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+ ---
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+ library_name: transformers
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+ tags: []
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+ ---
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+ # Model Card for Model ID
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+ <!-- Provide a quick summary of what the model is/does. -->
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+ ## Model Details
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+ This is the model card of a 🤗 transformers model that has been pushed on the Hub. This model card has been automatically generated.
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+ ## How to Get Started with the Model
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+ "bos_token": "<|begin_of_text|>",
2053
+ "chat_template": "{{- bos_token }}\n{%- if custom_tools is defined %}\n {%- set tools = custom_tools %}\n{%- endif %}\n{%- if not tools_in_user_message is defined %}\n {%- set tools_in_user_message = true %}\n{%- endif %}\n{%- if not date_string is defined %}\n {%- if strftime_now is defined %}\n {%- set date_string = strftime_now(\"%d %b %Y\") %}\n {%- else %}\n {%- set date_string = \"26 Jul 2024\" %}\n {%- endif %}\n{%- endif %}\n{%- if not tools is defined %}\n {%- set tools = none %}\n{%- endif %}\n\n{#- This block extracts the system message, so we can slot it into the right place. #}\n{%- if messages[0]['role'] == 'system' %}\n {%- set system_message = messages[0]['content']|trim %}\n {%- set messages = messages[1:] %}\n{%- else %}\n {%- set system_message = \"\" %}\n{%- endif %}\n\n{#- System message #}\n{{- \"<|start_header_id|>system<|end_header_id|>\\n\\n\" }}\n{%- if tools is not none %}\n {{- \"Environment: ipython\\n\" }}\n{%- endif %}\n{{- \"Cutting Knowledge Date: December 2023\\n\" }}\n{{- \"Today Date: \" + date_string + \"\\n\\n\" }}\n{%- if tools is not none and not tools_in_user_message %}\n {{- \"You have access to the following functions. To call a function, please respond with JSON for a function call.\" }}\n {{- 'Respond in the format {\"name\": function name, \"parameters\": dictionary of argument name and its value}.' }}\n {{- \"Do not use variables.\\n\\n\" }}\n {%- for t in tools %}\n {{- t | tojson(indent=4) }}\n {{- \"\\n\\n\" }}\n {%- endfor %}\n{%- endif %}\n{{- system_message }}\n{{- \"<|eot_id|>\" }}\n\n{#- Custom tools are passed in a user message with some extra guidance #}\n{%- if tools_in_user_message and not tools is none %}\n {#- Extract the first user message so we can plug it in here #}\n {%- if messages | length != 0 %}\n {%- set first_user_message = messages[0]['content']|trim %}\n {%- set messages = messages[1:] %}\n {%- else %}\n {{- raise_exception(\"Cannot put tools in the first user message when there's no first user message!\") }}\n{%- endif %}\n {{- '<|start_header_id|>user<|end_header_id|>\\n\\n' -}}\n {{- \"Given the following functions, please respond with a JSON for a function call \" }}\n {{- \"with its proper arguments that best answers the given prompt.\\n\\n\" }}\n {{- 'Respond in the format {\"name\": function name, \"parameters\": dictionary of argument name and its value}.' }}\n {{- \"Do not use variables.\\n\\n\" }}\n {%- for t in tools %}\n {{- t | tojson(indent=4) }}\n {{- \"\\n\\n\" }}\n {%- endfor %}\n {{- first_user_message + \"<|eot_id|>\"}}\n{%- endif %}\n\n{%- for message in messages %}\n {%- if not (message.role == 'ipython' or message.role == 'tool' or 'tool_calls' in message) %}\n {{- '<|start_header_id|>' + message['role'] + '<|end_header_id|>\\n\\n'+ message['content'] | trim + '<|eot_id|>' }}\n {%- elif 'tool_calls' in message %}\n {%- if not message.tool_calls|length == 1 %}\n {{- raise_exception(\"This model only supports single tool-calls at once!\") }}\n {%- endif %}\n {%- set tool_call = message.tool_calls[0].function %}\n {{- '<|start_header_id|>assistant<|end_header_id|>\\n\\n' -}}\n {{- '{\"name\": \"' + tool_call.name + '\", ' }}\n {{- '\"parameters\": ' }}\n {{- tool_call.arguments | tojson }}\n {{- \"}\" }}\n {{- \"<|eot_id|>\" }}\n {%- elif message.role == \"tool\" or message.role == \"ipython\" %}\n {{- \"<|start_header_id|>ipython<|end_header_id|>\\n\\n\" }}\n {%- if message.content is mapping or message.content is iterable %}\n {{- message.content | tojson }}\n {%- else %}\n {{- message.content }}\n {%- endif %}\n {{- \"<|eot_id|>\" }}\n {%- endif %}\n{%- endfor %}\n{%- if add_generation_prompt %}\n {{- '<|start_header_id|>assistant<|end_header_id|>\\n\\n' }}\n{%- endif %}\n",
2054
+ "clean_up_tokenization_spaces": true,
2055
+ "eos_token": "<|eot_id|>",
2056
+ "model_input_names": [
2057
+ "input_ids",
2058
+ "attention_mask"
2059
+ ],
2060
+ "model_max_length": 131072,
2061
+ "pad_token": "<|eot_id|>",
2062
+ "tokenizer_class": "PreTrainedTokenizerFast"
2063
+ }
ultravox_config.py ADDED
@@ -0,0 +1,166 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ import dataclasses
2
+ from enum import Enum
3
+ from typing import Any, Dict, List, Optional
4
+
5
+ import transformers
6
+
7
+
8
+ @dataclasses.dataclass
9
+ class LoraConfigSimplified:
10
+ """
11
+ Low Rank Approximation (LoRA) configuration.
12
+
13
+ Used for language and audio models separately.
14
+ """
15
+
16
+ # The rank of the approximation
17
+ r: int = 0
18
+ lora_alpha: float = 8
19
+ target_modules: Optional[List[str]] = dataclasses.field(
20
+ default_factory=lambda: ["k_proj", "q_proj", "linear_k", "linear_q"]
21
+ )
22
+
23
+
24
+ class LossFunction(str, Enum):
25
+ CrossEntropy = "ce"
26
+ KL_Divergence = "kl"
27
+
28
+
29
+ @dataclasses.dataclass
30
+ class LossConfig:
31
+ loss_function: LossFunction = LossFunction.KL_Divergence
32
+ kl_temperature: float = 2.0
33
+
34
+ @property
35
+ def requires_alt_fields(self):
36
+ return self.loss_function == LossFunction.KL_Divergence
37
+
38
+
39
+ class UltravoxConfig(transformers.PretrainedConfig):
40
+ r"""
41
+ This is the configuration class to store the configuration of a [`UltravoxForConditionalGeneration`]. It is used to instantiate an
42
+ Ultravox model according to the specified arguments, defining the model architecture.
43
+
44
+ Configuration objects inherit from [`PretrainedConfig`] and can be used to control the model outputs. Read the
45
+ documentation from [`PretrainedConfig`] for more information.
