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MICROSOFT OPEN SOURCED A 7B PARAMETER MODEL THAT TRANSCRIBES 60 MINUTES OF AUDIO IN A SINGLE PASS and it's completely free VIBEVOICE ASR no chunking, no context loss, full speaker diarization baked in not just speech to text..not a basic wrapper who spoke, when they spoke, exactly what they... show more

Rahul

144,231 subscribers

1,375,300 次观看 • 5 个月前 •via X (Twitter)

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32 条评论

Seth 的头像

Seth5 个月前

MIT means free code, not free compute. Self-hosting a 7B ASR means a GPU instance running 24/7. That's almost certainly more $/mo than a Deepgram bill. There's no Microsoft-hosted free API tier here.

Stef 的头像

Stef5 个月前

“Free” Who owns the data now? Who can parse and use that content now? Who can leverage that content now? Yeah… “free” is always a Trojan horse.

timdrchslr 的头像

timdrchslr5 个月前

Use groq with whisper large v3 turbo for only 0,04$ per hour, hours of audio transcribed in a couple of seconds!!

Nomad 的头像

Nomad5 个月前

Diarization baked in is cool, but 7B parameters kinda defeats the point. You can’t roll this out on device. Parakeet at 0.6b or Granite 1b are a better fit for os local

EventsLooped 的头像

EventsLooped5 个月前

Whisper models are also free to run on your machine

VC Intern 的头像

VC Intern5 个月前

we’re entering the phase where: speech becomes a solved layer the value won’t be transcription it’ll be what you do with it after

Alpha Batcher 的头像

Alpha Batcher5 个月前

i love free open sourced models >3 applause to Microsoft

Julian Goldie SEO 的头像

Julian Goldie SEO5 个月前

that’s actually a big drop

Sanchit 的头像

Sanchit5 个月前

oh wow i had been using elevenlabs API inside of kilo for some work but it was slightly costly. I am going to try this out. 7B is not a lot to run locally

tang | AI Product Maker 的头像

tang | AI Product Maker5 个月前

the 'no chunking' part is the real unlock. tested 3 ASR pipelines for a video workflow last quarter and chunked diarization always lost ~6-8% speaker accuracy when voices swapped at the chunk boundary. how does VibeVoice deal with overlapping speakers in the same window?

Quadcode AI 的头像

Quadcode AI5 个月前

funny how every week there's a revolutonary asr model that supposedly fixes everything at once if it actually nails long form audio and diarization cleanly then thats genuinely impressive but claims like that usually age weirdly

juaniam 的头像

juaniam5 个月前

Or you can use typrapp with whisper large v3 turbo running locally for free and a nice ui on top

flamz 的头像

flamz5 个月前

Nah, they are data farming peoples voices and you all are willingly giving it to them.

Layton Gott 的头像

Layton Gott5 个月前

It's not completely free...

Sebastian Buzdugan 的头像

Sebastian Buzdugan5 个月前

single pass is cool but diarization quality will decide if this actually ships

Max Bourke 的头像

Max Bourke5 个月前

Hmm single pass sounds good but 7b is pretty big for local transcription… be curious to see if quantized versions perform well.

Victor Tavernari  的头像

Victor Tavernari 5 个月前

@perssua @lucas_montano

Eric 的头像

Eric4 个月前

ignoring the caps lock hype, the actual signal here isn't the 60 minute context window. it's the native diarization.

CODIFY 的头像

CODIFY5 个月前

That’s ground-breaking!

Ben Dickey 的头像

Ben Dickey5 个月前

Wow,

Cyril Gupta 的头像

Cyril Gupta5 个月前

Microsoft just collapsed half the ASR stack. One-pass transcription + built-in diarization = no chunking, no glue code. Paired with Ollama → fully local, private voice pipelines. Free isn’t the headline. Owning the stack is.

Goldengmgm🐸 Яebel 🪖 的头像

Goldengmgm🐸 Яebel 🪖5 个月前

wen @AskVenice

Maxwell de Melo 的头像

Maxwell de Melo5 个月前

Nice!

