Alternatives to Gemini Live API

Google AI Studio is an all-in-one environment designed for building AI-first applications with Google’s latest models. It supports Gemini, Imagen, Veo, and Gemma, allowing developers to experiment across multiple modalities in one place. The platform emphasizes vibe coding, enabling users to describe what they want and let AI handle the technical heavy lifting. Developers can generate complete, production-ready apps using natural language instructions. One-click deployment makes it easy to move from prototype to live application. Google AI Studio includes a centralized dashboard for API keys, billing, and usage tracking. Detailed logs and rate-limit insights help teams operate efficiently. SDK support for Python, Node.js, and REST APIs ensures flexibility. Quickstart guides reduce onboarding time to minutes. Overall, Google AI Studio blends experimentation, vibe coding, and scalable production into a single workflow.

Dialogflow by Google Cloud is a natural-language understanding platform that allows you to create and integrate a conversational interface into your mobile, web, or device. It also makes it easy for you to integrate a bot, interactive voice response system, or other type of user interface into your app, web, or mobile application. Dialogflow allows you to create new ways for customers to interact with your product. Dialogflow can analyze input from customers in multiple formats, including text and audio (such as voice or phone calls). Dialogflow can also respond to customers via text or synthetic speech. Dialogflow CX, ES offer virtual agent services for chatbots or contact centers. Agent Assist can be used to assist human agents in contact centers that have them. Agent Assist offers real-time suggestions to human agents, even while they are talking with customers.

Gemini is Google’s intelligent AI platform built to support productivity, creativity, and learning across work, school, and everyday life. It allows users to ask questions, generate text, images, and videos, and explore ideas using conversational AI powered by Gemini 3. By integrating directly with Google Search, Gemini provides grounded answers and supports detailed follow-up discussions on complex topics. The platform includes advanced tools like Deep Research, which condenses hours of online research into structured reports in minutes. Gemini also enables real-time collaboration and spoken brainstorming through Gemini Live. Users can connect Gemini to Gmail, Google Docs, Calendar, Maps, and other Google services to complete tasks across multiple apps at once. Custom AI experts called Gems allow users to save instructions and tailor Gemini for specific roles or workflows. Gemini supports large file analysis with a long context window, making it capable of reviewing books, reports, and large codebases. Flexible subscription tiers offer different levels of access to models, credits, and creative tools. Gemini is available on web and mobile, making it accessible wherever users need intelligent assistance.

A compact model that excels in textual understanding and multimodal reasoning capabilities. The GPT-4o mini is designed to handle a wide array of tasks efficiently, thanks to its low cost and minimal latency, making it ideal for applications that require chaining or parallelizing multiple model calls, such as invoking several APIs simultaneously, processing extensive context like entire codebases or conversation histories, and providing swift, real-time text interactions for customer support chatbots. Currently, the API for GPT-4o mini accommodates both text and visual inputs, with plans to introduce support for text, images, videos, and audio in future updates. This model boasts an impressive context window of 128K tokens and can generate up to 16K output tokens per request, while its knowledge base is current as of October 2023. Additionally, the enhanced tokenizer shared with GPT-4o has made it more efficient in processing non-English text, further broadening its usability for diverse applications. As a result, GPT-4o mini stands out as a versatile tool for developers and businesses alike.

Amazon Polly is a service designed to convert written text into realistic speech, enabling the development of applications that can communicate vocally and fostering the creation of innovative speech-enabled products. Utilizing state-of-the-art deep learning technologies, Polly's Text-to-Speech (TTS) service produces natural-sounding human voices. With a variety of lifelike voices available in numerous languages, developers can create speech-enabled applications that are functional in diverse global markets. Beyond the Standard TTS voices, Amazon Polly also provides Neural Text-to-Speech (NTTS) voices, which enhance speech quality significantly through a novel machine learning technique. In addition, Polly's Neural TTS supports two distinct speaking styles: a Newscaster style designed for news narration and a Conversational style that is perfect for interactive communication scenarios such as telephony. This flexibility allows developers to tailor the auditory experience to fit their specific application needs.

GPT-4o, with the "o" denoting "omni," represents a significant advancement in the realm of human-computer interaction by accommodating various input types such as text, audio, images, and video, while also producing outputs across these same formats. Its capability to process audio inputs allows for responses in as little as 232 milliseconds, averaging 320 milliseconds, which closely resembles the response times seen in human conversations. In terms of performance, it maintains the efficiency of GPT-4 Turbo for English text and coding while showing marked enhancements in handling text in other languages, all while operating at a much faster pace and at a cost that is 50% lower via the API. Furthermore, GPT-4o excels in its ability to comprehend vision and audio, surpassing the capabilities of its predecessors, making it a powerful tool for multi-modal interactions. This innovative model not only streamlines communication but also broadens the possibilities for applications in diverse fields.

