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vllm.entrypoints.openai.speech_to_text

S module-attribute

S = TypeVar('S', bound=SpeechToTextSegment)

SpeechToTextResponse module-attribute

SpeechToTextResponse: TypeAlias = (
    TranscriptionResponse | TranslationResponse
)

SpeechToTextResponseVerbose module-attribute

SpeechToTextSegment module-attribute

SpeechToTextSegment: TypeAlias = (
    TranscriptionSegment | TranslationSegment
)

T module-attribute

T = TypeVar('T', bound=SpeechToTextResponse)

V module-attribute

logger module-attribute

logger = init_logger(__name__)

OpenAISpeechToText

Bases: OpenAIServing

Base class for speech-to-text operations like transcription and translation.

Source code in vllm/entrypoints/openai/speech_to_text.py
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class OpenAISpeechToText(OpenAIServing):
    """Base class for speech-to-text operations like transcription and
    translation."""

    def __init__(
        self,
        engine_client: EngineClient,
        models: OpenAIServingModels,
        *,
        request_logger: RequestLogger | None,
        return_tokens_as_token_ids: bool = False,
        task_type: Literal["transcribe", "translate"] = "transcribe",
        log_error_stack: bool = False,
        enable_force_include_usage: bool = False,
    ):
        super().__init__(
            engine_client=engine_client,
            models=models,
            request_logger=request_logger,
            return_tokens_as_token_ids=return_tokens_as_token_ids,
            log_error_stack=log_error_stack,
        )

        self.default_sampling_params = self.model_config.get_diff_sampling_param()
        self.task_type = task_type

        self.asr_config = self.model_cls.get_speech_to_text_config(
            self.model_config, task_type
        )

        self.enable_force_include_usage = enable_force_include_usage

        self.max_audio_filesize_mb = envs.VLLM_MAX_AUDIO_CLIP_FILESIZE_MB
        if self.model_cls.supports_segment_timestamp:
            self.tokenizer = cast(
                PreTrainedTokenizerBase,
                get_tokenizer(
                    tokenizer_name=self.model_config.tokenizer,
                    tokenizer_mode=self.model_config.tokenizer_mode,
                ),
            )

        if self.default_sampling_params:
            logger.info(
                "Overwriting default completion sampling param with: %s",
                self.default_sampling_params,
            )

    @cached_property
    def model_cls(self) -> type[SupportsTranscription]:
        from vllm.model_executor.model_loader import get_model_cls

        model_cls = get_model_cls(self.model_config)
        return cast(type[SupportsTranscription], model_cls)

    async def _preprocess_speech_to_text(
        self,
        request: SpeechToTextRequest,
        audio_data: bytes,
    ) -> tuple[list[PromptType], float]:
        # Validate request
        language = self.model_cls.validate_language(request.language)
        # Skip to_language validation to avoid extra logging for Whisper.
        to_language = (
            self.model_cls.validate_language(request.to_language)
            if request.to_language
            else None
        )

        if len(audio_data) / 1024**2 > self.max_audio_filesize_mb:
            raise ValueError("Maximum file size exceeded.")

        with io.BytesIO(audio_data) as bytes_:
            # NOTE resample to model SR here for efficiency. This is also a
            # pre-requisite for chunking, as it assumes Whisper SR.
            y, sr = librosa.load(bytes_, sr=self.asr_config.sample_rate)

        duration = librosa.get_duration(y=y, sr=sr)
        do_split_audio = (
            self.asr_config.allow_audio_chunking
            and duration > self.asr_config.max_audio_clip_s
        )
        chunks = [y] if not do_split_audio else self._split_audio(y, int(sr))
        prompts = []
        for chunk in chunks:
            # The model has control over the construction, as long as it
            # returns a valid PromptType.
            prompt = self.model_cls.get_generation_prompt(
                audio=chunk,
                stt_config=self.asr_config,
                model_config=self.model_config,
                language=language,
                task_type=self.task_type,
                request_prompt=request.prompt,
                to_language=to_language,
            )
            if request.response_format == "verbose_json":
                if not isinstance(prompt, dict):
                    raise ValueError(f"Expected prompt to be a dict,got {type(prompt)}")
                prompt_dict = cast(dict, prompt)
                decoder_prompt = prompt.get("decoder_prompt")
                if not isinstance(decoder_prompt, str):
                    raise ValueError(
                        f"Expected decoder_prompt to bestr, got {type(decoder_prompt)}"
                    )
                prompt_dict["decoder_prompt"] = decoder_prompt.replace(
                    "<|notimestamps|>", "<|0.00|>"
                )
            prompts.append(prompt)
        return prompts, duration

    def _get_verbose_segments(
        self,
        tokens: tuple,
        request: SpeechToTextRequest,
        segment_class: type[SpeechToTextSegment],
        start_time: float = 0,
    ) -> list[SpeechToTextSegment]:
        """
        Convert tokens to verbose segments.

