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Deprecated: please check out the new repository [DeepSpeech](https://github.com/PaddlePaddle/DeepSpeech).
# DeepSpeech2 on PaddlePaddle
*DeepSpeech2 on PaddlePaddle* is an open-source implementation of end-to-end Automatic Speech Recognition (ASR) engine, based on [Baidu's Deep Speech 2 paper](http://proceedings.mlr.press/v48/amodei16.pdf), with [PaddlePaddle](https://github.com/PaddlePaddle/Paddle) platform. Our vision is to empower both industrial application and academic research on speech recognition, via an easy-to-use, efficient and scalable implementation, including training, inference & testing module, distributed [PaddleCloud](https://github.com/PaddlePaddle/cloud) training, and demo deployment. Besides, several pre-trained models for both English and Mandarin are also released.
## Table of Contents
- [Installation](#installation)
- [Getting Started](#getting-started)
- [Data Preparation](#data-preparation)
- [Training a Model](#training-a-model)
- [Data Augmentation Pipeline](#data-augmentation-pipeline)
- [Inference and Evaluation](#inference-and-evaluation)
- [Running in Docker Container](#running-in-docker-container)
- [Distributed Cloud Training](#distributed-cloud-training)
- [Hyper-parameters Tuning](#hyper-parameters-tuning)
- [Training for Mandarin Language](#training-for-mandarin-language)
- [Trying Live Demo with Your Own Voice](#trying-live-demo-with-your-own-voice)
- [Released Models](#released-models)
- [Experiments and Benchmarks](#experiments-and-benchmarks)
- [Questions and Help](#questions-and-help)
## Installation
To avoid the trouble of environment setup, [running in docker container](#running-in-docker-container) is highly recommended. Otherwise follow the guidelines below to install the dependencies manually.
### Prerequisites
- Python 2.7 only supported
- PaddlePaddle the latest version (please refer to the [Installation Guide](https://github.com/PaddlePaddle/Paddle#installation))
### Setup
```bash
git clone https://github.com/PaddlePaddle/models.git
cd models/deep_speech_2
sh setup.sh
```
## Getting Started
Several shell scripts provided in `./examples` will help us to quickly give it a try, for most major modules, including data preparation, model training, case inference and model evaluation, with a few public dataset (e.g. [LibriSpeech](http://www.openslr.org/12/), [Aishell](http://www.openslr.org/33)). Reading these examples will also help you to understand how to make it work with your own data.
Some of the scripts in `./examples` are configured with 8 GPUs. If you don't have 8 GPUs available, please modify `CUDA_VISIBLE_DEVICES` and `--trainer_count`. If you don't have any GPU available, please set `--use_gpu` to False to use CPUs instead. Besides, if out-of-memory problem occurs, just reduce `--batch_size` to fit.
Let's take a tiny sampled subset of [LibriSpeech dataset](http://www.openslr.org/12/) for instance.
- Go to directory
```bash
cd examples/tiny
```
Notice that this is only a toy example with a tiny sampled subset of LibriSpeech. If you would like to try with the complete dataset (would take several days for training), please go to `examples/librispeech` instead.
- Prepare the data
```bash
sh run_data.sh
```
`run_data.sh` will download dataset, generate manifests, collect normalizer's statistics and build vocabulary. Once the data preparation is done, you will find the data (only part of LibriSpeech) downloaded in `~/.cache/paddle/dataset/speech/libri` and the corresponding manifest files generated in `./data/tiny` as well as a mean stddev file and a vocabulary file. It has to be run for the very first time you run this dataset and is reusable for all further experiments.
- Train your own ASR model
```bash
sh run_train.sh
```
`run_train.sh` will start a training job, with training logs printed to stdout and model checkpoint of every pass/epoch saved to `./checkpoints/tiny`. These checkpoints could be used for training resuming, inference, evaluation and deployment.
- Case inference with an existing model
```bash
sh run_infer.sh
```
`run_infer.sh` will show us some speech-to-text decoding results for several (default: 10) samples with the trained model. The performance might not be good now as the current model is only trained with a toy subset of LibriSpeech. To see the results with a better model, you can download a well-trained (trained for several days, with the complete LibriSpeech) model and do the inference:
```bash
sh run_infer_golden.sh
```
- Evaluate an existing model
```bash
sh run_test.sh
```
`run_test.sh` will evaluate the model with Word Error Rate (or Character Error Rate) measurement. Similarly, you can also download a well-trained model and test its performance:
```bash
sh run_test_golden.sh
```
More detailed information are provided in the following sections. Wish you a happy journey with the *DeepSpeech2 on PaddlePaddle* ASR engine!
## Data Preparation
### Generate Manifest
*DeepSpeech2 on PaddlePaddle* accepts a textual **manifest** file as its data set interface. A manifest file summarizes a set of speech data, with each line containing some meta data (e.g. filepath, transcription, duration) of one audio clip, in [JSON](http://www.json.org/) format, such as:
```
{"audio_filepath": "/home/work/.cache/paddle/Libri/134686/1089-134686-0001.flac", "duration": 3.275, "text": "stuff it into you his belly counselled him"}
{"audio_filepath": "/home/work/.cache/paddle/Libri/134686/1089-134686-0007.flac", "duration": 4.275, "text": "a cold lucid indifference reigned in his soul"}
```
To use your custom data, you only need to generate such manifest files to summarize the dataset. Given such summarized manifests, training, inference and all other modules can be aware of where to access the audio files, as well as their meta data including the transcription labels.
For how to generate such manifest files, please refer to `data/librispeech/librispeech.py`, which will download data and generate manifest files for LibriSpeech dataset.
### Compute Mean & Stddev for Normalizer
To perform z-score normalization (zero-mean, unit stddev) upon audio features, we have to estimate in advance the mean and standard deviation of the features, with some training samples:
```bash
python tools/compute_mean_std.py \
--num_samples 2000 \
--specgram_type linear \
--manifest_paths data/librispeech/manifest.train \
--output_path data/librispeech/mean_std.npz
```
It will compute the mean and standard deviation of power spectrum feature with 2000 random sampled audio clips listed in `data/librispeech/manifest.train` and save the results to `data/librispeech/mean_std.npz` for further usage.
### Build Vocabulary
A vocabulary of possible characters is required to convert the transcription into a list of token indices for training, and in decoding, to convert from a list of indices back to text again. Such a character-based vocabulary can be built with `tools/build_vocab.py`.
```bash
python tools/build_vocab.py \
--count_threshold 0 \
--vocab_path data/librispeech/eng_vocab.txt \
--manifest_paths data/librispeech/manifest.train
```
It will write a vocabuary file `data/librispeeech/eng_vocab.txt` with all transcription text in `data/librispeech/manifest.train`, without vocabulary truncation (`--count_threshold 0`).
### More Help
For more help on arguments:
```bash
python data/librispeech/librispeech.py --help
python tools/compute_mean_std.py --help
python tools/build_vocab.py --help
```
## Training a model
`train.py` is the main caller of the training module. Examples of usage are shown below.
- Start training from scratch with 8 GPUs:
```
CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 python train.py --trainer_count 8
```
- Start training from scratch with 16 CPUs:
```
python train.py --use_gpu False --trainer_count 16
```
- Resume training from a checkpoint:
```
CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 \
python train.py \
--init_model_path CHECKPOINT_PATH_TO_RESUME_FROM
```
For more help on arguments:
```bash
python train.py --help
```
or refer to `example/librispeech/run_train.sh`.
## Data Augmentation Pipeline
Data augmentation has often been a highly effective technique to boost the deep learning performance. We augment our speech data by synthesizing new audios with small random perturbation (label-invariant transformation) added upon raw audios. You don't have to do the syntheses on your own, as it is already embedded into the data provider and is done on the fly, randomly for each epoch during training.
Six optional augmentation components are provided to be selected, configured and inserted into the processing pipeline.
- Volume Perturbation
- Speed Perturbation
- Shifting Perturbation
- Online Bayesian normalization
- Noise Perturbation (need background noise audio files)
- Impulse Response (need impulse audio files)
In order to inform the trainer of what augmentation components are needed and what their processing orders are, it is required to prepare in advance an *augmentation configuration file* in [JSON](http://www.json.org/) format. For example:
```
[{
"type": "speed",
"params": {"min_speed_rate": 0.95,
"max_speed_rate": 1.05},
"prob": 0.6
},
{
"type": "shift",
"params": {"min_shift_ms": -5,
"max_shift_ms": 5},
"prob": 0.8
}]
```
When the `--augment_conf_file` argument of `trainer.py` is set to the path of the above example configuration file, every audio clip in every epoch will be processed: with 60% of chance, it will first be speed perturbed with a uniformly random sampled speed-rate between 0.95 and 1.05, and then with 80% of chance it will be shifted in time with a random sampled offset between -5 ms and 5 ms. Finally this newly synthesized audio clip will be feed into the feature extractor for further training.
For other configuration examples, please refer to `conf/augmenatation.config.example`.
Be careful when utilizing the data augmentation technique, as improper augmentation will do harm to the training, due to the enlarged train-test gap.
## Inference and Evaluation
### Prepare Language Model
A language model is required to improve the decoder's performance. We have prepared two language models (with lossy compression) for users to download and try. One is for English and the other is for Mandarin. Users can simply run this to download the preprared language models:
```bash
cd models/lm
sh download_lm_en.sh
sh download_lm_ch.sh
```
If you wish to train your own better language model, please refer to [KenLM](https://github.com/kpu/kenlm) for tutorials. Here we provide some tips to show how we preparing our English and Mandarin language models. You can take it as a reference when you train your own.
#### English LM
The English corpus is from the [Common Crawl Repository](http://commoncrawl.org) and you can download it from [statmt](http://data.statmt.org/ngrams/deduped_en). We use part en.00 to train our English language model. There are some preprocessing steps before training:
* Characters not in \[A-Za-z0-9\s'\] (\s represents whitespace characters) are removed and Arabic numbers are converted to English numbers like 1000 to one thousand.
* Repeated whitespace characters are squeezed to one and the beginning whitespace characters are removed. Notice that all transcriptions are lowercase, so all characters are converted to lowercase.
* Top 400,000 most frequent words are selected to build the vocabulary and the rest are replaced with 'UNKNOWNWORD'.
Now the preprocessing is done and we get a clean corpus to train the language model. Our released language model are trained with agruments '-o 5 --prune 0 1 1 1 1'. '-o 5' means the max order of language model is 5. '--prune 0 1 1 1 1' represents count thresholds for each order and more specifically it will prune singletons for orders two and higher. To save disk storage we convert the arpa file to 'trie' binary file with arguments '-a 22 -q 8 -b 8'. '-a' represents the maximum number of leading bits of pointers in 'trie' to chop. '-q -b' are quantization parameters for probability and backoff.
#### Mandarin LM
Different from the English language model, Mandarin language model is character-based where each token is a Chinese character. We use internal corpus to train the released Mandarin language models. The corpus contain billions of tokens. The preprocessing has tiny difference from English language model and main steps include:
* The beginning and trailing whitespace characters are removed.
* English punctuations and Chinese punctuations are removed.
* A whitespace character between two tokens is inserted.
Please notice that the released language models only contain Chinese simplified characters. After preprocessing done we can begin to train the language model. The key training arguments for small LM is '-o 5 --prune 0 1 2 4 4' and '-o 5' for large LM. Please refer above section for the meaning of each argument. We also convert the arpa file to binary file using default settings.
### Speech-to-text Inference
An inference module caller `infer.py` is provided to infer, decode and visualize speech-to-text results for several given audio clips. It might help to have an intuitive and qualitative evaluation of the ASR model's performance.
- Inference with GPU:
```bash
CUDA_VISIBLE_DEVICES=0 python infer.py --trainer_count 1
```
- Inference with CPUs:
```bash
python infer.py --use_gpu False --trainer_count 12
```
We provide two types of CTC decoders: *CTC greedy decoder* and *CTC beam search decoder*. The *CTC greedy decoder* is an implementation of the simple best-path decoding algorithm, selecting at each timestep the most likely token, thus being greedy and locally optimal. The [*CTC beam search decoder*](https://arxiv.org/abs/1408.2873) otherwise utilizes a heuristic breadth-first graph search for reaching a near global optimality; it also requires a pre-trained KenLM language model for better scoring and ranking. The decoder type can be set with argument `--decoding_method`.
For more help on arguments:
```
python infer.py --help
```
or refer to `example/librispeech/run_infer.sh`.
### Evaluate a Model
To evaluate a model's performance quantitatively, please run:
- Evaluation with GPUs:
```bash
CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 python test.py --trainer_count 8
```
- Evaluation with CPUs:
```bash
python test.py --use_gpu False --trainer_count 12
```
The error rate (default: word error rate; can be set with `--error_rate_type`) will be printed.
For more help on arguments:
```bash
python test.py --help
```
or refer to `example/librispeech/run_test.sh`.
## Hyper-parameters Tuning
The hyper-parameters $\alpha$ (language model weight) and $\beta$ (word insertion weight) for the [*CTC beam search decoder*](https://arxiv.org/abs/1408.2873) often have a significant impact on the decoder's performance. It would be better to re-tune them on the validation set when the acoustic model is renewed.
`tools/tune.py` performs a 2-D grid search over the hyper-parameter $\alpha$ and $\beta$. You must provide the range of $\alpha$ and $\beta$, as well as the number of their attempts.
- Tuning with GPU:
```bash
CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 \
python tools/tune.py \
--trainer_count 8 \
--alpha_from 1.0 \
--alpha_to 3.2 \
--num_alphas 45 \
--beta_from 0.1 \
--beta_to 0.45 \
--num_betas 8
```
- Tuning with CPU:
```bash
python tools/tune.py --use_gpu False
```
The grid search will print the WER (word error rate) or CER (character error rate) at each point in the hyper-parameters space, and draw the error surface optionally. A proper hyper-parameters range should include the global minima of the error surface for WER/CER, as illustrated in the following figure.
<p align="center">
<img src="docs/images/tuning_error_surface.png" width=550>
<br/>An example error surface for tuning on the dev-clean set of LibriSpeech
</p>
Usually, as the figure shows, the variation of language model weight ($\alpha$) significantly affect the performance of CTC beam search decoder. And a better procedure is to first tune on serveral data batches (the number can be specified) to find out the proper range of hyper-parameters, then change to the whole validation set to carray out an accurate tuning.
After tuning, you can reset $\alpha$ and $\beta$ in the inference and evaluation modules to see if they really help improve the ASR performance. For more help
```bash
python tune.py --help
```
or refer to `example/librispeech/run_tune.sh`.
## Running in Docker Container
Docker is an open source tool to build, ship, and run distributed applications in an isolated environment. A Docker image for this project has been provided in [hub.docker.com](https://hub.docker.com) with all the dependencies installed, including the pre-built PaddlePaddle, CTC decoders, and other necessary Python and third-party packages. This Docker image requires the support of NVIDIA GPU, so please make sure its availiability and the [nvidia-docker](https://github.com/NVIDIA/nvidia-docker) has been installed.
Take several steps to launch the Docker image:
- Download the Docker image
```bash
nvidia-docker pull paddlepaddle/models:deep-speech-2
```
- Clone this repository
```
git clone https://github.com/PaddlePaddle/models.git
```
- Run the Docker image
```bash
sudo nvidia-docker run -it -v $(pwd)/models:/models paddlepaddle/models:deep-speech-2 /bin/bash
```
Now go back and start from the [Getting Started](#getting-started) section, you can execute training, inference and hyper-parameters tuning similarly in the Docker container.
## Distributed Cloud Training
We also provide a cloud training module for users to do the distributed cluster training on [PaddleCloud](https://github.com/PaddlePaddle/cloud), to achieve a much faster training speed with multiple machines. To start with this, please first install PaddleCloud client and register a PaddleCloud account, as described in [PaddleCloud Usage](https://github.com/PaddlePaddle/cloud/blob/develop/doc/usage_cn.md#%E4%B8%8B%E8%BD%BD%E5%B9%B6%E9%85%8D%E7%BD%AEpaddlecloud).
Please take the following steps to submit a training job:
- Go to directory:
```bash
cd cloud
```
- Upload data:
Data must be uploaded to PaddleCloud filesystem to be accessed within a cloud job. `pcloud_upload_data.sh` helps do the data packing and uploading:
```bash
sh pcloud_upload_data.sh
```
Given input manifests, `pcloud_upload_data.sh` will:
- Extract the audio files listed in the input manifests.
- Pack them into a specified number of tar files.
- Upload these tar files to PaddleCloud filesystem.
- Create cloud manifests by replacing local filesystem paths with PaddleCloud filesystem paths. New manifests will be used to inform the cloud jobs of audio files' location and their meta information.
It should be done only once for the very first time to do the cloud training. Later, the data is kept persisitent on the cloud filesystem and reusable for further job submissions.
For argument details please refer to [Train DeepSpeech2 on PaddleCloud](https://github.com/PaddlePaddle/models/tree/develop/deep_speech_2/cloud).
- Configure training arguments:
Configure the cloud job parameters in `pcloud_submit.sh` (e.g. `NUM_NODES`, `NUM_GPUS`, `CLOUD_TRAIN_DIR`, `JOB_NAME` etc.) and then configure other hyper-parameters for training in `pcloud_train.sh` (just as what you do for local training).
For argument details please refer to [Train DeepSpeech2 on PaddleCloud](https://github.com/PaddlePaddle/models/tree/develop/deep_speech_2/cloud).
- Submit the job:
By running:
```bash
sh pcloud_submit.sh
```
a training job has been submitted to PaddleCloud, with the job name printed to the console.
- Get training logs
Run this to list all the jobs you have submitted, as well as their running status:
```bash
paddlecloud get jobs
```
Run this, the corresponding job's logs will be printed.
```bash
paddlecloud logs -n 10000 $REPLACED_WITH_YOUR_ACTUAL_JOB_NAME
```
For more information about the usage of PaddleCloud, please refer to [PaddleCloud Usage](https://github.com/PaddlePaddle/cloud/blob/develop/doc/usage_cn.md#提交任务).
For more information about the DeepSpeech2 training on PaddleCloud, please refer to
[Train DeepSpeech2 on PaddleCloud](https://github.com/PaddlePaddle/models/tree/develop/deep_speech_2/cloud).
## Training for Mandarin Language
The key steps of training for Mandarin language are same to that of English language and we have also provided an example for Mandarin training with Aishell in ```examples/aishell```. As mentioned above, please execute ```sh run_data.sh```, ```sh run_train.sh```, ```sh run_test.sh``` and ```sh run_infer.sh``` to do data preparation, training, testing and inference correspondingly. We have also prepared a pre-trained model (downloaded by ./models/aishell/download_model.sh) for users to try with ```sh run_infer_golden.sh``` and ```sh run_test_golden.sh```. Notice that, different from English LM, the Mandarin LM is character-based and please run ```tools/tune.py``` to find an optimal setting.
## Trying Live Demo with Your Own Voice
Until now, an ASR model is trained and tested qualitatively (`infer.py`) and quantitatively (`test.py`) with existing audio files. But it is not yet tested with your own speech. `deploy/demo_server.py` and `deploy/demo_client.py` helps quickly build up a real-time demo ASR engine with the trained model, enabling you to test and play around with the demo, with your own voice.
To start the demo's server, please run this in one console:
```bash
CUDA_VISIBLE_DEVICES=0 \
python deploy/demo_server.py \
--trainer_count 1 \
--host_ip localhost \
--host_port 8086
```
For the machine (might not be the same machine) to run the demo's client, please do the following installation before moving on.
For example, on MAC OS X:
```bash
brew install portaudio
pip install pyaudio
pip install pynput
```
Then to start the client, please run this in another console:
```bash
CUDA_VISIBLE_DEVICES=0 \
python -u deploy/demo_client.py \
--host_ip 'localhost' \
--host_port 8086
```
Now, in the client console, press the `whitespace` key, hold, and start speaking. Until finishing your utterance, release the key to let the speech-to-text results shown in the console. To quit the client, just press `ESC` key.
Notice that `deploy/demo_client.py` must be run on a machine with a microphone device, while `deploy/demo_server.py` could be run on one without any audio recording hardware, e.g. any remote server machine. Just be careful to set the `host_ip` and `host_port` argument with the actual accessible IP address and port, if the server and client are running with two separate machines. Nothing should be done if they are running on one single machine.
Please also refer to `examples/mandarin/run_demo_server.sh`, which will first download a pre-trained Mandarin model (trained with 3000 hours of internal speech data) and then start the demo server with the model. With running `examples/mandarin/run_demo_client.sh`, you can speak Mandarin to test it. If you would like to try some other models, just update `--model_path` argument in the script.  