46
+
47
+ Args:
48
+ audio_config (`Wav2Vec2Config`, *optional*):
49
+ Custom audio config or dict
50
+ text_config (`Union[AutoConfig, dict]`, *optional*):
51
+ The config object of the text backbone. Can be any of `LlamaConfig` or `MistralConfig`.
52
+ ignore_index (`int`, *optional*, defaults to -100):
53
+ The ignore index for the loss function.
54
+ audio_token_index (`int`, *optional*, defaults to 32000):
55
+ The audio token index to encode the audio prompt.
56
+ stack_factor (`int`, *optional*, defaults to 8):
57
+ Audio downsampling factor for the multimodal projector.
58
+ norm_init (`float`, *optional*, defaults to 0.4):
59
+ The initialization value for the layer normalization.
60
+ projector_act (`str`, *optional*, defaults to `"swiglu"`):
61
+ The activation function used by the multimodal projector.
62
+ text_model_lora_config (`LoraConfigSimplified`, *optional*):
63
+ The LoRA configuration for finetuning the text model.
64
+ audio_model_lora_config (`LoraConfigSimplified`, *optional*):
65
+ The LoRA configuration for finetuning the audio model.
66
+
67
+
68
+ Example:
69
+
70
+ ```python
71
+ >>> from transformers import UltravoxForConditionalGeneration, Wav2Vec2Config, UltravoxConfig, LlamaConfig
72
+
73
+ >>> # Initializing an audio encoder config
74
+ >>> audio_config = Wav2Vec2Config()
75
+
76
+ >>> # Initializing a Llama config
77
+ >>> text_config = LlamaConfig()
78
+
79
+ >>> # Initializing a default configuration
80
+ >>> configuration = UltravoxConfig(audio_config, text_config)
81
+
82
+ >>> # Initializing a completely untrained model from the configuration
83
+ >>> model = UltravoxForConditionalGeneration(configuration)
84
+
85
+ >>> # Accessing the model configuration
86
+ >>> configuration = model.config
87
+
88
+ >>> # Initialize a model from pretrained checkpoints and random projector weights
89
+ >>> config = UltravoxConfig(audio_model_id="facebook/wav2vec2-base-960h", text_model_id="meta-llama/Llama-2-7b-chat-hf")
90
+ ```"""
91
+
92
+ model_type = "ultravox"
93
+ is_composition = False
94
+
95
+ def __init__(
96
+ self,
97
+ audio_config: Optional[Dict[str, Any]] = None,
98
+ text_config: Optional[Dict[str, Any]] = None,
99
+ audio_model_id: Optional[str] = None,
100
+ text_model_id: Optional[str] = None,
101
+ ignore_index: int = -100,
102
+ hidden_size: int = 4096,
103
+ stack_factor: int = 8,
104
+ norm_init: float = 0.4,
105
+ projector_act: str = "swiglu",
106
+ text_model_lora_config: Optional[LoraConfigSimplified] = None,
107
+ audio_model_lora_config: Optional[LoraConfigSimplified] = None,
108
+ **kwargs,
109
+ ):
110
+ self.ignore_index = ignore_index
111
+
112
+ self.audio_model_id = audio_model_id
113
+ self.text_model_id = text_model_id
114
+
115
+ self.hidden_size = hidden_size
116
+ self.stack_factor = stack_factor
117
+ self.norm_init = norm_init
118
+ self.projector_act = projector_act
119
+
120
+ if text_model_id is not None:
121
+ self.text_config: transformers.LlamaConfig = (
122
+ transformers.AutoConfig.from_pretrained(text_model_id)
123
+ )
124
+ else:
125
+ text_config = text_config or {}
126
+ self.text_config = transformers.CONFIG_MAPPING[
127
+ text_config.get("model_type", "llama")
128
+ ](**text_config)
129
+
130
+ if audio_model_id is not None:
131
+ self.audio_config: transformers.PretrainedConfig = (
132
+ transformers.AutoConfig.from_pretrained(audio_model_id)
133
+ )
134
+ else:
135
+ audio_config = audio_config or {}
136
+ self.audio_config = transformers.CONFIG_MAPPING[
137
+ audio_config.get("model_type", "wav2vec2")
138
+ ](**audio_config)
139
+
140
+ self.text_model_lora_config = (
141
+ text_model_lora_config
142
+ if isinstance(text_model_lora_config, dict)
143
+ else dataclasses.asdict(text_model_lora_config or LoraConfigSimplified())
144
+ )
145
+ self.audio_model_lora_config = (
146
+ audio_model_lora_config
147
+ if isinstance(audio_model_lora_config, dict)
148
+ else dataclasses.asdict(audio_model_lora_config or LoraConfigSimplified())
149
+ )
150
+
151
+ self.vocab_size = self.text_config.vocab_size
152
+
153
+ self.initializer_range = self.text_config.initializer_range
154
+
155
+ super().__init__(**kwargs)
156
+
157
+ def to_diff_dict(self) -> Dict[str, Any]:
158
+ diff_dict = super().to_diff_dict()
159
+
160
+ # remove text_config and audio_config if text_model_id and audio_model_id are present
161
+ if self.text_model_id is not None:
162
+ diff_dict.pop("text_config", None)
163
+ if self.audio_model_id is not None:
164
+ diff_dict.pop("audio_config", None)
165
+
166
+ return diff_dict
ultravox_model.py ADDED
@@ -0,0 +1,644 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ import logging
2
+ from typing import Any, Dict, Optional, Set, Tuple, Union
3
+
4
+ import peft
5
+ import torch
6
+ import torch.nn as nn
7
+ import torch.nn.functional as F
8
+ import transformers
9
+ import transformers.activations
10
+ import transformers.modeling_outputs
11
+ import transformers.models
12
+ from transformers.models.whisper import modeling_whisper as whisper
13
+
14
+ # We must use relative import in this directory to allow uploading to HF Hub
15
+ # Even "from . import X" pattern doesn't work (undocumented and unclear why)
16
+ from .ultravox_config import LossConfig
17
+ from .ultravox_config import LossFunction
18
+ from .ultravox_config import UltravoxConfig
19
+
20
+
21
+ class UltravoxModel(transformers.LlamaPreTrainedModel):
22
+ """
23
+ The Ultravox model which consists of an audio encoder and a language model.
24
+
25
+ Audio input is processed by the audio encoder, then every `stack_factor` frames are stacked together and
26
+ projected to the language model's embedding space using a few linear layers.
27
+ The text is embedded by the language model as usual and then the audio and text embeddings are merged together.
28
+
29
+ A special token `<|audio|>` is used to indicate the start of the audio embeddings in the merged embeddings.
30
+
31
+ Parameters:
32
+ config: Model configuration class with all the parameters of the model.
33
+ """
34
+
35
+ config_class = UltravoxConfig
36
+ config: UltravoxConfig # for type hinting
37
+ # We minimize the weights in state_dict in order to reduce the size of the checkpoint
38
+ # The issue is that load_pretrained() uses state_dict() keys to know what keys are expected
39
+ # As such we have to tell is to ignore some keys that are not always in the model
40
+ _keys_to_ignore_on_load_unexpected = ["audio_tower.*", "language_model.*"]
41
+ # Usually we load encoder weights from a pretrained model, so we don't want to load the decoder weights
42
+ # Technically we never hit this issue because these keys are already removed from state_dict() however,
43
+ # but there's no harm in keeping it here for when we change that behavior.