        This method expects the model to produce
        timestamps as tokens (similar to Whisper).
        If the tokens do not include timestamp information,
        the segments may not be generated correctly.

        Note: Fields like avg_logprob, compression_ratio,
        and no_speech_prob are not supported
        in this implementation and will be None. See docs for details.
        """
        BASE_OFFSET = 0.02
        init_token = self.tokenizer.encode("<|0.00|>", add_special_tokens=False)[0]
        if tokens[-1] == self.tokenizer.eos_token_id:
            tokens = tokens[:-1]

        tokens_with_start = (init_token,) + tokens
        segments: list[SpeechToTextSegment] = []
        last_timestamp_start = 0

        if tokens_with_start[-2] < init_token and tokens_with_start[-1] >= init_token:
            tokens_with_start = tokens_with_start + (tokens_with_start[-1],)
        for idx, token in enumerate(tokens_with_start):
            # Timestamp tokens (e.g., <|0.00|>) are assumed to be sorted.
            # If the ordering is violated, this slicing may produce incorrect results.
            if (
                token >= init_token
                and idx != 0
                and tokens_with_start[idx - 1] >= init_token
            ):
                sliced_timestamp_tokens = tokens_with_start[last_timestamp_start:idx]
                start_timestamp = sliced_timestamp_tokens[0] - init_token
                end_timestamp = sliced_timestamp_tokens[-1] - init_token

                casting_segment = cast(
                    SpeechToTextSegment,
                    segment_class(
                        id=len(segments),
                        seek=start_time,
                        start=start_time + BASE_OFFSET * start_timestamp,
                        end=start_time + BASE_OFFSET * end_timestamp,
                        temperature=request.temperature,
                        text=self.tokenizer.decode(sliced_timestamp_tokens[1:-1]),
                        tokens=sliced_timestamp_tokens[1:-1],
                    ),
                )
                segments.append(casting_segment)
                last_timestamp_start = idx
        return segments

    async def _create_speech_to_text(
        self,
        audio_data: bytes,
        request: SpeechToTextRequest,
        raw_request: Request,
        response_class: type[T | V],
        stream_generator_method: Callable[..., AsyncGenerator[str, None]],
    ) -> T | V | AsyncGenerator[str, None] | ErrorResponse:
        """Base method for speech-to-text operations like transcription and
        translation."""
        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", "verbose_json"]:
            return self.create_error_response(
                ("Currently only support response_format")
                + ("`text`, `json` or `verbose_json`")
            )

        if (
            request.response_format == "verbose_json"
            and not self.model_cls.supports_segment_timestamp
        ):
            return self.create_error_response(
                f"Currently do not support verbose_json for {request.model}"
            )

        if request.response_format == "verbose_json" and request.stream:
            return self.create_error_response(
                "verbose_json format doesn't support streaming case"
            )
        request_id = f"{self.task_type}-{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 = self._maybe_get_adapters(request)

            prompts, duration_s = await self._preprocess_speech_to_text(
                request=request,
                audio_data=audio_data,
            )

        except ValueError as e:
            logger.exception("Error in preprocessing prompt inputs")
            return self.create_error_response(str(e))

        list_result_generator: list[AsyncGenerator[RequestOutput, None]] | None = None
        try:
            # Unlike most decoder-only models, whisper generation length is not
            # constrained by the size of the input audio, which is mapped to a
            # fixed-size log-mel-spectogram.
            default_max_tokens = self.model_config.max_model_len
            sampling_params = request.to_sampling_params(
                default_max_tokens, self.default_sampling_params
            )

            self._log_inputs(
                request_id,
                # It will not display special tokens like <|startoftranscript|>
                request.prompt,
                params=sampling_params,
                lora_request=lora_request,
            )

            list_result_generator = [
                self.engine_client.generate(
                    prompt,
                    sampling_params,
                    f"{request_id}_{i}",
                    lora_request=lora_request,
                )
                for i, prompt in enumerate(prompts)
            ]
        except ValueError as e:
            # TODO: Use a vllm-specific Validation Error
            return self.create_error_response(str(e))

