dailz
bc405c6d16
Step 1 of file-level refactor: prove the file->directory pattern with the cleanest possible extraction. - src/webrtc.rs: 913 -> 741 LOC - New src/webrtc/html_page.rs: 170-line HTML test page as pub(super) const - Parent module re-exports via `mod html_page; use html_page::HTML_PAGE;` so all references in handle_signaling stay unchanged. Verification (all green): - cargo build / cargo build --release - cargo test (79 lib + 3 integration = 82 pass, 1 ignored — unchanged) - cargo clippy --all-targets -- -D warnings - cargo fmt --check - cargo check --bin vaapi_import_bench --bin sw_encode_bench - Test count in webrtc.rs: 18 (unchanged from baseline) Oracle audit note: HTML_PAGE had a single use site (handle_signaling L257-258) and zero #[cfg(test)] references, so the extraction is provably behavior- preserving.
744 lines
26 KiB
Rust
744 lines
26 KiB
Rust
// WebRTC 传输模块 — 使用 str0m (Sans-IO) 将 H.264 编码帧推送到浏览器
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use std::io::{Read, Write};
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use std::net::{SocketAddr, TcpListener, UdpSocket};
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use std::time::{Duration, Instant};
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use anyhow::{bail, Result};
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use str0m::bwe::{Bitrate, BweKind};
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use str0m::change::SdpOffer;
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use str0m::format::Codec;
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use str0m::media::{Frequency, MediaKind, MediaTime, Mid, Pt};
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use str0m::net::{Protocol, Receive};
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use str0m::{Candidate, Event, IceConnectionState, Input, Output, Rtc, RtcConfig};
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/// Minimum interval between honored keyframe productions, regardless of source
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/// (PLI from viewer, connect event, resolution change). Prevents PLI storms
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/// from causing back-to-back IDRs that swamp the network with multi-hundred-KB
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/// bursts. See issue #23.
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const FORCED_KEYFRAME_MIN_INTERVAL: Duration = Duration::from_secs(1);
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mod html_page;
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use html_page::HTML_PAGE;
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// ── WebRTC 状态 ───────────────────────────────────────────────────────────
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pub struct WebRtcState {
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signal_listener: TcpListener,
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inner: Option<WebRtcInner>,
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fps: u32,
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}
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struct WebRtcInner {
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rtc: Rtc,
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socket: UdpSocket,
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udp_addr: SocketAddr,
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video_mid: Option<Mid>,
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video_pt: Option<Pt>,
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connected: bool,
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need_keyframe: bool,
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force_keyframe_to_encode: bool,
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last_forced_keyframe_at: Option<Instant>,
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current_bwe_estimate: Option<Bitrate>,
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rtp_clock: u32,
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buf: Vec<u8>,
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}
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impl WebRtcState {
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pub fn new(port: u16, fps: u32) -> Result<Self> {
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let signal_listener = TcpListener::bind(format!("0.0.0.0:{port}"))?;
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signal_listener.set_nonblocking(true)?;
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tracing::info!("WebRTC signaling on http://0.0.0.0:{port}/");
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Ok(Self {
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signal_listener,
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inner: None,
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fps,
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})
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}
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pub fn handle_signaling(&mut self) -> Result<bool> {
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let mut handled = false;
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loop {
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let (mut stream, _addr) = match self.signal_listener.accept() {
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Ok(s) => s,
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Err(ref e) if e.kind() == std::io::ErrorKind::WouldBlock => break,
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Err(e) => bail!("TCP accept error: {e}"),
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};
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handled = true;
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stream.set_nonblocking(true)?;
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let mut req = vec![0u8; 65536];
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let n = match stream.read(&mut req) {
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Ok(n) => n,
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Err(ref e) if e.kind() == std::io::ErrorKind::WouldBlock => continue,
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Err(e) => {
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tracing::warn!("TCP read error: {e}");
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continue;
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}
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};
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let req_str = String::from_utf8_lossy(&req[..n]);
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if req_str.starts_with("GET / ")
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|| req_str.starts_with("GET /sdp ")
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&& !req_str.contains("Content-Type: application/json")
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{
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let resp = format!(
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"HTTP/1.1 200 OK\r\nContent-Type: text/html\r\nContent-Length: {}\r\nConnection: close\r\n\r\n{}",
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HTML_PAGE.len(),
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HTML_PAGE
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);
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if let Err(e) = stream.write_all(resp.as_bytes()) {
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tracing::debug!("HTTP write error: {e}");
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}
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} else if req_str.starts_with("POST /sdp") {
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let body = extract_body(&req_str);
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if body.is_empty() {
