from typing import Iterable, Optional, Tuple import librosa import torch import torch.nn.functional as F import torchaudio from torch import Tensor, nn from transformers import PreTrainedModel, Qwen2Model from transformers.generation.utils import GenerationMixin from transformers.modeling_outputs import CausalLMOutputWithPast from .configuration_step_audio_2 import StepAudio2Config def _mel_filters(n_mels: int) -> torch.Tensor: """Load the mel filterbank matrix for projecting STFT into a Mel spectrogram.""" assert n_mels in {80, 128}, f"Unsupported n_mels: {n_mels}" if n_mels == 128: return torch.from_numpy(librosa.filters.mel(sr=16000, n_fft=400, n_mels=128)) else: return torch.from_numpy(librosa.filters.mel(sr=16000, n_fft=400, n_mels=80)) def load_audio(file_path, target_rate=16000, max_length=None): """ Open an audio file and read as mono waveform, resampling as necessary If max_length is provided, truncate the audio to that length """ waveform, sample_rate = torchaudio.load(file_path) if sample_rate != target_rate: waveform = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=target_rate)(waveform) audio = waveform[0] # get the first channel # Truncate audio if it exceeds max_length if max_length is not None and audio.shape[0] > max_length: audio = audio[:max_length] return audio def log_mel_spectrogram(audio, n_mels=128, padding=479, device=None): """ Compute the log-Mel spectrogram with specific padding for StepAudio """ if not torch.is_tensor(audio): if isinstance(audio, str): audio = load_audio(audio) audio = torch.from_numpy(audio) if device is not None: audio = audio.to(device) if padding > 0: audio = F.pad(audio, (0, padding)) window = torch.hann_window(400).to(audio.device) stft = torch.stft(audio, 400, 160, window=window, return_complex=True) magnitudes = stft[..., :-1].abs() ** 2 filters = _mel_filters(n_mels) mel_spec = filters @ magnitudes log_spec = torch.clamp(mel_spec, min=1e-10).log10() log_spec = torch.maximum(log_spec, log_spec.max() - 8.0) log_spec = (log_spec + 4.0) / 4.0 return log_spec def compute_token_num(max_feature_len): # First, audio goes through encoder: # 1. conv1: kernel=3, stride=1, padding=1 -> size unchanged # 2. conv2: kernel=3, stride=2, padding=1 -> size/2 # 3. avg_pooler: kernel=2, stride=2 -> size/2 max_feature_len = max_feature_len - 2 # remove padding encoder_output_dim = (max_feature_len + 1) // 2 // 2 # after conv2 and avg_pooler # Then through adaptor (parameters from config file): padding = 1 kernel_size = 3 # from config: audio_encoder_config.kernel_size stride = 2 # from config: audio_encoder_config.adapter_stride adapter_output_dim = (encoder_output_dim + 2 * padding - kernel_size) // stride + 1 return adapter_output_dim def make_non_pad_mask(lengths: torch.Tensor, max_len: int = 0) -> torch.Tensor: """Make mask tensor containing indices of non-padded part. The sequences in a batch may have different lengths. To enable batch computing, padding is need to make all sequence in same size. To avoid the padding part pass value to context dependent block such as attention or convolution , this padding part is masked. 