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import gradio as gr
from pydub import AudioSegment
from pydub.silence import detect_nonsilent
import numpy as np
import tempfile
import os
import noisereduce as nr
import json
import torch
from demucs import pretrained
from demucs.apply import apply_model
import torchaudio
from pathlib import Path
import matplotlib.pyplot as plt
from io import BytesIO
from PIL import Image
import zipfile
import datetime
import librosa
import warnings
from faster_whisper import WhisperModel
from mutagen.mp3 import MP3
from mutagen.id3 import ID3, TIT2, TPE1, TALB, TYER
from TTS.api import TTS
import pickle
# Suppress warnings
warnings.filterwarnings("ignore")
# === Helper Functions ===
def audiosegment_to_array(audio):
return np.array(audio.get_array_of_samples()), audio.frame_rate
def array_to_audiosegment(samples, frame_rate, channels=1):
return AudioSegment(
samples.tobytes(),
frame_rate=frame_rate,
sample_width=samples.dtype.itemsize,
channels=channels
)
# === Effect Functions ===
def apply_normalize(audio):
return audio.normalize()
def apply_noise_reduction(audio):
samples, frame_rate = audiosegment_to_array(audio)
reduced = nr.reduce_noise(y=samples, sr=frame_rate)
return array_to_audiosegment(reduced, frame_rate, channels=audio.channels)
def apply_compression(audio):
return audio.compress_dynamic_range()
def apply_reverb(audio):
reverb = audio - 10
return audio.overlay(reverb, position=1000)
def apply_pitch_shift(audio, semitones=-2):
new_frame_rate = int(audio.frame_rate * (2 ** (semitones / 12)))
samples = np.array(audio.get_array_of_samples())
resampled = np.interp(
np.arange(0, len(samples), 2 ** (semitones / 12)),
np.arange(len(samples)),
samples
).astype(np.int16)
return AudioSegment(
resampled.tobytes(),
frame_rate=new_frame_rate,
sample_width=audio.sample_width,
channels=audio.channels
)
def apply_echo(audio, delay_ms=500, decay=0.5):
echo = audio - 10
return audio.overlay(echo, position=delay_ms)
def apply_stereo_widen(audio, pan_amount=0.3):
left = audio.pan(-pan_amount)
right = audio.pan(pan_amount)
return AudioSegment.from_mono_audiosegments(left, right)
def apply_bass_boost(audio, gain=10):
return audio.low_pass_filter(100).apply_gain(gain)
def apply_treble_boost(audio, gain=10):
return audio.high_pass_filter(4000).apply_gain(gain)
def apply_noise_gate(audio, threshold=-50.0, attack=50, release=100):
samples = np.array(audio.get_array_of_samples())
rms = np.sqrt(np.mean(samples**2))
if rms < 1:
return audio
normalized = samples / np.max(np.abs(samples))
envelope = np.abs(normalized)
gated = np.where(envelope > threshold / 100, normalized, 0)
return array_to_audiosegment(gated * np.iinfo(np.int16).max, audio.frame_rate, channels=audio.channels)
def apply_limiter(audio, limit_dB=-1):
limiter = audio._spawn(audio.raw_data, overrides={"frame_rate": audio.frame_rate})
return limiter.apply_gain(limit_dB)
def apply_phaser(audio, rate=0.5, depth=0.7, feedback=0.2, mix=0.5):
return audio._spawn(audio.raw_data, overrides={"frame_rate": int(audio.frame_rate * rate)})
def apply_bitcrush(audio, bit_depth=8):
samples = np.array(audio.get_array_of_samples()).astype(np.float32)
max_val = np.iinfo(np.int16).max
crushed = ((samples / max_val) * (2 ** bit_depth)).astype(np.int16)
return array_to_audiosegment(crushed, audio.frame_rate, channels=audio.channels)
def apply_auto_gain(audio, target_dB=-20):
change = target_dB - audio.dBFS
return audio.apply_gain(change)
def apply_vocal_distortion(audio, intensity=0.3):
samples = np.array(audio.get_array_of_samples()).astype(np.float32)
distorted = samples + intensity * np.sin(samples * 2 * np.pi / 32768)
return array_to_audiosegment(distorted.astype(np.int16), audio.frame_rate, channels=audio.channels)
def apply_harmony(audio, shift_semitones=4):
shifted_up = apply_pitch_shift(audio, shift_semitones)
shifted_down = apply_pitch_shift(audio, -shift_semitones)
return audio.overlay(shifted_up).overlay(shifted_down)
def apply_stage_mode(audio):
processed = apply_reverb(audio)
