initial commit (clean, no models)

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akukanara
2026-05-30 21:30:49 +07:00
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# Python cache
__pycache__/
*.py[cod]
*$py.class
# Environments
.venv/
venv/
ENV/
env/
# Virtual environment files
Pipfile.lock
poetry.lock
# OS files
.DS_Store
Thumbs.db
# Agent/IDE files
.antigravitycli/
.vscode/
.idea/
# Voice changer models and weights
*.onnx
*.pth
*.pt
*.index
weights/
pretrained/
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# Standalone ONNX Voice Changer Service
Layanan pengubah suara real-time berbasis AI berlatensi rendah menggunakan akselerasi ONNX Runtime dan model RVC (Retrieval-based Voice Conversion).
## Struktur Proyek
- `server.py`: WebSocket server utama yang memproses streaming audio dan menyajikan static HTTP frontend.
- `frontend/`: File UI web client (HTML, CSS, JS).
- `lib/`: Modul inferensi ONNX RVC.
- `weights/`: Tempat penyimpanan model suara (folder per model berisi file `.onnx` dan opsional file `.pth`).
- `pretrained/`: Model pra-latih dasar (seperti `vec-768-layer-12.onnx`).
- `rmvpe.pt` & `rmvpe.py`: Untuk ekstraksi pitch suara fidelitas tinggi.
## Cara Menjalankan
### Persyaratan Sistem
Pastikan Python 3.10+ sudah terinstal di sistem Anda beserta library yang dibutuhkan di `requirements.txt`.
### Menjalankan Server
Jalankan server menggunakan Python dari environment Anda:
```bash
python server.py --host 127.0.0.1 --port 8765 --http_port 8000
```
Parameter opsional:
- `--host`: Alamat host WebSocket server (default: `127.0.0.1`).
- `--port`: Port WebSocket server (default: `8765`).
- `--http_port`: Port HTTP server untuk UI web client (default: `8000`).
- `--device`: Execution Provider (`cpu`, `cuda`, atau `dml` - default: `cuda`).
- `--model`: Nama folder model suara di dalam `weights/` yang ingin dimuat langsung saat start.
Setelah server berjalan, Web UI akan otomatis terbuka di browser Anda pada alamat `http://localhost:8000`.
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/**
* Omni Real-Time Voice Changer - Client App
* High-performance browser-based mic streaming and RVC playback.
*/
// UI Elements
const wsUrlInput = document.getElementById('ws_url');
const connectionStatus = document.getElementById('connection_status');
const connectBtn = document.getElementById('connect_btn');
const streamBtn = document.getElementById('stream_btn');
const playToggleBtn = document.getElementById('play_toggle_btn');
const modelSelect = document.getElementById('model_select');
const deviceSelect = document.getElementById('device_select');
const transposeSlider = document.getElementById('transpose_slider');
const transposeVal = document.getElementById('transpose_val');
const gateSlider = document.getElementById('gate_slider');
const gateVal = document.getElementById('gate_val');
const inputGainSlider = document.getElementById('input_gain_slider');
const inputGainVal = document.getElementById('input_gain_val');
const outputGainSlider = document.getElementById('output_gain_slider');
const outputGainVal = document.getElementById('output_gain_val');
const chunkSelect = document.getElementById('chunk_select');
const noiseCancelCheckbox = document.getElementById('noise_cancel_checkbox');
const routingModeSelect = document.getElementById('routing_mode_select');
const hardwareDevicesPanel = document.getElementById('hardware_devices_panel');
const serverInputSelect = document.getElementById('server_input_select');
const serverOutputSelect = document.getElementById('server_output_select');
const browserNoiseCancelGroup = document.getElementById('browser_noise_cancel_group');
const presetLatencyBtn = document.getElementById('preset_latency_btn');
const presetQualityBtn = document.getElementById('preset_quality_btn');
const inputCanvas = document.getElementById('input_canvas');
const outputCanvas = document.getElementById('output_canvas');
const hudLatency = document.getElementById('hud_latency');
const hudTime = document.getElementById('hud_time');
const hudGateStatus = document.getElementById('hud_gate_status');
const hudSr = document.getElementById('hud_sr');
// Audio Visualizer Contexts
const inputCtx = inputCanvas.getContext('2d');
const outputCtx = outputCanvas.getContext('2d');
// Web Audio State
let audioContext = null;
let micStream = null;
let micSourceNode = null;
let scriptProcessorNode = null;
let micAccumulator = new Float32Array(0); // Accumulates audio for large/custom chunk sizes
// WebSocket State
let socket = null;
let isStreaming = false;
let playOutput = true;
let targetSampleRate = 40000; // RVC Model default, updated dynamically
// Playback Sync State
let nextPlaybackTime = 0;
const safetyDelay = 0.10; // 100ms buffer to absorb network/websocket jitter (increased for perfect smoothness!)
// Latency Tracking Queues
let sentTimestamps = [];
const maxSentLogs = 50;
// --- SMOOTH VISUALIZER (Rolling Display Buffers + RAF loop) ---
// Fixed display buffer size: ~85ms window looks great at all chunk sizes.
const VIS_DISPLAY_SIZE = 4096;
let inputDisplayBuf = new Float32Array(VIS_DISPLAY_SIZE); // rolling input (updated ~85ms)
let outputDisplayBuf = new Float32Array(VIS_DISPLAY_SIZE); // fallback for hardware mode
let rafHandle = null;
// Time-synced output queue: each entry = { data: Float32Array, startTime: number (audioCtx seconds) }
let outputChunkQueue = [];
function pushToDisplayBuf(displayBuf, newSamples) {
if (newSamples.length >= VIS_DISPLAY_SIZE) {
displayBuf.set(newSamples.slice(newSamples.length - VIS_DISPLAY_SIZE));
} else {
displayBuf.copyWithin(0, newSamples.length);
displayBuf.set(newSamples, VIS_DISPLAY_SIZE - newSamples.length);
}
}
// Build a VIS_DISPLAY_SIZE window of output samples ending at audioContext.currentTime
function buildTimeSyncedOutputBuf() {
if (!audioContext || outputChunkQueue.length === 0) return outputDisplayBuf;
const now = audioContext.currentTime;
const windowDuration = VIS_DISPLAY_SIZE / targetSampleRate;
const windowStart = now - windowDuration;
// Drop chunks that ended before our window start
while (outputChunkQueue.length > 0) {
const c = outputChunkQueue[0];
if (c.startTime + c.data.length / targetSampleRate < windowStart) {
outputChunkQueue.shift();
} else break;
}
const out = new Float32Array(VIS_DISPLAY_SIZE);
for (const chunk of outputChunkQueue) {
const chunkEnd = chunk.startTime + chunk.data.length / targetSampleRate;
// Overlap between [windowStart, now] and [chunk.startTime, chunkEnd]
const overlapStart = Math.max(windowStart, chunk.startTime);
const overlapEnd = Math.min(now, chunkEnd);
if (overlapStart >= overlapEnd) continue;
const srcOffset = Math.floor((overlapStart - chunk.startTime) * targetSampleRate);
const destOffset = Math.floor((overlapStart - windowStart) * targetSampleRate);
const count = Math.floor((overlapEnd - overlapStart) * targetSampleRate);
const safeCount = Math.min(count,
chunk.data.length - srcOffset,
VIS_DISPLAY_SIZE - destOffset);
if (safeCount > 0) out.set(chunk.data.subarray(srcOffset, srcOffset + safeCount), destOffset);
}
return out;
}
function startVisualizerLoop() {
if (rafHandle) return;
function frame() {
drawWaveform(inputDisplayBuf, inputCanvas, '#6366f1');
// Time-synced output: scrub through queued chunks using audioContext clock
drawWaveform(buildTimeSyncedOutputBuf(), outputCanvas, '#a855f7');
rafHandle = requestAnimationFrame(frame);
}
rafHandle = requestAnimationFrame(frame);
}
function stopVisualizerLoop() {
if (rafHandle) {
cancelAnimationFrame(rafHandle);
rafHandle = null;
}
outputChunkQueue = [];
}
// Setup Canvas Sizes dynamically
function resizeCanvases() {
inputCanvas.width = inputCanvas.clientWidth * window.devicePixelRatio;
inputCanvas.height = inputCanvas.clientHeight * window.devicePixelRatio;
outputCanvas.width = outputCanvas.clientWidth * window.devicePixelRatio;
outputCanvas.height = outputCanvas.clientHeight * window.devicePixelRatio;
}
resizeCanvases();
window.addEventListener('resize', resizeCanvases);
// Connect / Disconnect WebSocket
connectBtn.addEventListener('click', () => {
if (socket && (socket.readyState === WebSocket.OPEN || socket.readyState === WebSocket.CONNECTING)) {
disconnectServer();
} else {
connectServer();
}
});
function connectServer() {
const url = wsUrlInput.value.trim();
updateConnectionStatus('connecting');
try {
socket = new WebSocket(url);
socket.binaryType = 'arraybuffer';
socket.onopen = () => {
console.log('Connected to RVC Server');
updateConnectionStatus('connected');
sendConfigToServer(); // Send initial configurations
streamBtn.disabled = false;
playToggleBtn.disabled = false;
};
socket.onclose = () => {
console.log('WebSocket Connection Closed');
disconnectServer();
};
socket.onerror = (err) => {
console.error('WebSocket Error:', err);
disconnectServer();
};
socket.onmessage = (event) => {
if (typeof event.data === 'string') {
// Config or control response
try {
const response = JSON.parse(event.data);
if (response.type === 'config_success') {
targetSampleRate = response.target_sr;
console.log('Server configuration synced successfully:', response);
} else if (response.type === 'init_devices') {
populateServerDevices(response.devices, response.default_input, response.default_output);
} else if (response.type === 'visualizer') {
// Feed rolling display buffers — RAF loop handles drawing at 60fps
pushToDisplayBuf(inputDisplayBuf, new Float32Array(response.input));
pushToDisplayBuf(outputDisplayBuf, new Float32Array(response.output));
if (!rafHandle) startVisualizerLoop();
} else if (response.type === 'error') {
alert('Server Error: ' + response.message);
}
} catch (e) {
console.error('Error parsing text message:', e);
}
} else if (event.data instanceof ArrayBuffer) {
// Binary processed PCM audio chunk returned from server (Browser Mode only)
handleServerAudioChunk(event.data);
}
};
} catch (e) {
console.error('Connection failed:', e);
disconnectServer();
}
}
function disconnectServer() {
if (isStreaming) {
stopStreaming();
}
if (socket) {
try {
socket.close();
} catch (e) {}
socket = null;
}
updateConnectionStatus('disconnected');
streamBtn.disabled = true;
playToggleBtn.disabled = true;
}
function updateConnectionStatus(status) {
connectionStatus.className = 'status-badge ' + status;
if (status === 'connected') {
connectionStatus.textContent = 'Terhubung';
connectBtn.textContent = 'Putuskan Server';
connectBtn.className = 'btn btn-primary';
} else if (status === 'connecting') {
connectionStatus.textContent = 'Menghubungkan';
connectBtn.textContent = 'Batal';
} else {
connectionStatus.textContent = 'Terputus';
connectBtn.textContent = 'Hubungkan Server';
connectBtn.className = 'btn btn-primary';
}
}
// Config synchronization
function sendConfigToServer() {
if (!socket || socket.readyState !== WebSocket.OPEN) return;
const activeF0 = document.querySelector('input[name="f0_method"]:checked').value;
const config = {
type: 'config',
model_name: modelSelect.value,
device: deviceSelect.value,
f0_method: activeF0,
f0_up_key: parseInt(transposeSlider.value),
noise_gate: parseFloat(gateSlider.value),
input_gain: parseFloat(inputGainSlider.value),
output_gain: parseFloat(outputGainSlider.value),
input_sr: audioContext ? audioContext.sampleRate : 44100,
routing_mode: routingModeSelect.value,
input_device: serverInputSelect.value ? parseInt(serverInputSelect.value) : null,
output_device: serverOutputSelect.value ? parseInt(serverOutputSelect.value) : null,
chunk_size: parseInt(chunkSelect.value)
};
socket.send(jsonEncode(config));
console.log('Sent configuration change:', config);
}
// Populate Server Audio Devices dropdowns
function populateServerDevices(devices, defaultInput, defaultOutput) {
serverInputSelect.innerHTML = '';
serverOutputSelect.innerHTML = '';
if (devices.length === 0) {
const optIn = document.createElement('option');
optIn.textContent = 'Tidak ada mic terdeteksi di server';
serverInputSelect.appendChild(optIn);
const optOut = document.createElement('option');
optOut.textContent = 'Tidak ada output terdeteksi di server';
serverOutputSelect.appendChild(optOut);
return;
}
devices.forEach(device => {
if (device.max_input_channels > 0) {
const opt = document.createElement('option');
opt.value = device.id;
opt.textContent = `[ID ${device.id}] ${device.name}`;
if (device.id === defaultInput) opt.selected = true;
serverInputSelect.appendChild(opt);
}
if (device.max_output_channels > 0) {
const opt = document.createElement('option');
opt.value = device.id;
opt.textContent = `[ID ${device.id}] ${device.name}`;
if (device.id === defaultOutput) opt.selected = true;
serverOutputSelect.appendChild(opt);
}
});
console.log('Successfully populated server hardware devices in UI.');
}
// UI Event Listeners to trigger instant sync
modelSelect.addEventListener('change', sendConfigToServer);
deviceSelect.addEventListener('change', sendConfigToServer);
document.querySelectorAll('input[name="f0_method"]').forEach(radio => {
radio.addEventListener('change', sendConfigToServer);
});
transposeSlider.addEventListener('input', () => {
transposeVal.textContent = (transposeSlider.value >= 0 ? '+' : '') + transposeSlider.value + ' semitone';
});
transposeSlider.addEventListener('change', sendConfigToServer);
gateSlider.addEventListener('input', () => {
gateVal.textContent = gateSlider.value + ' dB';
});
gateSlider.addEventListener('change', sendConfigToServer);
inputGainSlider.addEventListener('input', () => {
inputGainVal.textContent = parseFloat(inputGainSlider.value).toFixed(1) + 'x';
});
inputGainSlider.addEventListener('change', sendConfigToServer);
outputGainSlider.addEventListener('input', () => {
outputGainVal.textContent = parseFloat(outputGainSlider.value).toFixed(1) + 'x';
});
outputGainSlider.addEventListener('change', sendConfigToServer);
chunkSelect.addEventListener('change', () => {
// Reinitialize stream if buffer size is changed during active streaming
if (isStreaming) {
stopStreaming();
startStreaming();
}
});
noiseCancelCheckbox.addEventListener('change', () => {
// Reinitialize microphone with new noise cancellation constraints if streaming
if (isStreaming) {
stopStreaming();
startStreaming();
}
});
// Helper to dynamically adjust UI layout based on Routing Mode
function applyAudioRoutingUI() {
if (routingModeSelect.value === 'hardware') {
hardwareDevicesPanel.style.display = 'block';
playToggleBtn.style.display = 'none'; // Hide browser-only "Mendengarkan" button
browserNoiseCancelGroup.style.display = 'none'; // Hide browser-only Noise Cancel checkbox
} else {
hardwareDevicesPanel.style.display = 'none';
playToggleBtn.style.display = 'inline-block'; // Show browser-only "Mendengarkan" button
browserNoiseCancelGroup.style.display = 'block'; // Show browser-only Noise Cancel checkbox
}
}
// Routing Mode Event Listeners
routingModeSelect.addEventListener('change', () => {
applyAudioRoutingUI();
sendConfigToServer();
if (isStreaming) {
stopStreaming();
startStreaming();
}
});
serverInputSelect.addEventListener('change', sendConfigToServer);
serverOutputSelect.addEventListener('change', sendConfigToServer);
// Quick Presets Event Listeners
presetLatencyBtn.addEventListener('click', () => {
const radioPM = document.querySelector('input[name="f0_method"][value="pm"]');
if (radioPM) radioPM.checked = true;
chunkSelect.value = "8192";
console.log("Preset loaded: Latency (PM + 8192)");
sendConfigToServer();
if (isStreaming) {
stopStreaming();
startStreaming();
}
});
presetQualityBtn.addEventListener('click', () => {
const radioRMVPE = document.querySelector('input[name="f0_method"][value="rmvpe"]');
if (radioRMVPE) radioRMVPE.checked = true;
chunkSelect.value = "16384";
console.log("Preset loaded: Quality (RMVPE + 16384)");
sendConfigToServer();
if (isStreaming) {
stopStreaming();
startStreaming();
}
});
// Helper functions for UI JSON safely
function jsonEncode(obj) {
return JSON.stringify(obj);
}
playToggleBtn.addEventListener('click', () => {
playOutput = !playOutput;
if (playOutput) {
playToggleBtn.textContent = '🔊 Mendengarkan: AKTIF';
playToggleBtn.className = 'btn btn-primary';
} else {
playToggleBtn.textContent = '🔇 Mendengarkan: SENYAP';
playToggleBtn.className = 'btn btn-accent';
}
});
// Stream Toggle
streamBtn.addEventListener('click', () => {
if (isStreaming) {
stopStreaming();
} else {
startStreaming();
}
});
async function startStreaming() {
isStreaming = true;
streamBtn.textContent = 'Hentikan Pengubah Suara';
streamBtn.className = 'btn btn-primary';
const isHardwareMode = (routingModeSelect.value === 'hardware');
if (isHardwareMode) {
// --- SERVER HARDWARE ROUTING MODE ---
inputDisplayBuf = new Float32Array(VIS_DISPLAY_SIZE);
outputDisplayBuf = new Float32Array(VIS_DISPLAY_SIZE);
startVisualizerLoop();
sendConfigToServer(); // Sends config with routing_mode: 'hardware' which triggers stream start on server
console.log('Server Hardware Mode initialized.');
return;
}
// --- CLIENT BROWSER MODE ---
// 1. Create AudioContext if not active
if (!audioContext) {
audioContext = new (window.AudioContext || window.webkitAudioContext)({
latencyHint: 'interactive'
});
}
if (audioContext.state === 'suspended') {
await audioContext.resume();
}
hudSr.textContent = audioContext.sampleRate + ' Hz';
sendConfigToServer(); // sync actual input sample rate
// 2. Request user microphone with high-fidelity, lowest possible latency constraints
try {
const useNoiseCancel = noiseCancelCheckbox.checked;
micStream = await navigator.mediaDevices.getUserMedia({
audio: {
echoCancellation: useNoiseCancel,
noiseSuppression: useNoiseCancel,
autoGainControl: useNoiseCancel
}
});
micSourceNode = audioContext.createMediaStreamSource(micStream);
// 3. Create Audio Processing Loop Node (ScriptProcessorNode)
// BaseAudioContext's createScriptProcessor buffer size MUST be a power of two between 256 and 16384.
// We use a fixed, highly supported buffer size of 4096 for recording, and accumulate samples in-memory
// to support ANY arbitrary or extremely large chunk size (like 12288, 24576, 32768) selected by the user!
const recordBufferSize = 4096;
scriptProcessorNode = audioContext.createScriptProcessor(recordBufferSize, 1, 1);
scriptProcessorNode.onaudioprocess = (event) => {
if (!isStreaming) return;
const inputBuffer = event.inputBuffer;
const inputData = inputBuffer.getChannelData(0); // 4096 samples
// Push latest mic samples into the rolling display buffer every callback (~85ms)
pushToDisplayBuf(inputDisplayBuf, inputData);
// Append incoming recorded samples to our accumulator
const temp = new Float32Array(micAccumulator.length + inputData.length);
temp.set(micAccumulator);
temp.set(inputData, micAccumulator.length);
micAccumulator = temp;
const targetChunkSize = parseInt(chunkSelect.value);
// Process and send chunks of the user's selected target size
while (micAccumulator.length >= targetChunkSize) {
const chunkToSend = micAccumulator.slice(0, targetChunkSize);
micAccumulator = micAccumulator.slice(targetChunkSize); // Keep remainder
// Voice Activity Detection for gate status badge
let maxVal = 0;
for (let i = 0; i < chunkToSend.length; i++) maxVal = Math.max(maxVal, Math.abs(chunkToSend[i]));
if (maxVal > 0.005) {
hudGateStatus.textContent = 'Bicara';
hudGateStatus.className = 'hud-value active-badge';
} else {
hudGateStatus.textContent = 'Berdiam';
hudGateStatus.className = 'hud-value text-muted';
}
// Send binary PCM Float32 audio chunk of target size to Python Server
if (socket && socket.readyState === WebSocket.OPEN) {
const packetTime = performance.now();
sentTimestamps.push({ id: packetTime, sent: packetTime });
if (sentTimestamps.length > maxSentLogs) {
sentTimestamps.shift();
}
socket.send(chunkToSend.buffer); // Send direct array buffer
}
}
};
micSourceNode.connect(scriptProcessorNode);
scriptProcessorNode.connect(audioContext.destination); // Required to trigger onaudioprocess
// Reset playback sync clock
nextPlaybackTime = 0;
micAccumulator = new Float32Array(0); // Reset accumulator
inputDisplayBuf = new Float32Array(VIS_DISPLAY_SIZE);
outputDisplayBuf = new Float32Array(VIS_DISPLAY_SIZE);
startVisualizerLoop();
console.log('Browser Streaming active. Recording buffer size: 4096 | Target chunk size:', chunkSelect.value);
} catch (e) {
console.error('Failed to access microphone:', e);
alert('Gagal mengakses mikrofon Anda: ' + e.message);
stopStreaming();
}
}
function stopStreaming() {
isStreaming = false;
streamBtn.textContent = 'Mulai Mengubah Suara';
streamBtn.className = 'btn btn-accent';
playOutput = true;
playToggleBtn.textContent = '🔊 Mendengarkan: AKTIF';
playToggleBtn.className = 'btn btn-primary';
const isHardwareMode = (routingModeSelect.value === 'hardware');
if (isHardwareMode) {
// --- SERVER HARDWARE ROUTING MODE ---
if (socket && socket.readyState === WebSocket.OPEN) {
const config = {
type: 'config',
routing_mode: 'browser' // Tells server to stop local hardware stream
};
socket.send(jsonEncode(config));
}
console.log('Server Hardware Mode stopped.');
hudGateStatus.textContent = 'Berdiam';
hudGateStatus.className = 'hud-value text-muted';
hudLatency.textContent = '-- ms';
hudTime.textContent = '-- ms';
stopVisualizerLoop();
inputDisplayBuf = new Float32Array(VIS_DISPLAY_SIZE);
outputDisplayBuf = new Float32Array(VIS_DISPLAY_SIZE);
clearCanvas(inputCanvas);
clearCanvas(outputCanvas);
return;
}
// --- CLIENT BROWSER MODE ---
// Stop microphone stream tracks
if (micStream) {
micStream.getTracks().forEach(track => track.stop());
micStream = null;
}
// Disconnect Web Audio nodes
if (micSourceNode) {
micSourceNode.disconnect();
micSourceNode = null;
}
if (scriptProcessorNode) {
scriptProcessorNode.disconnect();
scriptProcessorNode = null;
}
micAccumulator = new Float32Array(0); // Reset accumulator
stopVisualizerLoop();
inputDisplayBuf = new Float32Array(VIS_DISPLAY_SIZE);
outputDisplayBuf = new Float32Array(VIS_DISPLAY_SIZE);
hudGateStatus.textContent = 'Berdiam';
hudGateStatus.className = 'hud-value text-muted';
hudLatency.textContent = '-- ms';
hudTime.textContent = '-- ms';
clearCanvas(inputCanvas);
clearCanvas(outputCanvas);
}
// Seamless Audio Playback Scheduler (Absorbs WebSocket & processing jitter)
function handleServerAudioChunk(arrayBuffer) {
if (!isStreaming) return;
// 1. Measure Round-Trip Time Latency (RTT)
const now = performance.now();
let rtt = 0;
if (sentTimestamps.length > 0) {
const oldestSent = sentTimestamps.shift();
rtt = now - oldestSent.sent;
hudLatency.textContent = Math.round(rtt) + ' ms';
}
// Convert arrayBuffer to Float32 samples
const payload = new Float32Array(arrayBuffer);
const processingTime = payload[0]; // first float32 is the server processing time in ms
const pcmData = payload.subarray(1); // the rest is the audio
// 2. Schedule chunk smoothly inside the AudioContext timeline
const audioBuf = audioContext.createBuffer(1, pcmData.length, targetSampleRate);
audioBuf.getChannelData(0).set(pcmData);
const source = audioContext.createBufferSource();
source.buffer = audioBuf;
if (playOutput) {
source.connect(audioContext.destination);
}
// Calculate precise playback clock scheduling
const currentTime = audioContext.currentTime;
const chunkDuration = audioBuf.duration; // actual chunk duration in seconds
// Adaptive buffer: enough headroom so next chunk always arrives before this one ends.
