tools / run_whisper_streaming.py
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#!/usr/bin/env python3
from transformers import (
WhisperForConditionalGeneration,
WhisperProcessor,
)
import torch
import re
import numpy as np
from datasets import load_dataset
device = "cpu"
dtype = torch.float32
processor = WhisperProcessor.from_pretrained("openai/whisper-tiny")
model = WhisperForConditionalGeneration.from_pretrained(
"openai/whisper-tiny", low_cpu_mem_usage=True, torch_dtype=dtype
)
model.to(device)
STREAMING_INTERVAL = 0.33 # in seconds
SAMPLING_RATE = 16_000
INTERVAL_LENGTH = int(STREAMING_INTERVAL * SAMPLING_RATE)
ds = load_dataset(
"hf-internal-testing/librispeech_asr_dummy", "clean", split="validation"
)
audio_array = np.concatenate([x["array"] for x in ds["audio"]])
# fake streaming by decoding every STREAMING_INTERVAL seconds
start_idx = 0
fully_decoded = ""
for end_idx in range(INTERVAL_LENGTH, audio_array.shape[-1], INTERVAL_LENGTH):
input_audio = audio_array[start_idx:end_idx]
processor_kwargs = (
{"padding": "longest", "truncation": False, "return_attention_mask": True}
if input_audio.shape[0] / SAMPLING_RATE > 30.0
else {}
)
inputs = processor(
input_audio,
sampling_rate=SAMPLING_RATE,
return_tensors="pt",
**processor_kwargs,
)
inputs = inputs.to(dtype=dtype, device=device)
tokens = model.generate(
**inputs,
return_timestamps=True,
)
sequences = processor.batch_decode(tokens, decode_with_timestamps=True)[0]
sequences_no_special = processor.batch_decode(tokens, skip_special_tokens=True)[0]
regex_search = re.findall(r"<\|[\d\.]+\|><\|[\d\.]+\|>", sequences)
regex_split = re.split(r"<\|[\d\.]+\|><\|[\d\.]+\|>", sequences)
# at least two timestamps seperations and 5 new words have to have been detected to cut input audio
if len(regex_search) > 1 and len("".join(regex_split[1:]).split()) > 5:
cut_idx = int(SAMPLING_RATE * float(regex_search[0].split("|><|")[0][2:]))
start_idx += cut_idx
fully_decoded += sequences_no_special
sequences_no_special = ""
print(fully_decoded + sequences_no_special)
print(f"Passed time: {end_idx / 16_000}")