Upload UltravoxPipeline
Browse files- README.md +9 -9
- config.json +2 -2
- generation_config.json +1 -1
- tokenizer.json +2 -2
- ultravox_config.py +1 -3
- ultravox_model.py +32 -24
- ultravox_processing.py +6 -1
README.md
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---
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language:
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- ar
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- de
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- tr
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- uk
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- zh
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license: mit
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library_name: transformers
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-
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- fixie-ai/librispeech_asr
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- fixie-ai/common_voice_17_0
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- fixie-ai/peoples_speech
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- fixie-ai/gigaspeech
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- fixie-ai/multilingual_librispeech
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- fixie-ai/wenetspeech
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- fixie-ai/covost2
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metrics:
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- bleu
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---
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---
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datasets:
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- fixie-ai/librispeech_asr
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- fixie-ai/common_voice_17_0
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- fixie-ai/peoples_speech
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- fixie-ai/gigaspeech
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- fixie-ai/multilingual_librispeech
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- fixie-ai/wenetspeech
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- fixie-ai/covost2
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language:
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- ar
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- de
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- tr
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- uk
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- zh
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library_name: transformers
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license: mit
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metrics:
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- bleu
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---
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config.json
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{
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"_name_or_path": "/
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"architectures": [
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"UltravoxModel"
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],
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"stack_factor": 8,
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"text_model_id": "meta-llama/Meta-Llama-3.1-8B-Instruct",
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"torch_dtype": "bfloat16",
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"transformers_version": "4.
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"vocab_size": 128256
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}
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{
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"_name_or_path": "/Users/zhuang/expts/2024-10-09-v0_4_1/stacking-4b/ultravox/artifacts/model-zhuang.2024-10-09-v0_4_1.stacking-4b.8c44a2e:v8",
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"architectures": [
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"UltravoxModel"
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],
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"stack_factor": 8,
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"text_model_id": "meta-llama/Meta-Llama-3.1-8B-Instruct",
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"torch_dtype": "bfloat16",
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"transformers_version": "4.44.0",
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"vocab_size": 128256
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}
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generation_config.json
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128009
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],
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"pad_token_id": 128009,
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"transformers_version": "4.
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}
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128009
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],
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"pad_token_id": 128009,
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"transformers_version": "4.44.0"
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}
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tokenizer.json
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version https://git-lfs.github.com/spec/v1
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oid sha256:
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size
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version https://git-lfs.github.com/spec/v1
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oid sha256:79e3e522635f3171300913bb421464a87de6222182a0570b9b2ccba2a964b2b4
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size 9085657
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ultravox_config.py
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target_modules: Optional[List[str]] = dataclasses.field(
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default_factory=lambda: ["k_proj", "q_proj", "linear_k", "linear_q"]
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)
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# A list of module names regex patterns to unfreeze. Only used if r == 0.
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unfreeze_layers: Optional[List[str]] = None
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class LossFunction(str, Enum):
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@dataclasses.dataclass
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class LossConfig:
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loss_function: LossFunction = LossFunction.
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kl_temperature: float = 2.0
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@property
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target_modules: Optional[List[str]] = dataclasses.field(
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default_factory=lambda: ["k_proj", "q_proj", "linear_k", "linear_q"]
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)
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class LossFunction(str, Enum):
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@dataclasses.dataclass
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class LossConfig:
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loss_function: LossFunction = LossFunction.KL_Divergence
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kl_temperature: float = 2.0
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@property
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ultravox_model.py
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import logging
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import re
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from typing import Any, Dict, Optional, Set, Tuple, Union
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import peft
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config_class = UltravoxConfig
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config: UltravoxConfig # for type hinting
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#
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# As such we have to tell is to ignore some keys that are not always in the model
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_keys_to_ignore_on_load_unexpected = ["audio_tower.*", "language_model.*"]
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# Usually we load encoder weights from a pretrained model, so we don't want to load the decoder weights
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# Technically we never hit this issue because these keys are already removed from state_dict() however,
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# but there's no harm in keeping it here for when we change that behavior.
