Spaces:
Build error
Build error
File size: 10,202 Bytes
9b43ccd ce7b54d |
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 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 |
import base64
import math
import os
import time
from multiprocessing import Pool
import gradio as gr
import numpy as np
import pytube
import requests
from processing_whisper import WhisperPrePostProcessor
from transformers.models.whisper.tokenization_whisper import TO_LANGUAGE_CODE
from transformers.pipelines.audio_utils import ffmpeg_read
title = "Whisper JAX: The Fastest Whisper API ⚡️"
description = """Whisper JAX is an optimised implementation of the [Whisper model](https://huggingface.co/openai/whisper-large-v2) by OpenAI. It runs on JAX with a TPU v4-8 in the backend. Compared to PyTorch on an A100 GPU, it is over [**70x faster**](https://github.com/sanchit-gandhi/whisper-jax#benchmarks), making it the fastest Whisper API available.
Note that using microphone or audio file requires the audio input to be transferred from the Gradio demo to the TPU, which for large audio files can be slow. We recommend using YouTube where possible, since this directly downloads the audio file to the TPU, skipping the file transfer step.
"""
API_URL = os.getenv("API_URL")
API_URL_FROM_FEATURES = os.getenv("API_URL_FROM_FEATURES")
article = "Whisper large-v2 model by OpenAI. Backend running JAX on a TPU v4-8 through the generous support of the [TRC](https://sites.research.google/trc/about/) programme. Whisper JAX [code](https://github.com/sanchit-gandhi/whisper-jax) and Gradio demo by 🤗 Hugging Face."
language_names = sorted(TO_LANGUAGE_CODE.keys())
CHUNK_LENGTH_S = 30
BATCH_SIZE = 16
NUM_PROC = 16
FILE_LIMIT_MB = 1000
def query(payload):
response = requests.post(API_URL, json=payload)
return response.json(), response.status_code
def inference(inputs, task=None, return_timestamps=False):
payload = {"inputs": inputs, "task": task, "return_timestamps": return_timestamps}
data, status_code = query(payload)
if status_code == 200:
text = data["text"]
else:
text = data["detail"]
timestamps = data.get("chunks")
if timestamps is not None:
timestamps = [
f"[{format_timestamp(chunk['timestamp'][0])} -> {format_timestamp(chunk['timestamp'][1])}] {chunk['text']}"
for chunk in timestamps
]
text = "\n".join(str(feature) for feature in timestamps)
return text
def chunked_query(payload):
response = requests.post(API_URL_FROM_FEATURES, json=payload)
return response.json()
def forward(batch, task=None, return_timestamps=False):
feature_shape = batch["input_features"].shape
batch["input_features"] = base64.b64encode(batch["input_features"].tobytes()).decode()
outputs = chunked_query(
{"batch": batch, "task": task, "return_timestamps": return_timestamps, "feature_shape": feature_shape}
)
outputs["tokens"] = np.asarray(outputs["tokens"])
return outputs
def identity(batch):
return batch
# Copied from https://github.com/openai/whisper/blob/c09a7ae299c4c34c5839a76380ae407e7d785914/whisper/utils.py#L50
def format_timestamp(seconds: float, always_include_hours: bool = False, decimal_marker: str = "."):
if seconds is not None:
milliseconds = round(seconds * 1000.0)
hours = milliseconds // 3_600_000
milliseconds -= hours * 3_600_000
minutes = milliseconds // 60_000
milliseconds -= minutes * 60_000
seconds = milliseconds // 1_000
milliseconds -= seconds * 1_000
hours_marker = f"{hours:02d}:" if always_include_hours or hours > 0 else ""
return f"{hours_marker}{minutes:02d}:{seconds:02d}{decimal_marker}{milliseconds:03d}"
else:
# we have a malformed timestamp so just return it as is
return seconds
if __name__ == "__main__":
processor = WhisperPrePostProcessor.from_pretrained("openai/whisper-large-v2")
stride_length_s = CHUNK_LENGTH_S / 6
chunk_len = round(CHUNK_LENGTH_S * processor.feature_extractor.sampling_rate)
stride_left = stride_right = round(stride_length_s * processor.feature_extractor.sampling_rate)
step = chunk_len - stride_left - stride_right
pool = Pool(NUM_PROC)
def tqdm_generate(inputs: dict, task: str, return_timestamps: bool, progress: gr.Progress):
inputs_len = inputs["array"].shape[0]
all_chunk_start_idx = np.arange(0, inputs_len, step)
num_samples = len(all_chunk_start_idx)
num_batches = math.ceil(num_samples / BATCH_SIZE)
dummy_batches = list(
range(num_batches)
) # Gradio progress bar not compatible with generator, see https://github.com/gradio-app/gradio/issues/3841
dataloader = processor.preprocess_batch(inputs, chunk_length_s=CHUNK_LENGTH_S, batch_size=BATCH_SIZE)
progress(0, desc="Pre-processing audio file...")
