Update app.py
Browse files
app.py
CHANGED
@@ -7,6 +7,9 @@ import os
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import re
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from pathlib import Path
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from pydub import AudioSegment
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def get_silence(duration_ms=1000):
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# Create silent audio segment with specified parameters
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@@ -14,11 +17,9 @@ def get_silence(duration_ms=1000):
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duration=duration_ms,
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frame_rate=24000 # 24kHz sampling rate
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)
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# Set audio parameters
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silent_audio = silent_audio.set_channels(1) # Mono
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silent_audio = silent_audio.set_sample_width(4) # 32-bit (4 bytes per sample)
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with tempfile.NamedTemporaryFile(delete=False, suffix=".mp3") as tmp_file:
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# Export with specific bitrate and codec parameters
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silent_audio.export(
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@@ -36,220 +37,183 @@ def get_silence(duration_ms=1000):
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# Get all available voices
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async def get_voices():
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async def generate_audio_with_voice_prefix(text_segment, default_voice, rate, pitch):
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"""Generates audio for a text segment, handling voice prefixes."""
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current_voice_full = default_voice
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current_voice_short = current_voice_full.split(" - ")[0] if current_voice_full else ""
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current_rate = rate
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current_pitch = pitch
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processed_text = text_segment.strip()
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current_voice_short = voice1F_short
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current_pitch = 25
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detect=1
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#processed_text = processed_text[2:].strip()
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elif processed_text.startswith("2F"):
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current_voice_short = voice2F_short
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#processed_text = processed_text[2:].strip()
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detect=1
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elif processed_text.startswith("3F"):
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current_voice_short = voice3F_short
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#processed_text = processed_text[2:].strip()
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detect=1
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elif processed_text.startswith("4F"):
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current_voice_short = voice4F_short
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#processed_text = processed_text[2:].strip()
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detect=1
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elif processed_text.startswith("1M"):
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current_voice_short = voice1_short
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#processed_text = processed_text[2:].strip()
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detect=1
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elif processed_text.startswith("2M"):
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current_voice_short = voice2_short
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#processed_text = processed_text[2:].strip()
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detect=1
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elif processed_text.startswith("3M"):
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current_voice_short = voice3_short
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#processed_text = processed_text[2:].strip()
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detect=1
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elif processed_text.startswith("4M"):
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current_voice_short = voice4_short
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#processed_text = processed_text[2:].strip()
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detect=1
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elif processed_text.startswith("1O"): # Old man voice
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current_voice_short = voice5_short
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current_pitch = -20
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current_rate = -10
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#processed_text = processed_text[2:].strip()
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detect=1
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elif processed_text.startswith("1C"): #Child voice
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current_voice_short = voice6_short
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#processed_text = processed_text[2:].strip()
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detect=1
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elif processed_text.startswith("1V"): #Female VN
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current_voice_short = voice7_short
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#processed_text = processed_text[2:].strip()
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detect=1
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elif processed_text.startswith("2V"):
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current_voice_short = voice8_short
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#processed_text = processed_text[2:].strip()
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detect=1
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elif processed_text.startswith("3V"): #Female VN
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current_voice_short = voice9F_short
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current_pitch = 25
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#processed_text = processed_text[2:].strip()
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detect=1
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elif processed_text.startswith("4V"):
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current_voice_short = voice9_short
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current_pitch = -20
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#processed_text = processed_text[2:].strip()
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detect=1
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#Looking for number following prefix, which are pitch values.
