Streaming ASR Architecture
Why batch ASR won’t work for voice assistants, and how streaming models transcribe speech as you speak in under 200ms.
Technical deep-dives into speech and audio ML systems, from research foundations to production deployment. Covers ASR, TTS, speaker recognition, and conversational AI with both theoretical insights and practical engineering.
Each post includes:
Automatic Speech Recognition (ASR):
Speech Classification:
Audio Processing:
Real-time Processing:
Text-to-Speech (TTS):
Speaker Technology:
Speech Separation:
Pipeline Optimization:
Conversational AI:
Model Optimization:
Speech MLOps:
Adaptive Systems:
Speech Model Design:
Below you’ll find all speech technology posts in chronological order:
Content created with the assistance of large language models and reviewed for technical accuracy.
Why batch ASR won’t work for voice assistants, and how streaming models transcribe speech as you speak in under 200ms.
How voice assistants recognize “turn on the lights” from raw audio in under 100ms without full ASR transcription.
How to transform raw audio waveforms into ML-ready features that capture speech characteristics for robust model training.
How voice assistants and video conferencing apps detect when you’re speaking vs silence, the critical first step in every speech pipeline.
How voice assistants recognize who’s speaking, the biometric authentication powering “Hey Alexa” and personalized experiences.
From text to natural speech: understanding modern neural TTS architectures that power Alexa, Google Assistant, and Siri.
Clean audio is the foundation of robust speech systems, master preprocessing pipelines that handle real-world noise and variability.
Build real-time speech processing pipelines that handle audio streams with minimal latency for live transcription and voice interfaces.
Build lightweight models that detect specific keywords in audio streams with minimal latency and power consumption for voice interfaces.
Build systems that enhance voice quality by removing noise, improving intelligibility, and optimizing audio for speech applications.
Separate overlapping speakers with 99%+ accuracy: Deep learning solves the cocktail party problem for meeting transcription and voice assistants.
Build production multi-speaker ASR systems: Combine speech recognition, speaker diarization, and overlap handling for real-world conversations.
Optimize speech pipeline throughput by allocating compute to bottleneck stages using greedy resource management.
Build production speech systems that combine multiple ASR/TTS models using backtracking-based selection strategies to achieve state-of-the-art accuracy.
Build production speaker diarization systems that cluster audio segments by speaker using embedding-based similarity and hash-based grouping.
Build production audio segmentation systems that detect boundaries in real-time using interval merging and temporal processing—the same principles from merge...
Design distributed training pipelines for large-scale speech models that efficiently handle hundreds of thousands of hours of sequential audio data.
Use audio augmentation techniques to make speech models robust to noise, accents, channels, and real-world conditions—built on the same matrix/tensor transfo...
Design experiment management systems tailored for speech research—tracking audio data, models, metrics, and multi-dimensional experiments at scale.
Design adaptive speech models that adjust in real-time to speakers, accents, noise, and domains—using the same greedy adaptation strategy as Jump Game and on...
Design neural architecture search systems for speech models that automatically discover optimal ASR/TTS architectures—using dynamic programming and path opti...
Strategies for building profitable speech recognition systems by optimizing the entire pipeline from signal processing to hardware.
Implementing the core decoding logic of modern Speech Recognition systems, handling alignment, blanks, and language models.
The breakthrough that allows us to treat audio like text, enabling GPT-style models for speech.
How do we know if the audio sounds “good” without asking a human?
Real-time ASR is hard. Offline ASR is big.
Goodbye HMMs. Goodbye Phonemes. Goodbye Lexicons. We are teaching the machine to Listen, Attend, and Spell.
“Play Call Me Maybe”. Did you mean the song, the video, or the contact named ‘Maybe’?
“From broad categories to fine-grained speech understanding.”
“Building recommendation and moderation systems for voice-based social platforms.”
“Orchestrating complex speech processing pipelines from audio ingestion to final output.”
“Finding ‘Jon’ when the user types ‘John’, or ‘Symphony’ when they say ‘Simfoni’.”
“Deploying speech models close to users for low-latency voice experiences.”
“The brain of a task-oriented dialogue system: remembering what the user wants.”
“Knowing when to listen and when to stop.”
“One model to rule them all: ASR, Translation, and Understanding.”
“From waveforms to words, and back again.”
“Tuning speech models for peak performance.”
“Giving machines a voice.”
“Hey Siri, Alexa, OK Google: The gateway to voice AI.”
“Who spoke when? The art of untangling voices.”
“Turning acoustic probabilities into coherent text.”
“Extracting clear speech from the noise of the real world.”
“Speaking with someone else’s voice.”
“Teaching machines to hear feelings.”