        if request.stream:
            return stream_generator_method(
                request, list_result_generator, request_id, request_metadata, duration_s
            )
        # Non-streaming response.
        total_segments = []
        text_parts = []
        try:
            assert list_result_generator is not None
            segments_types: dict[str, type[SpeechToTextSegment]] = {
                "transcribe": TranscriptionSegment,
                "translate": TranslationSegment,
            }
            segment_class: type[SpeechToTextSegment] = segments_types[self.task_type]
            text = ""
            for idx, result_generator in enumerate(list_result_generator):
                async for op in result_generator:
                    if request.response_format == "verbose_json":
                        segments: list[SpeechToTextSegment] = (
                            self._get_verbose_segments(
                                tokens=tuple(op.outputs[0].token_ids),
                                segment_class=segment_class,
                                request=request,
                                start_time=idx * self.asr_config.max_audio_clip_s,
                            )
                        )

                        total_segments.extend(segments)
                        text_parts.extend([seg.text for seg in segments])
                    else:
                        text_parts.append(op.outputs[0].text)
            text = "".join(text_parts)
            if self.task_type == "transcribe":
                final_response: ResponseType
                # add usage in TranscriptionResponse.
                usage = {
                    "type": "duration",
                    # rounded up as per openAI specs
                    "seconds": int(math.ceil(duration_s)),
                }
                if request.response_format != "verbose_json":
                    final_response = cast(
                        T, TranscriptionResponse(text=text, usage=usage)
                    )
                else:
                    final_response = cast(
                        V,
                        TranscriptionResponseVerbose(
                            text=text,
                            language=request.language,
                            duration=str(duration_s),
                            segments=total_segments,
                        ),
                    )
            else:
                # no usage in response for translation task
                if request.response_format != "verbose_json":
                    final_response = cast(T, TranslationResponse(text=text))
                else:
                    final_response = cast(
                        V,
                        TranslationResponseVerbose(
                            text=text,
                            language=request.language,
                            duration=str(duration_s),
                            segments=total_segments,
                        ),
                    )
            return final_response
        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))

    async def _speech_to_text_stream_generator(
        self,
        request: SpeechToTextRequest,
        list_result_generator: list[AsyncGenerator[RequestOutput, None]],
        request_id: str,
        request_metadata: RequestResponseMetadata,
        audio_duration_s: float,
        chunk_object_type: Literal["translation.chunk", "transcription.chunk"],
        response_stream_choice_class: type[TranscriptionResponseStreamChoice]
        | type[TranslationResponseStreamChoice],
        stream_response_class: type[TranscriptionStreamResponse]
        | type[TranslationStreamResponse],
    ) -> AsyncGenerator[str, None]:
        created_time = int(time.time())
        model_name = request.model

        completion_tokens = 0
        num_prompt_tokens = 0

        include_usage = self.enable_force_include_usage or request.stream_include_usage
        include_continuous_usage = (
            request.stream_continuous_usage_stats
            if include_usage and request.stream_continuous_usage_stats
            else False
        )

        try:
            for result_generator in list_result_generator:
                async for res in result_generator:
                    # On first result.
                    if res.prompt_token_ids is not None:
                        num_prompt_tokens = len(res.prompt_token_ids)
                        if audio_tokens := self.model_cls.get_num_audio_tokens(
                            audio_duration_s, self.asr_config, self.model_config
                        ):
                            num_prompt_tokens += audio_tokens

                    # We need to do it here, because if there are exceptions in
                    # the result_generator, it needs to be sent as the FIRST
                    # response (by the try...catch).

                    # Just one output (n=1) supported.
                    assert len(res.outputs) == 1
                    output = res.outputs[0]

                    delta_message = DeltaMessage(content=output.text)
                    completion_tokens += len(output.token_ids)

                    if output.finish_reason is None:
                        # Still generating, send delta update.
                        choice_data = response_stream_choice_class(delta=delta_message)
                    else:
                        # Model is finished generating.
                        choice_data = response_stream_choice_class(
                            delta=delta_message,
                            finish_reason=output.finish_reason,
                            stop_reason=output.stop_reason,
                        )

                    chunk = stream_response_class(
                        id=request_id,
                        object=chunk_object_type,
                        created=created_time,
                        choices=[choice_data],
                        model=model_name,
                    )

