diff --git a/benchmarks/qwen3-omni/README.md b/benchmarks/qwen3-omni/README.md
deleted file mode 100644
index dc282d052..000000000
--- a/benchmarks/qwen3-omni/README.md
+++ /dev/null
@@ -1,86 +0,0 @@
-# Benchmarks Guide
-
-This README explains how to (1) prepare benchmark datasets and (2) run the provided Qwen3-Omni benchmarks.
-
-## 1) Prepare the dataset (SeedTTS top100)
-
-```bash
-cd benchmarks/build_dataset
-pip install gdown
-
-# Download SeedTTS test set from Google Drive
-gdown  1GlSjVfSHkW3-leKKBlfrjuuTGqQ_xaLP
-
-# Extract
-tar -xf seedtts_testset.tar
-
-# Copy metadata and extract top-100 prompts
-cp seedtts_testset/en/meta.lst meta.lst
-python extract_prompts.py -i meta.lst -o top100.txt -n 100
-
-# (Optional) clean up to save space
-rm -rf seedtts_testset seedtts_testset.tar meta.lst
-```
-
-Artifacts:
-- `benchmarks/build_dataset/top100.txt` — 100 text prompts (one per line).
-
-## 2) Run benchmarks
-
-All commands assume repo root (`vllm-omni`).
-
-### A. Transformers benchmark (offline, HF Transformers)
-
-```
-bash benchmarks/qwen3-omni/transformers/eval_qwen3_moe_omni_transformers.sh
-```
-
-What it does:
-- Runs `qwen3_omni_moe_transformers.py` over `top100.txt` with `--num-prompts 100`.
-- Outputs to `benchmarks/qwen3-omni/transformers/benchmark_results/`:
-  - `perf_stats.json` — aggregated & per-prompt TPS/latency (thinker/talker/code2wav/overall).
-  - `results.json` — per-prompt outputs and audio paths.
-  - `audio/` — ~100 generated `.wav` files.
-
-Key checks:
-- `overall_tps` and `*_tps_avg` should be non-zero and reasonably stable.
-- Investigate any 0/NaN or unusually low TPS / long-tail latency.
-
-### B. vLLM Omni end-to-end benchmark (pipeline)
-
-```
-bash benchmarks/qwen3-omni/vllm_omni/eval_qwen3_moe_omni.sh
-```
-
-What it does:
-- Runs `examples/offline_inference/qwen3_omni/end2end.py` with `--log-stats`.
-- Uses `benchmarks/build_dataset/top100.txt` and writes to:
-  - Logs: `benchmarks/qwen3-omni/vllm_omni/logs/`
-    - `omni_pipeline_text.orchestrator.stats.jsonl` — per-stage latency stats.
-    - `omni_pipeline_text.overall.stats.jsonl` — end-to-end latency/TPS.
-    - `omni_pipeline_text.stage{0,1,2}.log` — per-stage detailed logs/errors.
-  - Outputs: `benchmarks/qwen3-omni/vllm_omni/outputs/` — ~100 text and `.wav` files.
-
-Key checks:
-- Overall stats: end-to-end latency/TPS should be reasonable.
-- Orchestrator stats: per-stage latency should be stable; investigate long tails.
-- Stage logs: ensure no errors and no unusually slow stages.
-
-
-## Performance snapshot
-
-The chart below summarizes our measured Qwen3-Omni MoE end-to-end benchmark, comparing vLLM-Omni against HF Transformers. It shows the overall throughput advantage for vLLM-Omni. These are actual experiment results—please refer to this performance when evaluating or reproducing the benchmark.
-
-![vLLM-Omni vs HF](./vllm-omni-vs-hf.png)
-
-## Directory layout
-- `benchmarks/build_dataset/` — dataset prep utilities (e.g., SeedTTS top100).
-- `benchmarks/<model>/vllm_omni/` — vLLM-Omni pipeline benchmarks, logs, outputs.
-- Add new tasks under `benchmarks/<model>/...` with the same pattern: `transformers/`, `vllm_omni/`, task-specific README, and (optionally) dataset prep notes.
-- `benchmarks/<model>/vllm-omni-vs-hf.png` — current performance snapshot (overall throughput comparison).
