Dynamically set the alsa buffer and period based on the device's reported min/max buffer and period sizes. In the event of failure use the device's defaults.
This should have no effect on devices that allow for reasonable buffer and period sizes but would allow us to be more forgiving with less reasonable devices or configurations.
Closes: https://github.com/librespot-org/librespot/issues/895
This makes `--device ?` only show compatible devices (ones that support 2 ch 44.1 Interleaved) and it shows what `librespot` format(s) they support.
This should be more useful to users as the info maps directly to `librespot`'s `--device` and `--format` options.
* Don't panic when parsing options. Instead list valid values and exit.
* Get rid of needless .expect in playback/src/audio_backend/mod.rs.
* Enforce reasonable ranges for option values (breaking).
* Don't evaluate options that would otherwise have no effect.
* Add pub const MIXERS to mixer/mod.rs very similar to the audio_backend's implementation. (non-breaking though)
* Use different option descriptions and error messages based on what backends are enabled at build time.
* Add a -q, --quiet option that changed the logging level to warn.
* Add a short name for every flag and option.
* Note removed options.
* Other misc cleanups.
While `Xoshiro256+` is faster on 64-bit, it has low linear complexity in the
lower three bits, which *are* used when generating dither.
Also, while `Xoshiro128StarStar` access one less variable from the heap,
multiplication is generally slower than addition in hardware.
* Make error messages more consistent and concise.
* `impl From<AlsaError> for io::Error` so `AlsaErrors` can be thrown to player as `io::Errors`. This little bit of boilerplate goes a long way to simplifying things further down in the code. And will make any needed future changes easier.
* Bonus: handle ALSA backend buffer sizing a little better.
* Improve error handling
* Harmonize `Seek`: Make the decoders and player use the same math for converting between samples and milliseconds
* Reduce duplicate calls: Make decoder seek in PCM, not ms
* Simplify decoder errors with `thiserror`
* Reuse the buffer for the life of the Alsa sink
* Don't depend on capacity being exact when sizing the buffer
* Always give the PCM a period's worth of audio even when draining the buffer
* Refactoring and code cleanup
* More meaningful error messages
* Use F32 if a user requests F64 (F64 is not supported by PulseAudio)
* Move all code that can fail to `start` where errors can be returned to prevent panics
* Use drain in `stop`
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
