refactor: 优化推流播放与录音，支持立体声
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Del Wang
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//! # Jitter Buffer - 抖动缓冲区
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//!
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//! 用于音频流的抖动缓冲和包重排序。
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//!
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//! ## 功能
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//! - 自适应缓冲区大小
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//! - 乱序包重排
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//! - 丢包检测和统计
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//! - 延迟统计
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use crate::net::protocol::AudioPacket;
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use std::cmp::Ordering;
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use std::collections::{BTreeMap, VecDeque};
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impl PartialEq for AudioPacket {
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fn eq(&self, other: &Self) -> bool {
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self.timestamp == other.timestamp
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}
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}
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impl Eq for AudioPacket {}
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impl Ord for AudioPacket {
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fn cmp(&self, other: &Self) -> Ordering {
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other.timestamp.cmp(&self.timestamp)
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}
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}
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impl PartialOrd for AudioPacket {
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fn partial_cmp(&self, other: &Self) -> Option<Ordering> {
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Some(self.cmp(other))
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}
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}
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/// Jitter Buffer 统计信息
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#[derive(Debug, Clone, Default)]
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pub struct JitterStats {
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/// 总接收包数
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pub received: u64,
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/// 总丢包数
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pub lost: u64,
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/// 总播放包数
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pub played: u64,
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/// 迟到的包数（到达时已过播放时间）
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pub late: u64,
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/// 重复包数
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pub duplicate: u64,
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/// 当前缓冲区大小（包数）
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pub buffer_size: usize,
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/// 最小延迟（微秒）
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pub min_delay: u128,
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/// 最大延迟（微秒）
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pub max_delay: u128,
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/// 平均延迟（微秒）
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pub avg_delay: u128,
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}
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impl JitterStats {
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/// 计算丢包率（百分比）
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pub fn loss_rate(&self) -> f64 {
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if self.received + self.lost == 0 {
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return 0.0;
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}
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(self.lost as f64 / (self.received + self.lost) as f64) * 100.0
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}
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}
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/// Jitter Buffer 配置
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#[derive(Debug, Clone)]
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pub struct JitterConfig {
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/// 最小缓冲区大小（包数）
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pub min_buffer_size: usize,
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/// 最大缓冲区大小（包数）
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pub max_buffer_size: usize,
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/// 目标缓冲区大小（包数）
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pub target_buffer_size: usize,
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/// 自适应调整间隔（包数）
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pub adapt_interval: usize,
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/// 最大容忍延迟（微秒）
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pub max_tolerable_delay: u128,
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}
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impl Default for JitterConfig {
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fn default() -> Self {
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Self {
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min_buffer_size: 2,
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max_buffer_size: 20,
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target_buffer_size: 5,
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adapt_interval: 50,
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max_tolerable_delay: 100_000, // 100ms
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}
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}
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}
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/// Jitter Buffer - 用于音频流的抖动缓冲
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pub struct JitterBuffer {
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/// 配置
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config: JitterConfig,
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/// 缓冲区（按时间戳排序）
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buffer: BTreeMap<u128, AudioPacket>,
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/// 统计信息
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stats: JitterStats,
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/// 期望的下一个序列号
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expected_seq: u32,
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/// 是否已接收第一个包
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first_packet_received: bool,
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/// 延迟样本窗口（用于自适应）
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delay_samples: VecDeque<u128>,
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/// 自适应计数器
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adapt_counter: usize,
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/// 上次播放的时间戳
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last_played_timestamp: u128,
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}
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impl JitterBuffer {
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/// 创建新的 Jitter Buffer
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pub fn new(config: JitterConfig) -> Self {
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Self {
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config,
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buffer: BTreeMap::new(),
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stats: JitterStats::default(),
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expected_seq: 0,
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first_packet_received: false,
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delay_samples: VecDeque::with_capacity(100),
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adapt_counter: 0,
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last_played_timestamp: 0,
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}
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}
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/// 使用默认配置创建
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pub fn default() -> Self {
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Self::new(JitterConfig::default())
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}
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/// 插入音频包
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pub fn push(&mut self, packet: AudioPacket, arrival_time: u128) {
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// 检测重复包
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if self.buffer.contains_key(&packet.timestamp) {
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self.stats.duplicate += 1;
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return;
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}
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// 初始化序列号
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if !self.first_packet_received {
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self.expected_seq = packet.seq.wrapping_add(1);
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self.first_packet_received = true;
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} else {
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// 检测丢包
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let seq_diff = packet.seq.wrapping_sub(self.expected_seq);
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if seq_diff > 0 && seq_diff < 1000 {
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// 允许一定的序列号跳跃（处理回环）
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self.stats.lost += seq_diff as u64;
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}
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self.expected_seq = packet.seq.wrapping_add(1);
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}
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// 检查是否迟到
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if self.last_played_timestamp > 0 && packet.timestamp < self.last_played_timestamp {
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self.stats.late += 1;
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return;
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}
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// 超过最大容忍延迟，直接丢弃
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if packet.timestamp < arrival_time
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&& arrival_time - packet.timestamp > self.config.max_tolerable_delay
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{
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self.stats.late += 1;
