feat: redesign audio networking for real-network resilience

Replaces the fire-and-forget datagram path with sequenced packets, a
per-peer jitter buffer, and persistent per-peer send tasks. Together these
fix three intertwined weaknesses that only showed up off localhost.

Packet format: every audio frame now carries a 4-byte little-endian
sequence number header ([seq][opus payload]), the basis for reordering and
loss detection.

Jitter buffer (core/jitter.rs): incoming packets are reordered by sequence
behind a fixed ~60ms playout delay. Missing sequences with later packets
already buffered are concealed via Opus PLC (decode(None)) -- a path the
decoder supported but nothing ever invoked. Underruns go idle and re-buffer
rather than concealing indefinitely. Covered by unit tests using real
encoded frames (reorder, gap-conceal, prime, late-drop).

Transport (network/iroh_impl.rs): each peer gets one long-lived send task
fed by a shallow bounded channel (drop-oldest on backpressure), instead of
spawning a throwaway task per peer per 20ms frame. Connections are now
established reactively on peer-join and torn down on peer-leave; the
lexicographically-lower EndpointId dials so a full-mesh pair forms exactly
one shared bidirectional connection instead of two racing ones. This also
removes the previous lock-held-across-connect().await serialization.

Opus decoder: PLC output is now sized to one 20ms frame, so concealment
synthesizes 20ms instead of a 120ms burst from the oversized max buffer.

Known follow-up (Tier 2): no reconnect on transient connection loss; a
send error currently retires the peer until they rejoin.

Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
This commit is contained in:
2026-05-29 15:48:43 -04:00
co-authored by Claude Opus 4.8
parent 7af0235736
commit 875e6e124c
5 changed files with 437 additions and 176 deletions
+28 -20
View File
@@ -31,44 +31,52 @@ impl AudioEncoder for OpusEncoder {
pub struct OpusDecoder {
decoder: Decoder,
channels: Channels,
/// Samples-per-channel of the frames we transmit (20ms @ 48kHz = 960).
/// Used to size the Packet Loss Concealment output, since libopus conceals
/// `frame_size` samples when given no input — passing the full max buffer
/// would synthesize a 120ms burst instead of a single 20ms frame.
frame_samples: usize,
}
impl OpusDecoder {
/// Creates a new Opus decoder.
/// Standard voice parameters: sample_rate = 48000, channels = Channels::Mono
pub fn new(sample_rate: u32, channels: Channels) -> Result<Self, CodecError> {
/// Standard voice parameters: sample_rate = 48000, channels = Channels::Mono.
/// `frame_samples` is the per-channel length of one transmitted frame (e.g. 960).
pub fn new(sample_rate: u32, channels: Channels, frame_samples: usize) -> Result<Self, CodecError> {
let decoder = Decoder::new(sample_rate, channels)
.map_err(|e| CodecError::Init(format!("Failed to create Opus decoder: {}", e)))?;
Ok(Self { decoder, channels })
Ok(Self { decoder, channels, frame_samples })
}
fn channels_count(&self) -> usize {
match self.channels {
Channels::Mono => 1,
Channels::Stereo => 2,
}
}
}
impl AudioDecoder for OpusDecoder {
fn decode(&mut self, compressed: Option<&[u8]>) -> Result<Vec<i16>, CodecError> {
// Maximum Opus frame size is 120ms. At 48kHz, this is 5760 samples per channel.
let channels_count = match self.channels {
Channels::Mono => 1,
Channels::Stereo => 2,
};
let max_samples = 5760 * channels_count;
let mut pcm = vec![0i16; max_samples];
let channels_count = self.channels_count();
let decoded_samples_per_channel = match compressed {
let (mut pcm, input): (Vec<i16>, &[u8]) = match compressed {
Some(data) if !data.is_empty() => {
// Normal decode
self.decoder.decode(data, &mut pcm, false)
.map_err(|e| CodecError::Decode(format!("Opus decoding failed: {}", e)))?
// Normal decode. Size the buffer to the maximum Opus frame (120ms =
// 5760 samples/channel); libopus decodes the packet's true duration.