The gpt-4o-mini-realtime-preview model is a streamlined and economical variant of GPT-4o, specifically crafted for real-time interaction in both speech and text formats with minimal delay. It is capable of processing both audio and text inputs and outputs, facilitating “speech in, speech out” dialogue experiences through a consistent WebSocket or WebRTC connection. In contrast to its larger counterparts in the GPT-4o family, this model currently lacks support for image and structured output formats, concentrating solely on immediate voice and text applications. Developers have the ability to initiate a real-time session through the /realtime/sessions endpoint to acquire a temporary key, allowing them to stream user audio or text and receive immediate responses via the same connection. This model belongs to the early preview family (version 2024-12-17) and is primarily designed for testing purposes and gathering feedback, rather than handling extensive production workloads. The usage comes with certain rate limitations and may undergo changes during the preview phase. Its focus on audio and text modalities opens up possibilities for applications like conversational voice assistants, enhancing user interaction in a variety of settings. As technology evolves, further enhancements and features may be introduced to enrich user experiences.

Gemini 2.5 Pro TTS represents Google's cutting-edge text-to-speech technology within the Gemini 2.5 series, designed to deliver high-quality and expressive speech synthesis tailored for structured audio generation needs. This model produces lifelike voice output that boasts improved expressiveness, tone modulation, pacing, and accurate pronunciation, allowing developers to specify style, accent, rhythm, and emotional subtleties through text prompts. Consequently, it is ideal for a variety of uses, including podcasts, audiobooks, customer support, educational tutorials, and multimedia storytelling that demand superior audio quality. Additionally, it accommodates both single and multiple speakers, facilitating varied voices and interactive dialogues within a single audio output, and supports speech synthesis in various languages while maintaining a consistent style. In contrast to faster alternatives like Flash TTS, the Pro TTS model focuses on delivering exceptional sound quality, rich expressiveness, and detailed control over voice characteristics. This emphasis on nuance and depth makes it a preferred choice for professionals seeking to enhance their audio content.

The Gemini 2.5 Flash TTS model represents the latest advancement in Google’s Gemini 2.5 series, focusing on rapid, low-latency speech synthesis that produces expressive and controllable audio output. This model introduces notable improvements in tonal variety and expressiveness, enabling developers to create speech that aligns more closely with style prompts, whether for storytelling, character portrayals, or other contexts, thus achieving a more authentic emotional depth. With its precision pacing feature, it can adjust the speed of speech based on the context, allowing for quicker delivery in certain sections while also slowing down for emphasis when required, following specific instructions. Additionally, it accommodates multi-speaker dialogues with consistent character voices, making it suitable for various scenarios such as podcasts, interviews, and conversational agents, while also enhancing multilingual capabilities to maintain each speaker's distinct tone and style across different languages. Optimized for reduced latency, Gemini 2.5 Flash TTS is particularly well-suited for interactive applications and real-time voice interfaces, ensuring a seamless user experience. This innovative model is set to redefine how developers implement voice technology in their projects.

Qwen3-Omni is a comprehensive multilingual omni-modal foundation model designed to handle text, images, audio, and video, providing real-time streaming responses in both textual and natural spoken formats. Utilizing a unique Thinker-Talker architecture along with a Mixture-of-Experts (MoE) framework, it employs early text-centric pretraining and mixed multimodal training, ensuring high-quality performance across all formats without compromising on text or image fidelity. This model is capable of supporting 119 different text languages, 19 languages for speech input, and 10 languages for speech output. Demonstrating exceptional capabilities, it achieves state-of-the-art performance across 36 benchmarks related to audio and audio-visual tasks, securing open-source SOTA on 32 benchmarks and overall SOTA on 22, thereby rivaling or equaling prominent closed-source models like Gemini-2.5 Pro and GPT-4o. To enhance efficiency and reduce latency in audio and video streaming, the Talker component leverages a multi-codebook strategy to predict discrete speech codecs, effectively replacing more cumbersome diffusion methods. Additionally, this innovative model stands out for its versatility and adaptability across a wide array of applications.

Google has unveiled enhanced Gemini audio models that greatly broaden the platform's functionalities for engaging and nuanced voice interactions, as well as real-time conversational AI, highlighted by the arrival of Gemini 2.5 Flash Native Audio and advancements in text-to-speech technology. The revamped native audio model supports live voice agents capable of managing intricate workflows, reliably adhering to detailed user directives, and facilitating smoother multi-turn dialogues by improving context retention from earlier exchanges. This upgrade is now accessible through Google AI Studio, Vertex AI, Gemini Live, and Search Live, allowing developers and products to create dynamic voice experiences such as smart assistants and corporate voice agents. Additionally, Google has refined the core Text-to-Speech (TTS) models within the Gemini 2.5 lineup to enhance expressiveness, tone modulation, pacing adjustments, and multilingual capabilities, resulting in synthesized speech that sounds increasingly natural. Furthermore, these innovations position Google's audio technology as a leader in the realm of conversational AI, driving forward the potential for more intuitive human-computer interactions.