-- `benchmarks/<model>/transformers/` — HF Transformers benchmarks (offline reference).
-
-## Troubleshooting
-- Make sure GPU/driver/FlashAttention2 requirements are met for the chosen model.
-- If downloads fail, confirm network access to Google Drive (`gdown`) and Hugging Face.
-- If audio files are missing, check for errors in stage logs or model generation.***
diff --git a/benchmarks/qwen3-omni/transformers/eval_qwen3_moe_omni_transformers.sh b/benchmarks/qwen3-omni/transformers/eval_qwen3_moe_omni_transformers.sh
deleted file mode 100644
index bae514fab..000000000
--- a/benchmarks/qwen3-omni/transformers/eval_qwen3_moe_omni_transformers.sh
+++ /dev/null
@@ -1,26 +0,0 @@
-#!/bin/bash
-# Qwen3-Omni Transformers Benchmark Evaluation Script
-# This script must be run from the vllm-omni root directory
-
-# Get the directory where this script is located
-SCRIPT_DIR="$(cd "$(dirname "${BASH_SOURCE[0]}")" && pwd)"
-
-# Navigate to vllm-omni root directory (4 levels up from script location)
-VLLM_OMNI_ROOT="$(cd "$SCRIPT_DIR/../../.." && pwd)"
-cd "$VLLM_OMNI_ROOT" || { echo "Error: Failed to navigate to vllm-omni directory"; exit 1; }
-
-echo "Working directory: $(pwd)"
-# Verify we're in the correct directory and run benchmark
-if [[ ! -f "benchmarks/qwen3-omni/transformers/qwen3_omni_moe_transformers.py" ]]; then
-    echo "Error: Not in vllm-omni root directory. Please run from vllm-omni folder."
-else
-    cd benchmarks/qwen3-omni/transformers
-
-    python qwen3_omni_moe_transformers.py --prompts-file ../../build_dataset/top100.txt --num-prompts 100
-
-    echo "Logs and outputs are saved to $(pwd)/benchmark_results:"
-    echo "  - perf_stats.json    Aggregated/per-prompt TPS and latency (thinker/talker/code2wav/overall)"
-    echo "  - results.json       Per-prompt outputs and audio paths"
-    echo "  - audio/             Generated wav files, there should be 100 wav file generated"
-    echo "Key checks: overall_tps and *_tps_avg should be non-zero and stable; investigate 0/NaN or unusually low TPS/long-tail latency."
-fi
diff --git a/benchmarks/qwen3-omni/transformers/qwen3_omni_moe_model.py b/benchmarks/qwen3-omni/transformers/qwen3_omni_moe_model.py
deleted file mode 100644
index 43b56f3e9..000000000
--- a/benchmarks/qwen3-omni/transformers/qwen3_omni_moe_model.py
+++ /dev/null
@@ -1,265 +0,0 @@
-import time
-
-import torch
-from transformers import Qwen3OmniMoeForConditionalGeneration
-
-
-class Qwen3OmniMoeForConditionalGenerationWithLogging(Qwen3OmniMoeForConditionalGeneration):
-    @torch.no_grad()
-    def generate(
-        self,
-        input_ids: torch.Tensor | None = None,
-        speaker: str = "Ethan",
-        use_audio_in_video: bool = False,
-        return_audio: bool | None = None,
-        thinker_max_new_tokens: int = 1024,
-        thinker_eos_token_id: int = 151645,
-        talker_max_new_tokens: int = 4096,
-        talker_do_sample: bool = True,
-        talker_top_k: int = 50,
-        talker_top_p: float = 1.0,
-        talker_temperature: float = 0.9,
-        talker_repetition_penalty: float = 1.05,
-        **kwargs,
-    ):
-        total_t0 = time.time()
-        perf_stats = {
-            "thinker_tokens": 0,
-            "thinker_time_s": 0.0,
-            "thinker_tps": 0.0,
-            "talker_tokens": 0,
-            "talker_time_s": 0.0,
-            "talker_tps": 0.0,
-            "code2wav_tokens": 0,
-            "code2wav_time_s": 0.0,
-            "code2wav_tps": 0.0,
-            "total_tokens": 0,
-            "total_time_s": 0.0,
-            "total_tps": 0.0,
-        }
-        if return_audio and not self.has_talker:
-            raise ValueError(
-                "Cannot use talker when talker module not initialized. "
-                "Use `enable_talker` method or set enable_talker in config "
-                "to enable talker."