(vec![0i16; 5760 * channels_count], data)
}
_ => {
// Packet Loss Concealment (PLC)
// In opus-rs, passing an empty slice triggers PLC.
self.decoder.decode(&[], &mut pcm, false)
.map_err(|e| CodecError::Decode(format!("Opus PLC decoding failed: {}", e)))?
// Packet Loss Concealment: an empty input makes libopus synthesize
// exactly `frame_samples` of concealment, so size the buffer to match.
(vec![0i16; self.frame_samples * channels_count], &[])
}
};
let total_samples = decoded_samples_per_channel * channels_count;
pcm.truncate(total_samples);
let decoded_per_channel = self.decoder.decode(input, &mut pcm, false)
.map_err(|e| CodecError::Decode(format!("Opus decoding failed: {}", e)))?;
pcm.truncate(decoded_per_channel * channels_count);
Ok(pcm)
}
}
+192
View File
@@ -0,0 +1,192 @@
//! Per-peer jitter buffer with Opus packet-loss concealment.
//!
//! Incoming audio arrives as unreliable QUIC datagrams that can be reordered,
//! duplicated, or dropped on real networks. Each packet carries a monotonic
//! sequence number (assigned by the sender). This buffer reorders packets by
//! sequence, holds a small fixed playout delay to absorb jitter, and — when a
//! sequence is missing but later packets have already arrived — synthesizes a
//! concealment frame via Opus PLC instead of emitting a click of silence.
use crate::codec::{AudioDecoder, CodecError, opus_impl::OpusDecoder};
use opus::Channels;
use std::collections::BTreeMap;
/// Samples per channel in one transmitted frame (20ms @ 48kHz mono).
pub const FRAME_SAMPLES: usize = 960;
/// How many frames to buffer before playout begins (~60ms). This is the
/// tolerance window for reordering and jitter; larger = more resilient but
/// more latency.
const TARGET_DELAY_FRAMES: usize = 3;
/// Hard cap on buffered frames (~640ms). If we ever exceed this we've fallen
/// badly behind, so we drop the oldest and resync rather than grow unbounded.
const MAX_BUFFERED_FRAMES: usize = 32;
pub struct JitterBuffer {
decoder: OpusDecoder,
/// Reorder window: sequence number -> encoded Opus payload.
packets: BTreeMap<u32, Vec<u8>>,
/// Next sequence we expect to play. `None` means idle/buffering: we are
/// waiting to accumulate `TARGET_DELAY_FRAMES` before (re)starting playout.
next_seq: Option<u32>,
}
/// Wrapping-aware "is `a` strictly before `b`" for sequence numbers.
fn seq_before(a: u32, b: u32) -> bool {
a != b && b.wrapping_sub(a) < (1 << 31)
}
impl JitterBuffer {
pub fn new() -> Result<Self, CodecError> {
Ok(Self {
decoder: OpusDecoder::new(48000, Channels::Mono, FRAME_SAMPLES)?,
packets: BTreeMap::new(),
next_seq: None,
})
}
/// Store a received packet, dropping ones we've already played past and
/// bounding total depth.
pub fn insert(&mut self, seq: u32, payload: Vec<u8>) {
// Too late: this sequence has already been played (or concealed).
if let Some(next) = self.next_seq
&& seq_before(seq, next)
{
return;
}
self.packets.insert(seq, payload);
while self.packets.len() > MAX_BUFFERED_FRAMES {
let oldest = *self.packets.keys().next().expect("non-empty");
self.packets.remove(&oldest);
// We've discarded backlog; resync the playout head to the new front.
self.next_seq = self.packets.keys().next().copied();
}
}
/// Produce the next 20ms PCM frame for playout, or `None` when idle or
/// still buffering (the caller should treat `None` as silence).
pub fn pop_frame(&mut self) -> Option<Vec<i16>> {
match self.next_seq {
None => {
// Buffering: start playout once we have enough to absorb jitter.
if self.packets.len() >= TARGET_DELAY_FRAMES {
self.next_seq = self.packets.keys().next().copied();
self.pop_frame()
} else {
None
}
}
Some(next) => {
if let Some(payload) = self.packets.remove(&next) {
self.next_seq = Some(next.wrapping_add(1));
self.decoder.decode(Some(&payload)).ok()
} else if self.packets.is_empty() {
// Underrun: the talker has gone quiet (or stopped). Go idle
// and re-buffer before resuming, rather than concealing forever.