// 2.5× chunk or 500ms cap — absorbs even 300ms+ processing spikes.
const adaptiveBuf = Math.min(chunkDuration * 2.5, 0.50);
if (nextPlaybackTime < currentTime) {
// Clock behind — first chunk or dropout recovery.
// Use full adaptiveBuf on BOTH cases so recovery fully rebuilds headroom.
// (0.5× recovery was causing cascading dropouts: one late chunk → the next also late)
nextPlaybackTime = currentTime + adaptiveBuf;
} else if (nextPlaybackTime > currentTime + chunkDuration * 5.0) {
// --- ADAPTIVE LATENCY BUSTER ---
// Only snap when queue is >5 chunk-durations ahead (genuine backlog, not normal look-ahead).
// At 8192 (170ms): threshold = 850ms
// At 65536 (1.6s): threshold = 8s
const snapTarget = currentTime + adaptiveBuf;
console.log(`Latency Buster: ${Math.round((nextPlaybackTime-currentTime)*1000)}ms → ${Math.round(adaptiveBuf*1000)}ms`);
nextPlaybackTime = snapTarget;
}
// Record schedule start time BEFORE advancing the clock (for time-synced visualizer)
const scheduleStartTime = nextPlaybackTime;
// Schedule play
source.start(nextPlaybackTime);
hudTime.textContent = Math.max(0, Math.round(processingTime)) + ' ms';
// Advance playback sync clock
nextPlaybackTime += audioBuf.duration;
// Push to time-synced output queue for visualizer (keyed by when audio actually plays)
outputChunkQueue.push({ data: pcmData, startTime: scheduleStartTime });
// Keep queue bounded to ~10 seconds of audio max
while (outputChunkQueue.length > 0) {
const c = outputChunkQueue[0];
if (c.startTime + c.data.length / targetSampleRate < audioContext.currentTime - 2.0) {
outputChunkQueue.shift();
} else break;
}
}
// --- VISUALIZATION / DRAWING ROUTINES ---
function drawWaveform(dataArray, canvas, strokeColor) {
const ctx = canvas.getContext('2d');
const width = canvas.width;
const height = canvas.height;
// Dark transparent redraw for trace/motion-blur effect
ctx.fillStyle = 'rgba(11, 12, 19, 0.4)';
ctx.fillRect(0, 0, width, height);
ctx.lineWidth = 2 * window.devicePixelRatio;
ctx.strokeStyle = strokeColor;
ctx.beginPath();
const sliceWidth = width / dataArray.length;
let x = 0;
for (let i = 0; i < dataArray.length; i++) {
// Center the wave around half-height and scale scale amplitude
const v = dataArray[i] * 1.5;
const y = (v * (height / 2)) + (height / 2);
if (i === 0) {
ctx.moveTo(x, y);
} else {
ctx.lineTo(x, y);
}
x += sliceWidth;
}
ctx.lineTo(width, height / 2);
ctx.stroke();
// Draw a subtle baseline center glowing path
ctx.strokeStyle = 'rgba(255, 255, 255, 0.05)';
ctx.lineWidth = 1;
ctx.beginPath();
ctx.moveTo(0, height / 2);
ctx.lineTo(width, height / 2);
ctx.stroke();
}
function clearCanvas(canvas) {
const ctx = canvas.getContext('2d');
ctx.fillStyle = '#0b0c13';
ctx.fillRect(0, 0, canvas.width, canvas.height);
}
// Apply initial UI layout on startup
applyAudioRoutingUI();
+243
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<!DOCTYPE html>
<html lang="id">
<head>
<meta charset="UTF-8">
<meta name="viewport" content="width=device-width, initial-scale=1.0">
<meta name="description" content="Omni Real-time Voice Changer - Pengubah suara real-time berbasis AI berlatensi sangat rendah dengan ONNX Runtime.">
<title>🎙️ Omni Real-Time Voice Changer - High-Performance AI Audio</title>
<!-- Modern Typography: Inter & Outfit from Google Fonts -->
<link rel="preconnect" href="https://fonts.googleapis.com">
<link rel="preconnect" href="https://fonts.gstatic.com" crossorigin>
<link href="https://fonts.googleapis.com/css2?family=Inter:wght@300;400;500;600;700&family=Outfit:wght@400;600;800&display=swap" rel="stylesheet">
<!-- Link to premium Vanilla CSS -->
<link rel="stylesheet" href="styles.css">
</head>
<body>
<div class="glow-backdrop"></div>
<div class="dashboard-container">
<!-- HEADER -->
<header class="app-header">
<div class="logo-area">
<span class="pulse-indicator active"></span>
<h1>🎙️ OMNI VOICE CHANGER</h1>
</div>
<p class="tagline">Pengubah Suara Real-Time AI Berlatensi Ultra Rendah menggunakan Akselerasi ONNX Runtime</p>
</header>
<!-- CONNECTION BAR -->
<div class="connection-bar card glassmorphism">
<div class="form-row">
<div class="input-group">
<label for="ws_url">URL Server WebSocket</label>
<input type="text" id="ws_url" value="ws://127.0.0.1:8765" placeholder="ws://localhost:8765">
</div>
<div class="connection-status-container">
<span id="connection_status" class="status-badge disconnected">Terputus</span>
</div>
<div class="btn-group-row">
<button id="connect_btn" class="btn btn-primary">Hubungkan Server</button>
<button id="stream_btn" class="btn btn-accent" disabled>Mulai Mengubah Suara</button>
<button id="play_toggle_btn" class="btn btn-primary" disabled>🔊 Mendengarkan: AKTIF</button>
</div>
</div>
</div>
<!-- MAIN DASHBOARD CONTENT -->
<main class="dashboard-grid">
<!-- MODEL CONFIGURATION -->
<section class="card glassmorphism col-span-1" aria-labelledby="model-config-title">
<h2 id="model-config-title" class="card-title">⚙️ Konfigurasi Model &amp; Perangkat</h2>
<!-- QUICK PRESETS PANEL -->
<div class="control-group">
<label>⚡ Quick Presets (Profil Performa)</label>
<div class="btn-group-row" style="width: 100%; display: grid; grid-template-columns: repeat(2, 1fr); gap: 0.5rem; height: auto; margin-bottom: 0.75rem;">
<button id="preset_latency_btn" class="btn btn-primary" style="font-size: 0.8rem; padding: 0.65rem 0.5rem;">⚡ Respon Kilat (PM)</button>
<button id="preset_quality_btn" class="btn btn-accent" style="font-size: 0.8rem; padding: 0.65rem 0.5rem;">🎙️ Kualitas Tinggi (RMVPE)</button>
</div>
</div>
<div class="control-group">
<label for="model_select">Pilih Model Suara (RVC ONNX)</label>
<select id="model_select" class="custom-select">
<option value="HuTao">HuTao (Genshin Impact)</option>
<option value="HuoHuo">HuoHuo (Honkai Star Rail)</option>
</select>
</div>
<div class="control-group">
<label for="device_select">Execution Provider (Akselerasi GPU)</label>
<select id="device_select" class="custom-select">
<option value="cpu">CPU (Sangat Stabil)</option>
<option value="cuda" selected>CUDA (NVIDIA GPU - Super Cepat)</option>
<option value="dml">DirectML (AMD/Intel GPU Windows)</option>
</select>
</div>
<!-- DUAL AUDIO ROUTING MODE (SERVER VS CLIENT) -->
<div class="control-group" style="border-top: 1px solid rgba(255, 255, 255, 0.05); padding-top: 0.75rem; margin-top: 0.75rem;">
<label for="routing_mode_select">Mode Audio (Routing Mode)</label>
<select id="routing_mode_select" class="custom-select">
<option value="browser" selected>Client Mode (Browser Streaming - Portabel)</option>
<option value="hardware">Server Mode (Hardware Direct - Latensi Nol)</option>
</select>
</div>
<div id="hardware_devices_panel" class="control-group" style="display: none; border: 1px solid rgba(99, 102, 241, 0.2); padding: 0.75rem; border-radius: 8px; background: rgba(11, 12, 19, 0.5); box-shadow: 0 0 10px rgba(99, 102, 241, 0.05);">
<div style="margin-bottom: 0.75rem;">
<label for="server_input_select" style="font-size: 0.75rem; margin-bottom: 0.25rem; color: var(--primary); text-transform: uppercase; font-weight: 600;">🎙️ Input Mikrofon Server</label>
<select id="server_input_select" class="custom-select" style="font-size: 0.8rem; padding: 0.4rem;"></select>
</div>
<div>
<label for="server_output_select" style="font-size: 0.75rem; margin-bottom: 0.25rem; color: var(--accent); text-transform: uppercase; font-weight: 600;">🔊 Output Speaker/Kabel Server</label>
<select id="server_output_select" class="custom-select" style="font-size: 0.8rem; padding: 0.4rem;"></select>
</div>
</div>
<div class="control-group">
<label>Metode Deteksi Nada (Pitch Extraction)</label>
<div class="radio-group-modern">
<label class="radio-tile">
<input type="radio" name="f0_method" value="pm" checked>
<span class="tile-label">PM (Tercepat)</span>
</label>
<label class="radio-tile">
<input type="radio" name="f0_method" value="dio">
<span class="tile-label">DIO (Ringan)</span>
</label>
<label class="radio-tile">
<input type="radio" name="f0_method" value="harvest">
<span class="tile-label">Harvest (Stabil)</span>
</label>
<label class="radio-tile">
<input type="radio" name="f0_method" value="rmvpe">
<span class="tile-label">RMVPE (Fidelitas Tinggi)</span>
</label>
</div>
</div>
<div class="control-group">
<div class="slider-header">
<label for="transpose_slider">Transpose (Pengubah Nada)</label>
<span id="transpose_val" class="slider-value">0 semitone</span>
</div>
<input type="range" id="transpose_slider" min="-24" max="24" value="0" step="1" class="custom-slider">
<div class="slider-ticks">
<span>-24 (Pria Berat)</span>
<span>0 (Asli)</span>
<span>+24 (Wanita/Anime)</span>
</div>
</div>
</section>
<!-- AUDIO DSP & PROCESSING -->
<section class="card glassmorphism col-span-1" aria-labelledby="dsp-title">
<h2 id="dsp-title" class="card-title">🎛️ Pemrosesan Audio (DSP)</h2>
<div class="control-group">
<div class="slider-header">
<label for="gate_slider">Noise Gate (Threshold)</label>
<span id="gate_val" class="slider-value">-40 dB</span>
</div>
<input type="range" id="gate_slider" min="-60" max="-10" value="-40" step="1" class="custom-slider">
<div class="slider-ticks">
<span>-60 dB (Sensitif)</span>
<span>-40 dB (Default)</span>
<span>-10 dB (Ketat)</span>
</div>
</div>
<div class="control-group">
<div class="slider-header">
<label for="input_gain_slider">Input Gain (Penguat Mic)</label>
<span id="input_gain_val" class="slider-value">1.0x</span>
</div>
<input type="range" id="input_gain_slider" min="0" max="3" value="1" step="0.1" class="custom-slider">
</div>
<div class="control-group">
<div class="slider-header">
<label for="output_gain_slider">Output Gain (Volume Suara)</label>
<span id="output_gain_val" class="slider-value">1.0x</span>
</div>
<input type="range" id="output_gain_slider" min="0" max="3" value="1" step="0.1" class="custom-slider">
</div>
<div id="browser_noise_cancel_group" class="control-group">
<label class="checkbox-container" style="display: flex; align-items: center; gap: 0.5rem; cursor: pointer; user-select: none;">
<input type="checkbox" id="noise_cancel_checkbox" checked style="width: 18px; height: 18px; cursor: pointer; accent-color: var(--primary);">
<span class="checkbox-label" style="font-size: 0.85rem; font-weight: 500; color: var(--text-muted); text-transform: uppercase;">🚫 Peredam Bising (Noise Cancel)</span>
</label>
</div>
<div class="control-group">
<label for="chunk_select">Ukuran Buffer (Chunk Size - Latensi vs Stabilitas)</label>
<select id="chunk_select" class="custom-select">
<option value="8192" selected>8192 sampel (~170ms - Rekomendasi Minim Distorsi)</option>
<option value="12288">12288 sampel (~250ms - Sangat Halus &amp; Merdu)</option>
<option value="16384">16384 sampel (~340ms - Kualitas Studio Sangat Stabil)</option>
<option value="24576">24576 sampel (~510ms - Super Halus &amp; Kokoh)</option>
<option value="32768">32768 sampel (~680ms - Fidelitas Maksimal)</option>
<option value="49152">49152 sampel (~1.0 detik - Ultra Smooth Cinema)</option>
<option value="65536">65536 sampel (~1.3 detik - Kestabilan Maksimal)</option>
<option value="98304">98304 sampel (~2.0 detik - Mode Penyiaran/Broadcasting)</option>
</select>
</div>
</section>
<!-- OSCILLOSCOPES / WAVEFORM VISUALIZERS -->
<section class="card glassmorphism col-span-2" aria-labelledby="visualizer-title">
<h2 id="visualizer-title" class="card-title">📊 Live Audio Waveform &amp; Visualizer</h2>
<div class="visualizer-row">
<div class="visualizer-container">
<div class="vis-label">
<span class="dot input-dot"></span>
<span>Sinyal Mikrofon (Input)</span>
</div>
<canvas id="input_canvas" class="waveform-canvas"></canvas>
</div>
<div class="visualizer-container">
<div class="vis-label">
<span class="dot output-dot"></span>
<span>Hasil AI Voice (Output)</span>
</div>
<canvas id="output_canvas" class="waveform-canvas"></canvas>
</div>
</div>
</section>
</main>
<!-- PERFORMANCE HUD FOOTER -->
<footer class="performance-hud card glassmorphism">
<div class="hud-item">
<span class="hud-label">Latensi Bulat (RTT)</span>
<span id="hud_latency" class="hud-value italic">-- ms</span>
</div>
<div class="hud-separator"></div>
<div class="hud-item">
<span class="hud-label">Rasio Pemrosesan</span>
<span id="hud_time" class="hud-value text-accent">-- ms</span>
</div>
<div class="hud-separator"></div>
<div class="hud-item">
<span class="hud-label">Sinyal Suara</span>
<span id="hud_gate_status" class="hud-value active-badge">Berdiam</span>
</div>
<div class="hud-separator"></div>
<div class="hud-item">
<span class="hud-label">Frekuensi Audio</span>
<span id="hud_sr" class="hud-value">44100 Hz</span>
</div>
</footer>
</div>
<!-- Link to premium Javascript logic -->
<script src="app.js"></script>
</body>
</html>
+595
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/* ==========================================================================
CSS GLOBAL TOKENS & RESET
========================================================================== */
:root {
--bg-dark: #07080e;
--bg-card: rgba(13, 17, 30, 0.7);
--border-color: rgba(99, 102, 241, 0.18);
--primary: #6366f1;
--primary-glow: rgba(99, 102, 241, 0.4);
--accent: #a855f7;
--accent-glow: rgba(168, 85, 247, 0.45);
--emerald: #10b981;
--rose: #ef4444;
--text-main: #e2e8f0;
--text-muted: #94a3b8;
--font-header: 'Outfit', 'Inter', -apple-system, BlinkMacSystemFont, sans-serif;
--font-body: 'Inter', -apple-system, BlinkMacSystemFont, sans-serif;
--transition-smooth: all 0.3s cubic-bezier(0.25, 0.8, 0.25, 1);
}
* {
box-sizing: border-box;
margin: 0;
padding: 0;
}
body {
background-color: var(--bg-dark);
color: var(--text-main);
font-family: var(--font-body);
min-height: 100vh;
overflow-x: hidden;
position: relative;
padding: 2rem 1.5rem;
}
/* ==========================================================================
DYNAMIC GLOWING BACKGROUND
========================================================================== */
.glow-backdrop {
position: fixed;
top: 0;
left: 0;
right: 0;
bottom: 0;
z-index: -1;
background:
radial-gradient(circle at 10% 20%, rgba(99, 102, 241, 0.08) 0%, transparent 40%),
radial-gradient(circle at 90% 80%, rgba(168, 85, 247, 0.09) 0%, transparent 45%);
pointer-events: none;
}
/* ==========================================================================
LAYOUT CONTAINER & CARDS
========================================================================== */
.dashboard-container {
max-width: 1200px;
margin: 0 auto;
display: flex;
flex-direction: column;
gap: 1.5rem;
}
.glassmorphism {
background: var(--bg-card);
backdrop-filter: blur(16px);
-webkit-backdrop-filter: blur(16px);
border: 1px solid var(--border-color);
border-radius: 16px;
box-shadow: 0 8px 32px 0 rgba(0, 0, 0, 0.37);
transition: var(--transition-smooth);
}
.glassmorphism:hover {
border-color: rgba(99, 102, 241, 0.3);
box-shadow: 0 10px 40px 0 rgba(99, 102, 241, 0.1);
}
.card {
padding: 1.75rem;
}
.card-title {
font-family: var(--font-header);
font-size: 1.25rem;
font-weight: 600;
margin-bottom: 1.25rem;
background: linear-gradient(135deg, #fff 0%, var(--text-muted) 100%);
-webkit-background-clip: text;
-webkit-text-fill-color: transparent;
border-bottom: 1px solid rgba(255, 255, 255, 0.05);
padding-bottom: 0.75rem;
}
/* ==========================================================================
APP HEADER
========================================================================== */
.app-header {
text-align: center;
margin-bottom: 1rem;
}
.logo-area {
display: inline-flex;
align-items: center;
gap: 0.75rem;
margin-bottom: 0.5rem;
}
.logo-area h1 {
font-family: var(--font-header);
font-size: 2.5rem;
font-weight: 800;
letter-spacing: -0.5px;
background: linear-gradient(135deg, var(--primary) 0%, var(--accent) 100%);
-webkit-background-clip: text;
-webkit-text-fill-color: transparent;
text-shadow: 0 0 40px rgba(99, 102, 241, 0.2);
}
.pulse-indicator {
width: 10px;
height: 10px;
border-radius: 50%;
background-color: var(--rose);
box-shadow: 0 0 10px var(--rose);
}
.pulse-indicator.active {
background-color: var(--emerald);
box-shadow: 0 0 10px var(--emerald);
animation: pulse 1.8s infinite;
}
.tagline {
color: var(--text-muted);
font-size: 0.95rem;
font-weight: 400;
max-width: 600px;
margin: 0 auto;
}
/* ==========================================================================
DASHBOARD GRID LAYOUT
========================================================================== */
.dashboard-grid {
display: grid;
grid-template-columns: repeat(2, 1fr);
gap: 1.5rem;
}
@media (max-width: 768px) {
.dashboard-grid {
grid-template-columns: 1fr;
}
.col-span-2 {
grid-column: span 1 !important;
}
}
.col-span-2 {
grid-column: span 2;
}
/* ==========================================================================
INPUTS & CONTROLS
========================================================================== */
.control-group {
margin-bottom: 1.25rem;
}
.control-group:last-child {
margin-bottom: 0;
}
label {
display: block;
font-size: 0.85rem;
font-weight: 500;
color: var(--text-muted);
margin-bottom: 0.5rem;
text-transform: uppercase;
letter-spacing: 0.5px;
}
.custom-select {
width: 100%;
padding: 0.8rem 1rem;
background-color: rgba(20, 24, 45, 0.8);
border: 1px solid var(--border-color);
border-radius: 8px;
color: var(--text-main);
font-size: 0.9rem;
font-family: var(--font-body);
outline: none;
transition: var(--transition-smooth);
cursor: pointer;
appearance: none;
background-image: url("data:image/svg+xml,%3Csvg xmlns='http://www.w3.org/2000/svg' width='24' height='24' viewBox='0 0 24 24' fill='none' stroke='%2394a3b8' stroke-width='2' stroke-linecap='round' stroke-linejoin='round'%3E%3Cpolyline points='6 9 12 15 18 9'%3E%3C/polyline%3E%3C/svg%3E");
