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Muse STT audio buffered before the websocket connection stays delayed for the whole stream

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#7,876 1 comentario 0 reacciones 0 asignados Ver en GitHub

Los mantenedores suelen responder en 1 día

Ya se ha fusionado un pull request relacionado.

  • #7890 de @DEVJHAWAR11 — fusionado

Evaluación

Dificultad
3/5
Tiempo estimado
1-2 días
Aptitud para principiantes
35/100
Tipo de issue
Error
Claridad
Bien especificado
Estado de actividad
Estancado
Stack tecnológico
python
Área
audio-video-rtc

Línea de trabajo

Start by reading SpeechStream._send_audio and _run in the Muse STT implementation, focusing on how _input_ch is queued while _connect_ws() runs and when the pacing clock starts. Reproduce the reported delays with the fake websocket and real-time frames described in the issue. Done means buffered packets are sent promptly after connection and subsequent packets arrive close to real time; an open pull request #7890 is already linked.

Escrito por el modelo de indexación a partir del texto del issue.

Descripción

bug
Bug Description

For Muse STT implementation, in SpeechStream._send_audio, pacing_origin is set when the first packet is sent. Every later packet then waits until pacing_origin + n × 80 ms before it is sent:

if pacing_origin is None:
    pacing_origin = loop.time()
deadline = pacing_origin + sent_duration

But before the first packet is sent, packets may already have stacked(between the session start and the Muse websocket connecting), since _run awaits _connect_ws() before starting _send_audio.
If x packets built up, every packet is sent x × 80 ms later than its original time, for the rest of the stream. The backlog is never caught up, because pacing never allows sending faster than real time.

As a result, Muse hears all audio late, and transcripts and end-of-turn events are late by the same amount.

Expected Behavior

Packets that stacked before the Muse websocket connects should be sent as soon as the connection is ready. After that, each packet should be sent once its audio is complete, so Muse receives audio close to real time no matter how long the connection took.

Reproduction Steps
**Repro:** running `_send_audio` with a fake websocket and frames arriving in real time:

| Connect time | Each packet sent late by |
| ------------ | ------------------------ |
| 0 ms         | 1 ms                     |
| 300 ms       | 221 ms                   |
| 1000 ms      | 921 ms                   |
Operating System

macos

Models Used

Muse, muse-voice-transcribe-1.0

Package Versions
livekit-agents==1.8.6
livekit-plugins-meta==1.8.6
Session/Room/Call IDs

No response

Proposed Solution
Send the buffered packets immediately, for example by skipping the sleep while `_input_ch` still has queued frames, or by starting the pacing clock when the first frame is pushed instead of when the first packet is sent.
Additional Context

No response

Screenshots and Recordings

No response

Lenguaje dominante
Python
Estrellas
14.6k
Forks
3.9k
Merge medio
3 d 18 h
PR fusionados (30 d)
193

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