Voice Live API now supports WebRTC (Preview)
We are thrilled to announce that Voice Live API now supports WebRTC (Web Real-Time Communication) connection, enabling low‑latency, real‑time voice interactions directly from web and mobile clients. Why WebRTC is required in real-time voice agent building WebRTC enables real‑time, bi‑directional audio and video streaming directly in the browser without plugins or native installs. Unlike WebSocket, which treat audio as generic data and require custom buffering and timing logic, WebRTC is purpose‑built for media‑aware streaming needed for responsive conversational experiences. Specifically: Lower latency: WebRTC is designed to minimize delay, making it more suitable for audio and video communication where low latency is critical for maintaining quality and synchronization. Built-in media handling: WebRTC has built-in support for audio and video codecs, providing optimized handling of media streams. Network resilience: WebRTC includes mechanisms for handling packet loss and jitter, which are essential for maintaining the quality of audio streams over unpredictable networks. How to set up WebRTC in Voice Live In a typical setup, the client establishes a WebSocket‑based control channel with the Voice Live API to exchange SDP offer and answer messages required for WebRTC session negotiation. Once negotiation completes, audio is transmitted over WebRTC RTP media tracks. Non‑audio events, such as voice activity and response lifecycle signals, are exchanged over WebRTC data channels alongside the media streams. Session configuration, control‑plane messages, and error notifications are delivered through the WebSocket control channel. When initiating a WebRTC call session, simply use the voice-live/realtime/calls endpoint instead of voice-live/realtime. For example: wss://<your-ai-foundry-resource-name>.services.ai.azure.com/voice-live/realtime/calls?api-version=2026-01-01-preview&model=gpt-realtime For more information, see the step-by-step instruction. Learn more Voice Live API is transforming how developers build voice-enabled agent systems by providing an integrated, scalable, and efficient solution. By combining speech recognition, generative AI, and text-to-speech functionalities into a unified interface, it addresses the challenges of traditional implementations, enabling faster development and superior user experiences. From streamlining customer service to enhancing education and public services, the opportunities are endless. The future of voice-first solutions is here—let’s build it together! Voice Live API introduction (video) Try Voice Live in Azure AI Foundry Voice Live API documents Voice Live quickstart Voice Live Agent code sample in GitHubPower Up Your Open WebUI with Azure AI Speech: Quick STT & TTS Integration
Introduction Ever found yourself wishing your web interface could really talk and listen back to you? With a few clicks (and a bit of code), you can turn your plain Open WebUI into a full-on voice assistant. In this post, you’ll see how to spin up an Azure Speech resource, hook it into your frontend, and watch as user speech transforms into text and your app’s responses leap off the screen in a human-like voice. By the end of this guide, you’ll have a voice-enabled web UI that actually converses with users, opening the door to hands-free controls, better accessibility, and a genuinely richer user experience. Ready to make your web app speak? Let’s dive in. Why Azure AI Speech? We use Azure AI Speech service in Open Web UI to enable voice interactions directly within web applications. This allows users to: Speak commands or input instead of typing, making the interface more accessible and user-friendly. Hear responses or information read aloud, which improves usability for people with visual impairments or those who prefer audio. Provide a more natural and hands-free experience especially on devices like smartphones or tablets. In short, integrating Azure AI Speech service into Open Web UI helps make web apps smarter, more interactive, and easier to use by adding speech recognition and voice output features. If you haven’t hosted Open WebUI already, follow my other step-by-step guide to host Ollama WebUI on Azure. Proceed to the next step if you have Open WebUI deployed already. Learn More about OpenWeb UI here. Deploy Azure AI Speech service in Azure. Navigate to the Azure Portal and search for Azure AI Speech on the Azure portal search bar. Create a new Speech Service by filling up the fields in the resource creation page. Click on “Create” to finalize the setup. After the resource has been deployed, click on “View resource” button and you should be redirected to the Azure AI Speech service page. The page should display the API Keys and Endpoints for Azure AI Speech services, which you can use in Open Web UI. Settings things up in Open Web UI Speech to Text settings (STT) Head to the Open Web UI Admin page > Settings > Audio. Paste the API Key obtained from the Azure AI Speech service page into the API key field below. Unless you use different Azure Region, or want to change the default configurations for the STT settings, leave all settings to blank. Text to Speech settings (TTS) Now, let's proceed with configuring the TTS Settings on OpenWeb UI by toggling the TTS Engine to Azure AI Speech option. Again, paste the API Key obtained from Azure AI Speech service page and leave all settings to blank. You can change the TTS Voice from the dropdown selection in the TTS settings as depicted in the image below: Click Save to reflect the change. Expected Result Now, let’s test if everything works well. Open a new chat / temporary chat on Open Web UI and click on the Call / Record button. The STT Engine (Azure AI Speech) should identify your voice and provide a response based on the voice input. To test the TTS feature, click on the Read Aloud (Speaker Icon) under any response from Open Web UI. The TTS Engine should reflect Azure AI Speech service! Conclusion And that’s a wrap! You’ve just given your Open WebUI the gift of capturing user speech, turning it into text, and then talking right back with Azure’s neural voices. Along the way you saw how easy it is to spin up a Speech resource in the Azure portal, wire up real-time transcription in the browser, and pipe responses through the TTS engine. From here, it’s all about experimentation. Try swapping in different neural voices or dialing in new languages. Tweak how you start and stop listening, play with silence detection, or add custom pronunciation tweaks for those tricky product names. Before you know it, your interface will feel less like a web page and more like a conversation partner.
Now in Foundry: Cohere Transcribe, Nanbeige 4.1-3B, and Octen Embedding
This week's Model Mondays edition spans three distinct layers of the AI application stack: Cohere's cohere-transcribe, a 2B Automatic Speech Recognition (ASR) model that ranks first on the Open ASR Leaderboard across 14 languages; Nanbeige's Nanbeige4.1-3B, a compact 3B reasoning model that outperforms models ten times its size on coding, math, and deep-search benchmarks; and Octen's Octen-Embedding-0.6B, a lightweight text embedding model that achieves strong retrieval scores across 100+ languages and industry-specific domains. Together, these three models illustrate how developers can build full AI pipelines—from audio ingestion to language reasoning to semantic retrieval—entirely with open-source models deployed through Microsoft Foundry. Each operates in a different modality and fills a distinct architectural role, making this week's selection especially well-suited for teams assembling production-grade systems across speech, text, and search. Models of the week Cohere's cohere-transcribe-03-2026 Model Specs Parameters / size: 2B Primary task: Automatic Speech Recognition (audio-to-text) Why it's interesting Top-ranked on the Open ASR Leaderboard: cohere-transcribe-03-2026 achieves a 5.42% average Word Error Rate (WER) across 8 English benchmark datasets as of March 26, 2026—placing it first among open models. It reaches 1.25% WER on LibriSpeech Clean and 8.15% on AMI (meeting transcription), demonstrating consistent accuracy across both clean speech and real-world, multi-speaker environments. Benchmarks: Open ASR Leaderboard. 