For more help on arguments:
```bash
python deploy/demo_server.py --help
python deploy/demo_client.py --help
```
## Released Models
#### Speech Model Released
Language | Model Name | Training Data | Hours of Speech
:-----------: | :------------: | :----------: | -------:
English | [LibriSpeech Model](http://cloud.dlnel.org/filepub/?uuid=17404caf-cf19-492f-9707-1fad07c19aae) | [LibriSpeech Dataset](http://www.openslr.org/12/) | 960 h
English | [BaiduEN8k Model](to-be-added) | Baidu Internal English Dataset | 8628 h
Mandarin | [Aishell Model](http://cloud.dlnel.org/filepub/?uuid=6c83b9d8-3255-4adf-9726-0fe0be3d0274) | [Aishell Dataset](http://www.openslr.org/33/) | 151 h
Mandarin | [BaiduCN1.2k Model](to-be-added) | Baidu Internal Mandarin Dataset | 1204 h
#### Language Model Released
Language Model | Training Data | Token-based | Size | Descriptions
:-------------:| :------------:| :-----: | -----: | :-----------------
[English LM](http://paddlepaddle.bj.bcebos.com/model_zoo/speech/common_crawl_00.prune01111.trie.klm) | [CommonCrawl(en.00)](http://web-language-models.s3-website-us-east-1.amazonaws.com/ngrams/en/deduped/en.00.deduped.xz) | Word-based | 8.3 GB | Pruned with 0 1 1 1 1; <br/> About 1.85 billion n-grams; <br/> 'trie' binary with '-a 22 -q 8 -b 8'
[Mandarin LM Small](http://cloud.dlnel.org/filepub/?uuid=d21861e4-4ed6-45bb-ad8e-ae417a43195e) | Baidu Internal Corpus | Char-based | 2.8 GB | Pruned with 0 1 2 4 4; <br/> About 0.13 billion n-grams; <br/> 'probing' binary with default settings
[Mandarin LM Large](http://cloud.dlnel.org/filepub/?uuid=245d02bb-cd01-4ebe-b079-b97be864ec37) | Baidu Internal Corpus | Char-based | 70.4 GB | No Pruning; <br/> About 3.7 billion n-grams; <br/> 'probing' binary with default settings
## Experiments and Benchmarks
#### Benchmark Results for English Models (Word Error Rate)
Test Set | LibriSpeech Model | BaiduEN8K Model
:--------------------- | ---------------: | -------------------:
LibriSpeech Test-Clean | 7.77 | 6.63
LibriSpeech Test-Other | 23.25 | 16.59
VoxForge American-Canadian | 12.52 |   7.46
VoxForge Commonwealth | 21.08 | 16.23
VoxForge European | 31.21 | 20.47
VoxForge Indian | 56.79 | 28.15
Baidu Internal Testset  |   47.73 |   8.92
#### Benchmark Results for Mandarin Model (Character Error Rate)
Test Set | Aishell Model | BaiduCN1.2k Model
:--------------------- | ---------------: | -------------------:
Baidu Internal Testset | - | 15.49
#### Acceleration with Multi-GPUs
We compare the training time with 1, 2, 4, 8, 16 Tesla K40m GPUs (with a subset of LibriSpeech samples whose audio durations are between 6.0 and 7.0 seconds). And it shows that a **near-linear** acceleration with multiple GPUs has been achieved. In the following figure, the time (in seconds) cost for training is printed on the blue bars.
<img src="docs/images/multi_gpu_speedup.png" width=450><br/>
| # of GPU | Acceleration Rate |
| -------- | --------------: |
| 1 | 1.00 X |
| 2 | 1.97 X |
| 4 | 3.74 X |
| 8 | 6.21 X |
|16 | 10.70 X |
`tools/profile.sh` provides such a profiling tool.
## Questions and Help
You are welcome to submit questions and bug reports in [Github Issues](https://github.com/PaddlePaddle/models/issues). You are also welcome to contribute to this project.
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浏览文件 @ 788f7011
# Train DeepSpeech2 on PaddleCloud
>Note:
>Please make sure [PaddleCloud Client](https://github.com/PaddlePaddle/cloud/blob/develop/doc/usage_cn.md#%E4%B8%8B%E8%BD%BD%E5%B9%B6%E9%85%8D%E7%BD%AEpaddlecloud) has be installed and current directory is `deep_speech_2/cloud/`
## Step 1: Upload Data
Provided with several input manifests, `pcloud_upload_data.sh` will pack and upload all the containing audio files to PaddleCloud filesystem, and also generate some corresponding manifest files with updated cloud paths.
Please modify the following arguments in `pcloud_upload_data.sh`:
- `IN_MANIFESTS`： Paths (in local filesystem) of manifest files containing the audio files to be uploaded. Multiple paths can be concatenated with a whitespace delimeter.
- `OUT_MANIFESTS`: Paths (in local filesystem) to write the updated output manifest files to. Multiple paths can be concatenated with a whitespace delimeter. The values of `audio_filepath` in the output manifests are updated with cloud filesystem paths.
- `CLOUD_DATA_DIR`: Directory (in PaddleCloud filesystem) to upload the data to. Don't forget to replace `USERNAME` in the default directory and make sure that you have the permission to write it.
- `NUM_SHARDS`: Number of data shards / parts (in tar files) to be generated when packing and uploading data. Smaller `num_shards` requires larger temoporal local disk space for packing data.
By running:
```
sh pcloud_upload_data.sh
```
all the audio files will be uploaded to PaddleCloud filesystem, and you will get modified manifests files in `OUT_MANIFESTS`.
You have to take this step only once, in the very first time you do the cloud training. Later on, the data is persisitent on the cloud filesystem and reusable for further job submissions.
## Step 2: Configure Training
Configure cloud training arguments in `pcloud_submit.sh`, with the following arguments:
- `TRAIN_MANIFEST`: Manifest filepath (in local filesystem) for training. Notice that the`audio_filepath` should be in cloud filesystem, like those generated by `pcloud_upload_data.sh`.
- `DEV_MANIFEST`: Manifest filepath (in local filesystem) for validation.
- `CLOUD_MODEL_DIR`: Directory (in PaddleCloud filesystem) to save the model parameters (checkpoints). Don't forget to replace `USERNAME` in the default directory and make sure that you have the permission to write it.
- `BATCH_SIZE`: Training batch size for a single node.
- `NUM_GPU`: Number of GPUs allocated for a single node.
- `NUM_NODE`: Number of nodes (machines) allocated for this job.
- `IS_LOCAL`: Set to False to enable parameter server, if using multiple nodes.
Configure other training hyper-parameters in `pcloud_train.sh` as you wish, just as what you can do in local training.
By running:
```
sh pcloud_submit.sh
```
you submit a training job to PaddleCloud. And you will see the job name when the submission is done.
## Step 3 Get Job Logs
Run this to list all the jobs you have submitted, as well as their running status:
```
paddlecloud get jobs
```
Run this, the corresponding job's logs will be printed.
```
paddlecloud logs -n 10000 $REPLACED_WITH_YOUR_ACTUAL_JOB_NAME
```
## More Help
For more information about the usage of PaddleCloud, please refer to [PaddleCloud Usage](https://github.com/PaddlePaddle/cloud/blob/develop/doc/usage_cn.md#提交任务).
deep_speech_2/cloud/_init_paths.py 已删除 100644 → 0
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"""Set up paths for DS2"""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import os.path
import sys
def add_path(path):
if path not in sys.path:
sys.path.insert(0, path)
this_dir = os.path.dirname(__file__)
proj_path = os.path.join(this_dir, '..')
add_path(proj_path)
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#! /usr/bin/env bash
TRAIN_MANIFEST="cloud/cloud_manifests/cloud.manifest.train"
DEV_MANIFEST="cloud/cloud_manifests/cloud.manifest.dev"
CLOUD_MODEL_DIR="./checkpoints"
BATCH_SIZE=512
NUM_GPU=8
NUM_NODE=1
IS_LOCAL="True"
JOB_NAME=deepspeech-`date +%Y%m%d%H%M%S`
DS2_PATH=${PWD%/*}
cp -f pcloud_train.sh ${DS2_PATH}
paddlecloud submit \
-image bootstrapper:5000/paddlepaddle/pcloud_ds2:latest \
-jobname ${JOB_NAME} \
-cpu ${NUM_GPU} \
-gpu ${NUM_GPU} \
-memory 64Gi \
-parallelism ${NUM_NODE} \
-pscpu 1 \
-pservers 1 \
-psmemory 64Gi \
-passes 1 \
-entry "sh pcloud_train.sh ${TRAIN_MANIFEST} ${DEV_MANIFEST} ${CLOUD_MODEL_DIR} ${NUM_GPU} ${BATCH_SIZE} ${IS_LOCAL}" \
${DS2_PATH}
rm ${DS2_PATH}/pcloud_train.sh
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#! /usr/bin/env bash
TRAIN_MANIFEST=$1
DEV_MANIFEST=$2
MODEL_PATH=$3
NUM_GPU=$4
BATCH_SIZE=$5
IS_LOCAL=$6
python ./cloud/split_data.py \
--in_manifest_path=${TRAIN_MANIFEST} \
--out_manifest_path='/local.manifest.train'
python ./cloud/split_data.py \
--in_manifest_path=${DEV_MANIFEST} \
--out_manifest_path='/local.manifest.dev'
mkdir ./logs
python -u train.py \
--batch_size=${BATCH_SIZE} \
--trainer_count=${NUM_GPU} \
--num_passes=200 \
--num_proc_data=${NUM_GPU} \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=2048 \
--num_iter_print=100 \
--learning_rate=5e-4 \
--max_duration=27.0 \
--min_duration=0.0 \
--use_sortagrad=True \
--use_gru=False \
--use_gpu=True \
--is_local=${IS_LOCAL} \
--share_rnn_weights=True \
--train_manifest='/local.manifest.train' \
--dev_manifest='/local.manifest.dev' \
--mean_std_path='data/librispeech/mean_std.npz' \
--vocab_path='data/librispeech/vocab.txt' \
--output_model_dir='./checkpoints' \
--output_model_dir=${MODEL_PATH} \
--augment_conf_path='conf/augmentation.config' \
--specgram_type='linear' \
--shuffle_method='batch_shuffle_clipped' \
2>&1 | tee ./logs/train.log
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#! /usr/bin/env bash
mkdir cloud_manifests
IN_MANIFESTS="../data/librispeech/manifest.train ../data/librispeech/manifest.dev-clean ../data/librispeech/manifest.test-clean"
OUT_MANIFESTS="cloud_manifests/cloud.manifest.train cloud_manifests/cloud.manifest.dev cloud_manifests/cloud.manifest.test"
CLOUD_DATA_DIR="/pfs/dlnel/home/USERNAME/deepspeech2/data/librispeech"
NUM_SHARDS=50
python upload_data.py \
--in_manifest_paths ${IN_MANIFESTS} \
--out_manifest_paths ${OUT_MANIFESTS} \
--cloud_data_dir ${CLOUD_DATA_DIR} \
--num_shards ${NUM_SHARDS}
if [ $? -ne 0 ]
then
echo "Upload Data Failed!"
exit 1
fi
echo "All Done."
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"""This tool is used for splitting data into each node of
paddlecloud. This script should be called in paddlecloud.
"""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import os
import json
import argparse
parser = argparse.ArgumentParser(description=__doc__)
parser.add_argument(
"--in_manifest_path",
type=str,
required=True,
help="Input manifest path for all nodes.")
parser.add_argument(
"--out_manifest_path",
type=str,
required=True,
help="Output manifest file path for current node.")
args = parser.parse_args()
def split_data(in_manifest_path, out_manifest_path):
with open("/trainer_id", "r") as f:
trainer_id = int(f.readline()[:-1])
with open("/trainer_count", "r") as f:
trainer_count = int(f.readline()[:-1])
out_manifest = []
for index, json_line in enumerate(open(in_manifest_path, 'r')):
if (index % trainer_count) == trainer_id:
out_manifest.append("%s\n" % json_line.strip())
with open(out_manifest_path, 'w') as f:
f.writelines(out_manifest)
if __name__ == '__main__':
split_data(args.in_manifest_path, args.out_manifest_path)
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"""This script is for uploading data for DeepSpeech2 training on paddlecloud.
Steps:
1. Read original manifests and extract local sound files.
2. Tar all local sound files into multiple tar files and upload them.
3. Modify original manifests with updated paths in cloud filesystem.
"""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import json
import os
import tarfile
import sys
import argparse
import shutil
from subprocess import call
import _init_paths
from data_utils.utils import read_manifest
parser = argparse.ArgumentParser(description=__doc__)
parser.add_argument(
"--in_manifest_paths",
default=[
"../datasets/manifest.train", "../datasets/manifest.dev",
"../datasets/manifest.test"
],
type=str,
nargs='+',
help="Local filepaths of input manifests to load, pack and upload."
"(default: %(default)s)")
parser.add_argument(
"--out_manifest_paths",
default=[
"./cloud.manifest.train", "./cloud.manifest.dev",
"./cloud.manifest.test"
],
type=str,
nargs='+',
help="Local filepaths of modified manifests to write to. "
"(default: %(default)s)")
parser.add_argument(
"--cloud_data_dir",
required=True,
type=str,
help="Destination directory on paddlecloud to upload data to.")
parser.add_argument(
"--num_shards",
default=10,
type=int,
help="Number of parts to split data to. (default: %(default)s)")
parser.add_argument(
"--local_tmp_dir",
default="./tmp/",
type=str,
help="Local directory for storing temporary data. (default: %(default)s)")
args = parser.parse_args()
def upload_data(in_manifest_path_list, out_manifest_path_list, local_tmp_dir,
upload_tar_dir, num_shards):
"""Extract and pack sound files listed in the manifest files into multple
tar files and upload them to padldecloud. Besides, generate new manifest
files with updated paths in paddlecloud.
"""
# compute total audio number
total_line = 0
for manifest_path in in_manifest_path_list:
with open(manifest_path, 'r') as f:
total_line += len(f.readlines())
line_per_tar = (total_line // num_shards) + 1
# pack and upload shard by shard
line_count, tar_file = 0, None
for manifest_path, out_manifest_path in zip(in_manifest_path_list,
out_manifest_path_list):
manifest = read_manifest(manifest_path)
out_manifest = []
for json_data in manifest:
sound_filepath = json_data['audio_filepath']
sound_filename = os.path.basename(sound_filepath)
if line_count % line_per_tar == 0:
if tar_file != None:
tar_file.close()
pcloud_cp(tar_path, upload_tar_dir)
os.remove(tar_path)
tar_name = 'part-%s-of-%s.tar' % (
str(line_count // line_per_tar).zfill(5),
str(num_shards).zfill(5))
tar_path = os.path.join(local_tmp_dir, tar_name)
tar_file = tarfile.open(tar_path, 'w')
tar_file.add(sound_filepath, arcname=sound_filename)
line_count += 1
json_data['audio_filepath'] = "tar:%s#%s" % (
os.path.join(upload_tar_dir, tar_name), sound_filename)
out_manifest.append("%s\n" % json.dumps(json_data))
with open(out_manifest_path, 'w') as f:
f.writelines(out_manifest)
pcloud_cp(out_manifest_path, upload_tar_dir)
tar_file.close()
pcloud_cp(tar_path, upload_tar_dir)
os.remove(tar_path)
def pcloud_mkdir(dir):
"""Make directory in PaddleCloud filesystem.
"""
if call(['paddlecloud', 'mkdir', dir]) != 0:
raise IOError("PaddleCloud mkdir failed: %s." % dir)
def pcloud_cp(src, dst):
"""Copy src from local filesytem to dst in PaddleCloud filesystem,
or downlowd src from PaddleCloud filesystem to dst in local filesystem.
"""
if call(['paddlecloud', 'cp', src, dst]) != 0:
raise IOError("PaddleCloud cp failed: from [%s] to [%s]." % (src, dst))
if __name__ == '__main__':
if not os.path.exists(args.local_tmp_dir):
os.makedirs(args.local_tmp_dir)
pcloud_mkdir(args.cloud_data_dir)
upload_data(args.in_manifest_paths, args.out_manifest_paths,
args.local_tmp_dir, args.cloud_data_dir, args.num_shards)
shutil.rmtree(args.local_tmp_dir)
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[
{
"type": "shift",
"params": {"min_shift_ms": -5,
"max_shift_ms": 5},
"prob": 1.0
}
]
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[
{
"type": "noise",
"params": {"min_snr_dB": 40,
"max_snr_dB": 50,
"noise_manifest_path": "datasets/manifest.noise"},
"prob": 0.6
},
{
"type": "impulse",
"params": {"impulse_manifest_path": "datasets/manifest.impulse"},
"prob": 0.5
},
{
"type": "speed",
"params": {"min_speed_rate": 0.95,
"max_speed_rate": 1.05},
"prob": 0.5
},
{
"type": "shift",
"params": {"min_shift_ms": -5,
"max_shift_ms": 5},
"prob": 1.0
},
{
"type": "volume",
"params": {"min_gain_dBFS": -10,
"max_gain_dBFS": 10},
"prob": 0.0
},
{
"type": "bayesian_normal",
"params": {"target_db": -20,
"prior_db": -20,
"prior_samples": 100},
"prob": 0.0
}
]
deep_speech_2/data/aishell/aishell.py 已删除 100644 → 0
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"""Prepare Aishell mandarin dataset
Download, unpack and create manifest files.
Manifest file is a json-format file with each line containing the
meta data (i.e. audio filepath, transcript and audio duration)
of each audio file in the data set.
"""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import os
import codecs
import soundfile
import json
import argparse
from data_utils.utility import download, unpack
DATA_HOME = os.path.expanduser('~/.cache/paddle/dataset/speech')
URL_ROOT = 'http://www.openslr.org/resources/33'
DATA_URL = URL_ROOT + '/data_aishell.tgz'
MD5_DATA = '2f494334227864a8a8fec932999db9d8'
parser = argparse.ArgumentParser(description=__doc__)
parser.add_argument(
"--target_dir",
default=DATA_HOME + "/Aishell",
type=str,
help="Directory to save the dataset. (default: %(default)s)")
parser.add_argument(
"--manifest_prefix",
default="manifest",
type=str,
help="Filepath prefix for output manifests. (default: %(default)s)")
args = parser.parse_args()
def create_manifest(data_dir, manifest_path_prefix):
print("Creating manifest %s ..." % manifest_path_prefix)
json_lines = []
transcript_path = os.path.join(data_dir, 'transcript',
'aishell_transcript_v0.8.txt')
transcript_dict = {}
for line in codecs.open(transcript_path, 'r', 'utf-8'):
line = line.strip()
if line == '': continue
audio_id, text = line.split(' ', 1)
# remove withespace
text = ''.join(text.split())
transcript_dict[audio_id] = text
data_types = ['train', 'dev', 'test']
for type in data_types:
audio_dir = os.path.join(data_dir, 'wav', type)
for subfolder, _, filelist in sorted(os.walk(audio_dir)):
for fname in filelist:
audio_path = os.path.join(subfolder, fname)
audio_id = fname[:-4]
# if no transcription for audio then skipped
if audio_id not in transcript_dict:
continue
audio_data, samplerate = soundfile.read(audio_path)
duration = float(len(audio_data) / samplerate)
text = transcript_dict[audio_id]
json_lines.append(
json.dumps(
{
'audio_filepath': audio_path,
'duration': duration,
'text': text
},
ensure_ascii=False))
manifest_path = manifest_path_prefix + '.' + type
with codecs.open(manifest_path, 'w', 'utf-8') as fout:
for line in json_lines:
fout.write(line + '\n')
def prepare_dataset(url, md5sum, target_dir, manifest_path):
"""Download, unpack and create manifest file."""
data_dir = os.path.join(target_dir, 'data_aishell')
if not os.path.exists(data_dir):
filepath = download(url, md5sum, target_dir)
unpack(filepath, target_dir)
# unpack all audio tar files
audio_dir = os.path.join(data_dir, 'wav')
for subfolder, _, filelist in sorted(os.walk(audio_dir)):
for ftar in filelist:
unpack(os.path.join(subfolder, ftar), subfolder, True)
else:
print("Skip downloading and unpacking. Data already exists in %s." %
target_dir)
create_manifest(data_dir, manifest_path)
def main():
if args.target_dir.startswith('~'):
args.target_dir = os.path.expanduser(args.target_dir)
prepare_dataset(
url=DATA_URL,
md5sum=MD5_DATA,
target_dir=args.target_dir,
manifest_path=args.manifest_prefix)
if __name__ == '__main__':
main()
deep_speech_2/data/librispeech/librispeech.py 已删除 100644 → 0
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"""Prepare Librispeech ASR datasets.
