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import dataclasses | |
import hashlib | |
import json | |
import os | |
import pathlib | |
import shutil | |
import subprocess | |
import tempfile | |
import time | |
from contextlib import nullcontext | |
from dataclasses import dataclass | |
from typing import List, Literal, Optional, Tuple, Type, Union | |
import torch | |
import tqdm | |
import tqdm.contrib.concurrent | |
import tyro | |
from huggingface_hub import snapshot_download | |
from fam.llm.adapters import FlattenedInterleavedEncodec2Codebook, TiltedEncodec | |
from fam.llm.decoders import Decoder, EncodecDecoder | |
from fam.llm.enhancers import BaseEnhancer, get_enhancer | |
from fam.llm.model import GPT, GPTConfig | |
from fam.llm.utils import check_audio_file, get_default_dtype, normalize_text | |
from fam.quantiser.audio.speaker_encoder.model import SpeakerEncoder | |
from fam.quantiser.text.tokenise import TrainedBPETokeniser | |
class InferenceConfig: | |
ckpt_path: str # path to checkpoint | |
output_dir: str | |
num_samples: int = 10 # number of samples to draw | |
seed: int = 1337 # random seed | |
device: str = "cuda" | |
dtype: str = "bfloat16" | |
compile: bool = False | |
init_from: str = "resume" # either 'resume' (from an out_dir) or a gpt2 variant (e.g. 'gpt2-xl') | |
def __str__(self): | |
field_strs = [] | |
for field in dataclasses.fields(self): | |
value = getattr(self, field.name) | |
field_strs.append(f" {field.name}: {value}") | |
return "InferenceConfig:\n" + "\n".join(field_strs) | |
class Model: | |
def __init__( | |
self, | |
config: InferenceConfig, | |
tokenizer_cls: Type[TrainedBPETokeniser], | |
decoder_cls: Type[Decoder], | |
data_adapter_fn, | |
use_kv_cache: Optional[Literal["vanilla"]] = None, | |
): | |
# TODO: disentangle the encodec stuff and numbers etc with rest of this code (esp at encoder-only / second stage model inference) | |
# TODO: remove magic number | |
self._encodec_codes_pad_token = 1024 | |
self._num_encodec_codebooks = 8 | |
self.config = config | |
self.use_kv_cache = use_kv_cache | |
torch.manual_seed(config.seed) | |
torch.cuda.manual_seed(config.seed) | |
torch.backends.cuda.matmul.allow_tf32 = True if config.dtype != "float32" else False # allow tf32 on matmul | |
torch.backends.cudnn.allow_tf32 = True if config.dtype != "float32" else False # allow tf32 on cudnn | |
device_type = "cuda" if "cuda" in config.device else "cpu" # for later use in torch.autocast | |
self.ptdtype = { | |
"float32": torch.float32, | |
"tfloat32": torch.float32, | |
"bfloat16": torch.bfloat16, | |
"float16": torch.float16, | |
}[config.dtype] | |
self._ctx = ( | |
nullcontext() if device_type == "cpu" else torch.amp.autocast(device_type=device_type, dtype=self.ptdtype) | |
) | |
self.use_bpe_tokenizer = False | |
self.load_meta = None | |
self.speaker_cond = None | |
self.meta = None | |
self.model = None | |
self.checkpoint_config = None | |
self.vocab_sizes = None | |
self.smodel = None | |
self._init_model() | |
self.tokenizer = tokenizer_cls(**self.meta["tokenizer"]) | |
self.decoder = decoder_cls( | |
tokeniser_decode_fn=self.tokenizer.decode, | |
output_dir=self.config.output_dir, | |
data_adapter_fn=data_adapter_fn, | |
) | |
def _init_model(self): | |
if self.config.init_from == "resume": | |
