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Speech & Audio AI

AI That Hears,
Understands & Speaks

Presear builds production speech and audio AI systems — automatic speech recognition, text-to-speech, speaker diarisation, and audio classification — with enterprise-grade accuracy across languages.

97.3%
Word Accuracy (WER)
50+
Languages Supported
85+
Audio AI Systems Deployed
AUDIO TRANSCRIPT

Technical Depth

Six Speech & Audio AI Capabilities We Build With

From transcription to synthesis to emotion understanding — here are the core techniques powering our audio AI systems.

Automatic Speech Recognition (ASR)

Building end-to-end ASR systems — from fine-tuned Whisper and wav2vec2 models to domain-adapted CTC and attention-based encoder-decoder architectures — optimised for low word error rate in noisy, accented, and domain-specific speech. We handle telephony-quality audio, spontaneous speech, and technical vocabulary with custom language models.

Whisper wav2vec2 CTC / Attention Custom LM

Text-to-Speech Synthesis (TTS)

Generating natural, expressive speech from text using neural TTS architectures — Tacotron2, FastSpeech2, VITS — with voice cloning capabilities and prosody control for brand-consistent synthetic voices. We fine-tune for Indian languages, regional accents, and domain-specific pronunciation to produce voices indistinguishable from native speakers.

FastSpeech2 VITS Voice Cloning Prosody Control

Speaker Diarisation & Identification

Segmenting multi-speaker audio into per-speaker segments and identifying individual speakers using x-vector and ECAPA-TDNN embeddings. Our diarisation pipelines handle overlapping speech, varying channel conditions, and unknown speaker counts — producing labelled transcripts that attribute every utterance to the correct speaker in real time.

ECAPA-TDNN x-vectors Overlap Detection

Emotion & Sentiment in Voice

Extracting paralinguistic signals — emotional state, sentiment polarity, stress level, and engagement intensity — from speech prosody, energy, and spectral features using deep learning classifiers. Deployed in call centre analytics and customer experience monitoring to surface emotional signals that text analysis alone misses.

Emotion Recognition Sentiment in Audio Paralinguistics

Audio Classification & Sound Event Detection

Classifying audio streams into event categories — machine anomalies, environmental sounds, keyword spotting, music genre, and scene classification — using CNN-based spectrogram classifiers and transformer-based audio models. We build low-latency always-on classifiers suitable for edge deployment in industrial IoT and security monitoring applications.

Spectrogram CNN Keyword Spotting Anomaly in Audio

Noise Cancellation & Enhancement

Removing background noise, reverberation, and channel distortion from degraded speech using deep learning spectral suppression models — RNNoise, FullSubNet, and custom architectures trained on domain-specific noise profiles. Essential preprocessing for downstream ASR, speaker ID, and voice analytics applications in real-world noisy environments.

RNNoise Spectral Suppression Dereverberation

Our Process

From Raw Audio to Deployed Speech Intelligence

A rigorous five-stage process. Click any step to explore what happens — and why it matters.

01
Audio Data Collection
02
Pre-processing & Augmentation
03
Acoustic Model Training
04
Language Model Integration
05
Deployment & Streaming API
Step 01 of 05

Audio Data Collection

We audit your existing audio assets — call recordings, dictation files, broadcast archives — and assess coverage across target languages, accents, speaking styles, and acoustic conditions. Where gaps exist, we design data collection protocols, speaker diversity requirements, and annotation standards to build the training corpus needed for production accuracy.

  • Existing audio asset audit and quality assessment
  • Speaker diversity and accent coverage planning
  • Annotation standards: transcription, speaker labelling, timestamps
  • Data collection protocols for domain-specific vocabulary
Step 02 of 05

Pre-processing & Augmentation

Converting raw recordings into training-ready feature representations — silence trimming, normalisation, VAD segmentation, feature extraction (MFCCs, mel spectrograms, raw waveforms) — and augmenting with noise injection, speed perturbation, and room impulse response convolution to improve robustness to real-world acoustic conditions.

