merge main + clippy

Author: b-ma

src/lib.rs

@@ -125,22 +125,30 @@ impl AtomicF64 {
 /// voice based applications, e.g. see phone bandwidth) and 96000Hz (for very high
 /// quality audio applications and spectrum manipulation).
 /// Most common sample rates for musical applications are 44100 and 48000.
-/// - see <https://webaudio.github.io/web-audio-api/#dom-baseaudiocontext-createbuffer-samplerate>
+///
+/// - see <https://webaudio.github.io/web-audio-api/#dom-baseaudiocontext-samplerate>
+/// > An implementation MUST support sample rates in at least the range 8000 to 96000.
 ///
 /// # Panics
 ///
 /// This function will panic if:
-/// - the given sample rate is zero
+/// - the given sample rate is lower than 4000 or greater than 192000
 ///
 #[track_caller]
 #[inline(always)]
 pub(crate) fn assert_valid_sample_rate(sample_rate: f32) {
-    // 1000 Hertz is a just a random cutoff, but it helps a if someone accidentally puts a
-    // timestamp in the sample_rate variable
+    // Arbitrary cutoffs defined as:
+    // min_sample_rate = min_required_in_spec / 4
+    // max_sample_rate = max_required_in_spec * 4
+    let min_sample_rate = 2_000.;
+    let max_sample_rate = 384_000.;
+
     assert!(
-        sample_rate > 1000.,
-        "NotSupportedError - Invalid sample rate: {:?}, should be greater than 1000",
-        sample_rate
+        sample_rate >= min_sample_rate && sample_rate <= max_sample_rate,
+        "NotSupportedError - Invalid sample rate: {:?}, should be in the range [{:?}, {:?}]",
+        sample_rate,
+        min_sample_rate,
+        max_sample_rate,
     );
 }
 
@@ -228,20 +236,18 @@ mod tests {
 
     #[test]
     #[should_panic]
-    fn test_invalid_sample_rate_zero() {
-        assert_valid_sample_rate(0.);
-    }
-
-    #[test]
-    #[should_panic]
-    fn test_invalid_sample_rate_subzero() {
-        assert_valid_sample_rate(-48000.);
+    fn test_invalid_sample_rate_too_small() {
+        // invalid lower value used in wpt check
+        // <the-audio-api/the-audiocontext-interface/audiocontextoptions.html>
+        assert_valid_sample_rate(1.);
     }
 
     #[test]
     #[should_panic]
-    fn test_invalid_sample_rate_too_small() {
-        assert_valid_sample_rate(100.);
+    fn test_invalid_sample_rate_too_big() {
+        // invalid upper value used in wpt check
+        // <the-audio-api/the-audiocontext-interface/audiocontextoptions.html>
+        assert_valid_sample_rate(1_000_000.);
     }
 
     #[test]

src/node/delay.rs

@@ -209,7 +209,8 @@ impl DelayNode {
         // We add one extra buffer in the ring buffer so that reader never read the
         // same entry in history as the writer, even if `delay_time === max_delay_time`
         // cf. test_max_delay_multiple_of_quantum_size and test_max_delay
-        let num_quanta = (max_delay_time * sample_rate / RENDER_QUANTUM_SIZE as f64).ceil() as usize;
+        let num_quanta =
+            (max_delay_time * sample_rate / RENDER_QUANTUM_SIZE as f64).ceil() as usize;
         let ring_buffer = Vec::with_capacity(num_quanta + 1);
 
         let shared_ring_buffer = Rc::new(RefCell::new(ring_buffer));
@@ -665,7 +666,7 @@ mod tests {
 
     #[test]
     fn test_audioparam_value_applies_immediately() {
-        let context = OfflineAudioContext::new(1, 128, 48000.);
+        let context = OfflineAudioContext::new(1, 128, 48_000.);
         let options = DelayOptions {
             delay_time: 0.12,
             ..Default::default()
@@ -677,7 +678,7 @@ mod tests {
     #[test]
     fn test_sample_accurate() {
         for delay_in_samples in [128., 131., 197.].iter() {
-            let sample_rate = 48000.;
+            let sample_rate = 48_000.;
             let mut context = OfflineAudioContext::new(1, 256, sample_rate);
 
             let delay = context.create_delay(2.);
@@ -705,7 +706,7 @@ mod tests {
     #[test]
     fn test_sub_sample_accurate_1() {
         let delay_in_samples = 128.5;
-        let sample_rate = 48000.;
+        let sample_rate = 48_000.;
         let mut context = OfflineAudioContext::new(1, 256, sample_rate);
 