# init from a model saved in a specific directory | |
checkpoint = torch.load(self.config.ckpt_path, map_location=self.config.device) | |
self.vocab_sizes = checkpoint["model_args"]["vocab_sizes"] | |
self.load_meta = False | |
self.speaker_cond = False | |
if "config" in checkpoint: | |
self.checkpoint_config = checkpoint["config"] | |
self.meta = checkpoint["meta"] | |
load_meta = True | |
if load_meta: | |
self.use_bpe_tokenizer = "stoi" not in self.meta or "itos" not in self.meta | |
self.speaker_cond = self.meta.get("speaker_cond") | |
if self.speaker_cond: | |
speaker_emb_size = self.meta["speaker_emb_size"] | |
model_args = checkpoint["model_args"] | |
if "causal" in self.checkpoint_config and self.checkpoint_config["causal"] is False: | |
self._encodec_ctx_window = model_args["block_size"] | |
gptconf = GPTConfig(**model_args) | |
# TODO: rename `speaker_emb_dim` to `speaker_emb_size`. | |
self.model = GPT(gptconf, speaker_emb_dim=speaker_emb_size if self.speaker_cond else None) | |
state_dict = checkpoint["model"] | |
unwanted_prefix = "_orig_mod." | |
for k, v in list(state_dict.items()): | |
if k.startswith(unwanted_prefix): | |
state_dict[k[len(unwanted_prefix) :]] = state_dict.pop(k) | |
self.model.load_state_dict(state_dict) | |
# model | |
self.model.eval() | |
self.model.to(self.config.device) | |
if self.config.compile: | |
from einops._torch_specific import allow_ops_in_compiled_graph | |
allow_ops_in_compiled_graph() | |
self.model = torch.compile(self.model) # type: ignore | |
if self.use_kv_cache is not None: | |
if "causal" in self.checkpoint_config and self.checkpoint_config["causal"] is False: | |
raise Exception("kv_cache not supported for non-causal models!") | |
if self.use_kv_cache == "vanilla": | |
self.model.enable_kv_cache() | |
else: | |
raise NotImplementedError(f"kv_cache type {self.use_kv_cache} not implemented!") | |
def causal_sample( | |
self, | |
*, | |
texts: list[str], | |
batch_size: int, | |
max_new_tokens: int, | |
temperature: Optional[float], | |
top_k: Optional[int], | |
top_p: Optional[float], | |
speaker_embs: Optional[torch.Tensor] = None, | |
guidance_scale: Optional[float] = None, | |
) -> list[torch.Tensor]: | |
""" | |
Returns list of torch.Tensors of tokens. Each tensor is of shape (1, c, t) where c is the number of codebooks. | |
Any flattening / inteleaving / tilting gets reversed before the output is returned. | |
""" | |
if speaker_embs is not None: | |
assert len(texts) == len(speaker_embs) | |
encoded_texts = [self.tokenizer.encode(text) for text in texts] | |
## create multiple hierarchies and get seq_lens | |
seq_lens = [] | |
xs = [] | |
for i, encoded_text in enumerate(encoded_texts): | |
encoded_text = torch.tensor([encoded_text], dtype=torch.long, device=self.config.device) | |
# TODO: remove magic number | |
xs.append( | |
torch.cat( | |
# [1st hierarchy of text, *remaining hierarchies of padded tokens] | |
# TODO: self.vocab_sizes should be from the model config? | |
[encoded_text, *[torch.ones_like(encoded_text) * 1024] * (len(self.vocab_sizes) - 1)], | |
dim=0, | |
).unsqueeze(0) | |
) # b x [(b=1, c, t)] | |
seq_lens.append(xs[-1].shape[-1]) | |
max_len = max(seq_lens) | |
assert len(xs) == len(seq_lens) | |