44
+ _keys_to_ignore_on_load_missing = ["audio_tower.*"]
45
+
46
+ def __init__(self, config: UltravoxConfig):
47
+ super().__init__(config)
48
+ self._register_load_state_dict_pre_hook(self._pre_load_state_dict_hook)
49
+
50
+ self.keep_params: Set[str] = set()
51
+ self.vocab_size = config.vocab_size
52
+
53
+ self.audio_tower = self._create_audio_tower(config)
54
+ self.multi_modal_projector = self._create_multi_modal_projector(config)
55
+ self.language_model = self._create_language_model(config)
56
+
57
+ # Determine no_split_modules dynamically to use with FSDP auto_wrap policy.
58
+ # FSDP throws an error if some of the layer types are not found in the model.
59
+ # This would be something like ["LlamaDecoderLayer", "WhisperEncoderLayer"]
60
+ self._no_split_modules = (self.language_model._no_split_modules or []) + (
61
+ self.audio_tower._no_split_modules or []
62
+ )
63
+
64
+ self.loss_config = LossConfig()
65
+ self.post_init()
66
+
67
+ def get_input_embeddings(self):
68
+ return self.language_model.get_input_embeddings()
69
+
70
+ def set_input_embeddings(self, value):
71
+ self.language_model.set_input_embeddings(value)
72
+
73
+ def get_output_embeddings(self):
74
+ return self.language_model.get_output_embeddings()
75
+
76
+ def set_output_embeddings(self, new_embeddings):
77
+ self.language_model.set_output_embeddings(new_embeddings)
78
+
79
+ def set_decoder(self, decoder):
80
+ self.language_model.set_decoder(decoder)
81
+
82
+ def get_decoder(self):
83
+ return self.language_model.get_decoder()
84
+
85
+ def tie_weights(self):
86
+ return self.language_model.tie_weights()
87
+
88
+ def set_loss_config(self, loss_config: LossConfig):
89
+ self.loss_config = loss_config
90
+
91
+ def _setup_cache(
92
+ self, cache_cls, max_batch_size: int, max_cache_len: Optional[int] = None
93
+ ):
94
+ self.language_model._setup_cache(cache_cls, max_batch_size, max_cache_len)
95
+
96
+ def _reorder_cache(self, past_key_values, beam_idx):
97
+ return self.language_model._reorder_cache(past_key_values, beam_idx)
98
+
99
+ def resize_token_embeddings(
100
+ self,
101
+ new_num_tokens: Optional[int] = None,
102
+ pad_to_multiple_of: Optional[int] = None,
103
+ ) -> nn.Embedding:
104
+ model_embeds = self.language_model.resize_token_embeddings(
105
+ new_num_tokens, pad_to_multiple_of
106
+ )
107
+ # update vocab size
108
+ self.config.text_config.vocab_size = model_embeds.num_embeddings
109
+ self.config.vocab_size = model_embeds.num_embeddings
110
+ self.vocab_size = model_embeds.num_embeddings
111
+ return model_embeds
112
+
113
+ def _compute_kl_loss(
114
+ self,
115
+ lm_output: transformers.modeling_outputs.CausalLMOutputWithPast,
116
+ labels: Optional[torch.Tensor] = None,
117
+ past_key_values: Optional[Union[Tuple, transformers.cache_utils.Cache]] = None,
118
+ alt_input_ids: Optional[torch.Tensor] = None,
119
+ alt_attention_mask: Optional[torch.Tensor] = None,
120
+ alt_labels: Optional[torch.Tensor] = None,
121
+ **kwargs,
122
+ ):
123
+ # disable gradient computation for the teacher model
124
+ with torch.no_grad():
125
+ # compute the teacher (text-only) model's distribution
126
+ alt_inputs_embeds = self.get_input_embeddings().forward(alt_input_ids)
127
+ alt_lm_output = self.language_model.forward(
128
+ inputs_embeds=alt_inputs_embeds,
129
+ labels=alt_labels,
130
+ attention_mask=alt_attention_mask,
131
+ past_key_values=past_key_values,
132
+ **kwargs,
133
+ )
134
+ # compute the KL divergence loss between the two models
135
+ kl_loss = F.kl_div(
136
+ F.log_softmax(
137
+ lm_output.logits[labels != -100] / self.loss_config.kl_temperature,
138
+ dim=-1,
139
+ ),
140
+ F.softmax(
141
+ alt_lm_output.logits[alt_labels != -100]
142
+ / self.loss_config.kl_temperature,
143
+ dim=-1,
144
+ ),
145
+ reduction="batchmean",
146
+ )
147
+ return {"loss": kl_loss}
148
+
149
+ def forward(
150
+ self,
151
+ input_ids: torch.Tensor,
152
+ audio_values: Optional[torch.FloatTensor] = None,
153
+ inputs_embeds: Optional[torch.FloatTensor] = None,
154
+ labels: Optional[torch.Tensor] = None,
155
+ attention_mask: Optional[torch.Tensor] = None,
156
+ audio_token_start_idx: Optional[torch.Tensor] = None,
157
+ audio_token_len: Optional[torch.Tensor] = None,
158
+ past_key_values: Optional[Union[Tuple, transformers.cache_utils.Cache]] = None,
159
+ # the alt_* fields are needed for KL divergence loss
160
+ alt_input_ids: Optional[torch.Tensor] = None,
161
+ alt_attention_mask: Optional[torch.Tensor] = None,
162
+ alt_labels: Optional[torch.Tensor] = None,
163
+ **kwargs,
164
+ ) -> Union[Tuple, transformers.modeling_outputs.CausalLMOutputWithPast]:
165
+ """
166
+ Forward pass for the Ultravox model.
167
+
168
+ `input_ids` are the tokenized text input. They are embedded by the language model as usual.
169
+ `audio_values` are processed by the audio encoder and then every `stack_factor` frames are stacked together and
170
+ projected to the language model's embedding space using a few linear layers.
171
+ The audio and text embeddings are merged together. A special token `<|audio|>` is used to indicate the start
172
+ of the audio embeddings in the merged embeddings.
173
+
174
+ Args:
175
+ input_ids: The tokenized text input.
176
+ audio_values: The processed audio values.
177
+ inputs_embeds: The embeddings for the input tokens.
178
+ labels: The tokenized text labels.
179
+ attention_mask: The attention mask for the input.
180
+ position_ids: The position ids for the input.
181
+ past_key_values: The past key value cache for the language model attention layers.
182
+ **kwargs: Additional keyword arguments. Passed directly to the language model.