Facundo Campo · Dev 的头像

Facundo Campo · Dev5 个月前

Es un cambio de rumbo!

DiffusionTales 的头像

DiffusionTales5 个月前

I use SubtitleEdit and use the audio to text (whisper).

Adam Jaworski 的头像

Adam Jaworski5 个月前

is it any good?

AIGoldenfinger 的头像

AIGoldenfinger5 个月前

17GB ASR Speech-to-Text model? Are you kidding me? I will run a Film Production Pipelines on 12GB GPU, that would generate the Images, The videos and also the Conversations And also the Music (the score). Microsoft Speech-to-text... ROFL - Get Out Of Here. Go fix your Windows MS

⭕💡 的头像

⭕💡5 个月前

When you say 🆓 what does it mean?

Terry Nunn 的头像

Terry Nunn5 个月前

Don’t just open source the weights, host it for us for free! No one normal person can afford the GPUs needed for this.

Tabish Gillani 的头像

Tabish Gillani5 个月前

Is it available on

Rey Neill 的头像

Rey Neill5 个月前

Hm

AI's Nest 的头像

AI's Nest5 个月前

We are entering an era where the most successful researchers won't be the best coders, but the best "Product Managers" of their own ideas

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🚨 JUST IN: MICROSOFT just open sourced a VOICE AI THAT TRANSCRIBES 60 MINUTES OF AUDIO in a single pass. 100% FREE. It knows who spoke. It knows when they spoke. It knows exactly what they said. All in one shot. No chunking. No context loss. It's called VibeVoice. Not a transcription tool. Not a basic speech to text wrapper. A frontier voice AI family with ASR, TTS, and real time streaming. All open source. All free. Here's what it actually does 👇 VibeVoice ASR - Speech Recognition: → Processes 60 minutes of continuous audio in a single pass → Never slices audio into chunks so global context is never lost → Identifies WHO spoke, WHEN they spoke and WHAT they said simultaneously → Supports customized hotwords for domain specific accuracy → Works in 50+ languages natively → Already adopted by Hugging Face Transformers library → Already being built on by the open source community BY PEOPLE WHO HAD NO IDEA THIS LEVEL OF ACCURACY WAS ALREADY FREE. VibeVoice TTS - Text to Speech: → Generates up to 90 minutes of speech in a single pass → Supports up to 4 distinct speakers in one conversation → Natural turn taking and speaker consistency throughout → Expressive speech that captures emotional nuances → Supports English, Chinese and multiple other languages VibeVoice Realtime - Streaming TTS: → Only 300 millisecond first audible latency → Streams text input in real time → 0.5B parameters so it actually deploys anywhere → Robust long form generation up to 10 minutes → Lightweight enough for production use today The core innovation nobody is talking about: Most voice AI models slice long audio into short chunks. Every time they slice, they lose context. Speaker tracking breaks. Semantic coherence breaks. Accuracy drops. VibeVoice uses continuous speech tokenizers running at an ultra low frame rate of 7.5 Hz. This preserves audio fidelity while dramatically boosting computational efficiency. The entire 60 minutes stays in context. Nothing gets lost. Nobody gets misidentified. The numbers: → VibeVoice ASR 7B - available now on Hugging Face → VibeVoice Realtime 0.5B - try it on Colab right now → 50+ supported languages → 11 distinct English voice styles → 9 multilingual speaker voices → Already integrated into Hugging Face Transformers → Finetuning code now available The wildest part? A voice powered input method called Vibing just built itself on top of VibeVoice ASR. Available on macOS and Windows right now. The open source community is already shipping products on top of this. 100% Open Source. Free to use. Free to fine tune. Free to build on. 🔖 Save this before your competitors find it first. 👇

