[Fix][Qwen3-TTS] Preserve ref_code decoder context for Base ICL

tests/model_executor/stage_input_processors/test_qwen3_tts_async_chunk.py

@@ -7,7 +7,10 @@
 import pytest
 import torch
 
-from vllm_omni.model_executor.stage_input_processors.qwen3_tts import talker2code2wav_async_chunk
+from vllm_omni.model_executor.stage_input_processors.qwen3_tts import (
+    talker2code2wav,
+    talker2code2wav_async_chunk,
+)
 
 _FRAME = [1, 2, 3, 4]  # 4-codebook frame
 _Q = len(_FRAME)  # num quantizers
@@ -29,6 +32,7 @@ def _tm(*, chunk_frames=25, left_context=25, initial_chunk=0):
     return SimpleNamespace(
         code_prompt_token_ids=defaultdict(list),
         put_req_chunk=defaultdict(int),
+        request_payload={},
         connector=SimpleNamespace(
             config={
                 "extra": {
@@ -152,3 +156,113 @@ def test_per_request_override_wins_over_stage_config():
     tm = _tm(initial_chunk=5)
     payload = _call(tm, "r-override2", n_frames=10, put_req=0, req_ic=15)
     assert payload is None
+
+
+def test_first_streaming_chunk_prepends_ref_code_context():
+    tm = _tm(initial_chunk=10)
+    rid = "r-ref"
+    tm.code_prompt_token_ids[rid] = [_FRAME[:] for _ in range(10)]
+    ref_code = torch.tensor([[9, 9, 9, 9], [8, 8, 8, 8]], dtype=torch.long)
+
+    payload = talker2code2wav_async_chunk(
+        transfer_manager=tm,
+        pooling_output={"audio_codes": torch.zeros((0,)), "ref_code": ref_code},
+        request=_req(rid, finished=False, initial_codec_chunk_frames=10),
+        is_finished=False,
+    )
+
+    assert payload is not None
+    assert payload["left_context_size"] == 2
+    assert len(payload["code_predictor_codes"]) == _Q * 12
+
+
+def test_ref_code_context_only_applies_to_first_streaming_chunk():
+    tm = _tm(initial_chunk=10)
+    rid = "r-ref2"
+    tm.code_prompt_token_ids[rid] = [_FRAME[:] for _ in range(20)]
+    tm.put_req_chunk[rid] = 1
+    ref_code = torch.tensor([[9, 9, 9, 9], [8, 8, 8, 8]], dtype=torch.long)
+
+    payload = talker2code2wav_async_chunk(
+        transfer_manager=tm,
+        pooling_output={"audio_codes": torch.zeros((0,)), "ref_code": ref_code},
+        request=_req(rid, finished=False, initial_codec_chunk_frames=10),
+        is_finished=False,
+    )
+
+    assert payload is not None
+    assert payload["left_context_size"] == 10
+    assert len(payload["code_predictor_codes"]) == _Q * 20
+
+
+def test_ref_code_context_can_be_buffered_before_first_emit():
+    tm = _tm(initial_chunk=10)
+    rid = "r-ref-buffered"
+    ref_code = torch.tensor([[9, 9, 9, 9], [8, 8, 8, 8]], dtype=torch.long)
+
+    first_payload = talker2code2wav_async_chunk(
+        transfer_manager=tm,
+        pooling_output={"audio_codes": torch.tensor([[1, 2, 3, 4]]), "ref_code": ref_code},
+        request=_req(rid, finished=False, initial_codec_chunk_frames=10),
+        is_finished=False,
+    )
+    assert first_payload is None
+    assert rid in tm.request_payload
+
+    for _ in range(8):
+        talker2code2wav_async_chunk(
+            transfer_manager=tm,
+            pooling_output={"audio_codes": torch.tensor([[1, 2, 3, 4]])},
+            request=_req(rid, finished=False, initial_codec_chunk_frames=10),
+            is_finished=False,
+        )
+
+    payload = talker2code2wav_async_chunk(
+        transfer_manager=tm,
+        pooling_output={"audio_codes": torch.tensor([[1, 2, 3, 4]])},
+        request=_req(rid, finished=False, initial_codec_chunk_frames=10),
+        is_finished=False,
+    )
+
+    assert payload is not None
+    assert payload["left_context_size"] == 2
+    assert len(payload["code_predictor_codes"]) == _Q * 12
+    assert rid not in tm.request_payload
+
+
+def test_non_async_processor_prepends_ref_code_and_sets_trim_context():
+    ref_code = torch.tensor([[9, 9, 9, 9], [8, 8, 8, 8]], dtype=torch.long)
+    audio_codes = torch.tensor(
+        [
+            [0, 0, 0, 0],
+            [1, 2, 3, 4],
+            [5, 6, 7, 8],
+        ],
+        dtype=torch.long,
+    )
+    output = SimpleNamespace(multimodal_output={"audio_codes": audio_codes, "ref_code": ref_code})
+    stage = SimpleNamespace(engine_outputs=[SimpleNamespace(outputs=[output])])
+
+    prompts = talker2code2wav(stage_list=[stage], engine_input_source=[0])
+
+    assert len(prompts) == 1
+    prompt = prompts[0]
+    assert prompt["additional_information"] == {"left_context_size": [2]}
+    assert prompt["prompt_token_ids"] == [
+        9,
+        8,
+        1,
+        5,
+        9,
+        8,
+        2,
+        6,
+        9,
+        8,
+        3,
+        7,
+        9,
+        8,
+        4,
+        8,
+    ]