## equalise the shapes in the batch. we can use torch.zeros as tokens > seq_lens will be masked out. | |
x = torch.zeros((len(encoded_texts), xs[0].shape[1], max_len), dtype=torch.long, device=self.config.device) | |
for i, _xs in enumerate(xs): | |
assert _xs.shape[-1] == seq_lens[i] | |
x[i, :, : seq_lens[i]] = _xs | |
## check that the input is correct | |
for i in range(x.shape[0]): | |
assert x[i, 0, : seq_lens[i]].tolist() == encoded_texts[i] | |
# TODO: remove magic number | |
if x.shape[1] > 1: | |
assert set(x[i, 1, : seq_lens[i]].tolist()) == set([1024]) | |
assert x.shape[0] == speaker_embs.shape[0] if speaker_embs is not None else True | |
if self.speaker_cond is False: | |
speaker_embs = None | |
# run sampling loop | |
with torch.no_grad(): | |
with self._ctx: # type: ignore | |
to_return = [] | |
for k in range(self.config.num_samples): | |
assert seq_lens is not None | |
assert batch_size is not None | |
if max(seq_lens) + max_new_tokens >= self.model.config.block_size: | |
raise Exception( | |
f"max_new_tokens {max_new_tokens} too large! Choose {self.model.config.block_size - max(seq_lens) - 1} instead." | |
) | |
y = self.model.generate( | |
x, | |
max_new_tokens, | |
seq_lens=seq_lens, | |
temperature=temperature, | |
top_k=top_k, | |
top_p=top_p, | |
speaker_embs=speaker_embs, | |
batch_size=batch_size, | |
guidance_scale=guidance_scale, | |
dtype=self.ptdtype, | |
end_of_audio_token=self.tokenizer.offset - 1, | |
end_of_text_token=self.tokenizer.eot_token, | |
) | |
for i in range(len(y)): | |
to_return.append(self.decoder.decode(tokens=y[i].tolist(), causal=True)) | |
return to_return | |
def non_causal_sample( | |
self, | |
*, | |
texts: list[str], | |
encodec_tokens: list[torch.Tensor], | |
batch_size: int, | |
top_k: Optional[int], | |
temperature: Optional[float], | |
speaker_embs: Optional[torch.Tensor] = None, | |
) -> list[str]: | |
""" | |
Returns paths to saved audio files. | |
""" | |
if speaker_embs is not None: | |
assert len(texts) == len(speaker_embs) | |
encoded_texts = [self.tokenizer.encode(text) for text in texts] | |
# setup input | |
# TODO: same code is used during data prep. refactor | |
padded_hierarchies_inputs = [] | |
for encoded_text, encodec_token in zip(encoded_texts, encodec_tokens): | |
x = torch.tensor(encoded_text, dtype=torch.long, device=self.config.device)[ | |
None, None, ... | |
] # (b=1, c=1, t) | |
# TODO: should only happen if decoder is encodecdeocder? | |
assert encodec_token.shape[0] == 1 | |
encodec_token = encodec_token[0].tolist() # (b=1, c, t) -> (c, t) | |
assert len(encodec_token) >= 1 and len(encodec_token) <= self._num_encodec_codebooks | |
## setup hierarchies of tokens | |
# TODO: refactor and merge with code in processing.py | |
text_tokens = encoded_text # (t,) | |
hierarchies_in = [] | |
hierarchies_in.append(text_tokens + encodec_token[0] + [self._encodec_codes_pad_token]) | |
hierarchies_in.append( | |
[self._encodec_codes_pad_token] * len(text_tokens) + encodec_token[1] + [self._encodec_codes_pad_token] | |
) | |
## adding padding / cutting to the right size as needed | |
# TODO: refactor and merge with code in processing.py | |
padded_hierarchies_input = [] | |
for _, t_hierarchy in enumerate(hierarchies_in): | |
assert len(t_hierarchy) == len(hierarchies_in[0]) | |
if len(t_hierarchy) < self._encodec_ctx_window: | |
padded_hierarchies_input.append( | |