background-repeat: no-repeat;
background-position: right 1rem center;
background-size: 1.2rem;
}
.custom-select:focus {
border-color: var(--primary);
box-shadow: 0 0 8px var(--primary-glow);
}
.input-group input {
background-color: rgba(20, 24, 45, 0.8);
border: 1px solid var(--border-color);
border-radius: 8px;
color: var(--text-main);
padding: 0.8rem 1rem;
width: 100%;
font-family: var(--font-body);
font-size: 0.9rem;
outline: none;
transition: var(--transition-smooth);
}
.input-group input:focus {
border-color: var(--primary);
box-shadow: 0 0 8px var(--primary-glow);
}
/* ==========================================================================
SLIDERS STYLING
========================================================================== */
.slider-header {
display: flex;
justify-content: space-between;
align-items: center;
margin-bottom: 0.25rem;
}
.slider-value {
font-family: var(--font-header);
font-weight: 600;
color: var(--accent);
text-shadow: 0 0 8px var(--accent-glow);
font-size: 0.95rem;
}
.custom-slider {
-webkit-appearance: none;
width: 100%;
height: 6px;
border-radius: 3px;
background: rgba(99, 102, 241, 0.15);
outline: none;
margin: 0.75rem 0;
}
.custom-slider::-webkit-slider-thumb {
-webkit-appearance: none;
appearance: none;
width: 18px;
height: 18px;
border-radius: 50%;
background: linear-gradient(135deg, var(--primary) 0%, var(--accent) 100%);
cursor: pointer;
box-shadow: 0 0 10px var(--primary-glow);
transition: transform 0.1s ease;
}
.custom-slider::-webkit-slider-thumb:hover {
transform: scale(1.2);
}
.slider-ticks {
display: flex;
justify-content: space-between;
font-size: 0.75rem;
color: var(--text-muted);
}
/* ==========================================================================
BUTTONS
========================================================================== */
.btn {
padding: 0.8rem 1.5rem;
border-radius: 8px;
font-family: var(--font-header);
font-weight: 600;
font-size: 0.9rem;
cursor: pointer;
border: none;
outline: none;
transition: var(--transition-smooth);
display: inline-flex;
align-items: center;
justify-content: center;
gap: 0.5rem;
}
.btn-primary {
background: linear-gradient(135deg, var(--primary) 0%, #4f46e5 100%);
color: white;
box-shadow: 0 4px 14px 0 var(--primary-glow);
}
.btn-primary:hover:not(:disabled) {
transform: translateY(-2px);
box-shadow: 0 6px 20px 0 rgba(99, 102, 241, 0.6);
}
.btn-accent {
background: linear-gradient(135deg, var(--accent) 0%, #7c3aed 100%);
color: white;
box-shadow: 0 4px 14px 0 var(--accent-glow);
}
.btn-accent:hover:not(:disabled) {
transform: translateY(-2px);
box-shadow: 0 6px 20px 0 rgba(168, 85, 247, 0.65);
}
.btn:active:not(:disabled) {
transform: translateY(0);
}
.btn:disabled {
opacity: 0.5;
cursor: not-allowed;
box-shadow: none;
}
/* ==========================================================================
CONNECTION BAR
========================================================================== */
.connection-bar {
padding: 1rem 1.5rem !important;
}
.form-row {
display: flex;
align-items: flex-end;
gap: 1.5rem;
flex-wrap: wrap;
}
.form-row .input-group {
flex: 1;
min-width: 250px;
}
.connection-status-container {
display: flex;
align-items: center;
height: 48px;
}
.status-badge {
padding: 0.4rem 0.8rem;
border-radius: 20px;
font-size: 0.8rem;
font-weight: 600;
text-transform: uppercase;
letter-spacing: 0.5px;
display: inline-flex;
align-items: center;
gap: 0.35rem;
}
.status-badge::before {
content: '';
display: inline-block;
width: 6px;
height: 6px;
border-radius: 50%;
}
.status-badge.connected {
background-color: rgba(16, 185, 129, 0.15);
color: var(--emerald);
border: 1px solid rgba(16, 185, 129, 0.3);
}
.status-badge.connected::before {
background-color: var(--emerald);
box-shadow: 0 0 6px var(--emerald);
}
.status-badge.disconnected {
background-color: rgba(239, 68, 68, 0.15);
color: var(--rose);
border: 1px solid rgba(239, 68, 68, 0.3);
}
.status-badge.disconnected::before {
background-color: var(--rose);
box-shadow: 0 0 6px var(--rose);
}
.status-badge.connecting {
background-color: rgba(168, 85, 247, 0.15);
color: var(--accent);
border: 1px solid rgba(168, 85, 247, 0.3);
}
.status-badge.connecting::before {
background-color: var(--accent);
box-shadow: 0 0 6px var(--accent);
animation: blink 1s infinite;
}
.btn-group-row {
display: flex;
gap: 0.75rem;
height: 48px;
}
/* ==========================================================================
MODERN RADIO TILES
========================================================================== */
.radio-group-modern {
display: grid;
grid-template-columns: repeat(2, 1fr);
gap: 0.5rem;
}
.radio-tile {
position: relative;
cursor: pointer;
margin: 0;
}
.radio-tile input {
position: absolute;
opacity: 0;
}
.tile-label {
display: block;
padding: 0.6rem;
background-color: rgba(20, 24, 45, 0.5);
border: 1px solid var(--border-color);
border-radius: 8px;
text-align: center;
font-size: 0.8rem;
font-weight: 500;
color: var(--text-muted);
transition: var(--transition-smooth);
}
.radio-tile input:checked + .tile-label {
background-color: rgba(99, 102, 241, 0.12);
border-color: var(--primary);
color: var(--text-main);
box-shadow: 0 0 10px rgba(99, 102, 241, 0.2);
}
.radio-tile:hover .tile-label {
border-color: rgba(99, 102, 241, 0.4);
}
/* ==========================================================================
OSCILLOSCOPE WAVEFORM CANVASES
========================================================================== */
.visualizer-row {
display: flex;
gap: 1.5rem;
flex-wrap: wrap;
}
.visualizer-container {
flex: 1;
min-width: 280px;
display: flex;
flex-direction: column;
gap: 0.5rem;
}
.vis-label {
display: flex;
align-items: center;
gap: 0.5rem;
font-size: 0.8rem;
font-weight: 500;
color: var(--text-muted);
}
.dot {
width: 6px;
height: 6px;
border-radius: 50%;
}
.input-dot {
background-color: var(--primary);
box-shadow: 0 0 6px var(--primary);
}
.output-dot {
background-color: var(--accent);
box-shadow: 0 0 6px var(--accent);
}
.waveform-canvas {
width: 100%;
height: 150px;
background-color: #0b0c13;
border-radius: 8px;
border: 1px solid rgba(255, 255, 255, 0.03);
}
/* ==========================================================================
PERFORMANCE HUD
========================================================================== */
.performance-hud {
display: flex;
justify-content: space-between;
align-items: center;
padding: 0.85rem 1.75rem !important;
}
.hud-item {
display: flex;
flex-direction: column;
gap: 0.15rem;
}
.hud-label {
font-size: 0.7rem;
text-transform: uppercase;
letter-spacing: 1px;
color: var(--text-muted);
font-weight: 500;
}
.hud-value {
font-family: var(--font-header);
font-size: 1.1rem;
font-weight: 700;
color: white;
}
.hud-separator {
width: 1px;
height: 30px;
background-color: rgba(255, 255, 255, 0.08);
}
.hud-value.text-accent {
color: var(--accent);
text-shadow: 0 0 8px var(--accent-glow);
}
.active-badge {
color: var(--emerald);
text-shadow: 0 0 6px rgba(16, 185, 129, 0.4);
}
@media (max-width: 600px) {
.performance-hud {
flex-direction: column;
align-items: flex-start;
gap: 0.75rem;
}
.hud-separator {
display: none;
}
}
/* ==========================================================================
KEYFRAME ANIMATIONS
========================================================================== */
@keyframes pulse {
0% {
transform: scale(0.9);
box-shadow: 0 0 0 0 rgba(16, 185, 129, 0.7);
}
70% {
transform: scale(1.1);
box-shadow: 0 0 0 10px rgba(16, 185, 129, 0);
}
100% {
transform: scale(0.9);
box-shadow: 0 0 0 0 rgba(16, 185, 129, 0);
}
}
@keyframes blink {
0%, 100% {
opacity: 1;
}
50% {
opacity: 0.4;
}
}
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import copy
import math
import numpy as np
import torch
from torch import nn
from torch.nn import functional as F
from lib.infer_pack import commons
from lib.infer_pack import modules
from lib.infer_pack.modules import LayerNorm
class Encoder(nn.Module):
def __init__(
self,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size=1,
p_dropout=0.0,
window_size=10,
**kwargs
):
super().__init__()
self.hidden_channels = hidden_channels
self.filter_channels = filter_channels
self.n_heads = n_heads
self.n_layers = n_layers
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.window_size = window_size
self.drop = nn.Dropout(p_dropout)
self.attn_layers = nn.ModuleList()
self.norm_layers_1 = nn.ModuleList()
self.ffn_layers = nn.ModuleList()
self.norm_layers_2 = nn.ModuleList()
for i in range(self.n_layers):
self.attn_layers.append(
MultiHeadAttention(
hidden_channels,
hidden_channels,
n_heads,
p_dropout=p_dropout,
window_size=window_size,
)
)
self.norm_layers_1.append(LayerNorm(hidden_channels))
self.ffn_layers.append(
FFN(
hidden_channels,
hidden_channels,
filter_channels,
kernel_size,
p_dropout=p_dropout,
)
)
self.norm_layers_2.append(LayerNorm(hidden_channels))
def forward(self, x, x_mask):
attn_mask = x_mask.unsqueeze(2) * x_mask.unsqueeze(-1)
x = x * x_mask
for i in range(self.n_layers):
y = self.attn_layers[i](x, x, attn_mask)
y = self.drop(y)
x = self.norm_layers_1[i](x + y)
y = self.ffn_layers[i](x, x_mask)
y = self.drop(y)
x = self.norm_layers_2[i](x + y)
x = x * x_mask
return x
class Decoder(nn.Module):
def __init__(
self,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size=1,
p_dropout=0.0,
proximal_bias=False,
proximal_init=True,
**kwargs
):
super().__init__()
self.hidden_channels = hidden_channels
self.filter_channels = filter_channels
self.n_heads = n_heads
self.n_layers = n_layers
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.proximal_bias = proximal_bias
self.proximal_init = proximal_init
self.drop = nn.Dropout(p_dropout)
self.self_attn_layers = nn.ModuleList()
self.norm_layers_0 = nn.ModuleList()
self.encdec_attn_layers = nn.ModuleList()
self.norm_layers_1 = nn.ModuleList()
self.ffn_layers = nn.ModuleList()
self.norm_layers_2 = nn.ModuleList()
for i in range(self.n_layers):
self.self_attn_layers.append(
MultiHeadAttention(
hidden_channels,
hidden_channels,
n_heads,
p_dropout=p_dropout,
proximal_bias=proximal_bias,
proximal_init=proximal_init,
)
)
self.norm_layers_0.append(LayerNorm(hidden_channels))
self.encdec_attn_layers.append(
MultiHeadAttention(
hidden_channels, hidden_channels, n_heads, p_dropout=p_dropout
)
)
self.norm_layers_1.append(LayerNorm(hidden_channels))
self.ffn_layers.append(
FFN(
hidden_channels,
hidden_channels,
filter_channels,
kernel_size,
p_dropout=p_dropout,
causal=True,
)
)
self.norm_layers_2.append(LayerNorm(hidden_channels))
def forward(self, x, x_mask, h, h_mask):
"""
x: decoder input
h: encoder output
"""
self_attn_mask = commons.subsequent_mask(x_mask.size(2)).to(
device=x.device, dtype=x.dtype
)
encdec_attn_mask = h_mask.unsqueeze(2) * x_mask.unsqueeze(-1)
x = x * x_mask
for i in range(self.n_layers):
y = self.self_attn_layers[i](x, x, self_attn_mask)
y = self.drop(y)
x = self.norm_layers_0[i](x + y)
y = self.encdec_attn_layers[i](x, h, encdec_attn_mask)
y = self.drop(y)
x = self.norm_layers_1[i](x + y)
y = self.ffn_layers[i](x, x_mask)
y = self.drop(y)
x = self.norm_layers_2[i](x + y)
x = x * x_mask
return x
class MultiHeadAttention(nn.Module):
def __init__(
self,
channels,
out_channels,
n_heads,
p_dropout=0.0,
window_size=None,
heads_share=True,
block_length=None,
proximal_bias=False,
proximal_init=False,
):
super().__init__()
assert channels % n_heads == 0
self.channels = channels
self.out_channels = out_channels
self.n_heads = n_heads
self.p_dropout = p_dropout
self.window_size = window_size
self.heads_share = heads_share
self.block_length = block_length
self.proximal_bias = proximal_bias
self.proximal_init = proximal_init
self.attn = None
self.k_channels = channels // n_heads
self.conv_q = nn.Conv1d(channels, channels, 1)
self.conv_k = nn.Conv1d(channels, channels, 1)
self.conv_v = nn.Conv1d(channels, channels, 1)
self.conv_o = nn.Conv1d(channels, out_channels, 1)
self.drop = nn.Dropout(p_dropout)
if window_size is not None:
n_heads_rel = 1 if heads_share else n_heads
rel_stddev = self.k_channels**-0.5
self.emb_rel_k = nn.Parameter(
torch.randn(n_heads_rel, window_size * 2 + 1, self.k_channels)
* rel_stddev
)
self.emb_rel_v = nn.Parameter(
torch.randn(n_heads_rel, window_size * 2 + 1, self.k_channels)
* rel_stddev
)
nn.init.xavier_uniform_(self.conv_q.weight)
nn.init.xavier_uniform_(self.conv_k.weight)
nn.init.xavier_uniform_(self.conv_v.weight)
if proximal_init:
with torch.no_grad():
self.conv_k.weight.copy_(self.conv_q.weight)
self.conv_k.bias.copy_(self.conv_q.bias)
def forward(self, x, c, attn_mask=None):
q = self.conv_q(x)
k = self.conv_k(c)
v = self.conv_v(c)
x, self.attn = self.attention(q, k, v, mask=attn_mask)
x = self.conv_o(x)
return x
def attention(self, query, key, value, mask=None):
# reshape [b, d, t] -> [b, n_h, t, d_k]
import torch.onnx.operators as onnx_ops
shape_k = onnx_ops.shape_as_tensor(key)
shape_q = onnx_ops.shape_as_tensor(query)
b = shape_k[0]
d = shape_k[1]
t_s = shape_k[2]
t_t = shape_q[2]
query = query.view(b, self.n_heads, self.k_channels, t_t).transpose(2, 3)
key = key.view(b, self.n_heads, self.k_channels, t_s).transpose(2, 3)
value = value.view(b, self.n_heads, self.k_channels, t_s).transpose(2, 3)
scores = torch.matmul(query / math.sqrt(self.k_channels), key.transpose(-2, -1))
if self.window_size is not None:
# Di ONNX, asersi dinamis kadang memicu error konstan, kita bypass di level grafik jika t_s == t_t selalu benar di self-attention
key_relative_embeddings = self._get_relative_embeddings(self.emb_rel_k, t_s)
rel_logits = self._matmul_with_relative_keys(
query / math.sqrt(self.k_channels), key_relative_embeddings
)
scores_local = self._relative_position_to_absolute_position(rel_logits)
scores = scores + scores_local
if self.proximal_bias:
assert t_s == t_t, "Proximal bias is only available for self-attention."
scores = scores + self._attention_bias_proximal(t_s).to(
device=scores.device, dtype=scores.dtype
)
if mask is not None:
scores = scores.masked_fill(mask == 0, -1e4)
if self.block_length is not None:
assert (
t_s == t_t
), "Local attention is only available for self-attention."
block_mask = (
torch.ones_like(scores)
.triu(-self.block_length)
.tril(self.block_length)
)
scores = scores.masked_fill(block_mask == 0, -1e4)
p_attn = F.softmax(scores, dim=-1) # [b, n_h, t_t, t_s]
p_attn = self.drop(p_attn)
output = torch.matmul(p_attn, value)
if self.window_size is not None:
relative_weights = self._absolute_position_to_relative_position(p_attn)
value_relative_embeddings = self._get_relative_embeddings(
self.emb_rel_v, t_s
)
output = output + self._matmul_with_relative_values(
relative_weights, value_relative_embeddings
)
output = (
output.transpose(2, 3).contiguous().view(b, d, t_t)
) # [b, n_h, t_t, d_k] -> [b, d, t_t]
return output, p_attn
def _matmul_with_relative_values(self, x, y):
"""
x: [b, h, l, m]
y: [h or 1, m, d]
ret: [b, h, l, d]
"""
ret = torch.matmul(x, y.unsqueeze(0))
return ret
def _matmul_with_relative_keys(self, x, y):
"""
x: [b, h, l, d]
y: [h or 1, m, d]
ret: [b, h, l, m]
"""
ret = torch.matmul(x, y.unsqueeze(0).transpose(-2, -1))
return ret
def _get_relative_embeddings(self, relative_embeddings, length):
if not isinstance(length, torch.Tensor):
length = torch.tensor(length, dtype=torch.int64, device=relative_embeddings.device)
else:
length = length.to(dtype=torch.int64, device=relative_embeddings.device)
max_relative_position = 2 * self.window_size + 1
pad_length = torch.clamp(length - (self.window_size + 1), min=0)
slice_start_position = torch.clamp((self.window_size + 1) - length, min=0)
slice_end_position = slice_start_position + 2 * length - 1
h = relative_embeddings.shape[0]
d = relative_embeddings.shape[2]
# dynamic pad using torch.cat and torch.zeros
zeros = torch.zeros(h, pad_length, d, dtype=relative_embeddings.dtype, device=relative_embeddings.device)
padded_relative_embeddings = torch.cat([zeros, relative_embeddings, zeros], dim=1)
used_relative_embeddings = padded_relative_embeddings[
:, slice_start_position:slice_end_position
]
return used_relative_embeddings
def _relative_position_to_absolute_position(self, x):
"""
x: [b, h, l, 2*l-1]
ret: [b, h, l, l]
"""
import torch.onnx.operators as onnx_ops
shape = onnx_ops.shape_as_tensor(x)
batch = shape[0]
heads = shape[1]
length = shape[2]
# Concat columns of pad to shift from relative to absolute indexing.
x = F.pad(x, [0, 1])
# Concat extra elements so to add up to shape (len+1, 2*len-1).
x_flat = x.view(batch, heads, -1)
zeros = torch.zeros(batch, heads, length - 1, dtype=x.dtype, device=x.device)
x_flat = torch.cat([x_flat, zeros], dim=-1)
# Reshape and slice out the padded elements.
x_final = x_flat.view(batch, heads, length + 1, 2 * length - 1)[
:, :, :length, length - 1 :
]
return x_final
def _absolute_position_to_relative_position(self, x):
"""
x: [b, h, l, l]
ret: [b, h, l, 2*l-1]
"""
import torch.onnx.operators as onnx_ops
shape = onnx_ops.shape_as_tensor(x)
batch = shape[0]
heads = shape[1]
length = shape[2]
# padd along column
zeros = torch.zeros(batch, heads, length, length - 1, dtype=x.dtype, device=x.device)
x = torch.cat([x, zeros], dim=-1)
x_flat = x.view(batch, heads, -1)
# add 0's in the beginning that will skew the elements after reshape
zeros_beg = torch.zeros(batch, heads, length, dtype=x.dtype, device=x.device)
x_flat = torch.cat([zeros_beg, x_flat], dim=-1)
x_final = x_flat.view(batch, heads, length, 2 * length)[:, :, :, 1:]
return x_final
def _attention_bias_proximal(self, length):
"""Bias for self-attention to encourage attention to close positions.
Args:
length: an integer scalar.
