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+# Adapted from https://github.com/fixie-ai/ultravox/blob/ecd58c4041030bae2ad15aa6bcf04ab43199ea02/ultravox/model/ultravox_model.py
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+"""PyTorch Ultravox model."""
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+
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+import itertools
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+import math
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+from array import array
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+from functools import lru_cache
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+from typing import (Iterable, List, Literal, Mapping, Optional, Tuple,
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+ TypedDict, Union, cast)
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+
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+import librosa
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+import numpy as np
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+import torch
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+import torch.utils.checkpoint
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+from torch import nn
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+from torch.nn import functional as F
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+from transformers.models.whisper import WhisperFeatureExtractor
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+from transformers.models.whisper.modeling_whisper import WhisperEncoder
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+
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+from aphrodite.attention import AttentionMetadata
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+from aphrodite.common.config import CacheConfig, MultiModalConfig
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+from aphrodite.common.sequence import (APHRODITE_TOKEN_ID_ARRAY_TYPE,
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+ SamplerOutput, SequenceData)
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+from aphrodite.inputs import INPUT_REGISTRY
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+from aphrodite.inputs.data import LLMInputs
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+from aphrodite.inputs.registry import InputContext
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+from aphrodite.modeling.layers.activation import SiluAndMul, get_act_fn
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+from aphrodite.modeling.layers.layernorm import RMSNorm
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+from aphrodite.modeling.model_loader.weight_utils import default_weight_loader
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+from aphrodite.modeling.models.interfaces import SupportsMultiModal
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+from aphrodite.modeling.models.utils import (filter_weights,
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+ init_aphrodite_registered_model,
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+ merge_multimodal_embeddings)
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+from aphrodite.modeling.sampling_metadata import SamplingMetadata
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+from aphrodite.multimodal import MULTIMODAL_REGISTRY
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+from aphrodite.multimodal.base import MultiModalInputs
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+from aphrodite.multimodal.utils import (cached_get_tokenizer,
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+ repeat_and_pad_placeholder_tokens)
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+from aphrodite.quantization.base_config import QuantizationConfig
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+from aphrodite.transformers_utils.configs.ultravox import UltravoxConfig
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+
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+_AUDIO_PLACEHOLDER_TOKEN = 128002
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+_AUDIO_TOKENS_PER_SECOND = 6.25
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+
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+
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+class UltravoxAudioFeatureInputs(TypedDict):
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+ type: Literal["audio_features"]
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+ data: Union[torch.Tensor, List[torch.Tensor]]
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+ """Shape: `(batch_size, 80, M)"""
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+
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+
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+class UltravoxAudioEmbeddingInputs(TypedDict):
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+ type: Literal["audio_embeds"]
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+ data: torch.Tensor
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+
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+
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+UltravoxAudioInputs = Union[UltravoxAudioFeatureInputs,
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+ UltravoxAudioEmbeddingInputs]
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+
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+
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+@lru_cache
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+def cached_feature_extractor(model_id: str) -> WhisperFeatureExtractor:
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+ return WhisperFeatureExtractor.from_pretrained(model_id)
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+
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+
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+def whisper_feature_extractor(ctx: InputContext) -> WhisperFeatureExtractor:
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+ return cached_feature_extractor(
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+ ctx.get_hf_config(UltravoxConfig).audio_model_id)
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+
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+
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+def get_ultravox_max_audio_tokens(ctx: InputContext):
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+ feature_extractor = whisper_feature_extractor(ctx)
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+ return math.ceil(feature_extractor.chunk_length * _AUDIO_TOKENS_PER_SECOND)
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+
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+
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+def dummy_data_for_ultravox(
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+ ctx: InputContext,
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+ seq_len: int,
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+ mm_counts: Mapping[str, int],
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+):
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+ feature_extractor = whisper_feature_extractor(ctx)
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+
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+ audio_count = mm_counts["audio"]
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+
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+ audio_token_ids = array(APHRODITE_TOKEN_ID_ARRAY_TYPE, [
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+ _AUDIO_PLACEHOLDER_TOKEN
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+ ]) * get_ultravox_max_audio_tokens(ctx) * audio_count
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+ other_token_ids = array(APHRODITE_TOKEN_ID_ARRAY_TYPE,
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+ [0]) * (seq_len - len(audio_token_ids))
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+
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+ audio_and_sr = (np.array([0.0] * feature_extractor.chunk_length), 1)
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+ mm_dict = {
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+ "audio":
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+ audio_and_sr if audio_count == 1 else [audio_and_sr] * audio_count
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+ }
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+
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+ return (SequenceData(audio_token_ids + other_token_ids), mm_dict)
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+
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+
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+def input_mapper_for_ultravox(ctx: InputContext, data: object):
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+ if isinstance(data, tuple):
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+ (audio, sr) = cast(Tuple[np.ndarray, Union[float, int]], data)
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+ feature_extractor = whisper_feature_extractor(ctx)
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+
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+ if sr != feature_extractor.sampling_rate:
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+ audio = librosa.resample(audio,
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+ orig_sr=sr,
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+ target_sr=feature_extractor.sampling_rate)
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+ sr = feature_extractor.sampling_rate
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+
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+ minimum_audio_length = feature_extractor.n_fft // 2 + 1
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+ if len(audio) < minimum_audio_length:
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+ # Not enough audio; pad it.
