From 208932210f118940cd88880c3f7ab52960903327 Mon Sep 17 00:00:00 2001
From: Yueqian Lin <70319226+linyueqian@users.noreply.github.com>
Date: Fri, 19 Jun 2026 00:05:26 -0700
Subject: [PATCH] [Chore][TTS] Reference recipes.vllm.ai recipes, remove dead
 code, and fix stale references (#4567)

Signed-off-by: Yueqian Lin <linyueqian@outlook.com>
---
 README.md                                     |   1 +
 docs/models/supported_models.md               |   5 +-
 docs/serving/speech_api.md                    |   7 +
 recipes/README.md                             |  11 +-
 .../entrypoints/openai/serving_speech.py      |   1 -
 .../entrypoints/openai/tts_adapters/base.py   |   3 -
 .../models/cosyvoice3/cosyvoice3.py           |  35 +----
 .../models/cosyvoice3/cosyvoice3_code2wav.py  |  73 ++++-------
 .../models/cosyvoice3/pipeline.py             |   6 +-
 .../model_executor/models/cosyvoice3/utils.py |  22 ----
 .../models/covo_audio/config_covo_audio.py    |   1 -
 .../models/covo_audio/token2wav.py            |   5 +-
 .../fish_speech/fish_speech_dac_decoder.py    |  18 ++-
 .../models/fish_speech/fish_speech_fast_ar.py |  11 +-
 .../models/fish_speech/prompt_utils.py        |  10 --
 .../models/glm_tts/glm_tts_dit.py             |   4 -
 .../configuration_higgs_audio_v2.py           |  12 --
 .../higgs_audio_v2/higgs_audio_v2_talker.py   |  35 +++--
 .../models/higgs_audio_v3/ref_audio_cache.py  | 111 ----------------
 .../models/ming_flash_omni/projectors.py      |  27 ----
 .../models/ming_flash_omni/text_processing.py |  91 -------------
 .../models/ming_tts/config_ming_tts.py        |   2 -
 .../models/ming_tts/constants.py              |   1 -
 .../models/ming_tts/ming_tts.py               |   4 -
 .../models/ming_tts/ming_tts_llm.py           |  14 +-
 .../models/moss_tts/audio_tokenizer.py        |   2 +-
 .../moss_tts/modeling_moss_tts_talker.py      |   4 +-
 .../configuration_moss_tts_nano.py            |  60 ---------
 .../moss_tts_nano/modeling_moss_tts_nano.py   |   1 -
 .../models/omnivoice/omnivoice.py             |   3 +-
 .../models/omnivoice/omnivoice_generator.py   |  23 ++--
 .../qwen3_tts/configuration_qwen3_tts.py      |  11 +-
 .../models/qwen3_tts/prompt_embeds_builder.py |   8 +-
 .../models/qwen3_tts/qwen3_tts_talker.py      |   7 -
 .../modeling_qwen3_tts_tokenizer_v2.py        |   5 -
 .../qwen3_tts/tokenizer_25hz/vq/core_vq.py    |   5 -
 .../models/step_audio2/step_audio2.py         |   8 +-
 .../step_audio2/step_audio2_constants.py      |  36 ------
 .../models/step_audio2/step_audio2_thinker.py |  30 +----
 .../step_audio2/step_audio2_token2wav.py      | 121 ------------------
 .../models/voxcpm2/minicpm4_paged.py          |  17 +--
 .../models/voxcpm2/voxcpm2_talker.py          |  43 +++----
 .../models/voxtral_tts/voxtral_tts.py         |   3 -
 .../voxtral_tts_audio_generation.py           |  26 +---
 .../voxtral_tts_audio_tokenizer.py            |   4 +-
 .../transformers_utils/processors/ming.py     |   1 -
 46 files changed, 141 insertions(+), 787 deletions(-)
 delete mode 100644 vllm_omni/model_executor/models/higgs_audio_v3/ref_audio_cache.py
 delete mode 100644 vllm_omni/model_executor/models/moss_tts_nano/configuration_moss_tts_nano.py

diff --git a/README.md b/README.md
index cf1658b45..e300b6b8d 100644
--- a/README.md
+++ b/README.md
@@ -68,6 +68,7 @@ Visit our [documentation](https://vllm-omni.readthedocs.io/en/latest/) to learn
 - [Installation](https://vllm-omni.readthedocs.io/en/latest/getting_started/installation/)
 - [Quickstart](https://vllm-omni.readthedocs.io/en/latest/getting_started/quickstart/)
 - [List of Supported Models](https://vllm-omni.readthedocs.io/en/latest/models/supported_models/)
+- [Deployment Recipes](https://recipes.vllm.ai) for vLLM-Omni model serving
 
 ## Contributing
 
diff --git a/docs/models/supported_models.md b/docs/models/supported_models.md
index 119f3bf68..e69f462cf 100644
--- a/docs/models/supported_models.md
+++ b/docs/models/supported_models.md
@@ -6,6 +6,8 @@ vLLM-Omni supports unified multimodal comprehension and generation models across
 
 If vLLM-Omni natively supports a model, its implementation can be found in <gh-file:vllm_omni/model_executor/models> and <gh-file:vllm_omni/diffusion/models>.
 
+For deployment recipes (TTS and multimodal), see [recipes.vllm.ai](https://recipes.vllm.ai) and the in-repo [`recipes/`](https://github.com/vllm-project/vllm-omni/tree/main/recipes) directory.
+
 ## List of Supported Models
 
 <style>
@@ -63,13 +65,14 @@ th {
 | `AudioXPipeline` | AudioX | `zhangj1an/AudioX` | ✅︎ | ✅︎ | | |
 | `Qwen3TTSForConditionalGeneration` | Qwen3-TTS-12Hz-1.7B-CustomVoice | `Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice` | ✅︎ | ✅︎ | ✅︎ | ✅︎ |
 | `Qwen3TTSForConditionalGeneration` | Qwen3-TTS-12Hz-1.7B-VoiceDesign | `Qwen/Qwen3-TTS-12Hz-1.7B-VoiceDesign` | ✅︎ | ✅︎ | ✅︎ | ✅︎ |
-| `Qwen3TTSForConditionalGeneration` | Qwen3-TTS-12Hz-1.7B-Base | `Qwen/Qwen3-TTS-12Hz-0.6B-Base` | ✅︎ | ✅︎ | ✅︎ | ✅︎ |
+| `Qwen3TTSForConditionalGeneration` | Qwen3-TTS-12Hz-1.7B-Base | `Qwen/Qwen3-TTS-12Hz-1.7B-Base` | ✅︎ | ✅︎ | ✅︎ | ✅︎ |
 | `MingTTSForConditionalGeneration` | Ming-omni-tts dense 0.5B | `inclusionAI/Ming-omni-tts-0.5B` | ✅︎ | | | |
 | `GLMTTSForConditionalGeneration` | GLM-TTS | `zai-org/GLM-TTS` | ✅︎ | | | |
 | `MossTTSNanoForCausalLM` | MOSS-TTS-Nano | `OpenMOSS-Team/MOSS-TTS-Nano` | ✅︎ | | | |
 | `MossTTSDelayModel` | MOSS-TTS, MOSS-TTSD, MOSS-SoundEffect, MOSS-VoiceGenerator | `OpenMOSS-Team/MOSS-VoiceGenerator` | ✅︎ | | | |
 | `MossTTSRealtime` | MOSS-TTS-Realtime | `OpenMOSS-Team/MOSS-TTS-Realtime` | ✅︎ | | | |
 | `HiggsAudioV2ForConditionalGeneration` | Higgs-Audio v2 | `bosonai/higgs-audio-v2-generation-3B-base` | ✅︎ | | | |
+| `HiggsMultimodalQwen3ForConditionalGeneration` | Higgs-Audio v3 (TTS) | `bosonai/higgs-audio-v3-tts-4b` | ✅︎ | | | |
 | `NextStep11Pipeline` | NextStep-1.1 | `stepfun-ai/NextStep-1.1` | ✅︎ | ✅︎ | | ✅︎ |
 | `MiMoAudioModel` | MiMo-Audio-7B-Instruct | `XiaomiMiMo/MiMo-Audio-7B-Instruct` | ✅︎ | ✅︎ | | |
 | `MiMoV2ASRForCausalLM` | MiMo-V2.5-ASR | `XiaomiMiMo/MiMo-V2.5-ASR` | ✅︎ | ✅︎ | | |
diff --git a/docs/serving/speech_api.md b/docs/serving/speech_api.md
index 935deba4f..b70c28297 100644
--- a/docs/serving/speech_api.md
+++ b/docs/serving/speech_api.md
@@ -9,6 +9,13 @@ vLLM-Omni provides an OpenAI-compatible API for text-to-speech (TTS) generation.
 
 See the [Supported Models](#supported-models) section below for the full list, including OmniVoice, VoxCPM2, and MOSS-TTS-Nano.
 
+!!! tip "Deployment recipes"
+    TTS deployment recipes are published at
+    [recipes.vllm.ai](https://recipes.vllm.ai) (e.g.
+    [Qwen3-TTS](https://recipes.vllm.ai/Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice),
+    [Higgs-Audio v3](https://recipes.vllm.ai/bosonai/higgs-audio-v3-tts-4b)).
+    The in-repo runbooks live under [`recipes/`](https://github.com/vllm-project/vllm-omni/tree/main/recipes).
+
 Each server instance runs a single model (specified at startup via `vllm serve <model> --omni`).
 
 ## Quick Start
diff --git a/recipes/README.md b/recipes/README.md
index 1f5b5be57..e3ee2bc56 100644
--- a/recipes/README.md
+++ b/recipes/README.md
@@ -12,6 +12,13 @@ that is aligned with
 that external repository as a reference for structure rather than the place to
 add files for this repo. Use one Markdown file per model family by default.
 
+> **Note:** The vLLM-Omni TTS recipes (12 models across 8 providers) are also
+> published on the rendered recipes site at
+> [recipes.vllm.ai](https://recipes.vllm.ai) (see
+> [vllm-project/recipes#554](https://github.com/vllm-project/recipes/pull/554)).
+> Individual pages resolve at `https://recipes.vllm.ai/<provider>/<model-id>`,
+> e.g. [`bosonai/higgs-audio-v3-tts-4b`](https://recipes.vllm.ai/bosonai/higgs-audio-v3-tts-4b).