OpenAI’s GPT-5 mini is a cost-efficient, faster version of the flagship GPT-5 model, designed to handle well-defined tasks and precise inputs with high reasoning capabilities. Supporting text and image inputs, GPT-5 mini can process and generate large amounts of content thanks to its extensive 400,000-token context window and a maximum output of 128,000 tokens. This model is optimized for speed, making it ideal for developers and businesses needing quick turnaround times on natural language processing tasks while maintaining accuracy. The pricing model offers significant savings, charging $0.25 per million input tokens and $2 per million output tokens, compared to the higher costs of the full GPT-5. It supports many advanced API features such as streaming responses, function calling, and fine-tuning, while excluding audio input and image generation capabilities. GPT-5 mini is compatible with a broad range of API endpoints including chat completions, real-time responses, and embeddings, making it highly flexible. Rate limits vary by usage tier, supporting from hundreds to tens of thousands of requests per minute, ensuring reliability for different scale needs. This model strikes a balance between performance and cost, suitable for applications requiring fast, high-quality AI interaction without extensive resource use.

The GPT-4 model represents a significant advancement in AI, being a large multimodal system capable of handling both text and image inputs while producing text outputs, which allows it to tackle complex challenges with a level of precision unmatched by earlier models due to its extensive general knowledge and enhanced reasoning skills. Accessible through the OpenAI API for subscribers, GPT-4 is also designed for chat interactions, similar to gpt-3.5-turbo, while proving effective for conventional completion tasks via the Chat Completions API. This state-of-the-art version of GPT-4 boasts improved features such as better adherence to instructions, JSON mode, consistent output generation, and the ability to call functions in parallel, making it a versatile tool for developers. However, it is important to note that this preview version is not fully prepared for high-volume production use, as it has a limit of 4,096 output tokens. Users are encouraged to explore its capabilities while keeping in mind its current limitations.

Google Cloud's Text-to-Speech API has unveiled Chirp 3, a feature that allows users to develop custom voice models by utilizing their own high-quality audio recordings. This innovation streamlines the process of generating unique voices for audio synthesis via the Cloud Text-to-Speech API, catering to both streaming and long-form text applications. Due to safety protocols, access to this voice cloning feature is limited to select users, and those interested in gaining access must reach out to the sales team for inclusion on the allowed list. The Instant Custom Voice capability supports a variety of languages, such as English (US), Spanish (US), and French (Canada), ensuring a broad reach for users. Moreover, this service is operational across multiple Google Cloud regions and offers a range of supported output formats, including LINEAR16, OGG_OPUS, PCM, ALAW, MULAW, and MP3, depending on the chosen API method. As voice technology continues to evolve, the possibilities for personalized audio experiences are expanding rapidly.

OpenAI’s GPT-5 nano is the most cost-effective and rapid variant of the GPT-5 series, tailored for tasks like summarization, classification, and other well-defined language problems. Supporting both text and image inputs, GPT-5 nano can handle extensive context lengths of up to 400,000 tokens and generate detailed outputs of up to 128,000 tokens. Its emphasis on speed makes it ideal for applications that require quick, reliable AI responses without the resource demands of larger models. With highly affordable pricing — just $0.05 per million input tokens and $0.40 per million output tokens — GPT-5 nano is accessible to a wide range of developers and businesses. The model supports key API functionalities including streaming responses, function calling, structured output, and fine-tuning capabilities. While it does not support web search or audio input, it efficiently handles code interpretation, image generation, and file search tasks. Rate limits scale with usage tiers to ensure reliable access across small to enterprise deployments. GPT-5 nano offers an excellent balance of speed, affordability, and capability for lightweight AI applications.

GPT-4, or Generative Pre-trained Transformer 4, is a highly advanced unsupervised language model that is anticipated for release by OpenAI. As the successor to GPT-3, it belongs to the GPT-n series of natural language processing models and was developed using an extensive dataset comprising 45TB of text, enabling it to generate and comprehend text in a manner akin to human communication. Distinct from many conventional NLP models, GPT-4 operates without the need for additional training data tailored to specific tasks. It is capable of generating text or responding to inquiries by utilizing only the context it creates internally. Demonstrating remarkable versatility, GPT-4 can adeptly tackle a diverse array of tasks such as translation, summarization, question answering, sentiment analysis, and more, all without any dedicated task-specific training. This ability to perform such varied functions further highlights its potential impact on the field of artificial intelligence and natural language processing.

Nova 2 Sonic is an innovative speech-to-speech model from Amazon that facilitates real-time voice interactions, seamlessly merging speech recognition, generation, and text processing into one cohesive system. This integration allows for natural and fluid conversations, effortlessly transitioning between spoken and written communication. With enhanced multilingual capabilities and a variety of expressive voice options, Nova 2 Sonic creates responses that are not only more lifelike but also display a deeper understanding of context. Its extensive one-million-token context window enables prolonged interactions while maintaining coherence with previous exchanges. Additionally, the model's ability to handle asynchronous tasks allows users to engage in conversation, switch topics, or pose follow-up inquiries without interrupting ongoing background processes, thereby creating a more dynamic and engaging voice interaction experience. Such advancements ensure that conversations feel less constrained by conventional turn-taking dialogue methods, paving the way for more immersive communication.