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+ audio = np.pad(audio, (0, minimum_audio_length - len(audio)))
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+
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+ return MultiModalInputs({
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+ "audio_features":
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+ feature_extractor(audio,
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+ sampling_rate=sr,
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+ padding="longest",
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+ return_tensors="pt")["input_features"]
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+ })
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+
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+ raise NotImplementedError(f"Unsupported data type: {type(data)}")
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+
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+
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+def input_processor_for_ultravox(ctx: InputContext, llm_inputs: LLMInputs):
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+ multi_modal_data = llm_inputs.get("multi_modal_data")
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+ if multi_modal_data is None or "audio" not in multi_modal_data:
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+ return llm_inputs
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+
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+ feature_extractor = whisper_feature_extractor(ctx)
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+ audio_data, sample_rate = multi_modal_data["audio"]
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+
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+ audio_length = audio_data.shape[0]
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+ if sample_rate != feature_extractor.sampling_rate:
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+ # Account for resampling.
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+ adjustment = feature_extractor.sampling_rate / sample_rate
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+ audio_length = math.ceil(adjustment * audio_length)
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+
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+ feature_extractor_output_length = math.ceil(
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+ (audio_length -
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+ (feature_extractor.hop_length - 1)) / feature_extractor.hop_length)
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+
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+ uv_config = ctx.get_hf_config(UltravoxConfig)
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+ audio_num_tokens = min(
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+ max(
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+ 1,
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+ math.ceil(feature_extractor_output_length /
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+ (uv_config.stack_factor * 2))),
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+ get_ultravox_max_audio_tokens(ctx))
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+ tokenizer = cached_get_tokenizer(ctx.model_config.tokenizer)
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+
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+ new_prompt, new_token_ids = repeat_and_pad_placeholder_tokens(
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+ tokenizer,
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+ llm_inputs.get("prompt"),
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+ llm_inputs["prompt_token_ids"],
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+ placeholder_token_id=_AUDIO_PLACEHOLDER_TOKEN,
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+ repeat_count=audio_num_tokens,
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+ )
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+
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+ # NOTE: Create a defensive copy of the original inputs
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+ return LLMInputs(prompt_token_ids=new_token_ids,
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+ prompt=new_prompt,
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+ multi_modal_data=multi_modal_data)
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+
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+
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+class StackAudioFrames(nn.Module):
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+ """
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+ Stack the audio embedding frames to reduce the sequence length by a factor
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+ of `stack_factor`.
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+ """
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+
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+ def __init__(self, stack_factor: int = 8):
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+ super().__init__()
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+ self.stack_factor = stack_factor
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+
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+ def forward(self, audio_embeds: torch.Tensor) -> torch.Tensor:
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+ B, T, C = audio_embeds.shape
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+ T_pad = (T + self.stack_factor -
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+ 1) // self.stack_factor * self.stack_factor
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+ audio_embeds = F.pad(audio_embeds, (0, 0, 0, T_pad - T))
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+ B, T, C = audio_embeds.shape
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+ audio_embeds = audio_embeds.view(B, T // self.stack_factor,
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+ C * self.stack_factor)
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+ return audio_embeds
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+
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+
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+class FlippedSiluAndMul(SiluAndMul):
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+ """Ultravox is trained with SwiGLU with flipped halves."""
