fix: eliminate playback crackle by pinning the PipeWire buffer to one quantum

The playback stream was negotiated with a ~256ms (12288-frame) maxsize
buffer. The sink drains the graph quantum (1024 frames) per cycle, so one
of our buffers lasted ~12 cycles and `process` was called only ~4x/sec,
each time asking us to fill all 12288 frames -- far more than the 200ms
(9600-sample) playout ring could ever hold. So ~half of every buffer was
silence-fill: a steady ~46% underrun, audible as constant crackle. This
is a consumer-side buffer-size bug, upstream of production pacing, which
is why earlier mixer-pacing attempts never moved the numbers.

Fix: pass an explicit SPA_TYPE_OBJECT_ParamBuffers param on connect,
pinning buffer size to one 1024-frame quantum (2048 bytes mono S16LE).
PipeWire now hands us a quantum-sized buffer ~47x/sec, the ring satisfies
every callback, and slice.len()/stride equals the quantum so we never
over-pull. A node.latency hint is added too (not load-bearing on its own
-- the hint alone changed nothing; the Buffers param is the fix). Note:
pipewire 0.9.2 only exposes feature v0_3_32, so Buffer::requested() is
unreachable -- pinning the buffer size is the available lever.

Verified with the probe (quantum=1024, 47 cb/s, underrun +0 steady) and
by ear: clean 440Hz tone, no clicks. Local playout path only -- not yet
verified on a live two-peer call.

Also in this commit (the investigation scaffolding that proved it out):
- Fill-paced mixer: production tracks the hardware clock via a shared
  exact ring-occupancy gauge (Arc<AtomicUsize>) kept near
  PLAYBACK_TARGET_SAMPLES, replacing the fixed 20ms timer that beat
  against the 1024 quantum.
- src/bin/audio_probe.rs: drives a sine through the real start_playback
  path with no network/mic, for isolating the local output stage.
- playout-health logging: quiet in normal use (logs only on underrun/
  dropped > 0); set PEERSPEAK_AUDIO_VERBOSE=1 for the per-second
  heartbeat (audio_probe sets it automatically).

Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>

src/audio/mod.rs

@@ -1,6 +1,18 @@
 use std::sync::mpsc::{Sender, Receiver};
+use std::sync::Arc;
+use std::sync::atomic::AtomicUsize;
 use thiserror::Error;
 
+/// Target depth of the playback ring buffer, in samples (48kHz mono).
+///
+/// The playout chain is paced to keep the ring near this level: production is
+/// driven by how fast PipeWire actually drains the ring (the hardware clock),
+/// not by a fixed software timer — which is what eliminates the producer/
+/// consumer beat that otherwise churns ~20% of audio into drops + silence.
+/// 2880 = 60ms = 3×20ms frames, comfortably above the 2048-sample max quantum
+/// so a single hardware pull can never empty the ring before the mixer refills.
+pub const PLAYBACK_TARGET_SAMPLES: usize = 2880;
+
 #[derive(Error, Debug)]
 pub enum AudioError {
     #[error("Failed to initialize audio backend: {0}")]
@@ -22,7 +34,18 @@ pub trait AudioBackend: Send + Sync {
 
     /// Starts playing back raw PCM audio to the output device (speaker),
     /// reading mixed/incoming chunks of samples from the provided Receiver.
-    fn start_playback(&self, rx: Receiver<Vec<i16>>, target_node: Option<String>) -> Result<(), AudioError>;
+    ///
+    /// `ring_fill` is updated with the playback ring's current occupancy (in
+    /// samples) as the device drains and the worker fills it. The caller (the
+    /// mixer) reads it to pace production to the hardware clock — produce only
+    /// while the ring is below [`PLAYBACK_TARGET_SAMPLES`] — instead of on a
+    /// fixed timer that beats against the device quantum.
+    fn start_playback(
+        &self,
+        rx: Receiver<Vec<i16>>,
+        target_node: Option<String>,
+        ring_fill: Arc<AtomicUsize>,
+    ) -> Result<(), AudioError>;
 
     /// Stops both capture and playback streams.
     fn stop(&self) -> Result<(), AudioError>;

src/audio/pipewire_impl.rs

@@ -1,7 +1,7 @@
 use crate::audio::{AudioBackend, AudioError};
 use std::sync::mpsc::{Sender, Receiver};
 use std::sync::{Arc, Mutex};
-use std::sync::atomic::{AtomicBool, Ordering};
+use std::sync::atomic::{AtomicBool, AtomicU64, AtomicUsize, Ordering};
 use std::thread::{self, JoinHandle};
 use std::time::Duration;
 use pipewire as pw;
@@ -63,7 +63,12 @@ impl AudioBackend for PipeWireBackend {
         Ok(())
     }
 