processed = apply_bass_boost(processed, gain=6)
return apply_limiter(processed, limit_dB=-2)
# === Vocal Isolation Helpers ===
def load_track_local(path, sample_rate, channels=2):
sig, rate = torchaudio.load(path)
if rate != sample_rate:
sig = torchaudio.functional.resample(sig, rate, sample_rate)
if channels == 1:
sig = sig.mean(0)
return sig
def save_track(path, wav, sample_rate):
path = Path(path)
torchaudio.save(str(path), wav, sample_rate)
def apply_vocal_isolation(audio_path):
model = pretrained.get_model(name='htdemucs')
wav = load_track_local(audio_path, model.samplerate, channels=2)
ref = wav.mean(0)
wav -= ref[:, None]
sources = apply_model(model, wav[None])[0]
wav += ref[:, None]
vocal_track = sources[3].cpu()
out_path = os.path.join(tempfile.gettempdir(), "vocals.wav")
save_track(out_path, vocal_track, model.samplerate)
return out_path
# === Stem Splitting (Drums, Bass, Other, Vocals) ===
def stem_split(audio_path):
model = pretrained.get_model(name='htdemucs')
wav = load_track_local(audio_path, model.samplerate, channels=2)
sources = apply_model(model, wav[None])[0]
output_dir = tempfile.mkdtemp()
stem_paths = []
for i, name in enumerate(['drums', 'bass', 'other', 'vocals']):
path = os.path.join(output_dir, f"{name}.wav")
save_track(path, sources[i].cpu(), model.samplerate)
stem_paths.append(gr.File(value=path))
return stem_paths
# === Preset Loader with Fallback ===
def load_presets():
try:
preset_files = [f for f in os.listdir("presets") if f.endswith(".json")]
presets = {}
for f in preset_files:
path = os.path.join("presets", f)
try:
with open(path, "r") as infile:
data = json.load(infile)
if "name" in data and "effects" in data:
presets[data["name"]] = data["effects"]
except json.JSONDecodeError:
print(f"Invalid JSON: {f}")
return presets
except FileNotFoundError:
print("Presets folder not found")
return {}
preset_choices = load_presets()
if not preset_choices:
preset_choices = {
"Default": [],
"Clean Podcast": ["Noise Reduction", "Normalize"],
"Podcast Mastered": ["Noise Reduction", "Normalize", "Compress Dynamic Range"],
"Radio Ready": ["Bass Boost", "Treble Boost", "Limiter"],
"Music Production": ["Reverb", "Stereo Widening", "Pitch Shift"],
"ASMR Creator": ["Noise Gate", "Auto Gain", "Low-Pass Filter"],
"Voiceover Pro": ["Vocal Isolation", "TTS", "EQ Match"],
"8-bit Retro": ["Bitcrusher", "Echo", "Mono Downmix"],
# 🎀 Vocalist Presets
"πŸŽ™ Clean Vocal": ["Noise Reduction", "Normalize", "High Pass Filter (80Hz)"],
"πŸ§ͺ Vocal Distortion": ["Vocal Distortion", "Reverb", "Compress Dynamic Range"],
"🎢 Singer's Harmony": ["Harmony", "Stereo Widening", "Pitch Shift"],
"🌫 ASMR Vocal": ["Auto Gain", "Low-Pass Filter (3000Hz)", "Noise Gate"],
"🎼 Stage Mode": ["Reverb", "Bass Boost", "Limiter"],
"🎡 Auto-Tune Style": ["Pitch Shift (+1 semitone)", "Normalize", "Treble Boost"]
}
preset_names = list(preset_choices.keys())
# === Waveform + Spectrogram Generator ===
def show_waveform(audio_file):
try:
audio = AudioSegment.from_file(audio_file)
samples = np.array(audio.get_array_of_samples())
plt.figure(figsize=(10, 2))
plt.plot(samples[:10000], color="blue")
plt.axis("off")
buf = BytesIO()
plt.savefig(buf, format="png", bbox_inches="tight", dpi=100)
plt.close()
buf.seek(0)
return Image.open(buf)
except Exception as e:
return None
def detect_genre(audio_path):
try:
y, sr = torchaudio.load(audio_path)
mfccs = librosa.feature.mfcc(y=y.numpy().flatten(), sr=sr, n_mfcc=13).mean(axis=1).reshape(1, -1)
return "Speech"
except Exception:
return "Unknown"
# === Session Info Export ===
def generate_session_log(audio_path, effects, isolate_vocals, export_format, genre):
log = {
"timestamp": str(datetime.datetime.now()),
"filename": os.path.basename(audio_path),
"effects_applied": effects,
"isolate_vocals": isolate_vocals,
"export_format": export_format,
"detected_genre": genre
}
return json.dumps(log, indent=2)
# === Main Processing Function with Status Updates ===
def process_audio(audio_file, selected_effects, isolate_vocals, preset_name, export_format):
status = "πŸ”Š Loading audio..."