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let resp = "HTTP/1.1 400 Bad Request\r\nConnection: close\r\n\r\nempty body";
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if let Err(e) = stream.write_all(resp.as_bytes()) {
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tracing::debug!("HTTP write error: {e}");
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}
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continue;
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}
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match WebRtcInner::new(self.fps).and_then(|mut new_inner| {
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let answer_json = new_inner.handle_sdp_offer(body.as_bytes())?;
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Ok((new_inner, answer_json))
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}) {
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Ok((new_inner, answer_json)) => {
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let replacing = self.inner.is_some();
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self.inner = Some(new_inner);
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if replacing {
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tracing::info!("Replaced WebRTC connection (old dropped)");
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} else {
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tracing::info!("New WebRTC connection");
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}
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let resp = format!(
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"HTTP/1.1 200 OK\r\nContent-Type: application/json\r\nContent-Length: {}\r\nConnection: close\r\n\r\n{}",
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answer_json.len(),
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answer_json
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);
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if let Err(e) = stream.write_all(resp.as_bytes()) {
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tracing::debug!("HTTP write error: {e}");
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}
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}
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Err(e) => {
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tracing::error!("SDP offer handling failed: {e}");
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let resp =
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"HTTP/1.1 500 Internal Server Error\r\nConnection: close\r\n\r\n";
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if let Err(e) = stream.write_all(resp.as_bytes()) {
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tracing::debug!("HTTP write error: {e}");
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}
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}
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}
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} else {
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let resp = "HTTP/1.1 404 Not Found\r\nConnection: close\r\n\r\n";
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if let Err(e) = stream.write_all(resp.as_bytes()) {
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tracing::debug!("HTTP write error: {e}");
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}
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}
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}
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Ok(handled)
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}
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pub fn poll_rtc(&mut self) -> Result<()> {
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if let Some(inner) = self.inner.as_mut() {
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if inner.poll_rtc()? {
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tracing::info!("WebRTC connection closed; clearing connection state");
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self.inner = None;
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}
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}
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Ok(())
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}
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pub fn feed_network(&mut self) -> Result<()> {
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if let Some(inner) = self.inner.as_mut() {
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inner.feed_network()?;
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}
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Ok(())
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}
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pub fn poll_and_feed(&mut self) -> Result<()> {
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self.poll_rtc()?;
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self.feed_network()?;
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self.poll_rtc()
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}
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pub fn write_h264_frame(&mut self, data: &[u8], pts_ticks: i64) -> Result<()> {
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let should_destroy = if let Some(inner) = self.inner.as_mut() {
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inner.write_h264_frame(data, pts_ticks)?
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} else {
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false
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};
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if should_destroy {
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tracing::info!("WebRTC connection failed during write; clearing connection state");
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self.inner = None;
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}
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Ok(())
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}
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pub fn is_connected(&self) -> bool {
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self.inner.as_ref().is_some_and(WebRtcInner::is_connected)
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}
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/// Returns the latest bandwidth estimation estimate in bits per second, if available.
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pub fn get_bwe_estimate(&self) -> Option<u64> {
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self.inner
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.as_ref()
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.and_then(|inner| inner.current_bwe_estimate.map(|b| b.as_u64()))
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}
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/// Internal keyframe request (connect, resolution change). Always honored,
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/// but updates last_forced_keyframe_at so a subsequent viewer PLI in the next
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/// second is throttled.
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pub fn set_need_keyframe(&mut self) {
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if let Some(inner) = self.inner.as_mut() {
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inner.set_need_keyframe();
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}
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}
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/// External keyframe request from viewer (PLI/FIR via str0m
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/// `Event::KeyframeRequest`). Rate-limited to FORCED_KEYFRAME_MIN_INTERVAL
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/// to prevent PLI storms from swamping the network with IDR bursts.