Utilize an API that leverages Google's advanced AI technologies to transform text into natural-sounding speech. With the foundation laid by DeepMind’s expertise in speech synthesis, this API offers voices that closely resemble human speech patterns. You can choose from an extensive selection of over 220 voices in more than 40 languages and their various dialects, such as Mandarin, Hindi, Spanish, Arabic, and Russian. Opt for the voice that best aligns with your user demographic and application requirements. Additionally, you have the opportunity to create a distinctive voice that embodies your brand across all customer interactions, rather than relying on a generic voice that might be used by other companies. By training a custom voice model with your own audio samples, you can achieve a more unique and authentic voice for your organization. This versatility allows you to define and select the voice profile that best matches your company while effortlessly adapting to any evolving voice demands without the necessity of re-recording new phrases. This capability ensures your brand maintains a consistent audio identity that resonates with your audience.

Qwen3-TTS represents an innovative collection of advanced text-to-speech models created by the Qwen team at Alibaba Cloud, released under the Apache-2.0 license, which delivers stable, expressive, and real-time speech output with functionalities like voice cloning, voice design, and precise control over prosody and acoustic features. This suite supports ten prominent languages—Chinese, English, Japanese, Korean, German, French, Russian, Portuguese, Spanish, and Italian—along with various dialect-specific voice profiles, enabling adaptive management of tone, speech rate, and emotional delivery tailored to text semantics and user instructions. The architecture of Qwen3-TTS incorporates efficient tokenization and a dual-track design, facilitating ultra-low-latency streaming synthesis, with the first audio packet generated in approximately 97 milliseconds, making it ideal for interactive and real-time applications. Additionally, the range of models available offers diverse capabilities, such as rapid three-second voice cloning, customization of voice timbres, and voice design based on given instructions, ensuring versatility for users in many different scenarios. This flexibility in design and performance highlights the model's potential for a wide array of applications in both commercial and personal contexts.

Gemini 3, the latest model from Google DeepMind, establishes a new standard for artificial intelligence by achieving cutting-edge reasoning capabilities and multimodal comprehension across various formats including text, images, and videos. It significantly outperforms its earlier version in critical AI assessments and showcases its strengths in intricate areas like scientific reasoning, advanced programming, spatial reasoning, and visual or video interpretation. The introduction of the innovative “Deep Think” mode takes performance to an even higher level, demonstrating superior reasoning abilities for exceptionally difficult tasks and surpassing the Gemini 3 Pro in evaluations such as Humanity’s Last Exam and ARC-AGI. Now accessible within Google’s ecosystem, Gemini 3 empowers users to engage in learning, developmental projects, and strategic planning with unprecedented sophistication. With context windows extending up to one million tokens and improved media-processing capabilities, along with tailored configurations for various tools, the model enhances precision, depth, and adaptability for practical applications, paving the way for more effective workflows across diverse industries. This advancement signals a transformative shift in how AI can be leveraged for real-world challenges.

AudioLM is an innovative audio language model designed to create high-quality, coherent speech and piano music by solely learning from raw audio data, eliminating the need for text transcripts or symbolic forms. It organizes audio in a hierarchical manner through two distinct types of discrete tokens: semantic tokens, which are derived from a self-supervised model to capture both phonetic and melodic structures along with broader context, and acoustic tokens, which come from a neural codec to maintain speaker characteristics and intricate waveform details. This model employs a series of three Transformer stages, initiating with the prediction of semantic tokens to establish the overarching structure, followed by the generation of coarse tokens, and culminating in the production of fine acoustic tokens for detailed audio synthesis. Consequently, AudioLM can take just a few seconds of input audio to generate seamless continuations that effectively preserve voice identity and prosody in speech, as well as melody, harmony, and rhythm in music. Remarkably, evaluations by humans indicate that the synthetic continuations produced are almost indistinguishable from actual recordings, demonstrating the technology's impressive authenticity and reliability. This advancement in audio generation underscores the potential for future applications in entertainment and communication, where realistic sound reproduction is paramount.

ElevenLabs has unveiled Scribe, a cutting-edge Automatic Speech Recognition (ASR) model that aims to provide remarkably accurate transcriptions in 99 different languages. This innovative system is tailored to effectively manage a wide range of real-world audio situations, featuring capabilities such as word-level timestamps, speaker identification, and audio-event tagging. In benchmark evaluations like FLEURS and Common Voice, Scribe has outperformed leading models, including Gemini 2.0 Flash, Whisper Large V3, and Deepgram Nova-3, achieving impressive word error rates of 98.7% for Italian and 96.7% for English. Additionally, Scribe shows a significant reduction in errors for languages that have often faced challenges, such as Serbian, Cantonese, and Malayalam, where competing models frequently report error rates above 40%. Furthermore, developers can easily incorporate Scribe into their applications via ElevenLabs' speech-to-text API, which returns structured JSON transcripts enriched with comprehensive annotations. This level of accessibility and performance is set to revolutionize the field of transcription and enhance the user experience across various applications.

