rustrtc 0.4.1

A high-performance real-time communication library — WebRTC, RTP/SRTP, T.38 Fax, and RTP Latching
Documentation 
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#[cfg_attr(feature = "std", allow(unused_imports))]
use crate::prelude::*;
pub mod depacketizer;
pub mod error;
pub mod frame;
#[cfg(feature = "std")]
pub mod gcc;
pub mod jitter_buffer;
#[cfg(feature = "std")]
pub mod packetizer;
#[cfg(feature = "std")]
pub mod pipeline;
pub mod spsc;
pub mod track;
#[cfg(feature = "std")]
pub mod twcc_feedback;

pub use depacketizer::{
    Depacketizer, H264Depacketizer, PassThroughDepacketizer, Vp9Depacketizer, Vp9Descriptor,
    parse_vp9_descriptor,
};
pub use error::{MediaError, MediaResult};
pub use frame::{AudioFrame, MediaKind, MediaSample, VideoFrame, VideoPixelFormat};
#[cfg(feature = "std")]
pub use jitter_buffer::JitterBuffer;
#[cfg(feature = "std")]
pub use packetizer::{Packetizer, Payloader, SimplePayloader, Vp8Payloader, Vp9Payloader};
#[cfg(feature = "std")]
pub use pipeline::{
    ChannelMediaSink, ChannelMediaSource, DynMediaSink, DynMediaSource, MediaSink, MediaSource,
    TrackMediaSink, TrackMediaSource, spawn_media_pump, track_from_source,
};
pub use spsc::SpscRing;
pub use track::{
    AudioStreamTrack, MediaRelay, MediaStreamTrack, RelayStreamTrack, SampleStreamSource,
    SampleStreamTrack, TrackState, VideoStreamTrack, sample_track,
};