MiMo-V2.5-ASR / src /mimo_audio /modeling_mimo_audio.py
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Initial Docker-based ASR demo (app.py + src + requirements)
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# Copyright 2025 Xiaomi Corporation.
import copy
import logging
from dataclasses import dataclass
from typing import List, Optional, Union, cast
import torch
import torch.distributed as dist
from torch import nn
from transformers import StoppingCriteria
from transformers.cache_utils import Cache, DynamicCache
from transformers.generation.streamers import BaseStreamer
from transformers.generation.utils import (
GenerateOutput,
GenerationConfig,
StoppingCriteriaList,
is_deepspeed_zero3_enabled,
)
from transformers.modeling_outputs import BaseModelOutputWithPast, ModelOutput
from transformers.models.qwen2.configuration_qwen2 import Qwen2Config
from transformers.models.qwen2.modeling_qwen2 import (
Qwen2Model,
Qwen2PreTrainedModel,
)
from transformers.utils import is_torchdynamo_compiling
logger = logging.getLogger(__name__)
class MiMoStopper(StoppingCriteria):
def __init__(
self,
group_size: int,
audio_channels: int,
stop_tokens: list[int] | None = None,
max_length: int | None = None,
min_length: int | None = None,
) -> None:
super().__init__()
self.group_size = group_size
self.audio_channels = audio_channels
self.step = (audio_channels + 1) * group_size
self.stop_token_ids = set(stop_tokens or [])
self.max_length = max_length
self.min_length = min_length or 0
def __call__(self, input_ids: torch.LongTensor, _scores: torch.FloatTensor):
is_done = False
cur_len = input_ids.shape[-1] // self.step
if self.max_length:
is_done |= cur_len >= self.max_length
if (self.stop_token_ids and
input_ids.shape[1] >= self.step and
cur_len >= self.min_length):
last_token = input_ids[0, -self.step].item()
is_done |= last_token in self.stop_token_ids
return torch.full(
(input_ids.shape[0],), is_done, device=input_ids.device, dtype=torch.bool
)
@dataclass
class MiMoSampler:
do_sample: bool | None = None
temperature: float | None = None
top_k: int | None = None
top_p: float | None = None
def process(self, scores: torch.Tensor):
if self.temperature is not None:
scores = scores / self.temperature
if self.top_k is not None and self.top_k > 0:
top_k = min(self.top_k, scores.shape[-1])
indices_to_remove = scores < torch.topk(scores, top_k)[0][:, -1]
scores = scores.masked_fill(indices_to_remove, float("-inf"))
if self.top_p is not None and 0.0 < self.top_p <= 1.0:
top_p = self.top_p if 0.0 < self.top_p <= 1.0 else 1.0
sorted_logits, sorted_indices = torch.sort(scores)
cumulative_probs = sorted_logits.softmax(dim=-1).cumsum(dim=-1)
sorted_indices_to_remove = cumulative_probs <= (1 - top_p)
sorted_indices_to_remove[:, -1] = 0
indices_to_remove = sorted_indices_to_remove.scatter(
1, sorted_indices, sorted_indices_to_remove
)
scores = scores.masked_fill(indices_to_remove, float("-inf"))
return scores
def sample(self, scores: torch.Tensor, removed_tokens: list[int] | None = None):
scores = self.process(scores)
for t in removed_tokens or []:
scores[:, t] = float("-inf")
if self.do_sample:
probs = scores.softmax(dim=-1)
return torch.multinomial(probs, num_samples=1).squeeze(-1)
return torch.argmax(scores, dim=-1)
@dataclass
class MiMoAudioOutput(ModelOutput):
text_logits: torch.FloatTensor | None = None
local_hidden_states: torch.FloatTensor | None = None
past_key_values: Cache | None = None
"""Downcast hidden states for local transformer generation"""
@dataclass