t_hierarchy + [self._encodec_codes_pad_token] * (self._encodec_ctx_window - len(t_hierarchy)) | |
) | |
elif len(t_hierarchy) > self._encodec_ctx_window: | |
padded_hierarchies_input.append(t_hierarchy[: self._encodec_ctx_window]) | |
else: | |
padded_hierarchies_input.append(t_hierarchy) | |
padded_hierarchies_inputs.append(padded_hierarchies_input) | |
## check that the input is correct | |
in_x = torch.tensor(padded_hierarchies_inputs, dtype=torch.long, device=self.config.device) | |
assert in_x.shape[0] == speaker_embs.shape[0] if speaker_embs is not None else True | |
if self.speaker_cond is False: | |
speaker_embs = None | |
# run sampling loop | |
with torch.no_grad(): | |
with self._ctx: # type: ignore | |
to_return = [] | |
for k in range(self.config.num_samples): | |
y = self.model.generate( | |
in_x, | |
None, | |
temperature=temperature, | |
top_k=top_k, | |
# TODO: handle separate top_p for this model explicitly | |
top_p=None, | |
speaker_embs=speaker_embs, | |
batch_size=batch_size, | |
guidance_scale=None, | |
) | |
b_tokens = torch.cat([in_x, y], dim=1) | |
for tokens in b_tokens: | |
try: | |
to_return.append(self.decoder.decode(tokens=tokens.tolist(), causal=False)) | |
except Exception as e: | |
print("failed to run MBD.") | |
print(f"reason: {str(e)}") | |
to_return.append(None) | |
return to_return | |
def __call__( | |
self, | |
*, | |
texts: list[str], | |
batch_size: int, | |
max_new_tokens: Optional[int], | |
top_k: Optional[int], | |
top_p: Optional[float], | |
temperature: Optional[float], | |
encodec_tokens: Optional[list[torch.Tensor]] = None, | |
speaker_embs: Optional[torch.Tensor] = None, | |
guidance_scale: Optional[float] = None, | |
): | |
if self.checkpoint_config.get("causal", True): | |
return self.causal_sample( | |
texts=texts, | |
batch_size=batch_size, | |
speaker_embs=speaker_embs, | |
guidance_scale=guidance_scale, | |
max_new_tokens=max_new_tokens, | |
top_k=top_k, | |
top_p=top_p, | |
temperature=temperature, | |
) | |
else: | |
assert encodec_tokens is not None | |
assert guidance_scale is None | |
assert max_new_tokens is None | |
assert top_p is None | |
return self.non_causal_sample( | |
texts=texts, | |
encodec_tokens=encodec_tokens, | |
batch_size=batch_size, | |
speaker_embs=speaker_embs, | |
top_k=top_k, | |
temperature=temperature, | |
) | |
def save_result_metadata(wav_path, ref_path, text, first_stage_ckpt_path, second_stage_ckpt_path): | |
if first_stage_ckpt_path is None or second_stage_ckpt_path is None: | |
return | |
json.dump( | |
{ | |
"speaker": ref_path, | |
"text": text, | |
}, | |
pathlib.Path(str(wav_path) + ".json").open("w"), | |
) | |
def get_cached_file(file_or_uri: str): | |
""" | |
If it's an s3 file, download it to a local temporary file and return that path. | |
Otherwise return the path as is. | |
""" | |
is_uri = file_or_uri.startswith("http") | |
cache_path = None | |
if is_uri: | |
ext = pathlib.Path(file_or_uri).suffix | |
# hash the file path to get the cache name | |
_cache_name = "audio_" + hashlib.md5(file_or_uri.encode("utf-8")).hexdigest() + ext | |
os.makedirs(os.path.expanduser("~/.cache/fam/"), exist_ok=True) | |
cache_path = os.path.expanduser(f"~/.cache/fam/{_cache_name}") | |
if not os.path.exists(cache_path): | |
command = f"curl -o {cache_path} {file_or_uri}" | |
