Implementing Captioned Audio Playback for Recording Review by Hearing-Impaired Supervisors
What This Guide Covers
You will configure Genesys Cloud Speech-to-Text processing, synchronize transcription timestamps with audio playback, and deploy a WCAG-compliant captioned review interface within Quality Management. The end result is a supervisor-facing playback environment where audio and text align precisely, enabling accurate quality scoring, compliance auditing, and performance coaching without auditory reliance.
Prerequisites, Roles & Licensing
- Licensing Tiers: Genesys Cloud CX 2 or CX 3 base license, Speech Analytics add-on, Workforce Engagement Management (WEM) license for Quality Management access.
- IAM Roles & Permissions:
Telephony > Recording > ViewQuality > Quality > ViewAnalytics > Conversation > ViewSpeech Analytics > Speech-to-Text > Configure
- OAuth Scopes:
view:recording,view:analytics:conversation,view:quality,view:speechanalytics:speech-to-text - External Dependencies: None required if using native Genesys Cloud STT. Third-party STT (Azure, Google, AWS) requires outbound HTTPS connectivity and corresponding IAM credentials if you elect to route transcription externally.
The Implementation Deep-Dive
1. Configure Speech-to-Text Processing & Transcription Storage
The foundation of captioned playback is deterministic transcription generation. Genesys Cloud processes audio asynchronously through its analytics pipeline. You must explicitly enable Speech-to-Text for the interaction types your supervisors review (voice, digital, or blended).
Navigate to Admin > Speech Analytics > Speech to Text or configure via API. The configuration defines language models, diarization thresholds, and redaction policies. You must enable speaker diarization to separate agent and customer utterances. Without diarization, the transcript renders as a single monolithic stream, which breaks accessibility compliance and forces supervisors to manually parse speaker turns.
Execute the following API call to lock down the STT configuration for your org:
PUT /api/v2/speechanalytics/speech-to-text-config
Authorization: Bearer <access_token>
Content-Type: application/json
{
"enabled": true,
"languageCode": "en-US",
"diarizationEnabled": true,
"speakerCount": 2,
"redactionEnabled": true,
"redactionTypes": ["PHI", "PCI", "EMAIL"],
"outputFormat": "WEBVTT_COMPATIBLE",
"confidenceThreshold": 0.75
}
The Trap: Setting confidenceThreshold too high (above 0.85) causes the engine to drop ambiguous phonemes, creating silent gaps in the caption stream. Supervisors interpret these gaps as missing audio, triggering false escalation workflows. Set the threshold between 0.70 and 0.75. You will retain critical context while allowing the UI to render placeholder tokens like [inaudible] for low-confidence segments, preserving timestamp continuity.
Architectural Reasoning: Genesys Cloud stores raw audio in immutable object storage and generates transcription artifacts in the analytics data lake. The outputFormat parameter dictates how the pipeline structures the JSON response. Using WEBVTT_COMPATIBLE ensures the timestamp fields (start, end) align with standard web media timebases, reducing client-side transformation overhead. You avoid writing custom timecode parsers when you render captions in the browser.
2. Align Timestamps & Handle Async Processing Delays
Transcription does not complete immediately after recording termination. The audio must upload, decompress, pass through the STT model, and undergo diarization and redaction. This pipeline typically takes 45 to 120 seconds depending on duration and queue depth. Your playback interface must handle this latency without blocking supervisor workflows.
You will implement a polling mechanism or webhook-driven state machine to track transcription readiness. The recording transcript endpoint returns a 404 until processing completes. Do not block the Quality scorecard UI on a synchronous transcript fetch. Instead, render the audio player immediately and attach a caption overlay that activates when the transcript becomes available.
