fix audio playback of 8Khz samples, further cut down on clones/copies

src/audio.rs

@@ -156,18 +156,16 @@ impl AudioSink {
     }
 
     fn process(&mut self, scope: &ProcessScope) -> Result<(), AudioError> {
-        let mut next_outbuf = match self.sample_src.try_recv() {
+        if let Ok(buf) = self.sample_src.try_recv() {
-            Ok(buf) => buf,
+            self.output_buf.extend(buf);
-            Err(tokio::sync::mpsc::error::TryRecvError::Empty) => return Ok(()),
+        }
-            Err(err) => return Err(err.into())
-        };
-        self.output_buf.append(&mut next_outbuf);
 
         if self.port.connected_count()? > 0 && !self.output_buf.is_empty() {
             let outbuf = self.port.as_mut_slice(scope);
-            let mut next_segment: Vec<f32> = self.output_buf.drain(0..(outbuf.len()).min(self.output_buf.len())).collect();
+            let mut next_segment: Vec<f32> = self.output_buf.drain(..(outbuf.len()).min(self.output_buf.len())).collect();
             let underrun = outbuf.len() - next_segment.len();
             if underrun > 0 {
+                log::warn!("Audio stream underrun: {} samples", underrun);
                 next_segment.extend(std::iter::repeat_n(0., underrun));
             }
 

src/sfx.rs

@@ -41,13 +41,23 @@ pub async fn start_sfx(audio_sink: AudioOutStream) -> SfxControl {
                             .probe(&hint, mss, fmt_opts, meta_opts)
                             .expect("Unsupported audio format");
                         let track = format.default_track(TrackType::Audio).expect("No audio track");
+                        let track_id = track.id;
                         let dec_opts = Default::default();
                         let mut decoder = symphonia::default::get_codecs()
                             .make_audio_decoder(
                                 track.codec_params.as_ref().expect("codec params missing").audio().unwrap(),
                                 &dec_opts
                             ).expect("Unsupported audio codec");
+
+                        let sample_rate = decoder.codec_params().sample_rate.unwrap();
+                        let channel_num = decoder.codec_params().channels.as_ref().unwrap().count();
+                        log::debug!("Resampling {} -> {}", sample_rate, audio_sink.sample_rate);
+                        // Our resampler works on a mono input
+                        let mut bitrate_resample = resampler::ResamplerFir::new_from_hz(channel_num, sample_rate, audio_sink.sample_rate, Default::default(), Default::default());
+
                         log::debug!("Starting stream");
+                        let mut audio_out_buf = vec![];
+                        let mut channel_bufs: Vec<f32> = vec![];
                         loop {
                             let packet = match format.next_packet() {
                                 Ok(Some(packet)) => packet,
@@ -55,19 +65,40 @@ pub async fn start_sfx(audio_sink: AudioOutStream) -> SfxControl {
                                 Err(err) => panic!()
                             };
 
+                            if packet.track_id != track_id {
+                                continue
+                            }
+
                             match decoder.decode_ref(&packet.as_packet_ref()) {
                                 Ok(samples) => {
-                                    let mut channel_bufs: Vec<f32> = vec![];
+                                    channel_bufs.resize(samples.samples_interleaved(), 0.);
-                                    samples.copy_to_vec_interleaved(&mut channel_bufs);
+                                    samples.copy_to_slice_interleaved(&mut channel_bufs);
-                                    let audio_out_buf: Vec<f32> = channel_bufs.windows(samples.byte_len_per_plane()).map(|channels| {
+
+                                    let mut resampled = [0.; 2048];
+                                    let (_, write_count) = bitrate_resample.resample(&channel_bufs, &mut resampled).unwrap();
+
+                                    // First we convert the audio feed from stereo down to mono by simple average
+                                    // TODO: This should be something smarter, like a saturating add..?
+                                    let mono_stream = resampled[..write_count].chunks(channel_num).map(|channels| {
                                         let total_volume = channels.iter().cloned().reduce(|a, b| a + b).unwrap_or_default();
-                                        total_volume / (channels.len() as f32)
+                                        total_volume / (channel_num as f32)
-                                    }).collect();
+                                    });
-                                    audio_sink.sink.send(audio_out_buf).await.unwrap();
+
+                                    // Then we write out the resampled audio to our staging buffer
+                                    audio_out_buf.extend(mono_stream);
+
+                                    // Once we have 1024 samples (jack default, I guess), we send it to the audio output
+                                    if audio_out_buf.len() >= 1024 {
+                                        audio_sink.sink.send(audio_out_buf).await.unwrap();
+                                        audio_out_buf = vec![];
+                                    }
                                 },
                                 Err(err) => panic!()
                             }
                         }
+                        if !audio_out_buf.is_empty() {
+                            audio_sink.sink.send(audio_out_buf).await.unwrap();
+                        }
                         log::debug!("Playback complete");
                     }
                 }