Roger Wang 2f0a0a17a4
[V1] Refactor model executable interface for multimodal models (#10570)
Signed-off-by: Roger Wang <ywang@roblox.com>
2024-11-26 20:46:11 +00:00

539 lines
21 KiB
Python

# Adapted from https://github.com/fixie-ai/ultravox/blob/ecd58c4041030bae2ad15aa6bcf04ab43199ea02/ultravox/model/ultravox_model.py
"""PyTorch Ultravox model."""
import math
from functools import cached_property, lru_cache
from typing import (Iterable, List, Literal, Mapping, Optional, Set, Tuple,
TypedDict, Union, cast)
import numpy as np
import torch
import torch.utils.checkpoint
from torch import nn
from torch.nn import functional as F
from transformers.models.whisper import WhisperFeatureExtractor
from transformers.models.whisper.modeling_whisper import WhisperEncoder
from vllm.attention import AttentionMetadata
from vllm.config import VllmConfig
from vllm.inputs import (INPUT_REGISTRY, DecoderOnlyInputs, DummyData,
InputContext, token_inputs)
from vllm.model_executor.layers.activation import SiluAndMul, get_act_fn
from vllm.model_executor.layers.layernorm import RMSNorm
from vllm.model_executor.layers.sampler import SamplerOutput, get_sampler
from vllm.model_executor.model_loader.loader import DefaultModelLoader
from vllm.model_executor.sampling_metadata import SamplingMetadata
from vllm.multimodal import (MULTIMODAL_REGISTRY, MultiModalKwargs,
NestedTensors)
from vllm.multimodal.utils import (cached_get_tokenizer,
consecutive_placeholder_ranges,
repeat_and_pad_placeholder_tokens)
from vllm.sequence import IntermediateTensors, SequenceData
from vllm.transformers_utils.configs.ultravox import UltravoxConfig
from vllm.utils import is_list_of
from .interfaces import SupportsMultiModal, SupportsPP
from .utils import (AutoWeightsLoader, WeightsMapper, flatten_bn,
init_vllm_registered_model, maybe_prefix,
merge_multimodal_embeddings_from_map)
_AUDIO_PLACEHOLDER_TOKEN = 128002
_AUDIO_TOKENS_PER_SECOND = 6.25
class UltravoxAudioFeatureInputs(TypedDict):
type: Literal["audio_features"]
data: NestedTensors
"""Shape: `(batch_size, num_audios, 80, M)`"""
class UltravoxAudioEmbeddingInputs(TypedDict):
type: Literal["audio_embeds"]
data: NestedTensors
"""Shape: `(batch_size, num_audios, audio_feature_size, hidden_size)`"""
UltravoxAudioInputs = Union[UltravoxAudioFeatureInputs,
UltravoxAudioEmbeddingInputs]
@lru_cache
def cached_feature_extractor(model_id: str) -> WhisperFeatureExtractor:
return WhisperFeatureExtractor.from_pretrained(model_id)
def whisper_feature_extractor(ctx: InputContext) -> WhisperFeatureExtractor:
return cached_feature_extractor(
ctx.get_hf_config(UltravoxConfig).audio_model_id)
def get_ultravox_max_audio_tokens(ctx: InputContext):
feature_extractor = whisper_feature_extractor(ctx)
return math.ceil(feature_extractor.chunk_length * _AUDIO_TOKENS_PER_SECOND)
def dummy_seq_data_for_ultravox(
ctx: InputContext,
seq_len: int,
audio_count: int,
):
audio_length = min(get_ultravox_max_audio_tokens(ctx),
seq_len // audio_count)
return SequenceData.from_prompt_token_counts(
(_AUDIO_PLACEHOLDER_TOKEN, audio_length * audio_count),
(0, seq_len - audio_length * audio_count)), {
"audio":
consecutive_placeholder_ranges(num_items=audio_count,
item_size=audio_length)
}
def dummy_audio_for_ultravox(
ctx: InputContext,
audio_count: int,
):
feature_extractor = whisper_feature_extractor(ctx)
audio_and_sr = (np.array([0.0] * feature_extractor.chunk_length), 1)
return {"audio": [audio_and_sr] * audio_count}
def dummy_data_for_ultravox(
ctx: InputContext,
seq_len: int,
mm_counts: Mapping[str, int],
):
audio_count = mm_counts["audio"]
seq_data, ranges = dummy_seq_data_for_ultravox(ctx, seq_len, audio_count)
mm_dict = dummy_audio_for_ultravox(ctx, audio_count)
return DummyData(seq_data, mm_dict, ranges)
def input_mapper_for_ultravox(ctx: InputContext, data: object):
if not isinstance(data, list):
data = [data]
if len(data) == 0:
return MultiModalKwargs()
# If the audio inputs are embeddings, no need for preprocessing
if is_list_of(data, torch.Tensor, check="all"):
return MultiModalKwargs({"audio_embeds": data})
audio_features = []
for audio_input in data:
if not isinstance(audio_input, tuple):
raise NotImplementedError(
f"Unsupported data type: {type(audio_input)}")
(audio, sr) = cast(Tuple[np.ndarray, Union[float, int]], audio_input)
feature_extractor = whisper_feature_extractor(ctx)
if sr != feature_extractor.sampling_rate:
try:
import librosa
except ImportError as exc:
raise ImportError(
"Please install vllm[audio] for audio support.") from exc
audio = librosa.resample(audio,
orig_sr=sr,
target_sr=feature_extractor.sampling_rate)
sr = feature_extractor.sampling_rate
minimum_audio_length = feature_extractor.n_fft // 2 + 1
if len(audio) < minimum_audio_length:
# Not enough audio; pad it.
audio = np.pad(audio, (0, minimum_audio_length - len(audio)))
single_audio_features = feature_extractor(
audio, sampling_rate=sr, padding="longest",
return_tensors="pt")["input_features"]
# Remove the batch dimension because we're wrapping it in a list.
audio_features.append(single_audio_features.squeeze(0))
return MultiModalKwargs({"audio_features": audio_features})
def input_processor_for_ultravox(ctx: InputContext, inputs: DecoderOnlyInputs):
multi_modal_data = inputs.get("multi_modal_data")
if multi_modal_data is None or "audio" not in multi_modal_data:
return inputs
if "multi_modal_placeholders" in inputs and "audio" in inputs[
"multi_modal_placeholders"]:
# The inputs already have placeholders.
return inputs
feature_extractor = whisper_feature_extractor(ctx)
audios = multi_modal_data["audio"]
if not isinstance(audios, list):
audios = [audios]
audio_token_counts = []
for audio in audios:
if isinstance(audio, torch.Tensor):
audio_num_tokens = audio.shape[1]
audio_token_counts.append(audio_num_tokens)
else:
audio_data, sample_rate = audio
audio_length = audio_data.shape[0]
if sample_rate != feature_extractor.sampling_rate:
# Account for resampling.
adjustment = feature_extractor.sampling_rate / sample_rate
audio_length = math.ceil(adjustment * audio_length)
feature_extractor_output_length = math.ceil(
(audio_length - (feature_extractor.hop_length - 1)) /
feature_extractor.hop_length)
uv_config = ctx.get_hf_config(UltravoxConfig)
audio_num_tokens = min(
max(
1,
math.ceil(feature_extractor_output_length /
(uv_config.stack_factor * 2))),
get_ultravox_max_audio_tokens(ctx))
audio_token_counts.append(audio_num_tokens)
tokenizer = cached_get_tokenizer(ctx.model_config.tokenizer)
new_prompt, new_token_ids, ranges = repeat_and_pad_placeholder_tokens(
tokenizer,
inputs.get("prompt"),
inputs["prompt_token_ids"],
placeholder_token_id=_AUDIO_PLACEHOLDER_TOKEN,
repeat_count=audio_token_counts,
)
# NOTE: Create a defensive copy of the original inputs
return token_inputs(prompt_token_ids=new_token_ids,
prompt=new_prompt,
multi_modal_data=multi_modal_data,
multi_modal_placeholders={"audio": ranges})
class StackAudioFrames(nn.Module):
"""
Stack the audio embedding frames to reduce the sequence length by a factor
of `stack_factor`.
"""
def __init__(self, stack_factor: int = 8):
super().__init__()
self.stack_factor = stack_factor
def forward(self, audio_embeds: torch.Tensor) -> torch.Tensor:
B, T, C = audio_embeds.shape
T_pad = (T + self.stack_factor -
1) // self.stack_factor * self.stack_factor
audio_embeds = F.pad(audio_embeds, (0, 0, 0, T_pad - T))
B, T, C = audio_embeds.shape
audio_embeds = audio_embeds.view(B, T // self.stack_factor,
C * self.stack_factor)
return audio_embeds
class FlippedSiluAndMul(SiluAndMul):
"""Ultravox is trained with SwiGLU with flipped halves."""