dataloader = pool.map(identity, dataloader)
model_outputs = []
start_time = time.time()
# iterate over our chunked audio samples
for batch, _ in zip(dataloader, progress.tqdm(dummy_batches, desc="Transcribing...")):
model_outputs.append(forward(batch, task=task, return_timestamps=return_timestamps))
runtime = time.time() - start_time
post_processed = processor.postprocess(model_outputs, return_timestamps=return_timestamps)
text = post_processed["text"]
timestamps = post_processed.get("chunks")
if timestamps is not None:
timestamps = [
f"[{format_timestamp(chunk['timestamp'][0])} -> {format_timestamp(chunk['timestamp'][1])}] {chunk['text']}"
for chunk in timestamps
]
text = "\n".join(str(feature) for feature in timestamps)
return text, runtime
def transcribe_chunked_audio(inputs, task, return_timestamps, progress=gr.Progress()):
progress(0, desc="Loading audio file...")
file_size_mb = os.stat(inputs).st_size / (1024 * 1024)
if file_size_mb > FILE_LIMIT_MB:
raise gr.Error(
f"File size exceeds file size limit. Got file of size {file_size_mb:.2f}MB for a limit of {FILE_LIMIT_MB}MB."
)
with open(inputs, "rb") as f:
inputs = f.read()
inputs = ffmpeg_read(inputs, processor.feature_extractor.sampling_rate)
inputs = {"array": inputs, "sampling_rate": processor.feature_extractor.sampling_rate}
text, runtime = tqdm_generate(inputs, task=task, return_timestamps=return_timestamps, progress=progress)
return text, runtime
def _return_yt_html_embed(yt_url):
video_id = yt_url.split("?v=")[-1]
HTML_str = (
f'<center> <iframe width="500" height="320" src="https://www.youtube.com/embed/{video_id}"> </iframe>'
" </center>"
)
return HTML_str
def transcribe_youtube(yt_url, task, return_timestamps, progress=gr.Progress(), max_filesize=75.0):
progress(0, desc="Loading audio file...")
html_embed_str = _return_yt_html_embed(yt_url)
try:
yt = pytube.YouTube(yt_url)
stream = yt.streams.filter(only_audio=True)[0]
except KeyError:
raise gr.Error("An error occurred while loading the YouTube video. Please try again.")
if stream.filesize_mb > max_filesize:
raise gr.Error(f"Maximum YouTube file size is {max_filesize}MB, got {stream.filesize_mb:.2f}MB.")
stream.download(filename="audio.mp3")
with open("audio.mp3", "rb") as f:
inputs = f.read()
inputs = ffmpeg_read(inputs, processor.feature_extractor.sampling_rate)
inputs = {"array": inputs, "sampling_rate": processor.feature_extractor.sampling_rate}
text, runtime = tqdm_generate(inputs, task=task, return_timestamps=return_timestamps, progress=progress)
return html_embed_str, text, runtime
microphone_chunked = gr.Interface(
fn=transcribe_chunked_audio,
inputs=[
gr.inputs.Audio(source="microphone", optional=True, type="filepath"),
gr.inputs.Radio(["transcribe", "translate"], label="Task", default="transcribe"),
gr.inputs.Checkbox(default=False, label="Return timestamps"),
],
outputs=[
gr.outputs.Textbox(label="Transcription").style(show_copy_button=True),
gr.outputs.Textbox(label="Transcription Time (s)"),
],
allow_flagging="never",
title=title,
description=description,
article=article,
)
audio_chunked = gr.Interface(
fn=transcribe_chunked_audio,
inputs=[
gr.inputs.Audio(source="upload", optional=True, label="Audio file", type="filepath"),
gr.inputs.Radio(["transcribe", "translate"], label="Task", default="transcribe"),
gr.inputs.Checkbox(default=False, label="Return timestamps"),
],
outputs=[
gr.outputs.Textbox(label="Transcription").style(show_copy_button=True),
gr.outputs.Textbox(label="Transcription Time (s)"),
],
allow_flagging="never",
title=title,
description=description,
article=article,
)
youtube = gr.Interface(
fn=transcribe_youtube,
inputs=[
gr.inputs.Textbox(lines=1, placeholder="Paste the URL to a YouTube video here", label="YouTube URL"),
gr.inputs.Radio(["transcribe", "translate"], label="Task", default="transcribe"),
gr.inputs.Checkbox(default=False, label="Return timestamps"),
],
outputs=[
gr.outputs.HTML(label="Video"),
gr.outputs.Textbox(label="Transcription").style(show_copy_button=True),
gr.outputs.Textbox(label="Transcription Time (s)"),
],
allow_flagging="never",
title=title,
examples=[["https://www.youtube.com/watch?v=m8u-18Q0s7I", "transcribe", False]],
cache_examples=False,
description=description,
article=article,
)
demo = gr.Blocks()
with demo:
gr.TabbedInterface([microphone_chunked, audio_chunked, youtube], ["Microphone", "Audio File", "YouTube"])
demo.queue(concurrency_count=3, max_size=5)
demo.launch(show_api=False, max_threads=10)
|