-            )
-        if return_audio is None:
-            return_audio = self.has_talker
-
-        shared_kwargs = {"use_audio_in_video": use_audio_in_video}
-        thinker_kwargs = {
-            "max_new_tokens": thinker_max_new_tokens,
-            "eos_token_id": thinker_eos_token_id,
-        }
-
-        talker_kwargs = {}
-        token2wav_kwargs = {}
-        if return_audio:
-            speaker_id = self.config.talker_config.speaker_id.get(speaker.lower())
-            if speaker_id is None:
-                raise NotImplementedError(f"Speaker {speaker} not implemented")
-            if input_ids.shape[0] != 1:
-                raise NotImplementedError("Qwen3-Omni currently does not support batched inference with audio output")
-            talker_suppressed_tokens = [
-                i
-                for i in range(
-                    self.config.talker_config.text_config.vocab_size - 1024,
-                    self.config.talker_config.text_config.vocab_size,
-                )
-                if i != self.config.talker_config.codec_eos_token_id
-            ]  # Suppress additional special tokens, should not be predicted
-            talker_kwargs = {
-                "max_new_tokens": talker_max_new_tokens,
-                "do_sample": talker_do_sample,
-                "top_k": talker_top_k,
-                "top_p": talker_top_p,
-                "temperature": talker_temperature,
-                "eos_token_id": self.config.talker_config.codec_eos_token_id,
-                "repetition_penalty": talker_repetition_penalty,
-                "suppress_tokens": talker_suppressed_tokens,
-                "output_hidden_states": True,
-                "return_dict_in_generate": True,
-            }
-            token2wav_kwargs = {}
-
-        for key, value in kwargs.items():
-            if key.startswith("thinker_"):
-                thinker_kwargs[key[len("thinker_") :]] = value
-            elif key.startswith("talker_"):
-                talker_kwargs[key[len("talker_") :]] = value
-            elif key.startswith("token2wav_"):
-                token2wav_kwargs[key[len("token2wav_") :]] = value
-            # Process special input values
-            elif key == "feature_attention_mask":
-                thinker_kwargs[key] = value
-                talker_kwargs["audio_feature_lengths"] = torch.sum(value, dim=1)
-            elif key in ("input_features", "attention_mask"):
-                thinker_kwargs[key] = value
-            # Put other key to shared kwargs
-            else:
-                shared_kwargs[key] = value
-
-        # Merge kwargs
-        for key, value in shared_kwargs.items():
-            if key not in thinker_kwargs:
-                thinker_kwargs[key] = value
-            if key not in talker_kwargs and key in ["image_grid_thw", "video_grid_thw", "video_second_per_grid"]:
-                talker_kwargs[key] = value
-            if key not in token2wav_kwargs:
-                token2wav_kwargs[key] = value
-
-        # 1. Generate from thinker module
-        generate_audio = return_audio and self.has_talker
-        if generate_audio:
-            thinker_kwargs["output_hidden_states"] = True
-            thinker_kwargs["return_dict_in_generate"] = True
-
-        t0 = time.time()
-        thinker_result = self.thinker.generate(input_ids=input_ids, **thinker_kwargs)
-        t1 = time.time()
-        perf_stats["thinker_time_s"] = max(0.0, t1 - t0)
-        try:
-            prompt_len = int(input_ids.shape[1]) if input_ids is not None else 0
-            total_len = int(thinker_result.sequences.shape[-1])
-            thinker_out_len = max(0, total_len - prompt_len)
-        except Exception:
-            thinker_out_len = 0
-        perf_stats["thinker_tokens"] = thinker_out_len
-        perf_stats["thinker_tps"] = (
-            (thinker_out_len / perf_stats["thinker_time_s"]) if perf_stats["thinker_time_s"] > 0 else 0.0
-        )
-
-        if not generate_audio:
-            perf_stats["total_tokens"] = perf_stats["thinker_tokens"]
-            perf_stats["total_time_s"] = time.time() - total_t0
-            perf_stats["total_tps"] = (
-                (perf_stats["total_tokens"] / perf_stats["total_time_s"]) if perf_stats["total_time_s"] > 0 else 0.0
-            )
-            # attach stats to self
-            setattr(self, "_perf_stats_last", perf_stats)
-            if not hasattr(self, "_perf_stats_history"):
-                setattr(self, "_perf_stats_history", [])
-            self._perf_stats_history.append(perf_stats)
-            return thinker_result, None
-
-        # 2. Prepare talker input
