Neural building blocks for speaker diarization: speech activity detection, speaker change detection, overlapped speech detection, speaker embedding

Overview

⚠️ Checkout develop branch to see what is coming in pyannote.audio 2.0:

Neural speaker diarization with pyannote-audio

pyannote.audio is an open-source toolkit written in Python for speaker diarization. Based on PyTorch machine learning framework, it provides a set of trainable end-to-end neural building blocks that can be combined and jointly optimized to build speaker diarization pipelines:

pyannote.audio also comes with pretrained models covering a wide range of domains for voice activity detection, speaker change detection, overlapped speech detection, and speaker embedding:

segmentation

Open In Colab

Installation

pyannote.audio only supports Python 3.7 (or later) on Linux and macOS. It might work on Windows but there is no garantee that it does, nor any plan to add official support for Windows.

The instructions below assume that pytorch has been installed using the instructions from https://pytorch.org.

$ pip install pyannote.audio==1.1.1

Documentation and tutorials

Until a proper documentation is released, note that part of the API is described in this tutorial.

Citation

If you use pyannote.audio please use the following citation

@inproceedings{Bredin2020,
  Title = {{pyannote.audio: neural building blocks for speaker diarization}},
  Author = {{Bredin}, Herv{\'e} and {Yin}, Ruiqing and {Coria}, Juan Manuel and {Gelly}, Gregory and {Korshunov}, Pavel and {Lavechin}, Marvin and {Fustes}, Diego and {Titeux}, Hadrien and {Bouaziz}, Wassim and {Gill}, Marie-Philippe},
  Booktitle = {ICASSP 2020, IEEE International Conference on Acoustics, Speech, and Signal Processing},
  Address = {Barcelona, Spain},
  Month = {May},
  Year = {2020},
}
Comments
  • [WIP] Multilabel Detection

    [WIP] Multilabel Detection

    This is a new PR for the VTC feature, this time based on a cleaner implem. I'm making a new PR as to keep the former branch "clean" (and prevent any mishaps).

    What is done:

    • renaming the SpeakerTracking task into a MultilabelDetection task
    • added MultilabelPipeline
    • update MultilabelFscore.report()
    • tested the new preprocessor

    What's to be done:

    • [ ] re-test the new implem on our clinical data (as well as the child data from @MarvinLvn )
    • [ ] maybe a couple of unit tests (especially for the preprocessor)
    • [ ] maybe make the aggregated "multilabel" fscore duration-based instead of file-based
    opened by hadware 29
  • Trying the diarization pipeline on random .wav files

    Trying the diarization pipeline on random .wav files

    Hey, as suggested by the detailed tutorials, i went through them and trained all the models required for the pipeline. The pipeline is working on the AMI dataset but when i try to reproduce the results on other .wav files sampled at 16k, mono, and 256bps, it is not able to diarize the audio. Here is the breif of what i actually did.

    1. Took a random meeting audio file, sampled at 16k , mono and 256bps
    2. renamed it to ES2003a and replaced it with actual ES2003a ( thought it as a turnaround of creating another database )
    3. ran all the pipelines ( sad,scd, emb, diarization )

    Output :

    1. Speaker activity detection works perfectly and is able to classify regions of speech.
    2. Speaker diarization does't works, everything is classified as 0

    can you please tell if its because of replacing the actual file that the pipeline is giving wrong outputs for the diarization, and whats a better way to test the pipeline on random audios.

    opened by saisumit 26
  • build error

    build error

    Hi, when I run pip install "pyannote.audio==0.3", I got the following error msg:

    In file included from _pysndfile.cpp:471:0: pysndfile.hh:55:21: fatal error: sndfile.h: No such file or directory #include <sndfile.h> ^ compilation terminated. error: command 'gcc' failed with exit status 1


    Failed building wheel for pysndfile Running setup.py clean for pysndfile Failed to build pysndfile

    cannot_reproduce 
    opened by ChristopherLu 24
  • Add support for file handle to pyannote.audio.core.io.Audio

    Add support for file handle to pyannote.audio.core.io.Audio

    This is not currently supported:

    from pyannote.audio.core.io import Audio
    from pyannote.core import Segment
    audio = Audio()
    with open('file.wav', 'rb') as f:
        waveform, sample_rate = audio(f)
    with open('file.wav', 'rb') as f:
        waveform, sample_rate = audio.crop(f, Segment(10, 20))
    

