From 2776a3104f076b1dbf33452ad88cb60036c39702 Mon Sep 17 00:00:00 2001
From: cai <cai@nbcai.cc>
Date: Fri, 10 Jul 2026 18:29:06 +0800
Subject: [PATCH] fix: decouple livekit audio drain from turn processing
---
src/main.rs | 595 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++-
1 files changed, 577 insertions(+), 18 deletions(-)
diff --git a/src/main.rs b/src/main.rs
index cf7cb4f..02f9fdb 100644
--- a/src/main.rs
+++ b/src/main.rs
@@ -1,7 +1,9 @@
+mod asr_realtime;
mod audio;
mod service;
use std::{
+ borrow::Cow,
env, fs,
path::{Path, PathBuf},
sync::{
@@ -12,6 +14,7 @@
};
use anyhow::{Context, Result, anyhow};
+use asr_realtime::{RealtimeAsrConfig, RealtimeAsrOutcome, RealtimeAsrUpload};
use audio::{AudioDiagnostics, load_pre_recorded_frames};
use base64::{Engine as _, engine::general_purpose};
use futures_util::StreamExt;
@@ -30,12 +33,18 @@
use reqwest::Client;
use serde::{Deserialize, Serialize};
use serde_json::json;
-use tokio::time::{sleep, sleep_until};
-use tokio::{sync::mpsc::UnboundedReceiver, task::JoinHandle};
+use tokio::time::{sleep, sleep_until, timeout};
+use tokio::{
+ sync::{mpsc, mpsc::UnboundedReceiver, watch},
+ task::JoinHandle,
+};
use tracing::{info, warn};
const USER_AUDIO_SAMPLE_RATE_HZ: u32 = 48_000;
const USER_AUDIO_NUM_CHANNELS: u16 = 1;
+const INBOUND_AUDIO_QUEUE_CAPACITY: usize = 100;
+const INBOUND_AUDIO_DROP_LOG_INTERVAL: u64 = 100;
+const AUDIO_DRAIN_STOP_GRACE: Duration = Duration::from_secs(2);
const DEFAULT_BOT_AUDIO_PROFILE: &str = "pcm-16k";
const LIVEKIT_48K_SAMPLE_RATE_HZ: u32 = 48_000;
const PCM_16K_SAMPLE_RATE_HZ: u32 = 16_000;
@@ -270,6 +279,11 @@
runtime_turn_bridge_url: Option<String>,
runtime_turn_bridge_token: Option<String>,
runtime_turn_bridge_mode: String,
+ runtime_asr_stream_enabled: bool,
+ runtime_asr_stream_url: Option<String>,
+ runtime_asr_realtime_enabled: bool,
+ runtime_asr_realtime_url: Option<String>,
+ runtime_asr_realtime_chunk_duration_ms: u64,
runtime_turn_artifact_dir: Option<String>,
runtime_session_nonce: Option<String>,
user_audio_observer_enabled: bool,
@@ -332,6 +346,11 @@
bridge_url: Option<String>,
bridge_token: Option<String>,
bridge_mode: String,
+ asr_stream_enabled: bool,
+ asr_stream_url: Option<String>,
+ asr_realtime_enabled: bool,
+ asr_realtime_url: Option<String>,
+ asr_realtime_chunk_duration_ms: u64,
artifact_dir: Option<String>,
runtime_session_nonce: Option<String>,
audio_debug_dump_dir: Option<String>,
@@ -343,6 +362,11 @@
bridge_url: config.runtime_turn_bridge_url.clone(),
bridge_token: config.runtime_turn_bridge_token.clone(),
bridge_mode: config.runtime_turn_bridge_mode.clone(),
+ asr_stream_enabled: config.runtime_asr_stream_enabled,
+ asr_stream_url: config.runtime_asr_stream_url.clone(),
+ asr_realtime_enabled: config.runtime_asr_realtime_enabled,