+
 Example layout:
 
 ```text
@@ -30,11 +37,12 @@ recipes/
 | [`audiox/AudioX.md`](./audiox/AudioX.md) | Offline + online unified text/video→audio diffusion | 1x L4 24GB |
 | [`Baidu/ERNIE-Image.md`](./Baidu/ERNIE-Image.md) | Text-to-image online serving (ERNIE-Image 8B) | 1x or 2x RTX 4090 24GB |
 | [`Bagel/BAGEL-7B-MoT.md`](./Bagel/BAGEL-7B-MoT.md) | Text-to-image with shared online/offline examples | 1x A100 80GB / 2x CUDA GPUs |
+| [`BosonAI/Higgs-Audio-V3-TTS.md`](./BosonAI/Higgs-Audio-V3-TTS.md) | Online + offline multilingual TTS with voice cloning | 1x H100 80GB |
 | [`ByteDance/Lance.md`](./ByteDance/Lance.md) | Unified AR+diffusion: text/img/video gen + understanding (Lance 3B) | 1x B300 / A100 80GB |
 | [`fishaudio/Fish-Speech-S2-Pro.md`](./fishaudio/Fish-Speech-S2-Pro.md) | Online serving for TTS | 1x A800 80GB |
 | [`Helios/Helios.md`](./Helios/Helios.md) | Text-to-video, image-to-video, and video-to-video generation | 1x NVIDIA H20 |
 | [`inclusionAI/Ming-flash-omni-2.0.md`](./inclusionAI/Ming-flash-omni-2.0.md) | Online serving for multimodal chat + standalone TTS | 4x H100 / 1x H100 80GB |
-| [`inclusionAI/Ming-omni-tts-0.5B.md`](./inclusionAI/Ming-omni-tts-0.5B.md) | Offline + online dense Ming TTS/audio generation | 1x AMD `gfx942` (ROCm 7.2) |
+| [`inclusionAI/Ming-omni-tts.md`](./inclusionAI/Ming-omni-tts.md) | Offline + online dense Ming TTS/audio generation | 1x H100 80GB / 1x AMD MI300X (ROCm 7.2) |
 | [`LTX/LTX-2.md`](./LTX/LTX-2.md) | Text-to-video and image-to-video serving | 1x H200 141GB |
 | [`LTX/LTX-2.3.md`](./LTX/LTX-2.3.md) | Text-to-video with audio generation (22B) | 1x GPU (96GB VRAM) |
 | [`mistralai/Voxtral-TTS.md`](./mistralai/Voxtral-TTS.md) | Online serving for TTS | 1x RTX 4090 24GB |
@@ -42,6 +50,7 @@ recipes/
 | [`cosmos3/Cosmos3-Super.md`](./cosmos3/Cosmos3-Super.md) | 64B T2I / T2V / I2V / V2V generation (+ optional audio) / Action policy | 8x H200/H100/A100 / 2x H200 / B300 |
 | [`OpenBMB/MiniCPM-o-4_5.md`](./OpenBMB/MiniCPM-o-4_5.md) | Online serving for omni multimodal chat (text / image / audio / video → text + 24 kHz speech) | 2x A100/H100 80GB / 3x mid-tier GPU / 8x RTX 4090 24GB |
 | [`OpenBMB/VoxCPM2.md`](./OpenBMB/VoxCPM2.md) | Online + offline TTS with native AR pipeline (48 kHz, 30+ languages) | 1x RTX 4090 24GB |
+| [`OpenMOSS/MOSS-TTS.md`](./OpenMOSS/MOSS-TTS.md) | Online + offline multilingual TTS (MOSS-TTS family, 8B) | 1x H100 80GB |
 | [`Qwen/Qwen-Image.md`](./Qwen/Qwen-Image.md) | Text-to-image serving with step-wise continuous batching replay and ModelOpt mixed FP8/NVFP4 | 1x A100 80GB / 2x B200 |
 | [`Qwen/Qwen-Image-2512.md`](./Qwen/Qwen-Image-2512.md) | Text-to-image serving with step-wise continuous batching replay and ModelOpt FP8 / mixed FP8/NVFP4 | 1x A800 80GB / 2x B200 |
 | [`Qwen/Qwen-Image-Edit.md`](./Qwen/Qwen-Image-Edit.md) | Text-guided single-image editing | 1x or 2x H200 141GB |
diff --git a/vllm_omni/entrypoints/openai/serving_speech.py b/vllm_omni/entrypoints/openai/serving_speech.py
index 1f7a6b79d..f867bfadb 100644
--- a/vllm_omni/entrypoints/openai/serving_speech.py
+++ b/vllm_omni/entrypoints/openai/serving_speech.py
@@ -672,7 +672,6 @@ class OmniOpenAIServingSpeech(OpenAIServing, AudioMixin):
         if model_stage in _OMNIVOICE_TTS_MODEL_STAGES:
             return "omnivoice"
         if model_stage in _COVO_AUDIO_MODEL_STAGES:
-            model_arch = getattr(self._tts_stage.engine_args, "model_arch", None)
             if model_arch and "CovoAudio" in model_arch:
                 return "covo_audio"
         if model_stage in _VOXCPM2_TTS_MODEL_STAGES:
diff --git a/vllm_omni/entrypoints/openai/tts_adapters/base.py b/vllm_omni/entrypoints/openai/tts_adapters/base.py
index b6d3d3e9f..ad753a91e 100644
--- a/vllm_omni/entrypoints/openai/tts_adapters/base.py
+++ b/vllm_omni/entrypoints/openai/tts_adapters/base.py
@@ -194,6 +194,3 @@ __all__ = [
     "SpeechServingContext",
     "TTSModelAdapter",
 ]
-
-# Type alias kept for readability in adapter signatures.
-NormalizeFn = Callable[["OpenAICreateSpeechRequest"], None]
diff --git a/vllm_omni/model_executor/models/cosyvoice3/cosyvoice3.py b/vllm_omni/model_executor/models/cosyvoice3/cosyvoice3.py
index 1a387cdcc..d32ea62f2 100644
--- a/vllm_omni/model_executor/models/cosyvoice3/cosyvoice3.py
+++ b/vllm_omni/model_executor/models/cosyvoice3/cosyvoice3.py
@@ -479,14 +479,6 @@ class CosyVoice3Model(
             # Keep additional information synchronized for async_chunk updates.
             self.enable_update_additional_information = True
 
-            # Expose streaming parameters
-            self.token_overlap_len = self.code2wav.token_overlap_len
-            self.mel_overlap_len = self.code2wav.mel_overlap_len
-            self.mel_window = self.code2wav.mel_window
-            self.mel_cache_len = self.code2wav.mel_cache_len
-            self.source_cache_len = self.code2wav.source_cache_len
-            self.speech_window = self.code2wav.speech_window
-            self._stream_audio_cache_by_req: dict[str, torch.Tensor] = {}
             self._stream_audio_cache_lock = Lock()
             self._stream_vocoder_cache_by_req: dict[str, dict[str, torch.Tensor]] = {}
         else:
@@ -511,27 +503,6 @@ class CosyVoice3Model(
         # Use parent's cache config - critical for PagedAttention to work correctly
         return parent_config.with_hf_config(qwen_hf_config, architectures=["Qwen2Model"])
 
-    @staticmethod
-    def _cross_fade_audio(audio: torch.Tensor, prev_tail: torch.Tensor) -> torch.Tensor:
-        """Blend previous chunk tail into current chunk head using a Hamming window.
-
-        This mirrors upstream CosyVoice's `fade_in_out(...)` semantics:
-        update the current head in-place using a 2*overlap window, then
-        concatenate the unchanged remainder.
-        """
-        if audio.numel() == 0 or prev_tail.numel() == 0:
-            return audio
-        overlap = min(int(audio.numel()), int(prev_tail.numel()))
-        if overlap <= 0:
-            return audio
-        window = torch.hamming_window(2 * overlap, periodic=False, dtype=audio.dtype, device=audio.device)
-        fade_in = window[:overlap]
-        fade_out = window[overlap:]
-        blended = audio[:overlap] * fade_in + prev_tail[-overlap:].to(device=audio.device, dtype=audio.dtype) * fade_out
-        if overlap == int(audio.numel()):
-            return blended
-        return torch.cat([blended, audio[overlap:]], dim=0)
-
     def _stitch_stream_audio(self, req_id: str | None, audio: torch.Tensor, stream_finished: bool) -> torch.Tensor:
         """Pass-through stitching for async_chunk.
 
@@ -539,11 +510,9 @@ class CosyVoice3Model(
         Applying an additional waveform-domain fade/cache step introduces either
         duplicated overlap (if no tail trim) or duration shrink (if tail trim).
         """
-        if req_id is not None and stream_finished and hasattr(self, "_stream_audio_cache_by_req"):
+        if req_id is not None and stream_finished and hasattr(self, "_stream_vocoder_cache_by_req"):
             with self._stream_audio_cache_lock:
-                self._stream_audio_cache_by_req.pop(req_id, None)
-                if hasattr(self, "_stream_vocoder_cache_by_req"):
-                    self._stream_vocoder_cache_by_req.pop(req_id, None)
+                self._stream_vocoder_cache_by_req.pop(req_id, None)
         return audio
 
     @staticmethod
diff --git a/vllm_omni/model_executor/models/cosyvoice3/cosyvoice3_code2wav.py b/vllm_omni/model_executor/models/cosyvoice3/cosyvoice3_code2wav.py
index cb3228c13..49e8e515b 100644
--- a/vllm_omni/model_executor/models/cosyvoice3/cosyvoice3_code2wav.py
+++ b/vllm_omni/model_executor/models/cosyvoice3/cosyvoice3_code2wav.py
@@ -11,8 +11,6 @@ This module contains the code2wav (token-to-waveform) stage which uses:
 
 from __future__ import annotations
 
-from contextlib import nullcontext
-
 import numpy as np
 import torch
 import torch.nn as nn
@@ -175,19 +173,18 @@ class CosyVoice3Code2Wav(nn.Module):
         prompt_token_len = torch.tensor([prompt_token.shape[1]], device=device, dtype=torch.int32)
         prompt_feat_len = torch.tensor([prompt_feat.shape[1]], device=device, dtype=torch.int32)
 
-        with nullcontext():
-            feat, _ = self.flow_model.inference(
-                token=token,
-                token_len=token_len,
-                prompt_token=prompt_token,
-                prompt_token_len=prompt_token_len,
-                prompt_feat=prompt_feat,
-                prompt_feat_len=prompt_feat_len,
-                embedding=embedding,
-                streaming=streaming,
-                finalize=finalize,
-                n_timesteps=n_timesteps,
-            )
+        feat, _ = self.flow_model.inference(
+            token=token,
+            token_len=token_len,
+            prompt_token=prompt_token,
+            prompt_token_len=prompt_token_len,
+            prompt_feat=prompt_feat,
+            prompt_feat_len=prompt_feat_len,
+            embedding=embedding,
+            streaming=streaming,
+            finalize=finalize,
+            n_timesteps=n_timesteps,
+        )
 
         trim_mel = max(0, int(token_offset_tokens)) * int(self.token_mel_ratio)
         if trim_mel > 0:
@@ -195,28 +192,6 @@ class CosyVoice3Code2Wav(nn.Module):
 
         return feat
 
-    @staticmethod
-    def _fade_speech(
-        speech: torch.Tensor,
-        prev_speech: torch.Tensor,
-    ) -> torch.Tensor:
-        """Blend previous speech tail into current speech head."""
-        if speech.numel() == 0 or prev_speech.numel() == 0:
-            return speech
-        overlap = min(int(speech.shape[-1]), int(prev_speech.shape[-1]))
-        if overlap <= 0:
-            return speech
-        window = torch.hamming_window(2 * overlap, periodic=False, dtype=speech.dtype, device=speech.device)
-        fade_in = window[:overlap].view(1, -1)
-        fade_out = window[overlap:].view(1, -1)
-        blended_head = (
-            speech[:, :overlap] * fade_in
-            + prev_speech[:, -overlap:].to(device=speech.device, dtype=speech.dtype) * fade_out
-        )
-        if overlap == int(speech.shape[-1]):
-            return blended_head
-        return torch.cat([blended_head, speech[:, overlap:]], dim=-1)
-
     @torch.inference_mode()
     def forward_streaming(
         self,
@@ -238,17 +213,16 @@ class CosyVoice3Code2Wav(nn.Module):
         suffix. That preserves causal look-right handling without double
         trimming or duplicated overlap at chunk boundaries.
         """
-        with nullcontext():
-            feat = self._forward_mel(
-                token=token,
-                prompt_token=prompt_token,
-                prompt_feat=prompt_feat,
-                embedding=embedding,
-                n_timesteps=n_timesteps,
-                token_offset_tokens=token_offset_tokens,
-                streaming=True,
-                finalize=finalize,
-            )
+        feat = self._forward_mel(
+            token=token,
+            prompt_token=prompt_token,
+            prompt_feat=prompt_feat,
+            embedding=embedding,
+            n_timesteps=n_timesteps,
+            token_offset_tokens=token_offset_tokens,
+            streaming=True,
+            finalize=finalize,
+        )
         hift_weight = self.hift.m_source.l_linear.weight
         chunk_mel = feat.to(device=hift_weight.device, dtype=hift_weight.dtype)
 
@@ -268,8 +242,7 @@ class CosyVoice3Code2Wav(nn.Module):
         if tts_mel.shape[-1] == 0:
             tts_speech = torch.zeros((chunk_mel.shape[0], 1, 0), device=chunk_mel.device, dtype=chunk_mel.dtype)
         else:
-            with nullcontext():
-                tts_speech, _ = self.hift.inference(speech_feat=tts_mel, finalize=finalize)
+            tts_speech, _ = self.hift.inference(speech_feat=tts_mel, finalize=finalize)
 
         tts_speech = tts_speech.reshape(tts_speech.shape[0], -1)
         speech_offset = max(0, min(speech_offset, int(tts_speech.shape[-1])))
diff --git a/vllm_omni/model_executor/models/cosyvoice3/pipeline.py b/vllm_omni/model_executor/models/cosyvoice3/pipeline.py
index ed35c93bd..32db8ef5b 100644
--- a/vllm_omni/model_executor/models/cosyvoice3/pipeline.py
+++ b/vllm_omni/model_executor/models/cosyvoice3/pipeline.py
@@ -4,8 +4,10 @@
 
 Stage 0: Talker   — text prompt → speech tokens (LLM autoregressive).
 Stage 1: Code2Wav — flow-matching decoder → acoustic features → waveform.
-  * ``sync_process_input_func`` runs when ``deploy.async_chunk=false``:
-    stage 1 builds full-sequence flow input via ``text2flow``.
+  * ``sync_process_input_func`` (``text2flow_token_only``) runs when
+    ``deploy.async_chunk=false``: stage 1 builds full-sequence flow input
+    from the emitted speech tokens. ``text2flow`` is the
+    ``custom_process_input_func`` variant.