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/// See issue #23.
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#[allow(dead_code)]
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pub fn request_keyframe_from_viewer(&mut self) {
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if let Some(inner) = self.inner.as_mut() {
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inner.request_keyframe_from_viewer();
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}
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}
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pub fn take_force_keyframe(&mut self) -> bool {
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if let Some(inner) = self.inner.as_mut() {
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let v = inner.force_keyframe_to_encode;
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inner.force_keyframe_to_encode = false;
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v
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} else {
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false
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}
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}
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}
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impl WebRtcInner {
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fn new(fps: u32) -> Result<Self> {
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let _ = fps;
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let mut rtc = RtcConfig::new()
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.enable_bwe(Some(Bitrate::mbps(5)))
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.build(Instant::now());
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let socket = UdpSocket::bind("0.0.0.0:0")?;
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socket.set_nonblocking(true)?;
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// Increase UDP send buffer to absorb IDR frame bursts (256KB IDR → ~145 RTP
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// packets in a single poll_rtc loop). Default Linux wmem is ~208KB which
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// causes EAGAIN on large keyframes. 2MB comfortably buffers several IDRs.
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const SND_BUF_REQ: usize = 2 * 1024 * 1024;
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// SAFETY: fd is a valid UDP socket; setsockopt/getsockopt with SOL_SOCKET +
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// SO_SNDBUF are safe on Linux. We check the return value and log the actual
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// kernel-assigned buffer (Linux may cap at wmem_max and/or double the value).
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unsafe {
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let fd = std::os::unix::io::AsRawFd::as_raw_fd(&socket);
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let val: libc::c_int = SND_BUF_REQ as libc::c_int;
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let ret = libc::setsockopt(
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fd,
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libc::SOL_SOCKET,
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libc::SO_SNDBUF,
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&val as *const libc::c_int as *const libc::c_void,
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std::mem::size_of::<libc::c_int>() as libc::socklen_t,
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);
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if ret < 0 {
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tracing::warn!(
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"setsockopt SO_SNDBUF failed (errno {})",
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std::io::Error::last_os_error()
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);
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}
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let mut actual: libc::c_int = 0;
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let mut actual_len: libc::socklen_t =
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std::mem::size_of::<libc::c_int>() as libc::socklen_t;
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let gret = libc::getsockopt(
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fd,
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libc::SOL_SOCKET,
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libc::SO_SNDBUF,
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&mut actual as *mut libc::c_int as *mut libc::c_void,
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&mut actual_len,
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);
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if gret == 0 {
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tracing::info!(
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"UDP send buffer: requested {}KB, actual {}KB",
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SND_BUF_REQ / 1024,
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actual / 1024,
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);
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}
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}
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let local_addr = socket.local_addr()?;
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let lan_ip = local_ip().unwrap_or_else(|| {
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tracing::debug!("Failed to detect LAN IP, falling back to 127.0.0.1");
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"127.0.0.1".to_string()
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});
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let candidate_addr: SocketAddr = format!("{lan_ip}:{}", local_addr.port()).parse()?;
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let candidate = Candidate::host(candidate_addr, "udp")
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.map_err(|e| anyhow::anyhow!("candidate: {e}"))?;
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rtc.add_local_candidate(candidate);
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tracing::info!("WebRTC UDP: {candidate_addr} (bound 0.0.0.0)");
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Ok(Self {
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rtc,
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socket,
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udp_addr: candidate_addr,
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video_mid: None,
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video_pt: None,
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connected: false,
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need_keyframe: false,
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force_keyframe_to_encode: false,
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last_forced_keyframe_at: None,
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current_bwe_estimate: None,
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rtp_clock: 0,
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buf: vec![0u8; 65535],
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})
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}
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fn handle_sdp_offer(&mut self, body: &[u8]) -> Result<String> {
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let offer: SdpOffer =
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serde_json::from_slice(body).map_err(|e| anyhow::anyhow!("parse SDP offer: {e}"))?;
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let answer = self
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.rtc
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.sdp_api()
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.accept_offer(offer)
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.map_err(|e| anyhow::anyhow!("accept_offer: {e}"))?;
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self.need_keyframe = true;
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self.force_keyframe_to_encode = true;
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tracing::info!("SDP exchange complete, waiting for ICE/DTLS...");
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self.discover_video_params();
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let answer_json =
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serde_json::to_vec(&answer).map_err(|e| anyhow::anyhow!("serialize answer: {e}"))?;
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String::from_utf8(answer_json).map_err(|e| anyhow::anyhow!("answer utf8: {e}"))
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}
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fn discover_video_params(&mut self) {
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let mid = match self.video_mid {
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Some(m) => m,
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None => {
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tracing::debug!("discover_video_params: no video_mid yet");
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return;
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}
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};
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self.video_pt = None;
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// Disable str0m's LeakyBucketPacer for this video stream. Default pacing
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// adds ~100ms send latency per large IDR; our 8Mbps cap + VBV already
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// provide rate control. BWE stays enabled for adaptation feedback.