Download, unpack and create manifest files.
Manifest file is a json-format file with each line containing the
meta data (i.e. audio filepath, transcript and audio duration)
of each audio file in the data set.
"""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import distutils.util
import os
import sys
import argparse
import soundfile
import json
import codecs
from data_utils.utility import download, unpack
URL_ROOT = "http://www.openslr.org/resources/12"
URL_TEST_CLEAN = URL_ROOT + "/test-clean.tar.gz"
URL_TEST_OTHER = URL_ROOT + "/test-other.tar.gz"
URL_DEV_CLEAN = URL_ROOT + "/dev-clean.tar.gz"
URL_DEV_OTHER = URL_ROOT + "/dev-other.tar.gz"
URL_TRAIN_CLEAN_100 = URL_ROOT + "/train-clean-100.tar.gz"
URL_TRAIN_CLEAN_360 = URL_ROOT + "/train-clean-360.tar.gz"
URL_TRAIN_OTHER_500 = URL_ROOT + "/train-other-500.tar.gz"
MD5_TEST_CLEAN = "32fa31d27d2e1cad72775fee3f4849a9"
MD5_TEST_OTHER = "fb5a50374b501bb3bac4815ee91d3135"
MD5_DEV_CLEAN = "42e2234ba48799c1f50f24a7926300a1"
MD5_DEV_OTHER = "c8d0bcc9cca99d4f8b62fcc847357931"
MD5_TRAIN_CLEAN_100 = "2a93770f6d5c6c964bc36631d331a522"
MD5_TRAIN_CLEAN_360 = "c0e676e450a7ff2f54aeade5171606fa"
MD5_TRAIN_OTHER_500 = "d1a0fd59409feb2c614ce4d30c387708"
parser = argparse.ArgumentParser(description=__doc__)
parser.add_argument(
"--target_dir",
default='~/.cache/paddle/dataset/speech/libri',
type=str,
help="Directory to save the dataset. (default: %(default)s)")
parser.add_argument(
"--manifest_prefix",
default="manifest",
type=str,
help="Filepath prefix for output manifests. (default: %(default)s)")
parser.add_argument(
"--full_download",
default="True",
type=distutils.util.strtobool,
help="Download all datasets for Librispeech."
" If False, only download a minimal requirement (test-clean, dev-clean"
" train-clean-100). (default: %(default)s)")
args = parser.parse_args()
def create_manifest(data_dir, manifest_path):
"""Create a manifest json file summarizing the data set, with each line
containing the meta data (i.e. audio filepath, transcription text, audio
duration) of each audio file within the data set.
"""
print("Creating manifest %s ..." % manifest_path)
json_lines = []
for subfolder, _, filelist in sorted(os.walk(data_dir)):
text_filelist = [
filename for filename in filelist if filename.endswith('trans.txt')
]
if len(text_filelist) > 0:
text_filepath = os.path.join(data_dir, subfolder, text_filelist[0])
for line in open(text_filepath):
segments = line.strip().split()
text = ' '.join(segments[1:]).lower()
audio_filepath = os.path.join(data_dir, subfolder,
segments[0] + '.flac')
audio_data, samplerate = soundfile.read(audio_filepath)
duration = float(len(audio_data)) / samplerate
json_lines.append(
json.dumps({
'audio_filepath': audio_filepath,
'duration': duration,
'text': text
}))
with codecs.open(manifest_path, 'w', 'utf-8') as out_file:
for line in json_lines:
out_file.write(line + '\n')
def prepare_dataset(url, md5sum, target_dir, manifest_path):
"""Download, unpack and create summmary manifest file.
"""
if not os.path.exists(os.path.join(target_dir, "LibriSpeech")):
# download
filepath = download(url, md5sum, target_dir)
# unpack
unpack(filepath, target_dir)
else:
print("Skip downloading and unpacking. Data already exists in %s." %
target_dir)
# create manifest json file
create_manifest(target_dir, manifest_path)
def main():
if args.target_dir.startswith('~'):
args.target_dir = os.path.expanduser(args.target_dir)
prepare_dataset(
url=URL_TEST_CLEAN,
md5sum=MD5_TEST_CLEAN,
target_dir=os.path.join(args.target_dir, "test-clean"),
manifest_path=args.manifest_prefix + ".test-clean")
prepare_dataset(
url=URL_DEV_CLEAN,
md5sum=MD5_DEV_CLEAN,
target_dir=os.path.join(args.target_dir, "dev-clean"),
manifest_path=args.manifest_prefix + ".dev-clean")
if args.full_download:
prepare_dataset(
url=URL_TRAIN_CLEAN_100,
md5sum=MD5_TRAIN_CLEAN_100,
target_dir=os.path.join(args.target_dir, "train-clean-100"),
manifest_path=args.manifest_prefix + ".train-clean-100")
prepare_dataset(
url=URL_TEST_OTHER,
md5sum=MD5_TEST_OTHER,
target_dir=os.path.join(args.target_dir, "test-other"),
manifest_path=args.manifest_prefix + ".test-other")
prepare_dataset(
url=URL_DEV_OTHER,
md5sum=MD5_DEV_OTHER,
target_dir=os.path.join(args.target_dir, "dev-other"),
manifest_path=args.manifest_prefix + ".dev-other")
prepare_dataset(
url=URL_TRAIN_CLEAN_360,
md5sum=MD5_TRAIN_CLEAN_360,
target_dir=os.path.join(args.target_dir, "train-clean-360"),
manifest_path=args.manifest_prefix + ".train-clean-360")
prepare_dataset(
url=URL_TRAIN_OTHER_500,
md5sum=MD5_TRAIN_OTHER_500,
target_dir=os.path.join(args.target_dir, "train-other-500"),
manifest_path=args.manifest_prefix + ".train-other-500")
if __name__ == '__main__':
main()
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"""Prepare CHiME3 background data.
Download, unpack and create manifest files.
Manifest file is a json-format file with each line containing the
meta data (i.e. audio filepath, transcript and audio duration)
of each audio file in the data set.
"""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import distutils.util
import os
import wget
import zipfile
import argparse
import soundfile
import json
from paddle.v2.dataset.common import md5file
DATA_HOME = os.path.expanduser('~/.cache/paddle/dataset/speech')
URL = "https://d4s.myairbridge.com/packagev2/AG0Y3DNBE5IWRRTV/?dlid=W19XG7T0NNHB027139H0EQ"
MD5 = "c3ff512618d7a67d4f85566ea1bc39ec"
parser = argparse.ArgumentParser(description=__doc__)
parser.add_argument(
"--target_dir",
default=DATA_HOME + "/chime3_background",
type=str,
help="Directory to save the dataset. (default: %(default)s)")
parser.add_argument(
"--manifest_filepath",
default="manifest.chime3.background",
type=str,
help="Filepath for output manifests. (default: %(default)s)")
args = parser.parse_args()
def download(url, md5sum, target_dir, filename=None):
"""Download file from url to target_dir, and check md5sum."""
if filename == None:
filename = url.split("/")[-1]
if not os.path.exists(target_dir): os.makedirs(target_dir)
filepath = os.path.join(target_dir, filename)
if not (os.path.exists(filepath) and md5file(filepath) == md5sum):
print("Downloading %s ..." % url)
wget.download(url, target_dir)
print("\nMD5 Chesksum %s ..." % filepath)
if not md5file(filepath) == md5sum:
raise RuntimeError("MD5 checksum failed.")
else:
print("File exists, skip downloading. (%s)" % filepath)
return filepath
def unpack(filepath, target_dir):
"""Unpack the file to the target_dir."""
print("Unpacking %s ..." % filepath)
if filepath.endswith('.zip'):
zip = zipfile.ZipFile(filepath, 'r')
zip.extractall(target_dir)
zip.close()
elif filepath.endswith('.tar') or filepath.endswith('.tar.gz'):
tar = zipfile.open(filepath)
tar.extractall(target_dir)
tar.close()
else:
raise ValueError("File format is not supported for unpacking.")
def create_manifest(data_dir, manifest_path):
"""Create a manifest json file summarizing the data set, with each line
containing the meta data (i.e. audio filepath, transcription text, audio
duration) of each audio file within the data set.
"""
print("Creating manifest %s ..." % manifest_path)
json_lines = []
for subfolder, _, filelist in sorted(os.walk(data_dir)):
for filename in filelist:
if filename.endswith('.wav'):
filepath = os.path.join(data_dir, subfolder, filename)
audio_data, samplerate = soundfile.read(filepath)
duration = float(len(audio_data)) / samplerate
json_lines.append(
json.dumps({
'audio_filepath': filepath,
'duration': duration,
'text': ''
}))
with open(manifest_path, 'w') as out_file:
for line in json_lines:
out_file.write(line + '\n')
def prepare_chime3(url, md5sum, target_dir, manifest_path):
"""Download, unpack and create summmary manifest file."""
if not os.path.exists(os.path.join(target_dir, "CHiME3")):
# download
filepath = download(url, md5sum, target_dir,
"myairbridge-AG0Y3DNBE5IWRRTV.zip")
# unpack
unpack(filepath, target_dir)
unpack(
os.path.join(target_dir, 'CHiME3_background_bus.zip'), target_dir)
unpack(
os.path.join(target_dir, 'CHiME3_background_caf.zip'), target_dir)
unpack(
os.path.join(target_dir, 'CHiME3_background_ped.zip'), target_dir)
unpack(
os.path.join(target_dir, 'CHiME3_background_str.zip'), target_dir)
else:
print("Skip downloading and unpacking. Data already exists in %s." %
target_dir)
# create manifest json file
create_manifest(target_dir, manifest_path)
def main():
prepare_chime3(
url=URL,
md5sum=MD5,
target_dir=args.target_dir,
manifest_path=args.manifest_filepath)
if __name__ == '__main__':
main()
deep_speech_2/data_utils/audio.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Contains the audio segment class."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import numpy as np
import io
import struct
import re
import soundfile
import resampy
from scipy import signal
import random
import copy
class AudioSegment(object):
"""Monaural audio segment abstraction.
:param samples: Audio samples [num_samples x num_channels].
:type samples: ndarray.float32
:param sample_rate: Audio sample rate.
:type sample_rate: int
:raises TypeError: If the sample data type is not float or int.
"""
def __init__(self, samples, sample_rate):
"""Create audio segment from samples.
Samples are convert float32 internally, with int scaled to [-1, 1].
"""
self._samples = self._convert_samples_to_float32(samples)
self._sample_rate = sample_rate
if self._samples.ndim >= 2:
self._samples = np.mean(self._samples, 1)
def __eq__(self, other):
"""Return whether two objects are equal."""
if type(other) is not type(self):
return False
if self._sample_rate != other._sample_rate:
return False
if self._samples.shape != other._samples.shape:
return False
if np.any(self.samples != other._samples):
return False
return True
def __ne__(self, other):
"""Return whether two objects are unequal."""
return not self.__eq__(other)
def __str__(self):
"""Return human-readable representation of segment."""
return ("%s: num_samples=%d, sample_rate=%d, duration=%.2fsec, "
"rms=%.2fdB" % (type(self), self.num_samples, self.sample_rate,
self.duration, self.rms_db))
@classmethod
def from_file(cls, file):
"""Create audio segment from audio file.
:param filepath: Filepath or file object to audio file.
:type filepath: basestring|file
:return: Audio segment instance.
:rtype: AudioSegment
"""
if isinstance(file, basestring) and re.findall(r".seqbin_\d+$", file):
return cls.from_sequence_file(file)
else:
samples, sample_rate = soundfile.read(file, dtype='float32')
return cls(samples, sample_rate)
@classmethod
def slice_from_file(cls, file, start=None, end=None):
"""Loads a small section of an audio without having to load
the entire file into the memory which can be incredibly wasteful.
:param file: Input audio filepath or file object.
:type file: basestring|file
:param start: Start time in seconds. If start is negative, it wraps
around from the end. If not provided, this function
reads from the very beginning.
:type start: float
:param end: End time in seconds. If end is negative, it wraps around
from the end. If not provided, the default behvaior is
to read to the end of the file.
:type end: float
:return: AudioSegment instance of the specified slice of the input
audio file.
:rtype: AudioSegment
:raise ValueError: If start or end is incorrectly set, e.g. out of
bounds in time.
"""
sndfile = soundfile.SoundFile(file)
sample_rate = sndfile.samplerate
duration = float(len(sndfile)) / sample_rate
start = 0. if start is None else start
end = 0. if end is None else end
if start < 0.0:
start += duration
if end < 0.0:
end += duration
if start < 0.0:
raise ValueError("The slice start position (%f s) is out of "
"bounds." % start)
if end < 0.0:
raise ValueError("The slice end position (%f s) is out of bounds." %
end)
if start > end:
raise ValueError("The slice start position (%f s) is later than "
"the slice end position (%f s)." % (start, end))
if end > duration:
raise ValueError("The slice end position (%f s) is out of bounds "
"(> %f s)" % (end, duration))
start_frame = int(start * sample_rate)
end_frame = int(end * sample_rate)
sndfile.seek(start_frame)
data = sndfile.read(frames=end_frame - start_frame, dtype='float32')
return cls(data, sample_rate)
@classmethod
def from_sequence_file(cls, filepath):
"""Create audio segment from sequence file. Sequence file is a binary
file containing a collection of multiple audio files, with several
header bytes in the head indicating the offsets of each audio byte data
chunk.
The format is:
4 bytes (int, version),
4 bytes (int, num of utterance),
4 bytes (int, bytes per header),
[bytes_per_header*(num_utterance+1)] bytes (offsets for each audio),
audio_bytes_data_of_1st_utterance,
audio_bytes_data_of_2nd_utterance,
......
Sequence file name must end with ".seqbin". And the filename of the 5th
utterance's audio file in sequence file "xxx.seqbin" must be
"xxx.seqbin_5", with "5" indicating the utterance index within this
sequence file (starting from 1).
:param filepath: Filepath of sequence file.
:type filepath: basestring
:return: Audio segment instance.
:rtype: AudioSegment
"""
# parse filepath
matches = re.match(r"(.+\.seqbin)_(\d+)", filepath)
if matches is None:
raise IOError("File type of %s is not supported" % filepath)
filename = matches.group(1)
fileno = int(matches.group(2))
# read headers
f = open(filename, 'rb')
version = f.read(4)
num_utterances = struct.unpack("i", f.read(4))[0]
bytes_per_header = struct.unpack("i", f.read(4))[0]
header_bytes = f.read(bytes_per_header * (num_utterances + 1))
header = [
struct.unpack("i", header_bytes[bytes_per_header * i:
bytes_per_header * (i + 1)])[0]
for i in range(num_utterances + 1)
]
# read audio bytes
f.seek(header[fileno - 1])
audio_bytes = f.read(header[fileno] - header[fileno - 1])
f.close()
# create audio segment
try:
return cls.from_bytes(audio_bytes)
except Exception as e:
samples = np.frombuffer(audio_bytes, dtype='int16')
return cls(samples=samples, sample_rate=8000)
@classmethod
def from_bytes(cls, bytes):
"""Create audio segment from a byte string containing audio samples.
:param bytes: Byte string containing audio samples.
:type bytes: str
:return: Audio segment instance.
:rtype: AudioSegment
"""
samples, sample_rate = soundfile.read(
io.BytesIO(bytes), dtype='float32')
return cls(samples, sample_rate)
@classmethod
def concatenate(cls, *segments):
"""Concatenate an arbitrary number of audio segments together.
:param *segments: Input audio segments to be concatenated.
:type *segments: tuple of AudioSegment
:return: Audio segment instance as concatenating results.
:rtype: AudioSegment
:raises ValueError: If the number of segments is zero, or if the
sample_rate of any segments does not match.
:raises TypeError: If any segment is not AudioSegment instance.
"""
# Perform basic sanity-checks.
if len(segments) == 0:
raise ValueError("No audio segments are given to concatenate.")
sample_rate = segments[0]._sample_rate
for seg in segments:
if sample_rate != seg._sample_rate:
raise ValueError("Can't concatenate segments with "
"different sample rates")
if type(seg) is not cls:
raise TypeError("Only audio segments of the same type "
"can be concatenated.")
samples = np.concatenate([seg.samples for seg in segments])
return cls(samples, sample_rate)
@classmethod
def make_silence(cls, duration, sample_rate):
"""Creates a silent audio segment of the given duration and sample rate.
:param duration: Length of silence in seconds.
:type duration: float
:param sample_rate: Sample rate.
:type sample_rate: float
:return: Silent AudioSegment instance of the given duration.
:rtype: AudioSegment
"""
samples = np.zeros(int(duration * sample_rate))
return cls(samples, sample_rate)
def to_wav_file(self, filepath, dtype='float32'):
"""Save audio segment to disk as wav file.
:param filepath: WAV filepath or file object to save the
audio segment.
:type filepath: basestring|file
:param dtype: Subtype for audio file. Options: 'int16', 'int32',
'float32', 'float64'. Default is 'float32'.
:type dtype: str
:raises TypeError: If dtype is not supported.
"""
samples = self._convert_samples_from_float32(self._samples, dtype)
subtype_map = {
'int16': 'PCM_16',
'int32': 'PCM_32',
'float32': 'FLOAT',
'float64': 'DOUBLE'
}
soundfile.write(
filepath,
samples,
self._sample_rate,
format='WAV',
subtype=subtype_map[dtype])
def superimpose(self, other):
"""Add samples from another segment to those of this segment
(sample-wise addition, not segment concatenation).
Note that this is an in-place transformation.
:param other: Segment containing samples to be added in.
:type other: AudioSegments
:raise TypeError: If type of two segments don't match.
:raise ValueError: If the sample rates of the two segments are not
equal, or if the lengths of segments don't match.
"""
if isinstance(other, type(self)):
raise TypeError("Cannot add segments of different types: %s "
"and %s." % (type(self), type(other)))
if self._sample_rate != other._sample_rate:
raise ValueError("Sample rates must match to add segments.")
if len(self._samples) != len(other._samples):
raise ValueError("Segment lengths must match to add segments.")
self._samples += other._samples
def to_bytes(self, dtype='float32'):
"""Create a byte string containing the audio content.
:param dtype: Data type for export samples. Options: 'int16', 'int32',
'float32', 'float64'. Default is 'float32'.
:type dtype: str
:return: Byte string containing audio content.
:rtype: str
"""
samples = self._convert_samples_from_float32(self._samples, dtype)
return samples.tostring()
def gain_db(self, gain):
"""Apply gain in decibels to samples.
Note that this is an in-place transformation.
:param gain: Gain in decibels to apply to samples.
:type gain: float|1darray
"""
self._samples *= 10.**(gain / 20.)
def change_speed(self, speed_rate):
"""Change the audio speed by linear interpolation.
Note that this is an in-place transformation.
:param speed_rate: Rate of speed change:
speed_rate > 1.0, speed up the audio;
speed_rate = 1.0, unchanged;
speed_rate < 1.0, slow down the audio;
speed_rate <= 0.0, not allowed, raise ValueError.
:type speed_rate: float
:raises ValueError: If speed_rate <= 0.0.
"""
if speed_rate <= 0:
raise ValueError("speed_rate should be greater than zero.")
old_length = self._samples.shape[0]
new_length = int(old_length / speed_rate)
old_indices = np.arange(old_length)
new_indices = np.linspace(start=0, stop=old_length, num=new_length)
self._samples = np.interp(new_indices, old_indices, self._samples)
def normalize(self, target_db=-20, max_gain_db=300.0):
"""Normalize audio to be of the desired RMS value in decibels.
Note that this is an in-place transformation.
:param target_db: Target RMS value in decibels. This value should be
less than 0.0 as 0.0 is full-scale audio.
:type target_db: float
:param max_gain_db: Max amount of gain in dB that can be applied for
normalization. This is to prevent nans when
attempting to normalize a signal consisting of
all zeros.
:type max_gain_db: float
:raises ValueError: If the required gain to normalize the segment to
the target_db value exceeds max_gain_db.
"""
gain = target_db - self.rms_db
if gain > max_gain_db:
raise ValueError(
"Unable to normalize segment to %f dB because the "
"the probable gain have exceeds max_gain_db (%f dB)" %
(target_db, max_gain_db))
self.gain_db(min(max_gain_db, target_db - self.rms_db))
def normalize_online_bayesian(self,
target_db,
prior_db,
prior_samples,
startup_delay=0.0):
"""Normalize audio using a production-compatible online/causal
algorithm. This uses an exponential likelihood and gamma prior to
make online estimates of the RMS even when there are very few samples.