    One has to do this instead:

    from pyannote.audio.core.io import Audio
    from pyannote.core import Segment
    audio = Audio()
    waveform, sample_rate = audio('file.wav')
    waveform, sample_rate = audio.crop('file.wav', Segment(10, 20))
    

    This is a limitation that might be problematic (e.g. with streamlit.file_uploader that returns a file handle)

    v2 
    opened by hbredin 20
  • ValueError: inconsistent

    ValueError: inconsistent "classes" (is ['non_change', 'change'], should be: ['non_speech', 'speech'])

    Describe the bug I'm trying to go through the diarization pipeline tutorial on my own data.

    I am trying to run "apply" on my own data and model for speaker change detection. I get an error that looks like it's trying to apply speech activity detection

    ValueError: inconsistent "classes" (is ['non_change', 'change'], should be: ['non_speech', 'speech'])

    To Reproduce Steps to reproduce the behavior:

    $ export EXP_DIR=tutorials/pipelines/speaker_diarization 
    $ pyannote-audio scd  apply --step=0.1 --pretrained="<path to>/tutorials/models/speaker_change_detection/train/myData.SpeakerDiarization.general.train/validate_segmentation_fscore/myData.SpeakerDiarization.general.train" --subset=train ${EXP_DIR} myData.SpeakerDiarization.general
    
    

    pyannote environment

    $ pip freeze | grep pyannote
    pyannote.core==4.1
    pyannote.database==4.0.1
    pyannote.metrics==3.0.1
    pyannote.pipeline==1.5.2
    

    Additional context I only prepared a development set called "train" right now - so I'm running on that. I successfully ran the SAD apply step before moving to SCD.

    wontfix 
    opened by danFromTelAviv 20
  • An error was encountered while loading

    An error was encountered while loading "pyannote/speaker-diarization"

    Hello,when i run the code :

    from pyannote.audio import Pipeline
    pipeline = Pipeline.from_pretrained("pyannote/speaker-diarization",
                                        use_auth_token="my_token")
    

    I get an error :

    Traceback (most recent call last):
      File "/home/dg/anaconda3/envs/pyannote/lib/python3.8/site-packages/huggingface_hub/utils/_errors.py", line 213, in hf_raise_for_status
        response.raise_for_status()
      File "/home/dg/anaconda3/envs/pyannote/lib/python3.8/site-packages/requests/models.py", line 1021, in raise_for_status
        raise HTTPError(http_error_msg, response=self)
    requests.exceptions.HTTPError: 403 Client Error: Forbidden for url: https://huggingface.co/pyannote/segmentation/resolve/2022.07/pytorch_model.bin
    

    whether I use the read token role or the write token role. Anyone else know how to fix it? Thx.

    opened by Zpadger 18
  • [WIP] Feat/vtc

    [WIP] Feat/vtc

    This is a working PR on the future VTC implementation inspired from @MarvinLvn 's work, and to be merged into the next release of pyannote-audio.

    Note: nothing has been done yet, this is just to get things started.

    wontfix 
    opened by hadware 17
  • Trying to finetune model for new speaker

    Trying to finetune model for new speaker

    I am trying to finetune models to support one more speaker, but it looks I am doing something wrong.

    I want to use "dia_hard" pipeline, so I need to finetune models: {sad_dihard, scd_dihard, emb_voxceleb}.

    For my speaker I have one WAV file with duration more then 1 hour.

    So, I created database.yml file:

    Databases:
       IK: /content/fine/kirilov/{uri}.wav
    
    Protocols:
        IK:
           SpeakerDiarization:
              kirilov:
                train:
                   uri: train.lst
                   annotation: train.rttm
                   annotated: train.uem
    

    and put additional files near database.yml:

    kirilov
    ├── database.yml
    ├── kirilov.wav
    ├── train.lst
    ├── train.rttm
    └── train.uem
    

    train.lst: kirilov

    train.rttm: SPEAKER kirilov 1 0.0 3600.0 <NA> <NA> Kirilov <NA> <NA>

    train.uem: kirilov NA 0.0 3600.0

    I assume it will say trainer to use kirilov.wav file and take 3600 seconds of audio from it to use for training.