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_keys_to_ignore_on_load_missing = ["audio_tower.*"]
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def __init__(self, config: UltravoxConfig):
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super().__init__(config)
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labels: Optional[torch.Tensor] = None,
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attention_mask: Optional[torch.Tensor] = None,
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audio_token_start_idx: Optional[torch.Tensor] = None,
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audio_token_len: Optional[torch.Tensor] = None,
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past_key_values: Optional[Union[Tuple, transformers.cache_utils.Cache]] = None,
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# the alt_* fields are needed for KL divergence loss
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# B x A/3200 x D
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audio_tower_output = self.audio_tower.forward(
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audio_values.to(self.audio_tower.dtype)
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).last_hidden_state
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audio_tower_output = audio_tower_output.to(inputs_embeds.dtype)
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audio_values: Optional[torch.FloatTensor] = None,
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audio_token_start_idx: Optional[torch.Tensor] = None,
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audio_token_len: Optional[torch.Tensor] = None,
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past_key_values: Optional[Union[Tuple, transformers.cache_utils.Cache]] = None,
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attention_mask: Optional[torch.Tensor] = None,
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inputs_embeds: Optional[torch.Tensor] = None,
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audio_token_start_idx - prefill_start_idx
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)
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model_input["audio_token_len"] = audio_token_len
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return model_input
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def push_to_hub(self, *args, **kwargs):
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self.merge_and_unload()
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return super().push_to_hub(*args, **kwargs)
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def save_pretrained(
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)
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# TODO: refactor common parts to a shared module
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def is_cache_empty(
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past_key_values: Optional[Union[Tuple, transformers.cache_utils.Cache]]
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) -> bool:
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"""
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Applies LoRA finetuning to the model. If the `r` parameter is set to 0, the model is frozen instead.
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"""
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unfreeze_layers = lora_config.pop("unfreeze_layers", None)
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lora_config = peft.LoraConfig(**lora_config or {})
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if lora_config.r == 0:
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# freeze the model entirely
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for
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re.match(layer, name) for layer in unfreeze_layers
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):
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param.requires_grad = False
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else:
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logging.info(f"Unfreezing layer: {name} with #{param.numel()} params")
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else:
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model = peft.get_peft_model(model, lora_config)
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return hidden_states
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-
class ModifiedWhisperEncoder(whisper.WhisperEncoder):
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"""
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Encoder portion of OpenAI's Whisper model.
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def forward(
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self,
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input_features,
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-
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head_mask=None,
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output_attentions=None,
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output_hidden_states=None,
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encoder_states = () if output_hidden_states else None
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all_attentions = () if output_attentions else None
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# check if head_mask has a correct number of layers specified if desired
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if head_mask is not None:
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assert head_mask.size()[0] == (
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layer_outputs = self._gradient_checkpointing_func(
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encoder_layer.__call__,
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hidden_states,
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-
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(head_mask[idx] if head_mask is not None else None),
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output_attentions,
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)
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else:
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layer_outputs = encoder_layer(
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hidden_states,
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-
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layer_head_mask=(
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head_mask[idx] if head_mask is not None else None
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),
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import logging
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from typing import Any, Dict, Optional, Set, Tuple, Union
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import peft
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config_class = UltravoxConfig
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config: UltravoxConfig # for type hinting
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+
# Usually we load encoder and LLM weights from a pretrained model separately, so they are allowed to be missing
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+
_keys_to_ignore_on_load_missing = ["audio_tower.*", "language_model.*"]
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def __init__(self, config: UltravoxConfig):
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super().__init__(config)
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labels: Optional[torch.Tensor] = None,
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attention_mask: Optional[torch.Tensor] = None,
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audio_token_start_idx: Optional[torch.Tensor] = None,
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+
audio_len: Optional[torch.Tensor] = None,
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audio_token_len: Optional[torch.Tensor] = None,
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past_key_values: Optional[Union[Tuple, transformers.cache_utils.Cache]] = None,
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# the alt_* fields are needed for KL divergence loss
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# B x A/3200 x D
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audio_tower_output = self.audio_tower.forward(
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audio_values.to(self.audio_tower.dtype),
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audio_len = audio_len
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).last_hidden_state
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audio_tower_output = audio_tower_output.to(inputs_embeds.dtype)
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audio_values: Optional[torch.FloatTensor] = None,
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audio_token_start_idx: Optional[torch.Tensor] = None,
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audio_token_len: Optional[torch.Tensor] = None,
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+
audio_len: Optional[torch.Tensor] = None,
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past_key_values: Optional[Union[Tuple, transformers.cache_utils.Cache]] = None,
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attention_mask: Optional[torch.Tensor] = None,
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inputs_embeds: Optional[torch.Tensor] = None,
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audio_token_start_idx - prefill_start_idx
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)
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model_input["audio_token_len"] = audio_token_len
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+
model_input["audio_len"] = audio_len
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return model_input
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def push_to_hub(self, *args, **kwargs):
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self.merge_and_unload()
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+
self.to(self.language_model.dtype)
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return super().push_to_hub(*args, **kwargs)
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def save_pretrained(
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)
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def is_cache_empty(
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past_key_values: Optional[Union[Tuple, transformers.cache_utils.Cache]]
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) -> bool:
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"""
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Applies LoRA finetuning to the model. If the `r` parameter is set to 0, the model is frozen instead.
