webrtc_srtp/stream.rs
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use tokio::sync::mpsc;
use util::marshal::*;
use util::Buffer;
use crate::error::{Error, Result};
/// Limit the buffer size to 1MB
pub const SRTP_BUFFER_SIZE: usize = 1000 * 1000;
/// Limit the buffer size to 100KB
pub const SRTCP_BUFFER_SIZE: usize = 100 * 1000;
/// Stream handles decryption for a single RTP/RTCP SSRC
#[derive(Debug)]
pub struct Stream {
ssrc: u32,
tx: mpsc::Sender<u32>,
pub(crate) buffer: Buffer,
is_rtp: bool,
}
impl Stream {
/// Create a new stream
pub fn new(ssrc: u32, tx: mpsc::Sender<u32>, is_rtp: bool) -> Self {
Stream {
ssrc,
tx,
// Create a buffer with a 1MB limit
buffer: Buffer::new(
0,
if is_rtp {
SRTP_BUFFER_SIZE
} else {
SRTCP_BUFFER_SIZE
},
),
is_rtp,
}
}
/// GetSSRC returns the SSRC we are demuxing for
pub fn get_ssrc(&self) -> u32 {
self.ssrc
}
/// Check if RTP is a stream.
pub fn is_rtp_stream(&self) -> bool {
self.is_rtp
}
/// Read reads and decrypts full RTP packet from the nextConn
pub async fn read(&self, buf: &mut [u8]) -> Result<usize> {
Ok(self.buffer.read(buf, None).await?)
}
/// ReadRTP reads and decrypts full RTP packet and its header from the nextConn
pub async fn read_rtp(&self, buf: &mut [u8]) -> Result<rtp::packet::Packet> {
if !self.is_rtp {
return Err(Error::InvalidRtpStream);
}
let n = self.buffer.read(buf, None).await?;
let mut b = &buf[..n];
let pkt = rtp::packet::Packet::unmarshal(&mut b)?;
Ok(pkt)
}
/// read_rtcp reads and decrypts full RTP packet and its header from the nextConn
pub async fn read_rtcp(
&self,
buf: &mut [u8],
) -> Result<Vec<Box<dyn rtcp::packet::Packet + Send + Sync>>> {
if self.is_rtp {
return Err(Error::InvalidRtcpStream);
}
let n = self.buffer.read(buf, None).await?;
let mut b = &buf[..n];
let pkt = rtcp::packet::unmarshal(&mut b)?;
Ok(pkt)
}
/// Close removes the ReadStream from the session and cleans up any associated state
pub async fn close(&self) -> Result<()> {
self.buffer.close().await;
let _ = self.tx.send(self.ssrc).await;
Ok(())
}
}