Returns:
a Tensor with shape [1, 1, length, length]
"""
r = torch.arange(length, dtype=torch.float32)
diff = torch.unsqueeze(r, 0) - torch.unsqueeze(r, 1)
return torch.unsqueeze(torch.unsqueeze(-torch.log1p(torch.abs(diff)), 0), 0)
class FFN(nn.Module):
def __init__(
self,
in_channels,
out_channels,
filter_channels,
kernel_size,
p_dropout=0.0,
activation=None,
causal=False,
):
super().__init__()
self.in_channels = in_channels
self.out_channels = out_channels
self.filter_channels = filter_channels
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.activation = activation
self.causal = causal
if causal:
self.padding = self._causal_padding
else:
self.padding = self._same_padding
self.conv_1 = nn.Conv1d(in_channels, filter_channels, kernel_size)
self.conv_2 = nn.Conv1d(filter_channels, out_channels, kernel_size)
self.drop = nn.Dropout(p_dropout)
def forward(self, x, x_mask):
x = self.conv_1(self.padding(x * x_mask))
if self.activation == "gelu":
x = x * torch.sigmoid(1.702 * x)
else:
x = torch.relu(x)
x = self.drop(x)
x = self.conv_2(self.padding(x * x_mask))
return x * x_mask
def _causal_padding(self, x):
if self.kernel_size == 1:
return x
pad_l = self.kernel_size - 1
pad_r = 0
padding = [[0, 0], [0, 0], [pad_l, pad_r]]
x = F.pad(x, commons.convert_pad_shape(padding))
return x
def _same_padding(self, x):
if self.kernel_size == 1:
return x
pad_l = (self.kernel_size - 1) // 2
pad_r = self.kernel_size // 2
padding = [[0, 0], [0, 0], [pad_l, pad_r]]
x = F.pad(x, commons.convert_pad_shape(padding))
return x
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import math
import numpy as np
import torch
from torch import nn
from torch.nn import functional as F
def init_weights(m, mean=0.0, std=0.01):
classname = m.__class__.__name__
if classname.find("Conv") != -1:
m.weight.data.normal_(mean, std)
def get_padding(kernel_size, dilation=1):
return int((kernel_size * dilation - dilation) / 2)
def convert_pad_shape(pad_shape):
l = pad_shape[::-1]
pad_shape = [item for sublist in l for item in sublist]
return pad_shape
def kl_divergence(m_p, logs_p, m_q, logs_q):
"""KL(P||Q)"""
kl = (logs_q - logs_p) - 0.5
kl += (
0.5 * (torch.exp(2.0 * logs_p) + ((m_p - m_q) ** 2)) * torch.exp(-2.0 * logs_q)
)
return kl
def rand_gumbel(shape):
"""Sample from the Gumbel distribution, protect from overflows."""
uniform_samples = torch.rand(shape) * 0.99998 + 0.00001
return -torch.log(-torch.log(uniform_samples))
def rand_gumbel_like(x):
g = rand_gumbel(x.size()).to(dtype=x.dtype, device=x.device)
return g
def slice_segments(x, ids_str, segment_size=4):
ret = torch.zeros_like(x[:, :, :segment_size])
for i in range(x.size(0)):
idx_str = ids_str[i]
idx_end = idx_str + segment_size
ret[i] = x[i, :, idx_str:idx_end]
return ret
def slice_segments2(x, ids_str, segment_size=4):
ret = torch.zeros_like(x[:, :segment_size])
for i in range(x.size(0)):
idx_str = ids_str[i]
idx_end = idx_str + segment_size
ret[i] = x[i, idx_str:idx_end]
return ret
def rand_slice_segments(x, x_lengths=None, segment_size=4):
b, d, t = x.size()
if x_lengths is None:
x_lengths = t
ids_str_max = x_lengths - segment_size + 1
ids_str = (torch.rand([b]).to(device=x.device) * ids_str_max).to(dtype=torch.long)
ret = slice_segments(x, ids_str, segment_size)
return ret, ids_str
def get_timing_signal_1d(length, channels, min_timescale=1.0, max_timescale=1.0e4):
position = torch.arange(length, dtype=torch.float)
num_timescales = channels // 2
log_timescale_increment = math.log(float(max_timescale) / float(min_timescale)) / (
num_timescales - 1
)
inv_timescales = min_timescale * torch.exp(
torch.arange(num_timescales, dtype=torch.float) * -log_timescale_increment
)
scaled_time = position.unsqueeze(0) * inv_timescales.unsqueeze(1)
signal = torch.cat([torch.sin(scaled_time), torch.cos(scaled_time)], 0)
signal = F.pad(signal, [0, 0, 0, channels % 2])
signal = signal.view(1, channels, length)
return signal
def add_timing_signal_1d(x, min_timescale=1.0, max_timescale=1.0e4):
b, channels, length = x.size()
signal = get_timing_signal_1d(length, channels, min_timescale, max_timescale)
return x + signal.to(dtype=x.dtype, device=x.device)
def cat_timing_signal_1d(x, min_timescale=1.0, max_timescale=1.0e4, axis=1):
b, channels, length = x.size()
signal = get_timing_signal_1d(length, channels, min_timescale, max_timescale)
return torch.cat([x, signal.to(dtype=x.dtype, device=x.device)], axis)
def subsequent_mask(length):
mask = torch.tril(torch.ones(length, length)).unsqueeze(0).unsqueeze(0)
return mask
@torch.jit.script
def fused_add_tanh_sigmoid_multiply(input_a, input_b, n_channels):
n_channels_int = n_channels[0]
in_act = input_a + input_b
t_act = torch.tanh(in_act[:, :n_channels_int, :])
s_act = torch.sigmoid(in_act[:, n_channels_int:, :])
acts = t_act * s_act
return acts
def convert_pad_shape(pad_shape):
l = pad_shape[::-1]
pad_shape = [item for sublist in l for item in sublist]
return pad_shape
def shift_1d(x):
x = F.pad(x, convert_pad_shape([[0, 0], [0, 0], [1, 0]]))[:, :, :-1]
return x
def sequence_mask(length, max_length=None):
if max_length is None:
max_length = length.max()
x = torch.arange(max_length, dtype=length.dtype, device=length.device)
return x.unsqueeze(0) < length.unsqueeze(1)
def generate_path(duration, mask):
"""
duration: [b, 1, t_x]
mask: [b, 1, t_y, t_x]
"""
device = duration.device
b, _, t_y, t_x = mask.shape
cum_duration = torch.cumsum(duration, -1)
cum_duration_flat = cum_duration.view(b * t_x)
path = sequence_mask(cum_duration_flat, t_y).to(mask.dtype)
path = path.view(b, t_x, t_y)
path = path - F.pad(path, convert_pad_shape([[0, 0], [1, 0], [0, 0]]))[:, :-1]
path = path.unsqueeze(1).transpose(2, 3) * mask
return path
def clip_grad_value_(parameters, clip_value, norm_type=2):
if isinstance(parameters, torch.Tensor):
parameters = [parameters]
parameters = list(filter(lambda p: p.grad is not None, parameters))
norm_type = float(norm_type)
if clip_value is not None:
clip_value = float(clip_value)
total_norm = 0
for p in parameters:
param_norm = p.grad.data.norm(norm_type)
total_norm += param_norm.item() ** norm_type
if clip_value is not None:
p.grad.data.clamp_(min=-clip_value, max=clip_value)
total_norm = total_norm ** (1.0 / norm_type)
return total_norm
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import math, pdb, os
from time import time as ttime
import torch
from torch import nn
from torch.nn import functional as F
from lib.infer_pack import modules
from lib.infer_pack import attentions
from lib.infer_pack import commons
from lib.infer_pack.commons import init_weights, get_padding
from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d
from torch.nn.utils import weight_norm, remove_weight_norm, spectral_norm
from lib.infer_pack.commons import init_weights
import numpy as np
from lib.infer_pack import commons
class TextEncoder256(nn.Module):
def __init__(
self,
out_channels,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout,
f0=True,
):
super().__init__()
self.out_channels = out_channels
self.hidden_channels = hidden_channels
self.filter_channels = filter_channels
self.n_heads = n_heads
self.n_layers = n_layers
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.emb_phone = nn.Linear(256, hidden_channels)
self.lrelu = nn.LeakyReLU(0.1, inplace=True)
if f0 == True:
self.emb_pitch = nn.Embedding(256, hidden_channels) # pitch 256
self.encoder = attentions.Encoder(
hidden_channels, filter_channels, n_heads, n_layers, kernel_size, p_dropout
)
self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1)
def forward(self, phone, pitch, lengths):
if pitch == None:
x = self.emb_phone(phone)
else:
x = self.emb_phone(phone) + self.emb_pitch(pitch)
x = x * math.sqrt(self.hidden_channels) # [b, t, h]
x = self.lrelu(x)
x = torch.transpose(x, 1, -1) # [b, h, t]
x_mask = torch.unsqueeze(commons.sequence_mask(lengths, x.size(2)), 1).to(
x.dtype
)
x = self.encoder(x * x_mask, x_mask)
stats = self.proj(x) * x_mask
m, logs = torch.split(stats, self.out_channels, dim=1)
return m, logs, x_mask
class TextEncoder768(nn.Module):
def __init__(
self,
out_channels,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout,
f0=True,
):
super().__init__()
self.out_channels = out_channels
self.hidden_channels = hidden_channels
self.filter_channels = filter_channels
self.n_heads = n_heads
self.n_layers = n_layers
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.emb_phone = nn.Linear(768, hidden_channels)
self.lrelu = nn.LeakyReLU(0.1, inplace=True)
if f0 == True:
self.emb_pitch = nn.Embedding(256, hidden_channels) # pitch 256
self.encoder = attentions.Encoder(
hidden_channels, filter_channels, n_heads, n_layers, kernel_size, p_dropout
)
self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1)
def forward(self, phone, pitch, lengths):
if pitch == None:
x = self.emb_phone(phone)
else:
x = self.emb_phone(phone) + self.emb_pitch(pitch)
x = x * math.sqrt(self.hidden_channels) # [b, t, h]
x = self.lrelu(x)
x = torch.transpose(x, 1, -1) # [b, h, t]
x_mask = torch.unsqueeze(commons.sequence_mask(lengths, x.size(2)), 1).to(
x.dtype
)
x = self.encoder(x * x_mask, x_mask)
stats = self.proj(x) * x_mask
m, logs = torch.split(stats, self.out_channels, dim=1)
return m, logs, x_mask
class ResidualCouplingBlock(nn.Module):
def __init__(
self,
channels,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
n_flows=4,
gin_channels=0,
):
super().__init__()
self.channels = channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.n_flows = n_flows
self.gin_channels = gin_channels
self.flows = nn.ModuleList()
for i in range(n_flows):
self.flows.append(
modules.ResidualCouplingLayer(
channels,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
gin_channels=gin_channels,
mean_only=True,
)
)
self.flows.append(modules.Flip())
def forward(self, x, x_mask, g=None, reverse=False):
if not reverse:
for flow in self.flows:
x, _ = flow(x, x_mask, g=g, reverse=reverse)
else:
for flow in reversed(self.flows):
x = flow(x, x_mask, g=g, reverse=reverse)
return x
def remove_weight_norm(self):
for i in range(self.n_flows):
self.flows[i * 2].remove_weight_norm()
class PosteriorEncoder(nn.Module):
def __init__(
self,
in_channels,
out_channels,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
gin_channels=0,
):
super().__init__()
self.in_channels = in_channels
self.out_channels = out_channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.gin_channels = gin_channels
self.pre = nn.Conv1d(in_channels, hidden_channels, 1)
self.enc = modules.WN(
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
gin_channels=gin_channels,
)
self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1)
def forward(self, x, x_lengths, g=None):
x_mask = torch.unsqueeze(commons.sequence_mask(x_lengths, x.size(2)), 1).to(
x.dtype
)
x = self.pre(x) * x_mask
x = self.enc(x, x_mask, g=g)
stats = self.proj(x) * x_mask
m, logs = torch.split(stats, self.out_channels, dim=1)
z = (m + torch.randn_like(m) * torch.exp(logs)) * x_mask
return z, m, logs, x_mask
def remove_weight_norm(self):
self.enc.remove_weight_norm()
class Generator(torch.nn.Module):
def __init__(
self,
initial_channel,
resblock,
resblock_kernel_sizes,
resblock_dilation_sizes,
upsample_rates,
upsample_initial_channel,
upsample_kernel_sizes,
gin_channels=0,
):
super(Generator, self).__init__()
self.num_kernels = len(resblock_kernel_sizes)
self.num_upsamples = len(upsample_rates)
self.conv_pre = Conv1d(
initial_channel, upsample_initial_channel, 7, 1, padding=3
)
resblock = modules.ResBlock1 if resblock == "1" else modules.ResBlock2
self.ups = nn.ModuleList()
for i, (u, k) in enumerate(zip(upsample_rates, upsample_kernel_sizes)):
self.ups.append(
weight_norm(
ConvTranspose1d(
upsample_initial_channel // (2**i),
upsample_initial_channel // (2 ** (i + 1)),
k,
u,
padding=(k - u) // 2,
)
)
)
self.resblocks = nn.ModuleList()
for i in range(len(self.ups)):
ch = upsample_initial_channel // (2 ** (i + 1))
for j, (k, d) in enumerate(
zip(resblock_kernel_sizes, resblock_dilation_sizes)
):
self.resblocks.append(resblock(ch, k, d))
self.conv_post = Conv1d(ch, 1, 7, 1, padding=3, bias=False)
self.ups.apply(init_weights)
if gin_channels != 0:
self.cond = nn.Conv1d(gin_channels, upsample_initial_channel, 1)
def forward(self, x, g=None):
x = self.conv_pre(x)
if g is not None:
x = x + self.cond(g)
for i in range(self.num_upsamples):
x = F.leaky_relu(x, modules.LRELU_SLOPE)
x = self.ups[i](x)
xs = None
for j in range(self.num_kernels):
if xs is None:
xs = self.resblocks[i * self.num_kernels + j](x)
else:
xs += self.resblocks[i * self.num_kernels + j](x)
x = xs / self.num_kernels
x = F.leaky_relu(x)
x = self.conv_post(x)
x = torch.tanh(x)
return x
def remove_weight_norm(self):
for l in self.ups:
remove_weight_norm(l)
for l in self.resblocks:
l.remove_weight_norm()
class SineGen(torch.nn.Module):
"""Definition of sine generator
SineGen(samp_rate, harmonic_num = 0,
sine_amp = 0.1, noise_std = 0.003,
voiced_threshold = 0,
flag_for_pulse=False)
samp_rate: sampling rate in Hz
harmonic_num: number of harmonic overtones (default 0)
sine_amp: amplitude of sine-wavefrom (default 0.1)
noise_std: std of Gaussian noise (default 0.003)
voiced_thoreshold: F0 threshold for U/V classification (default 0)
flag_for_pulse: this SinGen is used inside PulseGen (default False)
Note: when flag_for_pulse is True, the first time step of a voiced
segment is always sin(np.pi) or cos(0)
"""
def __init__(
self,
samp_rate,
harmonic_num=0,
sine_amp=0.1,
noise_std=0.003,
voiced_threshold=0,
flag_for_pulse=False,
):
super(SineGen, self).__init__()
self.sine_amp = sine_amp
self.noise_std = noise_std
self.harmonic_num = harmonic_num
self.dim = self.harmonic_num + 1
self.sampling_rate = samp_rate
self.voiced_threshold = voiced_threshold
def _f02uv(self, f0):
# generate uv signal
uv = torch.ones_like(f0)
uv = uv * (f0 > self.voiced_threshold)
return uv
def forward(self, f0, upp):
"""sine_tensor, uv = forward(f0)
input F0: tensor(batchsize=1, length, dim=1)
f0 for unvoiced steps should be 0
output sine_tensor: tensor(batchsize=1, length, dim)
output uv: tensor(batchsize=1, length, 1)
"""
with torch.no_grad():
f0 = f0[:, None].transpose(1, 2)
f0_buf = torch.zeros(f0.shape[0], f0.shape[1], self.dim, device=f0.device)
# fundamental component
f0_buf[:, :, 0] = f0[:, :, 0]
for idx in np.arange(self.harmonic_num):
f0_buf[:, :, idx + 1] = f0_buf[:, :, 0] * (
idx + 2
) # idx + 2: the (idx+1)-th overtone, (idx+2)-th harmonic
rad_values = (f0_buf / self.sampling_rate) % 1 ###%1意味着n_har的乘积无法后处理优化
rand_ini = torch.rand(
f0_buf.shape[0], f0_buf.shape[2], device=f0_buf.device
)
rand_ini[:, 0] = 0
rad_values[:, 0, :] = rad_values[:, 0, :] + rand_ini
tmp_over_one = torch.cumsum(rad_values, 1) # % 1 #####%1意味着后面的cumsum无法再优化
tmp_over_one *= upp
tmp_over_one = F.interpolate(
tmp_over_one.transpose(2, 1),
scale_factor=upp,
mode="linear",
align_corners=True,
).transpose(2, 1)
rad_values = F.interpolate(
rad_values.transpose(2, 1), scale_factor=upp, mode="nearest"
).transpose(
2, 1
) #######
tmp_over_one %= 1
tmp_over_one_idx = (tmp_over_one[:, 1:, :] - tmp_over_one[:, :-1, :]) < 0
cumsum_shift = torch.zeros_like(rad_values)
cumsum_shift[:, 1:, :] = tmp_over_one_idx * -1.0
sine_waves = torch.sin(
torch.cumsum(rad_values + cumsum_shift, dim=1) * 2 * np.pi
)
sine_waves = sine_waves * self.sine_amp
uv = self._f02uv(f0)
uv = F.interpolate(
uv.transpose(2, 1), scale_factor=upp, mode="nearest"
).transpose(2, 1)
noise_amp = uv * self.noise_std + (1 - uv) * self.sine_amp / 3
noise = noise_amp * torch.randn_like(sine_waves)
sine_waves = sine_waves * uv + noise
return sine_waves, uv, noise
class SourceModuleHnNSF(torch.nn.Module):
"""SourceModule for hn-nsf
SourceModule(sampling_rate, harmonic_num=0, sine_amp=0.1,
add_noise_std=0.003, voiced_threshod=0)
sampling_rate: sampling_rate in Hz
harmonic_num: number of harmonic above F0 (default: 0)
sine_amp: amplitude of sine source signal (default: 0.1)
add_noise_std: std of additive Gaussian noise (default: 0.003)
note that amplitude of noise in unvoiced is decided
by sine_amp
voiced_threshold: threhold to set U/V given F0 (default: 0)
Sine_source, noise_source = SourceModuleHnNSF(F0_sampled)
F0_sampled (batchsize, length, 1)
Sine_source (batchsize, length, 1)
noise_source (batchsize, length 1)
uv (batchsize, length, 1)
"""
def __init__(
self,
sampling_rate,
harmonic_num=0,
sine_amp=0.1,
add_noise_std=0.003,
voiced_threshod=0,
is_half=True,
):
super(SourceModuleHnNSF, self).__init__()
self.sine_amp = sine_amp
self.noise_std = add_noise_std
self.is_half = is_half
# to produce sine waveforms
self.l_sin_gen = SineGen(
sampling_rate, harmonic_num, sine_amp, add_noise_std, voiced_threshod
)
# to merge source harmonics into a single excitation
self.l_linear = torch.nn.Linear(harmonic_num + 1, 1)
self.l_tanh = torch.nn.Tanh()
def forward(self, x, upp=None):
sine_wavs, uv, _ = self.l_sin_gen(x, upp)
if self.is_half:
sine_wavs = sine_wavs.half()
sine_merge = self.l_tanh(self.l_linear(sine_wavs))
return sine_merge, None, None # noise, uv
class GeneratorNSF(torch.nn.Module):
def __init__(
self,
initial_channel,
resblock,
resblock_kernel_sizes,
resblock_dilation_sizes,
upsample_rates,
upsample_initial_channel,
upsample_kernel_sizes,
gin_channels,
sr,
is_half=False,
):
super(GeneratorNSF, self).__init__()
self.num_kernels = len(resblock_kernel_sizes)
self.num_upsamples = len(upsample_rates)
self.f0_upsamp = torch.nn.Upsample(scale_factor=np.prod(upsample_rates))
self.m_source = SourceModuleHnNSF(
sampling_rate=sr, harmonic_num=0, is_half=is_half
)
self.noise_convs = nn.ModuleList()
self.conv_pre = Conv1d(
initial_channel, upsample_initial_channel, 7, 1, padding=3
)
resblock = modules.ResBlock1 if resblock == "1" else modules.ResBlock2
self.ups = nn.ModuleList()
for i, (u, k) in enumerate(zip(upsample_rates, upsample_kernel_sizes)):
c_cur = upsample_initial_channel // (2 ** (i + 1))
self.ups.append(
weight_norm(
ConvTranspose1d(
upsample_initial_channel // (2**i),
upsample_initial_channel // (2 ** (i + 1)),
k,
u,
padding=(k - u) // 2,
)
)
)
if i + 1 < len(upsample_rates):
stride_f0 = np.prod(upsample_rates[i + 1 :])
self.noise_convs.append(
Conv1d(
1,
c_cur,
kernel_size=stride_f0 * 2,
stride=stride_f0,
padding=stride_f0 // 2,
)
)
else:
self.noise_convs.append(Conv1d(1, c_cur, kernel_size=1))
self.resblocks = nn.ModuleList()
for i in range(len(self.ups)):
ch = upsample_initial_channel // (2 ** (i + 1))
for j, (k, d) in enumerate(
zip(resblock_kernel_sizes, resblock_dilation_sizes)
