modeling_arkasr.py

from __future__ import annotations

from typing import Optional, List, Tuple, Union, Dict

import torch
from torch import Tensor, nn
from transformers import Qwen2ForCausalLM
from transformers.modeling_outputs import CausalLMOutputWithPast

from .configuration_arkasr import ArkasrConfig
from .modeling_audio import WhisperSpecialEncoder


class AudioMLPAdapter(nn.Module):
    def __init__(self, config: ArkasrConfig):
        super().__init__()
        whisper_config = config.whisper_config
        self.merge_factor = int(config.merge_factor)

        # 音频编码器
        self.whisper = WhisperSpecialEncoder(
            whisper_config,
            use_rope=getattr(config, "use_rope", False),
        )
        # 禁用 Whisper 自带 LayerNorm
        self.whisper.layer_norm = nn.Identity()
        self.layer_norm = nn.LayerNorm(whisper_config.hidden_size)

        act_fn_map = {
            "gelu": nn.GELU(),
            "relu": nn.ReLU(),
            "selu": nn.SELU(),
        }
        act = act_fn_map.get(getattr(config, "mlp_adapter_act", "gelu"), nn.GELU())

        input_dim = whisper_config.hidden_size * self.merge_factor
        output_dim = config.hidden_size

        self.adapting = nn.Sequential(
            nn.Linear(input_dim, output_dim * 2),
            act,
            nn.Linear(output_dim * 2, output_dim),
        )

    def forward(self, audios: Tensor) -> Tensor:
        """
        Args:
            audios: (B, mel, T) 或 (B, raw_len) —— 由 WhisperSpecialEncoder 决定
        Returns:
            adapted_features: (B, Seq_Audio, LLM_Hidden_Dim)
        """
        bsz = audios.size(0)

        encoded = self.whisper(audios)[0]  # (B, T, D)
        encoded = self.layer_norm(encoded)

        seq_len = encoded.size(1)
        if seq_len % self.merge_factor != 0:
            target_len = (seq_len // self.merge_factor) * self.merge_factor
            if target_len <= 0:
                # 极短音频兜底：pad 到 merge_factor
                target_len = self.merge_factor
                if seq_len < target_len:
                    pad_len = target_len - seq_len
                    pad = encoded.new_zeros((bsz, pad_len, encoded.size(-1)))
                    encoded = torch.cat([encoded, pad], dim=1)
            else:
                encoded = encoded[:, :target_len, :]

        encoded = encoded.reshape(bsz, -1, encoded.size(-1) * self.merge_factor)
        adapted = self.adapting(encoded)  # (B, T/k, hidden)
        return adapted


class ArkasrForConditionalGeneration(Qwen2ForCausalLM):
    config_class = ArkasrConfig
    _no_split_modules = ["WhisperSpecialEncoder"]

    def __init__(self, config: ArkasrConfig):
        super().__init__(config)
        self.audio_encoder = AudioMLPAdapter(config)

        self.audio_token_id = getattr(config, "audio_token_id", None)
        if self.audio_token_id is None:
            raise ValueError("`audio_token_id` must be defined in config.")

    @staticmethod
    def _cache_seq_len(past_key_values) -> int:
        if past_key_values is None:
            return 0
        if hasattr(past_key_values, "get_seq_length"):
            try:
                return int(past_key_values.get_seq_length())
            except Exception:
                return 0
        try:
            return int(past_key_values[0][0].shape[-2])
        except Exception:
            return 0

    def _inject_audio_embeddings_batch_encode_then_loop_scatter(
        self,
        input_ids: torch.LongTensor,          # (B, S)
        inputs_embeds: torch.FloatTensor,     # (B, S, H)
        audios: Tensor,                       # (B, ...)
    ) -> torch.FloatTensor:
        """
        先对「有 audio token 的样本」做一次 batch 音频编码，
        然后 for-loop 把每个样本的 audio features 按 audio_token 位置写回 inputs_embeds。

        好处：
        - encoder 只跑一次（快）
        - 写回按样本做，不会跨样本错位（稳）
        - 碰到某行没有 audio_token：直接跳过（TTS 行无影响）

        约束：
        - 每条样本的 audio_token 数量 n_i 需要和 audio_encoder 输出的 Sa 对齐。
          如果不对齐：这里采用截断/补零对齐到 n_i（不报错）。
        """
        B, S = input_ids.shape
        H = inputs_embeds.size(-1)
        device = inputs_embeds.device
        dtype = inputs_embeds.dtype

