Merge pull request webrtc-rs#3 from webrtc-rs/v0.3.0

pull v0.3.0beta to main

Cargo.toml

@@ -1,6 +1,6 @@
 [package]
 name = "examples"
-version = "0.2.1"
+version = "0.3.0"
 authors = ["Rain Liu <yliu@webrtc.rs>"]
 edition = "2018"
 description = "Examples of WebRTC.rs stack"
@@ -15,8 +15,7 @@ repository = "https://github.com/webrtc-rs/examples"
 
 
 [dev-dependencies]
-util = { package = "webrtc-util", version = "0.5.0" }
-webrtc = "0.2.2"
+webrtc = "0.3.0"
 tokio = { version = "1.12.0", features = ["full"] }
 env_logger = "0.9.0"
 clap = "2"
@@ -36,6 +35,11 @@ console-subscriber = { git = "https://github.com/tokio-rs/console", branch = "ma
 [profile.dev]
 opt-level = 0
 
+[[example]]
+name = "rc-cycle"
+path = "examples/rc-cycle/rc-cycle.rs"
+bench = false
+
 [[example]]
 name = "broadcast"
 path = "examples/broadcast/broadcast.rs"
@@ -77,13 +81,8 @@ path = "examples/insertable-streams/insertable-streams.rs"
 bench = false
 
 [[example]]
-name = "play-from-disk-vp8"
-path = "examples/play-from-disk-vp8/play-from-disk-vp8.rs"
-bench = false
-
-[[example]]
-name = "play-from-disk-vp9"
-path = "examples/play-from-disk-vp9/play-from-disk-vp9.rs"
+name = "play-from-disk-vpx"
+path = "examples/play-from-disk-vpx/play-from-disk-vpx.rs"
 bench = false
 
 [[example]]

examples/README.md

@@ -6,8 +6,7 @@ All examples are ported from [Pion](https://github.com/pion/webrtc/tree/master/e
 
 #### Media API
 - [x] [Reflect](reflect): The reflect example demonstrates how to have webrtc-rs send back to the user exactly what it receives using the same PeerConnection.
-- [x] [Play from Disk VP8](play-from-disk-vp8): The play-from-disk-vp8 example demonstrates how to send VP8 video to your browser from a file saved to disk.
-- [x] [Play from Disk VP9](play-from-disk-vp9): The play-from-disk-vp9 example demonstrates how to send VP9 video to your browser from a file saved to disk.
+- [x] [Play from Disk VPx](play-from-disk-vpx): The play-from-disk-vp8 example demonstrates how to send VP8/VP9 video to your browser from a file saved to disk.
 - [x] [Play from Disk H264](play-from-disk-h264): The play-from-disk-h264 example demonstrates how to send H264 video to your browser from a file saved to disk.
 - [x] [Play from Disk Renegotiation](play-from-disk-renegotiation): The play-from-disk-renegotiation example is an extension of the play-from-disk example, but demonstrates how you can add/remove video tracks from an already negotiated PeerConnection.
 - [x] [Insertable Streams](insertable-streams): The insertable-streams example demonstrates how webrtc-rs can be used to send E2E encrypted video and decrypt via insertable streams in the browser.

examples/broadcast/broadcast.rs

@@ -6,17 +6,17 @@ use tokio::time::Duration;
 use webrtc::api::interceptor_registry::register_default_interceptors;
 use webrtc::api::media_engine::MediaEngine;
 use webrtc::api::APIBuilder;
+use webrtc::ice_transport::ice_server::RTCIceServer;
 use webrtc::interceptor::registry::Registry;
-use webrtc::media::rtp::rtp_codec::RTPCodecType;
-use webrtc::media::rtp::rtp_receiver::RTCRtpReceiver;
-use webrtc::media::track::track_local::track_local_static_rtp::TrackLocalStaticRTP;
-use webrtc::media::track::track_local::{TrackLocal, TrackLocalWriter};
-use webrtc::media::track::track_remote::TrackRemote;
-use webrtc::peer::configuration::RTCConfiguration;
-use webrtc::peer::ice::ice_server::RTCIceServer;
-use webrtc::peer::peer_connection_state::RTCPeerConnectionState;
-use webrtc::peer::sdp::session_description::RTCSessionDescription;
+use webrtc::peer_connection::configuration::RTCConfiguration;
+use webrtc::peer_connection::peer_connection_state::RTCPeerConnectionState;
+use webrtc::peer_connection::sdp::session_description::RTCSessionDescription;
 use webrtc::rtcp::payload_feedbacks::picture_loss_indication::PictureLossIndication;
+use webrtc::rtp_transceiver::rtp_codec::RTPCodecType;
+use webrtc::rtp_transceiver::rtp_receiver::RTCRtpReceiver;
+use webrtc::track::track_local::track_local_static_rtp::TrackLocalStaticRTP;
+use webrtc::track::track_local::{TrackLocal, TrackLocalWriter};
+use webrtc::track::track_remote::TrackRemote;
 use webrtc::Error;
 