14 languages with a dedicated encoder-decoder architecture: The model uses a large Conformer encoder for acoustic representation extraction paired with a lightweight Transformer decoder for token generation, trained from scratch on 14 languages covering European, East Asian (Chinese Mandarin, Japanese, Korean, Vietnamese), and Arabic. Unlike general-purpose models adapted for ASR, this dedicated architecture makes it efficient without sacrificing accuracy. Long-form audio with automatic chunking: Audio longer than 35 seconds is automatically split into overlapping chunks and reassembled into a coherent transcript—no manual preprocessing required. Batched inference, punctuation control, and per-language configuration are all supported through the standard API. Try it Click on the window above, upload an audio file, and watch how quickly the model transcribes it for you. Or click the link to experiment with the Cohere Transcribe Space and record audio directly from your device. Use Case Prompt Pattern Meeting transcription Submit recorded audio with language tag; retrieve timestamped transcript per speaker turn Call center quality review Batch-process customer call recordings, extract transcript, pass to classification model Medical documentation Transcribe clinical encounters; feed transcript into summarization or structured note pipeline Multilingual content indexing Process podcasts or video audio in any of 14 supported languages; store as searchable text Sample prompt for a legal services deployment: You are building a contract negotiation assistant. A client submits a recorded audio of a 45-minute supplier negotiation call. Using the cohere-transcribe-03-2026 endpoint deployed in Microsoft Foundry, transcribe the call with punctuation enabled for the English audio. Once the transcript is available, pass it to a downstream language model with the following instruction: "Identify all pricing commitments, delivery deadlines, and liability clauses mentioned in this negotiation transcript. For each, note the speaker's position (client or supplier) and flag any terms that appear ambiguous or require legal review." Nanbeige's Nanbeige4.1-3B Model Specs Parameters / size: 3B Context length: 131,072 tokens Primary task: Text generation (reasoning, coding, tool use, deep search) Why it's interesting Reasoning performance that exceeds its size class: Nanbeige4.1-3B scores 76.9 on LiveCodeBench-V6, these results suggest that targeted post-training using Supervised Fine-Tuning (SFT) and Reinforcement Learning (RL) on a focused dataset can yield improvements that scale-based approaches cannot replicate at equivalent parameter counts. Read the technical report: https://huggingface.co/papers/2602.13367. Strong preference alignment at the 3B scale: On Arena-Hard-v2, Nanbeige4.1-3B scores 73.2, compared to 56.0 for Qwen3-32B and 60.2 for Qwen3-30B-A3B—both significantly larger models. This indicates that the model's outputs consistently match human preference for response quality and helpfulness, not just accuracy on structured tasks. Deep-search capability previously absent from small general models: On xBench-DeepSearch-2505, Nanbeige4.1-3B scores 75—matching search-specialized small agents. The model can sustain complex agentic tasks involving more than 500 sequential tool invocations, a capability gap that previously required either specialized search agents or significantly larger models. Native tool-use support: The model's chat template and generation pipeline natively support tool call formatting, making it straightforward to connect to external APIs and build multi-step agentic workflows without additional scaffolding. Try it Use Case Prompt Pattern Code review and fix Provide failing test + stack trace; ask model to diagnose root cause and write corrected implementation Competition-style math Submit problem as structured prompt; use temperature 0.6, top-p 0.95 for consistent reasoning steps Agentic task execution Provide tool definitions as JSON + goal; let model plan and execute tool calls sequentially Long-document Q&A Pass full document (up to 131K tokens) with targeted factual questions; extract structured answers Sample prompt for a software engineering deployment: You are automating pull request review for a backend engineering team. Using the Nanbeige4.1-3B endpoint deployed in Microsoft Foundry, provide the model with a unified diff of a proposed code change and the following system instruction: "You are a senior software engineer reviewing a pull request. For each modified function: (1) summarize what the change does, (2) identify any edge cases that are not handled, (3) flag any security or performance regressions relative to the original, and (4) suggest a specific improvement if one is warranted. Format your output as a structured list per function." Octen's Octen-Embedding-0.6B Model Specs Parameters / size: 0.6B Context length: 32,768 tokens Primary task: Text embeddings (semantic search, retrieval, similarity) Why it's interesting Retrieval performance above larger proprietary models at 0.6B: On the RTEB (Retrieval Text Embedding Benchmark) public leaderboard, Octen-Embedding-0.6B achieves a mean task score of 0.7241—above voyage-3.5 (0.7139), Cohere-embed-v4.0 (0.6534), and text-embedding-3-large (0.6110), despite being a fraction of their parameter count. The model is fine-tuned from Qwen3-Embedding-0.6B via Low-Rank Adaptation (LoRA), demonstrating that targeted fine-tuning on retrieval-specific data can close the gap with larger embedding models. Vertical domain coverage across legal, finance, healthcare, and code: Octen-Embedding-0.6B was trained with explicit coverage of domain-specific retrieval scenarios—legal document matching, financial report Q&A, clinical dialogue retrieval, and code search including SQL. This makes it suitable for regulated-industry applications where generic embedding models tend to underperform on specialized terminology. 32,768-token context for long-document retrieval: The extended context window supports encoding entire legal contracts, earnings reports, or clinical case notes as single embeddings—removing the need to chunk long documents and re-aggregate scores at query time, which can introduce ranking errors. 100+ language support with cross-lingual retrieval: The model handles multilingual and cross-lingual retrieval natively, with strong coverage across languages including English, Chinese, and other major languages via its Qwen3-based architecture—practical for global enterprise applications that span multiple languages. Use Case Prompt Pattern Semantic search Encode user query and document corpus; rank documents by cosine similarity to query embedding Legal precedent retrieval Embed case briefs and query with legal question; retrieve most semantically relevant precedents Cross-lingual document search Encode multilingual document set; submit query in any supported language for cross-lingual retrieval Financial Q&A pipeline Embed earnings reports or filings; retrieve relevant passages to ground downstream language model responses Sample prompt for a global enterprise knowledge base deployment: You are building a clinical decision support tool. Using the Octen-Embedding-0.6B endpoint deployed in Microsoft Foundry, embed a corpus of 10,000 clinical case notes at ingestion time and store the resulting 1024-dimensional vectors in a vector database. At query time, encode an incoming patient presentation summary and retrieve the 5 most semantically similar historical cases. Pass the retrieved cases and the current presentation to a language model with the following instruction: "Based on these five similar cases and their documented outcomes, summarize the most common treatment approaches and flag any cases where the outcome differed significantly from the initial prognosis." Getting started You can deploy open-source Hugging Face models directly in Microsoft Foundry by browsing the Hugging Face collection in the Foundry model catalog and deploying to managed endpoints in just a few clicks. You can also start from the Hugging Face Hub. First, select any supported model and then choose "Deploy on Microsoft Foundry", which brings you straight into Azure with secure, scalable inference already configured. Learn how to discover models and deploy them using Microsoft Foundry documentation: Follow along the Model Mondays series and access the GitHub to stay up to date on the latest Read Hugging Face on Azure docs Learn about one-click deployments from the Hugging Face Hub on Microsoft Foundry Explore models in Microsoft FoundryNow in Foundry: NVIDIA Nemotron-3-Super-120B-A12B, IBM Granite-4.0-1b-Speech, and Sarvam-105B
This week's Model Mondays edition highlights three models now available in Hugging Face collection on Microsoft Foundry: NVIDIA's Nemotron-3-Super-120B-A12B, a hybrid Latent Mixture-of-Experts (MOE) model with 12B active parameters and context handling up to 1 million tokens; IBM Granite's Granite-4.0-1b-Speech, a compact Automatic Speech Recognition (ASR) and Automatic Speech Translation (AST) model that achieves a 5.52% average Word Error Rate (WER) at 280× real-time speed with runtime keyword biasing for domain adaptation; and Sarvam's Sarvam-105B, a 105B Mixture-of-Experts (MoE) model with 10.3B active parameters optimized for complex reasoning and 22 Indian languages, with comparable agentic performance compared to other larger proprietary models on web search and task-planning benchmarks. Models of the week NVIDIA Nemotron-3-Super-120B-A12B Model Specs Parameters / size: 120B total with 12B active Context length: Up to 1M tokens Primary task: Text generation (reasoning, agentic workflows, long-context tasks, tool use, RAG) Why it's interesting (Spotlight) Hybrid Latent MoE architecture with selective attention: Nemotron-3-Super combines interleaved Mamba-2 state-space layers and sparse MoE layers with a select number of full attention layers—a design called Latent MoE. Tokens are routed into a smaller latent space for computation, which improves accuracy per parameter while keeping only 12B parameters active at inference time. Multi-Token Prediction (MTP) heads where the model simultaneously predicts multiple upcoming tokens during training enable native speculative decoding, reducing time-to-first-token on long outputs without a separate draft model. Configurable reasoning mode: The model supports toggling extended chain-of-thought reasoning on or off via the chat template flag enable_thinking. This lets developers suppress the reasoning trace for latency-sensitive tasks while keeping it available for high-stakes or multi-step agentic use cases without loading a separate model. Sustained 1M-token context reliability: On RULER, the standard long-context evaluation suite, Nemotron-3-Super achieves 91.75% at 1M tokens. This makes it practical for