def forward(self, x: torch.Tensor):
a, b = x.chunk(2, dim=-1)
flipped = torch.cat((b, a), dim=-1)
return super().forward(flipped)
class UltravoxProjector(nn.Module):
def __init__(self, config: UltravoxConfig):
super().__init__()
self.hidden_dim = config.hidden_size
self._pad_and_stack = StackAudioFrames(config.stack_factor)
dim = config.audio_config.hidden_size * config.stack_factor
self.ln_pre = RMSNorm(dim)
self.linear_1 = nn.Linear(dim, self.hidden_dim, bias=False)
dim = self.hidden_dim
if config.projector_act == "swiglu":
self.act = FlippedSiluAndMul()
dim = dim // 2
else:
self.act = get_act_fn(config.projector_act)
self.linear_2 = nn.Linear(dim,
config.text_config.hidden_size,
bias=False)
self.ln_post = RMSNorm(config.text_config.hidden_size)
def forward(self, audio_features: torch.Tensor) -> torch.Tensor:
audio_features = self._pad_and_stack(audio_features)
audio_features = self.ln_pre(audio_features)
hidden_states = self.linear_1(audio_features)
hidden_states = self.act(hidden_states)
hidden_states = self.linear_2(hidden_states)
hidden_states = self.ln_post(hidden_states)
return hidden_states
class ModifiedWhisperEncoder(WhisperEncoder):
"""
Encoder portion of OpenAI's Whisper model.
This implementation is a slightly modified version of HF Transformers'
Whisper Encoder, with only a few fixes:
1. base_model_prefix updated to allow for doing `.from_pretrained`
directly on the encoder
2. allow less than 30 second of audio padding to be passed in:
- relaxed ValueError check for `input_features` length to be less
than or equal to `expected_seq_length` instead of strictly equal
- embed_pos is now sliced to match the length of `inputs_embeds`
Original: https://github.com/huggingface/transformers/blob/main/src/transformers/models/whisper/modeling_whisper.py
See commentary: https://github.com/huggingface/transformers/issues/25744
"""
base_model_prefix = "model.encoder"
def forward(
self,
input_features,
):
expected_seq_length = (self.config.max_source_positions *
self.conv1.stride[0] * self.conv2.stride[0])
if input_features.shape[-1] > expected_seq_length:
raise ValueError(
f"Whisper expects the mel input features to be of length "
f"{expected_seq_length} or less, but found "
f"{input_features.shape[-1]}. Make sure to pad the input mel "
f"features to {expected_seq_length}.")
inputs_embeds = nn.functional.gelu(self.conv1(input_features))
inputs_embeds = nn.functional.gelu(self.conv2(inputs_embeds))
inputs_embeds = inputs_embeds.permute(0, 2, 1)
embed_pos = self.embed_positions.weight[:inputs_embeds.size(-2)]
hidden_states = inputs_embeds + embed_pos
hidden_states = nn.functional.dropout(hidden_states,
p=self.dropout,
training=self.training)
for encoder_layer in self.layers:
layer_outputs = encoder_layer(
hidden_states,
None,
layer_head_mask=None,
)
hidden_states = layer_outputs[0]
hidden_states = self.layer_norm(hidden_states)
return hidden_states
@MULTIMODAL_REGISTRY.register_input_mapper("audio", input_mapper_for_ultravox)
@MULTIMODAL_REGISTRY.register_max_multimodal_tokens(
"audio", get_ultravox_max_audio_tokens)
@INPUT_REGISTRY.register_dummy_data(dummy_data_for_ultravox)
@INPUT_REGISTRY.register_input_processor(input_processor_for_ultravox)
class UltravoxModel(nn.Module, SupportsMultiModal, SupportsPP):
def __init__(self, *, vllm_config: VllmConfig, prefix: str = ""):
super().__init__()
config = vllm_config.model_config.hf_config
multimodal_config = vllm_config.model_config.multimodal_config
self.config = config
self.multi_modal_config = multimodal_config
assert self.multi_modal_config
self.secondary_weights = []
self.audio_tower = ModifiedWhisperEncoder(config.audio_config)
if config.audio_model_id is not None:
# this prefix is not for initialization, but for loading weights
# note the trailing dot
self.secondary_weights.append(
DefaultModelLoader.Source(
model_or_path=config.audio_model_id,
revision=None,
prefix="audio_tower.",
))
self.multi_modal_projector = UltravoxProjector(config)
self.language_model = init_vllm_registered_model(
config.text_config,
vllm_config=vllm_config,
prefix=maybe_prefix(prefix, "language_model"))
if config.text_model_id is not None:
# this prefix is not for initialization, but for loading weights
# note the trailing dot
self.secondary_weights.append(
DefaultModelLoader.Source(model_or_path=config.text_model_id,
revision=None,
prefix="language_model."))