subprocess.run(command, shell=True, check=True) | |
else: | |
if os.path.exists(file_or_uri): | |
cache_path = file_or_uri | |
else: | |
raise FileNotFoundError(f"File {file_or_uri} not found!") | |
return cache_path | |
def get_cached_embedding(local_file_path: str, spkemb_model): | |
if not os.path.exists(local_file_path): | |
raise FileNotFoundError(f"File {local_file_path} not found!") | |
# hash the file path to get the cache name | |
_cache_name = "embedding_" + hashlib.md5(local_file_path.encode("utf-8")).hexdigest() + ".pt" | |
os.makedirs(os.path.expanduser("~/.cache/fam/"), exist_ok=True) | |
cache_path = os.path.expanduser(f"~/.cache/fam/{_cache_name}") | |
if not os.path.exists(cache_path): | |
spk_emb = spkemb_model.embed_utterance_from_file(local_file_path, numpy=False).unsqueeze(0) # (b=1, c) | |
torch.save(spk_emb, cache_path) | |
else: | |
spk_emb = torch.load(cache_path) | |
return spk_emb | |
def _sample_utterance_batch( | |
texts: list[str], | |
spk_cond_paths: list[Optional[str]], | |
spkemb_model, | |
first_stage_model, | |
second_stage_model, | |
enhancer: Optional[Union[Literal["df"], BaseEnhancer]], | |
first_stage_ckpt_path: str, | |
second_stage_ckpt_path: str, | |
guidance_scale: Optional[Tuple[float, float]], | |
max_new_tokens: int, | |
top_k: Optional[int], | |
top_p: Optional[float], | |
temperature: Optional[float], | |
batch_size: int = 128, | |
) -> List[str]: | |
speaker_embs = [] | |
refs = spk_cond_paths.copy() | |
# multithreaded loop to cache all the files | |
spk_cond_paths = tqdm.contrib.concurrent.thread_map( | |
get_cached_file, spk_cond_paths, desc="getting cached speaker ref files" | |
) | |
for i, (text, spk_cond_path) in tqdm.tqdm( | |
enumerate(zip(texts, spk_cond_paths)), total=len(texts), desc="calculating speaker embeddings" | |
): | |
texts[i] = normalize_text(text) | |
speaker_embs.append(get_cached_embedding(spk_cond_path, spkemb_model) if spk_cond_path else None) | |
b_speaker_embs = torch.cat(speaker_embs, dim=0) | |
start = time.time() | |
b_tokens = first_stage_model( | |
texts=texts, | |
speaker_embs=b_speaker_embs, | |
batch_size=batch_size, | |
guidance_scale=guidance_scale, | |
top_p=top_p, | |
top_k=top_k, | |
temperature=temperature, | |
max_new_tokens=max_new_tokens, | |
) | |
# TODO: set batch size for second stage model! | |
wav_files = second_stage_model( | |
texts=texts, | |
encodec_tokens=b_tokens, | |
speaker_embs=b_speaker_embs, | |
batch_size=batch_size, | |
guidance_scale=None, | |
top_p=None, | |
top_k=top_k, | |
temperature=temperature, | |
max_new_tokens=None, | |
) | |
for text, tokens, speaker_embs, ref_name, wav_file in zip(texts, b_tokens, b_speaker_embs, refs, wav_files): | |
if wav_file is None: | |
continue | |
with tempfile.NamedTemporaryFile(suffix=".wav") as enhanced_tmp: | |
if enhancer is not None: | |
enhancer = get_enhancer(enhancer) if isinstance(enhancer, str) else enhancer | |
enhancer(str(wav_file) + ".wav", enhanced_tmp.name) | |
# copy enhanced_tmp.name back to wav_file | |
print(f"copying enhanced file from {enhanced_tmp.name} to {str(wav_file) + '.wav'}.") | |
shutil.copy2(enhanced_tmp.name, str(wav_file) + ".wav") | |
save_result_metadata( | |
wav_file, | |
ref_name, | |
text, | |
first_stage_ckpt_path, | |
second_stage_ckpt_path, | |
) | |
