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voice-cat/tests/test_jitter_depth.cpp
Talon b44a200b95
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fix(audio): apply receive-side NR to stereo mic streams
The per-listener noise-reduction toggle (vc_set_remote_stream) did nothing
on Windows/macOS/iOS. The decode loop gated the RNNoise pass on
dec_channels == 1 as a proxy for "this stream is voice" (assuming
stereo => screen-share). The stereo-mic capture commit broke that: a stereo
mic with send-side NR off transmits stereo Opus, so the receiver decoded
two channels and skipped NR entirely. gain/mute have no channel guard, which
is why only NR appeared broken.

Thread the stream kind through init_recv_stream into RemoteStream::is_voice
(set from si.kind() == STREAM_MIC), gate receive NR on is_voice instead of
channel count, and fold a stereo voice frame to mono -> denoise -> duplicate
back across both channels in place (symmetric with the send-side downmix;
RNNoise is mono-only). Screen-audio shares are never denoised.

New test test_recv_noise_reduction drives AudioEngine and asserts a stereo
voice stream's noise floor collapses with NR on (RMS 1046 -> 0.1) while a
screen-audio share stays unchanged. ctest --preset dev green 29/29.
Docs: voice.md section 10. Shared-core fix; clients need only a rebuild.

Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-06-23 21:11:03 +02:00

145 lines
6.2 KiB
C++

/*
* test_jitter_depth — verifies the bounded-depth playout in AudioEngine::on_playback.
*
* Regression guard for the "latency keeps drifting backward, fixed only by rejoining" bug. The
* sender omits VAD/PTT/DTX silence from its timestamps (a compressed timeline), while the
* receiver's playout clock free-runs in real time. The old logic re-synced the playout clock to
* the *oldest* buffered frame and could only ever *add* standing latency (a reordered/late frame
* snapped the clock backward), with nothing to trim it — so latency ratcheted up across talkspurt
* gaps. The fix keeps the clock a bounded `target` behind the *newest* arrival and frame-skips to
* catch up, so depth stays bounded no matter the trigger.
*
* This drives many talkspurt/silence cycles with a compressed timeline plus a reordered straggler
* each cycle (which previously snapped the clock backward), and asserts the buffered depth
* (newest_ts - playout_ts) stays bounded while audio keeps playing. White-box via mix_for_test
* (no audio hardware needed), same pattern as test_plc_cap.
*/
#include <cmath>
#include <cstdint>
#include <cstdio>
#include <vector>
#if defined(VOICECAT_HAS_AUDIO) && defined(VOICECAT_HAS_OPUS)
#include "audio/audio_engine.h"
#include "codec/opus_codec.h"
static int g_failures = 0;
#define CHECK(cond) \
do { \
if (!(cond)) { \
std::printf("FAIL [%s:%d]: %s\n", __FILE__, __LINE__, #cond); \
++g_failures; \
} \
} while (0)
static double rms(const int16_t* pcm, int n) {
double sum = 0.0;
for (int i = 0; i < n; ++i) sum += static_cast<double>(pcm[i]) * pcm[i];
return std::sqrt(sum / n);
}
int main() {
voicecat::audio::AudioEngine engine;
voicecat::audio::AudioParams p;
p.sample_rate = 48000;
p.capture_channels = 1;
p.playback_channels = 2;
p.frame_ms = 20;
CHECK(engine.start(p)); // no capture_cb — headless safe
voicecat::codec::OpusParams op;
op.sample_rate = 48000;
op.frame_ms = 20;
op.stereo = false;
const int frame_samples = voicecat::codec::opus_frame_samples(op); // 960
// A loud sine, encoded once, reused for every pushed frame.
voicecat::codec::OpusEncoder enc;
CHECK(enc.init(op));
std::vector<int16_t> sine(static_cast<size_t>(frame_samples));
for (int i = 0; i < frame_samples; ++i) {
float t = static_cast<float>(i) / 48000.0f;
sine[i] = static_cast<int16_t>(std::sin(2.0f * 3.14159265f * 440.0f * t) * 20000.0f);
}
uint8_t opus_buf[1500];
const int opus_len = enc.encode(sine.data(), frame_samples, opus_buf, sizeof(opus_buf));
CHECK(opus_len > 0);
const uint32_t ssrc = 1;
engine.init_recv_stream(ssrc, op, /*user_id=*/0, /*stream_id=*/0, /*is_voice=*/false);
const uint32_t pb_frames = 480; // 10 ms hardware period
const int out_n = static_cast<int>(pb_frames) * 2; // stereo interleaved
std::vector<int16_t> out(static_cast<size_t>(out_n), 0);
auto mix_n = [&](int n) {
for (int i = 0; i < n; ++i) engine.mix_for_test(out.data(), pb_frames);
};
auto push = [&](uint32_t ts, bool marker) {
voicecat::audio::JitterBuffer::Frame f;
f.seq = 0;
f.timestamp = ts;
f.fec_present = false;
f.marker = marker;
f.payload.assign(opus_buf, opus_buf + opus_len);
engine.push_recv_frame(ssrc, std::move(f));
};
uint32_t ts = 1000; // arbitrary non-zero start
int32_t max_depth = 0;
double last_voice_rms = 0.0;
// Seed the stream (first frame is a talkspurt marker, like a real resume).
push(ts, /*marker=*/true);
ts += static_cast<uint32_t>(frame_samples);
mix_n(1);
// Drive the producer FASTER than the consumer: push one 960-sample frame per step but drain
// only 480 samples (one pb_frames callback) — i.e. arrivals outrun playout by ~480 samples a
// step, exactly the clock-drift / bursty-arrival condition that made latency ratchet up. Also
// inject a reordered straggler periodically (the old backward-snap trigger). The bounded-depth
// catch-up must keep the standing latency from growing without limit. Pre-fix (no catch-up,
// snap-to-oldest) the depth would climb to ~hundreds of frames here.
const int kSteps = 400;
const uint32_t kStraggler = 48000u * 250u / 1000u; // 250 ms behind the leading edge
for (int s = 0; s < kSteps; ++s) {
push(ts, /*marker=*/false);
ts += static_cast<uint32_t>(frame_samples);
if (s % 25 == 12) push(ts - kStraggler, /*marker=*/false); // reordered straggler
mix_n(1); // drain only 480 of the 960 produced — producer outruns consumer
int32_t d = engine.stream_playout_depth_samples(ssrc);
if (d > max_depth) max_depth = d;
last_voice_rms = std::max(last_voice_rms, rms(out.data(), out_n));
}
std::printf("jitter_depth: max_depth=%d samples (%.0f ms) voice_rms=%.1f\n", max_depth,
static_cast<double>(max_depth) * 1000.0 / 48000.0, last_voice_rms);
// Bounded: with catch-up the standing latency stays near the adaptive target, well under
// 200 ms even though arrivals outran playout for 400 steps (~4 s of pushed audio).
CHECK(max_depth > 0); // playout ran / depth observed
CHECK(max_depth < static_cast<int32_t>(48000 * 200 / 1000)); // bounded (was unbounded pre-fix)
CHECK(last_voice_rms > 1.0); // audio keeps playing
engine.remove_stream(ssrc);
engine.stop();
enc.destroy();
if (g_failures == 0) {
std::printf("jitter_depth: all checks passed\n");
return 0;
}
std::printf("jitter_depth: %d failure(s)\n", g_failures);
return 1;
}
#else
int main() {
std::printf("jitter_depth: SKIP (VOICECAT_HAS_AUDIO or VOICECAT_HAS_OPUS not defined)\n");
return 0;
}
#endif