  • Voice activity detection and segment boundary extraction
  • Spectral feature extraction: MFCCs, mel spectrograms
  • Noise injection, speed perturbation, and SpecAugment
  • Channel and microphone variability simulation
Step 03 of 05

Acoustic Model Training

Fine-tuning or training acoustic models — Whisper, wav2vec2, Conformer, or ESPnet architectures — on domain-specific speech data. We apply parameter-efficient fine-tuning techniques to minimise data requirements, track training with WER/CER metrics across diverse test partitions, and validate against real deployment conditions before considering a model production-ready.

  • Architecture selection: Whisper, wav2vec2, Conformer
  • Fine-tuning with domain-specific transcribed audio
  • WER/CER tracking across language and accent test sets
  • Experiment tracking and model registry management
Step 04 of 05

Language Model Integration

Integrating n-gram and neural language models tuned on domain text corpora — product names, medical terminology, legal vocabulary, financial instruments — to improve transcription accuracy for specialised vocabulary that acoustic models alone cannot handle reliably. Shallow fusion and deep fusion approaches are evaluated for each deployment context.

  • Domain text corpus collection and LM training
  • n-gram and neural LM shallow/deep fusion
  • Custom vocabulary and hotword boosting
  • Punctuation and inverse text normalisation
Step 05 of 05

Deployment & Streaming API

Packaging speech models for production — building streaming WebSocket APIs with WebRTC compatibility for real-time transcription, batch REST APIs for offline processing, and on-premise containerised deployments for data-sensitive environments. We build observability layers tracking latency, WER drift, and language distribution shifts over time.

  • Real-time streaming API: WebSocket + WebRTC
  • Batch transcription REST API with async job queuing
  • On-premise and air-gapped deployment support
  • WER drift monitoring and automated retraining triggers

Real-World Impact

Speech & Audio AI Problems We've Solved

Production audio AI deployments across industries — each delivering measurable accuracy, efficiency, and experience improvements.

Call Centre Voice Analytics

Finance / Telecom

Core Challenge

Call centres process millions of interactions monthly, but only 1–2% are manually reviewed for quality assurance. Critical compliance breaches, customer dissatisfaction signals, and agent coaching opportunities go undetected — while manual review creates backlog and inconsistent standards across teams and shifts.

Who Benefits

Banks, insurance companies, telecom operators, and BPO providers that need 100% call coverage for compliance monitoring, automated QA scoring, churn signal detection, and agent coaching — without expanding the QA headcount linearly with call volume.

ASR Transcription Sentiment Analysis Compliance Monitoring
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Medical Voice Transcription

Healthcare

Core Challenge

Clinicians spend up to 40% of their working hours on documentation — dictating notes, updating EHRs, and transcribing patient consultations. Manual transcription is expensive, creates documentation delays, and introduces errors in technical medical terminology that general-purpose ASR systems cannot handle reliably.

Who Benefits

Hospitals, specialist clinics, and health IT providers that need domain-adapted ASR for clinical dictation, SOAP note generation, and real-time transcription of patient consultations — integrated directly into existing EMR/EHR workflows.

Medical ASR EHR Integration HIPAA Compliance
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Voice-Controlled Industrial HMI

Manufacturing

Core Challenge

Factory floor operators wearing gloves, working in high-noise environments, and operating heavy machinery cannot safely interact with touchscreen HMI panels. Traditional voice command systems fail in industrial acoustic conditions — high ambient noise, machinery vibration, and domain-specific command vocabularies.

Who Benefits

Equipment manufacturers, automotive assembly plants, and industrial automation vendors that need hands-free, noise-robust voice control for machinery operation, quality data entry, and maintenance workflows — improving both safety and throughput.