self.make_empty_intermediate_tensors = (
self.language_model.make_empty_intermediate_tensors)
@cached_property
def sampler(self):
if hasattr(self.language_model, "sampler"):
return self.language_model.sampler
return get_sampler()
def _audio_features_to_embeddings(
self, input_features: torch.Tensor) -> torch.Tensor:
audio_input = input_features.to(self.audio_tower.dtype)
audio_features = self.audio_tower(audio_input)
audio_features = audio_features.to(self.audio_tower.dtype)
audio_embeddings = self.multi_modal_projector(audio_features)
return audio_embeddings
def _parse_and_validate_audio_input(
self, **kwargs: object) -> Optional[UltravoxAudioInputs]:
audio_features = kwargs.pop("audio_features", None)
audio_embeds = kwargs.pop("audio_embeds", None)
if audio_features is None and audio_embeds is None:
return None
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, list)):
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) -> NestedTensors:
if audio_input["type"] == "audio_embeds":
return audio_input["data"]
audio_features = audio_input["data"]
if isinstance(audio_features, torch.Tensor):
# Combine the B and N dimensions for the encoder/projector
flattened = flatten_bn(audio_features)
flattened_embeddings = self._audio_features_to_embeddings(
flattened)
# Restore the original dimensions
embeddings = flattened_embeddings.unflatten(
0, audio_features.shape[:2])
return embeddings
result = []
# TODO: Batch heterogeneous tensors through the encoder/projector
for audio_features_item in audio_features:
if isinstance(audio_features_item, torch.Tensor):
result.append(
self._audio_features_to_embeddings(audio_features_item))
else:
embeddings = [
# Add a batch dimension to embed it, then remove it.
self._audio_features_to_embeddings(tensor.unsqueeze(0)
).squeeze(0)
for tensor in audio_features_item
]
result.append(embeddings)
return result
def get_multimodal_embeddings(self, **kwargs) -> Optional[NestedTensors]:
audio_input = self._parse_and_validate_audio_input(**kwargs)
if audio_input is None:
return None
audio_embeddings = self._process_audio_input(audio_input)
return audio_embeddings
def get_input_embeddings(
self,
input_ids: torch.Tensor,
multimodal_embeddings: Optional[NestedTensors] = None,
attn_metadata: Optional[AttentionMetadata] = None,
) -> torch.Tensor:
inputs_embeds = self.language_model.get_input_embeddings(input_ids)
if multimodal_embeddings is not None:
# TODO(ywang96): use merge_multimodal_embeddings after
# v0 is deprecated
merge_multimodal_embeddings_from_map(
inputs_embeds, multimodal_embeddings,
attn_metadata.multi_modal_placeholder_index_maps["audio"])
return inputs_embeds
def forward(self,
input_ids: torch.Tensor,
positions: torch.Tensor,
kv_caches: List[torch.Tensor],
attn_metadata: AttentionMetadata,
intermediate_tensors: Optional[torch.Tensor] = None,
inputs_embeds: Optional[torch.Tensor] = None,
**kwargs) -> Union[torch.Tensor, IntermediateTensors]:
"""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:
audio_features: A batch of audio inputs [B, N, 80, M].
"""
if intermediate_tensors is not None:
inputs_embeds = None
# NOTE: In v1, inputs_embeds is always generated at model runner, this
# condition is for v0 compatibility.
elif inputs_embeds is None:
multimodal_embeddings = self.get_multimodal_embeddings(**kwargs)
# TODO(ywang96): remove attn_metadata from get_input_embeddings
# after v0 is deprecated
inputs_embeds = self.get_input_embeddings(input_ids,
multimodal_embeddings,
attn_metadata)
input_ids = None
hidden_states = self.language_model.model(input_ids,
positions,
kv_caches,
attn_metadata,
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]]) -> Set[str]:
hf_to_vllm_mapper = WeightsMapper(
orig_to_new_prefix={"audio_tower.model.encoder.": "audio_tower."})
loader = AutoWeightsLoader(self,
ignore_unexpected_prefixes=["audio_tower."])
return loader.load_weights(weights, mapper=hf_to_vllm_mapper)