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fix: 补齐 ATX 控制器缺失接口并完成全项目 clippy -D warnings 修复

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This commit is contained in:
mofeng-git
2026-02-10 21:37:33 +08:00
parent 72eb2c450d
commit 394baca938
64 changed files with 474 additions and 760 deletions
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7
src/webrtc/config.rs
View File

@@ -108,18 +108,15 @@ impl TurnServer {
/// Video codec preference
#[derive(Debug, Clone, Copy, PartialEq, Eq, Serialize, Deserialize)]
#[serde(rename_all = "lowercase")]
#[derive(Default)]
pub enum VideoCodec {
#[default]
H264,
VP8,
VP9,
AV1,
}
impl Default for VideoCodec {
fn default() -> Self {
Self::H264
}
}
impl std::fmt::Display for VideoCodec {
fn fmt(&self, f: &mut std::fmt::Formatter<'_>) -> std::fmt::Result {

1
src/webrtc/peer.rs
View File

@@ -93,7 +93,6 @@ impl PeerConnection {
urls: turn.urls.clone(),
username: turn.username.clone(),
credential: turn.credential.clone(),
..Default::default()
});
}

15
src/webrtc/rtp.rs
View File

@@ -330,9 +330,7 @@ impl OpusAudioTrack {
stream_id.to_string(),
));
Self {
track,
}
Self { track }
}
/// Get the underlying WebRTC track
@@ -365,13 +363,10 @@ impl OpusAudioTrack {
..Default::default()
};
self.track
.write_sample(&sample)
.await
.map_err(|e| {
error!("Failed to write Opus sample: {}", e);
AppError::WebRtcError(format!("Failed to write audio sample: {}", e))
})
self.track.write_sample(&sample).await.map_err(|e| {
error!("Failed to write Opus sample: {}", e);
AppError::WebRtcError(format!("Failed to write audio sample: {}", e))
})
}
}

2
src/webrtc/track.rs
View File

@@ -199,7 +199,7 @@ impl VideoTrack {
let data = frame.data();
let max_payload_size = 1200; // MTU - headers
let packet_count = (data.len() + max_payload_size - 1) / max_payload_size;
let packet_count = data.len().div_ceil(max_payload_size);
let mut bytes_sent = 0u64;
for i in 0..packet_count {

19
src/webrtc/universal_session.rs
View File

@@ -292,7 +292,6 @@ impl UniversalSession {
urls: turn.urls.clone(),
username: turn.username.clone(),
credential: turn.credential.clone(),
..Default::default()
});
}
@@ -430,7 +429,9 @@ impl UniversalSession {
let candidate = IceCandidate {
candidate: candidate_str,
sdp_mid: candidate_json.as_ref().and_then(|j| j.sdp_mid.clone()),
sdp_mline_index: candidate_json.as_ref().and_then(|j| j.sdp_mline_index),
sdp_mline_index: candidate_json
.as_ref()
.and_then(|j| j.sdp_mline_index),
username_fragment: candidate_json
.as_ref()
.and_then(|j| j.username_fragment.clone()),
@@ -615,20 +616,15 @@ impl UniversalSession {
};
// Verify codec matches
let frame_codec = match encoded_frame.codec {
VideoEncoderType::H264 => VideoEncoderType::H264,
VideoEncoderType::H265 => VideoEncoderType::H265,
VideoEncoderType::VP8 => VideoEncoderType::VP8,
VideoEncoderType::VP9 => VideoEncoderType::VP9,
};
let frame_codec = encoded_frame.codec;
if frame_codec != expected_codec {
continue;
}
// Debug log for H265 frames
if expected_codec == VideoEncoderType::H265 {
if encoded_frame.is_keyframe || frames_sent % 30 == 0 {
if expected_codec == VideoEncoderType::H265
&& (encoded_frame.is_keyframe || frames_sent.is_multiple_of(30)) {
debug!(
"[Session-H265] Received frame #{}: size={}, keyframe={}, seq={}",
frames_sent,
@@ -637,7 +633,6 @@ impl UniversalSession {
encoded_frame.sequence
);
}
}
// Ensure decoder starts from a keyframe and recover on gaps.
let mut gap_detected = false;
@@ -768,7 +763,7 @@ impl UniversalSession {
// 20ms at 48kHz = 960 samples
let samples = 960u32;
if let Err(e) = audio_track.write_packet(&opus_frame.data, samples).await {
if packets_sent % 100 == 0 {
if packets_sent.is_multiple_of(100) {
debug!("Failed to write audio packet: {}", e);
}
} else {

3
src/webrtc/video_track.rs
View File

@@ -285,7 +285,7 @@ impl UniversalVideoTrack {
}
/// Get current statistics
///
/// Write an encoded frame to the track
///
/// Handles codec-specific processing:
@@ -464,7 +464,6 @@ impl UniversalVideoTrack {
if let Err(e) = rtp_track.write_rtp(&packet).await {
trace!("H265 write_rtp failed: {}", e);
}
}
Ok(())

12
src/webrtc/webrtc_streamer.rs
View File

@@ -35,8 +35,8 @@ use tokio::sync::RwLock;
use tracing::{debug, info, trace, warn};
use crate::audio::{AudioController, OpusFrame};
use crate::events::EventBus;
use crate::error::{AppError, Result};
use crate::events::EventBus;
use crate::hid::HidController;
use crate::video::encoder::registry::EncoderBackend;
use crate::video::encoder::registry::VideoEncoderType;
@@ -270,7 +270,6 @@ impl WebRtcStreamer {
bitrate_preset: config.bitrate_preset,
fps: config.fps,
encoder_backend: config.encoder_backend,
..Default::default()
};
info!("Creating shared video pipeline for {:?}", codec);
@@ -311,7 +310,9 @@ impl WebRtcStreamer {
}
drop(pipeline_guard);
info!("Video pipeline stopped, but keeping capture config for new sessions");
info!(
"Video pipeline stopped, but keeping capture config for new sessions"
);
}
break;
}
@@ -926,10 +927,7 @@ impl WebRtcStreamer {
let pipeline = pipeline_for_callback.clone();
let sid = sid.clone();
tokio::spawn(async move {
info!(
"Requesting keyframe for session {} after reconnect",
sid
);
info!("Requesting keyframe for session {} after reconnect", sid);
pipeline.request_keyframe().await;
});
});