Fetch transcript readiness via:
GET /api/v2/recordings/transcripts/{recordingId}
Authorization: Bearer <access_token>
[
{
"start": 1200,
"end": 3400,
"text": "Thank you for calling support, how can I assist you today?",
"speaker": "agent",
"confidence": 0.94
},
{
"start": 3500,
"end": 6800,
"text": "I am experiencing a billing discrepancy on my latest invoice.",
"speaker": "customer",
"confidence": 0.89
}
]
Convert the JSON array to WebVTT format on the client side. WebVTT is the W3C standard for synchronized captions in HTML5 media players. The conversion requires millisecond-to-clock-time transformation:
function msToWebVTT(ms) {
const hours = Math.floor(ms / 3600000);
const minutes = Math.floor((ms % 3600000) / 60000);
const seconds = Math.floor((ms % 60000) / 1000);
const milliseconds = ms % 1000;
return `${hours.toString().padStart(2, '0')}:${minutes.toString().padStart(2, '0')}:${seconds.toString().padStart(2, '0')}.${milliseconds.toString().padStart(3, '0')}`;
}
function generateWebVTT(transcriptLines) {
let vtt = 'WEBVTT\n\n';
transcriptLines.forEach((line, index) => {
vtt += `${index + 1}\n${msToWebVTT(line.start)} --> ${msToWebVTT(line.end)}\n${line.text}\n\n`;
});
return vtt;
}
The Trap: Applying server-side WebVTT generation and caching the result indefinitely. Audio playback introduces codec decoding latency that varies by browser and device architecture. Server-side timecodes assume a zero-latency playback environment, causing captions to appear 200 to 500 milliseconds ahead of or behind the actual audio. Always generate WebVTT client-side and apply a dynamic offset correction based on the HTML5 <audio> element’s readyState and seeking events.
Architectural Reasoning: Client-side timecode alignment compensates for network jitter and media pipeline buffering. You attach a timeupdate listener to the audio player, calculate the delta between player.currentTime and the target caption timestamp, and apply a CSS transform or scroll adjustment to keep the active caption centered. This approach decouples transcription storage from playback rendering, ensuring WCAG 2.1 AA compliance across Chrome, Firefox, and Safari.
3. Implement Synchronized Caption Rendering in the Review Interface
The playback interface requires a dual-pane layout: audio controls on top, synchronized caption scroll below. You will use the HTML5 <audio> element paired with a <track> element for caption injection, or render captions in a dedicated div if you require custom styling for speaker attribution.
The native Quality Management application provides a transcript pane, but it lacks granular accessibility controls for hearing-impaired users. You will extend the review experience using the Genesys Cloud Web Components SDK or a custom React/Vue wrapper that mounts inside the Quality scorecard iframe.
Configure the audio player with caption injection:
<audio id="recordingPlayer" controls>
<source src="/api/v2/recordings/interactions/{recordingId}/media" type="audio/mpeg">
<track id="captionTrack" kind="captions" srclang="en" label="English Captions" default>
</audio>
<div id="captionContainer" aria-live="polite" aria-atomic="false">
<!-- Active caption renders here for screen reader compatibility -->
</div>
Inject the WebVTT blob into the track element:
const blob = new Blob([webvttString], { type: 'text/vtt' });
const url = URL.createObjectURL(blob);
document.getElementById('captionTrack').src = url;
Bind the timeupdate event to highlight the active caption line and scroll the container:
const player = document.getElementById('recordingPlayer');
const captions = document.querySelectorAll('#captionContainer .caption-line');
player.addEventListener('timeupdate', () => {
const currentTime = player.currentTime * 1000;
captions.forEach(line => {
const start = parseFloat(line.dataset.start);
const end = parseFloat(line.dataset.end);
if (currentTime >= start && currentTime <= end) {
line.classList.add('active');
line.scrollIntoView({ behavior: 'smooth', block: 'center' });
document.getElementById('captionContainer').textContent = line.textContent;
} else {
line.classList.remove('active');
}
});
});
The Trap: Relying solely on the native <track> element for visual caption rendering. The <track> element handles browser-native styling, which often lacks speaker differentiation, confidence indicators, or redaction masking. Hearing-impaired supervisors require visual speaker attribution to follow conversational turn-taking. Render captions in a controlled DOM structure instead of relying on browser default caption overlays. Use the <track> element only as a fallback for native accessibility APIs, while maintaining your own synchronized div for full visual control.
Architectural Reasoning: Custom DOM rendering grants you precise control over typography, contrast ratios, and speaker color-coding. You can inject CSS variables for high-contrast modes, apply font scaling for low-vision accommodations, and mask redacted segments with [REDACTED] tokens that maintain exact timing alignment. This separation of concerns ensures the UI meets Section 508 and WCAG 2.1 AA requirements without fighting browser caption rendering quirks.
4. Configure Quality Management Integration for Supervisor Access
Supervisors access recordings through Quality Management scorecards. You must configure the Quality app to surface the captioned player, restrict access by role, and ensure the transcript data loads before the scorecard renders.
Navigate to Admin > Quality > Scorecards and create a new scorecard template. Add a custom HTML widget or embed the playback component via the Quality app’s extension framework. Bind the recording ID from the quality session context to the player’s source URL.
