librtmp2 0.1.0

//! End-to-end smoke test for the real TCP socket glue: a `Server` listens on
//! loopback, a `Client` connects, performs the RTMP handshake + connect +
//! createStream + publish exchange over real sockets, and sends one video
//! frame that the server's `on_frame_cb` should observe.

use std::sync::atomic::{AtomicU64, AtomicUsize, Ordering};
use std::thread;
use std::time::{Duration, Instant};

use librtmp2::client::Client;
use librtmp2::server::Server;
use librtmp2::types::*;

static FRAMES_RECEIVED: AtomicUsize = AtomicUsize::new(0);

const SENT_FRAME_BYTE: u8 = 0xAB;
const SENT_FRAME_LEN: usize = 32;

fn on_frame(frame: &Frame) {
    if frame.size as usize == SENT_FRAME_LEN {
        FRAMES_RECEIVED.fetch_add(1, Ordering::SeqCst);
    }
}

fn allow_publish(_conn_id: u64, _app: &str, _stream_name: &str) -> bool {
    true
}

fn plain_config() -> ServerConfig {
    ServerConfig {
        max_connections: 8,
        chunk_size: 128,
        tls_enabled: 0,
        tls_cert_file: std::ptr::null(),
        tls_key_file: std::ptr::null(),
        tls_ca_file: std::ptr::null(),
        tls_insecure: 0,
    }
}

#[test]
fn server_client_publish_over_real_sockets() {
    FRAMES_RECEIVED.store(0, Ordering::SeqCst);

    let mut server = Server::new(plain_config()).unwrap();
    server.listen("127.0.0.1:19661").unwrap();
    server.on_frame_cb = Some(on_frame);
    server.on_publish_cb = Some(allow_publish);

    let (setup_tx, setup_rx) = std::sync::mpsc::channel();
    let client_thread = thread::spawn(move || {
        let mut client = Client::new();
        let result = (|| -> std::result::Result<(), librtmp2::types::ErrorCode> {
            client.connect("rtmp://127.0.0.1:19661/live/stream1")?;
            client.publish()?;

            let data = [SENT_FRAME_BYTE; SENT_FRAME_LEN];
            let frame = Frame {
                frame_type: FrameType::Video,
                timestamp: 0,
                composition_time: 0,
                size: data.len() as u32,
                data: data.as_ptr(),
                audio_codec: AudioCodec::default(),
                audio_sample_rate: 0,
                audio_channels: 0,
                audio_bit_depth: 0,
                audio_fourcc: FourCc::default(),
                video_codec: VideoCodec::H264,
                video_fourcc: FourCc::default(),
                video_frame_type: 1,
                is_metadata: 0,
            };
            client.send_frame(&frame)?;
            Ok(())
        })();
        let _ = setup_tx.send(result.is_ok());
        result.unwrap();
        thread::sleep(Duration::from_millis(200));
    });

    let deadline = Instant::now() + Duration::from_secs(5);
    loop {
        if let Ok(setup_ok) = setup_rx.try_recv() {
            assert!(setup_ok, "client setup failed");
        }
        if FRAMES_RECEIVED.load(Ordering::SeqCst) > 0 || Instant::now() >= deadline {
            break;
        }
        server.poll(20).unwrap();
    }

    client_thread.join().unwrap();
    assert!(
        FRAMES_RECEIVED.load(Ordering::SeqCst) > 0,
        "server never observed the published frame"
    );
}

static OBSERVED_CONN_ID: AtomicU64 = AtomicU64::new(0);

fn record_conn_id(conn_id: u64, _app: &str, _stream_name: &str) -> bool {
    OBSERVED_CONN_ID.store(conn_id, Ordering::SeqCst);
    true
}

/// `Server::set_conn_id_base` is what lets an integrator run two `Server`
/// instances (e.g. plaintext RTMP + RTMPS) in one process without their
/// auto-assigned `conn_id`s colliding. Verify the first connection accepted
/// after calling it actually gets the configured base rather than 1.
#[test]
fn set_conn_id_base_offsets_first_assigned_conn_id() {
    OBSERVED_CONN_ID.store(0, Ordering::SeqCst);
    const BASE: u64 = 1 << 40;

    let mut server = Server::new(plain_config()).unwrap();
    server.set_conn_id_base(BASE);
    server.listen("127.0.0.1:19662").unwrap();
    server.on_publish_cb = Some(record_conn_id);

    let (setup_tx, setup_rx) = std::sync::mpsc::channel();
    let client_thread = thread::spawn(move || {
        let mut client = Client::new();
        let result = (|| -> std::result::Result<(), librtmp2::types::ErrorCode> {
            client.connect("rtmp://127.0.0.1:19662/live/stream1")?;
            client.publish()?;
            Ok(())
        })();
        let _ = setup_tx.send(result.is_ok());
        result.unwrap();
        thread::sleep(Duration::from_millis(200));
    });

    let deadline = Instant::now() + Duration::from_secs(5);
    loop {
        if let Ok(setup_ok) = setup_rx.try_recv() {
            assert!(setup_ok, "client setup failed");
        }
        if OBSERVED_CONN_ID.load(Ordering::SeqCst) != 0 || Instant::now() >= deadline {
            break;
        }
        server.poll(20).unwrap();
    }

    client_thread.join().unwrap();
    assert_eq!(
        OBSERVED_CONN_ID.load(Ordering::SeqCst),
        BASE,
        "first accepted connection should have conn_id == configured base"
    );
}

