Update app.py
Browse files
app.py
CHANGED
@@ -7,38 +7,6 @@ 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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import librosa
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import soundfile as sf
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import numpy as np
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from pydub import AudioSegment
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from pydub.playback import play
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from scipy.signal import butter, lfilter # Ensure this line is present
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def apply_low_pass_filter(audio_segment, cutoff_freq, sample_rate, order=5):
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"""Applies a low-pass filter to a pydub AudioSegment."""
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audio_np = np.array(audio_segment.get_array_of_samples()).astype(np.float32) / (2**15 - 1)
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if audio_segment.channels == 2:
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audio_np = audio_np.reshape(-1, 2)
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nyquist_freq = 0.5 * sample_rate
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normalized_cutoff = cutoff_freq / nyquist_freq
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b, a = butter(order, normalized_cutoff, btype='low', analog=False)
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filtered_data = np.zeros_like(audio_np, dtype=np.float32)
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if audio_segment.channels == 1:
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filtered_data = lfilter(b, a, audio_np)
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else:
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for channel in range(audio_segment.channels):
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filtered_data[:, channel] = lfilter(b, a, audio_np[:, channel])
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filtered_data_int16 = (filtered_data * (2**15 - 1)).astype(np.int16)
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filtered_audio = AudioSegment(filtered_data_int16.tobytes(),
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frame_rate=sample_rate,
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sample_width=audio_segment.sample_width,
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channels=audio_segment.channels)
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return filtered_audio
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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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@@ -46,9 +14,11 @@ 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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@@ -66,212 +36,193 @@ 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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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
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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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if match:
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#
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current_pitch +=
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#
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processed_text =
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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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await communicate.save(audio_path)
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if os.path.exists(audio_path):
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audio = AudioSegment.from_mp3(audio_path)
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# Trim leading and trailing silence
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def detect_leading_silence(sound, silence_threshold=-50.0, chunk_size=10):
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trim_ms = 0
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assert chunk_size > 0 # to avoid infinite loop
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while sound[trim_ms:trim_ms+chunk_size].dBFS < silence_threshold and trim_ms < len(sound):
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trim_ms += chunk_size
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return trim_ms
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start_trim = detect_leading_silence(audio)
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end_trim = detect_leading_silence(audio.reverse())
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trimmed_audio = audio[start_trim:len(audio)-end_trim]
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trimmed_audio.export(audio_path, format="mp3") # Overwrite with trimmed version
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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,
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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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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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for segment_path in audio_segments:
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try:
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segment = AudioSegment.from_mp3(segment_path)
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combined_audio += segment
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os.remove(segment_path) # Clean up individual segment files
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except Exception as e:
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print(f"Error loading or combining audio segment {segment_path}: {e}")
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return None, None, None
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combined_audio_path = f"combined_audio_{start_time_ms}.mp3"
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try:
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combined_audio.export(combined_audio_path, format="mp3")
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return start_time_ms, [combined_audio_path], overall_duration_ms
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except Exception as e:
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print(f"Error exporting combined audio: {e}")
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return None, None, None
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return start_time_ms, [], overall_duration_ms # Return empty list if no audio generated
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async def transcript_to_speech(transcript_text, voice, rate, pitch, speed_adjustment_factor):
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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
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if
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)
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overall_duration_ms = None
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if next_line_start_time is not None:
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overall_duration_ms = next_line_start_time - start_time_ms
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start_time, audio_paths, duration = await process_transcript_line(line, next_line_start_time, voice, rate, pitch, overall_duration_ms, speed_adjustment_factor)
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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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total_generated_duration_ms = 0
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for path in audio_paths:
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if path:
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try:
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audio = AudioSegment.from_mp3(path)
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combined_line_audio += audio
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total_generated_duration_ms += len(audio)
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os.remove(path)
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except FileNotFoundError:
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print(f"Warning: Audio file not found: {path}")
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if combined_line_audio and overall_duration_ms is not None and overall_duration_ms > 0 and total_generated_duration_ms > overall_duration_ms:
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speed_factor = (total_generated_duration_ms / overall_duration_ms) * speed_adjustment_factor
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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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combined_line_audio = combined_line_audio.speedup(playback_speed=speed_factor)
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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 path:
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try:
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os.remove(path)
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except FileNotFoundError:
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pass # Clean up even if no timestamp
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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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# Apply the low-pass filter here
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cutoff_frequency = 3500 # 3.5 kHz (you can make this a user-configurable parameter later)
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filtered_final_audio = apply_low_pass_filter(final_audio, cutoff_frequency, final_audio.frame_rate)
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combined_audio_path = tempfile.mktemp(suffix=".mp3")
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filtered_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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The speed of the ENTIRE generated audio for a line will be adjusted to fit within this duration.