-    fn start_playback(&self, rx: Receiver<Vec<i16>>, target_node: Option<String>) -> Result<(), AudioError> {
+    fn start_playback(
+        &self,
+        rx: Receiver<Vec<i16>>,
+        target_node: Option<String>,
+        ring_fill: Arc<AtomicUsize>,
+    ) -> Result<(), AudioError> {
         let mut playback_guard = self.playback_state.lock().unwrap();
         if playback_guard.is_some() {
             return Err(AudioError::Stream("Playback already started".to_string()));
@@ -74,7 +79,7 @@ impl AudioBackend for PipeWireBackend {
         let thread = thread::Builder::new()
             .name("peerspeak-playback".to_string())
             .spawn(move || {
-                if let Err(e) = run_playback(cmd_rx, rx, target_node) {
+                if let Err(e) = run_playback(cmd_rx, rx, target_node, ring_fill) {
                     eprintln!("Playback thread error: {:?}", e);
                 }
             })
@@ -219,7 +224,12 @@ fn run_capture(cmd_rx: pw::channel::Receiver<()>, tx: Sender<Vec<i16>>, target_n
     Ok(())
 }
 
-fn run_playback(cmd_rx: pw::channel::Receiver<()>, rx: Receiver<Vec<i16>>, target_node: Option<String>) -> Result<(), AudioError> {
+fn run_playback(
+    cmd_rx: pw::channel::Receiver<()>,
+    rx: Receiver<Vec<i16>>,
+    target_node: Option<String>,
+    fill_gauge: Arc<AtomicUsize>,
+) -> Result<(), AudioError> {
     let mainloop = pw::main_loop::MainLoopRc::new(None)
         .map_err(|e| AudioError::Init(e.to_string()))?;
     let context = pw::context::ContextRc::new(&mainloop, None)
@@ -227,10 +237,46 @@ fn run_playback(cmd_rx: pw::channel::Receiver<()>, rx: Receiver<Vec<i16>>, targe
     let core = context.connect_rc(None)
         .map_err(|e| AudioError::Init(e.to_string()))?;
 
-    // Ring buffer setup: 9600 samples (200ms capacity for mono 48kHz)
+    // Ring buffer setup: 9600 samples (200ms capacity for mono 48kHz).
-    let rb = HeapRb::<i16>::new(9600);
+    const RING_CAPACITY: usize = 9600;
+    let rb = HeapRb::<i16>::new(RING_CAPACITY);
     let (mut producer, consumer) = rb.split();
 
