Spaces:
Runtime error
Runtime error
File size: 11,020 Bytes
6c82c95 |
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 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 |
import gradio as gr
import json
from difflib import Differ
import ffmpeg
import os
from pathlib import Path
import time
# Set true if you're using huggingface inference API API https://huggingface.co/inference-api
API_BACKEND = True
# MODEL = 'facebook/wav2vec2-large-960h-lv60-self'
# MODEL = "facebook/wav2vec2-large-960h"
MODEL = "facebook/wav2vec2-base-960h"
# MODEL = "patrickvonplaten/wav2vec2-large-960h-lv60-self-4-gram"
if API_BACKEND:
from dotenv import load_dotenv
import requests
import base64
import asyncio
load_dotenv(Path(".env"))
HF_TOKEN = os.environ["HF_TOKEN"]
headers = {"Authorization": f"Bearer {HF_TOKEN}"}
API_URL = f'https://api-inference.huggingface.co/models/{MODEL}'
else:
import torch
from transformers import pipeline
# is cuda available?
cuda = torch.device(
'cuda:0') if torch.cuda.is_available() else torch.device('cpu')
device = 0 if torch.cuda.is_available() else -1
speech_recognizer = pipeline(
task="automatic-speech-recognition",
model=f'{MODEL}',
tokenizer=f'{MODEL}',
framework="pt",
device=device,
)
videos_out_path = Path("./videos_out")
videos_out_path.mkdir(parents=True, exist_ok=True)
samples_data = sorted(Path('examples').glob('*.json'))
SAMPLES = []
for file in samples_data:
with open(file) as f:
sample = json.load(f)
SAMPLES.append(sample)
VIDEOS = list(map(lambda x: [x['video']], SAMPLES))
total_inferences_since_reboot = 0
total_cuts_since_reboot = 0
async def speech_to_text(video_file_path):
"""
Takes a video path to convert to audio, transcribe audio channel to text and char timestamps
Using https://huggingface.co/tasks/automatic-speech-recognition pipeline
"""
global total_inferences_since_reboot
if(video_file_path == None):
raise ValueError("Error no video input")
video_path = Path(video_file_path)
try:
# convert video to audio 16k using PIPE to audio_memory
audio_memory, _ = ffmpeg.input(video_path).output(
'-', format="wav", ac=1, ar='16k').overwrite_output().global_args('-loglevel', 'quiet').run(capture_stdout=True)
except Exception as e:
raise RuntimeError("Error converting video to audio")
last_time = time.time()
if API_BACKEND:
# Using Inference API https://huggingface.co/inference-api
# try twice, because the model must be loaded
for i in range(10):
for tries in range(4):
print(f'Transcribing from API attempt {tries}')
try:
inference_reponse = query_api(audio_memory)
transcription = inference_reponse["text"].lower()
timestamps = [[chunk["text"].lower(), chunk["timestamp"][0], chunk["timestamp"][1]]
for chunk in inference_reponse['chunks']]
total_inferences_since_reboot += 1
print("\n\ntotal_inferences_since_reboot: ",
total_inferences_since_reboot, "\n\n")
return (transcription, transcription, timestamps)
except:
if 'error' in inference_reponse and 'estimated_time' in inference_reponse:
wait_time = inference_reponse['estimated_time']
print("Waiting for model to load....", wait_time)
# wait for loading model
# 5 seconds plus for certanty
await asyncio.sleep(wait_time + 5.0)
elif 'error' in inference_reponse:
raise RuntimeError("Error Fetching API",
inference_reponse['error'])
else:
break
else:
raise RuntimeError(inference_reponse, "Error Fetching API")
else:
try:
print(f'Transcribing via local model')
output = speech_recognizer(
audio_memory, return_timestamps="char", chunk_length_s=10, stride_length_s=(4, 2))
transcription = output["text"].lower()
timestamps = [[chunk["text"].lower(), chunk["timestamp"][0].tolist(), chunk["timestamp"][1].tolist()]
for chunk in output['chunks']]
total_inferences_since_reboot += 1
print("\n\ntotal_inferences_since_reboot: ",
total_inferences_since_reboot, "\n\n")
return (transcription, transcription, timestamps)
except Exception as e:
raise RuntimeError("Error Running inference with local model", e)
def cut_timestamps_to_video(video_in, transcription, text_in, timestamps):
"""
Given original video input, text transcript + timestamps,
and edit ext cuts video segments into a single video
"""
global total_cuts_since_reboot
video_path = Path(video_in)
video_file_name = video_path.stem
if(video_in == None or text_in == None or transcription == None):
raise ValueError("Inputs undefined")
d = Differ()
# compare original transcription with edit text
diff_chars = d.compare(transcription, text_in)
# remove all text aditions from diff
filtered = list(filter(lambda x: x[0] != '+', diff_chars))