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#match = re.search(r'[A-Za-z]\d+', part) # Look for a letter followed by one or more digits
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match = re.search(r'[A-Za-z]+\-?\d+', processed_text) # Look for a letter(s) followed by an optional '-' and digits
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if match:
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if detect:
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processed_text = processed_text[2:]
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if processed_text:
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rate_str = f"{current_rate:+d}%"
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pitch_str = f"{current_pitch:+d}Hz"
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return None
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async def process_transcript_line(line, default_voice, rate, pitch):
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"""Processes a single transcript line with HH:MM:SS
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match = re.match(r'(\d{2}):(\d{2}):(\d{2})
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if match:
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start_time_ms = (
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int(
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int(
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int(
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int(
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)
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audio_segments = []
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split_parts = re.split(r'
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process_next = False
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for part in split_parts:
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if part == '"':
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process_next = not process_next
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continue
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if process_next and part.strip():
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audio_path = await generate_audio_with_voice_prefix(part, default_voice, rate, pitch)
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if audio_path:
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audio_segments.append(audio_path)
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elif not process_next and part.strip():
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audio_path = await generate_audio_with_voice_prefix(part, default_voice, rate, pitch
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if audio_path:
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audio_segments.append(audio_path)
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return None, None
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async def transcript_to_speech(transcript_text, voice, rate, pitch):
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if not transcript_text.strip():
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return None, gr.Warning("Please enter transcript text.")
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if not voice:
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return None, gr.Warning("Please select a voice.")
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lines = transcript_text.strip().split('\n')
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timed_audio_segments = []
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max_end_time_ms = 0
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for line in lines:
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start_time, audio_paths = await process_transcript_line(line, voice, rate, pitch)
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if start_time is not None and audio_paths:
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combined_line_audio = AudioSegment.empty()
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for path in audio_paths:
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if combined_line_audio:
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timed_audio_segments.append({'start': start_time, 'audio': combined_line_audio})
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max_end_time_ms = max(max_end_time_ms, start_time + len(combined_line_audio))
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elif audio_paths:
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for path in audio_paths:
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if not timed_audio_segments:
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return None, "No processable audio segments found."
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final_audio = AudioSegment.silent(duration=max_end_time_ms, frame_rate=24000)
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for segment in timed_audio_segments:
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final_audio = final_audio.overlay(segment['audio'], position=segment['start'])
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combined_audio_path = tempfile.mktemp(suffix=".mp3")
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final_audio.export(combined_audio_path, format="mp3")
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return combined_audio_path, None
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@spaces.GPU
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def tts_interface(transcript, voice, rate, pitch):
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audio, warning = asyncio.run(transcript_to_speech(transcript, voice, rate, pitch))
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return audio, warning
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async def create_demo():
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voices = await get_voices()
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default_voice = "en-US-AndrewMultilingualNeural - en-US (Male)"
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description = """
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Process timestamped text (HH:MM:SS
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Example:
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```
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00:00:00
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00:00:05
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```
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***************************************************************************************************
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1M = en-AU-WilliamNeural - en-AU (Male)
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demo = gr.Interface(
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fn=tts_interface,
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inputs=[
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gr.Textbox(label="Timestamped Text with Voice Changes", lines=10, placeholder='00:00:00
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gr.Dropdown(choices=[""] + list(voices.keys()), label="Select Default Voice", value=default_voice),
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gr.Slider(minimum=-50, maximum=50, value=0, label="Speech Rate Adjustment (%)", step=1),
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gr.Slider(minimum=-50, maximum=50, value=0, label="Pitch Adjustment (Hz)", step=1)
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],
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outputs=[
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gr.Audio(label="Generated Audio", type="filepath"),
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gr.Markdown(label="Warning", visible=False)
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],
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title="TTS with
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description=description,
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analytics_enabled=False,
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allow_flagging=False
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if __name__ == "__main__":
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demo = asyncio.run(create_demo())
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demo.launch()
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import re
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from pathlib import Path
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from pydub import AudioSegment
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import librosa
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import soundfile as sf
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import numpy as np
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def get_silence(duration_ms=1000):
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# Create silent audio segment with specified parameters
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duration=duration_ms,
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frame_rate=24000 # 24kHz sampling rate
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)
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# Set audio parameters
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silent_audio = silent_audio.set_channels(1) # Mono
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silent_audio = silent_audio.set_sample_width(4) # 32-bit (4 bytes per sample)
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with tempfile.NamedTemporaryFile(delete=False, suffix=".mp3") as tmp_file:
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# Export with specific bitrate and codec parameters
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silent_audio.export(
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# Get all available voices
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async def get_voices():
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try:
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voices = await edge_tts.list_voices()
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return {f"{v['ShortName']} - {v['Locale']} ({v['Gender']})": v['ShortName'] for v in voices}
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except Exception as e:
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print(f"Error listing voices: {e}")
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return {}
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async def generate_audio_with_voice_prefix(text_segment, default_voice, rate, pitch, target_duration_ms=None, speed_adjustment_factor=1.0):
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"""Generates audio for a text segment, handling voice prefixes and adjusting rate for duration."""