+    // Prefill to the target depth so playout starts at its steady-state level:
+    // the mixer keeps the ring near `PLAYBACK_TARGET_SAMPLES` by reading
+    // `fill_gauge`, so production tracks the PipeWire hardware clock instead of
+    // a fixed timer.
+    //
+    // `fill_gauge` is an EXACT occupancy counter, maintained by deltas: the
+    // worker `fetch_add`s every pushed sample, the RT callback `fetch_sub`s
+    // every popped one. We can't use ringbuf's `occupied_len()` for this — the
+    // split producer/consumer keep *cached* head/tail indices, so their length
+    // observers are approximate and stale, which would feed the mixer a fill
+    // reading that lies high and starve the ring. (The try_push/try_pop data
+    // path itself is exact; only the length observers are cached.)
+    for _ in 0..crate::audio::PLAYBACK_TARGET_SAMPLES {
+        let _ = producer.try_push(0);
+    }
+    fill_gauge.store(crate::audio::PLAYBACK_TARGET_SAMPLES, Ordering::Relaxed);
+
+    // Playout-health instrumentation (diagnostic). `underrun_samples` counts
+    // samples the RT callback had to substitute with silence because the ring
+    // was empty (clicks); `dropped_frames` counts whole frames the worker
+    // discarded because the ring was full (overrun). With clock-paced
+    // production both should stay at zero; a steady non-zero trend means the
+    // pacing isn't keeping up, bursts mean scheduling jitter. All RT-safe: the
+    // callback does a single wait-free fetch_add/store per quantum.
+    let underrun_samples = Arc::new(AtomicU64::new(0));
+    let dropped_frames = Arc::new(AtomicU64::new(0));
+    // Diagnostic: the per-cycle frame count the RT callback is asked to fill
+    // (`slice.len()/stride`) and how many callbacks fire per second. If the
+    // count is ~1024 (the graph quantum) at ~47/s the consumer is normal; if
+    // it's large (the buffer maxsize) we're over-pulling the whole slice
+    // instead of the requested quantum — the leading suspect for the ~46%
+    // underrun. See handoff.md 2026-05-31 PM entry.
+    let last_quantum = Arc::new(AtomicUsize::new(0));
+    let callback_count = Arc::new(AtomicU64::new(0));
+
     // Command receiver to quit main loop
     let mainloop_clone = mainloop.clone();
     let _cmd_recv = cmd_rx.attach(mainloop.loop_(), move |_| {
@@ -241,6 +287,10 @@ fn run_playback(cmd_rx: pw::channel::Receiver<()>, rx: Receiver<Vec<i16>>, targe
         *pw::keys::MEDIA_TYPE => "Audio",
         *pw::keys::MEDIA_CATEGORY => "Playback",
         *pw::keys::MEDIA_ROLE => "Communication",
+        // Low-latency hint (~21ms @ 48kHz). On its own this does NOT shrink the
+        // buffer — the real fix is the explicit Buffers param below — but it
+        // expresses the intended quantum for any node that honours it.
+        *pw::keys::NODE_LATENCY => "1024/48000",
     };
     if let Some(target) = target_node {
         props.insert("node.target", target);
@@ -251,10 +301,20 @@ fn run_playback(cmd_rx: pw::channel::Receiver<()>, rx: Receiver<Vec<i16>>, targe
 
     struct PlaybackUserData<C: ringbuf::traits::Consumer<Item = i16>> {
         consumer: C,
+        underrun_samples: Arc<AtomicU64>,
+        fill_gauge: Arc<AtomicUsize>,
+        last_quantum: Arc<AtomicUsize>,
+        callback_count: Arc<AtomicU64>,
     }
 
     let _listener = stream
-        .add_local_listener_with_user_data(PlaybackUserData { consumer })
+        .add_local_listener_with_user_data(PlaybackUserData {
+            consumer,
+            underrun_samples: underrun_samples.clone(),
+            fill_gauge: fill_gauge.clone(),
+            last_quantum: last_quantum.clone(),
+            callback_count: callback_count.clone(),
+        })
         .process(|stream, user_data| {
             if let Some(mut buffer) = stream.dequeue_buffer() {
                 let datas = buffer.datas_mut();
@@ -264,13 +324,36 @@ fn run_playback(cmd_rx: pw::channel::Receiver<()>, rx: Receiver<Vec<i16>>, targe
                     if let Some(slice) = data.data() {
                         let stride = 2; // S16LE Mono = 2 bytes per frame
                         let n_frames = slice.len() / stride;
+                        // Diagnostic: record the quantum the device asked for and
+                        // count the callback. Wait-free, RT-safe.
+                        user_data.last_quantum.store(n_frames, Ordering::Relaxed);
+                        user_data.callback_count.fetch_add(1, Ordering::Relaxed);
+                        let mut starved = 0u64;
                         for i in 0..n_frames {
-                            let val = user_data.consumer.try_pop().unwrap_or(0);
+                            let val = match user_data.consumer.try_pop() {
+                                Some(v) => v,
+                                None => {
+                                    starved += 1;
+                                    0
+                                }
+                            };
                             let bytes = val.to_le_bytes();
                             let start = i * stride;
                             slice[start] = bytes[0];
                             slice[start + 1] = bytes[1];
                         }
+                        if starved > 0 {
+                            // One wait-free atomic add per quantum — RT-safe.
+                            user_data.underrun_samples.fetch_add(starved, Ordering::Relaxed);
+                        }
+                        // Decrement the exact occupancy counter by the samples we
+                        // actually pulled (excluding underruns, which removed
+                        // nothing) so the mixer paces against true ring depth.
+                        // Wait-free fetch_sub, RT-safe.
+                        let popped = n_frames - starved as usize;
+                        if popped > 0 {
+                            user_data.fill_gauge.fetch_sub(popped, Ordering::Relaxed);
+                        }
                         total_size = n_frames * stride;
                     }
                     let chunk = data.chunk_mut();
@@ -301,7 +384,56 @@ fn run_playback(cmd_rx: pw::channel::Receiver<()>, rx: Receiver<Vec<i16>>, targe
     .0
     .into_inner();
 