class MiMoAudioConfig(Qwen2Config):
def __init__(
self,
*,
speech_vocab_size: str | int = "1025-1025-129-129-129-129-129-129",
speech_zeroemb_idx: str | int = "1024-1024-128-128-128-128-128-128",
delay_pattern: str = "0-1-2-3-4-5-6-7",
head_dim: int = 128,
group_size: int = 4,
audio_channels: int = 8,
local_dim: int = 1024,
local_layers: int = 16,
local_attn_heads: int = 64,
local_ffn_dim: int = 4096,
local_attn_dropout: float = 0.1,
input_local_layers: int = 6,
input_local_dim: int | None = None,
input_full_attention: bool | None = None,
**kwargs,
):
super().__init__(
**kwargs,
)
self.speech_vocab_size = speech_vocab_size
self.speech_zeroemb_idx = speech_zeroemb_idx
self.delay_pattern = delay_pattern
self.head_dim = head_dim
self.group_size = group_size
self.audio_channels = audio_channels
self.local_dim = local_dim
self.local_layers = local_layers
self.local_attn_heads = local_attn_heads
self.local_ffn_dim = local_ffn_dim
self.local_attn_dropout = local_attn_dropout
self.input_local_layers = input_local_layers
self.input_local_dim = input_local_dim or local_dim
self.input_full_attention = input_full_attention
def _parse_maybe_list(self, value: str | int, length: int) -> List[int]:
if isinstance(value, str) and "-" in value:
return [int(s) for s in value.split("-")]
return [int(value)] * length
def parsed_speech_empty_ids(self):
return self._parse_maybe_list(self.speech_zeroemb_idx, self.audio_channels)
def parsed_speech_vocab_sizes(self):
return self._parse_maybe_list(self.speech_vocab_size, self.audio_channels)
def parsed_delay_pattern(self):
return self._parse_maybe_list(self.delay_pattern, self.audio_channels)
def local_config(self):
config = copy.deepcopy(self)
config.hidden_size = self.local_dim
config.num_hidden_layers = self.local_layers
config.num_attention_heads = self.local_attn_heads
config.num_key_value_heads = self.local_attn_heads
config.head_dim = config.hidden_size // self.local_attn_heads
config.intermediate_size = self.local_ffn_dim
config.attention_dropout = self.local_attn_dropout
return config
def input_local_config(self):
config = copy.deepcopy(self)
config.hidden_size = self.input_local_dim
config.num_hidden_layers = self.input_local_layers
config.num_attention_heads = self.local_attn_heads
config.num_key_value_heads = self.local_attn_heads
config.head_dim = config.hidden_size // self.local_attn_heads
config.intermediate_size = config.hidden_size * 4
config.attention_dropout = self.local_attn_dropout
return config
@dataclass
class MiMoAudioArguments:
model_name_or_path: str
sosp_idx: int
eosp_idx: int
sostm_idx: int
eostm_idx: int
eot_idx: int
empty_idx: int
def to_dict(self):
return {
"model_name_or_path": self.model_name_or_path,
"sosp_idx": self.sosp_idx,
"eosp_idx": self.eosp_idx,
"sostm_idx": self.sostm_idx,
"eostm_idx": self.eostm_idx,
"eot_idx": self.eot_idx,
"empty_idx": self.empty_idx,
}
class MiMoAudioForCausalLM(Qwen2PreTrainedModel):
def __init__(
self,
config: MiMoAudioConfig | Qwen2Config,
args: MiMoAudioArguments | dict,
):
super().__init__(config)
config = (
MiMoAudioConfig(**vars(config))
if isinstance(config, Qwen2Config)
else config
)
args = MiMoAudioArguments(**args) if isinstance(args, dict) else args
self.config = config
self.args = args
self.model = Qwen2Model(config)
self.speech_vocab_sizes = config.parsed_speech_vocab_sizes()
self.speech_empty_ids = config.parsed_speech_empty_ids()
self.delay_pattern = config.parsed_delay_pattern()
self.group_size = config.group_size
self.audio_channels = config.audio_channels
