unarr/internal/engine/webrtc_stream.go
Deivid Soto 209ea38ecf feat(transcode): dynamic H.264 level + HW probe + capability reporting
Three related fixes around 4K-source transcoding that left the web
player stuck on "preparing session" with no useful diagnostics:

1. Dynamic -level:v derived from output height (hls.go, transcoder.go).
   The previous fixed "4.0" silently rejected anything taller than 1080p
   inside libx264 — "frame MB size > level limit", "DPB size > level
   limit" — and emitted unplayable segments. Helper H264LevelForHeight()
   now picks 4.0 / 5.0 / 5.1 / 6.0 from the actual encode height.

2. New `unarr probe-hwaccel` diagnostic command. Lists the HW encoders
   compiled into ffmpeg, the device files / drivers present, and the
   backend the daemon would actually pick today. Surfaces the canonical
   gotcha: a host with an RTX 3090 + nvidia-smi but a Homebrew ffmpeg
   built without --enable-nvenc still falls back to libx264 software.

3. Register payload now includes hwAccel + maxTranscodeHeight so the web
   side can suggest a smaller alternate quality before the user even
   tries to play a 4K source on a software-only host. Software-only =
   1080p cap, any HW backend = 2160p cap.
2026-05-08 15:57:02 +02:00

784 lines
23 KiB
Go

// Package engine — webrtc_stream.go implements the daemon side of the custom
// WebRTC byte-streaming protocol. The browser opens an RTCDataChannel via
// SDP exchange (signalled over the web's HTTP + SSE relay); this code:
//
// 1. Parses the browser's SDP offer.
// 2. Creates a pion PeerConnection bound to the configured ICE servers.
// 3. Answers + trickles its own ICE candidates back through the signal client.
// 4. On DataChannel open, sends a HELLO frame describing the file.
// 5. Services RangeReq frames by reading from disk and emitting RangeData
// chunks (16 KiB each) followed by a RangeEnd.
// 6. Honours app-level backpressure via SetBufferedAmountLowThreshold +
// OnBufferedAmountLow — Chromium closes a DataChannel when bufferedAmount
// exceeds 16 MiB, so we MUST pause the writer.
//
// No anacrolix, no torrent metadata. Just a peer-to-peer file server over
// WebRTC. Pass-through path; transcoding lives in transcoder.go (Fase 2.5).
package engine
import (
"context"
"encoding/json"
"errors"
"fmt"
"io"
"math"
"path/filepath"
"sync"
"sync/atomic"
"time"
"github.com/pion/webrtc/v4"
"github.com/torrentclaw/unarr/internal/agent"
"github.com/torrentclaw/unarr/internal/engine/wire"
)
// Tunables — values match the protocol spec in plan/clever-weaving-dove.md.
const (
// dcChunkPayload is the per-frame application payload size. Must match
// wire.MaxChunkPayload so RangeData frames fit one SCTP message.
dcChunkPayload = wire.MaxChunkPayload
// dcHighWatermark is the bufferedAmount cap above which the writer pauses.
// Chromium closes DCs above 16 MiB; pause well below.
dcHighWatermark = 8 << 20
// dcLowWatermark triggers OnBufferedAmountLow → resume the writer.
dcLowWatermark = 1 << 20
// rangeReqConcurrency is the cap on in-flight range responses per session.
rangeReqConcurrency = 4
// helloDeadline is the max wait for the DataChannel to open after answer.
helloDeadline = 30 * time.Second
)
// WebRTCStreamConfig describes a single browser ↔ daemon stream session.
type WebRTCStreamConfig struct {
SessionID string
FilePath string
FileName string
FileSize int64
ICEServers []webrtc.ICEServer
Signal *agent.Client
// Logger receives diagnostic events; a nil logger swallows everything.
Logger StreamLogger
// Transcode steers on-the-fly transcoding when source codecs are not
// browser-decodable (HEVC/AV1/AC3/DTS). Empty FFmpegPath disables it.
