Alternatives to Decibri

Compare Decibri alternatives for your business or organization using the curated list below. SourceForge ranks the best alternatives to Decibri in 2026. Compare features, ratings, user reviews, pricing, and more from Decibri competitors and alternatives in order to make an informed decision for your business.

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    Dessa

    Dessa

    Block

    Dessa is Block's speech-to-speech artificial intelligence model for conversational voice research and applications. It takes audio input and produces audio output directly, without converting speech into text as an intermediate step. Trained from scratch on human conversation rather than text, the model supports full-duplex interaction, listening and speaking at the same time, and runs locally on a MacBook Pro with an M4 Max chip or better. It captures conversational cues such as emotion, sarcasm, laughter, pauses, and turn-taking that can be lost in text-based pipelines. Dessa is intended for AI safety researchers and developers studying voice-model behavior, privacy, and real-time interaction. It is a speech model for research and development, not a general-purpose business application or hosted voice-agent platform.
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    Voxtral Mini Transcribe Realtime 3
    Voxtral Mini Transcribe Realtime 3 is a speech-to-text model delivered through Mistral’s API for applications that need live transcription. It streams audio over WebSocket and returns transcript events while a person is speaking, with configurable streaming delays. Developers can use it to add real-time captions, voice-assistant input, or transcription to voice-agent workflows. The model provides turn detection, speaker diarization, and context biasing for domain-specific vocabulary, and supports transcription across 45 languages and dialects. Applications can connect to its real-time endpoint and receive transcription events as audio arrives, rather than waiting for an entire recording to finish. Voxtral Mini Transcribe Realtime 3 is part of Mistral’s Voxtral audio model family.
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    Seeduplex

    Seeduplex

    ByteDance

    Seeduplex is a native full-duplex speech large language model built on a new “listen while speaking” framework for more natural, fluid, and precisely paced voice interaction. Unlike traditional half-duplex systems that alternate between listening and replying, it continuously receives and understands user-side audio, allowing it to listen and speak simultaneously while tracking the broader acoustic environment. Its high-precision interference suppression distinguishes genuine user interaction from background noise, broadcasts, navigation prompts, side conversations, and overlapping voices, reducing false responses and false interruptions in complex settings. Seeduplex also combines speech and semantic features for adaptive endpoint detection, helping it recognize when a user is thinking, hesitating, correcting themselves, or has actually finished speaking. It can wait patiently through reflective pauses, respond quickly once an utterance ends, and stop smoothly when interrupted.
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    Azure Voice Live API
    Azure Voice Live API is a fully managed solution for building low-latency, high-quality speech-to-speech agents through one unified interface. It combines speech recognition, generative AI, and text-to-speech, allowing developers to send audio input and receive audio output, synchronized avatar visuals, and action triggers without manually orchestrating separate backend components or deploying the underlying models. It supports more than 140 speech-to-text locales and over 600 standard voices across 150+ text-to-speech locales, with options for phrase lists, custom speech, custom voices, and brand-aligned avatars. Developers can choose among multiple generative AI models, including GPT-Realtime, GPT-5, GPT-4.1, GPT-4o, Phi, and compatible bring-your-own models, depending on the intelligence, speed, and latency required. Advanced conversational features include noise suppression, echo cancellation, robust interruption detection, and end-of-turn detection.
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    AssemblyAI

    AssemblyAI

    AssemblyAI

    Automatically convert audio and video files and live audio streams to text with AssemblyAI's speech-to-text APIs. Do more with audio intelligence, summarization, content moderation, topic detection, and more. Powered by cutting-edge AI models. From in-depth tutorials to detailed changelogs, to comprehensive documentation, AssemblyAI is focused on providing developers a great experience every step of the way. From core speech-to-text conversion to sentiment analysis, our simple API offers a full suite of solutions catered to all your business speech-to-text needs. We work with startups of all sizes, from early-stage startups to scale-ups, by providing cost-efficient speech-to-text solutions. We're built for scale. We process millions of audio files every day for hundreds of customers, including dozens of Fortune 500 enterprises. Universal-2: Our most advanced speech-to-text model captures the complexity of human speech for impeccable audio data that powers sharper insights.
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    Phonon