                    # handle usage stats if requested & if continuous
                    if include_continuous_usage:
                        chunk.usage = UsageInfo(
                            prompt_tokens=num_prompt_tokens,
                            completion_tokens=completion_tokens,
                            total_tokens=num_prompt_tokens + completion_tokens,
                        )

                    data = chunk.model_dump_json(exclude_unset=True)
                    yield f"data: {data}\n\n"

            # Once the final token is handled, if stream_options.include_usage
            # is sent, send the usage.
            if include_usage:
                final_usage = UsageInfo(
                    prompt_tokens=num_prompt_tokens,
                    completion_tokens=completion_tokens,
                    total_tokens=num_prompt_tokens + completion_tokens,
                )

                final_usage_chunk = stream_response_class(
                    id=request_id,
                    object=chunk_object_type,
                    created=created_time,
                    choices=[],
                    model=model_name,
                    usage=final_usage,
                )
                final_usage_data = final_usage_chunk.model_dump_json(
                    exclude_unset=True, exclude_none=True
                )
                yield f"data: {final_usage_data}\n\n"

            # report to FastAPI middleware aggregate usage across all choices
            request_metadata.final_usage_info = UsageInfo(
                prompt_tokens=num_prompt_tokens,
                completion_tokens=completion_tokens,
                total_tokens=num_prompt_tokens + completion_tokens,
            )

        except Exception as e:
            # TODO: Use a vllm-specific Validation Error
            logger.exception("Error in %s stream generator.", self.task_type)
            data = self.create_streaming_error_response(str(e))
            yield f"data: {data}\n\n"
        # Send the final done message after all response.n are finished
        yield "data: [DONE]\n\n"

    def _split_audio(
        self, audio_data: np.ndarray, sample_rate: int
    ) -> list[np.ndarray]:
        chunk_size = sample_rate * self.asr_config.max_audio_clip_s
        overlap_size = sample_rate * self.asr_config.overlap_chunk_second
        chunks = []
        i = 0
        while i < audio_data.shape[-1]:
            if i + chunk_size >= audio_data.shape[-1]:
                # handle last chunk
                chunks.append(audio_data[..., i:])
                break

            # Find the best split point in the overlap region
            search_start = i + chunk_size - overlap_size
            search_end = min(i + chunk_size, audio_data.shape[-1])
            split_point = self._find_split_point(audio_data, search_start, search_end)

            # Extract chunk up to the split point
            chunks.append(audio_data[..., i:split_point])
            i = split_point
        return chunks

    def _find_split_point(self, wav: np.ndarray, start_idx: int, end_idx: int) -> int:
        """Find the best point to split audio by
        looking for silence or low amplitude.
        Args:
            wav: Audio tensor [1, T]
            start_idx: Start index of search region
            end_idx: End index of search region
        Returns:
            Index of best splitting point
        """
        segment = wav[start_idx:end_idx]

        # Calculate RMS energy in small windows
        min_energy = math.inf
        quietest_idx = 0
        min_energy_window = self.asr_config.min_energy_split_window_size
        assert min_energy_window is not None
        for i in range(0, len(segment) - min_energy_window, min_energy_window):
            window = segment[i : i + min_energy_window]
            energy = (window**2).mean() ** 0.5
            if energy < min_energy:
                quietest_idx = i + start_idx
                min_energy = energy
        return quietest_idx

asr_config instance-attribute

asr_config = get_speech_to_text_config(
    model_config, task_type
)

default_sampling_params instance-attribute

default_sampling_params = get_diff_sampling_param()

enable_force_include_usage instance-attribute

enable_force_include_usage = enable_force_include_usage

max_audio_filesize_mb instance-attribute

max_audio_filesize_mb = VLLM_MAX_AUDIO_CLIP_FILESIZE_MB

model_cls cached property

task_type instance-attribute

task_type = task_type

tokenizer instance-attribute

tokenizer = cast(
    PreTrainedTokenizerBase,
    get_tokenizer(
        tokenizer_name=tokenizer,
        tokenizer_mode=tokenizer_mode,
    ),
)

