Upload UltravoxPipeline
Browse files- README.md +199 -0
- config.json +33 -0
- generation_config.json +11 -0
- model.safetensors +3 -0
- special_tokens_map.json +17 -0
- tokenizer.json +0 -0
- tokenizer_config.json +2063 -0
- ultravox_config.py +166 -0
- ultravox_model.py +644 -0
- ultravox_pipeline.py +127 -0
- ultravox_processing.py +210 -0
README.md
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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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### Model Description
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<!-- Provide a longer summary of what this model is. -->
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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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- **Developed by:** [More Information Needed]
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- **Funded by [optional]:** [More Information Needed]
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- **Shared by [optional]:** [More Information Needed]
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- **Model type:** [More Information Needed]
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- **Language(s) (NLP):** [More Information Needed]
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- **License:** [More Information Needed]
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- **Finetuned from model [optional]:** [More Information Needed]
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### Model Sources [optional]
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<!-- Provide the basic links for the model. -->
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- **Repository:** [More Information Needed]
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- **Paper [optional]:** [More Information Needed]
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- **Demo [optional]:** [More Information Needed]
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## Uses
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<!-- Address questions around how the model is intended to be used, including the foreseeable users of the model and those affected by the model. -->
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### Direct Use
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<!-- This section is for the model use without fine-tuning or plugging into a larger ecosystem/app. -->
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[More Information Needed]
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### Downstream Use [optional]
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<!-- This section is for the model use when fine-tuned for a task, or when plugged into a larger ecosystem/app -->
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[More Information Needed]
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### Out-of-Scope Use
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<!-- This section addresses misuse, malicious use, and uses that the model will not work well for. -->
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[More Information Needed]
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## Bias, Risks, and Limitations
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<!-- This section is meant to convey both technical and sociotechnical limitations. -->
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[More Information Needed]
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### Recommendations
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<!-- This section is meant to convey recommendations with respect to the bias, risk, and technical limitations. -->
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Users (both direct and downstream) should be made aware of the risks, biases and limitations of the model. More information needed for further recommendations.
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## How to Get Started with the Model
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Use the code below to get started with the model.
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[More Information Needed]
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## Training Details
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### Training Data
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<!-- This should link to a Dataset Card, perhaps with a short stub of information on what the training data is all about as well as documentation related to data pre-processing or additional filtering. -->
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[More Information Needed]
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### Training Procedure
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<!-- This relates heavily to the Technical Specifications. Content here should link to that section when it is relevant to the training procedure. -->
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#### Preprocessing [optional]
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[More Information Needed]
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#### Training Hyperparameters
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- **Training regime:** [More Information Needed] <!--fp32, fp16 mixed precision, bf16 mixed precision, bf16 non-mixed precision, fp16 non-mixed precision, fp8 mixed precision -->
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#### Speeds, Sizes, Times [optional]
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<!-- This section provides information about throughput, start/end time, checkpoint size if relevant, etc. -->
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[More Information Needed]
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## Evaluation
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<!-- This section describes the evaluation protocols and provides the results. -->
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### Testing Data, Factors & Metrics
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#### Testing Data
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<!-- This should link to a Dataset Card if possible. -->
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[More Information Needed]
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#### Factors
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<!-- These are the things the evaluation is disaggregating by, e.g., subpopulations or domains. -->
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[More Information Needed]
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#### Metrics
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<!-- These are the evaluation metrics being used, ideally with a description of why. -->
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[More Information Needed]
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### Results
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[More Information Needed]
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#### Summary
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## Model Examination [optional]
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<!-- Relevant interpretability work for the model goes here -->
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[More Information Needed]
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## Environmental Impact
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<!-- Total emissions (in grams of CO2eq) and additional considerations, such as electricity usage, go here. Edit the suggested text below accordingly -->
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Carbon emissions can be estimated using the [Machine Learning Impact calculator](https://mlco2.github.io/impact#compute) presented in [Lacoste et al. (2019)](https://arxiv.org/abs/1910.09700).