vllm_omni/entrypoints/openai/serving_speech.py

@@ -11,6 +11,7 @@
 import numpy as np
 from fastapi import Request, UploadFile
 from fastapi.responses import Response, StreamingResponse
+from transformers.utils.hub import cached_file
 from vllm.entrypoints.openai.engine.serving import OpenAIServing
 from vllm.logger import init_logger
 from vllm.multimodal.media import MediaConnector
@@ -120,7 +121,9 @@ def _load_codec_frame_rate(self) -> float | None:
         try:
             model_path = self.engine_client.model_config.model
             st_config_path = os.path.join(model_path, "speech_tokenizer", "config.json")
-            if os.path.exists(st_config_path):
+            if not os.path.exists(st_config_path):
+                st_config_path = cached_file(model_path, "speech_tokenizer/config.json")
+            if st_config_path is not None and os.path.exists(st_config_path):
                 with open(st_config_path) as f:
                     st_config = json.load(f)
                 output_sr = st_config.get("output_sample_rate")

vllm_omni/model_executor/models/qwen3_tts/qwen3_tts_talker.py

@@ -447,6 +447,7 @@ def make_omni_output(self, model_outputs: torch.Tensor | OmniOutput, **kwargs: A
             logger.warning_once("runtime_additional_information is deprecated, use model_intermediate_buffer")
         audio_codes_list: list[torch.Tensor] = []
         ref_code_len_list: list[torch.Tensor] = []
+        ref_code_tensor: torch.Tensor | None = None
         codec_streaming_list: list[torch.Tensor] = []
         for info in info_dicts:
             if not isinstance(info, dict):
@@ -459,6 +460,9 @@ def make_omni_output(self, model_outputs: torch.Tensor | OmniOutput, **kwargs: A
                     codec_streaming_list.append(
                         torch.full((int(ac.shape[0]),), int(cs), dtype=torch.int8, device=ac.device)
                     )
+            ref_code = info.get("ref_code")
+            if isinstance(ref_code, torch.Tensor) and ref_code.numel() > 0:
+                ref_code_tensor = ref_code
             ref_len = info.get("ref_code_len")
             if ref_len is None:
                 continue
@@ -487,6 +491,8 @@ def make_omni_output(self, model_outputs: torch.Tensor | OmniOutput, **kwargs: A
         mm: dict[str, torch.Tensor] = {"audio_codes": audio_codes}
         if ref_code_len_list:
             mm["ref_code_len"] = torch.cat(ref_code_len_list, dim=0)[:span_len]
+        if ref_code_tensor is not None:
+            mm["ref_code"] = [ref_code_tensor]
         if codec_streaming_list:
             mm["codec_streaming"] = torch.cat(codec_streaming_list, dim=0)[:span_len]
         return OmniOutput(text_hidden_states=hidden, multimodal_outputs=mm)
@@ -538,8 +544,8 @@ def preprocess(
             # Subsequent prefill rounds (multi-chunk): prompt_embeds_cpu is a Tensor stored by the first round.
             is_first_prefill = not isinstance(prompt_embeds_cpu, torch.Tensor) or prompt_embeds_cpu.ndim != 2
             if is_first_prefill:
-                full_prompt_embeds, tailing_text_hidden, tts_pad_embed, ref_code_len = self._build_prompt_embeds(
-                    task_type=task_type, info_dict=info_dict
+                full_prompt_embeds, tailing_text_hidden, tts_pad_embed, ref_code_len, ref_code = (
+                    self._build_prompt_embeds(task_type=task_type, info_dict=info_dict)
                 )
                 # Store full prompt embeddings + trailing queue on CPU for later chunks/steps.
                 prompt_embeds_cpu = full_prompt_embeds.detach().to("cpu").contiguous()
@@ -550,6 +556,8 @@ def preprocess(
                     "talker_prefill_offset": 0,
                     "codec_streaming": codec_streaming,
                 }
+                if isinstance(ref_code, torch.Tensor) and ref_code.numel() > 0:
+                    info_update["ref_code"] = ref_code.detach().to("cpu").contiguous()
                 if ref_code_len is not None:
                     info_update["ref_code_len"] = int(ref_code_len)
                 # Always return a span_len slice; if the scheduled placeholder is longer, pad with tts_pad_embed.
@@ -773,7 +781,6 @@ def _first(x: object, default: object) -> object:
 
