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447 lines
17 KiB
Python
447 lines
17 KiB
Python
# SPDX-License-Identifier: Apache-2.0
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# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
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# Adapted from:
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# https://github.com/vllm-project/vllm/blob/main/vllm/model_executor/models/voxtral.py
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# https://huggingface.co/mistralai/Voxtral-Mini-3B-2507
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#
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# Copyright 2025 Mistral AI and the HuggingFace Inc. team.
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# Licensed under the Apache License, Version 2.0.
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"""Inference-only Voxtral (speech-to-text) model."""
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import math
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from typing import Any, Iterable, List, Optional, Tuple
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import torch
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import torch.nn as nn
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from transformers import PretrainedConfig
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from sglang.srt.layers.activation import get_act_fn
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from sglang.srt.layers.linear import (
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ColumnParallelLinear,
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QKVParallelLinear,
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RowParallelLinear,
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)
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from sglang.srt.layers.quantization.base_config import QuantizationConfig
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from sglang.srt.managers.mm_utils import (
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MultiModalityDataPaddingPatternMultimodalTokens,
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general_mm_embed_routine,
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)
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from sglang.srt.managers.schedule_batch import (
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Modality,
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MultimodalDataItem,
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MultimodalInputs,
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)
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from sglang.srt.model_executor.forward_batch_info import ForwardBatch
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from sglang.srt.model_loader.weight_utils import default_weight_loader
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from sglang.srt.models.llama import LlamaForCausalLM
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class AudioLanguageAdapter(nn.Module):
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"""MLP projector: Linear -> GELU -> Linear (no bias)."""
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def __init__(self, hidden_size: int, dim: int) -> None:
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super().__init__()
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self.w_in = nn.Linear(hidden_size, dim, bias=False)
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self.gelu = nn.GELU()
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self.w_out = nn.Linear(dim, dim, bias=False)
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def forward(self, x: torch.Tensor) -> torch.Tensor:
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return self.w_out(self.gelu(self.w_in(x)))
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class VoxtralWhisperAttention(nn.Module):
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"""Multi-headed self-attention using plain SDPA (no KV cache).
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Note: HF Voxtral has bias on q_proj, v_proj, out_proj but NOT on k_proj.
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We use QKVParallelLinear with bias=True and create a zero bias for k_proj
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during weight loading.
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"""
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def __init__(
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self,
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embed_dim: int,
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num_heads: int,
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quant_config: Optional[QuantizationConfig] = None,
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):
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super().__init__()
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self.head_dim = embed_dim // num_heads
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self.scaling = self.head_dim**-0.5
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self.qkv_proj = QKVParallelLinear(
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embed_dim, self.head_dim, num_heads, quant_config=quant_config
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)
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# After TP split, the local head count lives on the linear layer
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self.num_heads = self.qkv_proj.num_heads
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self.out_proj = RowParallelLinear(
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embed_dim, embed_dim, bias=True, quant_config=quant_config
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)
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def forward(self, hidden_states: torch.Tensor) -> torch.Tensor:
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batch_size, seq_len, _ = hidden_states.shape
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qkv, _ = self.qkv_proj(hidden_states)
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q, k, v = qkv.chunk(3, dim=-1)
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q = q * self.scaling
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q = q.view(batch_size, seq_len, self.num_heads, self.head_dim).permute(
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0, 2, 1, 3
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)
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k = k.view(batch_size, seq_len, self.num_heads, self.head_dim).permute(
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0, 2, 1, 3
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)
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v = v.view(batch_size, seq_len, self.num_heads, self.head_dim).permute(
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0, 2, 1, 3
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)
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attn_output = torch.nn.functional.scaled_dot_product_attention(
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q, k, v, scale=1.0
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)
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attn_output = attn_output.permute(0, 2, 1, 3).reshape(
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batch_size, seq_len, self.num_heads * self.head_dim
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)
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attn_output, _ = self.out_proj(attn_output)
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return attn_output
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class VoxtralWhisperEncoderLayer(nn.Module):
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def __init__(
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self,
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config: PretrainedConfig,
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quant_config: Optional[QuantizationConfig] = None,
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):
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super().__init__()
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embed_dim = config.d_model
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self.self_attn = VoxtralWhisperAttention(
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embed_dim=embed_dim,
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num_heads=config.encoder_attention_heads,
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quant_config=quant_config,
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)
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self.self_attn_layer_norm = nn.LayerNorm(embed_dim)
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self.activation_fn = get_act_fn(
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getattr(config, "activation_function", "gelu"),
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quant_config=quant_config,
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)
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self.fc1 = ColumnParallelLinear(embed_dim, config.encoder_ffn_dim)
