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Model: kotoba-tech/kotoba-whisper-v1.0
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wandb

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---
license: apache-2.0
language: ja
library_name: transformers
tags:
- audio
- automatic-speech-recognition
- hf-asr-leaderboard
widget:
- example_title: CommonVoice 8.0 (Test Split)
src: >-
https://huggingface.co/datasets/japanese-asr/ja_asr.common_voice_8_0/resolve/main/sample.flac
- example_title: JSUT Basic 5000
src: >-
https://huggingface.co/datasets/japanese-asr/ja_asr.jsut_basic5000/resolve/main/sample.flac
- example_title: ReazonSpeech (Test Split)
src: >-
https://huggingface.co/datasets/japanese-asr/ja_asr.reazonspeech_test/resolve/main/sample.flac
pipeline_tag: automatic-speech-recognition
metrics:
- wer
- cer
model-index:
- name: kotoba-tech/kotoba-whisper-v1.0
results:
- task:
type: automatic-speech-recognition
dataset:
name: CommonVoice 8 (Japanese test set)
type: japanese-asr/ja_asr.common_voice_8_0
metrics:
- name: WER
type: WER
value: 59.2
- name: CER
type: CER
value: 9.4
- task:
type: automatic-speech-recognition
dataset:
name: ReazonSpeech (held out test set)
type: japanese-asr/ja_asr.reazonspeech_test
metrics:
- name: WER
type: WER
value: 56.6
- name: CER
type: CER
value: 12.2
- task:
type: automatic-speech-recognition
dataset:
name: JSUT Basic 5000
type: japanese-asr/ja_asr.jsut_basic5000
metrics:
- name: WER
type: WER
value: 64.3
- name: CER
type: CER
value: 8.5
datasets:
- japanese-asr/whisper_transcriptions.reazonspeech.large
- japanese-asr/whisper_transcriptions.reazonspeech.large.wer_10.0
- japanese-asr/whisper_transcriptions.reazonspeech.large.wer_10.0.vectorized
---
# Kotoba-Whisper (v1.0)
[**faster-whisper weight**](https://huggingface.co/kotoba-tech/kotoba-whisper-v1.0-faster), [**whisper.cpp weight**](https://huggingface.co/kotoba-tech/kotoba-whisper-v1.0-ggml), [**pipeline with stable-ts/punctuation**](https://huggingface.co/kotoba-tech/kotoba-whisper-v1.1)
***News:*** Newer version of this model is availabel at [https://huggingface.co/kotoba-tech/kotoba-whisper-v2.0](https://huggingface.co/kotoba-tech/kotoba-whisper-v2.0)!
_Kotoba-Whisper_ is a collection of distilled [Whisper](https://arxiv.org/abs/2212.04356) models for Japanese ASR, developed through the collaboration bewteen
[Asahi Ushio](https://asahiushio.com) and [Kotoba Technologies](https://twitter.com/kotoba_tech).
Following the original work of distil-whisper ([Robust Knowledge Distillation via Large-Scale Pseudo Labelling](https://arxiv.org/abs/2311.00430)),
we employ OpenAI's [Whisper large-v3](https://huggingface.co/openai/whisper-large-v3) as the teacher model, and the student model consists the full encoder of the
teacher large-v3 model and the decoder with two layers initialized from the first and last layer of the large-v3 model.
Kotoba-Whisper is **6.3x faster than large-v3**, while retaining as low error rate as the large-v3.
As the initial version, we release ***kotoba-whisper-v1.0*** trained on the `large` subset of [ReazonSpeech](https://huggingface.co/datasets/reazon-research/reazonspeech)
(the largest speech-transcription paired dataset in Japanese extracted from Japanese TV audio recordings),
which amounts 1,253 hours of audio with 16,861,235 characters of transcriptions (5 sec audio with 18 text tokens in average) after
those transcriptions more than 10 WER are removed (see [WER Filter](https://huggingface.co/distil-whisper/distil-large-v3#wer-filter) for detail).
The model was trained for 8 epochs with batch size 256 with sampling rate of 16kHz, and the training and evaluation code to reproduce kotoba-whisper is available at [https://github.com/kotoba-tech/kotoba-whisper](https://github.com/kotoba-tech/kotoba-whisper).
Kotoba-whisper-v1.0 achieves better CER and WER than the [openai/whisper-large-v3](https://huggingface.co/openai/whisper-large-v3) in the in-domain held-out test set
from ReazonSpeech, and achieves competitive CER and WER on the out-of-domain test sets including [JSUT basic 5000](https://sites.google.com/site/shinnosuketakamichi/publication/jsut) and
the Japanese subset from [CommonVoice 8.0](https://huggingface.co/datasets/common_voice) (see [Evaluation](#evaluation) for detail).
- ***CER***
| model | [CommonVoice 8 (Japanese test set)](https://huggingface.co/datasets/japanese-asr/ja_asr.common_voice_8_0) | [JSUT Basic 5000](https://huggingface.co/datasets/japanese-asr/ja_asr.jsut_basic5000) | [ReazonSpeech (held out test set)](https://huggingface.co/datasets/japanese-asr/ja_asr.reazonspeech_test) |
|:--------------------------------------------------------------------------------------------------------------------------------------------------|------------------------------------------------------------------------------------------------------------:|----------------------------------------------------------------------------------------:|------------------------------------------------------------------------------------------------------------:|
| [kotoba-tech/kotoba-whisper-v2.0](https://huggingface.co/kotoba-tech/kotoba-whisper-v2.0) | 9.2 | 8.4 | 11.6 |
| [kotoba-tech/kotoba-whisper-v1.0](https://huggingface.co/kotoba-tech/kotoba-whisper-v1.0) | 9.4 | 8.5 | 12.2 |
| [openai/whisper-large-v3](https://huggingface.co/openai/whisper-large-v3) | 8.5 | 7.1 | 14.9 |
| [openai/whisper-large-v2](https://huggingface.co/openai/whisper-large-v2) | 9.7 | 8.2 | 28.1 |
| [openai/whisper-large](https://huggingface.co/openai/whisper-large) | 10 | 8.9 | 34.1 |
| [openai/whisper-medium](https://huggingface.co/openai/whisper-medium) | 11.5 | 10 | 33.2 |
| [openai/whisper-base](https://huggingface.co/openai/whisper-base) | 28.6 | 24.9 | 70.4 |
| [openai/whisper-small](https://huggingface.co/openai/whisper-small) | 15.1 | 14.2 | 41.5 |
| [openai/whisper-tiny](https://huggingface.co/openai/whisper-tiny) | 53.7 | 36.5 | 137.9 |
- ***WER***
| model | [CommonVoice 8 (Japanese test set)](https://huggingface.co/datasets/japanese-asr/ja_asr.common_voice_8_0) | [JSUT Basic 5000](https://huggingface.co/datasets/japanese-asr/ja_asr.jsut_basic5000) | [ReazonSpeech (held out test set)](https://huggingface.co/datasets/japanese-asr/ja_asr.reazonspeech_test) |
|:--------------------------------------------------------------------------------------------------------------------------------------------------|------------------------------------------------------------------------------------------------------------:|----------------------------------------------------------------------------------------:|------------------------------------------------------------------------------------------------------------:|
| [kotoba-tech/kotoba-whisper-v2.0](https://huggingface.co/kotoba-tech/kotoba-whisper-v2.0) | 58.8 | 63.7 | 55.6 |
| [kotoba-tech/kotoba-whisper-v1.0](https://huggingface.co/kotoba-tech/kotoba-whisper-v1.0) | 59.2 | 64.3 | 56.4 |
| [openai/whisper-large-v3](https://huggingface.co/openai/whisper-large-v3) | 55.1 | 59.2 | 60.2 |
| [openai/whisper-large-v2](https://huggingface.co/openai/whisper-large-v2) | 59.3 | 63.2 | 74.1 |
| [openai/whisper-large](https://huggingface.co/openai/whisper-large) | 61.1 | 66.4 | 74.9 |
| [openai/whisper-medium](https://huggingface.co/openai/whisper-medium) | 63.4 | 69.5 | 76 |
| [openai/whisper-base](https://huggingface.co/openai/whisper-base) | 87.2 | 93 | 91.8 |
| [openai/whisper-small](https://huggingface.co/openai/whisper-small) | 74.2 | 81.9 | 83 |
| [openai/whisper-tiny](https://huggingface.co/openai/whisper-tiny) | 93.8 | 97.6 | 94.9 |
- ***Latency***: As kotoba-whisper uses the same architecture as [distil-whisper/distil-large-v3](https://huggingface.co/distil-whisper/distil-large-v3),
it inherits the benefit of the improved latency compared to [openai/whisper-large-v3](https://huggingface.co/openai/whisper-large-v3)
(**6.3x faster than large-v3**, see the table below taken from [distil-whisper/distil-large-v3](https://huggingface.co/distil-whisper/distil-large-v3)).
| Model | Params / M | Rel. Latency |
|----------------------------------------------------------------------------------------------|------------|--------------|
| **[kotoba-tech/kotoba-whisper-v1.0](https://huggingface.co/kotoba-tech/kotoba-whisper-v1.0)**| **756** | **6.3** |
| [openai/whisper-large-v3](https://huggingface.co/openai/whisper-large-v3) | 1550 | 1.0 |
## Transformers Usage
Kotoba-Whisper is supported in the Hugging Face 🤗 Transformers library from version 4.39 onwards. To run the model, first
install the latest version of Transformers.