Gemini Diffusion represents our cutting-edge research initiative aimed at redefining the concept of diffusion in the realm of language and text generation. Today, large language models serve as the backbone of generative AI technology. By employing a diffusion technique, we are pioneering a new type of language model that enhances user control, fosters creativity, and accelerates the text generation process. Unlike traditional models that predict text in a straightforward manner, diffusion models take a unique approach by generating outputs through a gradual refinement of noise. This iterative process enables them to quickly converge on solutions and make real-time corrections during generation. As a result, they demonstrate superior capabilities in tasks such as editing, particularly in mathematics and coding scenarios. Furthermore, by generating entire blocks of tokens simultaneously, they provide more coherent responses to user prompts compared to autoregressive models. Remarkably, the performance of Gemini Diffusion on external benchmarks rivals that of much larger models, while also delivering enhanced speed, making it a noteworthy advancement in the field. This innovation not only streamlines the generation process but also opens new avenues for creative expression in language-based tasks.

Gemini 2.0 represents a cutting-edge AI model created by Google, aimed at delivering revolutionary advancements in natural language comprehension, reasoning abilities, and multimodal communication. This new version builds upon the achievements of its earlier model by combining extensive language processing with superior problem-solving and decision-making skills, allowing it to interpret and produce human-like responses with enhanced precision and subtlety. In contrast to conventional AI systems, Gemini 2.0 is designed to simultaneously manage diverse data formats, such as text, images, and code, rendering it an adaptable asset for sectors like research, business, education, and the arts. Key enhancements in this model include improved contextual awareness, minimized bias, and a streamlined architecture that guarantees quicker and more consistent results. As a significant leap forward in the AI landscape, Gemini 2.0 is set to redefine the nature of human-computer interactions, paving the way for even more sophisticated applications in the future. Its innovative features not only enhance user experience but also facilitate more complex and dynamic engagements across various fields.

In 2024, the OpenAI Realtime API was unveiled, providing developers the capability to build applications that support instantaneous, low-latency interactions, exemplified by speech-to-speech conversations. This innovative API caters to various applications, including customer support systems, AI-driven voice assistants, and educational tools for language learning. Departing from earlier methods that necessitated the use of multiple models for speech recognition and text-to-speech tasks, the Realtime API integrates these functions into a single call, significantly enhancing the speed and fluidity of voice interactions in applications. As a result, developers can create more engaging and responsive user experiences.

Gemini Enterprise, an all-encompassing AI platform from Google Cloud, is designed to harness the full capabilities of Google’s sophisticated AI models, tools for creating agents, and enterprise-level access to data, seamlessly integrating these into daily workflows. This innovative solution features a cohesive chat interface that facilitates employee interaction with internal documents, applications, various data sources, and personalized AI agents. The foundation of Gemini Enterprise consists of six essential elements: the Gemini suite of large multimodal models, an agent orchestration workbench (previously known as Google Agentspace), ready-made starter agents, powerful data integration connectors for business systems, extensive security and governance frameworks, and a collaborative partner ecosystem for customized integrations. Built to scale across various departments and organizations, it empowers users to develop no-code or low-code agents capable of automating diverse tasks like research synthesis, customer service responses, code assistance, and contract analysis while adhering to corporate compliance regulations. Moreover, the platform is designed to enhance productivity and foster innovation within businesses, ensuring that users can leverage advanced AI technologies with ease.

Wan2.5-Preview arrives with a groundbreaking multimodal foundation that unifies understanding and generation across text, imagery, audio, and video. Its native multimodal design, trained jointly across diverse data sources, enables tighter modal alignment, smoother instruction execution, and highly coherent audio-visual output. Through reinforcement learning from human feedback, it continually adapts to aesthetic preferences, resulting in more natural visuals and fluid motion dynamics. Wan2.5 supports cinematic 1080p video generation with synchronized audio, including multi-speaker content, layered sound effects, and dynamic compositions. Creators can control outputs using text prompts, reference images, or audio cues, unlocking a new range of storytelling and production workflows. For still imagery, the model achieves photorealism, artistic versatility, and strong typography, plus professional-level chart and design rendering. Its editing tools allow users to perform conversational adjustments, merge concepts, recolor products, modify materials, and refine details at pixel precision. This preview marks a major leap toward fully integrated multimodal creativity powered by AI.

UntitledPen is an innovative platform that harnesses AI technology, allowing users to craft, enhance, and seamlessly convert text into lifelike, human-like voice-overs through sophisticated audio generation techniques. It boasts a user-friendly smart editor and a writing assistant designed for script creation, text refinement, and content enhancement in multiple languages. Users have the ability to easily transform text into speech or vice versa, select from various voice options, and tailor aspects such as tone, accent, and personality. With efficient commands that facilitate both writing and audio production, the platform also offers integrated voice editing tools for minor modifications. Ideal for applications like podcasts, videos, and presentations, it includes features for audio downloading and uploading, as well as intelligent transcription services to convert spoken words into polished written content. Currently available in open beta, UntitledPen encourages users to explore its features at no cost, providing an excellent opportunity to experience its full potential. The platform aims to redefine the way individuals interact with text and audio, making content creation more accessible and efficient than ever before.

