whisperX
whisper-asr-webservice
whisperX | whisper-asr-webservice | |
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24 | 11 | |
9,064 | 1,644 | |
- | - | |
8.4 | 7.8 | |
7 days ago | 9 days ago | |
Python | Python | |
BSD 4-Clause "Original" or "Old" License | MIT License |
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whisperX
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Easy video transcription and subtitling with Whisper, FFmpeg, and Python
It uses this, which does support diarization: https://github.com/m-bain/whisperX
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SOTA ASR Tooling: Long-Form Transcription
Author compared various whisper implementation
"We found that WhisperX is the best framework for transcribing long audio files efficiently and accurately. It’s much better than using the standard openai-whisper library."
https://github.com/m-bain/whisperX
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Deploying whisperX on AWS SageMaker as Asynchronous Endpoint
import os # Directory and file paths dir_path = './models-v1' inference_file_path = os.path.join(dir_path, 'code/inference.py') requirements_file_path = os.path.join(dir_path, 'code/requirements.txt') # Create the directory structure os.makedirs(os.path.dirname(inference_file_path), exist_ok=True) # Inference.py content inference_content = '''# inference.py # inference.py import io import json import logging import os import tempfile import time import boto3 import torch import whisperx DEVICE = 'cuda' if torch.cuda.is_available() else 'cpu' s3 = boto3.client('s3') def model_fn(model_dir, context=None): """ Load and return the WhisperX model necessary for audio transcription. """ print("Entering model_fn") logging.info("Loading WhisperX model") model = whisperx.load_model(whisper_arch=f"{model_dir}/guillaumekln/faster-whisper-large-v2", device=DEVICE, language="en", compute_type="float16", vad_options={'model_fp': f"{model_dir}/whisperx/vad/pytorch_model.bin"}) print("Loaded WhisperX model") print("Exiting model_fn with model loaded") return { 'model': model } def input_fn(request_body, request_content_type): """ Process and load audio from S3, given the request body containing S3 bucket and key. """ print("Entering input_fn") if request_content_type != 'application/json': raise ValueError("Invalid content type. Must be application/json") request = json.loads(request_body) s3_bucket = request['s3bucket'] s3_key = request['s3key'] # Download the file from S3 temp_file = tempfile.NamedTemporaryFile(delete=False) s3.download_file(Bucket=s3_bucket, Key=s3_key, Filename=temp_file.name) print(f"Downloaded audio from S3: {s3_bucket}/{s3_key}") print("Exiting input_fn") return temp_file.name def predict_fn(input_data, model, context=None): """ Perform transcription on the provided audio file and delete the file afterwards. """ print("Entering predict_fn") start_time = time.time() whisperx_model = model['model'] logging.info("Loading audio") audio = whisperx.load_audio(input_data) logging.info("Transcribing audio") transcription_result = whisperx_model.transcribe(audio, batch_size=16) try: os.remove(input_data) # input_data contains the path to the temp file print(f"Temporary file {input_data} deleted.") except OSError as e: print(f"Error: {input_data} : {e.strerror}") end_time = time.time() elapsed_time = end_time - start_time logging.info(f"Transcription took {int(elapsed_time)} seconds") print(f"Exiting predict_fn, processing took {int(elapsed_time)} seconds") return transcription_result def output_fn(prediction, accept, context=None): """ Prepare the prediction result for the response. """ print("Entering output_fn") if accept != "application/json": raise ValueError("Accept header must be application/json") response_body = json.dumps(prediction) print("Exiting output_fn with response prepared") return response_body, accept ''' # Write the inference.py file with open(inference_file_path, 'w') as file: file.write(inference_content) # Requirements.txt content requirements_content = '''speechbrain==0.5.16 faster-whisper==0.7.1 git+https://github.com/m-bain/whisperx.git@1b092de19a1878a8f138f665b1467ca21b076e7e ffmpeg-python ''' # Write the requirements.txt file with open(requirements_file_path, 'w') as file: file.write(requirements_content)
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OpenVoice: Versatile Instant Voice Cloning
Whisper doesn't, but WhisperX <https://github.com/m-bain/whisperX/> does. I am using it right now and it's perfectly serviceable.
For reference, I'm transcribing research-related podcasts, meaning speech doesn't overlap a lot, which would be a problem for WhisperX from what I understand. There's also a lot of accents, which are straining on Whisper (though it's also doing well), but surely help WhisperX. It did have issues with figuring number of speakers on it's own, but that wasn't a problem for my use case.
- FLaNK 15 Jan 2024
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Subtitle is now open-source
I've had good results with whisperx when I needed to generate captions. https://github.com/m-bain/whisperX
There is currently a problem with diarization, but otherwise, it is SOTA.