```bash
pip install --upgrade pip
pip install --upgrade transformers accelerate
```
### Short-Form Transcription
The model can be used with the [`pipeline`](https://huggingface.co/docs/transformers/main_classes/pipelines#transformers.AutomaticSpeechRecognitionPipeline)
class to transcribe short-form audio files (< 30-seconds) as follows:
```python
import torch
from transformers import pipeline
from datasets import load_dataset
# config
model_id = "kotoba-tech/kotoba-whisper-v1.0"
torch_dtype = torch.bfloat16 if torch.cuda.is_available() else torch.float32
device = "cuda:0" if torch.cuda.is_available() else "cpu"
model_kwargs = {"attn_implementation": "sdpa"} if torch.cuda.is_available() else {}
generate_kwargs = {"language": "ja", "task": "transcribe"}
# load model
pipe = pipeline(
"automatic-speech-recognition",
model=model_id,
torch_dtype=torch_dtype,
device=device,
model_kwargs=model_kwargs
)
# load sample audio
dataset = load_dataset("japanese-asr/ja_asr.reazonspeech_test", split="test")
sample = dataset[0]["audio"]
# run inference
result = pipe(sample, generate_kwargs=generate_kwargs)
print(result["text"])
```
- To transcribe a local audio file, simply pass the path to your audio file when you call the pipeline (make sure the audio is sampled in 16kHz):
```diff
- result = pipe(sample, generate_kwargs=generate_kwargs)
+ result = pipe("audio.mp3", generate_kwargs=generate_kwargs)
```
- For segment-level timestamps, pass the argument `return_timestamps=True` and return the `"chunks"` output:
```python
result = pipe(sample, return_timestamps=True, generate_kwargs=generate_kwargs)
print(result["chunks"])
```
***Sequential Long-Form:*** Kotoba-whisper is designed to be compatible with OpenAI's sequential long-form transcription algorithm. This algorithm uses a sliding window for buffered
inference of long audio files (> 30-seconds), and returns more accurate transcriptions compared to the [chunked long-form algorithm](#chunked-long-form).
As default, if long audio files are passed to the model, it will transcribes with the sequential long-form transcription.
The sequential long-form algorithm should be used in either of the following scenarios:
1. Transcription accuracy is the most important factor, and latency is less of a consideration
2. You are transcribing **batches** of long audio files, in which case the latency of sequential is comparable to chunked, while being up to 0.5% WER more accurate
If you are transcribing single long audio files and latency is the most important factor, you should use the chunked algorithm
described [below](#chunked-long-form). For a detailed explanation of the different algorithms, refer to Sections 5 of
the [Distil-Whisper paper](https://arxiv.org/pdf/2311.00430.pdf). The [`pipeline`](https://huggingface.co/docs/transformers/main_classes/pipelines#transformers.AutomaticSpeechRecognitionPipeline)
class can be used to transcribe long audio files with the sequential algorithm as follows:
### Chunked Long-Form
This algorithm should be used when a single large audio file is being transcribed and the fastest possible inference is required. In such circumstances,
the chunked algorithm is up to 9x faster than OpenAI's sequential long-form implementation (see Table 7 of the [Distil-Whisper paper](https://arxiv.org/pdf/2311.00430.pdf)).
To enable chunking, pass the `chunk_length_s` parameter to the `pipeline`. For distil-large-v3, a chunk length of 25-seconds
is optimal. To activate batching over long audio files, pass the argument `batch_size`:
```python
import torch
from transformers import pipeline
from datasets import load_dataset
# config
model_id = "kotoba-tech/kotoba-whisper-v1.0"
torch_dtype = torch.bfloat16 if torch.cuda.is_available() else torch.float32
device = "cuda:0" if torch.cuda.is_available() else "cpu"
model_kwargs = {"attn_implementation": "sdpa"} if torch.cuda.is_available() else {}
generate_kwargs = {"language": "japanese", "task": "transcribe"}
# load model
pipe = pipeline(
"automatic-speech-recognition",
model=model_id,
torch_dtype=torch_dtype,
device=device,
model_kwargs=model_kwargs,
batch_size=16
)
# load sample audio (concatenate instances to create a long audio)
dataset = load_dataset("japanese-asr/ja_asr.reazonspeech_test", split="test")
sample = {"array": np.concatenate([i["array"] for i in dataset[:20]["audio"]]), "sampling_rate": dataset[0]['audio']['sampling_rate']}
# run inference
result = pipe(sample, chunk_length_s=15, generate_kwargs=generate_kwargs)
print(result["text"])
```
### Transcription with Prompt
Kotoba-whisper can generate transcription with prompting as below:
```python
import re
import torch
from transformers import pipeline
from datasets import load_dataset
# config
model_id = "kotoba-tech/kotoba-whisper-v1.0"
torch_dtype = torch.bfloat16 if torch.cuda.is_available() else torch.float32
device = "cuda:0" if torch.cuda.is_available() else "cpu"
model_kwargs = {"attn_implementation": "sdpa"} if torch.cuda.is_available() else {}
generate_kwargs = {"language": "ja", "task": "transcribe"}
# load model
pipe = pipeline(
"automatic-speech-recognition",
model=model_id,
torch_dtype=torch_dtype,
device=device,
model_kwargs=model_kwargs
)
# load sample audio
dataset = load_dataset("japanese-asr/ja_asr.reazonspeech_test", split="test")
# --- Without prompt ---
text = pipe(dataset[10]["audio"], generate_kwargs=generate_kwargs)['text']
print(text)
# 81歳、力強い走りに変わってきます。
# --- With prompt ---: Let's change `81` to `91`.
prompt = "91歳"
generate_kwargs['prompt_ids'] = pipe.tokenizer.get_prompt_ids(prompt, return_tensors="pt").to(device)
text = pipe(dataset[10]["audio"], generate_kwargs=generate_kwargs)['text']
# currently the pipeline for ASR appends the prompt at the beginning of the transcription, so remove it
text = re.sub(rf"\A\s*{prompt}\s*", "", text)
print(text)
# あっぶったでもスルガさん、91歳、力強い走りに変わってきます。
```
### Additional Speed & Memory Improvements
You can apply additional speed and memory improvements to further reduce the inference speed and VRAM
requirements. These optimisations primarily target the attention kernel, swapping it from an eager implementation to a
more efficient flash attention version.
#### Flash Attention 2
We recommend using [Flash-Attention 2](https://huggingface.co/docs/transformers/main/en/perf_infer_gpu_one#flashattention-2)
if your GPU allows for it. To do so, you first need to install [Flash Attention](https://github.com/Dao-AILab/flash-attention):
```
pip install flash-attn --no-build-isolation
```
Then pass `attn_implementation="flash_attention_2"` to `from_pretrained`:
```diff
- model_kwargs = {"attn_implementation": "sdpa"} if torch.cuda.is_available() else {}
+ model_kwargs = {"attn_implementation": "flash_attention_2"} if torch.cuda.is_available() else {}
```
## Model Details
See [https://huggingface.co/distil-whisper/distil-large-v3#model-details](https://huggingface.co/distil-whisper/distil-large-v3#model-details).
## Training
Please refer to [https://github.com/kotoba-tech/kotoba-whisper](https://github.com/kotoba-tech/kotoba-whisper) for the model training detail.
Datasets used in distillation and the whole model variations can be found at [https://huggingface.co/japanese-asr](https://huggingface.co/japanese-asr).
## Evaluation
The following code-snippets demonstrates how to evaluate the kotoba-whisper model on the Japanese subset of the CommonVoice 8.0.
First, we need to install the required packages, including 🤗 Datasets to load the audio data, and 🤗 Evaluate to
perform the WER calculation:
```bash
pip install --upgrade pip
pip install --upgrade transformers datasets[audio] evaluate jiwer
```
Evaluation can then be run end-to-end with the following example:
```python
import torch
from transformers import pipeline
from datasets import load_dataset
from evaluate import load
from transformers.models.whisper.english_normalizer import BasicTextNormalizer
# model config
model_id = "kotoba-tech/kotoba-whisper-v1.0"
torch_dtype = torch.bfloat16 if torch.cuda.is_available() else torch.float32
device = "cuda:0" if torch.cuda.is_available() else "cpu"
model_kwargs = {"attn_implementation": "sdpa"} if torch.cuda.is_available() else {}
generate_kwargs = {"language": "japanese", "task": "transcribe"}
normalizer = BasicTextNormalizer()
# data config
dataset_name = "japanese-asr/ja_asr.reazonspeech_test"
audio_column = 'audio'
text_column = 'transcription'
# load model
pipe = pipeline(
"automatic-speech-recognition",
model=model_id,
torch_dtype=torch_dtype,
device=device,
model_kwargs=model_kwargs,
batch_size=16
)
# load the dataset and sample the audio with 16kHz
dataset = load_dataset(dataset_name, split="test")
transcriptions = pipe(dataset['audio'])
transcriptions = [normalizer(i['text']).replace(" ", "") for i in transcriptions]
references = [normalizer(i).replace(" ", "") for i in dataset['transcription']]
# compute the CER metric
cer_metric = load("cer")
cer = 100 * cer_metric.compute(predictions=transcriptions, references=references)
print(cer)
```
The huggingface links to the major Japanese ASR datasets for evaluation are summarized at [here](https://huggingface.co/collections/japanese-asr/japanese-asr-evaluation-dataset-66051a03d6ca494d40baaa26).
