9ae9441de4
Partial reads from the capture ring consumed samples that were then discarded when the send loop retried from buf[0]. For 20ms codecs this was invisible (single Oboe burst fills 960 samples in one read), but 40ms codecs (Opus6k, 1920 samples) needed 2 bursts — the first partial read consumed 960 real samples and threw them away. Result: Opus6k produced ~11 frames/s instead of 25 (~44% of expected). Fix: expose wzp_native_audio_capture_available() and check it before reading, matching the desktop capture_ring.available() pattern. Partial reads no longer occur because we only read when enough samples exist. Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
1706 lines
81 KiB
Rust
1706 lines
81 KiB
Rust
//! Call engine for the desktop app — wraps wzp-client audio + transport
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//! into a clean async interface for Tauri commands.
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//!
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//! Step C of the incremental Android rewrite: the module now compiles on
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//! Android too (previously cfg-gated out entirely in lib.rs), but the
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//! actual `CallEngine::start()` body uses CPAL via `wzp_client::audio_io`
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//! which is only available on desktop. On Android we expose a stub
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//! `start()` that returns an error, so the frontend's `connect` command
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//! still fails cleanly but the rest of the engine code links in.
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use std::net::SocketAddr;
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use std::sync::atomic::{AtomicBool, AtomicU8, AtomicU32, AtomicU64, Ordering};
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use std::sync::Arc;
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use std::time::Instant;
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use tauri::Emitter;
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use tokio::sync::Mutex;
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use tracing::{error, info};
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// CPAL audio I/O is only available on desktop (wzp-client's `audio` feature).
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#[cfg(not(target_os = "android"))]
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use wzp_client::audio_io::{AudioCapture, AudioPlayback};
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// Codec + handshake pipelines are platform-independent Rust (no CPAL
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// dependency) so they're available from wzp-client on both desktop and
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// Android (where wzp-client is pulled in with default-features=false).
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use wzp_client::call::{CallConfig, CallEncoder};
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use wzp_proto::traits::{AudioDecoder, QualityController};
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use wzp_proto::{AdaptiveQualityController, CodecId, MediaTransport, QualityProfile};
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const FRAME_SAMPLES_40MS: usize = 1920;
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/// Profile index mapping for the AtomicU8 adaptive-quality bridge.
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const PROFILE_NO_CHANGE: u8 = 0xFF;
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fn profile_to_index(p: &QualityProfile) -> u8 {
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match p.codec {
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CodecId::Opus64k => 0,
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CodecId::Opus48k => 1,
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CodecId::Opus32k => 2,
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CodecId::Opus24k => 3,
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CodecId::Opus6k => 4,
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CodecId::Codec2_1200 => 5,
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_ => 3, // default to GOOD
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}
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}
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fn index_to_profile(idx: u8) -> Option<QualityProfile> {
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match idx {
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0 => Some(QualityProfile::STUDIO_64K),
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1 => Some(QualityProfile::STUDIO_48K),
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2 => Some(QualityProfile::STUDIO_32K),
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3 => Some(QualityProfile::GOOD),
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4 => Some(QualityProfile::DEGRADED),
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5 => Some(QualityProfile::CATASTROPHIC),
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_ => None,
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}
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}
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/// Resolve a quality string from the UI to a QualityProfile.
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/// Returns None for "auto" (use default adaptive behavior).
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fn resolve_quality(quality: &str) -> Option<QualityProfile> {
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match quality {
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"good" | "opus" => Some(QualityProfile::GOOD),
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"degraded" | "opus6k" => Some(QualityProfile::DEGRADED),
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"catastrophic" | "codec2-1200" => Some(QualityProfile::CATASTROPHIC),
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"codec2-3200" => Some(QualityProfile {
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codec: CodecId::Codec2_3200,
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fec_ratio: 0.5,
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frame_duration_ms: 20,
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frames_per_block: 5,
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}),
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"studio-32k" => Some(QualityProfile::STUDIO_32K),
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"studio-48k" => Some(QualityProfile::STUDIO_48K),
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"studio-64k" => Some(QualityProfile::STUDIO_64K),
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_ => None, // "auto" or unknown
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}
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}
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/// Wrapper to make non-Sync audio handles safe to store in shared state.
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/// The audio handle is only accessed from the thread that created it (drop),
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/// never shared across threads — Sync is safe.
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#[allow(dead_code)]
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struct SyncWrapper(Box<dyn std::any::Any + Send>);
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unsafe impl Sync for SyncWrapper {}
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pub struct ParticipantInfo {
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pub fingerprint: String,
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pub alias: Option<String>,
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pub relay_label: Option<String>,
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}
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pub struct EngineStatus {
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pub mic_muted: bool,
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pub spk_muted: bool,
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pub participants: Vec<ParticipantInfo>,
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pub frames_sent: u64,
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pub frames_received: u64,
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pub audio_level: u32,
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pub call_duration_secs: f64,
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pub fingerprint: String,
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pub tx_codec: String,
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pub rx_codec: String,
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}
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pub struct CallEngine {
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running: Arc<AtomicBool>,
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mic_muted: Arc<AtomicBool>,
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spk_muted: Arc<AtomicBool>,
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participants: Arc<Mutex<Vec<ParticipantInfo>>>,
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frames_sent: Arc<AtomicU64>,
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frames_received: Arc<AtomicU64>,
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audio_level: Arc<AtomicU32>,
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tx_codec: Arc<Mutex<String>>,
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rx_codec: Arc<Mutex<String>>,
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transport: Arc<wzp_transport::QuinnTransport>,
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start_time: Instant,
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fingerprint: String,
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/// Keep audio handles alive for the duration of the call.
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/// Wrapped in SyncWrapper because AudioUnit isn't Sync.
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_audio_handle: SyncWrapper,
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}
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/// Phase 3b/3c DRED reconstruction state for a recv task.
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///
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/// Wraps the libopus 1.5 DRED decoder + two `DredState` buffers (scratch +
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/// cached last-good) + sequence tracking needed to fill packet-loss gaps
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/// with neural redundancy reconstruction. Lives inside the recv task of
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/// `CallEngine::start` and is reset on codec/profile switches.
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///
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/// The original Phase 3c port landed on `crates/wzp-android/src/engine.rs`,
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/// which turned out to be dead code on the Tauri mobile pipeline — the
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/// live Android audio recv path is in *this* file. This helper rehomes
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/// the same logic to the correct engine.
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struct DredRecvState {
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dred_decoder: wzp_codec::dred_ffi::DredDecoderHandle,
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scratch: wzp_codec::dred_ffi::DredState,
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last_good: wzp_codec::dred_ffi::DredState,
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last_good_seq: Option<u16>,
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expected_seq: Option<u16>,
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pub dred_reconstructions: u64,
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pub classical_plc_invocations: u64,
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/// Number of arriving Opus packets we have parsed for DRED so far —
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/// used to throttle the periodic "DRED state observed" log to one
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/// line every N packets so logcat doesn't drown.
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parses_total: u64,
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/// Counter of parses that yielded a non-zero `samples_available`.
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parses_with_data: u64,
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}
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impl DredRecvState {
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fn new() -> Self {
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Self {
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dred_decoder: wzp_codec::dred_ffi::DredDecoderHandle::new()
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.expect("opus_dred_decoder_create failed at call setup"),
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scratch: wzp_codec::dred_ffi::DredState::new()
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.expect("opus_dred_alloc failed at call setup (scratch)"),
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last_good: wzp_codec::dred_ffi::DredState::new()
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.expect("opus_dred_alloc failed at call setup (good state)"),
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last_good_seq: None,
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expected_seq: None,
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dred_reconstructions: 0,
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classical_plc_invocations: 0,
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parses_total: 0,
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parses_with_data: 0,
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}
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}
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/// Parse DRED side-channel data from an arriving Opus source packet
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/// into the scratch state; on success, swap it into the cached good
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/// state and record the sequence number as the new anchor.
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///
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/// Call this BEFORE `fill_gap_to` so the anchor reflects the freshest
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/// DRED source available for gap reconstruction.
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fn ingest_opus(&mut self, seq: u16, payload: &[u8]) {
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self.parses_total += 1;
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match self.dred_decoder.parse_into(&mut self.scratch, payload) {
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Ok(available) if available > 0 => {
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self.parses_with_data += 1;
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std::mem::swap(&mut self.scratch, &mut self.last_good);
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self.last_good_seq = Some(seq);
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// First successful parse on this call: log loudly so the
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// user can see "DRED is on the wire" in logcat. After
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// that, sample every 100th parse to confirm the window
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// is steady-state without drowning the log.
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let should_log = self.parses_with_data == 1
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|| self.parses_with_data % 100 == 0;
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if should_log && wzp_codec::dred_verbose_logs() {
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info!(
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seq,
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samples_available = available,
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ms = available / 48,
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parses_with_data = self.parses_with_data,
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parses_total = self.parses_total,
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"DRED state parsed from Opus packet"
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);
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}
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}
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_ => {
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// Packet carried no DRED data, or parse failed — keep
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// the cached good state (it may still cover upcoming
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// gaps from a warm-up period).
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}
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}
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}
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/// On an arriving packet with sequence `current_seq`, detect any gap
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/// from `expected_seq` to `current_seq - 1` and fill the missing
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/// frames via DRED reconstruction (if state covers them) or classical
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/// Opus PLC fallback. The `emit` callback is invoked once per
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/// reconstructed/concealed frame with a `&mut [i16]` slice of length
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/// `frame_samples`; the caller is responsible for AGC + playout.
