Commit Graph
3 Commits
Author SHA1 Message Date
gramps 102cb96b28 fix(audio): only count audio that actually reached the encoder
Second measurement of the same recording family (ty-20261112-1133-701-0000-2.mp4):

  wall clock (audio start -> audio stop)  6.477 s
  video: 386 frames                       6.455 s   correct
  audio: 294 packets                      6.252 s   226 ms SHORT

Same magnitude as the original bug, opposite sign — the clock fixed the tick-rate error and
exposed the next one underneath. Audio started 49 ms before the pump, ffmpeg connected ~80 ms
after, and every chunk written in that window was dropped (dropped writes 27). The loop still
counted those frames as produced, so the track lost exactly that much time.

Now: the clock does not start until the pipe is connected, and AudioClock.Advance counts a
frame only when its write reached the pipe. A dropped write stays owed and is paid off on the
next tick, so a start-up gap costs content (silence, which there was anyway) instead of
length.

Two measurements, two different signs, one lesson: the audio track's LENGTH has to be the
clock's, and anything that removes samples without removing time — a fast tick, a dropped
write — shows up as drift no static offset can fix.

674 tests (one notification-timing flake under full-suite load; passes alone), 0 warnings.
2026-11-12 11:39:20 -08:00
gramps f3d578c81b fix(audio): add per-5s live-loop telemetry to name the silent-recording stage
The recorded audio is full-length silent AAC (−91dB, 1124 frames/23.95s) —
the named pipe carried ~9.1MB of zeros the entire take. The mixer's loop
ran, the pipe connected, ffmpeg read it, and the file is full-length
silence. Sources started fine ('Audio: using system default mic...' logged),
no failure callbacks fired, and the existing integration test
(Mix_WithFiltersDuckAndGain_Lands_On_AudioPipe) proves the loop→pipe path
carries real audio when fed — the fault is capture-side.

Added permanent per-5s live-loop telemetry to startup.log so the next take
names the exact stage without new code:

  Audio live: pipe connected= dropped writes= micLevel= loopLevel= drained
  N/N samples peakMix N (per 5s)

- AudioMixer.FillAndMix now returns (MicRms, MicDrained, LoopDrained) for
  the accumulation; StartLive/StopLive log start/stop lines.
- NamedPipeAudioWriter.DroppedWrites: nonzero = audio dropped before ffmpeg
  connected (names 'pipe never connected' stage).

Likely root cause: WASAPI loopback captures the default render endpoint —
if audio plays on a non-default device the recording is silently silent.
The fix requires device enumeration + selection (slice 13 candidate).

Suite 290/291 — same sole pre-existing compositor pixel failure.
2026-09-10 10:55:13 -07:00
gramps 6d71acef8c TASK 9 audio milestone: real stream audio + voice filters + auto-duck + free TRAX music — the mixer now feeds the encoder real audio (ffmpeg reads a Windows named pipe, -f f32le -ar 48000 -ac 2 -i \.\pipe\ytllive_audio, replacing anullsrc silence; explicit -map 0:v -map 1:a via EncoderOptions.AudioPipeName), with honest gains reaching the stream (MicVolume scales mic, GameAudioVolume + mute scale loopback, Func<double> seams on AudioMixer), an always-on pure-C# voice chain on the mic BEFORE the meter and mix (LowShelfFilter 120Hz +4dB → HighShelfFilter 8kHz +3dB → NoiseGate 0.005/hysteresis 0.5 → Compressor 0.5 4:1, all TDF2), AutoDucker (mic RMS>0.02 → loopback ×0.25, attack 0.05/release 0.005), and TRAX free background music (MusicPlayer = MediaFoundationReader → VolumeWaveProvider16 at fixed 0.20 → WaveOutEvent via the new sibling NAudio.WinMM 2.2.1 package; plays to the default device so the existing loopback carries it, ducked with game; footer TRAX button with red/yellow/green status dot, left-click toggles/picks, right-click opens the OpenFileDialog picker, tooltip shows the track name; persisted via schema v9 single-row Music). Build-time deviations from the plan (recorded in ai.md + TASKS.md): WasapiCapture in NAudio 2.2.1 exposes no overridable GetDefaultMixFormat so sources run device mix format and the mixer's TinyResampler normalizes any rate to 48kHz (resampler IS the design); FfmpegEncoder.cs untouched — MainViewModel.StopStream calls _audioMixer.StopLive() (pipe EOF) before the frame pump stops; sound-bar relabelled 'Desktop Audio', IsGameAudioBarVisible = game sound OR music playing. Tests: new AudioPipelineTests (DSP/ring-buffer/ducker/resampler units + the ONE integration test Mix_WithFiltersDuckAndGain_Lands_On_AudioPipe reading real pipe bytes; ring-buffer overwrite bug found + fixed), FfmpegEncoderTests/LayoutStorePersistenceTests updated — 196 tests passing, 0 warnings 2026-08-14 18:09:26 -07:00