XXX WIP

Author: phip1611

src/audio_history.rs

@@ -1,20 +1,18 @@
+use crate::envelope_iterator::ENVELOPE_MIN_DURATION_MS;
 use core::cmp::Ordering;
 use core::time::Duration;
 use ringbuffer::{ConstGenericRingBuffer, RingBuffer};
 
-/// Default buffer size for [`AudioHistory`]. For a typical (and to be expected)
-/// sampling frequency of 44100Hz, this corresponds to 1000ms which is more than
-/// enough to detect beats that are typically at 60Hz, with a period duration
-/// of 20ms (or 20Hz at 50ms). This results in roughly 35kib of stack usage
-/// for the audio buffer.
-///
-/// TODO I plan to reduce this to 500, 300, or even 200ms to reduce memory usage.
-/// However, during development, a large ubffer size is necessary to tune the
-/// algorithm in a way that it reliably works for data analysis when the full
-/// sample is in memory. If that works reliably, I can test the library in a
-/// "data streaming approach" and discover beats one by one. However, for now,
-/// I keep it very basic and easy.
-pub const DEFAULT_BUFFER_SIZE: usize = 44100;
+const SAFE_MIN_DURATION_MS: usize = (ENVELOPE_MIN_DURATION_MS as f64 * 2.5) as usize;
+
+/// Based on the de-facto default sampling rate of 44100 Hz / 44.1 kHz.
+const DEFAULT_SAMPLES_PER_SECOND: usize = 44100;
+const MS_PER_SECOND: usize = 1000;
+
+/// Default buffer size for [`AudioHistory`]. The size is a trade-off between
+/// memory efficiency and effectiveness in detecting envelops properly.
+pub const DEFAULT_BUFFER_SIZE: usize =
+    (SAFE_MIN_DURATION_MS * DEFAULT_SAMPLES_PER_SECOND) / MS_PER_SECOND;
 
 /// Sample info with time context.
 #[derive(Copy, Clone, Debug)]
@@ -87,6 +85,7 @@ impl AudioHistory {
 
     /// Update the audio history with fresh samples. The audio samples are
     /// expected to be in mono channel, i.e., no stereo interleaving
+    /// TODO: Update consume iterator and consume enum: Interleaved or Mono
     pub fn update(&mut self, mono_samples: &[f32]) {
         if mono_samples.len() >= self.audio_buffer.capacity() {
             log::warn!(
@@ -183,9 +182,18 @@ impl AudioHistory {
 #[cfg(test)]
 mod tests {
     use super::*;
-    use crate::test_util::{sample_1_test_data, single_beat_test_data};
+    use crate::test_util::{sample_1_test_data};
     use std::prelude::v1::Vec;
 
+    #[test]
+    fn buffer_len_sane() {
+        let sampling_rate = 1.0 / DEFAULT_SAMPLES_PER_SECOND as f32;
+        let duration = Duration::from_secs_f32(sampling_rate * DEFAULT_BUFFER_SIZE as f32);
+        dbg!(duration);
+        assert!(duration.as_millis() > 10);
+        assert!(duration.as_millis() <= 1000);
+    }
+
     #[test]
     fn audio_duration_is_updated_properly() {
         let mut hist = AudioHistory::new(2.0);
@@ -277,18 +285,37 @@ mod tests {
         let mut hist = AudioHistory::new(1.0);
 
         hist.update(&[0.0]);
-        assert_eq!(hist.index_to_sample_info(0).duration_behind, Duration::from_secs(0));
+        assert_eq!(
+            hist.index_to_sample_info(0).duration_behind,
+            Duration::from_secs(0)
+        );
         hist.update(&[0.0]);
-        assert_eq!(hist.index_to_sample_info(0).duration_behind, Duration::from_secs(1));
-        assert_eq!(hist.index_to_sample_info(1).duration_behind, Duration::from_secs(0));
-
+        assert_eq!(
+            hist.index_to_sample_info(0).duration_behind,
+            Duration::from_secs(1)
+        );
+        assert_eq!(
+            hist.index_to_sample_info(1).duration_behind,
+            Duration::from_secs(0)
+        );
 
         hist.update(&[0.0].repeat(hist.data().capacity() * 2));
 
         let sample = hist.index_to_sample_info(0);
-        assert_eq!(hist.index_to_sample_info(0).duration_behind, Duration::from_secs_f32((DEFAULT_BUFFER_SIZE - 1) as f32));
-        assert_eq!(hist.index_to_sample_info(DEFAULT_BUFFER_SIZE - 10).duration_behind, Duration::from_secs_f32(9.0));
-        assert_eq!(hist.index_to_sample_info(DEFAULT_BUFFER_SIZE - 1).duration_behind, Duration::from_secs(0));
+        assert_eq!(
+            hist.index_to_sample_info(0).duration_behind,
+            Duration::from_secs_f32((DEFAULT_BUFFER_SIZE - 1) as f32)
+        );
+        assert_eq!(
+            hist.index_to_sample_info(DEFAULT_BUFFER_SIZE - 10)
+                .duration_behind,
+            Duration::from_secs_f32(9.0)
+        );
+        assert_eq!(
+            hist.index_to_sample_info(DEFAULT_BUFFER_SIZE - 1)
+                .duration_behind,
+            Duration::from_secs(0)
+        );
     }
 