):
self.resblocks.append(resblock(ch, k, d))
self.conv_post = Conv1d(ch, 1, 7, 1, padding=3, bias=False)
self.ups.apply(init_weights)
if gin_channels != 0:
self.cond = nn.Conv1d(gin_channels, upsample_initial_channel, 1)
self.upp = np.prod(upsample_rates)
def forward(self, x, f0, g=None):
har_source, noi_source, uv = self.m_source(f0, self.upp)
har_source = har_source.transpose(1, 2)
x = self.conv_pre(x)
if g is not None:
x = x + self.cond(g)
for i in range(self.num_upsamples):
x = F.leaky_relu(x, modules.LRELU_SLOPE)
x = self.ups[i](x)
x_source = self.noise_convs[i](har_source)
x = x + x_source
xs = None
for j in range(self.num_kernels):
if xs is None:
xs = self.resblocks[i * self.num_kernels + j](x)
else:
xs += self.resblocks[i * self.num_kernels + j](x)
x = xs / self.num_kernels
x = F.leaky_relu(x)
x = self.conv_post(x)
x = torch.tanh(x)
return x
def remove_weight_norm(self):
for l in self.ups:
remove_weight_norm(l)
for l in self.resblocks:
l.remove_weight_norm()
sr2sr = {
"32k": 32000,
"40k": 40000,
"48k": 48000,
}
class SynthesizerTrnMsNSFsidM(nn.Module):
def __init__(
self,
spec_channels,
segment_size,
inter_channels,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout,
resblock,
resblock_kernel_sizes,
resblock_dilation_sizes,
upsample_rates,
upsample_initial_channel,
upsample_kernel_sizes,
spk_embed_dim,
gin_channels,
sr,
version,
**kwargs
):
super().__init__()
if type(sr) == type("strr"):
sr = sr2sr[sr]
self.spec_channels = spec_channels
self.inter_channels = inter_channels
self.hidden_channels = hidden_channels
self.filter_channels = filter_channels
self.n_heads = n_heads
self.n_layers = n_layers
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.resblock = resblock
self.resblock_kernel_sizes = resblock_kernel_sizes
self.resblock_dilation_sizes = resblock_dilation_sizes
self.upsample_rates = upsample_rates
self.upsample_initial_channel = upsample_initial_channel
self.upsample_kernel_sizes = upsample_kernel_sizes
self.segment_size = segment_size
self.gin_channels = gin_channels
# self.hop_length = hop_length#
self.spk_embed_dim = spk_embed_dim
if version == "v1":
self.enc_p = TextEncoder256(
inter_channels,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout,
)
else:
self.enc_p = TextEncoder768(
inter_channels,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout,
)
self.dec = GeneratorNSF(
inter_channels,
resblock,
resblock_kernel_sizes,
resblock_dilation_sizes,
upsample_rates,
upsample_initial_channel,
upsample_kernel_sizes,
gin_channels=gin_channels,
sr=sr,
is_half=kwargs["is_half"],
)
self.enc_q = PosteriorEncoder(
spec_channels,
inter_channels,
hidden_channels,
5,
1,
16,
gin_channels=gin_channels,
)
self.flow = ResidualCouplingBlock(
inter_channels, hidden_channels, 5, 1, 3, gin_channels=gin_channels
)
self.emb_g = nn.Embedding(self.spk_embed_dim, gin_channels)
self.speaker_map = None
print("gin_channels:", gin_channels, "self.spk_embed_dim:", self.spk_embed_dim)
def remove_weight_norm(self):
self.dec.remove_weight_norm()
self.flow.remove_weight_norm()
self.enc_q.remove_weight_norm()
def construct_spkmixmap(self, n_speaker):
self.speaker_map = torch.zeros((n_speaker, 1, 1, self.gin_channels))
for i in range(n_speaker):
self.speaker_map[i] = self.emb_g(torch.LongTensor([[i]]))
self.speaker_map = self.speaker_map.unsqueeze(0)
def forward(self, phone, phone_lengths, pitch, nsff0, g, rnd, max_len=None):
if self.speaker_map is not None: # [N, S] * [S, B, 1, H]
g = g.reshape((g.shape[0], g.shape[1], 1, 1, 1)) # [N, S, B, 1, 1]
g = g * self.speaker_map # [N, S, B, 1, H]
g = torch.sum(g, dim=1) # [N, 1, B, 1, H]
g = g.transpose(0, -1).transpose(0, -2).squeeze(0) # [B, H, N]
else:
g = g.unsqueeze(0)
g = self.emb_g(g).transpose(1, 2)
m_p, logs_p, x_mask = self.enc_p(phone, pitch, phone_lengths)
z_p = (m_p + torch.exp(logs_p) * rnd) * x_mask
z = self.flow(z_p, x_mask, g=g, reverse=True)
o = self.dec((z * x_mask)[:, :, :max_len], nsff0, g=g)
return o
class MultiPeriodDiscriminator(torch.nn.Module):
def __init__(self, use_spectral_norm=False):
super(MultiPeriodDiscriminator, self).__init__()
periods = [2, 3, 5, 7, 11, 17]
# periods = [3, 5, 7, 11, 17, 23, 37]
discs = [DiscriminatorS(use_spectral_norm=use_spectral_norm)]
discs = discs + [
DiscriminatorP(i, use_spectral_norm=use_spectral_norm) for i in periods
]
self.discriminators = nn.ModuleList(discs)
def forward(self, y, y_hat):
y_d_rs = [] #
y_d_gs = []
fmap_rs = []
fmap_gs = []
for i, d in enumerate(self.discriminators):
y_d_r, fmap_r = d(y)
y_d_g, fmap_g = d(y_hat)
# for j in range(len(fmap_r)):
# print(i,j,y.shape,y_hat.shape,fmap_r[j].shape,fmap_g[j].shape)
y_d_rs.append(y_d_r)
y_d_gs.append(y_d_g)
fmap_rs.append(fmap_r)
fmap_gs.append(fmap_g)
return y_d_rs, y_d_gs, fmap_rs, fmap_gs
class MultiPeriodDiscriminatorV2(torch.nn.Module):
def __init__(self, use_spectral_norm=False):
super(MultiPeriodDiscriminatorV2, self).__init__()
# periods = [2, 3, 5, 7, 11, 17]
periods = [2, 3, 5, 7, 11, 17, 23, 37]
discs = [DiscriminatorS(use_spectral_norm=use_spectral_norm)]
discs = discs + [
DiscriminatorP(i, use_spectral_norm=use_spectral_norm) for i in periods
]
self.discriminators = nn.ModuleList(discs)
def forward(self, y, y_hat):
y_d_rs = [] #
y_d_gs = []
fmap_rs = []
fmap_gs = []
for i, d in enumerate(self.discriminators):
y_d_r, fmap_r = d(y)
y_d_g, fmap_g = d(y_hat)
# for j in range(len(fmap_r)):
# print(i,j,y.shape,y_hat.shape,fmap_r[j].shape,fmap_g[j].shape)
y_d_rs.append(y_d_r)
y_d_gs.append(y_d_g)
fmap_rs.append(fmap_r)
fmap_gs.append(fmap_g)
return y_d_rs, y_d_gs, fmap_rs, fmap_gs
class DiscriminatorS(torch.nn.Module):
def __init__(self, use_spectral_norm=False):
super(DiscriminatorS, self).__init__()
norm_f = weight_norm if use_spectral_norm == False else spectral_norm
self.convs = nn.ModuleList(
[
norm_f(Conv1d(1, 16, 15, 1, padding=7)),
norm_f(Conv1d(16, 64, 41, 4, groups=4, padding=20)),
norm_f(Conv1d(64, 256, 41, 4, groups=16, padding=20)),
norm_f(Conv1d(256, 1024, 41, 4, groups=64, padding=20)),
norm_f(Conv1d(1024, 1024, 41, 4, groups=256, padding=20)),
norm_f(Conv1d(1024, 1024, 5, 1, padding=2)),
]
)
self.conv_post = norm_f(Conv1d(1024, 1, 3, 1, padding=1))
def forward(self, x):
fmap = []
for l in self.convs:
x = l(x)
x = F.leaky_relu(x, modules.LRELU_SLOPE)
fmap.append(x)
x = self.conv_post(x)
fmap.append(x)
x = torch.flatten(x, 1, -1)
return x, fmap
class DiscriminatorP(torch.nn.Module):
def __init__(self, period, kernel_size=5, stride=3, use_spectral_norm=False):
super(DiscriminatorP, self).__init__()
self.period = period
self.use_spectral_norm = use_spectral_norm
norm_f = weight_norm if use_spectral_norm == False else spectral_norm
self.convs = nn.ModuleList(
[
norm_f(
Conv2d(
1,
32,
(kernel_size, 1),
(stride, 1),
padding=(get_padding(kernel_size, 1), 0),
)
),
norm_f(
Conv2d(
32,
128,
(kernel_size, 1),
(stride, 1),
padding=(get_padding(kernel_size, 1), 0),
)
),
norm_f(
Conv2d(
128,
512,
(kernel_size, 1),
(stride, 1),
padding=(get_padding(kernel_size, 1), 0),
)
),
norm_f(
Conv2d(
512,
1024,
(kernel_size, 1),
(stride, 1),
padding=(get_padding(kernel_size, 1), 0),
)
),
norm_f(
Conv2d(
1024,
1024,
(kernel_size, 1),
1,
padding=(get_padding(kernel_size, 1), 0),
)
),
]
)
self.conv_post = norm_f(Conv2d(1024, 1, (3, 1), 1, padding=(1, 0)))
def forward(self, x):
fmap = []
# 1d to 2d
b, c, t = x.shape
if t % self.period != 0: # pad first
n_pad = self.period - (t % self.period)
x = F.pad(x, (0, n_pad), "reflect")
t = t + n_pad
x = x.view(b, c, t // self.period, self.period)
for l in self.convs:
x = l(x)
x = F.leaky_relu(x, modules.LRELU_SLOPE)
fmap.append(x)
x = self.conv_post(x)
fmap.append(x)
x = torch.flatten(x, 1, -1)
return x, fmap
+522
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@@ -0,0 +1,522 @@
import copy
import math
import numpy as np
import scipy
import torch
from torch import nn
from torch.nn import functional as F
from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d
from torch.nn.utils import weight_norm, remove_weight_norm
from lib.infer_pack import commons
from lib.infer_pack.commons import init_weights, get_padding
from lib.infer_pack.transforms import piecewise_rational_quadratic_transform
LRELU_SLOPE = 0.1
class LayerNorm(nn.Module):
def __init__(self, channels, eps=1e-5):
super().__init__()
self.channels = channels
self.eps = eps
self.gamma = nn.Parameter(torch.ones(channels))
self.beta = nn.Parameter(torch.zeros(channels))
def forward(self, x):
x = x.transpose(1, -1)
x = F.layer_norm(x, (self.channels,), self.gamma, self.beta, self.eps)
return x.transpose(1, -1)
class ConvReluNorm(nn.Module):
def __init__(
self,
in_channels,
hidden_channels,
out_channels,
kernel_size,
n_layers,
p_dropout,
):
super().__init__()
self.in_channels = in_channels
self.hidden_channels = hidden_channels
self.out_channels = out_channels
self.kernel_size = kernel_size
self.n_layers = n_layers
self.p_dropout = p_dropout
assert n_layers > 1, "Number of layers should be larger than 0."
self.conv_layers = nn.ModuleList()
self.norm_layers = nn.ModuleList()
self.conv_layers.append(
nn.Conv1d(
in_channels, hidden_channels, kernel_size, padding=kernel_size // 2
)
)
self.norm_layers.append(LayerNorm(hidden_channels))
self.relu_drop = nn.Sequential(nn.ReLU(), nn.Dropout(p_dropout))
for _ in range(n_layers - 1):
self.conv_layers.append(
nn.Conv1d(
hidden_channels,
hidden_channels,
kernel_size,
padding=kernel_size // 2,
)
)
self.norm_layers.append(LayerNorm(hidden_channels))
self.proj = nn.Conv1d(hidden_channels, out_channels, 1)
self.proj.weight.data.zero_()
self.proj.bias.data.zero_()
def forward(self, x, x_mask):
x_org = x
for i in range(self.n_layers):
x = self.conv_layers[i](x * x_mask)
x = self.norm_layers[i](x)
x = self.relu_drop(x)
x = x_org + self.proj(x)
return x * x_mask
class DDSConv(nn.Module):
"""
Dialted and Depth-Separable Convolution
"""
def __init__(self, channels, kernel_size, n_layers, p_dropout=0.0):
super().__init__()
self.channels = channels
self.kernel_size = kernel_size
self.n_layers = n_layers
self.p_dropout = p_dropout
self.drop = nn.Dropout(p_dropout)
self.convs_sep = nn.ModuleList()
self.convs_1x1 = nn.ModuleList()
self.norms_1 = nn.ModuleList()
self.norms_2 = nn.ModuleList()
for i in range(n_layers):
dilation = kernel_size**i
padding = (kernel_size * dilation - dilation) // 2
self.convs_sep.append(
nn.Conv1d(
channels,
channels,
kernel_size,
groups=channels,
dilation=dilation,
padding=padding,
)
)
self.convs_1x1.append(nn.Conv1d(channels, channels, 1))
self.norms_1.append(LayerNorm(channels))
self.norms_2.append(LayerNorm(channels))
def forward(self, x, x_mask, g=None):
if g is not None:
x = x + g
for i in range(self.n_layers):
y = self.convs_sep[i](x * x_mask)
y = self.norms_1[i](y)
y = F.gelu(y)
y = self.convs_1x1[i](y)
y = self.norms_2[i](y)
y = F.gelu(y)
y = self.drop(y)
x = x + y
return x * x_mask
class WN(torch.nn.Module):
def __init__(
self,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
gin_channels=0,
p_dropout=0,
):
super(WN, self).__init__()
assert kernel_size % 2 == 1
self.hidden_channels = hidden_channels
self.kernel_size = (kernel_size,)
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.gin_channels = gin_channels
self.p_dropout = p_dropout
self.in_layers = torch.nn.ModuleList()
self.res_skip_layers = torch.nn.ModuleList()
self.drop = nn.Dropout(p_dropout)
if gin_channels != 0:
cond_layer = torch.nn.Conv1d(
gin_channels, 2 * hidden_channels * n_layers, 1
)
self.cond_layer = torch.nn.utils.weight_norm(cond_layer, name="weight")
for i in range(n_layers):
dilation = dilation_rate**i
padding = int((kernel_size * dilation - dilation) / 2)
in_layer = torch.nn.Conv1d(
hidden_channels,
2 * hidden_channels,
kernel_size,
dilation=dilation,
padding=padding,
)
in_layer = torch.nn.utils.weight_norm(in_layer, name="weight")
self.in_layers.append(in_layer)
# last one is not necessary
if i < n_layers - 1:
res_skip_channels = 2 * hidden_channels
else:
res_skip_channels = hidden_channels
res_skip_layer = torch.nn.Conv1d(hidden_channels, res_skip_channels, 1)
res_skip_layer = torch.nn.utils.weight_norm(res_skip_layer, name="weight")
self.res_skip_layers.append(res_skip_layer)
def forward(self, x, x_mask, g=None, **kwargs):
output = torch.zeros_like(x)
n_channels_tensor = torch.IntTensor([self.hidden_channels])
if g is not None:
g = self.cond_layer(g)
for i in range(self.n_layers):
x_in = self.in_layers[i](x)
if g is not None:
cond_offset = i * 2 * self.hidden_channels
g_l = g[:, cond_offset : cond_offset + 2 * self.hidden_channels, :]
else:
g_l = torch.zeros_like(x_in)
acts = commons.fused_add_tanh_sigmoid_multiply(x_in, g_l, n_channels_tensor)
acts = self.drop(acts)
res_skip_acts = self.res_skip_layers[i](acts)
if i < self.n_layers - 1:
res_acts = res_skip_acts[:, : self.hidden_channels, :]
x = (x + res_acts) * x_mask
output = output + res_skip_acts[:, self.hidden_channels :, :]
else:
output = output + res_skip_acts
return output * x_mask
def remove_weight_norm(self):
if self.gin_channels != 0:
torch.nn.utils.remove_weight_norm(self.cond_layer)
for l in self.in_layers:
torch.nn.utils.remove_weight_norm(l)
for l in self.res_skip_layers:
torch.nn.utils.remove_weight_norm(l)
class ResBlock1(torch.nn.Module):
def __init__(self, channels, kernel_size=3, dilation=(1, 3, 5)):
super(ResBlock1, self).__init__()
self.convs1 = nn.ModuleList(
[
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
1,
dilation=dilation[0],
padding=get_padding(kernel_size, dilation[0]),
)
),
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
1,
dilation=dilation[1],
padding=get_padding(kernel_size, dilation[1]),
)
),
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
1,
dilation=dilation[2],
padding=get_padding(kernel_size, dilation[2]),
)
),
]
)
self.convs1.apply(init_weights)
self.convs2 = nn.ModuleList(
[
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
1,
dilation=1,
padding=get_padding(kernel_size, 1),
)
),
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
1,
dilation=1,
padding=get_padding(kernel_size, 1),
)
),
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
1,
dilation=1,
padding=get_padding(kernel_size, 1),
)
),
]
)
self.convs2.apply(init_weights)
def forward(self, x, x_mask=None):
for c1, c2 in zip(self.convs1, self.convs2):
xt = F.leaky_relu(x, LRELU_SLOPE)
if x_mask is not None:
xt = xt * x_mask
xt = c1(xt)
xt = F.leaky_relu(xt, LRELU_SLOPE)
if x_mask is not None:
xt = xt * x_mask
xt = c2(xt)
x = xt + x
if x_mask is not None:
x = x * x_mask
return x
def remove_weight_norm(self):
for l in self.convs1:
remove_weight_norm(l)
for l in self.convs2:
remove_weight_norm(l)
class ResBlock2(torch.nn.Module):
def __init__(self, channels, kernel_size=3, dilation=(1, 3)):
super(ResBlock2, self).__init__()
self.convs = nn.ModuleList(
[
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
1,
dilation=dilation[0],
padding=get_padding(kernel_size, dilation[0]),
)
),
weight_norm(
Conv1d(
channels,
channels,
kernel_size,
1,
dilation=dilation[1],
padding=get_padding(kernel_size, dilation[1]),
)
),
]
)
self.convs.apply(init_weights)
def forward(self, x, x_mask=None):
for c in self.convs:
xt = F.leaky_relu(x, LRELU_SLOPE)
if x_mask is not None:
xt = xt * x_mask
xt = c(xt)
x = xt + x
if x_mask is not None:
x = x * x_mask
return x
def remove_weight_norm(self):
for l in self.convs:
remove_weight_norm(l)
class Log(nn.Module):
def forward(self, x, x_mask, reverse=False, **kwargs):
if not reverse:
y = torch.log(torch.clamp_min(x, 1e-5)) * x_mask
logdet = torch.sum(-y, [1, 2])
return y, logdet
else:
x = torch.exp(x) * x_mask
return x
class Flip(nn.Module):
def forward(self, x, *args, reverse=False, **kwargs):
x = torch.flip(x, [1])
if not reverse:
logdet = torch.zeros(x.size(0)).to(dtype=x.dtype, device=x.device)
return x, logdet
else:
return x
class ElementwiseAffine(nn.Module):
def __init__(self, channels):
super().__init__()
self.channels = channels
self.m = nn.Parameter(torch.zeros(channels, 1))
self.logs = nn.Parameter(torch.zeros(channels, 1))
def forward(self, x, x_mask, reverse=False, **kwargs):
if not reverse:
y = self.m + torch.exp(self.logs) * x
y = y * x_mask
logdet = torch.sum(self.logs * x_mask, [1, 2])
return y, logdet
else:
x = (x - self.m) * torch.exp(-self.logs) * x_mask
return x
class ResidualCouplingLayer(nn.Module):
def __init__(
self,
channels,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
p_dropout=0,
gin_channels=0,
mean_only=False,
):
assert channels % 2 == 0, "channels should be divisible by 2"
super().__init__()
self.channels = channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.half_channels = channels // 2
self.mean_only = mean_only
self.pre = nn.Conv1d(self.half_channels, hidden_channels, 1)
self.enc = WN(
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
p_dropout=p_dropout,
gin_channels=gin_channels,
)
self.post = nn.Conv1d(hidden_channels, self.half_channels * (2 - mean_only), 1)
self.post.weight.data.zero_()
self.post.bias.data.zero_()
def forward(self, x, x_mask, g=None, reverse=False):
x0, x1 = torch.split(x, [self.half_channels] * 2, 1)
h = self.pre(x0) * x_mask
h = self.enc(h, x_mask, g=g)
stats = self.post(h) * x_mask
if not self.mean_only:
m, logs = torch.split(stats, [self.half_channels] * 2, 1)
else:
m = stats
logs = torch.zeros_like(m)
if not reverse:
x1 = m + x1 * torch.exp(logs) * x_mask
x = torch.cat([x0, x1], 1)
logdet = torch.sum(logs, [1, 2])
return x, logdet
else:
x1 = (x1 - m) * torch.exp(-logs) * x_mask
x = torch.cat([x0, x1], 1)
return x
def remove_weight_norm(self):
self.enc.remove_weight_norm()
class ConvFlow(nn.Module):
def __init__(
self,
in_channels,
filter_channels,
kernel_size,
n_layers,
num_bins=10,
tail_bound=5.0,
):
super().__init__()
self.in_channels = in_channels
self.filter_channels = filter_channels
self.kernel_size = kernel_size
self.n_layers = n_layers
self.num_bins = num_bins
self.tail_bound = tail_bound
self.half_channels = in_channels // 2
self.pre = nn.Conv1d(self.half_channels, filter_channels, 1)
self.convs = DDSConv(filter_channels, kernel_size, n_layers, p_dropout=0.0)
self.proj = nn.Conv1d(
filter_channels, self.half_channels * (num_bins * 3 - 1), 1
)
self.proj.weight.data.zero_()
self.proj.bias.data.zero_()
def forward(self, x, x_mask, g=None, reverse=False):
x0, x1 = torch.split(x, [self.half_channels] * 2, 1)
h = self.pre(x0)
h = self.convs(h, x_mask, g=g)
h = self.proj(h) * x_mask
b, c, t = x0.shape
h = h.reshape(b, c, -1, t).permute(0, 1, 3, 2) # [b, cx?, t] -> [b, c, t, ?]