try:
audio = AudioSegment.from_file(audio_file)
status = "πŸ›  Applying effects..."
effect_map = {
"Noise Reduction": apply_noise_reduction,
"Compress Dynamic Range": apply_compression,
"Add Reverb": apply_reverb,
"Pitch Shift": lambda x: apply_pitch_shift(x),
"Echo": apply_echo,
"Stereo Widening": apply_stereo_widen,
"Bass Boost": apply_bass_boost,
"Treble Boost": apply_treble_boost,
"Normalize": apply_normalize,
"Noise Gate": lambda x: apply_noise_gate(x, threshold=-50.0),
"Limiter": lambda x: apply_limiter(x, limit_dB=-1),
"Phaser": lambda x: apply_phaser(x),
"Flanger": lambda x: apply_phaser(x, rate=1.2, depth=0.9, mix=0.7),
"Bitcrusher": lambda x: apply_bitcrush(x, bit_depth=8),
"Auto Gain": lambda x: apply_auto_gain(x, target_dB=-20),
"Vocal Distortion": lambda x: apply_vocal_distortion(x),
"Harmony": lambda x: apply_harmony(x),
"Stage Mode": apply_stage_mode
}
effects_to_apply = preset_choices.get(preset_name, selected_effects)
for effect_name in effects_to_apply:
if effect_name in effect_map:
audio = effect_map[effect_name](audio)
status = "πŸ’Ύ Saving final audio..."
with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f:
if isolate_vocals:
temp_input = os.path.join(tempfile.gettempdir(), "input.wav")
audio.export(temp_input, format="wav")
vocal_path = apply_vocal_isolation(temp_input)
final_audio = AudioSegment.from_wav(vocal_path)
else:
final_audio = audio
output_path = f.name
final_audio.export(output_path, format=export_format.lower())
waveform_image = show_waveform(output_path)
genre = detect_genre(output_path)
session_log = generate_session_log(audio_file, effects_to_apply, isolate_vocals, export_format, genre)
status = "πŸŽ‰ Done!"
return output_path, waveform_image, session_log, genre, status
except Exception as e:
status = f"❌ Error: {str(e)}"
return None, None, status, "", status
# === Batch Processing Function ===
def batch_process_audio(files, selected_effects, isolate_vocals, preset_name, export_format):
status = "πŸ”Š Loading files..."
try:
output_dir = tempfile.mkdtemp()
results = []
session_logs = []
for file in files:
processed_path, _, log, _, _ = process_audio(file.name, selected_effects, isolate_vocals, preset_name, export_format)
results.append(processed_path)
session_logs.append(log)
zip_path = os.path.join(output_dir, "batch_output.zip")
with zipfile.ZipFile(zip_path, 'w') as zipf:
for i, res in enumerate(results):
filename = f"processed_{i}.{export_format.lower()}"
zipf.write(res, filename)
zipf.writestr(f"session_info_{i}.json", session_logs[i])
return zip_path, "πŸ“¦ ZIP created successfully!"
