qwen2_audio.py 15.2 KB
Newer Older
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
# Copyright 2024 The Qwen team.
# Copyright 2023 The vLLM team.
# Copyright 2022 EleutherAI and the HuggingFace Inc. team. All rights reserved.
#
# This code is based on EleutherAI's GPT-NeoX library and the GPT-NeoX
# and OPT implementations in this library. It has been modified from its
# original forms to accommodate minor architectural differences compared
# to GPT-NeoX and OPT used by the Meta AI team that trained the model.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
#     http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
"""Inference-only Qwen2-Audio model compatible with HuggingFace weights."""
22
from functools import cached_property
23
24
from typing import (Iterable, List, Mapping, Optional, Set, Tuple, TypedDict,
                    Union)
25
26
27

import torch
import torch.nn as nn
28
from transformers import BatchFeature
29
30
31
32
from transformers.models.qwen2_audio import (Qwen2AudioConfig,
                                             Qwen2AudioEncoder,
                                             Qwen2AudioProcessor)
from transformers.models.whisper import WhisperFeatureExtractor
33
34

from vllm.attention import AttentionMetadata
35
from vllm.config import VllmConfig
Joe Runde's avatar
Joe Runde committed
36
from vllm.model_executor.layers.sampler import SamplerOutput, get_sampler
37
from vllm.model_executor.sampling_metadata import SamplingMetadata
38
from vllm.multimodal import MULTIMODAL_REGISTRY
39
40
from vllm.multimodal.inputs import (MultiModalFieldConfig, MultiModalKwargs,
                                    NestedTensors)
41
from vllm.multimodal.parse import AudioProcessorItems, MultiModalDataParser
42
from vllm.multimodal.processing import (BaseMultiModalProcessor,
43
44
                                        MultiModalDataItems, ProcessorInputs,
                                        PromptReplacement)
45
from vllm.sequence import IntermediateTensors
46
47

from .interfaces import SupportsMultiModal, SupportsPP
48
49
from .utils import (AutoWeightsLoader, init_vllm_registered_model,
                    maybe_prefix, merge_multimodal_embeddings)
50
51
52
53
54


# # === Audio Inputs === #
class Qwen2AudioInputs(TypedDict):
    input_features: torch.Tensor
55
    """Shape: `(num_audios, num_mel_bins, 3000)`"""
56
57

    feature_attention_mask: torch.Tensor
58
    """Shape: `(num_audios, 3000)`"""
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74


# === Audio Encoder === #


class Qwen2AudioMultiModalProjector(nn.Module):

    def __init__(self, audio_hidden_size: int, text_hidden_size: int):
        super().__init__()
        self.linear = nn.Linear(audio_hidden_size, text_hidden_size, bias=True)

    def forward(self, audio_features):
        hidden_states = self.linear(audio_features)
        return hidden_states


75
# From Qwen2AudioEncoder._get_feat_extract_output_lengths
76
def _get_feat_extract_output_lengths(input_lengths: torch.Tensor):
77
78
79
    feat_lengths = (input_lengths - 1) // 2 + 1
    output_lengths = (feat_lengths - 2) // 2 + 1
    return feat_lengths, output_lengths
80
81


82
class Qwen2AudioMultiModalProcessor(BaseMultiModalProcessor):
83

84
85
    def get_supported_mm_limits(self) -> Mapping[str, Optional[int]]:
        return {"audio": None}
86

87
88
89
90
91
92
    def get_mm_max_tokens_per_item(self) -> Mapping[str, int]:
        hf_config = self.ctx.get_hf_config(Qwen2AudioConfig)
        max_source_positions = hf_config.audio_config.max_source_positions
        max_output_lengths = (max_source_positions - 2) // 2 + 1

        return {"audio": max_output_lengths}
93

94
95
96
97
98
99
    def _get_hf_processor(
        self,
        *,
        # Ignored in initialization
        sampling_rate: Optional[int] = None,
    ) -> Qwen2AudioProcessor:
100
        return self.ctx.get_hf_processor(Qwen2AudioProcessor)
101