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+
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+ def forward(self, x: torch.Tensor):
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+ a, b = x.chunk(2, dim=-1)
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+ flipped = torch.cat((b, a), dim=-1)
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+ return super().forward(flipped)
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+
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+
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+class UltravoxProjector(nn.Module):
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+
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+ def __init__(self, config: UltravoxConfig):
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+ super().__init__()
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+ self.hidden_dim = config.hidden_size
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+ self._pad_and_stack = StackAudioFrames(config.stack_factor)
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+ dim = config.audio_config.hidden_size * config.stack_factor
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+ self.ln_pre = RMSNorm(dim)
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+ self.linear_1 = nn.Linear(dim, self.hidden_dim, bias=False)
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+ dim = self.hidden_dim
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+
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+ if config.projector_act == "swiglu":
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+ self.act = FlippedSiluAndMul()
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+ dim = dim // 2
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+ else:
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+ self.act = get_act_fn(config.projector_act)
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+
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+ self.linear_2 = nn.Linear(dim,
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+ config.text_config.hidden_size,
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+ bias=False)
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+ self.ln_post = RMSNorm(config.text_config.hidden_size)
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+
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+ def forward(self, audio_features: torch.Tensor) -> torch.Tensor:
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+ audio_features = self._pad_and_stack(audio_features)
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+ audio_features = self.ln_pre(audio_features)
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+ hidden_states = self.linear_1(audio_features)
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+ hidden_states = self.act(hidden_states)
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+ hidden_states = self.linear_2(hidden_states)
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+ hidden_states = self.ln_post(hidden_states)
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+ return hidden_states
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+
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+
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+class ModifiedWhisperEncoder(WhisperEncoder):
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+ """
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+ Encoder portion of OpenAI's Whisper model.
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+
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+ This implementation is a slightly modified version of HF Transformers'
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+ Whisper Encoder, with only a few fixes:
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+ 1. base_model_prefix updated to allow for doing `.from_pretrained`
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+ directly on the encoder
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+ 2. allow less than 30 second of audio padding to be passed in:
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+ - relaxed ValueError check for `input_features` length to be less
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+ than or equal to `expected_seq_length` instead of strictly equal
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+ - embed_pos is now sliced to match the length of `inputs_embeds`
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+
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+ Original: https://github.com/huggingface/transformers/blob/main/src/transformers/models/whisper/modeling_whisper.py
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+ See commentary: https://github.com/huggingface/transformers/issues/25744
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+ """
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+
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+ base_model_prefix = "model.encoder"
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+
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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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+ expected_seq_length = (self.config.max_source_positions *
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+ self.conv1.stride[0] * self.conv2.stride[0])
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+ if input_features.shape[-1] > expected_seq_length:
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+ raise ValueError(
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+ f"Whisper expects the mel input features to be of length "
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+ f"{expected_seq_length} or less, but found "
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+ f"{input_features.shape[-1]}. Make sure to pad the input mel "
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+ f"features to {expected_seq_length}.")