+ asr_realtime_url: config.runtime_asr_realtime_url.clone(),
+ asr_realtime_chunk_duration_ms: config.runtime_asr_realtime_chunk_duration_ms,
artifact_dir: config.runtime_turn_artifact_dir.clone(),
runtime_session_nonce: config.runtime_session_nonce.clone(),
audio_debug_dump_dir: config.audio_debug_dump_dir.clone(),
@@ -374,6 +398,32 @@
.as_deref()
.is_some_and(|value| value.trim_end_matches('/').ends_with("/stream"))
}
+
+ fn is_asr_stream_ready(&self) -> bool {
+ self.asr_stream_enabled
+ && self
+ .asr_stream_url
+ .as_ref()
+ .is_some_and(|value| !value.is_empty())
+ && self
+ .bridge_token
+ .as_ref()
+ .is_some_and(|value| !value.is_empty())
+ && self
+ .runtime_session_nonce
+ .as_ref()
+ .is_some_and(|value| !value.is_empty())
+ }
+
+ fn realtime_asr_config(&self) -> RealtimeAsrConfig {
+ RealtimeAsrConfig {
+ enabled: self.asr_realtime_enabled,
+ url: self.asr_realtime_url.clone(),
+ runtime_token: self.bridge_token.clone(),
+ runtime_session_nonce: self.runtime_session_nonce.clone(),
+ chunk_duration_ms: self.asr_realtime_chunk_duration_ms,
+ }
+ }
}
impl Config {
@@ -395,6 +445,14 @@
runtime_turn_bridge_token: optional_env("CV_RUNTIME_TURN_BRIDGE_TOKEN"),
runtime_turn_bridge_mode: env::var("CV_RUNTIME_TURN_BRIDGE_MODE")
.unwrap_or_else(|_| "json".to_string()),
+ runtime_asr_stream_enabled: bool_env("CV_RUNTIME_ASR_STREAM_ENABLED", false),
+ runtime_asr_stream_url: optional_env("CV_RUNTIME_ASR_STREAM_URL"),
+ runtime_asr_realtime_enabled: bool_env("CV_RUNTIME_ASR_REALTIME_ENABLED", false),
+ runtime_asr_realtime_url: optional_env("CV_RUNTIME_ASR_REALTIME_URL"),
+ runtime_asr_realtime_chunk_duration_ms: u64_env(
+ "CV_RUNTIME_ASR_REALTIME_CHUNK_DURATION_MS",
+ 200,
+ ),
runtime_turn_artifact_dir: optional_env("CV_RUNTIME_TURN_ARTIFACT_DIR"),
runtime_session_nonce: optional_env("CV_RUNTIME_SESSION_NONCE"),
user_audio_observer_enabled: bool_env("CV_ENABLE_USER_AUDIO_OBSERVER", true),
@@ -596,6 +654,7 @@
"traceId": trace_id,
"turnId": turn_id,
"eventName": event_name,
+ "eventWallTimeMs": current_time_millis(),
"result": result,
"reasonCode": reason_code,
"retryable": retryable,
@@ -750,6 +809,7 @@
call_id: &str,
trace_id: &str,
turn: FinishedSpeechTurn,
+ realtime_asr_result_ref: Option<String>,
) {
let turn_pipeline_started_at = Instant::now();
if !bridge_config.is_ready() {
@@ -806,9 +866,13 @@
"turnArtifactBytes": byte_size,
"frameCount": turn.frame_count,
"sampleCount": turn.sample_count,
- "endReason": turn.end_reason,
+ "endReason": turn.end_reason.as_str(),
}),
);
+ let asr_result_ref = match realtime_asr_result_ref {
+ Some(value) => Some(value),
+ None => request_asr_result_ref(http, bridge_config, call_id, trace_id, &turn).await,
+ };
if bridge_config.is_stream_mode() {
match request_turn_bridge_stream(
http,
@@ -819,6 +883,7 @@
&turn,
&path_ref,
byte_size,
+ asr_result_ref.as_deref(),
