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import spaces
import gradio as gr
import edge_tts
import asyncio
import tempfile
import os
import re
from pathlib import Path
from pydub import AudioSegment
import librosa
import soundfile as sf
import numpy as np
from pydub import AudioSegment
from pydub.playback import play
from scipy.signal import butter, lfilter # Ensure this line is present
def apply_low_pass_filter(audio_segment, cutoff_freq, sample_rate, order=5):
"""Applies a low-pass filter to a pydub AudioSegment."""
audio_np = np.array(audio_segment.get_array_of_samples()).astype(np.float32) / (2**15 - 1)
if audio_segment.channels == 2:
audio_np = audio_np.reshape(-1, 2)
nyquist_freq = 0.5 * sample_rate
normalized_cutoff = cutoff_freq / nyquist_freq
b, a = butter(order, normalized_cutoff, btype='low', analog=False)
filtered_data = np.zeros_like(audio_np, dtype=np.float32)
if audio_segment.channels == 1:
filtered_data = lfilter(b, a, audio_np)
else:
for channel in range(audio_segment.channels):
filtered_data[:, channel] = lfilter(b, a, audio_np[:, channel])
filtered_data_int16 = (filtered_data * (2**15 - 1)).astype(np.int16)
filtered_audio = AudioSegment(filtered_data_int16.tobytes(),
frame_rate=sample_rate,
sample_width=audio_segment.sample_width,
channels=audio_segment.channels)
return filtered_audio
def get_silence(duration_ms=1000):
# Create silent audio segment with specified parameters
silent_audio = AudioSegment.silent(
duration=duration_ms,
frame_rate=24000 # 24kHz sampling rate
)
# Set audio parameters
silent_audio = silent_audio.set_channels(1) # Mono
silent_audio = silent_audio.set_sample_width(4) # 32-bit (4 bytes per sample)
with tempfile.NamedTemporaryFile(delete=False, suffix=".mp3") as tmp_file:
# Export with specific bitrate and codec parameters
silent_audio.export(
tmp_file.name,
format="mp3",
bitrate="48k",
parameters=[
"-ac", "1", # Mono
"-ar", "24000", # Sample rate
"-sample_fmt", "s32", # 32-bit samples
"-codec:a", "libmp3lame" # MP3 codec
]
)
return tmp_file.name
# Get all available voices
async def get_voices():
try:
voices = await edge_tts.list_voices()
return {f"{v['ShortName']} - {v['Locale']} ({v['Gender']})": v['ShortName'] for v in voices}
except Exception as e:
print(f"Error listing voices: {e}")
return {}
async def generate_audio_with_voice_prefix(text_segment, default_voice, rate, pitch, overall_target_duration_ms=None, speed_adjustment_factor=1.0):
"""Generates audio for a text segment, handling voice prefixes and adjusting rate for duration."""
current_voice_full = default_voice
current_voice_short = current_voice_full.split(" - ")[0] if current_voice_full else ""
current_rate = rate
current_pitch = pitch
processed_text = text_segment.strip()
#print(f"Processing this text segment: '{processed_text}'") # Debug
voice_map = {
"1F": "en-GB-SoniaNeural",
"2M": "en-GB-RyanNeural",
"3M": "en-US-BrianMultilingualNeural",
"2F": "en-US-JennyNeural",
"1M": "en-AU-WilliamNeural",
"3F": "en-HK-YanNeural",
"4M": "en-GB-ThomasNeural",
"4F": "en-US-EmmaNeural",
"1O": "en-GB-RyanNeural", # Old Man
"1C": "en-GB-MaisieNeural", # Child
"1V": "vi-VN-HoaiMyNeural", # Vietnamese (Female)
"2V": "vi-VN-NamMinhNeural", # Vietnamese (Male)
"3V": "vi-VN-HoaiMyNeural", # Vietnamese (Female)
"4V": "vi-VN-NamMinhNeural", # Vietnamese (Male)
}
detect = 0
#iterate throught the voice map to see if a match if found, if found then set the voice
for prefix, voice_short in voice_map.items():
if processed_text.startswith(prefix):
current_voice_short = voice_short
if prefix in ["1F", "3F", "1V", "3V"]:
current_pitch = 0
elif prefix in ["1O", "4V"]:
current_pitch = -20
current_rate = -10
detect = 1
processed_text = processed_text[len(prefix):].strip() #this removes the Prefix and leave only number or text after it.
