Fixes a bug where right-clicking on a sample in the sidebar and clicking "Send to new SlicerT instance" under the Pattern Editor section would instead create the instance in the Song Editor.
This PR makes it so that the scale/key/chord in the Piano Roll is saved with the project.
Also, while I was at it, I made it so that the zoom, vertical zoom, snap settings, quantization, and note lengths are saved too.
Aaand I also decided to save the zoom and snap size in the Song Editor because why not :D
However, the zoom/snap/etc are saved in the lmmsrc file, since it doesn't really make sense to save them per-project.
Makes the following changes to the Lb302 plugin for increased functionality, maintainability, and readability:
- Add note velocity & note panning to Lb302
- Remove evil mutex+dynamic array combo in favor of a real-time safe ringbuffer queue
- Use non-static data member initializers where possible
- Use `std::numbers::pi` instead of `M_PI`
- Use `std::array` and `std::unique_ptr` instead of manual memory management
- Change plain-old-data classes with only public members to structs
- Change some ints to f_cnt_t to better communicate their purpose and avoid weird implicit casting nonsense
- Change floats to sample_t where appropriate to better communicate their purpose
- Find suitable homes for loose constants
- Prefix standard library math function calls with std::
- Ensure all floating-point literals intended to be floats end in 'f' and are not doubles to appease MSVC
- Change uses of SIGNAL and SLOT macros to whatever they should be post-Qt6 upgrade
- Tidy code to conform to current code conventions
Also adds Hardware.h for low-level performance tools:
- busyWaitHint(), supporting x86, ARM, and RISC-V
- hardware_destructive_interference_size polyfill, supporting x86, ARM, RISC-V, and PPC
- Moved the contents of denormals.h to Hardware.h, renamed disable_denormals() to disableDenormals()
- Added ARM support to disableDenormals()
* Adds a new native plugin for incoming waveform display
* Features the ability to pause, zoom in, set window size and set amplification (scale)
* Does not feature a pitch-tracking option
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Co-authored-by: Fawn <rubiefawn@gmail.com>
Co-authored-by: Sotonye Atemie <satemiej@gmail.com>
Co-authored-by: Dalton Messmer <messmer.dalton@gmail.com>
Co-authored-by: bratpeki <pkatic2003@gmail.com>
Removes construction of the unused default volume model from `SamplePlayHandle`. Frequent construction of this model was causing performance slowdowns due to the model having to find and create a journaling ID in a for loop inside `ProjectJournal::allocID`, leading to the exhaustion of available IDs very quickly. There were also general improvements made to `SamplePlayHandle` (using NSDMI, removing other unused members, etc).
Adds a check in finishRecordNote that verifies no notes of length 0 can be made, which was possible before.
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Co-authored-by: bratpeki <pkatic2003@gmail.com>
Migrates all MinGW 3rd-party dependencies to vcpkg except for Qt and libgig.
This enables the following for MinGW builds
- mp3/opus support
- per-note panning in Sf2 Player
- DirectSound and WASAPI audio backends in PortAudio
- LV2 plugin support
Opening the Settings would unconditionally try to start the JACK server
if it was not already running, even if the user was using an audio
backend other than JACK. When it failed to start the JACK server, it
would retry a few times which made opening the Settings take several
seconds. For me, it also broke my whole system's audio, requiring me
to restart pipewire.
This PR fixes that problem by passing the JackNoStartServer option to
jack_client_open() in the setupWidget used by the Settings.
AudioJack::initJackClient() continues to use the JackNullOption option,
so like before, it will still attempt to start the JACK server if you select
the JACK backend and restart LMMS.
I also moved some static free functions into a nearby anonymous
namespace, and I removed the server name argument from the
jack_client_open() calls since that variadic argument is only supposed
to be passed if the JackServerName option is passed.
Improves the tap tempo algorithm and usage, as well refactoring the code for better maintainability.
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Co-authored-by: Dalton Messmer <messmer.dalton@gmail.com>
This reworks the auto-quit feature by introducing a new AudioBuffer class which keeps track of which channels are currently silent as audio flows through the effects chain.
When track channels going into an effect's input are not marked as quiet, it is assumed a signal is present and the plugin needs to wake up if it is asleep due to auto-quit. After a plugin processes a buffer, the silence status is updated.
When the auto-quit setting is disabled (that is, when effects are always kept running), effects are always assumed to have input noise (a non-quiet signal present at the plugin inputs), which should result in the same behavior as before.
Benefits:
- The auto-quit system now closely follows how it is supposed to function by only waking plugins which have non-zero input rather than waking all plugins at once whenever an instrument plays a note or a sample track plays. This granularity better fits multi-channel plugins and pin connector routing where not all plugin inputs are connected to the same track channels. This means a sleeping plugin whose inputs are connected to channels 3/4 would not need to wake up if a signal is only present on channels 1/2.
- Silencing channels that are already known to be silent is a no-op
- Calculating the absolute peak sample value for a channel already known to be silent is a no-op
- The silence flags also could be useful for other purposes, such as adding visual indicators to represent how audio signals flow in and out of each plugin
- With a little more work, auto-quit could be enabled/disabled for plugins on an individual basis
- With a little more work, auto-quit could be implemented for instrument plugins
- AudioBuffer can be used with SharedMemory
- AudioBuffer could be used in plugins for their buffers
This new system works so long as the silence flags for each channel remain valid at each point along the effect chain. Modifying the buffers without an accompanying update of the silence flags could violate assumptions. Through unit tests, the correct functioning of AudioBuffer itself can be validated, but its usage in AudioBusHandle, Mixer, and a few other places where track channels are handled will need to be done with care.