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If there is no subsequent timestamp, the speed adjustment will be skipped.
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You can control the intensity of the speed adjustment using the "Speed Adjustment Factor" slider.
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Format: `HH:MM:SS,milliseconds "VoicePrefix Text" more text "AnotherVoicePrefix More Text"`
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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
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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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gr.Slider(minimum=0.5, maximum=1.5, value=1.0, step=0.05, label="Speed Adjustment Factor")
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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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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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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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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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voice1_full = "en-AU-WilliamNeural - en-AU (Male)"
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voice1_short = voice1_full.split(" - ")[0]
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voice1F_full ="en-GB-SoniaNeural - en-GB (Female)"
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voice1F_short = voice1F_full.split(" - ")[0]
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voice2_full = "en-GB-RyanNeural - en-GB (Male)"
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voice2_short = voice2_full.split(" - ")[0]
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voice2F_full = "en-US-JennyNeural - en-US (Female)"
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voice2F_short = voice2F_full.split(" - ")[0]
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voice3_full ="en-US-BrianMultilingualNeural - en-US (Male)" #good for reading
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voice3_short = voice3_full.split(" - ")[0]
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voice3F_full = "en-HK-YanNeural - en-HK (Female)"
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voice3F_short = voice3F_full.split(" - ")[0]
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voice4_full = "en-GB-ThomasNeural - en-GB (Male)"
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voice4_short = voice4_full.split(" - ")[0]
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voice4F_full ="en-US-EmmaNeural - en-US (Female)"
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voice4F_short = voice4F_full.split(" - ")[0]
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voice5_full = "en-GB-RyanNeural - en-GB (Male)" #Old Man
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voice5_short = voice5_full.split(" - ")[0]
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voice6_full = "en-GB-MaisieNeural - en-GB (Female)" #Child
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voice6_short = voice6_full.split(" - ")[0]
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voice7_full = "vi-VN-HoaiMyNeural - vi-VN (Female)" #Vietnamese
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voice7_short = voice7_full.split(" - ")[0]
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voice8_full = "vi-VN-NamMinhNeural - vi-VN (Male)" #Vietnamese
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voice8_short = voice8_full.split(" - ")[0]
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voice9F_full = "de-DE-SeraphinaMultilingualNeural - de-DE (Female)" #Vietnamese
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voice9F_short = voice7_full.split(" - ")[0]
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voice9_full = "ko-KR-HyunsuMultilingualNeural - ko-KR (Male)" #Vietnamese
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voice9_short = voice8_full.split(" - ")[0]
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detect=0
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if processed_text.startswith("1F"):
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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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84 |
+
current_voice_short = voice2F_short
|
85 |
+
#processed_text = processed_text[2:].strip()
|
86 |
+
detect=1
|
87 |
+
elif processed_text.startswith("3F"):
|
88 |
+
current_voice_short = voice3F_short
|
89 |
+
#processed_text = processed_text[2:].strip()
|
90 |
+
detect=1
|
91 |
+
elif processed_text.startswith("4F"):
|
92 |
+
current_voice_short = voice4F_short
|
93 |
+
#processed_text = processed_text[2:].strip()
|
94 |
+
detect=1
|
95 |
+
elif processed_text.startswith("1M"):
|
96 |
+
current_voice_short = voice1_short
|
97 |
+
#processed_text = processed_text[2:].strip()
|
98 |
+
detect=1
|
99 |
+
elif processed_text.startswith("2M"):
|
100 |
+
current_voice_short = voice2_short
|
101 |
+
#processed_text = processed_text[2:].strip()
|
102 |
+
detect=1
|
103 |
+
elif processed_text.startswith("3M"):
|
104 |
+
current_voice_short = voice3_short
|
105 |
+
#processed_text = processed_text[2:].strip()
|
106 |
+
detect=1
|
107 |
+
elif processed_text.startswith("4M"):
|
108 |
+
current_voice_short = voice4_short
|
109 |
+
#processed_text = processed_text[2:].strip()
|
110 |
+
detect=1
|
111 |
+
elif processed_text.startswith("1O"): # Old man voice
|
112 |
+
current_voice_short = voice5_short
|
113 |
+
current_pitch = -20
|
114 |
+
current_rate = -10
|
115 |
+
#processed_text = processed_text[2:].strip()
|
116 |
+
detect=1
|
117 |
+
elif processed_text.startswith("1C"): #Child voice
|
118 |
+
current_voice_short = voice6_short
|
119 |
+
#processed_text = processed_text[2:].strip()
|
120 |
+
detect=1
|
121 |
+
elif processed_text.startswith("1V"): #Female VN
|
122 |
+
current_voice_short = voice7_short
|
123 |
+
#processed_text = processed_text[2:].strip()
|
124 |
+
detect=1
|
125 |
+
elif processed_text.startswith("2V"):
|
126 |
+
current_voice_short = voice8_short
|
127 |
+
#processed_text = processed_text[2:].strip()
|
128 |
+
detect=1
|
129 |
+
elif processed_text.startswith("3V"): #Female VN
|
130 |
+
current_voice_short = voice9F_short
|
131 |
+
current_pitch = 25
|
132 |
+
#processed_text = processed_text[2:].strip()
|
133 |
+
detect=1
|
134 |
+
elif processed_text.startswith("4V"):
|
135 |
+
current_voice_short = voice9_short
|
136 |
+
current_pitch = -20
|
137 |
+
#processed_text = processed_text[2:].strip()
|
138 |
+
detect=1
|
139 |
+
#Looking for number following prefix, which are pitch values.
