ibrahimabdelaal
Add torchaudio dependency and improve UI layout
6d0fe97
import gradio as gr
import soundfile as sf
import torch
import numpy as np
from pathlib import Path
from transformers import AutoProcessor, AutoModel
import tempfile
import os
import spaces
import shutil
# Import helper functions from your existing code
from typing import List
def smart_text_split_arabic(text: str, max_length: int = 300) -> List[str]:
"""Intelligently split Arabic text into chunks while preserving context."""
if len(text) <= max_length:
return [text]
chunks = []
remaining_text = text.strip()
while remaining_text:
if len(remaining_text) <= max_length:
chunks.append(remaining_text)
break
chunk = remaining_text[:max_length]
split_point = -1
# Priority 1: Sentence endings
sentence_endings = ['.', '!', '?', '۔']
for i in range(len(chunk) - 1, max(0, max_length - 100), -1):
if chunk[i] in sentence_endings:
if i == len(chunk) - 1 or chunk[i + 1] == ' ':
split_point = i + 1
break
# Priority 2: Arabic clause separators
if split_point == -1:
arabic_separators = ['،', '؛', ':', ';', ',']
for i in range(len(chunk) - 1, max(0, max_length - 50), -1):
if chunk[i] in arabic_separators:
if i == len(chunk) - 1 or chunk[i + 1] == ' ':
split_point = i + 1
break
# Priority 3: Word boundaries
if split_point == -1:
for i in range(len(chunk) - 1, max(0, max_length - 30), -1):
if chunk[i] == ' ':
split_point = i + 1
break
if split_point == -1:
split_point = max_length
current_chunk = remaining_text[:split_point].strip()
if current_chunk:
chunks.append(current_chunk)
remaining_text = remaining_text[split_point:].strip()
return chunks
def apply_crossfade(audio1: np.ndarray, audio2: np.ndarray,
fade_duration: float = 0.1, sample_rate: int = 24000) -> np.ndarray:
"""Apply crossfade between two audio segments."""
fade_samples = int(fade_duration * sample_rate)
fade_samples = min(fade_samples, len(audio1), len(audio2))
if fade_samples <= 0:
return np.concatenate([audio1, audio2])
fade_out = np.linspace(1.0, 0.0, fade_samples)
fade_in = np.linspace(0.0, 1.0, fade_samples)
audio1_faded = audio1.copy()
audio2_faded = audio2.copy()
audio1_faded[-fade_samples:] *= fade_out
audio2_faded[:fade_samples] *= fade_in
overlap = audio1_faded[-fade_samples:] + audio2_faded[:fade_samples]
result = np.concatenate([
audio1_faded[:-fade_samples],
overlap,
audio2_faded[fade_samples:]
])
return result
def normalize_audio(audio: np.ndarray, target_rms: float = 0.1) -> np.ndarray:
"""Normalize audio to target RMS level."""
if len(audio) == 0:
return audio
current_rms = np.sqrt(np.mean(audio ** 2))
if current_rms > 1e-6:
scaling_factor = target_rms / current_rms
return audio * scaling_factor
return audio
def remove_silence(audio: np.ndarray, sample_rate: int = 24000,
silence_threshold: float = 0.01, min_silence_duration: float = 0.5) -> np.ndarray:
"""Remove long silences from audio."""
if len(audio) == 0:
return audio
frame_size = int(0.05 * sample_rate)
min_silence_frames = int(min_silence_duration / 0.05)
frames = []
for i in range(0, len(audio), frame_size):
frame = audio[i:i + frame_size]
if len(frame) < frame_size:
frames.append(frame)
break
rms = np.sqrt(np.mean(frame ** 2))
frames.append(frame if rms > silence_threshold else None)
result_frames = []
silence_count = 0
for frame in frames:
if frame is None:
silence_count += 1
else:
if silence_count > 0:
if silence_count >= min_silence_frames:
for _ in range(min(2, silence_count)):
result_frames.append(np.zeros(frame_size, dtype=np.float32))
else:
for _ in range(silence_count):
result_frames.append(np.zeros(frame_size, dtype=np.float32))
result_frames.append(frame)
silence_count = 0
if not result_frames:
return np.array([], dtype=np.float32)
return np.concatenate(result_frames)
# Global model instance
model_cache = {}
def load_model(model_id: str = "IbrahimSalah/Arabic-TTS-Spark"):
"""Load the TTS model (cached)."""
