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thealphamerc
commited on
Commit
•
7a97be1
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Parent(s):
Initial commit 🎉
Browse files- Output/audio.txt +1 -0
- Output/audio2.txt +1 -0
- Output/audio3.json +173 -0
- Output/audio3.txt +1 -0
- app.py +87 -0
- data/audio.wav +0 -0
- data/audio2.mp3 +0 -0
- data/audio3.wav +0 -0
- flagged/Audio file/0.wav +0 -0
- flagged/log.csv +2 -0
- requirements.txt +2 -0
- trans.py +122 -0
Output/audio.txt
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Quatlin, quatlin quatlin quatlin quatlin. Anti-six.
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Output/audio2.txt
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to gain life in all that...
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Output/audio3.json
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[
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{
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"seek": 0,
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"start": 0.0,
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"end": 1.52,
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"text": " Come and sit on a rock.",
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"tokens": [
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],
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"temperature": 0.0,
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"avg_logprob": -0.34572365704704733,
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"compression_ratio": 1.356164383561644,
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"no_speech_prob": 0.01958448439836502
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},
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"id": 1,
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"seek": 0,
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"start": 1.52,
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"end": 5.08,
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"text": " Overlooking the river's blow, he wears a hat and some glasses.",
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"tokens": [
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"temperature": 0.0,
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"avg_logprob": -0.34572365704704733,
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"compression_ratio": 1.356164383561644,
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"no_speech_prob": 0.01958448439836502
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"id": 2,
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"seek": 0,
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"start": 5.08,
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"end": 7.36,
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"text": " A smile on his face.",
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"tokens": [
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"temperature": 0.0,
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"compression_ratio": 1.356164383561644,
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"no_speech_prob": 0.01958448439836502
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},
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"id": 3,
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"seek": 0,
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"start": 7.36,
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"end": 8.56,
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"text": " He's not lost.",
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"tokens": [
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"temperature": 0.0,
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"avg_logprob": -0.34572365704704733,
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"compression_ratio": 1.356164383561644,
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"no_speech_prob": 0.01958448439836502
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},
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"seek": 0,
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"start": 8.56,
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"end": 10.4,
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"text": " The water rushes by.",
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"tokens": [
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"temperature": 0.0,
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"seek": 0,
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"start": 10.4,
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"end": 12.08,
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"text": " A constant sound.",
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"temperature": 0.0,
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"avg_logprob": -0.34572365704704733,
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"compression_ratio": 1.356164383561644,
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"no_speech_prob": 0.01958448439836502
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},
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"id": 6,
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"seek": 0,
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"start": 12.08,
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"end": 13.68,
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"text": " It takes in the view.",
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"temperature": 0.0,
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"avg_logprob": -0.34572365704704733,
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"compression_ratio": 1.356164383561644,
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"no_speech_prob": 0.01958448439836502
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},
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{
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"id": 7,
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"seek": 0,
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"start": 13.68,
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"end": 14.48,
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"text": " The mountains.",
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"temperature": 0.0,
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"avg_logprob": -0.34572365704704733,
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"compression_ratio": 1.356164383561644,
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"no_speech_prob": 0.01958448439836502
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}
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]
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Output/audio3.txt
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Come and sit on a rock. Overlooking the river's blow, he wears a hat and some glasses. A smile on his face. He's not lost. The water rushes by. A constant sound. It takes in the view. The mountains.
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app.py
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# Basic script for using the OpenAI Whisper model to transcribe a video file. You can uncomment whichever model you want to use.
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# Author: ThioJoe ( https://github.com/ThioJoe )
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# Required third party packages: whisper
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# See instructions for setup here: https://github.com/openai/whisper#setup
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# - You can use the below command to pull the repo and install dependencies, then just put this script in the repo directory:
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# pip install git+https://github.com/openai/whisper.git
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import whisper
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import io
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import time
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import os
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import json
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import pathlib
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# Choose model to use by uncommenting
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# modelName = "tiny.en"
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modelName = "base.en"
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# modelName = "small.en"
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# modelName = "medium.en"
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# modelName = "large-v2"
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# Other Variables
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# (bool) Whether to export the segment data to a json file. Will include word level timestamps if word_timestamps is True.
