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Microphone uplink (client → host's virtual mic, 0xCB) and a cleaner connect screen.
Mic (Rust-heavy, mirrors the audio playback path in reverse):
- crates/punktfunk-android/src/mic.rs: AAudio LowLatency **input** → realtime callback hands
captured f32 to a channel → a worker thread Opus-encodes 20 ms stereo frames (48 kHz, VOIP,
64 kbps) and calls NativeClient::send_mic. MicCapture owns the stream + encode thread (RAII stop).
- session.rs: SessionHandle gains a `mic` slot; nativeStartMic/nativeStopMic JNI (mirror of audio);
stopped in Drop. NativeBridge: the two externs.
- Settings: a `micEnabled` flag + a Microphone toggle in SettingsScreen that requests RECORD_AUDIO
(denied → stays off). StreamScreen starts the mic only if enabled AND the permission is held.
Connect-screen redesign:
- One scrollable Column (was a fixed centered layout that could clip with the new tab bar);
host rows render via forEach (no nested LazyColumn). Colored section labels ("Saved hosts",
"Discovered on the network", "Connect manually"), full-width host cards / fields / Connect button,
a header + subtitle, and a muted footer.
Verified live (emulator pf_phone -> home-worker-2): toggling mic requests RECORD_AUDIO; with it
granted, a session sends mic frames (client "mic: sent=250 … peak=0.439" — real audio) and the host
logs "client datagram stream ended … mic=276". Redesigned screen confirmed via screenshots.
Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
175 lines
7.0 KiB
Rust
175 lines
7.0 KiB
Rust
//! Android microphone uplink (android-only): capture mic PCM via AAudio (LowLatency **input**),
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//! Opus-encode 20 ms stereo frames, and push them to the host over the connector's mic plane
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//! (`send_mic` → 0xCB datagram). The mirror of [`crate::audio`] in reverse: AAudio's realtime input
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//! callback hands captured interleaved f32 to a channel; a worker thread we own does the Opus encode
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//! + send (encoding is too heavy for the realtime callback, exactly as decode is on the playback
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//! side). Format matches the host decoder + the Linux client: 48 kHz **stereo**, 20 ms, Opus VOIP.
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use ndk::audio::{
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AudioCallbackResult, AudioDirection, AudioFormat, AudioPerformanceMode, AudioSharingMode,
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AudioStream, AudioStreamBuilder,
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};
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use punktfunk_core::client::NativeClient;
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use std::collections::VecDeque;
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use std::ffi::c_void;
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use std::sync::atomic::{AtomicBool, AtomicU64, Ordering};
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use std::sync::mpsc::{sync_channel, Receiver, RecvTimeoutError, TrySendError};
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use std::sync::Arc;
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use std::time::{Duration, SystemTime, UNIX_EPOCH};
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const CHANNELS: usize = 2;
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const SAMPLE_RATE: i32 = 48_000;
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/// 20 ms per channel @ 48 kHz — the Linux client's frame; the host accepts ≤ 120 ms.
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const FRAME_SAMPLES: usize = 960;
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/// Captured-chunk hand-off depth (each ~ one burst); drops on overflow (best-effort uplink).
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const RING_CHUNKS: usize = 64;
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/// Opus VOIP target bitrate (speech; tunable).
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const MIC_BITRATE: i32 = 64_000;
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/// Owned by [`crate::session::SessionHandle`]: the live AAudio input stream + the encode thread.
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pub struct MicCapture {
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_stream: AudioStream, // dropping it stops + closes the AAudio input stream
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shutdown: Arc<AtomicBool>,
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join: Option<std::thread::JoinHandle<()>>,
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}
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impl MicCapture {
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/// Open AAudio (LowLatency, 48 kHz/stereo/f32) for **input** with a realtime callback that
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/// forwards captured PCM to a channel, then spawn the Opus encode + uplink thread. `None` on
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/// failure (the caller leaves the rest of the session streaming).
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pub fn start(client: Arc<NativeClient>) -> Option<MicCapture> {
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let (tx, rx) = sync_channel::<Vec<f32>>(RING_CHUNKS);
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let captured = Arc::new(AtomicU64::new(0));
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let cb_captured = captured.clone();
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let callback = move |_s: &AudioStream, data: *mut c_void, num_frames: i32| {
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let n = num_frames as usize * CHANNELS;
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// SAFETY: for an input stream AAudio provides `num_frames * channel_count` captured F32
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// samples at `data` (read-only for us).
