Better error handling in Alsa backend
* More consistent error messages
* Bail on fatal errors in player
* Capture and log the original error as a warning when trying to write to PCM before trying to recover
* Go back to 4 periods at 125ms.
* Deal strictly in period time and periods to set ALSA buffer.
* Rename `buffer` to `period_buffer`.
* Add comments and change some other var names to add clarity.
* Let ALSA calculate the size of `period_buffer`.
This caused quite a bump in CPU usage, which be acceptable if this
actually improved sound quality. However, it turns out that this
function only has one decimal precision, i.e. it would consider
all values from `0.50..0.60` (exclusive) as `0.5` which is in
error for our purposes.
* Remove deprecated use of std::u16::MAX
* Use `FromStr` for fallible `&str` conversions
* DRY up strings into constants
* Change `as_ref().map()` into `as_deref()`
* Use `Duration` for time constants and functions
* Optimize `Vec` with response times
* Move comments for `rustdoc` to parse
Dithering lowers digital-to-analog conversion ("requantization") error, linearizing output, lowering distortion and replacing it with a constant, fixed noise level, which is more pleasant to the ear than the distortion.
Guidance:
- On S24, S24_3 and S24, the default is to use triangular dithering. Depending on personal preference you may use Gaussian dithering instead; it's not as good objectively, but it may be preferred subjectively if you are looking for a more "analog" sound akin to tape hiss.
- Advanced users who know that they have a DAC without noise shaping have a third option: high-passed dithering, which is like triangular dithering except that it moves dithering noise up in frequency where it is less audible. Note: 99% of DACs are of delta-sigma design with noise shaping, so unless you have a multibit / R2R DAC, or otherwise know what you are doing, this is not for you.
- Don't dither or shape noise on S32 or F32. On F32 it's not supported anyway (there are no integer conversions and so no rounding errors) and on S32 the noise level is so far down that it is simply inaudible even after volume normalisation and control.
New command line option:
--dither DITHER Specify the dither algorithm to use - [none, gpdf,
tpdf, tpdf_hp]. Defaults to 'tpdf' for formats S16
S24, S24_3 and 'none' for other formats.
Notes:
This PR also features some opportunistic improvements. Worthy of mention are:
- matching reference Vorbis sample conversion techniques for lower noise
- a cleanup of the convert API
This is a squashed commit featuring the following:
Connect:
- Synchronize player volume with mixer volume on playback
- Fix step size on volume up/down events
- Remove no-op mixer started/stopped logic
Playback:
- Move from `connect` to `playback` crate
- Make cubic volume control available to all mixers with `--volume-ctrl cubic`
- Normalize volumes to `[0.0..1.0]` instead of `[0..65535]` for greater precision and performance (breaking)
- Add `--volume-range` option to set dB range and control `log` and `cubic` volume control curves
- Fix `log` and `cubic` volume controls to be mute at zero volume
Alsa mixer:
- Complete rewrite (breaking)
- Query card dB range for the `log` volume control unless specified otherwise
- Query dB range from Alsa softvol (previously only from hardware)
- Use `--device` name for `--mixer-card` unless specified otherwise
- Fix consistency for `cubic` between cards that report minimum volume as mute, and cards that report some dB value
- Fix `--volume-ctrl {linear|log}` to work as expected
- Removed `--mixer-linear-volume` option; use `--volume-ctrl linear` instead
fe37186 added the restriction that `Sink`s must be `Send`. It turned
out later that this restrictions was unnecessary, and since some
`Sink`s aren't `Send` yet, this restriction is lifted again.
librespot-org/librespot#601 refactored the `RodioSink` in order to make
it `Send`. These changes are partly reverted in favour of the initial
simpler design.
Furthermore, there were some compile errors in the gstreamer backend
which are hereby fixed.
- DRY-ups
- Remove incorrect optimization attempt in the libvorbis decoder,
that skewed 0.0 samples non-linear
- PortAudio and SDL backends do not support S24 output. The PortAudio
bindings could, but not through this API.
Usage: `--format {F32|S16}`. Default is F32.
- Implemented for all backends, except for JACK audio which itself
only supports 32-bit output at this time. Setting JACK audio to S16
will panic and instruct the user to set output to F32.
- The F32 default works fine for Rodio on macOS, but not on Raspian 10
with Alsa as host. Therefore users on Linux systems are warned to set
output to S16 in case of garbled sound with Rodio. This seems an issue
with cpal incorrectly detecting the output stream format.
- While at it, DRY up lots of code in the backends and by that virtue,
also enable OggData passthrough on the subprocess backend.
- I tested Rodio, ALSA, pipe and subprocess quite a bit, and call on
others to join in and test the other backends.
- Store and output samples as 32-bit floats instead of 16-bit integers.
This provides 24-25 bits of transparency, allowing for 42-48 dB of
headroom to do volume control and normalisation without throwing
away bits or dropping dynamic range below 96 dB CD quality.
- Perform volume control and normalisation in 64-bit arithmetic.
- Add a dynamic limiter with configurable threshold, attack time,
release or decay time, and steepness for the sigmoid transfer
function. This mimics the native Spotify limiter, offering greater
dynamic range than the old limiter, that just reduced overall gain
to prevent clipping.
- Make the configurable threshold also apply to the old limiter, which
is still available.
Resolves: librespot-org/librespot#608