-        thinker_embed = torch.cat([hidden_states[0] for hidden_states in thinker_result.hidden_states], dim=1).to(
-            self.talker.device
-        )  # [1 t d]
-        thinker_hidden = torch.cat(
-            [
-                hidden_states[self.config.talker_config.accept_hidden_layer]
-                for hidden_states in thinker_result.hidden_states
-            ],
-            dim=1,
-        ).to(self.talker.device)  # [1 t d]
-
-        im_start_indexes = torch.cat(
-            (
-                torch.nonzero(input_ids[0] == self.config.im_start_token_id).squeeze(),
-                torch.tensor([thinker_result.sequences.shape[-1]], device=input_ids.device, dtype=input_ids.dtype),
-            ),
-            dim=-1,
-        ).to(self.talker.device)  # Shape [n_starts + 1]; Take batch 0 since batched inference is not supported here.
-        multimodal_mask = (
-            (thinker_result.sequences == self.config.thinker_config.audio_token_id) |
-            (thinker_result.sequences == self.config.thinker_config.image_token_id) |
-            (thinker_result.sequences == self.config.thinker_config.video_token_id)
-        ).to(self.talker.device)  # [1 t] # fmt: skip
-
-        talker_special_tokens = torch.tensor(
-            [[self.config.tts_bos_token_id, self.config.tts_eos_token_id, self.config.tts_pad_token_id]],
-            device=self.thinker.device,
-            dtype=input_ids.dtype,
-        )
-        tts_bos_embed, tts_eos_embed, tts_pad_embed = (
-            self.talker.text_projection(self.thinker.get_input_embeddings()(talker_special_tokens))
-            .to(self.talker.device)
-            .chunk(3, dim=1)
-        )  # 3 * [1 1 d]
-
-        talker_input_embeds = []  # [1 t d]
-        talker_input_ids = []
-        # For every chatml parts
-        for i in range(len(im_start_indexes) - 1):
-            im_start_index = im_start_indexes[i]
-            segment_end_index = im_start_indexes[i + 1]
-            role_token = input_ids[0][im_start_index + 1]
-            # Talker should ignore thinker system prompt
-            if role_token == self.config.system_token_id:
-                continue
-            # Talker takes word embeddings for tokens and hidden state from `accept_hidden_layer` for multimodal inputs
-            elif role_token == self.config.user_token_id:
-                talker_user_part = self._get_talker_user_parts(
-                    im_start_index, segment_end_index, multimodal_mask, thinker_hidden, thinker_embed
-                )
-                talker_input_embeds.append(talker_user_part)
-                talker_input_ids.append(thinker_result.sequences[:, im_start_index:segment_end_index])
-            # Take assistant output (for now)
-            elif role_token == self.config.assistant_token_id and i == len(im_start_indexes) - 2:
-                talker_assistant_embeds, talker_assistant_ids, trailing_text_hidden = self._get_talker_assistant_parts(
-                    im_start_index,
-                    segment_end_index,
-                    speaker_id,
-                    thinker_embed,
-                    tts_pad_embed,
-                    tts_bos_embed,
-                    tts_eos_embed,
-                )
-                talker_input_embeds.append(talker_assistant_embeds)
-                talker_input_ids.append(talker_assistant_ids)
-            # History assistant output (ignore for now)
-            elif role_token == self.config.assistant_token_id and i != len(im_start_indexes) - 2:
-                continue
-            else:
-                raise AssertionError("Expect role id after <|im_start|> (assistant, user, system)")
-        talker_input_embed = torch.cat([embed.to(self.talker.device) for embed in talker_input_embeds], dim=1)
-        talker_input_id = torch.cat([embed.to(self.talker.device) for embed in talker_input_ids], dim=1)
-        t2 = time.time()
-        talker_result = self.talker.generate(
-            inputs_embeds=talker_input_embed,
-            trailing_text_hidden=trailing_text_hidden,
-            tts_pad_embed=tts_pad_embed,
-            talker_input_ids=talker_input_id,  # Not use input_ids to prevent repetition penalty out of bound
-            **talker_kwargs,
-        )
-        t3 = time.time()
-        perf_stats["talker_time_s"] = max(0.0, t3 - t2)
-        talker_codes = (
-            torch.stack([hid[-1] for hid in talker_result.hidden_states if hid[-1] is not None], dim=1)
-            .transpose(1, 2)
-            .to(self.code2wav.device)
-        )
-        try:
-            # codes shape: (B, num_quantizers, T). We log T as token length.