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current_voice_full = default_voice
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current_voice_short = current_voice_full.split(" - ")[0] if current_voice_full else ""
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current_rate = rate
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current_pitch = pitch
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processed_text = text_segment.strip()
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print(f"Processing this text segment: {processed_text}") # Debug
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voice_map = {
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"1F": "en-GB-SoniaNeural",
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"2M": "en-GB-RyanNeural",
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"3M": "en-US-BrianMultilingualNeural",
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"2F": "en-US-JennyNeural",
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"1M": "en-AU-WilliamNeural",
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"3F": "en-HK-YanNeural",
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"4M": "en-GB-ThomasNeural",
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"4F": "en-US-EmmaNeural",
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"1O": "en-GB-RyanNeural", # Old Man
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"1C": "en-GB-MaisieNeural", # Child
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"1V": "vi-VN-HoaiMyNeural", # Vietnamese (Female)
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"2V": "vi-VN-NamMinhNeural", # Vietnamese (Male)
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"3V": "vi-VN-HoaiMyNeural", # Vietnamese (Female)
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"4V": "vi-VN-NamMinhNeural", # Vietnamese (Male)
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}
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detect = 0
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for prefix, voice_short in voice_map.items():
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if processed_text.startswith(prefix):
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current_voice_short = voice_short
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if prefix in ["1F", "3F", "1V", "3V"]:
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current_pitch = 25
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elif prefix in ["1O", "4V"]:
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current_pitch = -20
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current_rate = -10
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detect = 1
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processed_text = processed_text[len(prefix):].strip()
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break
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match = re.search(r'([A-Za-z]+)-?(\d+)', processed_text)
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if match:
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prefix_pitch = match.group(1)
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number = int(match.group(2))
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if prefix_pitch in voice_map:
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current_pitch += number
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processed_text = re.sub(r'[A-Za-z]+-?\d+', '', processed_text, count=1).strip()
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elif detect:
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processed_text = processed_text.lstrip('-0123456789').strip() # Remove potential leftover numbers
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elif detect:
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processed_text = processed_text[2:].strip()
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if processed_text:
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rate_str = f"{current_rate:+d}%"
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pitch_str = f"{current_pitch:+d}Hz"
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try:
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communicate = edge_tts.Communicate(processed_text, current_voice_short, rate=rate_str, pitch=pitch_str)
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with tempfile.NamedTemporaryFile(delete=False, suffix=".mp3") as tmp_file:
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audio_path = tmp_file.name
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await communicate.save(audio_path)
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if target_duration_ms is not None and os.path.exists(audio_path):
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audio = AudioSegment.from_mp3(audio_path)
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audio_duration_ms = len(audio)
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#print(f"Generated audio duration: {audio_duration_ms}ms, Target duration: {target_duration_ms}ms") # Debug
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if audio_duration_ms > target_duration_ms and target_duration_ms > 0:
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speed_factor = (audio_duration_ms / target_duration_ms) * speed_adjustment_factor
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#print(f"Speed factor (after user adjustment): {speed_factor}") # Debug
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if speed_factor > 0:
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if speed_factor < 1.0:
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speed_factor = 1.0
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y, sr = librosa.load(audio_path, sr=None)
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y_stretched = librosa.effects.time_stretch(y, rate=speed_factor)
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sf.write(audio_path, y_stretched, sr)
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else:
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print("Generated audio is not longer than target duration, no speed adjustment.") # Debug
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return audio_path
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except Exception as e:
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print(f"Edge TTS error processing '{processed_text}': {e}")
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return None
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return None
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async def process_transcript_line(line, default_voice, rate, pitch, speed_adjustment_factor):
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"""Processes a single transcript line with HH:MM:SS,milliseconds - HH:MM:SS,milliseconds timestamp."""