break
#match = re.search(r'([A-Za-z]+)-?(\d+)', processed_text)
match = re.search(r"^(-?\d+)\s*(.*)", processed_text)
if match:
#prefix_pitch = match.group(1)
number = match.group(1)
print(f"Prefix match found.") # Debug
current_pitch += int(number)
#processed_text = re.sub(r'[A-Za-z]+-?\d+', '', processed_text, count=1).strip()
#processed_text = re.sub(r'([A-Za-z]+)([-]?\d*)', '', processed_text, count=1).strip()
processed_text = match.group(2)
#elif detect:
# processed_text = processed_text.lstrip('-0123456789').strip() # Remove potential leftover numbers
if processed_text:
rate_str = f"{current_rate:+d}%"
pitch_str = f"{current_pitch:+d}Hz"
print(f"Sending to Edge: '{processed_text}'") # Debug
try:
communicate = edge_tts.Communicate(processed_text, current_voice_short, rate=rate_str, pitch=pitch_str)
with tempfile.NamedTemporaryFile(delete=False, suffix=".mp3") as tmp_file:
audio_path = tmp_file.name
await communicate.save(audio_path)
if os.path.exists(audio_path):
audio = AudioSegment.from_mp3(audio_path)
# Trim leading and trailing silence
def detect_leading_silence(sound, silence_threshold=-50.0, chunk_size=10):
trim_ms = 0
assert chunk_size > 0 # to avoid infinite loop
while sound[trim_ms:trim_ms+chunk_size].dBFS < silence_threshold and trim_ms < len(sound):
trim_ms += chunk_size
return trim_ms
start_trim = detect_leading_silence(audio)
end_trim = detect_leading_silence(audio.reverse())
trimmed_audio = audio[start_trim:len(audio)-end_trim]
trimmed_audio.export(audio_path, format="mp3") # Overwrite with trimmed version
return audio_path
except Exception as e:
print(f"Edge TTS error processing '{processed_text}': {e}")
return None
return None
async def process_transcript_line(line, next_line_start_time, default_voice, rate, pitch, overall_duration_ms, speed_adjustment_factor):
"""Processes a single transcript line with HH:MM:SS,milliseconds timestamp."""
match = re.match(r'(\d{2}):(\d{2}):(\d{2}),(\d{3})\s+(.*)', line)
if match:
start_h, start_m, start_s, start_ms, text_parts = match.groups()
start_time_ms = (
int(start_h) * 3600000 +
int(start_m) * 60000 +
int(start_s) * 1000 +
int(start_ms)
)
audio_segments = []
split_parts = re.split(r'[“”"]', text_parts)
process_next = False
for part in split_parts:
if part == '"':
process_next = not process_next
continue
if process_next and part.strip():
audio_path = await generate_audio_with_voice_prefix(part, default_voice, rate, pitch, overall_duration_ms, speed_adjustment_factor)
if audio_path:
audio_segments.append(audio_path)
elif not process_next and part.strip():
audio_path = await generate_audio_with_voice_prefix(part, default_voice, rate, pitch, overall_duration_ms, speed_adjustment_factor)
if audio_path:
audio_segments.append(audio_path)
if audio_segments:
combined_audio = AudioSegment.empty()
for segment_path in audio_segments:
try:
segment = AudioSegment.from_mp3(segment_path)
combined_audio += segment
os.remove(segment_path) # Clean up individual segment files
except Exception as e:
print(f"Error loading or combining audio segment {segment_path}: {e}")
return None, None, None
combined_audio_path = f"combined_audio_{start_time_ms}.mp3"
try:
combined_audio.export(combined_audio_path, format="mp3")
return start_time_ms, [combined_audio_path], overall_duration_ms
except Exception as e:
print(f"Error exporting combined audio: {e}")
return None, None, None
return start_time_ms, [], overall_duration_ms # Return empty list if no audio generated
return None, None, None
async def transcript_to_speech(transcript_text, voice, rate, pitch, speed_adjustment_factor):
if not transcript_text.strip():
return None, gr.Warning("Please enter transcript text.")
if not voice:
return None, gr.Warning("Please select a voice.")
lines = transcript_text.strip().split('\n')
timed_audio_segments = []
max_end_time_ms = 0
for i, line in enumerate(lines):
next_line_start_time = None
if i < len(lines) - 1:
next_line_match = re.match(r'(\d{2}):(\d{2}):(\d{2}),(\d{3})\s+.*', lines[i+1])
if next_line_match:
nh, nm, ns, nms = next_line_match.groups()
next_line_start_time = (
int(nh) * 3600000 +
int(nm) * 60000 +
int(ns) * 1000 +
int(nms)
)
current_line_match = re.match(r'(\d{2}):(\d{2}):(\d{2}),(\d{3})\s+(.*)', line)
if current_line_match:
sh, sm, ss, sms, text_content = current_line_match.groups()
start_time_ms = (
int(sh) * 3600000 +
int(sm) * 60000 +
int(ss) * 1000 +
int(sms)
)
overall_duration_ms = None
if next_line_start_time is not None:
overall_duration_ms = next_line_start_time - start_time_ms
start_time, audio_paths, duration = await process_transcript_line(line, next_line_start_time, voice, rate, pitch, overall_duration_ms, speed_adjustment_factor)
if start_time is not None and audio_paths:
combined_line_audio = AudioSegment.empty()
total_generated_duration_ms = 0
for path in audio_paths:
if path:
try:
audio = AudioSegment.from_mp3(path)
combined_line_audio += audio
total_generated_duration_ms += len(audio)
os.remove(path)
except FileNotFoundError:
print(f"Warning: Audio file not found: {path}")
if combined_line_audio and overall_duration_ms is not None and overall_duration_ms > 0 and total_generated_duration_ms > overall_duration_ms:
speed_factor = (total_generated_duration_ms / overall_duration_ms) * speed_adjustment_factor
if speed_factor > 0:
if speed_factor < 1.0:
speed_factor = 1.0
combined_line_audio = combined_line_audio.speedup(playback_speed=speed_factor)
if combined_line_audio:
timed_audio_segments.append({'start': start_time, 'audio': combined_line_audio})
max_end_time_ms = max(max_end_time_ms, start_time + len(combined_line_audio))
elif audio_paths:
for path in audio_paths:
if path:
try:
os.remove(path)
except FileNotFoundError:
pass # Clean up even if no timestamp
if not timed_audio_segments:
return None, "No processable audio segments found."