self.lm_head = nn.Linear(config.hidden_size, config.vocab_size, bias=False)
# Construct local transformer
self.local_config = config.local_config()
self.local_transformer = Qwen2Model(self.local_config)
self.local_transformer.embed_tokens = None
# Add input local transformer if configured
self.input_local_config = config.input_local_config()
self.input_local_transformer = Qwen2Model(self.input_local_config)
self.input_local_transformer.embed_tokens = None
self.local_transformer_lm_heads = nn.ModuleList(
[
nn.Linear(
self.local_config.hidden_size,
self.speech_vocab_sizes[i],
bias=False,
)
for i in range(self.audio_channels)
]
)
self.speech_embeddings = nn.ModuleList(
[
nn.Embedding(
self.speech_vocab_sizes[i],
self.input_local_config.hidden_size,
padding_idx=self.speech_empty_ids[i],
)
for i in range(self.audio_channels)
]
)
if self.input_local_config.hidden_size != self.local_config.hidden_size:
self.speech_embeddings_to_local = nn.Linear(
self.input_local_config.hidden_size,
self.local_config.hidden_size,
bias=False,
)
else:
self.speech_embeddings_to_local = None
# Create speech_group_downcast_first for group_first_in_global_context
self.speech_group_downcast = nn.Linear(
self.input_local_config.hidden_size * config.group_size,
config.hidden_size,
bias=False,
)
self.hidden_states_downcast = nn.Linear(
config.hidden_size,
self.local_config.hidden_size,
bias=False,
)
# Initialize weights and apply final processing
self.post_init()
def apply_input_local_transformer(self, speech_embeddings: torch.Tensor):
B, T_groups, group_size, hidden_size = speech_embeddings.shape
# Process each group independently: [B*T//group_size, group_size, hidden_size]
input_embeddings = speech_embeddings.reshape(
B * T_groups, group_size, hidden_size
)
output: BaseModelOutputWithPast = self.input_local_transformer(
inputs_embeds=input_embeddings,
return_dict=True,
is_causal=not self.config.input_full_attention, # for SDPA
)
encoded_embeddings = output.last_hidden_state
# Reshape back to original format
# [B*T//group_size, group_size, hidden_size] -> [B, T//group_size, group_size, hidden_size]
encoded_embeddings = encoded_embeddings.reshape(
B, T_groups, group_size, hidden_size
)
return encoded_embeddings
def _prepare_input_embeds(
self,
input_ids: torch.LongTensor, # [B, audio_channels + 1, new_T]
):
B = input_ids.shape[0]
input_ids = input_ids.int()
group_size = self.config.group_size
text_input_ids = input_ids[:, 0, ::group_size]
speech_input_ids = (
input_ids[:, 1:, :]
.view(B, self.audio_channels, -1, group_size)
.transpose(1, 2)
) # [B, T//group_size, audio_channels, group_size]
is_speech = text_input_ids == self.args.empty_idx # [B, T//group_size]
speech_embeds = torch.zeros(
(
B,
is_speech.shape[1],
group_size,
self.input_local_config.hidden_size,
),
device=input_ids.device,
dtype=torch.bfloat16,
)
for idx in range(self.audio_channels):
cur_empty = self.speech_empty_ids[idx]
cur_embed = self.speech_embeddings[idx]
cur_speech_ids = speech_input_ids[:, :, idx, :]
cur_speech_embeds: torch.Tensor = cur_embed(cur_speech_ids)
# [B, T_groups, group_size, hidden_size]
cur_mask = cur_speech_ids == cur_empty
cur_speech_embeds.masked_fill_(cur_mask.unsqueeze(-1), 0.0)
speech_embeds += cur_speech_embeds
speech_embeds = speech_embeds * is_speech.unsqueeze(-1).unsqueeze(-1)
# Apply input local transformer if configured
speech_embeds = self.apply_input_local_transformer(speech_embeds)
speech_embeds = speech_embeds * is_speech.unsqueeze(-1).unsqueeze(-1)
T_groups = speech_embeds.shape[1]