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match = re.match(r'(\d{2}):(\d{2}):(\d{2}),(\d{3})\s+-\s+(\d{2}):(\d{2}):(\d{2}),(\d{3})\s+(.*)', line)
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if match:
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start_h, start_m, start_s, start_ms, end_h, end_m, end_s, end_ms, text_parts = match.groups()
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start_time_ms = (
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int(start_h) * 3600000 +
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int(start_m) * 60000 +
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int(start_s) * 1000 +
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int(start_ms)
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)
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end_time_ms = (
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int(end_h) * 3600000 +
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int(end_m) * 60000 +
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int(end_s) * 1000 +
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int(end_ms)
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)
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duration_ms = end_time_ms - start_time_ms
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audio_segments = []
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142 |
+
split_parts = re.split(r'[“”"]', text_parts)
|
|
|
143 |
process_next = False
|
144 |
for part in split_parts:
|
145 |
if part == '"':
|
146 |
process_next = not process_next
|
147 |
continue
|
148 |
if process_next and part.strip():
|
149 |
+
audio_path = await generate_audio_with_voice_prefix(part, default_voice, rate, pitch, duration_ms, speed_adjustment_factor)
|
150 |
if audio_path:
|
151 |
audio_segments.append(audio_path)
|
152 |
elif not process_next and part.strip():
|
153 |
+
audio_path = await generate_audio_with_voice_prefix(part, default_voice, rate, pitch, duration_ms, speed_adjustment_factor)
|
154 |
if audio_path:
|
155 |
audio_segments.append(audio_path)
|
156 |
+
return start_time_ms, audio_segments, duration_ms
|
157 |
+
return None, None, None
|
158 |
|
159 |
+
async def transcript_to_speech(transcript_text, voice, rate, pitch, speed_adjustment_factor):
|
|
|
|
|
|
|
160 |
if not transcript_text.strip():
|
161 |
return None, gr.Warning("Please enter transcript text.")
|
162 |
if not voice:
|
163 |
return None, gr.Warning("Please select a voice.")
|
|
|
164 |
lines = transcript_text.strip().split('\n')
|
165 |
timed_audio_segments = []
|
166 |
max_end_time_ms = 0
|
|
|
167 |
for line in lines:
|
168 |
+
start_time, audio_paths, duration = await process_transcript_line(line, voice, rate, pitch, speed_adjustment_factor)
|
169 |
if start_time is not None and audio_paths:
|
170 |
combined_line_audio = AudioSegment.empty()
|
171 |
+
current_time_ms = start_time
|
172 |
+
segment_duration = duration / len(audio_paths) if audio_paths else 0
|
173 |
for path in audio_paths:
|
174 |
+
if path: # Only process if audio_path is not None (meaning TTS was successful)
|
175 |
+
try:
|
176 |
+
audio = AudioSegment.from_mp3(path)
|
177 |
+
combined_line_audio += audio
|
178 |
+
os.remove(path)
|
179 |
+
except FileNotFoundError:
|
180 |
+
print(f"Warning: Audio file not found: {path}")
|
181 |
if combined_line_audio:
|
182 |
timed_audio_segments.append({'start': start_time, 'audio': combined_line_audio})
|
183 |
max_end_time_ms = max(max_end_time_ms, start_time + len(combined_line_audio))
|
184 |
elif audio_paths:
|
185 |
for path in audio_paths:
|
186 |
+
if path:
|
187 |
+
try:
|
188 |
+
os.remove(path)
|
189 |
+
except FileNotFoundError:
|
190 |
+
pass # Clean up even if no timestamp
|
191 |
if not timed_audio_segments:
|
192 |
return None, "No processable audio segments found."