For example, to evaluate the model on JSUT Basic5000, change the `dataset_name`:
```diff
- dataset_name = "japanese-asr/ja_asr.reazonspeech_test"
+ dataset_name = "japanese-asr/ja_asr.jsut_basic5000"
```
## Acknowledgements
* [OpenAI](https://openai.com/) for the Whisper [model](https://huggingface.co/openai/whisper-large-v3).
* Hugging Face 🤗 [Transformers](https://github.com/huggingface/transformers) for the model integration.
* Hugging Face 🤗 for the [Distil-Whisper codebase](https://github.com/huggingface/distil-whisper).
* [Reazon Human Interaction Lab](https://research.reazon.jp/) for the [ReazonSpeech dataset](https://huggingface.co/datasets/reazon-research/reazonspeech).

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from time import time
from pprint import pprint
import torch
from transformers import pipeline
from datasets import load_dataset
# config
generate_kwargs = {"language": "japanese", "task": "transcribe"}
model_id = "kotoba-tech/kotoba-whisper-v1.0"
# load model
pipe = pipeline(
"automatic-speech-recognition",
model=model_id,
torch_dtype=torch.float32
)
test_audio = [
"kotoba-whisper-eval/audio/manzai1.wav",
"kotoba-whisper-eval/audio/manzai2.wav",
"kotoba-whisper-eval/audio/manzai3.wav",
"kotoba-whisper-eval/audio/long_interview_1.wav",
]
elapsed = {}
for x in test_audio:
start = time()
transcription = pipe(x, generate_kwargs=generate_kwargs)
elapsed[x] = time() - start
pprint(transcription)
pprint(elapsed)

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# clone dataset
git clone https://huggingface.co/datasets/kotoba-tech/kotoba-whisper-eval
# convert to 16khz
ffmpeg -i kotoba-whisper-eval/audio/long_interview_1.mp3 -ar 16000 -ac 1 -c:a pcm_s16le kotoba-whisper-eval/audio/long_interview_1.wav
ffmpeg -i kotoba-whisper-eval/audio/manzai1.mp3 -ar 16000 -ac 1 -c:a pcm_s16le kotoba-whisper-eval/audio/manzai1.wav
ffmpeg -i kotoba-whisper-eval/audio/manzai2.mp3 -ar 16000 -ac 1 -c:a pcm_s16le kotoba-whisper-eval/audio/manzai2.wav
ffmpeg -i kotoba-whisper-eval/audio/manzai3.mp3 -ar 16000 -ac 1 -c:a pcm_s16le kotoba-whisper-eval/audio/manzai3.wav
# run the benchmark
wget https://huggingface.co/kotoba-tech/kotoba-whisper-v1.0/raw/main/benchmark.py
python benchmark.py

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{
"_name_or_path": "./distil-whisper-large-v3-ja-reazonspeech-large-init",
"activation_dropout": 0.0,
"activation_function": "gelu",
"apply_spec_augment": false,
"architectures": [
"WhisperForConditionalGeneration"
],
"attention_dropout": 0.0,
"begin_suppress_tokens": [
220,
50257
],
"bos_token_id": 50257,
"classifier_proj_size": 256,
"d_model": 1280,
"decoder_attention_heads": 20,
"decoder_ffn_dim": 5120,
"decoder_layerdrop": 0.0,
"decoder_layers": 2,
"decoder_start_token_id": 50258,
"dropout": 0.0,
"encoder_attention_heads": 20,
"encoder_ffn_dim": 5120,
"encoder_layerdrop": 0.0,
"encoder_layers": 32,
"eos_token_id": 50257,
"init_std": 0.02,
"is_encoder_decoder": true,
"mask_feature_length": 10,
"mask_feature_min_masks": 0,
"mask_feature_prob": 0.0,
"mask_time_length": 10,
"mask_time_min_masks": 2,
"mask_time_prob": 0.05,
"max_length": 448,
"max_source_positions": 1500,
"max_target_positions": 448,
"median_filter_width": 7,
"model_type": "whisper",
"num_hidden_layers": 32,
"num_mel_bins": 128,
"pad_token_id": 50256,
"scale_embedding": false,
"torch_dtype": "float32",
"transformers_version": "4.38.2",
"use_cache": true,
"use_weighted_layer_sum": false,
"vocab_size": 51866
}

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#!/usr/bin/env python
# coding=utf-8
# Copyright 2023 The HuggingFace Inc. team. All rights reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
"""
Initialise a student Whisper model from a pre-trained teacher model for
teacher-student distillation.
"""
import argparse
import copy
import logging
import os
import numpy as np
import torch
from transformers import GenerationConfig, WhisperForConditionalGeneration, WhisperProcessor
# https://stackoverflow.com/questions/71692354/facing-ssl-error-with-huggingface-pretrained-models
os.environ['CURL_CA_BUNDLE'] = ''
# disable warning message
os.environ['TOKENIZERS_PARALLELISM'] = 'false'
logger = logging.getLogger(__name__)
def parse_args():
parser = argparse.ArgumentParser(
description="Initialise a student Whisper model from a teacher model, copying the relevant layer weights and adjusting the processor as necessary."
)
parser.add_argument(
"--teacher_checkpoint",
type=str,
required=True,
help="The HF Hub ID of the teacher checkpoint.",
)
parser.add_argument(
"--subfolder",
type=str,
default="",
help="In case the relevant teacher weights are located inside a subfolder of the model repo on huggingface.co, you "
"can specify the folder name here.",
)
parser.add_argument(
"--encoder_layers",
type=int,
default=None,
help="Number of encoder layers to use in the student model. Defaults to all layers from the teacher.",
)
parser.add_argument(
"--decoder_layers",
type=int,
default=2,
help="Number of decoder layers to use in the student model. Defaults to 2 layers.",
)
parser.add_argument(
"--save_dir",
type=str,
required=True,
help="Where to save the student weights and processor.",
)
parser.add_argument(
"--push_to_hub",
type=bool,
required=False,
default=False,
help="Whether to push the student weights and processor to the Hub.",
)
parser.add_argument(
"--cache_dir",
type=str,
default=None,
help="Where to store the pretrained models downloaded from huggingface.co",
)
args = parser.parse_args()
return args
def init_student_model_from_teacher(
teacher_checkpoint,
encoder_layers=None,
decoder_layers=2,
save_dir=None,
push_to_hub=None,
cache_dir=None,
subfolder="",
):
teacher_model = WhisperForConditionalGeneration.from_pretrained(
teacher_checkpoint,
cache_dir=cache_dir,
subfolder=subfolder,
low_cpu_mem_usage=True,
)
processor = WhisperProcessor.from_pretrained(teacher_checkpoint)
generation_config = GenerationConfig.from_pretrained(teacher_checkpoint)
teacher_config = teacher_model.config
teacher_encoder_layers = teacher_config.encoder_layers
teacher_decoder_layers = teacher_config.decoder_layers
student_config = copy.deepcopy(teacher_config)
student_config.update(
{
"encoder_layers": encoder_layers if encoder_layers is not None else teacher_encoder_layers,
"decoder_layers": decoder_layers,
}
)
encoder_mapping = np.linspace(0, teacher_encoder_layers - 1, student_config.encoder_layers, dtype=int)
encoder_mapping[-1] = teacher_encoder_layers - 1
encoder_map = {}
for student_layer, teacher_layer in enumerate(encoder_mapping):
encoder_map[teacher_layer] = student_layer
decoder_mapping = np.linspace(0, teacher_decoder_layers - 1, student_config.decoder_layers, dtype=int)
decoder_mapping[-1] = teacher_decoder_layers - 1
decoder_map = {}
for student_layer, teacher_layer in enumerate(decoder_mapping):
decoder_map[teacher_layer] = student_layer
# init the student params from the teacher model
student_model = WhisperForConditionalGeneration(student_config)
missing_keys, unexpected_keys = student_model.load_state_dict(teacher_model.state_dict(), strict=False)
if len(missing_keys) > 0:
raise RuntimeError(
"Error(s) in loading state_dict for WhisperForConditionalGeneration. \n"
f"Missing key(s) in state_dict: {missing_keys}"
)
if decoder_layers == teacher_decoder_layers:
decoder_keys = [key for key in unexpected_keys if "model.decoder.layers" in key]
if len(decoder_keys) > 0:
raise RuntimeError(
"Error(s) in loading state_dict for WhisperForConditionalGeneration. \n"
f"Unexpected key(s) in state_dict: {decoder_keys}"
)
if encoder_layers == teacher_encoder_layers:
encoder_keys = [key for key in unexpected_keys if "model.encoder.layers" in key]
if len(encoder_keys) > 0:
raise RuntimeError(
"Error(s) in loading state_dict for WhisperForConditionalGeneration. \n"
f"Unexpected key(s) in state_dict: {encoder_keys}"
)
for layer in range(teacher_decoder_layers):
if layer in decoder_map:
# re-introduce pre-defined layers from the teacher
student_model.model.decoder.layers[decoder_map[layer]].load_state_dict(
teacher_model.model.decoder.layers[layer].state_dict()
)
if encoder_layers is not None:
for layer in range(teacher_encoder_layers):
if layer in encoder_map:
# re-introduce pre-defined layers from the teacher
student_model.model.encoder.layers[encoder_map[layer]].load_state_dict(
teacher_model.model.encoder.layers[layer].state_dict()
)
# remove the teacher params and model
del teacher_model
# save the converted weights and model
if save_dir is not None:
student_model.save_pretrained(save_dir)
# we also need to correctly save the processor and generation config
processor.save_pretrained(save_dir)
generation_config.save_pretrained(save_dir)
# check we can do a forward pass with the saved model - first load the weights and processor
logger.info("Checking we can load the saved model...")
