[Frontend] Add /v1/audio/transcriptions OpenAI API endpoint (#12909)

.buildkite/test-pipeline.yaml

@@ -117,7 +117,7 @@ steps:
   - pytest -v -s entrypoints/llm/test_generate.py # it needs a clean process
   - pytest -v -s entrypoints/llm/test_generate_multiple_loras.py # it needs a clean process
   - pytest -v -s entrypoints/llm/test_guided_generate.py # it needs a clean process
-  - pytest -v -s entrypoints/openai --ignore=entrypoints/openai/test_oot_registration.py
+  - pytest -v -s entrypoints/openai --ignore=entrypoints/openai/test_oot_registration.py --ignore=entrypoints/openai/correctness/
   - pytest -v -s entrypoints/test_chat_utils.py
   - pytest -v -s entrypoints/offline_mode # Needs to avoid interference with other tests
 
@@ -205,7 +205,7 @@ steps:
     - VLLM_USE_V1=1 pytest -v -s v1/e2e
     # Integration test for streaming correctness (requires special branch).
     - pip install -U git+https://github.com/robertgshaw2-neuralmagic/lm-evaluation-harness.git@streaming-api
-    - pytest -v -s entrypoints/openai/test_accuracy.py::test_lm_eval_accuracy_v1_engine
+    - pytest -v -s entrypoints/openai/correctness/test_lmeval.py::test_lm_eval_accuracy_v1_engine
 
 - label: Examples Test # 25min
   working_dir: "/vllm-workspace/examples"
@@ -339,6 +339,14 @@ steps:
   - export VLLM_WORKER_MULTIPROC_METHOD=spawn
   - bash ./run-tests.sh -c configs/models-small.txt -t 1
 
+- label: OpenAI API correctness
+  source_file_dependencies:
+  - csrc/
+  - vllm/entrypoints/openai/
+  - vllm/model_executor/models/whisper.py
+  commands: # LMEval+Transcription WER check
+  - pytest -s entrypoints/openai/correctness/
+
 - label: Encoder Decoder tests # 5min
   source_file_dependencies:
   - vllm/

docs/source/serving/openai_compatible_server.md

@@ -41,6 +41,8 @@ We currently support the following OpenAI APIs:
   - *Note: `parallel_tool_calls` and `user` parameters are ignored.*
 - [Embeddings API](#embeddings-api) (`/v1/embeddings`)
   - Only applicable to [embedding models](../models/pooling_models.md) (`--task embed`).
+- [Transcriptions API](#transcriptions-api) (`/v1/audio/transcriptions`)
+  - Only applicable to Automatic Speech Recognition (ASR) models (OpenAI Whisper) (`--task generate`).
 
 In addition, we have the following custom APIs:
 
@@ -296,6 +298,17 @@ For chat-like input (i.e. if `messages` is passed), these extra parameters are s
 :end-before: end-chat-embedding-extra-params
 :::
 
+(transcriptions-api)=
+
+### Transcriptions API
+
+Our Transcriptions API is compatible with [OpenAI's Transcriptions API](https://platform.openai.com/docs/api-reference/audio/createTranscription);
+you can use the [official OpenAI Python client](https://github.com/openai/openai-python) to interact with it.
+
+<!-- TODO: api enforced limits + uploading audios -->
+
+Code example: <gh-file:examples/online_serving/openai_transcription_client.py>
+
 (tokenizer-api)=
 
 ### Tokenizer API

examples/online_serving/openai_transcription_client.py

@@ -0,0 +1,23 @@
+# SPDX-License-Identifier: Apache-2.0
+from openai import OpenAI
+
+from vllm.assets.audio import AudioAsset
+
+mary_had_lamb = AudioAsset('mary_had_lamb').get_local_path()
+winning_call = AudioAsset('winning_call').get_local_path()
+
+# Modify OpenAI's API key and API base to use vLLM's API server.
+openai_api_key = "EMPTY"
+openai_api_base = "http://localhost:8000/v1"
+client = OpenAI(
+    api_key=openai_api_key,
+    base_url=openai_api_base,
+)
+with open(str(mary_had_lamb), "rb") as f:
+    transcription = client.audio.transcriptions.create(
+        file=f,
+        model="openai/whisper-large-v3",
+        language="en",
+        response_format="text",
+        temperature=0.0)
+    print("transcription result:", transcription)

requirements-common.txt

@@ -8,12 +8,11 @@ py-cpuinfo
 transformers >= 4.48.2  # Required for Bamba model and Transformers backend.
 tokenizers >= 0.19.1  # Required for Llama 3.
 protobuf # Required by LlamaTokenizer.
-fastapi >= 0.107.0, < 0.113.0; python_version < '3.9'
-fastapi >= 0.107.0, != 0.113.*, != 0.114.0; python_version >= '3.9'
+fastapi[standard] >= 0.107.0, < 0.113.0; python_version < '3.9'
+fastapi[standard]  >= 0.107.0, != 0.113.*, != 0.114.0; python_version >= '3.9'
 aiohttp
 openai >= 1.52.0 # Ensure modern openai package (ensure types module present and max_completion_tokens field support)
-uvicorn[standard]
-pydantic >= 2.9  # Required for fastapi >= 0.113.0
+pydantic >= 2.9
 prometheus_client >= 0.18.0
 pillow  # Required for image processing
 prometheus-fastapi-instrumentator >= 7.0.0

requirements-test.in

@@ -19,6 +19,7 @@ pqdm
 ray[adag]==2.40.0
 sentence-transformers # required for embedding tests
 soundfile # required for audio tests
+jiwer # required for audio tests
 timm # required for internvl test
 torch==2.5.1
 torchaudio==2.5.1