 #[tokio::main]
@@ -96,7 +96,7 @@ async fn main() -> Result<()> {
  let mut registry = Registry::new();
 
  // Use the default set of Interceptors
- registry = register_default_interceptors(registry, &mut m)?;
+ registry = register_default_interceptors(registry, &mut m).await?;
 
  // Create the API object with the MediaEngine
  let api = APIBuilder::new()
@@ -127,15 +127,15 @@ async fn main() -> Result<()> {
  let local_track_chan_tx = Arc::new(local_track_chan_tx);
  // Set a handler for when a new remote track starts, this handler copies inbound RTP packets,
  // replaces the SSRC and sends them back
- let pc = Arc::clone(&peer_connection);
+ let pc = Arc::downgrade(&peer_connection);
  peer_connection
  .on_track(Box::new(
  move |track: Option<Arc<TrackRemote>>, _receiver: Option<Arc<RTCRtpReceiver>>| {
  if let Some(track) = track {
  // Send a PLI on an interval so that the publisher is pushing a keyframe every rtcpPLIInterval
  // This is a temporary fix until we implement incoming RTCP events, then we would push a PLI only when a viewer requests it
  let media_ssrc = track.ssrc();
- let pc2 = Arc::clone(&pc);
+ let pc2 = pc.clone();
  tokio::spawn(async move {
  let mut result = Result::<usize>::Ok(0);
  while result.is_ok() {
@@ -144,10 +144,14 @@ async fn main() -> Result<()> {
 
  tokio::select! {
  _ = timeout.as_mut() =>{
- result = pc2.write_rtcp(&PictureLossIndication{
+ if let Some(pc) = pc2.upgrade(){
+ result = pc.write_rtcp(&[Box::new(PictureLossIndication{
  sender_ssrc: 0,
  media_ssrc,
- }).await.map_err(Into::into);
+ })]).await.map_err(Into::into);
+ }else{
+ break;
+ }
  }
  };
  }
@@ -237,7 +241,7 @@ async fn main() -> Result<()> {
  let mut registry = Registry::new();
 
  // Use the default set of Interceptors
- registry = register_default_interceptors(registry, &mut m)?;
+ registry = register_default_interceptors(registry, &mut m).await?;
 
  // Create the API object with the MediaEngine
  let api = APIBuilder::new()

examples/data-channels-close/data-channels-close.rs

@@ -8,14 +8,14 @@ use tokio::time::Duration;
 use webrtc::api::interceptor_registry::register_default_interceptors;
 use webrtc::api::media_engine::MediaEngine;
 use webrtc::api::APIBuilder;
-use webrtc::data::data_channel::data_channel_message::DataChannelMessage;
-use webrtc::data::data_channel::RTCDataChannel;
+use webrtc::data_channel::data_channel_message::DataChannelMessage;
+use webrtc::data_channel::RTCDataChannel;
+use webrtc::ice_transport::ice_server::RTCIceServer;
 use webrtc::interceptor::registry::Registry;
-use webrtc::peer::configuration::RTCConfiguration;
-use webrtc::peer::ice::ice_server::RTCIceServer;
-use webrtc::peer::peer_connection_state::RTCPeerConnectionState;
-use webrtc::peer::sdp::session_description::RTCSessionDescription;
-use webrtc::util::math_rand_alpha;
+use webrtc::peer_connection::configuration::RTCConfiguration;
+use webrtc::peer_connection::math_rand_alpha;
+use webrtc::peer_connection::peer_connection_state::RTCPeerConnectionState;
+use webrtc::peer_connection::sdp::session_description::RTCSessionDescription;
 