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exportTimestampData = True
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outputFolder = "Output"
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# ----- Select variables for transcribe method -----
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# audio: path to audio file
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verbose = True # (bool): Whether to display the text being decoded to the console. If True, displays all the details, If False, displays minimal details. If None, does not display anything
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language = "english" # Language of audio file
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# (bool): Extract word-level timestamps using the cross-attention pattern and dynamic time warping, and include the timestamps for each word in each segment.
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word_timestamps = False
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# initial_prompt="" # (optional str): Optional text to provide as a prompt for the first window. This can be used to provide, or "prompt-engineer" a context for transcription, e.g. custom vocabularies or proper nouns to make it more likely to predict those word correctly.
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# -------------------------------------------------------------------------
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print(f"Using Model: {modelName}")
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filePath = input("Path to File Being Transcribed: ")
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filePath = filePath.strip("\"")
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if not os.path.exists(filePath):
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print("Problem Getting File...")
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input("Press Enter to Exit...")
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exit()
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# If output folder does not exist, create it
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if not os.path.exists(outputFolder):
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os.makedirs(outputFolder)
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print("Created Output Folder.\n")
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# Get filename stem using pathlib (filename without extension)
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fileNameStem = pathlib.Path(filePath).stem
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resultFileName = f"{fileNameStem}.txt"
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jsonFileName = f"{fileNameStem}.json"
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model = whisper.load_model(modelName)
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start = time.time()
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# ---------------------------------------------------
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result = model.transcribe(audio=filePath, language=language,
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word_timestamps=word_timestamps, verbose=verbose)
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# ---------------------------------------------------
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end = time.time()
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elapsed = float(end - start)
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# Save transcription text to file
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print("\nWriting transcription to file...")
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with open(os.path.join(outputFolder, resultFileName), "w", encoding="utf-8") as file:
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file.write(result["text"])
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print("Finished writing transcription file.")
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# Sav
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# e the segments data to json file
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# if word_timestamps == True:
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if exportTimestampData == True:
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print("\nWriting segment data to file...")
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with open(os.path.join(outputFolder, jsonFileName), "w", encoding="utf-8") as file:
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segmentsData = result["segments"]
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json.dump(segmentsData, file, indent=4)
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print("Finished writing segment data file.")
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elapsedMinutes = str(round(elapsed/60, 2))
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print(f"\nElapsed Time With {modelName} Model: {elapsedMinutes} Minutes")
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input("Press Enter to exit...")
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exit()
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data/audio.wav
ADDED
Binary file (172 kB). View file
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data/audio2.mp3
ADDED
Binary file (35.4 kB). View file
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data/audio3.wav
ADDED
Binary file (695 kB). View file
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flagged/Audio file/0.wav
ADDED
Binary file (693 kB). View file
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flagged/log.csv
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Audio file,Transcription,timestamp
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Audio file/0.wav,No audio file submitted! Please upload an audio file before submitting your request.,2023-04-26 23:19:33.132801
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requirements.txt
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whisper
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gradio===3.27.0
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trans.py
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import logging
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from subprocess import call
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import gradio as gr
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import os
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# from transformers.pipelines.audio_utils import ffmpeg_read
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import whisper
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logger = logging.getLogger("whisper-jax-app")
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logger.setLevel(logging.INFO)
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11 |
+
ch = logging.StreamHandler()
|
12 |
+
ch.setLevel(logging.INFO)
|
13 |
+
formatter = logging.Formatter(
|
14 |
+
"%(asctime)s;%(levelname)s;%(message)s", "%Y-%m-%d %H:%M:%S")
|
15 |
+
ch.setFormatter(formatter)
|
16 |
+
logger.addHandler(ch)
|
17 |
+
|
18 |
+
|
19 |
+
BATCH_SIZE = 16
|
20 |
+
CHUNK_LENGTH_S = 30
|
21 |
+
NUM_PROC = 8
|
22 |
+
FILE_LIMIT_MB = 1000
|
23 |
+
YT_ATTEMPT_LIMIT = 3
|
24 |
+
|
25 |
+
|
26 |
+
def run_cmd(command):
|
27 |
+
try:
|
28 |
+
print(command)
|
29 |
+
call(command)
|
30 |
+
except KeyboardInterrupt:
|
31 |
+
print("Process interrupted")
|
32 |
+
sys.exit(1)
|
33 |
+
|
34 |
+
|
35 |
+
def inference(text):
|
36 |
+
cmd = ['tts', '--text', text]
|
37 |
+
run_cmd(cmd)
|
38 |
+
return 'tts_output.wav'
|
39 |
+
|
40 |
+
|
41 |
+
model = whisper.load_model("base")
|
42 |
+
|
43 |
+
inputs = gr.components.Audio(type="filepath", label="Add audio file")
|
44 |
+
outputs = gr.components.Textbox()
|
45 |
+
title = "Audio To text⚡️"
|
46 |
+
description = "An example of using TTS to generate speech from text."
