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import numpy as np
import soundfile as sf
import subprocess
import tempfile
import os
import gradio as gr
from scipy import signal
# ========== Processing Functions ==========
def convert_to_wav_float(input_file):
"""
Convert any input audio to 32-bit float WAV to preserve full dynamic range.
"""
temp_wav = tempfile.NamedTemporaryFile(suffix=".wav", delete=False)
temp_wav.close()
# PCM 32-bit little endian preserves float dynamic without clipping
subprocess.run([
"ffmpeg", "-y", "-i", input_file,
"-c:a", "pcm_f32le", "-f", "wav", temp_wav.name
], stdout=subprocess.DEVNULL, stderr=subprocess.DEVNULL, check=True)
return temp_wav.name
def apply_reverb_wet_only(audio, samplerate, reverb_args):
"""
Apply wet-only reverb using SoX to a single channel with custom reverb args.
"""
with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as tin, \
tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as tout:
sf.write(tin.name, audio, samplerate, subtype='FLOAT')
subprocess.run(
["sox", tin.name, tout.name, "reverb", "-w"] + reverb_args,
stdout=subprocess.DEVNULL, stderr=subprocess.DEVNULL, check=True
)
wet, _ = sf.read(tout.name, dtype='float32')
os.unlink(tin.name)
os.unlink(tout.name)
return wet
def sox_filter(audio, samplerate, filter_type, cutoff):
"""
Apply highpass or lowpass filter via SoX.
filter_type: 'highpass' or 'lowpass'; cutoff in Hz.
"""
with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as tin, \
tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as tout:
sf.write(tin.name, audio, samplerate, subtype='FLOAT')
subprocess.run(
["sox", tin.name, tout.name, filter_type, str(cutoff)],
stdout=subprocess.DEVNULL, stderr=subprocess.DEVNULL, check=True
)
out, _ = sf.read(tout.name, dtype='float32')
os.unlink(tin.name)
os.unlink(tout.name)
return out
def extract_phantom_center(input_file, rdf=0.99999):
"""
Returns FL (front left without centre), FR, and FC (phantom centre).
"""
wav = convert_to_wav_float(input_file)
data, fs = sf.read(wav, dtype='float32')
os.unlink(wav)
if data.ndim != 2 or data.shape[1] != 2:
raise gr.Error("Input must be stereo 2-channel")
L, R = data[:,0], data[:,1]
M = (L + R) / 2
nperseg = fs
noverlap = nperseg // 2
_, _, ZL = signal.stft(L, fs=fs, nperseg=nperseg, noverlap=noverlap)
_, _, ZR = signal.stft(R, fs=fs, nperseg=nperseg, noverlap=noverlap)
_, _, ZM = signal.stft(M, fs=fs, nperseg=nperseg, noverlap=noverlap)
Zc = np.minimum(np.abs(ZL), np.abs(ZR)) * np.exp(1j * np.angle(ZM))
Zl_res = ZL - Zc * rdf
Zr_res = ZR - Zc * rdf
_, FL = signal.istft(Zl_res, fs=fs, nperseg=nperseg, noverlap=noverlap)
_, FR = signal.istft(Zr_res, fs=fs, nperseg=nperseg, noverlap=noverlap)
_, FC = signal.istft(Zc, fs=fs, nperseg=nperseg, noverlap=noverlap)
return fs, FL[:len(L)], FR[:len(R)], FC[:len(M)]
def create_5_1_surround(input_file, preset="music"):
print("Starting Normal Processing")
p = gr.Progress()
# Preset-based parameters
# Reverberance (50%) HF-damping (50%) room-scale (100%) stereo-depth (100%) pre-delay (0ms) wet-gain (0dB)
if preset == "music":
hp_cutoff = 120
lfe_cutoff = 120
reverb_args = ['70', '40', '100', '95', '10', '-2']
elif preset == "speech":
hp_cutoff = 120
lfe_cutoff = 120
reverb_args = ['50', '99', '50', '70', '0', '0']
elif preset == "open":
hp_cutoff = 120
lfe_cutoff = 120
reverb_args = ['20', '50', '100', '100', '100', '0']
else:
raise gr.Error(f"Unknown preset: {preset}")
p((1,7),"Extracting Centre")# 1. Extract FL/FR/phantom centre
fs, FL, FR, FC = extract_phantom_center(input_file)
p((2,7),"Getting File")# 2. Get stereo original for reverb
wav = convert_to_wav_float(input_file)
stereo, _ = sf.read(wav, dtype='float32')
os.unlink(wav)
L_orig, R_orig = stereo[:, 0], stereo[:, 1]
p((3,7),"Reverb For Rear")# 3. Wet-only reverb with chosen settings
SL = apply_reverb_wet_only(L_orig, fs, reverb_args)
SR = apply_reverb_wet_only(R_orig, fs, reverb_args)
p((4,7),"Highpassing")# 4. Highpass filter everything except LFE
FL_hp = sox_filter(FL, fs, 'highpass', hp_cutoff)
