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Speaker Diarization

Documentation section for speaker related tasks can be found at:

Features of NeMo Speaker Diarization

  • Provides pretrained speaker embedding extractor models and VAD models.
  • Does not need to be tuned on dev-set while showing the better performance than AHC+PLDA method in general.
  • Estimates the number of speakers in the given session.
  • Provides example script for asr transcription with speaker labels.

Supported Pretrained Speaker Embedding Extractor models

Supported End-to-End Speaker Diarization models

Supported Pretrained VAD models

Supported ASR models

QuartzNet, CitriNet and Conformer-CTC models are supported. Recommended models HuggingFace:

Performance

Clustering Diarizer

Diarization Error Rate (DER) table of titanet_large.nemo model on well known evaluation datasets.

Evaluation Condition AMI(Lapel) AMI(MixHeadset) CH109 NIST SRE 2000
Domain Configuration Meeting Meeting Telephonic Telephonic
Oracle VAD
Known # of Speakers
1.28 1.07 0.56 5.62
Oracle VAD
Unknown # of Speakers
1.28 1.4 0.88 4.33
  • All models were tested using the domain specific .yaml files which can be found in conf/inference/ folder.
  • The above result is based on the oracle Voice Activity Detection (VAD) result.
  • This result is based on titanet_large.nemo model.

End-to-End Diarizer: Sortformer

Sortformer is a novel end-to-end neural model for speaker diarization that resolves the permutation problem by following the arrival-time order of speech segments from each speaker. Sortformer consists of a Fast-Conformer (NEST) encoder followed by a Transformer encoder with sigmoid outputs for each speaker. Both offline and streaming variants are available.

diar_sortformer_4spk-v1 (Offline)

An offline Sortformer diarizer with an 18-layer NEST encoder (L-size) and 18-layer Transformer encoder (hidden size 192), supporting up to 4 speakers. Trained on ~7,180 hours of real conversations and simulated audio mixtures.

from nemo.collections.asr.models import SortformerEncLabelModel
diar_model = SortformerEncLabelModel.from_pretrained("nvidia/diar_sortformer_4spk-v1")
diar_model.eval()
predicted_segments = diar_model.diarize(audio="/path/to/audio.wav", batch_size=1)

Diarization Error Rate (DER) β€” all evaluations include overlapping speech:

Dataset Collar DER (no PP) DER (with PP)
DIHARD3-Eval (<=4spk) 0.0s 16.28 14.76
CALLHOME-part2 (2spk) 0.25s 6.49 5.85
CALLHOME-part2 (3spk) 0.25s 10.01 8.46
CALLHOME-part2 (4spk) 0.25s 14.14 12.59
CH109 0.25s 6.27 6.86
diar_streaming_sortformer_4spk-v2 (Streaming)

A streaming version of Sortformer using an Arrival-Order Speaker Cache (AOSC) for consistent speaker tracking across chunks. Uses a 17-layer NEST encoder and 18-layer Transformer encoder. Supports configurable latency from ultra-low (0.32s) to very-high (30.4s).

from nemo.collections.asr.models import SortformerEncLabelModel
diar_model = SortformerEncLabelModel.from_pretrained("nvidia/diar_streaming_sortformer_4spk-v2")
diar_model.eval()
diar_model.sortformer_modules.chunk_len = 340
diar_model.sortformer_modules.chunk_right_context = 40
diar_model.sortformer_modules.fifo_len = 40
diar_model.sortformer_modules.spkcache_update_period = 300
predicted_segments = diar_model.diarize(audio="/path/to/audio.wav", batch_size=1)

Diarization Error Rate (DER) with post-processing β€” all evaluations include overlapping speech:

Dataset Collar 30.4s latency 10.0s latency 1.04s latency 0.32s latency
DIHARD III Eval (<=4spk) 0.0s 13.45 13.75 13.24 13.44
DIHARD III Eval (>=5spk) 0.0s 41.40 41.41 42.56 43.73
CALLHOME-part2 (2spk) 0.25s 5.34 6.05 6.57 6.91
CALLHOME-part2 (3spk) 0.25s 9.22 9.88 10.05 10.45
CALLHOME-part2 (4spk) 0.25s 11.29 11.72 12.44 13.70
CH109 0.25s 4.61 4.80 4.88 5.27
diar_streaming_sortformer_4spk-v2.1 (Streaming, improved)

