NeMo Speaker Diarization Configuration Files

Both training and inference of speaker diarization is configured by .yaml files. The diarizer section will generally require information about the dataset(s) being used, models used in this pipeline, as well as inference related parameters such as post processing of each models. The sections on this page cover each of these in more detail.

Note

For model details and deep understanding about configs, training, fine-tuning and evaluations, please refer to <NeMo_git_root>/tutorials/speaker_tasks/Speaker_Diarization_Inference.ipynb and <NeMo_git_root>/tutorials/speaker_tasks/Speaker_Diarization_Training.ipynb; for other applications such as possible integration with ASR, have a look at <NeMo_git_root>/tutorials/speaker_tasks/ASR_with_SpeakerDiarization.ipynb.

Currently, NeMo supports Multi-scale diarization decoder (MSDD) as a neural diarizer model. MSDD is a speaker diarization model based on initializing clustering and multi-scale segmentation input. Example configuration files for MSDD model training can be found in <NeMo_git_root>/examples/speaker_tasks/diarization/conf/neural_diarizer/.

  • Model name convention for MSDD: msdd_<number of scales>scl_<longest scale in decimal second (ds)>_<shortest scale in decimal second (ds)>_<overlap percentage of window shifting>Povl_<hidden layer size>x<number of LSTM layers>x<number of CNN output channels>x<repetition count of conv layer>

  • Example: msdd_5scl_15_05_50Povl_256x3x32x2.yaml has 5 scales, the longest scale is 1.5 sec, the shortest scale is 0.5 sec, with 50 percent overlap, hidden layer size is 256, 3 LSTM layers, 32 CNN channels, 2 repeated Conv layers

MSDD model checkpoint (.ckpt) and NeMo file (.nemo) contain speaker embedding model (TitaNet) and the speaker model is loaded along with standalone MSDD module. Note that MSDD models require more than one scale. Thus, the parameters in diarizer.speaker_embeddings.parameters should have more than one scale to function as a MSDD model.

General Diarizer Configuration

The items (OmegaConfig keys) directly under model determines segmentation and clustering related parameters. Multi-scale parameters (window_length_in_sec, shift_length_in_sec and multiscale_weights) are specified. max_num_of_spks, scale_n, soft_label_thres and emb_batch_size are set here and then assigned to dataset configurations.

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diarizer: out_dir: null oracle_vad: True # If True, uses RTTM files provided in manifest file to get speech activity (VAD) timestamps speaker_embeddings: model_path: ??? # .nemo local model path or pretrained model name (titanet_large is recommended) parameters: window_length_in_sec: [1.5,1.25,1.0,0.75,0.5] # Window length(s) in sec (floating-point number). either a number or a list. ex) 1.5 or [1.5,1.0,0.5] shift_length_in_sec: [0.75,0.625,0.5,0.375,0.25] # Shift length(s) in sec (floating-point number). either a number or a list. ex) 0.75 or [0.75,0.5,0.25] multiscale_weights: [1,1,1,1,1] # Weight for each scale. should be null (for single scale) or a list matched with window/shift scale count. ex) [0.33,0.33,0.33] save_embeddings: True # Save embeddings as pickle file for each audio input. num_workers: ${num_workers} # Number of workers used for data-loading. max_num_of_spks: 2 # Number of speakers per model. This is currently fixed at 2. scale_n: 5 # Number of scales for MSDD model and initializing clustering. soft_label_thres: 0.5 # Threshold for creating discretized speaker label from continuous speaker label in RTTM files. emb_batch_size: 0 # If this value is bigger than 0, corresponding number of embedding vectors are attached to torch graph and trained.

Dataset Configuration

Training, validation, and test parameters are specified using the train_ds, validation_ds, and test_ds sections in the configuration YAML file, respectively. The items such as num_spks, soft_label_thres and emb_batch_size follow the settings in model key. You may also leave fields such as the manifest_filepath or emb_dir blank, and then specify it via command-line interface. Note that test_ds is not used during training and only used for speaker diarization inference.