if "model" not in model_cache:
device = "cuda" if torch.cuda.is_available() else "cpu"
print(f"Loading model on {device}...")
processor = AutoProcessor.from_pretrained(model_id, trust_remote_code=True)
model = AutoModel.from_pretrained(model_id, trust_remote_code=True).eval().to(device)
processor.model = model
model_cache["model"] = model
model_cache["processor"] = processor
model_cache["device"] = device
print("Model loaded successfully!")
return model_cache["model"], model_cache["processor"], model_cache["device"]
@spaces.GPU(duration=120) # Request GPU for 120 seconds
def generate_speech(
text: str,
reference_audio,
reference_transcript: str,
temperature: float = 0.8,
top_p: float = 0.95,
max_chunk_length: int = 300,
crossfade_duration: float = 0.08,
progress=gr.Progress()
):
"""Generate speech from text using Spark TTS."""
try:
# Load model
progress(0.1, desc="Loading model...")
model, processor, device = load_model()
# Validate inputs
if not text.strip():
return None, "❌ Please enter text to synthesize."
if reference_audio is None:
return None, "❌ Please upload a reference audio file."
if not reference_transcript.strip():
return None, "❌ Please enter the reference transcript."
# Split text into chunks
progress(0.2, desc="Splitting text...")
text_chunks = smart_text_split_arabic(text, max_chunk_length)
audio_segments = []
sample_rate = None
# Generate audio for each chunk
for i, chunk in enumerate(text_chunks):
progress(0.2 + (0.6 * (i / len(text_chunks))), desc=f"Generating chunk {i+1}/{len(text_chunks)}...")
inputs = processor(
text=chunk.lower(),
prompt_speech_path=reference_audio,
prompt_text=reference_transcript,
return_tensors="pt"
).to(device)
global_tokens_prompt = inputs.pop("global_token_ids_prompt", None)
with torch.no_grad():
output_ids = model.generate(
**inputs,
max_new_tokens=8000,
do_sample=True,
temperature=temperature,
top_k=50,
top_p=top_p,
eos_token_id=processor.tokenizer.eos_token_id,
pad_token_id=processor.tokenizer.pad_token_id
)
output = processor.decode(
generated_ids=output_ids,
global_token_ids_prompt=global_tokens_prompt,
input_ids_len=inputs["input_ids"].shape[-1]
)
audio = output["audio"]
if isinstance(audio, torch.Tensor):
audio = audio.cpu().numpy()
if sample_rate is None:
sample_rate = output["sampling_rate"]
# Post-process
audio = normalize_audio(audio, target_rms=0.1)
audio = remove_silence(audio, sample_rate)
if len(audio) > 0:
audio_segments.append(audio)
if not audio_segments:
return None, "❌ No audio was generated."
# Concatenate segments
progress(0.9, desc="Concatenating audio...")
final_audio = audio_segments[0]
for i in range(1, len(audio_segments)):
final_audio = apply_crossfade(
final_audio, audio_segments[i],
fade_duration=crossfade_duration,
sample_rate=sample_rate
)
# Final normalization
final_audio = normalize_audio(final_audio, target_rms=0.1)
# Save to temporary file
with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as tmp_file:
sf.write(tmp_file.name, final_audio, sample_rate)
output_path = tmp_file.name
duration = len(final_audio) / sample_rate
status = f"✅ Generated {duration:.2f}s audio from {len(text_chunks)} chunks"
progress(1.0, desc="Complete!")
return output_path, status
except Exception as e:
import traceback
error_msg = f"❌ Error: {str(e)}\n{traceback.format_exc()}"
print(error_msg)
return None, error_msg
# Default examples
DEFAULT_REFERENCE_TEXT = "لَا يَمُرُّ يَوْمٌ إِلَّا وَأَسْتَقْبِلُ عِدَّةَ رَسَائِلَ، تَتَضَمَّنُ أَسْئِلَةً مُلِحَّةْ."
DEFAULT_TEXT = "تُسَاهِمُ التِّقْنِيَّاتُ الْحَدِيثَةُ فِي تَسْهِيلِ حَيَاةِ الْإِنْسَانِ، وَذَلِكَ مِنْ خِلَالِ تَطْوِيرِ أَنْظِمَةٍ ذَكِيَّةٍ تَعْتَمِدُ عَلَى الذَّكَاءِ الِاصْطِنَاعِيِّ."