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- **Hardware Type:** [More Information Needed]
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- **Hours used:** [More Information Needed]
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- **Cloud Provider:** [More Information Needed]
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- **Compute Region:** [More Information Needed]
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- **Carbon Emitted:** [More Information Needed]
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## Technical Specifications [optional]
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### Model Architecture and Objective
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[More Information Needed]
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### Compute Infrastructure
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[More Information Needed]
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#### Hardware
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[More Information Needed]
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#### Software
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[More Information Needed]
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## Citation [optional]
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<!-- If there is a paper or blog post introducing the model, the APA and Bibtex information for that should go in this section. -->
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**BibTeX:**
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[More Information Needed]
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**APA:**
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[More Information Needed]
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## Glossary [optional]
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<!-- If relevant, include terms and calculations in this section that can help readers understand the model or model card. -->
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[More Information Needed]
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## More Information [optional]
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[More Information Needed]
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## Model Card Authors [optional]
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[More Information Needed]
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## Model Card Contact
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[More Information Needed]
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config.json
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{
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"_name_or_path": "/Users/zhuang/expts/2024-10-14-baselines/stacking-1a/ultravox/artifacts/model-zhuang.2024-10-14-baselines.stacking-1a.284d6e4:v5",
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"architectures": [
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"UltravoxModel"
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],
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"audio_model_id": "openai/whisper-small",
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"auto_map": {
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"AutoConfig": "ultravox_config.UltravoxConfig",
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"AutoModel": "ultravox_model.UltravoxModel"
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},
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"custom_pipelines": {
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"ultravox-pipeline": {
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"impl": "ultravox_pipeline.UltravoxPipeline",
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"pt": [
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"AutoModel"
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],
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"tf": [],
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"type": "multimodal"
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}
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},
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"hidden_size": 4096,
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"ignore_index": -100,
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"initializer_range": 0.02,
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"model_type": "ultravox",
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"norm_init": 0.4,
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"pad_token_id": 128009,
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"projector_act": "swiglu",
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"stack_factor": 8,
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"text_model_id": "meta-llama/Llama-3.2-1B-Instruct",
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"torch_dtype": "bfloat16",
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"transformers_version": "4.44.0",
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"vocab_size": 128256
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}
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generation_config.json
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{
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"_from_model_config": true,
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"bos_token_id": 128000,
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"eos_token_id": [
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128001,
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128008,
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128009
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],
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"pad_token_id": 128009,
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"transformers_version": "4.44.0"
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}
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model.safetensors
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version https://git-lfs.github.com/spec/v1
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oid sha256:bf2dcacf5a1f00c10072a94f4729f8d89017908bd1444569e98b7bba4883a548
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size 58737112
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special_tokens_map.json
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{
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"bos_token": {
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"content": "<|begin_of_text|>",
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"lstrip": false,
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"normalized": false,
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"rstrip": false,
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"single_word": false
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},
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"eos_token": {
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"content": "<|eot_id|>",
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"lstrip": false,
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"normalized": false,
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"rstrip": false,
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"single_word": false
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},
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"pad_token": "<|eot_id|>"
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}
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tokenizer.json
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tokenizer_config.json
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1 |
+
{
|
2 |
+
"added_tokens_decoder": {
|
3 |
+
"128000": {
|
4 |
+
"content": "<|begin_of_text|>",
|
5 |
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|
6 |
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|
7 |
+
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|
8 |
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|
9 |
+
"special": true
|
10 |
+
},
|
11 |
+
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|
12 |
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"content": "<|end_of_text|>",
|
13 |
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|
14 |
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|
15 |
+
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|
16 |
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|
17 |
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"special": true
|
18 |
+
},
|
19 |
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|
20 |
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"content": "<|reserved_special_token_0|>",
|
21 |
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|
22 |
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|
23 |
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|
24 |
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|
25 |
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|
26 |
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|
27 |
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|
28 |
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"content": "<|reserved_special_token_1|>",
|
29 |
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|
30 |
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|
31 |
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|
32 |
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|
33 |
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|
34 |
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},
|
35 |
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|
36 |
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|
37 |
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|
38 |
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|
39 |
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|
40 |
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|
41 |
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|
42 |
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|
43 |
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|
44 |
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|
45 |
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|
46 |
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|
47 |
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|
48 |
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|
49 |
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|
50 |
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|
51 |
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|
52 |
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|
53 |
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|
54 |
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|
55 |
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|
56 |
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|
57 |
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|
58 |
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|
59 |
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|
60 |
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|
61 |
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62 |
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63 |
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64 |
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65 |
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|
66 |
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|
67 |
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|
68 |
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|
69 |
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70 |
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71 |
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72 |
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73 |
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74 |
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75 |
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76 |
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|
77 |
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78 |
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79 |
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|
80 |
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81 |
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|
82 |
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|
83 |
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|
84 |
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|
85 |
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86 |
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87 |
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88 |
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89 |
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90 |
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91 |
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92 |
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93 |
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94 |
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96 |
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97 |
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98 |
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99 |
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|
100 |
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|
101 |
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102 |
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103 |
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104 |
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105 |
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106 |
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|
107 |
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|
108 |
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|
109 |
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110 |
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111 |
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112 |
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113 |
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114 |
