Alternatives to Orpheus TTS
Compare Orpheus TTS alternatives for your business or organization using the curated list below. SourceForge ranks the best alternatives to Orpheus TTS in 2026. Compare features, ratings, user reviews, pricing, and more from Orpheus TTS competitors and alternatives in order to make an informed decision for your business.
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GPT-Live
OpenAI
GPT-Live is a new generation of voice models for natural human-AI interaction, now powering ChatGPT Voice. It is built to make talking with AI feel much more like having a real conversation through a full-duplex architecture, meaning it can listen and speak at the same time. During conversations, GPT-Live can show it is paying attention with short acknowledgments like “mhmm” or “yeah,” engage in quick back-and-forth, or stay quiet when the user needs a moment to think. Instead of processing separate turns one after another, GPT-Live continuously processes input while generating output, allowing it to decide many times per second whether to speak, keep listening, pause, interrupt, or invoke a tool. For questions that require web search, deeper reasoning, or more complex work, GPT-Live can delegate to a frontier model behind the scenes and bring the result back into the conversation when it is ready, while still maintaining the flow of the voice interaction. -
2
GPT-Live-1
OpenAI
GPT-Live-1 is one of the two new GPT-Live voice models rolling out to ChatGPT users globally, built to make talking with AI feel much more like having a real conversation. It is powered by a full-duplex architecture, so it can listen and speak at the same time instead of waiting for one rigid turn to end before the next begins. During conversations, GPT-Live-1 can show it is paying attention with short acknowledgments, engage in quick back-and-forth, pause when the user needs a moment to think, or stay quiet when asked to listen. It continuously processes input while generating output, allowing the model to decide many times per second whether to speak, keep listening, pause, interrupt, or invoke a tool. GPT-Live-1 also separates natural interaction from deeper work: when a question requires web search, reasoning, or more agentic capabilities, it can delegate the task to a frontier model behind the scenes and bring the result back when it is ready. -
3
GPT-Live-1 mini
OpenAI
GPT-Live-1 mini is one of the two GPT-Live voice models rolling out to ChatGPT users globally, designed to bring more natural, intelligent, and responsive voice interaction to everyday conversations. Built with the same full-duplex approach as GPT-Live, it can listen and speak at the same time instead of waiting for rigid turn-by-turn exchanges. The model continuously processes input while generating output, allowing it to decide many times per second whether to speak, keep listening, pause, interrupt, or invoke a tool. This makes conversations feel faster, smoother, and more natural, with active listening, quick back-and-forth, better timing, and fewer awkward interruptions when the user pauses to think. GPT-Live-1 mini also benefits from the new ChatGPT Voice experience, where users can interrupt with a question, ask ChatGPT to slow down, or tell it to stay quiet and listen. -
4
MARS6
CAMB.AI
CAMB.AI's MARS6 is a groundbreaking text-to-speech (TTS) model that has become the first speech model accessible on Amazon Web Services (AWS) Bedrock platform. This integration allows developers to incorporate advanced TTS capabilities into generative AI applications, facilitating the creation of enhanced voice assistants, engaging audiobooks, interactive media, and various audio-centric experiences. MARS6's advanced algorithms enable natural and expressive speech synthesis, setting a new standard for TTS conversion. Developers can access MARS6 directly through the Amazon Bedrock platform, ensuring seamless integration into applications and enhancing user engagement and accessibility. The inclusion of MARS6 in AWS Bedrock's diverse selection of foundation models underscores CAMB.AI's commitment to advancing machine learning and artificial intelligence, providing developers with vital tools to create rich audio experiences supported by AWS's reliable and scalable infrastructure. -
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Piper TTS
Rhasspy
Piper is a fast, local neural text-to-speech (TTS) system optimized for devices like the Raspberry Pi 4, designed to deliver high-quality speech synthesis without relying on cloud services. It utilizes neural network models trained with VITS and exported to ONNX Runtime, enabling efficient and natural-sounding speech generation. Piper supports a wide range of languages, including English (US and UK), Spanish (Spain and Mexico), French, German, and many others, with voices available for download. Users can run Piper via the command line or integrate it into Python applications using the piper-tts package. The system allows for real-time audio streaming, JSON input for batch processing, and supports multi-speaker models. Piper relies on espeak-ng for phoneme generation, converting text into phonemes before synthesizing speech. It is employed in various projects such as Home Assistant, Rhasspy 3, NVDA, and others.Starting Price: Free -
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Voxtral TTS
Mistral AI