Transcode TranscodeRuntime
// Quality overrides the cap from Transcode for this session. One of
// "2160p" | "1080p" | "720p" | "480p" | "original" | "" (= defer to
// Transcode defaults).
Quality string
}
// TranscodeRuntime carries the resolved ffmpeg/ffprobe paths + tunables so
// each session can decide whether to passthrough or pipe through ffmpeg.
type TranscodeRuntime struct {
FFmpegPath string
FFprobePath string
HWAccel HWAccel
Preset string
VideoBitrate string
AudioBitrate string
MaxHeight int
// Disabled forces passthrough for every file even when codecs are not
// browser-friendly. Useful when the user explicitly turns transcoding
// off in config.
Disabled bool
}
// StreamLogger is an injectable logger so tests can capture events.
type StreamLogger interface {
Infof(format string, args ...any)
Warnf(format string, args ...any)
Errorf(format string, args ...any)
}
type nopLogger struct{}
func (nopLogger) Infof(string, ...any) {}
func (nopLogger) Warnf(string, ...any) {}
func (nopLogger) Errorf(string, ...any) {}
func logger(l StreamLogger) StreamLogger {
if l == nil {
return nopLogger{}
}
return l
}
// qualityCap maps a session's Quality label to a (MaxHeight, VideoBitrate)
// pair. An empty label or "original" returns zero-values, signalling "no
// override" to the caller.
type qualityCap struct {
MaxHeight int
VideoBitrate string // ffmpeg -b:v string, e.g. "3500k"
}
func resolveQualityCap(label string) qualityCap {
switch label {
case "2160p":
return qualityCap{MaxHeight: 2160, VideoBitrate: "25000k"}
case "1080p":
return qualityCap{MaxHeight: 1080, VideoBitrate: "6000k"}
case "720p":
return qualityCap{MaxHeight: 720, VideoBitrate: "3500k"}
case "480p":
return qualityCap{MaxHeight: 480, VideoBitrate: "1500k"}
default:
// "original", "auto", "" → defer to config.
return qualityCap{}
}
}
// buildStreamSource picks between passthrough and transcoded source. ffprobe
// failure or missing ffmpeg falls back to passthrough — the browser surfaces
// a clearer codec error than us refusing to start.
//
// Quality override (cfg.Quality) can force a downscale even when the source
// codec is browser-friendly: a 4K h264 file watched on a phone with quality
// "720p" must transcode (otherwise we'd ship 4K bytes for a 6" screen).
func buildStreamSource(
ctx context.Context,
abs string,
displayName string,
cfg WebRTCStreamConfig,
log StreamLogger,
) (streamSource, error) {
tc := cfg.Transcode
qcap := resolveQualityCap(cfg.Quality)
if tc.Disabled || tc.FFmpegPath == "" || tc.FFprobePath == "" {
return newDiskFileSource(abs)
}
probe, err := ProbeFile(ctx, tc.FFprobePath, abs)
if err != nil {
log.Warnf("[wrtc %s] probe failed (%v) — passthrough", agent.ShortID(cfg.SessionID), err)
return newDiskFileSource(abs)
}
action := DecideAction(probe)
// Quality cap can promote a passthrough/remux decision into a full video
// transcode when the source resolution exceeds the requested cap.