self.next_seq = None;
None
} else {
// Gap with later packets already buffered: a packet was lost
// or reordered out of window. Conceal this frame via Opus PLC.
self.next_seq = Some(next.wrapping_add(1));
self.decoder.decode(None).ok()
}
}
}
}
/// True when nothing is buffered and playout is idle (talker silent).
pub fn is_idle(&self) -> bool {
self.next_seq.is_none() && self.packets.is_empty()
}
}
#[cfg(test)]
mod tests {
use super::*;
use crate::codec::AudioEncoder;
use crate::codec::opus_impl::OpusEncoder;
use opus::{Application, Channels};
/// A real, decodable Opus packet for one 20ms mono frame at amplitude `amp`.
fn frame(enc: &mut OpusEncoder, amp: i16) -> Vec<u8> {
let pcm: Vec<i16> = (0..FRAME_SAMPLES)
.map(|i| if i % 2 == 0 { amp } else { -amp })
.collect();
enc.encode(&pcm).unwrap()
}
#[test]
fn buffers_then_plays_in_order() {
let mut enc = OpusEncoder::new(48000, Channels::Mono, Application::Voip).unwrap();
let mut jb = JitterBuffer::new().unwrap();
// Below the target delay, playout hasn't primed yet.
jb.insert(0, frame(&mut enc, 1000));
assert!(jb.pop_frame().is_none());
// Reaching the target delay primes playout and yields the first frame.
jb.insert(1, frame(&mut enc, 1000));
jb.insert(2, frame(&mut enc, 1000));
assert_eq!(jb.pop_frame().map(|f| f.len()), Some(FRAME_SAMPLES));
assert_eq!(jb.pop_frame().map(|f| f.len()), Some(FRAME_SAMPLES));
assert_eq!(jb.pop_frame().map(|f| f.len()), Some(FRAME_SAMPLES));
// Drained: idle again.
assert!(jb.pop_frame().is_none());
assert!(jb.is_idle());
}
#[test]
fn reorders_out_of_order_arrivals() {
let mut enc = OpusEncoder::new(48000, Channels::Mono, Application::Voip).unwrap();
let mut jb = JitterBuffer::new().unwrap();
// Arrive scrambled but within the buffering window.
jb.insert(2, frame(&mut enc, 800));
jb.insert(0, frame(&mut enc, 800));
jb.insert(1, frame(&mut enc, 800));
// Three real frames come out (in sequence order), then idle.
assert!(jb.pop_frame().is_some());
assert!(jb.pop_frame().is_some());
assert!(jb.pop_frame().is_some());
assert!(jb.pop_frame().is_none());
}
#[test]
fn conceals_gap_when_later_packets_present() {
let mut enc = OpusEncoder::new(48000, Channels::Mono, Application::Voip).unwrap();
let mut jb = JitterBuffer::new().unwrap();
// Seq 2 is missing, but 0,1,3 arrive — enough to prime.
jb.insert(0, frame(&mut enc, 1200));
jb.insert(1, frame(&mut enc, 1200));
jb.insert(3, frame(&mut enc, 1200));
assert!(jb.pop_frame().is_some()); // seq 0
assert!(jb.pop_frame().is_some()); // seq 1
// seq 2 missing but seq 3 buffered -> Opus PLC produces a concealment frame.
let concealed = jb.pop_frame();
assert_eq!(concealed.map(|f| f.len()), Some(FRAME_SAMPLES));
assert!(jb.pop_frame().is_some()); // seq 3
assert!(jb.pop_frame().is_none());
}
#[test]
fn drops_packets_already_played() {
let mut enc = OpusEncoder::new(48000, Channels::Mono, Application::Voip).unwrap();
let mut jb = JitterBuffer::new().unwrap();
jb.insert(5, frame(&mut enc, 600));
jb.insert(6, frame(&mut enc, 600));
jb.insert(7, frame(&mut enc, 600));
assert!(jb.pop_frame().is_some()); // primes at seq 5, plays 5
assert!(jb.pop_frame().is_some()); // 6
// A straggler for an already-played sequence must be discarded.