    Phonon

    Gradium

    Phonon is an open-weight text-to-speech (TTS) model and inference runtime for developers adding speech generation to applications that need local, offline audio. It converts text into spoken audio and can stream speech as it is generated. The runtime provides Rust and Python interfaces, a Swift package for iOS and macOS, JavaScript for browsers, and Android libraries; it can also run as a self-hosted server with an OpenAI-compatible API. Depending on the model variant, it supports English, French, German, Spanish, and Portuguese. Phonon is intended for developers integrating speech into mobile, desktop, web, embedded, and server applications, including offline or privacy-sensitive use cases. The 31-million-parameter English model weights use the Apache 2.0 license; 40-million-parameter language-specific and multilingual weights use CC BY-NC 4.0. The runtime code is separately available under Apache 2.0 or MIT.
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    Silkwave Voice
    Silkwave Voice is a privacy-focused audio recording and transcription app for macOS. Record from your microphone, system audio, or both at once - with accurate, real-time transcription powered by Apple's on-device speech-to-text models. No cloud uploads, no subscriptions, no per-minute API costs. RECORD ANY AUDIO SOURCE • Microphone - voice notes, in-person meetings, dictation • System Audio - Zoom, Google Meet, Teams, YouTube, browser tabs • Both at once - capture your mic and remote participants simultaneously ON-DEVICE TRANSCRIPTION • Real-time speech-to-text using Apple's on-device models • 10 languages: Cantonese, Chinese, English, French, German, Italian, Japanese, Korean, Portuguese, Spanish • Completely local - no internet connection needed AI-POWERED SUMMARIES • Structured summaries with key topics, action items, and decisions • Powered by ChatGPT through Apple Intelligence - no API keys needed
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    Voxtral Small Transcribe 3
    Voxtral Small Transcribe 3 is a multilingual speech-to-text model for transcribing recorded audio. Mistral provides it through an audio transcription API and documents timestamps, speaker diarization, and context biasing, which allows developers to supply key phrases to help recognize domain-specific terms and names. The model automatically detects supported languages or accepts a language code, and supports more than 50 languages. It is designed for application developers building recorded-audio workflows, including meeting, interview, and call transcription, customer-service systems, and voice-agent pipelines. Mistral also describes self-deployment options for the Voxtral 3 family, including private-cloud and on-premises hosting.
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    StepAudio 3
    StepAudio 3 is StepFun’s next-generation audio model family, built to understand, generate, and interact through voice, sound, and music. The lineup includes StepAudio 3 Realtime for natural full-duplex conversation, StepAudio 3 ASR for speech recognition, StepAudio 3 TTS for speech synthesis, StepAudio 3 Gen for general-purpose audio generation, and StepAudio 3 Music for long-form music creation. Realtime is designed around a continuous listen-converse-think-act loop, understanding not only words but also hesitation, laughter, emotion, pauses, backchannels, and interruptions. It can think while speaking, reason through harder questions without breaking conversational flow, and use tools to complete tasks once it understands the user’s intent. StepAudio 3 Gen unifies zero-shot TTS, voice design, vocal generation, sound effects, music, and mixed audio generation within one framework, while StepAudio 3 Music supports text-controlled songs, instrumentals, vocal arrangement, and more.
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    Fish Audio

    Fish Audio

    Hanabi AI

    Fish Audio provides innovative AI-powered solutions for text-to-speech (TTS), voice cloning, and speech-to-text (STT) technologies. The platform is designed for businesses and developers looking to integrate high-quality, realistic voice synthesis into their applications. Fish Audio offers voice cloning tools that allow users to replicate voices, and its generative AI technology can produce expressive, natural-sounding speech in multiple languages. Additionally, Fish Audio supports an API for easy integration and has expanded capabilities with a voice activity detection feature. Whether for content creation, virtual assistants, or customer support, Fish Audio offers powerful solutions for a variety of industries.
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    Higgs Audio / Avatar
    Higgs Audio / Avatar is a family of foundation audio and avatar models designed to generate natural speech, understand tone, emotion, and intent, and give voice interactions a visual presence. The models support text-to-speech, speech-to-text, avatar generation, and automatic voice casting that selects an appropriate voice based on context, sentiment, and content. Built for real-world production, Higgs combines expressive generation, robust speech understanding, and flexible deployment for workloads where quality, latency, and reliability matter. High-accuracy multilingual speech recognition supports major languages, while voice cloning reproduces a speaker’s tone from short reference samples to maintain consistent brand voices across interactions. Sentiment detection reads emotional signals in speech to enable smarter routing, stronger analytics, and more context-aware agent behavior.
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    Boson AI

    Boson AI

    Boson AI

    Boson AI provides voice agents powered by foundation audio models, built to run in business workflows and learn from every call. Higgs Realtime enables live voice agents for support lines, sales calls, and product assistants that listen, reason, call tools, and respond in real time with low latency and natural speech-to-speech interaction. Higgs Audio and Avatar extend these capabilities with text-to-speech, speech-to-text, voice cloning, sentiment detection, and avatar generation, producing natural speech while understanding tone, emotion, and intent. The models support high-accuracy multilingual speech recognition, real-time translation, and expressive voice generation, while sentiment signals can improve routing, analytics, and context-aware agent behavior. Designed for real-world production, the platform emphasizes quality, latency, reliability, and flexible deployment across managed and self-serve environments.
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    Diffio AI

