jittr
1.0.0
此实现旨在对抗网络抖动并通过 udp 数据包创建可靠的媒体流。当尝试通过 udp(很可能是 rtp)持续传输音频时,无序数据包和变化的网络延迟是最大的问题之一。该数据结构缓冲数据包并对它们重新排序,同时引入尽可能小的延迟。
通过抖动缓冲区播放 udp/rtp 流中的 opus 数据包:
use async_std :: stream :: interval ;
use std :: time :: Duration ;
use jittr :: { JitterBuffer , Packet } ;
let mut rtp_stream = /* your rtp stream */ ;
/// Your Packet implementation
struct Opus { .. }
impl Packet for Opus { .. }
/// Create a jitter buffer for Opus packets
/// It can hold up to 10 packets before it starts to discard old packets
let mut jitter = JitterBuffer :: < Opus , 10 > :: new ( ) ;
/// Create an interval for packet playback
/// A typical length for audio packets is between 10 and 20ms
let mut playback = interval ( Duration :: from_millis ( 20 ) ) ;
loop {
futures :: select! {
_ = playback . next ( ) . fuse ( ) => {
let packet = jittr . pop ( ) ;
let pcm = /* Opus decodes audio if present or infers if none */
// Write pcm to speaker
} ,
rtp = rtp_stream . next ( ) . fuse ( ) => {
let opus : Opus = rtp . unwrap ( ) ;
jittr . push ( opus ) ;
} ,
}
}