__init__

__init__(
    engine_client: EngineClient,
    models: OpenAIServingModels,
    *,
    request_logger: RequestLogger | None,
    return_tokens_as_token_ids: bool = False,
    task_type: Literal[
        "transcribe", "translate"
    ] = "transcribe",
    log_error_stack: bool = False,
    enable_force_include_usage: bool = False,
)
Source code in vllm/entrypoints/openai/speech_to_text.py
def __init__(
    self,
    engine_client: EngineClient,
    models: OpenAIServingModels,
    *,
    request_logger: RequestLogger | None,
    return_tokens_as_token_ids: bool = False,
    task_type: Literal["transcribe", "translate"] = "transcribe",
    log_error_stack: bool = False,
    enable_force_include_usage: bool = False,
):
    super().__init__(
        engine_client=engine_client,
        models=models,
        request_logger=request_logger,
        return_tokens_as_token_ids=return_tokens_as_token_ids,
        log_error_stack=log_error_stack,
    )

    self.default_sampling_params = self.model_config.get_diff_sampling_param()
    self.task_type = task_type

    self.asr_config = self.model_cls.get_speech_to_text_config(
        self.model_config, task_type
    )

    self.enable_force_include_usage = enable_force_include_usage

    self.max_audio_filesize_mb = envs.VLLM_MAX_AUDIO_CLIP_FILESIZE_MB
    if self.model_cls.supports_segment_timestamp:
        self.tokenizer = cast(
            PreTrainedTokenizerBase,
            get_tokenizer(
                tokenizer_name=self.model_config.tokenizer,
                tokenizer_mode=self.model_config.tokenizer_mode,
            ),
        )

    if self.default_sampling_params:
        logger.info(
            "Overwriting default completion sampling param with: %s",
            self.default_sampling_params,
        )

_create_speech_to_text async

_create_speech_to_text(
    audio_data: bytes,
    request: SpeechToTextRequest,
    raw_request: Request,
    response_class: type[T | V],
    stream_generator_method: Callable[
        ..., AsyncGenerator[str, None]
    ],
) -> T | V | AsyncGenerator[str, None] | ErrorResponse

Base method for speech-to-text operations like transcription and translation.

Source code in vllm/entrypoints/openai/speech_to_text.py
async def _create_speech_to_text(
    self,
    audio_data: bytes,
    request: SpeechToTextRequest,
    raw_request: Request,
    response_class: type[T | V],
    stream_generator_method: Callable[..., AsyncGenerator[str, None]],
) -> T | V | AsyncGenerator[str, None] | ErrorResponse:
    """Base method for speech-to-text operations like transcription and
    translation."""
    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", "verbose_json"]:
        return self.create_error_response(
            ("Currently only support response_format")
            + ("`text`, `json` or `verbose_json`")
        )

    if (
        request.response_format == "verbose_json"
        and not self.model_cls.supports_segment_timestamp
    ):
        return self.create_error_response(
            f"Currently do not support verbose_json for {request.model}"
        )

    if request.response_format == "verbose_json" and request.stream:
        return self.create_error_response(
            "verbose_json format doesn't support streaming case"
        )
    request_id = f"{self.task_type}-{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 = self._maybe_get_adapters(request)

        prompts, duration_s = await self._preprocess_speech_to_text(
            request=request,
            audio_data=audio_data,
        )

    except ValueError as e:
        logger.exception("Error in preprocessing prompt inputs")
        return self.create_error_response(str(e))

    list_result_generator: list[AsyncGenerator[RequestOutput, None]] | None = None
    try:
        # Unlike most decoder-only models, whisper generation length is not
        # constrained by the size of the input audio, which is mapped to a
        # fixed-size log-mel-spectogram.
        default_max_tokens = self.model_config.max_model_len
        sampling_params = request.to_sampling_params(
            default_max_tokens, self.default_sampling_params
        )

        self._log_inputs(
            request_id,
            # It will not display special tokens like <|startoftranscript|>
            request.prompt,
            params=sampling_params,
            lora_request=lora_request,
        )

        list_result_generator = [
            self.engine_client.generate(
                prompt,
                sampling_params,
                f"{request_id}_{i}",
                lora_request=lora_request,
            )
            for i, prompt in enumerate(prompts)
        ]
    except ValueError as e:
        # TODO: Use a vllm-specific Validation Error
        return self.create_error_response(str(e))

    if request.stream:
        return stream_generator_method(
            request, list_result_generator, request_id, request_metadata, duration_s
        )
    # Non-streaming response.
    total_segments = []
    text_parts = []
    try:
        assert list_result_generator is not None
        segments_types: dict[str, type[SpeechToTextSegment]] = {
            "transcribe": TranscriptionSegment,
            "translate": TranslationSegment,
        }
        segment_class: type[SpeechToTextSegment] = segments_types[self.task_type]
        text = ""
        for idx, result_generator in enumerate(list_result_generator):
            async for op in result_generator:
                if request.response_format == "verbose_json":
                    segments: list[SpeechToTextSegment] = (
                        self._get_verbose_segments(
                            tokens=tuple(op.outputs[0].token_ids),
                            segment_class=segment_class,
                            request=request,
                            start_time=idx * self.asr_config.max_audio_clip_s,
                        )
                    )