full-document retrieval-augmented generation (RAG), long-form code analysis, and extended agentic sessions without chunking or windowing strategies. Try it Use cases Best practices Ultra‑long document ingestion & consolidation (e.g., end‑to‑end review of massive specs, logs, or multi‑volume manuals without chunking) Use the native 1M‑token context to avoid windowing strategies; feed full corpora in one pass to reduce stitching errors. Prefer default decoding for general analysis (NVIDIA recommends temperature≈1.0, top_p≈0.95) before tuning; this aligns with the model’s training and MTP‑optimized generation path. Leverage MTP for throughput (multi‑token prediction improves output speed on long outputs), making single‑pass synthesis practical at scale. Latency‑sensitive chat & tool‑calling at scale (e.g., high‑volume enterprise assistants where response time matters) Toggle reasoning traces intentionally via the chat template (enable_thinking on/off): turn off for low‑latency interactions; on for harder prompts where accuracy benefits from explicit reasoning. Use model‑recommended sampling for tool calls (many guides tighten temperature for tool use) to improve determinism while keeping top_p near 0.95. Rely on the LatentMoE + MTP design to sustain high tokens/sec under load instead of adding a draft model for speculative decoding. IBM Granite-4.0-1b-Speech Model Specs Parameters / size: ~1B Context length: 128K tokens (LLM backbone; audio processed per utterance through the speech encoder) Primary task: Multilingual Automatic Speech Recognition (ASR) and bidirectional Automatic Speech Translation (AST) Why it's interesting (Spotlight) Compact ASR with speculative decoding at near-real-time speed: At roughly 1B parameters, Granite-4.0-1b-Speech achieves a 5.52% average WER across eight English benchmarks at 280× real-time speed (RTFx—the ratio of audio duration processed to wall-clock time) on the Open ASR Leaderboard. Runtime keyword biasing for domain adaptation without fine-tuning: Granite-4.0-1b-Speech accepts a runtime keyword list—proper nouns, brand names, technical terms, acronyms—that adjusts decoding probabilities toward those terms. This allows domain-specific vocabulary to be injected at inference time rather than requiring a fine-tuning run, practical for legal transcription, medical dictation, or financial meeting notes where terminology changes across clients. Bidirectional speech translation across 6 languages in one model: Beyond ASR, the model supports translation both to and from English for French, German, Spanish, Portuguese, and Japanese, plus English-to-Italian and English-to-Mandarin. A single deployed endpoint handles ASR and AST tasks without routing audio to separate models, reducing infrastructure surface area. Try it Test the model in the Hugging Face space before deploying in Foundry here: Sarvam’s Sarvam-105B Model Specs Parameters / size: 105B total with 10.3B active (Mixture of Experts, BF16) Context length: 128K tokens (with YaRN-based long-context extrapolation, scale factor 40) Primary task: Text generation (reasoning, coding, agentic tasks, Indian language understanding) Why it's interesting (Spotlight) Broad Indian language coverage at scale: Sarvam-105B supports English and 22 Indian languages—Hindi, Bengali, Tamil, Telugu, Marathi, Gujarati, Kannada, Malayalam, Punjabi, Odia, Assamese, Urdu, Sanskrit, Maithili, Dogri, Manipuri, Santali, Kashmiri, Nepali, Sindhi, Konkani, and Tibetan—the broadest open-model coverage for this language set at this parameter range. Training explicitly prioritized the Indian context, resulting in reported state-of-the-art performance across these languages for models of comparable size. Strong agentic and web-search performance: Sarvam-105B scores 49.5% on BrowseComp (web research benchmark with search tool access)—substantially above GLM-4.5-Air (21.3%) and Qwen3-Next-80B-A3B-Thinking (38.0%). It also achieves 68.3% average on τ² Bench (multi-domain task-planning benchmark), above GPT-OSS-120B (65.8%) and GLM-4.5-Air (53.2%). This reflects training emphasis on multi-step agentic workflows in addition to standard reasoning. Try it Use cases Best practices Agentic web research & technical troubleshooting (multi-step reasoning, planning, troubleshooting) Use longer context when needed: the model is designed for long-context workflows (up to 128K context with YaRN-based extrapolation noted). Start from the model’s baseline decoding settings (as shown in the model’s sample usage) and adjust for your task: temperature ~0.8, top_p ~0.95, repetition_penalty ~1.0, and set an explicit max_new_tokens (sample shows 2048). Suggestion (general, not stated verbatim in the sources): For agentic tasks, keep the prompt structured (goal → constraints → tools available → required output format), and ask for a short plan + final answer to reduce wandering. Multilingual (Indic) customer support & content generation (English + 22 Indian languages; native-script / romanized / code-mixed inputs) Be explicit about the language/script you want back (e.g., Hindi in Devanagari vs romanized Hinglish), since training emphasized Indian languages and code-mixed/romanized inputs. Provide in-language examples (a short “good response” example in the target language/script) to anchor tone and terminology. (Suggestion—general best practice; not stated verbatim in sources.) Use the model’s baseline generation settings first (sample decoding params) and then tighten creativity for support use cases (e.g., lower temperature) if you see variability. Getting started You can deploy open-source Hugging Face models directly in Microsoft Foundry by browsing the Hugging Face collection in the Foundry model catalog and deploying to managed endpoints in just a few clicks. Or start from the Hugging Face Hub and choose the "Deploy on Microsoft Foundry" option, which brings you straight into Foundry. Learn how to discover models and deploy them using Microsoft Foundry here: Follow along the Model Mondays series and access the GitHub to stay up to date on the latest Read Hugging Face on Azure docs Learn about one-click deployments from the Hugging Face Hub on Microsoft Foundry Explore models in Microsoft FoundryNow in Foundry: VibeVoice-ASR, MiniMax M2.5, Qwen3.5-9B
This week's Model Mondays edition features two models that have just arrived in Microsoft Foundry: Microsoft's VibeVoice-ASR, a unified speech-to-text model that handles 60-minute audio files in a single pass with built-in speaker diarisation and timestamps, and MiniMaxAI's MiniMax-M2.5, a frontier agentic model that leads on coding and tool-use benchmarks with performance comparable to the strongest proprietary models at a fraction of their cost; and Qwen's Qwen3.5-9B, the largest of the Qwen3.5 Small Series. All three represent a shift toward long-context, multi-step capability: VibeVoice-ASR processes up to an hour of continuous audio without chunking; MiniMax-M2.5 handles complex, multi-phase agentic tasks more efficiently than its predecessor—completing SWE-Bench Verified 37% faster than M2.1 with 20% fewer tool-use rounds; and Qwen3.5-9B brings multimodal reasoning on consumer hardware that outperforms much larger models. Models of the week VibeVoice-ASR Model Specs Parameters / size: ~8.3B Primary task: Automatic Speech Recognition with diarisation and timestamps Why it's interesting 60-minute single-pass with full speaker attribution: VibeVoice-ASR processes up to 60 minutes of continuous audio without chunk-based segmentation—yielding structured JSON output with start/end timestamps, speaker IDs, and transcribed content for each segment. This eliminates the speaker-tracking drift and semantic discontinuities that chunk-based pipelines introduce at segment boundaries. Joint ASR, diarisation, and timestamps in one model: Rather than running separate systems for transcription, speaker separation, and timing, VibeVoice-ASR produces all three outputs in a single forward pass. Users can also inject customized hot words—proper nouns, technical terms, or domain-specific phrases—to improve recognition accuracy on specialized content without fine-tuning. Multilingual with native code-switching: Supports 50+ languages with no explicit language configuration required and handles code-switching within and across utterances natively. This makes it suitable for multilingual meetings and international call center recordings without pre-routing audio by language. Benchmarks: On the Open ASR Leaderboard, VibeVoice-ASR achieves an average WER of 7.77% across 8 English datasets (RTFx 51.80), including 2.20% on LibriSpeech Clean and 2.57% on TED-LIUM. On the MLC-Challenge multi-speaker benchmark: DER 4.28%, cpWER 11.48%, tcpWER 13.02%. Try it Use case What to build Best practices Long-form, multi-speaker transcription for meetings + compliance A transcription service that ingests up to 60 minutes of audio per request and returns structured segments with speaker IDs + start/end timestamps + transcript text (ready for search, summaries, or compliance review). Keep audio un-chunked (single-pass) to preserve speaker coherence and avoid stitching drift; rely on the model’s joint ASR, diarisation, and timestamping so you don’t need separate diarisation/timestamp pipelines or postprocessing. Multilingual + domain-specific transcription (global support, technical reviews) A global transcription workflow for multilingual meetings or call center recordings that outputs “who/when/what,” and supports vocabulary injection for product names, acronyms, and technical terms. Provide customized hot