    Diffio AI

    Diffio AI

    Diffio.ai is an AI-powered audio denoising solution built specifically for spoken-word content. It restores voices in podcasts, interviews, and calls by removing background noise, echo, and hiss while keeping speech clear, natural, and consistent.
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    Voxtral Mini TTS 3.0
    Voxtral Mini TTS 3.0 is Mistral AI's text-to-speech model for developers building voice-enabled applications and services. It generates speech from text in batch and real-time workflows, with preset or cloned voices, emotion and pace controls, and custom pronunciation. The model supports multilingual speech generation and dialect adaptation, making it suitable for conversational audio, localized content, spoken responses, and voice agents. Developers can access speech generation and voice-cloning capabilities through Mistral's API. Voxtral Mini TTS 3.0 is also part of Mistral's Voxtral 3 voice-agent stack, which combines speech recognition, reasoning, tool use, and generated speech; the Voxtral 3 stack can be self-deployed on customer infrastructure.
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    Baidu AI Cloud Speech-to-Text
    Baidu’s speech technology provides developers with such industry-leading capabilities as speech-to-text,text-to-speech, and speech wake-up. Combining with the NLP technology, it is applicable for several scenarios, including speech input, speech search, video subtitle, audio content analysis, calling center, book broadcasting, news broadcasting, and order broadcasting. It can convert a speech with a duration of fewer than 60 seconds to characters. It is applicable for mobile speech input, intelligent speech interaction, speech commands, and speech search. It can convert the audio stream into characters and return each sentence's start and end times. It is applicable for such scenarios as long-sentence speech input, audio and video subtitles, and meeting records. It can convert the audio files uploaded in batches into characters and return the recognition results within 12 hours. It is applicable for such scenarios as record quality check, and audio content analysis.
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    Note67

    Note67

    Note67

    Note67 is a privacy-centric meeting assistant designed for professionals who demand total control over their data. Unlike traditional transcription tools that rely on cloud processing, Note67 is an open-source, local-first application for macOS that captures audio, transcribes speech, and generates intelligent summaries entirely on your device. No audio or text ever leaves your machine, ensuring zero data leakage. Built with performance and security in mind, the application leverages the power of Rust and Tauri to deliver a lightweight, native experience. It integrates seamless local AI capabilities, utilizing Whisper for high-accuracy speech-to-text and Ollama for generating insightful meeting summaries using local Large Language Models (LLMs). Key Features: 100% Local Processing: Powered by on-device Whisper models, ensuring your audio and transcripts remain completely private.
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    AudioLM

    AudioLM

    Google

    AudioLM is a pure audio language model that generates high‑fidelity, long‑term coherent speech and piano music by learning from raw audio alone, without requiring any text transcripts or symbolic representations. It represents audio hierarchically using two types of discrete tokens, semantic tokens extracted from a self‑supervised model to capture phonetic or melodic structure and global context, and acoustic tokens from a neural codec to preserve speaker characteristics and fine waveform details, and chains three Transformer stages to predict first semantic tokens for high‑level structure, then coarse and finally fine acoustic tokens for detailed synthesis. The resulting pipeline allows AudioLM to condition on a few seconds of input audio and produce seamless continuations that retain voice identity, prosody, and recording conditions in speech or melody, harmony, and rhythm in music. Human evaluations show that synthetic continuations are nearly indistinguishable from real recordings.
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    Eleven v4

    Eleven v4

    ElevenLabs

    Eleven v4 is a text-to-speech model from ElevenLabs for generating expressive spoken audio from written scripts. It supports multi-speaker dialogue, emotional direction and audio-event tags, voice cloning, and consistent speaker identity across generated scenes. Creators, publishers, and production teams can use it for narration, audiobooks, character dialogue, video and game voiceovers, and multilingual projects. Developers can generate speech through the ElevenLabs web application or API; the related Eleven v4 Turbo model is intended for real-time speech and voice-agent scenarios. Eleven v4 supports 90+ languages. A free plan includes monthly credits for personal use, while paid plans provide higher credit limits and additional features.
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    Supavocal