print(f"time_to_synth_s: {time.time() - start}") | |
return [str(w) + ".wav" if not str(w).endswith(".wav") else str(w) for w in wav_files] | |
def sample_utterance( | |
text: str, | |
spk_cond_path: Optional[str], | |
spkemb_model, | |
first_stage_model, | |
second_stage_model, | |
enhancer: Optional[Union[Literal["df"], BaseEnhancer]], | |
first_stage_ckpt_path: str, | |
second_stage_ckpt_path: str, | |
guidance_scale: Optional[Tuple[float, float]], | |
max_new_tokens: int, | |
top_k: Optional[int], | |
top_p: Optional[float], | |
temperature: Optional[float], | |
) -> str: | |
# NOTE: supports max. 220 characters atm. | |
# Long form synthesis coming soon... | |
MAX_CHARS = 220 | |
if len(text) > MAX_CHARS: | |
print( | |
f"\n***WARNING: Max {MAX_CHARS} characters supported. Provided: {len(text)}. Truncating and generating speech...Can lead to unpredictable speech at the end.***" | |
) | |
return _sample_utterance_batch( | |
texts=[text], | |
spk_cond_paths=[spk_cond_path], | |
spkemb_model=spkemb_model, | |
first_stage_model=first_stage_model, | |
second_stage_model=second_stage_model, | |
enhancer=enhancer, | |
first_stage_ckpt_path=first_stage_ckpt_path, | |
second_stage_ckpt_path=second_stage_ckpt_path, | |
batch_size=1, | |
guidance_scale=guidance_scale, | |
max_new_tokens=max_new_tokens, | |
top_k=top_k, | |
top_p=top_p, | |
temperature=temperature, | |
)[0] | |
def build_models(config_first_stage, config_second_stage, model_dir, device, use_kv_cache): | |
smodel = SpeakerEncoder( | |
weights_fpath=os.path.join(model_dir, "speaker_encoder.pt"), device=device, eval=True, verbose=False | |
) | |
data_adapter = FlattenedInterleavedEncodec2Codebook(end_of_audio_token=1024) | |
llm_first_stage = Model( | |
config_first_stage, | |
TrainedBPETokeniser, | |
EncodecDecoder, | |
data_adapter_fn=data_adapter.decode, | |
use_kv_cache=use_kv_cache, | |
) | |
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 | |
) | |
return smodel, llm_first_stage, llm_second_stage | |
def get_first_stage_path(model_dir: str): | |
"""Absolute path to checkpoint for the first stage model.""" | |
return os.path.join(os.path.expanduser(model_dir), "first_stage.pt") | |
def get_second_stage_path(model_dir: str): | |
"""Absolute path to checkpoint for the second stage model.""" | |
return os.path.join(os.path.expanduser(model_dir), "second_stage.pt") | |
class SamplingControllerConfig: | |
""" | |
Sample from a trained model. | |
""" | |
spk_cond_path: str | |
"""Path to speaker reference file. Min. 30s of audio required. Supports both local paths & public URIs. Audio formats: wav, flac & mp3""" | |
huggingface_repo_id: str = "kotoba-tech/kotoba-speech-v0.1" | |
"""Absolute path to the model directory.""" | |
text: str = ( | |
"This is a demo of text to speech by MetaVoice-1B, an open-source foundational audio model by MetaVoice." | |
) | |
"""Text to synthesise.""" | |
num_samples: int = 1 | |
"""Number of samples to generate from each model.""" | |
max_new_tokens: int = 864 | |
"""Maximum number of new tokens to generate from the first stage model.""" | |
temperature: float = 1.0 | |
"""Temperature for sampling applied to both models.""" | |
top_k: Optional[int] = None | |
"""Top k for sampling applied to both models.""" | |
top_p: Optional[float] = 0.95 | |
"""Top p for sampling applied to first-stage model.""" | |
