This commit is contained in:
2026-05-28 23:14:39 -05:00
parent fa06e31b42
commit e87f30173d
2 changed files with 32 additions and 29 deletions

60
main.py
View File

@@ -6,9 +6,10 @@ import dataclasses
import numpy as np
from sklearn.cluster import AgglomerativeClustering
from sklearn.preprocessing import normalize
from speechbrain.dataio import audio_io
from speechbrain.dataio.preprocess import AudioNormalizer
from speechbrain.inference.speaker import EncoderClassifier
import torch
import torchaudio
import whisper
@dataclasses.dataclass
@@ -27,12 +28,24 @@ def srt_timestamp(milis):
return f"{h:02}:{m:02}:{s:02},{milis:03}"
def main():
parser = argparse.ArgumentParser()
parser = argparse.ArgumentParser(formatter_class=argparse.ArgumentDefaultsHelpFormatter)
parser.add_argument("input")
parser.add_argument("-o", "--output", default="output.srt")
parser.add_argument("-o", "--output", default="output.srt",
help="Output subtitle (SRT) file path")
parser.add_argument("-s", "--speakers", type=int, default=2,
help="Number of speakers in the audio")
parser.add_argument("-m", "--model", default='turbo', choices=whisper.available_models(),
help="Whisper model to use")
parser.add_argument("--gpu", action="store_true", help="run on GPU if available")
args = parser.parse_args()
model = whisper.load_model("turbo")
has_gpu = torch.cuda.is_available()
if args.gpu and not has_gpu:
print("GPU not available. Falling back to CPU")
dev = torch.device("cuda" if has_gpu and args.gpu else "cpu")
model = whisper.load_model("turbo").to(dev)
result = model.transcribe(args.input, word_timestamps=True, language="en")
segments = []
for seg in result["segments"]:
@@ -45,36 +58,35 @@ def main():
classifier = EncoderClassifier.from_hparams(
source="speechbrain/spkrec-ecapa-voxceleb",
run_opts={"device": str(dev)},
)
signal, sample_rate = torchaudio.load(args.input)
print(sample_rate)
# resample to 16khz
if sample_rate != 16000:
resampler = torchaudio.transforms.Resample(sample_rate, 16000)
signal = resampler(signal)
sample_rate = 16000
# convert to mono if needed
if signal.shape[0] > 1:
signal = torch.mean(signal, dim=0, keepdim=True)
signal, sample_rate = audio_io.load(args.input, channels_first=False)
audio_normalizer = AudioNormalizer()
signal = audio_normalizer(signal, sample_rate)
sample_rate = audio_normalizer.sample_rate
embedding_segments = []
embeddings = []
for seg in segments:
if seg.end - seg.start < 0.1:
# Segment is too short to bother trying to classify
continue
start_sample = int(seg.start * sample_rate)
end_sample = int(seg.end * sample_rate)
chunk = signal[:, start_sample:end_sample]
chunk = signal[start_sample:end_sample]
# normalize
chunk /= chunk.abs().max()
# SpeechBrain expects [batch, time]
emb = classifier.encode_batch(chunk)
emb = classifier.encode_batch(chunk.unsqueeze(0).to(dev))
emb = emb.squeeze().detach().cpu().numpy()
embeddings.append(emb)
embedding_segments.append(seg)
embeddings = np.vstack(embeddings)
embeddings = normalize(embeddings)
@@ -86,9 +98,10 @@ def main():
)
labels = clustering.fit_predict(embeddings)
for seg, label in zip(segments, labels):
for seg, label in zip(embedding_segments, labels):
seg.text = f"Speaker {label + 1}: {seg.text}"
print("Final transcription:")
with open(args.output, "w") as srt_file:
for i, seg in enumerate(segments, 1):
start = srt_timestamp(round(seg.start * 1000))
@@ -96,14 +109,5 @@ def main():
srt_file.write(f"{i}\n{start} --> {end}\n{seg.text}\n\n")
print(start, end, seg.text)
from sklearn.decomposition import PCA
import matplotlib.pyplot as plt
pca = PCA(n_components=2)
points = pca.fit_transform(embeddings)
plt.scatter(points[:,0], points[:,1])
plt.show()
if __name__ == "__main__":
main()