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+
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+ inputs_embeds = nn.functional.gelu(self.conv1(input_features))
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+ inputs_embeds = nn.functional.gelu(self.conv2(inputs_embeds))
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+
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+ inputs_embeds = inputs_embeds.permute(0, 2, 1)
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+ embed_pos = self.embed_positions.weight[:inputs_embeds.size(-2)]
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+
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+ hidden_states = inputs_embeds + embed_pos
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+ hidden_states = nn.functional.dropout(hidden_states,
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+ p=self.dropout,
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+ training=self.training)
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+
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+ for encoder_layer in self.layers:
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+ layer_outputs = encoder_layer(
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+ hidden_states,
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+ None,
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+ layer_head_mask=None,
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+ )
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+
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+ hidden_states = layer_outputs[0]
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+
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+ hidden_states = self.layer_norm(hidden_states)
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+ return hidden_states
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+
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+
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+@MULTIMODAL_REGISTRY.register_input_mapper("audio", input_mapper_for_ultravox)
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+@MULTIMODAL_REGISTRY.register_max_multimodal_tokens(
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+ "audio", get_ultravox_max_audio_tokens)
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+@INPUT_REGISTRY.register_dummy_data(dummy_data_for_ultravox)
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+@INPUT_REGISTRY.register_input_processor(input_processor_for_ultravox)
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+class UltravoxModel(nn.Module, SupportsMultiModal):
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+
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+ def __init__(self,
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+ config: UltravoxConfig,
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+ multimodal_config: MultiModalConfig,
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+ cache_config: Optional[CacheConfig] = None,
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+ quant_config: Optional["QuantizationConfig"] = None):
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+ super().__init__()
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+ self.config = config
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+ self.multi_modal_config = multimodal_config
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+ assert self.multi_modal_config
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+
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+ if config.audio_model_id is not None:
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+ self.audio_tower = ModifiedWhisperEncoder.from_pretrained(
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+ config.audio_model_id)
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+ else:
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+ self.audio_tower = ModifiedWhisperEncoder(config.audio_config)
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+ self.multi_modal_projector = UltravoxProjector(config)
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+ self.language_model = init_aphrodite_registered_model(
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+ config.text_config, cache_config, quant_config)
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+
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+ def _audio_features_to_embeddings(
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+ self, input_features: torch.Tensor) -> torch.Tensor:
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+ audio_input = input_features.to(self.audio_tower.dtype)
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+ audio_features = self.audio_tower(audio_input)
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+ audio_features = audio_features.to(self.audio_tower.dtype)
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+ audio_embeddings = self.multi_modal_projector(audio_features)
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+ return audio_embeddings
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+
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+ def _parse_and_validate_audio_input(
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+ self, **kwargs: object) -> Optional[UltravoxAudioInputs]:
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+ audio_features = kwargs.pop("audio_features", None)
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+ audio_embeds = kwargs.pop("audio_embeds", None)
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+
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+ if audio_features is None and audio_embeds is None:
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+ return None
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+
|
|
|
|
+ if audio_features is not None:
|
|
|
|
+ if not isinstance(audio_features, (torch.Tensor, list)):
|
|
|
|
+ raise ValueError("Incorrect type of audio features. "
|
|
|
|
+ f"Got type: {type(audio_features)}")
|
|
|
|
+
|
|
|
|
+ return UltravoxAudioFeatureInputs(type="audio_features",
|
|
|
|
+ data=audio_features)
|
|
|
|
+
|
|
|
|
+ if audio_embeds is not None:
|
|
|
|
+ if not isinstance(audio_embeds, torch.Tensor):
|
|
|
|
+ raise ValueError("Incorrect type of audio embeds. "
|
|
|
|
+ f"Got type: {type(audio_embeds)}")
|
|
|
|
+
|
|
|
|
+ return UltravoxAudioEmbeddingInputs(type="audio_embeds",
|
|
|
|
+ data=audio_embeds)
|
|
|
|
+
|
|
|
|
+ raise AssertionError("This line should be unreachable.")
|
|
|
|
+
|
|
|
|
+ def _process_audio_input(
|
|
|
|
+ self, audio_input: UltravoxAudioInputs
|
|
|
|
+ ) -> Union[torch.Tensor, List[torch.Tensor]]:
|
|
|
|
+ if audio_input["type"] == "audio_embeds":
|
|
|
|
+ return audio_input["data"]
|
|
|
|
+
|
|
|
|
+ audio_features = audio_input["data"]
|
|
|
|
+ if isinstance(audio_features, list):
|
|
|
|
+ # TODO: Batch these through the encoder/projector instead of
|
|
|
|
+ # serializing them.