🚨 JUST IN: MICROSOFT just open sourced a VOICE AI THAT TRANSCRIBES 60 MINUTES OF AUDIO in a single pass. 100% FREE. It knows who spoke. It knows when they spoke. It knows exactly what they said. All in one shot. No chunking. No context loss. It's called VibeVoice. Not a transcription tool. Not a basic speech to text wrapper. A frontier voice AI family with ASR, TTS, and real time streaming. All open source. All free. Here's what it actually does 👇 VibeVoice ASR - Speech Recognition: → Processes 60 minutes of continuous audio in a single pass → Never slices audio into chunks so global context is never lost → Identifies WHO spoke, WHEN they spoke and WHAT they said simultaneously → Supports customized hotwords for domain specific accuracy → Works in 50+ languages natively → Already adopted by Hugging Face Transformers library → Already being built on by the open source community BY PEOPLE WHO HAD NO IDEA THIS LEVEL OF ACCURACY WAS ALREADY FREE. VibeVoice TTS - Text to Speech: → Generates up to 90 minutes of speech in a single pass → Supports up to 4 distinct speakers in one conversation → Natural turn taking and speaker consistency throughout → Expressive speech that captures emotional nuances → Supports English, Chinese and multiple other languages VibeVoice Realtime - Streaming TTS: → Only 300 millisecond first audible latency → Streams text input in real time → 0.5B parameters so it actually deploys anywhere → Robust long form generation up to 10 minutes → Lightweight enough for production use today The core innovation nobody is talking about: Most voice AI models slice long audio into short chunks. Every time they slice, they lose context. Speaker tracking breaks. Semantic coherence breaks. Accuracy drops. VibeVoice uses continuous speech tokenizers running at an ultra low frame rate of 7.5 Hz. This preserves audio fidelity while dramatically boosting computational efficiency. The entire 60 minutes stays in context. Nothing gets lost. Nobody gets misidentified. The numbers: → VibeVoice ASR 7B - available now on Hugging Face → VibeVoice Realtime 0.5B - try it on Colab right now → 50+ supported languages → 11 distinct English voice styles → 9 multilingual speaker voices → Already integrated into Hugging Face Transformers → Finetuning code now available The wildest part? A voice powered input method called Vibing just built itself on top of VibeVoice ASR. Available on macOS and Windows right now. The open source community is already shipping products on top of this. 100% Open Source. Free to use. Free to fine tune. Free to build on. 🔖 Save this before your competitors find it first. 👇

Kanika

221,715 次观看 • 5 个月前

Microsoft did it again! Speech AI models have a major limitation. They slice long recordings into tiny chunks, lose track of who's speaking, and forget all context halfway through. This is exactly what Microsoft's VibeVoice solves. It's an open-source family of frontier voice AI models for both speech recognition and speech generation. Here's what it can do: > VibeVoice-ASR processes up to 60 minutes of audio in a single pass. No chunking. It outputs structured transcriptions with who spoke, when they spoke, and what they said. > You can feed it custom hotwords like names, technical jargon, or domain-specific terms. The model uses them to significantly improve accuracy on specialized content. > VibeVoice-TTS generates up to 90 minutes of multi-speaker speech with up to 4 distinct speakers. Natural turn-taking, emotional expression, all in one pass. > VibeVoice-Realtime is a 0.5B streaming TTS model with ~300ms first-audio latency. Small enough to deploy practically anywhere. All of this is powered by continuous speech tokenizers running at just 7.5 Hz. This ultra-low frame rate preserves audio quality while making long sequences computationally feasible. I have shared the link to the GitHub repo in the replies!

Microsoft did it again! Speech AI models have a major limitation. They slice long recordings into tiny chunks, lose track of who's speaking, and forget all context halfway through. This is exactly what Microsoft's VibeVoice solves. It's an open-source family of frontier voice AI models for both speech recognition and speech generation. Here's what it can do: > VibeVoice-ASR processes up to 60 minutes of audio in a single pass. No chunking. It outputs structured transcriptions with who spoke, when they spoke, and what they said. > You can feed it custom hotwords like names, technical jargon, or domain-specific terms. The model uses them to significantly improve accuracy on specialized content. > VibeVoice-TTS generates up to 90 minutes of multi-speaker speech with up to 4 distinct speakers. Natural turn-taking, emotional expression, all in one pass. > VibeVoice-Realtime is a 0.5B streaming TTS model with ~300ms first-audio latency. Small enough to deploy practically anywhere. All of this is powered by continuous speech tokenizers running at just 7.5 Hz. This ultra-low frame rate preserves audio quality while making long sequences computationally feasible. I have shared the link to the GitHub repo in the replies!