    Now I finetune the models, current folder is /content/fine/kirilov, so database.yml is taken from the current directory:

    !pyannote-audio sad train --pretrained=sad_dihard --subset=train --to=1 --parallel=4 "/content/fine/sad" IK.SpeakerDiarization.kirilov
    !pyannote-audio scd train --pretrained=scd_dihard --subset=train --to=1 --parallel=4 "/content/fine/scd" IK.SpeakerDiarization.kirilov
    !pyannote-audio emb train --pretrained=emb_voxceleb --subset=train --to=1 --parallel=4 "/content/fine/emb" IK.SpeakerDiarization.kirilov
    

    Output looks like:

    Using cache found in /root/.cache/torch/hub/pyannote_pyannote-audio_develop
    Loading labels: 0file [00:00, ?file/s]/usr/local/lib/python3.6/dist-packages/pyannote/database/protocol/protocol.py:128: UserWarning:
    
    Existing key "annotation" may have been modified.
    
    Loading labels: 1file [00:00, 20.49file/s]
    /usr/local/lib/python3.6/dist-packages/pyannote/audio/train/trainer.py:128: UserWarning:
    
    Did not load optimizer state (most likely because current training session uses a different loss than the one used for pre-training).
    
    2020-06-19 15:35:26.763592: I tensorflow/stream_executor/platform/default/dso_loader.cc:44] Successfully opened dynamic library libcudart.so.10.1
    Training:   0%|                                        | 0/1 [00:00<?, ?epoch/s]
    Epoch pyannote/pyannote-database#1:   0%|                                       | 0/29 [00:00<?, ?batch/s]
    Epoch pyannote/pyannote-database#1:   0%|                           | 0/29 [00:00<?, ?batch/s, loss=0.676]
    Epoch pyannote/pyannote-database#1:   3%|▋                  | 1/29 [00:00<00:26,  1.04batch/s, loss=0.676]
    

    Etc.

    And try to run pipeline with new .pt's:

    import os
    import torch
    from pyannote.audio.pipeline import SpeakerDiarization
    pipeline = SpeakerDiarization(embedding = "/content/fine/emb/train/IK.SpeakerDiarization.kirilov.train/weights/0001.pt", 
                                  sad_scores = "/content/fine/sad/train/IK.SpeakerDiarization.kirilov.train/weights/0001.pt",
                                  scd_scores = "/content/fine/scd/train/IK.SpeakerDiarization.kirilov.train/weights/0001.pt",
                                  method= "affinity_propagation")
    
    #params from dia_dihard\train\X.SpeakerDiarization.DIHARD_Official.development\params.yml
    pipeline.load_params("/content/drive/My Drive/pyannote/params.yml")
    FILE = {'audio': "/content/groundtruth/new.wav"}
    diarization = pipeline(FILE)
    diarization
    

    The result is that for my new.wav the whole audio is recognized as speaker talking without pauses. So I assume that the models were broken. And it does not matter if I train for 1 epoch or for 100.

    In case I use:

    1. 0000.pt - I assume these are the original models
    pipeline = SpeakerDiarization(embedding = "/content/fine/emb/train/IK.SpeakerDiarization.kirilov.train/weights/0000.pt", 
                                  sad_scores = "/content/fine/sad/train/IK.SpeakerDiarization.kirilov.train/weights/0000.pt",
                                  scd_scores = "/content/fine/scd/train/IK.SpeakerDiarization.kirilov.train/weights/0000.pt",
                                  method= "affinity_propagation")
    

    or

    1. weights from original models
    pipeline = SpeakerDiarization(embedding = "/content/drive/My Drive/pyannote/emb_voxceleb/train/X.SpeakerDiarization.VoxCeleb.train/weights/0326.pt", 
                                 sad_scores = "/content/drive/My Drive/pyannote/sad_dihard/sad_dihard/train/X.SpeakerDiarization.DIHARD_Official.train/weights/0231.pt",
                                 scd_scores = "/content/drive/My Drive/pyannote/scd_dihard/train/X.SpeakerDiarization.DIHARD_Official.train/weights/0421.pt",
                                 method= "affinity_propagation")
    

    everything is ok and the result is similar to

    pipeline = torch.hub.load('pyannote/pyannote-audio', 'dia_dihard')
    FILE = {'audio': "/content/groundtruth/new.wav"}
    diarization = pipeline(FILE)
    diarization
    

    Could you please advise what could be wrong with my training\finetuning process?

    opened by marlon-br 17
  • `b c t` vs. `b t c`?