    Supavocal

    Supavocal

    Supavocal is an AI voice studio for text-to-speech, voice cloning, and speech-to-text. Turn text into expressive, studio-quality speech, clone any voice from about 15 seconds of audio, and transcribe recordings to text. Teams use Supavocal for video voiceovers, audiobook narration, character voices for games and animation, conversational chatbots and voice agents, and transcription, with a voice API for developers.
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    Rekam AI

    Rekam AI

    Rekam AI

    Rekam AI is an all-in-one voice creation platform offering text to speech, speech to text, voice cloning, and AI voice generation. It uses high-quality, human-like voice models to transform written text into natural-sounding audio. Rekam AI provides a free text-to-speech tool that allows users to generate lifelike narration instantly. The platform includes a curated voice library with multiple male and female voices across accents and tones. Voice cloning enables users to create realistic digital voice replicas using short audio samples. Rekam AI also supports accurate speech-to-text transcription for meetings, interviews, and content creation. Overall, it serves as a complete voice studio for modern audio production.
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    Beey

    Beey

    NEWTON Technologies

    Beey is an application which transcribes audio or video recordings into text with great accuracy in a few minutes. Beey can recognize speech in 20 languages. The user-friendly editor provides further processing of the transcribed text, export to various formats, and creating automatic subtitles or translation. The editor includes a recording preview synchronized with the edited text, which is illustrated by the moving cursor position. Editor controls allow slowing down, speeding up the playback, or starting the playback from the selected cursor position. Beey offers several additional tools: Link, Splitter, Stream and Voice. Link allows transcribing the video/audio directly from global platforms, such as YouTube. Splitter is convenient for working with long content. It splits the original recording into shorter ones, and users can work with them separately. Stream can perform real-time transcription, and caption ongoing streams. Voice records and transcribes live speech.
    Starting Price: €7.50 EUR per hour
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    Higgs Realtime
    Higgs Realtime is a production-quality, real-time speech-to-speech model and API built for natural, continuous conversation. It is an end-to-end, instruction-tuned, audio-native model that can understand audio, text, or both and generate high-quality responses, while also functioning as a text LLM when given text alone. Designed for live voice agents, it follows conversations, handles interruptions, adapts when requests change mid-sentence, and carries multi-step workflows through to completion. The model is trained specifically for voice-agent reflexes such as natural turn-taking, conversational cadence, tone adaptation, spoken tool preambles, multi-turn state tracking, and robust instruction following through changing requests. Semantic turn detection helps distinguish a completed turn from a pause, while multilingual and code-switched understanding supports more than 100 languages without per-language setup.
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    TTSLabs

    TTSLabs

    TTSLabs

    TTSLabs gives streamers the ability to customize their text-to-speech donations, enable custom voices, add unique sound clips and more! Seamless management and playback of text-to-speech. Allows easy customization of prices, voices, clips, and more. 20 seconds of audio can be generated in less than 3 seconds, even on an entry-level CPU. Sync our desktop app to allow your moderators to control text-to-speech through Streamlabs or StreamElements dashboard. Viewers can check enabled alerts, voices, clips, and minimum values for text-to-speech. Contact us to get your own unique voice! Get access to your own and other voices on your stream! Dedicated desktop app, faster than real-time processing. Sync with Streamlabs and StreamElements, with custom guides for viewers.
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    Orate

