Spaces:
Runtime error
Runtime error
File size: 5,496 Bytes
7091430 172ec24 7091430 172ec24 7091430 172ec24 7091430 172ec24 2ad1599 7091430 26faa5f 7091430 80ca55c 7091430 172ec24 7091430 172ec24 7091430 ccb306d 7091430 ccb306d 7091430 80ca55c 7091430 ccb306d 7091430 7c1a8fa 7091430 7c1a8fa 172ec24 7091430 5d7014c 7091430 213b090 7091430 ccb306d 7091430 172ec24 |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 |
from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
from transformers.utils import is_flash_attn_2_available
from transformers.pipelines.audio_utils import ffmpeg_read
import torch
import gradio as gr
import time
BATCH_SIZE = 16
MAX_AUDIO_MINS = 30 # maximum audio input in minutes
device = "cuda:0" if torch.cuda.is_available() else "cpu"
torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32
use_flash_attention_2 = is_flash_attn_2_available()
model = AutoModelForSpeechSeq2Seq.from_pretrained(
"openai/whisper-large-v2", torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True, use_flash_attention_2=use_flash_attention_2
)
distilled_model = AutoModelForSpeechSeq2Seq.from_pretrained(
"distil-whisper/distil-large-v2", torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True, use_flash_attention_2=use_flash_attention_2
)
if not use_flash_attention_2:
# use flash attention from pytorch sdpa
model = model.to_bettertransformer()
distilled_model = distilled_model.to_bettertransformer()
processor = AutoProcessor.from_pretrained("openai/whisper-large-v2")
model.to(device)
distilled_model.to(device)
pipe = pipeline(
"automatic-speech-recognition",
model=model,
tokenizer=processor.tokenizer,
feature_extractor=processor.feature_extractor,
max_new_tokens=128,
chunk_length_s=30,
torch_dtype=torch_dtype,
device=device,
generate_kwargs={"language": "en", "task": "transcribe"},
)
pipe_forward = pipe._forward
distil_pipe = pipeline(
"automatic-speech-recognition",
model=distilled_model,
tokenizer=processor.tokenizer,
feature_extractor=processor.feature_extractor,
max_new_tokens=128,
chunk_length_s=15,
torch_dtype=torch_dtype,
device=device,
generate_kwargs={"language": "en", "task": "transcribe"},
)
distil_pipe_forward = distil_pipe._forward
def transcribe(inputs):
if inputs is None:
raise gr.Error("No audio file submitted! Please record or upload an audio file before submitting your request.")
with open(inputs, "rb") as f:
inputs = f.read()
inputs = ffmpeg_read(inputs, pipe.feature_extractor.sampling_rate)
audio_length_mins = len(inputs) / pipe.feature_extractor.sampling_rate / 60
if audio_length_mins > MAX_AUDIO_MINS:
raise gr.Error(
f"To ensure fair usage of the Space, the maximum audio length permitted is {MAX_AUDIO_MINS} minutes."
f"Got an audio of length {round(audio_length_mins, 3)} minutes."
)
inputs = {"array": inputs, "sampling_rate": pipe.feature_extractor.sampling_rate}
def _forward_distil_time(*args, **kwargs):
global distil_runtime
start_time = time.time()
result = distil_pipe_forward(*args, **kwargs)
distil_runtime = time.time() - start_time
distil_runtime = round(distil_runtime, 2)
return result
distil_pipe._forward = _forward_distil_time
distil_text = distil_pipe(inputs, batch_size=BATCH_SIZE)["text"]
yield distil_text, distil_runtime, None, None, None
def _forward_time(*args, **kwargs):
global runtime
start_time = time.time()
result = pipe_forward(*args, **kwargs)
runtime = time.time() - start_time
runtime = round(runtime, 2)
return result
pipe._forward = _forward_time
text = pipe(inputs, batch_size=BATCH_SIZE)["text"]
yield distil_text, distil_runtime, text, runtime
if __name__ == "__main__":
with gr.Blocks() as demo:
gr.HTML(
"""
<div style="text-align: center; max-width: 700px; margin: 0 auto;">
<div
style="
display: inline-flex; align-items: center; gap: 0.8rem; font-size: 1.75rem;
"
>
<h1 style="font-weight: 900; margin-bottom: 7px; line-height: normal;">
Whisper vs Distil-Whisper: Speed Comparison
</h1>
</div>
</div>
"""
)
gr.HTML(
f"""
Speed comparison between <a href="https://huggingface.co/openai/whisper-large-v2"> Whisper</a>
and <a href="https://huggingface.co/distil-whisper/distil-large-v2"> Distil-Whisper</a>. Both models use the <a href="https://huggingface.co/distil-whisper/distil-large-v2#long-form-transcription"> chunked long-form transcription algorithm</a>
in 🤗 Transformers with Flash Attention support. To ensure fair usage of the Space, we ask that audio
file inputs are kept to < 30 mins.
"""
)
audio = gr.components.Audio(type="filepath", label="Audio input")
button = gr.Button("Transcribe")
with gr.Row():
distil_runtime = gr.components.Textbox(label="Distil-Whisper Transcription Time (s)")
runtime = gr.components.Textbox(label="Whisper Transcription Time (s)")
with gr.Row():
distil_transcription = gr.components.Textbox(label="Distil-Whisper Transcription", show_copy_button=True)
transcription = gr.components.Textbox(label="Whisper Transcription", show_copy_button=True)
button.click(
fn=transcribe,
inputs=audio,
outputs=[distil_transcription, distil_runtime, transcription, runtime],
)
demo.queue(max_size=10).launch()
|