student_model = WhisperForConditionalGeneration.from_pretrained(
save_dir,
low_cpu_mem_usage=True,
)
processor = WhisperProcessor.from_pretrained(save_dir)
# define some random inputs
input_features = processor(np.ones(16000), sampling_rate=16000, return_tensors="pt").input_features
decoder_start_token_id = student_model.config.decoder_start_token_id
decoder_input_ids = torch.ones((input_features.shape[0], 1), dtype=torch.long) * decoder_start_token_id
# do a forward pass - outputs will be gibberish for the initialised model so we can't check them
# but we make can sure the model runs as expected
logger.info("Checking we can run the converted model forward...")
_ = student_model(input_features, decoder_input_ids=decoder_input_ids).logits
logger.info("Conversion successful!")
if push_to_hub:
student_model.push_to_hub(save_dir)
processor.push_to_hub(save_dir)
generation_config.push_to_hub(save_dir)
if __name__ == "__main__":
args = parse_args()
init_student_model_from_teacher(
teacher_checkpoint=args.teacher_checkpoint,
encoder_layers=args.encoder_layers,
decoder_layers=args.decoder_layers,
save_dir=args.save_dir,
push_to_hub=args.push_to_hub,
cache_dir=args.cache_dir,
subfolder=args.subfolder,
)

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generation_config.json Normal file
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{
"alignment_heads": [
[
7,
0
],
[
10,
17
],
[
12,
18
],
[
13,
12
],
[
16,
1
],
[
17,
14
],
[
19,
11
],
[
21,
4
],
[
24,
1
],
[
25,
6
]
],
"begin_suppress_tokens": [
220,
50257
],
"bos_token_id": 50257,
"decoder_start_token_id": 50258,
"eos_token_id": 50257,
"forced_decoder_ids": [
[
1,
null
],
[
2,
50360
]
],
"is_multilingual": true,
"lang_to_id": {
"<|af|>": 50327,
"<|am|>": 50334,
"<|ar|>": 50272,
"<|as|>": 50350,
"<|az|>": 50304,
"<|ba|>": 50355,
"<|be|>": 50330,
"<|bg|>": 50292,
"<|bn|>": 50302,
"<|bo|>": 50347,
"<|br|>": 50309,
"<|bs|>": 50315,
"<|ca|>": 50270,
"<|cs|>": 50283,
"<|cy|>": 50297,
"<|da|>": 50285,
"<|de|>": 50261,
"<|el|>": 50281,
"<|en|>": 50259,
"<|es|>": 50262,
"<|et|>": 50307,
"<|eu|>": 50310,
"<|fa|>": 50300,
"<|fi|>": 50277,
"<|fo|>": 50338,
"<|fr|>": 50265,
"<|gl|>": 50319,
"<|gu|>": 50333,
"<|haw|>": 50352,
"<|ha|>": 50354,
"<|he|>": 50279,
"<|hi|>": 50276,
"<|hr|>": 50291,
"<|ht|>": 50339,
"<|hu|>": 50286,
"<|hy|>": 50312,
"<|id|>": 50275,
"<|is|>": 50311,
"<|it|>": 50274,
"<|ja|>": 50266,
"<|jw|>": 50356,
"<|ka|>": 50329,
"<|kk|>": 50316,
"<|km|>": 50323,
"<|kn|>": 50306,
"<|ko|>": 50264,
"<|la|>": 50294,
"<|lb|>": 50345,
"<|ln|>": 50353,
"<|lo|>": 50336,
"<|lt|>": 50293,
"<|lv|>": 50301,
"<|mg|>": 50349,
"<|mi|>": 50295,
"<|mk|>": 50308,
"<|ml|>": 50296,
"<|mn|>": 50314,
"<|mr|>": 50320,
"<|ms|>": 50282,
"<|mt|>": 50343,
"<|my|>": 50346,
"<|ne|>": 50313,
"<|nl|>": 50271,
"<|nn|>": 50342,
"<|no|>": 50288,
"<|oc|>": 50328,
"<|pa|>": 50321,
"<|pl|>": 50269,
"<|ps|>": 50340,
"<|pt|>": 50267,
"<|ro|>": 50284,
"<|ru|>": 50263,
"<|sa|>": 50344,
"<|sd|>": 50332,
"<|si|>": 50322,
"<|sk|>": 50298,
"<|sl|>": 50305,
"<|sn|>": 50324,
"<|so|>": 50326,
"<|sq|>": 50317,
"<|sr|>": 50303,
"<|su|>": 50357,
"<|sv|>": 50273,
"<|sw|>": 50318,
"<|ta|>": 50287,
"<|te|>": 50299,
"<|tg|>": 50331,
"<|th|>": 50289,
"<|tk|>": 50341,
"<|tl|>": 50348,
"<|tr|>": 50268,
"<|tt|>": 50351,
"<|uk|>": 50280,
"<|ur|>": 50290,
"<|uz|>": 50337,
"<|vi|>": 50278,
"<|yi|>": 50335,
"<|yo|>": 50325,
"<|yue|>": 50358,
"<|zh|>": 50260
},
"max_initial_timestamp_index": 50,
"max_length": 448,
"no_timestamps_token_id": 50364,
"pad_token_id": 50257,
"prev_sot_token_id": 50362,
"return_timestamps": false,
"suppress_tokens": [
1,
2,
7,
8,
9,
10,
14,
25,
26,
27,
28,
29,
31,
58,
59,
60,
61,
62,
63,
90,
91,
92,
93,
359,
503,
522,
542,
873,
893,
902,
918,
922,
931,
1350,
1853,
1982,
2460,
2627,
3246,
3253,
3268,
3536,
3846,
3961,
4183,
4667,
6585,
6647,
7273,
9061,
9383,
10428,
10929,
11938,
12033,
12331,
12562,
13793,
14157,
14635,
15265,
15618,
16553,
16604,
18362,
18956,
20075,
21675,
22520,
26130,
26161,
26435,
28279,
29464,
31650,
32302,
32470,
36865,
42863,
47425,
49870,
50254,
50258,
50359,
50360,
50361,
50362,
50363
],
"task_to_id": {
"transcribe": 50360,
"translate": 50359
},
"transformers_version": "4.38.2"
}

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version https://git-lfs.github.com/spec/v1
oid sha256:1de8b4eb1b4c069060fc6da4f345c0d3a4153473c9f0a349554649064b4371a0
size 3025686376

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{
"chunk_length": 30,
"feature_extractor_type": "WhisperFeatureExtractor",
"feature_size": 128,
"hop_length": 160,
"n_fft": 400,
"n_samples": 480000,
"nb_max_frames": 3000,
"padding_side": "right",
"padding_value": 0.0,
"processor_class": "WhisperProcessor",
"return_attention_mask": false,
"sampling_rate": 16000
}

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#!/usr/bin/env python
# coding=utf-8
# Copyright 2023 The HuggingFace Inc. team. All rights reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
"""
Training the Whisper model for sequence to sequence speech recognition via teacher-student distillation.