    `b c t` vs. `b t c`?

    Issue by hbredin Friday Oct 30, 2020 at 16:46 GMT Originally opened as https://github.com/hbredin/pyannote-audio-v2/issues/54


    Which convention should we use?

    v2 
    opened by mogwai 16
  • Segmentation Fault when conducting change detection tutorial

    Segmentation Fault when conducting change detection tutorial

    Hi,

    Everything seems ok for the feature extraction tutorial. But when I train the model following exactly what the tutorial asks me to do for change detection, I got segmentation fault. What might be probably the reason. Thank you for your help.

    opened by Charliechen1 16
  • Cannot find my Pretrained model

    Cannot find my Pretrained model

    I successfully trained an sad model. I want to create sad scores as part of the speaker diarization pipeline. I thought I am passing the weights correctly to the pyannote-audio script but the model is never found and the script aborts. Here is the output of my bash script with tracing on.

    This is the error message I get.

    RuntimeError: Cannot find callable /misc/vlgscratch4/PichenyGroup/picheny/headcam/headcam-code-try2/models/speech_activity_detection/train/AMI.SpeakerDiarization.MixHeadset.train/weights/0101.pt in hubconf

    For readability, I have boldfaced the commands in the script.

    ++ export EXP_DIR=models/speaker_diarization ++ EXP_DIR=models/speaker_diarization ++ cd /misc/vlgscratch4/PichenyGroup/picheny/headcam/headcam-code-try2 ++ export PYANNOTE_DATABASE_CONFIG=/misc/vlgscratch4/PichenyGroup/picheny/headcam/headcam-code-try2/database.yml ++ PYANNOTE_DATABASE_CONFIG=/misc/vlgscratch4/PichenyGroup/picheny/headcam/headcam-code-try2/database.yml ++ sad_model=/misc/vlgscratch4/PichenyGroup/picheny/headcam/headcam-code-try2/models/speech_activity_detection/train/AMI.SpeakerDiarization.MixHeadset.train/weights/0101.pt ++ ls /misc/vlgscratch4/PichenyGroup/picheny/headcam/headcam-code-try2/models/speech_activity_detection/train/AMI.SpeakerDiarization.MixHeadset.train/weights/0101.pt /misc/vlgscratch4/PichenyGroup/picheny/headcam/headcam-code-try2/models/speech_activity_detection/train/AMI.SpeakerDiarization.MixHeadset.train/weights/0101.pt ++ pyannote-audio sad apply --step=0.1 --pretrained=/misc/vlgscratch4/PichenyGroup/picheny/headcam/headcam-code-try2/models/speech_activity_detection/train/AMI.SpeakerDiarization.MixHeadset.train/weights/0101.pt --subset=dev models/speaker_diarization AMI.SpeakerDiarization.MixHeadset Using cache found in /home/map22/.cache/torch/hub/pyannote_pyannote-audio_develop Traceback (most recent call last): File "/misc/vlgscratch4/PichenyGroup/picheny/anaconda3/envs/pyannote-feb32020/bin/pyannote-audio", line 8, in sys.exit(main()) File "/misc/vlgscratch4/PichenyGroup/picheny/anaconda3/envs/pyannote-feb32020/lib/python3.7/site-packages/pyannote/audio/applications/pyannote_audio.py", line 406, in main apply_pretrained(validate_dir, protocol, **params) File "/misc/vlgscratch4/PichenyGroup/picheny/anaconda3/envs/pyannote-feb32020/lib/python3.7/site-packages/pyannote/audio/applications/base.py", line 514, in apply_pretrained step=step) File "/misc/vlgscratch4/PichenyGroup/picheny/anaconda3/envs/pyannote-feb32020/lib/python3.7/site-packages/torch/hub.py", line 364, in load entry = _load_entry_from_hubconf(hub_module, model) File "/misc/vlgscratch4/PichenyGroup/picheny/anaconda3/envs/pyannote-feb32020/lib/python3.7/site-packages/torch/hub.py", line 237, in _load_entry_from_hubconf raise RuntimeError('Cannot find callable {} in hubconf'.format(model)) RuntimeError: Cannot find callable /misc/vlgscratch4/PichenyGroup/picheny/headcam/headcam-code-try2/models/speech_activity_detection/train/AMI.SpeakerDiarization.MixHeadset.train/weights/0101.pt in hubconf

    opened by picheny-nyu 15
  • TypeError: __init__() missing 1 required positional argument: 'signature' while reproducing change-detection tutorial

