diag(android-audio): aggressive logging across the whole Oboe pipeline
User confirmed: mac hears android, android does not hear mac. So Oboe
capture works end-to-end but Oboe playout on Android silently drops
audio even though QUIC forwards the packets. Archaeology on the legacy
wzp-android crate also revealed that the "last known good" Android audio
path NEVER used Oboe in production — it used Kotlin AudioRecord +
AudioTrack via JNI, and cpp/oboe_bridge.cpp was dead code. So every time
we've "tested" Oboe end-to-end this week was the first production use,
and any of its config knobs could be the bug.
Instrumenting every stage of the pipeline so one smoke-test log dump can
isolate the layer at fault:
C++ (oboe_bridge.cpp)
- Log the ACTUAL stream parameters after openStream for both capture
and playout (sample rate, channels, format, framesPerBurst,
framesPerDataCallback, bufferCapacityInFrames, sharing, perf mode).
Oboe may silently override values we requested — e.g. if we ask for
48kHz mono but the device gives us 44.1kHz stereo our 960-sample
frames are the wrong duration and the pipeline drifts.
- Capture callback: on cb#0 log sample range+RMS of the first frame
to prove we get real mic data (not zeros). Every 50 callbacks
(~1s at 20ms burst) log calls, numFrames, ring available_write,
bytes actually written, ring_full_drops, total_written.
- Playout callback: on cb#0 log numFrames + ring state. On the FIRST
non-empty read log sample range+RMS so we can tell if the samples
coming out of the ring are real audio or zeros. Every 50 callbacks
log calls, nonempty count, numFrames, ring available_read,
underrun_frames, total_played_real.
Rust wzp-native (src/lib.rs)
- wzp_native_audio_write_playout now logs the first 3 writes and then
every 50th: in_len, written, sample range, RMS, ring write/read
cursors before, available_read and available_write after. Reveals
ring-overflow and whether the engine is actually handing us audio.
- Minimal android logcat shim via __android_log_write extern — no
new crate dependency.
- AudioBackend grows a `playout_write_log_count` AtomicU64 to gate
the write-side log throttle.
Rust engine.rs (android branch)
- Recv task: log sample range + RMS for the first 3 decoded PCM
frames and then every 100th. Reveals whether decoder.decode is
producing real audio or silent buffers.
- Recv task: if audio_write_playout returns fewer samples than we
handed it (partial write → ring nearly full) warn about it in the
first 10 frames.
- Recv heartbeat every 2s: recv_fr, decoded_frames, last_decode_n,
last_written, written_samples, decode_errs, codec.
Expected flow in a healthy log:
capture cb#0: numFrames=960 range=[-1200..900] rms=180 ← mic OK
capture stream opened: actualSR=48000 Ch=1 ... ← no override
playout stream opened: actualSR=48000 Ch=1 ...
CallEngine::start invoked ... → connected → audio started
recv: first media packet received ...
recv: decoded PCM sample range decoded_frames=1 range=[-300..250] rms=92
playout WRITE #0: in_len=960 written=960 range=[-300..250] rms=92
playout FIRST nonempty read: to_read=960 range=[-300..250] rms=92
playout heartbeat: calls=50 nonempty=50 underrun=0 ...
recv heartbeat: decoded_frames=100 last_written=960 ...
If any of those are missing/zero we know the exact stage to fix.
This commit is contained in:
@@ -351,11 +351,34 @@ impl CallEngine {
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Ok(n) => {
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last_decode_n = n;
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decoded_frames += 1;
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// Log sample range for the first few decoded frames and periodically
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if decoded_frames <= 3 || decoded_frames % 100 == 0 {
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let slice = &pcm[..n];
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let (mut lo, mut hi, mut sumsq) = (i16::MAX, i16::MIN, 0i64);
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for &s in slice.iter() {
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if s < lo { lo = s; }
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if s > hi { hi = s; }
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sumsq += (s as i64) * (s as i64);
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}
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let rms = (sumsq as f64 / n as f64).sqrt() as i32;
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info!(
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decoded_frames,
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n,
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sample_lo = lo,
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sample_hi = hi,
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rms,
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codec = ?current_codec,
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"recv: decoded PCM sample range"
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);
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}
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agc.process_frame(&mut pcm[..n]);
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if !recv_spk.load(Ordering::Relaxed) {
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let w = crate::wzp_native::audio_write_playout(&pcm[..n]);
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last_written = w;
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written_samples = written_samples.saturating_add(w as u64);
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if w < n && decoded_frames <= 10 {
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tracing::warn!(n, w, "recv: partial playout write (ring nearly full)");
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}
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}
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}
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Err(e) => {
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@@ -378,6 +401,22 @@ impl CallEngine {
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}
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Err(_) => {}
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}
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// Heartbeat every 2s with decode+playout state
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if heartbeat.elapsed() >= std::time::Duration::from_secs(2) {
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let fr = recv_fr.load(Ordering::Relaxed);
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info!(
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recv_fr = fr,
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decoded_frames,
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last_decode_n,
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last_written,
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written_samples,
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decode_errs,
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codec = ?current_codec,
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"recv heartbeat (android)"
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);
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heartbeat = std::time::Instant::now();
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}
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}
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});
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