   * ``async_chunk_process_next_stage_input_func`` runs when
     ``deploy.async_chunk=true``: stage 0 streams codec chunks to stage 1
     through the shared-memory connector.
diff --git a/vllm_omni/model_executor/models/cosyvoice3/utils.py b/vllm_omni/model_executor/models/cosyvoice3/utils.py
index 5827e6bbb..d2c3c6865 100644
--- a/vllm_omni/model_executor/models/cosyvoice3/utils.py
+++ b/vllm_omni/model_executor/models/cosyvoice3/utils.py
@@ -13,8 +13,6 @@ from vllm_omni.utils.audio import mel_filter_bank
 
 logger = logging.getLogger(__name__)
 
-IGNORE_ID = -1
-
 
 def dynamic_range_compression_torch(x, c=1, clip_val=1e-5):
     return torch.log(torch.clamp(x, min=clip_val) * c)
@@ -182,26 +180,6 @@ def log_mel_spectrogram(
     return log_spec
 
 
-def extract_speech_token(prompt_wav, speech_tokenizer_session, device):
-    speech = load_wav(prompt_wav, 16000)
-    assert speech.shape[1] / 16000 <= 30, "do not support extract speech token for audio longer than 30s"
-    feat = log_mel_spectrogram(speech, n_mels=128)
-    speech_token = (
-        speech_tokenizer_session.run(
-            None,
-            {
-                speech_tokenizer_session.get_inputs()[0].name: feat.detach().cpu().numpy(),
-                speech_tokenizer_session.get_inputs()[1].name: np.array([feat.shape[2]], dtype=np.int32),
-            },
-        )[0]
-        .flatten()
-        .tolist()
-    )
-    speech_token = torch.tensor([speech_token], dtype=torch.int32).to(device)
-    speech_token_len = torch.tensor([speech_token.shape[1]], dtype=torch.int32).to(device)
-    return speech_token, speech_token_len
-
-
 def extract_spk_embedding(prompt_wav, campplus_session, device):
     speech = load_wav(prompt_wav, 16000)
     feat = kaldi.fbank(speech, num_mel_bins=80, dither=0, sample_frequency=16000)
diff --git a/vllm_omni/model_executor/models/covo_audio/config_covo_audio.py b/vllm_omni/model_executor/models/covo_audio/config_covo_audio.py
index 8153c2f38..b9b06857d 100644
--- a/vllm_omni/model_executor/models/covo_audio/config_covo_audio.py
+++ b/vllm_omni/model_executor/models/covo_audio/config_covo_audio.py
@@ -67,6 +67,5 @@ class CovoAudioCode2WavConfig(PretrainedConfig):
         self.inference = {
             "s_steps": 10,
             "cfg_alpha": 1.0,
-            "dynamic_the": False,
             "rescale_logits": False,
         }
diff --git a/vllm_omni/model_executor/models/covo_audio/token2wav.py b/vllm_omni/model_executor/models/covo_audio/token2wav.py
index 4a6ce4b7e..8a96e1f39 100644
--- a/vllm_omni/model_executor/models/covo_audio/token2wav.py
+++ b/vllm_omni/model_executor/models/covo_audio/token2wav.py
@@ -598,7 +598,7 @@ class Token2latentFlowMatching(nn.Module):
         latent_cond = torch.zeros(b, tgt_lens, self.target_dim).to(device)
         if prefix_target is not None:
             latent_cond[:, : prefix_target.size(1), :] = prefix_target
-        sample, trajectory = self.sample(
+        sample, _ = self.sample(
             tokens=token, audio=latent_cond, steps=s_steps, alpha=cfg_alpha, g_cond=ge, rescale_logits=rescale_logits
         )
         return sample
@@ -612,9 +612,8 @@ class Token2latentFlowMatching(nn.Module):
                 output = torch.cat([tokens, audio, z], dim=-1)
                 return self.vectorfield_forward(inputs=output, times=t.unsqueeze(0), self_attn_mask=None, g_cond=g_cond)
             tokens_empty = torch.zeros(*audio.shape[:2], self.model_dim, device=tokens.device, dtype=tokens.dtype)
-            audio_empty = audio
             tokens_t = torch.cat([tokens_empty, tokens], dim=0)
-            audio_t = torch.cat([audio_empty, audio], dim=0)
+            audio_t = torch.cat([audio, audio], dim=0)
             audio_noizy_t = torch.cat([z, z], dim=0)
             t_t = torch.stack([t, t], dim=0)
             c = g_cond
diff --git a/vllm_omni/model_executor/models/fish_speech/fish_speech_dac_decoder.py b/vllm_omni/model_executor/models/fish_speech/fish_speech_dac_decoder.py
index c6e9064f1..9627dc7ce 100644
--- a/vllm_omni/model_executor/models/fish_speech/fish_speech_dac_decoder.py
+++ b/vllm_omni/model_executor/models/fish_speech/fish_speech_dac_decoder.py
@@ -348,18 +348,16 @@ class FishSpeechDACDecoder(nn.Module):
 
             if req_ids.numel() < 1:
                 continue
-            flat = req_ids
-            n = flat.numel()
-            if n == 0 or n % q != 0:
-                if n > 0:
-                    logger.warning(
-                        "DAC decoder input_ids length %d not divisible by num_codebooks %d; returning empty audio.",
-                        n,
-                        q,
-                    )
+            n = req_ids.numel()
+            if n % q != 0:
+                logger.warning(
+                    "DAC decoder input_ids length %d not divisible by num_codebooks %d; returning empty audio.",
+                    n,
+                    q,
+                )
                 continue
             frames = n // q
-            codes_qf = flat.reshape(q, frames)
+            codes_qf = req_ids.reshape(q, frames)
             parsed_ctx_frames[i] = ctx_frames
             parsed_total_frames[i] = frames
             valid_codes_qf.append(codes_qf)
diff --git a/vllm_omni/model_executor/models/fish_speech/fish_speech_fast_ar.py b/vllm_omni/model_executor/models/fish_speech/fish_speech_fast_ar.py
index 46ceee498..eb056f16a 100644
--- a/vllm_omni/model_executor/models/fish_speech/fish_speech_fast_ar.py
+++ b/vllm_omni/model_executor/models/fish_speech/fish_speech_fast_ar.py
@@ -191,7 +191,11 @@ class _FastARMLP(nn.Module):
 
 
 class _FastARDecoderLayer(nn.Module):
-    """Transformer decoder layer for Fast AR (SDPA, no KV cache)."""
+    """Transformer decoder layer for Fast AR.
+
+    The batched ``forward`` runs SDPA over the full sequence and is cache-free,
+    while ``forward_one`` decodes a single step using a per-call KV cache.
+    """
 
     def __init__(self, config: FishSpeechFastARConfig, *, prefix: str = "") -> None:
         super().__init__()
@@ -519,7 +523,7 @@ class FishSpeechFastAR(nn.Module):
         embed_buf[:bsz, 0, :] = projected
 
         # Position 1: embedding of semantic code.
-        code_embed = self.fast_embeddings(semantic_code.clamp(min=0))
+        code_embed = self.fast_embeddings(semantic_code)
         embed_buf[:bsz, 1, :] = code_embed
 
         use_sampling = do_sample and temperature > 0
@@ -542,8 +546,7 @@ class FishSpeechFastAR(nn.Module):
             logits = self.fast_output(self.fast_norm(hidden_out))
 
             # Residual codebooks (step >= 1) only have 1024 entries.
-            if step >= 1:
-                logits = logits[:, :residual_codebook_size]
+            logits = logits[:, :residual_codebook_size]
 
             if use_sampling:
                 scaled = logits * inv_temperature
diff --git a/vllm_omni/model_executor/models/fish_speech/prompt_utils.py b/vllm_omni/model_executor/models/fish_speech/prompt_utils.py
index 8b8d8559e..3918a6f94 100644
--- a/vllm_omni/model_executor/models/fish_speech/prompt_utils.py
+++ b/vllm_omni/model_executor/models/fish_speech/prompt_utils.py
@@ -132,13 +132,3 @@ def estimate_fish_voice_clone_prompt_len_from_normalized(
         [0] * semantic_len,
     )
     return len(prompt_ids)
-
-
-def estimate_fish_voice_clone_prompt_len(tokenizer: Any, text: str, ref_text: str, semantic_len: int) -> int:
-    normalized_text, normalized_ref_text = normalize_fish_voice_clone_texts(text, ref_text)
-    return estimate_fish_voice_clone_prompt_len_from_normalized(
-        tokenizer,
-        normalized_text,
-        normalized_ref_text,
-        semantic_len,
-    )
diff --git a/vllm_omni/model_executor/models/glm_tts/glm_tts_dit.py b/vllm_omni/model_executor/models/glm_tts/glm_tts_dit.py
index 5cda6ec45..666998e9d 100644
--- a/vllm_omni/model_executor/models/glm_tts/glm_tts_dit.py
+++ b/vllm_omni/model_executor/models/glm_tts/glm_tts_dit.py
@@ -472,10 +472,6 @@ class GLMTTSDiT(nn.Module):
         self.num_heads = heads
         self.spkr_emb_adaln = spkr_emb_adaln
 
-        # Frame rates for block-causal mask token→mel conversion
-        self._input_frame_rate: float = 25.0
-        self._mel_framerate: float = 50.0
-
         # Speaker embedding dimension for AdaLN
         if spkr_emb_adaln:
             additional_dim = spkr_dim + 256 if use_wavlm_emb else spkr_dim
diff --git a/vllm_omni/model_executor/models/higgs_audio_v2/configuration_higgs_audio_v2.py b/vllm_omni/model_executor/models/higgs_audio_v2/configuration_higgs_audio_v2.py
index dc5eb12b4..b6eef8dc4 100644
--- a/vllm_omni/model_executor/models/higgs_audio_v2/configuration_higgs_audio_v2.py
+++ b/vllm_omni/model_executor/models/higgs_audio_v2/configuration_higgs_audio_v2.py
@@ -155,15 +155,3 @@ class HiggsAudioV2Config(PretrainedConfig):
     @property
     def num_real_codes(self) -> int:
         return self.audio_stream_bos_id  # i.e. 1024 for the boson-ai checkpoint
-
-    # The full vocab size of each per-codebook output head.
-    @property
-    def codebook_output_size(self) -> int:
-        return self.codebook_size  # i.e. 1026 for the boson-ai checkpoint
-
-    # Canonical MusicGen-style delay pattern used by
-    # `HiggsAudioV2DelayPatternLogitsProcessor`: codebook k starts emitting
-    # real codes only after k frames. Kept as a function so subclasses can
-    # override if upstream ever publishes a different convention.
-    def default_delay_pattern(self) -> list[int]:
-        return list(range(self.num_codebooks))
diff --git a/vllm_omni/model_executor/models/higgs_audio_v2/higgs_audio_v2_talker.py b/vllm_omni/model_executor/models/higgs_audio_v2/higgs_audio_v2_talker.py
index df636a4bb..be08163e4 100644
--- a/vllm_omni/model_executor/models/higgs_audio_v2/higgs_audio_v2_talker.py
+++ b/vllm_omni/model_executor/models/higgs_audio_v2/higgs_audio_v2_talker.py
@@ -821,12 +821,12 @@ class HiggsAudioV2TalkerForConditionalGeneration(nn.Module):
         except Exception:
             self._last_step_query_start_loc = None
 
-        # Voice-clone ref-audio substitution: at the prompt-side audio
-        # placeholder span we overwrite the text-embed lookups with the
-        # multi-codebook embedding of the encoded reference clip carried in
-        # ``model_intermediate_buffer``. No-op for plain-text requests and
-        # for decode steps (which fall through to the audio-feedback path).
         if input_ids is not None and inputs_embeds is None:
+            # Voice-clone ref-audio substitution: at the prompt-side audio
+            # placeholder span we overwrite the text-embed lookups with the
+            # multi-codebook embedding of the encoded reference clip carried in
+            # ``model_intermediate_buffer``. No-op for plain-text requests and
+            # for decode steps (which fall through to the audio-feedback path).
             info_dicts = kwargs.get("model_intermediate_buffer")
             if info_dicts is None:
                 info_dicts = kwargs.get("runtime_additional_information")
@@ -836,15 +836,14 @@ class HiggsAudioV2TalkerForConditionalGeneration(nn.Module):
                 info_dicts,
             )
 
-        # Audio-feedback substitution: vLLM's AR runner calls forward()
-        # with ``inputs_embeds=None`` and lets the model embed its own
-        # input_ids inline. ``embed_input_ids`` is only invoked via the
-        # supports_mm_inputs/prompt_embeds branches, neither of which fires
-        # for the talker. So we apply the substitution directly here: at
-        # decode positions where input_ids == audio_token_id, swap the text
-        # embedding with embed_audio_codes(last audio frame stashed in
-        # ``self._audio_state[req_id]`` by :meth:`sample`).