Akshay 🚀

45,256 次观看 • 6 个月前

Microsoft just killed voice subscriptions. 🤯 They quietly open-sourced VibeVoice, and it’s a total industry disruptor. Transcribe hour-long meetings or generate 90 minutes of natural, multi-speaker speech, all locally on your hardware for free. It handles 50+ languages and complex speaker tracking without the monthly fees. The era of expensive voice apps is officially over. I share updates like these in my free AI community on WhatsApp. Join here

Microsoft just killed voice subscriptions. 🤯 They quietly open-sourced VibeVoice, and it’s a total industry disruptor. Transcribe hour-long meetings or generate 90 minutes of natural, multi-speaker speech, all locally on your hardware for free. It handles 50+ languages and complex speaker tracking without the monthly fees. The era of expensive voice apps is officially over. I share updates like these in my free AI community on WhatsApp. Join here

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220,789 次观看 • 10 个月前

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30,416 次观看 • 2 个月前

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21,609 次观看 • 4 个月前

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CyrilXBT

37,847 次观看 • 5 个月前

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Andy Jassy

155,905 次观看 • 1 年前

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i found a website that gives you the best AI models for free, every single day 😳 claude, gpt, gemini, deepseek and 10+ more, all in one place. no subscription, no credit card. it's called poe. you get free credits every day, and they reset every 24 hours so you never really run out. what you can use for free: - claude, for writing and thinking through hard stuff - gpt, for everyday questions and chat - gemini, for long documents and images - deepseek, for tricky problems - and 10+ more, all in one login how to start (2 minutes): > 1. go to (or get the app on your phone or mac) > 2. sign up with google or email, no card needed > 3. you land straight in the chat, free credits already there > 4. pick a model on the left and start typing > 5. switch models anytime, the credits refill every day one simple tip to make your credits last: use the small fast models for easy stuff, and save the big ones for when you really need them. don't waste a powerful model on a yes or no question. what this saves you: - chatgpt plus: $20/mo - claude pro: $20/mo - gemini advanced: $20/mo that's $60 a month for three apps that mostly do the same thing. poe gives you all of them in one place, for free. one login. every top model. zero dollars. bookmark this so you stop paying for AI you can get for free

Isra

58,018 次观看 • 3 个月前

OpenAI's S2S preview is polished but it still thinks in steps. Speech → text → model → text → speech. That's not how humans converse. Introducing Hydra. A native speech-to-speech model that doesn't wait for turn-taking, doesn't flatten emotion into text, and doesn't break when you interrupt it mid-sentence. Hydra reasons asynchronously, speaks and listens simultaneously, and preserves emotion because it never leaves the audio domain. It's still in beta, but the shift is obvious. If you want early access, the link is in the comments. Here's a preview of what that looks like -

OpenAI's S2S preview is polished but it still thinks in steps. Speech → text → model → text → speech. That's not how humans converse. Introducing Hydra. A native speech-to-speech model that doesn't wait for turn-taking, doesn't flatten emotion into text, and doesn't break when you interrupt it mid-sentence. Hydra reasons asynchronously, speaks and listens simultaneously, and preserves emotion because it never leaves the audio domain. It's still in beta, but the shift is obvious. If you want early access, the link is in the comments. Here's a preview of what that looks like -

Sudarshan Kamath

329,378 次观看 • 6 个月前

💥Bill Maher SCHOOLS Canadian Comedian Tom Green on why Canada *DOESN'T* have free speech like America🇺🇸 GREEN: "We have freedom of speech … There are hate speech laws in Canada — so there are things you can't say ... Who would want to say that sh*t anyways?" MAHER: "That's not what free speech means … You don’t know what free speech is." "The Supreme Court ruled in our country that the Nazis could march in Skokie, Illinois — which they were marching in because it is a community of a lot of Holocaust survivors ... That is what free speech is about."