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self.fc2 = RowParallelLinear(config.encoder_ffn_dim, embed_dim)
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self.final_layer_norm = nn.LayerNorm(embed_dim)
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def forward(self, hidden_states: torch.Tensor) -> torch.Tensor:
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residual = hidden_states
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hidden_states = self.self_attn_layer_norm(hidden_states)
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hidden_states = self.self_attn(hidden_states)
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hidden_states = residual + hidden_states
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residual = hidden_states
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hidden_states = self.final_layer_norm(hidden_states)
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hidden_states, _ = self.fc1(hidden_states)
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hidden_states = self.activation_fn(hidden_states)
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hidden_states, _ = self.fc2(hidden_states)
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hidden_states = residual + hidden_states
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if hidden_states.dtype == torch.float16:
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clamp_value = torch.finfo(hidden_states.dtype).max - 1000
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hidden_states = torch.clamp(
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hidden_states, min=-clamp_value, max=clamp_value
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)
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return hidden_states
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class VoxtralWhisperEncoder(nn.Module):
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"""Whisper encoder (Conv1d + positional embed + transformer + layer norm)."""
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def __init__(
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self,
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config: PretrainedConfig,
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quant_config: Optional[QuantizationConfig] = None,
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):
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super().__init__()
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embed_dim = config.d_model
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self.conv1 = nn.Conv1d(config.num_mel_bins, embed_dim, kernel_size=3, padding=1)
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self.conv2 = nn.Conv1d(embed_dim, embed_dim, kernel_size=3, stride=2, padding=1)
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self.embed_positions = nn.Embedding(config.max_source_positions, embed_dim)
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self.layers = nn.ModuleList(
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[
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VoxtralWhisperEncoderLayer(config, quant_config)
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for _ in range(config.encoder_layers)
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]
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)
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self.layer_norm = nn.LayerNorm(embed_dim)
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def forward(self, input_features: torch.Tensor) -> torch.Tensor:
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"""
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Args:
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input_features: [batch, num_mel_bins, seq_len]
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Returns:
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[batch, seq_len // 2, d_model]
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"""
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inputs_embeds = torch.nn.functional.gelu(self.conv1(input_features))
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inputs_embeds = torch.nn.functional.gelu(self.conv2(inputs_embeds))
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inputs_embeds = inputs_embeds.permute(0, 2, 1)
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seq_len = inputs_embeds.shape[1]
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position_ids = torch.arange(seq_len, device=inputs_embeds.device)
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hidden_states = inputs_embeds + self.embed_positions(position_ids)
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for layer in self.layers:
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hidden_states = layer(hidden_states)
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hidden_states = self.layer_norm(hidden_states)
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return hidden_states
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class VoxtralForConditionalGeneration(nn.Module):
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"""Voxtral: Whisper encoder + MLP projector + Llama decoder.
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HF weight prefixes:
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audio_tower.* -> self.audio_tower (VoxtralWhisperEncoder)
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multi_modal_projector.* -> self.multi_modal_projector (AudioLanguageAdapter)
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language_model.* -> self.language_model (LlamaForCausalLM)
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"""
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def __init__(
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self,
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config: PretrainedConfig,
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quant_config: Optional[QuantizationConfig] = None,
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prefix: str = "",
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) -> None:
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super().__init__()
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self.config = config
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audio_config = config.audio_config
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text_config = config.text_config
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# Ensure text_config has rope_parameters (transformers v5 compatibility)
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if not hasattr(text_config, "rope_parameters"):
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text_config.rope_parameters = {
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"rope_type": getattr(text_config, "rope_type", "default"),
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"rope_theta": getattr(text_config, "rope_theta", 10000.0),
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}
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if getattr(text_config, "rope_scaling", None):
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text_config.rope_parameters.update(text_config.rope_scaling)
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# Infer downsample_factor: intermediate_size / hidden_size for HF format
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self.downsample_factor = getattr(
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audio_config,
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"downsample_factor",
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audio_config.intermediate_size // audio_config.hidden_size,
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)
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# Encoder (named audio_tower to match HF weight prefix directly)
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self.audio_tower = VoxtralWhisperEncoder(audio_config, quant_config)
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# Projector: input = d_model * downsample_factor, output = text_hidden_size
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adapter_input_dim = audio_config.d_model * self.downsample_factor
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self.multi_modal_projector = AudioLanguageAdapter(
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hidden_size=adapter_input_dim,
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dim=text_config.hidden_size,
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)
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# Language model
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self.language_model = LlamaForCausalLM(text_config, quant_config=quant_config)
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# Mel filter bank for raw waveform -> mel spectrogram
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self._init_mel_filters(audio_config)
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self.pattern = MultiModalityDataPaddingPatternMultimodalTokens()
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def _init_mel_filters(self, audio_config: PretrainedConfig):
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"""Initialize mel filter bank for mel spectrogram computation."""