if qcap.MaxHeight > 0 && probe.Height > 0 && probe.Height > qcap.MaxHeight && action != ActionTranscodeVideo {
log.Infof("[wrtc %s] quality=%s caps height %d→%d — forcing video transcode",
agent.ShortID(cfg.SessionID), cfg.Quality, probe.Height, qcap.MaxHeight)
action = ActionTranscodeVideo
}
if action == ActionPassthrough {
log.Infof("[wrtc %s] codec passthrough (%s + %s in %s)",
agent.ShortID(cfg.SessionID), probe.VideoCodec, probe.AudioCodec, probe.Container)
return newDiskFileSource(abs)
}
log.Infof("[wrtc %s] transcoding %s/%s/%s → h264+aac (%s, quality=%s)",
agent.ShortID(cfg.SessionID), probe.Container, probe.VideoCodec, probe.AudioCodec,
action, coalesce(cfg.Quality, "default"))
maxHeight := tc.MaxHeight
videoBitrate := tc.VideoBitrate
if qcap.MaxHeight > 0 {
maxHeight = qcap.MaxHeight
videoBitrate = qcap.VideoBitrate
}
opts := TranscodeOpts{
Action: action,
HWAccel: tc.HWAccel,
Preset: tc.Preset,
VideoBitrate: videoBitrate,
AudioBitrate: tc.AudioBitrate,
MaxHeight: maxHeight,
SourceHeight: probe.Height,
FFmpegPath: tc.FFmpegPath,
}
return newTranscodeSource(ctx, abs, probe, action, opts, displayName)
}
// RunWebRTCStream blocks until the session ends — either the DataChannel
// closes, the peer connection drops, or ctx is cancelled. Always returns a
// non-nil error explaining the termination reason.
func RunWebRTCStream(ctx context.Context, cfg WebRTCStreamConfig) error {
log := logger(cfg.Logger)
if cfg.SessionID == "" {
return errors.New("webrtc_stream: empty SessionID")
}
if cfg.FilePath == "" {
return errors.New("webrtc_stream: empty FilePath")
}
abs, err := filepath.Abs(cfg.FilePath)
if err != nil {
return fmt.Errorf("webrtc_stream: resolve path: %w", err)
}
displayName := cfg.FileName
if displayName == "" {
displayName = filepath.Base(abs)
}
// Decide passthrough vs transcoding. Probe is best-effort: if ffprobe
// is missing or fails we fall back to passthrough (the browser will
// surface a clearer error than us guessing wrong).
source, err := buildStreamSource(ctx, abs, displayName, cfg, log)
if err != nil {
return fmt.Errorf("webrtc_stream: build source: %w", err)
}
defer source.Close()
// 1. Build PeerConnection.
api := webrtc.NewAPI()
pc, err := api.NewPeerConnection(webrtc.Configuration{
ICEServers: cfg.ICEServers,
})
if err != nil {
return fmt.Errorf("webrtc_stream: new peer connection: %w", err)
}
defer pc.Close()
sessionCtx, cancelSession := context.WithCancel(ctx)
defer cancelSession()
// Stop the session when ICE drops permanently. "Disconnected" is
// transient per RFC 8445 (NAT rebind, brief packet loss) — wait for
// "Failed" or "Closed" before tearing down.
pc.OnICEConnectionStateChange(func(state webrtc.ICEConnectionState) {
log.Infof("[wrtc %s] ice=%s", agent.ShortID(cfg.SessionID), state.String())
switch state {
case webrtc.ICEConnectionStateFailed,
webrtc.ICEConnectionStateClosed:
cancelSession()
case webrtc.ICEConnectionStateUnknown,
webrtc.ICEConnectionStateNew,
webrtc.ICEConnectionStateChecking,
webrtc.ICEConnectionStateConnected,
webrtc.ICEConnectionStateCompleted,
webrtc.ICEConnectionStateDisconnected:
// Disconnected is transient (RFC 8445 — NAT rebind / packet loss);
// the others are normal progress states. Don't tear the session down.
}
})
// Trickle our ICE candidates back to the browser.
// PostSignal runs on its own goroutine so a slow signal server can't
// stall pion's ICE-gathering thread.
pc.OnICECandidate(func(c *webrtc.ICECandidate) {
if c == nil {
go func() {
_ = cfg.Signal.PostSignal(sessionCtx, cfg.SessionID, agent.SignalMessage{
Type: agent.SignalMsgCandidateEnd,
Payload: "",
})
}()
return
}
init := c.ToJSON()
payload, _ := json.Marshal(init)
go func() {
_ = cfg.Signal.PostSignal(sessionCtx, cfg.SessionID, agent.SignalMessage{
Type: agent.SignalMsgCandidate,
Payload: string(payload),
})
}()
})
// Browser is the offerer — we react to the DataChannel it creates.
