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Merge 3777d81112 into 82ca10cc2d
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Haquant
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commit
1dc3f21717
6 changed files with 260 additions and 9 deletions
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@ -4,8 +4,9 @@ Utils used for litellm.transcription() and litellm.atranscription()
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import hashlib
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import os
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from collections.abc import Sized
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from dataclasses import dataclass
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from typing import Final
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from typing import Final, Protocol
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from litellm.types.files import (
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AUDIO_FILE_TYPES,
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@ -16,6 +17,40 @@ from litellm.types.files import (
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)
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from litellm.types.utils import FileTypes
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# libsndfile uses SF_COUNT_MAX (2**63 - 1) for "unknown length"; no real upload gets near 2**62 frames
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_MAX_PLAUSIBLE_AUDIO_FRAMES: Final = 2**62
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_DECODE_BLOCK_FRAMES: Final = 1 << 16
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class _ReadableAudio(Protocol):
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def read(self, frames: int, dtype: str) -> Sized: ...
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def close(self) -> None: ...
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def _open_forward_only(file_content: bytes) -> _ReadableAudio:
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import io
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import soundfile as sf
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class _ForwardOnlySoundFile(sf.SoundFile): # pyright: ignore[reportUntypedBaseClass, reportUnknownMemberType] # soundfile ships no type stubs
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# soundfile sizes seekable reads from tell(), which libsndfile can't answer for an
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# unknown-length stream, so read it front to back instead
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def seekable(self) -> bool:
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return False
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return _ForwardOnlySoundFile(io.BytesIO(file_content))
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def _decoded_frame_count(file_content: bytes) -> int:
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"""Count frames by decoding, for audio whose header carries no length"""
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audio: Final = _open_forward_only(file_content)
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try:
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# iter() stops at the first empty block, i.e. the end of the stream
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return sum(iter(lambda: len(audio.read(_DECODE_BLOCK_FRAMES, "int16")), 0))
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finally:
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audio.close()
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@dataclass
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class ProcessedAudioFile:
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@ -323,13 +358,21 @@ def calculate_request_duration(file: FileTypes) -> float | None:
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# Extract duration using soundfile
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file_object: Final = io.BytesIO(file_content)
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with sf.SoundFile(file_object) as audio:
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duration: Final = len(audio) / audio.samplerate
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return duration
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reported_frames: Final[int] = len(audio)
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samplerate: Final[int] = audio.samplerate
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# libsndfile reports an unknown length (e.g. a FLAC with no sample count
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# in STREAMINFO) as SF_COUNT_MAX, which would bill as ~2^63 frames, so
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# count the real frames by decoding instead of trusting the header
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frames: Final = (
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reported_frames if reported_frames < _MAX_PLAUSIBLE_AUDIO_FRAMES else _decoded_frame_count(file_content)
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)
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except Exception:
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# Silently fail if duration extraction fails
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return None
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return frames / samplerate
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DEFAULT_SPEECH_MEDIA_TYPE: Final = "audio/mpeg"
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@ -7,6 +7,7 @@ API reference: https://docs.mistral.ai/api/#tag/audio/operation/audio_transcript
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from typing import Final
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import httpx
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from pydantic import BaseModel, ConfigDict, ValidationError
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from litellm.litellm_core_utils.audio_utils.utils import process_audio_file
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from litellm.llms.base_llm.audio_transcription.transformation import (
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@ -26,6 +27,26 @@ class MistralAudioTranscriptionException(BaseLLMException):
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pass
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class _MistralTranscriptionUsage(BaseModel):
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model_config = ConfigDict(frozen=True)
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prompt_audio_seconds: float | None = None
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class _MistralTranscriptionBody(BaseModel):
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model_config = ConfigDict(frozen=True)
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usage: _MistralTranscriptionUsage | None = None
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def _billed_audio_seconds(body: object) -> float | None:
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try:
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parsed: Final = _MistralTranscriptionBody.model_validate(body)
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except ValidationError:
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return None
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return parsed.usage.prompt_audio_seconds if parsed.usage is not None else None
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class MistralAudioTranscriptionConfig(BaseAudioTranscriptionConfig):
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def get_supported_openai_params(self, model: str) -> list[OpenAIAudioTranscriptionOptionalParams]:
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return [
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@ -147,5 +168,12 @@ class MistralAudioTranscriptionConfig(BaseAudioTranscriptionConfig):
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if "language" in response_json:
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response["language"] = response_json["language"]
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response.hidden_params = response_json
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# Mistral bills transcription on usage.prompt_audio_seconds, so cost uses it
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# instead of measuring the uploaded file locally
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billed_seconds: Final = _billed_audio_seconds(response_json)
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response.hidden_params = (
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response_json
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if billed_seconds is None
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else {**response_json, "audio_transcription_duration": billed_seconds}
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)
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return response
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@ -8024,7 +8024,7 @@ async def atranscription(*args, **kwargs) -> TranscriptionResponse:
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# a plain Transcription is a TranscriptionVerbose/Diarized type.
