Rework latency HUD: base all measurements on first user_transcript
Previous approach used TurnEndTime (from StopListening) which was never set in Server VAD mode. Now all latency measurements are anchored to TurnStartTime, captured when the first user_transcript arrives from the ElevenLabs server — the earliest client-side confirmation of user speech. Timeline: [user speaks] → STT → user_transcript(=T0) → agent_response_started → audio → playback Metrics shown: - Transcript>Gen: T0 → generation start (LLM think time) - Gen>Audio: generation start → first audio chunk (LLM + TTS) - Total: T0 → first audio chunk (full pipeline) - Pre-buffer: wait before playback - End-to-Ear: T0 → playback starts (user-perceived) Removed UserSpeechMs (unmeasurable without client-side VAD). Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
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@ -166,9 +166,9 @@ void UPS_AI_ConvAgent_ElevenLabsComponent::TickComponent(float DeltaTime, ELevel
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AudioPlaybackComponent->Play();
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AudioPlaybackComponent->Play();
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}
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}
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PlaybackStartTime = FPlatformTime::Seconds();
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PlaybackStartTime = FPlatformTime::Seconds();
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if (TurnEndTime > 0.0)
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if (TurnStartTime > 0.0)
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{
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{
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CurrentLatencies.EndToEarMs = static_cast<float>((PlaybackStartTime - TurnEndTime) * 1000.0);
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CurrentLatencies.EndToEarMs = static_cast<float>((PlaybackStartTime - TurnStartTime) * 1000.0);
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}
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}
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if (PreBufferStartTime > 0.0)
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if (PreBufferStartTime > 0.0)
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{
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{
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@ -279,7 +279,6 @@ void UPS_AI_ConvAgent_ElevenLabsComponent::TickComponent(float DeltaTime, ELevel
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bHardTimeoutFired = bHardTimeout && !bAgentResponseReceived;
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bHardTimeoutFired = bHardTimeout && !bAgentResponseReceived;
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bAgentSpeaking = false;
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bAgentSpeaking = false;
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bPreBuffering = false; // Cancel pending pre-buffer to prevent stale OnAudioPlaybackStarted.
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bPreBuffering = false; // Cancel pending pre-buffer to prevent stale OnAudioPlaybackStarted.
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TurnStartTime = FPlatformTime::Seconds(); // Mark start of next potential user turn.
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bAgentResponseReceived = false;
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bAgentResponseReceived = false;
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SilentTickCount = 0;
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SilentTickCount = 0;
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bShouldBroadcastStopped = true;
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bShouldBroadcastStopped = true;
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@ -1018,6 +1017,20 @@ void UPS_AI_ConvAgent_ElevenLabsComponent::HandleAudioReceived(const TArray<uint
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void UPS_AI_ConvAgent_ElevenLabsComponent::HandleTranscript(const FPS_AI_ConvAgent_TranscriptSegment_ElevenLabs& Segment)
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void UPS_AI_ConvAgent_ElevenLabsComponent::HandleTranscript(const FPS_AI_ConvAgent_TranscriptSegment_ElevenLabs& Segment)
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{
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{
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// In Server VAD mode, user_transcript is the first server confirmation that the
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// user spoke. Capture the time of the FIRST transcript this turn as TurnStartTime.
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// This is after STT completes (not real speech start), but it's the best client-side
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// approximation. The gap between TurnStartTime and GenerationStartTime then measures
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// the LLM think time (transcript received → agent starts generating).
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if (Segment.Speaker == TEXT("user"))
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{
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// "First transcript this turn" = TurnStartTime hasn't been updated since last generation.
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if (GenerationStartTime <= 0.0 || TurnStartTime <= GenerationStartTime)
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{
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TurnStartTime = FPlatformTime::Seconds();
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}
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}
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if (bEnableUserTranscript)
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if (bEnableUserTranscript)
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{
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{
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OnAgentTranscript.Broadcast(Segment);
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OnAgentTranscript.Broadcast(Segment);
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@ -1091,13 +1104,16 @@ void UPS_AI_ConvAgent_ElevenLabsComponent::HandleAgentResponseStarted()
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// Reset all latency measurements — new response cycle starts here.
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// Reset all latency measurements — new response cycle starts here.
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CurrentLatencies = FDebugLatencies();
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CurrentLatencies = FDebugLatencies();
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if (TurnStartTime > 0.0 && TurnEndTime > TurnStartTime)
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// Transcript→Gen: time between first user_transcript received and generation start.
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// In Server VAD, this captures the LLM "think time" after STT completed.
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// TurnStartTime is set in HandleTranscript when the first user_transcript arrives.
