mrq
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README.md
CHANGED
@@ -7,3 +7,34 @@ This repo catalogs my weights for use with my [VALL-E](https://github.com/e-c-k-
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The model currently is in a *semi-usable* state, and I'm releasing them now in hopes that it also helps jumpstart anyone else that wants to use them.
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To reiterate, this is ***by no means*** complete. I am not passing this off as competitive.
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The model currently is in a *semi-usable* state, and I'm releasing them now in hopes that it also helps jumpstart anyone else that wants to use them.
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To reiterate, this is ***by no means*** complete. I am not passing this off as competitive.
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## Models
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* `config.retnet.yaml` / `ar+nar-retnet-8`: The previously released weights.
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+ This configuration utilizes a RetNet (retention based transformer) as the underlying architecture due to a number of misleading interpretations with comparisons, for better or for worse.
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+ Prompt and response embeddings are summed (further RVQ levels gets the previous RVQ levels' embeddings factored in).
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+ Tokenizer is a homebrewed "naive" implementation.
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+ This model received the most training time between my 4070Ti, 7900XTX, and a few rental rigs to training further progress, entirely at `bfloat16` with `prodigyopt` (and a few optimizer restarts).
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+ The later part of training aimed to shuffle between speakers rather than the global pool of utterances to better focus on zero-shot performance. Due to this, I feel it achieved *decent* zero-shot performance.
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+ However, due to the dataset being aggressively trimmed under 12 seconds for memory savings during training, it suffers trying to inference non-short utterances. Additional training may fix this, the following models seemed to adapt well to longer utterances.
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+ Prior testing showed that longer prompt durations results in better utterances.
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* `config.llama.yaml` / `ar+nar-llama-8`: The most recent-ishly trained weights after learning from my mistakes.
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+ This configuration utilizes Llama's attention-based transformer as the underlying architecture, making use of creature comforts like RoPE, GQA, and memory-efficient attention (trained under `xformers`, shouldn't really affect things).
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+ Prompt and response embeddings are NOT summed (each RVQ level only attends to the current RVQ level).
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+ Utilizes a HF tokenizer for "optimal" vocab.
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+ The current RVQ level is included as a token as well to help guide NAR tasks better.
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+ This model received a few days of training on my 4xV100s, stepping up the duration window to *try* and better make the model inference for longer utterances.
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+ Some sessions end up training the current duration window for a few epochs, but I don't know how much it affected things.
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+ However, it seems to *only* do well with long utterances. Short utterances fumble. I believe further training with a variety of durations should allow the AR to handle a variety of durations.
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- I believe the "slowly stepping up the context length" only works for text, and not audio.
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+ Zero-shot performance leaves a bit to be desired, as it did not receive the special training prioritizing shuffling between speakers rather than the global pool of utterances.
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+ Testing showed that, despite also stepping up the prompt duration, it *really* likes three second prompts.
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+ Definitely needs additional training.
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* `config.llama.split.yaml` / `ar-llama-1` + `nar-llama-8`: The above model, but split and trained a little bit more.
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+ This experiment is to see whether the AR and NAR benefitted from being split up after enough pretraining, to un-"lobotomize" any penalties from attending to two different tasks (as the AR predicts the next token, and the NAR predicts the same token but a different level).
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+ I believe I trained each separate model an additional extra day for another additional audio-duration window for similar training lengths.
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+ I don't think audio quality differs a non-trivial amount to warrant splitting the model.
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There's a bunch of additional configurations (between the underlying arch, embedding modes, interleaving, and even a NAR-"only" model) that are to be further explored, but current experiments showed they either are not worth the additional performance penalties (interleaving) or fall flat (NAR-"only", chunked interleaving).
