sample_rate: 24_000 # 44_000 for dac audio_backend: "vocos" # or dac models: - name: "ar+nar" # vanity name size: "full" # model dimensionality resp_levels: 8 # RVQ levels this model targets prom_levels: 8 # should always be the above tasks: 8 # tasks this model can attend to, only tts is supported at the moment langs: 2 # languages this model supports, semi-unused at the moment tones: 1 # tones this model supports, currently unused arch_type: llama # underlying LLM arch to use, currently focusing on llama training: True # signals this model is to be trained version: 5 # helps keep backwards compatibility for when I add new things to the model attention: auto # attention mechanism to use, "auto" for safety dropout: 0.1 # percentage of the model to disable during training # factors for split loss values, remove to have a unified loss calculation loss_factors: text: 0.1 # text phoneme portion of the sequence prom: 0.0 # input prompt portion of the sequence resp: 1.0 # output audio portin of the sequence # experimental settings experimental: hf: False # uses vall_e.models.experimental, a wrapper around a HF model that could technically be used for non-pytorch backends later interleave: False # interleaves RVQ levels, only works with above for now audio_embedding_mode: "" # "" | "inclusive" | "exclusive", whether to utilize the audio backend's embeddings with the input embeddings audio_embedding_sums: False # whether the input embeddings include all prior RVQ levels (sums) or only the current one, further experimentation is needed to see if this matters hyperparameters: autotune: False autotune_params: start_profile_step: 1 end_profile_step: 50 num_tuning_micro_batch_sizes: 8 batch_size: 16 gradient_accumulation_steps: 4 gradient_clipping: 1.0 warmup_steps: 100 optimizer: Prodigy learning_rate: 1.0 torch_optimizer: True scheduler: "" # ScheduleFree torch_scheduler: True evaluation: batch_size: 8 frequency: 5000 size: 8 steps: 500 ar_temperature: 0.95 nar_temperature: 0.25 load_disabled_engines: True trainer: #no_logger: True ddp: False #check_for_oom: False iterations: 1_000_000 save_tag: step save_on_oom: True save_on_quit: True save_frequency: 250 export_on_save: True keep_last_checkpoints: 4 gradient_checkpointing: True strict_loading: False #load_state_dict: True #load_tag: "9500" #load_states: False #restart_step_count: True gc_mode: None # "global_step" weight_dtype: float32 # float16 or bfloat16 amp: False backend: deepspeed deepspeed: inferencing: True zero_optimization_level: 0 use_compression_training: False amp: False load_webui: False inference: backend: deepspeed normalize: False weight_dtype: float32 # float16 or bfloat16 amp: False optimizations: injects: False replace: True linear: False embedding: False optimizers: True bitsandbytes: False dadaptation: False bitnet: False fp8: False dataset: speaker_name_getter: "lambda p: f'{p.parts[-3]}_{p.parts[-2]}'" speaker_group_getter: "lambda p: f'{p.parts[-3]}'" speaker_languages: ja: [] use_hdf5: True use_metadata: True hdf5_flag: r validate: True workers: 2 cache: True duration_range: [3.0, 5.0] random_utterance: 1.0 max_prompts: 1 prompt_duration: 3.0 max_resps: 1 p_resp_append: 0.25 sample_type: path # path | speaker | group sample_order: duration # shuffle | duration sample_max_duration_batch: 0 # used when above = duration, 120 seconds per batch at 12GiB of VRAM works tasks_list: [ "tts" ] # , [ "tts", "tts-c", "ns", "sr", "tse", "cse", "nse", "tts"] training: [] validation: [] noise: []