Gemini 2.5, developed by Google DeepMind, represents a breakthrough in AI with enhanced reasoning capabilities and native multimodality, allowing it to process long context windows of up to one million tokens. The family includes three variants: Pro for complex coding tasks, Flash for fast general use, and Flash-Lite for high-volume, cost-efficient workflows. Gemini 2.5 models improve accuracy by thinking through diverse strategies and provide developers with adaptive controls to optimize performance and resource use. The models handle multiple input types—text, images, video, audio, and PDFs—and offer powerful tool use like search and code execution. Gemini 2.5 achieves state-of-the-art results across coding, math, science, reasoning, and multilingual benchmarks, outperforming its predecessors. It is accessible through Google AI Studio, Gemini API, and Vertex AI platforms. Google emphasizes responsible AI development, prioritizing safety and security in all applications. Gemini 2.5 enables developers to build advanced interactive simulations, automated coding, and other innovative AI-driven solutions.

Amazon Nova Sonic is an advanced speech-to-speech model that offers real-time, lifelike voice interactions while maintaining exceptional price efficiency. By integrating speech comprehension and generation into one cohesive model, it allows developers to craft engaging and fluid conversational AI solutions with minimal delay. This system fine-tunes its replies by analyzing the prosody of the input speech, including elements like rhythm and tone, which leads to more authentic conversations. Additionally, Nova Sonic features function calling and agentic workflows that facilitate interactions with external services and APIs, utilizing knowledge grounding with enterprise data through Retrieval-Augmented Generation (RAG). Its powerful speech understanding capabilities encompass both American and British English across a variety of speaking styles and acoustic environments, with plans to incorporate more languages in the near future. Notably, Nova Sonic manages interruptions from users seamlessly while preserving the context of the conversation, demonstrating its resilience against background noise interference and enhancing the overall user experience. This technology represents a significant leap forward in conversational AI, ensuring that interactions are not only efficient but also genuinely engaging.

Mirage represents a groundbreaking advancement as the first real-time, autoregressive model designed for transforming video into a new digital landscape instantly, requiring no pre-rendering. Utilizing cutting-edge Live-Stream Diffusion (LSD) technology, it achieves an impressive processing rate of 24 FPS with latency under 40 ms, which guarantees smooth and continuous video transformations while maintaining the integrity of motion and structure. Compatible with an array of inputs including webcams, gameplay, films, and live broadcasts, Mirage can dynamically incorporate text-prompted style modifications in real-time. Its sophisticated history-augmentation feature ensures that temporal coherence is upheld throughout the frames, effectively eliminating the common glitches associated with diffusion-only models. With GPU-accelerated custom CUDA kernels, it boasts performance that is up to 16 times faster than conventional techniques, facilitating endless streaming without interruptions. Additionally, it provides real-time previews for both mobile and desktop platforms, allows for effortless integration with any video source, and supports a variety of deployment options, enhancing accessibility for users. Overall, Mirage stands out as a transformative tool in the realm of digital video innovation.

Hume AI has unveiled Octave, an innovative text-to-speech platform that utilizes advanced language model technology to deeply understand and interpret word context, allowing it to produce speech infused with the right emotions, rhythm, and cadence. Unlike conventional TTS systems that simply vocalize text, Octave mimics the performance of a human actor, delivering lines with rich expression tailored to the content being spoken. Users are empowered to create a variety of unique AI voices by submitting descriptive prompts, such as "a skeptical medieval peasant," facilitating personalized voice generation that reflects distinct character traits or situational contexts. Moreover, Octave supports the adjustment of emotional tone and speaking style through straightforward natural language commands, enabling users to request changes like "speak with more enthusiasm" or "whisper in fear" for precise output customization. This level of interactivity enhances user experience by allowing for a more engaging and immersive auditory experience.

Gemini Flash represents a cutting-edge large language model developed by Google, specifically engineered for rapid, efficient language processing activities. As a part of the Gemini lineup from Google DeepMind, it is designed to deliver instantaneous responses and effectively manage extensive applications, proving to be exceptionally suited for dynamic AI-driven interactions like customer service, virtual assistants, and real-time chat systems. In addition to its impressive speed, Gemini Flash maintains a high standard of quality; it utilizes advanced neural architectures that guarantee responses are contextually appropriate, coherent, and accurate. Google has also integrated stringent ethical guidelines and responsible AI methodologies into Gemini Flash, providing it with safeguards to address and reduce biased outputs, thereby ensuring compliance with Google’s principles for secure and inclusive AI. With the capabilities of Gemini Flash, businesses and developers are empowered to implement agile, intelligent language solutions that can satisfy the requirements of rapidly evolving environments. This innovative model marks a significant step forward in the quest for sophisticated AI technologies that respect ethical considerations while enhancing user experience.

Murf API is a cutting-edge text-to-speech (TTS) solution that converts written content into highly realistic, human-like voiceovers with precision and ease. Designed for developers and businesses, it offers advanced features such as pitch and speed control, adjustable pauses, fine-tuned audio duration, and an extensive pronunciation library. With over 133 AI voices available in 20+ languages, including diverse regional accents, Murf API makes it simple to create localized and engaging audio content for global users. It supports multiple audio formats, including MP3, WAV, FLAC, ALAW, ULAW, and Base64, ensuring compatibility across different platforms. Backed by flexible, transparent pricing, strong security protocols, and detailed documentation, Murf API seamlessly integrates with websites, chatbots, IVR systems, and mobile applications.

