Three systems are usually used to create most production voice stacks. A model is used to transcribe, another separates the speakers and finally a detector determines whether or not the speaker has stopped speaking. With each handoff, latency is added and there’s a new mode of failure.
Muse Voice TranscribeMeta Superintelligence Labs announced this week that it has combined these three tasks into one auto-regressive model. Meta claims it is its first model of real-time sound perception. The model performs ASR streaming, speaker diarization of 20+ speakers and endpointing all in one go, without any post-processing.
Is it deployable? The API is hosted. It’s live on the Meta Model API As well as muse-voice-transcribe-1.0 Meta AI Mac and PC already has dictation powered by this technology, which costs $3.00 for 1,000 audio minutes (0.18% per hour). Muse Code. There are no weights available, therefore there is not a self-hosted route.
The foundation for streaming ASR
Muse Voice Transcribe comes from Muse Spark’s multimodal family. Audio comes in 80ms pieces at 12.5. Each chunk becomes a soft token.
It makes a binary selection after every chunk. Either it predicts or it doesn’t. The model can either emit a sound token or keep listening. If the model is able to predict . That token is then replaced by the audio data in the input. The stream will end when the audio is finished. When the token is placed, it flushes out all of the remaining audio without needing to request more.
The alignment of listening and writing is done by the same decoder, therefore there’s no need for a separate stage.
Adaptive delay with RL
It also determines how much context audio is behind each phrase, because the model has control over when it listens. Meta calls that gap ‘delay.’ The longer the delay, the more accurate and latency-increasing transcript will be.
Meta trains the trade-off instead of fixing it. Reinforcement Learning combines word error rates and delay rewards multiplicatively to produce a policy which varies the delay per word based on difficulty. Meta reports that the new model, based on time taken to complete the transcription, is ahead of previous systems such as Soniox Cartesia and ElevenLabs.
More tokens for Diarization, Endpointing and other Tokens
Meta has not added a second speaker-attribution model. It added tokens for special purposes to the stream.
For diarization, a A token indicates a speaker that could be switched, as well as a The tag identifies a speaker. Turn token is fired as soon as switchable, but speaker tag will be delayed until the end of chunk. One speaker’s audio can be broken up into multiple segments which resolve all to the same tag.
The endpointing of the arrow Marks the beginning of speech The user can mark the exact point that they finished. The ASR stream is used to train both tasks, with extra rewards on top.
Capabilities
Model was tested on more than 70 languages. 25 of them were extensively checked and are recommended for launch. The model is able to switch code both in a sentence as well as between sentences. This can be important for multilingual speakers that mix up languages during a clause. Language, keyword and context biasing can improve accuracy.
A practical difference is the ability to manage long contexts. Meta claims that the model supports native audio inputs exceeding an hour, and up to 20 speakers without any post-processing required.
Benchmarks
Meta is ranked first in Artificial Analysis of streaming speech-to text and public diarization benchmarks as of September 1, 2020.
The following are some of the ways to get in touch with us. Artificial Analysis AA-WER StreamingMuse Voice Transcribe records a final-transcripted WER of 3.1% at 0.16s from the end of the speech. Cartesia Ink-2 is 3.4% with semantic ends at 0.43s. ElevenLabs Scribe Realtime has a 3.6% accuracy at 0.14s. Cartesia Ink-2, with external endpoints, is the fastest (0.07s) but also the least accurate (4.0%). Muse Voice Transcribe, on the first partial transcript it records at 0.13s, 3.6% of WER.
Meta reported a diarization average error rate of 17.5% across AMI IHM, AMI SDM and VoxConverse. The five other systems on the chart are between 21,1% and 28,6%.
The price is also a major factor. With a price of $3.00 per 1000 minutes, this product is cheaper than Cartesia Ink-2 ($4.00) and less expensive than ElevenLabs Scribe Realtime or Deepgram Flux ($6.50).
Interactive explainer
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The SEQ parameter is used to specify the number of rows.[
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Endpointing is a way to tell a voice assistant when he or she has finished the task, and not simply paused.function type(i). No separate voice-activity model sits in the loop.’
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