-    let mut params = [Pod::from_bytes(&values).unwrap()];
+    // Explicit Buffers param — THE fix for the playback crackle. Without it
+    // PipeWire handed this stream a ~256ms (12288-frame) maxsize buffer and
+    // only called `process` ~4x/sec; each callback then asked us to fill all
+    // 12288 frames, far more than the 200ms ring could ever hold, so ~half of
+    // every buffer was silence (the crackle). Pinning the buffer size to one
+    // graph quantum (1024 frames = 2048 bytes mono S16LE) makes the device hand
+    // us a ~1024-frame buffer ~47x/sec, which the ring satisfies comfortably and
+    // keeps `slice.len()/stride` equal to the quantum so we never over-pull.
+    const QUANTUM_FRAMES: i32 = 1024;
+    const STRIDE: i32 = 2; // S16LE mono = 2 bytes/frame
+    let buffers_obj = pw::spa::pod::Object {
+        type_: pw::spa::utils::SpaTypes::ObjectParamBuffers.as_raw(),
+        id: pw::spa::param::ParamType::Buffers.as_raw(),
+        properties: vec![
+            // Let PipeWire pick the buffer count (>=2 for double-buffering).
+            pw::spa::pod::Property::new(
+                pw::spa::sys::SPA_PARAM_BUFFERS_buffers,
+                pw::spa::pod::Value::Choice(pw::spa::pod::ChoiceValue::Int(
+                    pw::spa::utils::Choice(
+                        pw::spa::utils::ChoiceFlags::empty(),
+                        pw::spa::utils::ChoiceEnum::Range { default: 8, min: 2, max: 64 },
+                    ),
+                )),
+            ),
+            pw::spa::pod::Property::new(
+                pw::spa::sys::SPA_PARAM_BUFFERS_blocks,
+                pw::spa::pod::Value::Int(1),
+            ),
+            pw::spa::pod::Property::new(
+                pw::spa::sys::SPA_PARAM_BUFFERS_size,
+                pw::spa::pod::Value::Int(QUANTUM_FRAMES * STRIDE),
+            ),
+            pw::spa::pod::Property::new(
+                pw::spa::sys::SPA_PARAM_BUFFERS_stride,
+                pw::spa::pod::Value::Int(STRIDE),
+            ),
+        ],
+    };
+    let buffers_values: Vec<u8> = pw::spa::pod::serialize::PodSerializer::serialize(
+        std::io::Cursor::new(Vec::new()),
+        &pw::spa::pod::Value::Object(buffers_obj),
+    )
+    .unwrap()
+    .0
+    .into_inner();
+
+    let mut params = [
+        Pod::from_bytes(&values).unwrap(),
+        Pod::from_bytes(&buffers_values).unwrap(),
+    ];
 
     stream.connect(
         spa::utils::Direction::Output,
@@ -316,25 +448,71 @@ fn run_playback(cmd_rx: pw::channel::Receiver<()>, rx: Receiver<Vec<i16>>, targe
     // Spawn a worker thread to read from rx and push to producer
     let running = Arc::new(AtomicBool::new(true));
     let running_clone = running.clone();
+    let worker_dropped = dropped_frames.clone();
+    let worker_fill = fill_gauge.clone();
     let worker_handle = thread::spawn(move || {
         while running_clone.load(Ordering::Relaxed) {
             if let Ok(frame) = rx.recv() {
+                // Drop the whole frame (rather than tearing it) only if it truly
+                // won't fit — checked against the exact occupancy counter, not
+                // the ring's stale length observer. With clock-paced production
+                // this should never fire.
+                if worker_fill.load(Ordering::Relaxed) + frame.len() > RING_CAPACITY {
+                    worker_dropped.fetch_add(1, Ordering::Relaxed);
+                    continue;
+                }
                 for &sample in &frame {
-                    // Try to push. If buffer is full, drop to avoid growing latency.
+                    let _ = producer.try_push(sample);
-                    if producer.try_push(sample).is_err() {
-                        break;
-                    }
                 }
+                worker_fill.fetch_add(frame.len(), Ordering::Relaxed);
             } else {
                 return;
             }
         }
     });
 