Note that this is an in-place transformation.
:param target_db: Target RMS value in decibels.
:type target_bd: float
:param prior_db: Prior RMS estimate in decibels.
:type prior_db: float
:param prior_samples: Prior strength in number of samples.
:type prior_samples: float
:param startup_delay: Default 0.0s. If provided, this function will
accrue statistics for the first startup_delay
seconds before applying online normalization.
:type startup_delay: float
"""
# Estimate total RMS online.
startup_sample_idx = min(self.num_samples - 1,
int(self.sample_rate * startup_delay))
prior_mean_squared = 10.**(prior_db / 10.)
prior_sum_of_squares = prior_mean_squared * prior_samples
cumsum_of_squares = np.cumsum(self.samples**2)
sample_count = np.arange(self.num_samples) + 1
if startup_sample_idx > 0:
cumsum_of_squares[:startup_sample_idx] = \
cumsum_of_squares[startup_sample_idx]
sample_count[:startup_sample_idx] = \
sample_count[startup_sample_idx]
mean_squared_estimate = ((cumsum_of_squares + prior_sum_of_squares) /
(sample_count + prior_samples))
rms_estimate_db = 10 * np.log10(mean_squared_estimate)
# Compute required time-varying gain.
gain_db = target_db - rms_estimate_db
self.gain_db(gain_db)
def resample(self, target_sample_rate, filter='kaiser_best'):
"""Resample the audio to a target sample rate.
Note that this is an in-place transformation.
:param target_sample_rate: Target sample rate.
:type target_sample_rate: int
:param filter: The resampling filter to use one of {'kaiser_best',
'kaiser_fast'}.
:type filter: str
"""
self._samples = resampy.resample(
self.samples, self.sample_rate, target_sample_rate, filter=filter)
self._sample_rate = target_sample_rate
def pad_silence(self, duration, sides='both'):
"""Pad this audio sample with a period of silence.
Note that this is an in-place transformation.
:param duration: Length of silence in seconds to pad.
:type duration: float
:param sides: Position for padding:
'beginning' - adds silence in the beginning;
'end' - adds silence in the end;
'both' - adds silence in both the beginning and the end.
:type sides: str
:raises ValueError: If sides is not supported.
"""
if duration == 0.0:
return self
cls = type(self)
silence = self.make_silence(duration, self._sample_rate)
if sides == "beginning":
padded = cls.concatenate(silence, self)
elif sides == "end":
padded = cls.concatenate(self, silence)
elif sides == "both":
padded = cls.concatenate(silence, self, silence)
else:
raise ValueError("Unknown value for the sides %s" % sides)
self._samples = padded._samples
def shift(self, shift_ms):
"""Shift the audio in time. If `shift_ms` is positive, shift with time
advance; if negative, shift with time delay. Silence are padded to
keep the duration unchanged.
Note that this is an in-place transformation.
:param shift_ms: Shift time in millseconds. If positive, shift with
time advance; if negative; shift with time delay.
:type shift_ms: float
:raises ValueError: If shift_ms is longer than audio duration.
"""
if abs(shift_ms) / 1000.0 > self.duration:
raise ValueError("Absolute value of shift_ms should be smaller "
"than audio duration.")
shift_samples = int(shift_ms * self._sample_rate / 1000)
if shift_samples > 0:
# time advance
self._samples[:-shift_samples] = self._samples[shift_samples:]
self._samples[-shift_samples:] = 0
elif shift_samples < 0:
# time delay
self._samples[-shift_samples:] = self._samples[:shift_samples]
self._samples[:-shift_samples] = 0
def subsegment(self, start_sec=None, end_sec=None):
"""Cut the AudioSegment between given boundaries.
Note that this is an in-place transformation.
:param start_sec: Beginning of subsegment in seconds.
:type start_sec: float
:param end_sec: End of subsegment in seconds.
:type end_sec: float
:raise ValueError: If start_sec or end_sec is incorrectly set, e.g. out
of bounds in time.
"""
start_sec = 0.0 if start_sec is None else start_sec
end_sec = self.duration if end_sec is None else end_sec
if start_sec < 0.0:
start_sec = self.duration + start_sec
if end_sec < 0.0:
end_sec = self.duration + end_sec
if start_sec < 0.0:
raise ValueError("The slice start position (%f s) is out of "
"bounds." % start_sec)
if end_sec < 0.0:
raise ValueError("The slice end position (%f s) is out of bounds." %
end_sec)
if start_sec > end_sec:
raise ValueError("The slice start position (%f s) is later than "
"the end position (%f s)." % (start_sec, end_sec))
if end_sec > self.duration:
raise ValueError("The slice end position (%f s) is out of bounds "
"(> %f s)" % (end_sec, self.duration))
start_sample = int(round(start_sec * self._sample_rate))
end_sample = int(round(end_sec * self._sample_rate))
self._samples = self._samples[start_sample:end_sample]
def random_subsegment(self, subsegment_length, rng=None):
"""Cut the specified length of the audiosegment randomly.
Note that this is an in-place transformation.
:param subsegment_length: Subsegment length in seconds.
:type subsegment_length: float
:param rng: Random number generator state.
:type rng: random.Random
:raises ValueError: If the length of subsegment is greater than
the origineal segemnt.
"""
rng = random.Random() if rng is None else rng
if subsegment_length > self.duration:
raise ValueError("Length of subsegment must not be greater "
"than original segment.")
start_time = rng.uniform(0.0, self.duration - subsegment_length)
self.subsegment(start_time, start_time + subsegment_length)
def convolve(self, impulse_segment, allow_resample=False):
"""Convolve this audio segment with the given impulse segment.
Note that this is an in-place transformation.
:param impulse_segment: Impulse response segments.
:type impulse_segment: AudioSegment
:param allow_resample: Indicates whether resampling is allowed when
the impulse_segment has a different sample
rate from this signal.
:type allow_resample: bool
:raises ValueError: If the sample rate is not match between two
audio segments when resample is not allowed.
"""
if allow_resample and self.sample_rate != impulse_segment.sample_rate:
impulse_segment.resample(self.sample_rate)
if self.sample_rate != impulse_segment.sample_rate:
raise ValueError("Impulse segment's sample rate (%d Hz) is not "
"equal to base signal sample rate (%d Hz)." %
(impulse_segment.sample_rate, self.sample_rate))
samples = signal.fftconvolve(self.samples, impulse_segment.samples,
"full")
self._samples = samples
def convolve_and_normalize(self, impulse_segment, allow_resample=False):
"""Convolve and normalize the resulting audio segment so that it
has the same average power as the input signal.
Note that this is an in-place transformation.
:param impulse_segment: Impulse response segments.
:type impulse_segment: AudioSegment
:param allow_resample: Indicates whether resampling is allowed when
the impulse_segment has a different sample
rate from this signal.
:type allow_resample: bool
"""
target_db = self.rms_db
self.convolve(impulse_segment, allow_resample=allow_resample)
self.normalize(target_db)
def add_noise(self,
noise,
snr_dB,
allow_downsampling=False,
max_gain_db=300.0,
rng=None):
"""Add the given noise segment at a specific signal-to-noise ratio.
If the noise segment is longer than this segment, a random subsegment
of matching length is sampled from it and used instead.
Note that this is an in-place transformation.
:param noise: Noise signal to add.
:type noise: AudioSegment
:param snr_dB: Signal-to-Noise Ratio, in decibels.
:type snr_dB: float
:param allow_downsampling: Whether to allow the noise signal to be
downsampled to match the base signal sample
rate.
:type allow_downsampling: bool
:param max_gain_db: Maximum amount of gain to apply to noise signal
before adding it in. This is to prevent attempting
to apply infinite gain to a zero signal.
:type max_gain_db: float
:param rng: Random number generator state.
:type rng: None|random.Random
:raises ValueError: If the sample rate does not match between the two
audio segments when downsampling is not allowed, or
if the duration of noise segments is shorter than
original audio segments.
"""
rng = random.Random() if rng is None else rng
if allow_downsampling and noise.sample_rate > self.sample_rate:
noise = noise.resample(self.sample_rate)
if noise.sample_rate != self.sample_rate:
raise ValueError("Noise sample rate (%d Hz) is not equal to base "
"signal sample rate (%d Hz)." % (noise.sample_rate,
self.sample_rate))
if noise.duration < self.duration:
raise ValueError("Noise signal (%f sec) must be at least as long as"
" base signal (%f sec)." %
(noise.duration, self.duration))
noise_gain_db = min(self.rms_db - noise.rms_db - snr_dB, max_gain_db)
noise_new = copy.deepcopy(noise)
noise_new.random_subsegment(self.duration, rng=rng)
noise_new.gain_db(noise_gain_db)
self.superimpose(noise_new)
@property
def samples(self):
"""Return audio samples.
:return: Audio samples.
:rtype: ndarray
"""
return self._samples.copy()
@property
def sample_rate(self):
"""Return audio sample rate.
:return: Audio sample rate.
:rtype: int
"""
return self._sample_rate
@property
def num_samples(self):
"""Return number of samples.
:return: Number of samples.
:rtype: int
"""
return self._samples.shape[0]
@property
def duration(self):
"""Return audio duration.
:return: Audio duration in seconds.
:rtype: float
"""
return self._samples.shape[0] / float(self._sample_rate)
@property
def rms_db(self):
"""Return root mean square energy of the audio in decibels.
:return: Root mean square energy in decibels.
:rtype: float
"""
# square root => multiply by 10 instead of 20 for dBs
mean_square = np.mean(self._samples**2)
return 10 * np.log10(mean_square)
def _convert_samples_to_float32(self, samples):
"""Convert sample type to float32.
Audio sample type is usually integer or float-point.
Integers will be scaled to [-1, 1] in float32.
"""
float32_samples = samples.astype('float32')
if samples.dtype in np.sctypes['int']:
bits = np.iinfo(samples.dtype).bits
float32_samples *= (1. / 2**(bits - 1))
elif samples.dtype in np.sctypes['float']:
pass
else:
raise TypeError("Unsupported sample type: %s." % samples.dtype)
return float32_samples
def _convert_samples_from_float32(self, samples, dtype):
"""Convert sample type from float32 to dtype.
Audio sample type is usually integer or float-point. For integer
type, float32 will be rescaled from [-1, 1] to the maximum range
supported by the integer type.
This is for writing a audio file.
"""
dtype = np.dtype(dtype)
output_samples = samples.copy()
if dtype in np.sctypes['int']:
bits = np.iinfo(dtype).bits
output_samples *= (2**(bits - 1) / 1.)
min_val = np.iinfo(dtype).min
max_val = np.iinfo(dtype).max
output_samples[output_samples > max_val] = max_val
output_samples[output_samples < min_val] = min_val
elif samples.dtype in np.sctypes['float']:
min_val = np.finfo(dtype).min
max_val = np.finfo(dtype).max
output_samples[output_samples > max_val] = max_val
output_samples[output_samples < min_val] = min_val
else:
raise TypeError("Unsupported sample type: %s." % samples.dtype)
return output_samples.astype(dtype)
deep_speech_2/data_utils/augmentor/augmentation.py 已删除 100644 → 0
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"""Contains the data augmentation pipeline."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import json
import random
from data_utils.augmentor.volume_perturb import VolumePerturbAugmentor
from data_utils.augmentor.shift_perturb import ShiftPerturbAugmentor
from data_utils.augmentor.speed_perturb import SpeedPerturbAugmentor
from data_utils.augmentor.noise_perturb import NoisePerturbAugmentor
from data_utils.augmentor.impulse_response import ImpulseResponseAugmentor
from data_utils.augmentor.resample import ResampleAugmentor
from data_utils.augmentor.online_bayesian_normalization import \
OnlineBayesianNormalizationAugmentor
class AugmentationPipeline(object):
"""Build a pre-processing pipeline with various augmentation models.Such a
data augmentation pipeline is oftern leveraged to augment the training
samples to make the model invariant to certain types of perturbations in the
real world, improving model's generalization ability.
The pipeline is built according the the augmentation configuration in json
string, e.g.
.. code-block::
[ {
"type": "noise",
"params": {"min_snr_dB": 10,
"max_snr_dB": 20,
"noise_manifest_path": "datasets/manifest.noise"},
"prob": 0.0
},
{
"type": "speed",
"params": {"min_speed_rate": 0.9,
"max_speed_rate": 1.1},
"prob": 1.0
},
{
"type": "shift",
"params": {"min_shift_ms": -5,
"max_shift_ms": 5},
"prob": 1.0
},
{
"type": "volume",
"params": {"min_gain_dBFS": -10,
"max_gain_dBFS": 10},
"prob": 0.0
},
{
"type": "bayesian_normal",
"params": {"target_db": -20,
"prior_db": -20,
"prior_samples": 100},
"prob": 0.0
}
]
This augmentation configuration inserts two augmentation models
into the pipeline, with one is VolumePerturbAugmentor and the other
SpeedPerturbAugmentor. "prob" indicates the probability of the current
augmentor to take effect. If "prob" is zero, the augmentor does not take
effect.
:param augmentation_config: Augmentation configuration in json string.
:type augmentation_config: str
:param random_seed: Random seed.
:type random_seed: int
:raises ValueError: If the augmentation json config is in incorrect format".
"""
def __init__(self, augmentation_config, random_seed=0):
self._rng = random.Random(random_seed)
self._augmentors, self._rates = self._parse_pipeline_from(
augmentation_config)
def transform_audio(self, audio_segment):
"""Run the pre-processing pipeline for data augmentation.
Note that this is an in-place transformation.
:param audio_segment: Audio segment to process.
:type audio_segment: AudioSegmenet|SpeechSegment
"""
for augmentor, rate in zip(self._augmentors, self._rates):
if self._rng.uniform(0., 1.) < rate:
augmentor.transform_audio(audio_segment)
def _parse_pipeline_from(self, config_json):
"""Parse the config json to build a augmentation pipelien."""
try:
configs = json.loads(config_json)
augmentors = [
self._get_augmentor(config["type"], config["params"])
for config in configs
]
rates = [config["prob"] for config in configs]
except Exception as e:
raise ValueError("Failed to parse the augmentation config json: "
"%s" % str(e))
return augmentors, rates
def _get_augmentor(self, augmentor_type, params):
"""Return an augmentation model by the type name, and pass in params."""
if augmentor_type == "volume":
return VolumePerturbAugmentor(self._rng, **params)
elif augmentor_type == "shift":
return ShiftPerturbAugmentor(self._rng, **params)
elif augmentor_type == "speed":
return SpeedPerturbAugmentor(self._rng, **params)
elif augmentor_type == "resample":
return ResampleAugmentor(self._rng, **params)
elif augmentor_type == "bayesian_normal":
return OnlineBayesianNormalizationAugmentor(self._rng, **params)
elif augmentor_type == "noise":
return NoisePerturbAugmentor(self._rng, **params)
elif augmentor_type == "impulse":
return ImpulseResponseAugmentor(self._rng, **params)
else:
raise ValueError("Unknown augmentor type [%s]." % augmentor_type)
deep_speech_2/data_utils/augmentor/base.py 已删除 100644 → 0
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"""Contains the abstract base class for augmentation models."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from abc import ABCMeta, abstractmethod
class AugmentorBase(object):
"""Abstract base class for augmentation model (augmentor) class.
All augmentor classes should inherit from this class, and implement the
following abstract methods.
"""
__metaclass__ = ABCMeta
@abstractmethod
def __init__(self):
pass
@abstractmethod
def transform_audio(self, audio_segment):
"""Adds various effects to the input audio segment. Such effects
will augment the training data to make the model invariant to certain
types of perturbations in the real world, improving model's
generalization ability.
Note that this is an in-place transformation.
:param audio_segment: Audio segment to add effects to.
:type audio_segment: AudioSegmenet|SpeechSegment
"""
pass
deep_speech_2/data_utils/augmentor/impulse_response.py 已删除 100644 → 0
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"""Contains the impulse response augmentation model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.augmentor.base import AugmentorBase
from data_utils.utility import read_manifest
from data_utils.audio import AudioSegment
class ImpulseResponseAugmentor(AugmentorBase):
"""Augmentation model for adding impulse response effect.
:param rng: Random generator object.
:type rng: random.Random
:param impulse_manifest_path: Manifest path for impulse audio data.
:type impulse_manifest_path: basestring
"""
def __init__(self, rng, impulse_manifest_path):
self._rng = rng
self._impulse_manifest = read_manifest(impulse_manifest_path)
def transform_audio(self, audio_segment):
"""Add impulse response effect.
Note that this is an in-place transformation.
:param audio_segment: Audio segment to add effects to.
:type audio_segment: AudioSegmenet|SpeechSegment
"""
impulse_json = self._rng.sample(self._impulse_manifest, 1)[0]
impulse_segment = AudioSegment.from_file(impulse_json['audio_filepath'])
audio_segment.convolve(impulse_segment, allow_resample=True)
deep_speech_2/data_utils/augmentor/noise_perturb.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Contains the noise perturb augmentation model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.augmentor.base import AugmentorBase
from data_utils.utility import read_manifest
from data_utils.audio import AudioSegment
class NoisePerturbAugmentor(AugmentorBase):
"""Augmentation model for adding background noise.
:param rng: Random generator object.
:type rng: random.Random
:param min_snr_dB: Minimal signal noise ratio, in decibels.
:type min_snr_dB: float
:param max_snr_dB: Maximal signal noise ratio, in decibels.
:type max_snr_dB: float
:param noise_manifest_path: Manifest path for noise audio data.
:type noise_manifest_path: basestring
"""
def __init__(self, rng, min_snr_dB, max_snr_dB, noise_manifest_path):
self._min_snr_dB = min_snr_dB
self._max_snr_dB = max_snr_dB
self._rng = rng
self._noise_manifest = read_manifest(manifest_path=noise_manifest_path)
def transform_audio(self, audio_segment):
"""Add background noise audio.
Note that this is an in-place transformation.
:param audio_segment: Audio segment to add effects to.
:type audio_segment: AudioSegmenet|SpeechSegment
"""
noise_json = self._rng.sample(self._noise_manifest, 1)[0]
if noise_json['duration'] < audio_segment.duration:
raise RuntimeError("The duration of sampled noise audio is smaller "
"than the audio segment to add effects to.")
diff_duration = noise_json['duration'] - audio_segment.duration
start = self._rng.uniform(0, diff_duration)
end = start + audio_segment.duration
noise_segment = AudioSegment.slice_from_file(
noise_json['audio_filepath'], start=start, end=end)
snr_dB = self._rng.uniform(self._min_snr_dB, self._max_snr_dB)
audio_segment.add_noise(
noise_segment, snr_dB, allow_downsampling=True, rng=self._rng)
deep_speech_2/data_utils/augmentor/online_bayesian_normalization.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Contain the online bayesian normalization augmentation model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.augmentor.base import AugmentorBase
class OnlineBayesianNormalizationAugmentor(AugmentorBase):
"""Augmentation model for adding online bayesian normalization.
:param rng: Random generator object.
:type rng: random.Random
:param target_db: Target RMS value in decibels.
:type target_db: float
:param prior_db: Prior RMS estimate in decibels.
:type prior_db: float
:param prior_samples: Prior strength in number of samples.
:type prior_samples: int
:param startup_delay: Default 0.0s. If provided, this function will
accrue statistics for the first startup_delay
seconds before applying online normalization.
:type starup_delay: float.
"""
def __init__(self,
rng,
target_db,
prior_db,
prior_samples,
startup_delay=0.0):
self._target_db = target_db
self._prior_db = prior_db
self._prior_samples = prior_samples
self._rng = rng
self._startup_delay = startup_delay
def transform_audio(self, audio_segment):
"""Normalizes the input audio using the online Bayesian approach.
Note that this is an in-place transformation.
:param audio_segment: Audio segment to add effects to.
:type audio_segment: AudioSegment|SpeechSegment
"""
audio_segment.normalize_online_bayesian(self._target_db, self._prior_db,
self._prior_samples,
self._startup_delay)
deep_speech_2/data_utils/augmentor/resample.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Contain the resample augmentation model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.augmentor.base import AugmentorBase
class ResampleAugmentor(AugmentorBase):
"""Augmentation model for resampling.
See more info here:
https://ccrma.stanford.edu/~jos/resample/index.html
:param rng: Random generator object.
:type rng: random.Random
:param new_sample_rate: New sample rate in Hz.