"""
# You can also adapt this script for your own distillation tasks. Pointers for this are left as comments.
import logging
import os
import re
import shutil
import sys
import time
# from multiprocessing import set_start_method
from dataclasses import dataclass, field
from functools import partial
from pathlib import Path
from typing import Any, Dict, List, Optional, Union
import datasets
import numpy as np
import torch
import torch.nn as nn
import transformers
from accelerate import Accelerator
from accelerate.logging import get_logger
from datasets import (
DatasetDict,
IterableDataset,
load_dataset,
)
from huggingface_hub import Repository, create_repo
from torch.utils.data import DataLoader
from tqdm import tqdm
from transformers import (
AddedToken,
HfArgumentParser,
Seq2SeqTrainingArguments,
WhisperConfig,
WhisperFeatureExtractor,
WhisperForConditionalGeneration,
WhisperProcessor,
WhisperTokenizerFast,
get_scheduler,
set_seed,
)
from transformers.modeling_outputs import BaseModelOutput
from transformers.utils import check_min_version
from transformers.utils.versions import require_version
# https://stackoverflow.com/questions/71692354/facing-ssl-error-with-huggingface-pretrained-models
os.environ['CURL_CA_BUNDLE'] = ''
# disable warning message
os.environ['TOKENIZERS_PARALLELISM'] = 'false'
# Will error if the minimal version of Transformers is not installed. Remove at your own risks.
check_min_version("4.34.0.dev0")
require_version("datasets>=2.14.6", "To fix: `pip install --upgrade datasets`")
logger = get_logger(__name__)
# set_start_method("spawn")
@dataclass
class ModelArguments:
"""
Arguments pertaining to which model/config/tokenizer we are going to distill from.
"""
model_name_or_path: str = field(
metadata={"help": "Path to pretrained Whisper model or model identifier from huggingface.co/models"}
)
teacher_model_name_or_path: str = field(
metadata={"help": "Path to pretrained teacher model or model identifier from huggingface.co/models"}
)
config_name: Optional[str] = field(
default=None,
metadata={"help": "Pretrained config name or path if not the same as model_name"},
)
tokenizer_name: Optional[str] = field(
default=None,
metadata={"help": "Pretrained tokenizer name or path if not the same as model_name"},
)
feature_extractor_name: Optional[str] = field(
default=None,
metadata={"help": "feature extractor name or path if not the same as model_name"},
)
cache_dir: Optional[str] = field(
default=None,
metadata={"help": "Where to store the pretrained models downloaded from huggingface.co"},
)
use_fast_tokenizer: bool = field(
default=True,
metadata={"help": "Whether to use one of the fast tokenizer (backed by the tokenizers library) or not."},
)
model_revision: str = field(
default="main",
metadata={"help": "The specific model version to use (can be a branch name, tag name or commit id)."},
)
subfolder: str = field(
default="",
metadata={
"help": "In case the relevant files are located inside a subfolder of the model repo on huggingface.co, you can"
"specify the folder name here."
},
)
token: str = field(
default=None,
metadata={
"help": (
"The token to use as HTTP bearer authorization for remote files. If not specified, will use the token "
"generated when running `huggingface-cli login` (stored in `~/.huggingface`)."
)
},
)
@dataclass
class DataTrainingArguments:
"""
Arguments pertaining to what data we are going to input our model for training and eval.
"""
train_dataset_name: str = field(
default=None,
metadata={
"help": "The name of the training dataset to use (via the datasets library). Load and combine "
"multiple datasets by separating dataset ids by a '+' symbol. For example, to load LibriSpeech "
"and Common Voice, set `train_dataset_name='librispeech_asr+common_voice'`."
},
)
train_dataset_config_name: Optional[str] = field(
default=None,
metadata={
"help": "The configuration name of the training dataset to use (via the datasets library). Load and combine "
"multiple datasets by separating dataset configs by a '+' symbol. Note that the order of the configs should "
"match the order of the datasets."
},
)
dataset_cache_dir: Optional[str] = field(
default=None,
metadata={"help": "Path to cache directory for saving and loading datasets"},
)
overwrite_cache: bool = field(
default=False,
metadata={"help": "Overwrite the cached training and evaluation sets"},
)
preprocessing_num_workers: Optional[int] = field(
default=None,
metadata={"help": "The number of processes to use for the preprocessing if using non-streaming mode."},
)
preprocessing_batch_size: Optional[int] = field(
default=256,
metadata={"help": "Number of examples per batch provided to the `prepare_dataset` function."},
)
max_label_length: int = field(
default=128,
metadata={"help": "Truncate transcriptions that are longer `max_label_length` tokens."},
)
train_split_name: str = field(
default="train",
metadata={
"help": "The name of the training data set split to use (via the datasets library). Defaults to 'train'"
},
)
timestamp_probability: float = field(
default=0.2, metadata={"help": "Probability for training on timestamped tokens if the data contains it."}
)
return_timestamps: bool = field(
default=False, metadata={"help": "Whether or not to predict timestamps in the generation step."}
)
language: str = field(
default=None,
metadata={
"help": (
"Language for multilingual distillation. This argument should be set for multilingual distillation "
"only. For English speech recognition, it should be left as `None`."
)
},
)
task: str = field(
default="transcribe",
metadata={
"help": "Task, either `transcribe` for speech recognition or `translate` for speech translation."
"This argument should be set for multilingual distillation only. For English speech recognition, it should be left as `None`."
},
)
wandb_project: str = field(
default="distil-whisper",
metadata={"help": "The name of the wandb project."},
)
skip_logmel_transformation: bool = field(
default=False, metadata={
"help": "Whether or not to transform log-mel transformation. No need to transform if the dataset contains"
"log mel feature, otherwise it's required."
}
)
logmel_dataset_name: Optional[str] = field(
default=None, metadata={"help": "To upload the dataset with the log-mel feature."}
)
@dataclass
class DistillationTrainingArguments(Seq2SeqTrainingArguments):
freeze_encoder: Optional[bool] = field(
default=False,
metadata={
"help": (
"Whether to freeze the entire encoder model. Only recommended when the entire encoder has been "
"copied from the teacher model."
)
},
)
temperature: Optional[float] = field(
default=2.0, metadata={"help": "Temperature to anneal the logits when computing the softmax."}
)
kl_weight: Optional[float] = field(
default=1.0,
metadata={
"help": (
"Weighting assigned to the MSE loss in the KD formulation. MSE loss is "
"computed between the teacher-student hidden states and attentions."
)
},
)
dtype: Optional[str] = field(
default="float32",
metadata={
"help": (
"The data type (dtype) in which to run training. One of `float32` (full-precision), "
"`float16` or `bfloat16` (both half-precision)."
)
},
)
@dataclass
class DataCollatorSpeechSeq2SeqWithPadding:
"""
Data collator that will dynamically pad the inputs received.
Args:
processor ([`Wav2Vec2Processor`])
The processor used for proccessing the data.
decoder_start_token_id (:obj: `int`)
The start-of-sequence token id of the decoder.
decoder_prev_token_id (:obj: `int`)
The start-of-prompt token id of the decoder
input_padding (:obj:`bool`, :obj:`str` or :class:`~transformers.tokenization_utils_base.PaddingStrategy`, `optional`, defaults to :obj:`True`):
Select a strategy to pad the returned input sequences (according to the model's padding side and padding index)
among:
* :obj:`True` or :obj:`'longest'`: Pad to the longest sequence in the batch (or no padding if only a single
sequence if provided).
* :obj:`'max_length'`: Pad to a maximum length specified with the argument :obj:`max_length` or to the
maximum acceptable input length for the model if that argument is not provided.
* :obj:`False` or :obj:`'do_not_pad'` (default): No padding (i.e., can statistics a batch with sequences of
different lengths).
target_padding (:obj:`bool`, :obj:`str` or :class:`~transformers.tokenization_utils_base.PaddingStrategy`, `optional`, defaults to :obj:`True`):
Select a strategy to pad the returned target sequences (according to the model's padding side and padding index).
See above for details.
max_target_length (:obj:`int`, `optional`):
Maximum length of the ``labels`` of the returned list and optionally padding length (see above).
"""
processor: Any
decoder_start_token_id: int
decoder_prev_token_id: int
input_padding: Union[bool, str] = "max_length"
target_padding: Union[bool, str] = "max_length"
max_target_length: Optional[int] = None
def __call__(self, features: List[Dict[str, Union[List[int], np.ndarray]]]) -> Dict[str, np.ndarray]:
# split inputs and labels since they have to be of different lengths and need
# different padding methods
model_input_name = self.processor.model_input_names[0]
# dataloader returns a list of features which we convert to a dict
input_features = {model_input_name: [feature[model_input_name] for feature in features]}
label_features = {"input_ids": [feature["labels"] for feature in features]}
# reformat list to dict and set to pytorch format
batch = self.processor.feature_extractor.pad(
input_features,
padding=self.input_padding,
return_tensors="pt",
)
labels_batch = self.processor.tokenizer.pad(
label_features,
max_length=self.max_target_length,
padding=self.target_padding,
return_tensors="pt",
)
# shift labels to the right to get decoder input ids
labels = labels_batch["input_ids"]
decoder_input_ids = labels[:, :-1]
labels = labels[:, 1:]
labels_mask = labels_batch.attention_mask[:, 1:]
# replace padding with -100 to ignore correctly when computing the loss
labels = labels.masked_fill(labels_mask.ne(1), -100)
# replace initial prompt tokens with -100 to ignore correctly when computing the loss
bos_index = torch.argmax((labels == self.decoder_start_token_id).long(), dim=1)
bos_index = torch.where(bos_index > 0, bos_index + 1, bos_index)
prompt_mask = torch.arange(labels.shape[1]) < bos_index[:, None]
labels = torch.where(prompt_mask, -100, labels)
batch["labels"] = labels
batch["decoder_input_ids"] = decoder_input_ids
return batch
def log_metric(
accelerator,
metrics: Dict,
train_time: float,
step: int,
epoch: int,
learning_rate: float = None,
prefix: str = "train",
):
"""Helper function to log all training/evaluation metrics with the correct prefixes and styling."""