turn_pipeline_started_at,
)
.await
@@ -882,6 +947,7 @@
&turn,
&path_ref,
byte_size,
+ asr_result_ref.as_deref(),
turn_pipeline_started_at,
)
.await
@@ -1003,6 +1069,251 @@
Ok((path_ref, byte_size))
}
+async fn request_asr_result_ref(
+ http: &Client,
+ bridge_config: &TurnBridgeConfig,
+ call_id: &str,
+ trace_id: &str,
+ turn: &FinishedSpeechTurn,
+) -> Option<String> {
+ if !bridge_config.is_asr_stream_ready() {
+ return None;
+ }
+ match request_asr_stream(http, bridge_config, call_id, trace_id, turn).await {
+ Ok(Some(asr_result_ref)) => {
+ emit_activity(
+ call_id,
+ trace_id,
+ Some(&turn.turn_id),
+ "asr_stream_ref_ready",
+ "ok",
+ None,
+ None,
+ json!({
+ "asrResultRefPresent": true,
+ "format": "pcm_s16le",
+ "sampleRate": 16000,
+ "channels": 1,
+ }),
+ );
+ Some(asr_result_ref)
+ }
+ Ok(None) => None,
+ Err(error) => {
+ warn!(
+ call_id = %call_id,
+ trace_id = %trace_id,
+ turn_id = %turn.turn_id,
+ error = %safe_error(&error.to_string()),
+ "runtime helper asr_stream_failed_fallback"
+ );
+ emit_activity(
+ call_id,
+ trace_id,
+ Some(&turn.turn_id),
+ "asr_stream_fallback",
+ "skipped",
+ Some("ASR_STREAM_INTERRUPTED"),
+ Some(true),
+ json!({
+ "fallbackReason": "asr_stream_request_failed",
+ "fallbackStage": "asr_stream",
+ }),
+ );
+ None
+ }
+ }
+}
+
+async fn request_asr_stream(
+ http: &Client,
+ bridge_config: &TurnBridgeConfig,
+ call_id: &str,
+ trace_id: &str,
+ turn: &FinishedSpeechTurn,
+) -> Result<Option<String>> {
+ let started_at = Instant::now();
+ let ndjson = build_asr_stream_ndjson(call_id, trace_id, turn, bridge_config)?;
+ let response = http
+ .post(bridge_config.asr_stream_url.as_deref().unwrap_or_default())
+ .header("Content-Type", "application/x-ndjson")
+ .header(
+ "X-CV-Runtime-Token",
+ bridge_config.bridge_token.as_deref().unwrap_or_default(),
+ )
+ .header("X-CV-Call-Id", call_id)
+ .header("X-CV-Trace-Id", trace_id)
+ .header(
+ "X-CV-Runtime-Session-Nonce",
+ bridge_config
+ .runtime_session_nonce
+ .as_deref()
+ .unwrap_or_default(),
+ )
+ .body(ndjson)
+ .send()
+ .await
+ .context("failed to post asr stream")?;
+ let status = response.status();
+ if !status.is_success() {
+ let body_len = response.text().await.map(|body| body.len()).unwrap_or(0);
+ warn!(
+ call_id = %call_id,
+ trace_id = %trace_id,
+ turn_id = %turn.turn_id,
+ http_status = status.as_u16(),
+ body_len,
+ "runtime helper asr_stream_http_failed"
+ );
+ return Ok(None);
+ }
+ let body: RuntimeTurnCommonResult<RuntimeAsrStreamResp> = response
+ .json()
+ .await
+ .context("failed to decode asr stream response")?;
+ if body.code != 0 {
+ warn!(
+ call_id = %call_id,
+ trace_id = %trace_id,
+ turn_id = %turn.turn_id,
+ code = body.code,
+ msg_len = body.msg.as_deref().unwrap_or_default().len(),
+ "runtime helper asr_stream_common_result_failed"
+ );
+ return Ok(None);