words (names / technical terms) in the request to improve recognition on specialized content; don’t require explicit language configuration—VibeVoice-ASR supports 50+ languages and code-switching, so you can avoid pre-routing audio by language. Read more about the model and try out the playground Microsoft for Hugging Face Spaces to try the model for yourself. MiniMax-M2.5 Model Specs Parameters / size: ~229B (FP8, Mixture of Experts) Primary task: Text generation (agentic coding, tool use, search) Why it's interesting? Leading coding benchmark performance: Scores 80.2% on SWE-Bench Verified and 51.3% on Multi-SWE-Bench across 10+ programming languages (Go, C, C++, TypeScript, Rust, Python, Java, and others). In evaluations across different agent harnesses, M2.5 scores 79.7% on Droid and 76.1% on OpenCode—both ahead of Claude Opus 4.6 (78.9% and 75.9% respectively). The model was trained across 200,000+ real-world coding environments covering the full development lifecycle: system design, environment setup, feature iteration, code review, and testing. Expert-level search and tool use: M2.5 achieves industry-leading performance in BrowseComp, Wide Search, and Real-world Intelligent Search Evaluation (RISE), laying a solid foundation for autonomously handling complex tasks. Professional office work: Achieves a 59.0% average win rate against other mainstream models in financial modeling, Word, and PowerPoint tasks, evaluated via the GDPval-MM framework with pairwise comparison by senior domain professionals (finance, law, social sciences). M2.5 was co-developed with these professionals to incorporate domain-specific tacit knowledge—rather than general instruction-following—into the model's training. Try it Use case What to build Best practices Agentic software engineering Multi‑file code refactors, CI‑gated patch generation, long‑running coding agents working across large repositories Start prompts with a clear architecture or refactor goal. Let the model plan before editing files, keep tool calls sequential, and break large changes into staged tasks to maintain state and coherence across long workflows. Autonomous productivity agents Research assistants, web‑enabled task agents, document and spreadsheet generation workflows Be explicit about intent and expected output format. Decompose complex objectives into smaller steps (search → synthesize → generate), and leverage the model’s long‑context handling for multi‑step reasoning and document creation. With these use cases and best practices in mind, the next step is translating them into a clear, bounded prompt that gives the model a specific goal and the right tools to act. The example below shows how a product or engineering team might frame an automated code review and implementation task, so the model can reason through the work step by step and return results that map directly back to the original requirement: “You're building an automated code review and feature implementation system for a backend engineering team. Deploy MiniMax-M2.5 in Microsoft Foundry with access to your repository's file system tools and test runner. Given a GitHub issue describing a new API endpoint requirement, have the model first write a functional specification decomposing the requirement into sub-tasks, then implement the endpoint across the relevant service files, write unit tests with at least 85% coverage, and return a pull request summary explaining each code change and its relationship to the original requirement. Flag any implementation decisions that deviate from the patterns found in the existing codebase.” Qwen3.5-9B Model Specs Parameters / size: 9B Context length: 262,144 tokens natively; extensible to 1,010,000 tokens Primary task: Image-text-to-text (multimodal reasoning) Why it’s interesting High intelligence density at small sizes: Qwen 3.5 Small models show large reasoning gains relative to parameter count, with the 4B and 9B variants outperforming other sub‑10B models on public reasoning benchmarks. Long‑context by default: Support for up to 262K tokens enables long‑document analysis, codebase review, and multi‑turn workflows without chunking. Native multimodal architecture: Vision is built into the model architecture rather than added via adapters, allowing small models (0.8B, 2B) to handle image‑text tasks efficiently. Open and deployable: Apache‑2.0 licensed models designed for local, edge, or cloud deployment scenarios. Benchmarks AI Model & API Providers Analysis | Artificial Analysis Try it Use case When to use Best‑practice prompt pattern Long‑context reasoning Analyzing full PDFs, long research papers, or large code repositories where chunking would lose context Set a clear goal and scope. Ask the model to summarize key arguments, surface contradictions, or trace decisions across the entire document before producing an output. Lightweight multimodal document understanding OCR‑driven workflows using screenshots, scanned forms, or mixed image‑text inputs Ground the task in the artifact. Instruct the model to first describe what it sees, then extract structured information, then answer follow‑up questions. With these best practices in mind, Qwen 3.5-9B demonstrates how compact, multimodal models can handle complex reasoning tasks without chunking or manual orchestration. The prompt below shows how an operations analyst might use the model to analyze a full report end‑to‑end: "You are assisting an operations analyst. Review the attached PDF report and extracted tables. Identify the three largest cost drivers, explain how they changed quarter‑over‑quarter, and flag any anomalies that would require follow‑up. If information is missing, state what data would be needed." Getting started You can deploy open-source Hugging Face models directly in Microsoft Foundry by browsing the Hugging Face collection in the Foundry model catalog and deploying to managed endpoints in just a few clicks. You can also start from the Hugging Face Hub. First, select any supported model and then choose "Deploy on Microsoft Foundry", which brings you straight into Azure with secure, scalable inference already configured. Learn how to discover models and deploy them using Microsoft Foundry documentation. Follow along the Model Mondays series and access the GitHub to stay up to date on the latest Read Hugging Face on Azure docs Learn about one-click deployments from the Hugging Face Hub on Microsoft Foundry Explore models in Microsoft FoundryBuilding Knowledge-Grounded Conversational AI Agents with Azure Speech Photo Avatars
From Chat to Presence: The Next Step in Conversational AI Chat agents are now embedded across nearly every industry, from customer support on websites to direct integrations inside business applications designed to boost efficiency and productivity. As these agents become more capable and more visible, user expectations are also rising: conversations should feel natural, trustworthy, and engaging. While text‑only chat agents work well for many scenarios, voice‑enabled agents take a meaningful step forward by introducing a clearer persona and a stronger sense of presence, making interactions feel more human and intuitive (see healow Genie success story). In domains such as Retail, Healthcare, Education, and Corporate Training, adding a visual dimension through AI avatars further elevates the experience. Pairing voice with a lifelike visual representation improves inclusiveness, reduces interaction friction, and helps users better contextualize conversations—especially in scenarios that rely on trust, guidance, or repeated engagement. To support these experiences, Microsoft offers two AI avatar options through Azure Speech: Video Avatars, which are generally available and provide full‑ or partial‑body immersive representations, and Photo Avatars, currently in public preview, which deliver a headshot‑style visual well suited for web‑based agents and digital twin scenarios. Both options support custom avatars, enabling organizations to reflect their brand identity rather than relying solely on generic representations (see W2M custom video avatar). Choosing between Video Avatars and Photo Avatars is less about preference and more about intent. Video Avatars offer higher visual fidelity and immersion but require more extensive onboarding, such as high-quality recorded video of an avatar talent. Photo Avatars, by contrast, can be created from a single image, enabling a lighter‑weight onboarding process while still delivering a human‑centered experience. The right choice depends on the desired interaction style, visual presence, and target deployment scenario. What this solution demonstrates In this post, I walk through how to integrate Azure Speech Photo Avatars — powered by Microsoft Research's VASA-1 model — into a knowledge‑grounded conversational AI agent built on Azure AI Search. The goal is to show how voice, visuals, and retrieval‑augmented generation (RAG) can come together to create a more natural and engaging agent experience. The solution exposes a web‑based interface where users can speak naturally to the AI agent using their voice. The agent responds in real time using synthesized speech, while live transcriptions of the conversation are displayed in the UI to improve clarity and accessibility. To help compare different interaction patterns, the sample application supports three modes: 1) Photo Avatar mode, which adds a lifelike visual presence. 2) Video Avatar mode, which provides a more immersive, full‑motion experience. 