requirements-test.txt

@@ -66,6 +66,7 @@ charset-normalizer==3.4.0
 click==8.1.7
     # via
     #   black
+    #   jiwer
     #   nltk
     #   ray
 colorama==0.4.6
@@ -187,6 +188,8 @@ jinja2==3.1.4
     # via
     #   datamodel-code-generator
     #   torch
+jiwer==3.0.5
+    # via -r requirements-test.in
 jmespath==1.0.1
     # via
     #   boto3
@@ -470,6 +473,8 @@ pyyaml==6.0.2
     #   timm
     #   transformers
     #   vocos
+rapidfuzz==3.12.1
+    # via jiwer
 ray[adag]==2.40.0
     # via -r requirements-test.in
 redis==5.2.0

tests/entrypoints/openai/correctness/test_lmeval.py

@@ -13,7 +13,7 @@ import pytest
 
 from vllm.platforms import current_platform
 
-from ...utils import RemoteOpenAIServer
+from ....utils import RemoteOpenAIServer
 
 MODEL_NAME = "Qwen/Qwen2-1.5B-Instruct"
 NUM_CONCURRENT = 500

tests/entrypoints/openai/correctness/test_transcription_api_correctness.py

@@ -0,0 +1,166 @@
+# SPDX-License-Identifier: Apache-2.0
+"""
+Evaluate Transcription API correctness by computing Word Error Rate (WER)
+on a given ASR dataset. When provided, it will also compare the WER against
+a baseline.
+This simulates real work usage of the API and makes sure that the frontend and
+AsyncLLMEngine are working correctly.
+"""
+import asyncio
+import io
+import time
+from statistics import mean, median
+from typing import List
+
+import librosa
+import pytest
+import soundfile
+import torch
+from datasets import load_dataset
+from evaluate import load
+from transformers import AutoTokenizer
+
+from ....utils import RemoteOpenAIServer
+
+
+def to_bytes(y, sr):
+    buffer = io.BytesIO()
+    soundfile.write(buffer, y, sr, format="WAV")
+    buffer.seek(0)
+    return buffer
+
+
+async def transcribe_audio(client, tokenizer, y, sr):
+    # Send loaded audio directly instead of loading from disk,
+    # dont account for that time though
+    with to_bytes(y, sr) as f:
+        start_time = time.perf_counter()
+        transcription = await client.audio.transcriptions.create(
+            file=f,
+            model=tokenizer.name_or_path,
+            language="en",
+            temperature=0.0,
+        )
+        end_time = time.perf_counter()
+        # NOTE there's no streaming in transcriptions, can't measure ttft
+    latency = end_time - start_time
+    num_output_tokens = len(
+        tokenizer(transcription.text, add_special_tokens=False).input_ids)
+    return latency, num_output_tokens, transcription.text
+
+
+async def bound_transcribe(model_name, sem, client, audio, reference):
+    tokenizer = AutoTokenizer.from_pretrained(model_name)
+    # Use semaphore to limit concurrent requests.
+    async with sem:
+        result = await transcribe_audio(client, tokenizer, *audio)
+        # Normalize *english* output/reference for evaluation.
+        out = tokenizer.normalize(result[2])
+        ref = tokenizer.normalize(reference)
+        return result[:2] + (out, ref)
+
+
+async def process_dataset(model, client, data, concurrent_request):
+    sem = asyncio.Semaphore(concurrent_request)
+
+    # Warmup call as the first `librosa.load` server-side is quite slow.
+    audio, sr = data[0]["audio"]["array"], data[0]["audio"]["sampling_rate"]
+    _ = await bound_transcribe(model, sem, client, (audio, sr), "")
+
+    tasks: List[asyncio.Task] = []
+    for sample in data:
+        audio, sr = sample["audio"]["array"], sample["audio"]["sampling_rate"]
+        task = asyncio.create_task(
+            bound_transcribe(model, sem, client, (audio, sr), sample["text"]))
+        tasks.append(task)
+    return await asyncio.gather(*tasks)
+
+
+def print_performance_metrics(results, total_time):
+    latencies = [res[0] for res in results]
+    total_tokens = sum([res[1] for res in results])
+
+    total = len(results)
+    print(f"Total Requests: {total}")
+    print(f"Successful Requests: {len(latencies)}")
+    print(f"Average Latency: {mean(latencies):.4f} seconds")
+    print(f"Median Latency: {median(latencies):.4f} seconds")
+    perc = sorted(latencies)[int(len(latencies) * 0.95) - 1]
+    print(f"95th Percentile Latency: {perc:.4f} seconds")
+    # Throughput
+    req_throughput = len(latencies) / total_time
+    print(f"Estimated req_Throughput: {req_throughput:.2f} requests/s")
+    throughput = total_tokens / total_time
+    print(f"Estimated Throughput: {throughput:.2f} tok/s")
+
+
+def add_duration(sample):
+    y, sr = sample['audio']["array"], sample['audio']["sampling_rate"]
+    sample['duration_ms'] = librosa.get_duration(y=y, sr=sr) * 1000
+    return sample
+
+
+def load_hf_dataset(dataset_repo: str, split='validation', **hf_kwargs):
+    ## Load and filter the dataset
+    dataset = load_dataset(dataset_repo, split=split, **hf_kwargs)
+    if 'duration_ms' not in dataset[0]:
+        # compute duration to filter
+        dataset = dataset.map(add_duration)
+
+    # Whisper max supported duration
+    dataset = dataset.filter(lambda example: example['duration_ms'] < 30000)
+    return dataset
+
+
+def run_evaluation(model: str,
+                   client,
+                   dataset,
+                   max_concurrent_reqs: int,
+                   n_examples: int = -1,
+                   print_metrics: bool = True):
+    if n_examples > 0:
+        dataset = dataset.select(range(n_examples))
+    start = time.perf_counter()
+    results = asyncio.run(
+        process_dataset(model, client, dataset, max_concurrent_reqs))
+    end = time.perf_counter()
+    total_time = end - start
+    print(f"Total Test Time: {total_time:.4f} seconds")
+    if print_metrics:
+        print_performance_metrics(results, total_time)
+    # Compute WER
+    predictions = [res[2] for res in results]
+    references = [res[3] for res in results]
+    wer = load("wer")
+    wer_score = 100 * wer.compute(references=references,
+                                  predictions=predictions)
+    print("WER:", wer_score)
+    return wer_score
+
+
+# alternatives "openai/whisper-large-v2", "openai/whisper-large-v3-turbo"..
+@pytest.mark.parametrize("model_name", ["openai/whisper-large-v3"])
+# Original dataset is 20GB+ in size, hence we use a pre-filtered slice.
+@pytest.mark.parametrize(
+    "dataset_repo", ["D4nt3/esb-datasets-earnings22-validation-tiny-filtered"])
+# NOTE: Expected WER measured with equivalent hf.transformers args:
+# whisper-large-v3 + esb-datasets-earnings22-validation-tiny-filtered.
+@pytest.mark.parametrize("expected_wer", [12.744980])
+def test_wer_correctness(model_name,
+                         dataset_repo,
+                         expected_wer,
+                         n_examples=-1,
+                         max_concurrent_request=None):
+    with RemoteOpenAIServer(model_name, ['--enforce-eager']) as remote_server:
+        dataset = load_hf_dataset(dataset_repo)
+
+        if not max_concurrent_request:
+            # No max concurrency
+            max_concurrent_request = n_examples if n_examples > 0\
+                else len(dataset)
+
+        client = remote_server.get_async_client()
+        wer = run_evaluation(model_name, client, dataset,
+                             max_concurrent_request, n_examples)
+        if expected_wer:
+            torch.testing.assert_close(wer, expected_wer, atol=1e-1, rtol=1e-2)

