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punktfunk/crates/punktfunk-android/src/mic.rs
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feat(android): mic uplink + connect-screen redesign
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>
2026-06-15 17:05:25 +02:00

175 lines
7.0 KiB
Rust

//! Android microphone uplink (android-only): capture mic PCM via AAudio (LowLatency **input**),
//! Opus-encode 20 ms stereo frames, and push them to the host over the connector's mic plane
//! (`send_mic` → 0xCB datagram). The mirror of [`crate::audio`] in reverse: AAudio's realtime input
//! callback hands captured interleaved f32 to a channel; a worker thread we own does the Opus encode
//! + send (encoding is too heavy for the realtime callback, exactly as decode is on the playback
//! side). Format matches the host decoder + the Linux client: 48 kHz **stereo**, 20 ms, Opus VOIP.
use ndk::audio::{
AudioCallbackResult, AudioDirection, AudioFormat, AudioPerformanceMode, AudioSharingMode,
AudioStream, AudioStreamBuilder,
};
use punktfunk_core::client::NativeClient;
use std::collections::VecDeque;
use std::ffi::c_void;
use std::sync::atomic::{AtomicBool, AtomicU64, Ordering};
use std::sync::mpsc::{sync_channel, Receiver, RecvTimeoutError, TrySendError};
use std::sync::Arc;
use std::time::{Duration, SystemTime, UNIX_EPOCH};
const CHANNELS: usize = 2;
const SAMPLE_RATE: i32 = 48_000;
/// 20 ms per channel @ 48 kHz — the Linux client's frame; the host accepts ≤ 120 ms.
const FRAME_SAMPLES: usize = 960;
/// Captured-chunk hand-off depth (each ~ one burst); drops on overflow (best-effort uplink).
const RING_CHUNKS: usize = 64;
/// Opus VOIP target bitrate (speech; tunable).
const MIC_BITRATE: i32 = 64_000;
/// Owned by [`crate::session::SessionHandle`]: the live AAudio input stream + the encode thread.
pub struct MicCapture {
_stream: AudioStream, // dropping it stops + closes the AAudio input stream
shutdown: Arc<AtomicBool>,
join: Option<std::thread::JoinHandle<()>>,
}
impl MicCapture {
/// Open AAudio (LowLatency, 48 kHz/stereo/f32) for **input** with a realtime callback that
/// forwards captured PCM to a channel, then spawn the Opus encode + uplink thread. `None` on
/// failure (the caller leaves the rest of the session streaming).
pub fn start(client: Arc<NativeClient>) -> Option<MicCapture> {
let (tx, rx) = sync_channel::<Vec<f32>>(RING_CHUNKS);
let captured = Arc::new(AtomicU64::new(0));
let cb_captured = captured.clone();
let callback = move |_s: &AudioStream, data: *mut c_void, num_frames: i32| {
let n = num_frames as usize * CHANNELS;
// SAFETY: for an input stream AAudio provides `num_frames * channel_count` captured F32
// samples at `data` (read-only for us).
let inp = unsafe { std::slice::from_raw_parts(data as *const f32, n) };
match tx.try_send(inp.to_vec()) {
Ok(()) | Err(TrySendError::Full(_)) => {} // drop-newest if the encoder lags
Err(TrySendError::Disconnected(_)) => return AudioCallbackResult::Stop,
}
cb_captured.fetch_add(num_frames as u64, Ordering::Relaxed);
AudioCallbackResult::Continue
};
let stream = AudioStreamBuilder::new()
.map_err(|e| log::error!("mic: AudioStreamBuilder::new: {e}"))
.ok()?
.direction(AudioDirection::Input)
.sample_rate(SAMPLE_RATE)
.channel_count(CHANNELS as i32)
.format(AudioFormat::PCM_Float)
.performance_mode(AudioPerformanceMode::LowLatency)
.sharing_mode(AudioSharingMode::Shared)
.data_callback(Box::new(callback))
.error_callback(Box::new(|_s, e| {
log::warn!("mic: AAudio error (device reroute/disconnect?): {e:?}");
}))
.open_stream()
.map_err(|e| log::error!("mic: open_stream (RECORD_AUDIO granted?): {e}"))
.ok()?;
if let Err(e) = stream.request_start() {
log::error!("mic: request_start: {e}");
return None;
}
log::info!(
"mic: AAudio input started rate={} ch={} fmt={:?}",
stream.sample_rate(),
stream.channel_count(),
stream.format(),
);
let shutdown = Arc::new(AtomicBool::new(false));
let sd = shutdown.clone();
let join = std::thread::Builder::new()
.name("pf-mic".into())
.spawn(move || encode_loop(client, rx, sd, captured))
.ok();
Some(MicCapture {
_stream: stream,
shutdown,
join,
})
}
}
impl Drop for MicCapture {
fn drop(&mut self) {
self.shutdown.store(true, Ordering::SeqCst);
if let Some(j) = self.join.take() {
let _ = j.join();
}
// `_stream` drops here → AAudio request_stop + close.
}
}
/// Consumer: drain captured f32 → accumulate → Opus `encode_float` 20 ms stereo frames → `send_mic`.
fn encode_loop(
client: Arc<NativeClient>,
rx: Receiver<Vec<f32>>,
shutdown: Arc<AtomicBool>,
captured: Arc<AtomicU64>,
) {
let mut enc = match opus::Encoder::new(
SAMPLE_RATE as u32,
opus::Channels::Stereo,
opus::Application::Voip,
) {
Ok(e) => e,
Err(e) => {
log::error!("mic: opus encoder init: {e} — mic disabled");
return;
}
};
let _ = enc.set_bitrate(opus::Bitrate::Bits(MIC_BITRATE));
let frame = FRAME_SAMPLES * CHANNELS;
let mut ring: VecDeque<f32> = VecDeque::with_capacity(frame * 4);
let mut out = vec![0u8; 4000]; // max Opus packet for a 20 ms frame fits easily
let mut seq: u32 = 0;
let mut sent: u64 = 0;
let mut peak = 0f32; // loudest |sample| since the last log — tells speech from silence
while !shutdown.load(Ordering::Relaxed) {
match rx.recv_timeout(Duration::from_millis(100)) {
Ok(chunk) => ring.extend(chunk),
Err(RecvTimeoutError::Timeout) => continue, // wake to re-check shutdown
Err(RecvTimeoutError::Disconnected) => break,
}
while ring.len() >= frame {
let pcm: Vec<f32> = ring.drain(..frame).collect();
for &s in &pcm {
peak = peak.max(s.abs());
}
match enc.encode_float(&pcm, &mut out) {
Ok(len) => {
let pts = SystemTime::now()
.duration_since(UNIX_EPOCH)
.map(|d| d.as_nanos() as u64)
.unwrap_or(0);
let _ = client.send_mic(seq, pts, out[..len].to_vec());
seq = seq.wrapping_add(1);
sent += 1;
if sent % 250 == 0 {
log::info!(
"mic: sent={sent} captured_frames={} peak={peak:.3}",
captured.load(Ordering::Relaxed),
);
peak = 0.0;
}
}
Err(e) => log::debug!("mic: opus encode: {e}"),
}
}
}
log::info!(
"mic: stopped (sent={sent} captured_frames={})",
captured.load(Ordering::Relaxed),
);
}