refactor: 删除部分多余的代码和注释

src/audio/encoder.rs

@@ -1,26 +1,19 @@
-//! Opus audio encoder for WebRTC
+//! Opus encoder.
 
 use audiopus::coder::GenericCtl;
 use audiopus::{coder::Encoder, Application, Bitrate, Channels, SampleRate};
 use bytes::Bytes;
-use std::time::Instant;
 use tracing::info;
 
 use super::capture::AudioFrame;
 use crate::error::{AppError, Result};
 
-/// Opus encoder configuration
 #[derive(Debug, Clone)]
 pub struct OpusConfig {
-    /// Sample rate (must be 8000, 12000, 16000, 24000, or 48000)
     pub sample_rate: u32,
-    /// Channels (1 or 2)
     pub channels: u32,
-    /// Target bitrate in bps
     pub bitrate: u32,
-    /// Application mode
     pub application: OpusApplication,
-    /// Enable forward error correction
     pub fec: bool,
 }
 
@@ -29,7 +22,7 @@ impl Default for OpusConfig {
         Self {
             sample_rate: 48000,
             channels: 2,
-            bitrate: 64000, // 64 kbps
+            bitrate: 64000,
             application: OpusApplication::Audio,
             fec: true,
         }
@@ -37,7 +30,6 @@ impl Default for OpusConfig {
 }
 
 impl OpusConfig {
-    /// Create config for voice (lower latency)
     pub fn voice() -> Self {
         Self {
             application: OpusApplication::Voip,
@@ -46,7 +38,6 @@ impl OpusConfig {
         }
     }
 
-    /// Create config for music (higher quality)
     pub fn music() -> Self {
         Self {
             application: OpusApplication::Audio,
@@ -82,30 +73,18 @@ impl OpusConfig {
     }
 }
 
-/// Opus application mode
 #[derive(Debug, Clone, Copy, PartialEq, Eq)]
 pub enum OpusApplication {
-    /// Voice over IP
     Voip,
-    /// General audio
     Audio,
-    /// Low delay mode
     LowDelay,
 }
 
-/// Encoded Opus frame
 #[derive(Debug, Clone)]
 pub struct OpusFrame {
-    /// Encoded Opus data
     pub data: Bytes,
-    /// Duration in milliseconds
     pub duration_ms: u32,
-    /// Sequence number
     pub sequence: u64,
-    /// Timestamp
-    pub timestamp: Instant,
-    /// RTP timestamp (samples)
-    pub rtp_timestamp: u32,
 }
 
 impl OpusFrame {
@@ -118,20 +97,14 @@ impl OpusFrame {
     }
 }
 
-/// Opus encoder
 pub struct OpusEncoder {
     config: OpusConfig,
     encoder: Encoder,
-    /// Output buffer
     output_buffer: Vec<u8>,
-    /// Frame counter for RTP timestamp
     frame_count: u64,
-    /// Samples per frame
-    samples_per_frame: u32,
 }
 
 impl OpusEncoder {
-    /// Create a new Opus encoder
     pub fn new(config: OpusConfig) -> Result<Self> {
         let sample_rate = config.to_audiopus_sample_rate();
         let channels = config.to_audiopus_channels();
@@ -140,7 +113,6 @@ impl OpusEncoder {
         let mut encoder = Encoder::new(sample_rate, channels, application)
             .map_err(|e| AppError::AudioError(format!("Failed to create Opus encoder: {:?}", e)))?;
 
-        // Configure encoder
         encoder
             .set_bitrate(Bitrate::BitsPerSecond(config.bitrate as i32))
             .map_err(|e| AppError::AudioError(format!("Failed to set bitrate: {:?}", e)))?;
@@ -151,9 +123,6 @@ impl OpusEncoder {
                 .map_err(|e| AppError::AudioError(format!("Failed to enable FEC: {:?}", e)))?;
         }
 
-        // Calculate samples per frame (20ms at sample_rate)
-        let samples_per_frame = config.sample_rate / 50;
-
         info!(
             "Opus encoder created: {}Hz {}ch {}bps",
             config.sample_rate, config.channels, config.bitrate
@@ -162,18 +131,11 @@ impl OpusEncoder {
         Ok(Self {
             config,
             encoder,
-            output_buffer: vec![0u8; 4000], // Max Opus frame size
+            output_buffer: vec![0u8; 4000],
             frame_count: 0,
-            samples_per_frame,
         })
     }
 
-    /// Create with default configuration
-    pub fn default_config() -> Result<Self> {
-        Self::new(OpusConfig::default())
-    }
-
-    /// Encode PCM audio data (S16LE interleaved)
     pub fn encode(&mut self, pcm_data: &[i16]) -> Result<OpusFrame> {
         let encoded_len = self
             .encoder
@@ -182,7 +144,6 @@ impl OpusEncoder {
 
         let samples = pcm_data.len() as u32 / self.config.channels;
         let duration_ms = (samples * 1000) / self.config.sample_rate;
-        let rtp_timestamp = (self.frame_count * self.samples_per_frame as u64) as u32;
 
         self.frame_count += 1;
 
@@ -190,27 +151,18 @@ impl OpusEncoder {
             data: Bytes::copy_from_slice(&self.output_buffer[..encoded_len]),
             duration_ms,
             sequence: self.frame_count - 1,
-            timestamp: Instant::now(),
-            rtp_timestamp,
         })
     }
 
-    /// Encode from AudioFrame
-    ///
-    /// Uses zero-copy conversion from bytes to i16 samples via bytemuck.
     pub fn encode_frame(&mut self, frame: &AudioFrame) -> Result<OpusFrame> {
-        // Zero-copy: directly cast bytes to i16 slice
-        // AudioFrame.data is S16LE format, which matches native little-endian i16
         let samples: &[i16] = bytemuck::cast_slice(&frame.data);
         self.encode(samples)
     }
 
-    /// Get encoder configuration
     pub fn config(&self) -> &OpusConfig {
         &self.config
     }
 
-    /// Reset encoder state
     pub fn reset(&mut self) -> Result<()> {
         self.encoder
             .reset_state()
@@ -219,7 +171,6 @@ impl OpusEncoder {
         Ok(())
     }
 
-    /// Set bitrate dynamically
     pub fn set_bitrate(&mut self, bitrate: u32) -> Result<()> {
         self.encoder
             .set_bitrate(Bitrate::BitsPerSecond(bitrate as i32))
@@ -228,15 +179,6 @@ impl OpusEncoder {
     }
 }
 
-/// Audio encoder statistics
-#[derive(Debug, Clone, Default)]
-pub struct EncoderStats {
-    pub frames_encoded: u64,
-    pub bytes_output: u64,
-    pub avg_frame_size: usize,
-    pub current_bitrate: u32,
-}
-
 #[cfg(test)]
 mod tests {
     use super::*;
@@ -261,13 +203,12 @@ mod tests {
         let config = OpusConfig::default();
         let mut encoder = OpusEncoder::new(config).unwrap();
 
-        // 20ms of stereo silence at 48kHz
         let silence = vec![0i16; 960 * 2];
         let result = encoder.encode(&silence);
         assert!(result.is_ok());
 
         let frame = result.unwrap();
         assert!(!frame.is_empty());
-        assert!(frame.len() < silence.len() * 2); // Should be compressed
+        assert!(frame.len() < silence.len() * 2);
     }
 }