Voxtral TTS is a state-of-the-art, multilingual text-to-speech model designed to generate highly realistic and emotionally expressive speech from text, combining strong contextual understanding with advanced speaker modeling to produce natural, human-like audio output. Built as a lightweight model with around 4 billion parameters, it delivers efficient performance while maintaining high quality, enabling scalable deployment for enterprise voice applications. It supports nine major languages and diverse dialects, and can adapt to new voices using only a short reference audio sample, capturing not just tone but also rhythm, pauses, intonation, and emotional nuance. Its zero-shot voice cloning capabilities allow it to replicate a speaker’s style without additional training, and it can even perform cross-lingual voice adaptation, generating speech in one language while preserving the accent of another. -
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Inworld TTS
Inworld
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.Starting Price: $0.005 per minute -
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Octave TTS
Hume AI
Hume AI has introduced Octave (Omni-capable Text and Voice Engine), a groundbreaking text-to-speech system that leverages large language model technology to understand and interpret the context of words, enabling it to generate speech with appropriate emotions, rhythm, and cadence, unlike traditional TTS models that merely read text, Octave acts akin to a human actor, delivering lines with nuanced expression based on the content. Users can create diverse AI voices by providing descriptive prompts, such as "a sarcastic medieval peasant," allowing for tailored voice generation that aligns with specific character traits or scenarios. Additionally, Octave offers the flexibility to modify the emotional delivery and speaking style through natural language instructions, enabling commands like "sound more enthusiastic" or "whisper fearfully" to fine-tune the output.Starting Price: $3 per month -
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MAI-Voice-2-Flash
Microsoft
MAI-Voice-2-Flash is Microsoft AI’s fast, efficient text-to-speech model for high-volume voice experiences where responsiveness is essential. It produces high-fidelity, natural, and expressive speech while preserving the prosody, acoustic quality, human-like rhythm, intonation, and emotional nuance of MAI-Voice-2. The model is optimized for real-time synthesis and runs twice as fast as MAI-Voice-2, making it suitable for voice agents, assistants, interactive applications, call centers, and IVR systems that must respond without noticeable delay. It supports 15 languages across 18 locales and includes a library of licensed, curated voices that can be used immediately. Developers can control speaking style and emotion through SSML, shaping delivery with expressions such as joy, excitement, empathy, sadness, whispering, or shouting to match different conversational situations and brand experiences. -
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ElevenLabs
ElevenLabs
The most realistic and versatile AI speech software, ever. Eleven brings the most compelling, rich and lifelike voices to creators and publishers seeking the ultimate tools for storytelling. Generate top-quality spoken audio in any voice and style with the most advanced and multipurpose AI speech tool out there. Our deep learning model renders human intonation and inflections with unprecedented fidelity and adjusts delivery based on context. Our AI model is built to grasp the logic and emotions behind words. And rather than generate sentences one-by-one, it’s always mindful of how each utterance ties to preceding and succeeding text. This zoomed-out perspective allows it to intonate longer fragments convincingly and with purpose. And finally you can do this with any voice you want.Starting Price: $1 per month -
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EVI 3
Hume AI
Hume AI's EVI 3 is a third-generation speech-language model that streams in user speech and forms natural, expressive speech and language responses. At conversational latency, it produces the same quality of speech as our text-to-speech model, Octave. Simultaneously, it responds with the same intelligence as the most advanced LLMs of similar latency. It also communicates with reasoning models and web search systems as it speaks, “thinking fast and slow” to match the intelligence of any frontier AI system. EVI 3 can instantly generate new voices and personalities instead of being limited to a handful of speakers. For instance, users can speak to any of the more than 100,000 custom voices already created on our text-to-speech platform, each with an inferred personality. No matter the voice, it responds with a wide range of emotions or styles, implicitly or on command.Starting Price: Free -
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Chatterbox
Resemble AI
Chatterbox is a free, open source voice cloning AI model developed by Resemble AI, licensed under MIT. It enables zero-shot voice cloning using just 5 seconds of reference audio, eliminating the need for training. The model offers expressive speech synthesis with unique emotion control, allowing users to adjust the intensity from monotone to dramatically expressive with a single parameter. Chatterbox supports accent control and text-based controllability, ensuring high-quality, human-like text-to-speech conversion. It operates with faster-than-real-time inference, making it suitable for real-time applications, voice assistants, and interactive media. The model is built for production and designed for developers, featuring simple installation via pip and comprehensive documentation. Chatterbox includes built-in watermarking using Resemble AI’s PerTh (Perceptual Threshold) Watermarker, embedding data imperceptibly to protect generated audio content.Starting Price: $5 per month -