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if let Some(stream_tx) = self.rtc.direct_api().stream_tx_by_mid(mid, None) {
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stream_tx.set_unpaced(true);
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}
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if let Some(writer) = self.rtc.writer(mid) {
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for pp in writer.payload_params() {
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tracing::debug!("Codec: pt={:?} spec={:?}", pp.pt(), pp.spec());
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if pp.spec().codec.is_video() && pp.spec().codec == Codec::H264 {
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self.video_pt = Some(pp.pt());
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tracing::info!("H.264 payload type: {:?}", pp.pt());
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break;
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}
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}
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}
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if self.video_pt.is_none() {
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tracing::warn!("discover_video_params: no H.264 codec found for mid={mid}");
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}
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}
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/// Unthrottled keyframe trigger. Always sets the keyframe flags and refreshes
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/// `last_forced_keyframe_at` so a follow-up viewer PLI within the next
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/// `FORCED_KEYFRAME_MIN_INTERVAL` is dropped.
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fn set_need_keyframe(&mut self) {
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self.need_keyframe = true;
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self.force_keyframe_to_encode = true;
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self.last_forced_keyframe_at = Some(Instant::now());
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}
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/// Throttled keyframe trigger used for viewer-originated PLI/FIR requests.
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/// Honored only if enough time has elapsed since the last forced keyframe.
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fn request_keyframe_from_viewer(&mut self) {
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let now = Instant::now();
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let should_honor = self
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.last_forced_keyframe_at
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.is_none_or(|last| now.duration_since(last) >= FORCED_KEYFRAME_MIN_INTERVAL);
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if should_honor {
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self.last_forced_keyframe_at = Some(now);
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self.need_keyframe = true;
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self.force_keyframe_to_encode = true;
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} else {
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tracing::warn!(
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"PLI throttled (last forced keyframe {:?} ago, min interval {:?})",
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self.last_forced_keyframe_at.map(|t| now.duration_since(t)),
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FORCED_KEYFRAME_MIN_INTERVAL
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);
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}
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}
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fn poll_rtc(&mut self) -> Result<bool> {
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loop {
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match self.rtc.poll_output() {
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Ok(Output::Transmit(t)) => {
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tracing::trace!("TX {} bytes -> {}", t.contents.len(), t.destination);
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if let Err(e) = self.socket.send_to(&t.contents, t.destination) {
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if e.kind() == std::io::ErrorKind::WouldBlock {
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tracing::debug!(
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"UDP send WouldBlock ({} bytes) — send buffer full",
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t.contents.len(),
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);
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} else {
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tracing::warn!("UDP send error to {}: {e}", t.destination);
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}
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}
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}
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Ok(Output::Event(e)) => {
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tracing::debug!("RTC event: {e:?}");
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match &e {
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Event::Connected => {
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tracing::info!("WebRTC connected!");
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self.connected = true;
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self.set_need_keyframe();
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self.discover_video_params();
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}
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Event::IceConnectionStateChange(IceConnectionState::Disconnected) => {
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tracing::warn!("WebRTC disconnected");
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self.connected = false;
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return Ok(true);