183
+ """
184
+ if inputs_embeds is None:
185
+ # B x T -> B x T x D
186
+ inputs_embeds = self.get_input_embeddings().forward(input_ids)
187
+
188
+ if audio_values is not None:
189
+ assert (
190
+ audio_token_start_idx is not None and audio_token_len is not None
191
+ ), "audio_token_start_idx and audio_token_len must be provided if audio_values are provided."
192
+ assert (
193
+ len(audio_token_start_idx) == len(audio_token_len) == len(audio_values)
194
+ ), "audio_token_start_idx, audio_token_len, and audio_values must have the same batch size."
195
+
196
+ # B x A/3200 x D
197
+ audio_tower_output = self.audio_tower.forward(
198
+ audio_values.to(self.audio_tower.dtype)
199
+ ).last_hidden_state
200
+ audio_tower_output = audio_tower_output.to(inputs_embeds.dtype)
201
+
202
+ audio_embeds = self.multi_modal_projector.forward(audio_tower_output)
203
+
204
+ # combine audio and text embeddings
205
+ for i, (audio, start, length) in enumerate(
206
+ zip(audio_embeds, audio_token_start_idx, audio_token_len)
207
+ ):
208
+ length = min(length, audio.shape[0])
209
+ inputs_embeds[i, start : start + length] = audio[:length]
210
+
211
+ lm_output = self.language_model.forward(
212
+ inputs_embeds=inputs_embeds,
213
+ labels=labels,
214
+ attention_mask=attention_mask,
215
+ past_key_values=past_key_values,
216
+ **kwargs,
217
+ )
218
+ if self.training:
219
+ if self.loss_config.loss_function == LossFunction.CrossEntropy:
220
+ return lm_output
221
+ elif self.loss_config.loss_function == LossFunction.KL_Divergence:
222
+ return self._compute_kl_loss(
223
+ lm_output=lm_output,
224
+ labels=labels,
225
+ past_key_values=past_key_values,
226
+ alt_input_ids=alt_input_ids,
227
+ alt_attention_mask=alt_attention_mask,
228
+ alt_labels=alt_labels,
229
+ **kwargs,
230
+ )
231
+ else:
232
+ raise ValueError(
233
+ f"Unsupported loss function: {self.loss_config.loss_function}"
234
+ )
235
+ else:
236
+ return lm_output
237
+
238
+ def prepare_inputs_for_generation(
239
+ self,
240
+ input_ids: torch.Tensor,
241
+ audio_values: Optional[torch.FloatTensor] = None,
242
+ audio_token_start_idx: Optional[torch.Tensor] = None,
243
+ audio_token_len: Optional[torch.Tensor] = None,
244
+ past_key_values: Optional[Union[Tuple, transformers.cache_utils.Cache]] = None,
245
+ attention_mask: Optional[torch.Tensor] = None,
246
+ inputs_embeds: Optional[torch.Tensor] = None,
247
+ cache_position: Optional[torch.Tensor] = None,
248
+ **kwargs,
249
+ ) -> Dict[str, Any]:
250
+ model_input = self.language_model.prepare_inputs_for_generation(
251
+ input_ids=input_ids,
252
+ past_key_values=past_key_values,
253
+ attention_mask=attention_mask,
254
+ inputs_embeds=inputs_embeds,
255
+ cache_position=cache_position,
256
+ **kwargs,
257
+ )
258
+
259
+ # include audio information in model_input only when it is needed during prefilling
260
+ # audio_token_start_idx should always be relative to the current cache position
261
+ prefill_start_idx = 0 if cache_position is None else cache_position[0]
262
+ if (
263
+ audio_values is not None
264
+ and audio_token_start_idx is not None
265
+ and prefill_start_idx <= torch.max(audio_token_start_idx)
266
+ ):
267
+ model_input["audio_values"] = audio_values
268
+ model_input["audio_token_start_idx"] = (
269
+ audio_token_start_idx - prefill_start_idx
270
+ )
271
+ model_input["audio_token_len"] = audio_token_len
272
+
273
+ return model_input
274
+
275
+ @classmethod
276
+ def _create_multi_modal_projector(
277
+ cls, config: UltravoxConfig
278
+ ) -> "UltravoxProjector":
279
+ projector = UltravoxProjector(config)
280
+ projector.to(config.torch_dtype)
281
+ return projector
282
+
283
+ @classmethod
284
+ def _create_audio_tower(
285
+ cls, config: UltravoxConfig
286
+ ) -> Union[transformers.Wav2Vec2Model, "ModifiedWhisperEncoder"]:
287
+ if config.audio_model_id is not None:
288
+ if "whisper" in config.audio_model_id is not None:
289
+ audio_tower = ModifiedWhisperEncoder.from_pretrained(
290
+ config.audio_model_id, torch_dtype=config.torch_dtype
291
+ )
292
+ else:
293
+ audio_tower = transformers.AutoModel.from_pretrained(
294
+ config.audio_model_id, torch_dtype=config.torch_dtype
295
+ )
296
+ else:
297
+ if "whisper" in config.audio_config._name_or_path:
298
+ audio_tower = ModifiedWhisperEncoder(config.audio_config)
299
+ else:
300
+ with transformers.modeling_utils.no_init_weights():
301
+ # we only ever use from_config if the weights are retrained, hence initializing is not
302
+ # required. This makes the model quite creation faster since init on CPU is quite slow.
303
+ audio_tower = transformers.AutoModel.from_config(
304
+ config.audio_config
305
+ )
306
+
307
+ if isinstance(
308
+ audio_tower,
309
+ (transformers.Wav2Vec2BertModel, transformers.WhisperModel),
310
+ ):
311
+ # For these models we only need the encoder part
312
+ # Wav2Vec2BertModel -> Wav2Vec2BertEncoder
313
+ # WhisperModel -> WhisperEncoder
314
+ audio_tower = audio_tower.encoder
315
+
316
+ audio_tower = apply_lora(audio_tower, config.audio_model_lora_config)
317
+ return audio_tower
318
+
319
+ @classmethod
320
+ def _create_language_model(
321
+ cls, config: UltravoxConfig
322
+ ) -> transformers.LlamaForCausalLM:
323
+ if config.text_model_id is not None:
324
+ language_model = transformers.AutoModelForCausalLM.from_pretrained(
325
+ config.text_model_id,
326
+ attn_implementation=config._attn_implementation,
327
+ torch_dtype=config.torch_dtype,
328
+ )
329
+ else:
330
+ with transformers.modeling_utils.no_init_weights():
331
+ # we only ever use from_config if the weights are retrained, hence initializing is not
332
+ # required. This makes the model quite creation faster since init on CPU is quite slow.
