xref: /webrtc/examples/examples/reflect/reflect.rs (revision 8ec0a1cb)

use anyhow::Result;
use clap::{AppSettings, Arg, Command};
use std::collections::HashMap;
use std::io::Write;
use std::sync::Arc;
use tokio::time::Duration;
use webrtc::api::interceptor_registry::register_default_interceptors;
use webrtc::api::media_engine::{MediaEngine, MIME_TYPE_OPUS, MIME_TYPE_VP8};
use webrtc::api::APIBuilder;
use webrtc::ice_transport::ice_server::RTCIceServer;
use webrtc::interceptor::registry::Registry;
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::{
    RTCRtpCodecCapability, RTCRtpCodecParameters, RTPCodecType,
};
use webrtc::track::track_local::track_local_static_rtp::TrackLocalStaticRTP;
use webrtc::track::track_local::{TrackLocal, TrackLocalWriter};

#[tokio::main]
async fn main() -> Result<()> {
    let mut app = Command::new("reflect")
        .version("0.1.0")
        .author("Rain Liu <[email protected]>")
        .about("An example of how to send back to the user exactly what it receives using the same PeerConnection.")
        .setting(AppSettings::DeriveDisplayOrder)
        .subcommand_negates_reqs(true)
        .arg(
            Arg::new("FULLHELP")
                .help("Prints more detailed help information")
                .long("fullhelp"),
        )
        .arg(
            Arg::new("debug")
                .long("debug")
                .short('d')
                .help("Prints debug log information"),
        ).arg(
            Arg::new("audio")
                .long("audio")
                .short('a')
                .help("Enable audio reflect"),
        ).arg(
            Arg::new("video")
                .long("video")
                .short('v')
                .help("Enable video reflect"),
        );

    let matches = app.clone().get_matches();

    if matches.is_present("FULLHELP") {
        app.print_long_help().unwrap();
        std::process::exit(0);
    }

    let audio = matches.is_present("audio");
    let video = matches.is_present("video");
    if !audio && !video {
        println!("one of audio or video must be enabled");
        std::process::exit(0);
    }
    let debug = matches.is_present("debug");
    if debug {
        env_logger::Builder::new()
            .format(|buf, record| {
                writeln!(
                    buf,
                    "{}:{} [{}] {} - {}",
                    record.file().unwrap_or("unknown"),
                    record.line().unwrap_or(0),
                    record.level(),
                    chrono::Local::now().format("%H:%M:%S.%6f"),
                    record.args()
                )
            })
            .filter(None, log::LevelFilter::Trace)
            .init();
    }

    // Everything below is the WebRTC-rs API! Thanks for using it ❤️.

    // Create a MediaEngine object to configure the supported codec
    let mut m = MediaEngine::default();

    // Setup the codecs you want to use.
    if audio {
        m.register_codec(
            RTCRtpCodecParameters {
                capability: RTCRtpCodecCapability {
                    mime_type: MIME_TYPE_OPUS.to_owned(),
                    ..Default::default()
                },
                payload_type: 120,
                ..Default::default()
            },
            RTPCodecType::Audio,
        )?;
    }

    // We'll use a VP8 and Opus but you can also define your own
    if video {
        m.register_codec(
            RTCRtpCodecParameters {
                capability: RTCRtpCodecCapability {
                    mime_type: MIME_TYPE_VP8.to_owned(),
                    clock_rate: 90000,
                    channels: 0,
                    sdp_fmtp_line: "".to_owned(),
                    rtcp_feedback: vec![],
                },
                payload_type: 96,
                ..Default::default()
            },
            RTPCodecType::Video,
        )?;
    }

    // Create a InterceptorRegistry. This is the user configurable RTP/RTCP Pipeline.
    // This provides NACKs, RTCP Reports and other features. If you use `webrtc.NewPeerConnection`
    // this is enabled by default. If you are manually managing You MUST create a InterceptorRegistry
    // for each PeerConnection.
    let mut registry = Registry::new();

    // Use the default set of Interceptors
    registry = register_default_interceptors(registry, &mut m)?;

    // Create the API object with the MediaEngine
    let api = APIBuilder::new()
        .with_media_engine(m)
        .with_interceptor_registry(registry)
        .build();

    // Prepare the configuration
    let config = RTCConfiguration {
        ice_servers: vec![RTCIceServer {
            urls: vec!["stun:stun.l.google.com:19302".to_owned()],
            ..Default::default()
        }],
        ..Default::default()
    };

    // Create a new RTCPeerConnection
    let peer_connection = Arc::new(api.new_peer_connection(config).await?);
    let mut output_tracks = HashMap::new();
    let mut media = vec![];
    if audio {
        media.push("audio");
    }
    if video {
        media.push("video");
    };
    for s in media {
        let output_track = Arc::new(TrackLocalStaticRTP::new(
            RTCRtpCodecCapability {
                mime_type: if s == "video" {
                    MIME_TYPE_VP8.to_owned()
                } else {
                    MIME_TYPE_OPUS.to_owned()
                },
                ..Default::default()
            },
            format!("track-{s}"),
            "webrtc-rs".to_owned(),
        ));