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if response is not None and not isinstance(response, Coroutine) and file is not None:
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existing_duration: Final = getattr(response, "duration", None)
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if existing_duration is None:
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if existing_duration is None and response.hidden_params.get("audio_transcription_duration") is None:
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calculated_duration: Final = calculate_request_duration(file)
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if calculated_duration is not None:
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response.hidden_params["audio_transcription_duration"] = calculated_duration
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@ -8295,12 +8295,12 @@ def transcription(
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# exposing it in the response body (see sync path comment above).
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if response is not None and not isinstance(response, Coroutine):
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existing_duration: Final = getattr(response, "duration", None)
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if existing_duration is None:
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response_hidden_params: Final = cast( # cast-ok: preserve dynamic mapping behavior
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dict[str, object], getattr(response, HIDDEN_PARAMS_ATTR)
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)
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if existing_duration is None and response_hidden_params.get("audio_transcription_duration") is None:
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calculated_duration: Final = calculate_request_duration(file)
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if calculated_duration is not None:
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response_hidden_params: Final = cast( # cast-ok: preserve dynamic mapping behavior
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dict[str, object], getattr(response, HIDDEN_PARAMS_ATTR)
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)
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response_hidden_params["audio_transcription_duration"] = calculated_duration
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if response is None:
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@ -276,6 +276,31 @@ class TestCalculateRequestDuration:
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wav_header
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), "File position should be restored to original position"
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@pytest.mark.skipif(
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os.environ.get("SKIP_AUDIO_TESTS") == "true",
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reason="Skipping audio tests - soundfile may not be available",
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)
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def test_flac_without_sample_count_counts_decoded_frames(self):
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"""
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A FLAC whose STREAMINFO has no total sample count (what ffmpeg writes to a
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pipe) makes libsndfile report SF_COUNT_MAX frames. The duration must come
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from the decoded audio, not a ~2^63 frame count and not a free 0 or None
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"""
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np = pytest.importorskip("numpy")
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sf = pytest.importorskip("soundfile")
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buffer = io.BytesIO()
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sf.write(buffer, np.zeros(1600, dtype="int16"), 16000, format="FLAC")
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known_length = buffer.getvalue()
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assert calculate_request_duration(known_length) == pytest.approx(0.1)
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# STREAMINFO starts at byte 8; the low 36 bits of bytes 18..25 hold the total sample count
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unknown_length = bytearray(known_length)
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packed = int.from_bytes(unknown_length[18:26], "big") & ~((1 << 36) - 1)
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unknown_length[18:26] = packed.to_bytes(8, "big")
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assert calculate_request_duration(bytes(unknown_length)) == pytest.approx(0.1)
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class TestGetAudioFileContentHash:
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"""Test the get_audio_file_content_hash function for cache key generation"""
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@ -2,11 +2,14 @@ import os
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from unittest.mock import MagicMock
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import httpx
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import pytest
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import litellm
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from litellm.llms.base_llm.audio_transcription.transformation import (
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BaseAudioTranscriptionConfig,
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)
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from litellm.llms.custom_httpx.http_handler import HTTPHandler
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from litellm.llms.mistral.audio_transcription.transformation import (
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MistralAudioTranscriptionConfig,
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)
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@ -191,3 +194,89 @@ def test_mistral_audio_transcription_response_transform_empty():
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assert isinstance(response, TranscriptionResponse)
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assert response.text == ""
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def test_mistral_audio_transcription_response_uses_prompt_audio_seconds_for_cost():
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"""Mistral bills on usage.prompt_audio_seconds, so that is the duration cost reads"""
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config = MistralAudioTranscriptionConfig()
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mock_response = MagicMock(spec=httpx.Response)
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mock_response.json.return_value = {
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"text": "hello",
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"usage": {"prompt_audio_seconds": 600, "total_tokens": 10},
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}
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response = config.transform_audio_transcription_response(mock_response)
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assert response.hidden_params["audio_transcription_duration"] == 600.0
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assert response.hidden_params["usage"] == {"prompt_audio_seconds": 600, "total_tokens": 10}
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assert getattr(response, "duration", None) is None
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@pytest.mark.parametrize(
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"usage",
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[
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{"prompt_audio_seconds": None},
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{"prompt_audio_seconds": "not-a-number"},
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"not-an-object",
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],
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)
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def test_mistral_audio_transcription_response_without_usable_usage_has_no_duration(usage):
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config = MistralAudioTranscriptionConfig()
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mock_response = MagicMock(spec=httpx.Response)
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mock_response.json.return_value = {"text": "hello", "usage": usage}
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response = config.transform_audio_transcription_response(mock_response)
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assert "audio_transcription_duration" not in response.hidden_params
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def test_mistral_transcription_cost_uses_provider_billed_seconds():
