feat: 完善架构优化性能

- 调整音视频架构,提升 RKMPP 编码 MJPEG-->H264 性能,同时解决丢帧马赛克问题;
- 删除多用户逻辑,只保留单用户,支持设置 web 单会话;
- 修复删除体验不好的的回退逻辑,前端页面菜单位置微调;
- 增加 OTG USB 设备动态调整功能;
- 修复 mdns 问题,webrtc 视频切换更顺畅。
This commit is contained in:
mofeng
2026-01-25 16:04:29 +08:00
parent 01e01430da
commit 1786b7689d
66 changed files with 4225 additions and 2936 deletions

View File

@@ -135,7 +135,8 @@ impl VideoStreamManager {
/// Set event bus for notifications
pub async fn set_event_bus(&self, events: Arc<EventBus>) {
*self.events.write().await = Some(events);
*self.events.write().await = Some(events.clone());
self.webrtc_streamer.set_event_bus(events).await;
}
/// Set configuration store
@@ -199,19 +200,20 @@ impl VideoStreamManager {
}
}
// Always reconnect frame source after initialization
// This ensures WebRTC has the correct frame_tx from the current capturer
if let Some(frame_tx) = self.streamer.frame_sender().await {
// Synchronize WebRTC config with actual capture format
let (format, resolution, fps) = self.streamer.current_video_config().await;
info!(
"Reconnecting frame source to WebRTC after init: {}x{} {:?} @ {}fps (receiver_count={})",
resolution.width, resolution.height, format, fps, frame_tx.receiver_count()
);
// Configure WebRTC capture source after initialization
let (device_path, resolution, format, fps, jpeg_quality) =
self.streamer.current_capture_config().await;
info!(
"WebRTC capture config after init: {}x{} {:?} @ {}fps",
resolution.width, resolution.height, format, fps
);
self.webrtc_streamer
.update_video_config(resolution, format, fps)
.await;
if let Some(device_path) = device_path {
self.webrtc_streamer
.update_video_config(resolution, format, fps)
.set_capture_device(device_path, jpeg_quality)
.await;
self.webrtc_streamer.set_video_source(frame_tx).await;
}
Ok(())
@@ -329,7 +331,7 @@ impl VideoStreamManager {
/// Ensure video capture is running (for WebRTC mode)
async fn ensure_video_capture_running(self: &Arc<Self>) -> Result<()> {
// Initialize streamer if not already initialized
// Initialize streamer if not already initialized (for config discovery)
if self.streamer.state().await == StreamerState::Uninitialized {
info!("Initializing video capture for WebRTC (ensure)");
if let Err(e) = self.streamer.init_auto().await {
@@ -338,29 +340,19 @@ impl VideoStreamManager {
}
}
// Start video capture if not streaming
if self.streamer.state().await != StreamerState::Streaming {
info!("Starting video capture for WebRTC (ensure)");
if let Err(e) = self.streamer.start().await {
error!("Failed to start video capture: {}", e);
return Err(e);
}
// Wait a bit for capture to stabilize
tokio::time::sleep(tokio::time::Duration::from_millis(100)).await;
}
// Reconnect frame source to WebRTC
if let Some(frame_tx) = self.streamer.frame_sender().await {
let (format, resolution, fps) = self.streamer.current_video_config().await;
info!(
"Reconnecting frame source to WebRTC: {}x{} {:?} @ {}fps",
resolution.width, resolution.height, format, fps
);
let (device_path, resolution, format, fps, jpeg_quality) =
self.streamer.current_capture_config().await;
info!(
"Configuring WebRTC capture: {}x{} {:?} @ {}fps",
resolution.width, resolution.height, format, fps
);
self.webrtc_streamer
.update_video_config(resolution, format, fps)
.await;
if let Some(device_path) = device_path {
self.webrtc_streamer
.update_video_config(resolution, format, fps)
.set_capture_device(device_path, jpeg_quality)
.await;
self.webrtc_streamer.set_video_source(frame_tx).await;
}
Ok(())
@@ -403,7 +395,6 @@ impl VideoStreamManager {
match current_mode {
StreamMode::Mjpeg => {
info!("Stopping MJPEG streaming");
// Only stop MJPEG distribution, keep video capture running for WebRTC
self.streamer.mjpeg_handler().set_offline();
if let Err(e) = self.streamer.stop().await {
warn!("Error stopping MJPEG streamer: {}", e);
@@ -458,10 +449,9 @@ impl VideoStreamManager {
}
}
StreamMode::WebRTC => {
// WebRTC mode: ensure video capture is running for H264 encoding
// WebRTC mode: configure direct capture for encoder pipeline
info!("Activating WebRTC mode");
// Initialize streamer if not already initialized
if self.streamer.state().await == StreamerState::Uninitialized {
info!("Initializing video capture for WebRTC");
if let Err(e) = self.streamer.init_auto().await {
@@ -470,77 +460,32 @@ impl VideoStreamManager {
}
}
// Auto-switch to non-compressed format if current format is MJPEG/JPEG
if let Some(device) = self.streamer.current_device().await {
let (current_format, resolution, fps) =
self.streamer.current_video_config().await;
if current_format.is_compressed() {
let available_formats: Vec<PixelFormat> =