:type new_sample_rate: int
"""
def __init__(self, rng, new_sample_rate):
self._new_sample_rate = new_sample_rate
self._rng = rng
def transform_audio(self, audio_segment):
"""Resamples the input audio to a target sample rate.
Note that this is an in-place transformation.
:param audio: Audio segment to add effects to.
:type audio: AudioSegment|SpeechSegment
"""
audio_segment.resample(self._new_sample_rate)
deep_speech_2/data_utils/augmentor/shift_perturb.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Contains the volume perturb augmentation model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.augmentor.base import AugmentorBase
class ShiftPerturbAugmentor(AugmentorBase):
"""Augmentation model for adding random shift perturbation.
:param rng: Random generator object.
:type rng: random.Random
:param min_shift_ms: Minimal shift in milliseconds.
:type min_shift_ms: float
:param max_shift_ms: Maximal shift in milliseconds.
:type max_shift_ms: float
"""
def __init__(self, rng, min_shift_ms, max_shift_ms):
self._min_shift_ms = min_shift_ms
self._max_shift_ms = max_shift_ms
self._rng = rng
def transform_audio(self, audio_segment):
"""Shift audio.
Note that this is an in-place transformation.
:param audio_segment: Audio segment to add effects to.
:type audio_segment: AudioSegmenet|SpeechSegment
"""
shift_ms = self._rng.uniform(self._min_shift_ms, self._max_shift_ms)
audio_segment.shift(shift_ms)
deep_speech_2/data_utils/augmentor/speed_perturb.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Contain the speech perturbation augmentation model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.augmentor.base import AugmentorBase
class SpeedPerturbAugmentor(AugmentorBase):
"""Augmentation model for adding speed perturbation.
See reference paper here:
http://www.danielpovey.com/files/2015_interspeech_augmentation.pdf
:param rng: Random generator object.
:type rng: random.Random
:param min_speed_rate: Lower bound of new speed rate to sample and should
not be smaller than 0.9.
:type min_speed_rate: float
:param max_speed_rate: Upper bound of new speed rate to sample and should
not be larger than 1.1.
:type max_speed_rate: float
"""
def __init__(self, rng, min_speed_rate, max_speed_rate):
if min_speed_rate < 0.9:
raise ValueError(
"Sampling speed below 0.9 can cause unnatural effects")
if max_speed_rate > 1.1:
raise ValueError(
"Sampling speed above 1.1 can cause unnatural effects")
self._min_speed_rate = min_speed_rate
self._max_speed_rate = max_speed_rate
self._rng = rng
def transform_audio(self, audio_segment):
"""Sample a new speed rate from the given range and
changes the speed of the given audio clip.
Note that this is an in-place transformation.
:param audio_segment: Audio segment to add effects to.
:type audio_segment: AudioSegment|SpeechSegment
"""
sampled_speed = self._rng.uniform(self._min_speed_rate,
self._max_speed_rate)
audio_segment.change_speed(sampled_speed)
deep_speech_2/data_utils/augmentor/volume_perturb.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Contains the volume perturb augmentation model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.augmentor.base import AugmentorBase
class VolumePerturbAugmentor(AugmentorBase):
"""Augmentation model for adding random volume perturbation.
This is used for multi-loudness training of PCEN. See
https://arxiv.org/pdf/1607.05666v1.pdf
for more details.
:param rng: Random generator object.
:type rng: random.Random
:param min_gain_dBFS: Minimal gain in dBFS.
:type min_gain_dBFS: float
:param max_gain_dBFS: Maximal gain in dBFS.
:type max_gain_dBFS: float
"""
def __init__(self, rng, min_gain_dBFS, max_gain_dBFS):
self._min_gain_dBFS = min_gain_dBFS
self._max_gain_dBFS = max_gain_dBFS
self._rng = rng
def transform_audio(self, audio_segment):
"""Change audio loadness.
Note that this is an in-place transformation.
:param audio_segment: Audio segment to add effects to.
:type audio_segment: AudioSegmenet|SpeechSegment
"""
gain = self._rng.uniform(self._min_gain_dBFS, self._max_gain_dBFS)
audio_segment.gain_db(gain)
deep_speech_2/data_utils/data.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Contains data generator for orgnaizing various audio data preprocessing
pipeline and offering data reader interface of PaddlePaddle requirements.
"""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import random
import tarfile
import multiprocessing
import numpy as np
import paddle.v2 as paddle
from threading import local
import atexit
from data_utils.utility import read_manifest
from data_utils.utility import xmap_readers_mp
from data_utils.augmentor.augmentation import AugmentationPipeline
from data_utils.featurizer.speech_featurizer import SpeechFeaturizer
from data_utils.speech import SpeechSegment
from data_utils.normalizer import FeatureNormalizer
class DataGenerator(object):
"""
DataGenerator provides basic audio data preprocessing pipeline, and offers
data reader interfaces of PaddlePaddle requirements.
:param vocab_filepath: Vocabulary filepath for indexing tokenized
transcripts.
:type vocab_filepath: basestring
:param mean_std_filepath: File containing the pre-computed mean and stddev.
:type mean_std_filepath: None|basestring
:param augmentation_config: Augmentation configuration in json string.
Details see AugmentationPipeline.__doc__.
:type augmentation_config: str
:param max_duration: Audio with duration (in seconds) greater than
this will be discarded.
:type max_duration: float
:param min_duration: Audio with duration (in seconds) smaller than
this will be discarded.
:type min_duration: float
:param stride_ms: Striding size (in milliseconds) for generating frames.
:type stride_ms: float
:param window_ms: Window size (in milliseconds) for generating frames.
:type window_ms: float
:param max_freq: Used when specgram_type is 'linear', only FFT bins
corresponding to frequencies between [0, max_freq] are
returned.
:types max_freq: None|float
:param specgram_type: Specgram feature type. Options: 'linear'.
:type specgram_type: str
:param use_dB_normalization: Whether to normalize the audio to -20 dB
before extracting the features.
:type use_dB_normalization: bool
:param num_threads: Number of CPU threads for processing data.
:type num_threads: int
:param random_seed: Random seed.
:type random_seed: int
:param keep_transcription_text: If set to True, transcription text will
be passed forward directly without
converting to index sequence.
:type keep_transcription_text: bool
:param num_conv_layers: The number of convolution layer, used to compute
the sequence length.
:type num_conv_layers: int
"""
def __init__(self,
vocab_filepath,
mean_std_filepath,
augmentation_config='{}',
max_duration=float('inf'),
min_duration=0.0,
stride_ms=10.0,
window_ms=20.0,
max_freq=None,
specgram_type='linear',
use_dB_normalization=True,
num_threads=multiprocessing.cpu_count() // 2,
random_seed=0,
keep_transcription_text=False,
num_conv_layers=2):
self._max_duration = max_duration
self._min_duration = min_duration
self._normalizer = FeatureNormalizer(mean_std_filepath)
self._augmentation_pipeline = AugmentationPipeline(
augmentation_config=augmentation_config, random_seed=random_seed)
self._speech_featurizer = SpeechFeaturizer(
vocab_filepath=vocab_filepath,
specgram_type=specgram_type,
stride_ms=stride_ms,
window_ms=window_ms,
max_freq=max_freq,
use_dB_normalization=use_dB_normalization)
self._num_threads = num_threads
self._rng = random.Random(random_seed)
self._keep_transcription_text = keep_transcription_text
self._epoch = 0
# for caching tar files info
self._local_data = local()
self._local_data.tar2info = {}
self._local_data.tar2object = {}
self._num_conv_layers = num_conv_layers
def process_utterance(self, filename, transcript):
"""Load, augment, featurize and normalize for speech data.
:param filename: Audio filepath
:type filename: basestring | file
:param transcript: Transcription text.
:type transcript: basestring
:return: Tuple of audio feature tensor and data of transcription part,
where transcription part could be token ids or text.
:rtype: tuple of (2darray, list)
"""
if filename.startswith('tar:'):
speech_segment = SpeechSegment.from_file(
self._subfile_from_tar(filename), transcript)
else:
speech_segment = SpeechSegment.from_file(filename, transcript)
self._augmentation_pipeline.transform_audio(speech_segment)
specgram, transcript_part = self._speech_featurizer.featurize(
speech_segment, self._keep_transcription_text)
specgram = self._normalizer.apply(specgram)
return specgram, transcript_part
def batch_reader_creator(self,
manifest_path,
batch_size,
min_batch_size=1,
padding_to=-1,
flatten=False,
sortagrad=False,
shuffle_method="batch_shuffle"):
"""
Batch data reader creator for audio data. Return a callable generator
function to produce batches of data.
Audio features within one batch will be padded with zeros to have the
same shape, or a user-defined shape.
:param manifest_path: Filepath of manifest for audio files.
:type manifest_path: basestring
:param batch_size: Number of instances in a batch.
:type batch_size: int
:param min_batch_size: Any batch with batch size smaller than this will
be discarded. (To be deprecated in the future.)
:type min_batch_size: int
:param padding_to: If set -1, the maximun shape in the batch
will be used as the target shape for padding.
Otherwise, `padding_to` will be the target shape.
:type padding_to: int
:param flatten: If set True, audio features will be flatten to 1darray.
:type flatten: bool
:param sortagrad: If set True, sort the instances by audio duration
in the first epoch for speed up training.
:type sortagrad: bool
:param shuffle_method: Shuffle method. Options:
'' or None: no shuffle.
'instance_shuffle': instance-wise shuffle.
'batch_shuffle': similarly-sized instances are
put into batches, and then
batch-wise shuffle the batches.
For more details, please see
``_batch_shuffle.__doc__``.
'batch_shuffle_clipped': 'batch_shuffle' with
head shift and tail
clipping. For more
details, please see
``_batch_shuffle``.
If sortagrad is True, shuffle is disabled
for the first epoch.
:type shuffle_method: None|str
:return: Batch reader function, producing batches of data when called.
:rtype: callable
"""
def batch_reader():
# read manifest
manifest = read_manifest(
manifest_path=manifest_path,
max_duration=self._max_duration,
min_duration=self._min_duration)
# sort (by duration) or batch-wise shuffle the manifest
if self._epoch == 0 and sortagrad:
manifest.sort(key=lambda x: x["duration"])
else:
if shuffle_method == "batch_shuffle":
manifest = self._batch_shuffle(
manifest, batch_size, clipped=False)
elif shuffle_method == "batch_shuffle_clipped":
manifest = self._batch_shuffle(
manifest, batch_size, clipped=True)
elif shuffle_method == "instance_shuffle":
self._rng.shuffle(manifest)
elif shuffle_method == None:
pass
else:
raise ValueError("Unknown shuffle method %s." %
shuffle_method)
# prepare batches
instance_reader = self._instance_reader_creator(manifest)
batch = []
for instance in instance_reader():
batch.append(instance)
if len(batch) == batch_size:
yield self._padding_batch(batch, padding_to, flatten)
batch = []
if len(batch) >= min_batch_size:
yield self._padding_batch(batch, padding_to, flatten)
self._epoch += 1
return batch_reader
@property
def feeding(self):
"""Returns data reader's feeding dict.
:return: Data feeding dict.
:rtype: dict
"""
feeding_dict = {
"audio_spectrogram": 0,
"transcript_text": 1,
"sequence_offset": 2,
"sequence_length": 3
}
for i in xrange(self._num_conv_layers):
feeding_dict["conv%d_index_range" % i] = len(feeding_dict)
return feeding_dict
@property
def vocab_size(self):
"""Return the vocabulary size.
:return: Vocabulary size.
:rtype: int
"""
return self._speech_featurizer.vocab_size
@property
def vocab_list(self):
"""Return the vocabulary in list.
:return: Vocabulary in list.
:rtype: list
"""
return self._speech_featurizer.vocab_list
def _parse_tar(self, file):
"""Parse a tar file to get a tarfile object
and a map containing tarinfoes
"""
result = {}
f = tarfile.open(file)
for tarinfo in f.getmembers():
result[tarinfo.name] = tarinfo
return f, result
def _subfile_from_tar(self, file):
"""Get subfile object from tar.
It will return a subfile object from tar file
and cached tar file info for next reading request.
"""
tarpath, filename = file.split(':', 1)[1].split('#', 1)
if 'tar2info' not in self._local_data.__dict__:
self._local_data.tar2info = {}
if 'tar2object' not in self._local_data.__dict__:
self._local_data.tar2object = {}
if tarpath not in self._local_data.tar2info:
object, infoes = self._parse_tar(tarpath)
self._local_data.tar2info[tarpath] = infoes
self._local_data.tar2object[tarpath] = object
return self._local_data.tar2object[tarpath].extractfile(
self._local_data.tar2info[tarpath][filename])
def _instance_reader_creator(self, manifest):
"""
Instance reader creator. Create a callable function to produce
instances of data.
Instance: a tuple of ndarray of audio spectrogram and a list of
token indices for transcript.
"""
def reader():
for instance in manifest:
yield instance
reader, cleanup_callback = xmap_readers_mp(
lambda instance: self.process_utterance(instance["audio_filepath"], instance["text"]),
reader,
self._num_threads,
4096,
order=True)
# register callback to main process
atexit.register(cleanup_callback)
return reader
def _padding_batch(self, batch, padding_to=-1, flatten=False):
"""
Padding audio features with zeros to make them have the same shape (or
a user-defined shape) within one bach.
If ``padding_to`` is -1, the maximun shape in the batch will be used
as the target shape for padding. Otherwise, `padding_to` will be the
target shape (only refers to the second axis).
If `flatten` is True, features will be flatten to 1darray.
"""
new_batch = []
# get target shape
max_length = max([audio.shape[1] for audio, text in batch])
if padding_to != -1:
if padding_to < max_length:
raise ValueError("If padding_to is not -1, it should be larger "
"than any instance's shape in the batch")
max_length = padding_to
# padding
for audio, text in batch:
padded_audio = np.zeros([audio.shape[0], max_length])
padded_audio[:, :audio.shape[1]] = audio
if flatten:
padded_audio = padded_audio.flatten()
# Stride size for conv0 is (3, 2)
# Stride size for conv1 to convN is (1, 2)
# Same as the network, hard-coded here
padded_instance = [padded_audio, text]
padded_conv0_h = (padded_audio.shape[0] - 1) // 2 + 1
padded_conv0_w = (padded_audio.shape[1] - 1) // 3 + 1
valid_w = (audio.shape[1] - 1) // 3 + 1
padded_instance += [
[0], # sequence offset, always 0
[valid_w], # valid sequence length
# Index ranges for channel, height and width
# Please refer scale_sub_region layer to see details
[1, 32, 1, padded_conv0_h, valid_w + 1, padded_conv0_w]
]
pre_padded_h = padded_conv0_h
for i in xrange(self._num_conv_layers - 1):
padded_h = (pre_padded_h - 1) // 2 + 1
pre_padded_h = padded_h
padded_instance += [
[1, 32, 1, padded_h, valid_w + 1, padded_conv0_w]
]
new_batch.append(padded_instance)
return new_batch
def _batch_shuffle(self, manifest, batch_size, clipped=False):
"""Put similarly-sized instances into minibatches for better efficiency
and make a batch-wise shuffle.
1. Sort the audio clips by duration.
2. Generate a random number `k`, k in [0, batch_size).
3. Randomly shift `k` instances in order to create different batches
for different epochs. Create minibatches.
4. Shuffle the minibatches.
:param manifest: Manifest contents. List of dict.
:type manifest: list
:param batch_size: Batch size. This size is also used for generate
a random number for batch shuffle.
:type batch_size: int
:param clipped: Whether to clip the heading (small shift) and trailing
(incomplete batch) instances.
:type clipped: bool
:return: Batch shuffled mainifest.
:rtype: list
"""
manifest.sort(key=lambda x: x["duration"])
shift_len = self._rng.randint(0, batch_size - 1)
batch_manifest = zip(*[iter(manifest[shift_len:])] * batch_size)
self._rng.shuffle(batch_manifest)
batch_manifest = [item for batch in batch_manifest for item in batch]
if not clipped:
res_len = len(manifest) - shift_len - len(batch_manifest)
batch_manifest.extend(manifest[-res_len:])
batch_manifest.extend(manifest[0:shift_len])
return batch_manifest
deep_speech_2/data_utils/featurizer/audio_featurizer.py 已删除 100644 → 0
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"""Contains the audio featurizer class."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import numpy as np
from data_utils.utility import read_manifest
from data_utils.audio import AudioSegment
from python_speech_features import mfcc
from python_speech_features import delta
class AudioFeaturizer(object):
"""Audio featurizer, for extracting features from audio contents of
AudioSegment or SpeechSegment.
Currently, it supports feature types of linear spectrogram and mfcc.
:param specgram_type: Specgram feature type. Options: 'linear'.
:type specgram_type: str
:param stride_ms: Striding size (in milliseconds) for generating frames.
:type stride_ms: float
:param window_ms: Window size (in milliseconds) for generating frames.
:type window_ms: float
:param max_freq: When specgram_type is 'linear', only FFT bins
corresponding to frequencies between [0, max_freq] are
returned; when specgram_type is 'mfcc', max_feq is the
highest band edge of mel filters.
:types max_freq: None|float
:param target_sample_rate: Audio are resampled (if upsampling or
downsampling is allowed) to this before
extracting spectrogram features.
:type target_sample_rate: float
:param use_dB_normalization: Whether to normalize the audio to a certain
decibels before extracting the features.
:type use_dB_normalization: bool
:param target_dB: Target audio decibels for normalization.
:type target_dB: float
"""
def __init__(self,
specgram_type='linear',
stride_ms=10.0,
window_ms=20.0,
max_freq=None,
target_sample_rate=16000,
use_dB_normalization=True,
target_dB=-20):
self._specgram_type = specgram_type
self._stride_ms = stride_ms
self._window_ms = window_ms
self._max_freq = max_freq
self._target_sample_rate = target_sample_rate
self._use_dB_normalization = use_dB_normalization
self._target_dB = target_dB
def featurize(self,
audio_segment,
allow_downsampling=True,
allow_upsampling=True):
"""Extract audio features from AudioSegment or SpeechSegment.
:param audio_segment: Audio/speech segment to extract features from.
:type audio_segment: AudioSegment|SpeechSegment
:param allow_downsampling: Whether to allow audio downsampling before
featurizing.
:type allow_downsampling: bool
:param allow_upsampling: Whether to allow audio upsampling before
featurizing.
:type allow_upsampling: bool
:return: Spectrogram audio feature in 2darray.
:rtype: ndarray
:raises ValueError: If audio sample rate is not supported.
"""
# upsampling or downsampling
if ((audio_segment.sample_rate > self._target_sample_rate and
allow_downsampling) or
(audio_segment.sample_rate < self._target_sample_rate and
allow_upsampling)):
audio_segment.resample(self._target_sample_rate)
if audio_segment.sample_rate != self._target_sample_rate:
raise ValueError("Audio sample rate is not supported. "
"Turn allow_downsampling or allow up_sampling on.")
# decibel normalization
if self._use_dB_normalization:
audio_segment.normalize(target_db=self._target_dB)
# extract spectrogram
return self._compute_specgram(audio_segment.samples,
audio_segment.sample_rate)
def _compute_specgram(self, samples, sample_rate):
"""Extract various audio features."""
if self._specgram_type == 'linear':
return self._compute_linear_specgram(
samples, sample_rate, self._stride_ms, self._window_ms,
self._max_freq)
elif self._specgram_type == 'mfcc':
return self._compute_mfcc(samples, sample_rate, self._stride_ms,
self._window_ms, self._max_freq)
else:
raise ValueError("Unknown specgram_type %s. "
"Supported values: linear." % self._specgram_type)
def _compute_linear_specgram(self,
samples,
sample_rate,
stride_ms=10.0,
window_ms=20.0,
max_freq=None,
eps=1e-14):
"""Compute the linear spectrogram from FFT energy."""
if max_freq is None:
max_freq = sample_rate / 2
if max_freq > sample_rate / 2:
raise ValueError("max_freq must be greater than half of "
"sample rate.")
if stride_ms > window_ms:
raise ValueError("Stride size must not be greater than "
"window size.")
stride_size = int(0.001 * sample_rate * stride_ms)
window_size = int(0.001 * sample_rate * window_ms)
specgram, freqs = self._specgram_real(
samples,
window_size=window_size,
stride_size=stride_size,
sample_rate=sample_rate)
ind = np.where(freqs <= max_freq)[0][-1] + 1
return np.log(specgram[:ind, :] + eps)
def _specgram_real(self, samples, window_size, stride_size, sample_rate):
"""Compute the spectrogram for samples from a real signal."""