102
103
    def _get_feature_extractor(self) -> WhisperFeatureExtractor:
        return self._get_hf_processor().feature_extractor  # type: ignore
104

105
    def _get_data_parser(self) -> MultiModalDataParser:
106
        feature_extractor = self._get_feature_extractor()
107
        return MultiModalDataParser(target_sr=feature_extractor.sampling_rate)
108

109
110
111
    def _call_hf_processor(
        self,
        prompt: str,
112
113
        mm_data: Mapping[str, object],
        mm_kwargs: Mapping[str, object],
114
    ) -> BatchFeature:
115
116
        mm_data = dict(mm_data)
        audios = mm_data.pop("audios", [])
117

118
        if audios:
119
            mm_data["audios"] = audios
120
121

            feature_extractor = self._get_feature_extractor()
122
123
            mm_kwargs = dict(
                **mm_kwargs,
124
125
126
127
128
129
                sampling_rate=feature_extractor.sampling_rate,
            )
        else:
            # NOTE: WhisperFeatureExtractor cannot handle empty list of audios
            pass

130
        processed_outputs = super()._call_hf_processor(
131
            prompt=prompt,
132
133
134
135
136
137
138
139
140
141
142
143
144
145
            mm_data=mm_data,
            mm_kwargs=mm_kwargs,
        )

        return processed_outputs

    def _get_mm_fields_config(
        self,
        hf_inputs: BatchFeature,
        hf_processor_mm_kwargs: Mapping[str, object],
    ) -> Mapping[str, MultiModalFieldConfig]:
        return dict(
            input_features=MultiModalFieldConfig.batched("audio"),
            feature_attention_mask=MultiModalFieldConfig.batched("audio"),
146
147
148
149
150
        )

    def _get_prompt_replacements(
        self,
        mm_items: MultiModalDataItems,
151
152
        hf_processor_mm_kwargs: Mapping[str, object],
        out_mm_kwargs: MultiModalKwargs,
153
154
155
156
    ) -> list[PromptReplacement]:
        hf_config = self.ctx.get_hf_config(Qwen2AudioConfig)
        placeholder = hf_config.audio_token_index

157
        feature_attention_mask = out_mm_kwargs.get("feature_attention_mask")
158
159
160
        if feature_attention_mask is None:
            audio_output_lengths = []
        else:
161
            assert isinstance(feature_attention_mask, torch.Tensor)
162
            _, audio_output_lens = _get_feat_extract_output_lengths(
163
164
                feature_attention_mask.sum(-1))

165
166
            audio_output_lengths = audio_output_lens.tolist()

167
        def get_replacement_qwen2_audio(item_idx: int):
168
169
170
171
172
173
174
175
176
            num_placeholders = audio_output_lengths[item_idx]
            if num_placeholders == 0:
                audios = mm_items.get_items("audio", AudioProcessorItems)
                audio = audios.get(item_idx)
                raise ValueError(
                    f"The audio {audio} (len={len(audio)}) is too short "
                    "to be represented inside the model")

            return [placeholder] * num_placeholders
177
178
179
180
181
182
183

        return [
            PromptReplacement(
                modality="audio",
                target=[placeholder],
                replacement=get_replacement_qwen2_audio,
            )
184
        ]
185

186
187
188
189
190
191
192
193
    def _always_apply_prompt_replacements(self) -> bool:
        # HF never applies prompt replacements, so we have to do it ourselves
        # _find_placeholders may incorrectly think that HF has already performed
        # processing for multi-audio input when the input audios are short
        # (the corresponding placeholders may take up fewer tokens than
        # the number of audio items)
        return True

194
195
196
197
    def _get_dummy_mm_inputs(
        self,
        mm_counts: Mapping[str, int],
    ) -> ProcessorInputs:
198
        feature_extractor = self._get_feature_extractor()
199