     #[test]

src/audio_input.rs

@@ -0,0 +1,23 @@
+pub enum AudioInput<'a> {
+    /// The audio input stream only consists of mono samples.
+    Mono(&'a [f32]),
+    /// The audio input streams consists of interleaved samples following a
+    /// LRLRLR scheme. This is typically the case for stereo channel audio.
+    InterleavedLR(&'a [f32]),
+}
+
+impl<'a> AudioInput<'a> {
+    pub fn iter(&self) -> impl Iterator<Item = f32> {
+        match self {
+            AudioInput::Mono(samples) => samples.iter(),
+            AudioInput::InterleavedLR(samples) => samples.chunks(2).map(|lr| (lr[0] + lr[1]) / 2.0)
+        }
+    }
+}
+
+#[cfg(test)]
+mod tests {
+
+    fn stereo_to:
+
+}

src/envelope_iterator.rs

@@ -8,7 +8,13 @@ const ENVELOPE_MIN_VALUE: f32 = 0.2;
 /// The factor by that a peak must be higher than the begin and end so that we
 /// in fact found an envelope that represents a beat.
 const ENVELOPE_PEAK_RATIO: f32 = 2.5;
-const ENVELOPE_MIN_DURATION: Duration = Duration::from_millis(100);
+
+pub(crate) const ENVELOPE_MIN_DURATION_MS: u64 = 160;
+
+/// Minimum realistic duration of an envelope. This value is the result of
+/// analyzing some waveforms in Audacity. Specifically, this results from an
+/// envelope of two beats very close to each other.
+const ENVELOPE_MIN_DURATION: Duration = Duration::from_millis(ENVELOPE_MIN_DURATION_MS);
 
 /// Iterates the envelopes of an audio signal. An envelope is the set of
 /// vibrations(? - german: Schwingungen) that characterize a beat. Its waveform
@@ -164,7 +170,6 @@ impl PartialEq for EnvelopeInfo {
 mod tests {
     use super::*;
     use crate::test_util::*;
-    use std::collections::{BTreeMap, BTreeSet};
     use std::vec::Vec;
 
     #[test]
@@ -244,12 +249,12 @@ mod tests {
         let mut hist = AudioHistory::new(header.sampling_rate as f32);
 
         // 5ms at 44.1kHz - realistic value for an audio input device
-        let mut begin_index = None;
+        /*let mut begin_index = None;
         for chunk in vec.chunks(250) {
             hist.update(chunk);
             let iter = EnvelopeIterator::new(&hist, begin_index);
             let maybe_envelop = iter.next();
-        }
+        }*/
         /*.flat_map(|chunks| {
             hist.update(chunks);
             let iter = EnvelopeIterator::new(&hist, None);

src/lib.rs

@@ -83,13 +83,14 @@ SOFTWARE.
 extern crate std;
 
 mod audio_history;
-mod root_iterator;
 mod envelope_iterator;
 mod max_min_iterator;
+mod root_iterator;
 
 // Todo export function to combine large audio stream with my code lololo
 
+mod sample;
 /// PRIVATE. For tests and helper binaries.
 #[cfg(test)]
 mod test_util;
-
+mod audio_input;

src/max_min_iterator.rs

@@ -84,7 +84,7 @@ mod tests {
             // I checked in Audacity whether the values returned by the code
             // make sense. Then, they became the reference for the test.
             [
-                (0, -0.07258828),
+                (0, -0.07258828), // TODO should not have zero here?!
                 (71, 0.5884732),
                 (323, -0.7123936),
                 (608, 0.599353),

src/root_iterator.rs

@@ -51,40 +51,25 @@ mod tests {
     use crate::test_util::*;
     use std::vec::Vec;
 
-    #[test]
-    fn find_roots_simple() {
-        let history = simple_test_data();
-        let iter = RootIterator::new(&history, None);
-        #[rustfmt::skip]
-        assert_eq!(
-            iter.map(|info| (info.value, info.index)).collect::<Vec<_>>(),
-            [
-                (-0.5, 5),
-                (0.5, 9),
-                (-0.5, 13),
-            ]
-        );
-    }
-
     #[test]
     fn find_roots_in_real_sample() {
         let history = single_beat_excerpt_test_data();
         let iter = RootIterator::new(&history, None);
         #[rustfmt::skip]
         assert_eq!(
-            iter.map(|info| (info.timestamp.as_secs_f32(), info.value)).collect::<Vec<_>>(),
+            iter.map(|info| (info.total_index, info.value)).collect::<Vec<_>>(),
             // I checked in Audacity whether the values returned by the code
             // make sense. Then, they became the reference for the test.
             [
-                (0.000181406, -0.002929777),
-                (0.004444445, 0.005798517),
-                (0.009818594, -0.004074221),
-                (0.017823128, 0.0008850368),
-                (0.025170067, -0.0017700735),
-                (0.032743763, 0.004379406),
-                (0.04111111, -0.0031891842),
-                (0.04918367, 0.0013428144),
-                (0.05732426, -0.0026245918)
+                (8, -0.002929777),
+                (196, 0.005798517),
+                (433, -0.004074221),
+                (786, 0.0008850368),
+                (1110, -0.0017700735),
+                (1444, 0.004379406),
+                (1813, -0.0031891842),
+                (2169, 0.0013428144),
+                (2528, -0.0026245918)
             ]
         );
     }