transformers 5.0.0rc1__py3-none-any.whl → 5.0.0rc2__py3-none-any.whl

transformers/models/glmasr/modular_glmasr.py

@@ -0,0 +1,433 @@
+# coding=utf-8
+# Copyright 2025 the HuggingFace Team. 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.
+
+from collections.abc import Callable
+from typing import Optional, Union
+
+import numpy as np
+
+from ...activations import ACT2FN
+from ...audio_utils import AudioInput, make_list_of_audio
+from ...cache_utils import Cache
+from ...feature_extraction_utils import BatchFeature
+from ...modeling_layers import GradientCheckpointingLayer
+from ...modeling_outputs import BaseModelOutput, CausalLMOutputWithPast
+from ...modeling_utils import ALL_ATTENTION_FUNCTIONS
+from ...processing_utils import Unpack
+from ...utils import TransformersKwargs, auto_docstring, is_torch_available, logging
+from ...utils.generic import check_model_inputs
+from ..audioflamingo3.modeling_audioflamingo3 import (
+    AudioFlamingo3ForConditionalGeneration,
+    AudioFlamingo3MultiModalProjector,
+    AudioFlamingo3PreTrainedModel,
+)
+from ..audioflamingo3.processing_audioflamingo3 import AudioFlamingo3Processor, AudioFlamingo3ProcessorKwargs
+from ..glm.modeling_glm import GlmRotaryEmbedding
+from ..llama.modeling_llama import LlamaAttention, eager_attention_forward, rotate_half
+from .configuration_glmasr import GlmAsrConfig, GlmAsrEncoderConfig
+
+
+if is_torch_available():
+    import torch
+    from torch import nn
+
+
+logger = logging.get_logger(__name__)
+
+
+class GlmAsrProcessorKwargs(AudioFlamingo3ProcessorKwargs): ...
+
+
+class GlmAsrProcessor(AudioFlamingo3Processor):
+    r"""
+    Constructs an GlmAsr processor which wraps an GlmAsr feature extractor and an GlmAsr
+    tokenizer into a single processor.
+
+    [`GlmAsrProcessor`] offers all the functionalities of [`WhisperFeatureExtractor`] and
+    [`Qwen2TokenizerFast`]. See the [`~GlmAsrProcessor.__call__`] for more information.
+
+    Args:
+            feature_extractor ([`WhisperFeatureExtractor`]):
+                The feature extractor is a required input.
+            tokenizer ([`Qwen2TokenizerFast`]):
+                The tokenizer is a required input.
+            chat_template (`Optional[str]`, *optional*):
+                The Jinja template to use for formatting the conversation. If not provided, the tokenizer's default chat
+                template will be used.
+            audio_token (`Optional[str]`, *optional*, defaults to `"<|pad|>`"):
+                Special token used to represent audio inputs in the chat template.
+            default_transcription_prompt (`str`, *optional*, defaults to `"Please transcribe this audio into text"`):
+                Default prompt to use for transcription tasks when applying transcription requests.
+            max_audio_len (`int`, *optional*, defaults to 655):
+                Maximum length of audio sequences in seconds. Audio longer than this will be truncated.
+                655 gives approximately 8192 tokens, corresponding to the maximum sequence length of the text model.
+    """
+
+    def __init__(
+        self,
+        feature_extractor,
+        tokenizer,
+        chat_template=None,
+        audio_token="<|pad|>",
+        default_transcription_prompt="Please transcribe this audio into text",
+        max_audio_len=655,
+    ):
+        super().__init__(
+            feature_extractor,
+            tokenizer,
+            chat_template=chat_template,
+            audio_token=audio_token,
+            default_transcription_prompt=default_transcription_prompt,
+            max_audio_len=max_audio_len,
+        )
+
+    def _get_audio_token_length(self, audio_lengths: "torch.Tensor") -> "torch.Tensor":
+        merge_factor = 4
+        for padding, kernel_size, stride in [(1, 3, 1), (1, 3, 2)]:
+            audio_lengths = (audio_lengths + 2 * padding - (kernel_size - 1) - 1) // stride + 1
+
+        num_tokens = (audio_lengths - merge_factor) // merge_factor + 1
+        return num_tokens
+
+    def apply_transcription_request(
+        self,
+        audio: Union[str, list[str], AudioInput],
+        prompt: Optional[Union[str, list[str]]] = None,
+        **kwargs: Unpack[GlmAsrProcessorKwargs],
+    ) -> BatchFeature:
+        """
+        Prepare inputs for automatic speech recognition without manually writing the default transcription prompt.
