- 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>
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.
* 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
- Switch from `lewton` to `Symphonia`. This is a pure Rust demuxer
and decoder in active development that supports a wide range of
formats, including Ogg Vorbis, MP3, AAC and FLAC for future HiFi
support. At the moment only Ogg Vorbis and MP3 are enabled; all
AAC files are DRM-protected.
- Bump MSRV to 1.51, required for `Symphonia`.
- Filter out all files whose format is not specified.
- Not all episodes seem to be encrypted. If we can't get an audio
key, try and see if we can play the file without decryption.
- After seeking, report the actual position instead of the target.
- Remove the 0xa7 bytes offset from `Subfile`, `Symphonia` does
not balk at Spotify's custom Ogg packet before it. This also
simplifies handling of formats other than Ogg Vorbis.
- When there is no next track to load, signal the UI that the
player has stopped. Before, the player would get stuck in an
infinite reloading loop when there was only one track in the
queue and that track could not be loaded.
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.
* 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`