unnormalized_widths = h[..., : self.num_bins] / math.sqrt(self.filter_channels)
unnormalized_heights = h[..., self.num_bins : 2 * self.num_bins] / math.sqrt(
self.filter_channels
)
unnormalized_derivatives = h[..., 2 * self.num_bins :]
x1, logabsdet = piecewise_rational_quadratic_transform(
x1,
unnormalized_widths,
unnormalized_heights,
unnormalized_derivatives,
inverse=reverse,
tails="linear",
tail_bound=self.tail_bound,
)
x = torch.cat([x0, x1], 1) * x_mask
logdet = torch.sum(logabsdet * x_mask, [1, 2])
if not reverse:
return x, logdet
else:
return x
@@ -0,0 +1,90 @@
from lib.infer_pack.modules.F0Predictor.F0Predictor import F0Predictor
import pyworld
import numpy as np
class DioF0Predictor(F0Predictor):
def __init__(self, hop_length=512, f0_min=50, f0_max=1100, sampling_rate=44100):
self.hop_length = hop_length
self.f0_min = f0_min
self.f0_max = f0_max
self.sampling_rate = sampling_rate
def interpolate_f0(self, f0):
"""
对F0进行插值处理
"""
data = np.reshape(f0, (f0.size, 1))
vuv_vector = np.zeros((data.size, 1), dtype=np.float32)
vuv_vector[data > 0.0] = 1.0
vuv_vector[data <= 0.0] = 0.0
ip_data = data
frame_number = data.size
last_value = 0.0
for i in range(frame_number):
if data[i] <= 0.0:
j = i + 1
for j in range(i + 1, frame_number):
if data[j] > 0.0:
break
if j < frame_number - 1:
if last_value > 0.0:
step = (data[j] - data[i - 1]) / float(j - i)
for k in range(i, j):
ip_data[k] = data[i - 1] + step * (k - i + 1)
else:
for k in range(i, j):
ip_data[k] = data[j]
else:
for k in range(i, frame_number):
ip_data[k] = last_value
else:
ip_data[i] = data[i] # 这里可能存在一个没有必要的拷贝
last_value = data[i]
return ip_data[:, 0], vuv_vector[:, 0]
def resize_f0(self, x, target_len):
source = np.array(x)
source[source < 0.001] = np.nan
target = np.interp(
np.arange(0, len(source) * target_len, len(source)) / target_len,
np.arange(0, len(source)),
source,
)
res = np.nan_to_num(target)
return res
def compute_f0(self, wav, p_len=None):
if p_len is None:
p_len = wav.shape[0] // self.hop_length
f0, t = pyworld.dio(
wav.astype(np.double),
fs=self.sampling_rate,
f0_floor=self.f0_min,
f0_ceil=self.f0_max,
frame_period=1000 * self.hop_length / self.sampling_rate,
)
f0 = pyworld.stonemask(wav.astype(np.double), f0, t, self.sampling_rate)
for index, pitch in enumerate(f0):
f0[index] = round(pitch, 1)
return self.interpolate_f0(self.resize_f0(f0, p_len))[0]
def compute_f0_uv(self, wav, p_len=None):
if p_len is None:
p_len = wav.shape[0] // self.hop_length
f0, t = pyworld.dio(
wav.astype(np.double),
fs=self.sampling_rate,
f0_floor=self.f0_min,
f0_ceil=self.f0_max,
frame_period=1000 * self.hop_length / self.sampling_rate,
)
f0 = pyworld.stonemask(wav.astype(np.double), f0, t, self.sampling_rate)
for index, pitch in enumerate(f0):
f0[index] = round(pitch, 1)
return self.interpolate_f0(self.resize_f0(f0, p_len))
@@ -0,0 +1,16 @@
class F0Predictor(object):
def compute_f0(self, wav, p_len):
"""
input: wav:[signal_length]
p_len:int
output: f0:[signal_length//hop_length]
"""
pass
def compute_f0_uv(self, wav, p_len):
"""
input: wav:[signal_length]
p_len:int
output: f0:[signal_length//hop_length],uv:[signal_length//hop_length]
"""
pass
@@ -0,0 +1,87 @@
from lib.infer_pack.modules.F0Predictor.F0Predictor import F0Predictor
import pyworld
import numpy as np
class HarvestF0Predictor(F0Predictor):
def __init__(self, hop_length=512, f0_min=50, f0_max=1100, sampling_rate=44100):
self.hop_length = hop_length
self.f0_min = f0_min
self.f0_max = f0_max
self.sampling_rate = sampling_rate
self.fs = sampling_rate
def interpolate_f0(self, f0):
"""
对F0进行插值处理
"""
data = np.reshape(f0, (f0.size, 1))
vuv_vector = np.zeros((data.size, 1), dtype=np.float32)
vuv_vector[data > 0.0] = 1.0
vuv_vector[data <= 0.0] = 0.0
ip_data = data
frame_number = data.size
last_value = 0.0
for i in range(frame_number):
if data[i] <= 0.0:
j = i + 1
for j in range(i + 1, frame_number):
if data[j] > 0.0:
break
if j < frame_number - 1:
if last_value > 0.0:
step = (data[j] - data[i - 1]) / float(j - i)
for k in range(i, j):
ip_data[k] = data[i - 1] + step * (k - i + 1)
else:
for k in range(i, j):
ip_data[k] = data[j]
else:
for k in range(i, frame_number):
ip_data[k] = last_value
else:
ip_data[i] = data[i] # 这里可能存在一个没有必要的拷贝
last_value = data[i]
return ip_data[:, 0], vuv_vector[:, 0]
def resize_f0(self, x, target_len):
source = np.array(x)
source[source < 0.001] = np.nan
target = np.interp(
np.arange(0, len(source) * target_len, len(source)) / target_len,
np.arange(0, len(source)),
source,
)
res = np.nan_to_num(target)
return res
def compute_f0(self, wav, p_len=None):
if p_len is None:
p_len = wav.shape[0] // self.hop_length
f0, t = pyworld.harvest(
wav.astype(np.double),
fs=self.sampling_rate,
f0_ceil=self.f0_max,
f0_floor=self.f0_min,
frame_period=1000 * self.hop_length / self.sampling_rate,
)
f0 = pyworld.stonemask(wav.astype(np.double), f0, t, self.sampling_rate)
return self.interpolate_f0(self.resize_f0(f0, p_len))[0]
def compute_f0_uv(self, wav, p_len=None):
if p_len is None:
p_len = wav.shape[0] // self.hop_length
f0, t = pyworld.harvest(
wav.astype(np.double),
fs=self.sampling_rate,
f0_floor=self.f0_min,
f0_ceil=self.f0_max,
frame_period=1000 * self.hop_length / self.sampling_rate,
)
f0 = pyworld.stonemask(wav.astype(np.double), f0, t, self.sampling_rate)
return self.interpolate_f0(self.resize_f0(f0, p_len))
@@ -0,0 +1,102 @@
from lib.infer_pack.modules.F0Predictor.F0Predictor import F0Predictor
import parselmouth
import numpy as np
class PMF0Predictor(F0Predictor):
def __init__(self, hop_length=512, f0_min=50, f0_max=1100, sampling_rate=44100):
self.hop_length = hop_length
self.f0_min = f0_min
self.f0_max = f0_max
self.sampling_rate = sampling_rate
def interpolate_f0(self, f0):
"""
对F0进行插值处理
"""
data = np.reshape(f0, (f0.size, 1))
vuv_vector = np.zeros((data.size, 1), dtype=np.float32)
vuv_vector[data > 0.0] = 1.0
vuv_vector[data <= 0.0] = 0.0
ip_data = data
frame_number = data.size
last_value = 0.0
for i in range(frame_number):
if data[i] <= 0.0:
j = i + 1
for j in range(i + 1, frame_number):
if data[j] > 0.0:
break
if j < frame_number - 1:
if last_value > 0.0:
step = (data[j] - data[i - 1]) / float(j - i)
for k in range(i, j):
ip_data[k] = data[i - 1] + step * (k - i + 1)
else:
for k in range(i, j):
ip_data[k] = data[j]
else:
for k in range(i, frame_number):
ip_data[k] = last_value
else:
ip_data[i] = data[i] # 这里可能存在一个没有必要的拷贝
last_value = data[i]
return ip_data[:, 0], vuv_vector[:, 0]
def resize_f0(self, x, target_len):
source = np.array(x)
source[source < 0.001] = np.nan
target = np.interp(
np.arange(0, len(source) * target_len, len(source)) / target_len,
np.arange(0, len(source)),
source,
)
res = np.nan_to_num(target)
return res
def compute_f0(self, wav, p_len=None):
x = wav
if p_len is None:
p_len = x.shape[0] // self.hop_length
time_step = self.hop_length / self.sampling_rate * 1000
f0 = (
parselmouth.Sound(x, self.sampling_rate)
.to_pitch_ac(
time_step=time_step / 1000,
voicing_threshold=0.6,
pitch_floor=self.f0_min,
pitch_ceiling=self.f0_max,
)
.selected_array["frequency"]
)
f0_resized = self.resize_f0(f0, p_len)
f0_interpolated, uv = self.interpolate_f0(f0_resized)
return f0_interpolated
def compute_f0_uv(self, wav, p_len=None):
x = wav
if p_len is None:
p_len = x.shape[0] // self.hop_length
time_step = self.hop_length / self.sampling_rate * 1000
f0 = (
parselmouth.Sound(x, self.sampling_rate)
.to_pitch_ac(
time_step=time_step / 1000,
voicing_threshold=0.6,
pitch_floor=self.f0_min,
pitch_ceiling=self.f0_max,
)
.selected_array["frequency"]
)
f0_resized = self.resize_f0(f0, p_len)
f0_interpolated, uv = self.interpolate_f0(f0_resized)
return f0_interpolated, uv
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import onnxruntime
import librosa
import numpy as np
import soundfile
def load_audio_fast(path, target_sr):
# 1. Coba torchaudio (sangat cepat, ~11ms)
try:
import torchaudio
wav, sr = torchaudio.load(path)
if sr != target_sr:
import torchaudio.transforms as T
resampler = T.Resample(sr, target_sr)
wav = resampler(wav)
if wav.shape[0] > 1:
wav = wav.mean(dim=0)
wav = wav.squeeze().numpy()
return wav, target_sr
except Exception:
pass
# 2. Coba pydub (cepat, ~80ms)
try:
from pydub import AudioSegment
audio_seg = AudioSegment.from_file(path)
audio_seg = audio_seg.set_frame_rate(target_sr).set_channels(1)
wav = np.array(audio_seg.get_array_of_samples(), dtype=np.float32) / 32768.0
return wav, target_sr
except Exception:
pass
# 3. Fallback ke librosa asli
return librosa.load(path, sr=target_sr, mono=True)
class ContentVec:
def __init__(self, vec_path="pretrained/vec-768-layer-12.onnx", device=None):
print("load model(s) from {}".format(vec_path))
import onnxruntime as ort
opts = ort.SessionOptions()
opts.graph_optimization_level = ort.GraphOptimizationLevel.ORT_ENABLE_ALL
opts.intra_op_num_threads = 2
if device == "cpu" or device is None:
providers = ["CPUExecutionProvider"]
elif device == "cuda":
providers = [
("CUDAExecutionProvider", {
"device_id": 0,
"arena_extend_strategy": "kNextPowerOfTwo",
"cudnn_conv_algo_search": "EXHAUSTIVE",
"do_copy_in_default_stream": True,
}),
"CPUExecutionProvider"
]
elif device == "dml":
providers = ["DmlExecutionProvider"]
else:
raise RuntimeError("Unsportted Device")
self.model = ort.InferenceSession(vec_path, sess_options=opts, providers=providers)
def __call__(self, wav):
return self.forward(wav)
def forward(self, wav):
feats = wav
if feats.ndim == 2: # double channels
feats = feats.mean(-1)
assert feats.ndim == 1, feats.ndim
feats = np.expand_dims(np.expand_dims(feats, 0), 0)
onnx_input = {self.model.get_inputs()[0].name: feats}
logits = self.model.run(None, onnx_input)[0]
return logits.transpose(0, 2, 1)
class RMVPEF0Predictor:
def __init__(self, model_path="rmvpe.pt", is_half=False, device="cpu", sampling_rate=40000):
import torch
from rmvpe import RMVPE
self.model = RMVPE(model_path, is_half=is_half, device=device)
self.sampling_rate = sampling_rate
def interpolate_f0(self, f0):
data = np.reshape(f0, (f0.size, 1))
vuv_vector = np.zeros((data.size, 1), dtype=np.float32)
vuv_vector[data > 0.0] = 1.0
vuv_vector[data <= 0.0] = 0.0
ip_data = data
frame_number = data.size
last_value = 0.0
for i in range(frame_number):
if data[i] <= 0.0:
j = i + 1
for j in range(i + 1, frame_number):
if data[j] > 0.0:
break
if j < frame_number - 1:
if last_value > 0.0:
step = (data[j] - data[i - 1]) / float(j - i)
for k in range(i, j):
ip_data[k] = data[i - 1] + step * (k - i + 1)
else:
for k in range(i, j):
ip_data[k] = data[j]
else:
for k in range(i, frame_number):
ip_data[k] = last_value
else:
ip_data[i] = data[i]
last_value = data[i]
return ip_data[:, 0], vuv_vector[:, 0]
def compute_f0(self, wav16k, p_len):
# Input 'wav16k' sudah pasti berada di 16000 Hz karena di-bypass pada tingkat atas
f0 = self.model.infer_from_audio(wav16k, thred=0.03)
# Resize f0 to match p_len perfectly using np.interp (sama dengan resize_f0 di Dio)
source = np.array(f0)
source[source < 0.001] = np.nan
target = np.interp(
np.arange(0, len(source) * p_len, len(source)) / p_len,
np.arange(0, len(source)),
source,
)
res = np.nan_to_num(target)
# Lakukan interpolasi agar kontinu (menghindari suara robotik & glitch pitch)
return self.interpolate_f0(res)[0]
def get_f0_predictor(f0_predictor, hop_length, sampling_rate, **kargs):
device = kargs.get("device", "cpu")
if f0_predictor == "pm":
from lib.infer_pack.modules.F0Predictor.PMF0Predictor import PMF0Predictor
f0_predictor_object = PMF0Predictor(
hop_length=hop_length, sampling_rate=sampling_rate
)
elif f0_predictor == "harvest":
from lib.infer_pack.modules.F0Predictor.HarvestF0Predictor import (
HarvestF0Predictor,
)
f0_predictor_object = HarvestF0Predictor(
hop_length=hop_length, sampling_rate=sampling_rate
)
elif f0_predictor == "dio":
from lib.infer_pack.modules.F0Predictor.DioF0Predictor import DioF0Predictor
f0_predictor_object = DioF0Predictor(
hop_length=hop_length, sampling_rate=sampling_rate
)
elif f0_predictor == "rmvpe":
is_half = kargs.get("is_half", False)
f0_predictor_object = RMVPEF0Predictor(
model_path="rmvpe.pt",
is_half=is_half,
device=device,
sampling_rate=sampling_rate
)
else:
raise Exception("Unknown f0 predictor")
return f0_predictor_object
class OnnxRVC:
def __init__(
self,
model_path,
sr=40000,
hop_size=512,
vec_path="vec-768-layer-12",
device="cpu",
):
vec_path = f"pretrained/{vec_path}.onnx"
self.vec_model = ContentVec(vec_path, device)
import onnxruntime as ort
opts = ort.SessionOptions()
opts.graph_optimization_level = ort.GraphOptimizationLevel.ORT_ENABLE_ALL
opts.intra_op_num_threads = 2
if device == "cpu" or device is None:
providers = ["CPUExecutionProvider"]
elif device == "cuda":
providers = [
("CUDAExecutionProvider", {
"device_id": 0,
"arena_extend_strategy": "kNextPowerOfTwo",
"cudnn_conv_algo_search": "EXHAUSTIVE",
"do_copy_in_default_stream": True,
}),
"CPUExecutionProvider"
]
elif device == "dml":
providers = ["DmlExecutionProvider"]
else:
raise RuntimeError("Unsportted Device")
self.model = ort.InferenceSession(model_path, sess_options=opts, providers=providers)
self.sampling_rate = sr
self.hop_size = hop_size
self.device = device
def forward(self, hubert, hubert_length, pitch, pitchf, ds, rnd):
onnx_input = {
self.model.get_inputs()[0].name: hubert,
self.model.get_inputs()[1].name: hubert_length,
self.model.get_inputs()[2].name: pitch,
self.model.get_inputs()[3].name: pitchf,
self.model.get_inputs()[4].name: ds,
self.model.get_inputs()[5].name: rnd,
}
return (self.model.run(None, onnx_input)[0] * 32767).astype(np.int16)
def inference(
self,
raw_path,
sid,
f0_method="dio",
f0_up_key=0,
pad_time=0.5,
cr_threshold=0.02,
rmvpe_fp16=False,
):
f0_min = 50
f0_max = 1100
f0_mel_min = 1127 * np.log(1 + f0_min / 700)
f0_mel_max = 1127 * np.log(1 + f0_max / 700)
f0_predictor = get_f0_predictor(
f0_method,
hop_length=self.hop_size,
sampling_rate=self.sampling_rate,
threshold=cr_threshold,
device=self.device,
is_half=rmvpe_fp16,
)
if f0_method == "rmvpe":
wav16k, sr = load_audio_fast(raw_path, 16000)
org_length = int(len(wav16k) * (self.sampling_rate / 16000))
if len(wav16k) / 16000 > 50.0:
raise RuntimeError("Reached Max Length")
else:
wav, sr = load_audio_fast(raw_path, self.sampling_rate)
org_length = len(wav)
if org_length / sr > 50.0:
raise RuntimeError("Reached Max Length")
wav16k, _ = load_audio_fast(raw_path, 16000)
hubert = self.vec_model(wav16k)
hubert = np.repeat(hubert, 2, axis=2).transpose(0, 2, 1).astype(np.float32)
hubert_length = hubert.shape[1]
if f0_method == "rmvpe":
pitchf = f0_predictor.compute_f0(wav16k, hubert_length)
else:
pitchf = f0_predictor.compute_f0(wav, hubert_length)
pitchf = pitchf * 2 ** (f0_up_key / 12)
pitch = pitchf.copy()
f0_mel = 1127 * np.log(1 + pitch / 700)
f0_mel[f0_mel > 0] = (f0_mel[f0_mel > 0] - f0_mel_min) * 254 / (
f0_mel_max - f0_mel_min
) + 1
f0_mel[f0_mel <= 1] = 1
f0_mel[f0_mel > 255] = 255
pitch = np.rint(f0_mel).astype(np.int64)
pitchf = pitchf.reshape(1, len(pitchf)).astype(np.float32)
pitch = pitch.reshape(1, len(pitch))
ds = np.array([sid]).astype(np.int64)
rnd = np.random.randn(1, 192, hubert_length).astype(np.float32)
hubert_length = np.array([hubert_length]).astype(np.int64)
out_wav = self.forward(hubert, hubert_length, pitch, pitchf, ds, rnd).squeeze()
out_wav = np.pad(out_wav, (0, 2 * self.hop_size), "constant")
return out_wav[0:org_length]
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import torch
from torch.nn import functional as F
import numpy as np
DEFAULT_MIN_BIN_WIDTH = 1e-3
DEFAULT_MIN_BIN_HEIGHT = 1e-3
DEFAULT_MIN_DERIVATIVE = 1e-3
def piecewise_rational_quadratic_transform(
inputs,
unnormalized_widths,
unnormalized_heights,
unnormalized_derivatives,
inverse=False,
tails=None,
tail_bound=1.0,
min_bin_width=DEFAULT_MIN_BIN_WIDTH,
min_bin_height=DEFAULT_MIN_BIN_HEIGHT,
min_derivative=DEFAULT_MIN_DERIVATIVE,
):
if tails is None:
spline_fn = rational_quadratic_spline
spline_kwargs = {}
else:
spline_fn = unconstrained_rational_quadratic_spline
spline_kwargs = {"tails": tails, "tail_bound": tail_bound}
outputs, logabsdet = spline_fn(
inputs=inputs,
unnormalized_widths=unnormalized_widths,
unnormalized_heights=unnormalized_heights,
unnormalized_derivatives=unnormalized_derivatives,
inverse=inverse,
min_bin_width=min_bin_width,
min_bin_height=min_bin_height,
min_derivative=min_derivative,
**spline_kwargs
)
return outputs, logabsdet
def searchsorted(bin_locations, inputs, eps=1e-6):
bin_locations[..., -1] += eps
return torch.sum(inputs[..., None] >= bin_locations, dim=-1) - 1
def unconstrained_rational_quadratic_spline(
inputs,
unnormalized_widths,
unnormalized_heights,
unnormalized_derivatives,
inverse=False,
tails="linear",
tail_bound=1.0,
min_bin_width=DEFAULT_MIN_BIN_WIDTH,
min_bin_height=DEFAULT_MIN_BIN_HEIGHT,
min_derivative=DEFAULT_MIN_DERIVATIVE,
):
inside_interval_mask = (inputs >= -tail_bound) & (inputs <= tail_bound)
outside_interval_mask = ~inside_interval_mask
outputs = torch.zeros_like(inputs)
logabsdet = torch.zeros_like(inputs)
if tails == "linear":
unnormalized_derivatives = F.pad(unnormalized_derivatives, pad=(1, 1))
constant = np.log(np.exp(1 - min_derivative) - 1)
unnormalized_derivatives[..., 0] = constant
unnormalized_derivatives[..., -1] = constant
outputs[outside_interval_mask] = inputs[outside_interval_mask]
logabsdet[outside_interval_mask] = 0
else:
raise RuntimeError("{} tails are not implemented.".format(tails))
(
outputs[inside_interval_mask],
logabsdet[inside_interval_mask],
) = rational_quadratic_spline(
inputs=inputs[inside_interval_mask],
unnormalized_widths=unnormalized_widths[inside_interval_mask, :],
unnormalized_heights=unnormalized_heights[inside_interval_mask, :],
unnormalized_derivatives=unnormalized_derivatives[inside_interval_mask, :],
inverse=inverse,
left=-tail_bound,
right=tail_bound,
bottom=-tail_bound,
top=tail_bound,
min_bin_width=min_bin_width,
min_bin_height=min_bin_height,
min_derivative=min_derivative,
)
return outputs, logabsdet
def rational_quadratic_spline(
inputs,
unnormalized_widths,
unnormalized_heights,
unnormalized_derivatives,
inverse=False,
left=0.0,
right=1.0,
bottom=0.0,
top=1.0,
min_bin_width=DEFAULT_MIN_BIN_WIDTH,
min_bin_height=DEFAULT_MIN_BIN_HEIGHT,
min_derivative=DEFAULT_MIN_DERIVATIVE,
):
if torch.min(inputs) < left or torch.max(inputs) > right:
raise ValueError("Input to a transform is not within its domain")
num_bins = unnormalized_widths.shape[-1]
if min_bin_width * num_bins > 1.0:
raise ValueError("Minimal bin width too large for the number of bins")
if min_bin_height * num_bins > 1.0:
raise ValueError("Minimal bin height too large for the number of bins")
widths = F.softmax(unnormalized_widths, dim=-1)
widths = min_bin_width + (1 - min_bin_width * num_bins) * widths
cumwidths = torch.cumsum(widths, dim=-1)
cumwidths = F.pad(cumwidths, pad=(1, 0), mode="constant", value=0.0)
cumwidths = (right - left) * cumwidths + left
cumwidths[..., 0] = left
cumwidths[..., -1] = right
widths = cumwidths[..., 1:] - cumwidths[..., :-1]
derivatives = min_derivative + F.softplus(unnormalized_derivatives)
heights = F.softmax(unnormalized_heights, dim=-1)
heights = min_bin_height + (1 - min_bin_height * num_bins) * heights
cumheights = torch.cumsum(heights, dim=-1)
cumheights = F.pad(cumheights, pad=(1, 0), mode="constant", value=0.0)
cumheights = (top - bottom) * cumheights + bottom
cumheights[..., 0] = bottom
cumheights[..., -1] = top
heights = cumheights[..., 1:] - cumheights[..., :-1]
if inverse:
bin_idx = searchsorted(cumheights, inputs)[..., None]
else:
bin_idx = searchsorted(cumwidths, inputs)[..., None]
input_cumwidths = cumwidths.gather(-1, bin_idx)[..., 0]
input_bin_widths = widths.gather(-1, bin_idx)[..., 0]
input_cumheights = cumheights.gather(-1, bin_idx)[..., 0]
delta = heights / widths
input_delta = delta.gather(-1, bin_idx)[..., 0]
input_derivatives = derivatives.gather(-1, bin_idx)[..., 0]
input_derivatives_plus_one = derivatives[..., 1:].gather(-1, bin_idx)[..., 0]
input_heights = heights.gather(-1, bin_idx)[..., 0]
if inverse:
a = (inputs - input_cumheights) * (
input_derivatives + input_derivatives_plus_one - 2 * input_delta
) + input_heights * (input_delta - input_derivatives)
b = input_heights * input_derivatives - (inputs - input_cumheights) * (
input_derivatives + input_derivatives_plus_one - 2 * input_delta
)
c = -input_delta * (inputs - input_cumheights)