+ }
+ let Some(data) = body.data else {
+ return Ok(None);
+ };
+ if data.status.as_deref() == Some("final") {
+ info!(
+ call_id = %call_id,
+ trace_id = %trace_id,
+ turn_id = %turn.turn_id,
+ chunk_count = data.chunk_count.unwrap_or_default(),
+ audio_bytes = data.audio_bytes.unwrap_or_default(),
+ asr_duration_ms = data.asr_duration_ms.unwrap_or_default(),
+ wall_ms = started_at.elapsed().as_millis() as u64,
+ provider = %data.provider_alias.as_deref().unwrap_or("unknown"),
+ text_len = data.text_len.unwrap_or_default(),
+ "runtime helper asr_stream_final"
+ );
+ return Ok(data.asr_result_ref);
+ }
+ emit_activity(
+ call_id,
+ trace_id,
+ Some(&turn.turn_id),
+ "asr_stream_fallback",
+ "skipped",
+ None,
+ Some(true),
+ json!({
+ "fallbackReason": data.fallback_reason,
+ "fallbackStage": data.fallback_stage,
+ "status": data.status,
+ }),
+ );
+ Ok(None)
+}
+
+fn build_asr_stream_ndjson(
+ call_id: &str,
+ trace_id: &str,
+ turn: &FinishedSpeechTurn,
+ bridge_config: &TurnBridgeConfig,
+) -> Result<String> {
+ let chunks = asr_pcm_16k_chunks(turn)?;
+ let mut seq = 1u64;
+ let mut lines = Vec::with_capacity(chunks.len() + 3);
+ lines.push(serde_json::to_string(&json!({
+ "event": "asr_stream_started",
+ "seq": seq,
+ "callId": call_id,
+ "traceId": trace_id,
+ "turnId": turn.turn_id.as_str(),
+ "tsMs": current_time_millis(),
+ "payload": {
+ "format": "pcm_s16le",
+ "sampleRate": 16000,
+ "channels": 1,
+ "runtimeSessionNonce": bridge_config.runtime_session_nonce.as_deref().unwrap_or_default(),
+ "providerHint": "volcengine",
+ }
+ }))?);
+ for (index, samples) in chunks.iter().enumerate() {
+ seq += 1;
+ let bytes = pcm_i16_to_le_bytes(samples);
+ let duration_ms = ((samples.len() as u64) * 1000 / 16_000).max(1);
+ lines.push(serde_json::to_string(&json!({
+ "event": "asr_audio_chunk",
+ "seq": seq,
+ "callId": call_id,
+ "traceId": trace_id,
+ "turnId": turn.turn_id.as_str(),
+ "tsMs": current_time_millis(),
+ "payload": {
+ "chunkSeq": index + 1,
+ "format": "pcm_s16le",
+ "sampleRate": 16000,
+ "channels": 1,
+ "durationMs": duration_ms,
+ "payloadBase64": general_purpose::STANDARD.encode(bytes),
+ }
+ }))?);
+ }
+ seq += 1;
+ lines.push(serde_json::to_string(&json!({
+ "event": "vad_speech_end",
+ "seq": seq,
+ "callId": call_id,
+ "traceId": trace_id,
+ "turnId": turn.turn_id.as_str(),
+ "tsMs": current_time_millis(),
+ "payload": {
+ "endReason": turn.end_reason.as_str(),
+ "speechDurationMs": turn.duration_ms,
+ }
+ }))?);
+ seq += 1;
+ lines.push(serde_json::to_string(&json!({
+ "event": "asr_stream_finish",
+ "seq": seq,
+ "callId": call_id,
+ "traceId": trace_id,
+ "turnId": turn.turn_id.as_str(),
+ "tsMs": current_time_millis(),
+ "payload": {
+ "finalChunkSeq": chunks.len(),
+ "audioDurationMs": turn.duration_ms,
+ }
+ }))?);
+ Ok(lines.join("\n") + "\n")