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Cartesia Sonic-3
Cartesia
Cartesia Sonic-3 is a real-time, streaming text-to-speech (TTS) model designed to generate ultra-realistic, expressive voice output with extremely low latency, enabling AI systems to speak as fluidly as humans in live interactions. Built on advanced state space model architecture, Sonic delivers high-quality speech while achieving near-instant response times, with audio generation beginning in as little as 40–100 milliseconds, making conversations feel seamless rather than delayed. It is optimized for conversational AI use cases, acting as the “voice layer” for AI agents by converting text into natural-sounding speech that includes emotional nuance such as excitement, empathy, or even laughter. It supports more than 40 languages with native-level voices and accent localization, allowing developers to build globally accessible applications with consistent quality across regions.Starting Price: $4 per month -
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Qwen3-TTS
Alibaba
Qwen3-TTS is an open source series of advanced text-to-speech models developed by the Qwen team at Alibaba Cloud under the Apache-2.0 license, offering stable, expressive, and real-time speech generation with features such as voice cloning, voice design, and fine-grained control of prosody and acoustic attributes. The models support 10 major languages, including Chinese, English, Japanese, Korean, German, French, Russian, Portuguese, Spanish, and Italian, and multiple dialectal voice profiles with adaptive control over tone, speaking rate, and emotional expression based on text semantics and instructions. Qwen3-TTS uses efficient tokenization and a dual-track architecture that enables ultra-low-latency streaming synthesis (first audio packet in ~97 ms), making it suitable for interactive and real-time use cases, and includes a range of models with different capabilities (e.g., rapid 3-second voice cloning, custom voice timbres, and instruction-based voice design).Starting Price: Free -
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Miso TTS
Miso TTS
Miso Labs builds emotive foundation models for voice, designed to help developers create voice agents that feel fast, warm, and human instead of robotic or delayed. Its flagship model, Miso TTS, is an 8-billion-parameter transformer model for state-of-the-art emotive speech and dialogue generation, with open source weights available on Hugging Face and API access coming soon. Miso is built for real-time conversational voice, responding in 110ms to preserve natural flow and avoid the awkward pauses common in AI voice agents. It supports one-shot voice cloning, allowing users to clone a voice from a ten-second audio clip while keeping the agent’s voice consistent from the first second of a call to the last. Miso Labs also emphasizes local and sovereign deployment, with open source models built for local use and on-premises hosting and support available for enterprise teams that need to keep sensitive data in-house. -
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Gemini 2.5 Pro TTS
Google
Gemini 2.5 Pro TTS is Google’s advanced text-to-speech model in the Gemini 2.5 family, optimized for high-quality, expressive, controllable speech synthesis for structured and professional audio generation tasks. The model delivers natural-sounding voice output with enhanced expressivity, tone control, pacing, and pronunciation fidelity, enabling developers to dictate style, accent, rhythm, and emotional nuance through text-based prompts, making it suitable for applications like podcasts, audiobooks, customer assistance, tutorials, and multimedia narration that require premium audio output. It supports both single-speaker and multi-speaker audio, allowing distinct voices and conversational flows in the same output, and can synthesize speech across multiple languages with consistent style adherence. Compared with lower-latency variants like Flash TTS, the Pro TTS model prioritizes sound quality, depth of expression, and nuanced control. -
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Simba 3.2
Speechify
Speechify’s text-to-speech API offers a family of Simba models for real-time voice generation across English, European languages, and broader multilingual use cases. Simba 3.2 is recommended for new English integrations, providing streaming-native synthesis, the lowest time to first byte, richer expressivity than earlier generations, and full support for SSML and emotion control. Simba 3.0 extends streaming-native speech to English, German, Spanish, French, Italian, and Brazilian Portuguese, with language selection handled through the request or voice locale. Simba Multilingual supports 35 locales across 30 languages, including mixed-language content and automatic language detection, while Simba English remains available as a legacy model for compatibility. Developers select a model through one parameter and can switch without changing the rest of the request structure, including voice, format, and SSML settings. -
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Google Cloud Text-to-Speech
Google
Convert text into natural-sounding speech using an API powered by Google’s AI technologies. Deploy Google’s groundbreaking technologies to generate speech with humanlike intonation. Built based on DeepMind’s speech synthesis expertise, the API delivers voices that are near human quality. Choose from a set of 220+ voices across 40+ languages and variants, including Mandarin, Hindi, Spanish, Arabic, Russian, and more. Pick the voice that works best for your user and application. Create a unique voice to represent your brand across all your customer touchpoints, instead of using a common voice shared with other organizations. Train a custom voice model using your own audio recordings to create a unique and more natural sounding voice for your organization. You can define and choose the voice profile that suits your organization and quickly adjust to changes in voice needs without needing to record new phrases. -