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{old β model}/ckpt/ar+nar-retnet-8/fp32.pth
RENAMED
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model/{config.split.yaml β config.llama-split.yaml}
RENAMED
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model/{config.yaml β config.llama.yaml}
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old/config.ar_nar.yaml β model/config.retnet.yaml
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noise: []
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speaker_name_getter: "lambda p: f'{p.parts[-3]}_{p.parts[-2]}'"
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use_hdf5: True
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use_metadata: True
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hdf5_flag: r
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validate: True
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workers: 2
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cache: True
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phones_range: [4, 256]
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duration_range: [1.0, 16.0]
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random_utterance: 1.0
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max_prompts: 3
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prompt_duration: 6.0
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sample_type: speaker
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tasks_list: [ "tts" ] # , [ "tts", "tts-c", "ns", "sr", "tse", "cse", "nse", "tts"]
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models:
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hyperparameters:
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optimizer: Prodigy
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torch_optimizer: True
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learning_rate: 1.0
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#scheduler_params:
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# cycle_first_step_size: 10_000
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# cycle_first_stair_count: 10_000
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# cycle_second_step_size: 15_000
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# cycle_second_stair_count: 15_000
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# decay_step_size: 5_000
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# cycle_min_lr: 2.5e-4 # 1.0e-5
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# cycle_max_lr: 2.5e-4 # 1.0e-4
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# decay_lr_rate: 0.0
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# cycle_min_mom: 0.90
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# cycle_max_mom: 0.99
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# decay_mom_rate: 0.0
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evaluation:
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batch_size: 16
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frequency:
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size: 16
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steps:
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ar_temperature: 0.95
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nar_temperature: 0.25
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load_disabled_engines: True
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trainer:
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iterations: 1_000_000
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save_tag: step
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save_on_oom: True
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save_on_quit: True
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save_frequency:
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export_on_save: True
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keep_last_checkpoints:
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aggressive_optimizations: False
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load_disabled_engines: False
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#load_state_dict: True
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#load_tag: "9500"
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#load_states: False
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#restart_step_count: True
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gc_mode: None # "global_step"
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weight_dtype: bfloat16
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amp:
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backend: deepspeed
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deepspeed:
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zero_optimization_level: 0
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use_compression_training:
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inference:
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normalize: False
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weight_dtype: bfloat16
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amp:
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sample_rate: 24_000
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audio_backend: vocos
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experimental: True
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models:
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- name: "ar+nar"
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size: "full"
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resp_levels: 8
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prom_levels: 8
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tasks: 8
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langs: 2
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tones: 1
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arch_type: retnet
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training: False
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version: 2
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dropout: 0.1
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audio_embedding_sums: True
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interleave: False
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experimental: False
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capabilities: ["ar", "nar"]
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hyperparameters:
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autotune: False
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autotune_params:
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start_profile_step: 1
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end_profile_step: 50
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num_tuning_micro_batch_sizes: 8
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batch_size: 16
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gradient_accumulation_steps: 8
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gradient_clipping: 1.0
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warmup_steps: 250
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optimizer: Prodigy
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learning_rate: 1.0
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torch_optimizer: True
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scheduler: "" # ScheduleFree
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torch_scheduler: True
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evaluation:
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batch_size: 16
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frequency: 1000
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size: 16
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steps: 500
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ar_temperature: 0.95
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nar_temperature: 0.25
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load_disabled_engines: True
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trainer:
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#no_logger: True
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ddp: False
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check_for_oom: False
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iterations: 1_000_000
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save_tag: step
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save_on_oom: True
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save_on_quit: True
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save_frequency: 500
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export_on_save: True
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keep_last_checkpoints: 8
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aggressive_optimizations: False
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load_disabled_engines: False
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gradient_checkpointing: True
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#load_state_dict: True
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strict_loading: False
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#load_tag: "9500"
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#load_states: False
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#restart_step_count: True
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gc_mode: None # "global_step"
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weight_dtype: bfloat16
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amp: True
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backend: deepspeed
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deepspeed:
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inferencing: True
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zero_optimization_level: 0
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use_compression_training: False
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amp: False
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load_webui: False
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inference:
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backend: deepspeed
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audio_backend: "vocos"
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normalize: False
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weight_dtype: bfloat16
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amp: True
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optimizations:
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injects: False
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replace: True
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linear: False
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embedding: False
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optimizers: True
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bitsandbytes: False
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dadaptation: False
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bitnet: False
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fp8: False
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dataset:
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speaker_name_getter: "lambda p: f'{p.parts[-3]}_{p.parts[-2]}'"
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speaker_group_getter: "lambda p: f'{p.parts[-3]}'"
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speaker_languages:
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ja: []
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use_hdf5: True
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use_metadata: True
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hdf5_flag: r
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validate: True
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workers: 6
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cache: True
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duration_range: [3.0, 16.0]
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random_utterance: 1.0
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max_prompts: 1
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prompt_duration_range: [3.0, 9.0]
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max_resps: 1
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p_resp_append: 0.25
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sample_type: path # path # speaker
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tasks_list: [ "tts" ] # , [ "tts", "tts-c", "ns", "sr", "tse", "cse", "nse", "tts"]
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training: []
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validation: []
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noise: []
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old/config.yaml
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