-        if input_ids is not None and inputs_embeds is None:
+            # Audio-feedback substitution: vLLM's AR runner calls forward()
+            # with ``inputs_embeds=None`` and lets the model embed its own
+            # input_ids inline. ``embed_input_ids`` is only invoked via the
+            # supports_mm_inputs/prompt_embeds branches, neither of which fires
+            # for the talker. So we apply the substitution directly here: at
+            # decode positions where input_ids == audio_token_id, swap the text
+            # embedding with embed_audio_codes(last audio frame stashed in
+            # ``self._audio_state[req_id]`` by :meth:`sample`).
             hidden_states = self._maybe_apply_audio_feedback(hidden_states, input_ids)
 
         audio_mask = self.audio_token_mask(input_ids) if input_ids is not None else None
@@ -1506,7 +1505,7 @@ class HiggsAudioV2TalkerForConditionalGeneration(nn.Module):
             sampled[~valid] = torch.argmax(logits[~valid].float(), dim=-1)
         return sampled
 
-    def _apply_audio_mode_bias(self, logits: torch.Tensor, sampling_metadata: Any) -> int:
+    def _apply_audio_mode_bias(self, logits: torch.Tensor, sampling_metadata: Any) -> None:
         """Mask non-audio tokens at audio-mode positions, in-place on ``logits``.
 
         Heuristic: per-request, find the last token seen so far (last of
@@ -1524,7 +1523,7 @@ class HiggsAudioV2TalkerForConditionalGeneration(nn.Module):
         the stock sampler.
         """
         if logits is None or logits.ndim != 2:
-            return 0
+            return
         audio_bos = int(self.config.audio_bos_token_id)
         audio_id = int(self.config.audio_token_id)
         # (PyTorch align, corrected): upstream NEVER lets the model
@@ -1542,7 +1541,6 @@ class HiggsAudioV2TalkerForConditionalGeneration(nn.Module):
         prompt_ids = getattr(sampling_metadata, "prompt_token_ids", None)
         output_ids = getattr(sampling_metadata, "output_token_ids", None)
         num_rows = int(logits.shape[0])
-        biased = 0
         # Fallback "previous token" source: read each request's LAST input
         # position from ``self._last_step_input_ids`` using ``query_start_loc``
         # from the forward context. The previous version of this code took
@@ -1619,7 +1617,6 @@ class HiggsAudioV2TalkerForConditionalGeneration(nn.Module):
                 mask[audio_eos_vocab] = row[audio_eos_vocab]
                 logits[i].copy_(mask)
                 slot_state["should_terminate"] = False
-                biased += 1
                 continue
             # Always force ``audio_token_id`` — upstream mirror.
             allowed: set[int] = {audio_id, *allowed_extra}
@@ -1629,9 +1626,7 @@ class HiggsAudioV2TalkerForConditionalGeneration(nn.Module):
                 if 0 <= tok < row.shape[-1]:
                     mask[tok] = row[tok]
             logits[i].copy_(mask)
-            biased += 1
         self._last_first_audio_after_bos = first_after_bos
-        return biased
 
     has_postprocess: bool = True
 
diff --git a/vllm_omni/model_executor/models/higgs_audio_v3/ref_audio_cache.py b/vllm_omni/model_executor/models/higgs_audio_v3/ref_audio_cache.py
deleted file mode 100644
index 2a21aafcd..000000000
--- a/vllm_omni/model_executor/models/higgs_audio_v3/ref_audio_cache.py
+++ /dev/null
@@ -1,111 +0,0 @@
-# SPDX-License-Identifier: Apache-2.0
-# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
-"""Bounded LRU cache for Higgs Audio v3 reference-audio codec encoding.
-
-Voice-clone requests that re-send the same ref_audio are otherwise re-encoded
-through the DAC encoder on every call. The cache keys by a stable content
-hash (size + sample rate + head/mid/tail sentinel bytes) so the same audio
-served via different URLs or repeated base64 payloads also hits warm.
-Tensors are cloned on get/put so callers and the cache cannot alias storage.
-"""
-
-from __future__ import annotations
-
-import hashlib
-import threading
-from collections import OrderedDict
-from collections.abc import Callable
-
-import numpy as np
-import torch
-
-__all__ = ["cached_encode_reference_audio", "cache_stats", "clear_cache"]
-
-_DEFAULT_MAX_ENTRIES = 128
-_SENTINEL_BYTES = 4096
-
-_cache: OrderedDict[str, torch.Tensor] = OrderedDict()
-_cache_lock = threading.Lock()
-_hits = 0
-_misses = 0
-
-
-def _content_hash(wav: np.ndarray, sr: int) -> str:
-    """Stable content hash. Head/mid/tail sentinel keeps cost O(1)."""
-    raw = wav.tobytes()
-    n = len(raw)
-    h = hashlib.sha256()
-    h.update(n.to_bytes(8, "little"))
-    h.update(int(sr).to_bytes(4, "little"))
-    if n == 0:
-        return h.hexdigest()
-    if n <= _SENTINEL_BYTES * 3:
-        h.update(raw)
-        return h.hexdigest()
-    h.update(raw[:_SENTINEL_BYTES])
-    mid = (n - _SENTINEL_BYTES) // 2
-    h.update(raw[mid : mid + _SENTINEL_BYTES])
-    h.update(raw[-_SENTINEL_BYTES:])
-    return h.hexdigest()
-
-
-def cached_encode_reference_audio(
-    wav: np.ndarray,
-    sr: int,
-    encode_fn: Callable[[np.ndarray, int], torch.Tensor],
-    max_entries: int = _DEFAULT_MAX_ENTRIES,
-) -> torch.Tensor:
-    """LRU-cached wrapper around ``encode_reference_audio``.
-
-    ``encode_fn`` is the actual codec encoder; passed in so this module
-    stays decoupled from the tokenizer import (avoids circular imports
-    during stage-input-processor discovery).
-
-    Returns a CPU tensor clone — never the cache's interior storage.
-    """
-    global _hits, _misses
-    key = _content_hash(wav, sr)
-    with _cache_lock:
-        cached = _cache.get(key)
-        if cached is not None:
-            _hits += 1
-            _cache.move_to_end(key)
-            return cached.clone()
-
-    # Miss: encode outside the lock so concurrent misses on different
-    # refs don't serialize on this single mutex.
-    codes = encode_fn(wav, sr)
-    codes_cpu = codes.detach().to("cpu").contiguous()
-    with _cache_lock:
-        if key in _cache:
-            # Concurrent miss winner already stored it; refresh LRU position
-            # and return that copy to avoid divergent tensors per request.
-            _cache.move_to_end(key)
-            _misses += 1  # we still paid the encode cost
-            return _cache[key].clone()
-        _cache[key] = codes_cpu
-        if len(_cache) > max_entries:
-            _cache.popitem(last=False)
-        _misses += 1
-    return codes_cpu.clone()
-
-
-def cache_stats() -> dict:
-    """Return (hits, misses, size, hit_rate) — for /metrics-style reporting."""
-    with _cache_lock:
-        total = _hits + _misses
-        return {
-            "hits": _hits,
-            "misses": _misses,
-            "size": len(_cache),
-            "hit_rate": (_hits / total) if total > 0 else 0.0,
-        }
-
-
-def clear_cache() -> None:
-    """Drop all cached entries. Used by tests."""
-    global _hits, _misses
-    with _cache_lock:
-        _cache.clear()
-        _hits = 0
-        _misses = 0
diff --git a/vllm_omni/model_executor/models/ming_flash_omni/projectors.py b/vllm_omni/model_executor/models/ming_flash_omni/projectors.py
index 42e53d1c6..03527ba63 100644
--- a/vllm_omni/model_executor/models/ming_flash_omni/projectors.py
+++ b/vllm_omni/model_executor/models/ming_flash_omni/projectors.py
@@ -108,20 +108,6 @@ class AudioProjector(nn.Module):
             layers.append(nn.Linear(llm_dim, llm_dim))
         self.proj = nn.Sequential(*layers)
 
-    def forward(self, x: torch.Tensor) -> torch.Tensor:
-        """Project audio features with temporal downsampling.
-
-        Args:
-            x: [B, T, audio_dim] audio encoder output (channel-last).
-
-        Returns:
-            [B, T', llm_dim] projected features (channel-last),
-            where T' = (T - ds_kernel_size + 2*(ds_kernel_size//2)) // ds_stride + 1.
-        """
-        # Conv1d expects [B, C, T], so transpose input
-        x = x.transpose(-1, -2)  # [B, audio_dim, T]
-        return self.proj(x)
-
     def forward_packed(
         self,
         packed: torch.Tensor,
@@ -155,19 +141,6 @@ class AudioProjector(nn.Module):
         packed_proj = mlp(packed_proj)
         return packed_proj, proj_lens
 
-    def compute_output_length(self, input_length: torch.Tensor) -> torch.Tensor:
-        """Compute output sequence length after Conv1d downsampling.
-
-        Args:
-            input_length: Original mel spectrogram lengths.
-
-        Returns:
-            Output lengths after both convolutions.
-        """
-        length = (input_length - 3 + 2 * 1) // 2 + 1
-        length = (length - self.ds_kernel_size + 2 * (self.ds_kernel_size // 2)) // self.ds_stride + 1
-        return length
-
     def load_weights(self, weights: Iterable[tuple[str, torch.Tensor]]) -> set[str]:
         params_dict = dict(self.named_parameters())
         loaded_params: set[str] = set()
diff --git a/vllm_omni/model_executor/models/ming_flash_omni/text_processing.py b/vllm_omni/model_executor/models/ming_flash_omni/text_processing.py
index 436b92f14..e0e8e73e6 100644
--- a/vllm_omni/model_executor/models/ming_flash_omni/text_processing.py
+++ b/vllm_omni/model_executor/models/ming_flash_omni/text_processing.py
@@ -15,19 +15,6 @@ from vllm.logger import init_logger
 
 logger = init_logger(__name__)
 
-# Ported from
-# https://github.com/inclusionAI/Ming/blob/e58533db227031990c5a6864dcf5f08fb53ed0d2/front/toolkit.py
-_TOKENIZE_PATTERN = re.compile(r"(?:[a-zA-Z]\.)+|[a-zA-Z]+(?:['\-][a-zA-Z]+)*|\d+(?:\.\d+)?|[\u4e00-\u9fff]|\s+|\S")
-
-
-def tokenize_mixed_text(text: str) -> list[str]:
-    return re.findall(_TOKENIZE_PATTERN, text)
-
-
-def tokenize_mixed_text_iterator(text: str):
-    for match in _TOKENIZE_PATTERN.finditer(text):
-        yield match.group(0)
-
 
 # Ported from
 # https://github.com/inclusionAI/Ming/blob/e58533db227031990c5a6864dcf5f08fb53ed0d2/front/text_segment_cut.py
@@ -218,84 +205,6 @@ def cut_text_by_semantic_length(
     return [f.replace(_DOT_PLACEHOLDER, ".") for f in result_fragments]
 
 
-# Streaming sentence boundary detection
-
-_RE_CJK = re.compile(r"[\u4e00-\u9fff]")
-_RE_DIGIT_LAST = re.compile(r"[0-9]")
-
-
-# Left for reference for now
-# Alternative to `cut_text_by_semantic_length`
-def detect_sentence_boundaries(
-    text: str,
-    max_length: int = 50,
-) -> list[str]:
-    """Accumulate tokens and flush at sentence boundaries.
-
-    Ported from the streaming sentence detection loop from the Ming repo
-    TTS branch, but operates on the full text since we have it available upfront.
-    """
-    sentences: list[str] = []
-    streaming_text: list[str] = []
-    count = 0
-
-    for ele in tokenize_mixed_text_iterator(text):
-        if len(ele) == 0:
-            continue
-
-        should_process = False
-        min_tokens = 12 if count == 0 else 8
-
-        if ele[-1] in "！？。，!?":
-            if len(streaming_text) >= min_tokens:
-                should_process = True
-            streaming_text.append(ele)
-
-        elif ele[-1] == ".":
-            if (
-                len(streaming_text) >= min_tokens
-                and streaming_text
-                and not _RE_DIGIT_LAST.search(streaming_text[-1][-1])
-            ):
-                should_process = True
-            streaming_text.append(ele)
-
-        elif ele[-1] == "\n":
-            if streaming_text:
-                joined = "".join(streaming_text)
-                if _RE_CJK.search(joined):
-                    if _RE_CJK.search(streaming_text[-1][-1]):
-                        ele = "，"
-                        streaming_text.append(ele)
-                else:
-                    if len(ele) > 1 and re.search(r"[a-zA-Z]", ele[-2]):
-                        ele = ele[:-1] + "."