|
140 |
+
#match = re.search(r'[A-Za-z]\d+', part) # Look for a letter followed by one or more digits
|
141 |
+
match = re.search(r'[A-Za-z]+\-?\d+', processed_text) # Look for a letter(s) followed by an optional '-' and digits
|
142 |
if match:
|
143 |
+
# Extract the prefix (e.g., '2F') and number (e.g., '-20')
|
144 |
+
prefix = ''.join([ch for ch in match.group() if ch.isalpha()]) # Extract letters (prefix)
|
145 |
+
number = int(''.join([ch for ch in match.group() if ch.isdigit() or ch == '-'])) # Extract digits (number)
|
146 |
+
current_pitch += number
|
147 |
+
# Step 2: Remove the found number from the string
|
148 |
+
new_text = re.sub(r'[A-Za-z]+\-?\d+', '', processed_text, count=1).strip() # Remove prefix and number (e.g., '2F-20')
|
149 |
+
#processed_text = new_text[2:] #cut out the prefix like 1F, 3M etc
|
150 |
+
processed_text = new_text[len(prefix):] # Dynamically remove the prefix part
|
151 |
+
else:
|
152 |
+
if detect:
|
153 |
+
processed_text = processed_text[2:]
|
154 |
if processed_text:
|
155 |
rate_str = f"{current_rate:+d}%"
|
156 |
pitch_str = f"{current_pitch:+d}Hz"
|
157 |
+
communicate = edge_tts.Communicate(processed_text, current_voice_short, rate=rate_str, pitch=pitch_str)
|
158 |
+
with tempfile.NamedTemporaryFile(delete=False, suffix=".mp3") as tmp_file:
|
159 |
+
audio_path = tmp_file.name
|
160 |
+
await communicate.save(audio_path)
|
161 |
+
return audio_path
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
162 |
return None
|
163 |
|
164 |
+
async def process_transcript_line(line, default_voice, rate, pitch):
|
165 |
+
"""Processes a single transcript line with HH:MM:SS.milliseconds timestamp and quoted text segments."""
|
166 |
+
match = re.match(r'(\d{2}):(\d{2}):(\d{2})\.(\d{3})\s+(.*)', line)
|
167 |
if match:
|
168 |
+
hours, minutes, seconds, milliseconds, text_parts = match.groups()
|
169 |
start_time_ms = (
|
170 |
+
int(hours) * 3600000 +
|
171 |
+
int(minutes) * 60000 +
|
172 |
+
int(seconds) * 1000 +
|
173 |
+
int(milliseconds)
|
174 |
)
|
175 |
audio_segments = []
|
176 |
+
split_parts = re.split(r'(")', text_parts) # Split by quote marks, keeping the quotes
|
177 |
+
|
178 |
process_next = False
|
179 |
for part in split_parts:
|
180 |
if part == '"':
|
181 |
process_next = not process_next
|
182 |
continue
|
183 |
if process_next and part.strip():
|
184 |
+
audio_path = await generate_audio_with_voice_prefix(part, default_voice, rate, pitch)
|
185 |
if audio_path:
|
186 |
audio_segments.append(audio_path)
|
187 |
elif not process_next and part.strip():
|
188 |
+
audio_path = await generate_audio_with_voice_prefix(part, default_voice, rate, pitch) # Process unquoted text with default voice
|
189 |
if audio_path:
|
190 |
audio_segments.append(audio_path)
|
191 |
|
192 |
+
return start_time_ms, audio_segments
|
193 |
+
return None, None
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
194 |
|
195 |
+
async def transcript_to_speech(transcript_text, voice, rate, pitch):
|
|
|
|
|
196 |
if not transcript_text.strip():
|
197 |
return None, gr.Warning("Please enter transcript text.")