    TypeError: __init__() missing 1 required positional argument: 'signature' while reproducing change-detection tutorial

    While following this tutorial https://github.com/pyannote/pyannote-audio/tree/89da05ea9d6de97da9bd21949a26ceb0042ef361/tutorials/change-detection

    and while executing this " pyannote-change-detection train \ ${EXPERIMENT_DIR} \ AMI.SpeakerDiarization.MixHeadset "

    File "/home/jashwanth/miniconda3/envs/py36-pyannote-audio/bin/pyannote-change-detection", line 8, in sys.exit(main()) File "/home/jashwanth/miniconda3/envs/py36-pyannote-audio/lib/python3.6/site-packages/pyannote/audio/applications/change_detection.py", line 380, in main train(protocol, experiment_dir, train_dir, subset=subset) File "/home/jashwanth/miniconda3/envs/py36-pyannote-audio/lib/python3.6/site-packages/pyannote/audio/applications/change_detection.py", line 202, in train generator = ChangeDetectionBatchGenerator(feature_extraction) File "/home/jashwanth/miniconda3/envs/py36-pyannote-audio/lib/python3.6/site-packages/pyannote/audio/generators/change.py", line 93, in init segment_generator) TypeError: init() missing 1 required positional argument: 'signature' please can anyone help me with this!!

    Thank you in advance

    opened by Jashwantherao 0
  • How to select threshold and min_cluster_size values in clustering after I finetuned embedding model ?

    How to select threshold and min_cluster_size values in clustering after I finetuned embedding model ?

    Hi. Thanks for sharing great repository. I have a question. I finetuned embedding model in speaker diarization pipeline. After that, I don't know how to set threshold and min_cluster_size params in config.yaml file. Can you give me some advices ?

    opened by dungnguyen98 0
  • Fixes for PytorchLightning >= 1.8

    Fixes for PytorchLightning >= 1.8

    Adjust to PytorchLightning API changes in version 1.8.0. I did some testing to make sure nothing broke, including model training/finetuning and loading from pretrained/HF-hub; however, my tests likely didn't cover everything.

    opened by entn-at 1
  • Support MLflow along with Tensorboard for logging segmentation task visualizations during validation

    Support MLflow along with Tensorboard for logging segmentation task visualizations during validation

    When using PyTorch-Lightning's MLFlowLogger instead of TensorBoardLogger during training of segmentation models, the current implementation fails because MLflow's experiment tracking client has a different method for logging figures than Tensorboard. Unfortunately, PyTorch-Lightning doesn't abstract away this logger API difference.

    Tested with MLflow and Tensorboard.

    opened by entn-at 1
  • Will it work on real time streaming data ?

    Will it work on real time streaming data ?

    I am currently running it on Apple M2 chip it is taking way much time comparing to Colab. Is there a way that pipeline could be modified to streaming data, and combined with some transcription service ?

    opened by ankurdhuriya 0
Releases(1.1.1)
Modular and extensible speech recognition library leveraging pytorch-lightning and hydra.

Lightning ASR Modular and extensible speech recognition library leveraging pytorch-lightning and hydra What is Lightning ASR • Installation • Get Star

Soohwan Kim 40 Sep 19, 2022
Python generation script for BitBirds

BitBirds generation script Intro This is published under MIT license, which means you can do whatever you want with it - entirely at your own risk. Pl

286 Dec 06, 2022
Blue Brain text mining toolbox for semantic search and structured information extraction

Blue Brain Search Source Code DOI Data & Models DOI Documentation Latest Release Python Versions License Build Status Static Typing Code Style Securit

The Blue Brain Project 29 Dec 01, 2022
Deeply Supervised, Layer-wise Prediction-aware (DSLP) Transformer for Non-autoregressive Neural Machine Translation

Non-Autoregressive Translation with Layer-Wise Prediction and Deep Supervision Training Efficiency We show the training efficiency of our DSLP model b

