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https://github.com/openai/whisper.git
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Merge e000892e137fb17d4ab43046c77853274eaf9ed2 into 517a43ecd132a2089d85f4ebc044728a71d49f6e
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commit
adf281518a
@ -1,19 +1,24 @@
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import os.path
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import numpy as np
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import pytest
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from whisper.audio import SAMPLE_RATE, load_audio, log_mel_spectrogram
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def test_audio():
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@pytest.mark.parametrize("read_bytes", [True, False])
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def test_audio(read_bytes):
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audio_path = os.path.join(os.path.dirname(__file__), "jfk.flac")
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audio = load_audio(audio_path)
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audio_input = audio_path
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if (read_bytes):
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with open(audio_path, 'rb') as f:
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audio_input = f.read()
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audio = load_audio(audio_input)
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assert audio.ndim == 1
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assert SAMPLE_RATE * 10 < audio.shape[0] < SAMPLE_RATE * 12
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assert 0 < audio.std() < 1
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mel_from_audio = log_mel_spectrogram(audio)
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mel_from_file = log_mel_spectrogram(audio_path)
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mel_from_file = log_mel_spectrogram(audio_input)
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assert np.allclose(mel_from_audio, mel_from_file)
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assert mel_from_audio.max() - mel_from_audio.min() <= 2.0
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@ -1,6 +1,6 @@
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import os
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from functools import lru_cache
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from subprocess import CalledProcessError, run
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from subprocess import CalledProcessError, run, PIPE
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from typing import Optional, Union
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import numpy as np
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@ -22,14 +22,14 @@ FRAMES_PER_SECOND = exact_div(SAMPLE_RATE, HOP_LENGTH) # 10ms per audio frame
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TOKENS_PER_SECOND = exact_div(SAMPLE_RATE, N_SAMPLES_PER_TOKEN) # 20ms per audio token
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def load_audio(file: str, sr: int = SAMPLE_RATE):
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def load_audio(file: Union[str, bytes], sr: int = SAMPLE_RATE):
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"""
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Open an audio file and read as mono waveform, resampling as necessary
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Parameters
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----------
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file: str
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The audio file to open
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file: Union[str, bytes]
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The audio file to open, or the bytes content of an audio file
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sr: int
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The sample rate to resample the audio if necessary
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@ -46,7 +46,6 @@ def load_audio(file: str, sr: int = SAMPLE_RATE):
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"ffmpeg",
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"-nostdin",
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"-threads", "0",
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"-i", file,
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"-f", "s16le",
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"-ac", "1",
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"-acodec", "pcm_s16le",
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@ -54,10 +53,18 @@ def load_audio(file: str, sr: int = SAMPLE_RATE):
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"-"
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]
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# fmt: on
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try:
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out = run(cmd, capture_output=True, check=True).stdout
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except CalledProcessError as e:
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raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e
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if isinstance(file, str):
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cmd += ["-i", file]
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try:
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out = run(cmd, capture_output=True, check=True).stdout
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except CalledProcessError as e:
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raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e
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else:
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cmd += ["-i", "-"]
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try:
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out = run(cmd, input=file, stdout=PIPE, stderr=PIPE, check=True).stdout
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except CalledProcessError as e:
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raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e
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return np.frombuffer(out, np.int16).flatten().astype(np.float32) / 32768.0
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@ -108,7 +115,7 @@ def mel_filters(device, n_mels: int) -> torch.Tensor:
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def log_mel_spectrogram(
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audio: Union[str, np.ndarray, torch.Tensor],
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audio: Union[str, bytes, np.ndarray, torch.Tensor],
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n_mels: int = 80,
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padding: int = 0,
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device: Optional[Union[str, torch.device]] = None,
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@ -136,7 +143,7 @@ def log_mel_spectrogram(
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A Tensor that contains the Mel spectrogram
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"""
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if not torch.is_tensor(audio):
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if isinstance(audio, str):
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if isinstance(audio, str) or isinstance(audio, bytes):
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audio = load_audio(audio)
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audio = torch.from_numpy(audio)
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@ -37,7 +37,7 @@ if TYPE_CHECKING:
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def transcribe(
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model: "Whisper",
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audio: Union[str, np.ndarray, torch.Tensor],
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audio: Union[str, bytes, np.ndarray, torch.Tensor],
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*,
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verbose: Optional[bool] = None,
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temperature: Union[float, Tuple[float, ...]] = (0.0, 0.2, 0.4, 0.6, 0.8, 1.0),
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@ -63,7 +63,7 @@ def transcribe(
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The Whisper model instance
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audio: Union[str, np.ndarray, torch.Tensor]
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The path to the audio file to open, or the audio waveform
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The path to the audio file to open, or the audio waveform, or the bytes content of an audio file
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verbose: bool
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Whether to display the text being decoded to the console. If True, displays all the details,
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