tests/entrypoints/openai/test_transcription_validation.py

@@ -0,0 +1,122 @@
+# SPDX-License-Identifier: Apache-2.0
+
+# imports for guided decoding tests
+import io
+import json
+
+import librosa
+import numpy as np
+import openai
+import pytest
+import soundfile as sf
+
+from vllm.assets.audio import AudioAsset
+
+from ...utils import RemoteOpenAIServer
+
+
+@pytest.fixture
+def mary_had_lamb():
+    path = AudioAsset('mary_had_lamb').get_local_path()
+    with open(str(path), "rb") as f:
+        yield f
+
+
+@pytest.fixture
+def winning_call():
+    path = AudioAsset('winning_call').get_local_path()
+    with open(str(path), "rb") as f:
+        yield f
+
+
+@pytest.mark.asyncio
+async def test_basic_audio(mary_had_lamb):
+    model_name = "openai/whisper-large-v3-turbo"
+    server_args = ["--enforce-eager"]
+    # Based on https://github.com/openai/openai-cookbook/blob/main/examples/Whisper_prompting_guide.ipynb.
+    prompt = "THE FIRST WORDS I SPOKE"
+    with RemoteOpenAIServer(model_name, server_args) as remote_server:
+        client = remote_server.get_async_client()
+        transcription = await client.audio.transcriptions.create(
+            model=model_name,
+            file=mary_had_lamb,
+            language="en",
+            response_format="text",
+            temperature=0.0)
+        out = json.loads(transcription)['text']
+        assert "Mary had a little lamb," in out
+        # This should "force" whisper to continue prompt in all caps
+        transcription_wprompt = await client.audio.transcriptions.create(
+            model=model_name,
+            file=mary_had_lamb,
+            language="en",
+            response_format="text",
+            prompt=prompt,
+            temperature=0.0)
+        out_capital = json.loads(transcription_wprompt)['text']
+        assert prompt not in out_capital
+
+
+@pytest.mark.asyncio
+async def test_bad_requests(mary_had_lamb):
+    model_name = "openai/whisper-small"
+    server_args = ["--enforce-eager"]
+    with RemoteOpenAIServer(model_name, server_args) as remote_server:
+        client = remote_server.get_async_client()
+
+        # invalid language
+        with pytest.raises(openai.BadRequestError):
+            await client.audio.transcriptions.create(model=model_name,
+                                                     file=mary_had_lamb,
+                                                     language="hh",
+                                                     temperature=0.0)
+
+        # Expect audio too long: repeat the timeseries
+        mary_had_lamb.seek(0)
+        audio, sr = librosa.load(mary_had_lamb)
+        repeated_audio = np.tile(audio, 10)
+        # Repeated audio to buffer
+        buffer = io.BytesIO()
+        sf.write(buffer, repeated_audio, sr, format='WAV')
+        buffer.seek(0)
+        with pytest.raises(openai.BadRequestError):
+            await client.audio.transcriptions.create(model=model_name,
+                                                     file=buffer,
+                                                     language="en",
+                                                     temperature=0.0)
+
+
+@pytest.mark.asyncio
+async def test_non_asr_model(winning_call):
+    # text to text model
+    model_name = "JackFram/llama-68m"
+    server_args = ["--enforce-eager"]
+    with RemoteOpenAIServer(model_name, server_args) as remote_server:
+        client = remote_server.get_async_client()
+        res = await client.audio.transcriptions.create(model=model_name,
+                                                       file=winning_call,
+                                                       language="en",
+                                                       temperature=0.0)
+        assert res.code == 400 and not res.text
+        assert res.message == "The model does not support Transcriptions API"
+
+
+@pytest.mark.asyncio
+async def test_completion_endpoints():
+    # text to text model
+    model_name = "openai/whisper-small"
+    server_args = ["--enforce-eager"]
+    with RemoteOpenAIServer(model_name, server_args) as remote_server:
+        client = remote_server.get_async_client()
+        res = await client.chat.completions.create(
+            model=model_name,
+            messages=[{
+                "role": "system",
+                "content": "You are a helpful assistant."
+            }])
+        assert res.code == 400
+        assert res.message == "The model does not support Chat Completions API"
+
+        res = await client.completions.create(model=model_name, prompt="Hello")
+        assert res.code == 400
+        assert res.message == "The model does not support Completions API"