3) Voice‑only mode, which focuses purely on speech‑to‑speech interaction. Key architectural components An end‑to‑end architecture for the solution is shown in the diagram below. The solution is composed of the following core services and building blocks: Microsoft Foundry — provides the platform for deploying, managing, and accessing the foundation models used by the application. Azure OpenAI — provides the Realtime API for speech‑to‑speech interaction in the voice‑only mode and the Chat Completions API used by backend services for reasoning and conversational responses. gpt‑4.1 — LLM used for reasoning tasks such as deciding when to invoke tool calls and summarizing responses. gpt-realtime-mini — LLM used for speech-to-speech interaction in the Voice-only mode. text‑embedding‑3‑large — LLM used for generating vector embeddings used in retrieval‑augmented generation. Azure Speech — delivers the real‑time speech‑to‑text (STT), text‑to‑speech (TTS), and AI avatars capabilities for both Photo Avatar and Video Avatar experiences. Azure Document Intelligence — extracts structured text, layout, and key information from source documents used to build the knowledge base. Azure AI Search — provides vector‑based retrieval to ground the language model with relevant, context‑aware content. Azure Container Apps — hosts the web UI frontend, backend services, and MCP server within a managed container runtime. Azure Container Apps Environment — defines a secure and isolated boundary for networking, scaling, and observability of the containerized workloads. Azure Container Registry — stores and manages Docker images used by the container applications. How you can try it yourself The complete sample implementation is available in the LiveChat AI Voice Assistant repository, which includes instructions for deploying the solution into your Azure environment. The repository uses Infrastructure as Code (IaC) deployment via Azure Developer CLI (azd) to orchestrate Azure resource provisioning and application deployment. Prerequisites: An Azure subscription with appropriate services and models' quota is required to deploy the solution. Getting the solution up and running in just three simple steps: Clone the repository and navigate to the project git clone https://github.com/mardianto-msft/azure-speech-ai-avatars.git cd azure-speech-ai-avatars Authenticate with Azure azd auth login Initialize and deploy the solution azd up Once deployed, you can access the sample application by opening the frontend service URL in a web browser. To demonstrate knowledge grounding, the sample includes source documents derived from Microsoft’s 2025 Annual Report and Shareholder Letter. These grounding documents can optionally be replaced with your own data, allowing the same architecture to be reused for domain‑specific or enterprise scenarios. When using the provided sample documents, you can ask questions such as: “How much was Microsoft’s net income in 2025?”, “What are Microsoft’s priorities according to the shareholder letter?”, “Who is Microsoft’s CEO?” Bringing Conversational AI Agents to Life This implementation of Azure Speech Photo Avatars serves as a practical starting point for building more engaging, knowledge‑grounded conversational AI agents. By combining voice interaction, visual presence, and retrieval‑augmented generation, Photo Avatars offer a lightweight yet powerful way to make AI agents feel more approachable, trustworthy, and human‑centered — especially in web‑based and enterprise scenarios. From here, the solution can be extended over time with capabilities such as long‑term memory, richer personalization, or more advanced multi‑agent orchestration. Whether used as a reference architecture or as the foundation for a production system, this approach demonstrates how Azure Speech Photo Avatars can help bridge the gap between conversational intelligence and meaningful user experience. By emphasizing accessibility, trust, and human‑centered design, it reflects Microsoft’s broader mission to empower every person and every organization on the planet to achieve more.What’s trending on Hugging Face: PubMedBERT Base Embeddings, Paraphrase Multilingual MiniLM, BGE-M3
The embedding model landscape has evolved beyond one-size-fits-all solutions. Today’s developers navigate a set of deliberate trade‑offs: domain specialization to improve accuracy in vertical applications, multilingual capabilities to support global use cases, and retrieval strategies that optimize performance at scale. Once a model demonstrates strong semantic performance, predictable behavior, and broad community support, it often becomes a trusted reference baseline that developers build around and deploy with confidence. This week, we’re not spotlighting models that are new to Microsoft Foundry. Instead, we’re turning our attention to models that have managed to stay relevant in a rapidly expanding sea of options. This week's Model Monday's edition highlights three Hugging Face models including NeuML's PubMedBERT Base Embeddings for domain-specific medical text understanding, Sentence Transformers' Paraphrase Multilingual MiniLM for lightweight cross-lingual semantic similarity, and BAAI's BGE-M3 for multi-functional long-context retrieval across 100+ languages. Models of the week NeuML: PubMedBERT Base Embeddings Model Specs Parameters / size: 109M Context length: 512 tokens Primary task: Embeddings (medical domain) Why it's interesting Domain-specific performance gains: Fine-tuned on PubMed title-abstract pairs, achieving 95.62% average Pearson correlation across medical benchmarks—outperforming general-purpose models like gte-base (95.37%), bge-base-en-v1.5 (93.78%), and all-MiniLM-L6-v2 (93.46%) on medical literature tasks Production-validated for medical RAG: With 141K downloads and deployment in 30+ medical AI applications, this model demonstrates consistent real-world performance for clinical research, drug discovery, and biomedical semantic search pipelines Built on Microsoft's BiomedNLP foundation: Extends BioMed BERT family with sentence-transformers mean pooling, creating 768-dimensional embeddings optimized for medical literature clustering and retrieval Try it Clinical research sample prompt: Industry specific sample prompt: You're building a clinical decision support system for oncology. Deploy PubMedBERT Base Embeddings in Microsoft Foundry to index 50,000 recent cancer research abstracts from PubMed. A physician queries: "What are the cardiotoxicity risks of combining checkpoint inhibitors with anthracycline chemotherapy in elderly patients?" Embed the query, retrieve the top 10 most semantically similar abstracts using cosine similarity, and return citations with PubMed IDs for evidence-based treatment planning. Sentence Transformers: Paraphrase Multilingual MiniLM L12 v2 Model Specs Parameters / size: 117M Context length: 128 tokens Primary task: Embeddings (multilingual, sentence similarity) Why it's interesting Multilingual adoption: Supports 50+ languages including Arabic, Chinese, Hebrew, Hindi, Japanese, Korean, Russian, Thai, and Vietnamese—with 18.4 million downloads last month demonstrating production-scale validation across global deployments Compact architecture for edge deployment: At 117M parameters producing 384-dimensional embeddings, this model balances multilingual coverage with inference efficiency, making it ideal for resource-constrained environments or high-throughput applications Sentence-BERT foundation: Based on the influential Sentence-BERT paper (Reimers & Gurevych, 2019), using siamese BERT networks with mean pooling to create semantically meaningful sentence embeddings for clustering, paraphrase detection, and cross-lingual search Community-proven versatility: With 299 fine-tuned variants and 100+ Spaces implementations, this model serves as a peer reviewed starting point for multilingual semantic similarity tasks, from customer support ticket routing to cross-lingual document retrieval Try it E-commerce sample prompt: You're building a global customer support platform for an e-commerce company operating in 30 countries. Deploy Paraphrase Multilingual MiniLM in Microsoft Foundry to process incoming support tickets in English, Spanish, French, German, Portuguese, Japanese, and Korean. Embed each ticket as a 384-dimensional vector and cluster by semantic similarity to automatically route issues to specialized teams (payment, shipping, returns, technical). Flag duplicate tickets with cosine similarity > 0.85 to prevent redundant responses. BAAI: BGE-M3 Model Specs Parameters / size: ~560M Context length: 8192 tokens Primary task: Embeddings (multi-functional: dense, sparse, multi-vector) Why it's interesting Three retrieval modes in one model: Uniquely supports dense retrieval (1024-dim embeddings), sparse retrieval (lexical matching like BM25), and multi-vector retrieval (ColBERT-style fine-grained matching)—enabling hybrid search pipelines without maintaining separate models or indexes Exceptional long-context capability: 8192-token context window handles full documents, legal contracts, research papers, and lengthy technical content—validated on MLDR (13-language document retrieval) and NarrativeQA (long-form question answering) benchmarks Multilingual dominance: Outperforms OpenAI embeddings on MIRACL multilingual retrieval across 13+ languages and demonstrates strong zero-shot cross-lingual transfer on MKQA. Try it Legal document search sample prompt: You're building a legal document search system for a multinational law firm. Deploy BGE-M3 in Microsoft Foundry to index 5,000 full-length commercial contracts (average 6,000 tokens each) in English, French, German, and Spanish. A lawyer queries: "Find all force majeure clauses that exclude liability for pandemics or global health emergencies." Use hybrid retrieval: (1) dense embeddings for semantic similarity to capture concept variations like "Act of God" or "unforeseen circumstances", (2) sparse retrieval for exact keyword matches on "force majeure", "pandemic", "health emergency". Combine scores with weighted sum (0.6 dense + 0.4 sparse) and return top 15 contract sections with clause numbers and jurisdiction metadata. Getting started You can deploy open-source Hugging Face models directly in Microsoft Foundry by browsing the Hugging Face collection in the Foundry model catalog and deploying to managed endpoints in just a few clicks. You can also start from the Hugging Face Hub. First, select any supported model and then choose "Deploy on Microsoft Foundry", which brings you straight into Azure with secure, scalable inference already configured. Learn how to discover models and deploy them using Microsoft Foundry documentation. Follow along the Model Mondays series and access the GitHub to stay up to date on the latest Read Hugging Face on Azure docs Learn about one-click deployments from the Hugging Face Hub on Microsoft Foundry Explore models in Microsoft FoundryNow in Foundry: Qwen3-Coder-Next, Qwen3-ASR-1.7B, Z-Image