FR_hp = sox_filter(FR, fs, 'highpass', hp_cutoff)
FC_hp = sox_filter(FC, fs, 'highpass', hp_cutoff)
SL_hp = sox_filter(SL, fs, 'highpass', hp_cutoff)
SR_hp = sox_filter(SR, fs, 'highpass', hp_cutoff)
p((5,7),"Extracting LFE")# 5. Lowpass for LFE
bass_sum = .5 * (L_orig + R_orig)
LFE = sox_filter(bass_sum, fs, 'lowpass', lfe_cutoff)
p((6,7),"Stacking")# 6. Stack and pad
channels = [FL_hp, FR_hp, FC_hp, LFE, SL_hp, SR_hp]
length = max(len(ch) for ch in channels)
def pad(x): return np.pad(x, (0, length - len(x)))
multich = np.column_stack([pad(ch) for ch in channels])
p((7,7),"Encoding")# 7. Write WAV and encode to OGG
out_wav = tempfile.NamedTemporaryFile(suffix='.wav', delete=False)
sf.write(out_wav.name, multich, fs, subtype='FLOAT')
out_wav.close()
out_ogg = tempfile.NamedTemporaryFile(suffix='.ogg', delete=False)
out_ogg.close()
subprocess.run([
"ffmpeg", "-y", "-i", out_wav.name,
"-c:a", "libvorbis", "-ac", "6", "-channel_layout", "5.1", out_ogg.name
], stdout=subprocess.DEVNULL, stderr=subprocess.DEVNULL, check=True)
os.unlink(out_wav.name)
return out_ogg.name
import mimetypes
import requests
import time
def send_mvsep_audio_job(
api_token: str,
audio_bytes: bytes,
filename: str,
sep_type: int = 34,
output_format: int = 2,
addopt1: str = None,
addopt2: str = None,
poll_interval_sec: int = 5
):
"""
Send audio to MVSep for source separation and wait for the result.
Args:
api_token (str): Your API token.
audio_bytes (bytes): Audio data (any format).
filename (str): Original filename, used for extension/MIME type.
sep_type (int): Separation type (e.g., 34 for karaoke).
output_format (int): Output format (e.g., 2 for FLAC).
addopt1 (str): Optional extra parameter 1.
addopt2 (str): Optional extra parameter 2.
poll_interval_sec (int): How often to check job status.
Returns:
dict: Completed result data from mvsep.com (including file URLs).
"""
# Step 1: Determine MIME type
mime_type, _ = mimetypes.guess_type(filename)
if not mime_type:
mime_type = "application/octet-stream" # fallback
# Step 2: Prepare request
url = "https://mvsep.com/api/separation/create"
files = {
'audiofile': (filename, audio_bytes, mime_type)
}
data = {
'api_token': api_token,
'sep_type': str(sep_type),
'output_format': str(output_format)
}
if addopt1:
data['add_opt1'] = str(addopt1)
if addopt2:
data['add_opt2'] = str(addopt2)
# Step 3: Send creation request
response = requests.post(url, files=files, data=data)
response.raise_for_status()
json_resp = response.json()
if not json_resp.get('success'):
error_msg = json_resp.get('data', {}).get('message', 'Unknown error')
print(json_resp)
raise gr.Error(f"API error: {error_msg}")
job_hash = json_resp['data']['hash']
print(f"Job submitted successfully. Hash: {job_hash}")
# Step 4: Poll until job is done
status_url = "https://mvsep.com/api/separation/get"
while True:
poll_resp = requests.get(status_url, params={'hash': job_hash})
poll_resp.raise_for_status()
poll_data = poll_resp.json()
status = poll_data.get('status')
print(f"Job status: {status}")
if status == 'done':
return poll_data.get('data', {})
elif status in ('failed', 'not_found'):
raise gr.Error(f"Job failed or not found: {poll_data.get('data', {}).get('message', '')}")
time.sleep(poll_interval_sec)
# Download WAV and preserve sample rate, with optional resampling to target_fs
def download_wav(url, target_fs=None):
r = requests.get(url)
r.raise_for_status()
temp = tempfile.NamedTemporaryFile(suffix=".wav", delete=False)
temp.write(r.content)
temp.close()
audio, sr = sf.read(temp.name, dtype='float32')
os.unlink(temp.name)
if target_fs and sr != target_fs:
# resample if needed
num_samples = int(len(audio) * target_fs / sr)
audio = signal.resample(audio, num_samples)
sr = target_fs
return audio, sr
# Smart mode workflow
def smart_mode_process(input_file, api_key, multi_singer=False):
print("Starting Smartmode")
p = gr.Progress()
import shutil
if not api_key:
raise gr.Error("An MVSep API Key Is Required For This. Get your key <a href=\"https://mvsep.com/user-api\">Here</a>. it's Free!")
