JitterBufferConfig

#include <icy/webrtc/jitterbuffer.h>

Receive-side jitter buffer behaviour for depacketized WebRTC media frames.

The jitter buffer sits after libdatachannel depacketization and before icey emits encoded AudioPacket/VideoPacket objects to downstream decoders or recorders. It reorders frames by RTP-derived media timestamp and delays release long enough to absorb moderate network jitter.

Public Attributes

Return	Name	Description
`bool`	`enabled`	False keeps the current zero-buffer receive path.
`std::int64_t`	`minDelayMs`	Base playout delay in milliseconds.
`std::int64_t`	`maxDelayMs`	Upper bound for the adaptive playout delay.
`double`	`adaptiveFactor`	Extra delay multiplier applied to observed inter-arrival variance.
`std::int64_t`	`tickIntervalMs`	Poll interval for releasing buffered frames.

enabled

bool enabled = false

False keeps the current zero-buffer receive path.

minDelayMs

std::int64_t minDelayMs = 20

Base playout delay in milliseconds.

maxDelayMs

std::int64_t maxDelayMs = 120

Upper bound for the adaptive playout delay.

adaptiveFactor

double adaptiveFactor = 1.5

Extra delay multiplier applied to observed inter-arrival variance.

tickIntervalMs

std::int64_t tickIntervalMs = 5

Poll interval for releasing buffered frames.