333
+ language_model = transformers.AutoModelForCausalLM.from_config(
334
+ config.text_config,
335
+ attn_implementation=config._attn_implementation,
336
+ torch_dtype=config.torch_dtype,
337
+ )
338
+
339
+ language_model = apply_lora(language_model, config.text_model_lora_config)
340
+ return language_model
341
+
342
+ def merge_and_unload(self):
343
+ if isinstance(self.language_model, peft.PeftModel):
344
+ self.language_model = self.language_model.merge_and_unload()
345
+ # no need to download base language model weights anymore, so we can remove the id
346
+ self.config.text_model_id = None
347
+ self.keep_params.update(
348
+ set(
349
+ [
350
+ f"language_model.{name}"
351
+ for name, _ in self.language_model.named_parameters()
352
+ ]
353
+ )
354
+ )
355
+
356
+ if isinstance(self.audio_tower, peft.PeftModel):
357
+ self.audio_tower = self.audio_tower.merge_and_unload()
358
+ # no need to download base audio model weights anymore, so we can remove the id
359
+ self.config.audio_model_id = None
360
+ self.keep_params.update(
361
+ set(
362
+ [
363
+ f"audio_tower.{name}"
364
+ for name, _ in self.audio_tower.named_parameters()
365
+ ]
366
+ )
367
+ )
368
+
369
+ for param in ["text_model_lora_config", "audio_model_lora_config"]:
370
+ if hasattr(self.config, param):
371
+ delattr(self.config, param)
372
+
373
+ def push_to_hub(self, *args, **kwargs):
374
+ self.merge_and_unload()
375
+ self.to(self.language_model.dtype)
376
+ return super().push_to_hub(*args, **kwargs)
377
+
378
+ def save_pretrained(
379
+ self, *args, state_dict: Optional[Dict[str, Any]] = None, **kwargs
380
+ ):
381
+ if state_dict is None:
382
+ state_dict = super().state_dict()
383
+
384
+ named_params = dict(self.named_parameters())
385
+
386
+ state_dict = {
387
+ k: v
388
+ for k, v in state_dict.items()
389
+ if k in self.keep_params
390
+ or (k in named_params and named_params[k].requires_grad)
391
+ }
392
+
393
+ super().save_pretrained(*args, state_dict=state_dict, **kwargs)
394
+
395
+ def _pre_load_state_dict_hook(self, state_dict: Dict[str, Any], *args, **kwargs):
396
+ self.keep_params.update(set(state_dict.keys()))
397
+
398
+ def print_trainable_parameters(self):
399
+ """
400
+ Prints the number of trainable parameters in the model (reuses Peft model's method)
401
+ """
402
+ count_params = peft.peft_model.PeftModel.get_nb_trainable_parameters
403
+
404
+ trainable_params, all_param = count_params(self)
405
+
406
+ logging.info(
407
+ f"trainable params: {trainable_params:,d} || all params: {all_param:,d}"
408
+ f" || trainable%: {100 * trainable_params / all_param:.1f}%"
409
+ )
410
+
411
+ lm_trainable_params, lm_all_params = count_params(self.language_model)
412
+ audio_trainable_params, audio_all_params = count_params(self.audio_tower)
413
+
414
+ projector_trainable_params = (
415
+ trainable_params - lm_trainable_params - audio_trainable_params
416
+ )
417
+ projector_all_params = all_param - lm_all_params - audio_all_params
418
+
419
+ logging.info(
420
+ f"Trainable%: "
421
+ f" LLM: {100 * lm_trainable_params / lm_all_params:.1f}%"
422
+ f" || Audio Encoder: {100 * audio_trainable_params / audio_all_params:.1f}%"
423
+ f" || Projector: {100 * projector_trainable_params / projector_all_params:.1f}%"
424
+ )
425
+
426
+
427
+ def is_cache_empty(
428
+ past_key_values: Optional[Union[Tuple, transformers.cache_utils.Cache]]
429
+ ) -> bool:
430
+ """
431
+ Check if the cache is empty.
432
+ """
433
+ if past_key_values is None:
434
+ return True
435
+ if isinstance(past_key_values, tuple):
436
+ return all(len(c) == 0 for c in past_key_values)
437
+ return past_key_values.get_seq_length() == 0
438
+
439
+
440
+ def apply_lora(model: torch.nn.Module, lora_config: dict) -> torch.nn.Module:
441
+ """
442
+ Applies LoRA finetuning to the model. If the `r` parameter is set to 0, the model is frozen instead.
443
+ """
444
+ lora_config = peft.LoraConfig(**lora_config or {})
445
+
446
+ if lora_config.r == 0:
447
+ # freeze the model entirely
448
+ for param in model.parameters():
449
+ param.requires_grad = False
450
+ else:
451
+ model = peft.get_peft_model(model, lora_config)
452
+
453
+ return model
454
+
455
+
456
+ class StackAudioFrames(nn.Module):
457
+ """
458
+ Stack the audio embedding frames to reduce the sequence length by a factor of `stack_factor`.
459
+
460
+ The number of output frames will be `ceil(T / stack_factor) + 1` where `T` is the number of input frames.
461
+ NOTE: the extra +1 is intentional: in case the number of audio tokens are over-estimated by the processor,
462
+ we want to make sure `processor.audio_token_replacement` (i.e. EOS) doesn't get leaked into the middle of embeddings.
463
+ In most cases this extra padding will get removed in the model's forward function so it has no effect.
464
+ """
465
+
466
+ def __init__(self, stack_factor: int = 8):
467
+ super().__init__()
468
+ self.stack_factor = stack_factor
469
+
470
+ def forward(self, audio_embeds: torch.Tensor) -> torch.Tensor:
471
+ B, T, C = audio_embeds.shape
472
+ T_pad = (T + self.stack_factor - 1) // self.stack_factor * self.stack_factor
473
+ audio_embeds = F.pad(audio_embeds, (0, 0, 0, T_pad - T + self.stack_factor))
474
+ B, T, C = audio_embeds.shape
475
+ audio_embeds = audio_embeds.view(
476
+ B, T // self.stack_factor, C * self.stack_factor
477
+ )
478
+ return audio_embeds
479
+
480
+
481
+ class RMSNorm(transformers.models.llama.modeling_llama.LlamaRMSNorm):
482
+ def __init__(self, hidden_size: int, init: float = 1, eps: float = 1e-6):
483
+ super().__init__(hidden_size=hidden_size, eps=eps)
484
+ self.weight.data.fill_(init)
485
+
486
+
487
+ class SwiGLU(nn.Module):
488
+ def forward(self, x):
489
+ x, gate = x.chunk(2, dim=-1)
490
+ return F.silu(gate) * x
491
+
492
+
493
+ class UltravoxProjector(nn.Sequential):
494
+ def __init__(self, config: UltravoxConfig):
495
+ super().__init__()
496
+ self.hidden_dim = config.hidden_size
497
+ self._pad_and_stack = StackAudioFrames(config.stack_factor)
498
+ dim = config.audio_config.hidden_size * config.stack_factor
499
+ self.ln_pre = RMSNorm(dim, init=config.norm_init)
500
+ self.linear_1 = nn.Linear(dim, self.hidden_dim, bias=False)
501
+ dim = self.hidden_dim
502
+ self.act = transformers.activations.get_activation(config.projector_act)
503
+ dim = dim // 2 if config.projector_act == "swiglu" else dim
504
+ self.linear_2 = nn.Linear(dim, config.text_config.hidden_size, bias=False)
505
+ self.ln_post = RMSNorm(config.text_config.hidden_size, init=config.norm_init)
506
+
507
+ def forward(self, audio_features: torch.Tensor) -> torch.Tensor:
508
+ audio_features = self._pad_and_stack(audio_features)
509
+ audio_features = self.ln_pre(audio_features)
510
+ hidden_states = self.linear_1(audio_features)
511
+ hidden_states = self.act(hidden_states)
512
+ hidden_states = self.linear_2(hidden_states)
513
+ hidden_states = self.ln_post(hidden_states)
514
+ return hidden_states
515
+
516
+
517
+ class ModifiedWhisperEncoder(whisper.WhisperEncoder):
518
+ """
519
+ Encoder portion of OpenAI's Whisper model.