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///
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/// Updates `expected_seq` to `current_seq + 1` on return.
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fn fill_gap_to<F>(
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&mut self,
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decoder: &mut wzp_codec::AdaptiveDecoder,
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current_seq: u16,
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frame_samples: usize,
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pcm_scratch: &mut [i16],
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mut emit: F,
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) where
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F: FnMut(&mut [i16]),
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{
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const MAX_GAP_FRAMES: u16 = 16;
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if let Some(expected) = self.expected_seq {
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let gap = current_seq.wrapping_sub(expected);
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if gap > 0 && gap <= MAX_GAP_FRAMES {
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let available = self.last_good.samples_available();
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for gap_idx in 0..gap {
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let missing_seq = expected.wrapping_add(gap_idx);
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let offset_samples = match self.last_good_seq {
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Some(anchor) => {
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let delta = anchor.wrapping_sub(missing_seq);
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if delta == 0 || delta > MAX_GAP_FRAMES {
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-1 // skip DRED, fall through to PLC
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} else {
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delta as i32 * frame_samples as i32
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}
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}
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None => -1,
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};
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let out = &mut pcm_scratch[..frame_samples];
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let reconstructed = if offset_samples > 0 && offset_samples <= available {
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decoder
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.reconstruct_from_dred(&self.last_good, offset_samples, out)
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.ok()
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} else {
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None
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};
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match reconstructed {
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Some(_n) => {
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self.dred_reconstructions += 1;
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// Log every DRED reconstruction (gated behind
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// the GUI verbose-logs toggle). When enabled,
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// we want to know exactly which gap was
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// filled and how the offset math played out.
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if wzp_codec::dred_verbose_logs() {
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info!(
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missing_seq,
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anchor_seq = ?self.last_good_seq,
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offset_samples,
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offset_ms = offset_samples / 48,
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samples_available = available,
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gap_size = gap,
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total_dred_recoveries = self.dred_reconstructions,
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"DRED reconstruction fired for missing frame"
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);
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}
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emit(out);
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}
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None => {
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if decoder.decode_lost(out).is_ok() {
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self.classical_plc_invocations += 1;
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// Log the first few classical PLC fills
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// and then sample, so we can see when
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// DRED couldn't cover a gap. The reason
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// is whichever check failed in the if
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// above (offset out of range, no good
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// state, or reconstruct error).
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if (self.classical_plc_invocations <= 3
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|| self.classical_plc_invocations % 50 == 0)
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&& wzp_codec::dred_verbose_logs()
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{
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info!(
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missing_seq,
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anchor_seq = ?self.last_good_seq,
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offset_samples,
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samples_available = available,
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total_classical_plc = self.classical_plc_invocations,
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"classical PLC fill (DRED could not cover gap)"
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);
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}
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emit(out);
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}
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}
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}
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}
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}
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}
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self.expected_seq = Some(current_seq.wrapping_add(1));
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}
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/// Invalidate sequence tracking on profile switch. The cached DRED
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/// state is tied to the old profile's frame rate so offsets would
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/// produce wrong reconstructions until the next good-state parse.
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fn reset_on_profile_switch(&mut self) {
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self.last_good_seq = None;
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self.expected_seq = None;
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}
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}
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impl CallEngine {
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/// Android engine path — uses the standalone `wzp-native` cdylib
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/// (loaded at startup via `crate::wzp_native::init()`) for Oboe-backed
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/// capture and playout instead of CPAL. Mirrors the desktop send/recv
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/// task structure otherwise.
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#[cfg(target_os = "android")]
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pub async fn start<F>(
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relay: String,
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room: String,
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alias: String,
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_os_aec: bool,
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quality: String,
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reuse_endpoint: Option<wzp_transport::Endpoint>,
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// Phase 3.5: caller did the dual-path race and picked a
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// winning transport (direct or relay). If Some, we skip
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// our own wzp_transport::connect step and use this
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// directly. If None, existing Phase 0 behavior.
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pre_connected_transport: Option<Arc<wzp_transport::QuinnTransport>>,
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// Phase 6: explicit flag for whether the agreed media path
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// is truly direct P2P (skip handshake) or relay-mediated
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// (must run handshake). Previously derived from
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// pre_connected_transport.is_some() which was WRONG: when
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// Phase 6 negotiated relay but delivered the relay transport
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// via pre_connected_transport, the engine skipped the
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// handshake → relay couldn't authenticate the participant
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// → silent call.
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is_direct_p2p: bool,
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// Phase 5.6: Tauri AppHandle for emitting call-debug
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// events from inside the send/recv tasks. Lets the
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// debug log pane show first-send/first-recv/heartbeat
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// events when the user has call debug logs enabled.
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app: tauri::AppHandle,
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event_cb: F,
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) -> Result<Self, anyhow::Error>
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where
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F: Fn(&str, &str) + Send + Sync + 'static,
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{
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let call_t0 = std::time::Instant::now();
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info!(
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%relay, %room, %alias, %quality,
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has_reuse = reuse_endpoint.is_some(),
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has_pre_connected = pre_connected_transport.is_some(),
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is_direct_p2p,
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t_ms = 0u128,
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"CallEngine::start (android) invoked"
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);
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let _ = rustls::crypto::ring::default_provider().install_default();
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|
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let relay_addr: SocketAddr = relay.parse()?;
|
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info!(%relay_addr, "resolved relay addr");
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|
|
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let seed = crate::load_or_create_seed()
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.map_err(|e| anyhow::anyhow!("identity: {e}"))?;
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let fp = seed.derive_identity().public_identity().fingerprint;
|
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let fingerprint = fp.to_string();
|
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info!(%fp, "identity loaded");
|
|
|
|
// Transport source: either the pre-connected one from the
|
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// dual-path race or build a fresh one here.
|
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let transport = if let Some(t) = pre_connected_transport {
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info!(t_ms = call_t0.elapsed().as_millis(), is_direct_p2p, "first-join diag: using pre-connected transport");
|
|
t
|
|
} else {
|
|
// QUIC transport + handshake (Phase 0 relay-only path).
|
|
//
|
|
// If a `reuse_endpoint` was passed in (the direct-call path, where we
|
|
// already opened a quinn::Endpoint for the signal connection), reuse
|
|
// it: a second quinn::Endpoint on Android silently fails to complete
|
|
// the QUIC handshake against the same relay. Reusing the existing
|
|
// socket lets quinn multiplex the signal + media connections on one
|
|
// UDP port.
|
|
let endpoint = if let Some(ep) = reuse_endpoint {
|
|
info!(local_addr = ?ep.local_addr().ok(), "reusing signal endpoint for media connection");
|
|
ep
|
|
} else {
|
|
let bind_addr: SocketAddr = "0.0.0.0:0".parse().unwrap();
|
|
let ep = wzp_transport::create_endpoint(bind_addr, None)
|
|
.map_err(|e| { error!("create_endpoint failed: {e}"); e })?;
|
|
info!(local_addr = ?ep.local_addr().ok(), "created new endpoint, dialing relay");
|
|
ep
|
|
};
|
|
let client_config = wzp_transport::client_config();
|
|
let conn = match tokio::time::timeout(
|
|
std::time::Duration::from_secs(10),
|
|
wzp_transport::connect(&endpoint, relay_addr, &room, client_config),
|
|
).await {
|
|
Ok(Ok(c)) => c,
|
|
Ok(Err(e)) => {
|
|
error!("connect failed: {e}");
|
|
return Err(e.into());
|
|
}
|
|
Err(_) => {
|
|
error!("connect TIMED OUT after 10s — QUIC handshake never completed. Relay may be unreachable from this endpoint.");
|
|
return Err(anyhow::anyhow!("QUIC connect timeout (10s)"));
|
|
}
|
|
};
|
|
info!(t_ms = call_t0.elapsed().as_millis(), "first-join diag: QUIC connection established, performing handshake");
|
|
Arc::new(wzp_transport::QuinnTransport::new(conn))
|
|
};
|
|
|
|
// The media handshake (CallOffer/CallAnswer + crypto key
|
|
// exchange) is a relay-specific protocol: the relay runs
|
|
// `accept_handshake` on its side. On a direct P2P
|
|
// connection the peer is a phone, not a relay — nobody on
|
|
// the other end handles the handshake. So skip it when
|
|
// is_direct_p2p. The QUIC transport already provides TLS
|
|
// encryption, and both peers' identities were verified
|
|
// through the signal channel (DirectCallOffer/Answer carry
|
|
// identity_pub + ephemeral_pub + signature).
|
|
if !is_direct_p2p {
|
|
let _session = wzp_client::handshake::perform_handshake(
|
|
&*transport,
|
|
&seed.0,
|
|
Some(&alias),
|
|
)
|
|
.await
|
|
.map_err(|e| { error!("perform_handshake failed: {e}"); e })?;
|
|
info!(t_ms = call_t0.elapsed().as_millis(), "first-join diag: connected to relay, handshake complete");
|
|
} else {
|
|
info!(t_ms = call_t0.elapsed().as_millis(), "first-join diag: direct P2P — skipping relay handshake (QUIC TLS is the encryption layer)");
|
|
}
|
|
event_cb("connected", &format!("joined room {room}"));
|
|
|
|
// Oboe audio via the wzp-native cdylib that was dlopen'd at
|
|
// startup. `wzp_native::audio_start()` brings up the capture +
|
|
// playout streams; send/recv tasks below pull/push PCM through
|
|
// the extern "C" bridge rings.