# extract strided windows
truncate_size = (len(samples) - window_size) % stride_size
samples = samples[:len(samples) - truncate_size]
nshape = (window_size, (len(samples) - window_size) // stride_size + 1)
nstrides = (samples.strides[0], samples.strides[0] * stride_size)
windows = np.lib.stride_tricks.as_strided(
samples, shape=nshape, strides=nstrides)
assert np.all(
windows[:, 1] == samples[stride_size:(stride_size + window_size)])
# window weighting, squared Fast Fourier Transform (fft), scaling
weighting = np.hanning(window_size)[:, None]
fft = np.fft.rfft(windows * weighting, axis=0)
fft = np.absolute(fft)
fft = fft**2
scale = np.sum(weighting**2) * sample_rate
fft[1:-1, :] *= (2.0 / scale)
fft[(0, -1), :] /= scale
# prepare fft frequency list
freqs = float(sample_rate) / window_size * np.arange(fft.shape[0])
return fft, freqs
def _compute_mfcc(self,
samples,
sample_rate,
stride_ms=10.0,
window_ms=20.0,
max_freq=None):
"""Compute mfcc from samples."""
if max_freq is None:
max_freq = sample_rate / 2
if max_freq > sample_rate / 2:
raise ValueError("max_freq must not be greater than half of "
"sample rate.")
if stride_ms > window_ms:
raise ValueError("Stride size must not be greater than "
"window size.")
# compute the 13 cepstral coefficients, and the first one is replaced
# by log(frame energy)
mfcc_feat = mfcc(
signal=samples,
samplerate=sample_rate,
winlen=0.001 * window_ms,
winstep=0.001 * stride_ms,
highfreq=max_freq)
# Deltas
d_mfcc_feat = delta(mfcc_feat, 2)
# Deltas-Deltas
dd_mfcc_feat = delta(d_mfcc_feat, 2)
# transpose
mfcc_feat = np.transpose(mfcc_feat)
d_mfcc_feat = np.transpose(d_mfcc_feat)
dd_mfcc_feat = np.transpose(dd_mfcc_feat)
# concat above three features
concat_mfcc_feat = np.concatenate(
(mfcc_feat, d_mfcc_feat, dd_mfcc_feat))
return concat_mfcc_feat
deep_speech_2/data_utils/featurizer/speech_featurizer.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Contains the speech featurizer class."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.featurizer.audio_featurizer import AudioFeaturizer
from data_utils.featurizer.text_featurizer import TextFeaturizer
class SpeechFeaturizer(object):
"""Speech featurizer, for extracting features from both audio and transcript
contents of SpeechSegment.
Currently, for audio parts, it supports feature types of linear
spectrogram and mfcc; for transcript parts, it only supports char-level
tokenizing and conversion into a list of token indices. Note that the
token indexing order follows the given vocabulary file.
:param vocab_filepath: Filepath to load vocabulary for token indices
conversion.
:type specgram_type: basestring
:param specgram_type: Specgram feature type. Options: 'linear', 'mfcc'.
:type specgram_type: str
:param stride_ms: Striding size (in milliseconds) for generating frames.
:type stride_ms: float
:param window_ms: Window size (in milliseconds) for generating frames.
:type window_ms: float
:param max_freq: When specgram_type is 'linear', only FFT bins
corresponding to frequencies between [0, max_freq] are
returned; when specgram_type is 'mfcc', max_freq is the
highest band edge of mel filters.
:types max_freq: None|float
:param target_sample_rate: Speech are resampled (if upsampling or
downsampling is allowed) to this before
extracting spectrogram features.
:type target_sample_rate: float
:param use_dB_normalization: Whether to normalize the audio to a certain
decibels before extracting the features.
:type use_dB_normalization: bool
:param target_dB: Target audio decibels for normalization.
:type target_dB: float
"""
def __init__(self,
vocab_filepath,
specgram_type='linear',
stride_ms=10.0,
window_ms=20.0,
max_freq=None,
target_sample_rate=16000,
use_dB_normalization=True,
target_dB=-20):
self._audio_featurizer = AudioFeaturizer(
specgram_type=specgram_type,
stride_ms=stride_ms,
window_ms=window_ms,
max_freq=max_freq,
target_sample_rate=target_sample_rate,
use_dB_normalization=use_dB_normalization,
target_dB=target_dB)
self._text_featurizer = TextFeaturizer(vocab_filepath)
def featurize(self, speech_segment, keep_transcription_text):
"""Extract features for speech segment.
1. For audio parts, extract the audio features.
2. For transcript parts, keep the original text or convert text string
to a list of token indices in char-level.
:param audio_segment: Speech segment to extract features from.
:type audio_segment: SpeechSegment
:return: A tuple of 1) spectrogram audio feature in 2darray, 2) list of
char-level token indices.
:rtype: tuple
"""
audio_feature = self._audio_featurizer.featurize(speech_segment)
if keep_transcription_text:
return audio_feature, speech_segment.transcript
text_ids = self._text_featurizer.featurize(speech_segment.transcript)
return audio_feature, text_ids
@property
def vocab_size(self):
"""Return the vocabulary size.
:return: Vocabulary size.
:rtype: int
"""
return self._text_featurizer.vocab_size
@property
def vocab_list(self):
"""Return the vocabulary in list.
:return: Vocabulary in list.
:rtype: list
"""
return self._text_featurizer.vocab_list
deep_speech_2/data_utils/featurizer/text_featurizer.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Contains the text featurizer class."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import os
import codecs
class TextFeaturizer(object):
"""Text featurizer, for processing or extracting features from text.
Currently, it only supports char-level tokenizing and conversion into
a list of token indices. Note that the token indexing order follows the
given vocabulary file.
:param vocab_filepath: Filepath to load vocabulary for token indices
conversion.
:type specgram_type: basestring
"""
def __init__(self, vocab_filepath):
self._vocab_dict, self._vocab_list = self._load_vocabulary_from_file(
vocab_filepath)
def featurize(self, text):
"""Convert text string to a list of token indices in char-level.Note
that the token indexing order follows the given vocabulary file.
:param text: Text to process.
:type text: basestring
:return: List of char-level token indices.
:rtype: list
"""
tokens = self._char_tokenize(text)
return [self._vocab_dict[token] for token in tokens]
@property
def vocab_size(self):
"""Return the vocabulary size.
:return: Vocabulary size.
:rtype: int
"""
return len(self._vocab_list)
@property
def vocab_list(self):
"""Return the vocabulary in list.
:return: Vocabulary in list.
:rtype: list
"""
return self._vocab_list
def _char_tokenize(self, text):
"""Character tokenizer."""
return list(text.strip())
def _load_vocabulary_from_file(self, vocab_filepath):
"""Load vocabulary from file."""
vocab_lines = []
with codecs.open(vocab_filepath, 'r', 'utf-8') as file:
vocab_lines.extend(file.readlines())
vocab_list = [line[:-1] for line in vocab_lines]
vocab_dict = dict(
[(token, id) for (id, token) in enumerate(vocab_list)])
return vocab_dict, vocab_list
deep_speech_2/data_utils/normalizer.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Contains feature normalizers."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import numpy as np
import random
from data_utils.utility import read_manifest
from data_utils.audio import AudioSegment
class FeatureNormalizer(object):
"""Feature normalizer. Normalize features to be of zero mean and unit
stddev.
if mean_std_filepath is provided (not None), the normalizer will directly
initilize from the file. Otherwise, both manifest_path and featurize_func
should be given for on-the-fly mean and stddev computing.
:param mean_std_filepath: File containing the pre-computed mean and stddev.
:type mean_std_filepath: None|basestring
:param manifest_path: Manifest of instances for computing mean and stddev.
:type meanifest_path: None|basestring
:param featurize_func: Function to extract features. It should be callable
with ``featurize_func(audio_segment)``.
:type featurize_func: None|callable
:param num_samples: Number of random samples for computing mean and stddev.
:type num_samples: int
:param random_seed: Random seed for sampling instances.
:type random_seed: int
:raises ValueError: If both mean_std_filepath and manifest_path
(or both mean_std_filepath and featurize_func) are None.
"""
def __init__(self,
mean_std_filepath,
manifest_path=None,
featurize_func=None,
num_samples=500,
random_seed=0):
if not mean_std_filepath:
if not (manifest_path and featurize_func):
raise ValueError("If mean_std_filepath is None, meanifest_path "
"and featurize_func should not be None.")
self._rng = random.Random(random_seed)
self._compute_mean_std(manifest_path, featurize_func, num_samples)
else:
self._read_mean_std_from_file(mean_std_filepath)
def apply(self, features, eps=1e-14):
"""Normalize features to be of zero mean and unit stddev.
:param features: Input features to be normalized.
:type features: ndarray
:param eps: added to stddev to provide numerical stablibity.
:type eps: float
:return: Normalized features.
:rtype: ndarray
"""
return (features - self._mean) / (self._std + eps)
def write_to_file(self, filepath):
"""Write the mean and stddev to the file.
:param filepath: File to write mean and stddev.
:type filepath: basestring
"""
np.savez(filepath, mean=self._mean, std=self._std)
def _read_mean_std_from_file(self, filepath):
"""Load mean and std from file."""
npzfile = np.load(filepath)
self._mean = npzfile["mean"]
self._std = npzfile["std"]
def _compute_mean_std(self, manifest_path, featurize_func, num_samples):
"""Compute mean and std from randomly sampled instances."""
manifest = read_manifest(manifest_path)
sampled_manifest = self._rng.sample(manifest, num_samples)
features = []
for instance in sampled_manifest:
features.append(
featurize_func(
AudioSegment.from_file(instance["audio_filepath"])))
features = np.hstack(features)
self._mean = np.mean(features, axis=1).reshape([-1, 1])
self._std = np.std(features, axis=1).reshape([-1, 1])
deep_speech_2/data_utils/speech.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Contains the speech segment class."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from data_utils.audio import AudioSegment
class SpeechSegment(AudioSegment):
"""Speech segment abstraction, a subclass of AudioSegment,
with an additional transcript.
:param samples: Audio samples [num_samples x num_channels].
:type samples: ndarray.float32
:param sample_rate: Audio sample rate.
:type sample_rate: int
:param transcript: Transcript text for the speech.
:type transript: basestring
:raises TypeError: If the sample data type is not float or int.
"""
def __init__(self, samples, sample_rate, transcript):
AudioSegment.__init__(self, samples, sample_rate)
self._transcript = transcript
def __eq__(self, other):
"""Return whether two objects are equal.
"""
if not AudioSegment.__eq__(self, other):
return False
if self._transcript != other._transcript:
return False
return True
def __ne__(self, other):
"""Return whether two objects are unequal."""
return not self.__eq__(other)
@classmethod
def from_file(cls, filepath, transcript):
"""Create speech segment from audio file and corresponding transcript.
:param filepath: Filepath or file object to audio file.
:type filepath: basestring|file
:param transcript: Transcript text for the speech.
:type transript: basestring
:return: Speech segment instance.
:rtype: SpeechSegment
"""
audio = AudioSegment.from_file(filepath)
return cls(audio.samples, audio.sample_rate, transcript)
@classmethod
def from_bytes(cls, bytes, transcript):
"""Create speech segment from a byte string and corresponding
transcript.
:param bytes: Byte string containing audio samples.
:type bytes: str
:param transcript: Transcript text for the speech.
:type transript: basestring
:return: Speech segment instance.
:rtype: Speech Segment
"""
audio = AudioSegment.from_bytes(bytes)
return cls(audio.samples, audio.sample_rate, transcript)
@classmethod
def concatenate(cls, *segments):
"""Concatenate an arbitrary number of speech segments together, both
audio and transcript will be concatenated.
:param *segments: Input speech segments to be concatenated.
:type *segments: tuple of SpeechSegment
:return: Speech segment instance.
:rtype: SpeechSegment
:raises ValueError: If the number of segments is zero, or if the
sample_rate of any two segments does not match.
:raises TypeError: If any segment is not SpeechSegment instance.
"""
if len(segments) == 0:
raise ValueError("No speech segments are given to concatenate.")
sample_rate = segments[0]._sample_rate
transcripts = ""
for seg in segments:
if sample_rate != seg._sample_rate:
raise ValueError("Can't concatenate segments with "
"different sample rates")
if type(seg) is not cls:
raise TypeError("Only speech segments of the same type "
"instance can be concatenated.")
transcripts += seg._transcript
samples = np.concatenate([seg.samples for seg in segments])
return cls(samples, sample_rate, transcripts)
@classmethod
def slice_from_file(cls, filepath, transcript, start=None, end=None):
"""Loads a small section of an speech without having to load
the entire file into the memory which can be incredibly wasteful.
:param filepath: Filepath or file object to audio file.
:type filepath: basestring|file
:param start: Start time in seconds. If start is negative, it wraps
around from the end. If not provided, this function
reads from the very beginning.
:type start: float
:param end: End time in seconds. If end is negative, it wraps around
from the end. If not provided, the default behvaior is
to read to the end of the file.
:type end: float
:param transcript: Transcript text for the speech. if not provided,
the defaults is an empty string.
:type transript: basestring
:return: SpeechSegment instance of the specified slice of the input
speech file.
:rtype: SpeechSegment
"""
audio = AudioSegment.slice_from_file(filepath, start, end)
return cls(audio.samples, audio.sample_rate, transcript)
@classmethod
def make_silence(cls, duration, sample_rate):
"""Creates a silent speech segment of the given duration and
sample rate, transcript will be an empty string.
:param duration: Length of silence in seconds.
:type duration: float
:param sample_rate: Sample rate.
:type sample_rate: float
:return: Silence of the given duration.
:rtype: SpeechSegment
"""
audio = AudioSegment.make_silence(duration, sample_rate)
return cls(audio.samples, audio.sample_rate, "")
@property
def transcript(self):
"""Return the transcript text.
:return: Transcript text for the speech.
:rtype: basestring
"""
return self._transcript
deep_speech_2/data_utils/utility.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Contains data helper functions."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import json
import codecs
import os
import tarfile
import time
from Queue import Queue
from threading import Thread
from multiprocessing import Process, Manager
from paddle.v2.dataset.common import md5file
def read_manifest(manifest_path, max_duration=float('inf'), min_duration=0.0):
"""Load and parse manifest file.
Instances with durations outside [min_duration, max_duration] will be
filtered out.
:param manifest_path: Manifest file to load and parse.
:type manifest_path: basestring
:param max_duration: Maximal duration in seconds for instance filter.
:type max_duration: float
:param min_duration: Minimal duration in seconds for instance filter.
:type min_duration: float
:return: Manifest parsing results. List of dict.
:rtype: list
:raises IOError: If failed to parse the manifest.
"""
manifest = []
for json_line in codecs.open(manifest_path, 'r', 'utf-8'):
try:
json_data = json.loads(json_line)
except Exception as e:
raise IOError("Error reading manifest: %s" % str(e))
if (json_data["duration"] <= max_duration and
json_data["duration"] >= min_duration):
manifest.append(json_data)
return manifest
def download(url, md5sum, target_dir):
"""Download file from url to target_dir, and check md5sum."""
if not os.path.exists(target_dir): os.makedirs(target_dir)
filepath = os.path.join(target_dir, url.split("/")[-1])
if not (os.path.exists(filepath) and md5file(filepath) == md5sum):
print("Downloading %s ..." % url)
os.system("wget -c " + url + " -P " + target_dir)
print("\nMD5 Chesksum %s ..." % filepath)
if not md5file(filepath) == md5sum:
raise RuntimeError("MD5 checksum failed.")
else:
print("File exists, skip downloading. (%s)" % filepath)
return filepath
def unpack(filepath, target_dir, rm_tar=False):
"""Unpack the file to the target_dir."""
print("Unpacking %s ..." % filepath)
tar = tarfile.open(filepath)
tar.extractall(target_dir)
tar.close()
if rm_tar == True:
os.remove(filepath)
class XmapEndSignal():
pass
def xmap_readers_mp(mapper, reader, process_num, buffer_size, order=False):
"""A multiprocessing pipeline wrapper for the data reader.
:param mapper: Function to map sample.
:type mapper: callable
:param reader: Given data reader.
:type reader: callable
:param process_num: Number of processes in the pipeline
:type process_num: int
:param buffer_size: Maximal buffer size.
:type buffer_size: int
:param order: Reserve the order of samples from the given reader.
:type order: bool
:return: The wrappered reader
:rtype: callable
"""
end_flag = XmapEndSignal()
# define a worker to read samples from reader to in_queue
def read_worker(reader, in_queue):
for sample in reader():
in_queue.put(sample)
in_queue.put(end_flag)
# define a worker to read samples from reader to in_queue with order flag
def order_read_worker(reader, in_queue):
for order_id, sample in enumerate(reader()):
in_queue.put((order_id, sample))
in_queue.put(end_flag)
# define a worker to handle samples from in_queue by mapper and put results
# to out_queue
def handle_worker(in_queue, out_queue, mapper):
sample = in_queue.get()
while not isinstance(sample, XmapEndSignal):
out_queue.put(mapper(sample))
sample = in_queue.get()
in_queue.put(end_flag)
out_queue.put(end_flag)
# define a worker to handle samples from in_queue by mapper and put results
# to out_queue with order
def order_handle_worker(in_queue, out_queue, mapper, out_order):
ins = in_queue.get()
while not isinstance(ins, XmapEndSignal):
order_id, sample = ins
result = mapper(sample)
while order_id != out_order[0]:
time.sleep(0.001)
out_queue.put(result)
out_order[0] += 1
ins = in_queue.get()
in_queue.put(end_flag)
out_queue.put(end_flag)
# define a thread worker to flush samples from Manager.Queue to Queue
# for acceleration
def flush_worker(in_queue, out_queue):
finish = 0
while finish < process_num:
sample = in_queue.get()
if isinstance(sample, XmapEndSignal):
finish += 1
else:
out_queue.put(sample)
out_queue.put(end_flag)
def cleanup():
# kill all sub process and threads
os._exit(0)
def xreader():
# prepare shared memory
manager = Manager()
in_queue = manager.Queue(buffer_size)
out_queue = manager.Queue(buffer_size)
out_order = manager.list([0])
# start a read worker in a process
target = order_read_worker if order else read_worker
p = Process(target=target, args=(reader, in_queue))
p.daemon = True
p.start()
# start handle_workers with multiple processes
target = order_handle_worker if order else handle_worker
args = (in_queue, out_queue, mapper, out_order) if order else (
in_queue, out_queue, mapper)
workers = [
Process(target=target, args=args) for _ in xrange(process_num)
]
for w in workers:
w.daemon = True
w.start()
# start a thread to read data from slow Manager.Queue
flush_queue = Queue(buffer_size)
t = Thread(target=flush_worker, args=(out_queue, flush_queue))
t.daemon = True
t.start()
# get results
sample = flush_queue.get()
while not isinstance(sample, XmapEndSignal):
yield sample
sample = flush_queue.get()
return xreader, cleanup
deep_speech_2/decoders/decoders_deprecated.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Contains various CTC decoders."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from itertools import groupby
import numpy as np
from math import log
import multiprocessing
def ctc_greedy_decoder(probs_seq, vocabulary):
"""CTC greedy (best path) decoder.
Path consisting of the most probable tokens are further post-processed to
remove consecutive repetitions and all blanks.
:param probs_seq: 2-D list of probabilities over the vocabulary for each
character. Each element is a list of float probabilities
for one character.
:type probs_seq: list
:param vocabulary: Vocabulary list.
:type vocabulary: list
:return: Decoding result string.
:rtype: baseline
"""
# dimension verification
for probs in probs_seq:
if not len(probs) == len(vocabulary) + 1:
raise ValueError("probs_seq dimension mismatchedd with vocabulary")
# argmax to get the best index for each time step
max_index_list = list(np.array(probs_seq).argmax(axis=1))
# remove consecutive duplicate indexes
index_list = [index_group[0] for index_group in groupby(max_index_list)]
# remove blank indexes
blank_index = len(vocabulary)
index_list = [index for index in index_list if index != blank_index]
# convert index list to string
return ''.join([vocabulary[index] for index in index_list])
def ctc_beam_search_decoder(probs_seq,
beam_size,
vocabulary,
cutoff_prob=1.0,
cutoff_top_n=40,
ext_scoring_func=None,
nproc=False):
"""CTC Beam search decoder.
It utilizes beam search to approximately select top best decoding
labels and returning results in the descending order.