1 for non-padded part and 0 for padded part. Parameters ---------- lengths (torch.Tensor): Batch of lengths (B,). Returns: ------- torch.Tensor: Mask tensor containing indices of padded part (B, max_T). Examples: >>> import torch >>> import s3tokenizer >>> lengths = torch.tensor([5, 3, 2]) >>> masks = s3tokenizer.make_non_pad_mask(lengths) masks = [[1, 1, 1, 1, 1], [1, 1, 1, 0, 0], [1, 1, 0, 0, 0]] """ batch_size = lengths.size(0) max_len = max_len if max_len > 0 else lengths.max().item() seq_range = torch.arange(0, max_len, dtype=torch.int64, device=lengths.device) seq_range_expand = seq_range.unsqueeze(0).expand(batch_size, max_len) seq_length_expand = lengths.unsqueeze(-1) mask = seq_range_expand >= seq_length_expand return ~mask def mask_to_bias(mask: torch.Tensor, dtype: torch.dtype) -> torch.Tensor: """Convert bool-tensor to float-tensor for flash attention. Parameters ---------- lengths (torch.Tensor): Batch of lengths (B, ?). Returns: ------- torch.Tensor: Mask tensor containing indices of padded part (B, ?). Examples: >>> import torch >>> import s3tokenizer >>> lengths = torch.tensor([5, 3, 2]) >>> masks = s3tokenizer.make_non_pad_mask(lengths) masks = [[1, 1, 1, 1, 1], [1, 1, 1, 0, 0], [1, 1, 0, 0, 0]] >>> new_masks = s3tokenizer.mask_to_bias(masks, torch.float32) new_masks = [[-0.0000e+00, -0.0000e+00, -0.0000e+00, -0.0000e+00, -0.0000e+00], [-0.0000e+00, -0.0000e+00, -0.0000e+00, -1.0000e+10, -1.0000e+10], [-0.0000e+00, -0.0000e+00, -1.0000e+10, -1.0000e+10, -1.0000e+10]] """ assert mask.dtype == torch.bool assert dtype in [torch.float32, torch.bfloat16, torch.float16] mask = mask.to(dtype) # attention mask bias # NOTE(Mddct): torch.finfo jit issues # chunk_masks = (1.0 - chunk_masks) * torch.finfo(dtype).min mask = (1.0 - mask) * -1.0e+10 return mask class LayerNorm(nn.LayerNorm): def forward(self, input: Tensor) -> Tensor: return super().forward(input).type(input.dtype) class Linear(nn.Linear): def forward(self, input: Tensor) -> Tensor: return F.linear( input, self.weight.to(input.dtype), None if self.bias is None else self.bias.to(input.dtype), ) class Conv1d(nn.Conv1d): def _conv_forward( self, input: Tensor, weight: Tensor, bias: Optional[Tensor] ) -> Tensor: return super()._conv_forward( input, weight.to(input.dtype), None if bias is None else bias.to(input.dtype) ) class MultiHeadAttention(nn.Module): def __init__(self, n_state: int, n_head: int): super().__init__() self.n_head = n_head self.query = Linear(n_state, n_state) self.key = Linear(n_state, n_state, bias=False) self.value = Linear(n_state, n_state) self.out = Linear(n_state, n_state) def forward( self, x: Tensor, mask: Optional[Tensor] = None, ): q = self.query(x) k = self.key(x) v = self.value(x) wv, qk = self.qkv_attention(q, k, v, mask) return self.out(wv), qk def qkv_attention( self, q: Tensor, k: Tensor, v: Tensor, mask: Optional[Tensor] = None ): _, T, D = q.shape scale = (D // self.n_head) ** -0.25 q = q.view(*q.shape[:2], self.n_head, -1).permute(0, 2, 1, 3) * scale k = k.view(*k.shape[:2], self.n_head, -1).permute(0, 2, 3, 1) * scale v = v.view(*v.shape[:2], self.n_head, -1).permute(0, 2, 1, 3) qk = q @ k # (B, n_head, T, T) if mask is not None: qk = qk + mask qk = qk.float() w = F.softmax(qk, dim=-1).to(q.dtype) return (w @ v).permute(0, 2, 1, 3).flatten(start_dim=2), qk.detach() class ResidualAttentionBlock(nn.Module): def __init__(self, n_state: int, n_head: int): super().__init__() self.attn = MultiHeadAttention(n_state, n_head) self.attn_ln = LayerNorm(n_state) n_mlp = n_state * 4 self.mlp = nn.Sequential( Linear(n_state, n_mlp), nn.GELU(), Linear(n_mlp, n_state) ) self.mlp_ln = LayerNorm(n_state) def forward( self, x: Tensor, mask: Optional[Tensor] = None, ): x = x + self.attn(self.attn_ln(x.contiguous()), mask=mask)[0] x = x + self.mlp(self.mlp_ln(x.contiguous())) return x class AudioEncoder(nn.Module): def __init__( self, n_mels: int, n_ctx: int, n_state: int, n_head: int, n_layer: int ): super().