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
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
(needed to be done together)
there was no changelog for gstreamer and far too many commits to check
compatibility, but compiling with the gstreamer backend feature works fine
with these new versions.
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.
* Enable gapless playback via runtime flag
* Set gapless playback as default, use `--disable-gapless` to turn it off
* Ensure sink restarts b/w tracks when gapless is disabled
Ensure the player events are emitted correctly.
Only call the external script on events we want to notify about.
Stop sink when loading to pause.
cargo fmt
- change communication between player and spirc to use player events channel.
- enhance player events channel
- have spirc send loading messages to Spotify
- enable preloading of tracks in the player