|
|
|
193 |
final_audio = AudioSegment.silent(duration=max_end_time_ms, frame_rate=24000)
|
194 |
for segment in timed_audio_segments:
|
195 |
final_audio = final_audio.overlay(segment['audio'], position=segment['start'])
|
|
|
196 |
combined_audio_path = tempfile.mktemp(suffix=".mp3")
|
197 |
final_audio.export(combined_audio_path, format="mp3")
|
198 |
return combined_audio_path, None
|
199 |
|
200 |
@spaces.GPU
|
201 |
+
def tts_interface(transcript, voice, rate, pitch, speed_adjustment_factor):
|
202 |
+
audio, warning = asyncio.run(transcript_to_speech(transcript, voice, rate, pitch, speed_adjustment_factor))
|
203 |
return audio, warning
|
204 |
|
205 |
async def create_demo():
|
206 |
voices = await get_voices()
|
207 |
default_voice = "en-US-AndrewMultilingualNeural - en-US (Male)"
|
208 |
description = """
|
209 |
+
Process timestamped text (HH:MM:SS,milliseconds - HH:MM:SS,milliseconds) with voice changes within quotes.
|
210 |
+
The duration specified in the timestamp will be used to adjust the speech rate so the generated audio fits within that time.
|
211 |
+
You can control the intensity of the speed adjustment using the "Speed Adjustment Factor" slider.
|
212 |
+
Format: `HH:MM:SS,milliseconds - HH:MM:SS,milliseconds "VoicePrefix Text" more text "AnotherVoicePrefix More Text"`
|
213 |
Example:
|
214 |
```
|
215 |
+
00:00:00,000 - 00:00:05,000 "This is the default voice." more default. "1F Now a female voice." and back to default.
|
216 |
+
00:00:05,500 - 00:00:10,250 "1C Yes," said the child, "it is fun!"
|
217 |
```
|
218 |
***************************************************************************************************
|
219 |
1M = en-AU-WilliamNeural - en-AU (Male)
|
|
|
235 |
demo = gr.Interface(
|
236 |
fn=tts_interface,
|
237 |
inputs=[
|
238 |
+
gr.Textbox(label="Timestamped Text with Voice Changes and Duration", lines=10, placeholder='00:00:00,000 - 00:00:05,000 "Text" more text "1F Different Voice"'),
|
239 |
gr.Dropdown(choices=[""] + list(voices.keys()), label="Select Default Voice", value=default_voice),
|
240 |
gr.Slider(minimum=-50, maximum=50, value=0, label="Speech Rate Adjustment (%)", step=1),
|
241 |
+
gr.Slider(minimum=-50, maximum=50, value=0, label="Pitch Adjustment (Hz)", step=1),
|
242 |
+
gr.Slider(minimum=0.5, maximum=1.5, value=1.0, step=0.05, label="Speed Adjustment Factor")
|
243 |
],
|
244 |
outputs=[
|
245 |
gr.Audio(label="Generated Audio", type="filepath"),
|
246 |
gr.Markdown(label="Warning", visible=False)
|
247 |
],
|
248 |
+
title="TTS with Duration-Aware Speed Adjustment and In-Quote Voice Switching",
|
249 |
description=description,
|
250 |
analytics_enabled=False,
|
251 |
allow_flagging=False
|
|
|
254 |
|
255 |
if __name__ == "__main__":
|
256 |
demo = asyncio.run(create_demo())
|
257 |
+
demo.launch()
|