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if (TurnStartTime > 0.0 && Now > TurnStartTime)
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{
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{
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CurrentLatencies.UserSpeechMs = static_cast<float>((TurnEndTime - TurnStartTime) * 1000.0);
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CurrentLatencies.STTToGenMs = static_cast<float>((Now - TurnStartTime) * 1000.0);
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}
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}
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GenerationStartTime = Now;
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GenerationStartTime = Now;
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CurrentLatencies.STTToGenMs = static_cast<float>((Now - TurnEndTime) * 1000.0);
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const double T = Now - SessionStartTime;
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const double T = Now - SessionStartTime;
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const double LatencyFromTurnEnd = Now - TurnEndTime;
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const double LatencyFromTurnEnd = Now - TurnEndTime;
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@ -1378,8 +1394,8 @@ void UPS_AI_ConvAgent_ElevenLabsComponent::EnqueueAgentAudio(const TArray<uint8>
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SilentTickCount = 0;
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SilentTickCount = 0;
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// Latency capture (always, for HUD display).
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// Latency capture (always, for HUD display).
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if (TurnEndTime > 0.0)
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if (TurnStartTime > 0.0)
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CurrentLatencies.TotalMs = static_cast<float>((AgentSpeakStart - TurnEndTime) * 1000.0);
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CurrentLatencies.TotalMs = static_cast<float>((AgentSpeakStart - TurnStartTime) * 1000.0);
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if (GenerationStartTime > 0.0)
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if (GenerationStartTime > 0.0)
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CurrentLatencies.GenToAudioMs = static_cast<float>((AgentSpeakStart - GenerationStartTime) * 1000.0);
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CurrentLatencies.GenToAudioMs = static_cast<float>((AgentSpeakStart - GenerationStartTime) * 1000.0);
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@ -1422,9 +1438,9 @@ void UPS_AI_ConvAgent_ElevenLabsComponent::EnqueueAgentAudio(const TArray<uint8>
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AudioPlaybackComponent->Play();
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AudioPlaybackComponent->Play();
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}
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}
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PlaybackStartTime = FPlatformTime::Seconds();
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PlaybackStartTime = FPlatformTime::Seconds();
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if (TurnEndTime > 0.0)
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if (TurnStartTime > 0.0)
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{
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{
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CurrentLatencies.EndToEarMs = static_cast<float>((PlaybackStartTime - TurnEndTime) * 1000.0);
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CurrentLatencies.EndToEarMs = static_cast<float>((PlaybackStartTime - TurnStartTime) * 1000.0);
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}
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}
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// No pre-buffer in this path (immediate playback).
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// No pre-buffer in this path (immediate playback).
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OnAudioPlaybackStarted.Broadcast();
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OnAudioPlaybackStarted.Broadcast();
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@ -1454,9 +1470,9 @@ void UPS_AI_ConvAgent_ElevenLabsComponent::EnqueueAgentAudio(const TArray<uint8>
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AudioPlaybackComponent->Play();
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AudioPlaybackComponent->Play();
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}
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}
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PlaybackStartTime = FPlatformTime::Seconds();
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PlaybackStartTime = FPlatformTime::Seconds();
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if (TurnEndTime > 0.0)
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if (TurnStartTime > 0.0)
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{
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{
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CurrentLatencies.EndToEarMs = static_cast<float>((PlaybackStartTime - TurnEndTime) * 1000.0);
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CurrentLatencies.EndToEarMs = static_cast<float>((PlaybackStartTime - TurnStartTime) * 1000.0);
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}
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}
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if (PreBufferStartTime > 0.0)
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if (PreBufferStartTime > 0.0)
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{
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{
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@ -1517,7 +1533,6 @@ void UPS_AI_ConvAgent_ElevenLabsComponent::StopAgentAudio()
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SilentTickCount = 0;
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SilentTickCount = 0;
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bWasSpeaking = true;
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bWasSpeaking = true;
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Now = FPlatformTime::Seconds();
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Now = FPlatformTime::Seconds();
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TurnStartTime = Now; // Mark start of next potential user turn.