                    total_segments.extend(segments)
                    text_parts.extend([seg.text for seg in segments])
                else:
                    text_parts.append(op.outputs[0].text)
        text = "".join(text_parts)
        if self.task_type == "transcribe":
            final_response: ResponseType
            # add usage in TranscriptionResponse.
            usage = {
                "type": "duration",
                # rounded up as per openAI specs
                "seconds": int(math.ceil(duration_s)),
            }
            if request.response_format != "verbose_json":
                final_response = cast(
                    T, TranscriptionResponse(text=text, usage=usage)
                )
            else:
                final_response = cast(
                    V,
                    TranscriptionResponseVerbose(
                        text=text,
                        language=request.language,
                        duration=str(duration_s),
                        segments=total_segments,
                    ),
                )
        else:
            # no usage in response for translation task
            if request.response_format != "verbose_json":
                final_response = cast(T, TranslationResponse(text=text))
            else:
                final_response = cast(
                    V,
                    TranslationResponseVerbose(
                        text=text,
                        language=request.language,
                        duration=str(duration_s),
                        segments=total_segments,
                    ),
                )
        return final_response
    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))

_find_split_point

_find_split_point(
    wav: ndarray, start_idx: int, end_idx: int
) -> int

Find the best point to split audio by looking for silence or low amplitude. Args: wav: Audio tensor [1, T] start_idx: Start index of search region end_idx: End index of search region Returns: Index of best splitting point

Source code in vllm/entrypoints/openai/speech_to_text.py
def _find_split_point(self, wav: np.ndarray, start_idx: int, end_idx: int) -> int:
    """Find the best point to split audio by
    looking for silence or low amplitude.
    Args:
        wav: Audio tensor [1, T]
        start_idx: Start index of search region
        end_idx: End index of search region
    Returns:
        Index of best splitting point
    """
    segment = wav[start_idx:end_idx]

    # Calculate RMS energy in small windows
    min_energy = math.inf
    quietest_idx = 0
    min_energy_window = self.asr_config.min_energy_split_window_size
    assert min_energy_window is not None
    for i in range(0, len(segment) - min_energy_window, min_energy_window):
        window = segment[i : i + min_energy_window]
        energy = (window**2).mean() ** 0.5
        if energy < min_energy:
            quietest_idx = i + start_idx
            min_energy = energy
    return quietest_idx

_get_verbose_segments

_get_verbose_segments(
    tokens: tuple,
    request: SpeechToTextRequest,
    segment_class: type[SpeechToTextSegment],
    start_time: float = 0,
) -> list[SpeechToTextSegment]

Convert tokens to verbose segments.

This method expects the model to produce timestamps as tokens (similar to Whisper). If the tokens do not include timestamp information, the segments may not be generated correctly.

Note: Fields like avg_logprob, compression_ratio, and no_speech_prob are not supported in this implementation and will be None. See docs for details.

Source code in vllm/entrypoints/openai/speech_to_text.py
def _get_verbose_segments(
    self,
    tokens: tuple,
    request: SpeechToTextRequest,
    segment_class: type[SpeechToTextSegment],
    start_time: float = 0,
) -> list[SpeechToTextSegment]:
    """
    Convert tokens to verbose segments.

    This method expects the model to produce
    timestamps as tokens (similar to Whisper).
    If the tokens do not include timestamp information,
    the segments may not be generated correctly.

    Note: Fields like avg_logprob, compression_ratio,
    and no_speech_prob are not supported
    in this implementation and will be None. See docs for details.
    """
    BASE_OFFSET = 0.02
    init_token = self.tokenizer.encode("<|0.00|>", add_special_tokens=False)[0]
    if tokens[-1] == self.tokenizer.eos_token_id:
        tokens = tokens[:-1]

    tokens_with_start = (init_token,) + tokens
    segments: list[SpeechToTextSegment] = []
    last_timestamp_start = 0

