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// Copyright 2025 LiveKit, Inc.
//
// Licensed under the Apache License, Version 2.0 (the "License");
// you may not use this file except in compliance with the License.
// You may obtain a copy of the License at
//
// http://www.apache.org/licenses/LICENSE-2.0
//
// Unless required by applicable law or agreed to in writing, software
// distributed under the License is distributed on an "AS IS" BASIS,
// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
// See the License for the specific language governing permissions and
// limitations under the License.
use crate::imp::audio_stream as stream_imp;
#[cfg(not(target_arch = "wasm32"))]
pub mod native {
use std::{
fmt::{Debug, Formatter},
pin::Pin,
task::{Context, Poll},
};
use livekit_runtime::Stream;
use super::stream_imp;
use crate::{audio_frame::AudioFrame, audio_track::RtcAudioTrack};
const DEFAULT_QUEUE_SIZE_FRAMES: usize = 10;
#[derive(Clone, Debug, Default)]
pub struct NativeAudioStreamOptions {
/// Maximum number of queued WebRTC sink frames after the audio callback.
///
/// Each queued frame corresponds to roughly 10 ms of decoded PCM audio
/// on the WebRTC sink path.
///
/// `None` uses the default bounded queue size of 10 frames. `Some(0)`
/// opts into unbounded buffering. Positive values bound the queue, and
/// the stream drops the oldest queued frames on overflow so latency
/// stays bounded.
///
/// If your application consumes both audio and video, keep the queue
/// sizing strategy coordinated across both streams. Using a much larger
/// queue, or unbounded buffering, for only one of them can increase
/// end-to-end latency for that stream and cause audio/video drift.
pub queue_size_frames: Option<usize>,
}
pub struct NativeAudioStream {
pub(crate) handle: stream_imp::NativeAudioStream,
}
impl Debug for NativeAudioStream {
fn fmt(&self, f: &mut Formatter) -> std::fmt::Result {
f.debug_struct("NativeAudioStream").field("track", &self.track()).finish()
}
}
impl NativeAudioStream {
pub fn new(audio_track: RtcAudioTrack, sample_rate: i32, num_channels: i32) -> Self {
Self {
handle: stream_imp::NativeAudioStream::new(
audio_track,
sample_rate,
num_channels,
Some(DEFAULT_QUEUE_SIZE_FRAMES),
),
}
}
pub fn with_options(
audio_track: RtcAudioTrack,
sample_rate: i32,
num_channels: i32,
options: NativeAudioStreamOptions,
) -> Self {
Self {
handle: stream_imp::NativeAudioStream::new(
audio_track,
sample_rate,
num_channels,
normalize_queue_size_frames(options.queue_size_frames),
),
}
}
pub fn track(&self) -> RtcAudioTrack {
self.handle.track()
}
pub fn close(&mut self) {
self.handle.close()
}
}
impl Stream for NativeAudioStream {
type Item = AudioFrame<'static>;
fn poll_next(self: Pin<&mut Self>, cx: &mut Context) -> Poll<Option<Self::Item>> {
Pin::new(&mut self.get_mut().handle).poll_next(cx)
}
}
fn normalize_queue_size_frames(queue_size_frames: Option<usize>) -> Option<usize> {
match queue_size_frames {
None => Some(DEFAULT_QUEUE_SIZE_FRAMES),
Some(0) => None,
Some(value) => Some(value),
}
}
}