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`.
* 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
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
Probably more simple than the previous approach which
doubles the code: Instead of implementing the `Open` trait,
we simply use custom SinkBuilder, one for the default host,
and one for the "jack" host.
A bunch of stuff got moved around; means of constructing audio output port
changed.
I simply used the commits, mostly from [1], to their examples to figure
out how to address the errors that resulted from compiling after the
version bump. It compiles cleanly again now.
[1]: https://github.com/RustAudio/rust-jack/pull/89
requires adding dependency on libpulse-simple-sys since the PulseAudio
simple components were moved to their own crate (the original version
did not stick to the one crate per one system library rule).
this fixes the licensing compatibility issue discussed in #539 ([1])
(the original v0.0.0 was LGPL-3.0 licensed, while v1.11 onwards are
'MIT OR Apache-2.0').
[1]: https://github.com/librespot-org/librespot/issues/539
This prevents over or underestimating of the period.
While it is unlikely, with comparitively small period
sizes overestimating can cause buffer underruns and
underestimating causes more writes than necessary.
It also properly accounts for the number of channels,
which I had overlooked.
Writing to the pcm more often than necessary is
just a waste of resources and depending
on the pcm it can have quite an impact
on performance. The pcm expects full periods
anyway.
* create Volume struct for use with Cache
* add "volume" file to Cache
* load cached volume on start, intial overrides cached overrides default
* amend volume_to_mixer function to cache the volume on every change
* pass cache to Spirc and SpircTask so volume_to_mixer has access
* rustfmt changes
* revert volume_to_mixer function and Spirc/SpircTask cache variable
* Volume implements Copy, pass by value instead of reference
* clamp volume to 100 if cached value exceeds limit
* convert Volume to u16 internally, use float and round to convert hex->dec
* convert initial_volume and ConnectConfig.volume to u16 as well
* add cache_volume function to SpircTask
* remove conversion to/from percentage on cached volume
* consolidate device.set_volume, mixer.set_volume, and caching
* streamline intial volume logic