+    // Diagnostic logger: once per second, report the ring fill and the delta in
+    // underrun samples / dropped frames since the last report. Quiet line (all
+    // zeros) means the local playout path is healthy; a steady non-zero trend is
+    // the clock-drift signature, bursts are scheduling jitter.
+    let log_running = running.clone();
+    let log_underrun = underrun_samples.clone();
+    let log_dropped = dropped_frames.clone();
+    let log_fill = fill_gauge.clone();
+    let log_quantum = last_quantum.clone();
+    let log_callbacks = callback_count.clone();
+    // In normal operation this stays quiet — it only logs a second where the
+    // playout actually glitched (underrun or dropped > 0). Set
+    // PEERSPEAK_AUDIO_VERBOSE=1 (the `audio_probe` tool does) to get the full
+    // once-per-second heartbeat for diagnostics.
+    let verbose = std::env::var_os("PEERSPEAK_AUDIO_VERBOSE").is_some();
+    let logger_handle = thread::spawn(move || {
+        let (mut last_u, mut last_d, mut last_c) = (0u64, 0u64, 0u64);
+        while log_running.load(Ordering::Relaxed) {
+            thread::sleep(Duration::from_secs(1));
+            let u = log_underrun.load(Ordering::Relaxed);
+            let d = log_dropped.load(Ordering::Relaxed);
+            let fill = log_fill.load(Ordering::Relaxed);
+            let q = log_quantum.load(Ordering::Relaxed);
+            let c = log_callbacks.load(Ordering::Relaxed);
+            let (du, dd, dc) = (u - last_u, d - last_d, c - last_c);
+            last_u = u;
+            last_d = d;
+            last_c = c;
+            if verbose || du > 0 || dd > 0 {
+                crate::log_msg(&format!(
+                    "playout-health: fill={fill} samples (~{}ms) | underrun +{du} samples/s (total {u}) | dropped +{dd} frames/s (total {d}) | quantum={q} frames, {dc} callbacks/s",
+                    fill / 48,
+                ));
+            }
+        }
+    });
+
     mainloop.run();
 
     running.store(false, Ordering::Relaxed);
     let _ = worker_handle.join();
+    let _ = logger_handle.join();
 