speech_grouped_embeds: torch.Tensor = self.speech_group_downcast(
speech_embeds.view(B, T_groups, -1)
) # [B, T_groups, hidden_size]
text_embeds: torch.Tensor = self.model.embed_tokens(text_input_ids)
text_zero_mask = text_input_ids == self.args.empty_idx
text_embeds.masked_fill_(text_zero_mask.unsqueeze(-1), 0.0)
return text_embeds + speech_grouped_embeds
def forward(
self,
input_ids: torch.LongTensor, # [B, audio_channels + 1, new_T]
attention_mask: torch.Tensor, # [B, T_group]
position_ids: torch.LongTensor, # [B, new_T_group]
past_key_values: Cache | None = None,
cache_position: torch.LongTensor | None = None, # [new_T_group]
**_kwargs,
):
inputs_embeds = self._prepare_input_embeds(input_ids)
outputs: BaseModelOutputWithPast = self.model(
attention_mask=attention_mask,
position_ids=position_ids,
past_key_values=past_key_values,
inputs_embeds=inputs_embeds,
use_cache=True,
return_dict=True,
cache_position=cache_position,
)
hidden_states = outputs.last_hidden_state # [B, new_T_group, hidden_size]
text_logits: torch.Tensor = self.lm_head(
hidden_states[:, -1:, :]
) # [B, 1, vocab_size]
shift_hidden_states: torch.Tensor = self.hidden_states_downcast(
hidden_states[:, -1:, :]
) # [B, 1, hidden_size]
return MiMoAudioOutput(
text_logits=text_logits,
local_hidden_states=shift_hidden_states,
past_key_values=outputs.past_key_values,
)
def local_forward(
self,
local_embeds: torch.FloatTensor, # [B, 1, hidden_size]
tokens_dtype: torch.dtype,
tokens_device: torch.device,
local_sampler: MiMoSampler | None = None,
):
B = local_embeds.shape[0]
delay_iters = self.group_size + max(self.delay_pattern)
past_key_values = DynamicCache()
local_tokens = torch.zeros(
(B, self.group_size, self.audio_channels),
dtype=tokens_dtype,
device=tokens_device,
)
if local_sampler is None:
local_sampler = MiMoSampler()
for t in range(delay_iters):
output: BaseModelOutputWithPast = self.local_transformer(
inputs_embeds=local_embeds,
past_key_values=past_key_values,
return_dict=True,
use_cache=True,
)
hidden_state = output.last_hidden_state
past_key_values = output.past_key_values
local_embeds = torch.zeros_like(local_embeds)
for idx in range(self.audio_channels):
cur_start = self.delay_pattern[idx]
cur_end = cur_start + self.group_size
cur_empty = self.speech_empty_ids[idx]
if cur_start <= t < cur_end:
cur_lm_head = self.local_transformer_lm_heads[idx]
cur_scores: torch.Tensor = cur_lm_head(hidden_state)[:, -1, :]
# [B, vocab_size]
cur_token = local_sampler.sample(
cur_scores,
[cur_empty],
)
local_tokens[:, t - cur_start, idx] = cur_token
cur_input_embed = self.speech_embeddings[idx](
cur_token.unsqueeze(1)
)
if self.speech_embeddings_to_local is not None:
cur_input_embed = self.speech_embeddings_to_local(
cur_input_embed
)
local_embeds += cur_input_embed
return local_tokens # [B, group_size, audio_channels]
def _prepare_attention_mask(
self, inputs: torch.Tensor, input_ids_length: int
) -> torch.Tensor:
# No information for attention mask inference -> return default attention mask
return torch.ones(
(inputs.shape[0], input_ids_length),
dtype=torch.bool,
device=inputs.device,
)
def prepare_inputs_for_generation(
self,
input_ids: torch.LongTensor,
past_key_values: Optional[Cache] = None,
attention_mask: Optional[torch.LongTensor] = None,
inputs_embeds: Optional[torch.FloatTensor] = None,
cache_position: Optional[torch.LongTensor] = None,
**kwargs,
):
"""
Prepare the model inputs for generation. In includes operations like computing the 4D attention mask or
slicing inputs given the existing cache.
