mirror of
https://dev.narayana.im/narayana/telegabber.git
synced 2026-08-05 04:07:07 +00:00
233 lines
6.2 KiB
Go
233 lines
6.2 KiB
Go
package audio
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import (
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"errors"
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"fmt"
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"sync"
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"sync/atomic"
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"time"
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"github.com/pion/webrtc/v4"
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log "github.com/sirupsen/logrus"
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)
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type XmppToTgOptions struct {
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Ntg NtgClient
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ChatID int64 // P2P: equals the Telegram peer's user_id
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PrimePackets uint16 // buffered before playout starts; default 2 (~40ms)
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Logger *log.Entry
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}
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// xmpp->tg half: pion RemoteTrack RTP -> opus decode -> ntgcalls
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// SendMicrophonePCM. One reader goroutine per HandleTrack, plus one
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// playout ticker spawned by Start.
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type XmppToTg struct {
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opts XmppToTgOptions
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log *log.Entry
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dec *Decoder
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playout *Playout
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stop chan struct{}
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wg sync.WaitGroup
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pcmBuf []int16 // decoder output (worst-case 120ms stereo)
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silence []byte // 10ms silence for underrun
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closed atomic.Bool
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started atomic.Bool
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// false until Start runs. Audio between pion OnTrack (during
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// SetRemoteDescription) and bridge OnEstablished is drained from
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// pion's per-track queue but NOT enqueued - otherwise every packet
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// from the setup window would back up and the ticker would replay
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// it at real time, manifesting as call-long lag.
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livePush atomic.Bool
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preLiveDiscards atomic.Int64
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capture *captureWriter
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}
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func NewXmppToTg(opts XmppToTgOptions) (*XmppToTg, error) {
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if opts.Ntg == nil {
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return nil, errors.New("audio: Ntg required")
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}
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if opts.ChatID == 0 {
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return nil, errors.New("audio: ChatID required")
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}
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logger := opts.Logger
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if logger == nil {
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logger = log.WithField("module", "audio")
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}
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logger = logger.WithFields(log.Fields{"dir": "xmpp->tg", "chat_id": opts.ChatID})
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dec, err := NewDecoder()
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if err != nil {
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return nil, err
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}
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h := &XmppToTg{
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opts: opts,
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log: logger,
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dec: dec,
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playout: NewPlayout(opts.PrimePackets),
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pcmBuf: make([]int16, MaxDecodeSamplesPerChannel*Channels),
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silence: make([]byte, NtgFrameBytes),
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stop: make(chan struct{}),
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}
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if cap, err := newCaptureWriter(
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fmt.Sprintf("xmpp-to-tg-%d-%d.bin", opts.ChatID, time.Now().Unix()),
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magicRTP,
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); err != nil {
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logger.WithError(err).Warn("capture: xmpp->tg writer disabled")
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} else if cap != nil {
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logger.Info("capture: xmpp->tg RTP capture enabled")
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h.capture = cap
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}
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return h, nil
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}
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// flips on the ntgcalls Capture source, the playout ticker, and livePush
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func (h *XmppToTg) Start() error {
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if !h.started.CompareAndSwap(false, true) {
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return nil
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}
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if err := h.opts.Ntg.SetExternalMicrophone(h.opts.ChatID, true, SampleRate, Channels); err != nil {
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return fmt.Errorf("SetExternalMicrophone(capture): %w", err)
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}
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h.livePush.Store(true)
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h.wg.Add(1)
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go h.tickerLoop()
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h.log.WithField("pre_live_discards", h.preLiveDiscards.Load()).