-            perf_stats["talker_tokens"] = int(talker_codes.shape[-1])
-        except Exception:
-            perf_stats["talker_tokens"] = 0
-        perf_stats["talker_tps"] = (
-            (perf_stats["talker_tokens"] / perf_stats["talker_time_s"]) if perf_stats["talker_time_s"] > 0 else 0.0
-        )
-        t4 = time.time()
-        talker_wavs = self.code2wav.chunked_decode(talker_codes, chunk_size=300, left_context_size=25).float()
-        t5 = time.time()
-        perf_stats["code2wav_time_s"] = max(0.0, t5 - t4)
-        perf_stats["code2wav_tokens"] = perf_stats["talker_tokens"]  # same T, not times 16
-        perf_stats["code2wav_tps"] = (
-            (perf_stats["code2wav_tokens"] / perf_stats["code2wav_time_s"])
-            if perf_stats["code2wav_time_s"] > 0
-            else 0.0
-        )
-        perf_stats["total_tokens"] = perf_stats["thinker_tokens"] + perf_stats["talker_tokens"]
-        perf_stats["total_time_s"] = time.time() - total_t0
-        perf_stats["total_tps"] = (
-            (perf_stats["total_tokens"] / perf_stats["total_time_s"]) if perf_stats["total_time_s"] > 0 else 0.0
-        )
-        setattr(self, "_perf_stats_last", perf_stats)
-        if not hasattr(self, "_perf_stats_history"):
-            setattr(self, "_perf_stats_history", [])
-        self._perf_stats_history.append(perf_stats)
-        return thinker_result, talker_wavs.float()
-
-
-__all__ = [
-    "Qwen3OmniMoeForConditionalGenerationWithLogging",
-]
diff --git a/benchmarks/qwen3-omni/transformers/qwen3_omni_moe_transformers.py b/benchmarks/qwen3-omni/transformers/qwen3_omni_moe_transformers.py
deleted file mode 100644
index 87d3de797..000000000
--- a/benchmarks/qwen3-omni/transformers/qwen3_omni_moe_transformers.py
+++ /dev/null
@@ -1,275 +0,0 @@
-import argparse
-import json
-import os
-
-import soundfile as sf
-from qwen3_omni_moe_model import Qwen3OmniMoeForConditionalGenerationWithLogging
-from qwen_omni_utils import process_mm_info
-from tqdm import tqdm
-from transformers import Qwen3OmniMoeProcessor
-
-MODEL_PATH = "Qwen/Qwen3-Omni-30B-A3B-Instruct"
-# MODEL_PATH = "Qwen/Qwen3-Omni-30B-A3B-Thinking"
-
-
-def load_prompts(prompts_file: str) -> list[str]:
-    """Load prompts from a text file, one prompt per line."""
-    prompts = []
-    with open(prompts_file, encoding="utf-8") as f:
-        for line in f:
-            line = line.strip()
-            if line:
-                prompts.append(line)
-    return prompts
-
-
-def run_benchmark(
-    model,
-    processor,
-    prompts: list[str],
-    output_dir: str = "benchmark_results",
-    speaker: str = "Ethan",
-    use_audio_in_video: bool = True,
-):
-    """
-    Run benchmark on a list of prompts and collect performance stats.