except Exception as e:
return None, f"❌ Batch processing failed: {str(e)}"
# === Transcribe & Edit Tab ===
whisper_model = WhisperModel("base")
def transcribe_audio(audio_path):
segments, info = whisper_model.transcribe(audio_path, beam_size=5)
text = " ".join([seg.text for seg in segments])
return text
# === TTS Voice Generator ===
tts = TTS(model_name="tts_models/en/ljspeech/tacotron2-DDC", progress_bar=False)
def generate_tts(text):
out_path = os.path.join(tempfile.gettempdir(), "tts_output.wav")
tts.tts_to_file(text=text, file_path=out_path)
return out_path
# === Save/Load Project File (.aiproj) ===
def save_project(audio_path, preset_name, effects):
project_data = {
"audio": AudioSegment.from_file(audio_path).raw_data,
"preset": preset_name,
"effects": effects
}
out_path = os.path.join(tempfile.gettempdir(), "project.aiproj")
with open(out_path, "wb") as f:
pickle.dump(project_data, f)
return out_path
def load_project(project_file):
with open(project_file.name, "rb") as f:
data = pickle.load(f)
return data["preset"], data["effects"]
# === Trim Silence Automatically (VAD) ===
def detect_silence(audio_file, silence_threshold=-50.0, min_silence_len=1000):
audio = AudioSegment.from_file(audio_file)
nonsilent_ranges = detect_nonsilent(
audio,
min_silence_len=int(min_silence_len),
silence_thresh=silence_threshold
)
if not nonsilent_ranges:
return audio.export(os.path.join(tempfile.gettempdir(), "trimmed.wav"), format="wav")
trimmed = audio[nonsilent_ranges[0][0]:nonsilent_ranges[-1][1]]
out_path = os.path.join(tempfile.gettempdir(), "trimmed.wav")
trimmed.export(out_path, format="wav")
return out_path
# === Mix Two Tracks ===
def mix_tracks(track1, track2, volume_offset=0):
a1 = AudioSegment.from_file(track1)
a2 = AudioSegment.from_file(track2)
mixed = a1.overlay(a2 - volume_offset)
out_path = os.path.join(tempfile.gettempdir(), "mixed.wav")
mixed.export(out_path, format="wav")
return out_path
# === Dummy Voice Cloning Tab – Works Locally Only ===
def clone_voice(*args):
return "⚠️ Voice cloning requires local install – use Python 3.9 or below"
# === Speaker Diarization ("Who Spoke When?") ===
try:
from pyannote.audio import Pipeline as DiarizationPipeline
from huggingface_hub import login
hf_token = os.getenv("HF_TOKEN")
if hf_token:
login(token=hf_token)
diarize_pipeline = DiarizationPipeline.from_pretrained("pyannote/speaker-diarization", use_auth_token=hf_token or True)
except Exception as e:
diarize_pipeline = None
print(f"⚠️ Failed to load diarization: {e}")
def diarize_and_transcribe(audio_path):
if not diarize_pipeline:
return "⚠️ Diarization pipeline not loaded – check HF token or install pyannote.audio"
# Run diarization
audio = AudioSegment.from_file(audio_path)
temp_wav = os.path.join(tempfile.gettempdir(), "diarize.wav")
audio.export(temp_wav, format="wav")
try:
from pyannote.audio import Pipeline as DiarizationPipeline
diarization = diarize_pipeline(temp_wav)
result = whisper.transcribe(temp_wav)
segments = []
for turn, _, speaker in diarization.itertracks(yield_label=True):
text = " ".join([seg["text"] for seg in result["segments"] if seg["start"] >= turn.start and seg["end"] <= turn.end])
segments.append({
"speaker": speaker,
"start": turn.start,
"end": turn.end,
"text": text
})
return segments
except Exception as e:
return f"⚠️ Diarization failed: {str(e)}"
# === UI ===
effect_options = [
"Noise Reduction",
"Compress Dynamic Range",
"Add Reverb",
"Pitch Shift",
"Echo",
"Stereo Widening",
"Bass Boost",
"Treble Boost",
"Normalize",
"Noise Gate",
"Limiter",
"Phaser",
"Flanger",
"Bitcrusher",
"Auto Gain",
"Vocal Distortion",
"Harmony",
"Stage Mode"
]
with gr.Blocks(title="AI Audio Studio", css="style.css") as demo:
gr.Markdown("## 🎧 Ultimate AI Audio Studio\nUpload, edit, export β€” powered by AI!")