 #[tokio::main]
 async fn main() -> Result<()> {
@@ -92,7 +92,7 @@ async fn main() -> Result<()> {
  let mut registry = Registry::new();
 
  // Use the default set of Interceptors
- registry = register_default_interceptors(registry, &mut m)?;
+ registry = register_default_interceptors(registry, &mut m).await?;
 
  // Create the API object with the MediaEngine
  let api = APIBuilder::new()
@@ -112,6 +112,8 @@ async fn main() -> Result<()> {
  // Create a new RTCPeerConnection
  let peer_connection = Arc::new(api.new_peer_connection(config).await?);
 
+ let (done_tx, mut done_rx) = tokio::sync::mpsc::channel::<()>(1);
+
  // Set the handler for Peer connection state
  // This will notify you when the peer has connected/disconnected
  peer_connection
@@ -123,7 +125,7 @@ async fn main() -> Result<()> {
  // Use webrtc.PeerConnectionStateDisconnected if you are interested in detecting faster timeout.
  // Note that the PeerConnection may come back from PeerConnectionStateDisconnected.
  println!("Peer Connection has gone to failed exiting");
- std::process::exit(0);
+ let _ = done_tx.try_send(());
  }
 
  Box::pin(async {})
@@ -226,7 +228,14 @@ async fn main() -> Result<()> {
  }
 
  println!("Press ctrl-c to stop");
- tokio::signal::ctrl_c().await.unwrap();
+ tokio::select! {
+ _ = done_rx.recv() => {
+ println!("received done signal!");
+ }
+ _ = tokio::signal::ctrl_c() => {
+ println!("");
+ }
+ };
 
  peer_connection.close().await?;
 

examples/data-channels-create/data-channels-create.rs

@@ -6,13 +6,13 @@ use tokio::time::Duration;
 use webrtc::api::interceptor_registry::register_default_interceptors;
 use webrtc::api::media_engine::MediaEngine;
 use webrtc::api::APIBuilder;
-use webrtc::data::data_channel::data_channel_message::DataChannelMessage;
+use webrtc::data_channel::data_channel_message::DataChannelMessage;
+use webrtc::ice_transport::ice_server::RTCIceServer;
 use webrtc::interceptor::registry::Registry;
-use webrtc::peer::configuration::RTCConfiguration;
-use webrtc::peer::ice::ice_server::RTCIceServer;
-use webrtc::peer::peer_connection_state::RTCPeerConnectionState;
-use webrtc::peer::sdp::session_description::RTCSessionDescription;
-use webrtc::util::math_rand_alpha;
+use webrtc::peer_connection::configuration::RTCConfiguration;
+use webrtc::peer_connection::math_rand_alpha;
+use webrtc::peer_connection::peer_connection_state::RTCPeerConnectionState;
+use webrtc::peer_connection::sdp::session_description::RTCSessionDescription;
 
 #[tokio::main]
 async fn main() -> Result<()> {
@@ -74,7 +74,7 @@ async fn main() -> Result<()> {
  let mut registry = Registry::new();
 
  // Use the default set of Interceptors
- registry = register_default_interceptors(registry, &mut m)?;
+ registry = register_default_interceptors(registry, &mut m).await?;
 
  // Create the API object with the MediaEngine
  let api = APIBuilder::new()
@@ -97,6 +97,8 @@ async fn main() -> Result<()> {
  // Create a datachannel with label 'data'
  let data_channel = peer_connection.create_data_channel("data", None).await?;
 
+ let (done_tx, mut done_rx) = tokio::sync::mpsc::channel::<()>(1);
+
  // Set the handler for Peer connection state
  // This will notify you when the peer has connected/disconnected
  peer_connection
@@ -108,7 +110,7 @@ async fn main() -> Result<()> {
  // Use webrtc.PeerConnectionStateDisconnected if you are interested in detecting faster timeout.
  // Note that the PeerConnection may come back from PeerConnectionStateDisconnected.
  println!("Peer Connection has gone to failed exiting");
- std::process::exit(0);
+ let _ = done_tx.try_send(());
  }
 