|
|
if !crate::wzp_native::is_loaded() {
|
|
return Err(anyhow::anyhow!(
|
|
"wzp-native not loaded — dlopen failed at startup"
|
|
));
|
|
}
|
|
|
|
// Fix D (task #37): explicit stop+start cycle on EVERY call
|
|
// start — not just rejoin. Empirically, the first call after
|
|
// app launch on Nothing Phone has the Oboe playout callback
|
|
// fire once (cb#0) and then stop draining the ring, causing
|
|
// written_samples to freeze at 7679 (ring capacity minus
|
|
// one burst). Rejoin (second call) always works because
|
|
// audio_stop tears down the streams and audio_start rebuilds
|
|
// them in a state that the audio driver accepts. By always
|
|
// running stop first (no-op on cold start when not yet
|
|
// started), we get the same "fresh rebuild" behavior on
|
|
// every call.
|
|
crate::wzp_native::audio_stop();
|
|
// Brief pause to let Android's audio routing + AudioManager
|
|
// settle after the stop. 50ms is enough for the driver to
|
|
// release the audio session; shorter risks the new start
|
|
// hitting a "device busy" on some HALs.
|
|
tokio::time::sleep(std::time::Duration::from_millis(50)).await;
|
|
|
|
// Set MODE_IN_COMMUNICATION right before audio starts — NOT at
|
|
// app launch. Setting it early hijacks system audio routing
|
|
// (music drops from BT A2DP to earpiece, etc.).
|
|
#[cfg(target_os = "android")]
|
|
{
|
|
if let Err(e) = crate::android_audio::set_audio_mode_communication() {
|
|
tracing::warn!("set_audio_mode_communication failed: {e}");
|
|
}
|
|
}
|
|
|
|
let t_pre_audio = call_t0.elapsed().as_millis();
|
|
if let Err(code) = crate::wzp_native::audio_start() {
|
|
return Err(anyhow::anyhow!("wzp_native_audio_start failed: code {code}"));
|
|
}
|
|
|
|
// Fix C (task #36): prime the playout ring with 20ms of
|
|
// silence immediately after audio_start so the Oboe playout
|
|
// callback has data to drain on its FIRST invocation. On
|
|
// devices where the callback only fires when the ring is
|
|
// non-empty (or where an empty-ring callback causes the
|
|
// stream to self-pause), this ensures the callback keeps
|
|
// running until real decoded audio arrives.
|
|
{
|
|
let silence = vec![0i16; 960]; // 20ms @ 48kHz mono
|
|
let _ = crate::wzp_native::audio_write_playout(&silence);
|
|
}
|
|
|
|
let t_audio_start_done = call_t0.elapsed().as_millis();
|
|
info!(
|
|
t_ms = t_audio_start_done,
|
|
audio_start_ms = t_audio_start_done.saturating_sub(t_pre_audio),
|
|
"first-join diag: wzp-native audio started (with stop+prime cycle)"
|
|
);
|
|
|
|
let running = Arc::new(AtomicBool::new(true));
|
|
let mic_muted = Arc::new(AtomicBool::new(false));
|
|
let spk_muted = Arc::new(AtomicBool::new(false));
|
|
let participants: Arc<Mutex<Vec<ParticipantInfo>>> = Arc::new(Mutex::new(vec![]));
|
|
let frames_sent = Arc::new(AtomicU64::new(0));
|
|
let frames_received = Arc::new(AtomicU64::new(0));
|
|
let audio_level = Arc::new(AtomicU32::new(0));
|
|
let tx_codec = Arc::new(Mutex::new(String::new()));
|
|
let rx_codec = Arc::new(Mutex::new(String::new()));
|
|
|
|
// Adaptive quality: shared pending-profile bridge between recv → send.
|
|
let pending_profile = Arc::new(AtomicU8::new(PROFILE_NO_CHANGE));
|
|
let auto_profile = resolve_quality(&quality).is_none();
|
|
|
|
// Send task — drain Oboe capture ring, Opus-encode, push to transport.
|
|
let send_t = transport.clone();
|
|
let send_r = running.clone();
|
|
let send_mic = mic_muted.clone();
|
|
let send_fs = frames_sent.clone();
|
|
let send_level = audio_level.clone();
|
|
let send_drops = Arc::new(AtomicU64::new(0));
|
|
let send_last_err: Arc<Mutex<Option<String>>> = Arc::new(Mutex::new(None));
|
|
let send_quality = quality.clone();
|
|
let send_tx_codec = tx_codec.clone();
|
|
let send_t0 = call_t0;
|
|
let send_app = app.clone();
|
|
let send_pending_profile = pending_profile.clone();
|
|
tokio::spawn(async move {
|
|
let profile = resolve_quality(&send_quality);
|
|
let config = match profile {
|
|
Some(p) => CallConfig {
|
|
noise_suppression: false,
|
|
suppression_enabled: false,
|
|
..CallConfig::from_profile(p)
|
|
},
|
|
None => CallConfig {
|
|
noise_suppression: false,
|
|
suppression_enabled: false,
|
|
..CallConfig::default()
|
|
},
|
|
};
|
|
let mut frame_samples = (config.profile.frame_duration_ms as usize) * 48;
|
|
info!(codec = ?config.profile.codec, frame_samples, t_ms = send_t0.elapsed().as_millis(), "first-join diag: send task spawned (android/oboe)");
|
|
*send_tx_codec.lock().await = format!("{:?}", config.profile.codec);
|
|
let mut encoder = CallEncoder::new(&config);
|
|
encoder.set_aec_enabled(false);
|
|
// Sized for max frame (40ms = 1920 samples) so profile
|
|
// switches between 20ms ↔ 40ms codecs don't need realloc.
|
|
let mut buf = vec![0i16; 1920];
|
|
|
|
// Continuous DRED tuning: poll quinn path stats every 25
|
|
// frames (~500 ms at 20 ms/frame) and adjust DRED duration +
|
|
// expected-loss hint based on real-time network conditions.
|
|
let mut dred_tuner = wzp_proto::DredTuner::new(config.profile.codec);
|
|
let mut frames_since_dred_poll: u32 = 0;
|
|
const DRED_POLL_INTERVAL: u32 = 25;
|
|
|
|
let mut heartbeat = std::time::Instant::now();
|
|
let mut last_rms: u32 = 0;
|
|
let mut last_pkt_bytes: usize = 0;
|
|
let mut short_reads: u64 = 0;
|
|
// First-join diagnostic: latch the wall-clock offset of the
|
|
// first full-frame capture read and the first non-zero RMS
|
|
// reading separately. The gap between them tells us how long
|
|
// Oboe input took to actually start delivering real samples
|
|
// after returning a "started" status from audio_start.
|
|
let mut first_full_read_logged = false;
|
|
let mut first_nonzero_rms_logged = false;
|
|
|
|
loop {
|
|
if !send_r.load(Ordering::Relaxed) {
|
|
break;
|
|
}
|
|
// Check ring has enough samples before reading to avoid
|
|
// partial reads that consume samples and then get
|
|
// overwritten on the next attempt (caused 40ms codecs
|
|
// like Opus6k to produce ~11 frames/s instead of 25).
|
|
if crate::wzp_native::audio_capture_available() < frame_samples {
|
|
short_reads += 1;
|
|
tokio::time::sleep(std::time::Duration::from_millis(5)).await;
|
|
continue;
|
|
}
|
|
let read = crate::wzp_native::audio_read_capture(&mut buf[..frame_samples]);
|
|
if read < frame_samples {
|
|
// Shouldn't happen after available() check, but guard anyway.
|
|
short_reads += 1;
|
|
continue;
|
|
}
|
|
if !first_full_read_logged {
|
|
info!(
|
|
t_ms = send_t0.elapsed().as_millis(),
|
|
short_reads_before = short_reads,
|
|
frame_samples,
|
|
"first-join diag: send first full capture frame read"
|
|
);
|
|
first_full_read_logged = true;
|
|
}
|
|
|
|
// RMS for UI meter
|
|
let sum_sq: f64 = buf[..frame_samples].iter().map(|&s| (s as f64) * (s as f64)).sum();
|
|
let rms = (sum_sq / frame_samples as f64).sqrt() as u32;
|
|
send_level.store(rms, Ordering::Relaxed);
|
|
last_rms = rms;
|
|
if !first_nonzero_rms_logged && rms > 0 {
|
|
info!(
|
|
t_ms = send_t0.elapsed().as_millis(),
|
|
rms,
|
|
"first-join diag: send first non-zero capture RMS"
|
|
);
|
|
first_nonzero_rms_logged = true;
|
|
}
|
|
|
|
if send_mic.load(Ordering::Relaxed) {
|
|
buf[..frame_samples].fill(0);
|
|
}
|
|
match encoder.encode_frame(&buf[..frame_samples]) {
|
|
Ok(pkts) => {
|
|
for pkt in &pkts {
|
|
last_pkt_bytes = pkt.payload.len();
|
|
if let Err(e) = send_t.send_media(pkt).await {
|
|
send_drops.fetch_add(1, Ordering::Relaxed);
|
|
let count = send_drops.load(Ordering::Relaxed);
|
|
if count <= 3 {
|
|
tracing::warn!("send_media error (dropping packet): {e}");
|
|
}
|
|
// Latch last error for heartbeat
|
|
if count == 1 {
|
|
*send_last_err.lock().await = Some(format!("{e}"));
|
|
}
|
|
}
|
|
}
|
|
let before = send_fs.fetch_add(1, Ordering::Relaxed);
|
|
if before == 0 {
|
|
// First encoded frame successfully handed
|
|
// to the transport. Useful for diagnosing
|
|
// 1-way audio: if this fires but the
|
|
// peer's media:first_recv never does,
|
|
// outbound is broken on our side.
