This change collects all those audio fetch parameters that were defined as
static constants into a dedicated struct, AudioFetchParams.
This struct can be read and set statically, allowing a user of the library to
modify those parameters without the need to recompile.
- Use variables directly in format strings.
As reported by clippy, variables can be used directly in the
`format!` string.
- Use rewind() instead of seeking to 0.
- Remove superfluous & and ref.
Signed-off-by: Petr Tesarik <petr@tesarici.cz>
- Improve responsiveness by downloading the smallest possible chunk
size when seeking or first loading.
- Improve download time and decrease CPU usage by downloading the
largest possible chunk size as throughput allows, still allowing
for reasonable seek responsiveness (~1 second).
- As a result, take refactoring opportunities: simplify prefetching
logic, download threading, command sending, and some ergonomics.
- Fix disappearing controls in the Spotify mobile UI while loading.
- Fix handling of seek, pause, and play commands while loading.
- Fix download rate calculation (don't use the Mercury rate).
- Fix ping time calculation under lock contention.
Special thanks to @eladyn for all of their help and suggestions.
* Add all player events to `player_event_handler.rs`
* Move event handler code to `player_event_handler.rs`
* Add session events
* Clean up and de-noise events and event firing
* Added metadata support via a TrackChanged event
* Add `event_handler_example.py`
* Handle invalid track start positions by just starting the track from the beginning
* Add repeat support to `spirc.rs`
* Add `disconnect`, `set_position_ms` and `set_volume` to `spirc.rs`
* Set `PlayStatus` to the correct value when Player is loading to avoid blanking out the controls when `self.play_status` is `LoadingPlay` or `LoadingPause` in `spirc.rs`
* Handle attempts to play local files better by basically ignoring attempts to load them in `handle_remote_update` in `spirc.rs`
* Add an event worker thread that runs async to the main thread(s) but sync to itself to prevent potential data races for event consumers.
* Get rid of (probably harmless) `.unwrap()` in `main.rs`
* Ensure that events are emited in a logical order and at logical times
* Handle invalid and disappearing devices better
* Ignore SpircCommands unless we're active with the exception of ShutDown
Better error handling.
Move the checking of the shell command to start so a proper error can be thrown if it's None.
Use write instead of write_all for finer grained error handling and the ability to attempt a restart on write errors.
Use try_wait to skip flushing and killing the process if it's already dead.
Stop the player on shutdown to *mostly* prevent write errors from spamming the logs during shutdown. Previously Ctrl+c always resulted in a write error.
This sets the name displayed by PulseAudio to Librespot - Instance Name if a name is given otherwise Librespot (the default name).
This also sets the correct "role" as per the docs:
https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/Developer/Clients/ApplicationProperties/
PA_PROP_MEDIA_ROLE
"This is a property of the actual streamed data, not so much the application"
Roles are used for policies, things like automatically muting a music player when a call comes in and whatnot.
For bonus points this also sets PULSE_PROP_application.icon_name to audio-x-generic so that we get a nice icon in the PulseAudio settings by our name instead of a missing icon placeholder.
This saves up to 1-2% CPU useage on a PI 4 depending on how much normalisation is actually being done.
* We don't need to test against EPSILON. The factor will never be over 1.0 in basic normalisation mode.
* Don't check the normalisation mode EVERY sample.
* Do as little math as possible by simplfiying all equations as much as possible (while still retaining the textbook equations in comments).
* Misc cleanup
* Update GStreamer backend to 0.18
* Don't manually go through all intermediate states when shutting down the GStreamer backend; that happens automatically
* Don't initialize GStreamer twice
* Use less stringly-typed API for configuring the appsrc
* Create our own main context instead of stealing the default one; if the application somewhere else uses the default main context this would otherwise fail in interesting ways
* Create GStreamer pipeline more explicitly instead of going via strings for everything
* Add an audioresample element before the sink in case the sink doesn't support the sample rate
* Remove unnecessary `as_bytes()` call
* Use a GStreamer bus sync handler instead of spawning a new thread with a mainloop; it's only used for printing errors or when the end of the stream is reached, which can also be done as well when synchronously handling messages.
* Change `expect()` calls to proper error returns wherever possible in GStreamer backend
* Store asynchronously reported error in GStreamer backend and return them on next write
* Update MSRV to 1.56
