refactor(video): 删除废弃视频流水线并收敛 MJPEG/WebRTC 编排与死代码

src/webrtc/h265_payloader.rs

@@ -37,12 +37,6 @@ const H265_NAL_SPS: u8 = 33;
 const H265_NAL_PPS: u8 = 34;
 const H265_NAL_AUD: u8 = 35;
 const H265_NAL_FILLER: u8 = 38;
-#[allow(dead_code)]
-const H265_NAL_SEI_PREFIX: u8 = 39; // PREFIX_SEI_NUT
-#[allow(dead_code)]
-const H265_NAL_SEI_SUFFIX: u8 = 40; // SUFFIX_SEI_NUT
-#[allow(dead_code)]
-const H265_NAL_AP: u8 = 48; // Aggregation Packet
 const H265_NAL_FU: u8 = 49; // Fragmentation Unit
 
 /// H.265 NAL header size
@@ -51,11 +45,6 @@ const H265_NAL_HEADER_SIZE: usize = 2;
 /// FU header size (1 byte after NAL header)
 const H265_FU_HEADER_SIZE: usize = 1;
 
-/// Fixed PayloadHdr for FU packets: Type=49, LayerID=0, TID=1
-/// This matches the rtp crate's FRAG_PAYLOAD_HDR
-#[allow(dead_code)]
-const FU_PAYLOAD_HDR: [u8; 2] = [0x62, 0x01];
-
 /// Fixed PayloadHdr for AP packets: Type=48, LayerID=0, TID=1
 /// This matches the rtp crate's AGGR_PAYLOAD_HDR
 const AP_PAYLOAD_HDR: [u8; 2] = [0x60, 0x01];

src/webrtc/mod.rs

@@ -9,7 +9,7 @@
 //!
 //! Architecture:
 //! ```text
-//! VideoCapturer (MJPEG/YUYV)
+//! V4L2 capture
 //!        |
 //!        v
 //! SharedVideoPipeline (decode -> convert -> encode)

src/webrtc/track.rs

@@ -262,8 +262,6 @@ impl Default for AudioTrackConfig {
 
 /// Audio track for WebRTC streaming
 pub struct AudioTrack {
-    #[allow(dead_code)]
-    config: AudioTrackConfig,
     /// RTP track
     track: Arc<TrackLocalStaticRTP>,
     /// Running flag
@@ -284,7 +282,6 @@ impl AudioTrack {
         let (running_tx, _) = watch::channel(false);
 
         Self {
-            config,
             track,
             running: Arc::new(running_tx),
         }