+}
+
+fn asr_pcm_16k_chunks(turn: &FinishedSpeechTurn) -> Result<Vec<Vec<i16>>> {
+ if turn.samples.is_empty() {
+ return Err(anyhow!("empty turn samples"));
+ }
+ let samples_16k: Vec<i16> = turn.samples.iter().step_by(3).copied().collect();
+ if samples_16k.is_empty() {
+ return Err(anyhow!("empty 16k asr samples"));
+ }
+ let samples_per_chunk = 320usize;
+ Ok(samples_16k
+ .chunks(samples_per_chunk)
+ .map(|chunk| chunk.to_vec())
+ .collect())
+}
+
+fn pcm_i16_to_le_bytes(samples: &[i16]) -> Vec<u8> {
+ let mut bytes = Vec::with_capacity(samples.len() * 2);
+ for sample in samples {
+ bytes.extend_from_slice(&sample.to_le_bytes());
+ }
+ bytes
+}
+
async fn request_turn_bridge_stream(
http: &Client,
bridge_config: &TurnBridgeConfig,
@@ -1012,6 +1323,7 @@
turn: &FinishedSpeechTurn,
path_ref: &str,
byte_size: u64,
+ asr_result_ref: Option<&str>,
turn_pipeline_started_at: Instant,
) -> Result<RuntimeTurnStreamOutcome> {
let bridge_started_at = Instant::now();
@@ -1019,7 +1331,7 @@
call_id: call_id.to_string(),
trace_id: trace_id.to_string(),
turn_id: turn.turn_id.clone(),
- audio_artifact: RuntimeTurnAudioArtifact {
+ audio_artifact: Some(RuntimeTurnAudioArtifact {
artifact_type: "local_file".to_string(),
path_ref: path_ref.to_string(),
format: "wav".to_string(),
@@ -1027,7 +1339,8 @@
channels: u32::from(USER_AUDIO_NUM_CHANNELS),
duration_ms: turn.duration_ms,
byte_size,
- },
+ }),
+ asr_result_ref: asr_result_ref.map(str::to_string),
};
let response = http
.post(bridge_config.bridge_url.as_deref().unwrap_or_default())
@@ -1620,6 +1933,7 @@
turn: &FinishedSpeechTurn,
path_ref: &str,
byte_size: u64,
+ asr_result_ref: Option<&str>,
turn_pipeline_started_at: Instant,
) -> Result<RuntimeTurnBridgeOutcome> {
let bridge_started_at = Instant::now();
@@ -1627,7 +1941,7 @@
call_id: call_id.to_string(),
trace_id: trace_id.to_string(),
turn_id: turn.turn_id.clone(),
- audio_artifact: RuntimeTurnAudioArtifact {
+ audio_artifact: Some(RuntimeTurnAudioArtifact {
artifact_type: "local_file".to_string(),
path_ref: path_ref.to_string(),
format: "wav".to_string(),
@@ -1635,7 +1949,8 @@
channels: u32::from(USER_AUDIO_NUM_CHANNELS),
duration_ms: turn.duration_ms,
byte_size,
- },
+ }),
+ asr_result_ref: asr_result_ref.map(str::to_string),
};
let response = http
.post(bridge_config.bridge_url.as_deref().unwrap_or_default())
@@ -1926,7 +2241,10 @@
#[serde(rename = "turnId")]
turn_id: String,
#[serde(rename = "audioArtifact")]
- audio_artifact: RuntimeTurnAudioArtifact,
+ #[serde(skip_serializing_if = "Option::is_none")]
+ audio_artifact: Option<RuntimeTurnAudioArtifact>,
+ #[serde(rename = "asrResultRef", skip_serializing_if = "Option::is_none")]
+ asr_result_ref: Option<String>,
}
#[derive(Serialize)]
@@ -1952,6 +2270,20 @@
data: Option<T>,
stage: Option<String>,
retryable: Option<bool>,
+}
+