This week's spotlight features three models from that demonstrate enterprise-grade AI across the full scope of modalities. From low latency coding agents to state-of-the-art multilingual speech recognition and foundation-quality image generation, these models showcase the breadth of innovation happening in open-source AI. Each model balances performance with practical deployment considerations, making them viable for production systems while pushing the boundaries of what's possible in their respective domains. This week's Model Mondays edition highlights Qwen3-Coder-Next, an 80B MoE model that activates only 3B parameters while delivering coding agent capabilities with 256k context; Qwen3-ASR-1.7B, which achieves state-of-the-art accuracy across 52 languages and dialects; and Z-Image from Tongyi-MAI, an undistilled text-to-image foundation model with full Classifier-Free Guidance support for professional creative workflows. Models of the week Qwen: Qwen3-Coder-Next Model Specs Parameters / size: 80B total (3B activated) Context length: 262,144 tokens Primary task: Text generation (coding agents, tool use) Why it's interesting Extreme efficiency: Activates only 3B of 80B parameters while delivering performance comparable to models with 10-20x more active parameters, making advanced coding agents viable for local deployment on consumer hardware Built for agentic workflows: Excels at long-horizon reasoning, complex tool usage, and recovering from execution failures, a critical capability for autonomous development that go beyond simple code completion Benchmarks: Competitive performance with significantly larger models on SWE-bench and coding benchmarks (Technical Report) Try it Use Case Prompt Pattern Code generation with tool use Provide task context, available tools, and execution environment details Long-context refactoring Include full codebase context within 256k window with specific refactoring goals Autonomous debugging Present error logs, stack traces, and relevant code with failure recovery instructions Multi-file code synthesis Describe architecture requirements and file structure expectations Financial services sample prompt: You are a coding agent for a fintech platform. Implement a transaction reconciliation service that processes batches of transactions, detects discrepancies between internal records and bank statements, and generates audit reports. Use the provided database connection tool, logging utility, and alert system. Handle edge cases including partial matches, timing differences, and duplicate transactions. Include unit tests with 90%+ coverage. Qwen: Qwen3-ASR-1.7B Model Specs Parameters / size: 1.7B Context length: 256 tokens (default), configurable up to 4096 Primary task: Automatic speech recognition (multilingual) Why it's interesting All-in-one multilingual capability: Single 1.7B model handles language identification plus speech recognition for 30 languages, 22 Chinese dialects, and English accents from multiple regions—eliminating the need to manage separate models per language Specialized audio versatility: Transcribes not just clean speech but singing voice, songs with background music, and extended audio files, expanding use cases beyond traditional ASR to entertainment and media workflows State-of-the-art accuracy: Outperforms GPT-4o, Gemini-2.5, and Whisper-large-v3 across multiple benchmarks. English: Tedlium 4.50 WER vs 7.69/6.15/6.84; Chinese: WenetSpeech 4.97/5.88 WER vs 15.30/14.43/9.86 (Technical Paper) Language ID included: 97.9% average accuracy across benchmark datasets for automatic language identification, eliminating the need for separate language detection pipelines Try it Use Case Prompt Pattern Multilingual transcription Send audio files via API with automatic language detection Call center analytics Process customer service recordings to extract transcripts and identify languages Content moderation Transcribe user-generated audio content across multiple languages Meeting transcription Convert multilingual meeting recordings to text for documentation Customer support sample prompt: Deploy Qwen3-ASR-1.7B to a Microsoft Foundry endpoint and transcribe multilingual customer service calls. Send audio files via API to automatically detect the language (from 52 supported options including 30 languages and 22 Chinese dialects) and generate accurate transcripts. Process calls from customers speaking English, Spanish, Mandarin, Cantonese, Arabic, French, and other languages without managing separate models per language. Use transcripts for quality assurance, compliance monitoring, and customer sentiment analysis. Tongyi-MAI: Z-Image Model Specs Parameters / size: 6B Context length: N/A (text-to-image) Primary task: Text-to-image generation Why it's interesting Undistilled foundation model: Full-capacity base without distillation preserves complete training signal with Classifier-Free Guidance support (a technique that improves prompt adherence and output quality), enabling complex prompt engineering and negative prompting that distilled models cannot achieve High output diversity: Generates distinct character identities in multi-person scenes with varied compositions, facial features, and lighting, critical for creative applications requiring visual variety rather than consistency Aesthetic versatility: Handles diverse visual styles from hyper-realistic photography to anime and stylized illustrations within a single model, supporting resolutions from 512×512 to 2048×2048 at any aspect ratio with 28-50 inference steps (Technical Paper) Try it Use Case Prompt Pattern Multilingual transcription Send audio files via API with automatic language detection Call center analytics Process customer service recordings to extract transcripts and identify languages Content moderation Transcribe user-generated audio content across multiple languages Meeting transcription Convert multilingual meeting recordings to text for documentation E-commerce sample prompt: Professional product photography of a modern ergonomic office chair in a bright Scandinavian-style home office. Natural window lighting from left, clean white desk with laptop and succulent plant, light oak hardwood floor. Chair positioned at 45-degree angle showing design details. Photorealistic, commercial photography, sharp focus, 85mm lens, f/2.8, soft shadows. Getting started You can deploy open‑source Hugging Face models directly in Microsoft Foundry by browsing the Hugging Face collection in the Foundry model catalog and deploying to managed endpoints in just a few clicks. You can also start from the Hugging Face Hub. First, select any supported model and then choose "Deploy on Microsoft Foundry", which brings you straight into Azure with secure, scalable inference already configured. Learn how to discover models and deploy them using Microsoft Foundry documentation. Follow along the Model Mondays series and access the GitHub to stay up to date on the latest Read Hugging Face on Azure docs Learn about one-click deployments from the Hugging Face Hub on Microsoft Foundry Explore models in Microsoft FoundryWhat is trending in Hugging Face on Microsoft Foundry? Feb, 2, 2026