tests/test_config.py

@@ -17,6 +17,7 @@ from vllm.platforms import current_platform
         ("jason9693/Qwen2.5-1.5B-apeach", "pooling", "classify"),
         ("cross-encoder/ms-marco-MiniLM-L-6-v2", "pooling", "score"),
         ("Qwen/Qwen2.5-Math-RM-72B", "pooling", "reward"),
+        ("openai/whisper-small", "transcription", "transcription"),
     ],
 )
 def test_auto_task(model_id, expected_runner_type, expected_task):

vllm/assets/audio.py

@@ -1,6 +1,7 @@
 # SPDX-License-Identifier: Apache-2.0
 
 from dataclasses import dataclass
+from pathlib import Path
 from typing import Literal
 from urllib.parse import urljoin
 
@@ -28,6 +29,10 @@ class AudioAsset:
                                             s3_prefix=ASSET_DIR)
         return librosa.load(audio_path, sr=None)
 
+    def get_local_path(self) -> Path:
+        return get_vllm_public_assets(filename=f"{self.name}.ogg",
+                                      s3_prefix=ASSET_DIR)
+
     @property
     def url(self) -> str:
         return urljoin(VLLM_S3_BUCKET_URL, f"{ASSET_DIR}/{self.name}.ogg")

vllm/config.py

@@ -54,17 +54,18 @@ _POOLING_MODEL_MAX_NUM_BATCHED_TOKENS = 32768
 _MULTIMODAL_MODEL_MAX_NUM_BATCHED_TOKENS = 5120
 
 TaskOption = Literal["auto", "generate", "embedding", "embed", "classify",
-                     "score", "reward"]
+                     "score", "reward", "transcription"]
 
 _ResolvedTask = Literal["generate", "embed", "classify", "score", "reward",
-                        "draft"]
+                        "draft", "transcription"]
 
-RunnerType = Literal["generate", "pooling", "draft"]
+RunnerType = Literal["generate", "pooling", "draft", "transcription"]
 
 _RUNNER_TASKS: Dict[RunnerType, List[_ResolvedTask]] = {
     "generate": ["generate"],
     "pooling": ["embed", "classify", "score", "reward"],
     "draft": ["draft"],
+    "transcription": ["transcription"],
 }
 
 _TASK_RUNNER: Dict[_ResolvedTask, RunnerType] = {
@@ -484,6 +485,8 @@ class ModelConfig:
             return "embed"
         if ModelRegistry.is_cross_encoder_model(architectures):
             return "score"
+        if ModelRegistry.is_transcription_model(architectures):
+            return "transcription"
 
         suffix_to_preferred_task: List[Tuple[str, _ResolvedTask]] = [
             # Other models follow this pattern
@@ -516,6 +519,8 @@ class ModelConfig:
         runner_support: Dict[RunnerType, bool] = {
             # NOTE: Listed from highest to lowest priority,
             # in case the model supports multiple of them
+            "transcription":
+            ModelRegistry.is_transcription_model(architectures),
             "generate": ModelRegistry.is_text_generation_model(architectures),
             "pooling": ModelRegistry.is_pooling_model(architectures),
         }

vllm/entrypoints/openai/api_server.py

@@ -16,10 +16,10 @@ from argparse import Namespace
 from contextlib import asynccontextmanager
 from functools import partial
 from http import HTTPStatus
-from typing import AsyncIterator, Dict, Optional, Set, Tuple, Union
+from typing import Annotated, AsyncIterator, Dict, Optional, Set, Tuple, Union
 
 import uvloop
-from fastapi import APIRouter, Depends, FastAPI, HTTPException, Request
+from fastapi import APIRouter, Depends, FastAPI, Form, HTTPException, Request
 from fastapi.exceptions import RequestValidationError
 from fastapi.middleware.cors import CORSMiddleware
 from fastapi.responses import JSONResponse, Response, StreamingResponse
@@ -61,6 +61,8 @@ from vllm.entrypoints.openai.protocol import (ChatCompletionRequest,
                                               ScoreRequest, ScoreResponse,
                                               TokenizeRequest,
                                               TokenizeResponse,
+                                              TranscriptionRequest,
+                                              TranscriptionResponse,
                                               UnloadLoraAdapterRequest)
 from vllm.entrypoints.openai.reasoning_parsers import ReasoningParserManager
 # yapf: enable
@@ -75,6 +77,8 @@ from vllm.entrypoints.openai.serving_rerank import JinaAIServingRerank
 from vllm.entrypoints.openai.serving_score import OpenAIServingScores
 from vllm.entrypoints.openai.serving_tokenization import (
     OpenAIServingTokenization)
+from vllm.entrypoints.openai.serving_transcription import (
+    OpenAIServingTranscription)
 from vllm.entrypoints.openai.tool_parsers import ToolParserManager
 from vllm.entrypoints.utils import with_cancellation
 from vllm.logger import init_logger
@@ -327,6 +331,10 @@ def tokenization(request: Request) -> OpenAIServingTokenization:
     return request.app.state.openai_serving_tokenization
 
 
+def transcription(request: Request) -> OpenAIServingTranscription:
+    return request.app.state.openai_serving_transcription
+
+
 def engine_client(request: Request) -> EngineClient:
     return request.app.state.engine_client
 