seed: int = 1337 | |
"""Random seed for sampling.""" | |
device: Literal["cuda", "cpu"] = "cuda" | |
"""Device to use for sampling.""" | |
dtype: Literal["bfloat16", "float16", "float32", "tfloat32"] = get_default_dtype() | |
"""Data type to use for sampling.""" | |
compile: bool = False | |
"""Whether to compile the model using PyTorch 2.0.""" | |
enhancer: Optional[Literal["df"]] = "df" | |
"""Enhancer to use for post-processing.""" | |
init_from: str = "resume" | |
"""Either 'resume' (from an out_dir) or a gpt2 variant (e.g. 'gpt2-xl').""" | |
use_kv_cache: Optional[Literal["vanilla"]] = "vanilla" | |
"""Type of kv caching to use for inference: 1) [none] no kv caching, 2) [vanilla] use torch attention with hand implemented kv-cache.""" | |
output_dir: str = "samples/" | |
"""Relative path to output directory""" | |
guidance_scale: Optional[Tuple[float, float]] = (3.0, 1.0) | |
"""Guidance scale for sampling: (speaker conditioning guidance_scale, prompt conditioning guidance scale).""" | |
batch_size: int = 128 | |
"""Batch size to use for sampling. Note that the batch size gets doubled when guidance is used. For H100, and 1B model, | |
1 w/ guidance and 1 w/o guidance work well (without kv-caching). With kv-caching, 128 (w/o guidance) and | |
64 (w/ guidance) works well.""" | |
if __name__ == "__main__": | |
# TODO: add support for batch sampling via CLI. Function has been implemented above. | |
sampling_config = tyro.cli(SamplingControllerConfig, use_underscores=True) | |
check_audio_file(sampling_config.spk_cond_path) | |
model_dir = snapshot_download(repo_id=sampling_config.huggingface_repo_id) | |
first_stage_ckpt_path = get_first_stage_path(model_dir) | |
second_stage_ckpt_path = get_second_stage_path(model_dir) | |
config_first_stage = InferenceConfig( | |
ckpt_path=first_stage_ckpt_path, | |
num_samples=sampling_config.num_samples, | |
seed=sampling_config.seed, | |
device=sampling_config.device, | |
dtype=sampling_config.dtype, | |
compile=sampling_config.compile, | |
init_from=sampling_config.init_from, | |
output_dir=sampling_config.output_dir, | |
) | |
config_second_stage = InferenceConfig( | |
ckpt_path=second_stage_ckpt_path, | |
num_samples=sampling_config.num_samples, | |
seed=sampling_config.seed, | |
device=sampling_config.device, | |
dtype=sampling_config.dtype, | |
compile=sampling_config.compile, | |
init_from=sampling_config.init_from, | |
output_dir=sampling_config.output_dir, | |
) | |
sampling_config.max_new_tokens *= ( | |
2 # deal with max_new_tokens for flattened interleaving! (should scale with num_codebooks?) | |
) | |
# define models | |
smodel, llm_first_stage, llm_second_stage = build_models( | |
config_first_stage, | |
config_second_stage, | |
model_dir=model_dir, | |
device=sampling_config.device, | |
use_kv_cache=sampling_config.use_kv_cache, | |
) | |
sample_utterance( | |
sampling_config.text, | |
os.path.expanduser(sampling_config.spk_cond_path), | |
smodel, | |
llm_first_stage, | |
llm_second_stage, | |
sampling_config.enhancer, | |
first_stage_ckpt_path, | |
second_stage_ckpt_path, | |
sampling_config.guidance_scale, | |
max_new_tokens=sampling_config.max_new_tokens, | |
top_k=sampling_config.top_k, | |
top_p=sampling_config.top_p, | |
temperature=sampling_config.temperature, | |
) | |