Delete TextGen/coqui.py
Browse files- TextGen/coqui.py +0 -389
TextGen/coqui.py
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import sys
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import io, os, stat
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import subprocess
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import random
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from zipfile import ZipFile
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import uuid
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import time
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import torch
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import torchaudio
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import numpy as np
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#update gradio to faster streaming
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#download for mecab
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os.system('python -m unidic download')
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# By using XTTS you agree to CPML license https://coqui.ai/cpml
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os.environ["COQUI_TOS_AGREED"] = "1"
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# langid is used to detect language for longer text
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# Most users expect text to be their own language, there is checkbox to disable it
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import langid
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import base64
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import csv
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from io import StringIO
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import datetime
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import re
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from scipy.io.wavfile import write
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from pydub import AudioSegment
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from TTS.api import TTS
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from TTS.tts.configs.xtts_config import XttsConfig
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from TTS.tts.models.xtts import Xtts
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from TTS.utils.generic_utils import get_user_data_dir
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HF_TOKEN = os.environ.get("HF_TOKEN")
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from huggingface_hub import HfApi
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# will use api to restart space on a unrecoverable error
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api = HfApi(token=HF_TOKEN)
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repo_id = "coqui/xtts"
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# This will trigger downloading model
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print("Downloading if not downloaded Coqui XTTS V2")
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from TTS.utils.manage import ModelManager
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model_name = "tts_models/multilingual/multi-dataset/xtts_v2"
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ModelManager().download_model(model_name)
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model_path = os.path.join(get_user_data_dir("tts"), model_name.replace("/", "--"))
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print("XTTS downloaded")
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config = XttsConfig()
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config.load_json(os.path.join(model_path, "config.json"))
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model = Xtts.init_from_config(config)
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model.load_checkpoint(
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config,
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checkpoint_path=os.path.join(model_path, "model.pth"),
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vocab_path=os.path.join(model_path, "vocab.json"),
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eval=True,
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use_deepspeed=True,
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)
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model.cuda()
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# This is for debugging purposes only
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DEVICE_ASSERT_DETECTED = 0
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DEVICE_ASSERT_PROMPT = None
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DEVICE_ASSERT_LANG = None
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supported_languages = config.languages
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def numpy_to_mp3(audio_array, sampling_rate):
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# Normalize audio_array if it's floating-point
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if np.issubdtype(audio_array.dtype, np.floating):
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max_val = np.max(np.abs(audio_array))
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audio_array = (audio_array / max_val) * 32767 # Normalize to 16-bit range
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audio_array = audio_array.astype(np.int16)
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# Create an audio segment from the numpy array
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audio_segment = AudioSegment(
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audio_array.tobytes(),
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frame_rate=sampling_rate,
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sample_width=audio_array.dtype.itemsize,
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channels=1
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)
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# Export the audio segment to MP3 bytes - use a high bitrate to maximise quality
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mp3_io = io.BytesIO()
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audio_segment.export(mp3_io, format="mp3", bitrate="320k")
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# Get the MP3 bytes
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mp3_bytes = mp3_io.getvalue()
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mp3_io.close()
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return mp3_bytes
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def predict(
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prompt,
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language,
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audio_file_pth,
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mic_file_path,
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use_mic,
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voice_cleanup,
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no_lang_auto_detect,
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agree,
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):
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if agree == True:
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if language not in supported_languages:
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gr.Warning(
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f"Language you put {language} in is not in is not in our Supported Languages, please choose from dropdown"
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)
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return (
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None,
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)
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language_predicted = langid.classify(prompt)[
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0
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].strip() # strip need as there is space at end!