-                    else:
-                        ele = ele[:-1]
-                    streaming_text.append(ele)
-
-            if len(streaming_text) >= min_tokens:
-                should_process = True
-        else:
-            streaming_text.append(ele)
-            continue
-
-        if should_process:
-            joined = "".join(streaming_text)
-            fragments = cut_text_by_semantic_length(joined, max_length)
-            sentences.extend(fragments)
-            streaming_text = []
-            count += 1
-
-    # Flush remaining
-    if streaming_text and re.search(r"[a-zA-Z\u4e00-\u9fff1-9]", "".join(streaming_text)):
-        joined = "".join(streaming_text)
-        fragments = cut_text_by_semantic_length(joined, max_length)
-        sentences.extend(fragments)
-
-    return sentences
-
-
 # number normalization for English. Ported from
 # https://github.com/inclusionAI/Ming/blob/e58533db227031990c5a6864dcf5f08fb53ed0d2/front/number_en.py
 
diff --git a/vllm_omni/model_executor/models/ming_tts/config_ming_tts.py b/vllm_omni/model_executor/models/ming_tts/config_ming_tts.py
index b42c4fdca..8ef8a3a0e 100644
--- a/vllm_omni/model_executor/models/ming_tts/config_ming_tts.py
+++ b/vllm_omni/model_executor/models/ming_tts/config_ming_tts.py
@@ -30,7 +30,6 @@ from .constants import (
     KEY_MAX_DECODE_STEPS,
     KEY_MIN_DECODE_STEPS,
     KEY_NEXT_EMBEDS,
-    KEY_PROMPT_LATENT_TAIL,
     KEY_PROMPT_LATENTS,
     KEY_REQUEST_ID,
     KEY_SIGMA,
@@ -184,7 +183,6 @@ __all__ = [
     "KEY_MAX_DECODE_STEPS",
     "KEY_MIN_DECODE_STEPS",
     "KEY_NEXT_EMBEDS",
-    "KEY_PROMPT_LATENT_TAIL",
     "KEY_PROMPT_LATENTS",
     "KEY_REQUEST_ID",
     "KEY_SIGMA",
diff --git a/vllm_omni/model_executor/models/ming_tts/constants.py b/vllm_omni/model_executor/models/ming_tts/constants.py
index 2f59e2094..4b3f527da 100644
--- a/vllm_omni/model_executor/models/ming_tts/constants.py
+++ b/vllm_omni/model_executor/models/ming_tts/constants.py
@@ -55,7 +55,6 @@ KEY_DECODE_STEP = "ming_decode_step"
 KEY_LAST_STOP_PROB = "ming_last_stop_prob"
 KEY_NEXT_EMBEDS = "ming_next_embeds"
 KEY_PROMPT_LATENTS = "ming_prompt_latents"
-KEY_PROMPT_LATENT_TAIL = "ming_prompt_latent_tail"
 KEY_SPEAKER_EMBEDDING = "ming_speaker_embedding"
 KEY_REQUEST_ID = "ming_request_id"
 KEY_CHUNK_ID = "ming_chunk_id"
diff --git a/vllm_omni/model_executor/models/ming_tts/ming_tts.py b/vllm_omni/model_executor/models/ming_tts/ming_tts.py
index 667aded27..4140b403b 100644
--- a/vllm_omni/model_executor/models/ming_tts/ming_tts.py
+++ b/vllm_omni/model_executor/models/ming_tts/ming_tts.py
@@ -16,7 +16,6 @@ from vllm.v1.sample.sampler import Sampler
 from vllm_omni.model_executor.custom_process_mixin import CustomProcessMixin
 
 from .audio_prep import (
-    _coerce_prompt_latents,
     _find_audio_placeholder_positions,
     _find_speaker_placeholder_positions,
     _initial_history,
@@ -32,7 +31,6 @@ from .config_ming_tts import (
     KEY_MAX_DECODE_STEPS,
     KEY_MIN_DECODE_STEPS,
     KEY_NEXT_EMBEDS,
-    KEY_PROMPT_LATENT_TAIL,
     KEY_REQUEST_ID,
     KEY_SIGMA,
     KEY_SPEAKER_EMBEDDING,
@@ -175,7 +173,6 @@ class MingTTSForConditionalGeneration(nn.Module, SupportsPP, CustomProcessMixin)
             dtype=torch.float32,
         )
         update[KEY_LATENT_HISTORY] = history.detach().to("cpu").contiguous()
-        update[KEY_PROMPT_LATENT_TAIL] = update[KEY_LATENT_HISTORY]
 
         speaker_embedding = info_dict.get(KEY_SPEAKER_EMBEDDING, info_dict.get("speaker_embedding"))
         speaker_embeddings = None
@@ -262,7 +259,6 @@ def _copy_runtime_controls(update: dict[str, Any], info_dict: dict[str, Any]) ->
 __all__ = [
     "MingTTSForConditionalGeneration",
     "_ModelSampleAdapter",
-    "_coerce_prompt_latents",
     "_find_audio_placeholder_positions",
     "_find_speaker_placeholder_positions",
     "_initial_history",
diff --git a/vllm_omni/model_executor/models/ming_tts/ming_tts_llm.py b/vllm_omni/model_executor/models/ming_tts/ming_tts_llm.py
index 92daad967..e7e59631f 100644
--- a/vllm_omni/model_executor/models/ming_tts/ming_tts_llm.py
+++ b/vllm_omni/model_executor/models/ming_tts/ming_tts_llm.py
@@ -3,7 +3,6 @@
 # Adopted from https://github.com/inclusionAI/Ming-omni-tts/blob/main/modeling_bailingmm.py
 from __future__ import annotations
 
-import warnings
 from collections.abc import Iterable
 from typing import Any
 
@@ -346,10 +345,10 @@ class MingLLMModel(nn.Module):
     def load_weights(self, weights: Iterable[tuple[str, torch.Tensor]]) -> set[str]:
         params_dict = dict(self.named_parameters(remove_duplicate=False))
         loaded_params = AutoWeightsLoader(self).load_weights(weights, mapper=self.hf_to_vllm_mapper)
-        _warn_missing_prefix("flowloss", params_dict, loaded_params, prefix="flowloss.", fatal=True)
-        _warn_missing_prefix("linear_proj_audio", params_dict, loaded_params, prefix="linear_proj_audio.", fatal=True)
-        _warn_missing_prefix("stop_head", params_dict, loaded_params, prefix="stop_head.", fatal=True)
-        _warn_missing_prefix("spk_head", params_dict, loaded_params, prefix="spk_head.", fatal=True)
+        _warn_missing_prefix("flowloss", params_dict, loaded_params, prefix="flowloss.")
+        _warn_missing_prefix("linear_proj_audio", params_dict, loaded_params, prefix="linear_proj_audio.")
+        _warn_missing_prefix("stop_head", params_dict, loaded_params, prefix="stop_head.")
+        _warn_missing_prefix("spk_head", params_dict, loaded_params, prefix="spk_head.")
         return loaded_params
 
 
@@ -383,7 +382,6 @@ def _warn_missing_prefix(
     params_dict: dict[str, nn.Parameter],
     loaded_params: set[str],
     prefix: str,
-    fatal: bool = False,
 ) -> None:
     missing = {key for key in params_dict if key.startswith(prefix)} - loaded_params
     if not missing:
@@ -392,9 +390,7 @@ def _warn_missing_prefix(
         f"MingLLMModel: {len(missing)} {module_name} params not loaded "
         f"(prefix={prefix}). First few: {sorted(missing)[:5]}"
     )
-    if fatal:
-        raise RuntimeError(msg)
-    warnings.warn(msg, stacklevel=3)
+    raise RuntimeError(msg)
 
 
 __all__ = [
diff --git a/vllm_omni/model_executor/models/moss_tts/audio_tokenizer.py b/vllm_omni/model_executor/models/moss_tts/audio_tokenizer.py
index a2cf75b07..89c52485a 100644
--- a/vllm_omni/model_executor/models/moss_tts/audio_tokenizer.py
+++ b/vllm_omni/model_executor/models/moss_tts/audio_tokenizer.py
@@ -78,7 +78,7 @@ class MossAudioTokenizerConfig(PretrainedConfig):
 
     @property
     def codebook_size(self) -> int:
-        return int(self.quantizer_kwargs.get("codebook_size", 4096))
+        return int(self.quantizer_kwargs.get("codebook_size", 1024))
 
     @property
     def frame_rate(self) -> float:
diff --git a/vllm_omni/model_executor/models/moss_tts/modeling_moss_tts_talker.py b/vllm_omni/model_executor/models/moss_tts/modeling_moss_tts_talker.py
index d40e22014..d32af7c90 100644
--- a/vllm_omni/model_executor/models/moss_tts/modeling_moss_tts_talker.py
+++ b/vllm_omni/model_executor/models/moss_tts/modeling_moss_tts_talker.py
@@ -31,8 +31,6 @@ from vllm_omni.model_executor.models.output_templates import OmniOutput
 
 logger = init_logger(__name__)
 
-_AUDIO_PAD_ID = 0  # padding id used before delay slot fires
-
 
 def _maybe_prefix(prefix: str, name: str) -> str:
     return f"{prefix}.{name}" if prefix else name
@@ -1058,7 +1056,7 @@ class MossTTSRealtimeTalkerForGeneration(nn.Module):
                     history_per_codebook=hist_per_cb,
                 ).squeeze(0)  # (n_vq,)
                 if int(state.get("step", 0)) < 5 or int(state.get("step", 0)) % 50 == 0:
-                    logger.info(
+                    logger.debug(
                         "[MossTTSRealtime make_omni] step=%d ch0=%d cursor=%d/%d",
                         int(state.get("step", 0)),
                         int(new_codes[0].item()),
diff --git a/vllm_omni/model_executor/models/moss_tts_nano/configuration_moss_tts_nano.py b/vllm_omni/model_executor/models/moss_tts_nano/configuration_moss_tts_nano.py
deleted file mode 100644
index a5a9f25ac..000000000
--- a/vllm_omni/model_executor/models/moss_tts_nano/configuration_moss_tts_nano.py
+++ /dev/null
@@ -1,60 +0,0 @@
-# SPDX-License-Identifier: Apache-2.0
-# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
-"""Configuration for MOSS-TTS-Nano in vLLM-Omni single-stage pipeline."""
-
-from transformers.configuration_utils import PretrainedConfig
-
-
-class MossTTSNanoConfig(PretrainedConfig):
-    """Config for MOSS-TTS-Nano (OpenMOSS-Team/MOSS-TTS-Nano).
-
-    The model is a Global-Local GPT2-based AR LM (0.1B) paired with the
-    MOSS-Audio-Tokenizer-Nano codec.  Both components run in a single
-    vLLM-Omni generation stage.
-
-    Relevant fields from config.json:
-      gpt2_config  – backbone config (n_layer=12, n_embd=768, n_head=12, ...)
-      n_vq         – number of RVQ codebooks (16)
-      audio_vocab_size – per-codebook vocabulary size (1024)
-      audio_tokenizer_pretrained_name_or_path – HF hub path for codec model
-    """
-
-    model_type = "moss_tts_nano"
-
-    def __init__(self, **kwargs):
-        gpt2_cfg = kwargs.pop("gpt2_config", None) or {}
-        if hasattr(gpt2_cfg, "to_dict"):
-            gpt2_cfg = gpt2_cfg.to_dict()
-
-        super().__init__(**kwargs)
-
-        # --- GPT2 backbone parameters (exposed at top level for vLLM) ---
-        self.hidden_size = gpt2_cfg.get("n_embd", 768)
-        self.num_hidden_layers = gpt2_cfg.get("n_layer", 12)
-        self.num_attention_heads = gpt2_cfg.get("n_head", 12)
-        self.num_key_value_heads = self.num_attention_heads  # no GQA in GPT2
-        self.head_dim = self.hidden_size // self.num_attention_heads
-        self.vocab_size = gpt2_cfg.get("vocab_size", 16384)
-        self.max_position_embeddings = gpt2_cfg.get("n_positions", 32768)
-        self.intermediate_size = gpt2_cfg.get("n_inner", self.hidden_size * 4)
-
-        # --- Audio codec parameters ---
-        self.n_vq: int = getattr(self, "n_vq", 16)
-        self.audio_vocab_size: int = getattr(self, "audio_vocab_size", 1024)
-        self.audio_start_token_id: int = getattr(self, "audio_start_token_id", 6)
-        self.audio_end_token_id: int = getattr(self, "audio_end_token_id", 7)
-        self.audio_user_slot_token_id: int = getattr(self, "audio_user_slot_token_id", 8)
-        self.audio_assistant_slot_token_id: int = getattr(self, "audio_assistant_slot_token_id", 9)
-        self.audio_tokenizer_pretrained_name_or_path: str = getattr(
-            self,
-            "audio_tokenizer_pretrained_name_or_path",
-            "OpenMOSS-Team/MOSS-Audio-Tokenizer-Nano",
-        )
-        self.audio_tokenizer_sample_rate: int = getattr(self, "audio_tokenizer_sample_rate", 48000)
-
-        # vLLM requires speculative_config to be absent or None
-        self.speculative_config = None
-
-    def get_text_config(self, **kwargs):
-        """Return self so vLLM uses our top-level config."""