# Load original
wav = convert_to_wav_float(input_file)
data, fs = sf.read(wav, dtype='float32')
os.unlink(wav)
p((0, 8), "Loading File")
if data.ndim != 2 or data.shape[1] != 2:
raise gr.Error("Expected stereo input (2 channels), got something else.")
L, R = data[:, 0], data[:, 1]
# Step 1: LFE from lowpass
p((1, 8), "Processing LFE")
bass = sox_filter(0.5 * (L + R), fs, 'lowpass', 120)
# Step 2: Highpass for crowd extraction
p((2, 8), "Extracting Crowd")
hp_left = sox_filter(L, fs, 'highpass', 120)
hp_right = sox_filter(R, fs, 'highpass', 120)
hp_stereo = np.column_stack([hp_left, hp_right])
music_buf = tempfile.NamedTemporaryFile(suffix=".flac", delete=False)
sf.write(music_buf.name, hp_stereo, fs, format='FLAC', subtype='PCM_16')
music_buf.close()
crowd_resp = send_mvsep_audio_job(
api_key, open(music_buf.name, 'rb').read(), os.path.basename(music_buf.name),
sep_type=34, output_format=2, addopt1=0
)
os.unlink(music_buf.name)
crowd, _ = download_wav(crowd_resp['files'][0]['url'], target_fs=fs)
other_after_crowd, _ = download_wav(crowd_resp['files'][1]['url'], target_fs=fs)
# Step 3: Speech, music, SFX separation from 'other_after_crowd'
p((3, 8), "Separating Speech, Music, and SFX")
demucs_input_buf = tempfile.NamedTemporaryFile(suffix=".flac", delete=False)
sf.write(demucs_input_buf.name, other_after_crowd, fs, format='FLAC', subtype='PCM_16')
demucs_input_buf.close()
demucs_resp = send_mvsep_audio_job(
api_key, open(demucs_input_buf.name, 'rb').read(), os.path.basename(demucs_input_buf.name),
sep_type=24, output_format=2
)
os.unlink(demucs_input_buf.name)
dialog, _ = download_wav(demucs_resp['files'][0]['url'], target_fs=fs)
sfx, _ = download_wav(demucs_resp['files'][2]['url'], target_fs=fs)
music, _ = download_wav(demucs_resp['files'][1]['url'], target_fs=fs)
# Step 4: Apply Reverb to the 'music' stem
p((4, 8), "Applying Reverb")
reverb_args = ['20', '50', '100', '100', '100', '0'] # open preset
reverb_L = apply_reverb_wet_only(music[:, 0], fs, reverb_args)
reverb_R = apply_reverb_wet_only(music[:, 1], fs, reverb_args)
reverb = np.column_stack([reverb_L, reverb_R])
# Step 5: Vocal Extraction from music
p((5, 8), "Extracting Vocals")
music_buf = tempfile.NamedTemporaryFile(suffix=".flac", delete=False)
sf.write(music_buf.name, music, fs, format='FLAC', subtype='PCM_16')
music_buf.close()
karaoke_resp = send_mvsep_audio_job(
api_key, open(music_buf.name, 'rb').read(), os.path.basename(music_buf.name),
sep_type=49, output_format=2, addopt1=3, addopt2=1
)
os.unlink(music_buf.name)
vocals_full, _ = download_wav(karaoke_resp['files'][0]['url'], target_fs=fs)
vocals_lead, _ = download_wav(karaoke_resp['files'][1]['url'], target_fs=fs)
vocals_back, _ = download_wav(karaoke_resp['files'][2]['url'], target_fs=fs)
instr, _ = download_wav(karaoke_resp['files'][3]['url'], target_fs=fs)
# Step 6: Phantom center on vocals (lead or full)
p((6, 8), "Phantom Center for Lead Vocals")
vl_buf = tempfile.NamedTemporaryFile(suffix=".wav", delete=False)