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Info("xmpp->tg half started")
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return nil
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}
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// spawns a reader for each pc.OnTrack
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func (h *XmppToTg) HandleTrack(t *webrtc.TrackRemote) {
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if t == nil || t.Kind() != webrtc.RTPCodecTypeAudio {
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return
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}
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h.log.WithFields(log.Fields{
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"codec": t.Codec().MimeType,
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"ssrc": t.SSRC(),
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}).Info("xmpp->tg: remote audio track received from pion")
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h.wg.Add(1)
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go h.reader(t)
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}
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// stops both pumps; ntgcalls Capture+Playback are released by Call.Close
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func (h *XmppToTg) Close() error {
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if !h.closed.CompareAndSwap(false, true) {
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return nil
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}
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close(h.stop)
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h.wg.Wait()
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h.capture.close()
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h.log.Info("xmpp->tg half closed")
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return nil
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}
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func (h *XmppToTg) reader(t *webrtc.TrackRemote) {
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defer h.wg.Done()
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// negotiated PT for this track; peer is free to pick anything in 96-127,
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// hardcoding 111 would silently drop opus from peers that picked another
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expectedPT := uint8(t.PayloadType())
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for {
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select {
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case <-h.stop:
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return
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default:
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}
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pkt, _, err := t.ReadRTP()
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if err != nil {
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return
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}
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if pkt.PayloadType != expectedPT {
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// CN/DTMF/RED can share an SSRC; libopus rejects them
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continue
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}
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if !h.livePush.Load() {
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// pre-Start: drain pion's per-track queue but don't enqueue;
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// see livePush docstring for the lag rationale
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h.preLiveDiscards.Add(1)
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continue
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}
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h.capture.writeRTP(pkt.SSRC, pkt.SequenceNumber, pkt.Timestamp, pkt.PayloadType, pkt.Payload)
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h.playout.Push(pkt)
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}
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}
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// cross-tick playout state; separated so tests can drive processTick
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// step by step without a time.Ticker
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type tickState struct {
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lastDecodedSamples int // samples/ch from last Decode; DecodePLC output matches
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pending10ms [][]byte // 10ms PCM chunks awaiting shipment, one per tick
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}
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func newTickState() *tickState {
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return &tickState{lastDecodedSamples: SamplesPerOpusFrame}
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}
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func (h *XmppToTg) tickerLoop() {
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defer h.wg.Done()
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ticker := time.NewTicker(NtgFrameMs * time.Millisecond)
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defer ticker.Stop()
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st := newTickState()
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for {
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select {
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case <-h.stop:
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return
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case <-ticker.C:
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}
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h.processTick(st)
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}
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}
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// one iteration of the playout loop
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func (h *XmppToTg) processTick(st *tickState) {
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if len(st.pending10ms) == 0 {
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pkt, plcCount, err := h.playout.PopOrSkip()
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if err != nil {
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// empty or still priming - feed silence
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h.sendToNtg(h.silence)
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return
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}
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for i := 0; i < plcCount; i++ {
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plcOut := h.pcmBuf[:st.lastDecodedSamples*Channels]
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if err := h.dec.DecodePLC(plcOut); err != nil {
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h.log.WithError(err).Warn("DecodePLC failed")
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break
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}
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st.pending10ms = appendChunks10ms(st.pending10ms, plcOut)
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}
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n, err := h.dec.Decode(pkt.Payload, h.pcmBuf)
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if err != nil {
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h.log.WithError(err).Warn("opus decode failed")
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h.sendToNtg(h.silence)
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return
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}
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st.lastDecodedSamples = n
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st.pending10ms = appendChunks10ms(st.pending10ms, h.pcmBuf[:n*Channels])
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}
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chunk := st.pending10ms[0]
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st.pending10ms = st.pending10ms[1:]
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h.sendToNtg(chunk)
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}
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// slice interleaved stereo PCM into 10ms chunks; partial trailing chunks
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// are dropped (libopus's 2.5/5ms frames aren't multiples of 10ms)
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func appendChunks10ms(dst [][]byte, pcm []int16) [][]byte {
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samplesPerChunk := SamplesPerNtgFrame * Channels
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for off := 0; off+samplesPerChunk <= len(pcm); off += samplesPerChunk {
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buf := make([]byte, NtgFrameBytes)
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PCMInt16ToBytes(pcm[off:off+samplesPerChunk], buf)
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dst = append(dst, buf)
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}
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return dst
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}
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func (h *XmppToTg) sendToNtg(pcm []byte) {
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if err := h.opts.Ntg.SendMicrophonePCM(h.opts.ChatID, pcm); err != nil {
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h.log.WithError(err).Debug("SendMicrophonePCM failed")
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}
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}
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