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Co-authored-by: Sotonye Atemie <sakertooth@gmail.com>
Allows detaching a window from LMMS's main window, making things like working on multiple screens easier.
The behavior of detached windows can be customized in the Settings.
Closes#1259
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Signed-off-by: Dalton Messmer <messmer.dalton@gmail.com>
Co-authored-by: Hyunjin Song <tteu.ingog@gmail.com>
Co-authored-by: Dalton Messmer <messmer.dalton@gmail.com>
Co-authored-by: SpomJ <mihail_a_m@mail.ru>
Improves performance when moving mixer channels to the left or right using simple Qt layout operations and swapping of values. This commit also addresses a deadlock in Mixer::mixToChannel by ensuring we lock and unlock the same channel regardless of any move operations occurring simultaneously on another thread.
When #7454 was merged, Timelines were refactored to emit a positionJumped signal whenever their internal tick position was changed via Timeline::setTicks. This was meant as an easy way to account for all situations where the user might drag or click or somehow forcefully set the position of a timeline mid-playback.
However, the current way timeline syncing is implemented between the song editor and piano roll (when doing record-play in the piano roll) is by constantly calling setTicks on the piano roll's timeline whenever the song editor's timeline moves. This inadvertantly means that positionJumped is being emitted, even when the timeline is tracking smoothly.
This PR attempts to address this issue by adding an optional parameter to Timeline::setTicks which allows the caller to opt out of emitting positionJumped. Ideally, we would have a better system for syncing timelines together, one which does not rely on forcefully calling setTicks. But for now, this should work.
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Co-authored-by: Sotonye Atemie <satemiej@gmail.com>
Requested by a user on the forum. Sometimes, it can get tricky when two clips are right next to each other, and you're trying to resize one of them. Either you resize the right one, or the other one starts resizing behind the other, so you end up having to move one over, then resize it, then move it back. And it's just not awesome.
However, this can be mitigated by simply increasing the clip resize grip width. Currently, it is at 4 px, but increasing it to 8 px makes it easier to specifically target which clip you want to resize. It also probably makes it easier to resize clips in general for users who may struggle to aim the mouse perfectly within that 4 pixel gap.
Partially addresses #8188
For some reason, when a knob is set to be a "volume knob", the max volume allowed via double-clicking and typing in a value is 6.0 dB. For knobs such as the amplifier knob in Audio File Processor which go up to 500% (~14 db), this is not enough.
This PR makes it so that the dialog shows the actual max dB value based on the underlying model's max value, instead of being hardcoded to 6.0 dB. This PR does not address the minimum value.
* Clarify VST support to VST2 in README
Updated VST support mention from VST(i) to VST2 in features section for clarity.
* Update README.md
Co-authored-by: Dalton Messmer <messmer.dalton@gmail.com>
* mention LV2 support
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Co-authored-by: Dalton Messmer <messmer.dalton@gmail.com>
Refactors the PortAudio backend to fix issues with DirectSound and MME crackling and not loading properly, as well as to improve code quality and maintainability.
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Co-authored-by: Dalton Messmer <messmer.dalton@gmail.com>
Fixes#8200. Scrolling to change velocity/panning can affect unselected notes if hovering over them in the bottom section of the piano roll. This PR addresses that.
This refactors the export dialog to no longer use the export_project.ui file and moves it into standard C++ Qt code. This also brings minor changes to the dialog, such as horizontal labels instead of vertical ones.
* reimplement getter and setter of FloatModelEditorBase::m_volumeKnob to avoid undo checkpoint creation
* convert FloatModelEditorBase::m_volumeKnob from BoolModel to bool
* clean up
* remove unnecessary inline keywords and fix formatting
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Co-authored-by: Sotonye Atemie <sakertooth@gmail.com>
Fixes#8190
When #7454 was merged, I added a check before emitting playbackPositionJumped to only emit when the song was playing, to prevent LFOs from being reset when the user dragged the timeline while the song was paused. However, I used Song::isPlaying(), which only returns true when the song is not exporting. This caused sample clips to not be updated when the playback position jumped back every loop, which means they just kept playing, ignoring the loop. To fix this, I have changed it to use m_playing, which is true for both exporting and normal playing.
Removes `SampleLoader`. File dialog functions were moved into `FileDialog`. Creation functions were moved into `SampleBuffer`.
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Co-authored-by: Dalton Messmer <messmer.dalton@gmail.com>
This PR modifies how the Detuning tool works in the piano roll to let the user modify the automation curve right on the notes. This is actually fairly simple, as automation clips already contain the functions for dragging and removing points like in the Automation Editor. So essentially all that's being done is calculating the relative pos of the mouse to the nearest note, and setting the drag value in the automation clip to that position and key.
You can still access the old automation editor version by shift-click
messmerd asked if the code could be made general for any future note parameter besides detuning, in preparation for CLAP per-note parameter automation. I have refactored the relevant functions to accept an enum type for the kind of note parameter, although currently only detuning is supported.
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Co-authored-by: Sotonye Atemie <sakertooth@gmail.com>
Co-authored-by: Sotonye Atemie <satemiej@gmail.com>
Co-authored-by: szeli1 <143485814+szeli1@users.noreply.github.com>