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let inp = unsafe { std::slice::from_raw_parts(data as *const f32, n) };
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match tx.try_send(inp.to_vec()) {
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Ok(()) | Err(TrySendError::Full(_)) => {} // drop-newest if the encoder lags
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Err(TrySendError::Disconnected(_)) => return AudioCallbackResult::Stop,
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}
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cb_captured.fetch_add(num_frames as u64, Ordering::Relaxed);
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AudioCallbackResult::Continue
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};
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let stream = AudioStreamBuilder::new()
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.map_err(|e| log::error!("mic: AudioStreamBuilder::new: {e}"))
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.ok()?
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.direction(AudioDirection::Input)
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.sample_rate(SAMPLE_RATE)
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.channel_count(CHANNELS as i32)
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.format(AudioFormat::PCM_Float)
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.performance_mode(AudioPerformanceMode::LowLatency)
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.sharing_mode(AudioSharingMode::Shared)
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.data_callback(Box::new(callback))
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.error_callback(Box::new(|_s, e| {
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log::warn!("mic: AAudio error (device reroute/disconnect?): {e:?}");
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}))
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.open_stream()
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.map_err(|e| log::error!("mic: open_stream (RECORD_AUDIO granted?): {e}"))
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.ok()?;
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if let Err(e) = stream.request_start() {
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log::error!("mic: request_start: {e}");
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return None;
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}
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log::info!(
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"mic: AAudio input started rate={} ch={} fmt={:?}",
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stream.sample_rate(),
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stream.channel_count(),
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stream.format(),
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);
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let shutdown = Arc::new(AtomicBool::new(false));
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let sd = shutdown.clone();
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let join = std::thread::Builder::new()
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.name("pf-mic".into())
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.spawn(move || encode_loop(client, rx, sd, captured))
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.ok();
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Some(MicCapture {
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_stream: stream,
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shutdown,
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join,
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})
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}
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}
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impl Drop for MicCapture {
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fn drop(&mut self) {
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self.shutdown.store(true, Ordering::SeqCst);
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if let Some(j) = self.join.take() {
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let _ = j.join();
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}
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// `_stream` drops here → AAudio request_stop + close.
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}
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}
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/// Consumer: drain captured f32 → accumulate → Opus `encode_float` 20 ms stereo frames → `send_mic`.
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fn encode_loop(
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client: Arc<NativeClient>,
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rx: Receiver<Vec<f32>>,
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shutdown: Arc<AtomicBool>,
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captured: Arc<AtomicU64>,
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) {
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let mut enc = match opus::Encoder::new(
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SAMPLE_RATE as u32,
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opus::Channels::Stereo,
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opus::Application::Voip,
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) {
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Ok(e) => e,
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Err(e) => {
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log::error!("mic: opus encoder init: {e} — mic disabled");
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return;
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}
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};
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let _ = enc.set_bitrate(opus::Bitrate::Bits(MIC_BITRATE));
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let frame = FRAME_SAMPLES * CHANNELS;
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let mut ring: VecDeque<f32> = VecDeque::with_capacity(frame * 4);
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let mut out = vec![0u8; 4000]; // max Opus packet for a 20 ms frame fits easily
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let mut seq: u32 = 0;
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let mut sent: u64 = 0;
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let mut peak = 0f32; // loudest |sample| since the last log — tells speech from silence
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while !shutdown.load(Ordering::Relaxed) {
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match rx.recv_timeout(Duration::from_millis(100)) {
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Ok(chunk) => ring.extend(chunk),
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Err(RecvTimeoutError::Timeout) => continue, // wake to re-check shutdown
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Err(RecvTimeoutError::Disconnected) => break,
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}
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while ring.len() >= frame {
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let pcm: Vec<f32> = ring.drain(..frame).collect();
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for &s in &pcm {
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peak = peak.max(s.abs());
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}
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match enc.encode_float(&pcm, &mut out) {
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Ok(len) => {
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let pts = SystemTime::now()
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.duration_since(UNIX_EPOCH)
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.map(|d| d.as_nanos() as u64)
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.unwrap_or(0);
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let _ = client.send_mic(seq, pts, out[..len].to_vec());
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seq = seq.wrapping_add(1);
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sent += 1;
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if sent % 250 == 0 {
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log::info!(
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"mic: sent={sent} captured_frames={} peak={peak:.3}",
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captured.load(Ordering::Relaxed),
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);
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peak = 0.0;
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}
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}
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Err(e) => log::debug!("mic: opus encode: {e}"),
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}
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}
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}
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log::info!(
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"mic: stopped (sent={sent} captured_frames={})",
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captured.load(Ordering::Relaxed),
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);
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}
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