Add the MIDI playback to Buffr Glitch

plugins/buffr_glitch/src/buffer.rs

@@ -14,18 +14,29 @@
 // You should have received a copy of the GNU General Public License
 // along with this program.  If not, see <https://www.gnu.org/licenses/>.
 
-use nih_plug::prelude::util;
+use nih_plug::prelude::*;
 
-/// A super simple ring buffer abstraction to store the last received audio. This needs to be able
-/// to store at least the number of samples that correspond to the period size of MIDI note 0.
+/// A super simple ring buffer abstraction that records audio into a recording ring buffer, and then
+/// copies audio to a playback buffer when a note is pressed so audio can be repeated while still
+/// recording audio for further key presses. This needs to be able to store at least the number of
+/// samples that correspond to the period size of MIDI note 0.
 #[derive(Debug, Default)]
 pub struct RingBuffer {
-    /// Sample buffers indexed by channel and sample index.
-    buffers: Vec<Vec<f32>>,
+    sample_rate: f32,
+
+    /// Sample ring buffers indexed by channel and sample index. These are always recorded to.
+    recording_buffers: Vec<Vec<f32>>,
     /// The positions within the sample buffers the next sample should be written to. Since all
     /// channels will be written to in lockstep we only need a single value here. It's incremented
     /// when writing a sample for the last channel.
     next_write_pos: usize,
+
+    /// When a key is pressed, audio gets copied from `recording_buffers` to these buffers so it can
+    /// be played back without interrupting the recording process. These buffers are resized to
+    /// match the length of the audio being played back.
+    playback_buffers: Vec<Vec<f32>>,
+    /// The current playback position in `playback_buffers`.
+    playback_buffer_pos: usize,
 }
 
 impl RingBuffer {
@@ -38,34 +49,93 @@ impl RingBuffer {
         let note_period_samples = (note_frequency.recip() * sample_rate).ceil() as usize;
         let buffer_len = note_period_samples.next_power_of_two();
 
-        self.buffers.resize_with(num_channels, Vec::new);
-        for buffer in self.buffers.iter_mut() {
+        // Used later to compute period sizes in samples based on frequencies
+        self.sample_rate = sample_rate;
+
+        self.recording_buffers.resize_with(num_channels, Vec::new);
+        for buffer in self.recording_buffers.iter_mut() {
             buffer.resize(buffer_len, 0.0);
         }
+
+        self.playback_buffers.resize_with(num_channels, Vec::new);
+        for buffer in self.playback_buffers.iter_mut() {
+            buffer.resize(buffer_len, 0.0);
+            // We need to reserve capacity for the playback buffers, but they're initially empty
+            buffer.resize(0, 0.0);
+        }
     }
 
     /// Zero out the buffers.
     pub fn reset(&mut self) {
-        for buffer in self.buffers.iter_mut() {
+        for buffer in self.recording_buffers.iter_mut() {
             buffer.fill(0.0);
         }
-
         self.next_write_pos = 0;
+
+        // The playback buffers don't need to be reset since they're always initialized before being
+        // used
     }
 
     /// Push a sample to the buffer. The write position is advanced whenever the last channel is
     /// written to.
     pub fn push(&mut self, channel_idx: usize, sample: f32) {
-        self.buffers[channel_idx][self.next_write_pos] = sample;
+        self.recording_buffers[channel_idx][self.next_write_pos] = sample;
 
         // TODO: This can be done more efficiently, but you really won't notice the performance
         //       impact here
-        if channel_idx == self.buffers.len() - 1 {
+        if channel_idx == self.recording_buffers.len() - 1 {
             self.next_write_pos += 1;
 