|
|
crate::emit_call_debug(
|
|
&send_app,
|
|
"media:first_send",
|
|
serde_json::json!({
|
|
"t_ms": send_t0.elapsed().as_millis() as u64,
|
|
"pkt_bytes": last_pkt_bytes,
|
|
}),
|
|
);
|
|
}
|
|
}
|
|
Err(e) => error!("encode: {e}"),
|
|
}
|
|
|
|
// Adaptive quality: check if recv task recommended a profile switch.
|
|
if auto_profile {
|
|
let p = send_pending_profile.swap(PROFILE_NO_CHANGE, Ordering::Acquire);
|
|
if p != PROFILE_NO_CHANGE {
|
|
if let Some(new_profile) = index_to_profile(p) {
|
|
let new_fs = (new_profile.frame_duration_ms as usize) * 48;
|
|
info!(to = ?new_profile.codec, frame_samples = new_fs, "auto: switching encoder profile (android)");
|
|
if encoder.set_profile(new_profile).is_ok() {
|
|
frame_samples = new_fs;
|
|
dred_tuner.set_codec(new_profile.codec);
|
|
*send_tx_codec.lock().await = format!("{:?}", new_profile.codec);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// DRED tuner: poll quinn path stats periodically and
|
|
// adjust encoder DRED duration + expected-loss hint.
|
|
frames_since_dred_poll += 1;
|
|
if frames_since_dred_poll >= DRED_POLL_INTERVAL {
|
|
frames_since_dred_poll = 0;
|
|
let snap = send_t.quinn_path_stats();
|
|
let pq = send_t.path_quality();
|
|
if let Some(tuning) = dred_tuner.update(
|
|
snap.loss_pct,
|
|
snap.rtt_ms,
|
|
pq.jitter_ms,
|
|
) {
|
|
encoder.apply_dred_tuning(tuning);
|
|
if wzp_codec::dred_verbose_logs() {
|
|
info!(
|
|
dred_frames = tuning.dred_frames,
|
|
dred_ms = tuning.dred_frames as u32 * 10,
|
|
expected_loss = tuning.expected_loss_pct,
|
|
quinn_loss = format!("{:.1}", snap.loss_pct),
|
|
quinn_rtt = snap.rtt_ms,
|
|
jitter = pq.jitter_ms,
|
|
spike = dred_tuner.spike_boost_active(),
|
|
"DRED tuner adjusted encoder"
|
|
);
|
|
}
|
|
}
|
|
}
|
|
|
|
// Heartbeat every 2s with capture+encode+send state
|
|
if heartbeat.elapsed() >= std::time::Duration::from_secs(2) {
|
|
let fs = send_fs.load(Ordering::Relaxed);
|
|
let drops = send_drops.load(Ordering::Relaxed);
|
|
info!(
|
|
frames_sent = fs,
|
|
last_rms,
|
|
last_pkt_bytes,
|
|
short_reads,
|
|
send_drops = drops,
|
|
"send heartbeat (android)"
|
|
);
|
|
// Phase 5.6: also emit to the GUI debug log
|
|
// when call debug is enabled. Helps diagnose
|
|
// 1-way audio — a stalled send heartbeat
|
|
// (frames_sent == 0 or last_rms == 0) tells
|
|
// you capture/mic is broken; a live one with
|
|
// no peer recv tells you outbound is being
|
|
// dropped somewhere in the media path.
|
|
let err_str = send_last_err.lock().await.clone();
|
|
crate::emit_call_debug(
|
|
&send_app,
|
|
"media:send_heartbeat",
|
|
serde_json::json!({
|
|
"frames_sent": fs,
|
|
"last_rms": last_rms,
|
|
"last_pkt_bytes": last_pkt_bytes,
|
|
"short_reads": short_reads,
|
|
"drops": drops,
|
|
"last_send_err": err_str,
|
|
}),
|
|
);
|
|
heartbeat = std::time::Instant::now();
|
|
}
|
|
}
|
|
});
|
|
|
|
// Recv task — decode incoming packets, push PCM into Oboe playout.
|
|
let recv_t = transport.clone();
|
|
let recv_r = running.clone();
|
|
let recv_spk = spk_muted.clone();
|
|
let recv_fr = frames_received.clone();
|
|
let recv_rx_codec = rx_codec.clone();
|
|
let recv_t0 = call_t0;
|
|
let recv_app = app.clone();
|
|
let pending_profile_recv = pending_profile.clone();
|
|
tokio::spawn(async move {
|
|
let initial_profile = resolve_quality(&quality).unwrap_or(QualityProfile::GOOD);
|
|
// Phase 3b/3c: use concrete AdaptiveDecoder (not Box<dyn
|
|
// AudioDecoder>) so we can call the inherent
|
|
// reconstruct_from_dred method on packet-loss gaps.
|
|
let mut decoder = wzp_codec::AdaptiveDecoder::new(initial_profile)
|
|
.expect("failed to create adaptive decoder");
|
|
let mut current_profile = initial_profile;
|
|
let mut current_codec = initial_profile.codec;
|
|
let mut agc = wzp_codec::AutoGainControl::new();
|
|
let mut pcm = vec![0i16; FRAME_SAMPLES_40MS];
|
|
// Phase 3b/3c DRED reconstruction state — see DredRecvState
|
|
// above for the full flow.
|
|
let mut dred_recv = DredRecvState::new();
|
|
let mut quality_ctrl = AdaptiveQualityController::new();
|
|
info!(codec = ?current_codec, t_ms = recv_t0.elapsed().as_millis(), "first-join diag: recv task spawned (android/oboe)");
|
|
// First-join diagnostic latches — see send task above for the
|
|
// sibling capture milestones.
|
|
let mut first_decode_logged = false;
|
|
let mut first_playout_write_logged = false;
|
|
|
|
// ─── Decoded-PCM recorder (debug) ────────────────────────────
|
|
// Dumps the first ~10 seconds of post-AGC PCM to a raw i16 LE
|
|
// file in the app's private data dir so we can adb pull it and
|
|
// play it back to prove the pipeline is producing real audio
|
|
// independent of Oboe routing. Convert locally with e.g.
|
|
// ffmpeg -f s16le -ar 48000 -ac 1 -i decoded.pcm decoded.wav
|
|
use std::io::Write;
|
|
let recorder_path = crate::APP_DATA_DIR
|
|
.get()
|
|
.map(|p| p.join("decoded.pcm"));
|
|
let mut recorder = match recorder_path.as_ref() {
|
|
Some(p) => match std::fs::File::create(p) {
|
|
Ok(f) => {
|
|
info!(path = %p.display(), "decoded-pcm recorder open");
|
|
Some(std::io::BufWriter::new(f))
|
|
}
|
|
Err(e) => {
|
|
tracing::warn!(path = %p.display(), error = %e, "decoded-pcm recorder open failed");
|
|
None
|
|
}
|
|
},
|
|
None => None,
|
|
};
|
|
let mut recorder_bytes: u64 = 0;
|
|
// Stop writing after ~10 seconds @ 48kHz mono i16 = ~960KB.
|
|
const RECORDER_MAX_BYTES: u64 = 48_000 * 2 * 10;
|
|
|
|
let mut heartbeat = std::time::Instant::now();
|
|
let mut decoded_frames: u64 = 0;
|
|
let mut written_samples: u64 = 0;
|
|
let mut last_decode_n: usize = 0;
|
|
let mut last_written: usize = 0;
|
|
let mut decode_errs: u64 = 0;
|
|
let mut first_packet_logged = false;
|
|
// Phase 5.6: media health watchdog — track consecutive
|
|
// heartbeat ticks where recv_fr hasn't advanced. If
|
|
// media doesn't arrive for 3 consecutive heartbeats
|
|
// (6s), emit a user-facing "media-degraded" call-event
|
|
// so the UI can show a warning like "No audio — try
|
|
// reconnecting?". Covers the case where P2P direct
|
|
// established but the underlying network path died
|
|
// (e.g., phone switched from WiFi to LTE mid-call).
|
|
let mut last_recv_fr_for_watchdog: u64 = 0;
|
|
let mut no_recv_ticks: u32 = 0;
|
|
let mut media_degraded_emitted = false;
|
|
|
|
loop {
|
|
if !recv_r.load(Ordering::Relaxed) {
|
|
break;
|
|
}
|
|
match tokio::time::timeout(
|
|
std::time::Duration::from_millis(100),
|
|
recv_t.recv_media(),
|
|
)
|
|
.await
|
|
{
|
|
Ok(Ok(Some(pkt))) => {
|
|
if !first_packet_logged {
|
|
info!(
|
|
t_ms = recv_t0.elapsed().as_millis(),
|
|
codec_id = ?pkt.header.codec_id,
|
|
payload_bytes = pkt.payload.len(),
|
|
is_repair = pkt.header.is_repair,
|
|
"first-join diag: recv first media packet"
|
|
);
|
|
first_packet_logged = true;
|
|
// Phase 5.6 GUI debug: first packet from
|
|
// the peer. Useful for diagnosing 1-way
|
|
// audio — if this fires and the peer
|
|
// never sees media:first_recv, our
|
|
// inbound path is fine and theirs is
|
|
// broken, and vice versa.