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"""
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lora_config = peft.LoraConfig(**lora_config or {})
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if lora_config.r == 0:
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+
# freeze the model entirely
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+
for param in model.parameters():
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param.requires_grad = False
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else:
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model = peft.get_peft_model(model, lora_config)
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return hidden_states
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+
class ModifiedWhisperEncoder(whisper.WhisperEncoder, transformers.modeling_utils.ModuleUtilsMixin):
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"""
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Encoder portion of OpenAI's Whisper model.
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def forward(
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self,
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input_features,
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+
audio_len=None,
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head_mask=None,
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output_attentions=None,
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output_hidden_states=None,
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encoder_states = () if output_hidden_states else None
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all_attentions = () if output_attentions else None
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+
attention_mask = None
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+
if audio_len != None:
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+
audio_feature_len = self._get_feat_extract_output_lengths(audio_len)
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+
batch_size = hidden_states.shape[0]
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+
max_seq_len = hidden_states.shape[1]
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+
attention_mask = (
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+
torch.arange(max_seq_len, device=hidden_states.device)[None, :]
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+
.expand(batch_size, -1)
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+
.lt(audio_feature_len.view(batch_size, 1))
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+
)
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+
attention_mask = self.get_extended_attention_mask(
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+
attention_mask,
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+
None,
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+
device=hidden_states.device,
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+
dtype=hidden_states.dtype,
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+
)
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+
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# check if head_mask has a correct number of layers specified if desired
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if head_mask is not None:
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assert head_mask.size()[0] == (
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layer_outputs = self._gradient_checkpointing_func(
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encoder_layer.__call__,
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hidden_states,
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+
attention_mask,
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(head_mask[idx] if head_mask is not None else None),
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output_attentions,
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)
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else:
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layer_outputs = encoder_layer(
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hidden_states,
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+
attention_mask,
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layer_head_mask=(
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head_mask[idx] if head_mask is not None else None
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),
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ultravox_processing.py
CHANGED
@@ -62,7 +62,7 @@ class UltravoxProcessor(transformers.ProcessorMixin):
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super().__init__(audio_processor=audio_processor, tokenizer=tokenizer)
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@classmethod
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-
def from_pretrained(cls, pretrained_model_name_or_path, **kwargs):
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config: UltravoxConfig = transformers.AutoConfig.from_pretrained(
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pretrained_model_name_or_path, **kwargs
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)
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sampling_rate=sampling_rate,
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padding="longest",
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max_length=audio_len,
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**kwargs,
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)
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if "input_features" in x:
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data["audio_values"] = x.input_features
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else:
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data["audio_values"] = x.input_values
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if text is not None:
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assert isinstance(
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super().__init__(audio_processor=audio_processor, tokenizer=tokenizer)
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@classmethod
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+
def from_pretrained(cls, pretrained_model_name_or_path: str, **kwargs):
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config: UltravoxConfig = transformers.AutoConfig.from_pretrained(
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pretrained_model_name_or_path, **kwargs
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)
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sampling_rate=sampling_rate,
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padding="longest",
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max_length=audio_len,
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+
return_attention_mask=True,
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**kwargs,
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)
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if "input_features" in x:
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data["audio_values"] = x.input_features
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else:
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data["audio_values"] = x.input_values
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+
if self.audio_padding == "max_length":
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+
data["audio_len"] = x.attention_mask.sum(-1) - 1
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+
else:
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+
data["audio_len"] = [data["audio_values"].shape[-1]]
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if text is not None:
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assert isinstance(
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