sf.write(vl_buf.name, vocals_full if multi_singer else vocals_lead, fs, subtype='FLOAT')
vl_buf.close()
_, FL_vl, FR_vl, FC_vl = extract_phantom_center(vl_buf.name)
os.unlink(vl_buf.name)
# Mix dialog into the centre channel
FC_vl += dialog[:, 0] if dialog.ndim == 2 else dialog
# Step 7: Mapping and stacking
p((7, 8), "Mapping Channels and Encoding")
def match_len(x, length): return np.pad(x, (0, length - len(x)))
lens = [len(FL_vl), len(FR_vl), len(FC_vl), len(bass), len(sfx), crowd.shape[0], vocals_back.shape[0], instr.shape[0], len(reverb)]
length = max(lens)
# FL and FR: Lead vocals + SFX + instruments
out_L = match_len(FL_vl, length) + match_len(sfx[:, 0], length) + match_len(instr[:, 0], length)
out_R = match_len(FR_vl, length) + match_len(sfx[:, 1], length) + match_len(instr[:, 1], length)
out_C = match_len(FC_vl, length)
out_LFE = match_len(bass, length)
# SL/SR: Use reverb output
SL = match_len(reverb[:, 0], length)
SR = match_len(reverb[:, 1], length)
if not multi_singer:
SL += match_len(vocals_back[:, 0], length)
SR += match_len(vocals_back[:, 1], length)
SL += match_len(crowd[:, 0], length)
SR += match_len(crowd[:, 1], length)
multich = np.column_stack([out_L, out_R, out_C, out_LFE, SL, SR])
out_wav = tempfile.NamedTemporaryFile(suffix='.wav', delete=False)
sf.write(out_wav.name, multich, fs, subtype='FLOAT')
out_wav.close()
out_ogg = tempfile.NamedTemporaryFile(suffix='.ogg', delete=False)
subprocess.run([
"ffmpeg", "-y", "-i", out_wav.name,
"-c:a", "libvorbis", "-ac", "6", "-channel_layout", "5.1", out_ogg.name
], stdout=subprocess.DEVNULL, stderr=subprocess.DEVNULL, check=True)
os.unlink(out_wav.name)
return out_ogg.name
# ========== Gradio UI ==========
with gr.Blocks(title="Stereo to 5.1 Surround") as demo:
gr.Markdown("# 🎧 Stereo to 5.1 Converter")
gr.Markdown("Convert A Stereo File Into Surround")
inp = gr.Audio(label="Upload stereo audio", type="filepath")
smart_mode = gr.Checkbox(label="Enable Smart Mode", value=False)
# Normal mode elements
preset = gr.Dropdown(
label="Select Preset",
choices=["music", "speech", "open"],
value="music"
)
btn = gr.Button("Convert to 5.1 OGG")
out = gr.File(label="Download 5.1 OGG")
# Smart mode section
with gr.Column(visible=False) as smart_section:
api_key = gr.Textbox(label="MVSep API Key", type="password")
multi_singer = gr.Checkbox(label="Multi Singer Mode", value=False)
smart_btn = gr.Button("Convert")
smart_out = gr.File(label="Output")
# Logic for toggling sections
def toggle_mode(enabled):
return (
gr.update(visible=not enabled), # preset
gr.update(visible=not enabled), # btn
gr.update(visible=not enabled), # out
gr.update(visible=enabled) # smart_section
)
smart_mode.change(
fn=toggle_mode,
inputs=[smart_mode],
outputs=[preset, btn, out, smart_section]
)
# Button functions
btn.click(fn=create_5_1_surround, inputs=[inp, preset], outputs=[out], concurrency_limit=10)
smart_btn.click(fn=smart_mode_process, inputs=[inp, api_key, multi_singer], outputs=[smart_out], concurrency_limit=20)
if __name__ == "__main__":
demo.launch(show_error=True) |