make start time sticky to sample time when inside floating point error zone, cf. buffer stitching wpt test

make start time sticky to sample time when inside floating point error zone
cf. buffer stitching from wpt

Author: b-ma

Cargo.toml

@@ -18,6 +18,7 @@ include = [
 rust-version = "1.71"
 
 [dependencies]
+almost = "0.2.0"
 arc-swap = "1.6"
 arrayvec = "0.7"
 cpal = { version = "0.15", optional = true }

src/node/audio_buffer_source.rs

@@ -633,12 +633,20 @@ impl AudioProcessor for AudioBufferSourceRenderer {
             for (i, playback_info) in playback_infos.iter_mut().enumerate() {
                 let current_time = block_time + i as f64 * dt;
 
+                // handle floating point errors due to start time computation
+                // cf. test_subsample_buffer_stitching
+                if !self.render_state.started {
+                    if almost::equal(current_time, self.start_time) {
+                        self.start_time = current_time;
+                    }
+                }
+
                 // Handle following cases:
                 // - we are before start time
                 // - we are after stop time
                 // - explicit duration (in buffer time reference) has been given and we have reached it
                 // Note that checking against buffer duration is done below to handle looping
-                if current_time < self.start_time
+                if (current_time < self.start_time)
                     || current_time >= self.stop_time
                     || self.render_state.buffer_time_elapsed >= self.duration
                 {
@@ -647,6 +655,7 @@ impl AudioProcessor for AudioBufferSourceRenderer {
 
                 // we have now reached start time
                 if !self.render_state.started {
+                    // println!("start, {}, {}, {}", current_time, self.start_time, (current_time - self.start_time).abs());
                     let delta = current_time - self.start_time;
                     // handle that start time may be between last sample and this one
                     self.offset += delta;
@@ -835,6 +844,7 @@ mod tests {
     use std::sync::{Arc, Mutex};
 
     use crate::context::{BaseAudioContext, OfflineAudioContext};
+    use crate::AudioBufferOptions;
     use crate::RENDER_QUANTUM_SIZE;
 
     use super::*;
@@ -1817,6 +1827,49 @@ mod tests {
         assert_float_eq!(channel[..], expected[..], abs_all <= 0.);
     }
 
+    #[test]
+    // ported from wpt: the-audiobuffersourcenode-interface/sub-sample-buffer-stitching.html
+    fn test_subsample_buffer_stitching() {
+        let sample_rate = 44_100.;
+        let buffer_rate = 44_100.;
+        let buffer_length = 30;
+        let frequency = 440.;
+
+        let length = buffer_length * 15;
+        let mut context = OfflineAudioContext::new(1, length, sample_rate);
+
+        let mut wave_signal = vec![0.; context.length()];
+        let omega = 2. * PI / buffer_rate * frequency;
+
+        wave_signal.iter_mut().enumerate().for_each(|(i, s)| {
+            *s = (omega * i as f32).sin();
+        });
+
+        // Slice the sine wave into many little buffers to be assigned to ABSNs
+        // that are started at the appropriate times to produce a final sine
+        // wave.
+        for k in (0..context.length()).step_by(buffer_length) {
+            let mut buffer = AudioBuffer::new(AudioBufferOptions {
+                number_of_channels: 1,
+                length: buffer_length,
+                sample_rate,
+            });
+            buffer.copy_to_channel(&wave_signal[k..k + buffer_length], 0);
+
+            let mut src = AudioBufferSourceNode::new(&context, AudioBufferSourceOptions {
+                buffer: Some(buffer),
+                ..Default::default()
+            });
+            src.connect(&context.destination());
+            src.start_at(k as f64 / buffer_rate as f64);
+        }
+
+        let result = context.start_rendering_sync();
+        let channel = result.get_channel_data(0);
+
+        assert_float_eq!(channel[..], wave_signal[..], abs_all <= 1e-9);
+    }
+
     #[test]
     fn test_onended_before_drop() {
         let sample_rate = 48_000.;