Audio Overhaul

scripts/daemon/lesavka-core.sh

@@ -93,11 +93,15 @@ printf '\x05\x01\x09\x02\xa1\x01\x09\x01\xa1\x00'\
 '\x05\x01\x09\x30\x09\x31\x09\x38\x15\x81\x25\x7f\x75\x08\x95\x03\x81\x06'\
 '\xc0\xc0' >"$G/functions/hid.usb1/report_desc"
 
-# # -- UAC2 Audio
+# ---------- UAC2 function – speaker + mic, 2×48 kHz stereo ---------
-# mkdir -p $G/functions/uac2.usb0
+mkdir -p "$G/functions/uac2.usb0"
-# echo 48000 > $G/functions/uac2.usb0/c_srate
+cfg="$G/functions/uac2.usb0"
-# echo 2     > $G/functions/uac2.usb0/c_ssize
+echo 48000  >"$cfg/s_spk/freq"
-# echo 2     > $G/functions/uac2.usb0/p_chmask
+echo 2      >"$cfg/s_spk/chs"   # L+R
+echo 16     >"$cfg/s_spk/strm_maxpacket"
+echo 48000  >"$cfg/c_mic/freq"
+echo 2      >"$cfg/c_mic/chs"
+echo 16     >"$cfg/c_mic/strm_maxpacket"
 
 # ----------------------- configuration -----------------------------
 mkdir -p "$G/configs/c.1/strings/0x409"
@@ -107,8 +111,12 @@ echo "Config 1" > "$G/configs/c.1/strings/0x409/configuration"
 # 6) Finally bind to first available UDC
 ln -s $G/functions/hid.usb0  $G/configs/c.1/
 ln -s $G/functions/hid.usb1  $G/configs/c.1/
-# ln -s $G/functions/uac2.usb0 $G/configs/c.1/
+ln -sf "$cfg" "$G/configs/c.1/"
 
 # 7) Finally bind to first available UDC
 echo "$UDC" > "$G/UDC"
-echo "[lesavka-core] 🎉 gadget ready on $UDC (keyboard: hidg0, mouse: hidg1)"
+echo "[lesavka-core] 🎉 gadget ready"
+echo "⌨️ hidg0 gadget ready on controller $UDC"
+echo "🖱️ hidg1 gadget ready on controller $UDC"
+echo "🎧 UAC2  gadget ready on controller $UDC"
+echo "🎤 UAC2  gadget ready on controller $UDC"

scripts/install/server.sh

@@ -4,7 +4,20 @@ set -euo pipefail
 ORIG_USER=${SUDO_USER:-$(id -un)}
 
 echo "==> 1a. Base packages"
-sudo pacman -Syq --needed --noconfirm git rustup protobuf gcc pipewire pipewire-pulse tailscale base-devel gstreamer gst-plugins-base gst-plugins-good gst-plugins-bad gst-plugins-ugly gst-libav
+sudo pacman -Syq --needed --noconfirm git \
+                                      rustup \
+                                      protobuf \
+                                      gcc \
+                                      pipewire \
+                                      pipewire-pulse \
+                                      tailscale \
+                                      base-devel \
+                                      gstreamer \
+                                      gst-plugins-base \
+                                      gst-plugins-good \
+                                      gst-plugins-bad \
+                                      gst-plugins-ugly \
+                                      gst-libav 
 if ! command -v yay >/dev/null 2>&1; then
   echo "==> 1b. installing yay from AUR ..."
   sudo -u "$ORIG_USER" bash -c '

server/src/audio.rs

@@ -1,19 +1,21 @@
+//  server/src/audio.rs
+#![forbid(unsafe_code)]
+
 use anyhow::Context;
 use futures_util::Stream;
 use gstreamer as gst;
 use gstreamer_app as gst_app;
+use gst::prelude::*;
 use lesavka_common::lesavka::AudioPacket;
 use tokio_stream::wrappers::ReceiverStream;
 use tonic::Status;
-use tracing::{debug, trace};
+use tracing::{debug, error, warn};
-use gst::prelude::*;
-
-const EAR_ID: [&str; 2] = ["l", "r"];
-const PIPE: &str = "appsrc name=audsrc is-live=true do-timestamp=true ! aacparse ! queue ! appsink name=asink emit-signals=true";
 