See the forward pass in the model documentation for expected arguments (different models might have different
requirements for e.g. `past_key_values`). This function should work as is for most LLMs.
"""
# 1. Handle BC:
model_inputs = {}
input_ids = input_ids.reshape(
input_ids.shape[0], -1, (self.audio_channels + 1) * self.config.group_size
).transpose(1, 2) # [B, audio_channels*group_size, T]
# - some models don't have `Cache` support (which implies they don't expect `cache_position` in `forward`)
if self._supports_cache_class:
model_inputs["cache_position"] = cache_position
# - `cache_position` was not a mandatory input in `prepare_inputs_for_generation` for those models, and this
# function may be called outside of `generate`. Handle most use cases by creating `cache_position` on the fly
# (this alternative is not as robust as calling `generate` and letting it create `cache_position`)
elif cache_position is None:
past_length = (
past_key_values[0][0].shape[2] if past_key_values is not None else 0
)
cache_position = torch.arange(
past_length,
input_ids.shape[2],
dtype=torch.long,
device=input_ids.device,
)
# 2. Generic cache-dependent input preparation
# If we have cache: let's slice `input_ids` through `cache_position`, to keep only the unprocessed tokens
# Exception 1: when passing input_embeds, input_ids may be missing entries
# Exception 2: some generation methods do special slicing of input_ids, so we don't need to do it here
# Exception 3: with synced GPUs cache_position may go out of bounds, but we only want dummy token in that case
if past_key_values is not None:
model_inputs["past_key_values"] = past_key_values
if (
inputs_embeds is not None or cache_position[-1] >= input_ids.shape[2]
): # Exception 1 or Exception 3
input_ids = input_ids[:, :, -cache_position.shape[0] :]
elif (
input_ids.shape[2] != cache_position.shape[0]
): # Default case (the "else", a no op, is Exception 2)
input_ids = input_ids[:, :, cache_position]
# 3. Prepare base model inputs
input_ids_key = (
"decoder_input_ids" if self.config.is_encoder_decoder else "input_ids"
)
# if `inputs_embeds` are passed, we only want to use them in the 1st generation step
if not self.config.is_encoder_decoder:
if inputs_embeds is not None and cache_position[0] == 0:
model_inputs[input_ids_key] = None
model_inputs["inputs_embeds"] = inputs_embeds
else:
# `clone` calls in this function ensure a consistent stride. See #32227
model_inputs[input_ids_key] = input_ids.clone(
memory_format=torch.contiguous_format
)
model_inputs["inputs_embeds"] = None
else:
model_inputs[input_ids_key] = input_ids.clone(
memory_format=torch.contiguous_format
)
# 4. Create missing `position_ids` on the fly
if attention_mask is not None and kwargs.get("position_ids") is None:
position_ids = attention_mask.long().cumsum(-1) - 1
position_ids.masked_fill_(attention_mask == 0, 1)
kwargs["position_ids"] = (
position_ids # placed in kwargs for further processing (see below)
)
# 5. Slice model inputs if it's an input that should have the same length as `input_ids`
for model_input_name in ["position_ids", "token_type_ids"]:
model_input: torch.Tensor = kwargs.get(model_input_name)
if model_input is not None:
if past_key_values:
model_input = model_input[:, -input_ids.shape[2] :]
model_input = model_input.clone(
memory_format=torch.contiguous_format
)
model_inputs[model_input_name] = model_input
if attention_mask is not None:
model_inputs["attention_mask"] = attention_mask
# 7. Forward ALL kwargs that are uninitialized (e.g. `use_cache`).
for key, value in kwargs.items():
if key not in model_inputs:
model_inputs[key] = value
if model_inputs[input_ids_key] is not None:
model_inputs[input_ids_key] = (
cast(torch.Tensor, model_inputs[input_ids_key])
.transpose(1, 2)
.reshape(input_ids.shape[0], -1, (self.audio_channels + 1))
.transpose(1, 2)
) # [B, audio_channels, T*group_size]
# 8. Remove unexpected `generate` inputs (TODO @joao: fix trainer and examples)
model_inputs.pop("labels", None)
return model_inputs
def _get_initial_cache_position(self, input_ids: torch.Tensor, model_kwargs: dict):
"""Calculates `cache_position` for the pre-fill stage based on `input_ids` and optionally past length"""
# `torch.compile`-friendly `torch.arange` from a shape -- the lines below are equivalent to `torch.arange`
if "inputs_embeds" in model_kwargs:
cache_position = (
torch.ones_like(
model_kwargs["inputs_embeds"][0, :, 0], dtype=torch.int64
).cumsum(0)
- 1
)
else:
cache_position = (
torch.ones(
(
input_ids.shape[1]
// (self.audio_channels + 1)
// self.config.group_size,
),
dtype=torch.int64,
device=input_ids.device,
).cumsum(0)
- 1
)
past_length = 0
if model_kwargs.get("past_key_values") is not None:
cache = model_kwargs["past_key_values"]
past_length = 0
if not isinstance(cache, Cache):
past_length = cache[0][0].shape[2]
elif (
hasattr(cache, "get_seq_length") and cache.get_seq_length() is not None
):
past_length = cache.get_seq_length()
# TODO(joao): this is not torch.compile-friendly, find a work-around. If the cache is not empty,
# end-to-end compilation will yield bad results because `cache_position` will be incorrect.
if not is_torchdynamo_compiling():
cache_position = cache_position[past_length:]
model_kwargs["cache_position"] = cache_position
return model_kwargs
@torch.inference_mode()
def generate(
self,
inputs: torch.Tensor | None = None,
generation_config: GenerationConfig | None = None,
stopping_criteria: StoppingCriteriaList | list | None = None,
streamer: BaseStreamer | None = None,
synced_gpus: bool | None = None,
global_sampler: MiMoSampler | None = None,
local_sampler: MiMoSampler | None = None,
warmup_run: bool | None = None,
**kwargs,
) -> Union[GenerateOutput, torch.LongTensor]:
generation_config, model_kwargs = self._prepare_generation_config(
generation_config, **kwargs
)
self._validate_model_kwargs(model_kwargs.copy())
# 2. Set generation parameters if not already defined
if synced_gpus is None:
if is_deepspeed_zero3_enabled() and dist.get_world_size() > 1:
synced_gpus = True
else:
synced_gpus = False
# 3. Define model inputs
input_ids, _model_input_name, model_kwargs = self._prepare_model_inputs(
inputs, generation_config.bos_token_id, model_kwargs
)
input_ids_length = input_ids.shape[-1]
input_ids_length //= self.group_size * (self.audio_channels + 1)
if streamer is not None:
streamer.put(input_ids.cpu())
if "attention_mask" not in model_kwargs:
model_kwargs["attention_mask"] = self._prepare_attention_mask(
inputs, input_ids_length
)
device = input_ids.device
self._prepare_special_tokens(generation_config, True, device=device)
model_kwargs["use_cache"] = True
model_kwargs["past_key_values"] = DynamicCache()
prepared_stopping_criteria = StoppingCriteriaList(
stopping_criteria if stopping_criteria is not None else []
)
prepared_stopping_criteria.append(
MiMoStopper(
self.group_size,
self.audio_channels,
max_length=generation_config.max_length,
)
)
stance = "default" if warmup_run else "eager_on_recompile"