final_audio = AudioSegment.silent(duration=max_end_time_ms, frame_rate=24000)
for segment in timed_audio_segments:
final_audio = final_audio.overlay(segment['audio'], position=segment['start'])
# Apply the low-pass filter here
cutoff_frequency = 3500 # 3.5 kHz (you can make this a user-configurable parameter later)
filtered_final_audio = apply_low_pass_filter(final_audio, cutoff_frequency, final_audio.frame_rate)
combined_audio_path = tempfile.mktemp(suffix=".mp3")
# Export the *filtered* audio here
filtered_final_audio.export(combined_audio_path, format="mp3")
return combined_audio_path, None
@spaces.GPU
def tts_interface(transcript, voice, rate, pitch, speed_adjustment_factor):
audio, warning = asyncio.run(transcript_to_speech(transcript, voice, rate, pitch, speed_adjustment_factor))
return audio, warning
async def create_demo():
voices = await get_voices()
default_voice = "en-US-AndrewMultilingualNeural - en-US (Male)"
description = """
Process timestamped text (HH:MM:SS,milliseconds) with voice changes within quotes.
The duration for each line is determined by the timestamp of the following line.
The speed of the ENTIRE generated audio for a line will be adjusted to fit within this duration.
If there is no subsequent timestamp, the speed adjustment will be skipped.
You can control the intensity of the speed adjustment using the "Speed Adjustment Factor" slider.
Format: `HH:MM:SS,milliseconds "VoicePrefix Text" more text "AnotherVoicePrefix More Text"`
Example:
```
00:00:00,000 "This is the default voice." more default. "1F Now a female voice." and back to default.
00:00:05,500 "1C Yes," said the child, "it is fun!"
```
***************************************************************************************************
1M = en-AU-WilliamNeural - en-AU (Male)
1F = en-GB-SoniaNeural - en-GB (Female)
2M = en-GB-RyanNeural - en-GB (Male)
2F = en-US-JennyNeural - en-US (Female)
3M = en-US-BrianMultilingualNeural - en-US (Male)
3F = en-HK-YanNeural - en-HK (Female)
4M = en-GB-ThomasNeural - en-GB (Male)
4F = en-US-EmmaNeural - en-US (Female)
1O = en-GB-RyanNeural - en-GB (Male) # Old Man
1C = en-GB-MaisieNeural - en-GB (Female) # Child
1V = vi-VN-HoaiMyNeural - vi-VN (Female) # Vietnamese (Female)
2V = vi-VN-NamMinhNeural - vi-VN (Male) # Vietnamese (Male)
3V = vi-VN-HoaiMyNeural - vi-VN (Female) # Vietnamese (Female)
4V = vi-VN-NamMinhNeural - vi-VN (Male) # Vietnamese (Male)
****************************************************************************************************
"""
demo = gr.Interface(
fn=tts_interface,
inputs=[
gr.Textbox(label="Timestamped Text with Voice Changes and Duration", lines=10, placeholder='00:00:00,000 "Text" more text "1F Different Voice"'),
gr.Dropdown(choices=[""] + list(voices.keys()), label="Select Default Voice", value=default_voice),
gr.Slider(minimum=-50, maximum=50, value=0, label="Speech Rate Adjustment (%)", step=1),
gr.Slider(minimum=-50, maximum=50, value=0, label="Pitch Adjustment (Hz)", step=1),
gr.Slider(minimum=0.5, maximum=1.5, value=1.0, step=0.05, label="Speed Adjustment Factor")
],
outputs=[
gr.Audio(label="Generated Audio", type="filepath"),
gr.Markdown(label="Warning", visible=False)
],
title="TTS with Line-Wide Duration Adjustment and In-Quote Voice Switching",
description=description,
analytics_enabled=False,
allow_flagging=False
)
return demo
if __name__ == "__main__":
demo = asyncio.run(create_demo())
demo.launch()
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