The implementation is based on Prefix Beam Search
(https://arxiv.org/abs/1408.2873), and the unclear part is
redesigned. Two important modifications: 1) in the iterative computation
of probabilities, the assignment operation is changed to accumulation for
one prefix may comes from different paths; 2) the if condition "if l^+ not
in A_prev then" after probabilities' computation is deprecated for it is
hard to understand and seems unnecessary.
:param probs_seq: 2-D list of probability distributions over each time
step, with each element being a list of normalized
probabilities over vocabulary and blank.
:type probs_seq: 2-D list
:param beam_size: Width for beam search.
:type beam_size: int
:param vocabulary: Vocabulary list.
:type vocabulary: list
:param cutoff_prob: Cutoff probability in pruning,
default 1.0, no pruning.
:type cutoff_prob: float
:param ext_scoring_func: External scoring function for
partially decoded sentence, e.g. word count
or language model.
:type external_scoring_func: callable
:param nproc: Whether the decoder used in multiprocesses.
:type nproc: bool
:return: List of tuples of log probability and sentence as decoding
results, in descending order of the probability.
:rtype: list
"""
# dimension check
for prob_list in probs_seq:
if not len(prob_list) == len(vocabulary) + 1:
raise ValueError("The shape of prob_seq does not match with the "
"shape of the vocabulary.")
# blank_id assign
blank_id = len(vocabulary)
# If the decoder called in the multiprocesses, then use the global scorer
# instantiated in ctc_beam_search_decoder_batch().
if nproc is True:
global ext_nproc_scorer
ext_scoring_func = ext_nproc_scorer
## initialize
# prefix_set_prev: the set containing selected prefixes
# probs_b_prev: prefixes' probability ending with blank in previous step
# probs_nb_prev: prefixes' probability ending with non-blank in previous step
prefix_set_prev = {'\t': 1.0}
probs_b_prev, probs_nb_prev = {'\t': 1.0}, {'\t': 0.0}
## extend prefix in loop
for time_step in xrange(len(probs_seq)):
# prefix_set_next: the set containing candidate prefixes
# probs_b_cur: prefixes' probability ending with blank in current step
# probs_nb_cur: prefixes' probability ending with non-blank in current step
prefix_set_next, probs_b_cur, probs_nb_cur = {}, {}, {}
prob_idx = list(enumerate(probs_seq[time_step]))
cutoff_len = len(prob_idx)
#If pruning is enabled
if cutoff_prob < 1.0 or cutoff_top_n < cutoff_len:
prob_idx = sorted(prob_idx, key=lambda asd: asd[1], reverse=True)
cutoff_len, cum_prob = 0, 0.0
for i in xrange(len(prob_idx)):
cum_prob += prob_idx[i][1]
cutoff_len += 1
if cum_prob >= cutoff_prob:
break
cutoff_len = min(cutoff_len, cutoff_top_n)
prob_idx = prob_idx[0:cutoff_len]
for l in prefix_set_prev:
if not prefix_set_next.has_key(l):
probs_b_cur[l], probs_nb_cur[l] = 0.0, 0.0
# extend prefix by travering prob_idx
for index in xrange(cutoff_len):
c, prob_c = prob_idx[index][0], prob_idx[index][1]
if c == blank_id:
probs_b_cur[l] += prob_c * (
probs_b_prev[l] + probs_nb_prev[l])
else:
last_char = l[-1]
new_char = vocabulary[c]
l_plus = l + new_char
if not prefix_set_next.has_key(l_plus):
probs_b_cur[l_plus], probs_nb_cur[l_plus] = 0.0, 0.0
if new_char == last_char:
probs_nb_cur[l_plus] += prob_c * probs_b_prev[l]
probs_nb_cur[l] += prob_c * probs_nb_prev[l]
elif new_char == ' ':
if (ext_scoring_func is None) or (len(l) == 1):
score = 1.0
else:
prefix = l[1:]
score = ext_scoring_func(prefix)
probs_nb_cur[l_plus] += score * prob_c * (
probs_b_prev[l] + probs_nb_prev[l])
else:
probs_nb_cur[l_plus] += prob_c * (
probs_b_prev[l] + probs_nb_prev[l])
# add l_plus into prefix_set_next
prefix_set_next[l_plus] = probs_nb_cur[
l_plus] + probs_b_cur[l_plus]
# add l into prefix_set_next
prefix_set_next[l] = probs_b_cur[l] + probs_nb_cur[l]
# update probs
probs_b_prev, probs_nb_prev = probs_b_cur, probs_nb_cur
## store top beam_size prefixes
prefix_set_prev = sorted(
prefix_set_next.iteritems(), key=lambda asd: asd[1], reverse=True)
if beam_size < len(prefix_set_prev):
prefix_set_prev = prefix_set_prev[:beam_size]
prefix_set_prev = dict(prefix_set_prev)
beam_result = []
for seq, prob in prefix_set_prev.items():
if prob > 0.0 and len(seq) > 1:
result = seq[1:]
# score last word by external scorer
if (ext_scoring_func is not None) and (result[-1] != ' '):
prob = prob * ext_scoring_func(result)
log_prob = log(prob)
beam_result.append((log_prob, result))
else:
beam_result.append((float('-inf'), ''))
## output top beam_size decoding results
beam_result = sorted(beam_result, key=lambda asd: asd[0], reverse=True)
return beam_result
def ctc_beam_search_decoder_batch(probs_split,
beam_size,
vocabulary,
num_processes,
cutoff_prob=1.0,
cutoff_top_n=40,
ext_scoring_func=None):
"""CTC beam search decoder using multiple processes.
:param probs_seq: 3-D list with each element as an instance of 2-D list
of probabilities used by ctc_beam_search_decoder().
:type probs_seq: 3-D list
:param beam_size: Width for beam search.
:type beam_size: int
:param vocabulary: Vocabulary list.
:type vocabulary: list
:param num_processes: Number of parallel processes.
:type num_processes: int
:param cutoff_prob: Cutoff probability in pruning,
default 1.0, no pruning.
:type cutoff_prob: float
:param num_processes: Number of parallel processes.
:type num_processes: int
:param ext_scoring_func: External scoring function for
partially decoded sentence, e.g. word count
or language model.
:type external_scoring_function: callable
:return: List of tuples of log probability and sentence as decoding
results, in descending order of the probability.
:rtype: list
"""
if not num_processes > 0:
raise ValueError("Number of processes must be positive!")
# use global variable to pass the externnal scorer to beam search decoder
global ext_nproc_scorer
ext_nproc_scorer = ext_scoring_func
nproc = True
pool = multiprocessing.Pool(processes=num_processes)
results = []
for i, probs_list in enumerate(probs_split):
args = (probs_list, beam_size, vocabulary, cutoff_prob, cutoff_top_n,
None, nproc)
results.append(pool.apply_async(ctc_beam_search_decoder, args))
pool.close()
pool.join()
beam_search_results = [result.get() for result in results]
return beam_search_results
deep_speech_2/decoders/scorer_deprecated.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""External Scorer for Beam Search Decoder."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import os
import kenlm
import numpy as np
class Scorer(object):
"""External scorer to evaluate a prefix or whole sentence in
beam search decoding, including the score from n-gram language
model and word count.
:param alpha: Parameter associated with language model. Don't use
language model when alpha = 0.
:type alpha: float
:param beta: Parameter associated with word count. Don't use word
count when beta = 0.
:type beta: float
:model_path: Path to load language model.
:type model_path: basestring
"""
def __init__(self, alpha, beta, model_path):
self._alpha = alpha
self._beta = beta
if not os.path.isfile(model_path):
raise IOError("Invaid language model path: %s" % model_path)
self._language_model = kenlm.LanguageModel(model_path)
# n-gram language model scoring
def _language_model_score(self, sentence):
#log10 prob of last word
log_cond_prob = list(
self._language_model.full_scores(sentence, eos=False))[-1][0]
return np.power(10, log_cond_prob)
# word insertion term
def _word_count(self, sentence):
words = sentence.strip().split(' ')
return len(words)
# reset alpha and beta
def reset_params(self, alpha, beta):
self._alpha = alpha
self._beta = beta
# execute evaluation
def __call__(self, sentence, log=False):
"""Evaluation function, gathering all the different scores
and return the final one.
:param sentence: The input sentence for evalutation
:type sentence: basestring
:param log: Whether return the score in log representation.
:type log: bool
:return: Evaluation score, in the decimal or log.
:rtype: float
"""
lm = self._language_model_score(sentence)
word_cnt = self._word_count(sentence)
if log == False:
score = np.power(lm, self._alpha) * np.power(word_cnt, self._beta)
else:
score = self._alpha * np.log(lm) + self._beta * np.log(word_cnt)
return score
deep_speech_2/decoders/swig/_init_paths.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Set up paths for DS2"""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import os.path
import sys
def add_path(path):
if path not in sys.path:
sys.path.insert(0, path)
this_dir = os.path.dirname(__file__)
# Add project path to PYTHONPATH
proj_path = os.path.join(this_dir, '..')
add_path(proj_path)
deep_speech_2/decoders/swig/ctc_beam_search_decoder.cpp 已删除 100644 → 0
浏览文件 @ 788f7011
#include "ctc_beam_search_decoder.h"
#include <algorithm>
#include <cmath>
#include <iostream>
#include <limits>
#include <map>
#include <utility>
#include "ThreadPool.h"
#include "fst/fstlib.h"
#include "decoder_utils.h"
#include "path_trie.h"
using FSTMATCH = fst::SortedMatcher<fst::StdVectorFst>;
std::vector<std::pair<double, std::string>> ctc_beam_search_decoder(
const std::vector<std::vector<double>> &probs_seq,
const std::vector<std::string> &vocabulary,
size_t beam_size,
double cutoff_prob,
size_t cutoff_top_n,
Scorer *ext_scorer) {
// dimension check
size_t num_time_steps = probs_seq.size();
for (size_t i = 0; i < num_time_steps; ++i) {
VALID_CHECK_EQ(probs_seq[i].size(),
vocabulary.size() + 1,
"The shape of probs_seq does not match with "
"the shape of the vocabulary");
}
// assign blank id
size_t blank_id = vocabulary.size();
// assign space id
auto it = std::find(vocabulary.begin(), vocabulary.end(), " ");
int space_id = it - vocabulary.begin();
// if no space in vocabulary
if ((size_t)space_id >= vocabulary.size()) {
space_id = -2;
}
// init prefixes' root
PathTrie root;
root.score = root.log_prob_b_prev = 0.0;
std::vector<PathTrie *> prefixes;
prefixes.push_back(&root);
if (ext_scorer != nullptr && !ext_scorer->is_character_based()) {
auto fst_dict = static_cast<fst::StdVectorFst *>(ext_scorer->dictionary);
fst::StdVectorFst *dict_ptr = fst_dict->Copy(true);
root.set_dictionary(dict_ptr);
auto matcher = std::make_shared<FSTMATCH>(*dict_ptr, fst::MATCH_INPUT);
root.set_matcher(matcher);
}
// prefix search over time
for (size_t time_step = 0; time_step < num_time_steps; ++time_step) {
auto &prob = probs_seq[time_step];
float min_cutoff = -NUM_FLT_INF;
bool full_beam = false;
if (ext_scorer != nullptr) {
size_t num_prefixes = std::min(prefixes.size(), beam_size);
std::sort(
prefixes.begin(), prefixes.begin() + num_prefixes, prefix_compare);
min_cutoff = prefixes[num_prefixes - 1]->score +
std::log(prob[blank_id]) - std::max(0.0, ext_scorer->beta);
full_beam = (num_prefixes == beam_size);
}
std::vector<std::pair<size_t, float>> log_prob_idx =
get_pruned_log_probs(prob, cutoff_prob, cutoff_top_n);
// loop over chars
for (size_t index = 0; index < log_prob_idx.size(); index++) {
auto c = log_prob_idx[index].first;
auto log_prob_c = log_prob_idx[index].second;
for (size_t i = 0; i < prefixes.size() && i < beam_size; ++i) {
auto prefix = prefixes[i];
if (full_beam && log_prob_c + prefix->score < min_cutoff) {
break;
}
// blank
if (c == blank_id) {
prefix->log_prob_b_cur =
log_sum_exp(prefix->log_prob_b_cur, log_prob_c + prefix->score);
continue;
}
// repeated character
if (c == prefix->character) {
prefix->log_prob_nb_cur = log_sum_exp(
prefix->log_prob_nb_cur, log_prob_c + prefix->log_prob_nb_prev);
}
// get new prefix
auto prefix_new = prefix->get_path_trie(c);
if (prefix_new != nullptr) {
float log_p = -NUM_FLT_INF;
if (c == prefix->character &&
prefix->log_prob_b_prev > -NUM_FLT_INF) {
log_p = log_prob_c + prefix->log_prob_b_prev;
} else if (c != prefix->character) {
log_p = log_prob_c + prefix->score;
}
// language model scoring
if (ext_scorer != nullptr &&
(c == space_id || ext_scorer->is_character_based())) {
PathTrie *prefix_to_score = nullptr;
// skip scoring the space
if (ext_scorer->is_character_based()) {
prefix_to_score = prefix_new;
} else {
prefix_to_score = prefix;
}
float score = 0.0;
std::vector<std::string> ngram;
ngram = ext_scorer->make_ngram(prefix_to_score);
score = ext_scorer->get_log_cond_prob(ngram) * ext_scorer->alpha;
log_p += score;
log_p += ext_scorer->beta;
}
prefix_new->log_prob_nb_cur =
log_sum_exp(prefix_new->log_prob_nb_cur, log_p);
}
} // end of loop over prefix
} // end of loop over vocabulary
prefixes.clear();
// update log probs
root.iterate_to_vec(prefixes);
// only preserve top beam_size prefixes
if (prefixes.size() >= beam_size) {
std::nth_element(prefixes.begin(),
prefixes.begin() + beam_size,
prefixes.end(),
prefix_compare);
for (size_t i = beam_size; i < prefixes.size(); ++i) {
prefixes[i]->remove();
}
}
} // end of loop over time
// score the last word of each prefix that doesn't end with space
if (ext_scorer != nullptr && !ext_scorer->is_character_based()) {
for (size_t i = 0; i < beam_size && i < prefixes.size(); ++i) {
auto prefix = prefixes[i];
if (!prefix->is_empty() && prefix->character != space_id) {
float score = 0.0;
std::vector<std::string> ngram = ext_scorer->make_ngram(prefix);
score = ext_scorer->get_log_cond_prob(ngram) * ext_scorer->alpha;
score += ext_scorer->beta;
prefix->score += score;
}
}
}
size_t num_prefixes = std::min(prefixes.size(), beam_size);
std::sort(prefixes.begin(), prefixes.begin() + num_prefixes, prefix_compare);
// compute aproximate ctc score as the return score, without affecting the
// return order of decoding result. To delete when decoder gets stable.
for (size_t i = 0; i < beam_size && i < prefixes.size(); ++i) {
double approx_ctc = prefixes[i]->score;
if (ext_scorer != nullptr) {
std::vector<int> output;
prefixes[i]->get_path_vec(output);
auto prefix_length = output.size();
auto words = ext_scorer->split_labels(output);
// remove word insert
approx_ctc = approx_ctc - prefix_length * ext_scorer->beta;
// remove language model weight:
approx_ctc -= (ext_scorer->get_sent_log_prob(words)) * ext_scorer->alpha;
}
prefixes[i]->approx_ctc = approx_ctc;
}
return get_beam_search_result(prefixes, vocabulary, beam_size);
}
std::vector<std::vector<std::pair<double, std::string>>>
ctc_beam_search_decoder_batch(
const std::vector<std::vector<std::vector<double>>> &probs_split,
const std::vector<std::string> &vocabulary,
size_t beam_size,
size_t num_processes,
double cutoff_prob,
size_t cutoff_top_n,
Scorer *ext_scorer) {
VALID_CHECK_GT(num_processes, 0, "num_processes must be nonnegative!");
// thread pool
ThreadPool pool(num_processes);
// number of samples
size_t batch_size = probs_split.size();
// enqueue the tasks of decoding
std::vector<std::future<std::vector<std::pair<double, std::string>>>> res;
for (size_t i = 0; i < batch_size; ++i) {
res.emplace_back(pool.enqueue(ctc_beam_search_decoder,
probs_split[i],
vocabulary,
beam_size,
cutoff_prob,
cutoff_top_n,
ext_scorer));
}
// get decoding results
std::vector<std::vector<std::pair<double, std::string>>> batch_results;
for (size_t i = 0; i < batch_size; ++i) {
batch_results.emplace_back(res[i].get());
}
return batch_results;
}
deep_speech_2/decoders/swig/ctc_beam_search_decoder.h 已删除 100644 → 0
浏览文件 @ 788f7011
#ifndef CTC_BEAM_SEARCH_DECODER_H_
#define CTC_BEAM_SEARCH_DECODER_H_
#include <string>
#include <utility>
#include <vector>
#include "scorer.h"
/* CTC Beam Search Decoder
* Parameters:
* probs_seq: 2-D vector that each element is a vector of probabilities
* over vocabulary of one time step.
* vocabulary: A vector of vocabulary.
* beam_size: The width of beam search.
* cutoff_prob: Cutoff probability for pruning.
* cutoff_top_n: Cutoff number for pruning.
* ext_scorer: External scorer to evaluate a prefix, which consists of
* n-gram language model scoring and word insertion term.
* Default null, decoding the input sample without scorer.
* Return:
* A vector that each element is a pair of score and decoding result,
* in desending order.
*/
std::vector<std::pair<double, std::string>> ctc_beam_search_decoder(
const std::vector<std::vector<double>> &probs_seq,
const std::vector<std::string> &vocabulary,
size_t beam_size,
double cutoff_prob = 1.0,
size_t cutoff_top_n = 40,
Scorer *ext_scorer = nullptr);
/* CTC Beam Search Decoder for batch data
* Parameters:
* probs_seq: 3-D vector that each element is a 2-D vector that can be used
* by ctc_beam_search_decoder().
* vocabulary: A vector of vocabulary.
* beam_size: The width of beam search.
* num_processes: Number of threads for beam search.
* cutoff_prob: Cutoff probability for pruning.
* cutoff_top_n: Cutoff number for pruning.
* ext_scorer: External scorer to evaluate a prefix, which consists of
* n-gram language model scoring and word insertion term.
* Default null, decoding the input sample without scorer.
* Return:
* A 2-D vector that each element is a vector of beam search decoding
* result for one audio sample.
*/
std::vector<std::vector<std::pair<double, std::string>>>
ctc_beam_search_decoder_batch(
const std::vector<std::vector<std::vector<double>>> &probs_split,
const std::vector<std::string> &vocabulary,
size_t beam_size,
size_t num_processes,
double cutoff_prob = 1.0,
size_t cutoff_top_n = 40,
Scorer *ext_scorer = nullptr);
#endif // CTC_BEAM_SEARCH_DECODER_H_
deep_speech_2/decoders/swig/ctc_greedy_decoder.cpp 已删除 100644 → 0
浏览文件 @ 788f7011
#include "ctc_greedy_decoder.h"
#include "decoder_utils.h"
std::string ctc_greedy_decoder(
const std::vector<std::vector<double>> &probs_seq,
const std::vector<std::string> &vocabulary) {
// dimension check
size_t num_time_steps = probs_seq.size();
for (size_t i = 0; i < num_time_steps; ++i) {
VALID_CHECK_EQ(probs_seq[i].size(),
vocabulary.size() + 1,
"The shape of probs_seq does not match with "
"the shape of the vocabulary");
}
size_t blank_id = vocabulary.size();
std::vector<size_t> max_idx_vec(num_time_steps, 0);
std::vector<size_t> idx_vec;
for (size_t i = 0; i < num_time_steps; ++i) {
double max_prob = 0.0;
size_t max_idx = 0;
const std::vector<double> &probs_step = probs_seq[i];
for (size_t j = 0; j < probs_step.size(); ++j) {
if (max_prob < probs_step[j]) {
max_idx = j;
max_prob = probs_step[j];
}
}
// id with maximum probability in current time step
max_idx_vec[i] = max_idx;
// deduplicate
if ((i == 0) || ((i > 0) && max_idx_vec[i] != max_idx_vec[i - 1])) {
idx_vec.push_back(max_idx_vec[i]);
}
}
std::string best_path_result;
for (size_t i = 0; i < idx_vec.size(); ++i) {
if (idx_vec[i] != blank_id) {
best_path_result += vocabulary[idx_vec[i]];
}
}
return best_path_result;
}
deep_speech_2/decoders/swig/ctc_greedy_decoder.h 已删除 100644 → 0
浏览文件 @ 788f7011
#ifndef CTC_GREEDY_DECODER_H
#define CTC_GREEDY_DECODER_H
#include <string>
#include <vector>
/* CTC Greedy (Best Path) Decoder
*
* Parameters:
* probs_seq: 2-D vector that each element is a vector of probabilities
* over vocabulary of one time step.