520
+
521
+ This implementation is a slightly modified version of HF Transformers' Whisper Encoder, with only a few fixes:
522
+ 1. base_model_prefix updated to allow for doing `.from_pretrained` directly on the encoder
523
+ 2. allow less than 30 second of audio padding to be passed in:
524
+ - relaxed ValueError check for `input_features` length to be less than or equal to `expected_seq_length` instead of strictly equal
525
+ - embed_pos is now sliced to match the length of `inputs_embeds`
526
+
527
+ Original: https://github.com/huggingface/transformers/blob/main/src/transformers/models/whisper/modeling_whisper.py
528
+ """
529
+
530
+ base_model_prefix = "model.encoder"
531
+ _no_split_modules = ["WhisperEncoderLayer"]
532
+
533
+ def forward(
534
+ self,
535
+ input_features,
536
+ attention_mask=None,
537
+ head_mask=None,
538
+ output_attentions=None,
539
+ output_hidden_states=None,
540
+ return_dict=None,
541
+ ):
542
+ expected_seq_length = (
543
+ self.config.max_source_positions
544
+ * self.conv1.stride[0]
545
+ * self.conv2.stride[0]
546
+ )
547
+ if input_features.shape[-1] > expected_seq_length:
548
+ raise ValueError(
549
+ f"Whisper expects the mel input features to be of length {expected_seq_length} or less, but found {input_features.shape[-1]}. Make sure to pad the input mel features to {expected_seq_length}."
550
+ )
551
+
552
+ output_attentions = (
553
+ output_attentions
554
+ if output_attentions is not None
555
+ else self.config.output_attentions
556
+ )
557
+ output_hidden_states = (
558
+ output_hidden_states
559
+ if output_hidden_states is not None
560
+ else self.config.output_hidden_states
561
+ )
562
+ return_dict = (
563
+ return_dict if return_dict is not None else self.config.use_return_dict
564
+ )
565
+ inputs_embeds = nn.functional.gelu(self.conv1(input_features))
566
+ inputs_embeds = nn.functional.gelu(self.conv2(inputs_embeds))
567
+
568
+ inputs_embeds = inputs_embeds.permute(0, 2, 1)
569
+ embed_pos = self.embed_positions.weight[: inputs_embeds.size(-2)]
570
+
571
+ hidden_states = inputs_embeds + embed_pos
572
+ hidden_states = nn.functional.dropout(
573
+ hidden_states, p=self.dropout, training=self.training
574
+ )
575
+
576
+ encoder_states = () if output_hidden_states else None
577
+ all_attentions = () if output_attentions else None
578
+
579
+ # check if head_mask has a correct number of layers specified if desired
580
+ if head_mask is not None:
581
+ assert head_mask.size()[0] == (
582
+ len(self.layers)
583
+ ), f"The head_mask should be specified for {len(self.layers)} layers, but it is for {head_mask.size()[0]}."
584
+
585
+ for idx, encoder_layer in enumerate(self.layers):
586
+ if output_hidden_states:
587
+ encoder_states = encoder_states + (hidden_states,)
588
+ # add LayerDrop (see https://arxiv.org/abs/1909.11556 for description)
589
+ to_drop = False
590
+ if self.training:
591
+ dropout_probability = torch.rand([])
592
+ if dropout_probability < self.layerdrop: # skip the layer
593
+ to_drop = True
594
+
595
+ if to_drop:
596
+ layer_outputs = (None, None)
597
+ else:
598
+ if self.gradient_checkpointing and self.training:
599
+ layer_outputs = self._gradient_checkpointing_func(
600
+ encoder_layer.__call__,
601
+ hidden_states,
602
+ None,
603
+ (head_mask[idx] if head_mask is not None else None),
604
+ output_attentions,
605
+ )
606
+ else:
607
+ layer_outputs = encoder_layer(
608
+ hidden_states,
609
+ None,
610
+ layer_head_mask=(
611
+ head_mask[idx] if head_mask is not None else None
612
+ ),
613
+ output_attentions=output_attentions,
614
+ )
615
+
616
+ hidden_states = layer_outputs[0]
617
+
618
+ if output_attentions:
619
+ all_attentions = all_attentions + (layer_outputs[1],)
620
+
621
+ hidden_states = self.layer_norm(hidden_states)
622
+ if output_hidden_states:
623
+ encoder_states = encoder_states + (hidden_states,)
624
+
625
+ if not return_dict:
626
+ return tuple(
627
+ v
628
+ for v in [hidden_states, encoder_states, all_attentions]
629
+ if v is not None
630
+ )
631
+ return transformers.modeling_outputs.BaseModelOutput(
632
+ last_hidden_state=hidden_states,
633
+ hidden_states=encoder_states,
634
+ attentions=all_attentions,
635
+ )
636
+
637
+
638
+ UltravoxConfig.register_for_auto_class()
639
+ UltravoxModel.register_for_auto_class()
640
+
641
+ transformers.AutoConfig.register("ultravox", UltravoxConfig)
642
+ transformers.AutoModel.register(UltravoxConfig, UltravoxModel)
643
+
644
+ transformers.activations.ACT2FN["swiglu"] = SwiGLU
ultravox_pipeline.py ADDED
@@ -0,0 +1,127 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ import logging
2
+ from typing import Any, Dict, List, Optional
3
+
4
+ import numpy as np
5
+ import transformers
6
+
7
+ # We must use relative import in this directory to allow uploading to HF Hub
8
+ # Even "from . import X" pattern doesn't work (undocumented and unclear why)
9
+ from .ultravox_model import UltravoxModel
10
+ from .ultravox_processing import UltravoxProcessor
11
+
12
+
13
+ class UltravoxPipeline(transformers.Pipeline):
14
+ def __init__(
15
+ self,
16
+ model: UltravoxModel,
17
+ tokenizer: Optional[transformers.PreTrainedTokenizerBase] = None,
18
+ audio_processor: Optional[transformers.ProcessorMixin] = None,
19
+ **kwargs
20
+ ):
21
+ if tokenizer is None:
22
+ try:
23
+ tokenizer = transformers.AutoTokenizer.from_pretrained(
24
+ model.config._name_or_path
25
+ )
26
+ except:
27
+ tokenizer = transformers.AutoTokenizer.from_pretrained(
28
+ model.config.text_model_id or model.config.text_config._name_or_path
29
+ )
30
+
31
+ if audio_processor is None:
32
+ audio_processor = transformers.AutoProcessor.from_pretrained(
33
+ model.config.audio_model_id or model.config.audio_config._name_or_path
34
+ )
35
+
36
+ self.processor = UltravoxProcessor(
37
+ audio_processor=audio_processor,
38
+ tokenizer=tokenizer,
39
+ stack_factor=model.config.stack_factor,
40
+ )
41
+
42
+ super().__init__(model=model, tokenizer=tokenizer, **kwargs)
43
+
44
+ def _sanitize_parameters(self, **kwargs):
45
+ generation_keys = ["temperature", "max_new_tokens", "repetition_penalty"]
46
+ generation_kwargs = {k: kwargs[k] for k in kwargs if k in generation_keys}
47
+ return {}, generation_kwargs, {}
48
+
49
+ def preprocess(self, inputs: Dict[str, Any]):
50
+ turns: list = inputs.get("turns", [])
51
+
52
+ audio = inputs.get("audio", None)
53
+ # Convert to float32 if needed.
