import os import subprocess import sys def install(package): if '=' in package: package_name, package_version = package.split('==') else: package_name = package package_version = None try: subprocess.check_call([sys.executable, "-m", "pip", "uninstall", "-y", package_name]) print(f"Successfully uninstalled {package}") except subprocess.CalledProcessError: print(f"Package {package} was not installed, proceeding with installation") subprocess.check_call([sys.executable, "-m", "pip", "install", package]) # install('pydantic==2.0.0') # install('gradio==4.44.0') # install('spacy==3.7') debug = False is_prod = True if os.environ.get('PROD_MODE') == 'local': is_prod = False else: debug = False import pickle import gradio as gr import os if not is_prod: import os os.environ['HF_HOME'] = '/proj/afosr/metavoice/cache' os.environ['TRANSFORMERS_CACHE'] = '/proj/afosr/metavoice/cache' os.environ['HF_DATASETS_CACHE'] = '/proj/afosr/metavoice/cache' os.environ['HF_METRICS_CACHE'] = '/proj/afosr/metavoice/cache' os.environ['HF_MODULES_CACHE'] = '/proj/afosr/metavoice/cache' ffmpeg_path = '/home/hc3295/ffmpegg_build/bin' os.environ['PATH'] += os.pathsep + ffmpeg_path import torch if not debug: import shutil import tempfile import time from pathlib import Path import librosa from huggingface_hub import snapshot_download from fam.llm.adapters import FlattenedInterleavedEncodec2Codebook from fam.llm.decoders import EncodecDecoder from fam.llm.fast_inference_utils import build_model, main from fam.llm.inference import ( EncodecDecoder, InferenceConfig, Model, TiltedEncodec, TrainedBPETokeniser, get_cached_embedding, get_cached_file, get_enhancer, ) from fam.llm.utils import ( check_audio_file, get_default_dtype, get_device, normalize_text, ) DESCRIPTION = "" if not torch.cuda.is_available(): DESCRIPTION += "\n
Running on CPU 🥶 This demo does not work on CPU.
" if torch.cuda.is_available(): if not debug: model_name = "metavoiceio/metavoice-1B-v0.1" seed = 1337 output_dir = "outputs" _dtype = get_default_dtype() _device = 'cuda:0' _model_dir = snapshot_download(repo_id=model_name) first_stage_adapter = FlattenedInterleavedEncodec2Codebook(end_of_audio_token=1024) output_dir = output_dir os.makedirs(output_dir, exist_ok=True) second_stage_ckpt_path = f"{_model_dir}/second_stage.pt" config_second_stage = InferenceConfig( ckpt_path=second_stage_ckpt_path, num_samples=1, seed=seed, device=_device, dtype=_dtype, compile=False, init_from="resume", output_dir=output_dir, ) data_adapter_second_stage = TiltedEncodec(end_of_audio_token=1024) llm_second_stage = Model( config_second_stage, TrainedBPETokeniser, EncodecDecoder, data_adapter_fn=data_adapter_second_stage.decode ) enhancer = get_enhancer("df") precision = {"float16": torch.float16, "bfloat16": torch.bfloat16}[_dtype] model, tokenizer, smodel, model_size = build_model( precision=precision, checkpoint_path=Path(f"{_model_dir}/first_stage.pt"), spk_emb_ckpt_path=Path(f"{_model_dir}/speaker_encoder.pt"), device=_device, compile=True, compile_prefill=True, ) def generate_sample(text, emo_dir = None, source_path = None, emo_path = None, neutral_path = None, strength = 0.1, top_p = 0.95, guidance_scale = 3.0, preset_dropdown = None, toggle = None): print('text', text) print('emo_dir', emo_dir) print('source_path', source_path) print('emo_path', emo_path) print('neutral_path', neutral_path) print('strength', strength) print('top_p', top_p) print('guidance_scale', guidance_scale) if toggle == RADIO_CHOICES[0]: source_path = PRESET_VOICES[preset_dropdown] source_path = get_cached_file(source_path) check_audio_file(source_path) source_emb = get_cached_embedding(source_path, smodel).to(device=_device, dtype=precision) if emo_dir == EMO_NAMES[0]: emo_path = get_cached_file(emo_path) check_audio_file(emo_path) emo_emb = get_cached_embedding(emo_path, smodel).to(device=_device, dtype=precision) neutral_path = get_cached_file(neutral_path) check_audio_file(neutral_path) neutral_emb = get_cached_embedding(neutral_path, smodel).to(device=_device, dtype=precision) emo_dir = emo_emb - neutral_emb emo_dir = emo_dir / torch.norm(emo_dir, p=2) else: emo_dir = torch.tensor(ALL_EMO_DIRS[emo_dir], device=_device, dtype=precision) edited_emb = source_emb + strength * emo_dir edited_emb = edited_emb.to(device=_device, dtype=precision) temperature=1.0 text = normalize_text(text) start = time.time() # first stage LLM tokens = main( model=model, tokenizer=tokenizer, model_size=model_size, prompt=text, spk_emb=edited_emb, top_p=torch.tensor(top_p, device=_device, dtype=precision), guidance_scale=torch.tensor(guidance_scale, device=_device, dtype=precision), temperature=torch.tensor(temperature, device=_device, dtype=precision), ) text_ids, extracted_audio_ids = first_stage_adapter.decode([tokens]) b_speaker_embs = edited_emb.unsqueeze(0) # second stage LLM + multi-band diffusion model wav_files = llm_second_stage( texts=[text], encodec_tokens=[torch.tensor(extracted_audio_ids, dtype=torch.int32, device=_device).unsqueeze(0)], speaker_embs=b_speaker_embs, batch_size=1, guidance_scale=None, top_p=None, top_k=200, temperature=1.0, max_new_tokens=None, ) wav_file = wav_files[0] with tempfile.NamedTemporaryFile(suffix=".wav") as enhanced_tmp: enhancer(str(wav_file) + ".wav", enhanced_tmp.name) shutil.copy2(enhanced_tmp.name, str(wav_file) + ".wav") print(f"\nSaved audio to {wav_file}.wav") output_path = str(wav_file) + ".wav" return output_path ALL_EMO_DIRS = pickle.load(open('all_emo_dirs.pkl', 'rb')) EMO_NAMES = ['Upload your own sample'] + list(ALL_EMO_DIRS.keys()) RADIO_CHOICES = ["Preset voices", "Upload your voice"] MAX_CHARS = 220 PRESET_VOICES = { # female "Bria": "https://cdn.themetavoice.xyz/speakers%2Fbria.mp3", # male "Alex": "https://cdn.themetavoice.xyz/speakers/alex.mp3", "Jacob": "https://cdn.themetavoice.xyz/speakers/jacob.wav", } def denormalise_top_p(top_p): # returns top_p in the range [0.9, 1.0] return round(0.9 + top_p / 100, 2) def denormalise_guidance(guidance): # returns guidance in the range [1.0, 3.0] return 1 + ((guidance - 1) * (3 - 1)) / (5 - 1) def _check_file_size(path): if not path: return filesize = os.path.getsize(path) filesize_mb = filesize / 1024 / 1024 if filesize_mb >= 50: raise gr.Error(f"Please upload a sample less than 20MB for voice cloning. Provided: {round(filesize_mb)} MB") def _handle_edge_cases(to_say, upload_target): if not to_say: raise gr.Error("Please provide text to synthesise") if len(to_say) > MAX_CHARS: gr.Warning( f"Max {MAX_CHARS} characters allowed. Provided: {len(to_say)} characters. Truncating and generating speech...Result at the end can be unstable as a result." ) if not upload_target: return check_audio_file(upload_target) # check file duration to be atleast 30s _check_file_size(upload_target) def tts(to_say, top_p, guidance, toggle, preset_dropdown, upload_target): try: d_top_p = denormalise_top_p(top_p) d_guidance = denormalise_guidance(guidance) _handle_edge_cases(to_say, upload_target) to_say = to_say if len(to_say) < MAX_CHARS else to_say[:MAX_CHARS] return TTS_MODEL.synthesise( text=to_say, spk_ref_path=PRESET_VOICES[preset_dropdown] if toggle == RADIO_CHOICES[0] else upload_target, top_p=d_top_p, guidance_scale=d_guidance, ) except Exception as e: raise gr.Error(f"Something went wrong. Reason: {str(e)}") def change_voice_selection_layout(choice): if choice == RADIO_CHOICES[0]: return [gr.update(visible=True), gr.update(visible=False)] return [gr.update(visible=False), gr.update(visible=True)] def change_emotion_selection_layout(choice): if choice == EMO_NAMES[0]: return [gr.update(visible=True), gr.update(visible=True), gr.update(visible=True)] else: return [gr.update(visible=False), gr.update(visible=False), gr.update(visible=False)] title = """