    Orate

    Orate

    Orate is an AI toolkit for speech that enables developers to create realistic, human-like speech and transcribe audio through a unified API compatible with leading AI providers such as OpenAI, ElevenLabs, and AssemblyAI. The platform offers text-to-speech functionality, allowing users to convert text into lifelike speech using a simple API that integrates seamlessly with various providers. For instance, by importing the 'speak' function from Orate and the desired provider, developers can generate speech from text prompts. Additionally, Orate provides speech-to-text capabilities, transforming spoken words into meaningful text with unparalleled accuracy, speed, and reliability. By importing the 'transcribe' function and the chosen provider, users can transcribe audio files into text. The toolkit also supports speech-to-speech transformations, enabling users to change the voice of their audio using a straightforward voice-to-voice API compatible with leading AI providers.
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    Rev AI
    Rev AI is a speech-to-text API platform that helps developers convert prerecorded audio and real-time audio streams into accurate transcripts. The platform is designed for accuracy, speed, global scale, and developer-friendly implementation. Rev AI supports speech-to-text across 57+ languages with grammar, punctuation, formatting, and low word error rates. Its proprietary models are trained using a large library of human-verified speech data to improve precision across voices, accents, and use cases. Rev AI also includes AI insights such as language identification, sentiment analysis, topic extraction, summarization, translation, and precise word-level timestamps. Built for developers and enterprises, Rev AI helps teams turn speech into searchable, analyzable, and actionable text.
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    Phonexia Speech Platform
    Phonexia offers a comprehensive portfolio of cutting-edge speech recognition and voice biometrics technologies ready to meet any commercial and governmental scenarios. Powered by the latest advancements in artificial intelligence, acoustics, phonetics, and voice biometrics science, Phonexia products are extremely accurate, fast, and scalable. Phonexia’s AI-powered solutions let you build voicebots, verify a speaker’s identity based on voice biometrics, transcribe speech to text, and search for speakers and context in large amounts of audio. Secure access to your clients’ data conveniently with voice biometric authentication and detect fraud attempts natively. Phonexia offers a comprehensive portfolio of cutting-edge speech recognition and voice biometrics technologies ready to meet any commercial and governmental scenarios. Powered by the latest advancements in artificial intelligence, acoustics, phonetics, and voice biometrics science.
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    Grok Voice 3.0
    Grok Voice 3.0 is a speech-to-speech model offered through the Grok Voice Agent API for developers building real-time voice applications. It accepts audio and text and returns spoken audio or text through a WebSocket session, supporting conversational turn-taking, interruptions, and function calling. Developers can connect the model to custom functions and server-side tools, including web search, X search, collections, and remote MCP servers. The API is intended for interactive voice agents such as customer-support and service-call experiences, and is also used for voice interactions in Tesla vehicles and Grok Bot. Billing is usage-based, with audio charged per minute and text input events billed separately. The API documentation includes setup instructions and examples for Python and JavaScript clients.
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    Azure Speech to Text
    Quickly and accurately transcribe audio to text in more than 85 languages and variants. Customize models to enhance accuracy for domain-specific terminology. Get more value from spoken audio by enabling search or analytics on transcribed text or facilitating action, all in your preferred programming language. Get accurate audio to text transcriptions with state-of-the-art speech recognition. Add specific words to your base vocabulary or build your own speech-to-text models. Run Speech to Text anywhere, in the cloud or at the edge in containers. Access the same robust technology that powers speech recognition across Microsoft products. Convert audio to text from a range of sources, including microphones, audio files, and blob storage. Use speaker diarisation to determine who said what and when. Get readable transcripts with automatic formatting and punctuation. Tailor your speech models to understand organization- and industry-specific terminology.
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    Voisi

    Voisi

    Teknikforce

    Voisi is an innovative AI-powered toolkit that revolutionizes the way you create, manage, and utilize voice and language content. Ideal for businesses, educators, content creators, and developers, Voisi offers a comprehensive suite of tools designed to enhance and streamline your audio and linguistic needs. Whether you're looking to generate lifelike speech from text, transcribe spoken words into written form, or translate audio across multiple languages, Voisi provides state-of-the-art solutions that are both powerful and easy to use. Features of Voisi: Text-to-Speech Conversion: Voisi enables users to convert written text into natural, human-like speech in a variety of languages and accents. This feature is perfect for creating voice-overs, narrations, and interactive voice responses. Speech-to-Text Transcription: Transform audio files into text quickly and accurately.
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    Dograh

    Dograh

    Dograh

    Dograh is an open source, self-hostable voice agent platform with a no-code workflow builder for creating production voice agents. Teams can choose their own inbound channels, speech-to-text, LLM, text-to-speech, and telephony providers, or replace the traditional cascade with speech-to-speech models for direct audio-in, audio-out conversations with natural turn-taking, interruption handling, and low latency. The platform supports inbound and outbound calling, widgets, telephony integrations, observability, traces, real-time analytics, hybrid pre-recorded voice plus TTS, and more than 70 languages. Its MCP server lets Claude Code, Cursor, OpenClaw, Codex, and other agent runtimes create, modify, and deploy voice agents directly from development environments. Dograh can run on your own servers, inside a private cloud or VPC, or in a managed environment, with support for models hosted entirely within your perimeter.
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    Gemini Audio
    Gemini Audio is a set of advanced real-time audio models built on Gemini's architecture, designed to enable natural, fluid voice interaction and expressive audio generation through simple language prompts. It supports conversational experiences where users can speak, listen, and interact with AI in a seamless loop, combining understanding, reasoning, and response generation in audio form. It is capable of both analyzing and generating audio, allowing applications such as speech-to-text transcription, translation, speaker identification, emotion detection, and detailed audio content analysis. They are optimized for low-latency, real-time use cases, making them suitable for live assistants, voice agents, and interactive systems that require continuous, multi-turn dialogue. Gemini Audio also integrates advanced capabilities like function calling, enabling the model to trigger external tools and incorporate real-time data into responses.
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    Picovoice