with torch.compiler.set_stance(stance):
return self.slm_sample(
input_ids,
stopping_criteria=prepared_stopping_criteria,
generation_config=generation_config,
synced_gpus=synced_gpus,
streamer=streamer,
global_sampler=global_sampler,
local_sampler=local_sampler,
**model_kwargs,
)
def slm_sample(
self,
input_ids: torch.LongTensor,
stopping_criteria: StoppingCriteriaList,
generation_config: GenerationConfig,
synced_gpus: bool,
streamer: BaseStreamer | None,
global_sampler: MiMoSampler | None = None,
local_sampler: MiMoSampler | None = None,
**model_kwargs,
) -> torch.LongTensor:
max_length = generation_config.max_length
B, cur_len = input_ids.shape
cur_len //= self.group_size * (self.audio_channels + 1)
initial_len = cur_len
this_peer_finished = False
unfinished_sequences = torch.ones(B, dtype=torch.long, device=input_ids.device)
min_length = 0
stop_token_ids = set()
for criterion in stopping_criteria:
if isinstance(criterion, MiMoStopper):
if criterion.min_length is not None:
min_length = max(min_length, criterion.min_length)
stop_token_ids.update(criterion.stop_token_ids)
model_kwargs = self._get_initial_cache_position(input_ids, model_kwargs)
while self._has_unfinished_sequences(
this_peer_finished,
synced_gpus,
device=input_ids.device,
cur_len=cur_len,
max_length=max_length,
):
# prepare model inputs
model_inputs = self.prepare_inputs_for_generation(input_ids, **model_kwargs)
# forward pass to get next token
if (
cast(torch.Tensor, model_inputs["input_ids"]).shape[2]
!= self.group_size
):
# prefill run
with torch.compiler.set_stance("force_eager"):
outputs: MiMoAudioOutput = self(**model_inputs)
else:
outputs: MiMoAudioOutput = self(**model_inputs)
if synced_gpus and this_peer_finished:
continue # don't waste resources running the code we don't need
text_logits: torch.Tensor = outputs.text_logits[:, -1, :].clone()
# [B, vocab_size]
removed_tokens = None
if cur_len < min_length:
removed_tokens = list(stop_token_ids)
next_text_tokens = global_sampler.sample(text_logits, removed_tokens=removed_tokens)
# [B]
local_hidden_states = outputs.local_hidden_states
# Only Supports batch_size=1 here
if next_text_tokens[0] != self.args.empty_idx:
zero_embed_tensor = torch.tensor(
self.speech_empty_ids,
device=next_text_tokens.device,
dtype=input_ids.dtype,
)
next_speech_tokens = zero_embed_tensor.view(
1, 1, self.audio_channels
).expand(B, self.config.group_size, -1)
else:
next_speech_tokens = self.local_forward(
local_embeds=local_hidden_states,
tokens_dtype=next_text_tokens.dtype,
tokens_device=next_text_tokens.device,
local_sampler=local_sampler,
)
next_text_tokens = next_text_tokens.reshape(B, 1, 1).expand(
-1, self.group_size, -1
) # [B, group_size, 1]
# generate speech tokens
next_tokens = torch.cat(
(next_text_tokens, next_speech_tokens), dim=-1
).reshape(B, -1) # [B, group_size * (audio_channels + 1)]
input_ids = torch.cat(
[input_ids, next_tokens], dim=-1
) # [B, T*group_size*vq]
if streamer is not None:
streamer.put(next_tokens.cpu())
model_kwargs = self._update_model_kwargs_for_generation(
outputs,
model_kwargs,
is_encoder_decoder=self.config.is_encoder_decoder,
)
unfinished_sequences = unfinished_sequences & ~stopping_criteria(
input_ids, None
)
this_peer_finished = unfinished_sequences.max() == 0
cur_len += 1
# This is needed to properly delete outputs.logits which may be very large for first iteration
# Otherwise a reference to outputs is kept which keeps the logits alive in the next iteration
del outputs
if streamer is not None:
streamer.end()
input_ids = input_ids[:B]
return input_ids