# Path to default reference audio
DEFAULT_REFERENCE_AUDIO = "reference.wav"
# Create Gradio interface
with gr.Blocks(title="Arabic TTS - Spark", theme=gr.themes.Soft()) as demo:
gr.Markdown("""
# 🎙️ Arabic Text-to-Speech | Spark Model
High-quality Arabic TTS with voice cloning. **Diacritized text (تشكيل) required.**
**Model:** [IbrahimSalah/Arabic-TTS-Spark](https://huggingface.co/IbrahimSalah/Arabic-TTS-Spark)
""")
with gr.Row():
with gr.Column(scale=1):
text_input = gr.Textbox(
label="📝 Text to Synthesize (Arabic with Tashkeel)",
placeholder="أَدْخِلْ نَصًّا عَرَبِيًّا مُشَكَّلًا هُنَا...",
lines=6,
value=DEFAULT_TEXT
)
with gr.Row():
with gr.Column():
gr.Markdown("**🎵 Reference Audio**")
reference_audio = gr.Audio(
label="",
type="filepath",
value=DEFAULT_REFERENCE_AUDIO
)
with gr.Column():
reference_transcript = gr.Textbox(
label="📄 Reference Transcript (with Tashkeel)",
placeholder="النص المقابل للصوت المرجعي...",
lines=4,
value=DEFAULT_REFERENCE_TEXT
)
with gr.Accordion("⚙️ Advanced Settings", open=False):
with gr.Row():
temperature = gr.Slider(0.1, 1.5, value=0.8, step=0.1, label="Temperature")
top_p = gr.Slider(0.1, 1.0, value=0.95, step=0.05, label="Top P")
with gr.Row():
max_chunk = gr.Slider(100, 500, value=300, step=50, label="Max Chunk Length")
crossfade = gr.Slider(0.01, 0.2, value=0.08, step=0.01, label="Crossfade (s)")
generate_btn = gr.Button("🎤 Generate Speech", variant="primary", size="lg")
with gr.Column(scale=1):
output_audio = gr.Audio(label="🔊 Generated Speech", type="filepath")
status_text = gr.Textbox(label="Status", interactive=False, lines=2)
gr.Markdown("""
### ℹ️ Requirements
- **Diacritized text is required** (تشكيل/تشكيل)
- Reference audio: 5-30 seconds, clear speech
- Use AI (ChatGPT/Claude) or [online tools](https://tahadz.com/mishkal) to add diacritics
### 🔗 Resources
- [Model Card](https://huggingface.co/IbrahimSalah/Arabic-TTS-Spark)
- [F5-TTS Arabic](https://huggingface.co/IbrahimSalah/Arabic-F5-TTS-v2)
- [Report Issues](https://huggingface.co/IbrahimSalah/Arabic-TTS-Spark/discussions)
""")
# Examples
with gr.Accordion("📚 Examples", open=False):
gr.Examples(
examples=[
[DEFAULT_TEXT, DEFAULT_REFERENCE_AUDIO, DEFAULT_REFERENCE_TEXT],
["السَّلَامُ عَلَيْكُمْ وَرَحْمَةُ اللَّهِ وَبَرَكَاتُهُ، كَيْفَ حَالُكَ الْيَوْمَ؟", DEFAULT_REFERENCE_AUDIO, DEFAULT_REFERENCE_TEXT],
["الذَّكَاءُ الِاصْطِنَاعِيُّ يُغَيِّرُ الْعَالَمَ بِسُرْعَةٍ كَبِيرَةٍ وَيُسَاهِمُ فِي تَطْوِيرِ حُلُولٍ مُبْتَكَرَةٍ.", DEFAULT_REFERENCE_AUDIO, DEFAULT_REFERENCE_TEXT]
],
inputs=[text_input, reference_audio, reference_transcript]
)
generate_btn.click(
fn=generate_speech,
inputs=[text_input, reference_audio, reference_transcript, temperature, top_p, max_chunk, crossfade],
outputs=[output_audio, status_text]
)
if __name__ == "__main__":
demo.queue(max_size=20)
demo.launch()