+/// “Speaker” stream coming **from** the remote host (UAC2‑gadget playback
+/// endpoint) **towards** the client.
 pub struct AudioStream {
-    _pipe: gst::Pipeline,
+    _pipeline: gst::Pipeline,
-    inner: ReceiverStream<Result<AudioPacket, Status>>,
+    inner:     ReceiverStream<Result<AudioPacket, Status>>,
 }
 
 impl Stream for AudioStream {
@@ -26,51 +28,90 @@ impl Stream for AudioStream {
     }
 }
 
-// pub async fn eye_ear(
+impl Drop for AudioStream {
-//     dev: &str,
+    fn drop(&mut self) {
-//     id:  u32,
+        let _ = self._pipeline.set_state(gst::State::Null);
-// ) -> anyhow::Result<AudioStream> {
+    }
-//     let ear = EAR_ID[id as usize];
+}
-//     gst::init().context("gst init")?;
-//     let desc = format!(
-//         "v4l2src device=\"{dev}\" io-mode=mmap do-timestamp=true ! \
-//          video/mpegts,systemstream=true,packetsize=188 ! \
-//          tsdemux name=d  d.audio_0 ! queue ! \
-//          aacparse ! queue ! \
-//          appsink name=asink emit-signals=true max-buffers=64 drop=true"
-//     );
-//     let pipe: gst::Pipeline = gst::parse::launch(&desc)?
-//         .downcast()
-//         .expect("pipeline");
 
-//     let sink = pipe.by_name("asink").expect("asink").downcast::<gst_app::AppSink>().unwrap();
+/*───────────────────────────────────────────────────────────────────────────*/
-//     let (tx, rx) = tokio::sync::mpsc::channel(8192);
+/*  eye_ear() – capture from ALSA (“speaker”) and push AAC AUs via gRPC     */
+/*───────────────────────────────────────────────────────────────────────────*/
 
-//     sink.set_callbacks(
+pub async fn eye_ear(alsa_dev: &str, id: u32) -> anyhow::Result<AudioStream> {
-//         gst_app::AppSinkCallbacks::builder()
+    //   NB: one *logical* speaker → id==0.  A 2nd logical stream could be
-//             .new_sample(move |s| {
+    //   added later (for multi‑channel) without changing the client.
-//                 let samp = s.pull_sample().map_err(|_| gst::FlowError::Eos)?;
+    gst::init().context("gst init")?;
-//                 let buf  = samp.buffer().ok_or(gst::FlowError::Error)?;
-//                 let map  = buf.map_readable().map_err(|_| gst::FlowError::Error)?;
-//                 let pts  = buf.pts().unwrap_or(gst::ClockTime::ZERO).nseconds()/1_000;
-//                 let pkt  = AudioPacket { id, pts, data: map.as_slice().to_vec() };
-//                 debug!(target:"lesavka_server::audio",
-//                        eye = id, bytes = pkt.data.len(), pts = pkt.pts,
-//                        "⬆️  pushed audio sample ear-{ear}");
-//                 let _ = tx.try_send(Ok(pkt));
-//                 Ok(gst::FlowSuccess::Ok)
-//             }).build()
-//     );
 