-        return self
diff --git a/vllm_omni/model_executor/models/moss_tts_nano/modeling_moss_tts_nano.py b/vllm_omni/model_executor/models/moss_tts_nano/modeling_moss_tts_nano.py
index f52577f67..705960982 100644
--- a/vllm_omni/model_executor/models/moss_tts_nano/modeling_moss_tts_nano.py
+++ b/vllm_omni/model_executor/models/moss_tts_nano/modeling_moss_tts_nano.py
@@ -172,7 +172,6 @@ class MossTTSNanoForGeneration(nn.Module):
     has_preprocess = False
     has_postprocess = False
     enable_update_additional_information = True
-    inject_omni_request_id_into_runtime_info = True
 
     def __init__(self, *, vllm_config: VllmConfig, prefix: str = "") -> None:
         super().__init__()
diff --git a/vllm_omni/model_executor/models/omnivoice/omnivoice.py b/vllm_omni/model_executor/models/omnivoice/omnivoice.py
index c5026086e..43c496a1c 100644
--- a/vllm_omni/model_executor/models/omnivoice/omnivoice.py
+++ b/vllm_omni/model_executor/models/omnivoice/omnivoice.py
@@ -379,13 +379,12 @@ class OmniVoiceModel(
         target_ids = torch.full((num_codebooks, target_len), mask_id, dtype=torch.long, device=device)
 
         # Conditional: [text] [ref_audio?] [target_mask]
+        cond_audio_start = text_ids.shape[1]
         if ref_audio_tokens is not None:
             ref_tokens = ref_audio_tokens.to(device)  # [8, T_ref]
             cond_ids = torch.cat([text_ids, ref_tokens, target_ids], dim=1)
-            cond_audio_start = text_ids.shape[1]
         else:
             cond_ids = torch.cat([text_ids, target_ids], dim=1)
-            cond_audio_start = text_ids.shape[1]
 
         cond_len = cond_ids.shape[1]
 
diff --git a/vllm_omni/model_executor/models/omnivoice/omnivoice_generator.py b/vllm_omni/model_executor/models/omnivoice/omnivoice_generator.py
index fc5befeb9..7eacfb9bb 100644
--- a/vllm_omni/model_executor/models/omnivoice/omnivoice_generator.py
+++ b/vllm_omni/model_executor/models/omnivoice/omnivoice_generator.py
@@ -6,8 +6,9 @@ OmniVoice Generator (Stage 0) - Iterative unmasking with Qwen3 backbone.
 Generates 8-codebook audio tokens from text via 32-step non-autoregressive
 iterative masked prediction with classifier-free guidance.
 
-Uses vLLM-Omni's DiffusionAttention for optimized full (bidirectional) attention
-via FlashAttention/SageAttention/SDPA backends.
+Uses full bidirectional attention computed directly with PyTorch SDPA
+(torch.nn.functional.scaled_dot_product_attention); no auto-selected
+FlashAttention/SageAttention/DiffusionAttention backend is used.
 """
 
 from __future__ import annotations
@@ -237,7 +238,7 @@ def _gumbel_sample(logits: torch.Tensor, temperature: float, generator: torch.Ge
 
 
 # ---------------------------------------------------------------------------
-# Qwen3-style transformer blocks using DiffusionAttention
+# Qwen3-style transformer blocks using PyTorch SDPA
 # ---------------------------------------------------------------------------
 
 
@@ -256,7 +257,7 @@ class OmniVoiceRMSNorm(nn.Module):
 
 
 class OmniVoiceAttention(nn.Module):
-    """Qwen3-style GQA attention using DiffusionAttention backend."""
+    """Qwen3-style GQA attention using PyTorch SDPA (full bidirectional)."""
 
     def __init__(self, config: OmniVoiceConfig):
         super().__init__()
@@ -315,18 +316,17 @@ class OmniVoiceAttention(nn.Module):
         v = v.permute(0, 2, 1, 3)
 
         # Convert [B, 1, S, S] bool mask to float mask for SDPA
+        # (0.0 where attend, -inf where masked)
         sdpa_mask = None
         if attention_mask is not None:
-            sdpa_mask = attention_mask.to(dtype=q.dtype)
-            sdpa_mask = sdpa_mask.masked_fill(~attention_mask, float("-inf"))
-            sdpa_mask = sdpa_mask.masked_fill(attention_mask, 0.0)
+            sdpa_mask = torch.zeros_like(attention_mask, dtype=q.dtype).masked_fill_(~attention_mask, float("-inf"))
 
         out = F.scaled_dot_product_attention(
             q,
             k,
             v,
             attn_mask=sdpa_mask,
-            scale=1.0 / math.sqrt(self.head_dim),
+            scale=self.scale,
         )
 
         # Back to (batch, seq, heads * head_dim)
@@ -351,7 +351,7 @@ class OmniVoiceMLP(nn.Module):
 
 
 class OmniVoiceTransformerBlock(nn.Module):
-    """Single Qwen3 transformer block with DiffusionAttention."""
+    """Single Qwen3 transformer block with PyTorch SDPA attention."""
 
     def __init__(self, config: OmniVoiceConfig):
         super().__init__()
@@ -643,10 +643,9 @@ class OmniVoiceGenerator(nn.Module):
     - 32-step iterative unmasking with classifier-free guidance
 
     Optimizations:
-    - DiffusionAttention (FlashAttn/SageAttn/SDPA auto-selected)
-    - TeaCache / Cache-DiT compatible (hook-based, non-intrusive)
+    - Full bidirectional attention via PyTorch SDPA (no auto-selected
+      FlashAttn/SageAttn/DiffusionAttention backend)
     - regionally_compile() compatible for torch.compile on repeated blocks
-    - Sequence parallelism via SP hooks for multi-GPU
     """
 
     # For regionally_compile() support
diff --git a/vllm_omni/model_executor/models/qwen3_tts/configuration_qwen3_tts.py b/vllm_omni/model_executor/models/qwen3_tts/configuration_qwen3_tts.py
index 8bb921a1e..28ca4f362 100644
--- a/vllm_omni/model_executor/models/qwen3_tts/configuration_qwen3_tts.py
+++ b/vllm_omni/model_executor/models/qwen3_tts/configuration_qwen3_tts.py
@@ -27,7 +27,7 @@ class Qwen3TTSSpeakerEncoderConfig(PretrainedConfig):
     Args:
         mel_dim (`int`, *optional*, defaults to 128):
             The dimension of the input mel-spectrogram.
-        enc_dim (`int`, *optional*, defaults to 192):
+        enc_dim (`int`, *optional*, defaults to 1024):
             The dimension of the final speaker embedding.
         enc_channels (`list[int]`, *optional*, defaults to `[512, 512, 512, 512, 1536]`):
             A list of output channels for each TDNN/SERes2Net layer in the encoder.
@@ -80,11 +80,11 @@ class Qwen3TTSTalkerCodePredictorConfig(PretrainedConfig):
 
 
     Args:
-        vocab_size (`int`, *optional*, defaults to 151936):
+        vocab_size (`int`, *optional*, defaults to 2048):
             Vocabulary size of the Qwen3TTSTalkerCodePredictor model.
             Defines the number of different tokens that can be represented by the
             `inputs_ids` passed when calling [`Qwen3TTSTalkerCodePredictorModel`]
-        hidden_size (`int`, *optional*, defaults to 4096):
+        hidden_size (`int`, *optional*, defaults to 1024):
             Dimension of the hidden representations.
         intermediate_size (`int`, *optional*, defaults to 22016):
             Dimension of the MLP representations.
@@ -271,11 +271,11 @@ class Qwen3TTSTalkerConfig(PretrainedConfig):
 
 
     Args:
-        vocab_size (`int`, *optional*, defaults to 151936):
+        vocab_size (`int`, *optional*, defaults to 3072):
             Vocabulary size of the Qwen3TTSTalker model.
             Defines the number of different tokens that can be represented by the
             `inputs_ids` passed when calling [`Qwen3TTSTalkerModel`]
-        hidden_size (`int`, *optional*, defaults to 2048):
+        hidden_size (`int`, *optional*, defaults to 1024):
             Dimension of the hidden representations.
         intermediate_size (`int`, *optional*, defaults to 6144):
             Dimension of the MLP representations.
@@ -436,7 +436,6 @@ class Qwen3TTSTalkerConfig(PretrainedConfig):
             self.rope_scaling["rope_type"] = self.rope_scaling["type"]
 
         if code_predictor_config is None:
-            code_predictor_config = {}
             self.code_predictor_config = Qwen3TTSTalkerCodePredictorConfig()
             logger.info("code_predictor_config is None. Initializing code_predictor model with default values")
         elif isinstance(code_predictor_config, Qwen3TTSTalkerCodePredictorConfig):
diff --git a/vllm_omni/model_executor/models/qwen3_tts/prompt_embeds_builder.py b/vllm_omni/model_executor/models/qwen3_tts/prompt_embeds_builder.py
index b0294b725..0d8f03294 100644
--- a/vllm_omni/model_executor/models/qwen3_tts/prompt_embeds_builder.py
+++ b/vllm_omni/model_executor/models/qwen3_tts/prompt_embeds_builder.py
@@ -1320,9 +1320,7 @@ class Qwen3TTSPromptEmbedsBuilder:
             # Keep it at least 1D; embedding on a 0-d tensor can return 1D.
             spk_tensor = torch.tensor([spk_id], device=input_ids.device, dtype=torch.long)
             spk_embed = codec_embed(spk_tensor)
-            if spk_embed.ndim == 1:
-                spk_embed = spk_embed.view(1, 1, -1)
-            elif spk_embed.ndim == 2:
+            if spk_embed.ndim in (1, 2):
                 spk_embed = spk_embed.view(1, 1, -1)
             speaker_embed = spk_embed
             codec_input = torch.cat([codec_input_0, speaker_embed, codec_input_1], dim=1)
@@ -1523,9 +1521,7 @@ class Qwen3TTSPromptEmbedsBuilder:
 
                 ref_code_len: int | None = None
                 if isinstance(ref_code, list):
-                    if ref_code and isinstance(ref_code[0], list):
-                        ref_code_len = len(ref_code)
-                    elif ref_code:
+                    if ref_code:
                         ref_code_len = len(ref_code)
                 elif hasattr(ref_code, "shape"):
                     try:
diff --git a/vllm_omni/model_executor/models/qwen3_tts/qwen3_tts_talker.py b/vllm_omni/model_executor/models/qwen3_tts/qwen3_tts_talker.py
index 64eb3d253..c1bb7c049 100644
--- a/vllm_omni/model_executor/models/qwen3_tts/qwen3_tts_talker.py
+++ b/vllm_omni/model_executor/models/qwen3_tts/qwen3_tts_talker.py
@@ -286,8 +286,6 @@ class Qwen3TTSTalkerForConditionalGeneration(nn.Module):
             )
         self._codec_eos_token_id = int(getattr(self.talker_config, "codec_eos_token_id", -1))
 
-        self._eos_logit_bias: float = 0.0
-
         self.have_multimodal_outputs = True
         self.has_preprocess = True
         self.has_postprocess = True
@@ -517,11 +515,6 @@ class Qwen3TTSTalkerForConditionalGeneration(nn.Module):
