The Web Speech API currently does not expose the acoustic timing bounds of recognized speech segments, making it difficult to associate transcribed text with media timelines or generate synchronized subtitle cues. We propose adding speechStartTime and speechEndTime (double, in seconds relative to stream start) attributes on the SpeechRecognitionResult interface.
The Web Speech API currently does not expose the acoustic start and end timestamps of recognized speech utterances on SpeechRecognitionResult. While SpeechRecognitionEvent.timeStamp conveys when the DOM event occurred, it provides no information about the acoustic boundaries or duration of the speech segment within the source audio stream. Exposing speechStartTime and speechEndTime (in seconds as a double, matching HTMLMediaElement.currentTime, BaseAudioContext.currentTime, and VTTCue) addresses critical media timeline association capabilities that are currently impossible on the web: 1. Automated Subtitling & Caption Tracks: Constructing frame-accurate subtitle cues (VTTCue(start, end, text)) for recorded or streaming audio/video. 2. Interactive "Click-to-Seek" Navigation: Enabling users to click a phrase in a transcript and immediately seek playback to the exact moment it was spoken. 3. Conferencing & Media Recording Alignment: Synchronizing real-time captions with recorded media tracks or WebRTC playout delay buffers. 4. Text-Based Media Editing: Accurately cutting, splicing, or rearranging audio and video based on transcript text.
Samples: https://alan33d.github.io/web-speech-demos/live_meeting_recording_demo.html https://alan33d.github.io/web-speech-demos/webrtc_caption_sync_test.html
Explainers: https://github.com/WebAudio/web-speech-api/blob/main/explainers/speech-recognition-result-timestamps.md