#[test]
#[should_panic(expected = "conn_id base must be non-zero")]
fn set_conn_id_base_rejects_zero() {
    let mut server = Server::new(plain_config()).unwrap();
    server.set_conn_id_base(0);
}

#[test]
#[should_panic(expected = "conn_id base must leave room")]
fn set_conn_id_base_rejects_exhausted_counter() {
    let mut server = Server::new(plain_config()).unwrap();
    server.set_conn_id_base(u64::MAX);
}

#[test]
fn listener_fds_exposes_every_bound_listener_and_stop_clears_them() {
    let mut server = Server::new(plain_config()).unwrap();
    assert!(server.tls_ctx.is_none());

    server.listen("127.0.0.1:0").unwrap();
    server.listen("127.0.0.1:0").unwrap();

    let fds = server.listener_fds();
    assert_eq!(fds.len(), 2, "each listen() call should expose its fd");
    assert!(fds.iter().all(|fd| *fd >= 0));
    assert_ne!(fds[0], fds[1], "listeners must expose distinct fds");
    assert_eq!(server.server_fd, fds[0]);

    server.stop();
    assert_eq!(server.server_fd, -1);
    assert!(server.listener_fds().is_empty());
}

static PLAYER_FRAMES_RECEIVED: AtomicUsize = AtomicUsize::new(0);

fn on_player_frame(frame: &Frame) {
    if frame.size as usize == SENT_FRAME_LEN {
        PLAYER_FRAMES_RECEIVED.fetch_add(1, Ordering::SeqCst);
    }
}

/// A publisher on one listener and a player on a *different* listener of the
/// same `Server` must still relay to each other — the whole point of binding
/// multiple listeners (e.g. plaintext RTMP + RTMPS) on one `Server` instead
/// of running two separate `Server`s is that they share one relay/connection
/// list. Running two separate `Server`s instead would silently drop this
/// cross-listener case, since each would only relay within its own
/// `connections`.
#[test]
fn publisher_and_player_relay_across_different_listeners() {
    PLAYER_FRAMES_RECEIVED.store(0, Ordering::SeqCst);

    let mut server = Server::new(plain_config()).unwrap();
    server.listen("127.0.0.1:19663").unwrap();
    server.listen("127.0.0.1:19664").unwrap();

    let (setup_tx, setup_rx) = std::sync::mpsc::channel();
    let client_thread = thread::spawn(move || {
        let result = (|| -> std::result::Result<(), librtmp2::types::ErrorCode> {
            let mut publisher = Client::new();
            publisher.connect("rtmp://127.0.0.1:19663/live/relaytest")?;
            publisher.publish()?;

            let mut player = Client::new();
            player.on_frame_cb = Some(on_player_frame);
            player.connect("rtmp://127.0.0.1:19664/live/relaytest")?;
            player.play()?;

            // Give the server a moment to process the play authorization and
            // enable relay for this connection before the frame is sent.
            thread::sleep(Duration::from_millis(100));

            let data = [SENT_FRAME_BYTE; SENT_FRAME_LEN];
            let frame = Frame {
                frame_type: FrameType::Video,
                timestamp: 0,
                composition_time: 0,
                size: data.len() as u32,
                data: data.as_ptr(),
                audio_codec: AudioCodec::default(),
                audio_sample_rate: 0,
                audio_channels: 0,
                audio_bit_depth: 0,
                audio_fourcc: FourCc::default(),
                video_codec: VideoCodec::H264,
                video_fourcc: FourCc::default(),
                video_frame_type: 1,
                is_metadata: 0,
            };
            publisher.send_frame(&frame)?;

            let deadline = Instant::now() + Duration::from_secs(5);
            while PLAYER_FRAMES_RECEIVED.load(Ordering::SeqCst) == 0 && Instant::now() < deadline {
                player.poll(50)?;
            }
            Ok(())
        })();
        let _ = setup_tx.send(result.is_ok());
        result.unwrap();
    });

    let deadline = Instant::now() + Duration::from_secs(6);
    loop {
        if let Ok(setup_ok) = setup_rx.try_recv() {
            assert!(setup_ok, "publisher/player setup failed");
            break;
        }
        if Instant::now() >= deadline {
            break;
        }
        server.poll(20).unwrap();
    }

    client_thread.join().unwrap();
    assert!(
        PLAYER_FRAMES_RECEIVED.load(Ordering::SeqCst) > 0,
        "player on a different listener never received the publisher's frame \
         — relay must be shared across every listener on one Server"
    );
}