@@ -520,6 +528,31 @@ async def create_score_v1(request: ScoreRequest, raw_request: Request):
     return await create_score(request, raw_request)
 
 
+@router.post("/v1/audio/transcriptions")
+@with_cancellation
+async def create_transcriptions(request: Annotated[TranscriptionRequest,
+                                                   Form()],
+                                raw_request: Request):
+
+    handler = transcription(raw_request)
+    if handler is None:
+        return base(raw_request).create_error_response(
+            message="The model does not support Transcriptions API")
+
+    audio_data = await request.file.read()
+    generator = await handler.create_transcription(audio_data, request,
+                                                   raw_request)
+
+    if isinstance(generator, ErrorResponse):
+        return JSONResponse(content=generator.model_dump(),
+                            status_code=generator.code)
+
+    elif isinstance(generator, TranscriptionResponse):
+        return JSONResponse(content=generator.model_dump())
+
+    return StreamingResponse(content=generator, media_type="text/event-stream")
+
+
 @router.post("/rerank", dependencies=[Depends(validate_json_request)])
 @with_cancellation
 async def do_rerank(request: RerankRequest, raw_request: Request):
@@ -832,6 +865,12 @@ async def init_app_state(
         chat_template=resolved_chat_template,
         chat_template_content_format=args.chat_template_content_format,
     )
+    state.openai_serving_transcription = OpenAIServingTranscription(
+        engine_client,
+        model_config,
+        state.openai_serving_models,
+        request_logger=request_logger,
+    ) if model_config.runner_type == "transcription" else None
     state.task = model_config.task
 
 

vllm/entrypoints/openai/protocol.py

@@ -8,9 +8,10 @@ from argparse import Namespace
 from typing import Any, ClassVar, Dict, List, Literal, Optional, Set, Union
 
 import torch
+from fastapi import UploadFile
 from pydantic import (BaseModel, ConfigDict, Field, TypeAdapter,
                       ValidationInfo, field_validator, model_validator)
-from typing_extensions import Annotated
+from typing_extensions import Annotated, TypeAlias
 
 from vllm.entrypoints.chat_utils import ChatCompletionMessageParam
 from vllm.logger import init_logger
@@ -1426,3 +1427,163 @@ class LoadLoraAdapterRequest(BaseModel):
 class UnloadLoraAdapterRequest(BaseModel):
     lora_name: str
     lora_int_id: Optional[int] = Field(default=None)
+
+
+## Protocols for Audio
+AudioResponseFormat: TypeAlias = Literal["json", "text", "srt", "verbose_json",
+                                         "vtt"]
+
+
+class TranscriptionRequest(OpenAIBaseModel):
+    # Ordered by official OpenAI API documentation
+    #https://platform.openai.com/docs/api-reference/audio/createTranscription
+
+    file: UploadFile
+    """
+    The audio file object (not file name) to transcribe, in one of these
+    formats: flac, mp3, mp4, mpeg, mpga, m4a, ogg, wav, or webm.
+    """
+
+    model: str
+    """ID of the model to use.
+    """
+
+    language: Optional[str] = None
+    """The language of the input audio.
+
+    Supplying the input language in
+    [ISO-639-1](https://en.wikipedia.org/wiki/List_of_ISO_639-1_codes) format
+    will improve accuracy and latency.
+    """
+
+    prompt: str = Field(default="")
+    """An optional text to guide the model's style or continue a previous audio
+    segment.
+
+    The [prompt](https://platform.openai.com/docs/guides/speech-to-text#prompting)
+    should match the audio language.
+    """
+
+    response_format: AudioResponseFormat = Field(default="json")
+    """
+    The format of the output, in one of these options: `json`, `text`, `srt`,
+    `verbose_json`, or `vtt`.
+    """
+
+    ## TODO (varun) : Support if set to 0, certain thresholds are met !!
+    temperature: float = Field(default=0.0)
+    """The sampling temperature, between 0 and 1.
+
+    Higher values like 0.8 will make the output more random, while lower values
+    like 0.2 will make it more focused / deterministic. If set to 0, the model
+    will use [log probability](https://en.wikipedia.org/wiki/Log_probability)
+    to automatically increase the temperature until certain thresholds are hit.
+    """
+
+    timestamp_granularities: List[Literal["word", "segment"]] = Field(
+        alias="timestamp_granularities[]", default=[])
+    """The timestamp granularities to populate for this transcription.
+
+    `response_format` must be set `verbose_json` to use timestamp granularities.
+    Either or both of these options are supported: `word`, or `segment`. Note:
+    There is no additional latency for segment timestamps, but generating word
+    timestamps incurs additional latency.
+    """
+
+    # Default sampling parameters for transcription requests.
+    _DEFAULT_SAMPLING_PARAMS: dict = {
+        "temperature": 0,
+    }
+
+    def to_sampling_params(
+            self,
+            default_max_tokens: int,
+            default_sampling_params: Optional[dict] = None) -> SamplingParams:
+        # TODO(#9845): remove max_tokens when field is removed from OpenAI API
+        max_tokens = default_max_tokens
+
+        if default_sampling_params is None:
+            default_sampling_params = {}
+        # Default parameters
+        if (temperature := self.temperature) is None:
+            temperature = default_sampling_params.get(
+                "temperature", self._DEFAULT_SAMPLING_PARAMS["temperature"])
+
+        return SamplingParams.from_optional(temperature=temperature,
+                                            max_tokens=max_tokens)
+
+
+# Transcription response objects
+class TranscriptionResponse(OpenAIBaseModel):
+    text: str
+    """The transcribed text."""
+
+
+class TranscriptionWord(OpenAIBaseModel):
+    end: float
+    """End time of the word in seconds."""
+
+    start: float
+    """Start time of the word in seconds."""
+
+    word: str
+    """The text content of the word."""
+
+
+class TranscriptionSegment(OpenAIBaseModel):
+    id: int
+    """Unique identifier of the segment."""
+
+    avg_logprob: float
+    """Average logprob of the segment.
+
+    If the value is lower than -1, consider the logprobs failed.
+    """
+
+    compression_ratio: float
+    """Compression ratio of the segment.
+
+    If the value is greater than 2.4, consider the compression failed.
+    """
+
+    end: float
+    """End time of the segment in seconds."""
+
+    no_speech_prob: float
+    """Probability of no speech in the segment.
+
+    If the value is higher than 1.0 and the `avg_logprob` is below -1, consider
+    this segment silent.
+    """
+
+    seek: int
+    """Seek offset of the segment."""
+
+    start: float
+    """Start time of the segment in seconds."""
+
+    temperature: float
+    """Temperature parameter used for generating the segment."""
+
+    text: str
+    """Text content of the segment."""
+
+    tokens: List[int]
+    """Array of token IDs for the text content."""
+
+
+class TranscriptionResponseVerbose(OpenAIBaseModel):
+    duration: str
+    """The duration of the input audio."""
+
+    language: str
+    """The language of the input audio."""
+
+    text: str
+    """The transcribed text."""
+
+    segments: Optional[List[TranscriptionSegment]] = None
+    """Segments of the transcribed text and their corresponding details."""
+
+    words: Optional[List[TranscriptionWord]] = None
+    """Extracted words and their corresponding timestamps."""