200
201
        sampling_rate = feature_extractor.sampling_rate
        audio_len = feature_extractor.chunk_length * sampling_rate
202
        num_audios = mm_counts.get("audio", 0)
203

204
205
206
207
        mm_data = {
            "audio":
            self._get_dummy_audios(length=audio_len, num_audios=num_audios)
        }
208
209

        return ProcessorInputs(
210
211
            prompt_text="<|AUDIO|>" * num_audios,
            mm_data=mm_data,
212
213
214
215
        )


@MULTIMODAL_REGISTRY.register_processor(Qwen2AudioMultiModalProcessor)
216
217
218
class Qwen2AudioForConditionalGeneration(nn.Module, SupportsMultiModal,
                                         SupportsPP):

219
    def __init__(self, *, vllm_config: VllmConfig, prefix: str = ""):
220
        super().__init__()
221
222
223
        config = vllm_config.model_config.hf_config
        quant_config = vllm_config.quant_config
        multimodal_config = vllm_config.model_config.multimodal_config
224
225
226
227
228
229
230
231
232
        self.config = config
        self.multimodal_config = multimodal_config

        self.audio_tower = Qwen2AudioEncoder(config.audio_config)
        self.multi_modal_projector = Qwen2AudioMultiModalProjector(
            config.audio_config.d_model, config.text_config.hidden_size)

        self.quant_config = quant_config

233
234
235
236
237
238
        self.language_model = init_vllm_registered_model(
            vllm_config=vllm_config,
            hf_config=config.text_config,
            prefix=maybe_prefix(prefix, "language_model"),
            architectures=["Qwen2ForCausalLM"],
        )
239
240
241
242

        self.make_empty_intermediate_tensors = (
            self.language_model.make_empty_intermediate_tensors)

243
244
245
246
247
248
249
    @cached_property
    def sampler(self):
        if hasattr(self.language_model, "sampler"):
            return self.language_model.sampler

        return get_sampler()

250
    def _validate_and_reshape_mm_tensor(self, mm_input: object,
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
                                        name: str) -> torch.Tensor:
        if not isinstance(mm_input, (torch.Tensor, list)):
            raise ValueError(f"Incorrect type of {name}. "
                             f"Got type: {type(mm_input)}")
        if isinstance(mm_input, torch.Tensor):
            return torch.concat(list(mm_input))
        else:
            return torch.concat(mm_input)

    def _parse_and_validate_audio_input(
            self, **kwargs: object) -> Optional[Qwen2AudioInputs]:
        input_features = kwargs.pop('input_features', None)
        feature_attention_mask = kwargs.pop('feature_attention_mask', None)
        if input_features is None:
            return None
        input_features = self._validate_and_reshape_mm_tensor(
            input_features, 'input_features')
        feature_attention_mask = self._validate_and_reshape_mm_tensor(
            feature_attention_mask, 'feature_attention_mask')
        if not isinstance(input_features, (torch.Tensor, list)):
            raise ValueError("Incorrect type of audio input features. "
                             f"Got type: {type(input_features)}")
        return Qwen2AudioInputs(input_features=input_features,
                                feature_attention_mask=feature_attention_mask)

    def _process_audio_input(self,
                             audio_input: Qwen2AudioInputs) -> torch.Tensor:

        input_features = audio_input["input_features"]
        feature_attention_mask = audio_input["feature_attention_mask"]

        audio_feat_lengths, audio_output_lengths = (
            self.audio_tower._get_feat_extract_output_lengths(
                feature_attention_mask.sum(-1)))