  Box::pin(async {})
@@ -180,7 +182,14 @@ async fn main() -> Result<()> {
  peer_connection.set_remote_description(answer).await?;
 
  println!("Press ctrl-c to stop");
- tokio::signal::ctrl_c().await.unwrap();
+ tokio::select! {
+ _ = done_rx.recv() => {
+ println!("received done signal!");
+ }
+ _ = tokio::signal::ctrl_c() => {
+ println!("");
+ }
+ };
 
  peer_connection.close().await?;
 

examples/data-channels-detach-create/data-channels-detach-create.rs

@@ -8,12 +8,12 @@ use webrtc::api::interceptor_registry::register_default_interceptors;
 use webrtc::api::media_engine::MediaEngine;
 use webrtc::api::setting_engine::SettingEngine;
 use webrtc::api::APIBuilder;
+use webrtc::ice_transport::ice_server::RTCIceServer;
 use webrtc::interceptor::registry::Registry;
-use webrtc::peer::configuration::RTCConfiguration;
-use webrtc::peer::ice::ice_server::RTCIceServer;
-use webrtc::peer::peer_connection_state::RTCPeerConnectionState;
-use webrtc::peer::sdp::session_description::RTCSessionDescription;
-use webrtc::util::math_rand_alpha;
+use webrtc::peer_connection::configuration::RTCConfiguration;
+use webrtc::peer_connection::math_rand_alpha;
+use webrtc::peer_connection::peer_connection_state::RTCPeerConnectionState;
+use webrtc::peer_connection::sdp::session_description::RTCSessionDescription;
 
 const MESSAGE_SIZE: usize = 1500;
 
@@ -77,7 +77,7 @@ async fn main() -> Result<()> {
  let mut registry = Registry::new();
 
  // Use the default set of Interceptors
- registry = register_default_interceptors(registry, &mut m)?;
+ registry = register_default_interceptors(registry, &mut m).await?;
 
  // Since this behavior diverges from the WebRTC API it has to be
  // enabled using a settings engine. Mixing both detached and the
@@ -109,6 +109,8 @@ async fn main() -> Result<()> {
  // Create a datachannel with label 'data'
  let data_channel = peer_connection.create_data_channel("data", None).await?;
 
+ let (done_tx, mut done_rx) = tokio::sync::mpsc::channel::<()>(1);
+
  // Set the handler for Peer connection state
  // This will notify you when the peer has connected/disconnected
  peer_connection
@@ -120,7 +122,7 @@ async fn main() -> Result<()> {
  // Use webrtc.PeerConnectionStateDisconnected if you are interested in detecting faster timeout.
  // Note that the PeerConnection may come back from PeerConnectionStateDisconnected.
  println!("Peer Connection has gone to failed exiting");
- std::process::exit(0);
+ let _ = done_tx.try_send(());
  }
 
  Box::pin(async {})
@@ -189,15 +191,22 @@ async fn main() -> Result<()> {
  peer_connection.set_remote_description(answer).await?;
 
  println!("Press ctrl-c to stop");
- tokio::signal::ctrl_c().await.unwrap();
+ tokio::select! {
+ _ = done_rx.recv() => {
+ println!("received done signal!");
+ }
+ _ = tokio::signal::ctrl_c() => {
+ println!("");
+ }
+ };
 
  peer_connection.close().await?;
 
  Ok(())
 }
 
 // read_loop shows how to read from the datachannel directly
-async fn read_loop(d: Arc<webrtc::webrtc_data::data_channel::DataChannel>) -> Result<()> {
+async fn read_loop(d: Arc<webrtc::data::data_channel::DataChannel>) -> Result<()> {
  let mut buffer = vec![0u8; MESSAGE_SIZE];
  loop {
  let n = match d.read(&mut buffer).await {
@@ -216,7 +225,7 @@ async fn read_loop(d: Arc<webrtc::webrtc_data::data_channel::DataChannel>) -> Re
 }
 
 // write_loop shows how to write to the datachannel directly
-async fn write_loop(d: Arc<webrtc::webrtc_data::data_channel::DataChannel>) -> Result<()> {
+async fn write_loop(d: Arc<webrtc::data::data_channel::DataChannel>) -> Result<()> {
  let mut result = Result::<usize>::Ok(0);
  while result.is_ok() {
  let timeout = tokio::time::sleep(Duration::from_secs(5));