        // Add this newly created track to the PeerConnection
        let rtp_sender = peer_connection
            .add_track(Arc::clone(&output_track) as Arc<dyn TrackLocal + Send + Sync>)
            .await?;

        // Read incoming RTCP packets
        // Before these packets are returned they are processed by interceptors. For things
        // like NACK this needs to be called.
        let m = s.to_owned();
        tokio::spawn(async move {
            let mut rtcp_buf = vec![0u8; 1500];
            while let Ok((_, _)) = rtp_sender.read(&mut rtcp_buf).await {}
            println!("{m} rtp_sender.read loop exit");
            Result::<()>::Ok(())
        });

        output_tracks.insert(s.to_owned(), output_track);
    }

    // Wait for the offer to be pasted
    let line = signal::must_read_stdin()?;
    let desc_data = signal::decode(line.as_str())?;
    let offer = serde_json::from_str::<RTCSessionDescription>(&desc_data)?;

    // Set the remote SessionDescription
    peer_connection.set_remote_description(offer).await?;

    // 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::downgrade(&peer_connection);
    peer_connection.on_track(Box::new(move |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();

        if track.kind() == RTPCodecType::Video {
            let pc2 = pc.clone();
            tokio::spawn(async move {
                let mut result = Result::<usize>::Ok(0);
                while result.is_ok() {
                    let timeout = tokio::time::sleep(Duration::from_secs(3));
                    tokio::pin!(timeout);

                    tokio::select! {
                        _ = timeout.as_mut() =>{
                            if let Some(pc) = pc2.upgrade(){
                                result = pc.write_rtcp(&[Box::new(PictureLossIndication{
                                        sender_ssrc: 0,
                                        media_ssrc,
                                })]).await.map_err(Into::into);
                            }else{
                                break;
                            }
                        }
                    };
                }
            });
        }

        let kind = if track.kind() == RTPCodecType::Audio {
            "audio"
        } else {
            "video"
        };
        let output_track = if let Some(output_track) = output_tracks.get(kind) {
            Arc::clone(output_track)
        } else {
            println!("output_track not found for type = {kind}");
            return Box::pin(async {});
        };

        let output_track2 = Arc::clone(&output_track);
        tokio::spawn(async move {
            println!(
                "Track has started, of type {}: {}",
                track.payload_type(),
                track.codec().capability.mime_type
            );
            // Read RTP packets being sent to webrtc-rs
            while let Ok((rtp, _)) = track.read_rtp().await {
                if let Err(err) = output_track2.write_rtp(&rtp).await {
                    println!("output track write_rtp got error: {err}");
                    break;
                }
            }

            println!(
                "on_track finished, of type {}: {}",
                track.payload_type(),
                track.codec().capability.mime_type
            );
        });

        Box::pin(async {})
    }));

    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.on_peer_connection_state_change(Box::new(move |s: RTCPeerConnectionState| {
        println!("Peer Connection State has changed: {s}");

        if s == RTCPeerConnectionState::Failed {
            // Wait until PeerConnection has had no network activity for 30 seconds or another failure. It may be reconnected using an ICE Restart.
            // 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");
            let _ = done_tx.try_send(());
        }

        Box::pin(async {})
    }));

    // Create an answer
    let answer = peer_connection.create_answer(None).await?;

    // Create channel that is blocked until ICE Gathering is complete
    let mut gather_complete = peer_connection.gathering_complete_promise().await;

    // Sets the LocalDescription, and starts our UDP listeners
    peer_connection.set_local_description(answer).await?;

    // Block until ICE Gathering is complete, disabling trickle ICE
    // we do this because we only can exchange one signaling message
    // in a production application you should exchange ICE Candidates via OnICECandidate
    let _ = gather_complete.recv().await;

    // Output the answer in base64 so we can paste it in browser
    if let Some(local_desc) = peer_connection.local_description().await {
        let json_str = serde_json::to_string(&local_desc)?;
        let b64 = signal::encode(&json_str);
        println!("{b64}");
    } else {
        println!("generate local_description failed!");
    }

    println!("Press ctrl-c to stop");
    //let timeout = tokio::time::sleep(Duration::from_secs(20));
    //tokio::pin!(timeout);

    tokio::select! {
        //_ = timeout.as_mut() => {
        //    println!("received timeout signal!");
        //}
        _ = done_rx.recv() => {
            println!("received done signal!");
        }
        _ = tokio::signal::ctrl_c() => {
            println!();
        }
    };

    peer_connection.close().await?;

    Ok(())
}