    Picovoice

    Picovoice

    Picovoice is the first and only ubiquitous on-device voice AI platform. Picovoice offers speech-to-text, voice search, wake word, Speech-to-Intent (intent detection) and voice activity detection engines. Its stack can run on anything from embedded devices to web browsers, providing an immersive experience not achievable by any Big Tech.
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    Hume Audio Decisions API
    Hume Audio Decisions API is a hosted API for developers building voice applications and teams evaluating speech and sound. It accepts an audio recording, optional context, and user-defined questions, then returns probabilities for answers in structured form. Questions can cover observable audio facts, such as the number of speakers or background speech, and listener judgments about tone, naturalness, empathy, clarity, or frustration. Teams can use the results to review support calls, test voice agents and text-to-speech output, label speech corpora, compare recordings, and identify examples for escalation or model training. The API works as a decision layer alongside speech recognition and voice generation, allowing application code to use its scores and probabilities in custom workflows. API access is requested through Hume, and billing is based on audio, question, and context input tokens.
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    Inworld TTS
    Inworld TTS is a state-of-the-art text-to-speech platform designed to deliver ultra-realistic, context-aware speech synthesis and precise voice-cloning capabilities at a radically accessible price. The flagship model, TTS-1, is optimized for real-time applications and supports low-latency streaming (first audio chunk in ≈200 ms) as well as multiple languages (including English, Spanish, French, Korean, Chinese, and more). Developers can use instant zero-shot voice cloning (5-15 seconds of audio) or professional fine-tuned cloning, add voice-tags for emotion, style, and non-verbal sounds, and switch languages while preserving voice identity. The larger TTS-1-Max model (in preview) offers even more expressive speech and multilingual strength. The platform supports both API and portal access, streaming or batch mode, and is designed for everything from interactive voice agents and gaming characters to branded audio experiences.
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    KugelAudio

    KugelAudio

    KugelAudio

    KugelAudio is the most realistic speech AI platform, combining text-to-speech, speech-to-text, and voice-to-voice in one stack. With 39-50ms inference latency (lowest on the market), 30-second voice cloning, on-premises deployment, and industry-leading accuracy on email addresses, IBANs, and phone numbers, it's built for production voice applications where quality and compliance matter. It's a strong fit for voice bots and conversational agents that need to handle structured data without misreads, real-time applications requiring sub-50ms latency, and regulated industries like banking, insurance, healthcare, and the public sector that need on-premises or EU-sovereign deployment. Beyond enterprise voice automation, KugelAudio also powers branded voice experiences through natural cloning from 30 seconds of audio, multilingual products across over 30 languages German, English, French, and Italian, and media or content production needing the most realistic synthetic voices available.
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    Echo Live

    Echo Live

    Escalera Labs S.L.

    Change your voice live for games, Discord and streams. Echo Live combines AI voice changing, a soundboard, text-to-speech and Voice Lab on Windows 10/11 x64 and Apple Silicon Mac. Build custom effects with pitch, formants, harmonizer, reverb and delay. Import sound clips and trigger them with hotkeys. Turn text into spoken clips with downloadable models. Guided setup helps configure your microphone and audio routing. Voice processing runs locally; performance varies by hardware and model. Interface languages: English, German, Spanish, French, Portuguese, Japanese, Korean, Traditional Chinese, Simplified Chinese, Polish, Russian, Italian and Turkish. Some text falls back to English. Speech-language support depends separately on the selected model. Download at voicechanger.live. A free Echo account is required. Start with the free version; optional paid Pro features are available.
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    ai-coustics

    ai-coustics

    ai-coustics

    ai-coustics is a Berlin-based startup building the audio intelligence layer for Voice AI. Founded by researchers in audio, acoustics, and machine learning, the company focuses on the fundamental reliability problem that causes voice systems to fail outside controlled environments. Rather than competing with ASR, LLMs, or TTS, ai-coustics makes them reliable. Its SDK and model infrastructure sit between real-world sound and machine understanding, conditioning raw audio into stable, machine-ready input optimized for downstream behavior. The company’s Quail model family delivers real-time speech enhancement, speaker isolation, and voice activity detection designed specifically for production of Voice AI. ai-coustics powers voice agents, transcription pipelines, and telephony systems, and is natively integrated in LiveKit and Pipecat. Its mission is to make audio input reliable and measurable, so voice systems can operate with confidence where real people actually speak.
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    Grok Voice Think Fast 2.0
    Grok Voice Think Fast 2.0 is xAI’s flagship voice model for building real-time assistants, phone agents, and interactive voice systems that stream audio and text bidirectionally over WebSocket. Developers can configure system instructions, high or no reasoning effort, built-in or custom voices, automatic server-side voice activity detection, silence duration, idle re-engagement, playback speed, and session resumption after temporary disconnects. It accepts PCM, G.711 μ-law, G.711 A-law, and Opus audio through JSON or raw binary frames, with configurable PCM sample rates from telephone quality to 48 kHz. It supports more than 20 languages with native-quality accents, automatic language detection, natural responses in the speaker’s language, and seamless code-switching. Language hints and up to 100 key terms improve transcription of regional speech, names, products, codes, addresses, and specialized terminology, while pronunciation replacements correct spoken output.
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    Vocallab AI