jb.insert(5, frame(&mut enc, 600));
assert_eq!(jb.packets.len(), 1); // only seq 7 remains buffered
}
}
+73 -79
View File
@@ -1,7 +1,9 @@
pub mod messages;
pub mod jitter;
use crate::audio::{AudioBackend, pipewire_impl::PipeWireBackend};
use crate::codec::{AudioEncoder, AudioDecoder, opus_impl::{OpusEncoder, OpusDecoder}};
use crate::codec::{AudioEncoder, opus_impl::OpusEncoder};
use crate::core::jitter::{JitterBuffer, FRAME_SAMPLES};
use crate::network::{
NetworkTransport, RoomState, PeerState, RoomEvent, PeerSpeakTicket,
iroh_impl::IrohTransport,
@@ -12,7 +14,7 @@ use crate::core::messages::{CoreCommand, UiEvent};
use iroh::{Endpoint, EndpointId, endpoint::presets, protocol::Router};
use iroh_gossip::net::Gossip;
use tokio::sync::{mpsc, Mutex};
use std::collections::{HashMap, VecDeque};
use std::collections::HashMap;
use std::sync::Arc;
use std::sync::atomic::{AtomicBool, Ordering};
use std::time::Duration;
@@ -189,7 +191,7 @@ async fn run_core_loop(
continue;
}
let queues: Arc<Mutex<HashMap<EndpointId, VecDeque<i16>>>> = Arc::new(Mutex::new(HashMap::new()));
let jitter: Arc<Mutex<HashMap<EndpointId, JitterBuffer>>> = Arc::new(Mutex::new(HashMap::new()));
// 1. Capture & encoding thread
let is_muted_clone = is_muted.clone();
@@ -197,8 +199,6 @@ async fn run_core_loop(
let ptt_active_clone = ptt_active.clone();
let noise_gate_threshold_clone = noise_gate_threshold.clone();
let transport_clone = transport.clone();
let room_state_clone = room_state.clone();
let tokio_handle = tokio::runtime::Handle::current();
let capture_thread = std::thread::spawn(move || {
use opus::{Channels, Application};
@@ -209,6 +209,9 @@ async fn run_core_loop(
return;
}
};
// Per-sender packet sequence number, prepended to every frame so
// receivers can reorder and conceal loss. Wraps after ~years.
let mut seq: u32 = 0;
while let Ok(pcm) = capture_rx.recv() {
if is_muted_clone.load(Ordering::Relaxed) {
@@ -217,7 +220,7 @@ async fn run_core_loop(
if ptt_mode_clone.load(Ordering::Relaxed) && !ptt_active_clone.load(Ordering::Relaxed) {
continue;
}
let ng_bits = noise_gate_threshold_clone.load(Ordering::Relaxed);
let ng_thresh = f32::from_bits(ng_bits);
if ng_thresh > 0.0001 {
@@ -233,26 +236,22 @@ async fn run_core_loop(
}
if let Ok(encoded) = encoder.encode(&pcm) {
let bytes = bytes::Bytes::from(encoded);
let active = room_state_clone.active_peers();
for (peer_id, _) in active {
let transport = transport_clone.clone();
let bytes = bytes.clone();
tokio_handle.spawn(async move {
if let Err(e) = transport.send_datagram(peer_id, bytes).await {
crate::log_msg(&format!("Failed to send datagram to peer {:?}: {:?}", peer_id, e));
}
});
}
// Frame on the wire: [seq: u32 LE][opus payload].
let mut packet = Vec::with_capacity(4 + encoded.len());
packet.extend_from_slice(&seq.to_le_bytes());
packet.extend_from_slice(&encoded);
seq = seq.wrapping_add(1);
transport_clone.broadcast(bytes::Bytes::from(packet));
}
}
});
// 2. Receiver & decoding task
// 2. Receiver task: parse the sequence header and hand each packet
// to that peer's jitter buffer. Decoding happens later, on the
// playout side, so loss can be concealed at the right moment.