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Cartesia Sonic-3.5
Cartesia
Sonic 3.5 is Cartesia’s fastest, most natural text-to-speech model, built for expressive, real-time voice generation with sub-90ms latency and native support for 42 languages. It is designed to follow transcripts faithfully, voice confirmation codes, and heteronyms correctly without preprocessing, and stay expressive enough to carry a real conversation. It supports languages intended to deliver native-quality speech. Sonic 3.5 focuses on clean audio across every language and voice, with no artifacts to edit out, making it practical for production voice experiences where quality, speed, and consistency matter. Its expressive conversational delivery provides strong pacing and real emotional range, tuned for support and agent transcripts. Alphanumerics such as order numbers, phone numbers, IDs, and emails are spoken naturally in every language, while context-aware English pronunciation helps words like read, bass, and bow land correctly from the surrounding text. -
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Gemini 2.5 Flash TTS
Google
Gemini 2.5 Flash TTS is the latest text-to-speech (TTS) model variant in Google’s Gemini 2.5 lineup, designed for faster, low-latency speech synthesis with expressive, controllable audio output. It offers significant enhancements in tone versatility and expressivity so that developers can generate speech that better matches style prompts, from storytelling narrations to character voices, with more natural emotional range. It features precision pacing, which allows it to adjust speech tempo based on context, delivering faster sections or slowing for emphasis more accurately according to instructions. It also supports multi-speaker dialogues with consistent character voices for scenarios like podcasts, interviews, or conversational agents, and improved multilingual handling so each speaker’s unique tone and style persist across languages. Gemini 2.5 Flash TTS is optimized for lower latency, making it ideal for interactive applications and real-time voice interfaces. -
21
Qwen-Audio-3.0-TTS-Flash
Alibaba
Qwen-Audio-3.0-TTS-Flash is the real-time variant of Qwen-Audio-3.0-TTS, tuned for interactive applications with first-packet latency at the 300 ms level. It supports 16 languages, along with improved fidelity for several Chinese dialects. Across multilingual evaluations, Flash delivers the lowest average WER/CER in the family at 3.87, showing strong intelligibility while preserving speaker identity across diverse languages. Developers can guide delivery with plain-language instructions instead of manually adjusting acoustic parameters, controlling emotion, role, scenario, pace, projection, and tone through simple prompts. Inline tags add precise non-verbal details, making the model well-suited to conversational agents, narration, games, dubbing, and other expressive speech experiences. Voice cloning is designed to work with imperfect reference audio; targeted acoustic simulation suppresses noise and reverberation while retaining the original speaker’s timbre. -
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MiniMax Speech 2.8
MiniMax
MiniMax Speech 2.8 is a next-generation AI speech model built to make synthetic voice feel alive, expressive, and deeply human. It focuses on performance in real-world voice agent scenarios, combining ultra-fast response, richer emotional expression, cleaner audio, and stronger cross-lingual performance for products that need natural spoken interaction. Speech 2.8 is designed to reduce the distance between AI voice and real human communication, giving developers and creators more control over how a voice sounds, reacts, and carries meaning. It supports flexible emotion control, allowing users to shape delivery with moods, tone, and expressive direction instead of relying on flat or robotic speech. It can produce speech with more natural pauses, cadence, emphasis, and emotional texture, helping AI characters, assistants, narrators, and interactive agents sound more believable across longer conversations. -
23
Qwen-Audio-3.0-TTS-Plus
Alibaba
Qwen-Audio-3.0-TTS-Plus is the high-quality variant of Qwen-Audio-3.0-TTS, optimized for naturalness and timbre fidelity when output quality matters more than speed. It supports 16 languages, plus improved fidelity for several Chinese dialects. The model delivers strong multilingual intelligibility and ranks first in speaker similarity across all supported languages, helping cloned voices remain recognizable and consistent across linguistic contexts. Developers can direct delivery through ordinary natural-language instructions instead of manually tuning acoustic parameters, controlling emotion, role, scenario, pacing, projection, and tone with simple prompts. Inline tags provide fine-grained control over breaths, laughter, emotional shifts, and other non-verbal details, making the model useful for narration, games, character dialogue, and dubbing. -
24
Llama 2
Meta