|
198 |
if not voice:
|
199 |
return None, gr.Warning("Please select a voice.")
|
200 |
+
|
201 |
lines = transcript_text.strip().split('\n')
|
202 |
timed_audio_segments = []
|
203 |
max_end_time_ms = 0
|
204 |
|
205 |
+
for line in lines:
|
206 |
+
start_time, audio_paths = await process_transcript_line(line, voice, rate, pitch)
|
207 |
+
if start_time is not None and audio_paths:
|
208 |
+
combined_line_audio = AudioSegment.empty()
|
209 |
+
for path in audio_paths:
|
210 |
+
try:
|
211 |
+
audio = AudioSegment.from_mp3(path)
|
212 |
+
combined_line_audio += audio
|
213 |
+
os.remove(path)
|
214 |
+
except FileNotFoundError:
|
215 |
+
print(f"Warning: Audio file not found: {path}")
|
216 |
+
|
217 |
+
if combined_line_audio:
|
218 |
+
timed_audio_segments.append({'start': start_time, 'audio': combined_line_audio})
|
219 |
+
max_end_time_ms = max(max_end_time_ms, start_time + len(combined_line_audio))
|
220 |
+
elif audio_paths:
|
221 |
+
for path in audio_paths:
|
222 |
+
try:
|
223 |
+
os.remove(path)
|
224 |
+
except FileNotFoundError:
|
225 |
+
pass # Clean up even if no timestamp
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
226 |
|
227 |
if not timed_audio_segments:
|
228 |
return None, "No processable audio segments found."
|
|
|
230 |
final_audio = AudioSegment.silent(duration=max_end_time_ms, frame_rate=24000)
|
231 |
for segment in timed_audio_segments:
|
232 |
final_audio = final_audio.overlay(segment['audio'], position=segment['start'])
|
|
|
|
|
|
|
|
|
233 |
|
234 |
combined_audio_path = tempfile.mktemp(suffix=".mp3")
|
235 |
+
final_audio.export(combined_audio_path, format="mp3")
|
|
|
236 |
return combined_audio_path, None
|
237 |
|
238 |
@spaces.GPU
|
239 |
+
def tts_interface(transcript, voice, rate, pitch):
|
240 |
+
audio, warning = asyncio.run(transcript_to_speech(transcript, voice, rate, pitch))
|
241 |
return audio, warning
|
242 |
|
243 |
async def create_demo():
|
244 |
voices = await get_voices()
|
245 |
default_voice = "en-US-AndrewMultilingualNeural - en-US (Male)"
|
246 |
description = """
|
247 |
+
Process timestamped text (HH:MM:SS.milliseconds) with voice changes within quotes.
|
248 |
+
Format: `HH:MM:SS.milliseconds "VoicePrefix Text" more text "AnotherVoicePrefix More Text"`
|
|
|
|
|
|
|
|
|
249 |
Example:
|
250 |
```
|
251 |
+
00:00:00.000 "This is the default voice." more default. "1F Now a female voice." and back to default.
|
252 |
+
00:00:05.000 "1C Yes," said the child, "it is fun!"
|
253 |
```
|
254 |
***************************************************************************************************
|
255 |
1M = en-AU-WilliamNeural - en-AU (Male)
|
|
|
271 |
demo = gr.Interface(
|
272 |
fn=tts_interface,
|
273 |
inputs=[
|
274 |
+
gr.Textbox(label="Timestamped Text with Voice Changes", lines=10, placeholder='00:00:00.000 "Text" more text "1F Different Voice"'),
|
275 |
gr.Dropdown(choices=[""] + list(voices.keys()), label="Select Default Voice", value=default_voice),
|
276 |
gr.Slider(minimum=-50, maximum=50, value=0, label="Speech Rate Adjustment (%)", step=1),
|
277 |
+
gr.Slider(minimum=-50, maximum=50, value=0, label="Pitch Adjustment (Hz)", step=1)
|
|
|
278 |
],
|
279 |
outputs=[
|
280 |
gr.Audio(label="Generated Audio", type="filepath"),
|
281 |
gr.Markdown(label="Warning", visible=False)
|
282 |
],
|
283 |
+
title="TTS with HH:MM:SS.milliseconds and In-Quote Voice Switching",
|
284 |
description=description,
|
285 |
analytics_enabled=False,
|
286 |
allow_flagging=False
|
|
|
289 |
|
290 |
if __name__ == "__main__":
|
291 |
demo = asyncio.run(create_demo())
|
292 |
+
demo.launch()
|