|
|
|
|
+ return [
|
|
|
|
+ self._audio_features_to_embeddings(
|
|
|
|
+ features.unsqueeze(0)).squeeze(0)
|
|
|
|
+ for features in audio_features
|
|
|
|
+ ]
|
|
|
|
+ else:
|
|
|
|
+ return self._audio_features_to_embeddings(audio_features)
|
|
|
|
+
|
|
|
|
+ def forward(self, input_ids: torch.Tensor, positions: torch.Tensor,
|
|
|
|
+ kv_caches: List[torch.Tensor],
|
|
|
|
+ attn_metadata: AttentionMetadata,
|
|
|
|
+ intermediate_tensors: Optional[torch.Tensor],
|
|
|
|
+ **kwargs) -> SamplerOutput:
|
|
|
|
+ """Run forward pass for Ultravox
|
|
|
|
+
|
|
|
|
+ One key thing to understand is the `input_ids` already accounts for the
|
|
|
|
+ positions of the to-be-inserted audio embeddings. The to-be-inserted
|
|
|
|
+ audio has a size that is essentially 6.25 tokens per second of audio.
|
|
|
|
+
|
|
|
|
+ This way, the `positions` and `attn_metadata` are consistent
|
|
|
|
+ with the `input_ids`.
|
|
|
|
+
|
|
|
|
+ Args:
|
|
|
|
+ input_features: A batch of audio inputs, [1, 80, M].
|
|
|
|
+ """
|
|
|
|
+ audio_input = self._parse_and_validate_audio_input(**kwargs)
|
|
|
|
+ if audio_input is not None:
|
|
|
|
+ audio_embeddings = self._process_audio_input(audio_input)
|
|
|
|
+ inputs_embeds = self.language_model.model.get_input_embeddings(
|
|
|
|
+ input_ids)
|
|
|
|
+
|
|
|
|
+ inputs_embeds = merge_multimodal_embeddings(
|
|
|
|
+ input_ids, inputs_embeds, audio_embeddings,
|
|
|
|
+ _AUDIO_PLACEHOLDER_TOKEN)
|
|
|
|
+ input_ids = None
|
|
|
|
+ else:
|
|
|
|
+ inputs_embeds = None
|
|
|
|
+
|
|
|
|
+ hidden_states = self.language_model.model(
|
|
|
|
+ input_ids=input_ids,
|
|
|
|
+ positions=positions,
|
|
|
|
+ kv_caches=kv_caches,
|
|
|
|
+ attn_metadata=attn_metadata,
|
|
|
|
+ intermediate_tensors=intermediate_tensors,
|
|
|
|
+ inputs_embeds=inputs_embeds)
|
|
|
|
+ return hidden_states
|
|
|
|
+
|
|
|
|
+ def compute_logits(self, hidden_states: torch.Tensor,
|
|
|
|
+ sampling_metadata: SamplingMetadata) -> torch.Tensor:
|
|
|
|
+ return self.language_model.compute_logits(hidden_states,
|
|
|
|
+ sampling_metadata)
|
|
|
|
+
|
|
|
|
+ def sample(
|
|
|
|
+ self,
|
|
|
|
+ logits: torch.Tensor,
|
|
|
|
+ sampling_metadata: SamplingMetadata,
|
|
|
|
+ ) -> Optional[SamplerOutput]:
|
|
|
|
+ return self.language_model.sample(logits, sampling_metadata)
|
|
|
|
+
|
|
|
|
+ def load_weights(self, weights: Iterable[Tuple[str, torch.Tensor]]):
|
|
|
|
+ # prepare weight iterators for components
|
|
|
|
+ projector_weights, llm_weights = itertools.tee(weights, 2)
|
|
|
|
+
|
|
|
|
+ # load projector weights
|
|
|
|
+ projector_weights = filter_weights(projector_weights,
|
|
|
|
+ "multi_modal_projector")
|
|
|
|
+ projector_params_dict = dict(
|
|
|
|
+ self.multi_modal_projector.named_parameters())
|
|
|
|
+ for name, loaded_weight in projector_weights:
|
|
|
|
+ param = projector_params_dict[name]
|
|
|
|
+ weight_loader = getattr(param, "weight_loader",
|
|
|
|
+ default_weight_loader)
|
|
|
|
+ weight_loader(param, loaded_weight)
|
|
|
|
+
|
|
|
|
+ # load llm backbone
|
|
|
|
+ llm_weights = filter_weights(llm_weights, "language_model")
|
|
|
|
+ self.language_model.load_weights(llm_weights)
|