vllm/entrypoints/openai/serving_engine.py

@@ -31,7 +31,8 @@ from vllm.entrypoints.openai.protocol import (ChatCompletionRequest,
                                               ErrorResponse, RerankRequest,
                                               ScoreRequest,
                                               TokenizeChatRequest,
-                                              TokenizeCompletionRequest)
+                                              TokenizeCompletionRequest,
+                                              TranscriptionRequest)
 from vllm.entrypoints.openai.serving_models import OpenAIServingModels
 from vllm.entrypoints.openai.tool_parsers import ToolParser
 # yapf: enable
@@ -57,7 +58,8 @@ CompletionLikeRequest = Union[CompletionRequest, DetokenizeRequest,
 ChatLikeRequest = Union[ChatCompletionRequest, EmbeddingChatRequest,
                         TokenizeChatRequest]
 
-AnyRequest = Union[CompletionLikeRequest, ChatLikeRequest]
+AnyRequest = Union[CompletionLikeRequest, ChatLikeRequest,
+                   TranscriptionRequest]
 
 
 class TextTokensPrompt(TypedDict):

vllm/entrypoints/openai/serving_transcription.py

@@ -0,0 +1,305 @@
+# SPDX-License-Identifier: Apache-2.0
+import asyncio
+import io
+from typing import AsyncGenerator, Optional, Union, cast
+
+from fastapi import Request
+
+from vllm.config import ModelConfig
+from vllm.engine.protocol import EngineClient
+from vllm.entrypoints.logger import RequestLogger
+from vllm.entrypoints.openai.protocol import (ErrorResponse,
+                                              RequestResponseMetadata,
+                                              TranscriptionRequest,
+                                              TranscriptionResponse,
+                                              TranscriptionResponseVerbose)
+from vllm.entrypoints.openai.serving_engine import OpenAIServing
+from vllm.entrypoints.openai.serving_models import OpenAIServingModels
+from vllm.inputs.data import PromptType
+from vllm.logger import init_logger
+from vllm.outputs import RequestOutput
+from vllm.utils import PlaceholderModule
+
+try:
+    import librosa
+except ImportError:
+    librosa = PlaceholderModule("librosa")  # type: ignore[assignment]
+
+logger = init_logger(__name__)
+
+# From https://platform.openai.com/docs/guides/speech-to-text/supported-languages#supported-languages
+# TODO these configs should live somewhere with the model so we can support
+# additional ones
+
+ISO639_1_SUPPORTED_LANGS = {
+    "af": "Afrikaans",
+    "ar": "Arabic",
+    "hy": "Armenian",
+    "az": "Azerbaijani",
+    "be": "Belarusian",
+    "bs": "Bosnian",
+    "bg": "Bulgarian",
+    "ca": "Catalan",
+    "zh": "Chinese",
+    "hr": "Croatian",
+    "cs": "Czech",
+    "da": "Danish",
+    "nl": "Dutch",
+    "en": "English",
+    "et": "Estonian",
+    "fi": "Finnish",
+    "fr": "French",
+    "gl": "Galician",
+    "de": "German",
+    "el": "Greek",
+    "he": "Hebrew",
+    "hi": "Hindi",
+    "hu": "Hungarian",
+    "is": "Icelandic",
+    "id": "Indonesian",
+    "it": "Italian",
+    "ja": "Japanese",
+    "kn": "Kannada",
+    "kk": "Kazakh",
+    "ko": "Korean",
+    "lv": "Latvian",
+    "lt": "Lithuanian",
+    "mk": "Macedonian",
+    "ms": "Malay",
+    "mr": "Marathi",
+    "mi": "Maori",
+    "ne": "Nepali",
+    "no": "Norwegian",
+    "fa": "Persian",
+    "pl": "Polish",
+    "pt": "Portuguese",
+    "ro": "Romanian",
+    "ru": "Russian",
+    "sr": "Serbian",
+    "sk": "Slovak",
+    "sl": "Slovenian",
+    "es": "Spanish",
+    "sw": "Swahili",
+    "sv": "Swedish",
+    "tl": "Tagalog",
+    "ta": "Tamil",
+    "th": "Thai",
+    "tr": "Turkish",
+    "uk": "Ukrainian",
+    "ur": "Urdu",
+    "vi": "Vietnamese",
+    "cy": "Welsh"
+}
+ISO639_1_OTHER_LANGS = {
+    "lo": "Lao",
+    "jw": "Javanese",
+    "tk": "Turkmen",
+    "yi": "Yiddish",
+    "so": "Somali",
+    "bn": "Bengali",
+    "nn": "Norwegian Nynorsk",
+    "si": "Sinhala",
+    "yo": "Yoruba",
+    "sa": "Sanskrit",
+    "mi": "Māori",
+    "fo": "Faroese",  # codespell:ignore
+    "mt": "Maltese",
+    "tg": "Tajik",
+    "mg": "Malagasy",
+    "haw": "Hawaiian",
+    "km": "Khmer",
+    "br": "Breton",
+    "ps": "Pashto",
+    "ln": "Lingala",
+    "la": "Latin",
+    "ml": "Malayalam",
+    "sq": "Albanian",
+    "su": "Sundanese",
+    "eu": "Basque",
+    "ka": "Georgian",
+    "uz": "Uzbek",
+    "sn": "Shona",
+    "ht": "Haitian",
+    "as": "Assamese",
+    "mn": "Mongolian",
+    "te": "Telugu",
+    "pa": "Panjabi",
+    "tt": "Tatar",
+    "gu": "Gujarati",
+    "oc": "Occitan",
+    "ha": "Hausa",
+    "ba": "Bashkir",
+    "my": "Burmese",
+    "sd": "Sindhi",
+    "am": "Amharic",
+    "lb": "Luxembourgish",
+    "bo": "Tibetan"
+}
+
+# As per https://platform.openai.com/docs/guides/speech-to-text#overview.
+# TODO configurable
+MAX_AUDIO_CLIP_FILESIZE_MB = 25
+# TODO get from processor.feature_extractor.chunk_length
+MAX_AUDIO_CLIP_DURATION_S = 30
+
+
+class OpenAIServingTranscription(OpenAIServing):
+
+    def __init__(
+        self,
+        engine_client: EngineClient,
+        model_config: ModelConfig,
+        models: OpenAIServingModels,
+        *,
+        request_logger: Optional[RequestLogger],
+        return_tokens_as_token_ids: bool = False,
+    ):
+        super().__init__(engine_client=engine_client,
+                         model_config=model_config,
+                         models=models,
+                         request_logger=request_logger,
+                         return_tokens_as_token_ids=return_tokens_as_token_ids)
+
+        diff_sampling_param = self.model_config.get_diff_sampling_param()
+        if diff_sampling_param:
+            logger.info(
+                "Overwriting default completion sampling param with: %s",
