refactor funasr model. (#36108)

Signed-off-by: zixiao <shunli.dsl@alibaba-inc.com>
Co-authored-by: zixiao <shunli.dsl@alibaba-inc.com>
Co-authored-by: Isotr0py <mozf@mail2.sysu.edu.cn>

vllm/model_executor/models/funasr.py

@@ -51,7 +51,6 @@ from vllm.multimodal.processing import (
 )
 from vllm.transformers_utils.processor import cached_processor_from_config
 from vllm.transformers_utils.processors.funasr_processor import FunASRFeatureExtractor
-from vllm.utils.jsontree import json_map_leaves
 from vllm.utils.tensor_schema import TensorSchema, TensorShape
 
 from .interfaces import (
@@ -611,6 +610,10 @@ class FunASRAudioInputs(TensorSchema):
         list[torch.Tensor] | None,
         TensorShape("b"),
     ]
+    fake_token_lengths: Annotated[
+        list[torch.Tensor] | None,
+        TensorShape("b"),
+    ]
 
 
 class FunASREncoder(nn.Module):
@@ -732,9 +735,6 @@ class FunASRProcessingInfo(BaseProcessingInfo):
     def get_target_channels(self) -> int:
         return 1
 
-    def get_num_audio_tokens(self) -> int:
-        return self.get_hf_config().max_source_positions
-
 
 class FunASRDummyInputsBuilder(BaseDummyInputsBuilder[FunASRProcessingInfo]):
     def get_dummy_text(self, mm_counts: Mapping[str, int]) -> str:
@@ -798,7 +798,7 @@ class FunASRMultiModalProcessor(BaseMultiModalProcessor[FunASRProcessingInfo]):
         return dict(
             input_features=MultiModalFieldConfig.batched("audio"),
             speech_lengths=MultiModalFieldConfig.batched("audio"),
-            fake_token_len=MultiModalFieldConfig.batched("audio"),
+            fake_token_lengths=MultiModalFieldConfig.batched("audio"),
         )
 
     def _get_prompt_updates(
@@ -812,22 +812,16 @@ class FunASRMultiModalProcessor(BaseMultiModalProcessor[FunASRProcessingInfo]):
 
         out_mm_data = out_mm_kwargs.get_data()
 
-        fake_token_len = out_mm_data.get("fake_token_len")
-        if fake_token_len is None:
+        fake_token_lengths = out_mm_data.get("fake_token_lengths")
+        if fake_token_lengths is None:
             audio_output_lengths = []
         else:
-            assert isinstance(fake_token_len, torch.Tensor)
+            assert isinstance(fake_token_lengths, torch.Tensor)
 
-            audio_output_lengths = fake_token_len.tolist()
+            audio_output_lengths = fake_token_lengths.tolist()
 
         def get_replacement_qwen2_audio(item_idx: int):
-            if audio_output_lengths:
-                num_features = audio_output_lengths[item_idx]
-            else:
-                audio_embeds = out_mm_data["audio_embeds"][item_idx]
-                assert len(audio_embeds.shape) == 2, "audio_embeds must be a 2D tensor"
-                num_features = audio_embeds.shape[0]
-
+            num_features = audio_output_lengths[item_idx]
             return [audio_token_id] * num_features
 
         return [
@@ -847,21 +841,16 @@ class FunASRMultiModalProcessor(BaseMultiModalProcessor[FunASRProcessingInfo]):
 class FunASRForConditionalGeneration(
     nn.Module, SupportsTranscription, SupportsMultiModal
 ):
-    packed_modules_mapping = {
-        "self_attn.qkv_proj": [
-            "self_attn.q_proj",
-            "self_attn.k_proj",
-            "self_attn.v_proj",
-        ],
-        "encoder_attn.kv_proj": ["encoder_attn.k_proj", "encoder_attn.v_proj"],
-    }
-
     hf_to_vllm_mapper = WeightsMapper(
         orig_to_new_substr={
             "linear_q.": "q_proj.",
             "linear_k.": "k_proj.",
             "linear_v.": "v_proj.",
             "linear_out.": "out_proj.",
+            "audio_adaptor.": "model.encoder.audio_adaptor.",
+            "audio_encoder.": "model.encoder.audio_encoder.",
+            "llm.model.": "model.decoder.",
+            "llm.lm_head": "lm_head",
         }
     )
 