+
+        Args:
+            audio (`str`, `list[str]`, `np.ndarray`, `torch.Tensor`, `list[np.ndarray]`, `list[torch.Tensor]`):
+                Audio to transcribe. Strings are interpreted as local paths or URLs and will be loaded automatically by
+                the chat template loader; NumPy arrays and PyTorch tensors are forwarded directly.
+            prompt (`str` or `list[str]`, *optional*):
+                Custom prompt(s) to include in the user turn. A list must be the same length as the batch. When `None`,
+                each sample uses `"Transcribe the input speech."`.
+            **kwargs:
+                Additional keyword arguments forwarded to [`~AudioFlamingo3Processor.apply_chat_template`] (for example
+                `text_kwargs`, `audio_kwargs`, ...).
+
+        Returns:
+            [`BatchFeature`]: Processor outputs ready to be passed to [`AudioFlamingo3ForConditionalGeneration.generate`].
+
+        """
+
+        if isinstance(audio, str):
+            audio_items: list[Union[str, np.ndarray]] = [audio]
+        elif isinstance(audio, (list, tuple)) and audio and all(isinstance(el, str) for el in audio):
+            audio_items = list(audio)
+        else:
+            audio_items = list(make_list_of_audio(audio))
+            if is_torch_available():
+                audio_items = [el.detach().cpu().numpy() if isinstance(el, torch.Tensor) else el for el in audio_items]
+
+        batch_size = len(audio_items)
+        if batch_size == 0:
+            raise ValueError("`audio` must contain at least one sample.")
+
+        if prompt is None:
+            prompts = [self.default_transcription_prompt] * batch_size
+        elif isinstance(prompt, str):
+            prompts = [prompt] * batch_size
+        elif isinstance(prompt, (list, tuple)):
+            if len(prompt) != batch_size:
+                raise ValueError(
+                    f"Received {len(prompt)} prompt(s) for {batch_size} audio sample(s); counts must match."
+                )
+            prompts = []
+            for item in prompt:
+                if item is None:
+                    prompts.append(self.default_transcription_prompt)
+                elif isinstance(item, str):
+                    prompts.append(item)
+                else:
+                    raise TypeError("Each prompt must be a string or `None`.")
+        else:
+            raise TypeError("`prompt` must be a string, a sequence of strings, or `None`.")
+
+        conversations = [
+            [
+                {
+                    "role": "user",
+                    "content": [
+                        {"type": "audio", "path": audio_item}
+                        if isinstance(audio_item, str)
+                        else {"type": "audio", "audio": audio_item},
+                        {"type": "text", "text": prompt_text},
+                    ],
+                }
+            ]
+            for prompt_text, audio_item in zip(prompts, audio_items)
+        ]
+
+        return self.apply_chat_template(
+            conversations,
+            tokenize=True,
+            add_generation_prompt=True,
+            return_dict=True,
+            **kwargs,
+        )
+
+
+class GlmAsrRotaryEmbedding(GlmRotaryEmbedding): ...