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"""
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End to end through litellm.transcription: the seconds Mistral reports win over the
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local file measurement (about 17s for this wav), so spend matches what Mistral bills
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"""
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billed_seconds = 30
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wav_path = os.path.join(
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os.path.dirname(__file__),
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"../../../../..",
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"tests",
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"llm_translation",
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"gettysburg.wav",
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)
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with open(wav_path, "rb") as f:
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audio_bytes = f.read()
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def respond(request: httpx.Request) -> httpx.Response:
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return httpx.Response(
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200,
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json={
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"model": "voxtral-mini-2602",
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"text": "a tone",
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"language": None,
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"segments": [],
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"usage": {"prompt_audio_seconds": billed_seconds, "total_tokens": 3},
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},
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request=request,
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)
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client = HTTPHandler(client=httpx.Client(transport=httpx.MockTransport(respond)))
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response = litellm.transcription(
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model="mistral/voxtral-mini-2602",
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file=("gettysburg.wav", audio_bytes, "audio/wav"),
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api_key="fake-key",
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client=client,
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)
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assert response.hidden_params["audio_transcription_duration"] == float(billed_seconds)
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cost_per_second = litellm.get_model_info("mistral/voxtral-mini-2602")["input_cost_per_second"]
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assert cost_per_second
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cost = litellm.completion_cost(
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completion_response=response,
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model="mistral/voxtral-mini-2602",
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call_type="transcription",
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)
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assert cost == pytest.approx(billed_seconds * cost_per_second)
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@ -1266,3 +1266,69 @@ class TestSpendTracking:
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litellm.model_cost = original_model_cost
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litellm.get_model_info.cache_clear()
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@staticmethod
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def _wav_bytes() -> bytes:
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import os
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wav_path = os.path.join(
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os.path.dirname(__file__), "../../../../..", "tests", "llm_translation", "gettysburg.wav"
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)
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with open(wav_path, "rb") as f:
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return f.read()
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@staticmethod
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def _responses_without_duration() -> Dict[str, List[httpx.Response]]:
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# Soniox omits audio_duration_ms, so the handler stores a None duration
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return {
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"POST https://api.soniox.com/v1/files": [_make_response({"id": "file_1"})],
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"POST https://api.soniox.com/v1/transcriptions": [_make_response({"id": "tx_1"})],
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"GET https://api.soniox.com/v1/transcriptions/tx_1": [_make_response({"status": "completed"})],
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"GET https://api.soniox.com/v1/transcriptions/tx_1/transcript": [
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_make_response({"text": "four score", "tokens": []})
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],
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"DELETE https://api.soniox.com/v1/transcriptions/tx_1": [_make_response({})],
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"DELETE https://api.soniox.com/v1/files/file_1": [_make_response({})],
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}
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def test_should_fall_back_to_local_duration_when_soniox_omits_it(self, monkeypatch):
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import litellm
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from litellm.litellm_core_utils.audio_utils.utils import calculate_request_duration
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monkeypatch.setattr("time.sleep", lambda *_: None)
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audio = self._wav_bytes()
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local_duration = calculate_request_duration(audio)
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assert local_duration is not None and local_duration > 0
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resp = litellm.transcription(
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model="soniox/stt-async-v4",
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file=("gettysburg.wav", audio, "audio/wav"),
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api_key="sk-test",
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client=_MockSyncClient(self._responses_without_duration()),
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)
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assert resp._hidden_params["audio_transcription_duration"] == pytest.approx(local_duration)
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cost = litellm.completion_cost(completion_response=resp, model="soniox/stt-async-v4", call_type="transcription")
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assert cost > 0
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def test_should_fall_back_to_local_duration_when_soniox_omits_it_async(self, monkeypatch):
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import litellm
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from litellm.litellm_core_utils.audio_utils.utils import calculate_request_duration
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async def _no_sleep(*_a, **_kw):
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return None
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monkeypatch.setattr(asyncio, "sleep", _no_sleep)
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audio = self._wav_bytes()
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local_duration = calculate_request_duration(audio)
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assert local_duration is not None and local_duration > 0
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resp = asyncio.new_event_loop().run_until_complete(
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litellm.atranscription(
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model="soniox/stt-async-v4",
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file=("gettysburg.wav", audio, "audio/wav"),
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api_key="sk-test",
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client=_MockAsyncClient(self._responses_without_duration()),
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)
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)
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assert resp._hidden_params["audio_transcription_duration"] == pytest.approx(local_duration)
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