54
+ if isinstance(audio, np.ndarray):
55
+ if audio.dtype == np.float64:
56
+ audio = audio.astype(np.float32)
57
+ elif audio.dtype == np.int16:
58
+ audio = audio.astype(np.float32) / np.float32(32768.0)
59
+ elif audio.dtype == np.int32:
60
+ audio = audio.astype(np.float32) / np.float32(2147483648.0)
61
+
62
+ if audio is not None and (len(turns) == 0 or turns[-1]["role"] != "user"):
63
+ prompt = inputs.get("prompt", "<|audio|>")
64
+ if "<|audio|>" not in prompt:
65
+ logging.warning(
66
+ "Prompt does not contain '<|audio|>', appending '<|audio|>' to the end of the prompt."
67
+ )
68
+
69
+ prompt += " <|audio|>"
70
+ turns.append({"role": "user", "content": prompt})
71
+
72
+ text = self.processor.tokenizer.apply_chat_template(
73
+ turns, add_generation_prompt=True, tokenize=False
74
+ )
75
+
76
+ if "sampling_rate" not in inputs and audio is not None:
77
+ logging.warning(
78
+ "No sampling rate provided, using default of 16kHz. We highly recommend providing the correct sampling rate."
79
+ )
80
+
81
+ output = self.processor(
82
+ text=text,
83
+ audio=audio,
84
+ sampling_rate=inputs.get("sampling_rate", 16000),
85
+ )
86
+ if "audio_values" in output:
87
+ output["audio_values"] = output["audio_values"].to(self.model.dtype)
88
+
89
+ return output
90
+
91
+ def _forward(
92
+ self,
93
+ model_inputs: Dict[str, Any],
94
+ temperature: Optional[float] = None,
95
+ max_new_tokens: Optional[int] = None,
96
+ repetition_penalty: float = 1.1,
97
+ ) -> List[int]:
98
+ temperature = temperature or None
99
+ do_sample = temperature is not None
100
+
101
+ terminators = [self.tokenizer.eos_token_id]
102
+ if "<|eot_id|>" in self.tokenizer.added_tokens_encoder:
103
+ terminators.append(self.tokenizer.convert_tokens_to_ids("<|eot_id|>"))
104
+
105
+ input_len = model_inputs["input_ids"].shape[1]
106
+
107
+ outputs = self.model.generate(
108
+ **model_inputs,
109
+ do_sample=do_sample,
110
+ temperature=temperature,
111
+ max_new_tokens=max_new_tokens,
112
+ repetition_penalty=repetition_penalty,
113
+ eos_token_id=terminators
114
+ )
115
+ return outputs[0][input_len:]
116
+
117
+ def postprocess(self, model_outputs) -> str:
118
+ output_text = self.tokenizer.decode(model_outputs, skip_special_tokens=True)
119
+ return output_text
120
+
121
+
122
+ transformers.pipelines.PIPELINE_REGISTRY.register_pipeline(
123
+ "ultravox-pipeline",
124
+ pipeline_class=UltravoxPipeline,
125
+ pt_model=transformers.AutoModel,
126
+ type="multimodal",
127
+ )
ultravox_processing.py ADDED
@@ -0,0 +1,210 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ from typing import Optional, Union
2
+
3
+ import numpy as np
4
+ import torch
5
+ import transformers
6
+
7
+ from .ultravox_config import UltravoxConfig
8
+
9
+
10
+ class UltravoxProcessor(transformers.ProcessorMixin):
11
+ """
12
+ Constructs an Ultravox processor which wraps an audio processor and a tokenizer into a single processor.
13
+
14
+ Args:
15
+ audio_processor: The audio processor for the audio encoder.
16
+ tokenizer: The tokenizer for the language model.
17
+ """
18
+
19
+ attributes = ["audio_processor", "tokenizer"]
20
+ audio_processor_class = (
21
+ "Wav2Vec2Processor",
22
+ "SeamlessM4TFeatureExtractor",
23
+ "WhisperProcessor",
24
+ )
25
+ tokenizer_class = (
26
+ "PreTrainedTokenizer",
27
+ "PreTrainedTokenizerFast",
28
+ )
29
+
30
+ tokenizer: transformers.PreTrainedTokenizerBase
31
+ audio_processor: transformers.ProcessorMixin
32
+
33
+ def __init__(
34
+ self,
35
+ audio_processor=None,
36
+ tokenizer=None,
37
+ audio_padding: str = "longest",
38
+ encoder_ds_factor: int = 320,
39
+ stack_factor: int = 8,
40
+ audio_placeholder: str = "<|audio|>",
41
+ ):
42
+ """
43
+ Args:
44
+ audio_processor: The audio processor for the audio encoder.
45
+ tokenizer: The tokenizer for the language model.
46
+ audio_padding: The padding strategy for the audio encoder.
47
+ encoder_ds_factor: The downsample factor of the audio encoder.
48
+ stack_factor: The factor by which the audio encoder output is stacked in the multimodal projector.
49
+ audio_placeholder: The placeholder for the audio in the text.
50
+ """
51
+ self.audio_padding = audio_padding
52
+ self.encoder_ds_factor = encoder_ds_factor
53
+ self.stack_factor = stack_factor
54
+ self.audio_placeholder = audio_placeholder
55
+ self.audio_token_replacement = tokenizer.eos_token
56
+ assert (
57
+ self.audio_token_replacement is not None
58
+ ), "The tokenizer has no EOS token. Cannot recover."