# filter timestamps to be removed
# timestamps_to_cut = [b for (a,b) in zip(filtered, timestamps_var) if a[0]== '-' ]
# return diff tokes and cutted video!!
# groupping character timestamps so there are less cuts
idx = 0
grouped = {}
for(a, b) in zip(filtered, timestamps):
if a[0] != '-':
if idx in grouped:
grouped[idx].append(b)
else:
grouped[idx] = []
grouped[idx].append(b)
else:
idx += 1
# after grouping, gets the lower and upter start and time for each group
timestamps_to_cut = [[v[0][1], v[-1][2]] for v in grouped.values()]
between_str = '+'.join(
map(lambda t: f'between(t,{t[0]},{t[1]})', timestamps_to_cut))
if timestamps_to_cut:
video_file = ffmpeg.input(video_in)
video = video_file.video.filter(
"select", f'({between_str})').filter("setpts", "N/FRAME_RATE/TB")
audio = video_file.audio.filter(
"aselect", f'({between_str})').filter("asetpts", "N/SR/TB")
output_video = f'./videos_out/{video_file_name}.mp4'
ffmpeg.concat(video, audio, v=1, a=1).output(
output_video).overwrite_output().global_args('-loglevel', 'quiet').run()
else:
output_video = video_in
tokens = [(token[2:], token[0] if token[0] != " " else None)
for token in filtered]
total_cuts_since_reboot += 1
print("\n\ntotal_cuts_since_reboot: ", total_cuts_since_reboot, "\n\n")
return (tokens, output_video)
def query_api(audio_bytes: bytes):
"""
Query for Huggingface Inference API for Automatic Speech Recognition task
"""
payload = json.dumps({
"inputs": base64.b64encode(audio_bytes).decode("utf-8"),
"parameters": {
"return_timestamps": "char",
"chunk_length_s": 10,
"stride_length_s": [4, 2]
},
"options": {"use_gpu": False}
}).encode("utf-8")
response = requests.request(
"POST", API_URL, headers=headers, data=payload)
json_reponse = json.loads(response.content.decode("utf-8"))
return json_reponse
# ---- Gradio Layout -----
video_in = gr.Video(label="Video file")
text_in = gr.Textbox(label="Transcription", lines=10, interactive=True)
video_out = gr.Video(label="Video Out")
diff_out = gr.HighlightedText(label="Cuts Diffs", combine_adjacent=True)
examples = gr.components.Dataset(
components=[video_in], samples=VIDEOS, type="index")
demo = gr.Blocks(enable_queue=True, css='''
#cut_btn, #reset_btn { align-self:stretch; }
#\\31 3 { max-width: 540px; }
.output-markdown {max-width: 65ch !important;}
''')
demo.encrypt = False
with demo:
transcription_var = gr.Variable()
timestamps_var = gr.Variable()
with gr.Row():
with gr.Column():
gr.Markdown('''
# Edit Video By Editing Text
This project is a quick proof of concept of a simple video editor where the edits
are made by editing the audio transcription.
Using the [Huggingface Automatic Speech Recognition Pipeline](https://huggingface.co/tasks/automatic-speech-recognition)
with a fine tuned [Wav2Vec2 model using Connectionist Temporal Classification (CTC)](https://huggingface.co/facebook/wav2vec2-large-960h-lv60-self)
you can predict not only the text transcription but also the [character or word base timestamps](https://huggingface.co/docs/transformers/v4.19.2/en/main_classes/pipelines#transformers.AutomaticSpeechRecognitionPipeline.__call__.return_timestamps)
''')
with gr.Row():
examples.render()
def load_example(id):
video = SAMPLES[id]['video']
transcription = SAMPLES[id]['transcription'].lower()
timestamps = SAMPLES[id]['timestamps']
return (video, transcription, transcription, timestamps)
examples.click(
load_example,
inputs=[examples],
outputs=[video_in, text_in, transcription_var, timestamps_var],
queue=False)
with gr.Row():
with gr.Column():
video_in.render()
transcribe_btn = gr.Button("Transcribe Audio")
transcribe_btn.click(speech_to_text, [video_in], [
text_in, transcription_var, timestamps_var])
with gr.Row():
gr.Markdown('''
### Now edit as text
After running the video transcription, you can make cuts to the text below (only cuts, not additions!)''')
with gr.Row():
with gr.Column():
text_in.render()
with gr.Row():
cut_btn = gr.Button("Cut to video", elem_id="cut_btn")
# send audio path and hidden variables
cut_btn.click(cut_timestamps_to_video, [
video_in, transcription_var, text_in, timestamps_var], [diff_out, video_out])
reset_transcription = gr.Button(
"Reset to last trascription", elem_id="reset_btn")
reset_transcription.click(
lambda x: x, transcription_var, text_in)
with gr.Column():
video_out.render()
diff_out.render()
with gr.Row():
gr.Markdown('''
#### Video Credits
1. [Cooking](https://vimeo.com/573792389)
1. [Shia LaBeouf "Just Do It"](https://www.youtube.com/watch?v=n2lTxIk_Dr0)
1. [Mark Zuckerberg & Yuval Noah Harari in Conversation](https://www.youtube.com/watch?v=Boj9eD0Wug8)
''')
if __name__ == "__main__":
demo.launch(debug=True)
|