|
|
crate::emit_call_debug(
|
|
&recv_app,
|
|
"media:first_recv",
|
|
serde_json::json!({
|
|
"t_ms": recv_t0.elapsed().as_millis() as u64,
|
|
"codec": format!("{:?}", pkt.header.codec_id),
|
|
"payload_bytes": pkt.payload.len(),
|
|
"is_repair": pkt.header.is_repair,
|
|
}),
|
|
);
|
|
}
|
|
if !pkt.header.is_repair && pkt.header.codec_id != CodecId::ComfortNoise {
|
|
{
|
|
let mut rx = recv_rx_codec.lock().await;
|
|
let codec_name = format!("{:?}", pkt.header.codec_id);
|
|
if *rx != codec_name { *rx = codec_name; }
|
|
}
|
|
if pkt.header.codec_id != current_codec {
|
|
let new_profile = match pkt.header.codec_id {
|
|
CodecId::Opus24k => QualityProfile::GOOD,
|
|
CodecId::Opus6k => QualityProfile::DEGRADED,
|
|
CodecId::Opus32k => QualityProfile::STUDIO_32K,
|
|
CodecId::Opus48k => QualityProfile::STUDIO_48K,
|
|
CodecId::Opus64k => QualityProfile::STUDIO_64K,
|
|
CodecId::Codec2_1200 => QualityProfile::CATASTROPHIC,
|
|
CodecId::Codec2_3200 => QualityProfile {
|
|
codec: CodecId::Codec2_3200,
|
|
fec_ratio: 0.5, frame_duration_ms: 20, frames_per_block: 5,
|
|
},
|
|
other => QualityProfile { codec: other, ..QualityProfile::GOOD },
|
|
};
|
|
info!(from = ?current_codec, to = ?pkt.header.codec_id, "recv: switching decoder");
|
|
let _ = decoder.set_profile(new_profile);
|
|
current_profile = new_profile;
|
|
current_codec = pkt.header.codec_id;
|
|
// Phase 3c: new profile → offsets in the
|
|
// cached DRED state are invalid; reset.
|
|
dred_recv.reset_on_profile_switch();
|
|
}
|
|
|
|
// Phase 3b/3c DRED flow for Opus packets:
|
|
// 1. parse DRED from this packet → last_good
|
|
// 2. detect gap back to expected_seq and
|
|
// reconstruct missing frames via DRED
|
|
// (or classical PLC if no state covers)
|
|
// 3. then decode the current packet normally
|
|
// (unchanged fall-through below)
|
|
//
|
|
// Codec2 packets skip DRED entirely — libopus
|
|
// can't reconstruct them and the parse is a
|
|
// no-op.
|
|
if pkt.header.codec_id.is_opus() {
|
|
dred_recv.ingest_opus(pkt.header.seq, &pkt.payload);
|
|
let frame_samples_now = (48_000
|
|
* current_profile.frame_duration_ms as usize)
|
|
/ 1000;
|
|
let spk_muted_flag = recv_spk.load(Ordering::Relaxed);
|
|
dred_recv.fill_gap_to(
|
|
&mut decoder,
|
|
pkt.header.seq,
|
|
frame_samples_now,
|
|
&mut pcm,
|
|
|samples| {
|
|
agc.process_frame(samples);
|
|
if !spk_muted_flag {
|
|
let _ = crate::wzp_native::audio_write_playout(samples);
|
|
}
|
|
},
|
|
);
|
|
}
|
|
|
|
// Adaptive quality: ingest quality reports from peer
|
|
if let Some(ref qr) = pkt.quality_report {
|
|
if let Some(new_profile) = quality_ctrl.observe(qr) {
|
|
let idx = profile_to_index(&new_profile);
|
|
info!(to = ?new_profile.codec, "auto: quality adapter recommends switch");
|
|
pending_profile_recv.store(idx, Ordering::Release);
|
|
}
|
|
}
|
|
|
|
match decoder.decode(&pkt.payload, &mut pcm) {
|
|
Ok(n) => {
|
|
last_decode_n = n;
|
|
decoded_frames += 1;
|
|
if !first_decode_logged {
|
|
info!(
|
|
t_ms = recv_t0.elapsed().as_millis(),
|
|
n,
|
|
codec = ?current_codec,
|
|
"first-join diag: recv first successful decode"
|
|
);
|
|
first_decode_logged = true;
|
|
}
|
|
// Log sample range for the first few decoded frames and periodically
|
|
if decoded_frames <= 3 || decoded_frames % 100 == 0 {
|
|
let slice = &pcm[..n];
|
|
let (mut lo, mut hi, mut sumsq) = (i16::MAX, i16::MIN, 0i64);
|
|
for &s in slice.iter() {
|
|
if s < lo { lo = s; }
|
|
if s > hi { hi = s; }
|
|
sumsq += (s as i64) * (s as i64);
|
|
}
|
|
let rms = (sumsq as f64 / n as f64).sqrt() as i32;
|
|
info!(
|
|
decoded_frames,
|
|
n,
|
|
sample_lo = lo,
|
|
sample_hi = hi,
|
|
rms,
|
|
codec = ?current_codec,
|
|
"recv: decoded PCM sample range"
|
|
);
|
|
}
|
|
agc.process_frame(&mut pcm[..n]);
|
|
|
|
// Dump to debug recorder before playout
|
|
// so we capture post-AGC samples that
|
|
// are exactly what we hand to Oboe.
|
|
if let Some(rec) = recorder.as_mut() {
|
|
if recorder_bytes < RECORDER_MAX_BYTES {
|
|
let slice = &pcm[..n];
|
|
// SAFETY: i16 is Plain Old Data;
|
|
// writing its little-endian bytes
|
|
// is well-defined on all targets
|
|
// we build for.
|
|
let byte_slice: &[u8] = unsafe {
|
|
std::slice::from_raw_parts(
|
|
slice.as_ptr() as *const u8,
|
|
slice.len() * 2,
|
|
)
|
|
};
|
|
let _ = rec.write_all(byte_slice);
|
|
recorder_bytes = recorder_bytes
|
|
.saturating_add(byte_slice.len() as u64);
|
|
if recorder_bytes >= RECORDER_MAX_BYTES {
|
|
let _ = rec.flush();
|
|
info!(recorder_bytes, "decoded-pcm recorder: stopped after limit");
|
|
}
|
|
}
|
|
}
|
|
|
|
if !recv_spk.load(Ordering::Relaxed) {
|
|
let w = crate::wzp_native::audio_write_playout(&pcm[..n]);
|
|
if !first_playout_write_logged {
|
|
info!(
|
|
t_ms = recv_t0.elapsed().as_millis(),
|
|
n,
|
|
w,
|
|
"first-join diag: recv first playout-ring write"
|
|
);
|
|
first_playout_write_logged = true;
|
|
}
|
|
last_written = w;
|
|
written_samples = written_samples.saturating_add(w as u64);
|
|
if w < n && decoded_frames <= 10 {
|
|
tracing::warn!(n, w, "recv: partial playout write (ring nearly full)");
|
|
}
|
|
} else if decoded_frames <= 3 || decoded_frames % 100 == 0 {
|
|
// User clicked spk-mute — log it so we don't chase ghost bugs
|
|
tracing::info!(decoded_frames, "recv: spk_muted=true, skipping playout write");
|
|
}
|
|
}
|
|
Err(e) => {
|
|
decode_errs += 1;
|
|
if decode_errs <= 3 {
|
|
tracing::warn!("decode error: {e}");
|
|
}
|
|
}
|
|
}
|
|
}
|
|
recv_fr.fetch_add(1, Ordering::Relaxed);
|
|
}
|
|
Ok(Ok(None)) => break,
|
|
Ok(Err(e)) => {
|
|
let msg = e.to_string();
|
|
if msg.contains("closed") || msg.contains("reset") {
|
|
error!("recv fatal: {e}");
|
|
break;
|
|
}
|
|
}
|
|
Err(_) => {}
|
|
}
|
|
|
|
// Heartbeat every 2s with decode+playout state
|
|
if heartbeat.elapsed() >= std::time::Duration::from_secs(2) {
|
|
let fr = recv_fr.load(Ordering::Relaxed);
|
|
if wzp_codec::dred_verbose_logs() {
|
|
info!(
|
|
recv_fr = fr,
|
|
decoded_frames,
|
|
last_decode_n,
|
|
last_written,
|
|
written_samples,
|
|
decode_errs,
|
|
codec = ?current_codec,
|
|
dred_recv = dred_recv.dred_reconstructions,
|
|
classical_plc = dred_recv.classical_plc_invocations,
|
|
dred_parses_with_data = dred_recv.parses_with_data,
|
|
dred_parses_total = dred_recv.parses_total,
|
|
"recv heartbeat (android)"
|
|
);
|
|
} else {
|
|
info!(
|
|
recv_fr = fr,
|
|
decoded_frames,
|
|
last_decode_n,
|
|
last_written,
|
|
written_samples,
|
|
decode_errs,
|
|
codec = ?current_codec,
|
|
"recv heartbeat (android)"
|
|
);
|
|
}
|
|
// Phase 5.6: compact GUI debug emit.