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self._window_size = getattr(audio_config, "window_size", 400)
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self._hop_length = getattr(audio_config, "hop_length", 160)
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self._sampling_rate = getattr(audio_config, "sampling_rate", 16000)
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try:
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from mistral_common.audio import mel_filter_bank
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except ImportError:
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raise ImportError(
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"mistral_common is required for Voxtral. "
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"Install it with: pip install mistral_common"
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)
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mel_filters = mel_filter_bank(
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num_frequency_bins=1 + self._window_size // 2,
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num_mel_bins=audio_config.num_mel_bins,
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min_frequency=0.0,
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max_frequency=8000.0,
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sampling_rate=self._sampling_rate,
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)
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self.register_buffer(
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"mel_filters", torch.tensor(mel_filters, dtype=torch.float32)
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)
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@property
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def _conv_downsample_factor(self) -> int:
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return self.audio_tower.conv1.stride[0] * self.audio_tower.conv2.stride[0]
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@property
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def _chunk_size(self) -> int:
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return (
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self.config.audio_config.max_source_positions * self._conv_downsample_factor
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)
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def _compute_mel_spectrogram(self, audio_waveform: torch.Tensor) -> torch.Tensor:
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"""Compute log-mel spectrogram from raw waveform using STFT."""
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window = torch.hann_window(self._window_size, device=audio_waveform.device)
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stft = torch.stft(
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audio_waveform,
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self._window_size,
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self._hop_length,
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window=window,
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return_complex=True,
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)
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magnitudes = stft[..., :-1].abs() ** 2
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mel_spec = self.mel_filters.T @ magnitudes
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log_spec = torch.clamp(mel_spec, min=1e-10).log10()
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log_spec_max = log_spec.max()
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log_spec = torch.maximum(log_spec, log_spec_max - 8.0)
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log_spec = (log_spec + 4.0) / 4.0
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return log_spec
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def _encode_audio(self, audio_waveforms: List[torch.Tensor]) -> List[torch.Tensor]:
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"""Encode raw audio waveforms through mel spectrogram + whisper encoder."""
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dtype = self.audio_tower.conv1.weight.dtype
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device = self.audio_tower.conv1.weight.device
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chunked_features: List[torch.Tensor] = []
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chunks_per_example: List[int] = []
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chunk_size = self._chunk_size
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# Pad raw audio to a multiple of chunk_samples so that silence is
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# properly converted to mel features (matching HF VoxtralProcessor).
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chunk_samples = chunk_size * self._hop_length
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for waveform in audio_waveforms:
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waveform = waveform.to(device=device, dtype=torch.float32)
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n_samples = waveform.shape[-1]
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target_samples = chunk_samples * math.ceil(n_samples / chunk_samples)
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if target_samples > n_samples:
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waveform = torch.nn.functional.pad(
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waveform, (0, target_samples - n_samples)
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)
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mel = self._compute_mel_spectrogram(waveform)
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chunks = mel.split(chunk_size, dim=-1)
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chunked_features.extend(chunks)
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chunks_per_example.append(len(chunks))
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if not chunked_features:
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return []
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input_embeds = torch.stack(chunked_features).to(dtype)
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encoder_out = self.audio_tower(input_embeds)
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results = []
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chunk_idx = 0
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for n_chunks in chunks_per_example:
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result = encoder_out[chunk_idx : chunk_idx + n_chunks].flatten(0, 1)
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results.append(result)
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chunk_idx += n_chunks
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return results
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def pad_input_ids(self, input_ids: List[int], mm_inputs: MultimodalInputs):
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return self.pattern.pad_input_tokens(input_ids, mm_inputs)
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def get_audio_feature(self, items: List[MultimodalDataItem]) -> torch.Tensor:
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"""Encode audio waveforms -> downsample -> project."""