     Ok(())
 }

src/bin/audio_probe.rs

@@ -0,0 +1,119 @@
+//! Audio playout diagnostic probe.
+//!
+//! Drives a phase-continuous sine tone through the *real* PipeWire playback path
+//! (`PipeWireBackend::start_playback`), using the *same* fill-paced production
+//! the production mixer uses (`core/mod.rs`): generate a frame only while the
+//! playback ring is below `PLAYBACK_TARGET_SAMPLES`, so production tracks the
+//! PipeWire hardware clock. No network, no microphone — this isolates the local
+//! output path so we can confirm the clock-paced playout is glitch-free.
+//!
+//! Use your ears on the tone (any click/pop is a glitch) together with the
+//! `playout-health:` lines tailed to stdout:
+//!   - all-zero health lines + clean tone  → local playout is healthy; any
+//!     crackle on real calls is upstream (per-peer jitter buffer), not this ring.
+//!   - steady `underrun +N samples/s` (or steady `dropped +N frames/s`) with a
+//!     slowly drifting `fill`  → clock drift (producer vs hardware clock).
+//!   - random bursts correlated with system load  → scheduling jitter.
+//!
+//! Run:  cargo run --bin audio_probe -- [freq_hz] [seconds] [target_node]
+//!   e.g. cargo run --release --bin audio_probe -- 440 30
+
+use std::io::{BufRead, BufReader, Seek, SeekFrom};
+use std::sync::Arc;
+use std::sync::atomic::AtomicUsize;
+use std::sync::mpsc;
+use std::time::Duration;
+
+use peerspeak::audio::AudioBackend;
+use peerspeak::audio::pipewire_impl::PipeWireBackend;
+use peerspeak::core::jitter::FRAME_SAMPLES; // 960 samples = 20ms @ 48kHz mono
+
+const SAMPLE_RATE: f32 = 48_000.0;
+
+#[tokio::main]
+async fn main() {
+    let mut args = std::env::args().skip(1);
+    let freq: f32 = args.next().and_then(|s| s.parse().ok()).unwrap_or(440.0);
+    let secs: u64 = args.next().and_then(|s| s.parse().ok()).unwrap_or(30);
+    let target_node: Option<String> = args.next();
+
+    // The playout-health logger is quiet in normal operation (it only logs
+    // glitches); ask it for the full once-per-second heartbeat so the probe can
+    // show the steady-state numbers.
+    // SAFETY: set before any playback thread starts, so no concurrent env read.
+    unsafe { std::env::set_var("PEERSPEAK_AUDIO_VERBOSE", "1") };
+
+    println!("audio_probe: {freq} Hz tone for {secs}s through the real playback path.");
+    println!("Listen for clicks/pops; watch the playout-health lines below.\n");
+
+    // Tail the app log (where playout-health lines land) to stdout in the
+    // background so it's all in one terminal.
+    spawn_log_tailer();
+
+    let backend = PipeWireBackend::new();
+    let (tx, rx) = mpsc::channel::<Vec<i16>>();
+    let ring_fill = Arc::new(AtomicUsize::new(0));
+    if let Err(e) = backend.start_playback(rx, target_node, ring_fill.clone()) {
+        eprintln!("failed to start playback: {e}");
+        return;
+    }
+
+    // Phase-continuous sine, generated one 20ms frame at a time, fill-paced
+    // exactly like the production mixer: only produce while the ring is below
+    // target, so production tracks the PipeWire hardware clock.
+    use std::sync::atomic::Ordering;
+    let deadline = tokio::time::Instant::now() + Duration::from_secs(secs);
+    let mut n: u64 = 0; // running sample index keeps phase continuous across frames
+    while tokio::time::Instant::now() < deadline {
+        if ring_fill.load(Ordering::Relaxed) >= peerspeak::audio::PLAYBACK_TARGET_SAMPLES {
+            tokio::time::sleep(Duration::from_millis(2)).await;
+            continue;
+        }
+        let mut frame = Vec::with_capacity(FRAME_SAMPLES);
+        for _ in 0..FRAME_SAMPLES {
+            let t = n as f32 / SAMPLE_RATE;
+            // 0.25 amplitude: clearly audible but not harsh.
+            let sample = (0.25 * i16::MAX as f32 * (2.0 * std::f32::consts::PI * freq * t).sin()) as i16;
+            frame.push(sample);
+            n += 1;
+        }
+        if tx.send(frame).is_err() {
+            eprintln!("playback channel closed early");
+            break;
+        }
+    }
+
+    // Let the ring drain, then stop.
+    tokio::time::sleep(Duration::from_millis(300)).await;
+    let _ = backend.stop();
+    println!("\naudio_probe: done.");
+}
+
+/// Open the app log, seek to the end, and echo new lines (the `playout-health:`
+/// reports) to stdout once they appear.
+fn spawn_log_tailer() {
+    let path = peerspeak::log_file_path();
+    std::thread::spawn(move || {
+        // Wait for the file to exist (first log_msg creates it).
+        let file = loop {
+            if let Ok(f) = std::fs::File::open(&path) {
+                break f;
+            }
+            std::thread::sleep(Duration::from_millis(100));
+        };
+        let mut reader = BufReader::new(file);
+        let _ = reader.seek(SeekFrom::End(0));
+        loop {
+            let mut line = String::new();
+            match reader.read_line(&mut line) {
+                Ok(0) => std::thread::sleep(Duration::from_millis(150)),
+                Ok(_) => {
+                    if line.contains("playout-health:") {
+                        print!("{line}");
+                    }
+                }
+                Err(_) => std::thread::sleep(Duration::from_millis(150)),
+            }
+        }
+    });
+}

src/core/mod.rs

@@ -17,7 +17,7 @@ use iroh_gossip::net::Gossip;
 use tokio::sync::{mpsc, Mutex};
 use std::collections::HashMap;
 use std::sync::Arc;
-use std::sync::atomic::{AtomicBool, Ordering};
+use std::sync::atomic::{AtomicBool, AtomicUsize, Ordering};
 use std::time::Duration;
 
 pub struct CoreController {
@@ -213,7 +213,11 @@ async fn run_core_loop(
                     continue;
                 }
 
-                if let Err(e) = audio_backend.start_playback(playback_rx, output_device) {
+                // Shared gauge: PipeWire publishes the playback ring's live depth
+                // here (drain side + fill side); the mixer reads it to pace
+                // production to the hardware clock instead of a fixed timer.
+                let ring_fill = Arc::new(AtomicUsize::new(0));
+                if let Err(e) = audio_backend.start_playback(playback_rx, output_device, ring_fill.clone()) {
                     let _ = ui_tx.send(UiEvent::Error(format!("Failed to start playback: {}", e))).await;
                     let _ = audio_backend.stop();
                     let _ = room_state.leave().await;
@@ -314,28 +318,41 @@ async fn run_core_loop(
                     }
                 });
 