59
+ if tokenizer.pad_token_id is None:
60
+ tokenizer.pad_token_id = tokenizer.eos_token_id
61
+
62
+ super().__init__(audio_processor=audio_processor, tokenizer=tokenizer)
63
+
64
+ @classmethod
65
+ def from_pretrained(cls, pretrained_model_name_or_path, **kwargs):
66
+ config: UltravoxConfig = transformers.AutoConfig.from_pretrained(
67
+ pretrained_model_name_or_path, **kwargs
68
+ )
69
+ audio_processor = transformers.AutoProcessor.from_pretrained(
70
+ config.audio_model_id
71
+ or config.audio_config._name_or_path
72
+ or "facebook/wav2vec2-base-960h"
73
+ )
74
+
75
+ tokenizer = transformers.AutoTokenizer.from_pretrained(
76
+ pretrained_model_name_or_path, **kwargs
77
+ )
78
+ tokenizer.padding_side = "left"
79
+ tokenizer.pad_token = tokenizer.eos_token
80
+
81
+ return cls(
82
+ audio_processor=audio_processor,
83
+ tokenizer=tokenizer,
84
+ stack_factor=config.stack_factor,
85
+ )
86
+
87
+ def __call__(
88
+ self,
89
+ text: Optional[str] = None,
90
+ audio: Optional[Union[np.ndarray, torch.Tensor]] = None,
91
+ sampling_rate: Optional[int] = None,
92
+ return_tensors: Optional[
93
+ Union[str, transformers.TensorType]
94
+ ] = transformers.TensorType.PYTORCH,
95
+ **kwargs,
96
+ ) -> transformers.BatchFeature:
97
+ """
98
+ Main method to prepare for the model one text sequence and audio. This method forwards the `text`
99
+ and `kwargs` arguments to PreTrainedTokenizerFast's [`~PreTrainedTokenizerFast.__call__`] if `text` is not `None` to encode
100
+ the text. To prepare the audio(s), this method forwards the `audio`, `sampling_rate` and `kwargs` arguments to
101
+ audio processor's [`~Wav2Vec2Processor.__call__`] if `audio` is not `None`. Please refer to the docstring
102
+ of the above two methods for more information.
103
+
104
+ Args:
105
+ text (`str`, `List[str]`):
106
+ The sequence to be encoded. Sequence can be a string or (pretokenized string).
107
+ audio (`np.ndarray`, `torch.Tensor`, `List[np.ndarray]`, `List[torch.Tensor]`):
108
+ The audio to be prepared. Audio can be NumPy array or PyTorch tensor. In case of a
109
+ NumPy array/PyTorch tensor, each audio should be of shape (C, T), where C is a number of channels, and T the
110
+ sample length of the audio.
111
+ sampling_rate (`int`, *optional*, defaults to 16000):
112
+ Sampling rate of the input audio. We expect 16kHz audio. Don't change this value unless you know what
113
+ you are doing.
114
+ return_tensors (`str` or [`~utils.TensorType`], *optional*):
115
+ If set, will return tensors of a particular framework. Acceptable values are:
116
+
117
+ - `'tf'`: Return TensorFlow `tf.constant` objects.
118
+ - `'pt'`: Return PyTorch `torch.Tensor` objects.
119
+ - `'np'`: Return NumPy `np.ndarray` objects.
120
+ - `'jax'`: Return JAX `jnp.ndarray` objects.
121
+
122
+ Returns:
123
+ [`BatchFeature`]: A [`BatchFeature`] with the following fields:
124
+
125
+ - **input_ids** -- List of token ids to be fed to a model. Returned when `text` is not `None`.
126
+ - **attention_mask** -- List of indices specifying which tokens should be attended to by the model (when
127
+ `return_attention_mask=True` or if *"attention_mask"* is in `self.model_input_names` and if `text` is not
128
+ `None`).
129
+ - **audio_values** -- Processed audio values to be fed to a model. Returned when `audio` is not `None`.
130
+ - **audio_token_len** -- Predicted number of audio frames: this value is guaranteed to be a close upper bound.
131
+ Returned when `audio` is not `None`.
132
+ - **audio_token_start_idx** -- The index in the tokenized text where the audio starts. Returned when `audio` is not `None`.
133
+ """
134
+ # TODO: Add support for multiple audio and text inputs.
135
+ data = {}
136
+ audio_embed_frames = 0
137
+ if audio is not None and len(audio) > 0:
138
+ if self.audio_padding == "max_length":
139
+ # 30 seconds is the expected length for Whisper
140
+ assert sampling_rate is not None, "Sampling rate must be provided."
141
+ audio_len = 30 * sampling_rate
142
+ else:
143
+ audio_len = audio.shape[-1]
144
+ # It's guaranteed that the number of frames is less than or equal to this amount.
145
+ # For Whisper this is exact AFAICT, but for Wav2Vec2 it's an upper bound.
146
+ # Currently, StackAudioFrames makes sure an over-estimation won't cause issues by padding the audio embeddings.
147
+ nb_encoder_frames = int(round(audio_len / self.encoder_ds_factor + 1e-4))
148
+ audio_embed_frames = int(np.ceil(nb_encoder_frames / self.stack_factor))
149
+ data["audio_token_len"] = [audio_embed_frames]
150
+
151
+ # Main audio processing. The processor is model-specific.
152
+ x = self.audio_processor(
153
+ audio,
154
+ sampling_rate=sampling_rate,
155
+ padding="longest",
156
+ max_length=audio_len,
157
+ **kwargs,
158
+ )
159
+ if "input_features" in x:
160
+ data["audio_values"] = x.input_features
161
+ else:
162
+ data["audio_values"] = x.input_values
163
+
164
+ if text is not None:
165
+ assert isinstance(
166
+ text, str
167
+ ), "Text must be a string. Batch mode not supported yet."
168
+ if self.audio_placeholder in text:
169
+ if "audio_token_len" not in data:
170
+ raise ValueError(
171
+ f"audio must be provided when using audio placeholder ({self.audio_placeholder}) in text."
172
+ )
173
+
174
+ start_idx = len(
175
+ self.tokenizer.encode(
176
+ text[: text.index(self.audio_placeholder)],
177
+ add_special_tokens=False,
178
+ )
179
+ )
180
+ data["audio_token_start_idx"] = [start_idx]
181
+
182
+ # Replace the audio placeholder with the audio token.
183
+ # e.g. "Transcribe\n<|audio|>" -> "Transcribe </s></s></s></s></s></s></s></s>"
184
+ # where the number of </s> is the number of audio frames.
185
+ text = text.replace(
186
+ self.audio_placeholder,
187
+ self.audio_token_replacement * audio_embed_frames,
188
+ )
189
+
190
+ # Special tokens like BOS should already have been added by the caller.
191
+ data.update(self.tokenizer([text], add_special_tokens=False, **kwargs))
192
+
193
+ return transformers.BatchFeature(data=data, tensor_type=return_tensors)
194
+
195
+ def batch_decode(self, *args, **kwargs):
196
+ return self.tokenizer.batch_decode(*args, **kwargs)
197
+
198
+ def decode(self, *args, **kwargs):
199
+ return self.tokenizer.decode(*args, **kwargs)
200
+
201
+ @property
202
+ def model_input_names(self):
203
+ tokenizer_input_names = self.tokenizer.model_input_names
204
+ audio_processor_input_names = self.audio_processor.model_input_names
205
+ return list(set(tokenizer_input_names + audio_processor_input_names))
206
+
207
+
208
+ UltravoxProcessor.register_for_auto_class()
209
+
210
+ transformers.AutoProcessor.register(UltravoxConfig, UltravoxProcessor)