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115 |
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116 |
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117 |
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118 |
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119 |
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120 |
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121 |
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122 |
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123 |
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|
124 |
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|
125 |
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126 |
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127 |
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128 |
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129 |
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130 |
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131 |
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132 |
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133 |
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134 |
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135 |
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136 |
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137 |
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138 |
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139 |
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|
140 |
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141 |
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142 |
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143 |
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144 |
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145 |
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|
146 |
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|
147 |
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|
148 |
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|
149 |
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150 |
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151 |
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152 |
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|
153 |
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154 |
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155 |
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|
156 |
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|
157 |
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158 |
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|
159 |
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160 |
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161 |
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162 |
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|
163 |
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|
164 |
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165 |
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166 |
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167 |
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|
168 |
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|
169 |
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|
170 |
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|
171 |
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|
172 |
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|
173 |
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174 |
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175 |
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176 |
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177 |
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178 |
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179 |
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|
180 |
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|
181 |
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|
182 |
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183 |
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184 |
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|
185 |
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186 |
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|
187 |
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|
188 |
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|
189 |
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190 |
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191 |
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192 |
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|
193 |
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194 |
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|
195 |
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|
196 |
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197 |
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|
198 |
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|
199 |
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|
200 |
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|
201 |
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|
202 |
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|
203 |
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|
204 |
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|
205 |
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|
206 |
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|
207 |
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|
208 |
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|
209 |
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|
210 |
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|
211 |
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|
212 |
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|
213 |
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|
214 |
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|
215 |
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|
216 |
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217 |
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|
218 |
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|
219 |
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|
220 |
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|
221 |
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|
222 |
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223 |
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224 |
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225 |
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226 |
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|
227 |
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|
228 |
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|
229 |
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|
230 |
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|
231 |
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232 |
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|
233 |
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|
234 |
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|
235 |
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|
236 |
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237 |
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238 |
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239 |
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240 |
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241 |
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242 |
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243 |
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|
244 |
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245 |
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248 |
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250 |
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251 |
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|
252 |
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253 |
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254 |
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256 |
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257 |
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258 |
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259 |
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|
260 |
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261 |
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262 |
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263 |
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264 |
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266 |
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267 |
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268 |
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269 |
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270 |
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271 |
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272 |
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273 |
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274 |
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275 |
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|
276 |
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277 |
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278 |
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279 |
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280 |
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281 |
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282 |
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283 |
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284 |
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285 |
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286 |
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287 |
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288 |
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289 |
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290 |
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291 |
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|
292 |
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293 |
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294 |
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295 |
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296 |
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297 |
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298 |
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299 |
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|
300 |
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301 |
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302 |
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303 |
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304 |
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305 |
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306 |
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307 |
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|
2051 |
+
},
|
2052 |
+
"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 @@
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
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|
|
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|
|
|
|
|
|
|
|
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|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
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|
|
|
|
|
|
|
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|
|
|
|
|
|
|
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|
|
|
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|
|
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|
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|
|
|
|
|
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|
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|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
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|
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|
|
|
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|
|
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|
|
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|
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|
|
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|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
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 @@
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
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|
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 @@
|
|
|
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|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
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 @@
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
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|
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|
|
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|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
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|
|
|
|
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|
|
|
|
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|
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|
|
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|
|
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|
|
|
|
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|
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|
|
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|
|
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|
|
|
|
|
|
|
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|
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|
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|
|
|
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|
|
|
|
|
|
|
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|
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|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
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)
|