-                // 3. Mixing & level extraction loop task. Every 20ms, pull one
+                // 3. Mixing & level extraction loop task. Production is paced by
-                //    concealed frame per peer from its jitter buffer, apply
+                //    the playback ring's fill level (the PipeWire hardware clock),
-                //    per-peer volume, sum, and hand the mix to playback.
+                //    NOT a fixed software timer: we produce a 20ms frame only when
+                //    the ring is below its target depth, so the long-run mix rate
+                //    auto-matches the device drain rate and the producer/consumer
+                //    beat (which otherwise churns ~20% of audio) disappears. Each
+                //    produced frame pulls one concealed frame per peer from its
+                //    jitter buffer, applies per-peer volume, and sums.
                 let jitter_mixer = jitter.clone();
                 let is_deafened_clone = is_deafened.clone();
                 let peer_volumes_mixer = peer_volumes.clone();
                 let ui_tx_mixer = ui_tx.clone();
+                let ring_fill_mixer = ring_fill.clone();
                 let mixer_task = tokio::spawn(async move {
-                    let mut interval = tokio::time::interval(Duration::from_millis(20));
+                    // When the ring is at/above target we have nothing to do; nap
-                    interval.set_missed_tick_behavior(tokio::time::MissedTickBehavior::Skip);
+                    // briefly and re-check. Short enough (relative to the ~60ms
+                    // target and ~21ms device quantum) that we always refill well
+                    // before the ring can run dry.
+                    const IDLE_NAP: Duration = Duration::from_millis(2);
 
-                    // The 20ms mix cadence is fixed by playback, but pushing a
+                    // Pushing a level event per frame floods the UI runtime at
-                    // level event every tick floods the UI runtime at ~50/sec. We
+                    // ~50/sec. We peak-hold per-peer levels across this many
-                    // peak-hold per-peer levels across this many ticks and emit
+                    // produced frames and emit once per window (~10/sec) —
-                    // once per window (~10/sec) — peak-hold so a brief transient
+                    // peak-hold so a brief transient still lights the indicator.
-                    // inside the window still lights the speaking indicator.
+                    const LEVEL_EMIT_FRAMES: u32 = 5;
-                    const LEVEL_EMIT_TICKS: u32 = 5;
                     let mut level_peaks: HashMap<EndpointId, f32> = HashMap::new();
-                    let mut ticks_since_emit: u32 = 0;
+                    let mut frames_since_emit: u32 = 0;
 
                     loop {
-                        interval.tick().await;
+                        // Pace to the hardware clock: only produce while the ring
+                        // is draining below target. Otherwise yield and re-check.
+                        if ring_fill_mixer.load(Ordering::Relaxed) >= crate::audio::PLAYBACK_TARGET_SAMPLES {
+                            tokio::time::sleep(IDLE_NAP).await;
+                            continue;
+                        }
 
                         let current_volumes = peer_volumes_mixer.lock().await.clone();
                         let mut peer_frames = Vec::new();
@@ -389,11 +406,11 @@ async fn run_core_loop(
                         }
 
                         // Emit coalesced peaks once per window, then reset.
-                        ticks_since_emit += 1;
+                        frames_since_emit += 1;
-                        if ticks_since_emit >= LEVEL_EMIT_TICKS {
+                        if frames_since_emit >= LEVEL_EMIT_FRAMES {
                             let levels: Vec<(EndpointId, f32)> = level_peaks.drain().collect();
                             let _ = ui_tx_mixer.send(UiEvent::AudioLevels(levels)).await;
-                            ticks_since_emit = 0;
+                            frames_since_emit = 0;
                         }
                     }
                 });

src/lib.rs

@@ -24,6 +24,12 @@ fn log_path() -> &'static PathBuf {
     })
 }
 
+/// The resolved log file path. Exposed so diagnostic tools (e.g. the audio
+/// probe) can tail the same log the app writes to.
+pub fn log_file_path() -> PathBuf {
+    log_path().clone()
+}
+
 pub fn log_msg(msg: &str) {
     if let Ok(mut file) = std::fs::OpenOptions::new()
         .create(true)