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}
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Event::MediaAdded(ma) => {
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tracing::info!("Media added: mid={} kind={:?}", ma.mid, ma.kind);
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if ma.kind == MediaKind::Video {
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if let Some(media) = self.rtc.media(ma.mid) {
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if media.direction().is_sending() && self.video_mid.is_none() {
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self.video_mid = Some(ma.mid);
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tracing::info!("Captured video mid: {}", ma.mid);
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self.discover_video_params();
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}
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}
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}
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}
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Event::KeyframeRequest(_) => {
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tracing::info!("received keyframe request from viewer");
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self.request_keyframe_from_viewer();
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}
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Event::EgressBitrateEstimate(est) => {
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let bitrate = match est {
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BweKind::Twcc(b) => *b,
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BweKind::Remb(_, b) => *b,
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_ => {
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tracing::debug!("BWE estimate: unrecognized kind, skipping");
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continue;
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}
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};
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tracing::info!("BWE estimate: {bitrate}");
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self.current_bwe_estimate = Some(bitrate);
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}
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_ => {
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tracing::debug!("WebRTC event: {:?}", e);
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}
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}
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}
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Ok(Output::Timeout(_t)) => break,
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Err(e) => {
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tracing::error!("rtc.poll_output error: {e}");
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self.connected = false;
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return Ok(true);
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}
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}
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}
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Ok(false)
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}
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fn feed_network(&mut self) -> Result<()> {
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let mut recv_count = 0u32;
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loop {
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match self.socket.recv_from(&mut self.buf) {
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|
Ok((n, source)) => {
|
|
recv_count += 1;
|
|
if recv_count <= 5 {
|
|
tracing::trace!("UDP recv {} bytes from {}", n, source);
|
|
}
|
|
let input = Input::Receive(
|
|
Instant::now(),
|
|
Receive {
|
|
proto: Protocol::Udp,
|
|
source,
|
|
destination: self.udp_addr,
|
|
contents: self.buf[..n]
|
|
.try_into()
|
|
.map_err(|e| anyhow::anyhow!("receive contents: {e}"))?,
|
|
},
|
|
);
|
|
self.rtc.handle_input(input).map_err(|e| {
|
|
anyhow::anyhow!("handle_input({n} bytes from {source}): {e}")
|
|
})?;
|
|
}
|
|
Err(ref e) if e.kind() == std::io::ErrorKind::WouldBlock => break,
|
|
Err(ref e) if e.kind() == std::io::ErrorKind::Interrupted => continue,
|
|
Err(e) => bail!("UDP recv error: {e}"),
|
|
}
|
|
}
|
|
|
|
self.rtc
|
|
.handle_input(Input::Timeout(Instant::now()))
|
|
.map_err(|e| anyhow::anyhow!("handle timeout: {e}"))?;
|
|
|
|
Ok(())
|
|
}
|
|
|
|
fn write_h264_frame(&mut self, data: &[u8], pts_ticks: i64) -> Result<bool> {
|
|
if !self.connected {
|
|
return Ok(false);
|
|
}
|
|
|
|
let mid = match self.video_mid {
|
|
Some(m) => m,
|
|
None => {
|
|
tracing::debug!("write_h264: no video_mid");
|
|
return Ok(false);
|
|
}
|
|
};
|
|
let pt = match self.video_pt {
|
|
Some(p) => p,
|
|
None => {
|
|
tracing::debug!("write_h264: no video_pt");
|
|
return Ok(false);
|
|
}
|
|
};
|
|
|
|
if self.need_keyframe {
|
|
if !is_idr_nalu(data) {
|
|
tracing::debug!(
|
|
"write_h264: skipping non-IDR frame ({} bytes), waiting for keyframe",
|
|
data.len()
|
|
);
|
|
return Ok(false);
|
|
}
|
|
tracing::info!(
|
|
"write_h264: got IDR keyframe ({} bytes), starting playback",
|
|
data.len()
|
|
);
|
|
self.need_keyframe = false;
|
|
}
|
|
|
|
let rtp_timestamp = rtp_timestamp_from_pts_ticks(pts_ticks);
|
|
self.rtp_clock = rtp_timestamp as u32;
|
|
let rtp_time = MediaTime::new(rtp_timestamp, Frequency::NINETY_KHZ);
|
|
|
|
let writer = match self.rtc.writer(mid) {
|
|
Some(w) => w,
|
|
None => {
|
|
tracing::debug!("write_h264: no writer for mid={mid}");
|
|
return Ok(false);
|
|
}
|
|
};
|
|
|
|
tracing::debug!(
|
|
"write_h264: {} bytes, pt={:?}, rtp={}",
|
|
data.len(),
|
|
pt,
|
|
self.rtp_clock
|
|
);
|
|
writer
|
|
.write(pt, Instant::now(), rtp_time, data)
|
|
.map_err(|e| anyhow::anyhow!("writer.write: {e}"))?;
|
|
|
|
let should_destroy = self.poll_rtc()?;
|
|
|
|
Ok(should_destroy)
|
|
}
|
|
|
|
fn is_connected(&self) -> bool {
|
|
self.connected
|
|
}
|
|
}
|
|
|
|
/// Convert PTS in 90kHz media-clock ticks to RTP MediaTime ticks (u64).