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return;
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}
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let delay = if packet.timestamp >= arrival_time {
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packet.timestamp - arrival_time
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} else {
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0 // 包到达时已经过了播放时间
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};
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self.update_delay_stats(delay);
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// 插入缓冲区
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self.buffer.insert(packet.timestamp, packet);
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self.stats.received += 1;
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self.stats.buffer_size = self.buffer.len();
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// 自适应调整
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self.adapt_counter += 1;
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if self.adapt_counter >= self.config.adapt_interval {
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self.adapt_buffer_size();
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self.adapt_counter = 0;
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}
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}
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/// 获取下一个应该播放的包
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///
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/// # Arguments
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/// * `current_time` - 当前服务器时间
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///
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/// # Returns
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/// * `Some(packet)` - 如果有包应该播放
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/// * `None` - 如果缓冲区为空或还没到播放时间
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pub fn pop(&mut self, current_time: u128) -> Option<AudioPacket> {
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// 如果缓冲区为空，直接返回
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if self.buffer.is_empty() {
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return None;
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}
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// 检查缓冲区是否达到最小大小
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if self.buffer.len() < self.config.target_buffer_size && !self.should_drain() {
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return None;
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}
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// 获取最早的包
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if let Some((×tamp, _)) = self.buffer.iter().next() {
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// 检查是否到达播放时间
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if current_time >= timestamp {
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let packet = self.buffer.remove(×tamp).unwrap();
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self.last_played_timestamp = timestamp;
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self.stats.played += 1;
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self.stats.buffer_size = self.buffer.len();
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return Some(packet);
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}
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}
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None
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}
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/// 强制获取下一个包（无论时间）
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pub fn pop_next(&mut self) -> Option<AudioPacket> {
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if let Some((×tamp, _)) = self.buffer.iter().next() {
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let packet = self.buffer.remove(×tamp).unwrap();
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self.last_played_timestamp = timestamp;
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self.stats.played += 1;
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self.stats.buffer_size = self.buffer.len();
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return Some(packet);
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}
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None
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}
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/// 查看下一个包的播放时间（不移除）
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pub fn peek_next_timestamp(&self) -> Option<u128> {
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self.buffer.keys().next().copied()
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}
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/// 获取缓冲区大小
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pub fn len(&self) -> usize {
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self.buffer.len()
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}
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/// 检查缓冲区是否为空
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pub fn is_empty(&self) -> bool {
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self.buffer.is_empty()
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}
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/// 获取统计信息
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pub fn stats(&self) -> &JitterStats {
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&self.stats
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}
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/// 重置统计信息
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pub fn reset_stats(&mut self) {
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self.stats = JitterStats::default();
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self.stats.buffer_size = self.buffer.len();
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}
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/// 清空缓冲区
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pub fn clear(&mut self) {
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self.buffer.clear();
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self.stats.buffer_size = 0;
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self.first_packet_received = false;
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self.last_played_timestamp = 0;
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}
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/// 检查是否应该排空缓冲区（处理长时间没有新包的情况）
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fn should_drain(&self) -> bool {
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// 如果缓冲区有包且已经等了很久，就开始播放
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!self.buffer.is_empty() && self.buffer.len() >= self.config.min_buffer_size
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}
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/// 更新延迟统计
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fn update_delay_stats(&mut self, delay: u128) {
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// 更新最小/最大延迟
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if self.stats.received == 0 {
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self.stats.min_delay = delay;
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self.stats.max_delay = delay;
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self.stats.avg_delay = delay;
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} else {
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self.stats.min_delay = self.stats.min_delay.min(delay);
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self.stats.max_delay = self.stats.max_delay.max(delay);
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// 滑动平均
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self.stats.avg_delay = (self.stats.avg_delay * 9 + delay) / 10;
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}
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// 保存延迟样本用于自适应
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self.delay_samples.push_back(delay);
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if self.delay_samples.len() > 100 {
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self.delay_samples.pop_front();
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}
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}
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/// 自适应调整缓冲区大小
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fn adapt_buffer_size(&mut self) {
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if self.delay_samples.len() < 10 {
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return;
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}
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// 计算延迟方差（抖动）
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let avg = self.stats.avg_delay;
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let variance: f64 = self
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.delay_samples
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.iter()
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.map(|&d| {
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let diff = d as i128 - avg as i128;
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(diff * diff) as f64
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})
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.sum::<f64>()
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/ self.delay_samples.len() as f64;
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let jitter = variance.sqrt();
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// 根据抖动调整目标缓冲区大小
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// 抖动大 -> 增加缓冲区
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// 抖动小 -> 减少缓冲区
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let target = if jitter > 50_000.0 {
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// 高抖动（>50ms 标准差）
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self.config.target_buffer_size + 2
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} else if jitter < 10_000.0 {
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// 低抖动（<10ms 标准差）
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self.config.target_buffer_size.saturating_sub(1)
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} else {
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self.config.target_buffer_size
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};
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// 限制在最小/最大范围内
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self.config.target_buffer_size = target
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.max(self.config.min_buffer_size)
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.min(self.config.max_buffer_size);
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}
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}
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