__init__() self.conv1 = Conv1d(n_mels, n_state, kernel_size=3, padding=1) self.conv2 = Conv1d(n_state, n_state, kernel_size=3, stride=2, padding=1) self.positional_embedding = nn.Embedding(n_ctx, n_state) self.positional_embedding.requires_grad_(False) self.blocks: Iterable[ResidualAttentionBlock] = nn.ModuleList( [ResidualAttentionBlock(n_state, n_head) for _ in range(n_layer)] ) self.avg_pooler = nn.AvgPool1d(2, stride=2) self.after_norm = LayerNorm(n_state) self.gradient_checkpointing = False def forward(self, x: Tensor, x_len: Tensor) -> Tuple[Tensor, Tensor]: T = x.size(-1) x = F.gelu(self.conv1(x)) x = F.gelu(self.conv2(x)) x = x.permute(0, 2, 1) # (B, T // 2, n_state) mask = make_non_pad_mask(x_len, T).unsqueeze(1) # (B, 1, T) mask = mask_to_bias(mask[:, :, (T + 1) % 2::2], x.dtype) # (B, 1, T // 2) x = (x + self.positional_embedding.weight[:x.shape[1], :]).to(x.dtype) for block in self.blocks: if self.gradient_checkpointing and self.training: x = torch.utils.checkpoint.checkpoint(block, x, mask.unsqueeze(1)) else: x = block(x, mask.unsqueeze(1)) x = x.permute(0, 2, 1) x = self.avg_pooler(x) x = x.permute(0, 2, 1) x_len = (x_len + 1) // 2 // 2 x = self.after_norm(x.contiguous()) return x, x_len class Adaptor(nn.Module): def __init__( self, n_state: int = 1280, n_hidden: int = 3072, kernel_size: int = 7, stride: int = 4 ): super().__init__() self.stride = stride if self.stride != -1: # print("self.stride: {}".format(self.stride)) self.conv = Conv1d(n_state, n_state, kernel_size, stride, padding=1) self.linear1 = nn.Linear(n_state, 2048) self.relu = nn.ReLU() self.linear2 = nn.Linear(2048, n_hidden) self.gradient_checkpointing = False def forward(self, x: Tensor) -> Tuple[Tensor]: T = x.size(-1) if self.stride != -1: if self.gradient_checkpointing and self.training: x = torch.utils.checkpoint.checkpoint(self.conv, x.permute(0, 2, 1)) x = x.permute(0, 2, 1) else: x = x.permute(0, 2, 1) x = F.gelu(self.conv(x)) x = x.permute(0, 2, 1) if self.gradient_checkpointing and self.training: x = torch.utils.checkpoint.checkpoint(self.linear1, x) x = torch.utils.checkpoint.checkpoint(self.relu, x) x = torch.utils.checkpoint.checkpoint(self.linear2, x) else: x = self.linear1(x) x = self.relu(x) x = self.linear2(x) return x class StepAudio2ForCausalLM(PreTrainedModel, GenerationMixin): config_class = StepAudio2Config main_input_name = "input_ids" # Important: Add this attribute to make HF recognize it as a model with generation capability # _keys_to_ignore_on_load_missing = ["lm_head.weight"] supports_gradient_checkpointing = True # 新增,声明支持gradient checkpointing def __init__(self, config: StepAudio2Config): super().