Speech AI¶
Turning speech into text (ASR), text into speech (TTS), and building voice-driven AI applications.
Overview¶
Voice interfaces are built from a pipeline of three stages, often wrapped around an LLM:
flowchart LR
A[๐๏ธ Audio in] --> B[ASR<br/>speech โ text]
B --> C[๐ง LLM<br/>understand & respond]
C --> D[TTS<br/>text โ speech]
D --> E[๐ Audio out]
- ASR (Automatic Speech Recognition) โ transcribe audio to text. Whisper is the open standard; hosted APIs are often faster and simpler.
- TTS (Text-to-Speech) โ synthesize natural-sounding speech from text.
- Diarization โ figure out who spoke when (speaker labels).
Learning Objectives¶
By the end of this section you will be able to:
- Transcribe audio with an ASR model and handle timestamps.
- Choose between local (Whisper) and hosted transcription.
- Build a simple voice-assistant loop (ASR โ LLM โ TTS).
- Understand latency budgets for real-time voice.
Quick taste: transcribe audio¶
transcribe.py
# Using the open-source Whisper model locally (CPU works, GPU is faster).
import whisper
model = whisper.load_model("base") # tiny/base/small/medium/large
result = model.transcribe("meeting.mp3")
print(result["text"])
Best Practices¶
- โ Match model size to your latency and accuracy needs โ bigger isn't always worth it.
- โ Stream audio for real-time UX; batch for offline transcription.
- โ Clean audio (noise reduction, correct sample rate) before ASR.
Common Mistakes¶
- โ Ignoring latency โ a 3-second round trip feels broken in a voice conversation.
- โ Sending raw, noisy audio and blaming the model for poor transcripts.
- โ Forgetting to handle silence and interruptions in live voice apps.
๐ Help build this section¶
Claim a topic by opening an issue:
- โ Speech-to-Text (ASR) โ Whisper, timestamps, model sizing ๐ก
[WANTED]Real-time voice pipelines โ streaming, VAD, barge-in ๐ด- โ Text-to-Speech (TTS) โ local vs hosted, streaming, SSML ๐ก
[WANTED]Speaker diarization โ who said what ๐ด[WANTED]Meeting assistant example โ record โ transcribe โ summarize ๐ก