Noise-Robust ASR Keyword Spotting Edge Deployment
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Multilingual Customer Assistant

Retail

Core Challenge

Retailers serving multilingual markets need voice assistants that handle code-switching, regional accents, and colloquial phrasing across multiple languages simultaneously. Standard ASR models degrade significantly on Indian languages, regional dialects, and mixed-language queries that are common in real customer interactions.

Who Benefits

Retail chains, e-commerce platforms, and consumer service companies that serve linguistically diverse customer bases — needing voice assistants that work for Hindi, Tamil, Bengali, and other Indian languages at the same quality level as English.

Multilingual ASR Code-Switching Conversational AI
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Powered By

Our Speech & Audio AI Technology Ecosystem

Industry-standard frameworks, pre-trained models, and audio processing libraries — chosen for accuracy, speed, and production reliability.

Whisper ASR Foundation
wav2vec2 Self-Supervised ASR
ESPnet E2E Speech
SpeechBrain Speech Toolkit
Kaldi ASR Framework
DeepSpeech Mozilla ASR
Coqui TTS Text-to-Speech
PyTorch Audio Audio Deep Learning
librosa Feature Extraction
WebRTC Real-Time Audio
FastAPI Serving API
Docker Containerisation

Frequently Asked

Speech & Audio AI Questions

Answers to the questions product leaders, data science teams, and compliance officers ask before starting a speech AI engagement with Presear Softwares.

Ask Our Speech AI Team
Which languages do you support?
Our ASR systems support 50+ languages with production-grade accuracy, including all major European languages, Mandarin, Japanese, Arabic, and a comprehensive set of Indian languages: Hindi, Tamil, Telugu, Kannada, Malayalam, Bengali, Marathi, Gujarati, Punjabi, and Odia. For TTS, we support voice synthesis in 30+ languages. For less-resourced languages or regional dialects, we design data collection programs to build the training corpus required for target accuracy levels.
How do you handle accents and code-switching?
Accent robustness is built into the training pipeline through accent-stratified data collection, acoustic model fine-tuning on target accent distributions, and test set evaluation across accent groups. For code-switching (e.g., Hindi-English, Tamil-English), we train multi-lingual models on mixed-language corpora and deploy language identification as a preprocessing step to route audio to the optimal model. The approach is calibrated against your specific user population's linguistic patterns, not a generic multilingual benchmark.
Can it run in real-time with low latency?
Yes. Our streaming ASR architecture delivers partial transcriptions with first-token latency under 300ms and word-level latency under 600ms for most configurations. We achieve this through chunked audio streaming, CTC beam search with early emission, and model quantisation for inference acceleration. For use cases requiring sub-100ms latency, we combine a lightweight always-on keyword spotter with a full ASR model triggered on detection — reducing compute cost without sacrificing accuracy on full utterances.
Do you offer on-premise deployment?
Yes. All our speech AI systems are designed for flexible deployment from the outset. On-premise deployment is fully supported via Docker/Kubernetes containers that run on your own hardware, enabling air-gapped operation with no audio data leaving your network. This is the standard choice for healthcare, banking, legal, and government deployments where data residency and regulatory compliance requirements prevent cloud processing. On-prem performance is identical to cloud deployment for the same hardware configuration.
How do you fine-tune for domain-specific terminology?
Domain adaptation combines two complementary approaches. First, we fine-tune the acoustic model on domain-specific speech examples — clinical dictations, financial call recordings, legal proceedings — to improve phoneme-level recognition of specialised pronunciation. Second, we train or shallow-fuse a domain language model on relevant text corpora (medical literature, financial reports, product catalogues) to boost the probability of domain vocabulary in the decoding lattice. Together, these reduce terminology error rates by 40–70% compared to a generic model baseline.
Speech & Audio AI

Ready to Build Speech AI
That Performs in the Real World?

Partner with Presear Softwares to build speech and audio AI systems with enterprise-grade accuracy — domain-adapted, multilingual, and designed to deliver value from day one.