-//     pipe.set_state(gst::State::Playing)?;
+    /*──────────── pipeline description ────────────
-//     Ok(AudioStream{ _pipe: pipe, inner: ReceiverStream::new(rx) })
+     *
-// }
+     *    ALSA (UAC2 gadget)       AAC+ADTS            AppSink
+     *   ┌───────────┐ raw 48 kHz ┌─────────┐ AU/ADTS ┌──────────┐
+     *   │  alsasrc  │────────────► voaacenc │────────► appsink   │
+     *   └───────────┘            └─────────┘         └──────────┘
+     */
+    let desc = format!(
+        "alsasrc name=audsrc device=\"{alsa_dev}\" do-timestamp=true ! \
+         audio/x-raw,channels=2,rate=48000 ! \
+         voaacenc bitrate=192000 ! aacparse ! queue ! \
+         appsink name=asink emit-signals=true max-buffers=64 drop=true"
+    );
+
+    let pipeline: gst::Pipeline = gst::parse::launch(&desc)?
+        .downcast()
+        .expect("pipeline");
+
+    let sink: gst_app::AppSink = pipeline
+        .by_name("asink")
+        .expect("asink")
+        .downcast()
+        .expect("appsink");
+
+    let (tx, rx) = tokio::sync::mpsc::channel(8192);
+
+    /*──────────── callbacks ────────────*/
+    sink.set_callbacks(
+        gst_app::AppSinkCallbacks::builder()
+            .new_sample(move |s| {
+                let sample = s.pull_sample().map_err(|_| gst::FlowError::Eos)?;
+                let buffer = sample.buffer().ok_or(gst::FlowError::Error)?;
+                let map    = buffer.map_readable().map_err(|_| gst::FlowError::Error)?;
+
+                static CNT: std::sync::atomic::AtomicU64 =
+                    std::sync::atomic::AtomicU64::new(0);
+                let n = CNT.fetch_add(1, std::sync::atomic::Ordering::Relaxed);
+                if n < 10 || n % 300 == 0 {
+                    debug!("🔊 eye‑ear #{n}: {} bytes", map.len());
+                }
+
+                let pts_us = buffer
+                    .pts()
+                    .unwrap_or(gst::ClockTime::ZERO)
+                    .nseconds() / 1_000;
+
+                // push non‑blocking; drop oldest on overflow
+                if tx.try_send(Ok(AudioPacket {
+                    id,
+                    pts: pts_us,
+                    data: map.as_slice().to_vec(),
+                })).is_err() {
+                    static DROPS: std::sync::atomic::AtomicU64 =
+                        std::sync::atomic::AtomicU64::new(0);
+                    let d = DROPS.fetch_add(1, std::sync::atomic::Ordering::Relaxed);
+                    if d % 300 == 0 {
+                        warn!("🔊💔 dropped {d} audio AUs (client too slow)");
+                    }
+                }
+                Ok(gst::FlowSuccess::Ok)
+            })
+            .build(),
+    );
+
+    pipeline.set_state(gst::State::Playing)
+        .context("starting audio pipeline")?;
 
-pub async fn eye_ear(_dev: &str, _id: u32) -> anyhow::Result<AudioStream> {
-    use tokio_stream::StreamExt;
-    let (_tx, rx) = tokio::sync::mpsc::channel(1);
     Ok(AudioStream {
-        _pipe: gst::Pipeline::new(),
+        _pipeline: pipeline,
-        inner: ReceiverStream::new(rx), // always empty
+        inner:     ReceiverStream::new(rx),
     })
 }

server/src/main.rs

@@ -186,15 +186,17 @@ impl Relay for Handler {
         &self,
         req: Request<MonitorRequest>,
     ) -> Result<Response<Self::CaptureAudioStream>, Status> {
-        // let id  = req.into_inner().id;
+        // Only one speaker stream for now; both 0/1 → same ALSA dev.
-        // let dev = match id { 0 => "/dev/lesavka_l_eye",
+        let _id  = req.into_inner().id;
-        //                      1 => "/dev/lesavka_r_eye",
+        // Allow override (`LESAVKA_ALSA_DEV=hw:2,0` for debugging).
-        //                      _ => return Err(Status::invalid_argument("monitor id must be 0 or 1")) };
+        let dev = std::env::var("LESAVKA_ALSA_DEV")
-        // let s = audio::eye_ear(dev, id)
+            .unwrap_or_else(|_| "hw:UAC2Gadget,0".into());
-        //     .await
+
-        //     .map_err(|e| Status::internal(format!("{e:#}")))?;
+        let s = audio::eye_ear(&dev, 0)
-        // Ok(Response::new(Box::pin(s)))
+            .await
-        Err(Status::unimplemented("GC311 has no embedded audio"))
+            .map_err(|e| Status::internal(format!("{e:#}")))?;
+
+        Ok(Response::new(Box::pin(s)))
     }
 
     /*────────────── USB-reset RPC ───────────*/