forked from Karylab-cklius/vllm
Signed-off-by: Saddss <28726669061@qq.com> Signed-off-by: Nick Hill <nickhill123@gmail.com> Signed-off-by: Yifan Qiao <yifanqiao@inferact.ai> Co-authored-by: Yifan Qiao <yifanqiao@inferact.ai> Co-authored-by: Saddss <28726669061@qq.com> Co-authored-by: Nick Hill <nickhill123@gmail.com>
236 lines
8.1 KiB
Python
236 lines
8.1 KiB
Python
# SPDX-License-Identifier: Apache-2.0
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# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
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import pytest
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import pytest_asyncio
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from mistral_common.protocol.transcription.request import (
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StreamingMode,
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TranscriptionRequest,
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)
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from mistral_common.tokens.tokenizers.audio import Audio
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from mistral_common.tokens.tokenizers.mistral import MistralTokenizer
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from mistral_common.tokens.tokenizers.tekken import SpecialTokenPolicy
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from vllm import LLM, SamplingParams
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from vllm.assets.audio import AudioAsset
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from vllm.engine.arg_utils import AsyncEngineArgs
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from vllm.utils.math_utils import cdiv
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from vllm.v1.engine.async_llm import AsyncLLM
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from vllm.v1.kv_cache_interface import SlidingWindowSpec
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from ....utils import ROCM_ENGINE_KWARGS
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MODEL_NAME = "mistralai/Voxtral-Mini-4B-Realtime-2602"
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AUDIO_LAYER_NAME = "whisper_encoder.whisper_encoder.layers.0.layers.self_attn.attn"
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ENGINE_CONFIG = {
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"model": MODEL_NAME,
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"max_model_len": 8192,
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"max_num_seqs": 4,
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"limit_mm_per_prompt": {"audio": 1},
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"config_format": "mistral",
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"load_format": "mistral",
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"tokenizer_mode": "mistral",
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"enforce_eager": True,
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"gpu_memory_utilization": 0.9,
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**ROCM_ENGINE_KWARGS,
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}
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EXPECTED_TEXT = [
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(
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" First words I spoke in the original phonograph. "
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"A little piece of practical poetry. Mary had a little lamb,"
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" its fleece was quite a slow, and everywhere that Mary went, "
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"the lamb was sure to go."
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),
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(
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" And the 0-1 pitch on the way to Edgar Martinez. Swung on"
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" the line. Down the left field line for OBS. Here comes Joy. "
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"Here is Junior to third base. They're going to wave him in. "
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"The throw to the plate will be late. The Mariners are going"
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" to play. For the American League Championship, "
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"I don't believe it. It just continues. My, oh, my."
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),
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]
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def _normalize(texts: list[str]) -> list[str]:
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# The model occasionally transcribes "OBS" as "a base hit" and
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# "oh, my" as "oh my", but both are acoustically valid. Normalise so
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# the assertion is stable across runs and hardware.
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texts[1] = texts[1].replace("a base hit", "OBS").replace("oh my", "oh, my")
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return texts
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def assert_encoder_kv_cache_spec(engine: LLM) -> None:
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vllm_config = engine.llm_engine.vllm_config
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audio_config = vllm_config.model_config.hf_config.audio_config
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kv_cache_specs_per_rank = engine.llm_engine.model_executor.get_kv_cache_specs()
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assert len(kv_cache_specs_per_rank) == 1
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kv_cache_specs = kv_cache_specs_per_rank[0]
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assert AUDIO_LAYER_NAME in kv_cache_specs, kv_cache_specs.keys()
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spec = kv_cache_specs[AUDIO_LAYER_NAME]
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assert audio_config.sliding_window == 750
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assert audio_config.block_pool_size == 4
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assert isinstance(spec, SlidingWindowSpec)
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assert spec.block_size == 16
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assert spec.num_kv_heads == 128
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# cdiv(750, 4) == 188 pooled tokens cover the model's window; the extra
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# +1 is an eviction margin (see whisper_causal.py get_kv_cache_spec).