let transport_recv = transport.clone();
let queues_recv = queues.clone();
let jitter_recv = jitter.clone();
let datagram_task = tokio::spawn(async move {
use opus::Channels;
let mut datagram_rx = match transport_recv.receive_datagrams().await {
Ok(rx) => rx,
Err(e) => {
@@ -261,37 +260,34 @@ async fn run_core_loop(
}
};
let mut decoders: HashMap<EndpointId, OpusDecoder> = HashMap::new();
while let Some((from_peer, bytes)) = datagram_rx.recv().await {
crate::log_msg(&format!("Received datagram from peer={:?}, len={}", from_peer, bytes.len()));
let decoder = match decoders.entry(from_peer) {
if bytes.len() < 4 {
continue; // malformed: missing sequence header
}
let seq = u32::from_le_bytes([bytes[0], bytes[1], bytes[2], bytes[3]]);
let payload = bytes[4..].to_vec();
let mut guard = jitter_recv.lock().await;
let buffer = match guard.entry(from_peer) {
std::collections::hash_map::Entry::Occupied(entry) => entry.into_mut(),
std::collections::hash_map::Entry::Vacant(entry) => {
match OpusDecoder::new(48000, Channels::Mono) {
Ok(dec) => entry.insert(dec),
match JitterBuffer::new() {
Ok(jb) => entry.insert(jb),
Err(e) => {
crate::log_msg(&format!("Failed to initialize decoder for {:?}: {:?}", from_peer, e));
crate::log_msg(&format!("Failed to init jitter buffer for {:?}: {:?}", from_peer, e));
continue;
}
}
}
};
match decoder.decode(Some(&bytes)) {
Ok(pcm) => {
let mut guard = queues_recv.lock().await;
let queue = guard.entry(from_peer).or_insert_with(VecDeque::new);
queue.extend(pcm);
}
Err(e) => {
crate::log_msg(&format!("Failed to decode packet from {:?}: {:?}", from_peer, e));
}
}
buffer.insert(seq, payload);
}
});
// 3. Mixing & level extraction loop task
let queues_mixer = queues.clone();
// 3. Mixing & level extraction loop task. Every 20ms, pull one
// concealed frame per peer from its jitter buffer, apply
// per-peer volume, sum, and hand the mix to playback.
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();
@@ -302,55 +298,49 @@ async fn run_core_loop(
loop {
interval.tick().await;
let mut guard = queues_mixer.lock().await;
let mut mixed = vec![0i16; 960];
let current_volumes = peer_volumes_mixer.lock().await.clone();
let mut active_levels = Vec::new();
let mut peer_frames = Vec::new();
let current_volumes = peer_volumes_mixer.lock().await.clone();
for (&peer_id, queue) in guard.iter_mut() {
let mut frame = vec![0i16; 960];
let len = queue.len();
if len >= 960 {
if len > 9600 {
let drain = len - 960;
queue.drain(0..drain);
}
for sample in frame.iter_mut() {
*sample = queue.pop_front().unwrap_or(0);
}
} else {
for sample in frame.iter_mut().take(len) {
*sample = queue.pop_front().unwrap_or(0);
{
let mut guard = jitter_mixer.lock().await;
for (&peer_id, buffer) in guard.iter_mut() {
// `None` means idle/buffering: contribute nothing
// and report a zero level so the UI shows idle.