+                diff_sampling_param)
+
+    async def _preprocess_transcription(
+        self,
+        request: TranscriptionRequest,
+        audio_data: bytes,
+    ) -> PromptType:
+        # Validate request
+        # TODO language should be optional and can be guessed.
+        # For now we default to en. See
+        # https://github.com/huggingface/transformers/blob/main/src/transformers/models/whisper/generation_whisper.py#L1520
+        lang_token = f"<|{request.language}|>" if request.language else "<|en|>"
+        if request.language:
+            if request.language in ISO639_1_SUPPORTED_LANGS:
+                pass
+            elif request.language in ISO639_1_OTHER_LANGS:
+                logger.warning(
+                    "The selected language %s has limited accuracy with"
+                    " reported WER>=0.5. Results may be less accurate "
+                    "for this choice.", request.language)
+            else:
+                raise ValueError(
+                    f"Unsupported language: {request.language}."
+                    "Language should be one of:" +
+                    f" {list(ISO639_1_SUPPORTED_LANGS.values())}" +
+                    f"or {list(ISO639_1_OTHER_LANGS.values())}")
+
+        if len(audio_data) / 1024**2 > MAX_AUDIO_CLIP_FILESIZE_MB:
+            raise ValueError("Maximum file size exceeded.")
+
+        with io.BytesIO(audio_data) as bytes_:
+            y, sr = librosa.load(bytes_)
+        if librosa.get_duration(y=y, sr=sr) > MAX_AUDIO_CLIP_DURATION_S:
+            raise ValueError(
+                f"Maximum clip duration ({MAX_AUDIO_CLIP_DURATION_S}s) "
+                "exceeded.")
+
+        prompt = {
+            "encoder_prompt": {
+                "prompt": "",
+                "multi_modal_data": {
+                    "audio": (y, sr),
+                },
+            },
+            "decoder_prompt":
+            f"<|startoftranscript|>{lang_token}<|transcribe|><|notimestamps|>{request.prompt}"
+        }
+        return cast(PromptType, prompt)
+
+    # TODO (varun) : Make verbose response work !
+    async def create_transcription(
+        self, audio_data: bytes, request: TranscriptionRequest,
+        raw_request: Request
+    ) -> Union[TranscriptionResponse, TranscriptionResponseVerbose,
+               ErrorResponse]:
+        """Transcription API similar to OpenAI's API.
+
+        See https://platform.openai.com/docs/api-reference/audio/createTranscription
+        for the API specification. This API mimics the OpenAI transcription API.
+        """
+        error_check_ret = await self._check_model(request)
+        if error_check_ret is not None:
+            return error_check_ret
+
+        # If the engine is dead, raise the engine's DEAD_ERROR.
+        # This is required for the streaming case, where we return a
+        # success status before we actually start generating text :).
+        if self.engine_client.errored:
+            raise self.engine_client.dead_error
+
+        if request.response_format not in ['text', 'json']:
+            return self.create_error_response(
+                "Currently only support response_format `text` or `json`")
+
+        # TODO cmpl->transcription?
+        request_id = f"cmpl-{self._base_request_id(raw_request)}"
+
+        request_metadata = RequestResponseMetadata(request_id=request_id)
+        if raw_request:
+            raw_request.state.request_metadata = request_metadata
+
+        try:
+            (
+                lora_request,
+                prompt_adapter_request,
+            ) = self._maybe_get_adapters(request)
+
+            if lora_request:
+                return self.create_error_response(
+                    "Currently do not support LoRA for Transcription.")
+            if prompt_adapter_request:
+                return self.create_error_response(
+                    "Currently do not support PromptAdapter for Transcription."
+                )
+
+            prompt = await self._preprocess_transcription(
+                request=request,
+                audio_data=audio_data,
+            )
+
+        except ValueError as e:
+            logger.exception("Error in preprocessing prompt inputs")
+            return self.create_error_response(str(e))
+
+        result_generator: Optional[AsyncGenerator[RequestOutput, None]] = None
+        try:
+            # TODO(rob): subtract len of tokenized prompt.
+            default_max_tokens = self.model_config.max_model_len
+            default_params = self.model_config.get_diff_sampling_param()
+            sampling_params = request.to_sampling_params(
+                default_max_tokens, default_params)
+
+            self._log_inputs(
+                request_id,
+                prompt['decoder_prompt'],  # type: ignore
+                params=sampling_params,
+                lora_request=None,
+                prompt_adapter_request=None)
+
+            result_generator = self.engine_client.generate(
+                prompt,
+                sampling_params,
+                request_id,
+            )
+        except ValueError as e:
+            # TODO: Use a vllm-specific Validation Error
+            return self.create_error_response(str(e))
+
+        # TODO(rob): figure out a way to pipe streaming in.
+        # Non-streaming response.
+        try:
+            async for op in result_generator:
+                result = op
+            return TranscriptionResponse(text=result.outputs[0].text)
+        except asyncio.CancelledError:
+            return self.create_error_response("Client disconnected")
+        except ValueError as e:
+            # TODO: Use a vllm-specific Validation Error
+            return self.create_error_response(str(e))