+
+
+def apply_rotary_pos_emb(q, k, cos, sin, position_ids=None, unsqueeze_dim=1):
+    cos = cos.unsqueeze(unsqueeze_dim)
+    sin = sin.unsqueeze(unsqueeze_dim)
+
+    rotary_dim = cos.shape[-1]
+    q_rot, q_pass = q[..., :rotary_dim], q[..., rotary_dim:]
+    k_rot, k_pass = k[..., :rotary_dim], k[..., rotary_dim:]
+
+    # Apply rotary embeddings on the first half or full tensor
+    q_embed = (q_rot * cos) + (rotate_half(q_rot) * sin)
+    k_embed = (k_rot * cos) + (rotate_half(k_rot) * sin)
+
+    # Concatenate back to full shape
+    q_embed = torch.cat([q_embed, q_pass], dim=-1)
+    k_embed = torch.cat([k_embed, k_pass], dim=-1)
+    return q_embed, k_embed
+
+
+class GlmAsrAttention(LlamaAttention):
+    def __init__(self, config: GlmAsrConfig, layer_idx: int):
+        super().__init__(config, layer_idx)
+        self.is_causal = False
+        self.q_proj = nn.Linear(config.hidden_size, config.num_attention_heads * self.head_dim, bias=True)
+        self.k_proj = nn.Linear(config.hidden_size, config.num_key_value_heads * self.head_dim, bias=False)
+        self.v_proj = nn.Linear(config.hidden_size, config.num_key_value_heads * self.head_dim, bias=True)
+        self.o_proj = nn.Linear(config.num_attention_heads * self.head_dim, config.hidden_size, bias=True)
+
+    def forward(
+        self,
+        hidden_states: torch.Tensor,
+        position_embeddings: Optional[tuple[torch.Tensor, torch.Tensor]] = None,
+        **kwargs: Unpack[TransformersKwargs],
+    ) -> tuple[torch.Tensor, torch.Tensor]:
+        input_shape = hidden_states.shape[:-1]
+        hidden_shape = (*input_shape, -1, self.head_dim)
+
+        query_states = self.q_proj(hidden_states).view(hidden_shape).transpose(1, 2)
+        key_states = self.k_proj(hidden_states).view(hidden_shape).transpose(1, 2)
+        value_states = self.v_proj(hidden_states).view(hidden_shape).transpose(1, 2)
+
+        cos, sin = position_embeddings
+        query_states, key_states = apply_rotary_pos_emb(query_states, key_states, cos, sin)
+
+        attention_interface: Callable = eager_attention_forward
+        if self.config._attn_implementation != "eager":
+            attention_interface = ALL_ATTENTION_FUNCTIONS[self.config._attn_implementation]
+
+        attn_output, attn_weights = attention_interface(
+            self,
+            query_states,
+            key_states,
+            value_states,
+            attention_mask=None,
+            dropout=0.0 if not self.training else self.attention_dropout,
+            scaling=self.scaling,
+            **kwargs,
+        )
+
+        attn_output = attn_output.reshape(*input_shape, -1).contiguous()
+        attn_output = self.o_proj(attn_output)
+        return attn_output, attn_weights
+
+
+class GlmAsrMLP(nn.Module):
+    def __init__(self, config):
+        super().__init__()
+        self.fc1 = nn.Linear(config.hidden_size, config.intermediate_size)
+        self.fc2 = nn.Linear(config.intermediate_size, config.hidden_size)
+        self.act_fn = ACT2FN[config.hidden_act]
+
+    def forward(self, hidden_states: torch.Tensor):
+        hidden_states = self.fc1(hidden_states)
+        hidden_states = self.act_fn(hidden_states)
+        hidden_states = self.fc2(hidden_states)
+        return hidden_states
+
+
+class GlmAsrEncoderLayer(GradientCheckpointingLayer):
+    def __init__(self, config: GlmAsrConfig, layer_idx: int):
+        super().__init__()
+        self.hidden_size = config.hidden_size
+
+        self.self_attn = GlmAsrAttention(config=config, layer_idx=layer_idx)
+
+        self.mlp = GlmAsrMLP(config)
+        self.input_layernorm = nn.LayerNorm(config.hidden_size)
+        self.post_attention_layernorm = nn.LayerNorm(config.hidden_size)
+
+    def forward(
+        self,
+        hidden_states: torch.Tensor,
+        position_embeddings: Optional[tuple[torch.Tensor, torch.Tensor]] = None,
+        **kwargs: Unpack[TransformersKwargs],
+    ) -> torch.Tensor:
+        residual = hidden_states
+        hidden_states = self.input_layernorm(hidden_states)
+        # Self Attention
+        hidden_states, _ = self.self_attn(
+            hidden_states=hidden_states,
+            position_embeddings=position_embeddings,
+            **kwargs,
+        )
+        hidden_states = residual + hidden_states
+
+        # Fully Connected
+        residual = hidden_states
+        hidden_states = self.post_attention_layernorm(hidden_states)
+        hidden_states = self.mlp(hidden_states)
+        hidden_states = residual + hidden_states
+        return hidden_states
+
+
+class GlmAsrPreTrainedModel(AudioFlamingo3PreTrainedModel): ...