let Some(mut frame) = buffer.pop_frame() else {
active_levels.push((peer_id, 0.0));
continue;
};
let vol = current_volumes.get(&peer_id).copied().unwrap_or(1.0);
if (vol - 1.0).abs() > f32::EPSILON {
for sample in frame.iter_mut() {
*sample = (*sample as f32 * vol).clamp(i16::MIN as f32, i16::MAX as f32) as i16;
}
}
let sum_sq: f32 = frame.iter().map(|&x| (x as f32).powi(2)).sum();
let rms = (sum_sq / frame.len().max(1) as f32).sqrt();
let level = (rms / 32768.0).clamp(0.0, 1.0);
active_levels.push((peer_id, level));
peer_frames.push(frame);
}
let vol = current_volumes.get(&peer_id).copied().unwrap_or(1.0);
for sample in frame.iter_mut() {
*sample = (*sample as f32 * vol).clamp(i16::MIN as f32, i16::MAX as f32) as i16;
}
// Calculate speaking level (RMS normalized)
let sum_sq: f32 = frame.iter().map(|&x| (x as f32).powi(2)).sum();
let rms = (sum_sq / 960.0).sqrt();
let level = (rms / 32768.0).clamp(0.0, 1.0);
active_levels.push((peer_id, level));
peer_frames.push(frame);
}
let mut mixed = vec![0i16; FRAME_SAMPLES];
if !peer_frames.is_empty() {
for i in 0..960 {
let mut sum = 0i32;
for f in &peer_frames {
sum += f[i] as i32;
}
mixed[i] = sum.clamp(i16::MIN as i32, i16::MAX as i32) as i16;
for (i, out) in mixed.iter_mut().enumerate() {
let sum: i32 = peer_frames
.iter()
.map(|f| f.get(i).copied().unwrap_or(0) as i32)
.sum();
*out = sum.clamp(i16::MIN as i32, i16::MAX as i32) as i16;
}
}
let frame_to_send = if is_deafened_clone.load(Ordering::Relaxed) {
vec![0i16; 960]
vec![0i16; FRAME_SAMPLES]
} else {
mixed
};
@@ -372,16 +362,20 @@ async fn run_core_loop(
}
};
let ui_tx_events = ui_tx.clone();
let queues_events = queues.clone();
let jitter_events = jitter.clone();
let transport_events = transport.clone();
let event_task = tokio::spawn(async move {
while let Some(event) = room_events.recv().await {
match event {
RoomEvent::PeerJoined(peer_id, state) => {
queues_events.lock().await.entry(peer_id).or_insert_with(VecDeque::new);
// Establish the audio connection as soon as the peer
// is known (the transport dedupes the full-mesh race).
transport_events.connect_peer(peer_id).await;
let _ = ui_tx_events.send(UiEvent::PeerJoined { id: peer_id, state }).await;
}
RoomEvent::PeerLeft(peer_id) => {
queues_events.lock().await.remove(&peer_id);
transport_events.disconnect_peer(peer_id).await;
jitter_events.lock().await.remove(&peer_id);
let _ = ui_tx_events.send(UiEvent::PeerLeft { id: peer_id }).await;
}
RoomEvent::PeerUpdated(peer_id, state) => {
+133 -75
View File
@@ -2,16 +2,98 @@ use crate::network::{NetworkTransport, NetError};
use iroh::{Endpoint, EndpointId};
use iroh::endpoint::Connection;
use bytes::Bytes;
use tokio::sync::{mpsc, Mutex};
use tokio::sync::mpsc;
use tokio::sync::mpsc::Receiver;
use std::sync::Arc;
use std::sync::{Arc, Mutex as StdMutex};
use std::collections::HashMap;
use async_trait::async_trait;
#[derive(Debug, Clone)]
pub struct AudioProtocol {
const AUDIO_ALPN: &[u8] = b"peerspeak-audio";
/// Per-peer datagram send queue depth. Audio is real-time, so a backlog is
/// useless latency — keep it shallow and drop the oldest frame when full.
const SEND_QUEUE_DEPTH: usize = 8;
/// State shared between the transport and its protocol handler so both inbound
/// (accepted) and outbound (dialed) connections register the same way.
struct Shared {
/// Sync-lockable send handles, so `broadcast` can fan out from the (non-async)
/// capture/encode thread without touching the Tokio runtime.
senders: StdMutex<HashMap<EndpointId, mpsc::Sender<Bytes>>>,
/// Per-peer task handles + a retained connection clone. Keeping the clone
/// alive is what stops iroh from closing an accepted connection once the
/// `accept()` future returns.
peers: tokio::sync::Mutex<HashMap<EndpointId, PeerTasks>>,
incoming_tx: mpsc::Sender<(EndpointId, Bytes)>,
connections: Arc<Mutex<HashMap<EndpointId, Connection>>>,
}
struct PeerTasks {
send_task: tokio::task::JoinHandle<()>,
read_task: tokio::task::JoinHandle<()>,
_conn: Connection,
}
impl Shared {
/// Register a live connection: spin up its send loop (datagrams out) and
/// read loop (datagrams in). Idempotent — a second registration for an
/// already-known peer is ignored so we never run duplicate loops.
