diff --git a/.gitignore b/.gitignore
new file mode 100644
index 0000000..721cb1b
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1,5 @@
+settings.toml
+audio.mp3
+uploads/*
+models/*
+venv
\ No newline at end of file
diff --git a/README.md b/README.md
index 90c4c36..5809929 100644
--- a/README.md
+++ b/README.md
@@ -1 +1,22 @@
-# local_transcription
\ No newline at end of file
+# local_transcription
+
+## Settings
+
+Make sure to set the IP, port, and HuggingFace API key in the `settings.toml` file. The huggingface API key is used to fetch the models, everything else is ran locally.
+
+```toml
+[host]
+ip="192.168.0.99"
+port=5063
+
+[settings]
+upload_folder="uploads"
+model_folder="models"
+hf_token="HUGGINGFACE API KEY"
+```
+
+## Usage
+
+`python host.py` on the server
+
+`python client_upload.py audio.mp3` on the client
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diff --git a/client_upload.py b/client_upload.py
new file mode 100644
index 0000000..5dfbf92
--- /dev/null
+++ b/client_upload.py
@@ -0,0 +1,29 @@
+import requests
+import sys
+import toml
+
+settings = toml.load('settings.toml')
+
+def upload_audio_for_transcription(api_url, file_path):
+    """
+    Uploads an audio file to the transcription API and returns the response.
+
+    Parameters:
+    - api_url: The URL of the transcription API endpoint.
+    - file_path: The path to the audio file to be uploaded.
+
+    Returns:
+    - A dictionary with the transcription result or error message.
+    """
+    files = {'file': open(file_path, 'rb')}
+    response = requests.post(api_url, files=files)
+    if response.status_code == 200:
+        return response.json()
+    else:
+        return {"error": f"Failed to transcribe audio. Status code: {response.status_code}, Message: {response.text}"}
+
+# Example usage
+api_url = f"http://{settings['host']['ip']}:{settings['host']['port']}/transcribe"
+file_path = sys.argv[1]
+result = upload_audio_for_transcription(api_url, file_path)
+print(result)
diff --git a/host.py b/host.py
new file mode 100644
index 0000000..9b611f2
--- /dev/null
+++ b/host.py
@@ -0,0 +1,66 @@
+from flask import Flask, request, jsonify
+import whisperx
+import gc
+import torch
+from werkzeug.utils import secure_filename
+import os
+import toml
+
+settings = toml.load('settings.toml')
+
+app = Flask(__name__)
+ALLOWED_EXTENSIONS = {'mp3', 'wav', 'm4a'}
+app.config['UPLOAD_FOLDER'] = settings['settings']['upload_folder']
+
+def allowed_file(filename):
+    return '.' in filename and filename.rsplit('.', 1)[1].lower() in ALLOWED_EXTENSIONS
+
+device = "cuda"
+batch_size = 16
+language = "en"
+compute_type = "float16"
+model_dir = settings['settings']['model_folder']
+HF_TOKEN = settings['settings']['hf_token']
+
+@app.route('/transcribe', methods=['POST'])
+def transcribe_audio():
+    if 'file' not in request.files:
+        return jsonify(error="No file part"), 400
+    file = request.files['file']
+    if file.filename == '':
+        return jsonify(error="No selected file"), 400
+    if file and allowed_file(file.filename):
+        if file.filename is not None:
+          filename = secure_filename(file.filename)
+          file_path = os.path.join(app.config['UPLOAD_FOLDER'], filename)
+          file.save(file_path)
+
+          # Transcribe with Whisper
+          model = whisperx.load_model("large-v2", device, language=language, compute_type=compute_type, download_root=model_dir)
+          audio = whisperx.load_audio(file_path)
+          result = model.transcribe(audio, batch_size=batch_size)
+          print(result["segments"])  # before alignment
+
+          # Align Whisper output
+          model_a, metadata = whisperx.load_align_model(language_code=result["language"], device=device)
+          result = whisperx.align(result["segments"], model_a, metadata, audio, device, return_char_alignments=False)
+          print(result["segments"])  # after alignment
+
+        # Assign speaker labels
+        diarize_model = whisperx.DiarizationPipeline(use_auth_token=HF_TOKEN, device=device)
+        diarize_segments = diarize_model(audio)
+        result = whisperx.assign_word_speakers(diarize_segments, result)
+
+        # Clean-up
+        gc.collect()
+        torch.cuda.empty_cache()
+        del model, model_a, diarize_model
+
+        os.remove(file_path)  # Remove the uploaded file after processing
+
+        return jsonify(result["segments"])
+
+    return jsonify(error="Invalid file type"), 400
+
+if __name__ == '__main__':
+    app.run(debug=True, host=settings['host']['ip'], port=settings['host']['port'])
diff --git a/requirements.txt b/requirements.txt
new file mode 100644
index 0000000..a82019d
--- /dev/null
+++ b/requirements.txt
@@ -0,0 +1,5 @@
+pytorch
+torchvision
+torchaudio
+git+https://github.com/m-bain/whisperx.git
+flask
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diff --git a/settings.toml.example b/settings.toml.example
new file mode 100644
index 0000000..4bf8dc8
--- /dev/null
+++ b/settings.toml.example
@@ -0,0 +1,8 @@
+[host]
+ip="192.168.0.99"
+port=5063
+
+[settings]
+upload_folder="uploads"
+model_folder="models"
+hf_token="HUGGINGFACE API KEY"
\ No newline at end of file
diff --git a/whisperx_test.py b/whisperx_test.py
new file mode 100644
index 0000000..831488b
--- /dev/null
+++ b/whisperx_test.py
@@ -0,0 +1,43 @@
+import whisperx
+import gc
+
+device = "cuda"
+audio_file = "audio.mp3"
+batch_size = 16
+language = "en"
+compute_type = "float16"
+
+with open("hf_token.txt", "r") as f:
+    HF_TOKEN = f.read()
+
+# 1. Transcribe with original whisper (batched)
+# save model to local path (optional)
+model_dir = "./models/"
+model = whisperx.load_model("large-v2", device, language="en", compute_type=compute_type, download_root=model_dir)
+
+audio = whisperx.load_audio(audio_file)
+result = model.transcribe(audio, batch_size=batch_size)
+print(result["segments"]) # before alignment
+
+# delete model if low on GPU resources
+# import gc; gc.collect(); torch.cuda.empty_cache(); del model
+
+# 2. Align whisper output
+model_a, metadata = whisperx.load_align_model(language_code=result["language"], device=device)
+result = whisperx.align(result["segments"], model_a, metadata, audio, device, return_char_alignments=False)
+
+print(result["segments"]) # after alignment
+
+# delete model if low on GPU resources
+# import gc; gc.collect(); torch.cuda.empty_cache(); del model_a
+
+# 3. Assign speaker labels
+diarize_model = whisperx.DiarizationPipeline(use_auth_token=HF_TOKEN, device=device)
+
+# add min/max number of speakers if known
+diarize_segments = diarize_model(audio)
+# diarize_model(audio, min_speakers=min_speakers, max_speakers=max_speakers)
+
+result = whisperx.assign_word_speakers(diarize_segments, result)
+print(diarize_segments)
+print(result["segments"]) # segments are now assigned speaker IDs
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