Chatterbox, an open-source voice cloning AI model created by Resemble AI and distributed under the MIT license, allows users to perform zero-shot voice cloning with just a five-second sample of reference audio, thereby removing the requirement for extensive training. This innovative model provides expressive speech synthesis that features emotion control, enabling users to modify the expressiveness of the voice from a dull tone to a highly dramatic one using a single adjustable parameter. Additionally, Chatterbox allows for accent modulation and offers text-based control, which guarantees a high-quality and human-like text-to-speech output. With its faster-than-real-time inference capabilities, it is well-suited for applications requiring immediate responses, such as voice assistants and interactive media experiences. Designed with developers in mind, the model supports easy installation via pip and comes with thorough documentation. Furthermore, Chatterbox integrates built-in watermarking through Resemble AI’s PerTh (Perceptual Threshold) Watermarker, which discreetly embeds data to safeguard the authenticity of generated audio. This combination of features makes Chatterbox a powerful tool for creating versatile and realistic voice applications. The model's emphasis on user control and quality further enhances its appeal in various creative and professional fields.

Piper is a rapidly operating, localized neural text-to-speech (TTS) system that is particularly optimized for devices like the Raspberry Pi 4, aiming to provide top-notch speech synthesis capabilities without the dependence on cloud infrastructure. It employs neural network models developed with VITS and subsequently exported to ONNX Runtime, which facilitates both efficient and natural-sounding speech production. Supporting a diverse array of languages, Piper includes English (both US and UK dialects), Spanish (from Spain and Mexico), French, German, and many others, with downloadable voice options available. Users have the flexibility to operate Piper through command-line interfaces or integrate it seamlessly into Python applications via the piper-tts package. The system boasts features such as real-time audio streaming, JSON input for batch processing, and compatibility with multi-speaker models, enhancing its versatility. Additionally, Piper makes use of espeak-ng for phoneme generation, transforming text into phonemes before generating speech. It has found applications in various projects, including Home Assistant, Rhasspy 3, and NVDA, among others, illustrating its adaptability across different platforms and use cases. With its emphasis on local processing, Piper appeals to users looking for privacy and efficiency in their speech synthesis solutions.

Gemini's inherent multimodal capabilities allow for the conversion of various input types into diverse output forms. From its inception, Gemini has been developed with a strong emphasis on responsibility, implementing safeguards and collaborating with partners to enhance its safety and inclusivity. You can seamlessly incorporate Gemini models into your applications using Google AI Studio and Google Cloud Vertex AI, enabling a wide range of innovative uses. This integration facilitates a more dynamic interaction with technology across different platforms and applications.

Utterly Voice is an innovative application that allows for highly customizable voice dictation and comprehensive computer control, enabling a truly hands-free computing experience. With this tool, users can perform a variety of tasks such as typing, editing, executing keyboard shortcuts, managing windows, scrolling through content, controlling the mouse, and even creating macros, all through voice commands. It is designed to be compatible with both Windows 10 and 11 and currently supports English, with future plans to incorporate additional languages. The application features several speech recognizers and models, including Vosk, Microsoft Azure, Deepgram, Google Cloud Speech-to-Text V1, and Whisper, giving users a broad selection to meet their needs. Users can effortlessly input individual characters, alphanumeric data, or even code while enjoying the flexibility provided by extensive customization options through text configuration files. Enhanced mouse control techniques, adjustable voice commands, and tailored speech recognition settings significantly improve the overall user experience, making Utterly Voice a powerful tool for anyone looking to optimize their computing through voice interaction. Overall, this application not only increases productivity but also aims to make technology more accessible to a wider audience.

Experience unparalleled security and privacy along with the most advanced iteration of ChatGPT to date. 1. Customer data and prompts are excluded from model training processes. 2. Data is securely encrypted both at rest using AES-256 and during transit with TLS 1.2 or higher. 3. Compliance with SOC 2 standards is ensured. 4. A dedicated admin console simplifies bulk management of members. 5. Features like SSO and Domain Verification enhance security. 6. An analytics dashboard provides insights into usage patterns. 7. Users enjoy unlimited, high-speed access to GPT-4 alongside Advanced Data Analysis capabilities*. 8. With 32k token context windows, you can input four times longer texts and retain memory. 9. Easily shareable chat templates facilitate collaboration within your organization. 10. This comprehensive suite of features ensures that your team operates seamlessly and securely.