__init__(config) if isinstance(config.torch_dtype, str): dtype = getattr(torch, config.torch_dtype) else: dtype = config.torch_dtype self.model = Qwen2Model(config.text_config) self.bf16 = dtype==torch.bfloat16 self.encoder = AudioEncoder( config.audio_encoder_config.n_mels, config.audio_encoder_config.n_audio_ctx, config.audio_encoder_config.n_audio_state, config.audio_encoder_config.n_audio_head, config.audio_encoder_config.n_audio_layer ) self.adapter = Adaptor( config.audio_encoder_config.n_audio_state, config.audio_encoder_config.llm_dim, config.audio_encoder_config.kernel_size, config.audio_encoder_config.adapter_stride ) if self.bf16: self.encoder = self.encoder.bfloat16() self.adapter = self.adapter.bfloat16() self.lm_head = torch.nn.Linear( config.hidden_size, config.vocab_size, bias=False, dtype=dtype ) self.post_init() def forward( self, input_ids=None, wavs=None, wav_lens=None, attention_mask=None, **kwargs ): hidden_states = self.model.embed_tokens(input_ids) if wavs is not None: if self.bf16: wavs = wavs.bfloat16() out, feat_lens = self.encoder(wavs, wav_lens) out = self.adapter(out) feat_lens = (feat_lens - 1) // 2 + 1 insert_location = torch.nonzero(input_ids == 151688) insert_location[:,1] += 1 for idx in range(len(insert_location)): i,s = insert_location[idx] hidden_states[i][s : s+feat_lens[idx]] = out[idx][:feat_lens[idx]] x = self.model(inputs_embeds=hidden_states, attention_mask=attention_mask)[0] logits = self.lm_head(x) return CausalLMOutputWithPast( logits=logits, past_key_values=None, hidden_states=None, attentions=None ) def get_input_embeddings(self): """Return the model's input embeddings - required for GenerationMixin""" return self.model.embed_tokens def get_output_embeddings(self): """Return the model's output embeddings (LM head) - required for GenerationMixin""" return self.lm_head def prepare_inputs_for_generation(self, input_ids, attention_mask=None, **kwargs): """Prepare inputs for generation - required for GenerationMixin""" # Keep the wavs and wav_lens from the initial call wavs = kwargs.get("wavs", None) wav_lens = kwargs.get("wav_lens", None) # For generation steps after the first, we don't need to process audio again # because the audio tokens have already been replaced in the input sequence if "past_key_values" in kwargs and kwargs["past_key_values"] is not None: # We're in a generation step, no need to process audio again return { "input_ids": input_ids, "attention_mask": attention_mask, "past_key_values": kwargs.get("past_key_values") } # First generation step, include audio processing return { "input_ids": input_ids, "attention_mask": attention_mask, "wavs": wavs, "wav_lens": wav_lens } def _reorder_cache(self, past_key_values, beam_idx): """Reorder the cache for beam search - required for GenerationMixin if using beam search""" # If you're not using past_key_values or beam search, this can be a simple pass-through # Otherwise implement according to your model's cache structure return past_key_values def _set_gradient_checkpointing(self, module, value=False): # For Qwen2Model if hasattr(self.model, 'gradient_checkpointing'): self.model.gradient_checkpointing = value # Add the missing _gradient_checkpointing_func method to Qwen2Model # This is what Qwen2Model tries to use when gradient_checkpointing=True if value and not hasattr(self.model, '_gradient_checkpointing_func'): def _gradient_checkpointing_func(module_to_run, *args, **kwargs): # This function wraps torch.utils.checkpoint.checkpoint # and is used by Qwen2Model to perform checkpointing return torch.utils.checkpoint.checkpoint(module_to_run, *args, **kwargs) self.model._gradient_checkpointing_func = _gradient_checkpointing_func # For custom encoder and adapter if hasattr(self.encoder, 'gradient_checkpointing'): self.encoder.gradient_checkpointing = value if hasattr(self.adapter, 'gradient_checkpointing'): self.adapter.gradient_checkpointing = value