+#[derive(Deserialize)]
+#[serde(rename_all = "camelCase")]
+struct RuntimeAsrStreamResp {
+ status: Option<String>,
+ asr_result_ref: Option<String>,
+ chunk_count: Option<u64>,
+ audio_bytes: Option<u64>,
+ asr_duration_ms: Option<u64>,
+ provider_alias: Option<String>,
+ text_len: Option<u64>,
+ fallback_reason: Option<String>,
+ fallback_stage: Option<String>,
}
#[derive(Default)]
@@ -2144,6 +2476,68 @@
i32::from(USER_AUDIO_NUM_CHANNELS),
);
let started_at = Instant::now();
+ let (frame_tx, mut frame_rx) =
+ mpsc::channel::<DrainedUserAudioFrame>(INBOUND_AUDIO_QUEUE_CAPACITY);
+ let (drain_shutdown_tx, mut drain_shutdown_rx) = watch::channel(false);
+ let drain_call_id = call_id.clone();
+ let drain_trace_id = trace_id.clone();
+ let drain_task = tokio::spawn(async move {
+ let mut received_frame_count: u64 = 0;
+ let mut dropped_frame_count: u64 = 0;
+ loop {
+ tokio::select! {
+ changed = drain_shutdown_rx.changed() => {
+ match changed {
+ Ok(()) if *drain_shutdown_rx.borrow() => break,
+ Ok(()) => {}
+ Err(_) => break,
+ }
+ }
+ maybe_frame = stream.next() => {
+ let Some(frame) = maybe_frame else {
+ break;
+ };
+ received_frame_count = received_frame_count.saturating_add(1);
+ let drained = DrainedUserAudioFrame {
+ frame_index: received_frame_count,
+ captured_elapsed_ms: started_at.elapsed().as_millis() as u64,
+ frame: AudioFrame {
+ data: Cow::Owned(frame.data.as_ref().to_vec()),
+ sample_rate: frame.sample_rate,
+ num_channels: frame.num_channels,
+ samples_per_channel: frame.samples_per_channel,
+ },
+ };
+ match frame_tx.try_send(drained) {
+ Ok(()) => {}
+ Err(mpsc::error::TrySendError::Full(_)) => {
+ dropped_frame_count = dropped_frame_count.saturating_add(1);
+ if dropped_frame_count % INBOUND_AUDIO_DROP_LOG_INTERVAL == 1 {
+ warn!(
+ call_id = %drain_call_id,
+ trace_id = %drain_trace_id,
+ dropped_frame_count,
+ received_frame_count,
+ queue_capacity = INBOUND_AUDIO_QUEUE_CAPACITY,
+ "runtime helper inbound_audio_queue_full_dropping_newest"
+ );
+ }
+ }
+ Err(mpsc::error::TrySendError::Closed(_)) => break,
+ }
+ }
+ }
+ }
+ stream.close();
+ info!(
+ call_id = %drain_call_id,
+ trace_id = %drain_trace_id,
+ received_frame_count,
+ dropped_frame_count,
+ queue_capacity = INBOUND_AUDIO_QUEUE_CAPACITY,
+ "runtime helper user_audio_drain_ended"
+ );
+ });
let mut frame_count: u64 = 0;
let mut sample_count: u64 = 0;
let mut simple_vad = if simple_vad_enabled {
@@ -2151,11 +2545,13 @@
} else {
None
};
+ let mut realtime_asr_upload: Option<RealtimeAsrUpload> = None;
- while let Some(frame) = stream.next().await {
- frame_count += 1;
+ while let Some(drained) = frame_rx.recv().await {
+ let frame = drained.frame;
+ frame_count = drained.frame_index;
sample_count += u64::from(frame.samples_per_channel) * u64::from(frame.num_channels);
- let elapsed_ms = started_at.elapsed().as_millis() as u64;