-
-    Args:
-        model: The Qwen3OmniMoe model
-        processor: The Qwen3OmniMoe processor
-        prompts: List of text prompts to process
-        output_dir: Directory to save results
-        speaker: Speaker voice for audio output
-        use_audio_in_video: Whether to use audio in video
-
-    Returns:
-        tuple: (aggregated_stats, results, audio_outputs)
-            - aggregated_stats: dict with aggregated performance statistics
-            - results: list of dicts with per-prompt results
-            - audio_outputs: list of audio tensors/arrays (or None if no audio)
-    """
-    os.makedirs(output_dir, exist_ok=True)
-    audio_dir = os.path.join(output_dir, "audio")
-    os.makedirs(audio_dir, exist_ok=True)
-
-    all_stats = []
-    results = []
-    audio_outputs = []
-
-    for idx, prompt in enumerate(tqdm(prompts, desc="Processing prompts")):
-        conversation = [
-            {
-                "role": "user",
-                "content": [{"type": "text", "text": prompt}],
-            },
-        ]
-
-        # Preparation for inference
-        text = processor.apply_chat_template(conversation, add_generation_prompt=True, tokenize=False)
-        audios, images, videos = process_mm_info(conversation, use_audio_in_video=use_audio_in_video)
-        inputs = processor(
-            text=text,
-            audio=audios,
-            images=images,
-            videos=videos,
-            return_tensors="pt",
-            padding=True,
-            use_audio_in_video=use_audio_in_video,
-        )
-        inputs = inputs.to(model.device).to(model.dtype)
-
-        # Inference: Generation of the output text and audio
-        text_ids, audio = model.generate(
-            **inputs, speaker=speaker, thinker_return_dict_in_generate=True, use_audio_in_video=use_audio_in_video
-        )
-
-        # Decode output text
-        output_text = processor.batch_decode(
-            text_ids.sequences[:, inputs["input_ids"].shape[1] :],
-            skip_special_tokens=True,
-            clean_up_tokenization_spaces=False,
-        )[0]
-
-        # Collect performance stats
-        perf_stats = None
-        if hasattr(model, "_perf_stats_last"):
-            perf_stats = model._perf_stats_last.copy()
-            perf_stats["prompt_idx"] = idx
-            perf_stats["prompt"] = prompt
-            all_stats.append(perf_stats)
-
-        # Save audio and collect audio output
-        audio_path = None
-        audio_data = None
-        if audio is not None:
-            audio_data = audio.reshape(-1).detach().cpu().numpy()
-            audio_path = os.path.join(audio_dir, f"output_{idx:04d}.wav")
-            sf.write(
-                audio_path,
-                audio_data,
-                samplerate=24000,
-            )
-            audio_outputs.append(audio_data)
-        else:
-            audio_outputs.append(None)
-
-        # Save result
-        result = {
-            "idx": idx,
-            "prompt": prompt,
-            "output": output_text,
-            "audio_path": audio_path,
-            "perf_stats": perf_stats,
-        }
-        results.append(result)
-
-    # Aggregate statistics
-    aggregated_stats = aggregate_stats(all_stats)
-
-    # Save all results
-    results_path = os.path.join(output_dir, "results.json")
-    with open(results_path, "w", encoding="utf-8") as f:
-        json.dump(results, f, ensure_ascii=False, indent=2)
-
-    # Save aggregated stats
-    stats_path = os.path.join(output_dir, "perf_stats.json")
-    with open(stats_path, "w", encoding="utf-8") as f:
-        json.dump({"aggregated": aggregated_stats, "per_prompt": all_stats}, f, ensure_ascii=False, indent=2)
-
-    # Count saved audio files
-    num_audio_saved = sum(1 for a in audio_outputs if a is not None)
-    print(f"\nSaved {num_audio_saved} audio files to {audio_dir}/")
-
-    return aggregated_stats, results, audio_outputs
-
-
-def aggregate_stats(all_stats: list[dict]) -> dict:
-    """Aggregate performance statistics from multiple runs."""