The next generation of our open source large language model. This release includes model weights and starting code for pretrained and fine-tuned Llama language models — ranging from 7B to 70B parameters. Llama 2 pretrained models are trained on 2 trillion tokens, and have double the context length than Llama 1. Its fine-tuned models have been trained on over 1 million human annotations. Llama 2 outperforms other open source language models on many external benchmarks, including reasoning, coding, proficiency, and knowledge tests. Llama 2 was pretrained on publicly available online data sources. The fine-tuned model, Llama-2-chat, leverages publicly available instruction datasets and over 1 million human annotations. We have a broad range of supporters around the world who believe in our open approach to today’s AI — companies that have given early feedback and are excited to build with Llama 2.Starting Price: Free -
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Azure AI Speech
Microsoft
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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Realtime TTS-2
Inworld
Realtime TTS-2 from Inworld AI is a new generation of voice model built for real-time conversation: a voice model that feels as human as it sounds. It hears the full audio of an exchange, picks up the user’s tone, pacing, and emotional state, then takes voice direction in plain English, the way developers prompt an LLM. Instead of generating speech in isolation, it listens to prior turns of the exchange, so tone and pacing carry forward, and the same line can land differently after a joke than after bad news. Voice Direction lets developers steer delivery like a director would steer a voice actor, using natural-language descriptions rather than fixed emotion presets or sliders. Inline nonverbals like [sigh], [breathe], and [laugh] can be placed inside the text, and the model renders them as audio events. Realtime TTS-2 preserves one voice identity across more than 100 languages, including mid-utterance language switches.Starting Price: $25 per month -
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Phi-4-reasoning
Microsoft
Phi-4-reasoning is a 14-billion parameter transformer-based language model optimized for complex reasoning tasks, including math, coding, algorithmic problem solving, and planning. Trained via supervised fine-tuning of Phi-4 on carefully curated "teachable" prompts and reasoning demonstrations generated using o3-mini, it generates detailed reasoning chains that effectively leverage inference-time compute. Phi-4-reasoning incorporates outcome-based reinforcement learning to produce longer reasoning traces. It outperforms significantly larger open-weight models such as DeepSeek-R1-Distill-Llama-70B and approaches the performance levels of the full DeepSeek-R1 model across a wide range of reasoning tasks. Phi-4-reasoning is designed for environments with constrained computing or latency. Fine-tuned with synthetic data generated by DeepSeek-R1, it provides high-quality, step-by-step problem solving. -
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Grok Text to Speech (TTS)
SpaceXAI
Grok Text to Speech (TTS) is a standalone audio API built to help developers generate fast, natural, and expressive speech from text. Built on the same stack that powers Grok Voice, Tesla vehicles, and Starlink customer support, the API makes it straightforward to integrate high-quality voice generation into applications such as voice agents, accessibility tools, podcasts, assistants, customer experiences, and interactive audio products. Grok TTS can turn long-form text into speech through a REST API or generate speech in real time through a WebSocket API, giving developers flexibility for both batch audio generation and live conversational experiences. It is designed around expressive delivery, not just flat narration, with fine-grained control through simple inline and wrapping speech tags. Developers can add natural prosody and emotion using tags, allowing lifelike delivery without complex markup. -
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MAI-Voice-2
Microsoft AI
MAI-Voice-2 is Microsoft AI’s most expressive and natural-sounding text-to-speech model to date, built for production voice experiences where fidelity, language coverage, speaker consistency, and emotional range directly shape the user experience. It is designed for assistants, customer support, audiobooks, accessibility experiences, games, podcasts, courses, simulations, and creator workflows where voice quality must sound natural, fluid, and trustworthy. It expands from English-only support to 15 languages while maintaining naturalness and expressiveness, with support for English, Italian, French, German, Hindi, Spanish, Portuguese, Korean, Chinese, Turkish, Russian, Thai, Dutch, Romanian, and Hungarian. MAI-Voice-2 offers granular emotion control through tags such as sad, whispered, and excited, along with role-based expressive speech for experiences like motivational trainers, sports commentators, or character voices. -
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Mistral 7B
Mistral AI
Mistral 7B is a 7.3-billion-parameter language model that outperforms larger models like Llama 2 13B across various benchmarks. It employs Grouped-Query Attention (GQA) for faster inference and Sliding Window Attention (SWA) to efficiently handle longer sequences. Released under the Apache 2.0 license, Mistral 7B is accessible for deployment across diverse platforms, including local environments and major cloud services. Additionally, a fine-tuned version, Mistral 7B Instruct, demonstrates enhanced performance in instruction-following tasks, surpassing models like Llama 2 13B Chat.Starting Price: Free -
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Gemini 3.1 Flash TTS