+ let elapsed_ms = drained.captured_elapsed_ms;
if frame_count == 1 {
info!(
call_id = %call_id,
@@ -2183,7 +2579,8 @@
if let Some(vad) = simple_vad.as_mut() {
if vad_enabled_gate.load(Ordering::Acquire) {
- if let Some(turn) = vad.observe_frame(
+ let was_in_speech = vad.in_speech;
+ let turn = vad.observe_frame(
&call_id,
&trace_id,
&participant_alias,
@@ -2191,7 +2588,65 @@
frame_count,
elapsed_ms,
&frame,
- ) {
+ );
+ let is_in_speech = vad.in_speech;
+
+ if !was_in_speech && is_in_speech {
+ let turn_id = format!("turn-{:04}", vad.turn_index);
+ match RealtimeAsrUpload::start(
+ http.clone(),
+ turn_bridge_config.realtime_asr_config(),
+ &call_id,
+ &trace_id,
+ &turn_id,
+ &vad.speech_samples,
+ ) {
+ Ok(upload) => {
+ info!(
+ call_id = %call_id,
+ trace_id = %trace_id,
+ turn_id = %turn_id,
+ "runtime helper asr_realtime_session_started"
+ );
+ realtime_asr_upload = Some(upload);
+ }
+ Err(error) if turn_bridge_config.asr_realtime_enabled => {
+ warn!(
+ call_id = %call_id,
+ trace_id = %trace_id,
+ turn_id = %turn_id,
+ error = %safe_error(&error.to_string()),
+ "runtime helper asr_realtime_start_failed_fallback"
+ );
+ }
+ Err(_) => {}
+ }
+ } else if was_in_speech {
+ let push_failed = realtime_asr_upload
+ .as_mut()
+ .and_then(|upload| upload.push_48k_samples(frame.data.as_ref()).err());
+ if let Some(error) = push_failed {
+ warn!(
+ call_id = %call_id,
+ trace_id = %trace_id,
+ error = %safe_error(&error.to_string()),
+ "runtime helper asr_realtime_upload_failed_fallback"
+ );
+ if let Some(upload) = realtime_asr_upload.take() {
+ tokio::spawn(async move {
+ upload.cancel("upload_backpressure").await;
+ });
+ }
+ }
+ }
+
+ if let Some(turn) = turn {
+ let realtime_asr_result_ref = match realtime_asr_upload.take() {
+ Some(upload) => {
+ finish_realtime_asr_upload(upload, &call_id, &trace_id, &turn).await
+ }
+ None => None,
+ };
handle_finished_turn(
&http,
&turn_bridge_config,
@@ -2199,10 +2654,22 @@
&call_id,
&trace_id,
turn,
+ realtime_asr_result_ref,
)
.await;
+ } else if was_in_speech && !is_in_speech {
+ if let Some(upload) = realtime_asr_upload.take() {
+ tokio::spawn(async move {
+ upload.cancel("speech_too_short").await;
+ });
+ }
}
} else {
+ if let Some(upload) = realtime_asr_upload.take() {
+ tokio::spawn(async move {
+ upload.cancel("vad_disabled").await;
+ });
+ }
vad.observe_disabled_frame(
&call_id,
&trace_id,
@@ -2223,9 +2690,41 @@
&track_sid_alias,
started_at.elapsed().as_millis() as u64,
) {
- handle_finished_turn(&http, &turn_bridge_config, &sink, &call_id, &trace_id, turn)
- .await;
+ let realtime_asr_result_ref = match realtime_asr_upload.take() {
+ Some(upload) => {
+ finish_realtime_asr_upload(upload, &call_id, &trace_id, &turn).await
+ }
+ None => None,
+ };
+ handle_finished_turn(
+ &http,
+ &turn_bridge_config,