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# tts expects chinese as zh-cn
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if language_predicted == "zh":
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# we use zh-cn
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language_predicted = "zh-cn"
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print(f"Detected language:{language_predicted}, Chosen language:{language}")
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# After text character length 15 trigger language detection
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if len(prompt) > 15:
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# allow any language for short text as some may be common
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# If user unchecks language autodetection it will not trigger
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# You may remove this completely for own use
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if language_predicted != language and not no_lang_auto_detect:
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# Please duplicate and remove this check if you really want this
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# Or auto-detector fails to identify language (which it can on pretty short text or mixed text)
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gr.Warning(
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f"It looks like your text isn’t the language you chose , if you’re sure the text is the same language you chose, please check disable language auto-detection checkbox"
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)
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return (
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None,
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)
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if use_mic == True:
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if mic_file_path is not None:
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speaker_wav = mic_file_path
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else:
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gr.Warning(
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"Please record your voice with Microphone, or uncheck Use Microphone to use reference audios"
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)
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return (
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None,
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)
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else:
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speaker_wav = audio_file_pth
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# Filtering for microphone input, as it has BG noise, maybe silence in beginning and end
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# This is fast filtering not perfect
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# Apply all on demand
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lowpassfilter = denoise = trim = loudness = True
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if lowpassfilter:
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lowpass_highpass = "lowpass=8000,highpass=75,"
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else:
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lowpass_highpass = ""
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if trim:
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# better to remove silence in beginning and end for microphone
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trim_silence = "areverse,silenceremove=start_periods=1:start_silence=0:start_threshold=0.02,areverse,silenceremove=start_periods=1:start_silence=0:start_threshold=0.02,"
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else:
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trim_silence = ""
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if voice_cleanup:
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try:
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out_filename = (
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speaker_wav + str(uuid.uuid4()) + ".wav"
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) # ffmpeg to know output format
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# we will use newer ffmpeg as that has afftn denoise filter
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shell_command = f"./ffmpeg -y -i {speaker_wav} -af {lowpass_highpass}{trim_silence} {out_filename}".split(
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" "
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)
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command_result = subprocess.run(
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[item for item in shell_command],
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capture_output=False,
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text=True,
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check=True,
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)
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speaker_wav = out_filename
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print("Filtered microphone input")
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except subprocess.CalledProcessError:
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# There was an error - command exited with non-zero code
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print("Error: failed filtering, use original microphone input")
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else:
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speaker_wav = speaker_wav
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if len(prompt) < 2:
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gr.Warning("Please give a longer prompt text")
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return (
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None,
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)
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if len(prompt) > 1000:
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gr.Warning(
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"Text length limited to 200 characters for this demo, please try shorter text. You can clone this space and edit code for your own usage"
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)
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return (
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None,
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)
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global DEVICE_ASSERT_DETECTED
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if DEVICE_ASSERT_DETECTED:
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global DEVICE_ASSERT_PROMPT
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global DEVICE_ASSERT_LANG
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# It will likely never come here as we restart space on first unrecoverable error now
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print(
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f"Unrecoverable exception caused by language:{DEVICE_ASSERT_LANG} prompt:{DEVICE_ASSERT_PROMPT}"
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)
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# HF Space specific.. This error is unrecoverable need to restart space
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space = api.get_space_runtime(repo_id=repo_id)
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if space.stage != "BUILDING":
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api.restart_space(repo_id=repo_id)
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else:
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print("TRIED TO RESTART but space is building")
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try:
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metrics_text = ""
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t_latent = time.time()
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# note diffusion_conditioning not used on hifigan (default mode), it will be empty but need to pass it to model.inference
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try:
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(
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gpt_cond_latent,
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speaker_embedding,
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) = model.get_conditioning_latents(audio_path=speaker_wav, gpt_cond_len=30, gpt_cond_chunk_len=4, max_ref_length=60)
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except Exception as e:
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print("Speaker encoding error", str(e))
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gr.Warning(
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"It appears something wrong with reference, did you unmute your microphone?"
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)
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return (
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None,
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)
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latent_calculation_time = time.time() - t_latent
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# metrics_text=f"Embedding calculation time: {latent_calculation_time:.2f} seconds\n"
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# temporary comma fix
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prompt = re.sub("([^\x00-\x7F]|\w)(\.|\。|\?)", r"\1 \2\2", prompt)
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wav_chunks = []
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## Direct mode
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"""
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print("I: Generating new audio...")