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train_ds: manifest_filepath: ??? emb_dir: ??? sample_rate: ${sample_rate} num_spks: ${model.max_num_of_spks} soft_label_thres: ${model.soft_label_thres} labels: null batch_size: ${batch_size} emb_batch_size: ${model.emb_batch_size} shuffle: True validation_ds: manifest_filepath: ??? emb_dir: ??? sample_rate: ${sample_rate} num_spks: ${model.max_num_of_spks} soft_label_thres: ${model.soft_label_thres} labels: null batch_size: 2 emb_batch_size: ${model.emb_batch_size} shuffle: False test_ds: manifest_filepath: null emb_dir: null sample_rate: 16000 num_spks: ${model.max_num_of_spks} soft_label_thres: ${model.soft_label_thres} labels: null batch_size: 2 shuffle: False seq_eval_mode: False

Pre-processor Configuration

In the MSDD configuration, pre-processor configuration follows the pre-processor of the embedding extractor model.

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preprocessor: _target_: nemo.collections.asr.modules.AudioToMelSpectrogramPreprocessor normalize: "per_feature" window_size: 0.025 sample_rate: ${sample_rate} window_stride: 0.01 window: "hann" features: 80 n_fft: 512 frame_splicing: 1 dither: 0.00001

Model Architecture Configurations

The hyper-parameters for MSDD models are under the msdd_module key. The model architecture can be changed by setting up the weighting_scheme and context_vector_type. The detailed explanation for architecture can be found in the Models page.

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msdd_module: _target_: nemo.collections.asr.modules.msdd_diarizer.MSDD_module num_spks: ${model.max_num_of_spks} # Number of speakers per model. This is currently fixed at 2. hidden_size: 256 # Hidden layer size for linear layers in MSDD module num_lstm_layers: 3 # Number of stacked LSTM layers dropout_rate: 0.5 # Dropout rate cnn_output_ch: 32 # Number of filters in a conv-net layer. conv_repeat: 2 # Determines the number of conv-net layers. Should be greater or equal to 1. emb_dim: 192 # Dimension of the speaker embedding vectors scale_n: ${model.scale_n} # Number of scales for multiscale segmentation input weighting_scheme: 'conv_scale_weight' # Type of weighting algorithm. Options: ('conv_scale_weight', 'attn_scale_weight') context_vector_type: 'cos_sim' # Type of context vector: options. Options: ('cos_sim', 'elem_prod')

Loss Configurations

Neural diarizer uses a binary cross entropy (BCE) loss. A set of weights for negative (absence of the speaker’s speech) and positive (presence of the speaker’s speech) can be provided to the loss function.

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loss: _target_: nemo.collections.asr.losses.bce_loss.BCELoss weight: null # Weight for binary cross-entropy loss. Either `null` or list type input. (e.g. [0.5,0.5])

Example configuration files for speaker diarization inference can be found in <NeMo_git_root>/examples/speaker_tasks/diarization/conf/inference/. Choose a yaml file that fits your targeted domain. For example, if you want to diarize audio recordings of telephonic speech, choose diar_infer_telephonic.yaml.

The configurations for all the components of diarization inference are included in a single file named diar_infer_<domain>.yaml. Each .yaml file has a few different sections for the following modules: VAD, Speaker Embedding, Clustering and ASR.

In speaker diarization inference, the datasets provided in manifest format denote the data that you would like to perform speaker diarization on.

Diarizer Configurations

An example diarizer Hydra configuration could look like:

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diarizer: manifest_filepath: ??? out_dir: ??? oracle_vad: False # If True, uses RTTM files provided in manifest file to get speech activity (VAD) timestamps collar: 0.25 # Collar value for scoring ignore_overlap: True # Consider or ignore overlap segments while scoring

Under diarizer key, there are vad, speaker_embeddings, clustering and asr keys containing configurations for the inference of the corresponding modules.

Configurations for Voice Activity Detector

Parameters for VAD model are provided as in the following Hydra config example.

