avoid copying data

examples/playogg_from_disk/src/main.rs

@@ -2,17 +2,16 @@ use livekit::{
     options::TrackPublishOptions,
     track::{LocalAudioTrack, LocalTrack, TrackSource},
     webrtc::{
-        audio_source::native::NativeAudioSource,
         audio_source::native::EncodedAudioSource,
+        audio_source::native::NativeAudioSource,
         prelude::{AudioFrame_u8, AudioSourceOptions, RtcAudioSource},
     },
     Room, RoomOptions,
 };
-use std::{env, sync::Arc, time::Duration, error::Error};
 use std::fs::File;
-use std::io::{BufReader};
-use webrtc_media::io::ogg_reader::{OggReader};
-
+use std::io::BufReader;
+use std::{env, error::Error, sync::Arc, time::Duration};
+use webrtc_media::io::ogg_reader::OggReader;
 
 #[tokio::main]
 async fn main() -> Result<(), Box<dyn Error>> {
@@ -40,9 +39,7 @@ async fn main() -> Result<(), Box<dyn Error>> {
         header.version,
     );
 
-    let (room, mut rx) = Room::connect(&url, &token, RoomOptions::default())
-        .await
-        .unwrap();
+    let (room, mut rx) = Room::connect(&url, &token, RoomOptions::default()).await.unwrap();
     let room = Arc::new(room);
     log::info!("Connected to room: {} - {}", room.name(), room.sid());
 
@@ -63,12 +60,9 @@ async fn main() -> Result<(), Box<dyn Error>> {
     room.local_participant()
         .publish_track(
             LocalTrack::Audio(track),
-            TrackPublishOptions {
-                source: TrackSource::Microphone,
-                ..Default::default()
-            },
-        ).await?;
-
+            TrackPublishOptions { source: TrackSource::Microphone, ..Default::default() },
+        )
+        .await?;
 
     // Play the wav file and disconnect
     tokio::spawn({
@@ -78,24 +72,14 @@ async fn main() -> Result<(), Box<dyn Error>> {
             log::info!("num_channels: {}", header.channels);
 
             while let Ok((page_data, page_header)) = reader.parse_next_page() {
-
                 let frame_size = page_data.len();
                 let mut audio_frame = AudioFrame_u8 {
-                    data: vec![0u8; 0].into(),
+                    data: page_data.freeze().into(),
                     num_channels: header.channels as u32,
                     sample_rate: header.sample_rate,
                     samples_per_channel: (frame_size / header.channels as usize) as u32,
                 };
 
-                let mut bytes = 0;
-                for d in page_data.iter() {
-                    bytes += 1;
-                    audio_frame.data.push(*d);
-                }
-
-                log::info!("samples_per_channel: {}, bytes: {}, frame_size: {}",
-                    audio_frame.samples_per_channel, bytes, frame_size);
-
                 source.capture_frame(&audio_frame).await.unwrap();
             }
 

webrtc-sys/src/audio_track.cpp

@@ -85,7 +85,7 @@ void NativeAudioSink::OnData(const void* audio_data,
                              int sample_rate,
                              size_t number_of_channels,
                              size_t number_of_frames) {
-  RTC_CHECK_EQ(16, bits_per_sample);
+  // RTC_CHECK_EQ(16, bits_per_sample);
   rust::Slice<const int16_t> data(static_cast<const int16_t*>(audio_data),
                                   number_of_channels * number_of_frames);
   observer_->on_data(data, sample_rate, number_of_channels, number_of_frames);
@@ -150,7 +150,7 @@ void AudioTrackSource::InternalSource::on_captured_frame_u8(
     size_t number_of_frames) {
   webrtc::MutexLock lock(&mutex_);
   for (auto sink : sinks_) {
-    sink->OnData(data.data(), 8, 2 * sample_rate, number_of_channels,
+    sink->OnData(data.data(), 16,  sample_rate, number_of_channels,
                  number_of_frames);
   }
 }