|
47 |
+
article = ""
|
48 |
+
examples = [
|
49 |
+
[""]
|
50 |
+
]
|
51 |
+
|
52 |
+
|
53 |
+
def transcribe(inputs):
|
54 |
+
print('Inputs: ', inputs)
|
55 |
+
# print('Text: ', text)
|
56 |
+
# progress(0, desc="Loading audio file...")
|
57 |
+
if inputs is None:
|
58 |
+
logger.warning("No audio file")
|
59 |
+
return "No audio file submitted! Please upload an audio file before submitting your request."
|
60 |
+
file_size_mb = os.stat(inputs).st_size / (1024 * 1024)
|
61 |
+
if file_size_mb > FILE_LIMIT_MB:
|
62 |
+
logger.warning("Max file size exceeded")
|
63 |
+
return f"File size exceeds file size limit. Got file of size {file_size_mb:.2f}MB for a limit of {FILE_LIMIT_MB}MB."
|
64 |
+
|
65 |
+
# with open(inputs, "rb") as f:
|
66 |
+
# inputs = f.read()
|
67 |
+
|
68 |
+
# load audio and pad/trim it to fit 30 seconds
|
69 |
+
result = model.transcribe(audio=inputs, language='hindi',
|
70 |
+
word_timestamps=False, verbose=True)
|
71 |
+
# ---------------------------------------------------
|
72 |
+
|
73 |
+
print(result["text"])
|
74 |
+
return result["text"]
|
75 |
+
|
76 |
+
|
77 |
+
audio_chunked = gr.Interface(
|
78 |
+
fn=transcribe,
|
79 |
+
inputs=inputs,
|
80 |
+
outputs=outputs,
|
81 |
+
allow_flagging="never",
|
82 |
+
title=title,
|
83 |
+
description=description,
|
84 |
+
article=article,
|
85 |
+
)
|
86 |
+
|
87 |
+
microphone_chunked = gr.Interface(
|
88 |
+
fn=transcribe,
|
89 |
+
inputs=[
|
90 |
+
gr.inputs.Audio(source="microphone",
|
91 |
+
optional=True, type="filepath"),
|
92 |
+
],
|
93 |
+
outputs=[
|
94 |
+
gr.outputs.Textbox(label="Transcription").style(
|
95 |
+
show_copy_button=True),
|
96 |
+
],
|
97 |
+
allow_flagging="never",
|
98 |
+
title=title,
|
99 |
+
description=description,
|
100 |
+
article=article,
|
101 |
+
)
|
102 |
+
|
103 |
+
demo = gr.Blocks()
|
104 |
+
with demo:
|
105 |
+
gr.TabbedInterface([audio_chunked, microphone_chunked], [
|
106 |
+
"Audio File", "Microphone"])
|
107 |
+
demo.queue(concurrency_count=1, max_size=5)
|
108 |
+
demo.launch(show_api=False)
|
109 |
+
|
110 |
+
|
111 |
+
# gr.Interface(
|
112 |
+
# inference,
|
113 |
+
# inputs,
|
114 |
+
# outputs,
|
115 |
+
# verbose=True,
|
116 |
+
# title=title,
|
117 |
+
# description=description,
|
118 |
+
# article=article,
|
119 |
+
# examples=examples,
|
120 |
+
# enable_queue=True,
|
121 |
+
|
122 |
+
# ).launch(share=True, debug=True)
|