|
|
///
|
|
/// With WebRTC encoder time_base = 1/90000, pts_ticks ARE RTP timestamps.
|
|
/// No fps-based conversion needed. Returned as u64 to feed MediaTime::new
|
|
/// without premature 13.25-hour u32 wrap; str0m handles RTP u32 wrap internally.
|
|
pub fn rtp_timestamp_from_pts_ticks(pts_ticks: i64) -> u64 {
|
|
pts_ticks.max(0) as u64
|
|
}
|
|
|
|
// ── 工具函数 ──────────────────────────────────────────────────────────────
|
|
|
|
/// 从 HTTP 请求中提取 body（在 \r\n\r\n 之后）
|
|
fn extract_body(req: &str) -> &str {
|
|
if let Some(idx) = req.find("\r\n\r\n") {
|
|
req.get(idx + 4..).unwrap_or("")
|
|
} else {
|
|
""
|
|
}
|
|
}
|
|
|
|
fn local_ip() -> Option<String> {
|
|
std::net::UdpSocket::bind("0.0.0.0:0").ok().and_then(|s| {
|
|
s.connect("1.1.1.1:80").ok()?;
|
|
let addr = s.local_addr().ok()?;
|
|
drop(s);
|
|
let ip = addr.ip().to_string();
|
|
if ip == "0.0.0.0" || ip.starts_with("127.") {
|
|
return None;
|
|
}
|
|
Some(ip)
|
|
})
|
|
}
|
|
|
|
fn is_idr_nalu(data: &[u8]) -> bool {
|
|
let mut i = 0;
|
|
while i < data.len() {
|
|
let tail = &data[i..];
|
|
if tail.starts_with(&[0, 0, 0, 1]) {
|
|
let Some(&header) = tail.get(4) else { break };
|
|
if header & 0x1F == 5 {
|
|
return true;
|
|
}
|
|
i += 5;
|
|
} else if tail.starts_with(&[0, 0, 1]) {
|
|
let Some(&header) = tail.get(3) else { break };
|
|
if header & 0x1F == 5 {
|
|
return true;
|
|
}
|
|
i += 4;
|
|
} else {
|
|
i += 1;
|
|
}
|
|
}
|
|
false
|
|
}
|
|
|
|
#[cfg(test)]
|
|
mod tests {
|
|
use super::*;
|
|
|
|
#[test]
|
|
fn empty_data() {
|
|
assert!(!is_idr_nalu(&[]));
|
|
}
|
|
|
|
#[test]
|
|
fn short_data_no_start_code() {
|
|
assert!(!is_idr_nalu(&[0]));
|
|
assert!(!is_idr_nalu(&[0, 0]));
|
|
assert!(!is_idr_nalu(&[1, 2, 3]));
|
|
}
|
|
|
|
#[test]
|
|
fn three_byte_start_code_no_nal_header() {
|
|
assert!(!is_idr_nalu(&[0, 0, 1]));
|
|
}
|
|
|
|
#[test]
|
|
fn four_byte_start_code_no_nal_header() {
|
|
assert!(!is_idr_nalu(&[0, 0, 0, 1]));
|
|
}
|
|
|
|
#[test]
|
|
fn three_byte_start_code_idr_at_tail() {
|
|
assert!(is_idr_nalu(&[0, 0, 1, 0x65]));
|
|
assert!(!is_idr_nalu(&[0, 0, 1, 0x01]));
|
|
}
|
|
|
|
#[test]
|
|
fn four_byte_start_code_idr_at_tail() {
|
|
assert!(is_idr_nalu(&[0, 0, 0, 1, 0x65]));
|
|
assert!(!is_idr_nalu(&[0, 0, 0, 1, 0x01]));