dcReady := make(chan *webrtc.DataChannel, 1)
pc.OnDataChannel(func(dc *webrtc.DataChannel) {
log.Infof("[wrtc %s] data channel '%s' open", agent.ShortID(cfg.SessionID), dc.Label())
select {
case dcReady <- dc:
default:
// Browser opened a second DC — ignore, we only serve one.
log.Warnf("[wrtc %s] extra data channel ignored", agent.ShortID(cfg.SessionID))
}
})
// 2. Drive the SDP exchange. Any error from the loop (browser sent
// "bye", signal stream closed, etc.) cancels the session so we don't
// dangle on the DC waiting for a peer that's already gone.
sdpDone := make(chan error, 1)
go func() {
err := runSDPExchange(sessionCtx, pc, cfg)
sdpDone <- err
if err != nil && sessionCtx.Err() == nil {
log.Infof("[wrtc %s] signal loop ended: %v", agent.ShortID(cfg.SessionID), err)
cancelSession()
}
}()
// 3. Wait for either SDP error or DataChannel open.
var dc *webrtc.DataChannel
select {
case err := <-sdpDone:
if err != nil {
return fmt.Errorf("sdp exchange: %w", err)
}
// SDP complete — wait for the DC.
select {
case dc = <-dcReady:
case <-time.After(helloDeadline):
return errors.New("webrtc_stream: data channel never opened")
case <-sessionCtx.Done():
return sessionCtx.Err()
}
case dc = <-dcReady:
// DC opened before SDP loop reported done (typical: the loop keeps
// running to ferry remote ICE candidates).
case <-sessionCtx.Done():
return sessionCtx.Err()
}
// 4. Wire up the data channel pump.
pump := newDataChannelPump(dc, source, log, cancelSession)
dc.OnOpen(pump.onOpen)
dc.OnMessage(pump.onMessage)
dc.OnClose(func() {
log.Infof("[wrtc %s] data channel closed", agent.ShortID(cfg.SessionID))
cancelSession()
})
<-sessionCtx.Done()
pump.shutdown()
return sessionCtx.Err()
}
// runSDPExchange consumes signal events from the browser and answers the SDP
// offer. Keeps running for the lifetime of sessionCtx so trickle candidates
// flow in both directions. Reopens the SSE stream on every clean close — the
// server caps each response at ~25 s.
func runSDPExchange(ctx context.Context, pc *webrtc.PeerConnection, cfg WebRTCStreamConfig) error {
gotOffer := false
for ctx.Err() == nil {
stream, err := cfg.Signal.OpenSignalStream(ctx, cfg.SessionID)
if err != nil {
if ctx.Err() != nil {
return ctx.Err()
}
return fmt.Errorf("open signal stream: %w", err)
}
err = consumeSignalStream(ctx, pc, cfg, stream, &gotOffer)
stream.Close()
if err != nil {
return err
}
}
return ctx.Err()
}
// consumeSignalStream drains a single SSE connection until it closes or
// produces a hard error. Returns nil on a clean server-side disconnect so the
// caller can reopen.
func consumeSignalStream(
ctx context.Context,
pc *webrtc.PeerConnection,
cfg WebRTCStreamConfig,
stream *agent.SignalEventStream,
gotOffer *bool,
) error {
for {
select {
case <-ctx.Done():
return ctx.Err()
case msg, ok := <-stream.Events():
if !ok {
if err := stream.Err(); err != nil {
return fmt.Errorf("signal stream: %w", err)
}
return nil
}
if err := handleSignal(ctx, pc, cfg, msg, gotOffer); err != nil {
return err
}
}
}
}
func handleSignal(
ctx context.Context,
pc *webrtc.PeerConnection,
cfg WebRTCStreamConfig,
msg agent.SignalMessage,
gotOffer *bool,
) error {
switch msg.Type {
case agent.SignalMsgAnswer:
// Browser is the offerer in our protocol — we never expect an answer
// from the other side. Drop silently (also satisfies exhaustive lint).