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Insanely Fast Whisper: Transcribe 300 minutes of audio in less than 98 seconds
https://github.com/m-bain/whisperX/issues/569
WhisperX with the new model. It's not fast.
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Distil-Whisper: distilled version of Whisper that is 6 times faster, 49% smaller
How much faster in real wall-clock time is this in batched data than https://github.com/m-bain/whisperX ?
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whisper self hosted what's the most cost-efficient way
Checkout whisperx
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Whisper Turbo: transcribe 20x faster than realtime using Rust and WebGPU
Neat to see a new implementation, although I'll note that for those looking for a drop-in replacement for the whisper library, I believe that both faster-whisper https://github.com/guillaumekln/faster-whisper and https://github.com/m-bain/whisperX are easier (PyTorch-based, doesn't require a web browser), and a lot faster (WhisperX is up to 70X realtime).
whisper-asr-webservice
- How I converted a podcast into a knowledge base using Orama search and OpenAI whisper and Astro
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Bazarr AI subs
Check https://github.com/openai/whisper & https://github.com/ahmetoner/whisper-asr-webservice
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Bulk download subtitles
I see that bazarr had already been mentioned. If there are no subtitles available, you can also generate the subtitles by connecting bazarr to the AI model whisper which you can self host locally. I run everything in containers, tried it a few times and it works quite well for me! It does however use some computational resources to generate the subtitles, how long processing takes depends on the chosen model accuracy.
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Writeout.ai – Transcribe and translate any audio files. Free and open source
You (essentially) need GPU but here you go:
https://github.com/ahmetoner/whisper-asr-webservice
For your requirements the medium.en model (max) should be satisfactory.
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Whispers AI Modular Future
What utilities related to Whisper do you wish existed? What have you had to build yourself?
On the end user application side, I wish there was something that let me pick a podcast of my choosing, get it fully transcribed, and get an embeddings search plus answer q&a on top of that podcast or set of chosen podcasts. I've seen ones for specific podcasts, but I'd like one where I can choose the podcast. (Probably won't build it)
Also on the end user side, I wish there was an Otter alternative (still paid $30/mo, but unlimited minutes per month) that had longer transcription limits. (Started building this, not much interest from users though)
Things I've seen on the dev tool side:
Gladia (API call version of Whisper)
Whisper.cpp
Whisper webservice (https://github.com/ahmetoner/whisper-asr-webservice) - via this thread
Live microphone demo (not real time, it still does it in chunks) https://github.com/mallorbc/whisper_mic
Streamlit UI https://github.com/hayabhay/whisper-ui
Whisper playground https://github.com/saharmor/whisper-playground
Real time whisper https://github.com/shirayu/whispering
Whisper as a service https://github.com/schibsted/WAAS
Improved timestamps and speaker identification https://github.com/m-bain/whisperX
MacWhisper https://goodsnooze.gumroad.com/l/macwhisper
Crossplatform desktop Whisper that supports semi-realtime https://github.com/chidiwilliams/buzz
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I made a free transcription service powered by Whisper AI
I think there's been talk to do speaker diarization with whisper-asr-webservice[0] which is also written in python and should be able to make use of goodies such as pyannote-audio, py-webrtcvad, etc.
Whisper is great but at the point we get to kludging various things together it starts to make more sense to use something like Nvidia NeMo[1] which was built with all of this in mind and more
[0] - https://github.com/ahmetoner/whisper-asr-webservice
[1] - https://github.com/NVIDIA/NeMo
- whisper-asr-webservice-client - A self-hosted OpenAI Whisper API client
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Show HN: A self-hosted OpenAI Whisper API client
(read the docs in the repo)
In terms of me not storing your data for this (I don't) I guess you'll just have to trust me?
[0] - https://github.com/ahmetoner/whisper-asr-webservice
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[P] OpenAI Whisper ASR Webservice API released
For more details: https://github.com/ahmetoner/whisper-asr-webservice
What are some alternatives?
whisper.cpp - Port of OpenAI's Whisper model in C/C++
whisper - Robust Speech Recognition via Large-Scale Weak Supervision
faster-whisper - Faster Whisper transcription with CTranslate2
generate-subtitles - Generate transcripts for audio and video content with a user friendly UI, powered by Open AI's Whisper with automatic translations and download videos automatically with yt-dlp integration
insanely-fast-whisper - Incredibly fast Whisper-large-v3
whisper-asr-webservice-client
openai-whisper-cpu - Improving transcription performance of OpenAI Whisper for CPU based deployment
gitbar-2023 - New release of gitbar website
ControlNet - Let us control diffusion models!