                 if ref_code_len is None and estimate_ref_code_len is not None:
                     ref_code_len = estimate_ref_code_len(info.get("ref_audio"))
-
                 if ref_code_len is None:
                     raise ValueError(
                         "Base in-context voice cloning requires either `voice_clone_prompt.ref_code` "
@@ -1183,7 +1190,7 @@ def _build_prompt_embeds(
         *,
         task_type: str,
         info_dict: dict[str, Any],
-    ) -> tuple[torch.Tensor, torch.Tensor, torch.Tensor, int | None]:
+    ) -> tuple[torch.Tensor, torch.Tensor, torch.Tensor, int | None, torch.Tensor | None]:
         text = (info_dict.get("text") or [""])[0]
         language = (info_dict.get("language") or ["Auto"])[0]
         non_streaming_mode_val = info_dict.get("non_streaming_mode")
@@ -1268,6 +1275,7 @@ def _build_prompt_embeds(
         # Speaker embedding/token (task-dependent)
         speaker_embed = None
         ref_code_len: int | None = None
+        ref_code_prompt: torch.Tensor | None = None
 
         def _as_singleton(x: object) -> object:
             if isinstance(x, list):
@@ -1333,6 +1341,8 @@ def _normalize_voice_clone_prompt(raw: object) -> dict[str, object] | None:
                 wav_np, sr = self._normalize_ref_audio(ref_audio_list[0])
                 ref_code_t = self._encode_ref_audio_to_code(wav_np, sr).to(device=input_ids.device)
                 ref_code_len = int(ref_code_t.shape[0])
+            if isinstance(ref_code_t, torch.Tensor):
+                ref_code_prompt = ref_code_t
 
             # Speaker embedding: use prompt embed if provided; otherwise extract from audio.
             spk = None
@@ -1521,6 +1531,7 @@ def _normalize_voice_clone_prompt(raw: object) -> dict[str, object] | None:
             trailing_text_hidden.squeeze(0),  # [T, H]
             tts_pad_embed.squeeze(0),  # [1, H]
             ref_code_len,
+            ref_code_prompt.contiguous() if isinstance(ref_code_prompt, torch.Tensor) else None,
         )
 
     def load_weights(self, weights: Iterable[tuple[str, torch.Tensor]]) -> set[str]:

vllm_omni/model_executor/stage_input_processors/qwen3_tts.py

@@ -27,13 +27,24 @@ def talker2code2wav(
         # Filter zero-padded frames (EOS/invalid steps), matching _extract_last_frame behavior
         valid_mask = audio_codes.any(dim=1)
         audio_codes = audio_codes[valid_mask]
+        ref_code = output.multimodal_output.get("ref_code")
+        if isinstance(ref_code, list):
+            ref_code = ref_code[0] if ref_code else None
+        if isinstance(ref_code, torch.Tensor) and ref_code.numel() > 0:
+            ref_code = ref_code.to(torch.long).cpu().contiguous()
+            ref_code_len = int(ref_code.shape[0])
+            audio_codes = torch.cat([ref_code.to(audio_codes.device), audio_codes], dim=0)
+        else:
+            ref_code_len = 0
         # Code2Wav expects codebook-major flat: [Q*num_frames]
         codec_codes = audio_codes.transpose(0, 1).cpu().reshape(-1).tolist()
+        additional_information = {"left_context_size": [ref_code_len]} if ref_code_len > 0 else None
         code2wav_inputs.append(
             OmniTokensPrompt(
                 prompt_token_ids=codec_codes,
                 multi_modal_data=None,
                 mm_processor_kwargs=None,
+                additional_information=additional_information,
             )
         )
     return code2wav_inputs
@@ -61,12 +72,19 @@ def talker2code2wav_async_chunk(
 ) -> dict[str, Any] | None:
     request_id = request.external_req_id
     finished = bool(is_finished or request.is_finished())
+    request_payload = getattr(transfer_manager, "request_payload", None)
+    if request_payload is None:
+        request_payload = {}
+        transfer_manager.request_payload = request_payload
 
     if isinstance(pooling_output, dict):
         frame = _extract_last_frame(pooling_output)
         if frame is not None:
             codec_codes = frame.cpu().tolist()
             transfer_manager.code_prompt_token_ids[request_id].append(codec_codes)
+        ref_code = pooling_output.get("ref_code")
+        if isinstance(ref_code, torch.Tensor) and ref_code.numel() > 0 and request_payload.get(request_id) is None:
+            request_payload[request_id] = ref_code.to(torch.long).cpu().contiguous()
     elif not finished:
         # Some steps may not produce pooling_output. Only flush on finish.
         return None
@@ -139,6 +157,14 @@ def talker2code2wav_async_chunk(
         left_context_size = max(0, int(end_index - context_length))
         window_frames = transfer_manager.code_prompt_token_ids[request_id][-end_index:]
 
+    # Match offline Base ICL decoding by prepending ref_code only once to the first decoder window.
+    if transfer_manager.put_req_chunk[request_id] == 0:
+        ref_code = request_payload.pop(request_id, None)
+        if isinstance(ref_code, torch.Tensor) and ref_code.numel() > 0:
+            ref_frames = ref_code.tolist()
+            window_frames = ref_frames + window_frames
+            left_context_size += len(ref_frames)
+
     # Pack context + chunk into codebook-major flat codes for adapter.
     code_predictor_codes = torch.tensor(window_frames).transpose(0, 1).reshape(-1).tolist()