fix: 修复开始播放音频时爆音的问题

apps/hello/src/audio/player.rs

@@ -1,7 +1,7 @@
 #![cfg(target_os = "linux")]
 use crate::config::AudioConfig;
-use alsa::pcm::{Access, Format, HwParams, PCM};
 use alsa::Direction;
+use alsa::pcm::{Access, Format, HwParams, PCM};
 use anyhow::{Context, Result};
 
 pub struct AudioPlayer {
@@ -38,6 +38,10 @@ impl AudioPlayer {
             }
         }
     }
+
+    pub fn prepare(&self) -> Result<()> {
+        self.pcm.prepare().context("Failed to prepare PCM")
+    }
 }
 
 fn setup_pcm(pcm: &PCM, sample_rate: u32, channels: u16) -> Result<()> {

apps/hello/src/stereo_core/master.rs

@@ -78,15 +78,6 @@ pub async fn run_master(master_role: ChannelRole) -> Result<()> {
         ..AudioConfig::default()
     };
 
-    let mut left_encoder = OpusCodec::new(&AudioConfig {
-        channels: 1,
-        ..config.clone()
-    })?;
-    let mut right_encoder = OpusCodec::new(&AudioConfig {
-        channels: 1,
-        ..config.clone()
-    })?;
-
     let mut current_player_channels = 0;
     let mut player: Option<AudioPlayer> = None;
 
@@ -113,6 +104,16 @@ pub async fn run_master(master_role: ChannelRole) -> Result<()> {
                 }
             };
 
+            // 每个新流开始时，重置编码器状态以避免残留音频导致爆音
+            let mut left_encoder = OpusCodec::new(&AudioConfig {
+                channels: 1,
+                ..config.clone()
+            })?;
+            let mut right_encoder = OpusCodec::new(&AudioConfig {
+                channels: 1,
+                ..config.clone()
+            })?;
+
             loop {
                 // 从 FIFO 读取
                 if let Err(_) = fifo.read_exact(&mut raw_buf).await {
@@ -234,7 +235,7 @@ pub async fn run_master(master_role: ChannelRole) -> Result<()> {
 
                 seq += 1;
             }
-            
+
             // 重置流计时
             stream_start_ts = 0;
         }

apps/hello/src/stereo_core/slave.rs

@@ -138,6 +138,7 @@ async fn handle_connection(role: ChannelRole) -> Result<()> {
     println!("✅ 主节点已连接，音频串流中...");
     let mut pcm_buf = vec![0i16; config.frame_size];
     let mut last_seq: Option<u32> = None;
+    let mut last_packet_time = now_us();
 
     loop {
         // 检查 TCP 是否已断开
@@ -147,6 +148,15 @@ async fn handle_connection(role: ChannelRole) -> Result<()> {
 
         // 填充 Jitter Buffer
         while let Ok(pkt) = audio_rx.try_recv() {
+            let now = now_us();
+            // 如果超过 500ms 没有收到包，认为是新流开始，重置状态
+            if now - last_packet_time > 500_000 {
+                jitter.clear();
+                last_seq = None;
+                codec = OpusCodec::new(&config)?;
+                let _ = player.prepare();
+            }
+            last_packet_time = now;
             jitter.push(pkt);
         }