|
|
}
|
|
|
|
#[test]
|
|
fn idr_in_middle_of_frame() {
|
|
let data: Vec<u8> = [
|
|
&[0, 0, 0, 1, 0x67][..], // SPS
|
|
&[0, 0, 0, 1, 0x68][..], // PPS
|
|
&[0, 0, 0, 1, 0x65][..], // IDR
|
|
]
|
|
.concat();
|
|
assert!(is_idr_nalu(&data));
|
|
}
|
|
|
|
#[test]
|
|
fn no_idr_in_frame() {
|
|
let data: Vec<u8> = [
|
|
&[0, 0, 0, 1, 0x67][..], // SPS
|
|
&[0, 0, 0, 1, 0x68][..], // PPS
|
|
]
|
|
.concat();
|
|
assert!(!is_idr_nalu(&data));
|
|
}
|
|
|
|
#[test]
|
|
fn mixed_start_code_lengths() {
|
|
let data: Vec<u8> = [
|
|
&[0, 0, 0, 1, 0x67][..], // SPS (4-byte start code)
|
|
&[0, 0, 1, 0x65][..], // IDR (3-byte start code)
|
|
]
|
|
.concat();
|
|
assert!(is_idr_nalu(&data));
|
|
}
|
|
|
|
#[test]
|
|
fn all_zeros() {
|
|
assert!(!is_idr_nalu(&[0, 0, 0, 0, 0, 0, 0, 0]));
|
|
}
|
|
|
|
// ── Task 5: BWE estimate handling ──
|
|
|
|
#[test]
|
|
fn bitrate_mbps_conversion() {
|
|
let b = Bitrate::mbps(5);
|
|
assert!(b.as_u64() > 0, "5 Mbps should be > 0 bps");
|
|
assert_eq!(b.as_u64(), 5_000_000, "5 Mbps should be 5,000,000 bps");
|
|
}
|
|
|
|
#[test]
|
|
fn bitrate_bps_conversion() {
|
|
let b = Bitrate::bps(1_234_567);
|
|
assert_eq!(b.as_u64(), 1_234_567);
|
|
}
|
|
|
|
#[test]
|
|
fn default_bwe_is_5mbps() {
|
|
let default = Bitrate::mbps(5);
|
|
let bps = default.as_u64();
|
|
assert_eq!(bps, 5_000_000);
|
|
}
|
|
|
|
// ── RTP timestamp conversion (issue #24) ──
|
|
|
|
#[test]
|
|
fn rtp_timestamp_zero_pts() {
|
|
assert_eq!(rtp_timestamp_from_pts_ticks(0), 0);
|
|
}
|
|
|
|
#[test]
|
|
fn rtp_timestamp_one_frame_at_60fps() {
|
|
// 16.7ms at 90kHz = ~1500 ticks. Real time maps directly to ticks now.
|
|
assert_eq!(rtp_timestamp_from_pts_ticks(1500), 1500);
|
|
}
|
|
|
|
#[test]
|
|
fn rtp_timestamp_one_second() {
|
|
// 1 second at 90kHz = 90000 ticks
|
|
assert_eq!(rtp_timestamp_from_pts_ticks(90_000), 90_000);
|
|
}
|
|
|
|
#[test]
|
|
fn rtp_timestamp_negative_clamps_to_zero() {
|
|
assert_eq!(rtp_timestamp_from_pts_ticks(-5), 0);
|
|
}
|
|
|
|
#[test]
|
|
fn rtp_timestamp_u64_no_truncation() {
|
|
// Value above u32::MAX should NOT truncate when feeding MediaTime
|
|
let large = u32::MAX as i64 + 1000;
|
|
assert_eq!(rtp_timestamp_from_pts_ticks(large), large as u64);
|
|
}
|
|
}
|