log_metrics = {}
for k, v in metrics.items():
log_metrics[f"{prefix}/{k}"] = v
log_metrics[f"{prefix}/time"] = train_time
log_metrics[f"{prefix}/epoch"] = epoch
if learning_rate is not None:
log_metrics[f"{prefix}/learning_rate"] = learning_rate
accelerator.log(log_metrics, step=step)
def get_layers_to_supervise(student_layers: int, teacher_layers: int) -> Dict:
"""Helper function to map the student layer i to the teacher layer j whose statistics we'd like them to emulate. Used
for MSE loss terms in distillation (hidden-states and activations). Student layers are paired with teacher layers
in equal increments, e.g. for a 12-layer model distilled to a 3-layer model, student layer 0 emulates teacher layer
3 (such that it behaves like the first 4 teacher layers), student layer 1 emulates teacher layer 7, and student layer
2 emulates teacher layer 11. This mapping is summarised by the dictionary: {0: 3, 1: 7, 2: 11}, which is precisely
the statistics of this function for the arguments (student_layers=3, teacher_layers=12)."""
layer_intervals = np.linspace(teacher_layers // student_layers - 1, teacher_layers - 1, student_layers, dtype=int)
layer_intervals[-1] = teacher_layers - 1
layer_map = {}
for student_layer, teacher_layer in enumerate(layer_intervals):
layer_map[student_layer] = teacher_layer
return layer_map
def sorted_checkpoints(output_dir=None, checkpoint_prefix="checkpoint") -> List[str]:
"""Helper function to sort saved checkpoints from oldest to newest."""
ordering_and_checkpoint_path = []
glob_checkpoints = [str(x) for x in Path(output_dir).glob(f"{checkpoint_prefix}-*") if os.path.isdir(x)]
for path in glob_checkpoints:
regex_match = re.match(f".*{checkpoint_prefix}-([0-9]+)", path)
if regex_match is not None and regex_match.groups() is not None:
ordering_and_checkpoint_path.append((int(regex_match.groups()[0]), path))
checkpoints_sorted = sorted(ordering_and_checkpoint_path)
checkpoints_sorted = [checkpoint[1] for checkpoint in checkpoints_sorted]
return checkpoints_sorted
def rotate_checkpoints(save_total_limit=None, output_dir=None, checkpoint_prefix="checkpoint") -> None:
"""Helper function to delete old checkpoints."""
if save_total_limit is None or save_total_limit <= 0:
return
# Check if we should delete older checkpoint(s)
checkpoints_sorted = sorted_checkpoints(output_dir=output_dir, checkpoint_prefix=checkpoint_prefix)
if len(checkpoints_sorted) <= save_total_limit:
return
number_of_checkpoints_to_delete = max(0, len(checkpoints_sorted) - save_total_limit)
checkpoints_to_be_deleted = checkpoints_sorted[:number_of_checkpoints_to_delete]
for checkpoint in checkpoints_to_be_deleted:
logger.info(f"Deleting older checkpoint [{checkpoint}] due to args.save_total_limit")
shutil.rmtree(checkpoint, ignore_errors=True)
_RE_CHECKPOINT = re.compile(r"^checkpoint-(\d+)-epoch-(\d+)$")
def get_last_checkpoint(folder):
content = os.listdir(folder)
checkpoints = [
path
for path in content
if _RE_CHECKPOINT.search(path) is not None and os.path.isdir(os.path.join(folder, path))
]
if len(checkpoints) == 0:
return
return os.path.join(folder, max(checkpoints, key=lambda x: int(_RE_CHECKPOINT.search(x).groups()[0])))
def get_parameter_names(model, forbidden_layer_types, forbidden_module=None):
"""
Returns the names of the model parameters that are not inside a forbidden layer or forbidden module.
Can be used to get a subset of parameter names for decay masks, or to exclude parameters from an optimiser
(e.g. if the module is frozen).
"""
result = []
for name, child in model.named_children():
result += [
f"{name}.{n}"
for n in get_parameter_names(child, forbidden_layer_types, forbidden_module)
if not (
isinstance(child, tuple(forbidden_layer_types))
or (child in tuple(forbidden_module) if forbidden_module is not None else False)
)
]
# Add model specific parameters (defined with nn.Parameter) since they are not in any child.
result += list(model._parameters.keys())
return result
def main():
# 1. Parse input arguments
# We keep distinct sets of args, for cleaner separation of model/data/training related args
parser = HfArgumentParser((ModelArguments, DataTrainingArguments, DistillationTrainingArguments))
if len(sys.argv) == 2 and sys.argv[1].endswith(".json"):
# If we pass only one argument to the script and it's the path to a json file,
# let's parse it to get our arguments.
model_args, data_args, training_args = parser.parse_json_file(json_file=os.path.abspath(sys.argv[1]))
else:
model_args, data_args, training_args = parser.parse_args_into_dataclasses()
# 2. Initialize the accelerator
# We will let the accelerator handle device placement for us in this example
# We simply have to specify the training precision and any trackers being used
# We'll use the same dtype arguments as our JAX/Flax training script and convert
# it to accelerate format
# The teacher model can safely be cast to the dtype of training since we don't
# update the params
if training_args.dtype == "float16":
mixed_precision = "fp16"
teacher_dtype = torch.float16
elif training_args.dtype == "bfloat16":
mixed_precision = "bf16"
teacher_dtype = torch.bfloat16
else:
mixed_precision = "no"
teacher_dtype = torch.float32
accelerator = Accelerator(
gradient_accumulation_steps=training_args.gradient_accumulation_steps,
mixed_precision=mixed_precision,
log_with=training_args.report_to,
project_dir=training_args.output_dir,
)
accelerator.init_trackers(project_name=data_args.wandb_project)
# 3. Set-up basic logging
# Create one log on every process with the configuration for debugging
logging.basicConfig(
format="%(asctime)s - %(levelname)s - %(name)s - %(message)s",
datefmt="%m/%d/%Y %H:%M:%S",
level=logging.INFO,
)
# Log a small summary on each proces
logger.warning(
f"Process rank: {training_args.local_rank}, device: {training_args.device}, n_gpu: {training_args.n_gpu}, "
f"distributed training: {training_args.parallel_mode.value == 'distributed'}, 16-bits training: {training_args.fp16}"
)
# Set the verbosity to info of the Transformers logger (on main process only)
if accelerator.is_local_main_process:
datasets.utils.logging.set_verbosity_warning()
transformers.utils.logging.set_verbosity_info()
else:
datasets.utils.logging.set_verbosity_error()
transformers.utils.logging.set_verbosity_error()
logger.info("Training/evaluation parameters %s", training_args)
# 4. Detecting last checkpoint and eventually continue from last checkpoint
last_checkpoint = None
if os.path.isdir(training_args.output_dir) and training_args.do_train and not training_args.overwrite_output_dir:
last_checkpoint = get_last_checkpoint(training_args.output_dir)
if last_checkpoint is None and len(os.listdir(training_args.output_dir)) > 0:
raise ValueError(
f"Output directory ({training_args.output_dir}) already exists and is not empty. "
"Use --overwrite_output_dir to overcome."
)
elif last_checkpoint is not None and training_args.resume_from_checkpoint is None:
logger.info(
f"Checkpoint detected, resuming training at {last_checkpoint}. To avoid this behavior, change "
"the `--output_dir` or add `--overwrite_output_dir` to train from scratch."