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}
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}
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}
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}
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@ -2430,31 +2445,27 @@ void UPS_AI_ConvAgent_ElevenLabsComponent::DrawLatencyHUD() const
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return (Ms > 0.0f) ? FString::Printf(TEXT("%.0f ms"), Ms) : FString(TEXT("---"));
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return (Ms > 0.0f) ? FString::Printf(TEXT("%.0f ms"), Ms) : FString(TEXT("---"));
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};
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};
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// Title
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// Title — all times measured from first user_transcript received
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GEngine->AddOnScreenDebugMessage(BaseKey, DisplayTime, TitleColor,
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GEngine->AddOnScreenDebugMessage(BaseKey, DisplayTime, TitleColor,
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TEXT("=== Latency HUD ==="));
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TEXT("=== Latency (from transcript) ==="));
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// 1. User speech duration
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// 1. Transcript → Generation: LLM think time after STT completed
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GEngine->AddOnScreenDebugMessage(BaseKey + 1, DisplayTime, ValueColor,
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GEngine->AddOnScreenDebugMessage(BaseKey + 1, DisplayTime, ValueColor,
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FString::Printf(TEXT(" User speech: %s"), *Fmt(CurrentLatencies.UserSpeechMs)));
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FString::Printf(TEXT(" Transcript>Gen: %s"), *Fmt(CurrentLatencies.STTToGenMs)));
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// 2. STT + Network: end of speech → server starts generating
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// 2. Generation → Audio: LLM + TTS processing
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GEngine->AddOnScreenDebugMessage(BaseKey + 2, DisplayTime, ValueColor,
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GEngine->AddOnScreenDebugMessage(BaseKey + 2, DisplayTime, ValueColor,
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FString::Printf(TEXT(" STT + Network: %s"), *Fmt(CurrentLatencies.STTToGenMs)));
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FString::Printf(TEXT(" Gen>Audio: %s"), *Fmt(CurrentLatencies.GenToAudioMs)));
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// 3. LLM + TTS: server generating → first audio chunk received
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// 3. Total: transcript → first audio chunk (full pipeline)
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GEngine->AddOnScreenDebugMessage(BaseKey + 3, DisplayTime, ValueColor,
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GEngine->AddOnScreenDebugMessage(BaseKey + 3, DisplayTime, HighlightColor,
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FString::Printf(TEXT(" LLM + TTS: %s"), *Fmt(CurrentLatencies.GenToAudioMs)));
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// 4. Total round-trip: end of speech → first audio chunk
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GEngine->AddOnScreenDebugMessage(BaseKey + 4, DisplayTime, HighlightColor,
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FString::Printf(TEXT(" Total: %s"), *Fmt(CurrentLatencies.TotalMs)));
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FString::Printf(TEXT(" Total: %s"), *Fmt(CurrentLatencies.TotalMs)));
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// 5. Pre-buffer wait before playback
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// 4. Pre-buffer wait before playback
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GEngine->AddOnScreenDebugMessage(BaseKey + 5, DisplayTime, ValueColor,
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GEngine->AddOnScreenDebugMessage(BaseKey + 4, DisplayTime, ValueColor,
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FString::Printf(TEXT(" Pre-buffer: %s"), *Fmt(CurrentLatencies.PreBufferMs)));
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FString::Printf(TEXT(" Pre-buffer: %s"), *Fmt(CurrentLatencies.PreBufferMs)));
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// 6. End-to-ear: end of speech → audio playback starts (user-perceived latency)
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// 5. End-to-ear: transcript → audio playback starts (user-perceived)
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GEngine->AddOnScreenDebugMessage(BaseKey + 6, DisplayTime, HighlightColor,
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GEngine->AddOnScreenDebugMessage(BaseKey + 5, DisplayTime, HighlightColor,
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FString::Printf(TEXT(" End-to-Ear: %s"), *Fmt(CurrentLatencies.EndToEarMs)));
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FString::Printf(TEXT(" End-to-Ear: %s"), *Fmt(CurrentLatencies.EndToEarMs)));
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}
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}
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@ -640,17 +640,16 @@ private:
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double GenerationStartTime = 0.0; // Set in HandleAgentResponseStarted — server starts generating.
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double GenerationStartTime = 0.0; // Set in HandleAgentResponseStarted — server starts generating.
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double PlaybackStartTime = 0.0; // Set when audio playback actually starts (post pre-buffer).
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double PlaybackStartTime = 0.0; // Set when audio playback actually starts (post pre-buffer).
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// Current-turn latency measurements (ms). Reset when the user starts a new turn
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// Current-turn latency measurements (ms). Reset in HandleAgentResponseStarted.
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// (StopListening). Each field is populated as the corresponding event fires.
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// Each field is populated as the corresponding event fires.
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// Zero means "not yet measured this turn".
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// Zero means "not yet measured this turn".
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struct FDebugLatencies
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struct FDebugLatencies
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{
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{
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float UserSpeechMs = 0.0f; // How long the user spoke (TurnStart → TurnEnd)
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float STTToGenMs = 0.0f; // First user_transcript → agent_response_started (LLM think time)
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float STTToGenMs = 0.0f; // TurnEnd → server starts generating (network + STT)
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float GenToAudioMs = 0.0f; // agent_response_started → first audio chunk (LLM + TTS)
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float GenToAudioMs = 0.0f; // Server generating → first audio chunk (LLM + TTS)
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float TotalMs = 0.0f; // First user_transcript → first audio chunk (full pipeline)
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float TotalMs = 0.0f; // TurnEnd → first audio chunk (full round-trip)
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float PreBufferMs = 0.0f; // Pre-buffer wait before playback starts
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float PreBufferMs = 0.0f; // Pre-buffer wait before playback
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float EndToEarMs = 0.0f; // First user_transcript → audio playback starts (user-perceived)
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float EndToEarMs = 0.0f; // TurnEnd → audio playback starts (user-perceived)
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};
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};
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FDebugLatencies CurrentLatencies;
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FDebugLatencies CurrentLatencies;
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