    Vocallab AI

    Vocallab AI

    Vocallab AI is a professional text-to-speech platform that delivers high-quality, realistic AI-generated voices for all your audio content needs. Transform written text into natural-sounding speech with advanced voice synthesis technology tailored for creators and businesses. Features: • Text to Speech: Turns your written documents or scripts into clear spoken audio. • Natural Voices: Creates lifelike AI voices that sound human instead of robotic. • Professional Quality: Provides high fidelity audio suitable for business and creative projects. • Voice Synthesis: Uses advanced technology to generate realistic and expressive speech. • Content Creation: Helps you easily produce audio for videos, presentations, and more.
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    SpeechText.AI

    SpeechText.AI

    SpeechText.AI

    Transcribe audio and video into text. Get accurate transcriptions of podcasts with domain-specific speech recognition. SpeechText.AI is a powerful artificial intelligence software for speech to text conversion and audio transcription. Upload audio or video files. AI transcription software supports various file formats and transcribes from speech to text in any language. Select domain. Select industry domain and audio type from predefined categories to improve the recognition accuracy of domain-specific words. Transcribe. Our speech transcription engine uses state-of-the-art deep neural network models to convert from audio to text with close to human accuracy. Edit & Export. Search, modify and verify audio transcriptions using interactive editing tools. Export your content in different formats. Why SpeechText.AI? Set of amazing features to help you transcribe audio and video in seconds. Speech recognition. Powerful speech-to-text tech.
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    Voiser

    Voiser

    Voiser

    Voiser is an innovative AI-powered voice technology tool that revolutionizes the way we interact with audio content. With its seamless text-to-speech feature, Voiser effortlessly converts written text into natural and expressive speech, offering a wide range of possibilities with its 550 voice options in 75 languages. This enables businesses and individuals to create captivating voiceovers, engaging podcasts, and interactive virtual assistants that resonate with global audiences. On the other hand, Voiser's speech-to-text capability provides an accurate transcription of spoken words, including audio and video transcription, streamlining workflows and enhancing productivity. Additionally, Voiser offers a talking avatar feature, adding a visual and interactive element to content, and the ability to create personalized experiences through voice cloning. With Voiser, language barriers are broken, time is saved, and exceptional audio experiences are crafted to make a lasting impact.
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    Azure AI Speech
    Build voice-enabled apps confidently and quickly with the Speech SDK. Transcribe speech to text with high accuracy, produce natural-sounding text-to-speech voices, translate spoken audio, and use speaker recognition during conversations. Create custom models tailored to your app with Speech studio. Get state-of-the-art speech to text, lifelike text to speech, and award-winning speaker recognition. Your data stays yours, your speech input is not logged during processing. Create custom voices, add specific words to your base vocabulary, or build your own models. Run Speech anywhere, in the cloud or at the edge in containers. Quickly and accurately transcribe audio in more than 92 languages and variants. Gain customer insights with call center transcription, improve experiences with voice-enabled assistants, capture key discussions in meetings and more. Use text to speech to create apps and services that speak conversationally, choosing from more than 215 voices, and 60 languages.
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    Gradium

    Gradium

    Gradium

    Gradium is a voice AI platform for developers and organizations building speech-enabled applications and conversational agents. Its hosted APIs provide streaming text-to-speech, speech-to-text, speech translation, and speech-to-speech, with WebSocket and REST interfaces, real-time turn detection, pronunciation controls, and voice design or cloning features. A browser-based studio lets users try the models and manage voices, while documentation and SDK guides support application development. Use cases include voice agents, customer-service automation, live transcription, multilingual voice experiences, and spoken content generation. Gradium also publishes Phonon, an open-weight text-to-speech model designed for local, offline use in applications and on-device systems. Plans use credits for speech generation, transcription, and translation.
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    AVS Audio Editor
    Record audio data from various inputs like microphone, vinyl records, and other input lines on a sound card. Extract and edit audio from your video files. Remove noise and irritating sounds like roaring, hissing, crackling, etc. Turn written text into a natural sounding voice with Text-to-speech function. Select between 20 built-in effects and filters including delay, flanger, chorus, reverb, reverse, echo and more. Mix audio and blend several audio tracks together. Edit all popular formats MP3, FLAC, WAV, M4A, WMA, AAC, MP2, AMR, OGG, etc.
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    AnyToSpeech