         let delay = context.create_delay(2.);
@@ -733,7 +734,7 @@ mod tests {
     #[test]
     fn test_sub_sample_accurate_2() {
         let delay_in_samples = 128.8;
-        let sample_rate = 48000.;
+        let sample_rate = 48_000.;
         let mut context = OfflineAudioContext::new(1, 256, sample_rate);
 
         let delay = context.create_delay(2.);
@@ -761,7 +762,7 @@ mod tests {
     #[test]
     fn test_multichannel() {
         let delay_in_samples = 128.;
-        let sample_rate = 48000.;
+        let sample_rate = 48_000.;
         let mut context = OfflineAudioContext::new(2, 2 * 128, sample_rate);
 
         let delay = context.create_delay(2.);
@@ -794,7 +795,7 @@ mod tests {
     #[test]
     fn test_input_number_of_channels_change() {
         let delay_in_samples = 128.;
-        let sample_rate = 48000.;
+        let sample_rate = 48_000.;
         let mut context = OfflineAudioContext::new(2, 3 * 128, sample_rate);
 
         let delay = context.create_delay(2.);
@@ -838,7 +839,7 @@ mod tests {
     fn test_node_stays_alive_long_enough() {
         // make sure there are no hidden order problem
         for _ in 0..10 {
-            let sample_rate = 48000.;
+            let sample_rate = 48_000.;
             let mut context = OfflineAudioContext::new(1, 5 * 128, sample_rate);
 
             // Set up a source that starts only after 5 render quanta.
@@ -873,7 +874,7 @@ mod tests {
     #[test]
     fn test_subquantum_delay() {
         for i in 0..128 {
-            let sample_rate = 48000.;
+            let sample_rate = 48_000.;
             let mut context = OfflineAudioContext::new(1, 128, sample_rate);
 
             let delay = context.create_delay(1.);
@@ -900,7 +901,7 @@ mod tests {
 
     #[test]
     fn test_min_delay_when_in_loop() {
-        let sample_rate = 480000.;
+        let sample_rate = 48_000.;
         let mut context = OfflineAudioContext::new(1, 256, sample_rate);
 
         let delay = context.create_delay(1.);
@@ -950,8 +951,8 @@ mod tests {
             let mut tone_buffer = context.create_buffer(1, tone_length, sample_rate);
             let tone_data = tone_buffer.get_channel_data_mut(0);
 
-            for i in 0..tone_data.len() {
-                tone_data[i] = (tone_frequency * 2.0 * PI * i as f32 / sample_rate).sin();
+            for (i, s) in tone_data.iter_mut().enumerate() {
+                *s = (tone_frequency * 2.0 * PI * i as f32 / sample_rate).sin();
             }
 
             let mut buffer_source = context.create_buffer_source();
@@ -971,14 +972,14 @@ mod tests {
             let delay_time_frames = (delay_time_seconds * sample_rate as f64) as usize;
             let tone_length_frames = (tone_length_seconds * sample_rate as f64) as usize;
 
-            for i in 0..rendered.len() {
+            for (i, s) in rendered.iter().enumerate() {
                 if i < delay_time_frames {
-                    assert_eq!(rendered[i], 0.);
+                    assert_eq!(*s, 0.);
                 } else if i >= delay_time_frames && i < delay_time_frames + tone_length_frames {
                     let j = i - delay_time_frames;
-                    assert_eq!(rendered[i], source[j]);
+                    assert_eq!(*s, source[j]);
                 } else {
-                    assert_eq!(rendered[i], 0.);
+                    assert_eq!(*s, 0.);
                 }
             }
         }
@@ -991,7 +992,7 @@ mod tests {
         // that everything works even if order of processing is not guaranteed
         // (i.e. when delay is in a loop)
         for _ in 0..10 {
-            let sample_rate = 480000.;
+            let sample_rate = 48_000.;
             let mut context = OfflineAudioContext::new(1, 256, sample_rate);
 
             // this will be internally clamped to 128 * sample_rate
@@ -1032,7 +1033,7 @@ mod tests {
     #[test]
     fn test_max_delay_multiple_of_quantum_size_1() {
         // set delay and max delay time exactly 1 render quantum
-        let sample_rate = 48000.;
+        let sample_rate = 48_000.;
         let mut context = OfflineAudioContext::new(1, 256, sample_rate);
 
         let max_delay = 128. / sample_rate;
@@ -1087,7 +1088,7 @@ mod tests {
 
     #[test]
     fn test_subquantum_delay_dynamic_lifetime() {
-        let sample_rate = 48000.;
+        let sample_rate = 48_000.;
         let mut context = OfflineAudioContext::new(1, 3 * 128, sample_rate);
 
         // Setup a source that emits for 120 frames, so it deallocates after the first render