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#!D:\GitDownload\SupThirdParty\audioldm2\venv\Scripts\python.exe | |
import os | |
import torch | |
import logging | |
from audioldm2 import text_to_audio, build_model, save_wave, get_time, read_list | |
import argparse | |
os.environ["TOKENIZERS_PARALLELISM"] = "true" | |
matplotlib_logger = logging.getLogger('matplotlib') | |
matplotlib_logger.setLevel(logging.WARNING) | |
parser = argparse.ArgumentParser() | |
parser.add_argument( | |
"-t", | |
"--text", | |
type=str, | |
required=False, | |
default="", | |
help="Text prompt to the model for audio generation", | |
) | |
parser.add_argument( | |
"--transcription", | |
type=str, | |
required=False, | |
default="", | |
help="Transcription for Text-to-Speech", | |
) | |
parser.add_argument( | |
"-tl", | |
"--text_list", | |
type=str, | |
required=False, | |
default="", | |
help="A file that contains text prompt to the model for audio generation", | |
) | |
parser.add_argument( | |
"-s", | |
"--save_path", | |
type=str, | |
required=False, | |
help="The path to save model output", | |
default="./output", | |
) | |
parser.add_argument( | |
"--model_name", | |
type=str, | |
required=False, | |
help="The checkpoint you gonna use", | |
default="audioldm_48k", | |
choices=["audioldm_48k", "audioldm_16k_crossattn_t5", "audioldm2-full", "audioldm2-music-665k", | |
"audioldm2-full-large-1150k", "audioldm2-speech-ljspeech", "audioldm2-speech-gigaspeech"] | |
) | |
parser.add_argument( | |
"-d", | |
"--device", | |
type=str, | |
required=False, | |
help="The device for computation. If not specified, the script will automatically choose the device based on your environment.", | |
default="auto", | |
) | |
parser.add_argument( | |
"-b", | |
"--batchsize", | |
type=int, | |
required=False, | |
default=1, | |
help="Generate how many samples at the same time", | |
) | |
parser.add_argument( | |
"--ddim_steps", | |
type=int, | |
required=False, | |
default=200, | |
help="The sampling step for DDIM", | |
) | |
parser.add_argument( | |
"-gs", | |
"--guidance_scale", | |
type=float, | |
required=False, | |
default=3.5, | |
help="Guidance scale (Large => better quality and relavancy to text; Small => better diversity)", | |
) | |
parser.add_argument( | |
"-dur", | |
"--duration", | |
type=float, | |
required=False, | |
default=10.0, | |
help="The duration of the samples", | |
) | |
parser.add_argument( | |
"-n", | |
"--n_candidate_gen_per_text", | |
type=int, | |
required=False, | |
default=3, | |
help="Automatic quality control. This number control the number of candidates (e.g., generate three audios and choose the best to show you). A Larger value usually lead to better quality with heavier computation", | |
) | |
parser.add_argument( | |
"--seed", | |
type=int, | |
required=False, | |
default=0, | |
help="Change this value (any integer number) will lead to a different generation result.", | |
) | |
args = parser.parse_args() | |
torch.set_float32_matmul_precision("high") | |
save_path = os.path.join(args.save_path, get_time()) | |
text = args.text | |
random_seed = args.seed | |
duration = args.duration | |
sample_rate = 16000 | |
if ("audioldm2" in args.model_name): | |
print( | |
"Warning: For AudioLDM2 we currently only support 10s of generation. Please use audioldm_48k or audioldm_16k_crossattn_t5 if you want a different duration.") | |
duration = 10 | |
if ("48k" in args.model_name): | |
sample_rate = 48000 | |
guidance_scale = args.guidance_scale | |
n_candidate_gen_per_text = args.n_candidate_gen_per_text | |
transcription = args.transcription | |
if (transcription): | |
if "speech" not in args.model_name: | |
print( | |
"Warning: You choose to perform Text-to-Speech by providing the transcription.However you do not choose the correct model name (audioldm2-speech-gigaspeech or audioldm2-speech-ljspeech).") | |
print("Warning: We will use audioldm2-speech-gigaspeech by default") | |
args.model_name = "audioldm2-speech-gigaspeech" | |
if (not text): | |
print( | |
"Warning: You should provide text as a input to describe the speaker. Use default (A male reporter is speaking)") | |
text = "A female reporter is speaking full of emotion" | |
os.makedirs(save_path, exist_ok=True) | |
audioldm2 = build_model(model_name=args.model_name, device=args.device) | |
if (args.text_list): | |
print("Generate audio based on the text prompts in %s" % args.text_list) | |
prompt_todo = read_list(args.text_list) | |
else: | |
prompt_todo = [text] | |
for text in prompt_todo: | |
if ("|" in text): | |
text, name = text.split("|") | |
else: | |
name = text[:128] | |
if (transcription): | |
name += "-TTS-%s" % transcription | |
waveform = text_to_audio( | |
audioldm2, | |
text, | |
transcription=transcription, # To avoid the model to ignore the last vocab | |
seed=random_seed, | |
duration=duration, | |
guidance_scale=guidance_scale, | |
ddim_steps=args.ddim_steps, | |
n_candidate_gen_per_text=n_candidate_gen_per_text, | |
batchsize=args.batchsize, | |
) | |
save_wave(waveform, save_path, name=name, samplerate=sample_rate) | |