When PipeWire drops the stream during pause, the graph reverts to the
DAC's default rate (e.g. 44100Hz). The pre-stream force-rate metadata
can be ignored when no streams are active. Re-applying after the stream
is connected forces PipeWire to reconfigure the graph at the correct
sample rate.
Also fix unused import warning for MusicBrainzSharedState in commands_v2
by using the import instead of the full path in factory_reset.
Only write to gain atomic on initial measurement. Refinements now
update the cache only, preventing audible volume changes within a
single track. Inter-track normalization still works via cached values.
When ALSA backend detects hardware doesn't support a track's sample rate
(e.g. 176400Hz on a DAC that supports up to 192000Hz but not 176.4kHz),
automatically re-request the track from Qobuz at Hi-Res quality instead
of Ultra Hi-Res. This avoids resampling and maintains bit-perfect playback
by getting a version at a rate the hardware natively supports.
Adds device_supports_sample_rate() and get_device_supported_rates()
public API to qbz-audio for hardware capability queries via /proc/asound.
When transitioning from ALSA Direct to CPAL fallback (e.g. for rate
resampling), PipeWire auto-re-acquires the device after ALSA Direct
releases it. CPAL then can't open the device.
Suspend PipeWire sinks before creating CPAL stream in exclusive mode,
same pattern used in ALSA Direct DeviceBusy handler.
When ALSA Direct opens a PCM device and fails to set the rate, the
USB DAC doesn't release the device fast enough for CPAL to open it
afterwards. This caused "device busy" errors even with a delay.
New approach: read /proc/asound/cardN/stream0 to check supported
rates BEFORE opening the device. If the rate isn't supported, skip
ALSA Direct entirely and fall through to CPAL (which has rate
fallback logic). The device is never touched at the wrong rate.
When ALSA Direct opens a PCM device and fails to configure the rate,
the PCM handle is dropped but the kernel may not release the device
instantly. CPAL then fails with "Device busy" when trying to open it.
Add 100ms sleep after InvalidParams to let the kernel fully release
the device before CPAL attempts to open it. Also fix error messages
to show the effective (fallback) rate instead of the original rate.
When ALSA Direct can't handle a rate (e.g. 176.4kHz) and falls back
to CPAL, CPAL also fails because the hardware doesn't support it.
Port find_best_fallback_rate from PipeWire backend: queries supported
rates via CPAL, picks highest rate in same family (44.1k or 48k)
that's <= requested. 176.4kHz on a 96kHz-max DAC now falls back to
88.2kHz and rodio resamples, instead of failing entirely.
When hardware doesn't natively support a sample rate (e.g. 176.4kHz),
return None from try_create_direct_stream instead of a hard error.
This lets the player fall through to CPAL/rodio which can resample.
Applied to both hw and plughw paths — InvalidParams on rate now
triggers graceful fallback instead of killing playback.
When switching from PipeWire to ALSA Direct, PipeWire holds the ALSA
device open causing EBUSY on the first snd_pcm_open. Now detects
DeviceBusy, suspends the PipeWire sink via pactl, waits 200ms for
release, then retries. Applied to both hw and plughw paths.
cpal 0.15 returned "ALSA" from HostId::name(), cpal 0.17 returns "Alsa"
(now uses stringify! on the enum variant). Use eq_ignore_ascii_case()
for the comparison so it works regardless of cpal version.
Query the DAC's supported sample rates from /proc/asound/cardN/stream0
(maps PipeWire sink to ALSA card via pactl). If the track's native rate
isn't supported (e.g., 176.4kHz), fall back to the nearest rate in the
same family (e.g., 88.2kHz). Rodio resamples from track rate to stream
rate automatically, so audio plays instead of producing silence.
rodio 0.22's with_supported_config() resets buffer_size to Default via
..Default::default(), so our BufferSize settings in StreamConfig were
never reaching CPAL. MixerDeviceSink has zero internal buffering, making
CPAL's buffer the ONLY buffer between the mixer and audio hardware.
Chain .with_buffer_size(Fixed(sample_rate/10)) AFTER with_supported_config()
to actually pass ~100ms buffer to CPAL, matching old vendored cpal behavior.
PipeWire manages its own buffer scheduling via quantum. Forcing a fixed
buffer size (sample_rate / 10 = 19200 frames at 192kHz) conflicts with
PipeWire's internal scheduling and causes buffer underruns that produce
audible noise.
BufferSize::Fixed is kept only for:
- Exclusive mode (512 frames, low latency)
- ALSA backend (direct ALSA, not through PipeWire)
rodio 0.22's MixerDeviceSink has an internal mixer thread that cannot
synchronize with PipeWire's forced quantum scheduling. This caused
massive buffer underruns (600+) at sample rates >= 88.2kHz when Force
Bit Perfect was enabled.
clock.force-rate alone is sufficient for bit-perfect sample rate
switching. Logs confirm zero underruns at 88.2kHz and minimal underruns
at 192kHz (only during heavy I/O) when quantum is not forced.
Replace BufferSize::Default with BufferSize::Fixed(sample_rate / 10)
for non-exclusive mode. This preserves the 100ms period tuning that was
previously applied in the vendored cpal ALSA backend, reducing CPU
wakeups during playback.
- OutputStream/OutputStreamHandle -> DeviceSinkBuilder/MixerDeviceSink
- Sink -> Player (as RodioPlayer)
- current_frame_len -> current_span_len in all Source impls
- Source::channels() returns NonZero<u16>, sample_rate() returns NonZero<u32>
- SamplesBuffer no longer generic, constructor takes NonZero args
- SampleRate is now u32 type alias (no tuple struct)
- Decoder::new_mp4 no longer takes Mp4Type hint
- Applied across both crates/ workspace and src-tauri/ legacy code
Add experimental toggle to force PipeWire clock.force-quantum alongside
clock.force-rate for true bit-perfect playback. Quantum is matched to
sample rate (1024 for <=48kHz, 2048 for <=96kHz, etc). Both values
reset to 0 on stop and pause-suspend to avoid affecting other apps.
Requires DAC Passthrough to be enabled, PipeWire backend only.
- add Spectral Ribbon analyzer/render path and immersive panel
- tune ribbon density/interpolation and bars edge-bin rendering
- apply artist blacklist filtering to discover index home containers
- include supporting i18n/settings/player wiring changes
cx
Extract player/mod.rs, playback_engine.rs, and streaming_source.rs to
qbz-player crate. Also adds settings.rs and visualizer module to
qbz-audio (Tauri-specific code removed, core functionality retained).
Changes:
- Add settings.rs to qbz-audio with AudioSettings, AudioSettingsStore
- Add visualizer/ to qbz-audio with RingBuffer, TappedSource, VisualizerTap
- Add player module files to qbz-player
- Update all imports from crate::audio to qbz_audio
- Add serde_json dependency to qbz-audio
Full workspace compiles with cargo check.
Create the foundational workspace structure for the multi-crate
architecture refactor. This phase establishes placeholder crates
that will be populated in subsequent phases.
Crates created:
- qbz-models: Shared types, events, and traits (Tier 0)
- qbz-audio: Audio backend system (Tier 1)
- qbz-player: Playback engine and queue (Tier 1)
- qbz-qobuz: Qobuz API client (Tier 1)
- qbz-core: Orchestrator (Tier 3)
All crates compile successfully with placeholder implementations.
The workspace uses shared dependencies via workspace.dependencies.
cc
vicrodh