Google
Gemini 3.1 Flash TTS is Google’s latest text-to-speech model designed to deliver highly expressive, controllable, and scalable AI-generated speech for developers and enterprises. Available in Google AI Studio and Gemini Enterprise Agent Platform, it focuses on precise control over how audio is generated, allowing users to shape delivery through natural language prompts and an extensive system of more than 200 audio tags that define pacing, tone, emotion, and style. It supports over 70 languages and regional variants, along with a library of 30 prebuilt voices, enabling users to generate speech ranging from professional narration to conversational or stylized performances. Developers can embed instructions directly into text inputs to guide vocal expression, combining pacing, emotion, and pauses in a structured prompting framework that produces nuanced, high-fidelity audio output. Gemini 3.1 Flash TTS is optimized for real-world applications. -
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Phi-2
Microsoft
We are now releasing Phi-2, a 2.7 billion-parameter language model that demonstrates outstanding reasoning and language understanding capabilities, showcasing state-of-the-art performance among base language models with less than 13 billion parameters. On complex benchmarks Phi-2 matches or outperforms models up to 25x larger, thanks to new innovations in model scaling and training data curation. With its compact size, Phi-2 is an ideal playground for researchers, including for exploration around mechanistic interpretability, safety improvements, or fine-tuning experimentation on a variety of tasks. We have made Phi-2 available in the Azure AI Studio model catalog to foster research and development on language models. -
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Phi-4-mini-reasoning
Microsoft
Phi-4-mini-reasoning is a 3.8-billion parameter transformer-based language model optimized for mathematical reasoning and step-by-step problem solving in environments with constrained computing or latency. Fine-tuned with synthetic data generated by the DeepSeek-R1 model, it balances efficiency with advanced reasoning ability. Trained on over one million diverse math problems spanning multiple levels of difficulty from middle school to Ph.D. level, Phi-4-mini-reasoning outperforms its base model on long sentence generation across various evaluations and surpasses larger models like OpenThinker-7B, Llama-3.2-3B-instruct, and DeepSeek-R1. It features a 128K-token context window and supports function calling, enabling integration with external tools and APIs. Phi-4-mini-reasoning can be quantized using Microsoft Olive or Apple MLX Framework for deployment on edge devices such as IoT, laptops, and mobile devices. -
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Cartesia Sonic
Cartesia
Sonic is the fastest, ultra-realistic generative voice API, powered by our next-gen state space model and purpose-built for developers. With a time-to-first audio of 90ms, Sonic is the fastest generative voice model, with best-in-class quality and controllability. Built for streaming using our first-of-its-kind low-latency state space model stack. Fine-grained control over pitch, speed, emotion, and pronunciation. Sonic ranks #1 in quality in independent evaluations of quality. Sonic supports seamless speech in 13 languages, with more added to every release. From Japanese to German, any language you need, we’ve got it. Localize a given voice to any accent or language. Power support experiences that delight your customers. Bring your storytelling to life with immersive voices. Create content that engages viewers and drives clicks. Narrate content for podcasts, news, and publishing, and empower healthcare with voices that patients trust.Starting Price: $5 per month -
35
Azure Text to Speech
Microsoft
Build apps and services that speak naturally. Differentiate your brand with a customized, realistic voice generator, and access voices with different speaking styles and emotional tones to fit your use case—from text readers and talkers to customer support chatbots. Enable fluid, natural-sounding text to speech that matches the intonation and emotion of human voices. Tune voice output for your scenarios by easily adjusting rate, pitch, pronunciation, pauses, and more. Engage global audiences by using 400 neural voices across 140 languages and variants. Bring your scenarios like text readers and voice-enabled assistants to life with highly expressive and human-like voices. Neural Text to Speech supports several speaking styles including newscast, customer service, shouting, whispering, and emotions like cheerful and sad. -
36
Google has released updated Gemini audio models that significantly expand the platform’s capabilities for natural, expressive voice interactions and real-time conversational AI with the introduction of Gemini 2.5 Flash Native Audio and improved text-to-speech technology. The updated native audio model powers live voice agents that can handle complex workflows, follow detailed user instructions more reliably, and maintain smoother multi-turn conversations by better recalling context from previous turns. It is now available across Google AI Studio,Gemini Enterprise Agent Platform, Gemini Live, and Search Live, enabling developers and products to build interactive voice experiences such as intelligent assistants and enterprise voice agents. In addition to the real-time voice improvements, Google enhanced the underlying Text-to-Speech (TTS) models in the Gemini 2.5 family to offer greater expressivity, tone control, pacing adjustments, and multilingual support.