+ &sink,
+ &call_id,
+ &trace_id,
+ turn,
+ realtime_asr_result_ref,
+ )
+ .await;
}
+ }
+ if let Some(upload) = realtime_asr_upload.take() {
+ upload.cancel("stream_end").await;
+ }
+ let _ = drain_shutdown_tx.send(true);
+ let mut drain_task = drain_task;
+ if timeout(AUDIO_DRAIN_STOP_GRACE, &mut drain_task)
+ .await
+ .is_err()
+ {
+ drain_task.abort();
+ let _ = drain_task.await;
+ warn!(
+ call_id = %call_id,
+ trace_id = %trace_id,
+ stop_grace_ms = AUDIO_DRAIN_STOP_GRACE.as_millis() as u64,
+ "runtime helper user_audio_drain_stop_timeout"
+ );
}
info!(
@@ -2239,6 +2738,64 @@
"runtime helper user_audio_stream_ended"
);
})
+}
+
+struct DrainedUserAudioFrame {
+ frame_index: u64,
+ captured_elapsed_ms: u64,
+ frame: AudioFrame<'static>,
+}
+
+async fn finish_realtime_asr_upload(
+ upload: RealtimeAsrUpload,
+ call_id: &str,
+ trace_id: &str,
+ turn: &FinishedSpeechTurn,
+) -> Option<String> {
+ match upload.finish(turn.duration_ms, &turn.end_reason).await {
+ Ok(RealtimeAsrOutcome {
+ status,
+ asr_result_ref,
+ provider_alias,
+ partial_count,
+ fallback_reason,
+ fallback_stage,
+ chunk_count,
+ audio_bytes,
+ wall_ms,
+ }) => {
+ info!(
+ call_id = %call_id,
+ trace_id = %trace_id,
+ turn_id = %turn.turn_id,
+ status = %status,
+ provider_alias = ?provider_alias,
+ partial_count,
+ chunk_count,
+ audio_bytes,
+ wall_ms,
+ asr_result_ref_present = asr_result_ref.is_some(),
+ fallback_reason = ?fallback_reason,
+ fallback_stage = ?fallback_stage,
+ "runtime helper asr_realtime_finished"
+ );
+ if status == "final" {
+ asr_result_ref
+ } else {
+ None
+ }
+ }
+ Err(error) => {
+ warn!(
+ call_id = %call_id,
+ trace_id = %trace_id,
+ turn_id = %turn.turn_id,
+ error = %safe_error(&error.to_string()),
+ "runtime helper asr_realtime_failed_fallback"
+ );
+ None
+ }
+ }
}
#[derive(Clone)]
@@ -2258,8 +2815,8 @@
rms_threshold: f64_env("CV_VAD_RMS_THRESHOLD", 0.012),
peak_threshold: f64_env("CV_VAD_PEAK_THRESHOLD", 0.08),
start_frames: u32_env("CV_VAD_START_FRAMES", 5).max(1),
- end_silence_ms: u64_env("CV_VAD_END_SILENCE_MS", 700).max(100),
- min_speech_ms: u64_env("CV_VAD_MIN_SPEECH_MS", 300).max(1),
+ end_silence_ms: u64_env("CV_VAD_END_SILENCE_MS", 400).max(100),
+ min_speech_ms: u64_env("CV_VAD_MIN_SPEECH_MS", 250).max(1),
max_turn_ms: u64_env("CV_VAD_MAX_TURN_MS", 10_000).max(1_000),
initial_ignore_ms: u64_env("CV_VAD_INITIAL_IGNORE_MS", 500),
}
@@ -2372,7 +2929,9 @@
return None;
}
- let voiced = rms >= self.config.rms_threshold || peak >= self.config.peak_threshold;
+ // Align with cb-sdk's energy-based segmentation: peak is diagnostic only,
+ // otherwise isolated spikes can keep a turn open until max_turn_ms.
+ let voiced = rms >= self.config.rms_threshold;
if !self.in_speech {
self.remember_pre_speech_frame(frame);
}
--
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