Open‑source AI is moving fast, with important breakthroughs in reasoning, agentic systems, multimodality, and efficiency emerging every day. Hugging Face has been a leading platform where researchers, startups, and developers share and discover new models. Microsoft Foundry brings these trending Hugging Face models into a production‑ready experience, where developers can explore, evaluate, and deploy them within their Azure environment. Our weekly Model Monday’s series highlights Hugging Face models available in Foundry, focusing on what matters most to developers: why a model is interesting, where it fits, and how to put it to work quickly. This week’s Model Mondays edition highlights three Hugging Face models, including a powerful Mixture-of-Experts model from Z. AI designed for lightweight deployment, Meta’s unified foundation model for image and video segmentation, and MiniMax’s latest open-source agentic model optimized for complex workflows. Models of the week Z.AI’s GLM-4.7-flash Model Basics Model name: zai-org/GLM-4.7-Flash Parameters / size: 30B total -3B active Default settings: 131,072 max new tokens Primary task: Agentic, Reasoning and Coding Why this model matters Why it’s interesting: It utilizes a Mixture-of-Experts (MoE) architecture (30B total parameters and 3B active parameters) to offer a new option for lightweight deployment. It demonstrates strong performance on logic and reasoning benchmarks, outperforming similar sized models like gpt-oss-20b on AIME 25 and GPQA benchmarks. It supports advanced inference features like "Preserved Thinking" mode for multi-turn agentic tasks. Best‑fit use cases: Lightweight local deployment, multi-turn agentic tasks, and logical reasoning applications. What’s notable: From the Foundry catalog, users can deploy on a A100 instance or unsloth/GLM-4.7-Flash-GGUF on a CPU. ource SOTA scores among models of comparable size. Additionally, compared to similarly sized models, GLM-4.7-Flash demonstrates superior frontend and backend development capabilities. Click to see more: https://docs.z.ai Try it Use case Best‑practice prompt pattern Agentic coding (multi‑step repo work, debugging, refactoring) Treat the model as an autonomous coding agent, not a snippet generator. Explicitly require task decomposition and step‑by‑step execution, then a single consolidated result. Long‑context agent workflows (local or low‑cost autonomous agents) Call out long‑horizon consistency and context preservation. Instruct the model to retain earlier assumptions and decisions across turns. Now that you know GLM‑4.7‑Flash works best when you give it a clear goal and let it reason through a bounded task, here’s an example prompt that a product or engineering team might use to identify risks and propose mitigations: You are a software reliability analyst for a mid‑scale SaaS platform. Review recent incident reports, production logs, and customer issues to uncover edge‑case failures outside normal usage (e.g., rare inputs, boundary conditions, timing/concurrency issues, config drift, or unexpected feature interactions). Prioritize low‑frequency, high‑impact risks that standard testing misses. Recommend minimal, low‑cost fixes (validation, guardrails, fallback logic, or documentation). Deliver a concise executive summary with sections: Observed Edge Cases, Root Causes, User Impact, Recommended Lightweight Fixes, and Validation Steps. Meta's Segment Anything 3 (SAM3) Model Basics Model name: facebook/sam3 Parameters / size: 0.9B Primary task: Mask Generation, Promptable Concept Segmentation (PCS) Why this model matters Why it’s interesting: It handles a vastly larger set of open-vocabulary prompts than SAM 2, and unifies image and video segmentation capabilities. It includes a "SAM 3 Tracker" mode that acts as a drop-in replacement for SAM 2 workflows with improved performance. Best‑fit use cases: Open-vocabulary object detection, video object tracking, and automatic mask generation What’s notable: Introduces Promptable Concept Segmentation (PCS), allowing users to find all matching objects (e.g., "dial") via text prompt rather than just single instances. Try it This model enables users to identify specific objects within video footage and isolate them over extended periods. With just one line of code, it is possible to detect multiple similar objects simultaneously. The accompanying GIF demonstrates how SAM3 efficiently highlights players wearing white on the field as they appear and disappear from view. Additional examples are available at the following repository: https://github.com/facebookresearch/sam3/blob/main/assets/player.gif Use case Best‑practice prompt pattern Agentic coding (multi‑step repo work, debugging, refactoring) Treat SAM 3 as a concept detector, not an interactive click tool. Use short, concrete noun‑phrase concept prompts instead of describing the scene or asking questions. Example prompt: “yellow school bus” or “shipping containers”. Avoid verbs or full sentences. Video segmentation + object tracking Specify the same concept prompt once, then apply it across the video sequence. Do not restate the prompt per frame. Let the model maintain identity continuity. Example: “person wearing a red jersey”. Hard‑to‑name or visually subtle objects Use exemplar‑based prompts (image region or box) when text alone is ambiguous. Optionally combine positive and negative exemplars to refine the concept. Avoid over‑constraining with long descriptions. Using the GIF above as a leading example, here is a prompt that shows how SAM 3 turns raw sports footage into structured, reusable data. By identifying and tracking players based on visual concepts like jersey color so that sports leagues can turn tracked data into interactive experiences where automated player identification can relay stats, fun facts, etc when built into a larger application. Here is a prompt that will allow you to start identifying specific players across video: Act as a sports analytics operator analyzing football match footage. Segment and track all football players wearing blue jerseys across the video. Generate pixel‑accurate segmentation masks for each player and assign persistent instance IDs that remain stable during camera movement, zoom, and player occlusion. Exclude referees, opposing team jerseys, sidelines, and crowd. Output frame‑level masks and tracking metadata suitable for overlays, player statistics, and downstream analytics pipelines. MiniMax AI's MiniMax-M2.1 Model Basics Model name: MiniMaxAI/MiniMax-M2.1 Parameters / size: 229B-10B Active Default settings: 200,000 max new tokens Primary task: Agentic and Coding Why this model matters Why it’s interesting: It is optimized for robustness in coding, tool use, and long-horizon planning, outperforming Claude Sonnet 4.5 in multilingual scenarios. It excels in full-stack application development, capable of architecting apps "from zero to one”. Previous coding models focused on Python optimization, M2.1 brings enhanced capabilities in Rust, Java, Golang, C++, Kotlin, Objective-C, TypeScript, JavaScript, and other languages. The model delivers exceptional stability across various coding agent frameworks. Best‑fit use cases: Lightweight local deployment, multi-turn agentic tasks, and logical reasoning applications. What’s notable: The release of open-source weights for M2.1 delivers a massive leap over M2 on software engineering leaderboards. https://www.minimax.io/ Try it Use case Best‑practice prompt pattern End‑to‑end agentic coding (multi‑file edits, run‑fix loops) Treat the model as an autonomous coding agent, not a snippet generator. Explicitly require task decomposition and step‑by‑step execution, then a single consolidated result. Long‑horizon tool‑using agents (shell, browser, Python) Explicitly request stepwise planning and sequential tool use. M2.1’s interleaved thinking and improved instruction‑constraint handling are designed for complex, multi‑step analytical tasks that require evidence tracking and coherent synthesis, not conversational back‑and‑forth. Long‑context reasoning & analysis (large documents / logs) Declare the scope and desired output structure up front. MiniMax‑M2.1 performs best when the objective and final artifact are clear, allowing it to manage long context and maintain coherence. Because MiniMax‑M2.1 is designed to act as a long‑horizon analytical agent, it shines when you give it a clear end goal and let it work through large volumes of information—here’s a prompt a risk or compliance team could use in practice: You are a financial risk analysis agent. Analyze the following transaction logs and compliance policy documents to identify potential regulatory violations and systemic risk patterns. Plan your approach before executing. Work through the data step by step, referencing evidence where relevant. Deliver a final report with the following sections: Key Risk Patterns Identified, Supporting Evidence, Potential Regulatory Impact, Recommended Mitigations. Your response should be a complete, executive-ready report, not a conversational draft. Getting started You can deploy open‑source Hugging Face models directly in Microsoft Foundry by browsing the Hugging Face collection in the Foundry model catalog and deploying to managed endpoints in just a few clicks. You can also start from the Hugging Face Hub. First, select any supported model and then choose "Deploy on Microsoft Foundry", which brings you straight into Azure with secure, scalable inference already configured. Learn how to discover models and deploy them using Microsoft Foundry documentation. Follow along the Model Mondays series and access the GitHub to stay up to date on the latest Read Hugging Face on Azure docs Learn about one-click deployments from the Hugging Face Hub on Microsoft Foundry Explore models in Microsoft Foundry
Introducing Dragon HD Omni: Azure Speech New Voice Type Now in Preview via Microsoft Foundry