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vad: model_path: null # .nemo local model path or pretrained model name or none external_vad_manifest: null # This option is provided to use external vad and provide its speech activity labels for speaker embeddings extraction. Only one of model_path or external_vad_manifest should be set parameters: # Tuned parameters for CH109 (using the 11 multi-speaker sessions as dev set) window_length_in_sec: 0.15 # Window length in sec for VAD context input shift_length_in_sec: 0.01 # Shift length in sec for generate frame level VAD prediction smoothing: "median" # False or type of smoothing method (eg: median) overlap: 0.875 # Overlap ratio for overlapped mean/median smoothing filter onset: 0.4 # Onset threshold for detecting the beginning and end of a speech offset: 0.7 # Offset threshold for detecting the end of a speech pad_onset: 0.05 # Adding durations before each speech segment pad_offset: -0.1 # Adding durations after each speech segment min_duration_on: 0.2 # Threshold for small non_speech deletion min_duration_off: 0.2 # Threshold for short speech segment deletion filter_speech_first: True

Configurations for Speaker Embedding in Diarization

Parameters for speaker embedding model are provided in the following Hydra config example. Note that multiscale parameters either accept list or single floating point number.

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speaker_embeddings: model_path: ??? # .nemo local model path or pretrained model name (titanet_large, ecapa_tdnn or speakerverification_speakernet) parameters: window_length_in_sec: 1.5 # Window length(s) in sec (floating-point number). Either a number or a list. Ex) 1.5 or [1.5,1.25,1.0,0.75,0.5] shift_length_in_sec: 0.75 # Shift length(s) in sec (floating-point number). Either a number or a list. Ex) 0.75 or [0.75,0.625,0.5,0.375,0.25] multiscale_weights: null # Weight for each scale. should be null (for single scale) or a list matched with window/shift scale count. Ex) [1,1,1,1,1] save_embeddings: False # Save embeddings as pickle file for each audio input.

Configurations for Clustering in Diarization

Parameters for clustering algorithm are provided in the following Hydra config example.

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clustering: parameters: oracle_num_speakers: False # If True, use num of speakers value provided in the manifest file. max_num_speakers: 20 # Max number of speakers for each recording. If oracle_num_speakers is passed, this value is ignored. enhanced_count_thres: 80 # If the number of segments is lower than this number, enhanced speaker counting is activated. max_rp_threshold: 0.25 # Determines the range of p-value search: 0 < p <= max_rp_threshold. sparse_search_volume: 30 # The higher the number, the more values will be examined with more time.

Configurations for Diarization with ASR

The following configuration needs to be appended under diarizer to run ASR with diarization to get a transcription with speaker labels.

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asr: model_path: ??? # Provide NGC cloud ASR model name. stt_en_conformer_ctc_* models are recommended for diarization purposes. parameters: asr_based_vad: False # if True, speech segmentation for diarization is based on word-timestamps from ASR inference. asr_based_vad_threshold: 50 # threshold (multiple of 10ms) for ignoring the gap between two words when generating VAD timestamps using ASR based VAD. asr_batch_size: null # Batch size can be dependent on each ASR model. Default batch sizes are applied if set to null. lenient_overlap_WDER: True # If true, when a word falls into speaker-overlapped regions, consider the word as a correctly diarized word. decoder_delay_in_sec: null # Native decoder delay. null is recommended to use the default values for each ASR model. word_ts_anchor_offset: null # Offset to set a reference point from the start of the word. Recommended range of values is [-0.05 0.2]. word_ts_anchor_pos: "start" # Select which part of the word timestamp we want to use. The options are: 'start', 'end', 'mid'. fix_word_ts_with_VAD: False # Fix the word timestamp using VAD output. You must provide a VAD model to use this feature. colored_text: False # If True, use colored text to distinguish speakers in the output transcript. print_time: True # If True, the start of the end time of each speaker turn is printed in the output transcript. break_lines: False # If True, the output transcript breaks the line to fix the line width (default is 90 chars) ctc_decoder_parameters: # Optional beam search decoder (pyctcdecode) pretrained_language_model: null # KenLM model file: .arpa model file or .bin binary file. beam_width: 32 alpha: 0.5 beta: 2.5 realigning_lm_parameters: # Experimental feature arpa_language_model: null # Provide a KenLM language model in .arpa format. min_number_of_words: 3 # Min number of words for the left context. max_number_of_words: 10 # Max number of words for the right context. logprob_diff_threshold: 1.2 # The threshold for the difference between two log probability values from two hypotheses.

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