        batch_size, _, max_mel_seq_len = input_features.shape
        max_seq_len = (max_mel_seq_len - 2) // 2 + 1
        # Create a sequence tensor of shape (batch_size, max_seq_len)
        seq_range = (torch.arange(
            0,
            max_seq_len,
            dtype=audio_feat_lengths.dtype,
            device=audio_feat_lengths.device).unsqueeze(0).expand(
                batch_size, max_seq_len))
        lengths_expand = audio_feat_lengths.unsqueeze(-1).expand(
            batch_size, max_seq_len)
        # Create mask
        padding_mask = seq_range >= lengths_expand

        audio_attention_mask_ = padding_mask.view(
            batch_size, 1, 1, max_seq_len).expand(batch_size, 1, max_seq_len,
                                                  max_seq_len)
        audio_attention_mask = audio_attention_mask_.to(
            dtype=self.audio_tower.conv1.weight.dtype,
            device=self.audio_tower.conv1.weight.device)
        audio_attention_mask[audio_attention_mask_] = float("-inf")

        audio_outputs = self.audio_tower(input_features,
                                         attention_mask=audio_attention_mask)
        selected_audio_feature = audio_outputs.last_hidden_state
        audio_features = self.multi_modal_projector(selected_audio_feature)
        num_audios, max_audio_tokens, embed_dim = audio_features.shape
        audio_features_mask = torch.arange(max_audio_tokens).expand(
            num_audios, max_audio_tokens
        ).to(audio_output_lengths.device) < audio_output_lengths.unsqueeze(1)
        masked_audio_features = audio_features[audio_features_mask].view(
            -1, embed_dim)

        return masked_audio_features

321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
    def get_multimodal_embeddings(self, **kwargs) -> Optional[NestedTensors]:
        audio_input = self._parse_and_validate_audio_input(**kwargs)
        if audio_input is None:
            return None
        masked_audio_features = self._process_audio_input(audio_input)
        return masked_audio_features

    def get_input_embeddings(
        self,
        input_ids: torch.Tensor,
        multimodal_embeddings: Optional[NestedTensors] = None,
    ) -> torch.Tensor:
        inputs_embeds = self.language_model.get_input_embeddings(input_ids)
        if multimodal_embeddings is not None:
            inputs_embeds = merge_multimodal_embeddings(
                input_ids, inputs_embeds, multimodal_embeddings,
                self.config.audio_token_index)
        return inputs_embeds

340
341
342
343
344
345
346
    def forward(
        self,
        input_ids: torch.Tensor,
        positions: torch.Tensor,
        kv_caches: List[torch.Tensor],
        attn_metadata: AttentionMetadata,
        intermediate_tensors: Optional[IntermediateTensors] = None,
347
        inputs_embeds: Optional[torch.Tensor] = None,
348
349
        **kwargs: object,
    ) -> Union[torch.Tensor, IntermediateTensors]:
350

351
352
        if intermediate_tensors is not None:
            inputs_embeds = None
353
354
355
356
357
358
359
360
361

        # 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)
            inputs_embeds = self.get_input_embeddings(input_ids,
                                                      multimodal_embeddings)
            input_ids = None

362
363
364
365
366
367
        hidden_states = self.language_model.model(input_ids,
                                                  positions,
                                                  kv_caches,
                                                  attn_metadata,
                                                  intermediate_tensors,
                                                  inputs_embeds=inputs_embeds)
368
369
        return hidden_states

370
371
372
373
374
375
376
    def compute_logits(
        self,
        hidden_states: torch.Tensor,
        sampling_metadata: SamplingMetadata,
    ) -> Optional[torch.Tensor]:
        return self.language_model.compute_logits(hidden_states,
                                                  sampling_metadata)
377
378
379
380
381
382

    def sample(
        self,
        logits: torch.Tensor,
        sampling_metadata: SamplingMetadata,
    ) -> Optional[SamplerOutput]:
383
        return self.language_model.sample(logits, sampling_metadata)
384

385
386
    def load_weights(self, weights: Iterable[Tuple[str,
                                                   torch.Tensor]]) -> Set[str]:
387
388
        loader = AutoWeightsLoader(self)
        return loader.load_weights(weights)