Chenyang Huang 37 Jan 04, 2023
A website which allows you to play with the GPT-2 transformer

transformers A website which allows you to play with the GPT-2 model Built with ❤️ by raphtlw Table of contents Model Setup About Contributors Model T

raphtlw 2 Jan 27, 2022
GCRC: A Gaokao Chinese Reading Comprehension dataset for interpretable Evaluation

GCRC GCRC: A New Challenging MRC Dataset from Gaokao Chinese for Explainable Eva

Yunxiao Zhao 5 Nov 04, 2022
Data manipulation and transformation for audio signal processing, powered by PyTorch

torchaudio: an audio library for PyTorch The aim of torchaudio is to apply PyTorch to the audio domain. By supporting PyTorch, torchaudio follows the

1.9k Jan 08, 2023
An easy to use Natural Language Processing library and framework for predicting, training, fine-tuning, and serving up state-of-the-art NLP models.

Welcome to AdaptNLP A high level framework and library for running, training, and deploying state-of-the-art Natural Language Processing (NLP) models

Novetta 407 Jan 03, 2023
Contract Understanding Atticus Dataset

Contract Understanding Atticus Dataset This repository contains code for the Contract Understanding Atticus Dataset (CUAD), a dataset for legal contra

The Atticus Project 273 Dec 17, 2022
Refactored version of FastSpeech2

Refactored version of FastSpeech2. An implementation of Microsoft's "FastSpeech 2: Fast and High-Quality End-to-End Text to Speech"

ILJI CHOI 10 May 26, 2022
The code for the Subformer, from the EMNLP 2021 Findings paper: "Subformer: Exploring Weight Sharing for Parameter Efficiency in Generative Transformers", by Machel Reid, Edison Marrese-Taylor, and Yutaka Matsuo

Subformer This repository contains the code for the Subformer. To help overcome this we propose the Subformer, allowing us to retain performance while

Machel Reid 10 Dec 27, 2022
Download videos from YouTube/Twitch/Twitter right in the Windows Explorer, without installing any shady shareware apps

youtube-dl and ffmpeg Windows Explorer Integration Download videos from YouTube/Twitch/Twitter and more (any platform that is supported by youtube-dl)

Wolfgang 226 Dec 30, 2022
Simple telegram bot to convert files into direct download link.you can use telegram as a file server 🪁

TGCLOUD 🪁 Simple telegram bot to convert files into direct download link.you can use telegram as a file server 🪁 Features Easy to Deploy Heroku Supp

Mr.Acid dev 6 Oct 18, 2022
Transformer-XL: Attentive Language Models Beyond a Fixed-Length Context

Transformer-XL: Attentive Language Models Beyond a Fixed-Length Context This repository contains the code in both PyTorch and TensorFlow for our paper

Zhilin Yang 3.3k Dec 28, 2022
This repository is home to the Optimus data transformation plugins for various data processing needs.

Transformers Optimus's transformation plugins are implementations of Task and Hook interfaces that allows execution of arbitrary jobs in optimus. To i

Open Data Platform 37 Dec 14, 2022
TFPNER: Exploration on the Named Entity Recognition of Token Fused with Part-of-Speech

TFPNER TFPNER: Exploration on the Named Entity Recognition of Token Fused with Part-of-Speech Named entity recognition (NER), which aims at identifyin

1 Feb 07, 2022
DaCy: The State of the Art Danish NLP pipeline using SpaCy

DaCy: A SpaCy NLP Pipeline for Danish DaCy is a Danish preprocessing pipeline trained in SpaCy. At the time of writing it has achieved State-of-the-Ar

Kenneth Enevoldsen 71 Jan 06, 2023
Unet-TTS: Improving Unseen Speaker and Style Transfer in One-shot Voice Cloning

Unet-TTS: Improving Unseen Speaker and Style Transfer in One-shot Voice Cloning English | 中文 ❗ Now we provide inferencing code and pre-training models

164 Jan 02, 2023
Multilingual finetuning of Machine Translation model on low-resource languages. Project for Deep Natural Language Processing course.

Low-resource-Machine-Translation This repository contains the code for the project relative to the course Deep Natural Language Processing. The goal o

Andrea Cavallo 3 Jun 22, 2022
Knowledge Graph,Question Answering System,基于知识图谱和向量检索的医疗诊断问答系统

Knowledge Graph,Question Answering System,基于知识图谱和向量检索的医疗诊断问答系统

wangle 823 Dec 28, 2022