vllm/model_executor/models/interfaces.py

@@ -441,3 +441,30 @@ def supports_cross_encoding(
     model: Union[Type[object], object],
 ) -> Union[TypeIs[Type[SupportsCrossEncoding]], TypeIs[SupportsCrossEncoding]]:
     return is_pooling_model(model) and _supports_cross_encoding(model)
+
+
+@runtime_checkable
+class SupportsTranscription(Protocol):
+    """The interface required for all models that support transcription."""
+
+    supports_transcription: ClassVar[Literal[True]] = True
+
+
+@overload
+def supports_transcription(
+        model: Type[object]) -> TypeIs[Type[SupportsTranscription]]:
+    ...
+
+
+@overload
+def supports_transcription(model: object) -> TypeIs[SupportsTranscription]:
+    ...
+
+
+def supports_transcription(
+    model: Union[Type[object], object],
+) -> Union[TypeIs[Type[SupportsTranscription]], TypeIs[SupportsTranscription]]:
+    if isinstance(model, type):
+        return isinstance(model, SupportsTranscription)
+
+    return isinstance(model, SupportsTranscription)

vllm/model_executor/models/registry.py

@@ -22,7 +22,7 @@ from vllm.logger import init_logger
 
 from .interfaces import (has_inner_state, is_attention_free, is_hybrid,
                          supports_cross_encoding, supports_multimodal,
-                         supports_pp)
+                         supports_pp, supports_transcription)
 from .interfaces_base import is_text_generation_model
 
 logger = init_logger(__name__)
@@ -224,6 +224,7 @@ class _ModelInfo:
     has_inner_state: bool
     is_attention_free: bool
     is_hybrid: bool
+    supports_transcription: bool
 
     @staticmethod
     def from_model_cls(model: Type[nn.Module]) -> "_ModelInfo":
@@ -237,7 +238,7 @@ class _ModelInfo:
             has_inner_state=has_inner_state(model),
             is_attention_free=is_attention_free(model),
             is_hybrid=is_hybrid(model),
-        )
+            supports_transcription=supports_transcription(model))
 
 
 class _BaseRegisteredModel(ABC):
@@ -485,6 +486,13 @@ class _ModelRegistry:
         model_cls, _ = self.inspect_model_cls(architectures)
         return model_cls.is_hybrid
 
+    def is_transcription_model(
+        self,
+        architectures: Union[str, List[str]],
+    ) -> bool:
+        model_cls, _ = self.inspect_model_cls(architectures)
+        return model_cls.supports_transcription
+
 
 ModelRegistry = _ModelRegistry({
     model_arch:

vllm/model_executor/models/whisper.py

@@ -31,7 +31,7 @@ from vllm.multimodal.audio import resample_audio
 from vllm.sequence import SequenceData
 from vllm.transformers_utils.processor import cached_get_processor
 
-from .interfaces import SupportsMultiModal
+from .interfaces import SupportsMultiModal, SupportsTranscription
 from .utils import AutoWeightsLoader, WeightsMapper, make_layers
 
 logger = init_logger(__name__)
@@ -637,7 +637,8 @@ def input_mapper_for_whisper(
 @MULTIMODAL_REGISTRY.register_input_mapper("audio", input_mapper_for_whisper)
 @MULTIMODAL_REGISTRY.register_max_multimodal_tokens(
     "audio", get_max_whisper_audio_tokens)
-class WhisperForConditionalGeneration(nn.Module, SupportsMultiModal):
+class WhisperForConditionalGeneration(nn.Module, SupportsTranscription,
+                                      SupportsMultiModal):
     packed_modules_mapping = {
         "self_attn.qkv_proj": [
             "self_attn.q_proj",