Dragon HD Omni is Microsoft Azure Speech’s newest text‑to‑speech generation, delivering over 700 high‑quality voices with enhanced expressiveness, multi‑lingual fluency, and multi‑style control — all through a unified model built in Microsoft Foundry. It removes common developer pain points such as unnatural voice prosody, limited language coverage, and heavy SSML tuning effort. The result is a powerful value proposition: faster integration, richer user experiences, and production‑ready voice output with minimal effort. Azure speech offers a broad range of unique voices for applications like virtual agents, audiobooks, podcasts, and speech-to-speech tasks. Demo video 700+ prebuilt voices Dragon HD Omni offers a range of prebuilt voices with distinct personas and emotions, supporting diverse use cases from agent-based applications to content creation. These voices unlock endless possibilities, empowering users to enhance end-to-end applications. Full update for previous generation voices Dragon HD Omni merges a wide range of prebuilt voices into one, improving contextual adaptation, prosody, expression, and keeping each voice's unique character. This technology delivers more accurate, flexible, and lifelike speech for a variety of uses. Dragon HD Omni raises the standard for natural AI voices across customer service, accessibility, and creative projects, advancing human-computer interaction. You can explore some voices from voice list, such as: "en-US-Ava:DragonHDOmniLatestNeural" "en-US-Andrew:DragonHDOmniLatestNeural" "en-US-Dana:DragonHDOmniLatestNeural" "en-US-Caleb:DragonHDOmniLatestNeural" "zh-CN-Xiaoyue:DragonHDOmniLatestNeural" "zh-CN-Yunqi:DragonHDOmniLatestNeural" "en-US-Phoebe:DragonHDOmniLatestNeural" "en-US-Lewis:DragonHDOmniLatestNeural" They will be available to try directly via Speech Playground - Microsoft Foundry Or, you can use this voice name format by adding the suffix `:DragonHDOmniLatestNeural` to try the Omni version of the given voice via direct SSML call. For example: Previous neural voice Omni version voice name de-DE-ConradNeural de-DE-Conrad:DragonHDOmniLatestNeural AI-Generated Voices Dragon HD Omni now features nearly 300 brand‑new AI‑generated voices, carefully designed to deliver an unprecedented range of vocal diversity. These voices aren’t just more of the same — they’re built to give you choice, flexibility, and creative control. With variations across: Gender – male, female, and non‑binary options Age – youthful, mature, and senior tones Pitch & tone – from warm and friendly to authoritative and professional This expanded library means you can: Personalize experiences for different audiences, whether you’re building an educational app, a customer support bot, or a storytelling platform. Strengthen brand identity by selecting voices that reflect your company’s personality and values. Increase inclusivity with diverse vocal styles that resonate across cultures and communities. Unlock creativity by experimenting with unique voice personalities for podcasts, games, or immersive experiences. Speaker name – Description Sample en-us-graphiterhodium - A bold and dramatic male voice en-us-olivepoivre - An adult female voice that is calm and soothing. Check the full Dragon HD Omni voice list at here. Styles control Standard Azure voices have limited styles due to extensive tuning requirements. The Dragon HD Omni introduces automatic style prediction using natural language descriptions, enabling advanced customization, broader style support, reduced cost, and improved expressiveness. In the initial release, styles will launch for en-US-Ava and en-US-Andrew. Supported styles angry, chill surfer, confused, curious, determined, disgusted, embarrassed, emo teenager, empathetic, encouraging, excited, fearful, friendly, grateful, joyful, mad scientist, meditative, narration, neutral, new yorker, news, reflective, regretful, relieved, sad, santa, shy, soft voice, surprised Note that style result will be strongly influenced by the input content. SSML example <speak version="1.0" xmlns="http://www.w3.org/2001/10/synthesis" xmlns:mstts="http://www.w3.org/2001/mstts" xml:lang="en-US"> <voice name="en-us-ava:DragonHDOmniLatestNeural"> <mstts:express-as style="cheerful"> Wow! What an amazing day! I feel so full of energy, and everything around me seems brighter. My voice is bubbling with excitement, and I can’t stop smiling. I’m ready to take on anything that comes my way—let’s celebrate this wonderful moment together! </mstts:express-as> </voice> </speak> Multilingual and Accents All Dragon HD Omni voices support multiple languages, with the capability that can automatically predicting and generating output based on the input text. Additionally, you may utilize the tag to adjust speaking languages and accents, such as fr-FR for French, de-DE for German, etc. For a comprehensive list of supported languages and their associated syntax and attributes, please refer to the lang element. SSML example <speak version="1.0" xmlns="http://www.w3.org/2001/10/synthesis" xmlns:mstts="http://www.w3.org/2001/mstts" xml:lang="en-US"><voice name="en-us-ava:Dragon HD OmniLatestNeural"><lang xml:lang="fr-FR"> Bonjour ! Ce matin, j’ai pris un café au jardin du Luxembourg. Il faisait frais, mais très agréable. Ensuite, j’ai acheté une baguette et quelques macarons. Paris est vraiment charmant.</lang> </voice> </speak> Word Boundary Event Support Dragon HD Omni supports the word boundary event, which allows developers to track the precise timing of each word as it is spoken. This feature is essential for applications requiring word-level synchronization, such as karaoke, real-time captioning, or interactive voice experiences. When the event fires, it provides: Text: The word spoken AudioOffset: The time offset in the audio stream (milliseconds) TextOffset: The position of the word in the input text Example: Python Sample Using Wordboundary Event in Azure Speech SDK import azure.cognitiveservices.speech as speechsdk def word_boundary_cb(evt): print(f"Word: '{evt.text}', AudioOffset: {evt.audio_offset / 10000}ms, TextOffset: {evt.text_offset}") speech_config = speechsdk.SpeechConfig(subscription="YourSubscriptionKey", region="YourServiceRegion") synthesizer = speechsdk.SpeechSynthesizer(speech_config=speech_config) synthesizer.synthesis_word_boundary.connect(word_boundary_cb) ssml = """ <speak version="1.0" xmlns="http://www.w3.org/2001/10/synthesis" xmlns:mstts="http://www.w3.org/2001/mstts" xml:lang="en-US"> <voice name="en-us-ava:DragonHDOmniLatestNeural"> Hello Azure, welcome to Dragon HD Omni! </voice> </speak> """ result = synthesizer.speak_ssml_async(ssml).get() Sample Output: Word: 'Hello', AudioOffset: 110.0ms, TextOffset: 182 Word: 'Azure', AudioOffset: 590.0ms, TextOffset: 188 Word: ',', AudioOffset: 1110.0ms, TextOffset: 193 Word: 'welcome', AudioOffset: 1270.0ms, TextOffset: 195 Word: 'to', AudioOffset: 1750.0ms, TextOffset: 203 Word: 'Dragon HD Omni', AudioOffset: 1910.0ms, TextOffset: 206 Word: '!', AudioOffset: 2750.0ms, TextOffset: 216 Parameters Dragon HD Omni supports advanced parameter tuning to help you customize voice output for different scenarios. This guide explains each parameter in simple terms and provides recommendations for adjusting them based on your goals. Overview Parameter Default Range Purpose temperature 0.7 0.3 – 1.0 Controls creativity vs. stability top_p 0.7 0.3 – 1.0 Filters output for diversity top_k 22 1 – 50 Limits number of options considered cfg_scale 1.4 1.0 – 2.0 Adjusts relevance and speech speed Tuning for Expressiveness vs. Stability Higher values for temperature, top_p, and top_k result in more expressive, emotionally varied speech. Lower values produce more stable and predictable output. Recommendation: To increase expressiveness, raise all three parameters together. Keep top_p equal to temperature for best results. Tuning for Speed and Contextual Relevance cfg_scale affects how quickly the voice speaks and how well it aligns with the context. Higher values (e.g., 1.8–2.0): faster speech, stronger contextual relevance. Lower values (e.g., 1.0–1.2): slower speech, less contextual alignment. Suggested Tuning Strategies Goal Suggested Adjustment More expressive Increase temperature, top_p, and top_k together More stable Lower temperature first, then adjust top_p if needed Faster & relevant Increase cfg_scale Slower & neutral Decrease cfg_scale The following table describes the usage of the parameters above: Single parameter: <speak version="1.0" xmlns="http://www.w3.org/2001/10/synthesis" xmlns:mstts="http://www.w3.org/2001/mstts" xml:lang="en-US"> <voice name="en-us-ava:Dragon HD OmniLatestNeural" parameters="top_p=0.8"> Hello Azure! </voice> </speak> Multiple parameters: <speak version="1.0" xmlns="http://www.w3.org/2001/10/synthesis" xmlns:mstts="http://www.w3.org/2001/mstts" xml:lang="en-US"> <voice name="en-us-ava:Dragon HD OmniLatestNeural" parameters="top_p=0.8;top_k=22;temperature=0.7;cfg_scale=1.2"> Hello Azure! Hello Azure! </voice> </speak> Get Started In our ongoing journey to enhance multilingual capabilities in text to speech (TTS) technology, we strive to deliver the best voices to empower your applications. Our voices are designed to be incredibly adaptive, seamlessly switching between languages based on the text input. They deliver natural-sounding speech with precise pronunciation and prosody, making them invaluable for applications like language learning, travel guidance, and international business communication. Microsoft offers an extensive portfolio of over 600 neural voices, covering more than 150 languages and locales. These TTS voices can quickly add read-aloud functionality for a more accessible app design or provide a voice to chatbots, elevating the conversational experience for users. With the Custom Neural Voice capability, businesses can also create unique and distinctive brand voices effortlessly. With these advancements, we continue to push the boundaries of what’s possible in TTS technology, ensuring that our users have access to the most versatile, high-quality voices for their needs. For more information Try our demo to listen to existing neural voices Add Text to speech to your apps today Apply for access to Custom Neural Voice Join Discord to collaborate and share feedback Contact us ttsvoicefeedback@microsoft.com