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Llama
Meta
Llama (Large Language Model Meta AI) is a state-of-the-art foundational large language model designed to help researchers advance their work in this subfield of AI. Smaller, more performant models such as Llama enable others in the research community who don’t have access to large amounts of infrastructure to study these models, further democratizing access in this important, fast-changing field. Training smaller foundation models like Llama is desirable in the large language model space because it requires far less computing power and resources to test new approaches, validate others’ work, and explore new use cases. Foundation models train on a large set of unlabeled data, which makes them ideal for fine-tuning for a variety of tasks. We are making Llama available at several sizes (7B, 13B, 33B, and 65B parameters) and also sharing a Llama model card that details how we built the model in keeping with our approach to Responsible AI practices. -
38
Phi-4-mini-flash-reasoning
Microsoft
Phi-4-mini-flash-reasoning is a 3.8 billion‑parameter open model in Microsoft’s Phi family, purpose‑built for edge, mobile, and other resource‑constrained environments where compute, memory, and latency are tightly limited. It introduces the SambaY decoder‑hybrid‑decoder architecture with Gated Memory Units (GMUs) interleaved alongside Mamba state‑space and sliding‑window attention layers, delivering up to 10× higher throughput and a 2–3× reduction in latency compared to its predecessor without sacrificing advanced math and logic reasoning performance. Supporting a 64 K‑token context length and fine‑tuned on high‑quality synthetic data, it excels at long‑context retrieval, reasoning tasks, and real‑time inference, all deployable on a single GPU. Phi-4-mini-flash-reasoning is available today via Azure AI Foundry, NVIDIA API Catalog, and Hugging Face, enabling developers to build fast, scalable, logic‑intensive applications. -
39
Zyphra Zonos
Zyphra
Zyphra is excited to announce the release of Zonos-v0.1 beta, featuring two expressive and real-time text-to-speech models with high-fidelity voice cloning. We are releasing our 1.6B transformer and 1.6B hybrid under an Apache 2.0 license. It is difficult to quantitatively measure quality in the audio domain; we find that Zonos’ generation quality matches or exceeds that of leading proprietary TTS model providers. Further, we believe that openly releasing models of this caliber will significantly advance TTS research. Zonos model weights are available on Huggingface, and sample inference code for the models is available on our GitHub. You can also access Zonos through our model playground and API with simple and competitive flat-rate pricing. We have found that quantitative evaluations struggle to measure the quality of outputs in the audio domain, so for demonstration, we present a number of samples of Zonos vs both proprietary models.Starting Price: $0.02 per minute -
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MiniMax Audio
MiniMax
MiniMax Audio is an AI-driven audio generation platform that transforms text into realistic speech across 50+ languages, offering over 300 expressive voices, including regional accents like American, Cantonese, Dutch, German, Czech, Japanese, and more, while supporting advanced features such as emotion adjustment, speed, pitch customization, and noise isolation to clean up audio tracks. Users can quickly generate lifelike audio samples via long-text mode, URL input, or voice cloning, capturing a unique voice in as little as 10 seconds, without needing transcription. The underlying technology incorporates cutting-edge AI such as transformer-based TTS models, a learnable speaker encoder, and Flow-VAE architectures, enabling zero- or one-shot voice cloning with high fidelity and expressive control, and it ranks at the top of public voice cloning benchmarks.Starting Price: Free -
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Oumi
Oumi
Oumi is a fully open source platform that streamlines the entire lifecycle of foundation models, from data preparation and training to evaluation and deployment. It supports training and fine-tuning models ranging from 10 million to 405 billion parameters using state-of-the-art techniques such as SFT, LoRA, QLoRA, and DPO. The platform accommodates both text and multimodal models, including architectures like Llama, DeepSeek, Qwen, and Phi. Oumi offers tools for data synthesis and curation, enabling users to generate and manage training datasets effectively. For deployment, it integrates with popular inference engines like vLLM and SGLang, ensuring efficient model serving. The platform also provides comprehensive evaluation capabilities across standard benchmarks to assess model performance. Designed for flexibility, Oumi can run on various environments, from local laptops to cloud infrastructures such as AWS, Azure, GCP, and Lambda.Starting Price: Free -
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Replica
Replica
Replica Studios provides cutting edge text to speech, and speech to speech solutions in multiple languages for creative professionals, with fully licensed AI models safe for commercial use. Replica Studios offers two products: Replica Voice Director: Generate voice overs and dialogue instantly with text to speech OR speech to speech, while also managing the scripts for your project where it’s all tracked in one place. Access thousands of unique, natural-sounding, expressive AI voices tailored for specific projects or brands, such as content creators, audiobooks, corporate videos, educational content, games, and open-world games. Replica Voice Lab: Design unique human quality AI voices that can perform in multiple languages in seconds with Replica Studios Voice Lab. Blend up to 5 voice personas to create unique voices, with unique and interesting styles and accents. Multi Language Support: Localize and dub your content using our multi-lingual generative AI voice generator.Starting Price: $10 per month -