An improved streaming Sortformer with greater robustness for meeting speech scenarios. Same architecture as v2 but with additional training on meeting corpora (DiPCo, AliMeeting, NOTSOFAR1) and forced-alignment-based ground-truth RTTMs for AMI and AliMeeting.

from nemo.collections.asr.models import SortformerEncLabelModel
diar_model = SortformerEncLabelModel.from_pretrained("nvidia/diar_streaming_sortformer_4spk-v2.1")
diar_model.eval()
diar_model.sortformer_modules.chunk_len = 340
diar_model.sortformer_modules.chunk_right_context = 40
diar_model.sortformer_modules.fifo_len = 40
diar_model.sortformer_modules.spkcache_update_period = 300
predicted_segments = diar_model.diarize(audio="/path/to/audio.wav", batch_size=1)

Diarization Error Rate (DER) β€” all evaluations include overlapping speech:

Telephonic and General-Purpose Speech:

Dataset Collar 30.4s latency 1.04s latency
DIHARD III Eval (<=4spk) 0.0s 14.84 15.09
DIHARD III Eval (full) 0.0s 19.49 20.21
CALLHOME-part2 (full) 0.25s 10.10 11.19
CH109 0.25s 5.04 5.09

Meeting Speech (v2.1 vs v2 comparison):

Dataset Collar v2.1 (30.4s) v2 (30.4s) v2.1 (1.04s) v2 (1.04s)
AliMeeting Test (near) 0.0s 11.73 19.63 12.60 19.98
AliMeeting Test (far) 0.0s 13.55 21.09 15.60 22.09
AMI Test (IHM) 0.0s 15.90 22.39 16.67 25.11
AMI Test (SDM) 0.0s 17.80 28.56 20.57 31.34
NOTSOFAR1 Eval SC (full) 0.0s 27.07 33.43 28.75 34.52
Example inference script for end-to-end diarizer
python neural_diarizer/e2e_diarize_speech.py \
    model_path=<path to .nemo or HuggingFace model name> \
    dataset_manifest=<path to manifest file> \
    batch_size=1

Run Speaker Diarization on Your Audio Files

Example script for clustering diarizer: with system-VAD

  python clustering_diarizer/offline_diar_infer.py \
    diarizer.manifest_filepath=<path to manifest file> \
    diarizer.out_dir='demo_output' \
    diarizer.speaker_embeddings.parameters.save_embeddings=False \
    diarizer.vad.model_path=<pretrained model name or path to .nemo> \
    diarizer.speaker_embeddings.model_path=<pretrained speaker embedding model name or path to .nemo> 

If you have oracle VAD files and groundtruth RTTM files for evaluation: Provide rttm files in the input manifest file and enable oracle_vad as shown below.

...
    diarizer.oracle_vad=True \
...

Arguments

      To run speaker diarization on your audio recordings, you need to prepare the following file.

  • diarizer.manifest_filepath: Path to manifest file

Example: manifest.json

{"audio_filepath": "/path/to/audio_file", "offset": 0, "duration": null, label: "infer", "text": "-", "num_speakers": null, "rttm_filepath": "/path/to/rttm/file", "uem_filepath"="/path/to/uem/filepath"}

Mandatory fields are audio_filepath, offset, duration, label:"infer" and text: <ground truth or "-" > , and the rest are optional keys which can be passed based on the type of evaluation

Some of important options in config file:

  • diarizer.vad.model_path: voice activity detection model name or path to the model

Specify the name of VAD model, then the script will download the model from NGC. Currently, we have 'vad_multilingual_marblenet', 'vad_marblenet' and 'vad_telephony_marblenet' as options for VAD models.

diarizer.vad.model_path='vad_multilingual_marblenet'