         # Mask out invalid codec ids using the pre-built constant buffer.
         logits = logits.masked_fill(self._codec_disallowed_mask, float("-inf"))
 
-        if self._eos_logit_bias != 0.0:
-            eos_id = self._codec_eos_token_id
-            if 0 <= eos_id < logits.shape[-1]:
-                logits[:, eos_id] = logits[:, eos_id] + self._eos_logit_bias
-
         return logits
 
     # -------------------- Omni multimodal output plumbing --------------------
diff --git a/vllm_omni/model_executor/models/qwen3_tts/tokenizer_12hz/modeling_qwen3_tts_tokenizer_v2.py b/vllm_omni/model_executor/models/qwen3_tts/tokenizer_12hz/modeling_qwen3_tts_tokenizer_v2.py
index 6eb78eeae..447823abb 100644
--- a/vllm_omni/model_executor/models/qwen3_tts/tokenizer_12hz/modeling_qwen3_tts_tokenizer_v2.py
+++ b/vllm_omni/model_executor/models/qwen3_tts/tokenizer_12hz/modeling_qwen3_tts_tokenizer_v2.py
@@ -912,11 +912,6 @@ class Qwen3TTSTokenizerV2Decoder(Qwen3TTSTokenizerV2DecoderPreTrainedModel):
             self._cudagraph_wrapper.compile_shapes,
         )
 
-    def disable_cudagraph(self):
-        self._cudagraph_enabled = False
-        self._cudagraph_wrapper = None
-        logger.info("CUDA Graph disabled for decoder")
-
     def forward(self, codes):
         if codes.shape[1] != self.config.num_quantizers:
             raise ValueError(f"Expected {self.config.num_quantizers} layer of codes, got {codes.shape[1]}")
diff --git a/vllm_omni/model_executor/models/qwen3_tts/tokenizer_25hz/vq/core_vq.py b/vllm_omni/model_executor/models/qwen3_tts/tokenizer_25hz/vq/core_vq.py
index 5609abb39..401401453 100644
--- a/vllm_omni/model_executor/models/qwen3_tts/tokenizer_25hz/vq/core_vq.py
+++ b/vllm_omni/model_executor/models/qwen3_tts/tokenizer_25hz/vq/core_vq.py
@@ -33,7 +33,6 @@
 
 import random
 import typing as tp
-from math import ceil
 
 import numpy as np
 import torch
@@ -45,10 +44,6 @@ from vllm.logger import init_logger
 logger = init_logger(__name__)
 
 
-def round_up_multiple(num, mult):
-    return ceil(num / mult) * mult
-
-
 def default(val: tp.Any, d: tp.Any) -> tp.Any:
     return val if val is not None else d
 
diff --git a/vllm_omni/model_executor/models/step_audio2/step_audio2.py b/vllm_omni/model_executor/models/step_audio2/step_audio2.py
index 82e5609ff..0a94ab9de 100644
--- a/vllm_omni/model_executor/models/step_audio2/step_audio2.py
+++ b/vllm_omni/model_executor/models/step_audio2/step_audio2.py
@@ -26,17 +26,17 @@ class StepAudio2ForConditionalGeneration(nn.Module, SupportsMultiModal, Supports
     Step-Audio2 Main Controller
 
     Manages two-stage inference pipeline:
-    - Stage 1 (Thinker): Audio understanding and token generation
-    - Stage 2 (Token2Wav): Audio token to waveform synthesis
+    - Stage 0 (Thinker): Audio understanding and token generation
+    - Stage 1 (Token2Wav): Audio token to waveform synthesis
 
     Usage:
-        # Stage 1: Thinker
+        # Stage 0: Thinker
         model = StepAudio2ForConditionalGeneration(
             vllm_config=config,
             model_stage="thinker"
         )
 
-        # Stage 2: Token2Wav
+        # Stage 1: Token2Wav
         model = StepAudio2ForConditionalGeneration(
             vllm_config=config,
             model_stage="token2wav"
diff --git a/vllm_omni/model_executor/models/step_audio2/step_audio2_constants.py b/vllm_omni/model_executor/models/step_audio2/step_audio2_constants.py
index 802cd63be..2f2918a1b 100644
--- a/vllm_omni/model_executor/models/step_audio2/step_audio2_constants.py
+++ b/vllm_omni/model_executor/models/step_audio2/step_audio2_constants.py
@@ -42,39 +42,6 @@ class StepAudio2TokenConfig:
         return self.audio_start + self.audio_vocab_size - 1
 
 
-@dataclass(frozen=True)
-class StepAudio2EncoderConfig:
-    """
-    Step-Audio2 audio encoder architecture configuration.
-
-    These values define the encoder structure and should match your model weights.
-    """
-
-    # Mel spectrogram parameters
-    n_mels: int = 128
-    """Number of mel frequency bins"""
-
-    # Encoder architecture
-    n_audio_ctx: int = 1500
-    """Audio context length (max sequence length)"""
-
-    n_audio_state: int = 512
-    """Audio encoder hidden state dimension"""
-
-    n_audio_head: int = 8
-    """Number of attention heads in encoder"""
-
-    n_audio_layer: int = 6
-    """Number of encoder layers"""
-
-    # Adapter parameters
-    kernel_size: int = 3
-    """Convolution kernel size for adapter"""
-
-    adapter_stride: int = 2
-    """Stride for adapter convolution (downsampling factor)"""
-
-
 @dataclass(frozen=True)
 class StepAudio2Token2WavConfig:
     """Token2Wav synthesis configuration."""
@@ -141,7 +108,6 @@ class StepAudio2ModelConfig:
 
 # Default configuration instances
 DEFAULT_TOKEN_CONFIG = StepAudio2TokenConfig()
-DEFAULT_ENCODER_CONFIG = StepAudio2EncoderConfig()
 DEFAULT_TOKEN2WAV_CONFIG = StepAudio2Token2WavConfig()
 DEFAULT_STREAM_CONFIG = StepAudio2StreamConfig()
 DEFAULT_MODEL_CONFIG = StepAudio2ModelConfig()
@@ -162,12 +128,10 @@ STEP_AUDIO2_DEFAULT_PROMPT_WAV = "default_female.wav"
 
 __all__ = [
     "StepAudio2TokenConfig",
-    "StepAudio2EncoderConfig",
     "StepAudio2Token2WavConfig",
     "StepAudio2StreamConfig",
     "StepAudio2ModelConfig",
     "DEFAULT_TOKEN_CONFIG",
-    "DEFAULT_ENCODER_CONFIG",
     "DEFAULT_TOKEN2WAV_CONFIG",
     "DEFAULT_STREAM_CONFIG",
     "DEFAULT_MODEL_CONFIG",
diff --git a/vllm_omni/model_executor/models/step_audio2/step_audio2_thinker.py b/vllm_omni/model_executor/models/step_audio2/step_audio2_thinker.py
index 7fd0bf4fb..c92f23978 100644
--- a/vllm_omni/model_executor/models/step_audio2/step_audio2_thinker.py
+++ b/vllm_omni/model_executor/models/step_audio2/step_audio2_thinker.py
@@ -609,7 +609,7 @@ class StepAudio2ThinkerForConditionalGeneration(nn.Module, SupportsMultiModal, S
     Architecture:
         AudioEncoder (6 layers, 512 hidden) →
         Adaptor (512 → LLM dim) →
-        Qwen2 LLM (vocab=64012)
+        Qwen2 LLM (vocab=158720)
     """
 
     def __init__(self, *, vllm_config: VllmConfig, prefix: str = ""):
@@ -868,31 +868,3 @@ class StepAudio2ThinkerForConditionalGeneration(nn.Module, SupportsMultiModal, S
     def has_audio_output(token_ids: list[int], audio_start: int = DEFAULT_TOKEN_CONFIG.audio_start) -> bool:
         """Check if generated tokens contain audio tokens"""
         return any(tid >= audio_start for tid in token_ids)
-
-
-class StepAudio2OutputProcessor:
-    """Helper class to process Step-Audio2 outputs"""
-
-    @staticmethod
-    def process_output(output_ids: torch.Tensor, tokenizer, remove_audio_padding: bool = True) -> dict:
-        """Process model output and separate text/audio tokens"""
-        if isinstance(output_ids, torch.Tensor):
-            output_ids = output_ids.squeeze().tolist()
-
-        # Separate tokens
-        text_tokens, audio_tokens = StepAudio2ThinkerForConditionalGeneration.separate_tokens(output_ids)
-
-        # Remove audio padding if requested
-        if remove_audio_padding and audio_tokens:
-            audio_tokens = [t for t in audio_tokens if t < DEFAULT_TOKEN_CONFIG.audio_eos]
-
-        # Decode text
-        text = tokenizer.decode(text_tokens, skip_special_tokens=False)
-
-        return {
-            "text": text,
-            "text_tokens": text_tokens,
-            "audio_tokens": audio_tokens,
-            "has_audio": len(audio_tokens) > 0,
-            "all_tokens": output_ids,
-        }
diff --git a/vllm_omni/model_executor/models/step_audio2/step_audio2_token2wav.py b/vllm_omni/model_executor/models/step_audio2/step_audio2_token2wav.py
index a59ba3fef..86109ad9c 100644
--- a/vllm_omni/model_executor/models/step_audio2/step_audio2_token2wav.py
+++ b/vllm_omni/model_executor/models/step_audio2/step_audio2_token2wav.py
@@ -94,8 +94,6 @@ class StepAudio2Token2WavCore(nn.Module):
         self.mel_cache_len = DEFAULT_STREAM_CONFIG.mel_cache_len
         self.source_cache_len = STREAM_SOURCE_CACHE_LEN
         self.speech_window: torch.Tensor | None = None  # created lazily on device
-        self.stream_cache: dict | None = None  # flow model causal cache
-        self.hift_cache_dict: dict = {}  # HiFT vocoder overlap cache
 
     def _ensure_models_loaded(self):
         """Lazy load models on first use"""
@@ -253,125 +251,6 @@ class StepAudio2Token2WavCore(nn.Module):
         else:
             return wav
 
-    # ------------------------------------------------------------------
-    # Streaming (chunked) inference
-    # ------------------------------------------------------------------
-
-    def setup_stream(self, prompt_wav: str) -> None:
-        """Initialize flow + HiFT caches for chunked streaming inference.
-
-        Must be called once before the first ``stream_chunk()`` call for a
-        given utterance.
-        """
-        if prompt_wav not in self.cache:
-            self.cache[prompt_wav] = self._prepare_prompt(prompt_wav)
-        (
-            prompt_speech_tokens,
-            _,
-            spk_emb,
-            prompt_mels,
-            _,
-        ) = self.cache[prompt_wav]
-
-        # Lazily create the Hamming window on the correct device
-        if self.speech_window is None or self.speech_window.device != self.device:
-            self.speech_window = torch.from_numpy(np.hamming(2 * self.source_cache_len)).to(
-                device=self.device, dtype=torch.float32
-            )
-
-        # Flow model: setup causal cache with prompt tokens + lookahead
-        pre_lookahead = DEFAULT_STREAM_CONFIG.pre_lookahead_len
-        self.stream_cache = self.flow.setup_cache(
-            torch.cat([prompt_speech_tokens, prompt_speech_tokens[:, :pre_lookahead]], dim=1),
-            prompt_mels,
-            spk_emb,
-            n_timesteps=self.n_timesteps,
-        )
-
-        # HiFT vocoder: empty overlap buffers
-        self.hift_cache_dict = {
-            "mel": torch.zeros(1, prompt_mels.shape[2], 0, device=self.device),
-            "source": torch.zeros(1, 1, 0, device=self.device),
-            "speech": torch.zeros(1, 0, device=self.device),
-        }
-
-    def stream_chunk(
-        self,
-        audio_tokens: list[int],
-        prompt_wav: str,
-        last_chunk: bool = False,
-    ) -> torch.Tensor:
-        """Process one chunk of audio tokens and return a waveform segment.
-
-        Args:
-            audio_tokens: Token IDs for this chunk (includes lookahead for
-                non-last chunks).
-            prompt_wav: Path to the prompt wav (for cached speaker embeddings).
-            last_chunk: Whether this is the final chunk of the utterance.
-
-        Returns:
-            1-D float tensor of audio samples (24 kHz).
-        """
-        if self.stream_cache is None:
-            raise ValueError("stream_cache not initialised – call setup_stream() first")
-
-        if prompt_wav not in self.cache:
-            self.cache[prompt_wav] = self._prepare_prompt(prompt_wav)
-        _, _, spk_emb, prompt_mels, _ = self.cache[prompt_wav]
-
-        token_tensor = torch.tensor([audio_tokens], dtype=torch.int32, device=self.device)
-
-        with torch.amp.autocast(
-            str(self.device.type),
-            dtype=torch.float16 if self.float16 else torch.float32,
-        ):
-            chunk_mel, self.stream_cache = self.flow.inference_chunk(
-                token=token_tensor,
-                spk=spk_emb,
-                cache=self.stream_cache,
-                last_chunk=last_chunk,
-                n_timesteps=self.n_timesteps,
-            )
-
-        # Trim estimator attention cache to avoid unbounded growth
-        keep = DEFAULT_STREAM_CONFIG.estimator_cache_keep
-        est_att = self.stream_cache.get("estimator_att_cache")
-        if est_att is not None and est_att.shape[4] > (prompt_mels.shape[1] + keep):
-            self.stream_cache["estimator_att_cache"] = torch.cat(
-                [est_att[:, :, :, :, : prompt_mels.shape[1]], est_att[:, :, :, :, -keep:]],
-                dim=4,
-            )
-
-        # ---- HiFT vocoder with overlap-add ----
-        hift_cache_mel = self.hift_cache_dict["mel"]
-        hift_cache_source = self.hift_cache_dict["source"]
-        hift_cache_speech = self.hift_cache_dict["speech"]
-
-        mel = torch.cat([hift_cache_mel, chunk_mel], dim=2)
-        speech, source = self.hift(mel, hift_cache_source)
-
-        # Cross-fade overlap region for smooth concatenation
-        if hift_cache_speech.shape[-1] > 0:
-            speech = fade_in_out(speech, hift_cache_speech, self.speech_window)
-
-        # Update vocoder cache for the next chunk
-        self.hift_cache_dict = {
-            "mel": mel[..., -self.mel_cache_len :].clone().detach(),
-            "source": source[:, :, -self.source_cache_len :].clone().detach(),
-            "speech": speech[:, -self.source_cache_len :].clone().detach(),
-        }
-
-        # For non-last chunks, trim the tail that will be re-synthesised
-        if not last_chunk:
-            speech = speech[:, : -self.source_cache_len]
-
-        return speech.squeeze(0)  # [samples]
-
-    def reset_stream(self) -> None:
-        """Clear all streaming state after an utterance is complete."""
