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Overview

Whisper can extract word-level timestamps using cross-attention patterns and dynamic time warping (DTW). This feature enables precise synchronization between transcribed text and the original audio.
Word-level timestamps are extracted using the cross-attention mechanism, which aligns the decoder’s attention to specific audio frames for each word.

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Basic Usage

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CLI

Enable word timestamps with the --word_timestamps flag: 
whisper audio.mp3 --word_timestamps True
The output JSON file will include word-level timing information: 
whisper audio.mp3 --word_timestamps True --output_format json

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Python API

Set word_timestamps=True when calling transcribe(): 
import whisper

model = whisper.load_model("turbo")
result = model.transcribe("audio.mp3", word_timestamps=True)

# Access word-level timestamps
for segment in result["segments"]:
    for word in segment["words"]:
        print(f"{word['word']} [{word['start']:.2f}s - {word['end']:.2f}s]")

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Output Format

When word timestamps are enabled, each segment includes a words list: 
{
    "id": 0,
    "start": 0.0,
    "end": 5.5,
    "text": " Hello, how are you?",
    "words": [
        {
            "word": " Hello,",
            "start": 0.0,
            "end": 0.5,
            "probability": 0.95
        },
        {
            "word": " how",
            "start": 0.5,
            "end": 0.8,
            "probability": 0.98
        },
        {
            "word": " are",
            "start": 0.8,
            "end": 1.0,
            "probability": 0.97
        },
        {
            "word": " you?",
            "start": 1.0,
            "end": 1.5,
            "probability": 0.96
        }
    ]
}
Each word entry contains:
  • word: The word text (including leading/trailing spaces and punctuation)
  • start: Start time in seconds
  • end: End time in seconds
  • probability: Average token probability for the word

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Punctuation Handling

Whisper automatically merges punctuation marks with adjacent words:

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Prepended Punctuation

These marks are merged with the next word: 
prepend_punctuations = "\"'¿([{-"
Example: "Hello → treated as one word

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Appended Punctuation

These marks are merged with the previous word: 
append_punctuations = "\"\'。,，!！?？:：\")]}、"
Example: world! → treated as one word

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Custom Punctuation Rules

result = model.transcribe(
    "audio.mp3",
    word_timestamps=True,
    prepend_punctuations="\"'¿([{-",
    append_punctuations="\"\'。,，!！?？:：\")]}、"
)

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Advanced Options

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Subtitle Formatting

Word timestamps enable advanced subtitle formatting: 
whisper video.mp4 \
  --word_timestamps True \
  --max_line_width 50 \
  --max_line_count 2 \
  --highlight_words True \
  --output_format srt

--max_line_width 50
Maximum number of characters per line before breaking.

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Hallucination Detection

Skip silent periods when hallucinations are detected: 
whisper audio.mp3 \
  --word_timestamps True \
  --hallucination_silence_threshold 2.0

result = model.transcribe(
    "audio.mp3",
    word_timestamps=True,
    hallucination_silence_threshold=2.0  # Skip 2+ seconds of silence
)
This helps prevent the model from generating text during long silent periods.

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Use Cases

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1. Precise Subtitle Generation

import whisper

def generate_word_level_subtitles(audio_file: str, output_file: str):
    model = whisper.load_model("turbo")
    result = model.transcribe(audio_file, word_timestamps=True)
    
    with open(output_file, "w", encoding="utf-8") as f:
        subtitle_index = 1
        for segment in result["segments"]:
            for word in segment["words"]:
                start = format_timestamp(word["start"])
                end = format_timestamp(word["end"])
                text = word["word"].strip()
                
                f.write(f"{subtitle_index}\n")
                f.write(f"{start} --> {end}\n")
                f.write(f"{text}\n\n")
                subtitle_index += 1

def format_timestamp(seconds: float) -> str:
    hours = int(seconds // 3600)
    minutes = int((seconds % 3600) // 60)
    secs = int(seconds % 60)
    millis = int((seconds % 1) * 1000)
    return f"{hours:02d}:{minutes:02d}:{secs:02d},{millis:03d}"

generate_word_level_subtitles("video.mp4", "subtitles.srt")