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assert spec.sliding_window == cdiv(750, 4) + 1 == 189
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assert (
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spec.max_admission_blocks_per_request(
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max_in_flight_tokens=1,
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max_model_len=vllm_config.model_config.max_model_len,
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)
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== 13
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)
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@pytest.fixture
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def audio_assets() -> list[AudioAsset]:
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return [AudioAsset("mary_had_lamb"), AudioAsset("winning_call")]
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@pytest.fixture
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def tokenizer() -> MistralTokenizer:
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return MistralTokenizer.from_hf_hub(MODEL_NAME)
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@pytest_asyncio.fixture
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async def async_engine():
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gpu_memory_utilization = ENGINE_CONFIG.get("gpu_memory_utilization", 0.9)
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from vllm.platforms import current_platform
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if current_platform.is_rocm():
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from tests.utils import wait_for_rocm_memory_to_settle
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wait_for_rocm_memory_to_settle(threshold_ratio=1.0 - gpu_memory_utilization)
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engine_args = AsyncEngineArgs(**ENGINE_CONFIG)
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llm = AsyncLLM.from_engine_args(engine_args)
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try:
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yield llm
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finally:
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shutdown_timeout = 60.0 if current_platform.is_rocm() else None
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llm.shutdown(timeout=shutdown_timeout)
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del llm
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import torch
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torch._dynamo.reset()
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from vllm.distributed import cleanup_dist_env_and_memory
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cleanup_dist_env_and_memory()
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from tests.utils import wait_for_rocm_memory_to_settle
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wait_for_rocm_memory_to_settle(threshold_ratio=1.0 - gpu_memory_utilization)
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def test_voxtral_realtime_forward(audio_assets, tokenizer, vllm_runner, monkeypatch):
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monkeypatch.setenv("VLLM_ENABLE_V1_MULTIPROCESSING", "0")
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vllm_kwargs = {**ENGINE_CONFIG}
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vllm_kwargs["model_name"] = vllm_kwargs.pop("model")
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with vllm_runner(**vllm_kwargs) as vllm_model:
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assert_encoder_kv_cache_spec(vllm_model.llm)
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audio_config = tokenizer.instruct_tokenizer.tokenizer.audio
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def from_file(file_path: str):
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audio = Audio.from_file(file_path, strict=False)
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req = TranscriptionRequest(
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audio=audio.to_base64(audio.format),
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streaming=StreamingMode.OFFLINE,
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language=None,
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)
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tokenized = tokenizer.instruct_tokenizer.encode_transcription(req)
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return (tokenized.tokens, tokenized.audios[0].audio_array)
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tokenized_list = [
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from_file(audio_asset.get_local_path()) for audio_asset in audio_assets
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]
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inputs = []
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sampling_params = []
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for tokens, audio_array in tokenized_list:
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num_samples = audio_array.shape[0]
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max_tokens = audio_config.num_audio_tokens(num_samples) - len(tokens) - 1
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sampling_params.append(
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SamplingParams(temperature=0.0, max_tokens=max_tokens)
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)
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input_dict = {
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"multi_modal_data": {"audio": [(audio_array, None)]},
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"prompt_token_ids": tokens,
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}
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inputs.append(input_dict)
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outputs = vllm_model.llm.generate(
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inputs,
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sampling_params=sampling_params,
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)
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texts = _normalize([out.outputs[0].text for out in outputs])
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for i, (got, expected) in enumerate(zip(texts, EXPECTED_TEXT)):
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assert got == expected, (
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f"Output mismatch at index {i}:\n"
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f" got: {got!r}\n"
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f" expected: {expected!r}"
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)
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@pytest.mark.asyncio
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async def test_voxtral_realtime_generator(audio_assets, tokenizer, async_engine):
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# Lazy import to avoid CUDA-reinitialization error
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from vllm.model_executor.models.voxtral_realtime import VoxtralRealtimeBuffer
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sampling_params = SamplingParams(temperature=0.0, max_tokens=1)
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audio_config = tokenizer.instruct_tokenizer.audio_encoder.audio_config
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output_tokens_list = []
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for i, audio_asset in enumerate(audio_assets):
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output_tokens = []
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audio = Audio.from_file(audio_asset.get_local_path(), strict=False)
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req = TranscriptionRequest(
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streaming=StreamingMode.OFFLINE,
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audio=audio.to_base64(audio.format),
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language=None,
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)
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audio_enc = tokenizer.encode_transcription(req)
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buffer = VoxtralRealtimeBuffer(audio_config, audio_enc.tokens)
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await buffer.append_audio(audio_enc.audios[0].audio_array)
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await buffer.append_audio(None)
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request_id = f"session-{i}"
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async for resp in async_engine.generate(
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prompt=buffer.get_input_stream(),
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sampling_params=sampling_params,
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request_id=request_id,
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):
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tokens = resp.outputs[0].token_ids[-1:]
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output_tokens.extend(tokens)
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await buffer.append_tokens(tokens)
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output_tokens_list.append(output_tokens)
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texts = _normalize(
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[
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tokenizer.decode(
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output_tokens, special_token_policy=SpecialTokenPolicy.IGNORE
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)
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for output_tokens in output_tokens_list
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]
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)
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for i, (got, expected) in enumerate(zip(texts, EXPECTED_TEXT)):
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assert got == expected, (
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f"Output mismatch at index {i}:\n"
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f" got: {got!r}\n"
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f" expected: {expected!r}"
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)
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