Instead, you can also download the model from vad_multilingual_marblenet, vad_marblenet and vad_telephony_marblenet and specify the full path name to the model as below.

diarizer.vad.model_path='path/to/vad_multilingual_marblenet.nemo'

  • diarizer.speaker_embeddings.model_path: speaker embedding model name

Specify the name of speaker embedding model, then the script will download the model from NGC. Currently, we support 'titanet_large', 'ecapa_tdnn' and 'speakerverification_speakernet'.

diarizer.speaker_embeddings.model_path='titanet_large'

You could also download *.nemo files from this link and specify the full path name to the speaker embedding model file (*.nemo).

diarizer.speaker_embeddings.model_path='path/to/titanet_large.nemo'

  • diarizer.speaker_embeddings.parameters.multiscale_weights: multiscale diarization

Multiscale diarization system employs multiple scales at the same time to obtain a finer temporal resolution. To use multiscale feature, at least two scales and scale weights should be provided. The scales should be provided in descending order, from the longest scale to the base scale (the shortest). If multiple scales are provided, multiscale_weights must be provided in list format. The following example shows how multiscale parameters are specified and the recommended parameters.

Example script: single-scale and multiscale

Single-scale setting:

  python offline_diar_infer.py \
     ... <other parameters> ...
     parameters.window_length_in_sec=1.5 \
     parameters.shift_length_in_sec=0.75 \
     parameters.multiscale_weights=null \

Multiscale setting (base scale - window_length 0.5 s and shift_length 0.25):

  python offline_diar_infer.py \
     ... <other parameters> ...
     parameters.window_length_in_sec=[1.5,1.0,0.5] \
     parameters.shift_length_in_sec=[0.75,0.5,0.25] \
     parameters.multiscale_weights=[0.33,0.33,0.33] \

Run Speech Recognition with Clustering based Speaker Diarization

Using the script offline_diar_with_asr_infer.py, you can transcribe your audio recording with speaker labels as shown below:

[00:03.34 - 00:04.46] speaker_0: back from the gym oh good how's it going 
[00:04.46 - 00:09.96] speaker_1: oh pretty well it was really crowded today yeah i kind of assumed everyone would be at the shore uhhuh
[00:12.10 - 00:13.97] speaker_0: well it's the middle of the week or whatever so

Currently, offline clustering diarization inference supports ConformerCTC ASR models (e.g.,stt_en_conformer_ctc_large).

Example script

python offline_diar_with_asr_infer.py \
    diarizer.manifest_filepath=<path to manifest file> \
    diarizer.out_dir='demo_asr_output' \
    diarizer.speaker_embeddings.model_path=<pretrained model name or path to .nemo> \
    diarizer.asr.model_path=<pretrained model name or path to .nemo> \
    diarizer.speaker_embeddings.parameters.save_embeddings=False \
    diarizer.asr.parameters.asr_based_vad=True

If you have reference rttm files or oracle number of speaker information, you can provide those file paths and number of speakers in the manifest file path and pass diarizer.clustering.parameters.oracle_num_speakers=True as shown in the following example.

python offline_diar_with_asr_infer.py \
    diarizer.manifest_filepath=<path to manifest file> \
    diarizer.out_dir='demo_asr_output' \
    diarizer.speaker_embeddings.model_path=<pretrained model name or path to .nemo> \
    diarizer.asr.model_path=<pretrained model name or path to .nemo> \
    diarizer.speaker_embeddings.parameters.save_embeddings=False \
    diarizer.asr.parameters.asr_based_vad=True \
    diarizer.clustering.parameters.oracle_num_speakers=True

Output folders

The above script will create a folder named ./demo_asr_output/. For example, in ./demo_asr_output/, you can check the results as below.

./asr_with_diar
β”œβ”€β”€ pred_rttms
    └── my_audio1.json
    └── my_audio1.txt
    └── my_audio1.rttm
    └── my_audio1_gecko.json
β”‚
└── speaker_outputs
    └── oracle_vad_manifest.json
    └── subsegments_scale2_cluster.label
    └── subsegments_scale0.json
    └── subsegments_scale1.json
    └── subsegments_scale2.json
... 

my_audio1.json file contains word-by-word json output with speaker label and time stamps. We also provide a json output file for gecko tool, where you can visualize the diarization result along with the ASR output.