)
# 5. Handle the repository creation
if accelerator.is_main_process:
if training_args.push_to_hub:
# Retrieve of infer repo_name
repo_name = training_args.hub_model_id
if repo_name is None:
repo_name = Path(training_args.output_dir).absolute().name
# Create repo and retrieve repo_id
repo_id = create_repo(repo_name, exist_ok=True, token=training_args.hub_token).repo_id
# Clone repo locally
repo = Repository(training_args.output_dir, clone_from=repo_id, token=training_args.hub_token)
with open(os.path.join(training_args.output_dir, ".gitignore"), "w+") as gitignore:
if "wandb" not in gitignore:
gitignore.write("wandb\n")
elif training_args.output_dir is not None:
os.makedirs(training_args.output_dir, exist_ok=True)
accelerator.wait_for_everyone()
# 7. Load pretrained model, tokenizer, and feature extractor
feature_extractor = WhisperFeatureExtractor.from_pretrained(
(model_args.feature_extractor_name if model_args.feature_extractor_name else model_args.model_name_or_path),
cache_dir=model_args.cache_dir,
revision=model_args.model_revision,
token=model_args.token,
)
config = WhisperConfig.from_pretrained(
(model_args.config_name if model_args.config_name else model_args.model_name_or_path),
cache_dir=model_args.cache_dir,
revision=model_args.model_revision,
token=model_args.token,
)
tokenizer = WhisperTokenizerFast.from_pretrained(
(model_args.tokenizer_name if model_args.tokenizer_name else model_args.model_name_or_path),
cache_dir=model_args.cache_dir,
use_fast=model_args.use_fast_tokenizer,
revision=model_args.model_revision,
token=model_args.token,
)
# override timestamp tokens until tokenizer issues are fixed in transformers
timestamps = [AddedToken("<|%.2f|>" % (i * 0.02), lstrip=False, rstrip=False) for i in range(1500 + 1)]
tokenizer.add_tokens(timestamps)
teacher_model = WhisperForConditionalGeneration.from_pretrained(
model_args.teacher_model_name_or_path,
cache_dir=model_args.cache_dir,
token=model_args.token,
low_cpu_mem_usage=True,
torch_dtype=teacher_dtype,
)
student_model = WhisperForConditionalGeneration.from_pretrained(
model_args.model_name_or_path,
config=config,
cache_dir=model_args.cache_dir,
revision=model_args.model_revision,
subfolder=model_args.subfolder,
token=model_args.token,
low_cpu_mem_usage=True,
)
if student_model.config.decoder_start_token_id is None or teacher_model.config.decoder_start_token_id is None:
raise ValueError(
f"Make sure that `config.decoder_start_token_id` is correctly defined for both the "
f"student and teacher model. Got {student_model.config.decoder_start_token_id} for the "
f"student and {teacher_model.config.decoder_start_token_id} for the teacher."
)
share_hidden_states = training_args.freeze_encoder and student_model.config.d_model == teacher_model.config.d_model
# enable gradient checkpointing if necessary
if training_args.gradient_checkpointing:
student_model.gradient_checkpointing_enable()
# freeze student encoder if necessary
if training_args.freeze_encoder:
student_model.freeze_encoder()
student_model.model.encoder.gradient_checkpointing = False
if hasattr(teacher_model.generation_config, "is_multilingual") and teacher_model.generation_config.is_multilingual:
# We need to set the language and task ids for previously multilingual checkpoints
tokenizer.set_prefix_tokens(language=data_args.language, task=data_args.task, predict_timestamps=False)
student_model.generation_config.update(
**{
"language": data_args.language,
"task": data_args.task,
}
)
elif data_args.language is not None:
raise ValueError(
"Setting language token for an English-only checkpoint is not permitted. The language argument should "
"only be set for multilingual checkpoints."
)
# 8. Create a single speech processor - make sure all processes wait until data is saved
if accelerator.is_main_process:
feature_extractor.save_pretrained(training_args.output_dir)
tokenizer.save_pretrained(training_args.output_dir)
# save the config and generation config as well
config.save_pretrained(training_args.output_dir)
student_model.generation_config.save_pretrained(training_args.output_dir)
accelerator.wait_for_everyone()
processor = WhisperProcessor.from_pretrained(training_args.output_dir)
# 10. Preprocessing the datasets: we need to read the audio files as arrays and tokenize the targets.
set_seed(training_args.seed)
training_datasets = DatasetDict(
{
"train": load_dataset(
data_args.train_dataset_name,
data_args.train_dataset_config_name,
split=data_args.train_split_name,
trust_remote_code=True,
cache_dir=data_args.dataset_cache_dir,
token=model_args.token,
num_proc=data_args.preprocessing_num_workers
)
}
)
return_timestamps = data_args.return_timestamps if data_args.timestamp_probability > 0 else False
decoder_start_token_id = student_model.config.decoder_start_token_id # <|startoftranscript|>
decoder_prev_token_id = tokenizer.all_special_ids[-3] # <|startofprev|>
if not data_args.skip_logmel_transformation:
def prepare_train_dataset(batch):
"""Pre-process the raw dataset: Convert the audio arrays to log-mel spectrogram inputs"""
audio = [sample["array"] for sample in batch["audio"]]
inputs = feature_extractor(audio, sampling_rate=feature_extractor.sampling_rate)
batch["input_features"] = inputs.input_features
return batch
map_fn_train = partial(
training_datasets["train"].map,
keep_in_memory=True,
function=prepare_train_dataset,
remove_columns=["audio", "text", "whisper_transcript"],
batched=True,
batch_size=data_args.preprocessing_batch_size,
)
training_datasets = DatasetDict({
"train": map_fn_train(
num_proc=data_args.preprocessing_num_workers,
desc="obtain log-mel feature from audio"
)
})
if data_args.logmel_dataset_name:
try:
training_datasets.push_to_hub(
data_args.logmel_dataset_name, config_name=data_args.train_dataset_config_name
)
except Exception:
logger.exception(f"Failed to push dataset to {data_args.logmel_dataset_name}.")
# 12. Define Training Schedule
# Store some constants
per_device_train_batch_size = int(training_args.per_device_train_batch_size)
train_batch_size = per_device_train_batch_size * accelerator.num_processes
gradient_accumulation_steps = int(training_args.gradient_accumulation_steps)
if training_args.max_steps < 0:
num_epochs = int(training_args.num_train_epochs)
steps_per_epoch = len(training_datasets["train"]) // (train_batch_size * gradient_accumulation_steps)
total_train_steps = steps_per_epoch * num_epochs
elif training_args.max_steps > 0:
logger.info("max_steps is given, it will override any value given in num_train_epochs")
total_train_steps = int(training_args.max_steps)
# Setting a very large number of epochs so we go as many times as necessary over the iterator.
num_epochs = sys.maxsize
steps_per_epoch = total_train_steps
else:
raise ValueError("max_steps must be specified when training with a streaming (iterable) dataset")
# 13. Define optimizer, LR scheduler, collator
decay_parameters = get_parameter_names(
student_model,
[nn.LayerNorm],
forbidden_module=[student_model.model.encoder] if training_args.freeze_encoder else None,
)
decay_parameters = [name for name in decay_parameters if "bias" not in name]
optimizer_grouped_parameters = [
{
"params": [param for name, param in student_model.named_parameters() if name in decay_parameters],
"weight_decay": training_args.weight_decay,
},
{
"params": [param for name, param in student_model.named_parameters() if name not in decay_parameters],
"weight_decay": 0.0,
},
]
optimizer = torch.optim.AdamW(
params=optimizer_grouped_parameters,
lr=training_args.learning_rate,
betas=(training_args.adam_beta1, training_args.adam_beta2),
eps=training_args.adam_epsilon,
)
# LR scheduler gets stepped by `num_processes` each time -> account for this in warmup / total steps
lr_scheduler = get_scheduler(
name=training_args.lr_scheduler_type,
optimizer=optimizer,
num_warmup_steps=training_args.warmup_steps * accelerator.num_processes,
num_training_steps=total_train_steps * accelerator.num_processes,
)
max_label_length = (
data_args.max_label_length if data_args.max_label_length is not None else student_model.config.max_length
)
data_collator = DataCollatorSpeechSeq2SeqWithPadding(
processor=processor,
decoder_start_token_id=decoder_start_token_id,
decoder_prev_token_id=decoder_prev_token_id,
input_padding="longest",
target_padding="max_length",
max_target_length=max_label_length,
)
# 14. Define generation arguments - we need to do this before we wrap the models in DDP
# so that we can still access the configs
num_beams = (
training_args.generation_num_beams
if training_args.generation_num_beams is not None
else getattr(student_model.generation_config, "num_beams", 1)
)
gen_kwargs = {
"max_length": max_label_length,
"num_beams": num_beams,
"return_timestamps": return_timestamps,
}
if hasattr(teacher_model.generation_config, "is_multilingual") and teacher_model.generation_config.is_multilingual:
# forcing the language and task tokens helps multilingual models in their generations
gen_kwargs.update({"language": data_args.language, "task": data_args.task})
