86d00c43d1
Drops the FILTER_SAMPLE_RATE const dependency from the filter's
creation path so the audio thread can run at whatever rate the
real sink negotiates, not unconditionally 48 kHz. Closes one of
the two output-edge resamples PLAN §3.1's F5 caveat called out
— content matching the real-sink rate now passes through the
limiter without an output-side resample elevating its true peaks.
Phase A (foundation)
- `Filter::create(core, init, sample_rate)` takes the rate as a
runtime parameter. `DEFAULT_SAMPLE_RATE` keeps 48 kHz as the
fallback constant; `FILTER_SAMPLE_RATE` is kept as a
back-compat alias.
- `build_format_pod_bytes(sample_rate)` parameterised so the SPA
Format the filter advertises matches the chosen rate.
- `FilterBundle.sample_rate` exposed so the AGC controller and
`runtime` can size their own state.
- New `LimiterConfig::sanitize_for_rate(sample_rate)` caps the
oversample factor so the internal (post-upsample) rate stays
≤ 192 kHz: 48 k base → 4× = 192 k; 96 k → 2× = 192 k; 192 k
→ 1× = 192 k. Keeps the FIR cost from doubling each time the
base rate doubles, with negligible loss of true-peak detection
quality at high base rates (the signal already has plenty of
bandwidth). Two regression tests lock the math in.
Phase B (data plumbing)
- `SinkInfo` (wire-level) gains an optional `sample_rate`
field. `headroom status` now reports the processed sink's
running rate and the real sink's native rate — useful for
debugging "did the daemon actually match my hardware?"
without resorting to `pw-link`.
- `state::RealSink.sample_rate` populated by the registry
watcher from two sources:
- The `audio.rate` property (many virtual sinks expose it).
- A `Format`-param listener bound to the real sink's `Node`
proxy (ALSA sinks only expose the rate in the negotiated
Format, not in their property dict). New
`install_real_sink_format_listener` mirrors the
channelVolumes-listener pattern Layer A already uses.
Listener cleaned up in `on_global_remove` when the real
sink departs.
- `state::DaemonState.filter_sample_rate` mirrors the bus
filter's currently-running rate; surfaced in `status`.
- Layer A's block-period constant becomes a runtime function
(`layer_a_block_dt_s(sample_rate)`) so 96 k / 192 k hardware
gets correctly-scaled controller time-constants.
Known gap: filter created at boot uses whatever rate is known at
that moment. For ALSA sinks the Format listener fires ~tens of ms
*after* the registry capture — by which time the filter is
already created at the fallback rate. The next commit (Phase C)
rebuilds the filter when the listener delivers a rate different
from what the filter is running at.
Verified
- 191 tests pass (was 189; +2 for the new
`sanitize_for_rate` cases); clippy clean at -D warnings
--all-targets.
- Live: cold-boot against a 48 kHz Mbox shows
`status.sinks.processed.sample_rate = 48000` +
`status.sinks.real.sample_rate = 48000`, daemon log records
"creating filter at real-sink-matched rate initial_rate=48000"
and "real sink Format negotiated; updating sample_rate
new_rate=48000" within ~55 ms of each other. For sinks where
`audio.rate` IS in props (some virtual sinks) the rate is
captured before filter creation.
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| benches | ||
| src | ||
| Cargo.toml | ||
| README.md | ||
headroom-dsp
DSP kernels for Headroom. Pure Rust, no dependencies.
Limiter— feed-forward true-peak brickwall with configurable oversampling (1/2/4/8×), lookahead, hold, and release.Compressor— log-domain feed-forward with peak or RMS detector, soft knee, attack/release, and optional auto-makeup.AttackRelease— exponential envelope follower (peak / inverse-gain modes).DelayLine,SlidingMaxBuffer,PolyphaseUpsampler,PolyphaseDownsampler— supporting building blocks.
All processors are allocation-free in their process_* methods.
Construction allocates; do not construct in the audio thread.
License
MPL-2.0.