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audio_waveforms = [item.feature for item in items]
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audio_embeddings = self._encode_audio(audio_waveforms)
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# Downsample: reshape to merge adjacent frames
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for i, emb in enumerate(audio_embeddings):
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seq_len, dim = emb.shape
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audio_embeddings[i] = emb.reshape(
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seq_len // self.downsample_factor,
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dim * self.downsample_factor,
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)
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# Project through adapter
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packed = torch.cat(audio_embeddings, dim=0)
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packed = self.multi_modal_projector(packed)
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return packed
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def forward(
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self,
|
|
input_ids: torch.Tensor,
|
|
positions: torch.Tensor,
|
|
forward_batch: ForwardBatch,
|
|
**kwargs: Any,
|
|
) -> torch.Tensor:
|
|
hidden_states = general_mm_embed_routine(
|
|
input_ids=input_ids,
|
|
forward_batch=forward_batch,
|
|
language_model=self.language_model,
|
|
data_embedding_funcs={
|
|
Modality.AUDIO: self.get_audio_feature,
|
|
},
|
|
positions=positions,
|
|
)
|
|
return hidden_states
|
|
|
|
def get_language_model(self) -> nn.Module:
|
|
return self.language_model
|
|
|
|
def load_weights(self, weights: Iterable[Tuple[str, torch.Tensor]]):
|
|
encoder_stacked = [
|
|
("qkv_proj", "q_proj", "q"),
|
|
("qkv_proj", "k_proj", "k"),
|
|
("qkv_proj", "v_proj", "v"),
|
|
]
|
|
|
|
encoder_dict = dict(self.audio_tower.named_parameters())
|
|
projector_dict = dict(self.multi_modal_projector.named_parameters())
|
|
|
|
# Collect all weights; synthesise missing k_proj bias as zeros.
|
|
weights_list = list(weights)
|
|
extra_weights = []
|
|
for name, w in weights_list:
|
|
if name.startswith("audio_tower.") and ".self_attn.k_proj.weight" in name:
|
|
bias_name = name.replace(".weight", ".bias")
|
|
if not any(n == bias_name for n, _ in weights_list):
|
|
extra_weights.append(
|
|
(bias_name, torch.zeros(w.shape[0], dtype=w.dtype))
|
|
)
|
|
weights_list.extend(extra_weights)
|
|
|
|
def llm_weights_generator():
|
|
for name, w in weights_list:
|
|
# Encoder weights
|
|
if name.startswith("audio_tower."):
|
|
trimmed = name[len("audio_tower.") :]
|
|
loaded = False
|
|
for param_name, weight_name, shard_id in encoder_stacked:
|
|
if f".{weight_name}." in trimmed:
|
|
stacked_name = trimmed.replace(weight_name, param_name)
|
|
if stacked_name in encoder_dict:
|
|
param = encoder_dict[stacked_name]
|
|
param.weight_loader(param, w, shard_id)
|
|
loaded = True
|
|
break
|
|
if not loaded and trimmed in encoder_dict:
|
|
param = encoder_dict[trimmed]
|
|
weight_loader = getattr(
|
|
param, "weight_loader", default_weight_loader
|
|
)
|
|
weight_loader(param, w)
|
|
continue
|
|
|
|
# Projector weights
|
|
if name.startswith("multi_modal_projector."):
|
|
trimmed = name[len("multi_modal_projector.") :]
|
|
trimmed = trimmed.replace("linear_1.", "w_in.").replace(
|
|
"linear_2.", "w_out."
|
|
)
|
|
if trimmed in projector_dict:
|
|
param = projector_dict[trimmed]
|
|
default_weight_loader(param, w)
|
|
continue
|
|
|
|
# LLM weights
|
|
if name.startswith("language_model."):
|
|
name = name[len("language_model.") :]
|
|
yield (name, w)
|
|
|
|
self.language_model.load_weights(llm_weights_generator())
|
|
|
|
|
|
EntryClass = [VoxtralForConditionalGeneration]
|