        # 找到哪些样本需要注入
        mask = (input_ids == self.audio_token_id)  # (B, S)
        per_counts = mask.sum(dim=1)               # (B,)
        need_idx = (per_counts > 0).nonzero(as_tuple=False).squeeze(1)  # (K,)

        if need_idx.numel() == 0:
            return inputs_embeds

        # 只编码需要注入的那部分音频（K, ...)
        audios_sub = audios.index_select(0, need_idx)
        feats_sub = self.audio_encoder(audios_sub)  # (K, Sa, H)

        # 写回：逐样本替换（写回操作本身几乎不耗时）
        feats_sub = feats_sub.to(device=device, dtype=dtype)
        Sa = feats_sub.size(1)

        # 逐个样本注入
        for k in range(need_idx.numel()):
            i = int(need_idx[k].item())
            n_i = int(per_counts[i].item())
            if n_i <= 0:
                continue

            feat_i = feats_sub[k]  # (Sa, H)

            # 对齐到该样本的 audio token 数 n_i
            if Sa < n_i:
                pad = feat_i.new_zeros((n_i - Sa, H))
                feat_i_use = torch.cat([feat_i, pad], dim=0)
            elif Sa > n_i:
                feat_i_use = feat_i[:n_i]
            else:
                feat_i_use = feat_i

            pos_i = mask[i].nonzero(as_tuple=False).squeeze(1)  # (n_i,)
            # 写回 embeddings
            inputs_embeds[i, pos_i, :] = feat_i_use

        return inputs_embeds

    def forward(
        self,
        input_ids: Optional[torch.LongTensor] = None,
        audios: Optional[Tensor] = None,
        attention_mask: Optional[Tensor] = None,
        position_ids: Optional[Tensor] = None,
        past_key_values: Optional[List[torch.FloatTensor]] = None,
        inputs_embeds: Optional[torch.FloatTensor] = None,
        use_cache: Optional[bool] = None,
        labels: Optional[torch.LongTensor] = None,
        output_attentions: Optional[bool] = None,
        output_hidden_states: Optional[bool] = None,
        logits_to_keep: int | torch.Tensor = 0,
        **kwargs,
    ) -> Union[Tuple, CausalLMOutputWithPast]:

        if inputs_embeds is None:
            if input_ids is None:
                raise ValueError("Either `input_ids` or `inputs_embeds` must be provided.")
            inputs_embeds = self.model.embed_tokens(input_ids)

        # 只在首步（past_len==0）注入，避免 generation 后续重复 encode
        past_len = self._cache_seq_len(past_key_values)
        if audios is not None and input_ids is not None and past_len == 0:
            inputs_embeds = self._inject_audio_embeddings_batch_encode_then_loop_scatter(
                input_ids=input_ids,
                inputs_embeds=inputs_embeds,
                audios=audios,
            )

        outputs = self.model(
            input_ids=None,
            attention_mask=attention_mask,
            position_ids=position_ids,
            past_key_values=past_key_values,
            inputs_embeds=inputs_embeds,
            use_cache=use_cache,
            output_attentions=output_attentions,
            output_hidden_states=output_hidden_states,
        )

        hidden_states = outputs[0]

        # logits（避免重复算 lm_head）
        if isinstance(logits_to_keep, int) and logits_to_keep > 0:
            hidden_for_logits = hidden_states[:, -logits_to_keep:, :]
        elif isinstance(logits_to_keep, torch.Tensor):
            hidden_for_logits = hidden_states[:, logits_to_keep, :]
        else:
            hidden_for_logits = hidden_states

        logits = self.lm_head(hidden_for_logits)

        loss = None
        if labels is not None:
            loss = self.loss_function(
                logits=logits,
                labels=labels,
                vocab_size=self.config.vocab_size,
                **kwargs,
            )

        return CausalLMOutputWithPast(
            loss=loss,
            logits=logits,
            past_key_values=outputs.past_key_values,
            hidden_states=outputs.hidden_states,
            attentions=outputs.attentions,
        )

    def prepare_inputs_for_generation(
        self,
        input_ids,
        past_key_values=None,
        attention_mask=None,
        inputs_embeds=None,
        **kwargs,
    ):
        past_len = self._cache_seq_len(past_key_values)
        if past_len > 0:
            input_ids = input_ids[:, -1:]

        model_inputs = {
            "input_ids": input_ids,
            "past_key_values": past_key_values,
            "use_cache": kwargs.get("use_cache"),
            "attention_mask": attention_mask,
            # audios 透传；forward 内 past_len==0 才注入，所以后续 step 不会重复编码
            "audios": kwargs.get("audios", None),
        }

        if inputs_embeds is not None and past_key_values is None:
            model_inputs["inputs_embeds"] = inputs_embeds
            del model_inputs["input_ids"]

        return model_inputs


__all__ = ["ArkasrForConditionalGeneration", "AudioMLPAdapter"]