device.formats.iter().map(|f| f.format).collect();
// Determine if using hardware encoding
let is_hardware = self.webrtc_streamer.is_hardware_encoding().await;
if let Some(recommended) =
PixelFormat::recommended_for_encoding(&available_formats, is_hardware)
{
info!(
"Auto-switching from {:?} to {:?} for WebRTC encoding (hardware={})",
current_format, recommended, is_hardware
);
let device_path = device.path.to_string_lossy().to_string();
if let Err(e) = self
.streamer
.apply_video_config(&device_path, recommended, resolution, fps)
.await
{
warn!("Failed to auto-switch format for WebRTC: {}, keeping current format", e);
}
}
}
}
// Start video capture if not streaming
if self.streamer.state().await != StreamerState::Streaming {
info!("Starting video capture for WebRTC");
if let Err(e) = self.streamer.start().await {
error!("Failed to start video capture for WebRTC: {}", e);
return Err(e);
}
}
// Wait a bit for capture to stabilize
tokio::time::sleep(tokio::time::Duration::from_millis(100)).await;
// Connect frame source to WebRTC with correct format
if let Some(frame_tx) = self.streamer.frame_sender().await {
// Synchronize WebRTC config with actual capture format
let (format, resolution, fps) = self.streamer.current_video_config().await;
info!(
"Connecting frame source to WebRTC pipeline: {}x{} {:?} @ {}fps",
resolution.width, resolution.height, format, fps
);
self.webrtc_streamer
.update_video_config(resolution, format, fps)
.await;
self.webrtc_streamer.set_video_source(frame_tx).await;
// Publish WebRTCReady event - frame source is now connected
let codec = self.webrtc_streamer.current_video_codec().await;
let is_hardware = self.webrtc_streamer.is_hardware_encoding().await;
self.publish_event(SystemEvent::WebRTCReady {
transition_id: Some(transition_id.clone()),
codec: codec_to_string(codec),
hardware: is_hardware,
})
let (device_path, resolution, format, fps, jpeg_quality) =
self.streamer.current_capture_config().await;
info!(
"Configuring WebRTC capture pipeline: {}x{} {:?} @ {}fps",
resolution.width, resolution.height, format, fps
);
self.webrtc_streamer
.update_video_config(resolution, format, fps)
.await;
if let Some(device_path) = device_path {
self.webrtc_streamer
.set_capture_device(device_path, jpeg_quality)
.await;
} else {
warn!(
"No frame source available for WebRTC - sessions may fail to receive video"
);
warn!("No capture device configured for WebRTC");
}
let codec = self.webrtc_streamer.current_video_codec().await;
let is_hardware = self.webrtc_streamer.is_hardware_encoding().await;
self.publish_event(SystemEvent::WebRTCReady {
transition_id: Some(transition_id.clone()),
codec: codec_to_string(codec),
hardware: is_hardware,
})
.await;
info!("WebRTC mode activated (sessions created on-demand)");
}
}
@@ -587,36 +532,34 @@ impl VideoStreamManager {
.update_video_config(resolution, format, fps)
.await;
// Restart video capture for WebRTC (it was stopped during config change)
info!("Restarting video capture for WebRTC after config change");
if let Err(e) = self.streamer.start().await {
error!("Failed to restart video capture for WebRTC: {}", e);
return Err(e);
let (device_path, actual_resolution, actual_format, actual_fps, jpeg_quality) =
self.streamer.current_capture_config().await;
if actual_format != format || actual_resolution != resolution || actual_fps != fps {
info!(
"Actual capture config differs from requested, updating WebRTC: {}x{} {:?} @ {}fps",
actual_resolution.width, actual_resolution.height, actual_format, actual_fps
);
self.webrtc_streamer
.update_video_config(actual_resolution, actual_format, actual_fps)
.await;
}
// Wait a bit for capture to stabilize
tokio::time::sleep(tokio::time::Duration::from_millis(100)).await;
// Reconnect frame source with the new capturer
if let Some(frame_tx) = self.streamer.frame_sender().await {
// Note: update_video_config was already called above with the requested config,
// but verify that actual capture matches
let (actual_format, actual_resolution, actual_fps) =
self.streamer.current_video_config().await;
if actual_format != format || actual_resolution != resolution || actual_fps != fps {
info!(
"Actual capture config differs from requested, updating WebRTC: {}x{} {:?} @ {}fps",
actual_resolution.width, actual_resolution.height, actual_format, actual_fps
);
self.webrtc_streamer
.update_video_config(actual_resolution, actual_format, actual_fps)
.await;
}
info!("Reconnecting frame source to WebRTC after config change");
self.webrtc_streamer.set_video_source(frame_tx).await;
if let Some(device_path) = device_path {
info!("Configuring direct capture for WebRTC after config change");