# --- Single File Studio ---
with gr.Tab("🎡 Single File Studio"):
gr.Interface(
fn=process_audio,
inputs=[
gr.Audio(label="Upload Audio", type="filepath"),
gr.CheckboxGroup(choices=effect_options, label="Apply Effects in Order"),
gr.Checkbox(label="Isolate Vocals After Effects"),
gr.Dropdown(choices=preset_names, label="Select Preset", value=preset_names[0] if preset_names else None),
gr.Dropdown(choices=["MP3", "WAV"], label="Export Format", value="MP3")
],
outputs=[
gr.Audio(label="Processed Audio", type="filepath"),
gr.Image(label="Waveform Preview"),
gr.Textbox(label="Session Log (JSON)", lines=5),
gr.Textbox(label="Detected Genre", lines=1),
gr.Textbox(label="Status", value="βœ… Ready", lines=1)
],
title="Edit One File at a Time",
description="Apply effects, preview waveform, and get full session log.",
flagging_mode="never",
submit_btn="Process Audio",
clear_btn=None
)
# --- Batch Processing ---
with gr.Tab("πŸ”Š Batch Processing"):
gr.Interface(
fn=batch_process_audio,
inputs=[
gr.File(label="Upload Multiple Files", file_count="multiple"),
gr.CheckboxGroup(choices=effect_options, label="Apply Effects in Order"),
gr.Checkbox(label="Isolate Vocals After Effects"),
gr.Dropdown(choices=preset_names, label="Select Preset", value=preset_names[0]),
gr.Dropdown(choices=["MP3", "WAV"], label="Export Format", value="MP3")
],
outputs=[
gr.File(label="Download ZIP of All Processed Files"),
gr.Textbox(label="Status", value="βœ… Ready", lines=1)
],
title="Batch Audio Processor",
description="Upload multiple files, apply effects in bulk, and download all results in a single ZIP.",
flagging_mode="never",
submit_btn="Process All Files",
clear_btn=None
)
# --- Remix Mode ---
with gr.Tab("πŸŽ› Remix Mode"):
gr.Interface(
fn=stem_split,
inputs=gr.Audio(label="Upload Music Track", type="filepath"),
outputs=[
gr.File(label="Vocals"),
gr.File(label="Drums"),
gr.File(label="Bass"),
gr.File(label="Other")
],
title="Split Into Drums, Bass, Vocals, and More",
description="Use AI to separate musical elements like vocals, drums, and bass.",
flagging_mode="never",
clear_btn=None
)
# --- Transcribe & Edit Tab ===
with gr.Tab("πŸ“ Transcribe & Edit"):
gr.Interface(
fn=transcribe_audio,
inputs=gr.Audio(label="Upload Audio", type="filepath"),
outputs=gr.Textbox(label="Transcribed Text", lines=10),
title="Transcribe & Edit Spoken Content",
description="Convert voice to text and edit it before exporting again."
)
# --- Vocal Presets for Singers ===
with gr.Tab("🎀 Vocal Presets for Singers"):
gr.Interface(
fn=process_audio,
inputs=[
gr.Audio(label="Upload Vocal Track", type="filepath"),
gr.CheckboxGroup(choices=[
"Noise Reduction",
"Normalize",
"Compress Dynamic Range",
"Bass Boost",
"Treble Boost",
"Reverb",
"Auto Gain",
"Vocal Distortion",
"Harmony",
"Stage Mode"
]),
gr.Checkbox(label="Isolate Vocals After Effects"),
gr.Dropdown(choices=[
"πŸŽ™ Clean Vocal",
"πŸ§ͺ Vocal Distortion",
"🎢 Singer's Harmony",
"🌫 ASMR Vocal",
"🎼 Stage Mode",
"🎡 Auto-Tune Style"
], label="Select Vocal Preset", value="Default"),
gr.Dropdown(choices=["MP3", "WAV"], label="Export Format", value="MP3")
],
outputs=[
gr.Audio(label="Processed Vocal", type="filepath"),
gr.Image(label="Waveform Preview"),
gr.Textbox(label="Session Log (JSON)", lines=5),
gr.Textbox(label="Detected Genre", lines=1),
gr.Textbox(label="Status", value="βœ… Ready", lines=1)
],
title="Create Studio-Quality Vocal Tracks",
description="Apply singer-friendly presets and effects to enhance vocals.",
allow_flagging="never"
)
# --- Voice Cloning (Local Only) ===
with gr.Tab("🎭 Voice Cloning (Local Only)"):
gr.Interface(
fn=clone_voice,
inputs=[
gr.File(label="Source Voice Clip"),
gr.File(label="Target Voice Clip"),
gr.Textbox(label="Text to Clone", lines=5)
],
outputs=gr.Audio(label="Cloned Output", type="filepath"),
title="Replace One Voice With Another",
description="Clone voice from source to target speaker using AI"
)
# --- Speaker Diarization (Who Spoke When?) ===
if diarize_pipeline:
with gr.Tab("πŸ§β€β™‚οΈ Who Spoke When?"):
gr.Interface(
fn=diarize_and_transcribe,
inputs=gr.Audio(label="Upload Interview/Podcast", type="filepath"),
outputs=gr.JSON(label="Diarized Transcript"),
title="Split By Speaker + Transcribe",
description="Detect speakers and transcribe their speech automatically."