-    if not all_stats:
-        return {}
-
-    keys = [
-        "thinker_tokens",
-        "thinker_time_s",
-        "thinker_tps",
-        "talker_tokens",
-        "talker_time_s",
-        "talker_tps",
-        "code2wav_tokens",
-        "code2wav_time_s",
-        "code2wav_tps",
-        "total_tokens",
-        "total_time_s",
-        "total_tps",
-    ]
-
-    aggregated = {
-        "num_samples": len(all_stats),
-    }
-
-    for key in keys:
-        values = [s.get(key, 0) for s in all_stats if key in s]
-        if values:
-            aggregated[f"{key}_sum"] = sum(values)
-            aggregated[f"{key}_avg"] = sum(values) / len(values)
-            aggregated[f"{key}_min"] = min(values)
-            aggregated[f"{key}_max"] = max(values)
-
-    # Calculate overall throughput
-    total_tokens = aggregated.get("total_tokens_sum", 0)
-    total_time = aggregated.get("total_time_s_sum", 0)
-    if total_time > 0:
-        aggregated["overall_tps"] = total_tokens / total_time
-
-    return aggregated
-
-
-def print_stats(stats: dict):
-    """Print performance statistics in a formatted way."""
-    print("\n" + "=" * 60)
-    print("Performance Statistics Summary")
-    print("=" * 60)
-
-    print(f"\nNumber of samples: {stats.get('num_samples', 0)}")
-
-    print("\n--- Thinker ---")
-    print(f"  Total tokens:  {stats.get('thinker_tokens_sum', 0):.0f}")
-    print(f"  Total time:    {stats.get('thinker_time_s_sum', 0):.2f}s")
-    print(f"  Avg TPS:       {stats.get('thinker_tps_avg', 0):.2f}")
-    print(f"  Min TPS:       {stats.get('thinker_tps_min', 0):.2f}")
-    print(f"  Max TPS:       {stats.get('thinker_tps_max', 0):.2f}")
-
-    print("\n--- Talker ---")
-    print(f"  Total tokens:  {stats.get('talker_tokens_sum', 0):.0f}")
-    print(f"  Total time:    {stats.get('talker_time_s_sum', 0):.2f}s")
-    print(f"  Avg TPS:       {stats.get('talker_tps_avg', 0):.2f}")
-    print(f"  Min TPS:       {stats.get('talker_tps_min', 0):.2f}")
-    print(f"  Max TPS:       {stats.get('talker_tps_max', 0):.2f}")
-
-    print("\n--- Code2Wav ---")
-    print(f"  Total tokens:  {stats.get('code2wav_tokens_sum', 0):.0f}")
-    print(f"  Total time:    {stats.get('code2wav_time_s_sum', 0):.2f}s")
-    print(f"  Avg TPS:       {stats.get('code2wav_tps_avg', 0):.2f}")
-    print(f"  Min TPS:       {stats.get('code2wav_tps_min', 0):.2f}")
-    print(f"  Max TPS:       {stats.get('code2wav_tps_max', 0):.2f}")
-
-    print("\n--- Overall ---")
-    print(f"  Total tokens:  {stats.get('total_tokens_sum', 0):.0f}")
-    print(f"  Total time:    {stats.get('total_time_s_sum', 0):.2f}s")
-    print(f"  Overall TPS:   {stats.get('overall_tps', 0):.2f}")
-    print(f"  Avg TPS:       {stats.get('total_tps_avg', 0):.2f}")
-    print(f"  Min TPS:       {stats.get('total_tps_min', 0):.2f}")
-    print(f"  Max TPS:       {stats.get('total_tps_max', 0):.2f}")
-
-    print("=" * 60 + "\n")
-
-
-def main():
-    parser = argparse.ArgumentParser(description="Qwen3-Omni Benchmark Script")
-    parser.add_argument(
-        "--prompts-file",
-        type=str,
-        default="benchmark/build_dataset/top100.txt",
-        help="Path to the prompts file (one prompt per line)",
-    )
-    parser.add_argument(
-        "--output-dir", type=str, default="benchmark_results", help="Directory to save benchmark results"
-    )
-    parser.add_argument("--model-path", type=str, default=MODEL_PATH, help="Path to the model")
-    parser.add_argument("--speaker", type=str, default="Ethan", help="Speaker voice for audio output")
-    parser.add_argument("--num-prompts", type=int, default=None, help="Number of prompts to process (default: all)")
-    args = parser.parse_args()
-
-    # Load model and processor
-    print(f"Loading model from {args.model_path}...")
