decode the MPEG video. The jitterbuffer is inserted into the pipeline to smooth out network jitter and to reorder the out-of-order RTP packets.

## Synopsis

### Element Information

plugin | rtpmanager
author | Philippe Kalaf <philippe.kalaf@collabora.co.uk>, Wim Taymans <wim.taymans@gmail.com>
class | Filter/Network/RTP

### Element Pads

name | sink
direction | sink
presence | always
details | application/x-rtp

name | sink_rtcp
direction | sink
presence | request
details | application/x-rtcp

name | src
direction | source
presence | always
details | application/x-rtp

## Functions

## Types and Values

### struct GstRtpJitterBuffer

    struct GstRtpJitterBuffer;

Opaque jitterbuffer structure.

## Property Details

### The `“do-lost”` property

      “do-lost”                  gboolean

Send out a GstRTPPacketLost event downstream when a packet is considered lost.

Flags: Read / Write

Default value: FALSE

### The `“drop-on-latency”` property

      “drop-on-latency”          gboolean

Drop oldest buffers when the queue is completely filled.

Flags: Read / Write

Default value: FALSE

### The `“latency”` property

      “latency”                  guint

The maximum latency of the jitterbuffer. Packets will be kept in the buffer for at most this time.

Flags: Read / Write

Default value: 200

### The `“ts-offset”` property

      “ts-offset”                gint64

Adjust GStreamer output buffer timestamps in the jitterbuffer with offset. This is mainly used to ensure interstream synchronisation.

Flags: Read / Write

Default value: 0

### The `“mode”` property

      “mode”                     RTPJitterBufferMode

Control the buffering and timestamping mode used by the jitterbuffer.

Flags: Read / Write

Default value: Slave receiver to sender clock

### The `“percent”` property

      “percent”                  gint

The percent of the jitterbuffer that is filled.

Flags: Read

Allowed values: [0,100]

Default value: 0

### The `“do-retransmission”` property

      “do-retransmission”        gboolean

Send out a GstRTPRetransmission event upstream when a packet is considered late and should be retransmitted.

Flags: Read / Write

Default value: FALSE

Since: 1.2

### The `“rtx-delay”` property

      “rtx-delay”                gint

When a packet did not arrive at the expected time, wait this extra amount of time before sending a retransmission event.

When -1 is used, the max jitter will be used as extra delay.

Flags: Read / Write

Allowed values: >= G_MAXULONG

Default value: -1

Since: 1.2

### The `“rtx-delay-reorder”` property

      “rtx-delay-reorder”        gint

Assume that a retransmission event should be sent when we see this much packet reordering.

When -1 is used, the value will be estimated based on observed packet reordering.

Flags: Read / Write

Allowed values: >= G_MAXULONG

Default value: 3

Since: 1.2

### The `“rtx-retry-period”` property

      “rtx-retry-period”         gint

The amount of time to try to get a retransmission.

When -1 is used, the value will be estimated based on the jitterbuffer latency and the observed round trip time.

Flags: Read / Write

Allowed values: >= G_MAXULONG

Default value: -1

Since: 1.2

### The `“rtx-retry-timeout”` property

      “rtx-retry-timeout”        gint

Retry sending a transmission event after this timeout in ms (-1 automatic).

Flags: Read / Write

Allowed values: >= G_MAXULONG

Default value: -1

### The `“stats”` property

      “stats”                    GstStructure *

Various jitterbuffer statistics. This property returns a GstStructure with name application/x-rtp-jitterbuffer-stats with the following fields:

- guint64 `"rtx-count"`: the number of retransmissions requested.

- guint64 `"rtx-success-count"`: the number of successful retransmissions.

- gdouble `"rtx-per-packet"`: average number of RTX per packet.

- guint64 `"rtx-rtt"`: average round trip time per RTX.

Flags: Read

Since: 1.4

### The `“rtx-min-delay”` property

      “rtx-min-delay”            guint

When a packet did not arrive at the expected time, wait at least this extra amount of time before sending a retransmission event.

Flags: Read / Write

Default value: 0

Since: 1.6

### The `“rtx-min-retry-timeout”` property

      “rtx-min-retry-timeout”    gint

Minimum timeout between sending a transmission event in ms (-1 automatic).

Flags: Read / Write

Allowed values: >= G_MAXULONG

Default value: -1

### The `“rtx-max-retries”` property

      “rtx-max-retries”          gint

The maximum number of retries to request a retransmission.

This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested. When -1 is used, the number of retransmission request will not be limited.

Flags: Read / Write

Allowed values: >= G_MAXULONG

Default value: -1

Since: 1.6

### The `“rtx-next-seqnum”` property

      “rtx-next-seqnum”          gboolean

Estimate when the next packet should arrive and schedule a retransmission request for it. This is, when packet N arrives, a GstRTPRetransmission event is schedule for packet N+1. So it will be requested if it does not arrive at the expected time. The expected time is calculated using the dts of N and the packet spacing.

Flags: Read / Write

Default value: TRUE

Since: 1.6

### The `“max-dropout-time”` property

      “max-dropout-time”         guint

The maximum time (milliseconds) of missing packets tolerated.

Flags: Read / Write

Default value: 60000

### The `“max-misorder-time”` property

      “max-misorder-time”        guint

The maximum time (milliseconds) of misordered packets tolerated.

Flags: Read / Write

Default value: 2000

### The `“max-rtcp-rtp-time-diff”` property

      “max-rtcp-rtp-time-diff”   gint

The maximum amount of time in ms that the RTP time in the RTCP SRs is allowed to be ahead of the last RTP packet we received. Use -1 to disable ignoring of RTCP packets.

Flags: Read / Write

Allowed values: >= G_MAXULONG

Default value: 1000

Since: 1.8

## Signal Details

### The `“clear-pt-map”` signal

    void
    user_function (GstRtpJitterBuffer *buffer,
                   gpointer            user_data)

Invalidate the clock-rate as obtained with the “request-pt-map” signal.

#### Parameters

buffer | the object which received the signal |
user_data | user data set when the signal handler was connected. |

Flags: Action

### The `“handle-sync”` signal

    void
    user_function (GstRtpJitterBuffer *buffer,
                   GstStructure       *struct,
                   gpointer            user_data)

Be notified of new sync values.

#### Parameters

buffer | the object which received the signal |
struct | a GstStructure containing sync values. |
user_data | user data set when the signal handler was connected. |

Flags: Run Last

### The `“on-npt-stop”` signal

    void
    user_function (GstRtpJitterBuffer *buffer,
                   gpointer            user_data)

Signal that the jitterbufer has pushed the RTP packet that corresponds to the npt-stop position.

#### Parameters

buffer | the object which received the signal |
user_data | user data set when the signal handler was connected. |

Flags: Run Last

### The `“request-pt-map”` signal

    GstCaps*
    user_function (GstRtpJitterBuffer *buffer,
                   guint               pt,
                   gpointer            user_data)

Request the payload type as GstCaps for `pt` .

#### Parameters

buffer | the object which received the signal |
pt | the pt |
user_data | user data set when the signal handler was connected. |

Flags: Run Last

### The `“set-active”` signal

    guint64
    user_function (GstRtpJitterBuffer *buffer,
                   gboolean            Returns,
                   guint64             arg2,
                   gpointer            user_data)

Start pushing out packets with the given base time. This signal is only useful in buffering mode.

#### Parameters

buffer | the object which received the signal |
user_data | user data set when the signal handler was connected. |

#### Returns

the time of the last pushed packet.

Flags: Action