return nil
case agent.SignalMsgOffer:
if *gotOffer {
return nil // ignore duplicates
}
var offer webrtc.SessionDescription
if err := json.Unmarshal([]byte(msg.Payload), &offer); err != nil {
return fmt.Errorf("decode offer: %w", err)
}
if err := pc.SetRemoteDescription(offer); err != nil {
return fmt.Errorf("set remote description: %w", err)
}
answer, err := pc.CreateAnswer(nil)
if err != nil {
return fmt.Errorf("create answer: %w", err)
}
if err := pc.SetLocalDescription(answer); err != nil {
return fmt.Errorf("set local description: %w", err)
}
// Send back the local description *with* gathered candidates so far —
// remaining candidates trickle separately via OnICECandidate.
ld := pc.LocalDescription()
payload, _ := json.Marshal(ld)
if err := cfg.Signal.PostSignal(ctx, cfg.SessionID, agent.SignalMessage{
Type: agent.SignalMsgAnswer,
Payload: string(payload),
}); err != nil {
return fmt.Errorf("post answer: %w", err)
}
*gotOffer = true
case agent.SignalMsgCandidate:
if !*gotOffer {
// Browser may trickle candidates before we've seen the offer in
// rare race conditions — drop. Browser will retransmit.
return nil
}
var init webrtc.ICECandidateInit
if err := json.Unmarshal([]byte(msg.Payload), &init); err != nil {
return fmt.Errorf("decode candidate: %w", err)
}
if err := pc.AddICECandidate(init); err != nil {
return fmt.Errorf("add ice candidate: %w", err)
}
case agent.SignalMsgCandidateEnd:
// No-op — pion gathers complete on its own.
case agent.SignalMsgBye:
return errors.New("browser sent bye")
}
return nil
}
// dataChannelPump owns the DC + stream source and serves wire-protocol frames.
type dataChannelPump struct {
dc *webrtc.DataChannel
source streamSource
log StreamLogger
cancel context.CancelFunc
// Flow control: writers wait on resumeCh when bufferedAmount goes high.
paused atomic.Bool
resumeCh chan struct{}
// Active range responses keyed by stream_id so CANCEL frames can stop them.
activeMu sync.Mutex
active map[uint32]context.CancelFunc
// Bound concurrent in-flight responses.
sem chan struct{}
// closed once shutdown() has been called.
closed atomic.Bool
}
func newDataChannelPump(
dc *webrtc.DataChannel,
source streamSource,
log StreamLogger,
cancel context.CancelFunc,
) *dataChannelPump {
p := &dataChannelPump{
dc: dc,
source: source,
log: log,
cancel: cancel,
resumeCh: make(chan struct{}, 1),
active: make(map[uint32]context.CancelFunc),
sem: make(chan struct{}, rangeReqConcurrency),
}
dc.SetBufferedAmountLowThreshold(dcLowWatermark)
dc.OnBufferedAmountLow(p.onBufferedAmountLow)
return p
}
func (p *dataChannelPump) onOpen() {
// Use estimated size for transcoded streams so the browser scrubber has
// something to anchor on. Real size is reflected by Range responses as
// ffmpeg writes more bytes; the estimate just bootstraps the UI.
announceSize := p.source.EstimatedSize()
transcoding := p.source.Transcoded()
// Browsers refuse to start playback when Content-Length is 0. If we don't
// have a duration estimate (e.g. ffprobe couldn't tag the source), declare
// a large sentinel so the browser issues range requests; the Transcoding
// flag tells it the value is provisional.
if transcoding && announceSize <= 0 {
announceSize = math.MaxInt64
}
// Seekable=true even for transcoded sources because we read from a tmp
// file (random access). Seek backwards just works; seek forward beyond
// what ffmpeg has produced will block briefly inside ReadAt.