-    model = Qwen3OmniMoeForConditionalGenerationWithLogging.from_pretrained(
-        args.model_path,
-        dtype="auto",
-        device_map="auto",
-        attn_implementation="flash_attention_2",
-    )
-    processor = Qwen3OmniMoeProcessor.from_pretrained(args.model_path)
-
-    # Benchmark mode
-    print(f"Loading prompts from {args.prompts_file}...")
-    prompts = load_prompts(args.prompts_file)
-
-    if args.num_prompts:
-        prompts = prompts[: args.num_prompts]
-
-    print(f"Running benchmark on {len(prompts)} prompts...")
-
-    aggregated_stats, results, audio_outputs = run_benchmark(
-        model=model,
-        processor=processor,
-        prompts=prompts,
-        output_dir=args.output_dir,
-        speaker=args.speaker,
-    )
-
-    print_stats(aggregated_stats)
-    print(f"\nResults saved to {args.output_dir}/")
-
-
-if __name__ == "__main__":
-    main()
diff --git a/benchmarks/qwen3-omni/vllm-omni-vs-hf.png b/benchmarks/qwen3-omni/vllm-omni-vs-hf.png
deleted file mode 100644
index e47079335..000000000
Binary files a/benchmarks/qwen3-omni/vllm-omni-vs-hf.png and /dev/null differ
diff --git a/benchmarks/qwen3-omni/vllm_omni/eval_qwen3_moe_omni.sh b/benchmarks/qwen3-omni/vllm_omni/eval_qwen3_moe_omni.sh
deleted file mode 100644
index e4c83e975..000000000
--- a/benchmarks/qwen3-omni/vllm_omni/eval_qwen3_moe_omni.sh
+++ /dev/null
@@ -1,37 +0,0 @@
-#!/bin/bash
-# Qwen3-Omni Benchmark Evaluation Script
-# This script must be run from the vllm-omni root directory
-
-# Get the directory where this script is located
-SCRIPT_DIR="$(cd "$(dirname "${BASH_SOURCE[0]}")" && pwd)"
-
-# Navigate to vllm-omni root directory (4 levels up from script location)
-VLLM_OMNI_ROOT="$(cd "$SCRIPT_DIR/../../.." && pwd)"
-cd "$VLLM_OMNI_ROOT" || { echo "Error: Failed to navigate to vllm-omni directory"; exit 1; }
-
-echo "Working directory: $(pwd)"
-
-# Verify we're in the correct directory and run benchmark
-if [[ ! -d "benchmarks/qwen3-omni/vllm_omni" ]]; then
-    echo "Error: Not in vllm-omni root directory. Please run from vllm-omni folder."
-else
-    log_dir=benchmarks/qwen3-omni/vllm_omni/logs
-    outputs_dir=benchmarks/qwen3-omni/vllm_omni/outputs
-    end2end_script_path=examples/offline_inference/qwen3_omni/end2end.py
-    build_dataset_path=benchmarks/build_dataset/top100.txt
-
-    python $end2end_script_path --output-wav $outputs_dir \
-                      --query-type text \
-                      --txt-prompts $build_dataset_path \
-                      --log-stats \
-                      --log-dir $log_dir
-    echo "Logs and outputs are saved in ${log_dir} and ${outputs_dir} respectively:"
-    echo "  - omni_pipeline_text                       run dir/base name"
-    echo "  - omni_pipeline_text.orchestrator.stats.jsonl  orchestrator-stage latency stats"
-    echo "  - omni_pipeline_text.overall.stats.jsonl       overall latency/TPS stats"
-    echo "  - omni_pipeline_text.stage0.log                per-stage detailed logs"
-    echo "  - omni_pipeline_text.stage1.log"
-    echo "  - omni_pipeline_text.stage2.log"
-    echo "Key checks: overall.stats.jsonl for end-to-end latency/TPS; orchestrator.stats.jsonl for stable per-stage latency; stage*.log for errors or long tails."
-    echo "  - outputs/             Generated txt and wav files, there should be 100 text and wav files generated respectively"
-fi