@@ -969,9 +958,6 @@ class FunASRForConditionalGeneration(
         )
         return decoder_outputs
 
-    def get_language_model(self) -> torch.nn.Module:
-        return self.model.decoder
-
     def embed_multimodal(self, **kwargs: object) -> MultiModalEmbeddings:
         audio_input = self._parse_and_validate_audio_input(**kwargs)
 
@@ -1002,15 +988,12 @@ class FunASRForConditionalGeneration(
     def _parse_and_validate_audio_input(self, **kwargs: object) -> FunASRAudioInputs:
         input_features = kwargs.pop("input_features", None)
         speech_lengths = kwargs.pop("speech_lengths", None)
-
-        if input_features is not None:
-            input_features = json_map_leaves(lambda x: x.to(self.dtype), input_features)
-
-        if speech_lengths is not None:
-            speech_lengths = json_map_leaves(lambda x: x.to(self.dtype), speech_lengths)
+        fake_token_lengths = kwargs.pop("fake_token_lengths", None)
 
         return FunASRAudioInputs(
-            input_features=input_features, speech_lengths=speech_lengths
+            input_features=input_features,
+            speech_lengths=speech_lengths,
+            fake_token_lengths=fake_token_lengths,
         )
 
     def compute_logits(self, hidden_states: torch.Tensor) -> torch.Tensor:
@@ -1022,22 +1005,4 @@ class FunASRForConditionalGeneration(
             self,
         )
 
-        # add fake zeros bias for k_proj to state_dict
-        weights = _create_fake_bias_for_k_proj(weights)
         return loader.load_weights(weights, mapper=self.hf_to_vllm_mapper)
-
-
-def _create_fake_bias_for_k_proj(
-    weights: Iterable[tuple[str, torch.Tensor]],
-) -> Iterable[tuple[str, torch.Tensor]]:
-    """
-    Create full zeros bias for k_proj weight in self-attn and x-attn layers.
-    So that the bias for k_proj in qkv_proj can be initialized with zeros.
-    """
-    for name, weight in weights:
-        if name.endswith(".k_proj.weight"):
-            bias = torch.zeros(weight.size(0))
-            bias_name = name.replace("weight", "bias")
-            yield from [(name, weight), (bias_name, bias)]
-        else:
-            yield name, weight

vllm/model_executor/models/qwen3_omni_moe_thinker.py

@@ -1794,7 +1794,7 @@ class Qwen3OmniMoeThinkerForConditionalGeneration(
             return []
 
         # The result multimodal_embeddings is tuple of tensors, with each
-        # tensor correspoending to a multimodal data item (image or video).
+        # tensor corresponding to a multimodal data item (image or video).
         multimodal_embeddings: tuple[torch.Tensor, ...] = ()
 
         # NOTE: It is important to iterate over the keys in this dictionary

vllm/transformers_utils/processors/funasr_processor.py

@@ -370,7 +370,7 @@ class FunASRFeatureExtractor(SequenceFeatureExtractor):
         )
         olens = 1 + (speech_lengths - 3 + 2 * 1) // 2
         olens = 1 + (olens - 3 + 2 * 1) // 2
-        fake_token_len = (olens - 1) // 2 + 1
+        fake_token_lengths = (olens - 1) // 2 + 1
         if isinstance(input_features[0], list):
             padded_inputs["input_features"] = [
                 np.asarray(feature, dtype=np.float32) for feature in input_features
@@ -382,8 +382,10 @@ class FunASRFeatureExtractor(SequenceFeatureExtractor):
         if return_tensors is not None:
             padded_inputs = padded_inputs.convert_to_tensors(return_tensors)
 
+        fake_token_lengths = torch.clamp(fake_token_lengths, min=1)
+
         padded_inputs["speech_lengths"] = speech_lengths
-        padded_inputs["fake_token_len"] = fake_token_len
+        padded_inputs["fake_token_lengths"] = fake_token_lengths
 
         return padded_inputs
 
@@ -471,7 +473,7 @@ class FunASRProcessor(ProcessorMixin):
             for sample in text:
                 replace_str = []
                 while self.audio_token in sample:
-                    num_audio_tokens = inputs["fake_token_len"].item()
+                    num_audio_tokens = inputs["fake_token_lengths"].item()
 
                     expanded_audio_token = self.audio_token * num_audio_tokens