seekable := true
hello := wire.HelloPayload{
FileSize: uint64(announceSize),
Transcoding: transcoding,
Seekable: seekable,
FileName: p.source.FileName(),
}
payload := wire.EncodeHello(hello)
frame := wire.EncodeFrame(wire.Header{
Type: wire.FrameHello,
Flags: wire.HelloFlags(transcoding, seekable),
StreamID: 0,
Length: uint32(len(payload)),
}, payload)
if err := p.dc.Send(frame); err != nil {
p.log.Errorf("send hello: %v", err)
p.cancel()
}
}
func (p *dataChannelPump) onMessage(msg webrtc.DataChannelMessage) {
if len(msg.Data) < wire.HeaderSize {
p.log.Warnf("dc: short frame %d bytes", len(msg.Data))
return
}
hdr, err := wire.DecodeHeader(msg.Data[:wire.HeaderSize])
if err != nil {
p.log.Warnf("dc: bad header: %v", err)
return
}
payload := msg.Data[wire.HeaderSize:]
if uint32(len(payload)) != hdr.Length {
p.log.Warnf("dc: payload length mismatch: hdr=%d got=%d", hdr.Length, len(payload))
return
}
switch hdr.Type {
case wire.FrameRangeReq:
req, err := wire.DecodeRangeReq(payload)
if err != nil {
p.log.Warnf("dc: bad range_req: %v", err)
return
}
go p.serveRange(hdr.StreamID, req)
case wire.FrameCancel:
p.cancelStream(hdr.StreamID)
case wire.FramePing:
p.sendSimpleFrame(wire.FramePong, hdr.StreamID, nil)
case wire.FramePong:
// no-op
default:
p.log.Warnf("dc: unknown frame type 0x%02x", hdr.Type)
}
}
func (p *dataChannelPump) cancelStream(streamID uint32) {
p.activeMu.Lock()
cancel, ok := p.active[streamID]
delete(p.active, streamID)
p.activeMu.Unlock()
if ok {
cancel()
}
}
func (p *dataChannelPump) sendSimpleFrame(t wire.FrameType, streamID uint32, payload []byte) {
frame := wire.EncodeFrame(wire.Header{
Type: t,
StreamID: streamID,
Length: uint32(len(payload)),
}, payload)
if err := p.dc.Send(frame); err != nil {
p.log.Warnf("dc: send type=0x%02x: %v", t, err)
}
}
func (p *dataChannelPump) serveRange(streamID uint32, req wire.RangeReqPayload) {
if p.closed.Load() {
return
}
// Bound concurrency.
select {
case p.sem <- struct{}{}:
case <-time.After(5 * time.Second):
p.log.Warnf("dc: range_req sid=%d dropped (concurrency cap)", streamID)
p.sendRangeEnd(streamID, 1)
return
}
defer func() { <-p.sem }()
// Reject offsets above MaxInt64 — uint64→int64 narrowing would wrap to a
// negative value and bypass the bounds check, then ReadAt would be called
// with a negative offset.
currentSize := p.source.Size()
finalSize := p.source.EstimatedSize()
if req.Offset > math.MaxInt64 {
p.sendRangeEnd(streamID, 2) // out of range
return
}
// For transcoded streams `currentSize` grows over time; only reject when
// the offset is past the *estimated* final size.
if int64(req.Offset) >= finalSize && p.source.Final() {
p.sendRangeEnd(streamID, 2)
return
}
want := int64(req.Length)
if req.Length > math.MaxInt64 {
want = 0 // treat absurd length as "remainder of file"
}
// Cap by *final* size, not currentSize. For a still-transcoding stream
// currentSize grows over time and ReadAt below already blocks until
// ffmpeg produces the requested bytes (with a deadline). If we cap
// `want` by currentSize here we'll send an empty RangeEnd whenever the
// browser asks for bytes faster than ffmpeg writes them — which is
// always true on the first few seconds — and the browser then aborts
// playback with "Format error".
cap := finalSize
if !p.source.Final() && cap < int64(req.Offset)+1 {
// Estimate too small: serve as much as the browser asked for and
// let ReadAt block.