    AnyToSpeech

    AnyToSpeech

    AnyToSpeech is a text-to-speech online platform built to convert any text into audio instantly, creating audiobooks, MP3 files, podcasts, and voiceovers effortlessly. It turns plain text, documents, PDFs, DOCX, TXT files, webpages, PowerPoint presentations, images, and more into natural-sounding audio with multiple AI voices, accents, tones, and vibes. Users can quickly turn any text into a human-like voice through a simple interface, choose from hundreds of different voice and vibe combinations, and download the result as an MP3 file or listen directly in the browser. AnyToSpeech also includes PDF to MP3 for transforming documents, books, and research papers into audio content; URL to Speech for listening to articles and blogs on the go; Image to Speech for extracting text from signs, documents, screenshots, and images; and Image Translation for extracting text from images, translating it to 30+ languages, and converting the translation to speech.
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    Gemini 3.5 Live Translate
    Gemini 3.5 Live Translate is Google’s latest audio model for live speech-to-speech translation, delivering near real-time translation in more than 70 languages. The model automatically detects multilingual input and generates smooth, natural-sounding translated speech that preserves the speaker’s intonation, pacing, and pitch. Unlike turn-by-turn translation systems that wait for someone to finish speaking before responding, Gemini 3.5 Live Translate processes speech as it streams and generates translated audio continuously, balancing the need for context with the need to stay in sync. It stays only a few seconds behind the speaker throughout a session, helping conversations feel more fluid and natural, without awkward pauses. It is built for multilingual calls, meetings, lessons, broadcasts, live interpretation, dubbing, simultaneous translation, and voice translation applications.
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    Bland Speech v3
    Bland Speech v3 is a text-to-speech model designed to produce voice audio that sounds like a real person, especially for phone-call experiences where overly polished delivery can feel artificial. It preserves human details such as breaths, stumbles, pauses, laughs, and throat-clearing through performance tags that are performed instead of read aloud. Users can provide their own script or use Director to describe a dialogue and have Bland write the words, timing, and delivery before generating speech. It supports voice cloning: an instant clone can be created from roughly 10 seconds of audio, while professional cloning uses 30 minutes or more of verified audio, with users confirming that every voice is theirs to use. Bland Speech is available through a web studio and a single /v1/speak API endpoint using bearer-key authentication. Audio streams through HTTP chunked transfer or WebSocket and returns PCM16 WAV at 44.1 kHz.
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    GPT‑Realtime‑Whisper
    GPT-Realtime-Whisper is OpenAI’s streaming transcription model built for low-latency speech-to-text experiences in live products. It transcribes audio as people speak, helping voice-enabled apps feel faster, more responsive, and more natural, from captions that appear in the moment to meeting notes that keep up with the conversation. It makes live speech usable inside business workflows as it happens, so teams can power captions for meetings, classrooms, broadcasts, and events, generate notes and summaries while conversations are still in progress, build voice agents that need to understand users continuously, and create faster follow-up workflows for high-volume spoken interactions. It is part of a new generation of real-time voice models in the API that can reason, translate, and transcribe as people speak, moving real-time audio beyond simple call-and-response toward voice interfaces that can listen, translate, transcribe, and take action as a conversation unfolds.
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    Blogcast

    Blogcast

    Blogcast

    Generate clear, natural-sounding speech from your blog posts and content for podcasts, videos, and more using text-to-speech technology. No microphone is required! Blogcast generates audio from any text-based content. Create a podcast, download the raw audio files or use a simple embed on your site. Enhance WordPress posts, Medium articles, and website content with audio to expand your reach. Quickly create voice-over tracks for YouTube videos without hiring expensive talent. Generate podcast episodes as new articles are posted. Explain concepts and provide audio for courses and online training. Add audio to product explainers, demos, and support materials. Publish audio chapters from existing book content. Convert your articles into clear, natural-sounding audio using AI-powered text-to-speech technology. Add articles from a URL or RSS feed and automatically fetch and convert new articles as they are published.
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    HaloVoice

    HaloVoice

    Halo AI Labs

    HaloVoice is a real-time speech-to-speech AI tool that translates your voice instantly for streaming, gaming, and online meetings. It works seamlessly with platforms like OBS, Discord, Zoom, Slack, Teams, and more—offering multiple voices and personas, plus voice cloning, with low latency and high audio quality.