Fix absn sub sample scheduling

src/node/audio_buffer_source.rs

@@ -529,6 +529,15 @@ impl AudioProcessor for AudioBufferSourceRenderer {
             self.render_state.is_aligned = false;
         }
 
+        // If some user defined end of rendering, i.e. explicit stop_time or duration,
+        // is within this render quantum force slow track as well. It might imply
+        // resampling e.g. if stop_time is between 2 samples
+        if buffer_time + block_duration > self.duration
+            || block_time + block_duration > self.stop_time
+        {
+            self.render_state.is_aligned = false;
+        }
+
         if self.render_state.is_aligned {
             // ---------------------------------------------------------------
             // Fast track
@@ -537,22 +546,9 @@ impl AudioProcessor for AudioBufferSourceRenderer {
                 self.render_state.started = true;
             }
 
-            // check if buffer ends within this block
-            if buffer_time + block_duration > buffer_duration
-                || buffer_time + block_duration > self.duration
-                || block_time + block_duration > self.stop_time
-            {
-                let end_index = if block_time + block_duration > self.stop_time
-                    || buffer_time + block_duration > self.duration
-                {
-                    let dt = (self.stop_time - block_time).min(self.duration - buffer_time);
-                    let end_buffer_time = buffer_time + dt;
-                    let end_index = (end_buffer_time * sample_rate).round() as usize;
-                    end_index.min(buffer.length())
-                } else {
-                    buffer.length()
-                };
-
+            // buffer ends within this block
+            if buffer_time + block_duration > buffer_duration {
+                let end_index = buffer.length();
                 // In case of a loop point in the middle of the block, this value will
                 // be used to recompute `buffer_time` according to the actual loop point.
                 let mut loop_point_index: Option<usize> = None;
@@ -651,8 +647,9 @@ impl AudioProcessor for AudioBufferSourceRenderer {
 
                 // we have now reached start time
                 if !self.render_state.started {
+                    let delta = current_time - self.start_time;
                     // handle that start time may be between last sample and this one
-                    self.offset += current_time - self.start_time;
+                    self.offset += delta;
 
                     if is_looping && computed_playback_rate >= 0. && self.offset >= actual_loop_end
                     {
@@ -664,7 +661,8 @@ impl AudioProcessor for AudioBufferSourceRenderer {
                         self.offset = actual_loop_start;
                     }
 
-                    buffer_time = self.offset;
+                    buffer_time = self.offset * computed_playback_rate;
+                    self.render_state.buffer_time_elapsed = delta * computed_playback_rate;
                     self.render_state.started = true;
                 }
 
@@ -676,7 +674,6 @@ impl AudioProcessor for AudioBufferSourceRenderer {
                         }
 
                         // playback began after loop, and playhead is now prior to the loop end
-                        // @note - only possible when playback_rate < 0 (?)
                         if self.offset >= actual_loop_end && buffer_time < actual_loop_end {
                             self.render_state.entered_loop = true;
                         }
@@ -1292,7 +1289,7 @@ mod tests {
     }
 
     #[test]
-    fn test_with_duration_fast_track() {
+    fn test_with_duration_0() {
         let sample_rate = 48_000.;
         let mut context = OfflineAudioContext::new(1, RENDER_QUANTUM_SIZE, sample_rate);
 
@@ -1315,7 +1312,7 @@ mod tests {
     }
 
     #[test]
-    fn test_with_duration_slow_track() {
+    fn test_with_duration_1() {
         let sample_rate = 48_000.;
         let mut context = OfflineAudioContext::new(1, RENDER_QUANTUM_SIZE, sample_rate);
 
@@ -1343,6 +1340,49 @@ mod tests {
         assert_float_eq!(channel[..], expected[..], abs_all <= 0.);
     }
 
+    #[test]
+    // port from wpt - sub-sample-scheduling.html / sub-sample-grain
+    fn test_with_duration_2() {
+        let sample_rate = 32_768.;
+        let mut context = OfflineAudioContext::new(1, RENDER_QUANTUM_SIZE, sample_rate);
+
+        let mut buffer = context.create_buffer(1, RENDER_QUANTUM_SIZE, sample_rate);
+        buffer.copy_to_channel(&[1.; RENDER_QUANTUM_SIZE], 0);
+
+        let start_grain_index = 3.1;
+        let end_grain_index = 37.2;
+
+        let mut src = context.create_buffer_source();
+        src.connect(&context.destination());
+        src.set_buffer(buffer);
+
+        src.start_at_with_offset_and_duration(
+            start_grain_index / sample_rate as f64,
+            0.,
+            (end_grain_index - start_grain_index) / sample_rate as f64,
+        );
+
+        let result = context.start_rendering_sync();
+        let channel = result.get_channel_data(0);
+
+        let mut expected = [1.; RENDER_QUANTUM_SIZE];
+        for s in expected
+            .iter_mut()
+            .take(start_grain_index.floor() as usize + 1)
+        {
+            *s = 0.;
+        }
+        for s in expected
+            .iter_mut()
+            .take(RENDER_QUANTUM_SIZE)
+            .skip(end_grain_index.ceil() as usize)
+        {
+            *s = 0.;
+        }
+
+        assert_float_eq!(channel[..], expected[..], abs_all <= 0.);
+    }
+
     #[test]
     fn test_with_offset() {
         // offset always bypass slow track