Configure role-based access using IAM policies:
{
"policyName": "Quality_Supervisor_Captioned_Review",
"permissions": [
"view:quality",
"view:recording",
"view:analytics:conversation",
"view:speechanalytics:speech-to-text"
],
"roles": ["Quality_Supervisor", "WEM_Manager"]
}
Embed the player in the Quality scorecard layout XML:
<scorecard>
<section name="Recording Review">
<widget type="custom-html" id="captionedPlayer">
<config>
<recordingId field="interactionId"/>
<enableCaptions value="true"/>
<diarizationVisible value="true"/>
</config>
</widget>
</section>
</scorecard>
The Trap: Assigning Quality permissions without verifying WEM license allocation. Genesys Cloud enforces license-based UI rendering. If a supervisor lacks a WEM license, the Quality app returns a 403 Forbidden response, and the captioned player fails to initialize. Always validate license assignment in Admin > Users > Licenses before deploying scorecard templates. Use the GET /api/v2/users/{userId} endpoint to verify licenseType includes WEM or CX2/CX3 with WEM add-on.
Architectural Reasoning: Quality Management operates on a session-based context model. The scorecard engine passes interaction metadata through a secure iframe message bus. By binding the recording ID at the scorecard level, you eliminate cross-origin fetch issues and leverage Genesys Cloud’s internal CDN for media delivery. This reduces latency and ensures the captioned player inherits the supervisor’s session authentication, preventing unauthorized recording access.
Validation, Edge Cases & Troubleshooting
Edge Case 1: Transcription Latency Causing Playback Desynchronization
The Failure Condition: Supervisors press play, but captions appear 3 to 5 seconds behind the audio. The desynchronization worsens as the recording progresses.
The Root Cause: The STT pipeline processes audio in 10-second chunks. If the recording contains extended silence or low-amplitude segments, the engine delays chunk finalization. The client-side player does not account for this processing lag, causing timestamp drift.
The Solution: Implement a client-side timebase correction algorithm. Track the delta between player.currentTime and the expected caption timestamp every 500 milliseconds. If the delta exceeds 150 milliseconds, apply a linear interpolation offset to subsequent caption rendering. Expose a manual sync slider in the UI labeled Caption Offset (ms) to allow supervisors to fine-tune alignment when network conditions vary.
Edge Case 2: Multi-Party Speaker Diarization Failures
The Failure Condition: Two speakers overlap, and the transcript merges their utterances into a single line with incorrect speaker attribution. Captions display [customer] for agent speech, breaking comprehension.
The Root Cause: Diarization models rely on voice fingerprinting and energy thresholds. Overlapping speech, background noise, or similar vocal profiles cause the model to misclassify speaker turns. The confidence field drops below 0.65, but the UI does not flag low-confidence segments.
The Solution: Filter transcript lines by confidence threshold before rendering. If confidence < 0.70, append a visual indicator like (low confidence) and highlight the line in amber. Enable speakerCount dynamic adjustment in the STT config. For high-compliance environments, route low-confidence recordings to a manual review queue using the POST /api/v2/quality/interaction-queues endpoint. Supervisors can correct speaker attribution directly in the caption editor, and the corrections feed back into the analytics pipeline for model retraining.
Edge Case 3: API Pagination & Transcript Chunking Limits
The Failure Condition: Recordings longer than 45 minutes return truncated transcripts. The final 30 percent of the recording plays without captions.
The Root Cause: The GET /api/v2/recordings/transcripts/{recordingId} endpoint enforces a default page size of 500 lines. Long recordings exceed this limit, and the client does not paginate. The API returns only the first page, silently dropping subsequent segments.
The Solution: Implement cursor-based pagination. The response includes a nextPageToken field. Loop through pages until nextPageToken is null:
async function fetchFullTranscript(recordingId, token = null) {
const url = token
? `/api/v2/recordings/transcripts/${recordingId}?pageToken=${token}`
: `/api/v2/recordings/transcripts/${recordingId}`;
const response = await fetch(url, { headers: { Authorization: `Bearer ${token}` } });
const data = await response.json();
if (data.nextPageToken) {
return [...data.lines, ...(await fetchFullTranscript(recordingId, data.nextPageToken))];
}
return data.lines;
}
Cache the complete transcript array in sessionStorage to avoid repeated API calls during scorecard navigation. Apply rate limiting headers (Retry-After) to prevent 429 Throttling errors during bulk review sessions.