|
|
// recv_fr == 0 over time indicates inbound
|
|
// media is not reaching the client — either
|
|
// nothing is being sent by the peer, or the
|
|
// transport is dropping packets, or we're
|
|
// connected to the wrong side of the media
|
|
// path. Combined with the peer's send_heartbeat
|
|
// from the other log, this tells us exactly
|
|
// where 1-way audio breaks.
|
|
crate::emit_call_debug(
|
|
&recv_app,
|
|
"media:recv_heartbeat",
|
|
serde_json::json!({
|
|
"recv_fr": fr,
|
|
"decoded_frames": decoded_frames,
|
|
"last_written": last_written,
|
|
"written_samples": written_samples,
|
|
"decode_errs": decode_errs,
|
|
"codec": format!("{:?}", current_codec),
|
|
}),
|
|
);
|
|
|
|
// Media health watchdog: if recv_fr hasn't
|
|
// advanced in 3 consecutive heartbeats (6s) and
|
|
// we've been "connected" for at least 4s (give
|
|
// the first few frames time to arrive), emit a
|
|
// user-facing "media-degraded" event so the UI
|
|
// can show "No audio — connection may be lost".
|
|
if fr == last_recv_fr_for_watchdog {
|
|
no_recv_ticks += 1;
|
|
} else {
|
|
no_recv_ticks = 0;
|
|
if media_degraded_emitted {
|
|
// Was degraded but recovered — clear
|
|
// the banner.
|
|
media_degraded_emitted = false;
|
|
let _ = recv_app.emit(
|
|
"call-event",
|
|
serde_json::json!({
|
|
"kind": "media-recovered",
|
|
}),
|
|
);
|
|
crate::emit_call_debug(
|
|
&recv_app,
|
|
"media:recovered",
|
|
serde_json::json!({}),
|
|
);
|
|
}
|
|
}
|
|
last_recv_fr_for_watchdog = fr;
|
|
|
|
if no_recv_ticks >= 3 && !media_degraded_emitted {
|
|
media_degraded_emitted = true;
|
|
tracing::warn!(
|
|
recv_fr = fr,
|
|
no_recv_ticks,
|
|
"media watchdog: no inbound packets for 6s"
|
|
);
|
|
let _ = recv_app.emit(
|
|
"call-event",
|
|
serde_json::json!({
|
|
"kind": "media-degraded",
|
|
}),
|
|
);
|
|
crate::emit_call_debug(
|
|
&recv_app,
|
|
"media:no_recv_timeout",
|
|
serde_json::json!({
|
|
"recv_fr": fr,
|
|
"no_recv_ticks": no_recv_ticks,
|
|
}),
|
|
);
|
|
}
|
|
|
|
heartbeat = std::time::Instant::now();
|
|
}
|
|
}
|
|
});
|
|
|
|
// Signal task (presence + quality directives).
|
|
let sig_t = transport.clone();
|
|
let sig_r = running.clone();
|
|
let sig_p = participants.clone();
|
|
let sig_pending_profile = pending_profile.clone();
|
|
let event_cb = Arc::new(event_cb);
|
|
let sig_cb = event_cb.clone();
|
|
tokio::spawn(async move {
|
|
loop {
|
|
if !sig_r.load(Ordering::Relaxed) {
|
|
break;
|
|
}
|
|
match tokio::time::timeout(
|
|
std::time::Duration::from_millis(200),
|
|
sig_t.recv_signal(),
|
|
)
|
|
.await
|
|
{
|
|
Ok(Ok(Some(wzp_proto::SignalMessage::RoomUpdate {
|
|
participants: parts,
|
|
..
|
|
}))) => {
|
|
let mut seen = std::collections::HashSet::new();
|
|
let unique: Vec<ParticipantInfo> = parts
|
|
.into_iter()
|
|
.filter(|p| seen.insert((p.fingerprint.clone(), p.alias.clone())))
|
|
.map(|p| ParticipantInfo {
|
|
fingerprint: p.fingerprint,
|
|
alias: p.alias,
|
|
relay_label: p.relay_label,
|
|
})
|
|
.collect();
|
|
let count = unique.len();
|
|
*sig_p.lock().await = unique;
|
|
sig_cb("room-update", &format!("{count} participants"));
|
|
}
|
|
Ok(Ok(Some(wzp_proto::SignalMessage::QualityDirective {
|
|
recommended_profile,
|
|
reason,
|
|
}))) => {
|
|
let idx = profile_to_index(&recommended_profile);
|
|
info!(
|
|
codec = ?recommended_profile.codec,
|
|
reason = reason.as_deref().unwrap_or(""),
|
|
"relay quality directive: switching profile"
|
|
);
|
|
sig_pending_profile.store(idx, Ordering::Release);
|
|
}
|
|
Ok(Ok(Some(_))) => {}
|
|
Ok(Ok(None)) => break,
|
|
Ok(Err(_)) => break,
|
|
Err(_) => {}
|
|
}
|
|
}
|
|
});
|
|
|
|
Ok(Self {
|
|
running,
|
|
mic_muted,
|
|
spk_muted,
|
|
participants,
|
|
frames_sent,
|
|
frames_received,
|
|
audio_level,
|
|
transport,
|
|
start_time: Instant::now(),
|
|
fingerprint,
|
|
tx_codec,
|
|
rx_codec,
|
|
// No CPAL / VPIO handle to keep alive on Android — wzp_native
|
|
// is a static dlopen'd library, the audio streams live inside
|
|
// the standalone cdylib's process-global singleton.
|
|
_audio_handle: SyncWrapper(Box::new(())),
|
|
})
|
|
}
|
|
|
|
#[cfg(not(target_os = "android"))]
|
|
pub async fn start<F>(
|
|
relay: String,
|
|
room: String,
|
|
alias: String,
|
|
_os_aec: bool,
|
|
quality: String,
|
|
reuse_endpoint: Option<wzp_transport::Endpoint>,
|
|
// Phase 3.5: caller did the dual-path race and picked a
|
|
// winning transport. If Some, skip our own connect step.
|
|
pre_connected_transport: Option<Arc<wzp_transport::QuinnTransport>>,
|
|
// Phase 6: explicit is_direct_p2p flag (see android branch).
|
|
is_direct_p2p: bool,
|
|
_app: tauri::AppHandle,
|
|
event_cb: F,
|
|
) -> Result<Self, anyhow::Error>
|
|
where
|
|
F: Fn(&str, &str) + Send + Sync + 'static,
|
|
{
|
|
info!(
|
|
%relay, %room, %alias, %quality,
|
|
has_reuse = reuse_endpoint.is_some(),
|
|
has_pre_connected = pre_connected_transport.is_some(),
|
|
is_direct_p2p,
|
|
"CallEngine::start (desktop) invoked"
|
|
);
|
|
let _ = rustls::crypto::ring::default_provider().install_default();
|
|
|
|
let relay_addr: SocketAddr = relay.parse()?;
|
|
|
|
let seed = crate::load_or_create_seed()
|
|
.map_err(|e| anyhow::anyhow!("identity: {e}"))?;
|
|
let fp = seed.derive_identity().public_identity().fingerprint;
|
|
let fingerprint = fp.to_string();
|
|
info!(%fp, "identity loaded");
|
|
|
|
// Transport source: either pre-connected or fresh.
|
|
let transport = if let Some(t) = pre_connected_transport {
|
|
info!(
|
|
is_direct_p2p,
|
|
remote = %t.remote_address(),
|
|
max_datagram = ?t.max_datagram_size(),
|
|
"using pre-connected transport"
|
|
);
|
|
t
|
|
} else {
|
|
// Connect — reuse the signal endpoint if the direct-call path gave
|
|
// us one, otherwise create a fresh one (SFU room join path).
|
|
let endpoint = if let Some(ep) = reuse_endpoint {
|
|
info!(local_addr = ?ep.local_addr().ok(), "reusing signal endpoint for media connection");
|
|
ep
|
|
} else {
|
|
let bind_addr: SocketAddr = "0.0.0.0:0".parse().unwrap();
|
|
let ep = wzp_transport::create_endpoint(bind_addr, None)
|
|
.map_err(|e| { error!("create_endpoint failed: {e}"); e })?;
|
|
info!(local_addr = ?ep.local_addr().ok(), "created new endpoint, dialing relay");
|
|
ep
|
|
};
|
|
let client_config = wzp_transport::client_config();
|
|
let conn = wzp_transport::connect(&endpoint, relay_addr, &room, client_config)
|
|
.await
|
|
.map_err(|e| { error!("connect failed: {e}"); e })?;
|
|
info!("QUIC connection established, performing handshake");
|
|
Arc::new(wzp_transport::QuinnTransport::new(conn))
|
|
};
|
|
|
|
// Handshake — relay-specific. Direct P2P connections skip
|
|
// this because the peer is a phone, not a relay with an
|
|
// accept_handshake handler. See the android branch's
|
|
// comment for the full rationale.
|
|
if !is_direct_p2p {
|
|
let _session = wzp_client::handshake::perform_handshake(
|
|
&*transport,
|
|
&seed.0,
|
|
Some(&alias),
|
|
)
|
|
.await
|
|
.map_err(|e| { error!("perform_handshake failed: {e}"); e })?;
|
|
} else {
|
|
info!("direct P2P — skipping relay handshake (QUIC TLS is the encryption layer)");
|
|
}
|
|
|
|
info!("connected to relay, handshake complete");
|
|
event_cb("connected", &format!("joined room {room}"));
|
|
|
|
// Audio I/O — VPIO (OS AEC) on macOS, plain CPAL otherwise.