+
+
+# TODO: @eustlb, this is what WhisperEncoder should look like
+class GlmAsrEncoder(GlmAsrPreTrainedModel):
+    config: GlmAsrEncoderConfig
+    main_input_name = "input_features"
+    input_modalities = "audio"
+    _no_split_modules = ["GlmAsrEncoderLayer"]
+
+    def __init__(self, config: GlmAsrEncoderConfig):
+        super().__init__(config)
+        self.conv1 = nn.Conv1d(config.num_mel_bins, config.hidden_size, kernel_size=3, padding=1)
+        self.conv2 = nn.Conv1d(config.hidden_size, config.hidden_size, kernel_size=3, stride=2, padding=1)
+
+        self.layers = nn.ModuleList(
+            [GlmAsrEncoderLayer(config, layer_idx) for layer_idx in range(config.num_hidden_layers)]
+        )
+        self.norm = nn.LayerNorm(config.hidden_size)
+        self.rotary_emb = GlmAsrRotaryEmbedding(config=config)
+        self.gradient_checkpointing = False
+        self.post_init()
+
+    @check_model_inputs
+    @auto_docstring
+    def forward(self, input_features, **kwargs: Unpack[TransformersKwargs]):
+        inputs_embeds = nn.functional.gelu(self.conv1(input_features))
+        inputs_embeds = nn.functional.gelu(self.conv2(inputs_embeds))
+        inputs_embeds = inputs_embeds.transpose(1, 2)
+
+        hidden_states = inputs_embeds
+        position_embeddings = self.rotary_emb(
+            hidden_states, position_ids=torch.arange(hidden_states.shape[1], device=hidden_states.device)[None, :]
+        )
+
+        for encoder_layer in self.layers:
+            hidden_states = encoder_layer(hidden_states, position_embeddings=position_embeddings, **kwargs)
+
+        hidden_states = self.norm(hidden_states)
+        return BaseModelOutput(last_hidden_state=hidden_states)
+
+
+class GlmAsrMultiModalProjector(AudioFlamingo3MultiModalProjector):
+    def __init__(self, config: GlmAsrConfig):
+        super().__init__()
+        self.linear_1 = nn.Linear(config.audio_config.intermediate_size, config.text_config.hidden_size * 2)
+        self.linear_2 = nn.Linear(config.text_config.hidden_size * 2, config.text_config.hidden_size)
+
+
+@auto_docstring(
+    custom_intro="""
+    The GlmAsr model which consists of a fine-tuned Whisper encoder, a multi-modal projector and a Llama language model.