-            if self.next_write_pos == self.buffers[0].len() {
+            if self.next_write_pos == self.recording_buffers[0].len() {
                 self.next_write_pos = 0;
             }
         }
     }
+
+    /// Prepare the playback buffers to play back audio at the specified frequency. This copies
+    /// audio from the ring buffers to the playback buffers.
+    pub fn prepare_playback(&mut self, frequency: f32) {
+        let note_period_samples = (frequency.recip() * self.sample_rate).ceil() as usize;
+
+        // We'll copy the last `note_period_samples` samples from the recording ring buffers to the
+        // playback buffers
+        nih_debug_assert!(note_period_samples <= self.playback_buffers[0].capacity());
+        for (playback_buffer, recording_buffer) in self
+            .playback_buffers
+            .iter_mut()
+            .zip(self.recording_buffers.iter())
+        {
+            playback_buffer.resize(note_period_samples, 0.0);
+
+            // Keep in mind we'll need to go `note_period_samples` samples backwards in the
+            // recording buffer
+            let copy_num_from_start = usize::min(note_period_samples, self.next_write_pos);
+            let copy_num_from_end = note_period_samples - copy_num_from_start;
+            playback_buffer[0..copy_num_from_end]
+                .copy_from_slice(&recording_buffer[recording_buffer.len() - copy_num_from_end..]);
+            playback_buffer[copy_num_from_end..]
+                .copy_from_slice(&recording_buffer[..copy_num_from_start]);
+        }
+
+        // Reading from the buffer should always start at the beginning
+        self.playback_buffer_pos = 0;
+    }
+
+    /// Return a sample from the playback buffer. The playback position is advanced whenever the
+    /// last channel is written to. When the playback position reaches the end of the buffer it
+    /// wraps around.
+    pub fn next_playback_sample(&mut self, channel_idx: usize) -> f32 {
+        let sample = self.playback_buffers[channel_idx][self.playback_buffer_pos];
+
+        // TODO: Same as the above
+        if channel_idx == self.playback_buffers.len() - 1 {
+            self.playback_buffer_pos += 1;
+
+            if self.playback_buffer_pos == self.playback_buffers[0].len() {
+                self.playback_buffer_pos = 0;
+            }
+        }
+
+        sample
+    }
 }

plugins/buffr_glitch/src/lib.rs

@@ -26,8 +26,14 @@ struct BuffrGlitch {
     /// The ring buffer we'll write samples to. When a key is held down, we'll stop writing samples
     /// and instead keep reading from this buffer until the key is released.
     buffer: buffer::RingBuffer,
+
+    /// The MIDI note ID of the last note, if a note pas pressed.
+    //
+    // TODO: Add polyphony support, this is just a quick proof of concept.
+    midi_note_id: Option<u8>,
 }
 
+// TODO: Normalize option
 #[derive(Params)]
 struct BuffrGlitchParams {}
 
@@ -38,6 +44,8 @@ impl Default for BuffrGlitch {
 
             sample_rate: 1.0,
             buffer: buffer::RingBuffer::default(),
+
+            midi_note_id: None,
         }
     }
 }
@@ -88,19 +96,63 @@ impl Plugin for BuffrGlitch {
 
     fn reset(&mut self) {
         self.buffer.reset();
+        self.midi_note_id = None;
     }
 
     fn process(
         &mut self,
         buffer: &mut Buffer,
         _aux: &mut AuxiliaryBuffers,
-        _context: &mut impl ProcessContext<Self>,
+        context: &mut impl ProcessContext<Self>,
     ) -> ProcessStatus {
-        for channel_samples in buffer.iter_samples() {
+        let mut next_event = context.next_event();
+
+        for (sample_idx, channel_samples) in buffer.iter_samples().enumerate() {
+            // TODO: Split blocks based on events when adding polyphony, this is just a simple proof
+            //       of concept
+            while let Some(event) = next_event {
+                if event.timing() > sample_idx as u32 {
+                    break;
+                }
+
+                match event {
+                    NoteEvent::NoteOn { note, .. } => {
+                        // We don't keep a stack of notes right now. At some point we'll want to
+                        // make this polyphonic anyways.
+                        // TOOD: Also add an option to use velocity or poly pressure
+                        self.midi_note_id = Some(note);
+
+                        // We'll copy audio to the playback buffer to match the pitch of the note
+                        // that was just played
+                        self.buffer.prepare_playback(util::midi_note_to_freq(note));
+                    }
+                    NoteEvent::NoteOff { note, .. } if self.midi_note_id == Some(note) => {
+                        // A NoteOff for the currently playing note immediately ends playback
+                        self.midi_note_id = None;
+                    }
+                    _ => (),
+                }
+
+                next_event = context.next_event();
+            }
+
+            // When a note is being held, we'll replace the input audio with the looping contents of
+            // the playback buffer
+            if self.midi_note_id.is_some() {
+                for (channel_idx, sample) in channel_samples.into_iter().enumerate() {
+                    // New audio still needs to be recorded when the note is held to prepare for new
+                    // notes
+                    // TODO: At some point also handle polyphony here
+                    self.buffer.push(channel_idx, *sample);
+
+                    *sample = self.buffer.next_playback_sample(channel_idx);
+                }
+            } else {
                 for (channel_idx, sample) in channel_samples.into_iter().enumerate() {
                     self.buffer.push(channel_idx, *sample);
                 }
             }
+        }
 
         ProcessStatus::Normal
     }