Merge pull request #317 from orottier/feature/fix-media-element

Fix broken resampling in MediaElement, support more media types

Author: orottier

Cargo.toml

@@ -16,7 +16,7 @@ rust-version = "1.70"
 arc-swap = "1.6"
 arrayvec = "0.7"
 cpal = { version = "0.15.0", optional = true }
-creek = "1.0"
+creek = "1.1"
 crossbeam-channel = "0.5"
 cubeb = { version = "0.10.0", optional = true }
 dasp_sample = "0.11"
@@ -54,12 +54,9 @@ mp3 = ["symphonia/mp3", "creek/decode-mp3"]
 ogg = ["symphonia/ogg", "symphonia/vorbis", "creek/decode-ogg", "creek/decode-vorbis"]
 flac = ["symphonia/flac", "creek/decode-flac"]
 wav = ["symphonia/wav", "symphonia/pcm", "creek/decode-wav", "creek/decode-pcm"]
-# TODO(m4a): Add "creek/decode-aac" after <https://github.com/MeadowlarkDAW/creek/pull/22> has been merged
-aac = ["symphonia/aac"]
-# TODO(m4a): Add "creek/decode-isomp4" after <https://github.com/MeadowlarkDAW/creek/pull/22> has been merged
-m4a = ["aac", "symphonia/isomp4"]
-# TODO(alac): Add "creek/decode-alac" and "creek/decode-isomp4" after <https://github.com/MeadowlarkDAW/creek/pull/22> has been merged
-alac = ["symphonia/alac", "symphonia/isomp4"]
+aac = ["symphonia/aac", "creek/decode-aac"]
+m4a = ["aac", "symphonia/isomp4", "creek/decode-isomp4"]
+alac = ["symphonia/alac", "symphonia/isomp4", "creek/decode-alac", "creek/decode-isomp4"]
 cpal = ["dep:cpal"]
 cubeb = ["dep:cubeb"]
 cpal-jack = ["cpal", "cpal/jack"]

src/media_element.rs

@@ -11,6 +11,7 @@ use crate::{AtomicF64, AudioBuffer, RENDER_QUANTUM_SIZE};
 /// Real time safe audio stream
 pub(crate) struct RTSStream {
     stream: ReadDiskStream<SymphoniaDecoder>,
+    number_of_channels: usize,
     current_time: Arc<AtomicF64>,
     receiver: Receiver<MediaElementAction>,
     loop_: Arc<AtomicBool>,
@@ -54,6 +55,7 @@ impl MediaElement {
             0,                  // The frame in the file to start reading from.
             Default::default(), // Use default read stream options.
         )?;
+        let number_of_channels = read_disk_stream.info().num_channels as usize;
 
         // Cache the start of the file into cache with index `0`.
         let _ = read_disk_stream.cache(0, 0);
@@ -79,6 +81,7 @@ impl MediaElement {
 
         let rts_stream = RTSStream {
             stream: read_disk_stream,
+            number_of_channels,
             current_time: Arc::clone(&current_time),
             receiver,
             loop_: Arc::clone(&loop_),
@@ -161,7 +164,10 @@ impl Iterator for RTSStream {
         }
 
         if self.paused.load(Ordering::SeqCst) {
-            let silence = AudioBuffer::from(vec![vec![0.; RENDER_QUANTUM_SIZE]], sample_rate);
+            let silence = AudioBuffer::from(
+                vec![vec![0.; RENDER_QUANTUM_SIZE]; self.number_of_channels],
+                sample_rate,
+            );
             return Some(Ok(silence));
         }
 

tests/media_element.rs

@@ -0,0 +1,27 @@
+use web_audio_api::context::{AudioContext, AudioContextOptions, BaseAudioContext};
+use web_audio_api::node::AudioNode;
+use web_audio_api::MediaElement;
+
+#[test]
+fn test_media_element_source_progress() {
+    let options = AudioContextOptions {
+        sink_id: "none".into(),
+        ..AudioContextOptions::default()
+    };
+    let context = AudioContext::new(options);
+
+    let mut media = MediaElement::new("samples/major-scale.ogg").unwrap();
+    media.set_loop(true);
+
+    let src = context.create_media_element_source(&mut media);
+    src.connect(&context.destination());
+    media.play();
+
+    // TODO improve test setup of online AudioContext
+    // <https://github.com/orottier/web-audio-api-rs/issues/323>
+    // Sleep for a bit, make sure the audio thread has started
+    std::thread::sleep(std::time::Duration::from_millis(100));
+
+    // assert the media has progressed
+    assert!(media.current_time() > 0.05);
+}