💥Bill Maher SCHOOLS Canadian Comedian Tom Green on why Canada DOESN'T have free speech like America🇺🇸 GREEN: "We have freedom of speech … There are hate speech laws in Canada — so there are things you can't say ... Who would want to say that sh*t anyways?" MAHER: "That's not what free speech means … You don’t know what free speech is." "The Supreme Court ruled in our country that the Nazis could march in Skokie, Illinois — which they were marching in because it is a community of a lot of Holocaust survivors ... That is what free speech is about."

Jason Cohen 🇺🇸

1,443,938 次观看 • 1 年前

NVIDIA just removed the biggest friction point in Voice AI 🤯 They've open-sourced PersonaPlex-7B, a full-duplex conversational model that can listen and speak at the same time. Instead of waiting for you to finish talking, it uses a dual-stream architecture to process and respond in real-time. 100% Open-Source and Free.

NVIDIA just removed the biggest friction point in Voice AI 🤯 They've open-sourced PersonaPlex-7B, a full-duplex conversational model that can listen and speak at the same time. Instead of waiting for you to finish talking, it uses a dual-stream architecture to process and respond in real-time. 100% Open-Source and Free.

Simplifying AI

278,970 次观看 • 8 个月前

Kimi K3 dropped 3 days ago and i already used it to build my own PUBG the model that just took the number one coding spot on earth and it runs on a free tier no team no engine i just fed it free open models and assets that are already public and told it what i wanted it built the entire game itself the shooting the recoil the hit detection all of it came straight out of the model i barely wrote a line what youre watching here is the deathmatch mode and its already live battle royale is next im deep in it and the graphics get reworked right after i know it looks raw right now i wont pretend otherwise but the endgame is a build that runs perfect on phones and the moment its on mobile you drop ads in and it earns on its own the part that still gets me is my whole cost for this was a single subscription

Kimi K3 dropped 3 days ago and i already used it to build my own PUBG the model that just took the number one coding spot on earth and it runs on a free tier no team no engine i just fed it free open models and assets that are already public and told it what i wanted it built the entire game itself the shooting the recoil the hit detection all of it came straight out of the model i barely wrote a line what youre watching here is the deathmatch mode and its already live battle royale is next im deep in it and the graphics get reworked right after i know it looks raw right now i wont pretend otherwise but the endgame is a build that runs perfect on phones and the moment its on mobile you drop ads in and it earns on its own the part that still gets me is my whole cost for this was a single subscription

savip.

68,984 次观看 • 2 个月前

You can now generate real-time speech that sounds conversational. Microsoft just open-sourced VibeVoice, a real-time text-to-speech system with ~300 ms first audio latency and streaming input. It handles long conversations without falling apart. 𝗧𝗵𝗶𝘀 𝗺𝗼𝗱𝗲𝗹 𝗴𝗲𝗻𝗲𝗿𝗮𝘁𝗲𝘀 𝗹𝗼𝗻𝗴, 𝗺𝘂𝗹𝘁𝗶-𝘀𝗽𝗲𝗮𝗸𝗲𝗿 𝘀𝗽𝗲𝗲𝗰𝗵. It produces up to 90 minutes of audio. It supports up to four distinct speakers. Turn-taking stays consistent over long sessions. 𝗜𝘁 𝘄𝗼𝗿𝗸𝘀 𝗯𝘆 𝗿𝗲𝗱𝘂𝗰𝗶𝗻𝗴 𝘁𝗶𝗺𝗲 𝗿𝗲𝘀𝗼𝗹𝘂𝘁𝗶𝗼𝗻. Audio compresses into semantic and acoustic tokens. They run at 7.5 Hz instead of frame-level audio. A language model predicts structure. A diffusion head restores acoustic detail. 𝗜𝘁 𝗮𝗹𝗹𝗼𝘄𝘀 𝗹𝗼𝘄-𝗹𝗮𝘁𝗲𝗻𝗰𝘆 𝘀𝘁𝗿𝗲𝗮𝗺𝗶𝗻𝗴 𝗮𝘂𝗱𝗶𝗼. The real-time variant streams text incrementally. First speech arrives in ~300 ms. A WebSocket demo shows live generation. The code is MIT-licensed and research-only. The repo already passed 20k GitHub stars.