async fn register(self: &Arc<Self>, peer_id: EndpointId, conn: Connection) {
{
let peers = self.peers.lock().await;
if peers.contains_key(&peer_id) {
return;
}
}
let (send_tx, mut send_rx) = mpsc::channel::<Bytes>(SEND_QUEUE_DEPTH);
let conn_send = conn.clone();
let send_task = tokio::spawn(async move {
while let Some(data) = send_rx.recv().await {
if conn_send.send_datagram(data).is_err() {
break;
}
}
});
let conn_read = conn.clone();
let incoming_tx = self.incoming_tx.clone();
let read_task = tokio::spawn(async move {
while let Ok(bytes) = conn_read.read_datagram().await {
if incoming_tx.send((peer_id, bytes)).await.is_err() {
break;
}
}
});
self.senders.lock().unwrap().insert(peer_id, send_tx);
self.peers.lock().await.insert(
peer_id,
PeerTasks { send_task, read_task, _conn: conn },
);
crate::log_msg(&format!("Transport: registered peer {:?}", peer_id));
}
async fn remove(&self, peer_id: EndpointId) {
self.senders.lock().unwrap().remove(&peer_id);
if let Some(tasks) = self.peers.lock().await.remove(&peer_id) {
tasks.send_task.abort();
tasks.read_task.abort();
// Dropping `_conn` (the last retained clone) closes the connection.
crate::log_msg(&format!("Transport: removed peer {:?}", peer_id));
}
}
}
#[derive(Clone)]
pub struct AudioProtocol {
shared: Arc<Shared>,
}
impl std::fmt::Debug for AudioProtocol {
fn fmt(&self, f: &mut std::fmt::Formatter<'_>) -> std::fmt::Result {
f.debug_struct("AudioProtocol").finish_non_exhaustive()
}
}
impl iroh::protocol::ProtocolHandler for AudioProtocol {
@@ -20,24 +102,11 @@ impl iroh::protocol::ProtocolHandler for AudioProtocol {
connection: Connection,
) -> impl std::future::Future<Output = Result<(), iroh::protocol::AcceptError>> + Send {
let peer_id = connection.remote_id();
let incoming_tx = self.incoming_tx.clone();
let connections = self.connections.clone();
let shared = self.shared.clone();
async move {
connections.lock().await.insert(peer_id, connection.clone());
loop {
match connection.read_datagram().await {
Ok(bytes) => {
if incoming_tx.send((peer_id, bytes)).await.is_err() {
break;
}
}
Err(_) => {
connections.lock().await.remove(&peer_id);
break;
}
}
}
// Register and return: the retained connection clone in `PeerTasks`
// keeps the connection open after this future resolves.
shared.register(peer_id, connection).await;
Ok(())
}
}
@@ -45,81 +114,70 @@ impl iroh::protocol::ProtocolHandler for AudioProtocol {
pub struct IrohTransport {
endpoint: Endpoint,
connections: Arc<Mutex<HashMap<EndpointId, Connection>>>,
incoming_tx: mpsc::Sender<(EndpointId, Bytes)>,
incoming_rx: Mutex<Option<mpsc::Receiver<(EndpointId, Bytes)>>>,
self_id: EndpointId,
shared: Arc<Shared>,
incoming_rx: tokio::sync::Mutex<Option<mpsc::Receiver<(EndpointId, Bytes)>>>,
}
impl IrohTransport {
pub fn new(endpoint: Endpoint) -> (Self, AudioProtocol) {
let (incoming_tx, incoming_rx) = mpsc::channel(1000);
let connections = Arc::new(Mutex::new(HashMap::new()));
let self_id = endpoint.id();
let audio_proto = AudioProtocol {
incoming_tx: incoming_tx.clone(),
connections: connections.clone(),
};
let shared = Arc::new(Shared {
senders: StdMutex::new(HashMap::new()),
peers: tokio::sync::Mutex::new(HashMap::new()),
incoming_tx,
});
let protocol = AudioProtocol { shared: shared.clone() };
let transport = Self {
endpoint,
connections,
incoming_tx,
incoming_rx: Mutex::new(Some(incoming_rx)),
self_id,
shared,
incoming_rx: tokio::sync::Mutex::new(Some(incoming_rx)),
};
(transport, audio_proto)
(transport, protocol)
}
}
#[async_trait]
impl NetworkTransport for IrohTransport {
async fn send_datagram(&self, peer_id: EndpointId, data: Bytes) -> Result<(), NetError> {
let mut conns = self.connections.lock().await;
let conn = if let Some(conn) = conns.get(&peer_id) {
conn.clone()
} else {
// Establish a new connection.
