NeMo / nemo /collections /speechlm2 /modules /speech_generation.py
dlxj
update nemo==2.8.0.rc0
f5d2dd3
# Copyright (c) 2025, NVIDIA CORPORATION. All rights reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
import torch
from omegaconf import DictConfig
from torch import nn
from nemo.collections.tts.modules import transformer_2501
from nemo.core.classes.module import NeuralModule
class TransformerARSpeechDecoder(NeuralModule):
def __init__(
self,
speech_decoder_parms: DictConfig,
lantent_dim: int,
num_audio_codebooks: int,
num_audio_tokens_per_codebook: int,
):
super().__init__()
self.use_input_cache = False
self.speech_decoder_parms = speech_decoder_parms
self.lantent_dim = lantent_dim
self.num_audio_codebooks = num_audio_codebooks
self.num_audio_tokens_per_codebook = num_audio_tokens_per_codebook
# optional configs
self.cfg_unconditional_prob = self.speech_decoder_parms.pop("cfg_unconditional_prob", None)
self.cfg_scale = self.speech_decoder_parms.pop("cfg_scale", 2.5)
self.cond_on_prev_audio_tokens = self.speech_decoder_parms.pop("cond_on_prev_audio_tokens", True)
self.detach_input = self.speech_decoder_parms.pop("detach_input", False)
# projection to adapt llm embeddings into the same shape of speech decoder expected input
if lantent_dim != self.speech_decoder_parms["d_model"]:
self.input_proj = nn.Linear(lantent_dim, self.speech_decoder_parms["d_model"])
else:
self.input_proj = None
# instanciate T5-TTS decoder to full compatibility and potentialy load pretrained model
self.t5_decoder = transformer_2501.Transformer(**self.speech_decoder_parms)
# projection to predict audio codes
self.final_proj = nn.Linear(
self.speech_decoder_parms["d_model"], num_audio_codebooks * num_audio_tokens_per_codebook
)
# create embeddings for encode input tokens
if self.cond_on_prev_audio_tokens:
audio_embeddings = []
for _ in range(self.num_audio_codebooks):
audio_embeddings.append(
nn.Embedding(num_audio_tokens_per_codebook, self.speech_decoder_parms["d_model"])
)
self.audio_embeddings = nn.ModuleList(audio_embeddings)
def forward(self, hidden_states, speech_mask, input_audio_tokens=None, return_raw_logits=False):
# Megatron LLM parallel training returns T, B, F so reshape it
# T, B, F = hidden_states.size()
hidden_states = hidden_states.transpose(0, 1).contiguous() # .reshape(B, T, F) # from [T, B, F] to [B, T, F]
# input cache needed due our transformer kv cache implementation expect the whole left context
if self.use_input_cache:
if self.cache["hidden_states"] is None:
self.cache["hidden_states"] = hidden_states
else:
self.cache["hidden_states"] = torch.cat([self.cache["hidden_states"], hidden_states], dim=1)
hidden_states = self.cache["hidden_states"]
if self.cache["speech_mask"] is None:
self.cache["speech_mask"] = speech_mask
else:
self.cache["speech_mask"] = torch.cat([self.cache["speech_mask"], speech_mask], dim=1)
speech_mask = self.cache["speech_mask"]
if self.cache["input_audio_tokens"] is None:
self.cache["input_audio_tokens"] = input_audio_tokens
else:
self.cache["input_audio_tokens"] = torch.cat(
[self.cache["input_audio_tokens"], input_audio_tokens], dim=1
)
input_audio_tokens = self.cache["input_audio_tokens"]
if self.detach_input:
hidden_states = hidden_states.detach()
# map hidden states to the shape of the
if self.input_proj is not None:
speech_decoder_input = self.input_proj(hidden_states)
else:
speech_decoder_input = hidden_states
# workaround for inference, because during inference speech_mask will be None
if speech_mask is None:
speech_mask = torch.ones(
(speech_decoder_input.size(0), speech_decoder_input.size(1)),
device=speech_decoder_input.device,
dtype=torch.bool,
)
if self.cfg_unconditional_prob:
if self.training:
# if training drop the "text" conditioning in a percentage of batch
if torch.rand(1).item() < self.cfg_unconditional_prob:
# make the whole batch zeros to the unconditional model
# ToDo: move it to cache to need to just create a 1 frame tensor in inference
speech_decoder_input = torch.zeros_like(speech_decoder_input)
else:
# if inference or evaluation create a zero tensor for speech decoder input and concatenate it to compute unconditional logits
speech_decoder_input_zeros = torch.zeros_like(speech_decoder_input)
speech_decoder_input = torch.cat([speech_decoder_input, speech_decoder_input_zeros], dim=0)
# duplicate mask to match the new shape
speech_mask = torch.cat([speech_mask, speech_mask], dim=0)
# if cond on prev tokens enabled, so duplicate the tokens to the new shape
if self.cond_on_prev_audio_tokens:
input_audio_tokens = torch.cat([input_audio_tokens, input_audio_tokens], dim=0)
if self.cond_on_prev_audio_tokens:
if self.detach_input:
input_audio_tokens = input_audio_tokens.detach()
audio_tokens_embedded = self.embed_audio_tokens(
input_audio_tokens.transpose(1, 2).contiguous()
) # (B, T', E)
speech_decoder_input = speech_decoder_input + audio_tokens_embedded
decoder_out = self.t5_decoder(x=speech_decoder_input, x_mask=speech_mask)['output']
# if it is true we need to return just the last autoregressive step, it is valid because for 1 frame input we produce 1 frame ouput
if self.use_input_cache:
decoder_out = decoder_out[:, -1:, :]
# get the logits of all codebooks
all_code_logits = self.final_proj(decoder_out)
# if using cfg and it is in inference or evaluation mix unconditional and coditional logits
if self.cfg_unconditional_prob and not self.training:
batch_size = all_code_logits.size(0) // 2
cond_logits = all_code_logits[:batch_size]
uncond_logits = all_code_logits[batch_size:]
all_code_logits = (1 - self.cfg_scale) * uncond_logits + self.cfg_scale * cond_logits
if return_raw_logits:
return all_code_logits
# convert the logits from the single projection to a list with logits separated by codebook
all_codebook_logits = self.all_logits_to_each_codebooks_logits(all_code_logits)
return all_codebook_logits, all_code_logits
def sample_codes_from_logits(self, all_code_logits_t, temperature=0.7, topk=80):
# all_code_logits_t: (B, num_codebooks * num_tokens_per_codebook), logits at a given timestep
all_preds = []
for idx in range(self.num_audio_codebooks):
si = idx * self.num_audio_tokens_per_codebook
ei = si + self.num_audio_tokens_per_codebook
codebook_logits = all_code_logits_t[:, si:ei] # (B, num_tokens_per_codebook)
codebook_logits_topk = torch.topk(codebook_logits, topk, dim=-1)[0] # (B, topk)
indices_to_remove = codebook_logits < codebook_logits_topk[:, -1].unsqueeze(
-1
) # (B, num_tokens_per_codebook)
codebook_logits_rescored = codebook_logits.clone()
codebook_logits_rescored[indices_to_remove] = float('-inf')
codebook_probs = torch.softmax(codebook_logits / temperature, dim=-1) # (B, num_tokens_per_codebook)
codebook_preds = torch.multinomial(codebook_probs, 1) # (B, 1)
all_preds.append(codebook_preds)
all_preds = torch.cat(all_preds, dim=1).long() # (B, num_codebooks)
return all_preds
def all_logits_to_each_codebooks_logits(self, logits):
all_codebook_logits = []
for idx in range(self.num_audio_codebooks):
si = idx * self.num_audio_tokens_per_codebook
ei = si + self.num_audio_tokens_per_codebook
codebook_logits = logits[:, :, si:ei] # (B, num_tokens_per_codebook)
# B, T, F = codebook_logits.size()
codebook_logits = codebook_logits.transpose(
0, 1
).contiguous() # .reshape(T, B, F) # transpose for compatibility with megatron format
all_codebook_logits.append(codebook_logits)
return all_codebook_logits
def embed_audio_tokens(self, audio_tokens):
# Add and average the embeddings of the audio tokens across the codebooks
audio_embedding = None
for c in range(self.num_audio_codebooks):
embedding = self.audio_embeddings[c](audio_tokens[:, c, :])
if audio_embedding is None:
audio_embedding = embedding
else:
audio_embedding = audio_embedding + embedding
audio_embedding = audio_embedding / audio_tokens.size(1)
return audio_embedding
def reset_input_and_kv_cache(self, use_cache):
if use_cache:
print("Enabling input and KV cache!")
self.use_input_cache = use_cache
self.cache = self._init_cache()
self.t5_decoder.reset_cache(use_cache=use_cache)
@staticmethod
def _init_cache():
return {
'hidden_states': None,
'speech_mask': None,
'input_audio_tokens': None,
}