|
|
// The audio handle must be stored in CallEngine to keep streams alive.
|
|
let (capture_ring, playout_ring, audio_handle): (_, _, Box<dyn std::any::Any + Send>) =
|
|
if _os_aec {
|
|
#[cfg(target_os = "macos")]
|
|
{
|
|
match wzp_client::audio_vpio::VpioAudio::start() {
|
|
Ok(v) => {
|
|
let cr = v.capture_ring().clone();
|
|
let pr = v.playout_ring().clone();
|
|
info!("using VoiceProcessingIO (OS AEC)");
|
|
(cr, pr, Box::new(v))
|
|
}
|
|
Err(e) => {
|
|
info!("VPIO failed ({e}), falling back to CPAL");
|
|
let capture = AudioCapture::start()?;
|
|
let playback = AudioPlayback::start()?;
|
|
let cr = capture.ring().clone();
|
|
let pr = playback.ring().clone();
|
|
(cr, pr, Box::new((capture, playback)))
|
|
}
|
|
}
|
|
}
|
|
#[cfg(not(target_os = "macos"))]
|
|
{
|
|
info!("OS AEC not available on this platform, using CPAL");
|
|
let capture = AudioCapture::start()?;
|
|
let playback = AudioPlayback::start()?;
|
|
let cr = capture.ring().clone();
|
|
let pr = playback.ring().clone();
|
|
(cr, pr, Box::new((capture, playback)))
|
|
}
|
|
} else {
|
|
let capture = AudioCapture::start()?;
|
|
let playback = AudioPlayback::start()?;
|
|
let cr = capture.ring().clone();
|
|
let pr = playback.ring().clone();
|
|
(cr, pr, Box::new((capture, playback)))
|
|
};
|
|
|
|
let running = Arc::new(AtomicBool::new(true));
|
|
let mic_muted = Arc::new(AtomicBool::new(false));
|
|
let spk_muted = Arc::new(AtomicBool::new(false));
|
|
let participants: Arc<Mutex<Vec<ParticipantInfo>>> = Arc::new(Mutex::new(vec![]));
|
|
let frames_sent = Arc::new(AtomicU64::new(0));
|
|
let frames_received = Arc::new(AtomicU64::new(0));
|
|
let audio_level = Arc::new(AtomicU32::new(0));
|
|
let tx_codec = Arc::new(Mutex::new(String::new()));
|
|
let rx_codec = Arc::new(Mutex::new(String::new()));
|
|
|
|
// Adaptive quality: shared pending-profile bridge between recv → send.
|
|
let pending_profile = Arc::new(AtomicU8::new(PROFILE_NO_CHANGE));
|
|
let auto_profile = resolve_quality(&quality).is_none();
|
|
|
|
// Send task
|
|
let send_t = transport.clone();
|
|
let send_r = running.clone();
|
|
let send_mic = mic_muted.clone();
|
|
let send_fs = frames_sent.clone();
|
|
let send_level = audio_level.clone();
|
|
let send_drops = Arc::new(AtomicU64::new(0));
|
|
let send_quality = quality.clone();
|
|
let send_tx_codec = tx_codec.clone();
|
|
let send_pending_profile = pending_profile.clone();
|
|
tokio::spawn(async move {
|
|
let profile = resolve_quality(&send_quality);
|
|
let config = match profile {
|
|
Some(p) => CallConfig {
|
|
noise_suppression: false,
|
|
suppression_enabled: false,
|
|
..CallConfig::from_profile(p)
|
|
},
|
|
None => CallConfig {
|
|
noise_suppression: false,
|
|
suppression_enabled: false,
|
|
..CallConfig::default()
|
|
},
|
|
};
|
|
let mut frame_samples = (config.profile.frame_duration_ms as usize) * 48;
|
|
info!(codec = ?config.profile.codec, frame_samples, "send task starting");
|
|
*send_tx_codec.lock().await = format!("{:?}", config.profile.codec);
|
|
let mut encoder = CallEncoder::new(&config);
|
|
encoder.set_aec_enabled(false); // OS AEC or none
|
|
let mut buf = vec![0i16; 1920]; // max frame (40ms)
|
|
|
|
// Continuous DRED tuning (same as Android send task).
|
|
let mut dred_tuner = wzp_proto::DredTuner::new(config.profile.codec);
|
|
let mut frames_since_dred_poll: u32 = 0;
|
|
const DRED_POLL_INTERVAL: u32 = 25;
|
|
|
|
loop {
|
|
if !send_r.load(Ordering::Relaxed) {
|
|
break;
|
|
}
|
|
if capture_ring.available() < frame_samples {
|
|
tokio::time::sleep(std::time::Duration::from_millis(5)).await;
|
|
continue;
|
|
}
|
|
capture_ring.read(&mut buf[..frame_samples]);
|
|
|
|
// Compute RMS audio level for UI meter
|
|
{
|
|
let pcm = &buf[..frame_samples];
|
|
let sum_sq: f64 = pcm.iter().map(|&s| (s as f64) * (s as f64)).sum();
|
|
let rms = (sum_sq / pcm.len() as f64).sqrt() as u32;
|
|
send_level.store(rms, Ordering::Relaxed);
|
|
}
|
|
|
|
if send_mic.load(Ordering::Relaxed) {
|
|
buf[..frame_samples].fill(0);
|
|
}
|
|
match encoder.encode_frame(&buf[..frame_samples]) {
|
|
Ok(pkts) => {
|
|
for pkt in &pkts {
|
|
if let Err(e) = send_t.send_media(pkt).await {
|
|
// Transient congestion (Blocked) — drop packet, keep going
|
|
send_drops.fetch_add(1, Ordering::Relaxed);
|
|
if send_drops.load(Ordering::Relaxed) <= 3 {
|
|
tracing::warn!("send_media error (dropping packet): {e}");
|
|
}
|
|
}
|
|
}
|
|
send_fs.fetch_add(1, Ordering::Relaxed);
|
|
}
|
|
Err(e) => error!("encode: {e}"),
|
|
}
|
|
|
|
// Adaptive quality: check if recv task recommended a profile switch.
|
|
if auto_profile {
|
|
let p = send_pending_profile.swap(PROFILE_NO_CHANGE, Ordering::Acquire);
|
|
if p != PROFILE_NO_CHANGE {
|
|
if let Some(new_profile) = index_to_profile(p) {
|
|
let new_fs = (new_profile.frame_duration_ms as usize) * 48;
|
|
info!(to = ?new_profile.codec, frame_samples = new_fs, "auto: switching encoder profile (desktop)");
|
|
if encoder.set_profile(new_profile).is_ok() {
|
|
frame_samples = new_fs;
|
|
dred_tuner.set_codec(new_profile.codec);
|
|
*send_tx_codec.lock().await = format!("{:?}", new_profile.codec);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// DRED tuner: poll quinn path stats periodically.
|
|
frames_since_dred_poll += 1;
|
|
if frames_since_dred_poll >= DRED_POLL_INTERVAL {
|
|
frames_since_dred_poll = 0;
|
|
let snap = send_t.quinn_path_stats();
|
|
let pq = send_t.path_quality();
|
|
if let Some(tuning) = dred_tuner.update(
|
|
snap.loss_pct,
|
|
snap.rtt_ms,
|
|
pq.jitter_ms,
|
|
) {
|
|
encoder.apply_dred_tuning(tuning);
|
|
}
|
|
}
|
|
}
|
|
});
|
|
|
|
// Recv task (direct playout with auto codec switch)
|
|
let recv_t = transport.clone();
|
|
let recv_r = running.clone();
|
|
let recv_spk = spk_muted.clone();
|
|
let recv_fr = frames_received.clone();
|
|
let recv_rx_codec = rx_codec.clone();
|
|
let pending_profile_recv = pending_profile.clone();
|
|
tokio::spawn(async move {
|
|
let initial_profile = resolve_quality(&quality).unwrap_or(QualityProfile::GOOD);
|
|
// Phase 3b/3c: concrete AdaptiveDecoder (not Box<dyn>) so we
|
|
// can call reconstruct_from_dred. Same reasoning as the
|
|
// Android recv path above.