cap = int64(req.Offset) + want
}
if int64(req.Offset) >= cap && p.source.Final() {
// Past true end of a finished file.
p.sendRangeEnd(streamID, 0)
return
}
remaining := cap - int64(req.Offset)
if remaining < 0 {
remaining = 0
}
if want <= 0 || want > remaining {
want = remaining
}
p.log.Infof("dc: range_req sid=%d offset=%d wantReq=%d wantServe=%d currentSize=%d final=%v",
streamID, req.Offset, req.Length, want, currentSize, p.source.Final())
if want <= 0 {
// Only happens for a finished file when offset is at/past EOF.
p.sendRangeEnd(streamID, 0)
return
}
ctx, cancel := context.WithCancel(context.Background())
p.activeMu.Lock()
if p.active == nil {
p.activeMu.Unlock()
cancel()
p.sendRangeEnd(streamID, 3)
return
}
p.active[streamID] = cancel
p.activeMu.Unlock()
defer func() {
p.activeMu.Lock()
delete(p.active, streamID)
p.activeMu.Unlock()
cancel()
}()
buf := make([]byte, dcChunkPayload)
offset := int64(req.Offset)
end := offset + want
for offset < end {
if ctx.Err() != nil || p.closed.Load() {
return
}
// Wait if the DC is buffering too much.
if err := p.waitForLowWater(ctx); err != nil {
return
}
chunkLen := int64(len(buf))
if end-offset < chunkLen {
chunkLen = end - offset
}
n, rerr := p.source.ReadAt(buf[:chunkLen], offset)
if n > 0 {
// EOF on a short read means this is the final chunk — flag it so the
// browser doesn't wait for more data before processing RangeEnd.
isLast := offset+int64(n) >= end || rerr == io.EOF
if err := p.sendRangeData(streamID, buf[:n], isLast); err != nil {
p.log.Warnf("dc: send range_data sid=%d: %v", streamID, err)
return
}
offset += int64(n)
}
if rerr != nil {
if rerr == io.EOF {
break
}
p.log.Errorf("dc: read sid=%d: %v", streamID, rerr)
p.sendRangeEnd(streamID, 3)
return
}
}
p.sendRangeEnd(streamID, 0)
}
func (p *dataChannelPump) sendRangeData(streamID uint32, data []byte, last bool) error {
var flags uint8
if last {
flags |= wire.FlagLastChunk
}
frame := wire.EncodeFrame(wire.Header{
Type: wire.FrameRangeData,
Flags: flags,
StreamID: streamID,
Length: uint32(len(data)),
}, data)
return p.dc.Send(frame)
}
func (p *dataChannelPump) sendRangeEnd(streamID uint32, status uint32) {
payload := wire.EncodeRangeEnd(wire.RangeEndPayload{Status: status})
p.sendSimpleFrame(wire.FrameRangeEnd, streamID, payload)
}
func (p *dataChannelPump) waitForLowWater(ctx context.Context) error {
if p.dc.BufferedAmount() < dcHighWatermark {
return nil
}
p.paused.Store(true)
for {
// Drain any stale resume signal first.
select {
case <-p.resumeCh:
default:
}
if p.dc.BufferedAmount() < dcHighWatermark {
p.paused.Store(false)
return nil
}
select {
case <-ctx.Done():
return ctx.Err()
case <-p.resumeCh:
case <-time.After(500 * time.Millisecond):
// Belt-and-braces poll in case OnBufferedAmountLow misses a fire.
}
}
}
func (p *dataChannelPump) onBufferedAmountLow() {
if !p.paused.Load() {
return
}
select {
case p.resumeCh <- struct{}{}:
default:
}
}
func (p *dataChannelPump) shutdown() {
if !p.closed.CompareAndSwap(false, true) {
return
}
p.activeMu.Lock()
for _, cancel := range p.active {
cancel()
}
p.active = nil
p.activeMu.Unlock()
}