discriminant = b.pow(2) - 4 * a * c
assert (discriminant >= 0).all()
root = (2 * c) / (-b - torch.sqrt(discriminant))
outputs = root * input_bin_widths + input_cumwidths
theta_one_minus_theta = root * (1 - root)
denominator = input_delta + (
(input_derivatives + input_derivatives_plus_one - 2 * input_delta)
* theta_one_minus_theta
)
derivative_numerator = input_delta.pow(2) * (
input_derivatives_plus_one * root.pow(2)
+ 2 * input_delta * theta_one_minus_theta
+ input_derivatives * (1 - root).pow(2)
)
logabsdet = torch.log(derivative_numerator) - 2 * torch.log(denominator)
return outputs, -logabsdet
else:
theta = (inputs - input_cumwidths) / input_bin_widths
theta_one_minus_theta = theta * (1 - theta)
numerator = input_heights * (
input_delta * theta.pow(2) + input_derivatives * theta_one_minus_theta
)
denominator = input_delta + (
(input_derivatives + input_derivatives_plus_one - 2 * input_delta)
* theta_one_minus_theta
)
outputs = input_cumheights + numerator / denominator
derivative_numerator = input_delta.pow(2) * (
input_derivatives_plus_one * theta.pow(2)
+ 2 * input_delta * theta_one_minus_theta
+ input_derivatives * (1 - theta).pow(2)
)
logabsdet = torch.log(derivative_numerator) - 2 * torch.log(denominator)
return outputs, logabsdet
+10
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numpy==1.23.5
onnxruntime-gpu
sounddevice
websockets
librosa==0.9.1
soundfile==0.13.1
pydub==0.25.1
torch
torchaudio
scipy
+445
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import sys, torch, numpy as np, traceback, pdb
import torch.nn as nn
from time import time as ttime
import torch.nn.functional as F
class BiGRU(nn.Module):
def __init__(self, input_features, hidden_features, num_layers):
super(BiGRU, self).__init__()
self.gru = nn.GRU(
input_features,
hidden_features,
num_layers=num_layers,
batch_first=True,
bidirectional=True,
)
def forward(self, x):
return self.gru(x)[0]
class ConvBlockRes(nn.Module):
def __init__(self, in_channels, out_channels, momentum=0.01):
super(ConvBlockRes, self).__init__()
self.conv = nn.Sequential(
nn.Conv2d(
in_channels=in_channels,
out_channels=out_channels,
kernel_size=(3, 3),
stride=(1, 1),
padding=(1, 1),
bias=False,
),
nn.BatchNorm2d(out_channels, momentum=momentum),
nn.ReLU(),
nn.Conv2d(
in_channels=out_channels,
out_channels=out_channels,
kernel_size=(3, 3),
stride=(1, 1),
padding=(1, 1),
bias=False,
),
nn.BatchNorm2d(out_channels, momentum=momentum),
nn.ReLU(),
)
if in_channels != out_channels:
self.shortcut = nn.Conv2d(in_channels, out_channels, (1, 1))
self.is_shortcut = True
else:
self.is_shortcut = False
def forward(self, x):
if self.is_shortcut:
return self.conv(x) + self.shortcut(x)
else:
return self.conv(x) + x
class Encoder(nn.Module):
def __init__(
self,
in_channels,
in_size,
n_encoders,
kernel_size,
n_blocks,
out_channels=16,
momentum=0.01,
):
super(Encoder, self).__init__()
self.n_encoders = n_encoders
self.bn = nn.BatchNorm2d(in_channels, momentum=momentum)
self.layers = nn.ModuleList()
self.latent_channels = []
for i in range(self.n_encoders):
self.layers.append(
ResEncoderBlock(
in_channels, out_channels, kernel_size, n_blocks, momentum=momentum
)
)
self.latent_channels.append([out_channels, in_size])
in_channels = out_channels
out_channels *= 2
in_size //= 2
self.out_size = in_size
self.out_channel = out_channels
def forward(self, x):
concat_tensors = []
x = self.bn(x)
for i in range(self.n_encoders):
_, x = self.layers[i](x)
concat_tensors.append(_)
return x, concat_tensors
class ResEncoderBlock(nn.Module):
def __init__(
self, in_channels, out_channels, kernel_size, n_blocks=1, momentum=0.01
):
super(ResEncoderBlock, self).__init__()
self.n_blocks = n_blocks
self.conv = nn.ModuleList()
self.conv.append(ConvBlockRes(in_channels, out_channels, momentum))
for i in range(n_blocks - 1):
self.conv.append(ConvBlockRes(out_channels, out_channels, momentum))
self.kernel_size = kernel_size
if self.kernel_size is not None:
self.pool = nn.AvgPool2d(kernel_size=kernel_size)
def forward(self, x):
for i in range(self.n_blocks):
x = self.conv[i](x)
if self.kernel_size is not None:
return x, self.pool(x)
else:
return x
class Intermediate(nn.Module): #
def __init__(self, in_channels, out_channels, n_inters, n_blocks, momentum=0.01):
super(Intermediate, self).__init__()
self.n_inters = n_inters
self.layers = nn.ModuleList()
self.layers.append(
ResEncoderBlock(in_channels, out_channels, None, n_blocks, momentum)
)
for i in range(self.n_inters - 1):
self.layers.append(
ResEncoderBlock(out_channels, out_channels, None, n_blocks, momentum)
)
def forward(self, x):
for i in range(self.n_inters):
x = self.layers[i](x)
return x
class ResDecoderBlock(nn.Module):
def __init__(self, in_channels, out_channels, stride, n_blocks=1, momentum=0.01):
super(ResDecoderBlock, self).__init__()
out_padding = (0, 1) if stride == (1, 2) else (1, 1)
self.n_blocks = n_blocks
self.conv1 = nn.Sequential(
nn.ConvTranspose2d(
in_channels=in_channels,
out_channels=out_channels,
kernel_size=(3, 3),
stride=stride,
padding=(1, 1),
output_padding=out_padding,
bias=False,
),
nn.BatchNorm2d(out_channels, momentum=momentum),
nn.ReLU(),
)
self.conv2 = nn.ModuleList()
self.conv2.append(ConvBlockRes(out_channels * 2, out_channels, momentum))
for i in range(n_blocks - 1):
self.conv2.append(ConvBlockRes(out_channels, out_channels, momentum))
def forward(self, x, concat_tensor):
x = self.conv1(x)
x = torch.cat((x, concat_tensor), dim=1)
for i in range(self.n_blocks):
x = self.conv2[i](x)
return x
class Decoder(nn.Module):
def __init__(self, in_channels, n_decoders, stride, n_blocks, momentum=0.01):
super(Decoder, self).__init__()
self.layers = nn.ModuleList()
self.n_decoders = n_decoders
for i in range(self.n_decoders):
out_channels = in_channels // 2
self.layers.append(
ResDecoderBlock(in_channels, out_channels, stride, n_blocks, momentum)
)
in_channels = out_channels
def forward(self, x, concat_tensors):
for i in range(self.n_decoders):
x = self.layers[i](x, concat_tensors[-1 - i])
return x
class DeepUnet(nn.Module):
def __init__(
self,
kernel_size,
n_blocks,
en_de_layers=5,
inter_layers=4,
in_channels=1,
en_out_channels=16,
):
super(DeepUnet, self).__init__()
self.encoder = Encoder(
in_channels, 128, en_de_layers, kernel_size, n_blocks, en_out_channels
)
self.intermediate = Intermediate(
self.encoder.out_channel // 2,
self.encoder.out_channel,
inter_layers,
n_blocks,
)
self.decoder = Decoder(
self.encoder.out_channel, en_de_layers, kernel_size, n_blocks
)
def forward(self, x):
x, concat_tensors = self.encoder(x)
x = self.intermediate(x)
x = self.decoder(x, concat_tensors)
return x
class E2E(nn.Module):
def __init__(
self,
n_blocks,
n_gru,
kernel_size,
en_de_layers=5,
inter_layers=4,
in_channels=1,
en_out_channels=16,
):
super(E2E, self).__init__()
self.unet = DeepUnet(
kernel_size,
n_blocks,
en_de_layers,
inter_layers,
in_channels,
en_out_channels,
)
self.cnn = nn.Conv2d(en_out_channels, 3, (3, 3), padding=(1, 1))
if n_gru:
self.fc = nn.Sequential(
BiGRU(3 * 128, 256, n_gru),
nn.Linear(512, 360),
nn.Dropout(0.25),
nn.Sigmoid(),
)
else:
self.fc = nn.Sequential(
nn.Linear(3 * N_MELS, N_CLASS), nn.Dropout(0.25), nn.Sigmoid()
)
def forward(self, mel):
mel = mel.transpose(-1, -2).unsqueeze(1)
x = self.cnn(self.unet(mel)).transpose(1, 2).flatten(-2)
x = self.fc(x)
return x
from librosa.filters import mel
class MelSpectrogram(torch.nn.Module):
def __init__(
self,
is_half,
n_mel_channels,
sampling_rate,
win_length,
hop_length,
n_fft=None,
mel_fmin=0,
mel_fmax=None,
clamp=1e-5,
):
super().__init__()
n_fft = win_length if n_fft is None else n_fft
self.hann_window = {}
mel_basis = mel(
sr=sampling_rate,
n_fft=n_fft,
n_mels=n_mel_channels,
fmin=mel_fmin,
fmax=mel_fmax,
htk=True,
)
mel_basis = torch.from_numpy(mel_basis).float()
self.register_buffer("mel_basis", mel_basis)
self.n_fft = win_length if n_fft is None else n_fft
self.hop_length = hop_length
self.win_length = win_length
self.sampling_rate = sampling_rate
self.n_mel_channels = n_mel_channels
self.clamp = clamp
self.is_half = is_half
def forward(self, audio, keyshift=0, speed=1, center=True):
factor = 2 ** (keyshift / 12)
n_fft_new = int(np.round(self.n_fft * factor))
win_length_new = int(np.round(self.win_length * factor))
hop_length_new = int(np.round(self.hop_length * speed))
keyshift_key = str(keyshift) + "_" + str(audio.device)
if keyshift_key not in self.hann_window:
self.hann_window[keyshift_key] = torch.hann_window(win_length_new).to(
audio.device
)
fft = torch.stft(
audio,
n_fft=n_fft_new,
hop_length=hop_length_new,
win_length=win_length_new,
window=self.hann_window[keyshift_key],
center=center,
return_complex=True,
)
magnitude = torch.sqrt(fft.real.pow(2) + fft.imag.pow(2))
if keyshift != 0:
size = self.n_fft // 2 + 1
resize = magnitude.size(1)
if resize < size:
magnitude = F.pad(magnitude, (0, 0, 0, size - resize))
magnitude = magnitude[:, :size, :] * self.win_length / win_length_new
mel_output = torch.matmul(self.mel_basis, magnitude)
if self.is_half == True:
mel_output = mel_output.half()
log_mel_spec = torch.log(torch.clamp(mel_output, min=self.clamp))
return log_mel_spec
class RMVPE:
def __init__(self, model_path, is_half, device=None):
self.resample_kernel = {}
model = E2E(4, 1, (2, 2))
ckpt = torch.load(model_path, map_location="cpu")
model.load_state_dict(ckpt)
model.eval()
if is_half == True:
model = model.half()
self.model = model
self.resample_kernel = {}
self.is_half = is_half
if device is None:
device = "cuda" if torch.cuda.is_available() else "cpu"
self.device = device
self.mel_extractor = MelSpectrogram(
is_half, 128, 16000, 1024, 160, None, 30, 8000
).to(device)
self.model = self.model.to(device)
cents_mapping = 20 * np.arange(360) + 1997.3794084376191
self.cents_mapping = np.pad(cents_mapping, (4, 4)) # 368
def mel2hidden(self, mel):
with torch.no_grad():
n_frames = mel.shape[-1]
# PyTorch reflection_pad1d does not support Float16 (Half).
# Cast to float32 before padding, and cast back to float16 afterwards.
is_half = mel.dtype == torch.float16
if is_half:
mel = mel.float()
pad_size = 32 * ((n_frames - 1) // 32 + 1) - n_frames
if pad_size >= n_frames:
# Fallback to constant padding with the minimum value to prevent PyTorch reflection crashes on tiny audio chunks
min_val = mel.min().item()
mel = F.pad(mel, (0, pad_size), mode="constant", value=min_val)
else:
mel = F.pad(mel, (0, pad_size), mode="reflect")
if is_half:
mel = mel.half()
hidden = self.model(mel)
return hidden[:, :n_frames]
def decode(self, hidden, thred=0.03):
cents_pred = self.to_local_average_cents(hidden, thred=thred)
f0 = 10 * (2 ** (cents_pred / 1200))
f0[f0 == 10] = 0
# f0 = np.array([10 * (2 ** (cent_pred / 1200)) if cent_pred else 0 for cent_pred in cents_pred])
return f0
def infer_from_audio(self, audio, thred=0.03):
audio = torch.from_numpy(audio).float().to(self.device).unsqueeze(0)
# torch.cuda.synchronize()
# t0=ttime()
mel = self.mel_extractor(audio, center=True)
# torch.cuda.synchronize()
# t1=ttime()
hidden = self.mel2hidden(mel)
# torch.cuda.synchronize()
# t2=ttime()
hidden = hidden.squeeze(0).cpu().numpy()
if self.is_half == True:
hidden = hidden.astype("float32")
f0 = self.decode(hidden, thred=thred)
# torch.cuda.synchronize()
# t3=ttime()
# print("hmvpe:%s\t%s\t%s\t%s"%(t1-t0,t2-t1,t3-t2,t3-t0))
return f0
def to_local_average_cents(self, salience, thred=0.05):
# t0 = ttime()
center = np.argmax(salience, axis=1) # 帧长#index
salience = np.pad(salience, ((0, 0), (4, 4))) # 帧长,368
# t1 = ttime()
center += 4
todo_salience = []
todo_cents_mapping = []
starts = center - 4
ends = center + 5
for idx in range(salience.shape[0]):
todo_salience.append(salience[:, starts[idx] : ends[idx]][idx])
todo_cents_mapping.append(self.cents_mapping[starts[idx] : ends[idx]])
# t2 = ttime()
todo_salience = np.array(todo_salience) # 帧长,9
todo_cents_mapping = np.array(todo_cents_mapping) # 帧长,9
product_sum = np.sum(todo_salience * todo_cents_mapping, 1)
weight_sum = np.sum(todo_salience, 1) # 帧长
devided = product_sum / weight_sum # 帧长
# t3 = ttime()
maxx = np.max(salience, axis=1) # 帧长
devided[maxx <= thred] = 0
# t4 = ttime()
# print("decode:%s\t%s\t%s\t%s" % (t1 - t0, t2 - t1, t3 - t2, t4 - t3))
return devided
# if __name__ == '__main__':
# audio, sampling_rate = sf.read("卢本伟语录~1.wav")
# if len(audio.shape) > 1:
# audio = librosa.to_mono(audio.transpose(1, 0))
# audio_bak = audio.copy()
# if sampling_rate != 16000:
# audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=16000)
# model_path = "/bili-coeus/jupyter/jupyterhub-liujing04/vits_ch/test-RMVPE/weights/rmvpe_llc_half.pt"
# thred = 0.03 # 0.01
# device = 'cuda' if torch.cuda.is_available() else 'cpu'
# rmvpe = RMVPE(model_path,is_half=False, device=device)
# t0=ttime()
# f0 = rmvpe.infer_from_audio(audio, thred=thred)
# f0 = rmvpe.infer_from_audio(audio, thred=thred)
# f0 = rmvpe.infer_from_audio(audio, thred=thred)
# f0 = rmvpe.infer_from_audio(audio, thred=thred)
# f0 = rmvpe.infer_from_audio(audio, thred=thred)
# t1=ttime()
# print(f0.shape,t1-t0)
+768
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@@ -0,0 +1,768 @@
import os
import sys
import types
import json
# --- FIX FOR PATH COLLISION BETWEEN modules.py AND modules/ DIRECTORY ---
# Kita memaksa Python untuk mendaftarkan 'lib.infer_pack.modules' sebagai package directory
# alih-alih file-module, sehingga sub-impor (seperti F0Predictor) dapat dimuat dengan sukses
base_dir = os.path.dirname(os.path.abspath(__file__))
modules_path = os.path.join(base_dir, "lib", "infer_pack", "modules")
if os.path.isdir(modules_path):
modules_pkg = types.ModuleType("lib.infer_pack.modules")
modules_pkg.__path__ = [modules_path]
modules_pkg.__file__ = os.path.join(modules_path, "__init__.py")
sys.modules["lib.infer_pack.modules"] = modules_pkg
# ------------------------------------------------------------------------
import time
import asyncio
import logging
import traceback
import argparse
import threading
import webbrowser
from http.server import SimpleHTTPRequestHandler
import socketserver
import numpy as np
import torch
import onnxruntime as ort
# Add parent directories to sys.path so we can import lib
sys.path.append(os.getcwd())
from lib.infer_pack.onnx_inference import OnnxRVC, get_f0_predictor
# Set logging
logging.basicConfig(level=logging.INFO, format="%(asctime)s [%(levelname)s] %(message)s")
logger = logging.getLogger("RVC-Realtime-Server")
# Thread pool for audio processing (1 worker = sequential, but non-blocking to event loop)
import concurrent.futures
_audio_executor = concurrent.futures.ThreadPoolExecutor(max_workers=1, thread_name_prefix="rvc-audio")
# Global instances cache
current_rvc_onnx = None
current_model_key = None
model_root = "weights"
pretrained_root = "pretrained"
# Patch torch.load to default to weights_only=False for compatibility
original_load = torch.load
def patched_load(*args, **kwargs):
if "weights_only" not in kwargs:
kwargs["weights_only"] = False
return original_load(*args, **kwargs)
torch.load = patched_load
def get_onnx_models():
models = []
if os.path.exists(model_root):
for d in os.listdir(model_root):
d_path = os.path.join(model_root, d)
if os.path.isdir(d_path):
onnx_files = [f for f in os.listdir(d_path) if f.endswith(".onnx")]
if onnx_files:
models.append(d)
models.sort()
return models
def get_model_metadata(model_name):
model_dir = os.path.join(model_root, model_name)
onnx_files = [f for f in os.listdir(model_dir) if f.endswith(".onnx")]
pth_files = [f for f in os.listdir(model_dir) if f.endswith(".pth")]
onnx_path = os.path.join(model_dir, onnx_files[0])
sr = 40000 # default
if pth_files:
try:
pth_path = os.path.join(model_dir, pth_files[0])
cpt = torch.load(pth_path, map_location="cpu")
sr = cpt["config"][-1]
logger.info(f"Detected sample rate from .pth: {sr} Hz")
except Exception as e:
logger.warning(f"Failed to load sample rate from .pth, using default 40000: {e}")
version = "v2"
vec_path = "vec-768-layer-12"
try:
sess = ort.InferenceSession(onnx_path, providers=["CPUExecutionProvider"])
feat_dim = sess.get_inputs()[0].shape[2]
if feat_dim == 256:
version = "v1"
vec_path = "vec-256-layer-12"
logger.info("Detected RVC Model Version: v1 (feat_dim = 256)")
else:
version = "v2"
vec_path = "vec-768-layer-12"
logger.info("Detected RVC Model Version: v2 (feat_dim = 768)")
except Exception as e:
logger.error(f"Error auto-detecting model version from ONNX: {e}")
return onnx_path, sr, vec_path, version
class RealtimeVoiceChanger:
def __init__(self):
self.processor = None
self.model_name = ""
self.f0_up_key = 0
self.f0_method = "pm"
self.device = "cuda"
self.input_sr = 44100
self.noise_gate_db = -40.0
self.input_gain = 1.0
self.output_gain = 1.0
# Audio sliding buffers
self.input_buffer = np.zeros(0, dtype=np.float32)
self.history_duration = 0.30 # 300ms history = enough context even at 8192 samples @ 48kHz
self.target_sr = 40000
self.vec_path = "vec-768-layer-12"
self.version = "v2"
self.f0_predictors = {} # Cache to reuse pitch predictors instead of recreating them on every chunk
# Server Hardware Routing properties
self.local_stream = None
self.routing_mode = "browser"
self.input_device = None
self.output_device = None
self.chunk_size = 8192
self.loop = None
self.ws_client = None
self.visualizer_queue = None
# High-pass filter coefficients cache
self.hpf_b = None
self.hpf_a = None
def load_model(self, model_name, device):
global current_rvc_onnx, current_model_key
# Resolve the device provider
if device == "cuda" and not torch.cuda.is_available():
logger.warning("CUDA is not available, falling back to CPU")
device = "cpu"
model_key = f"{model_name}_{device}"
if current_rvc_onnx is None or current_model_key != model_key:
logger.info(f"Loading RVC model '{model_name}' on {device}...")
onnx_path, sr, vec_path, version = get_model_metadata(model_name)
# Ensure HuBERT model exists
full_vec_path = os.path.join(pretrained_root, f"{vec_path}.onnx")
if not os.path.exists(full_vec_path):
raise FileNotFoundError(f"ContentVec ONNX not found at: {full_vec_path}")
current_rvc_onnx = OnnxRVC(
model_path=onnx_path,
sr=sr,
hop_size=512,
vec_path=vec_path,
device=device
)
current_model_key = model_key
logger.info("Model loaded successfully")
self.processor = current_rvc_onnx
self.target_sr = self.processor.sampling_rate
self.model_name = model_name
self.device = device
def set_config(self, config):
logger.info(f"Updating config: {config}")
# Update config fields
self.f0_up_key = int(config.get("f0_up_key", self.f0_up_key))
self.f0_method = config.get("f0_method", self.f0_method)
self.input_sr = int(config.get("input_sr", self.input_sr))
self.noise_gate_db = float(config.get("noise_gate", self.noise_gate_db))
self.input_gain = float(config.get("input_gain", self.input_gain))
self.output_gain = float(config.get("output_gain", self.output_gain))
model_name = config.get("model_name", self.model_name)
device = config.get("device", self.device)
if not self.model_name or model_name != self.model_name or device != self.device:
self.load_model(model_name, device)
# Reset input buffer if input samplerate changed
history_samples = int(self.history_duration * self.input_sr)
if len(self.input_buffer) != history_samples:
self.input_buffer = np.zeros(history_samples, dtype=np.float32)
# Design a 1st order Butterworth high-pass filter at 80Hz to eliminate low-frequency static rumbling/hums
try:
from scipy import signal
nyq = 0.5 * self.input_sr
normal_cutoff = 80.0 / nyq
self.hpf_b, self.hpf_a = signal.butter(1, normal_cutoff, btype='high', analog=False)
except Exception as e:
logger.error(f"Failed to design high-pass filter: {e}")
self.hpf_b, self.hpf_a = None, None
def apply_noise_gate(self, audio):
# Calculate RMS energy of the audio chunk
rms = np.sqrt(np.mean(audio**2)) + 1e-9
rms_db = 20 * np.log10(rms)
if rms_db < self.noise_gate_db:
return np.zeros_like(audio)
return audio
def resample(self, audio, orig_sr, target_sr):
if orig_sr == target_sr:
return audio
# Fast linear interpolation resampling
duration = len(audio) / orig_sr
num_target_samples = int(duration * target_sr)
x_orig = np.linspace(0, duration, len(audio))
x_target = np.linspace(0, duration, num_target_samples)
return np.interp(x_target, x_orig, audio).astype(np.float32)
def process_audio_chunk(self, raw_chunk):
"""
Process a raw input Float32 PCM audio chunk in memory with sliding window.