You can now generate real-time speech that sounds conversational. Microsoft just open-sourced VibeVoice, a real-time text-to-speech system with ~300 ms first audio latency and streaming input. It handles long conversations without falling apart. 𝗧𝗵𝗶𝘀 𝗺𝗼𝗱𝗲𝗹 𝗴𝗲𝗻𝗲𝗿𝗮𝘁𝗲𝘀 𝗹𝗼𝗻𝗴, 𝗺𝘂𝗹𝘁𝗶-𝘀𝗽𝗲𝗮𝗸𝗲𝗿 𝘀𝗽𝗲𝗲𝗰𝗵. It produces up to 90 minutes of audio. It supports up to four distinct speakers. Turn-taking stays consistent over long sessions. 𝗜𝘁 𝘄𝗼𝗿𝗸𝘀 𝗯𝘆 𝗿𝗲𝗱𝘂𝗰𝗶𝗻𝗴 𝘁𝗶𝗺𝗲 𝗿𝗲𝘀𝗼𝗹𝘂𝘁𝗶𝗼𝗻. Audio compresses into semantic and acoustic tokens. They run at 7.5 Hz instead of frame-level audio. A language model predicts structure. A diffusion head restores acoustic detail. 𝗜𝘁 𝗮𝗹𝗹𝗼𝘄𝘀 𝗹𝗼𝘄-𝗹𝗮𝘁𝗲𝗻𝗰𝘆 𝘀𝘁𝗿𝗲𝗮𝗺𝗶𝗻𝗴 𝗮𝘂𝗱𝗶𝗼. The real-time variant streams text incrementally. First speech arrives in ~300 ms. A WebSocket demo shows live generation. The code is MIT-licensed and research-only. The repo already passed 20k GitHub stars.

Lior Alexander

61,122 次观看 • 8 个月前

NVIDIA just removed one of the biggest friction points in Voice AI. PersonaPlex-7B is an open-source, full-duplex conversational model. Free, open source (MIT), with open model weights on Hugging Face 🤗 Links to repo and weights in 🧵↓ The traditional ASR → LLM → TTS pipeline forces rigid turn-taking. It’s efficient, but it never feels natural. PersonaPlex-7B changes that. This NVIDIA model can listen and speak at the same time. It runs directly on continuous audio tokens with a dual-stream transformer, generating text and audio in parallel instead of passing control between components. That unlocks: → instant back-channel responses → interruptions that feel human → real conversational rhythm Persona control is fully zero-shot! If you’re building low-latency assistants or support agents, this is a big step forward 🔥

NVIDIA just removed one of the biggest friction points in Voice AI. PersonaPlex-7B is an open-source, full-duplex conversational model. Free, open source (MIT), with open model weights on Hugging Face 🤗 Links to repo and weights in 🧵↓ The traditional ASR → LLM → TTS pipeline forces rigid turn-taking. It’s efficient, but it never feels natural. PersonaPlex-7B changes that. This NVIDIA model can listen and speak at the same time. It runs directly on continuous audio tokens with a dual-stream transformer, generating text and audio in parallel instead of passing control between components. That unlocks: → instant back-channel responses → interruptions that feel human → real conversational rhythm Persona control is fully zero-shot! If you’re building low-latency assistants or support agents, this is a big step forward 🔥

Charly Wargnier ♨️

566,178 次观看 • 8 个月前