-        self.stream_cache = None
-        self.hift_cache_dict = {}
-
     # ------------------------------------------------------------------
     # Per-request streaming (does NOT mutate self – uses external state)
     # ------------------------------------------------------------------
diff --git a/vllm_omni/model_executor/models/voxcpm2/minicpm4_paged.py b/vllm_omni/model_executor/models/voxcpm2/minicpm4_paged.py
index b87ec5aaf..13f1d0937 100644
--- a/vllm_omni/model_executor/models/voxcpm2/minicpm4_paged.py
+++ b/vllm_omni/model_executor/models/voxcpm2/minicpm4_paged.py
@@ -65,7 +65,6 @@ class _PagedMiniCPM4Attention(nn.Module):
         prefix: str = "",
     ) -> None:
         super().__init__()
-        self.layer_idx = layer_idx
         self.hidden_size = hidden_size
         self.num_heads = num_attention_heads
         self.head_dim = kv_channels if kv_channels else hidden_size // num_attention_heads
@@ -318,18 +317,12 @@ class MiniCPM4PagedForVoxCPM2(nn.Module):
                 ).detach()
 
     def compile_selective(self) -> list[str]:
-        """Compile the full model forward as one graph.
+        """Compile the full model ``forward`` as one graph.
 
-        Earlier versions compiled ``layer.mlp`` + ``layer.self_attn.o_proj``
-        (PR #2690) and then the whole ``layer`` (perf/voxcpm2-streaming-vae).
-        Both still paid one Dynamo dispatch per layer per decode step.
-        V3 profiling showed 1,332 per-layer dispatches (~28 layers × ~47
-        decode steps) costing ~726 ms of CPU self-time for a long prompt.
-
-        Compiling ``forward`` at the model level lets Dynamo unroll the
-        28-layer Python loop inside the graph. Graph breaks at
-        PagedAttention produce sub-graphs but Dynamo memoises the whole
-        trace once, so the per-step dispatch drops from 28 to just a few.
+        ``torch.compile`` is applied at the model level so Dynamo unrolls the
+        per-layer Python loop inside the graph. Graph breaks at PagedAttention
+        produce sub-graphs, but Dynamo memoises the whole trace once, so the
+        per-decode-step dispatch drops to just a few instead of one per layer.
         """
         if self._compiled_layers:
             return []
diff --git a/vllm_omni/model_executor/models/voxcpm2/voxcpm2_talker.py b/vllm_omni/model_executor/models/voxcpm2/voxcpm2_talker.py
index 3fcc683f3..9505bfbee 100644
--- a/vllm_omni/model_executor/models/voxcpm2/voxcpm2_talker.py
+++ b/vllm_omni/model_executor/models/voxcpm2/voxcpm2_talker.py
@@ -312,7 +312,6 @@ class _RequestState:
     decode_step_count: int = 0
     request_start_time: float = 0.0
     prefill_completed: bool = False
-    prefill_text: str = ""
     prompt_cache: dict | None = None
     prefill_masks: tuple | None = None
     is_stopping: bool = False
@@ -2510,7 +2509,6 @@ class VoxCPM2TalkerForConditionalGeneration(nn.Module):
                 token_ids = token_ids[1:]
 
             state = self._get_or_create_state(req_id)
-            state.prefill_text = ""
             state.decode_pad = None
             state.prefill_completed = False
             state.decode_step_count = 0
@@ -2572,29 +2570,26 @@ class VoxCPM2TalkerForConditionalGeneration(nn.Module):
                 )
 
             if state.prompt_cache is None and (ref_audio or (prompt_audio and prompt_text)):
-                if state.prompt_cache is None:
-                    try:
-                        state.prompt_cache = self._build_prompt_cache(
-                            ref_audio=ref_audio,
-                            prompt_audio=prompt_audio,
-                            prompt_text=prompt_text,
+                try:
+                    state.prompt_cache = self._build_prompt_cache(
+                        ref_audio=ref_audio,
+                        prompt_audio=prompt_audio,
+                        prompt_text=prompt_text,
+                    )
+                    if (
+                        voice_name
+                        and state.prompt_cache is not None
+                        and state.prompt_cache.get("mode") == "reference"
+                        and "ref_audio_feat" in state.prompt_cache
+                    ):
+                        _key = self._speaker_cache.make_cache_key(
+                            voice_name, model_type="voxcpm2", created_at=_created_at
                         )
-                        if (
-                            voice_name
-                            and state.prompt_cache is not None
-                            and state.prompt_cache.get("mode") == "reference"
-                            and "ref_audio_feat" in state.prompt_cache
-                        ):
-                            _key = self._speaker_cache.make_cache_key(
-                                voice_name, model_type="voxcpm2", created_at=_created_at
-                            )
-                            self._speaker_cache.put(
-                                _key, {"ref_audio_feat": state.prompt_cache["ref_audio_feat"].cpu()}
-                            )
-                            logger.debug("Speaker cache STORE for VoxCPM2 speaker '%s'", voice_name)
-                    except Exception as e:
-                        logger.warning("build_prompt_cache failed: %s; falling back to zero-shot", e)
-                        state.prompt_cache = None
+                        self._speaker_cache.put(_key, {"ref_audio_feat": state.prompt_cache["ref_audio_feat"].cpu()})
+                        logger.debug("Speaker cache STORE for VoxCPM2 speaker '%s'", voice_name)
+                except Exception as e:
+                    logger.warning("build_prompt_cache failed: %s; falling back to zero-shot", e)
+                    state.prompt_cache = None
 
             inputs = self._build_prefill_inputs(token_ids, dev, req_id)
             tts = self.tts
diff --git a/vllm_omni/model_executor/models/voxtral_tts/voxtral_tts.py b/vllm_omni/model_executor/models/voxtral_tts/voxtral_tts.py
index d12034c24..8b6872cfe 100644
--- a/vllm_omni/model_executor/models/voxtral_tts/voxtral_tts.py
+++ b/vllm_omni/model_executor/models/voxtral_tts/voxtral_tts.py
@@ -244,9 +244,6 @@ class VoxtralTTSForConditionalGeneration(
         # Delegate to generation model for multimodal processing
         return self.model.embed_multimodal(**kwargs)
 
-    def last_index_of(self, list, value):
-        return len(list) - 1 - list[::-1].index(value)
-
     def forward(
         self,
         input_ids: torch.Tensor,
diff --git a/vllm_omni/model_executor/models/voxtral_tts/voxtral_tts_audio_generation.py b/vllm_omni/model_executor/models/voxtral_tts/voxtral_tts_audio_generation.py
index 2e50e34f3..ce005936c 100644
--- a/vllm_omni/model_executor/models/voxtral_tts/voxtral_tts_audio_generation.py
+++ b/vllm_omni/model_executor/models/voxtral_tts/voxtral_tts_audio_generation.py
@@ -246,7 +246,6 @@ class BidirectionalAttention(nn.Module):
 
         self.n_local_heads: int = args.n_heads
         self.n_local_kv_heads: int = args.n_kv_heads
-        self.repeats = self.n_local_heads
         self.layer_id = layer_id
 
         self.head_dim = args.head_dim
@@ -276,7 +275,6 @@ class BidirectionalAttention(nn.Module):
             bias=args.use_biases,
         )
 
-        self.softmax_scale: float = self.args.head_dim**-0.5
         self.repeats = self.n_local_heads // self.n_local_kv_heads
 
     def _native_attention(
@@ -420,13 +418,6 @@ class FlowMatchingAudioTransformer(nn.Module):
         self.acoustic_transformer_args = args
         assert isinstance(self.acoustic_transformer_args, AcousticTransformerArgs)
 
-        # currently assuming always 1 semantic codebook + N acoustic codebooks
-        self.num_non_acoustic_embeddings = 1
-        self.num_acoustic_codebooks = len(self.model_args.get_codebook_sizes()) - self.num_non_acoustic_embeddings
-
-        # flow matching utils
-        self.sigma = args.sigma
-
         # codebook sizes
         acoustic_codebook_sizes = self.model_args.get_codebook_sizes(
             pad_to_multiple=None, include_special_tokens=False
@@ -715,7 +706,6 @@ class VoxtralTTSProcessorAdapter:
                 )
 
             text_tokens_for_audio = list[torch.Tensor]()
-            assert audios is not None
             audio_tokens_pt = list[torch.Tensor]()
             for audio_token_array in audio_tokens:
                 assert isinstance(audio_token_array, np.ndarray)
@@ -1008,7 +998,7 @@ class VoxtralTTSAudioGenerationForConditionalGeneration(nn.Module, SupportsMulti
 
         if audio_arrays is not None:
             if not isinstance(audio_arrays, (torch.Tensor, list)):
-                raise ValueError(f"Incorrect type of images. Got type: {type(audio_arrays)}")
+                raise ValueError(f"Incorrect type of audio_arrays. Got type: {type(audio_arrays)}")
             if isinstance(audio_arrays, torch.Tensor) and audio_arrays.dim() == 3:
                 audio_arrays = flatten_bn(audio_arrays)
             if isinstance(audio_arrays, torch.Tensor):
@@ -1053,24 +1043,11 @@ class VoxtralTTSAudioGenerationForConditionalGeneration(nn.Module, SupportsMulti
         fake_logits[~is_eos, self.audio_tok_id] = 1.0
         return fake_logits
 
-    # TODO(chenyo): Remove this
-    def compute_logits(
-        self,
-        hidden_states: torch.Tensor,
-    ) -> torch.Tensor | None:
-        text_logits = self.language_model.compute_logits(
-            hidden_states,
-        )
-        assert text_logits is not None
-        return text_logits
-
     def compute_mm_logits(
         self,
         hidden_states: torch.Tensor,
         cfg_alpha: torch.Tensor,
     ) -> tuple[torch.Tensor, torch.Tensor | None]:
-        audio_codes = None
-        mm_tokens = None
         audio_codes = self.acoustic_transformer(
             llm_hidden=hidden_states,
             cfg_alpha=cfg_alpha,
@@ -1088,7 +1065,6 @@ class VoxtralTTSAudioGenerationForConditionalGeneration(nn.Module, SupportsMulti
         return fake_eos, mm_tokens
 
     def load_weights(self, weights: Iterable[tuple[str, torch.Tensor]]) -> set[str]:
-        # fmt: on
         remapping_rules = [
             (r"^acoustic_transformer\.(.*)$", r"\1"),  # noqa: E501
             (r"^audio_tokenizer\.(.*)$", r"\1"),  # noqa: E501
diff --git a/vllm_omni/model_executor/models/voxtral_tts/voxtral_tts_audio_tokenizer.py b/vllm_omni/model_executor/models/voxtral_tts/voxtral_tts_audio_tokenizer.py
index fc753a58f..57a86a5e9 100644
--- a/vllm_omni/model_executor/models/voxtral_tts/voxtral_tts_audio_tokenizer.py
+++ b/vllm_omni/model_executor/models/voxtral_tts/voxtral_tts_audio_tokenizer.py
@@ -652,7 +652,7 @@ class TransformerBlock(nn.Module):
     def layer_id(self) -> int:
         return self._layer_id
 
-    def forward(self, x: torch.Tensor) -> tuple[torch.Tensor, torch.Tensor | None, torch.Tensor | None]:
+    def forward(self, x: torch.Tensor) -> torch.Tensor:
         r = self.attention.forward(self.attention_norm(x))
         if self.post_attention_norm is not None:
             r = self.post_attention_norm(r)
@@ -757,7 +757,7 @@ class VoxtralTTSAudioTokenizer(nn.Module):
             # transformer
             layer_args = deepcopy(args)
             layer_args.attn_sliding_window_size = cur_window_size
-            assert layer_args.qk_norm, "qk_norm must be True for decoder"
+            assert layer_args.qk_norm, "qk_norm must be True for encoder"
             encoder_transformer = Transformer(
                 args=layer_args,
                 n_layers=n_layers,
diff --git a/vllm_omni/transformers_utils/processors/ming.py b/vllm_omni/transformers_utils/processors/ming.py
index 685909029..b32f10d12 100644
--- a/vllm_omni/transformers_utils/processors/ming.py
+++ b/vllm_omni/transformers_utils/processors/ming.py
@@ -143,7 +143,6 @@ class MingWhisperFeatureExtractor(FeatureExtractionMixin):
         #      (conv1 has stride=1 and does not change T)
         #   2. AudioProjector Conv1d: kernel=3, stride=2, padding=1
         # Combined: T → ((T-1)//2 + 1 - 1)//2 + 1
-        # See also: AudioProjector.compute_output_length()
         encoder_feats_lengths = ((audio_feats_lengths - 3 + 2 * 1) // 2 + 1 - 3 + 2 * 1) // 2 + 1
         audio_feats = torch.cat(audio_feat_list, dim=0).unsqueeze(0)  # [1, T_total, n_mels]