    if tokens_with_start[-2] < init_token and tokens_with_start[-1] >= init_token:
        tokens_with_start = tokens_with_start + (tokens_with_start[-1],)
    for idx, token in enumerate(tokens_with_start):
        # Timestamp tokens (e.g., <|0.00|>) are assumed to be sorted.
        # If the ordering is violated, this slicing may produce incorrect results.
        if (
            token >= init_token
            and idx != 0
            and tokens_with_start[idx - 1] >= init_token
        ):
            sliced_timestamp_tokens = tokens_with_start[last_timestamp_start:idx]
            start_timestamp = sliced_timestamp_tokens[0] - init_token
            end_timestamp = sliced_timestamp_tokens[-1] - init_token

            casting_segment = cast(
                SpeechToTextSegment,
                segment_class(
                    id=len(segments),
                    seek=start_time,
                    start=start_time + BASE_OFFSET * start_timestamp,
                    end=start_time + BASE_OFFSET * end_timestamp,
                    temperature=request.temperature,
                    text=self.tokenizer.decode(sliced_timestamp_tokens[1:-1]),
                    tokens=sliced_timestamp_tokens[1:-1],
                ),
            )
            segments.append(casting_segment)
            last_timestamp_start = idx
    return segments

_preprocess_speech_to_text async

_preprocess_speech_to_text(
    request: SpeechToTextRequest, audio_data: bytes
) -> tuple[list[PromptType], float]
Source code in vllm/entrypoints/openai/speech_to_text.py
async def _preprocess_speech_to_text(
    self,
    request: SpeechToTextRequest,
    audio_data: bytes,
) -> tuple[list[PromptType], float]:
    # Validate request
    language = self.model_cls.validate_language(request.language)
    # Skip to_language validation to avoid extra logging for Whisper.
    to_language = (
        self.model_cls.validate_language(request.to_language)
        if request.to_language
        else None
    )

    if len(audio_data) / 1024**2 > self.max_audio_filesize_mb:
        raise ValueError("Maximum file size exceeded.")

    with io.BytesIO(audio_data) as bytes_:
        # NOTE resample to model SR here for efficiency. This is also a
        # pre-requisite for chunking, as it assumes Whisper SR.
        y, sr = librosa.load(bytes_, sr=self.asr_config.sample_rate)

    duration = librosa.get_duration(y=y, sr=sr)
    do_split_audio = (
        self.asr_config.allow_audio_chunking
        and duration > self.asr_config.max_audio_clip_s
    )
    chunks = [y] if not do_split_audio else self._split_audio(y, int(sr))
    prompts = []
    for chunk in chunks:
        # The model has control over the construction, as long as it
        # returns a valid PromptType.
        prompt = self.model_cls.get_generation_prompt(
            audio=chunk,
            stt_config=self.asr_config,
            model_config=self.model_config,
            language=language,
            task_type=self.task_type,
            request_prompt=request.prompt,
            to_language=to_language,
        )
        if request.response_format == "verbose_json":
            if not isinstance(prompt, dict):
                raise ValueError(f"Expected prompt to be a dict,got {type(prompt)}")
            prompt_dict = cast(dict, prompt)
            decoder_prompt = prompt.get("decoder_prompt")
            if not isinstance(decoder_prompt, str):
                raise ValueError(
                    f"Expected decoder_prompt to bestr, got {type(decoder_prompt)}"
                )
            prompt_dict["decoder_prompt"] = decoder_prompt.replace(
                "<|notimestamps|>", "<|0.00|>"
            )
        prompts.append(prompt)
    return prompts, duration

_speech_to_text_stream_generator async

_speech_to_text_stream_generator(
    request: SpeechToTextRequest,
    list_result_generator: list[
        AsyncGenerator[RequestOutput, None]
    ],
    request_id: str,
    request_metadata: RequestResponseMetadata,
    audio_duration_s: float,
    chunk_object_type: Literal[
        "translation.chunk", "transcription.chunk"
    ],
    response_stream_choice_class: type[
        TranscriptionResponseStreamChoice
    ]
    | type[TranslationResponseStreamChoice],
    stream_response_class: type[TranscriptionStreamResponse]
    | type[TranslationStreamResponse],
) -> AsyncGenerator[str, None]
Source code in vllm/entrypoints/openai/speech_to_text.py
async def _speech_to_text_stream_generator(
    self,
    request: SpeechToTextRequest,
    list_result_generator: list[AsyncGenerator[RequestOutput, None]],
    request_id: str,
    request_metadata: RequestResponseMetadata,
    audio_duration_s: float,
    chunk_object_type: Literal["translation.chunk", "transcription.chunk"],
    response_stream_choice_class: type[TranscriptionResponseStreamChoice]
    | type[TranslationResponseStreamChoice],
    stream_response_class: type[TranscriptionStreamResponse]
    | type[TranslationStreamResponse],
) -> AsyncGenerator[str, None]:
    created_time = int(time.time())
    model_name = request.model