43
Chirp 3
Google
Google Cloud's Text-to-Speech API introduces Chirp 3, enabling users to create personalized voice models using their own high-quality audio recordings. This feature facilitates the rapid generation of custom voices, which can be utilized to synthesize audio through the Cloud Text-to-Speech API, supporting both streaming and long-form text. Access to this voice cloning capability is restricted to allow-listed users due to safety considerations; interested parties should contact the sales team to be added to the allowed list. Instant Custom Voice creation and synthesis are supported in various languages, including English (US), Spanish (US), and French (Canada), among others. It is available in multiple Google Cloud regions, and supported output formats include LINEAR16, OGG_OPUS, PCM, ALAW, MULAW, and MP3, depending on the API method used. -
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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.Starting Price: $1 -
45
Athene-V2
Nexusflow
Athene-V2 is Nexusflow's latest 72-billion-parameter model suite, fine-tuned from Qwen 2.5 72B, designed to compete with GPT-4o across key capabilities. This suite includes Athene-V2-Chat-72B, a state-of-the-art chat model that matches GPT-4o in multiple benchmarks, excelling in chat helpfulness (Arena-Hard), code completion (ranking #2 on bigcode-bench-hard), mathematics (MATH), and precise long log extraction. Additionally, Athene-V2-Agent-72B balances chat and agent functionalities, offering concise, directive responses and surpassing GPT-4o in Nexus-V2 function calling benchmarks focused on complex enterprise-level use cases. These advancements underscore the industry's shift from merely scaling model sizes to specialized customization, illustrating how targeted post-training processes can finely optimize models for distinct skills and applications. -
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Entry Point AI
Entry Point AI
Entry Point AI is the modern AI optimization platform for proprietary and open source language models. Manage prompts, fine-tunes, and evals all in one place. When you reach the limits of prompt engineering, it’s time to fine-tune a model, and we make it easy. Fine-tuning is showing a model how to behave, not telling. It works together with prompt engineering and retrieval-augmented generation (RAG) to leverage the full potential of AI models. Fine-tuning can help you to get better quality from your prompts. Think of it like an upgrade to few-shot learning that bakes the examples into the model itself. For simpler tasks, you can train a lighter model to perform at or above the level of a higher-quality model, greatly reducing latency and cost. Train your model not to respond in certain ways to users, for safety, to protect your brand, and to get the formatting right. Cover edge cases and steer model behavior by adding examples to your dataset.Starting Price: $49 per month -
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aiOla
aiOla
aiOla is a deep tech Conversational, Voice, and Speech AI lab with an enterprise-level automatic speech recognition (ASR) foundation model, Text-to-speech (TTS) technology and Natural Language Understanding (NLU). It’s designed to help enterprises and developers adapt speech technologies to any process, whether through seamless API integration or an intuitive in-house app. aiOla is revolutionizing enterprise operations with enterprise level Conversational AI. We specialize in speech-to-text and text-to-speech AI that deliver unmatched accuracy (95%), specialized in specific jargon, in any language, accent, vertical, or acoustic environment. From empowering frontline workers with hands-free workflows to enabling voice AI agents with enterprise-grade ASR and TTS, aiOla seamlessly integrates into workflows, internal apps and products. -
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Code Llama
Meta
Code Llama is a large language model (LLM) that can use text prompts to generate code. Code Llama is state-of-the-art for publicly available LLMs on code tasks, and has the potential to make workflows faster and more efficient for current developers and lower the barrier to entry for people who are learning to code. Code Llama has the potential to be used as a productivity and educational tool to help programmers write more robust, well-documented software. Code Llama is a state-of-the-art LLM capable of generating code, and natural language about code, from both code and natural language prompts. Code Llama is free for research and commercial use. Code Llama is built on top of Llama 2 and is available in three models: Code Llama, the foundational code model; Codel Llama - Python specialized for Python; and Code Llama - Instruct, which is fine-tuned for understanding natural language instructions.Starting Price: Free -
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Gemini 3.5 Live Translate
Google
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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Tune Studio
NimbleBox
Tune Studio is an intuitive and versatile platform designed to streamline the fine-tuning of AI models with minimal effort. It empowers users to customize pre-trained machine learning models to suit their specific needs without requiring extensive technical expertise. With its user-friendly interface, Tune Studio simplifies the process of uploading datasets, configuring parameters, and deploying fine-tuned models efficiently. Whether you're working on NLP, computer vision, or other AI applications, Tune Studio offers robust tools to optimize performance, reduce training time, and accelerate AI development, making it ideal for both beginners and advanced users in the AI space.Starting Price: $10/user/month