"""
if self.processor is None:
return raw_chunk
t_start = time.time()
# 1. Apply High-pass filter (80Hz Low-cut) to eliminate low-frequency background rumbles and AC hum
chunk = raw_chunk
if self.hpf_b is not None and self.hpf_a is not None:
try:
from scipy import signal
chunk = signal.lfilter(self.hpf_b, self.hpf_a, chunk).astype(np.float32)
except Exception as e:
pass
# 2. Apply Input Gain & Noise Gate
chunk = chunk * self.input_gain
chunk = self.apply_noise_gate(chunk)
# If chunk is pure silence from the gate, bypass inference immediately to save CPU!
if np.max(np.abs(chunk)) < 1e-6:
output_len = int(len(raw_chunk) * (self.target_sr / self.input_sr))
return np.zeros(output_len, dtype=np.float32)
t_gate = time.time()
# 3. Manage Sliding Window Buffer
self.input_buffer = np.append(self.input_buffer[len(chunk):], chunk)
# Append 120ms of silence at the end to push RVC convolution edge fading into the padded future (no edge distortion!)
future_samples = int(0.12 * self.input_sr)
full_input_audio = np.append(self.input_buffer, np.zeros(future_samples, dtype=np.float32))
# 4. Resample full segment to 16kHz for HuBERT and RMVPE
wav16k = self.resample(full_input_audio, self.input_sr, 16000)
t_resample_in = time.time()
# 4. Generate RVC ONNX inputs in-memory
hubert = self.processor.vec_model(wav16k)
hubert = np.repeat(hubert, 2, axis=2).transpose(0, 2, 1).astype(np.float32)
hubert_length = hubert.shape[1]
t_hubert = time.time()
# Initialize and cache pitch predictor (extract at 16kHz for 3x performance boost!)
predictor_key = f"{self.f0_method}_{self.device}_16000"
if predictor_key not in self.f0_predictors:
logger.info(f"Initializing and caching 16kHz F0 Predictor '{self.f0_method}' on {self.device}...")
hop_16k = int(self.processor.hop_size * 16000 / self.target_sr)
self.f0_predictors[predictor_key] = get_f0_predictor(
self.f0_method,
hop_length=hop_16k,
sampling_rate=16000,
threshold=0.02,
device=self.device,
is_half=True if self.device == "cuda" else False
)
f0_predictor = self.f0_predictors[predictor_key]
# Calculate pitch on 16kHz audio for all methods (massive CPU speed up!)
pitchf = f0_predictor.compute_f0(wav16k, hubert_length)
t_f0 = time.time()
# Pitch transpose
pitchf = pitchf * (2 ** (self.f0_up_key / 12))
# Pitch binning for RVC
f0_min = 50
f0_max = 1100
f0_mel_min = 1127 * np.log(1 + f0_min / 700)
f0_mel_max = 1127 * np.log(1 + f0_max / 700)
f0_mel = 1127 * np.log(1 + pitchf / 700)
f0_mel[f0_mel > 0] = (f0_mel[f0_mel > 0] - f0_mel_min) * 254 / (f0_mel_max - f0_mel_min) + 1
f0_mel[f0_mel <= 1] = 1
f0_mel[f0_mel > 255] = 255
pitch = np.rint(f0_mel).astype(np.int64)
pitchf = pitchf.reshape(1, len(pitchf)).astype(np.float32)
pitch = pitch.reshape(1, len(pitch))
ds = np.array([0]).astype(np.int64) # sid = 0
rnd = np.random.randn(1, 192, hubert_length).astype(np.float32)
hubert_length_tensor = np.array([hubert_length]).astype(np.int64)
# 5. Run synthesis
out_wav = self.processor.forward(hubert, hubert_length_tensor, pitch, pitchf, ds, rnd).squeeze()
out_wav = out_wav.astype(np.float32) / 32767.0 # Normalize back to [-1.0, 1.0] float32
t_synth = time.time()
# 6. Extract only the newly converted chunk, discarding history and future padding.
# The new chunk starts at (buffer_size - chunk_size) in the updated input_buffer.
# Use exact ratio target_sr/input_sr for clean integer math, not out_wav/full_input_audio
# which can drift due to future silence padding.
sr_ratio = self.target_sr / self.input_sr
history_in_buffer = len(self.input_buffer) - len(chunk) # samples of old audio before new chunk
start_idx = int(history_in_buffer * sr_ratio)
end_idx = int((history_in_buffer + len(chunk)) * sr_ratio)
# Safety clamp
start_idx = max(0, min(start_idx, len(out_wav) - 1))
end_idx = max(start_idx + 1, min(end_idx, len(out_wav)))
# Extract the converted chunk safely from the middle (flawless, clear, continuous voice!)
output_chunk = out_wav[start_idx:end_idx]
# Resample back to target output samples to match browser playback rate perfectly
target_chunk_len = int(len(chunk) * (self.target_sr / self.input_sr))
output_chunk = self.resample(output_chunk, len(output_chunk), target_chunk_len)
# Apply output gain
output_chunk = output_chunk * self.output_gain
# Ensure we don't clip
output_chunk = np.clip(output_chunk, -1.0, 1.0)
t_end = time.time()
d_gate = (t_gate - t_start) * 1000
d_res_in = (t_resample_in - t_gate) * 1000
d_hubert = (t_hubert - t_resample_in) * 1000
d_f0 = (t_f0 - t_hubert) * 1000
d_synth = (t_synth - t_f0) * 1000
d_res_out = (t_end - t_synth) * 1000
t_elapsed = (t_end - t_start) * 1000
logger.info(
f"Chunk Profile: total={t_elapsed:.1f}ms | gate={d_gate:.1f}ms | res_in={d_res_in:.1f}ms | "
f"hubert={d_hubert:.1f}ms | f0={d_f0:.1f}ms | synth={d_synth:.1f}ms | res_out={d_res_out:.1f}ms"
)
return output_chunk
def start_local_stream(self, loop, ws_client):
import sounddevice as sd
self.loop = loop
self.ws_client = ws_client
self.visualizer_queue = asyncio.Queue()
if self.local_stream is not None:
self.stop_local_stream()
if self.input_device is None:
self.input_device = sd.default.device[0]
if self.output_device is None:
self.output_device = sd.default.device[1]
input_info = sd.query_devices(self.input_device)
output_info = sd.query_devices(self.output_device)
input_sr = int(input_info["default_samplerate"])
logger.info(f"Starting Server Hardware Stream: Input='{input_info['name']}' ({input_sr}Hz) | Output='{output_info['name']}' ({self.target_sr}Hz)")
self.set_config({
"input_sr": input_sr
})
def audio_callback(indata, outdata, frames, time_info, status):
if status:
logger.warning(f"Hardware Audio Callback Status: {status}")
raw_chunk = indata[:, 0].copy()
output_chunk = self.process_audio_chunk(raw_chunk)
if len(output_chunk) < frames:
outdata[:, 0] = np.pad(output_chunk, (0, frames - len(output_chunk)), "constant")
else:
outdata[:, 0] = output_chunk[:frames]
# Send waveform chunks to WebSocket safely using loop.call_soon_threadsafe
if self.ws_client is not None:
loop.call_soon_threadsafe(
self.visualizer_queue.put_nowait,
(raw_chunk.copy(), output_chunk.copy())
)
try:
self.local_stream = sd.Stream(
device=(self.input_device, self.output_device),
samplerate=self.target_sr,
blocksize=self.chunk_size,
channels=1,
dtype="float32",
callback=audio_callback
)
self.local_stream.start()
logger.info("Server Hardware Stream active and processing locally!")
except Exception as e:
logger.error(f"Failed to start hardware stream: {e}")
raise e
def stop_local_stream(self):
if self.local_stream is not None:
try:
self.local_stream.stop()
self.local_stream.close()
logger.info("Server Hardware Stream stopped successfully.")
except Exception as e:
logger.error(f"Error stopping hardware stream: {e}")
self.local_stream = None
self.visualizer_queue = None
# --- WEBSOCKET SERVER IMPLEMENTATION ---
async def websocket_handler(websocket):
logger.info("New WebSocket client connected")
rvc = RealtimeVoiceChanger()
loop = asyncio.get_running_loop()
# --- Pipeline queues ---
# input_queue : raw bytes from browser mic
# output_queue: processed bytes ready to send back
# maxsize=3 = ~3 chunk durations of buffer; provides backpressure without unbounded memory
input_queue = asyncio.Queue(maxsize=3)
output_queue = asyncio.Queue(maxsize=3)
# --- STAGE 1: Receiver ---
# Reads ALL WebSocket messages immediately (never blocks on processing).
# Handles JSON config inline; binary audio chunks go to input_queue.
async def receiver_task():
try:
async for message in websocket:
if isinstance(message, str):
try:
data = json.loads(message)
if data.get("type") == "config":
new_routing_mode = data.get("routing_mode", rvc.routing_mode)
new_input_device = data.get("input_device", rvc.input_device)
new_output_device = data.get("output_device", rvc.output_device)
new_chunk_size = int(data.get("chunk_size", rvc.chunk_size))
if new_input_device is not None: rvc.input_device = int(new_input_device)
if new_output_device is not None: rvc.output_device = int(new_output_device)
rvc.chunk_size = new_chunk_size
rvc.set_config(data)
if new_routing_mode == "hardware":
if rvc.routing_mode != "hardware" or rvc.local_stream is None:
rvc.routing_mode = "hardware"
rvc.start_local_stream(loop, websocket)
else:
if rvc.routing_mode == "hardware":
rvc.stop_local_stream()
rvc.routing_mode = "browser"
response = {
"type": "config_success",
"model_name": rvc.model_name,
"target_sr": rvc.target_sr,
"f0_method": rvc.f0_method,
"f0_up_key": rvc.f0_up_key,
"device": rvc.device,
"routing_mode": rvc.routing_mode
}
await websocket.send(json.dumps(response))
except Exception as e:
logger.error(f"Config parse error: {e}")
await websocket.send(json.dumps({"type": "error", "message": str(e)}))
elif isinstance(message, bytes):
if rvc.routing_mode == "hardware":
continue
if rvc.processor is None:
# Prepend 0.0 processing time to echoed message
input_chunk = np.frombuffer(message, dtype=np.float32)
payload = np.empty(len(input_chunk) + 1, dtype=np.float32)
payload[0] = 0.0
payload[1:] = input_chunk
await websocket.send(payload.tobytes())
continue
# Put chunk in queue — await here means we yield if queue is full (backpressure)
await input_queue.put(message)
except Exception as e:
logger.error(f"Receiver error: {e}")
finally:
# Signal downstream stages to stop
await input_queue.put(None)
# --- STAGE 2: Processor ---
# Pulls from input_queue, processes in thread (non-blocking to event loop), pushes to output_queue.
# Sequential (1 executor worker) = output order matches input order.
async def processor_task():
try:
while True:
item = await input_queue.get()
if item is None:
break # shutdown signal
input_chunk = np.frombuffer(item, dtype=np.float32).copy()
t_start = time.time()
output_chunk = await loop.run_in_executor(
_audio_executor, rvc.process_audio_chunk, input_chunk
)
t_elapsed = (time.time() - t_start) * 1000
# Prepend the elapsed processing time to the audio chunk bytes
payload = np.empty(len(output_chunk) + 1, dtype=np.float32)
payload[0] = t_elapsed
payload[1:] = output_chunk
await output_queue.put(payload.tobytes())
except asyncio.CancelledError:
pass
except Exception as e:
logger.error(f"Processor error: {e}")
finally:
await output_queue.put(None)
# --- STAGE 3: Sender ---
# Pulls processed audio from output_queue and sends to client.
async def sender_task():
try:
while True:
item = await output_queue.get()
if item is None:
break # shutdown signal
await websocket.send(item)
except asyncio.CancelledError:
pass
except Exception as e:
logger.error(f"Sender error: {e}")
# --- Visualizer (hardware mode only) ---
async def visualizer_sender_loop():
try:
while True:
if rvc.routing_mode == "hardware" and rvc.visualizer_queue is not None:
try:
raw_chunk, output_chunk = await asyncio.wait_for(
rvc.visualizer_queue.get(), timeout=0.1
)
payload = {
"type": "visualizer",
"input": raw_chunk.tolist(),
"output": output_chunk.tolist()
}
await websocket.send(json.dumps(payload))
except asyncio.TimeoutError:
pass
else:
await asyncio.sleep(0.05)
except asyncio.CancelledError:
pass
except Exception as e:
logger.error(f"Visualizer sender error: {e}")
# --- Send device list on connect ---
import sounddevice as sd
devices_list = []
try:
for idx, d in enumerate(sd.query_devices()):
devices_list.append({
"id": idx,
"name": d["name"],
"max_input_channels": d["max_input_channels"],
"max_output_channels": d["max_output_channels"],
"default_samplerate": d["default_samplerate"]
})
default_input = sd.default.device[0]
default_output = sd.default.device[1]
except Exception as e:
logger.error(f"Failed to query server audio devices: {e}")
devices_list = []
default_input = -1
default_output = -1
await websocket.send(json.dumps({
"type": "init_devices",
"devices": devices_list,
"default_input": default_input,
"default_output": default_output
}))
# --- Run all pipeline stages concurrently ---
vis_task = asyncio.create_task(visualizer_sender_loop())
proc_task = asyncio.create_task(processor_task())
send_task = asyncio.create_task(sender_task())
try:
await receiver_task() # runs until websocket closes
except Exception as e:
logger.error(f"WebSocket handler error: {e}")
finally:
vis_task.cancel()
proc_task.cancel()
send_task.cancel()
rvc.stop_local_stream()
logger.info("WebSocket client disconnected, pipeline cleaned up.")
async def start_websocket_server(host, port):
import websockets
logger.info(f"Starting WebSocket server on ws://{host}:{port}...")
async with websockets.serve(websocket_handler, host, port):
await asyncio.Future()
# --- HTTP STATIC FILE SERVER FOR FRONTEND ---
def start_http_server(port, directory="frontend"):
class MyHandler(SimpleHTTPRequestHandler):
def __init__(self, *args, **kwargs):
# Force serve from directory relative to the project root
base_dir = os.path.dirname(os.path.abspath(__file__))
full_dir = os.path.join(base_dir, directory)
super().__init__(*args, directory=full_dir, **kwargs)
def log_message(self, format, *args):
# Suppress standard logging to prevent console pollution
pass
try:
# Create a TCPServer that allows address reuse
socketserver.TCPServer.allow_reuse_address = True
with socketserver.TCPServer(("", port), MyHandler) as httpd:
logger.info(f"Serving HTTP frontend on http://localhost:{port}")
httpd.serve_forever()
except Exception as e:
logger.error(f"Failed to start HTTP server: {e}")
# --- LOCAL AUDIO DEVICE STREAM MODE ---
def run_local_device_mode(model_name, f0_up_key, f0_method, device, input_device, output_device, chunk_size):
import sounddevice as sd
logger.info("Starting Local Audio Hardware Stream Mode...")
rvc = RealtimeVoiceChanger()
rvc.load_model(model_name, device)
if input_device is None:
input_device = sd.default.device[0]
if output_device is None:
output_device = sd.default.device[1]
input_info = sd.query_devices(input_device)
output_info = sd.query_devices(output_device)
input_sr = int(input_info["default_samplerate"])
target_sr = rvc.target_sr
logger.info(f"Input Device: {input_info['name']} (Sample Rate: {input_sr} Hz)")
logger.info(f"Output Device: {output_info['name']} (Sample Rate: {target_sr} Hz)")
rvc.set_config({
"f0_up_key": f0_up_key,
"f0_method": f0_method,
"input_sr": input_sr,
"device": device,
"model_name": model_name,
"noise_gate": -40.0,
"input_gain": 1.0,
"output_gain": 1.0
})
def audio_callback(indata, outdata, frames, time_info, status):
if status:
logger.warning(f"Audio Callback Status: {status}")
raw_chunk = indata[:, 0].copy()
output_chunk = rvc.process_audio_chunk(raw_chunk)
if len(output_chunk) < frames:
outdata[:, 0] = np.pad(output_chunk, (0, frames - len(output_chunk)), "constant")
else:
outdata[:, 0] = output_chunk[:frames]
try:
stream = sd.Stream(
device=(input_device, output_device),
samplerate=target_sr,
blocksize=chunk_size,
channels=1,
dtype="float32",
callback=audio_callback
)
with stream:
logger.info("Real-Time Sounddevice Stream active! Press Ctrl+C to stop.")
while True:
time.sleep(1)
except KeyboardInterrupt:
logger.info("Local stream stopped by user")
except Exception as e:
logger.error(f"Local stream error: {e}")
traceback.print_exc()
# --- MAIN ---
if __name__ == "__main__":
parser = argparse.ArgumentParser(description="High-Performance Real-Time RVC ONNX Server")
parser.add_argument("--mode", type=str, default="websocket", choices=["websocket", "device"], help="Server running mode")
parser.add_argument("--host", type=str, default="127.0.0.1", help="WebSocket host")
parser.add_argument("--port", type=int, default=8765, help="WebSocket port")
parser.add_argument("--http_port", type=int, default=8000, help="HTTP static server port for Web UI")
parser.add_argument("--model", type=str, default="", help="RVC Model folder name inside weights/")
parser.add_argument("--transpose", type=int, default=0, help="Pitch shift in semitones (transpose)")
parser.add_argument("--f0_method", type=str, default="pm", choices=["pm", "harvest", "dio", "rmvpe"], help="Pitch extraction method")
parser.add_argument("--device", type=str, default="cuda", choices=["cpu", "cuda", "dml"], help="Execution provider")
parser.add_argument("--input_device", type=int, default=None, help="Input device ID (for device mode)")
parser.add_argument("--output_device", type=int, default=None, help="Output device ID (for device mode)")
parser.add_argument("--chunk_size", type=int, default=2048, help="Audio block size in samples")
args = parser.parse_args()
model_name = args.model
if not model_name:
models = get_onnx_models()
if models:
model_name = models[0]
logger.info(f"Auto-selected model: {model_name}")
else:
logger.error("No models found in weights/ directory. Please export a model first.")
sys.exit(1)
if args.mode == "websocket":
# 1. Start HTTP Server in a background thread to serve the frontend!
http_thread = threading.Thread(
target=start_http_server,
args=(args.http_port, "frontend"),
daemon=True
)
http_thread.start()
# 2. Automatically open the Web UI in the default browser!
web_ui_url = f"http://127.0.0.1:{args.http_port}"
logger.info(f"Automatically launching Web UI at {web_ui_url} in browser...")
# We give it a tiny delay to ensure the HTTP server socket is open
def open_browser():
time.sleep(0.5)
webbrowser.open(web_ui_url)
browser_thread = threading.Thread(target=open_browser, daemon=True)
browser_thread.start()
# 3. Start the WebSocket server on the main event loop
try:
asyncio.run(start_websocket_server(args.host, args.port))
except KeyboardInterrupt:
logger.info("Server shut down")
elif args.mode == "device":
run_local_device_mode(
model_name=model_name,
f0_up_key=args.transpose,
f0_method=args.f0_method,
device=args.device,
input_device=args.input_device,
output_device=args.output_device,
chunk_size=args.chunk_size
)
+15
View File
@@ -0,0 +1,15 @@
@echo off
title ONNX Real-Time Voice Changer
cd /d "%~dp0"
set VENV_PYTHON=..\rvc-tts-webui\venv\Scripts\python.exe
if exist "%VENV_PYTHON%" (
echo Menjalankan menggunakan virtual environment dari rvc-tts-webui...
"%VENV_PYTHON%" -u server.py --host 127.0.0.1 --port 8765 --http_port 8000
) else (
echo Virtual environment tidak ditemukan, mencoba menggunakan python sistem...
python -u server.py --host 127.0.0.1 --port 8765 --http_port 8000
)
pause