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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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os.system('python -m unidic download') |
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os.environ["COQUI_TOS_AGREED"] = "1" |
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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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api = HfApi(token=HF_TOKEN) |
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repo_id = "coqui/xtts" |
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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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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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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 |
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audio_array = audio_array.astype(np.int16) |
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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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mp3_io = io.BytesIO() |
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audio_segment.export(mp3_io, format="mp3", bitrate="320k") |
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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() |
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if language_predicted == "zh": |
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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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if len(prompt) > 15: |
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if language_predicted != language and not no_lang_auto_detect: |
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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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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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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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) |
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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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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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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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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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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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prompt = re.sub("([^\x00-\x7F]|\w)(\.|\。|\?)", r"\1 \2\2", prompt) |
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wav_chunks = [] |
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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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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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except RuntimeError as e: |
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if "device-side assert" in str(e): |
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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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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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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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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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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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