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2. Audio-Text Alignment for Editing

import whisper

def find_word_position(audio_file: str, search_word: str):
    model = whisper.load_model("turbo")
    result = model.transcribe(audio_file, word_timestamps=True)
    
    matches = []
    for segment in result["segments"]:
        for word in segment["words"]:
            if search_word.lower() in word["word"].lower():
                matches.append({
                    "word": word["word"],
                    "start": word["start"],
                    "end": word["end"],
                    "probability": word["probability"]
                })
    
    return matches

# Find all instances of "artificial intelligence"
matches = find_word_position("podcast.mp3", "artificial")
for match in matches:
    print(f"{match['word']} at {match['start']:.2f}s (confidence: {match['probability']:.2%})")

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3. Karaoke-Style Lyrics Display

import whisper
import time
import os

def display_lyrics_realtime(audio_file: str):
    model = whisper.load_model("turbo")
    result = model.transcribe(audio_file, word_timestamps=True)
    
    # Flatten all words
    all_words = []
    for segment in result["segments"]:
        all_words.extend(segment["words"])
    
    print("Starting playback...\n")
    start_time = time.time()
    
    for word in all_words:
        # Wait until word should be displayed
        while time.time() - start_time < word["start"]:
            time.sleep(0.01)
        
        # Display word
        print(word["word"], end="", flush=True)
        
        # Optional: clear after word ends
        # time.sleep(word["end"] - word["start"])
    
    print("\n\nPlayback complete!")

# Note: This doesn't actually play audio, just simulates timing
# Combine with an audio player for real karaoke

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4. Speech Segmentation and Analysis

import whisper
import numpy as np

def analyze_speech_rate(audio_file: str):
    model = whisper.load_model("turbo")
    result = model.transcribe(audio_file, word_timestamps=True)
    
    word_durations = []
    for segment in result["segments"]:
        for word in segment["words"]:
            duration = word["end"] - word["start"]
            word_count = len(word["word"].strip().split())
            if word_count > 0:
                word_durations.append(duration / word_count)
    
    if word_durations:
        avg_duration = np.mean(word_durations)
        words_per_minute = 60 / avg_duration if avg_duration > 0 else 0
        
        print(f"Average word duration: {avg_duration:.3f}s")
        print(f"Estimated speech rate: {words_per_minute:.1f} words/minute")
    
    return word_durations

analyze_speech_rate("speech.mp3")

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5. Create Clickable Transcript

import whisper
import json

def create_interactive_transcript(audio_file: str, output_html: str):
    model = whisper.load_model("turbo")
    result = model.transcribe(audio_file, word_timestamps=True)
    
    html = """
    <!DOCTYPE html>
    <html>
    <head>
        <title>Interactive Transcript</title>
        <style>
            .word { cursor: pointer; }
            .word:hover { background-color: yellow; }
        </style>
    </head>
    <body>
        <audio id="audio" controls>
            <source src="{audio_file}" type="audio/mpeg">
        </audio>
        <div id="transcript">
    """.format(audio_file=audio_file)
    
    for segment in result["segments"]:
        for word in segment["words"]:
            html += f'<span class="word" onclick="seek({word["start"]})">{word["word"]}</span>'
    
    html += """
        </div>
        <script>
            function seek(time) {
                document.getElementById('audio').currentTime = time;
                document.getElementById('audio').play();
            }
        </script>
    </body>
    </html>
    """
    
    with open(output_html, "w", encoding="utf-8") as f:
        f.write(html)

create_interactive_transcript("podcast.mp3", "transcript.html")

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Limitations and Considerations

Word-level timestamps on translations may not be reliable, as the timing is based on the source language but the text is in English.

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Accuracy Factors

  • Word timestamps are more accurate for clear speech with minimal background noise
  • Fast speech or overlapping speakers can reduce accuracy
  • The probability field indicates confidence for each word
  • Very short words (< 0.133s) or very long words (> 2.0s) may indicate alignment issues

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Performance Impact

Enabling word timestamps:
  • Increases processing time (requires cross-attention analysis and DTW)
  • Uses additional memory for storing alignment data
  • Is most noticeable on longer audio files

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Best Practices

  1. Use higher-quality models (medium, large) for better timestamp accuracy
  2. Filter out low-probability words for critical applications
  3. Validate timestamps against actual audio for important use cases
  4. Consider segment-level timestamps for less critical applications
# Filter low-confidence words
result = model.transcribe("audio.mp3", word_timestamps=True)

for segment in result["segments"]:
    high_confidence_words = [
        word for word in segment["words"]
        if word["probability"] > 0.8
    ]
    print([w["word"] for w in high_confidence_words])

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