+    """
+)
+class GlmAsrForConditionalGeneration(AudioFlamingo3ForConditionalGeneration):
+    def get_audio_features(
+        self, input_features: torch.FloatTensor, input_features_mask: torch.Tensor
+    ) -> torch.FloatTensor:
+        audio_outputs = self.audio_tower(input_features)
+        audio_hidden_states = audio_outputs.last_hidden_state
+        audio_hidden_states = audio_hidden_states.reshape(
+            input_features.shape[0], -1, self.config.audio_config.intermediate_size
+        )
+        audio_embeds = self.multi_modal_projector(audio_hidden_states)
+
+        audio_lengths = input_features_mask.sum(-1)
+        for padding, kernel_size, stride in [(1, 3, 1), (1, 3, 2)]:
+            audio_lengths = (audio_lengths + 2 * padding - (kernel_size - 1) - 1) // stride + 1
+        merge_factor = 4
+        post_lengths = (audio_lengths - merge_factor) // merge_factor + 1
+
+        valid_mask = torch.arange(audio_embeds.shape[1], device=post_lengths.device)[None, :] < post_lengths[:, None]
+        audio_embeds = audio_embeds[valid_mask.to(audio_embeds.device)]
+        return audio_embeds
+
+    def forward(
+        self,
+        input_ids: Optional[torch.LongTensor] = None,
+        input_features: Optional[torch.FloatTensor] = None,
+        input_features_mask: Optional[torch.Tensor] = None,
+        attention_mask: Optional[torch.Tensor] = None,
+        position_ids: Optional[torch.LongTensor] = None,
+        past_key_values: Optional[Cache] = None,
+        inputs_embeds: Optional[torch.FloatTensor] = None,
+        labels: Optional[torch.LongTensor] = None,
+        use_cache: Optional[bool] = None,
+        cache_position: Optional[torch.LongTensor] = None,
+        logits_to_keep: Union[int, torch.Tensor] = 0,
+        **kwargs: Unpack[TransformersKwargs],
+    ) -> CausalLMOutputWithPast:
+        r"""
+        input_features_mask (`torch.Tensor` of shape `(batch_size, feature_sequence_length)`):
+            Mask to avoid performing attention on padding feature indices. Mask values selected in `[0, 1]`:
+
+            - 1 for tokens that are **not masked**,
+            - 0 for tokens that are **masked**.
+        labels (`torch.LongTensor` of shape `(batch_size, sequence_length)`, *optional*):
+            Labels for computing the masked language modeling loss. Indices should either be in `[0, ...,
+            config.vocab_size]` or -100 (see `input_ids` docstring). Tokens with indices set to `-100` are ignored
+            (masked), the loss is only computed for the tokens with labels in `[0, ..., config.vocab_size]`.
+
+        Example:
+
+        ```python
+        >>> from transformers import GlmAsrForConditionalGeneration, AutoProcessor
+
+        >>> model_id = "zai-org/GLM-ASR-Nano-2512"
+        >>> processor = AutoProcessor.from_pretrained(model_id)
+        >>> model = GlmAsrForConditionalGeneration.from_pretrained(model_id, dtype="auto", device_map="auto")
+        >>> inputs = processor.apply_transcription_request("https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/bcn_weather.mp3")
+
+        >>> inputs = inputs.to(model.device, dtype=model.dtype)
+
+        >>> outputs = model.generate(**inputs, do_sample=False, max_new_tokens=500)
+
+        >>> decoded_outputs = processor.batch_decode(outputs[:, inputs.input_ids.shape[1] :], skip_special_tokens=True)
+        >>> print(decoded_outputs)
+        ```"""
+        return super().forward(
+            input_ids=input_ids,
+            attention_mask=attention_mask,
+            position_ids=position_ids,
+            past_key_values=past_key_values,
+            inputs_embeds=inputs_embeds,
+            labels=labels,
+            use_cache=use_cache,
+            cache_position=cache_position,
+            logits_to_keep=logits_to_keep,
+            **kwargs,
+        )
+
+
+__all__ = ["GlmAsrEncoder", "GlmAsrForConditionalGeneration", "GlmAsrProcessor", "GlmAsrPreTrainedModel"]