// We use the same audio ALPN: b"peerspeak-audio"
let alpn = b"peerspeak-audio";
let conn = self.endpoint.connect(peer_id, alpn).await
.map_err(|e| NetError::Connection(e.to_string()))?;
async fn connect_peer(&self, peer_id: EndpointId) {
// Deterministic initiator: only the lexicographically-lower id dials, so
// a full-mesh pair forms exactly one shared connection instead of two
// racing ones. The higher id waits for the inbound `accept()`.
if self.self_id.to_string() >= peer_id.to_string() {
return;
}
if self.shared.peers.lock().await.contains_key(&peer_id) {
return;
}
conns.insert(peer_id, conn.clone());
match self.endpoint.connect(peer_id, AUDIO_ALPN).await {
Ok(conn) => self.shared.register(peer_id, conn).await,
Err(e) => crate::log_msg(&format!("Transport: dial to {:?} failed: {:?}", peer_id, e)),
}
}
let incoming_tx_inner = self.incoming_tx.clone();
let connections_inner = self.connections.clone();
let conn_clone = conn.clone();
async fn disconnect_peer(&self, peer_id: EndpointId) {
self.shared.remove(peer_id).await;
}
tokio::spawn(async move {
loop {
match conn_clone.read_datagram().await {
Ok(bytes) => {
if incoming_tx_inner.send((peer_id, bytes)).await.is_err() {
break;
}
}
Err(_) => {
connections_inner.lock().await.remove(&peer_id);
break;
}
}
}
});
conn
};
conn.send_datagram(data)
.map_err(|e| NetError::Connection(e.to_string()))?;
Ok(())
fn broadcast(&self, data: Bytes) {
let senders = self.shared.senders.lock().unwrap();
for tx in senders.values() {
// Drop on a full queue: stale audio is worthless, and we must never
// block the encode thread on a slow peer.
let _ = tx.try_send(data.clone());
}
}
async fn receive_datagrams(&self) -> Result<Receiver<(EndpointId, Bytes)>, NetError> {
let mut rx_guard = self.incoming_rx.lock().await;
if let Some(rx) = rx_guard.take() {
Ok(rx)
} else {
Err(NetError::Other("Datagram receiver already subscribed".to_string()))
}
rx_guard
.take()
.ok_or_else(|| NetError::Other("Datagram receiver already subscribed".to_string()))
}
}
+11 -2
View File
@@ -69,8 +69,17 @@ impl FromStr for PeerSpeakTicket {
#[async_trait]
pub trait NetworkTransport: Send + Sync {
/// Send a low-latency unreliable datagram to a specific peer (for audio).
async fn send_datagram(&self, peer_id: EndpointId, data: Bytes) -> Result<(), NetError>;
/// Establish (or ensure) a connection to a peer and set up its send path.
/// Idempotent; safe to call again for an already-connected peer.
async fn connect_peer(&self, peer_id: EndpointId);
/// Tear down the connection and send path for a peer that has left.
async fn disconnect_peer(&self, peer_id: EndpointId);
/// Fan a single audio datagram out to every connected peer. Non-blocking:
/// per-peer queues drop the oldest-pending frame when full, so a slow link
/// can never stall the capture/encode thread. Callable from any thread.
fn broadcast(&self, data: Bytes);
/// Subscribes to incoming datagrams from any peer.
async fn receive_datagrams(&self) -> Result<Receiver<(EndpointId, Bytes)>, NetError>;