)
# --- TTS Voice Generator ===
with gr.Tab("πŸ’¬ TTS Voice Generator"):
gr.Interface(
fn=generate_tts,
inputs=gr.Textbox(label="Enter Text", lines=5),
outputs=gr.Audio(label="Generated Speech", type="filepath"),
title="Text-to-Speech Generator",
description="Type anything and turn it into natural-sounding speech."
)
# --- Auto-Save / Resume Sessions ===
session_state = gr.State()
def save_or_resume_session(audio, preset, effects, action="save"):
if action == "save":
return {"audio": audio, "preset": preset, "effects": effects}, None, None, None
elif action == "load" and isinstance(audio, dict):
return (
None,
audio.get("audio"),
audio.get("preset"),
audio.get("effects")
)
return None, None, None, None
with gr.Tab("🧾 Auto-Save & Resume"):
gr.Markdown("Save your current state and resume later.")
action_radio = gr.Radio(["save", "load"], label="Action", value="save")
audio_input = gr.Audio(label="Upload or Load Audio", type="filepath")
preset_dropdown = gr.Dropdown(choices=preset_names, label="Used Preset", value=preset_names[0] if preset_names else None)
effect_checkbox = gr.CheckboxGroup(choices=effect_options, label="Applied Effects")
action_btn = gr.Button("Save or Load Session")
session_data = gr.State()
loaded_audio = gr.Audio(label="Loaded Audio", type="filepath")
loaded_preset = gr.Dropdown(choices=preset_names, label="Loaded Preset")
loaded_effects = gr.CheckboxGroup(choices=effect_options, label="Loaded Effects")
action_btn.click(
fn=save_or_resume_session,
inputs=[audio_input, preset_dropdown, effect_checkbox, action_radio],
outputs=[session_data, loaded_audio, loaded_preset, loaded_effects]
)
# --- VAD – Detect & Remove Silence ===
with gr.Tab("βœ‚οΈ Trim Silence Automatically"):
gr.Interface(
fn=detect_silence,
inputs=[
gr.File(label="Upload Track"),
gr.Slider(minimum=-100, maximum=-10, value=-50, label="Silence Threshold (dB)"),
gr.Number(label="Min Silence Length (ms)", value=1000)
],
outputs=gr.File(label="Trimmed Output"),
title="Auto-Detect & Remove Silence",
description="Detect and trim silence at start/end or between words"
)
# --- Load/Save Project File (.aiproj) ===
with gr.Tab("πŸ“ Save/Load Project"):
gr.Interface(
fn=save_project,
inputs=[
gr.File(label="Original Audio"),
gr.Dropdown(choices=preset_names, label="Used Preset", value=preset_names[0]),
gr.CheckboxGroup(choices=effect_options, label="Applied Effects")
],
outputs=gr.File(label="Project File (.aiproj)"),
title="Save Everything Together",
description="Save your session, effects, and settings in one file to reuse later."
)
gr.Interface(
fn=load_project,
inputs=gr.File(label="Upload .aiproj File"),
outputs=[
gr.Dropdown(choices=preset_names, label="Loaded Preset"),
gr.CheckboxGroup(choices=effect_options, label="Loaded Effects")
],
title="Resume Last Project",
description="Load your saved session"
)
# --- Mix Two Tracks ===
with gr.Tab(" remix mode"),
gr.Interface(
fn=mix_tracks,
inputs=[
gr.File(label="Main Track"),
gr.File(label="Background Track"),
gr.Slider(minimum=-10, maximum=10, value=0, label="Volume Offset (dB)")
],
outputs=gr.File(label="Mixed Output"),
title="Overlay Two Tracks",
description="Mix, blend, or subtract two audio files."
)
demo.launch()