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t0 = time.time()
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out = model.inference(
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prompt,
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language,
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gpt_cond_latent,
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speaker_embedding,
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repetition_penalty=5.0,
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temperature=0.75,
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)
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inference_time = time.time() - t0
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print(f"I: Time to generate audio: {round(inference_time*1000)} milliseconds")
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metrics_text+=f"Time to generate audio: {round(inference_time*1000)} milliseconds\n"
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real_time_factor= (time.time() - t0) / out['wav'].shape[-1] * 24000
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print(f"Real-time factor (RTF): {real_time_factor}")
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metrics_text+=f"Real-time factor (RTF): {real_time_factor:.2f}\n"
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torchaudio.save("output.wav", torch.tensor(out["wav"]).unsqueeze(0), 24000)
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"""
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print("I: Generating new audio in streaming mode...")
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t0 = time.time()
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chunks = model.inference_stream(
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prompt,
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language,
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gpt_cond_latent,
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speaker_embedding,
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repetition_penalty=7.0,
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temperature=0.85,
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)
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first_chunk = True
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for i, chunk in enumerate(chunks):
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if first_chunk:
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first_chunk_time = time.time() - t0
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metrics_text += f"Latency to first audio chunk: {round(first_chunk_time*1000)} milliseconds\n"
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first_chunk = False
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# Convert chunk to numpy array and return it
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chunk_np = chunk.cpu().numpy()
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print('chunk',i)
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yield (24000, chunk_np)
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wav_chunks.append(chunk)
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print(f"Received chunk {i} of audio length {chunk.shape[-1]}")
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inference_time = time.time() - t0
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print(
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f"I: Time to generate audio: {round(inference_time*1000)} milliseconds"
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)
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# metrics_text += (
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# f"Time to generate audio: {round(inference_time*1000)} milliseconds\n"
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#)
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except RuntimeError as e:
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if "device-side assert" in str(e):
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# cannot do anything on cuda device side error, need tor estart
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print(
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f"Exit due to: Unrecoverable exception caused by language:{language} prompt:{prompt}",
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flush=True,
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)
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gr.Warning("Unhandled Exception encounter, please retry in a minute")
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print("Cuda device-assert Runtime encountered need restart")
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if not DEVICE_ASSERT_DETECTED:
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DEVICE_ASSERT_DETECTED = 1
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DEVICE_ASSERT_PROMPT = prompt
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DEVICE_ASSERT_LANG = language
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# just before restarting save what caused the issue so we can handle it in future
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# Uploading Error data only happens for unrecovarable error
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error_time = datetime.datetime.now().strftime("%d-%m-%Y-%H:%M:%S")
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error_data = [
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error_time,
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prompt,
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language,
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audio_file_pth,
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mic_file_path,
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use_mic,
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voice_cleanup,
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no_lang_auto_detect,
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agree,
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]
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error_data = [str(e) if type(e) != str else e for e in error_data]
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print(error_data)
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print(speaker_wav)
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write_io = StringIO()
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csv.writer(write_io).writerows([error_data])
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csv_upload = write_io.getvalue().encode()
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filename = error_time + "_" + str(uuid.uuid4()) + ".csv"
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print("Writing error csv")
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error_api = HfApi()
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error_api.upload_file(
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path_or_fileobj=csv_upload,
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path_in_repo=filename,
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repo_id="coqui/xtts-flagged-dataset",
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repo_type="dataset",
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)
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# speaker_wav
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print("Writing error reference audio")
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speaker_filename = (
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error_time + "_reference_" + str(uuid.uuid4()) + ".wav"
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)
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error_api = HfApi()
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error_api.upload_file(
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path_or_fileobj=speaker_wav,
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path_in_repo=speaker_filename,
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repo_id="coqui/xtts-flagged-dataset",
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repo_type="dataset",
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)
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364 |
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# HF Space specific.. This error is unrecoverable need to restart space
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space = api.get_space_runtime(repo_id=repo_id)
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if space.stage != "BUILDING":
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api.restart_space(repo_id=repo_id)
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else:
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print("TRIED TO RESTART but space is building")
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else:
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if "Failed to decode" in str(e):
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print("Speaker encoding error", str(e))
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gr.Warning(
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"It appears something wrong with reference, did you unmute your microphone?"
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)
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else:
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print("RuntimeError: non device-side assert error:", str(e))
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gr.Warning("Something unexpected happened please retry again.")
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return (
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None,
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)
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384 |
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else:
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gr.Warning("Please accept the Terms & Condition!")
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return (
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None,
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)
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