use audio resampler on the write_to_disk example (#68)

examples/save_to_disk/src/main.rs

@@ -2,6 +2,7 @@ use bytes::{BufMut, BytesMut};
 use futures::StreamExt;
 use livekit::prelude::*;
 use livekit::webrtc::audio_stream::native::NativeAudioStream;
+use livekit::webrtc::native::audio_resampler;
 use std::env;
 use tokio::fs::File;
 use tokio::io::{AsyncWriteExt, BufWriter};
@@ -13,7 +14,7 @@ const FILE_PATH: &str = "record.wav";
 pub struct WavHeader {
     pub sample_rate: u32,
     pub bit_depth: u16,
-    pub num_channels: u16,
+    pub num_channels: u32,
 }
 
 pub struct WavWriter {
@@ -41,9 +42,9 @@ impl WavWriter {
     }
 
     fn write_header(&mut self) -> Result<(), std::io::Error> {
-        let byte_rate = (self.header.sample_rate
+        let byte_rate = self.header.sample_rate
             * self.header.bit_depth as u32
-            * self.header.num_channels as u32);
+            * self.header.num_channels as u32;
 
         let block_align = byte_rate as u16 / self.header.sample_rate as u16;
 
@@ -53,7 +54,7 @@ impl WavWriter {
         self.data.put_slice(b"fmt ");
         self.data.put_u32_le(16); // Subchunk1Size (16 for PCM)
         self.data.put_u16_le(1); // AudioFormat (1 for PCM)
-        self.data.put_u16_le(self.header.num_channels);
+        self.data.put_u16_le(self.header.num_channels as u16);
         self.data.put_u32_le(self.header.sample_rate);
         self.data.put_u32_le(byte_rate);
         self.data.put_u16_le(block_align);
@@ -115,21 +116,30 @@ async fn record_track(audio_track: RemoteAudioTrack) -> Result<(), std::io::Erro
     println!("Recording track {:?}", audio_track.sid());
     let rtc_track = audio_track.rtc_track();
 
-    // TODO(theomonnom): Remove hardcoded values
     let header = WavHeader {
         sample_rate: 48000,
         bit_depth: 16,
-        num_channels: 1,
+        num_channels: 2,
     };
 
+    let mut resampler = audio_resampler::AudioResampler::default();
     let mut wav_writer = WavWriter::create(FILE_PATH, header).await?;
     let mut audio_stream = NativeAudioStream::new(rtc_track);
 
     let max_record = 5 * header.sample_rate * header.num_channels as u32;
     let mut sample_count = 0;
     'recv_loop: while let Some(frame) = audio_stream.next().await {
-        for sample in frame.data {
+        let data = resampler.remix_and_resample(
-            wav_writer.write_sample(sample).await.unwrap();
+            &frame.data,
+            frame.samples_per_channel,
+            frame.num_channels,
+            frame.sample_rate,
+            header.num_channels,
+            header.sample_rate,
+        );
+
+        for sample in data {
+            wav_writer.write_sample(*sample).await.unwrap();
             sample_count += 1;
 
             if sample_count >= max_record {

livekit-ffi/src/server/mod.rs

@@ -611,7 +611,7 @@ impl FfiServer {
 
     fn new_audio_resampler(
         &'static self,
-        new_resampler: proto::NewAudioResamplerRequest,
+        _: proto::NewAudioResamplerRequest,
     ) -> FfiResult<proto::NewAudioResamplerResponse> {
         let resampler = audio_resampler::AudioResampler::default();
         let resampler = Arc::new(Mutex::new(resampler));

livekit-webrtc/src/native/audio_resampler.rs

@@ -33,7 +33,7 @@ impl AudioResampler {
                 dst_sample_rate as i32,
             );
 
-            std::slice::from_raw_parts(self.sys_handle.data(), len)
+            std::slice::from_raw_parts(self.sys_handle.data(), len / 2)
         }
     }
 }

webrtc-sys/src/audio_resampler.cpp

@@ -33,7 +33,7 @@ size_t AudioResampler::remix_and_resample(const int16_t* src,
   webrtc::voe::RemixAndResample(src, samples_per_channel, num_channels,
                                 sample_rate, &resampler_, &frame_);
 
-  return frame_.num_channels() * frame_.samples_per_channel();
+  return frame_.num_channels() * frame_.samples_per_channel() * sizeof(int16_t);
 }
 
 const int16_t* AudioResampler::data() const {