|
|
let mut decoder = wzp_codec::AdaptiveDecoder::new(initial_profile)
|
|
.expect("failed to create adaptive decoder");
|
|
let mut current_profile = initial_profile;
|
|
let mut current_codec = initial_profile.codec;
|
|
let mut agc = wzp_codec::AutoGainControl::new();
|
|
let mut pcm = vec![0i16; FRAME_SAMPLES_40MS]; // big enough for any codec
|
|
let mut dred_recv = DredRecvState::new();
|
|
let mut quality_ctrl = AdaptiveQualityController::new();
|
|
|
|
loop {
|
|
if !recv_r.load(Ordering::Relaxed) {
|
|
break;
|
|
}
|
|
match tokio::time::timeout(
|
|
std::time::Duration::from_millis(100),
|
|
recv_t.recv_media(),
|
|
)
|
|
.await
|
|
{
|
|
Ok(Ok(Some(pkt))) => {
|
|
if !pkt.header.is_repair && pkt.header.codec_id != CodecId::ComfortNoise {
|
|
// Track RX codec
|
|
{
|
|
let mut rx = recv_rx_codec.lock().await;
|
|
let codec_name = format!("{:?}", pkt.header.codec_id);
|
|
if *rx != codec_name { *rx = codec_name; }
|
|
}
|
|
// Auto-switch decoder if incoming codec differs
|
|
if pkt.header.codec_id != current_codec {
|
|
let new_profile = match pkt.header.codec_id {
|
|
CodecId::Opus24k => QualityProfile::GOOD,
|
|
CodecId::Opus6k => QualityProfile::DEGRADED,
|
|
CodecId::Opus32k => QualityProfile::STUDIO_32K,
|
|
CodecId::Opus48k => QualityProfile::STUDIO_48K,
|
|
CodecId::Opus64k => QualityProfile::STUDIO_64K,
|
|
CodecId::Codec2_1200 => QualityProfile::CATASTROPHIC,
|
|
CodecId::Codec2_3200 => QualityProfile {
|
|
codec: CodecId::Codec2_3200,
|
|
fec_ratio: 0.5, frame_duration_ms: 20, frames_per_block: 5,
|
|
},
|
|
other => QualityProfile { codec: other, ..QualityProfile::GOOD },
|
|
};
|
|
info!(from = ?current_codec, to = ?pkt.header.codec_id, "recv: switching decoder");
|
|
let _ = decoder.set_profile(new_profile);
|
|
current_profile = new_profile;
|
|
current_codec = pkt.header.codec_id;
|
|
dred_recv.reset_on_profile_switch();
|
|
}
|
|
|
|
// Phase 3b/3c: parse DRED + fill gaps before
|
|
// decoding the current packet. See the Android
|
|
// start() recv task for full commentary.
|
|
if pkt.header.codec_id.is_opus() {
|
|
dred_recv.ingest_opus(pkt.header.seq, &pkt.payload);
|
|
let frame_samples_now = (48_000
|
|
* current_profile.frame_duration_ms as usize)
|
|
/ 1000;
|
|
let spk_muted_flag = recv_spk.load(Ordering::Relaxed);
|
|
dred_recv.fill_gap_to(
|
|
&mut decoder,
|
|
pkt.header.seq,
|
|
frame_samples_now,
|
|
&mut pcm,
|
|
|samples| {
|
|
agc.process_frame(samples);
|
|
if !spk_muted_flag {
|
|
playout_ring.write(samples);
|
|
}
|
|
},
|
|
);
|
|
}
|
|
|
|
// Adaptive quality: ingest quality reports from peer
|
|
if let Some(ref qr) = pkt.quality_report {
|
|
if let Some(new_profile) = quality_ctrl.observe(qr) {
|
|
let idx = profile_to_index(&new_profile);
|
|
info!(to = ?new_profile.codec, "auto: quality adapter recommends switch");
|
|
pending_profile_recv.store(idx, Ordering::Release);
|
|
}
|
|
}
|
|
|
|
if let Ok(n) = decoder.decode(&pkt.payload, &mut pcm) {
|
|
agc.process_frame(&mut pcm[..n]);
|
|
if !recv_spk.load(Ordering::Relaxed) {
|
|
playout_ring.write(&pcm[..n]);
|
|
}
|
|
}
|
|
}
|
|
recv_fr.fetch_add(1, Ordering::Relaxed);
|
|
}
|
|
Ok(Ok(None)) => break,
|
|
Ok(Err(e)) => {
|
|
let msg = e.to_string();
|
|
if msg.contains("closed") || msg.contains("reset") {
|
|
error!("recv fatal: {e}");
|
|
break;
|
|
}
|
|
}
|
|
Err(_) => {}
|
|
}
|
|
}
|
|
});
|
|
|
|
// Signal task (presence + quality directives)
|
|
let sig_t = transport.clone();
|
|
let sig_r = running.clone();
|
|
let sig_p = participants.clone();
|
|
let sig_pending_profile = pending_profile.clone();
|
|
let event_cb = Arc::new(event_cb);
|
|
let sig_cb = event_cb.clone();
|
|
tokio::spawn(async move {
|
|
loop {
|
|
if !sig_r.load(Ordering::Relaxed) {
|
|
break;
|
|
}
|
|
match tokio::time::timeout(
|
|
std::time::Duration::from_millis(200),
|
|
sig_t.recv_signal(),
|
|
)
|
|
.await
|
|
{
|
|
Ok(Ok(Some(wzp_proto::SignalMessage::RoomUpdate {
|
|
participants: parts,
|
|
..
|
|
}))) => {
|
|
let mut seen = std::collections::HashSet::new();
|
|
let unique: Vec<ParticipantInfo> = parts
|
|
.into_iter()
|
|
.filter(|p| seen.insert((p.fingerprint.clone(), p.alias.clone())))
|
|
.map(|p| ParticipantInfo {
|
|
fingerprint: p.fingerprint,
|
|
alias: p.alias,
|
|
relay_label: p.relay_label,
|
|
})
|
|
.collect();
|
|
let count = unique.len();
|
|
*sig_p.lock().await = unique;
|
|
sig_cb("room-update", &format!("{count} participants"));
|
|
}
|
|
Ok(Ok(Some(wzp_proto::SignalMessage::QualityDirective {
|
|
recommended_profile,
|
|
reason,
|
|
}))) => {
|
|
let idx = profile_to_index(&recommended_profile);
|
|
info!(
|
|
codec = ?recommended_profile.codec,
|
|
reason = reason.as_deref().unwrap_or(""),
|
|
"relay quality directive: switching profile"
|
|
);
|
|
sig_pending_profile.store(idx, Ordering::Release);
|
|
}
|
|
Ok(Ok(Some(_))) => {}
|
|
Ok(Ok(None)) => break,
|
|
Ok(Err(_)) => break,
|
|
Err(_) => {}
|
|
}
|
|
}
|
|
});
|
|
|
|
Ok(Self {
|
|
running,
|
|
mic_muted,
|
|
spk_muted,
|
|
participants,
|
|
frames_sent,
|
|
frames_received,
|
|
audio_level,
|
|
transport,
|
|
start_time: Instant::now(),
|
|
fingerprint,
|
|
tx_codec,
|
|
rx_codec,
|
|
_audio_handle: SyncWrapper(audio_handle),
|
|
})
|
|
}
|
|
|
|
pub fn toggle_mic(&self) -> bool {
|
|
let was = self.mic_muted.load(Ordering::Relaxed);
|
|
self.mic_muted.store(!was, Ordering::Relaxed);
|
|
!was
|
|
}
|
|
|
|
pub fn toggle_speaker(&self) -> bool {
|
|
let was = self.spk_muted.load(Ordering::Relaxed);
|
|
self.spk_muted.store(!was, Ordering::Relaxed);
|
|
!was
|
|
}
|
|
|
|
pub async fn status(&self) -> EngineStatus {
|
|
let participants = {
|
|
let parts = self.participants.lock().await;
|
|
parts
|
|
.iter()
|
|
.map(|p| ParticipantInfo {
|
|
fingerprint: p.fingerprint.clone(),
|
|
alias: p.alias.clone(),
|
|
relay_label: p.relay_label.clone(),
|
|
})
|
|
.collect()
|
|
}; // lock dropped here
|
|
EngineStatus {
|
|
mic_muted: self.mic_muted.load(Ordering::Relaxed),
|
|
spk_muted: self.spk_muted.load(Ordering::Relaxed),
|
|
participants,
|
|
frames_sent: self.frames_sent.load(Ordering::Relaxed),
|
|
frames_received: self.frames_received.load(Ordering::Relaxed),
|
|
audio_level: self.audio_level.load(Ordering::Relaxed),
|
|
call_duration_secs: self.start_time.elapsed().as_secs_f64(),
|
|
fingerprint: self.fingerprint.clone(),
|
|
tx_codec: self.tx_codec.lock().await.clone(),
|
|
rx_codec: self.rx_codec.lock().await.clone(),
|
|
}
|
|
}
|
|
|
|
pub async fn stop(self) {
|
|
self.running.store(false, Ordering::SeqCst);
|
|
self.transport.close().await.ok();
|
|
// On Android, the Oboe capture/playout streams live inside the
|
|
// wzp-native cdylib as a process-global singleton. Explicitly stop
|
|
// them here so the mic + speaker are released between calls, matching
|
|
// the desktop behaviour where dropping _audio_handle tears down CPAL.
|
|
#[cfg(target_os = "android")]
|
|
{
|
|
crate::wzp_native::audio_stop();
|
|
// Release the BT SCO communication device so Android can
|
|
// route media (video, music) back to BT A2DP. Without this,
|
|
// setCommunicationDevice locks BT to SCO mode and other apps
|
|
// can't use the headset for media playback until reboot.
|
|
if let Err(e) = crate::android_audio::stop_bluetooth_sco() {
|
|
tracing::warn!("stop_bluetooth_sco on call end failed: {e}");
|
|
}
|
|
// Restore MODE_NORMAL so other apps' audio routes normally.
|
|
if let Err(e) = crate::android_audio::set_audio_mode_normal() {
|
|
tracing::warn!("set_audio_mode_normal failed: {e}");
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
impl Drop for CallEngine {
|
|
fn drop(&mut self) {
|
|
// Safety net: if stop() was never called (crash, app
|
|
// backgrounding), signal tasks to exit so they don't
|
|
// spin on a dropped transport.
|
|
self.running.store(false, Ordering::SeqCst);
|
|
}
|
|
}
|