Example: ./demo_asr_output/pred_rttms/my_audio1.json

{
    "status": "Success",
    "session_id": "my_audio1",
    "transcription": "back from the gym oh good ...",
    "speaker_count": 2,
    "words": [
        {
            "word": "back",
            "start_time": 0.44,
            "end_time": 0.56,
            "speaker_label": "speaker_0"
        },
...
        {
            "word": "oh",
            "start_time": 1.74,
            "end_time": 1.88,
            "speaker_label": "speaker_1"
        },
        {
            "word": "good",
            "start_time": 2.08,
            "end_time": 3.28,
            "speaker_label": "speaker_1"
        },

*.txt files in pred_rttms folder contain transcriptions with speaker labels and corresponding time.

Example: ./demo_asr_output/pred_rttms/my_audio1.txt

[00:03.34 - 00:04.46] speaker_0: back from the gym oh good how's it going
[00:04.46 - 00:09.96] speaker_1: pretty well it was really crowded today yeah i kind of assumed everylonewould be at the shore uhhuh
[00:12.10 - 00:13.97] speaker_0: well it's the middle of the week or whatever so
[00:13.97 - 00:15.78] speaker_1: but it's the fourth of july mm
[00:16.90 - 00:21.80] speaker_0: so yeah people still work tomorrow do you have to work tomorrow did you drive off yesterday

In speaker_outputs folder we have three kinds of files as follows:

  • oracle_vad_manifest.json file contains oracle VAD labels that are extracted from RTTM files.
  • subsegments_scale<scale_index>.json is a manifest file for subsegments, which includes segment-by-segment start and end time with original wav file path. In multi-scale mode, this file is generated for each <scale_index>.
  • subsegments_scale<scale_index>_cluster.label file contains the estimated cluster labels for each segment. This file is only generated for the base scale index in multi-scale diarization mode.

Optional Features for Speech Recognition with Speaker Diarization

Beam Search Decoder

Beam-search decoder can be applied to CTC based ASR models. To use this feature, pyctcdecode should be installed. pyctcdecode supports word timestamp generation and can be applied to speaker diarization. pyctcdecode also requires KenLM and KenLM is recommended to be installed using PyPI. Install pyctcdecode in your environment with the following commands:

pip install pyctcdecode
pip install https://github.com/kpu/kenlm/archive/master.zip

You should provide a trained KenLM language model to use pyctcdecode. Binary or .arpa format can be provided to hydra configuration as below.

  python offline_diar_with_asr_infer.py \
    ... <other parameters> ...
    diarizer.asr.ctc_decoder_parameters.pretrained_language_model="/path/to/kenlm_language_model.binary"

You can download publicly available language models (.arpa files) at KALDI Tedlium Language Models. Download 4-gram Big ARPA and provide the model path.

The following CTC decoder parameters can be modified to optimize the performance.
diarizer.asr.ctc_decoder_parameters.beam_width (default: 32)
diarizer.asr.ctc_decoder_parameters.alpha (default: 0.5)
diarizer.asr.ctc_decoder_parameters.beta (default: 2.5)

Realign Words with a Language Model (Experimental)

Diarization result with ASR transcript can be enhanced by applying a language model. To use this feature, python package arpa should be installed.

pip install arpa

diarizer.asr.realigning_lm_parameters.logprob_diff_threshold can be modified to optimize the diarization performance (default value is 1.2). The lower the threshold, the more changes are expected to be seen in the output transcript.

arpa package also uses KenLM language models as in pyctcdecode. You can download publicly available 4-gram Big ARPA model and provide the model path to hydra configuration as follows.

python offline_diar_with_asr_infer.py \
    ... <other parameters> ...
    diarizer.asr.realigning_lm_parameters.logprob_diff_threshold=1.2 \
    diarizer.asr.realigning_lm_parameters.arpa_language_model="/path/to/4gram_big.arpa"\