    completion_tokens = 0
    num_prompt_tokens = 0

    include_usage = self.enable_force_include_usage or request.stream_include_usage
    include_continuous_usage = (
        request.stream_continuous_usage_stats
        if include_usage and request.stream_continuous_usage_stats
        else False
    )

    try:
        for result_generator in list_result_generator:
            async for res in result_generator:
                # On first result.
                if res.prompt_token_ids is not None:
                    num_prompt_tokens = len(res.prompt_token_ids)
                    if audio_tokens := self.model_cls.get_num_audio_tokens(
                        audio_duration_s, self.asr_config, self.model_config
                    ):
                        num_prompt_tokens += audio_tokens

                # We need to do it here, because if there are exceptions in
                # the result_generator, it needs to be sent as the FIRST
                # response (by the try...catch).

                # Just one output (n=1) supported.
                assert len(res.outputs) == 1
                output = res.outputs[0]

                delta_message = DeltaMessage(content=output.text)
                completion_tokens += len(output.token_ids)

                if output.finish_reason is None:
                    # Still generating, send delta update.
                    choice_data = response_stream_choice_class(delta=delta_message)
                else:
                    # Model is finished generating.
                    choice_data = response_stream_choice_class(
                        delta=delta_message,
                        finish_reason=output.finish_reason,
                        stop_reason=output.stop_reason,
                    )

                chunk = stream_response_class(
                    id=request_id,
                    object=chunk_object_type,
                    created=created_time,
                    choices=[choice_data],
                    model=model_name,
                )

                # handle usage stats if requested & if continuous
                if include_continuous_usage:
                    chunk.usage = UsageInfo(
                        prompt_tokens=num_prompt_tokens,
                        completion_tokens=completion_tokens,
                        total_tokens=num_prompt_tokens + completion_tokens,
                    )

                data = chunk.model_dump_json(exclude_unset=True)
                yield f"data: {data}\n\n"

        # Once the final token is handled, if stream_options.include_usage
        # is sent, send the usage.
        if include_usage:
            final_usage = UsageInfo(
                prompt_tokens=num_prompt_tokens,
                completion_tokens=completion_tokens,
                total_tokens=num_prompt_tokens + completion_tokens,
            )

            final_usage_chunk = stream_response_class(
                id=request_id,
                object=chunk_object_type,
                created=created_time,
                choices=[],
                model=model_name,
                usage=final_usage,
            )
            final_usage_data = final_usage_chunk.model_dump_json(
                exclude_unset=True, exclude_none=True
            )
            yield f"data: {final_usage_data}\n\n"

        # report to FastAPI middleware aggregate usage across all choices
        request_metadata.final_usage_info = UsageInfo(
            prompt_tokens=num_prompt_tokens,
            completion_tokens=completion_tokens,
            total_tokens=num_prompt_tokens + completion_tokens,
        )

    except Exception as e:
        # TODO: Use a vllm-specific Validation Error
        logger.exception("Error in %s stream generator.", self.task_type)
        data = self.create_streaming_error_response(str(e))
        yield f"data: {data}\n\n"
    # Send the final done message after all response.n are finished
    yield "data: [DONE]\n\n"

_split_audio

_split_audio(
    audio_data: ndarray, sample_rate: int
) -> list[ndarray]
Source code in vllm/entrypoints/openai/speech_to_text.py
def _split_audio(
    self, audio_data: np.ndarray, sample_rate: int
) -> list[np.ndarray]:
    chunk_size = sample_rate * self.asr_config.max_audio_clip_s
    overlap_size = sample_rate * self.asr_config.overlap_chunk_second
    chunks = []
    i = 0
    while i < audio_data.shape[-1]:
        if i + chunk_size >= audio_data.shape[-1]:
            # handle last chunk
            chunks.append(audio_data[..., i:])
            break

        # Find the best split point in the overlap region
        search_start = i + chunk_size - overlap_size
        search_end = min(i + chunk_size, audio_data.shape[-1])
        split_point = self._find_split_point(audio_data, search_start, search_end)

        # Extract chunk up to the split point
        chunks.append(audio_data[..., i:split_point])
        i = split_point
    return chunks