Inworld TTS stands out as a cutting-edge text-to-speech solution that provides exceptionally realistic and context-aware speech synthesis alongside advanced voice-cloning features, all at an incredibly affordable price. Its leading model, TTS-1, is tailored for real-time usage, boasting low-latency streaming capabilities—where the first audio segment is available in about 200 milliseconds—and supports a wide array of languages such as English, Spanish, French, Korean, Chinese, and several others. Developers have the flexibility to utilize instant zero-shot voice cloning, requiring only 5 to 15 seconds of audio input, or opt for more detailed fine-tuned cloning, enabling the addition of voice-tags that convey emotion, style, and non-verbal cues, while also allowing for language switching without losing the unique voice identity. For those seeking even greater expressiveness and multilingual capabilities, the TTS-1-Max model is currently in preview, offering enhanced features. The platform accommodates various access methods, including API and portal options, and can operate in either streaming or batch modes, making it suitable for a diverse range of applications such as interactive voice agents, gaming characters, and bespoke audio branding experiences. With its versatility and advanced technology, Inworld TTS is poised to revolutionize how we interact with synthetic voices.

Voxtral models represent cutting-edge open-source systems designed for speech understanding, available in two sizes: a larger 24 B variant aimed at production-scale use and a smaller 3 B variant suitable for local and edge applications, both of which are provided under the Apache 2.0 license. These models excel in delivering precise transcription while featuring inherent semantic comprehension, accommodating long-form contexts of up to 32 K tokens and incorporating built-in question-and-answer capabilities along with structured summarization. They automatically detect languages across a range of major tongues and enable direct function-calling to activate backend workflows through voice commands. Retaining the textual strengths of their Mistral Small 3.1 architecture, Voxtral can process audio inputs of up to 30 minutes for transcription tasks and up to 40 minutes for comprehension, consistently surpassing both open-source and proprietary competitors in benchmarks like LibriSpeech, Mozilla Common Voice, and FLEURS. Users can access Voxtral through downloads on Hugging Face, API endpoints, or by utilizing private on-premises deployments, and the model also provides options for domain-specific fine-tuning along with advanced features tailored for enterprise needs, thus enhancing its applicability across various sectors.

VideoPoet is an innovative modeling technique that transforms any autoregressive language model or large language model (LLM) into an effective video generator. It comprises several straightforward components. An autoregressive language model is trained across multiple modalities—video, image, audio, and text—to predict the subsequent video or audio token in a sequence. The training framework for the LLM incorporates a range of multimodal generative learning objectives, such as text-to-video, text-to-image, image-to-video, video frame continuation, inpainting and outpainting of videos, video stylization, and video-to-audio conversion. Additionally, these tasks can be combined to enhance zero-shot capabilities. This straightforward approach demonstrates that language models are capable of generating and editing videos with impressive temporal coherence, showcasing the potential for advanced multimedia applications. As a result, VideoPoet opens up exciting possibilities for creative expression and automated content creation.

Hume AI's EVI 3 represents a cutting-edge advancement in speech-language technology, seamlessly streaming user speech to create natural and expressive verbal responses. It achieves conversational latency while maintaining the same level of speech quality as our text-to-speech model, Octave, and simultaneously exhibits the intelligence comparable to leading LLMs operating at similar speeds. In addition, it collaborates with reasoning models and web search systems, allowing it to “think fast and slow,” thereby aligning its cognitive capabilities with those of the most sophisticated AI systems available. Unlike traditional models constrained to a limited set of voices, EVI 3 has the ability to instantly generate a vast array of new voices and personalities, engaging users with over 100,000 custom voices already available on our text-to-speech platform, each accompanied by a distinct inferred personality. Regardless of the chosen voice, EVI 3 can convey a diverse spectrum of emotions and styles, either implicitly or explicitly upon request, enhancing user interaction. This versatility makes EVI 3 an invaluable tool for creating personalized and dynamic conversational experiences.

ChatGPT by OpenAI is a versatile AI conversational platform that provides assistance in writing, learning, brainstorming, code generation, and problem-solving across a wide range of topics. Available for free with optional Plus and Pro subscription plans, it supports real-time text and voice interactions on web browsers and mobile apps. Users can leverage ChatGPT to create content, summarize meetings, debug code, analyze data, and even generate images using integrated tools like DALL·E 3. The platform is accessible via desktop and mobile devices and offers personalized workflows through custom GPTs and projects. Advanced plans unlock deeper research capabilities, extended limits, and access to cutting-edge AI models like GPT-4o and OpenAI o1 pro mode. ChatGPT integrates search capabilities for real-time information and enables collaboration through features like Canvas for project editing. It caters to students, professionals, hobbyists, and developers seeking efficient, AI-driven support. OpenAI continually updates ChatGPT with new tools and enhanced usability.

AudioTextHub is a powerful, free online text-to-speech platform that uses advanced AI voice synthesis to transform text into natural-sounding, expressive speech within seconds. It offers a diverse library of more than 500 voices spanning multiple languages and regional accents, making it ideal for a global audience. Users can personalize the speech output by adjusting speed, pitch, and emphasis, ensuring the audio matches their specific style or requirements. The platform is optimized for fast, high-quality audio generation, helping content creators, educators, and developers save time and increase efficiency. Its easy-to-use API enables smooth integration of text-to-speech features into websites and applications. AudioTextHub prioritizes security, guaranteeing that all text data is processed confidentially and safely. The platform is suitable for accessibility projects, e-learning, podcasting, and more. Its combination of flexibility, speed, and natural voice quality makes it a top choice for transforming written content into engaging audio.