# 15. Prepare everything with accelerate
student_model, teacher_model, optimizer, lr_scheduler = accelerator.prepare(
student_model, teacher_model, optimizer, lr_scheduler
)
def kl_divergence(target_distribution, log_predicted_distribution, labels):
kl_loss = nn.KLDivLoss(reduction="none")
divergence = kl_loss(log_predicted_distribution, target_distribution)
# ignore padded tokens from divergence, i.e. where labels are not set to -100
padding_mask = labels >= 0
padding_mask = padding_mask.unsqueeze(-1)
divergence = divergence * padding_mask
# take the average over the mini-batch
divergence = divergence.sum() / padding_mask.sum()
return divergence
# Define gradient update step fn
def train_step(batch, temperature=2.0,):
student_model.train()
teacher_model.eval()
student_outputs = student_model(**batch)
with torch.no_grad():
if share_hidden_states:
# if the student and teacher share the same frozen encoder then we don't have to recompute the
# encoder hidden-states for the teacher model, we can just re-use from the student
encoder_outputs = BaseModelOutput(student_outputs.encoder_last_hidden_state)
teacher_outputs = teacher_model(encoder_outputs=encoder_outputs, labels=batch["labels"])
else:
# do the full forward pass for the teacher model (encoder + decoder)
teacher_outputs = teacher_model(**batch)
# CE (data) loss
ce_loss = student_outputs.loss
# rescale distribution by temperature to ensure gradients scale correctly
teacher_distribution = nn.functional.softmax(teacher_outputs.logits / temperature, dim=-1)
# log softmax of student predictions for numerical stability
student_distribution = nn.functional.log_softmax(student_outputs.logits / temperature, dim=-1)
# KL-divergence loss (scaled by temperature)
kl_loss = kl_divergence(teacher_distribution, student_distribution, batch["labels"]) * temperature**2
# use Distil-Whisper formulation (fix weight of CE loss and tune KL weight, 1 as default)
loss = 0.8 * ce_loss + training_args.kl_weight * kl_loss
metrics = {"loss": loss, "ce_loss": ce_loss, "kl_loss": kl_loss}
return loss, metrics
logger.info("***** Running training *****")
logger.info(f" Num examples = {total_train_steps * train_batch_size * gradient_accumulation_steps}")
logger.info(" Instantaneous batch size per device =" f" {training_args.per_device_train_batch_size}")
logger.info(" Gradient accumulation steps =" f" {gradient_accumulation_steps}")
logger.info(
f" Total train batch size (w. parallel & distributed) = {train_batch_size * gradient_accumulation_steps}"
)
logger.info(f" Total optimization steps = {total_train_steps}")
# ======================== Training ================================
train_time = 0
train_start = time.time()
steps_trained_progress_bar = tqdm(
range(total_train_steps), desc="Train steps ... ", position=0, disable=not accelerator.is_local_main_process
)
continue_training = True
epochs_trained = 0
cur_step = 0
checkpoint = None
if training_args.resume_from_checkpoint is not None:
checkpoint = training_args.resume_from_checkpoint
elif last_checkpoint is not None:
checkpoint = last_checkpoint
if checkpoint is not None:
accelerator.load_state(checkpoint)
# Find num steps and epoch from saved state string pattern
pattern = r"checkpoint-(\d+)-epoch-(\d+)"
match = re.search(pattern, checkpoint)
cur_step = int(match.group(1))
epochs_trained = int(match.group(2))
logger.info(" Continuing training from checkpoint, will skip to saved global_step")
logger.info(f" Continuing training from epoch {epochs_trained}")
logger.info(f" Continuing training from global step {cur_step}")
steps_trained_progress_bar.update(cur_step)
for epoch in range(0, epochs_trained):
training_datasets["train"] = training_datasets["train"].shuffle(training_args.seed)
if training_args.max_steps < 0:
# we know exactly the number of steps per epoch, so can skip through the required number of batches
resume_step = (cur_step - epochs_trained * steps_per_epoch) * gradient_accumulation_steps
else:
# Currently we don't know how many steps we've taken in the current epoch
# So we just shuffle the dataset one extra time and start from a fresh epoch
# This is "good enough" for our purposes but not fully correct
resume_step = None
training_datasets["train"] = training_datasets["train"].shuffle(training_args.seed)
else:
resume_step = None
for epoch in range(epochs_trained, num_epochs):
training_datasets["train"] = training_datasets["train"].shuffle(training_args.seed)
train_dataloader = DataLoader(
training_datasets["train"],
collate_fn=data_collator,
batch_size=per_device_train_batch_size,
num_workers=training_args.dataloader_num_workers,
pin_memory=training_args.dataloader_pin_memory,
)
train_dataloader = accelerator.prepare(train_dataloader)
if hasattr(train_dataloader, "dataset") and isinstance(train_dataloader.dataset, IterableDataset):
train_dataloader.dataset.set_epoch(epoch)
if resume_step is not None:
# Skip the first N batches in the dataloader when resuming from a checkpoint
train_dataloader = accelerator.skip_first_batches(train_dataloader, resume_step)
resume_step = None
for batch in train_dataloader:
with accelerator.accumulate(student_model):
loss, train_metric = train_step(batch, temperature=training_args.temperature)
accelerator.backward(loss)
if accelerator.sync_gradients:
accelerator.clip_grad_norm_(student_model.parameters(), training_args.max_grad_norm)
optimizer.step()
lr_scheduler.step()
optimizer.zero_grad()
# Check if the accelerator has performed an optimization step behind the scenes
if accelerator.sync_gradients:
steps_trained_progress_bar.update(1)
cur_step += 1
if cur_step % training_args.logging_steps == 0:
steps_trained_progress_bar.write(
f"Step... ({cur_step} / {total_train_steps} | Loss:"
f" {train_metric['loss']}, Learning Rate:"
f" {lr_scheduler.get_last_lr()[0]})"
)
log_metric(
accelerator,
metrics=train_metric,
learning_rate=lr_scheduler.get_last_lr()[0],
train_time=train_time + time.time() - train_start,
step=cur_step,
epoch=epoch,
prefix="train",
)
# save checkpoint and weights after each save_steps and at the end of training
if (cur_step % training_args.save_steps == 0) or cur_step == total_train_steps:
intermediate_dir = os.path.join(training_args.output_dir, f"checkpoint-{cur_step}-epoch-{epoch}")
accelerator.save_state(output_dir=intermediate_dir)
accelerator.wait_for_everyone()
if accelerator.is_main_process:
rotate_checkpoints(training_args.save_total_limit, output_dir=training_args.output_dir)
if cur_step == total_train_steps:
# un-wrap student model for save
student_model = accelerator.unwrap_model(student_model)
student_model.save_pretrained(training_args.output_dir)
# re-wrap student model for final eval
student_model = accelerator.prepare(student_model)
if training_args.push_to_hub:
repo.push_to_hub(
commit_message=f"Saving train state of step {cur_step}",
blocking=False,
)
# break condition
if cur_step == total_train_steps:
continue_training = False
break
if not continue_training:
break
accelerator.end_training()
if __name__ == "__main__":
main()

139
special_tokens_map.json Normal file
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@@ -0,0 +1,139 @@
{
"additional_special_tokens": [
"<|startoftranscript|>",
"<|en|>",
"<|zh|>",
"<|de|>",
"<|es|>",
"<|ru|>",
"<|ko|>",
"<|fr|>",
"<|ja|>",
"<|pt|>",
"<|tr|>",
"<|pl|>",
"<|ca|>",
"<|nl|>",
"<|ar|>",
"<|sv|>",
"<|it|>",
"<|id|>",
"<|hi|>",
"<|fi|>",
"<|vi|>",
"<|he|>",
"<|uk|>",
"<|el|>",
"<|ms|>",
"<|cs|>",
"<|ro|>",
"<|da|>",
"<|hu|>",
"<|ta|>",
"<|no|>",
"<|th|>",
"<|ur|>",
"<|hr|>",
"<|bg|>",
"<|lt|>",
"<|la|>",
"<|mi|>",
"<|ml|>",
"<|cy|>",
"<|sk|>",
"<|te|>",
"<|fa|>",
"<|lv|>",
"<|bn|>",
"<|sr|>",
"<|az|>",
"<|sl|>",
"<|kn|>",
"<|et|>",
"<|mk|>",
"<|br|>",
"<|eu|>",
"<|is|>",
"<|hy|>",
"<|ne|>",
"<|mn|>",
"<|bs|>",
"<|kk|>",
"<|sq|>",
"<|sw|>",
"<|gl|>",
"<|mr|>",
"<|pa|>",
"<|si|>",
"<|km|>",
"<|sn|>",
"<|yo|>",
"<|so|>",
"<|af|>",
"<|oc|>",
"<|ka|>",
"<|be|>",
"<|tg|>",
"<|sd|>",
"<|gu|>",
"<|am|>",
"<|yi|>",
"<|lo|>",
"<|uz|>",
"<|fo|>",
"<|ht|>",
"<|ps|>",
"<|tk|>",
"<|nn|>",
"<|mt|>",
"<|sa|>",
"<|lb|>",
"<|my|>",
"<|bo|>",
"<|tl|>",
"<|mg|>",
"<|as|>",
"<|tt|>",
"<|haw|>",
"<|ln|>",
"<|ha|>",
"<|ba|>",
"<|jw|>",
"<|su|>",
"<|yue|>",
"<|translate|>",
"<|transcribe|>",
"<|startoflm|>",
"<|startofprev|>",
"<|nospeech|>",
"<|notimestamps|>"
],
"bos_token": {
"content": "<|endoftext|>",
"lstrip": false,
"normalized": false,
"rstrip": false,
"single_word": false
},
"eos_token": {
"content": "<|endoftext|>",
"lstrip": false,
"normalized": false,
"rstrip": false,
"single_word": false
},
"pad_token": {
"content": "<|endoftext|>",
"lstrip": false,
"normalized": false,
"rstrip": false,
"single_word": false
},
"unk_token": {
"content": "<|endoftext|>",
"lstrip": false,
"normalized": false,
"rstrip": false,
"single_word": false
}
}

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vocab.json Normal file

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