self.webrtc_streamer
.set_capture_device(device_path, jpeg_quality)
.await;
} else {
warn!("No frame source available after config change");
warn!("No capture device configured for WebRTC after config change");
}
let codec = self.webrtc_streamer.current_video_codec().await;
let is_hardware = self.webrtc_streamer.is_hardware_encoding().await;
self.publish_event(SystemEvent::WebRTCReady {
transition_id: None,
codec: codec_to_string(codec),
hardware: is_hardware,
})
.await;
}
Ok(())
@@ -631,22 +574,23 @@ impl VideoStreamManager {
self.streamer.start().await?;
}
StreamMode::WebRTC => {
// Ensure video capture is running
// Ensure device is initialized for config discovery
if self.streamer.state().await == StreamerState::Uninitialized {
self.streamer.init_auto().await?;
}
if self.streamer.state().await != StreamerState::Streaming {
self.streamer.start().await?;
}
// Connect frame source with correct format
if let Some(frame_tx) = self.streamer.frame_sender().await {
// Synchronize WebRTC config with actual capture format
let (format, resolution, fps) = self.streamer.current_video_config().await;
// Synchronize WebRTC config with current capture config
let (device_path, resolution, format, fps, jpeg_quality) =
self.streamer.current_capture_config().await;
self.webrtc_streamer
.update_video_config(resolution, format, fps)
.await;
if let Some(device_path) = device_path {
self.webrtc_streamer
.update_video_config(resolution, format, fps)
.set_capture_device(device_path, jpeg_quality)
.await;
self.webrtc_streamer.set_video_source(frame_tx).await;
} else {
warn!("No capture device configured for WebRTC");
}
}
}
@@ -764,13 +708,6 @@ impl VideoStreamManager {
self.streamer.is_streaming().await
}
/// Get frame sender for video frames
pub async fn frame_sender(
&self,
) -> Option<tokio::sync::broadcast::Sender<crate::video::frame::VideoFrame>> {
self.streamer.frame_sender().await
}
/// Subscribe to encoded video frames from the shared video pipeline
///
/// This allows RustDesk (and other consumers) to receive H264/H265/VP8/VP9
@@ -781,10 +718,10 @@ impl VideoStreamManager {
/// Returns None if video capture cannot be started or pipeline creation fails.
pub async fn subscribe_encoded_frames(
&self,
) -> Option<
tokio::sync::broadcast::Receiver<crate::video::shared_video_pipeline::EncodedVideoFrame>,
> {
// 1. Ensure video capture is initialized
) -> Option<tokio::sync::mpsc::Receiver<std::sync::Arc<
crate::video::shared_video_pipeline::EncodedVideoFrame,
>>> {
// 1. Ensure video capture is initialized (for config discovery)
if self.streamer.state().await == StreamerState::Uninitialized {
tracing::info!("Initializing video capture for encoded frame subscription");
if let Err(e) = self.streamer.init_auto().await {
@@ -796,28 +733,9 @@ impl VideoStreamManager {
}
}
// 2. Ensure video capture is running (streaming)
if self.streamer.state().await != StreamerState::Streaming {
tracing::info!("Starting video capture for encoded frame subscription");
if let Err(e) = self.streamer.start().await {
tracing::error!("Failed to start video capture for encoded frames: {}", e);
return None;
}
// Wait for capture to stabilize
tokio::time::sleep(tokio::time::Duration::from_millis(100)).await;
}
// 3. Get frame sender from running capture
let frame_tx = match self.streamer.frame_sender().await {
Some(tx) => tx,
None => {
tracing::warn!("Cannot subscribe to encoded frames: no frame sender available");
return None;
}
};
// 4. Synchronize WebRTC config with actual capture format
let (format, resolution, fps) = self.streamer.current_video_config().await;
// 2. Synchronize WebRTC config with capture config
let (device_path, resolution, format, fps, jpeg_quality) =
self.streamer.current_capture_config().await;
tracing::info!(
"Connecting encoded frame subscription: {}x{} {:?} @ {}fps",
resolution.width,
@@ -828,14 +746,17 @@ impl VideoStreamManager {
self.webrtc_streamer
.update_video_config(resolution, format, fps)
.await;
if let Some(device_path) = device_path {
self.webrtc_streamer
.set_capture_device(device_path, jpeg_quality)
.await;
} else {
tracing::warn!("No capture device configured for encoded frames");
return None;
}
// 5. Use WebRtcStreamer to ensure the shared video pipeline is running
// This will create the pipeline if needed
match self
.webrtc_streamer
.ensure_video_pipeline_for_external(frame_tx)
.await
{
// 3. Use WebRtcStreamer to ensure the shared video pipeline is running
match self.webrtc_streamer.ensure_video_pipeline_for_external().await {
Ok(pipeline) => Some(pipeline.subscribe()),
Err(e) => {
tracing::error!("Failed to start shared video pipeline: {}", e);