* vocabulary: A vector of vocabulary.
* Return:
* The decoding result in string
*/
std::string ctc_greedy_decoder(
const std::vector<std::vector<double>>& probs_seq,
const std::vector<std::string>& vocabulary);
#endif // CTC_GREEDY_DECODER_H
deep_speech_2/decoders/swig/decoder_utils.cpp 已删除 100644 → 0
浏览文件 @ 788f7011
#include "decoder_utils.h"
#include <algorithm>
#include <cmath>
#include <limits>
std::vector<std::pair<size_t, float>> get_pruned_log_probs(
const std::vector<double> &prob_step,
double cutoff_prob,
size_t cutoff_top_n) {
std::vector<std::pair<int, double>> prob_idx;
for (size_t i = 0; i < prob_step.size(); ++i) {
prob_idx.push_back(std::pair<int, double>(i, prob_step[i]));
}
// pruning of vacobulary
size_t cutoff_len = prob_step.size();
if (cutoff_prob < 1.0 || cutoff_top_n < cutoff_len) {
std::sort(
prob_idx.begin(), prob_idx.end(), pair_comp_second_rev<int, double>);
if (cutoff_prob < 1.0) {
double cum_prob = 0.0;
cutoff_len = 0;
for (size_t i = 0; i < prob_idx.size(); ++i) {
cum_prob += prob_idx[i].second;
cutoff_len += 1;
if (cum_prob >= cutoff_prob || cutoff_len >= cutoff_top_n) break;
}
}
prob_idx = std::vector<std::pair<int, double>>(
prob_idx.begin(), prob_idx.begin() + cutoff_len);
}
std::vector<std::pair<size_t, float>> log_prob_idx;
for (size_t i = 0; i < cutoff_len; ++i) {
log_prob_idx.push_back(std::pair<int, float>(
prob_idx[i].first, log(prob_idx[i].second + NUM_FLT_MIN)));
}
return log_prob_idx;
}
std::vector<std::pair<double, std::string>> get_beam_search_result(
const std::vector<PathTrie *> &prefixes,
const std::vector<std::string> &vocabulary,
size_t beam_size) {
// allow for the post processing
std::vector<PathTrie *> space_prefixes;
if (space_prefixes.empty()) {
for (size_t i = 0; i < beam_size && i < prefixes.size(); ++i) {
space_prefixes.push_back(prefixes[i]);
}
}
std::sort(space_prefixes.begin(), space_prefixes.end(), prefix_compare);
std::vector<std::pair<double, std::string>> output_vecs;
for (size_t i = 0; i < beam_size && i < space_prefixes.size(); ++i) {
std::vector<int> output;
space_prefixes[i]->get_path_vec(output);
// convert index to string
std::string output_str;
for (size_t j = 0; j < output.size(); j++) {
output_str += vocabulary[output[j]];
}
std::pair<double, std::string> output_pair(-space_prefixes[i]->approx_ctc,
output_str);
output_vecs.emplace_back(output_pair);
}
return output_vecs;
}
size_t get_utf8_str_len(const std::string &str) {
size_t str_len = 0;
for (char c : str) {
str_len += ((c & 0xc0) != 0x80);
}
return str_len;
}
std::vector<std::string> split_utf8_str(const std::string &str) {
std::vector<std::string> result;
std::string out_str;
for (char c : str) {
if ((c & 0xc0) != 0x80) // new UTF-8 character
{
if (!out_str.empty()) {
result.push_back(out_str);
out_str.clear();
}
}
out_str.append(1, c);
}
result.push_back(out_str);
return result;
}
std::vector<std::string> split_str(const std::string &s,
const std::string &delim) {
std::vector<std::string> result;
std::size_t start = 0, delim_len = delim.size();
while (true) {
std::size_t end = s.find(delim, start);
if (end == std::string::npos) {
if (start < s.size()) {
result.push_back(s.substr(start));
}
break;
}
if (end > start) {
result.push_back(s.substr(start, end - start));
}
start = end + delim_len;
}
return result;
}
bool prefix_compare(const PathTrie *x, const PathTrie *y) {
if (x->score == y->score) {
if (x->character == y->character) {
return false;
} else {
return (x->character < y->character);
}
} else {
return x->score > y->score;
}
}
void add_word_to_fst(const std::vector<int> &word,
fst::StdVectorFst *dictionary) {
if (dictionary->NumStates() == 0) {
fst::StdVectorFst::StateId start = dictionary->AddState();
assert(start == 0);
dictionary->SetStart(start);
}
fst::StdVectorFst::StateId src = dictionary->Start();
fst::StdVectorFst::StateId dst;
for (auto c : word) {
dst = dictionary->AddState();
dictionary->AddArc(src, fst::StdArc(c, c, 0, dst));
src = dst;
}
dictionary->SetFinal(dst, fst::StdArc::Weight::One());
}
bool add_word_to_dictionary(
const std::string &word,
const std::unordered_map<std::string, int> &char_map,
bool add_space,
int SPACE_ID,
fst::StdVectorFst *dictionary) {
auto characters = split_utf8_str(word);
std::vector<int> int_word;
for (auto &c : characters) {
if (c == " ") {
int_word.push_back(SPACE_ID);
} else {
auto int_c = char_map.find(c);
if (int_c != char_map.end()) {
int_word.push_back(int_c->second);
} else {
return false; // return without adding
}
}
}
if (add_space) {
int_word.push_back(SPACE_ID);
}
add_word_to_fst(int_word, dictionary);
return true; // return with successful adding
}
deep_speech_2/decoders/swig/decoder_utils.h 已删除 100644 → 0
浏览文件 @ 788f7011
#ifndef DECODER_UTILS_H_
#define DECODER_UTILS_H_
#include <utility>
#include "fst/log.h"
#include "path_trie.h"
const float NUM_FLT_INF = std::numeric_limits<float>::max();
const float NUM_FLT_MIN = std::numeric_limits<float>::min();
// inline function for validation check
inline void check(
bool x, const char *expr, const char *file, int line, const char *err) {
if (!x) {
std::cout << "[" << file << ":" << line << "] ";
LOG(FATAL) << "\"" << expr << "\" check failed. " << err;
}
}
#define VALID_CHECK(x, info) \
check(static_cast<bool>(x), #x, __FILE__, __LINE__, info)
#define VALID_CHECK_EQ(x, y, info) VALID_CHECK((x) == (y), info)
#define VALID_CHECK_GT(x, y, info) VALID_CHECK((x) > (y), info)
#define VALID_CHECK_LT(x, y, info) VALID_CHECK((x) < (y), info)
// Function template for comparing two pairs
template <typename T1, typename T2>
bool pair_comp_first_rev(const std::pair<T1, T2> &a,
const std::pair<T1, T2> &b) {
return a.first > b.first;
}
// Function template for comparing two pairs
template <typename T1, typename T2>
bool pair_comp_second_rev(const std::pair<T1, T2> &a,
const std::pair<T1, T2> &b) {
return a.second > b.second;
}
// Return the sum of two probabilities in log scale
template <typename T>
T log_sum_exp(const T &x, const T &y) {
static T num_min = -std::numeric_limits<T>::max();
if (x <= num_min) return y;
if (y <= num_min) return x;
T xmax = std::max(x, y);
return std::log(std::exp(x - xmax) + std::exp(y - xmax)) + xmax;
}
// Get pruned probability vector for each time step's beam search
std::vector<std::pair<size_t, float>> get_pruned_log_probs(
const std::vector<double> &prob_step,
double cutoff_prob,
size_t cutoff_top_n);
// Get beam search result from prefixes in trie tree
std::vector<std::pair<double, std::string>> get_beam_search_result(
const std::vector<PathTrie *> &prefixes,
const std::vector<std::string> &vocabulary,
size_t beam_size);
// Functor for prefix comparsion
bool prefix_compare(const PathTrie *x, const PathTrie *y);
/* Get length of utf8 encoding string
* See: http://stackoverflow.com/a/4063229
*/
size_t get_utf8_str_len(const std::string &str);
/* Split a string into a list of strings on a given string
* delimiter. NB: delimiters on beginning / end of string are
* trimmed. Eg, "FooBarFoo" split on "Foo" returns ["Bar"].
*/
std::vector<std::string> split_str(const std::string &s,
const std::string &delim);
/* Splits string into vector of strings representing
* UTF-8 characters (not same as chars)
*/
std::vector<std::string> split_utf8_str(const std::string &str);
// Add a word in index to the dicionary of fst
void add_word_to_fst(const std::vector<int> &word,
fst::StdVectorFst *dictionary);
// Add a word in string to dictionary
bool add_word_to_dictionary(
const std::string &word,
const std::unordered_map<std::string, int> &char_map,
bool add_space,
int SPACE_ID,
fst::StdVectorFst *dictionary);
#endif // DECODER_UTILS_H
deep_speech_2/decoders/swig/decoders.i 已删除 100644 → 0
浏览文件 @ 788f7011
%module swig_decoders
%{
#include "scorer.h"
#include "ctc_greedy_decoder.h"
#include "ctc_beam_search_decoder.h"
#include "decoder_utils.h"
%}
%include "std_vector.i"
%include "std_pair.i"
%include "std_string.i"
%import "decoder_utils.h"
namespace std {
%template(DoubleVector) std::vector<double>;
%template(IntVector) std::vector<int>;
%template(StringVector) std::vector<std::string>;
%template(VectorOfStructVector) std::vector<std::vector<double> >;
%template(FloatVector) std::vector<float>;
%template(Pair) std::pair<float, std::string>;
%template(PairFloatStringVector) std::vector<std::pair<float, std::string> >;
%template(PairDoubleStringVector) std::vector<std::pair<double, std::string> >;
%template(PairDoubleStringVector2) std::vector<std::vector<std::pair<double, std::string> > >;
%template(DoubleVector3) std::vector<std::vector<std::vector<double> > >;
}
%template(IntDoublePairCompSecondRev) pair_comp_second_rev<int, double>;
%template(StringDoublePairCompSecondRev) pair_comp_second_rev<std::string, double>;
%template(DoubleStringPairCompFirstRev) pair_comp_first_rev<double, std::string>;
%include "scorer.h"
%include "ctc_greedy_decoder.h"
%include "ctc_beam_search_decoder.h"
deep_speech_2/decoders/swig/path_trie.cpp 已删除 100644 → 0
浏览文件 @ 788f7011
#include "path_trie.h"
#include <algorithm>
#include <limits>
#include <memory>
#include <utility>
#include <vector>
#include "decoder_utils.h"
PathTrie::PathTrie() {
log_prob_b_prev = -NUM_FLT_INF;
log_prob_nb_prev = -NUM_FLT_INF;
log_prob_b_cur = -NUM_FLT_INF;
log_prob_nb_cur = -NUM_FLT_INF;
score = -NUM_FLT_INF;
ROOT_ = -1;
character = ROOT_;
exists_ = true;
parent = nullptr;
dictionary_ = nullptr;
dictionary_state_ = 0;
has_dictionary_ = false;
matcher_ = nullptr;
}
PathTrie::~PathTrie() {
for (auto child : children_) {
delete child.second;
}
}
PathTrie* PathTrie::get_path_trie(int new_char, bool reset) {
auto child = children_.begin();
for (child = children_.begin(); child != children_.end(); ++child) {
if (child->first == new_char) {
break;
}
}
if (child != children_.end()) {
if (!child->second->exists_) {
child->second->exists_ = true;
child->second->log_prob_b_prev = -NUM_FLT_INF;
child->second->log_prob_nb_prev = -NUM_FLT_INF;
child->second->log_prob_b_cur = -NUM_FLT_INF;
child->second->log_prob_nb_cur = -NUM_FLT_INF;
}
return (child->second);
} else {
if (has_dictionary_) {
matcher_->SetState(dictionary_state_);
bool found = matcher_->Find(new_char);
if (!found) {
// Adding this character causes word outside dictionary
auto FSTZERO = fst::TropicalWeight::Zero();
auto final_weight = dictionary_->Final(dictionary_state_);
bool is_final = (final_weight != FSTZERO);
if (is_final && reset) {
dictionary_state_ = dictionary_->Start();
}
return nullptr;
} else {
PathTrie* new_path = new PathTrie;
new_path->character = new_char;
new_path->parent = this;
new_path->dictionary_ = dictionary_;
new_path->dictionary_state_ = matcher_->Value().nextstate;
new_path->has_dictionary_ = true;
new_path->matcher_ = matcher_;
children_.push_back(std::make_pair(new_char, new_path));
return new_path;
}
} else {
PathTrie* new_path = new PathTrie;
new_path->character = new_char;
new_path->parent = this;
children_.push_back(std::make_pair(new_char, new_path));
return new_path;
}
}
}
PathTrie* PathTrie::get_path_vec(std::vector<int>& output) {
return get_path_vec(output, ROOT_);
}
PathTrie* PathTrie::get_path_vec(std::vector<int>& output,
int stop,
size_t max_steps) {
if (character == stop || character == ROOT_ || output.size() == max_steps) {
std::reverse(output.begin(), output.end());
return this;
} else {
output.push_back(character);
return parent->get_path_vec(output, stop, max_steps);
}
}
void PathTrie::iterate_to_vec(std::vector<PathTrie*>& output) {
if (exists_) {
log_prob_b_prev = log_prob_b_cur;
log_prob_nb_prev = log_prob_nb_cur;
log_prob_b_cur = -NUM_FLT_INF;
log_prob_nb_cur = -NUM_FLT_INF;
score = log_sum_exp(log_prob_b_prev, log_prob_nb_prev);
output.push_back(this);
}
for (auto child : children_) {
child.second->iterate_to_vec(output);
}
}
void PathTrie::remove() {
exists_ = false;
if (children_.size() == 0) {
auto child = parent->children_.begin();
for (child = parent->children_.begin(); child != parent->children_.end();
++child) {
if (child->first == character) {
parent->children_.erase(child);
break;
}
}
if (parent->children_.size() == 0 && !parent->exists_) {
parent->remove();
}
delete this;
}
}
void PathTrie::set_dictionary(fst::StdVectorFst* dictionary) {
dictionary_ = dictionary;
dictionary_state_ = dictionary->Start();
has_dictionary_ = true;
}
using FSTMATCH = fst::SortedMatcher<fst::StdVectorFst>;
void PathTrie::set_matcher(std::shared_ptr<FSTMATCH> matcher) {
matcher_ = matcher;
}
deep_speech_2/decoders/swig/path_trie.h 已删除 100644 → 0
浏览文件 @ 788f7011
#ifndef PATH_TRIE_H
#define PATH_TRIE_H
#include <algorithm>
#include <limits>
#include <memory>
#include <utility>
#include <vector>
#include "fst/fstlib.h"
/* Trie tree for prefix storing and manipulating, with a dictionary in
* finite-state transducer for spelling correction.
*/
class PathTrie {
public:
PathTrie();
~PathTrie();
// get new prefix after appending new char
PathTrie* get_path_trie(int new_char, bool reset = true);
// get the prefix in index from root to current node
PathTrie* get_path_vec(std::vector<int>& output);
// get the prefix in index from some stop node to current nodel
PathTrie* get_path_vec(std::vector<int>& output,
int stop,
size_t max_steps = std::numeric_limits<size_t>::max());
// update log probs
void iterate_to_vec(std::vector<PathTrie*>& output);
// set dictionary for FST
void set_dictionary(fst::StdVectorFst* dictionary);
void set_matcher(std::shared_ptr<fst::SortedMatcher<fst::StdVectorFst>>);
bool is_empty() { return ROOT_ == character; }
// remove current path from root
void remove();
float log_prob_b_prev;
float log_prob_nb_prev;
float log_prob_b_cur;
float log_prob_nb_cur;
float score;
float approx_ctc;
int character;
PathTrie* parent;
private:
int ROOT_;
bool exists_;
bool has_dictionary_;
std::vector<std::pair<int, PathTrie*>> children_;
// pointer to dictionary of FST
fst::StdVectorFst* dictionary_;
fst::StdVectorFst::StateId dictionary_state_;
// true if finding ars in FST
std::shared_ptr<fst::SortedMatcher<fst::StdVectorFst>> matcher_;
};
#endif // PATH_TRIE_H
deep_speech_2/decoders/swig/scorer.cpp 已删除 100644 → 0
浏览文件 @ 788f7011
#include "scorer.h"
#include <unistd.h>
#include <iostream>
#include "lm/config.hh"
#include "lm/model.hh"
#include "lm/state.hh"
#include "util/string_piece.hh"
#include "util/tokenize_piece.hh"
#include "decoder_utils.h"
using namespace lm::ngram;
Scorer::Scorer(double alpha,
double beta,
const std::string& lm_path,
const std::vector<std::string>& vocab_list) {
this->alpha = alpha;
this->beta = beta;
dictionary = nullptr;
is_character_based_ = true;
language_model_ = nullptr;
max_order_ = 0;
dict_size_ = 0;
SPACE_ID_ = -1;
setup(lm_path, vocab_list);
}
Scorer::~Scorer() {
if (language_model_ != nullptr) {
delete static_cast<lm::base::Model*>(language_model_);
}
if (dictionary != nullptr) {
delete static_cast<fst::StdVectorFst*>(dictionary);
}
}
void Scorer::setup(const std::string& lm_path,
const std::vector<std::string>& vocab_list) {
// load language model
load_lm(lm_path);
// set char map for scorer
set_char_map(vocab_list);
// fill the dictionary for FST
if (!is_character_based()) {
fill_dictionary(true);
}
}
void Scorer::load_lm(const std::string& lm_path) {
const char* filename = lm_path.c_str();
VALID_CHECK_EQ(access(filename, F_OK), 0, "Invalid language model path");
RetriveStrEnumerateVocab enumerate;
lm::ngram::Config config;
config.enumerate_vocab = &enumerate;
language_model_ = lm::ngram::LoadVirtual(filename, config);
max_order_ = static_cast<lm::base::Model*>(language_model_)->Order();
vocabulary_ = enumerate.vocabulary;
for (size_t i = 0; i < vocabulary_.size(); ++i) {
if (is_character_based_ && vocabulary_[i] != UNK_TOKEN &&
vocabulary_[i] != START_TOKEN && vocabulary_[i] != END_TOKEN &&
get_utf8_str_len(enumerate.vocabulary[i]) > 1) {
is_character_based_ = false;
}
}
}
double Scorer::get_log_cond_prob(const std::vector<std::string>& words) {
lm::base::Model* model = static_cast<lm::base::Model*>(language_model_);
double cond_prob;
lm::ngram::State state, tmp_state, out_state;
// avoid to inserting <s> in begin
model->NullContextWrite(&state);
for (size_t i = 0; i < words.size(); ++i) {
lm::WordIndex word_index = model->BaseVocabulary().Index(words[i]);
// encounter OOV
if (word_index == 0) {
return OOV_SCORE;
}
cond_prob = model->BaseScore(&state, word_index, &out_state);
tmp_state = state;
state = out_state;
out_state = tmp_state;
}
// return log10 prob
return cond_prob;
}
double Scorer::get_sent_log_prob(const std::vector<std::string>& words) {
std::vector<std::string> sentence;
if (words.size() == 0) {
for (size_t i = 0; i < max_order_; ++i) {
sentence.push_back(START_TOKEN);
}
} else {
for (size_t i = 0; i < max_order_ - 1; ++i) {
sentence.push_back(START_TOKEN);
}
sentence.insert(sentence.end(), words.begin(), words.end());
}
sentence.push_back(END_TOKEN);
return get_log_prob(sentence);
}
double Scorer::get_log_prob(const std::vector<std::string>& words) {
assert(words.size() > max_order_);
double score = 0.0;
for (size_t i = 0; i < words.size() - max_order_ + 1; ++i) {
std::vector<std::string> ngram(words.begin() + i,
words.begin() + i + max_order_);
score += get_log_cond_prob(ngram);
}
return score;
}
void Scorer::reset_params(float alpha, float beta) {
this->alpha = alpha;
this->beta = beta;
}
std::string Scorer::vec2str(const std::vector<int>& input) {
std::string word;
for (auto ind : input) {
word += char_list_[ind];
}
return word;
}
std::vector<std::string> Scorer::split_labels(const std::vector<int>& labels) {
if (labels.empty()) return {};
std::string s = vec2str(labels);
std::vector<std::string> words;
if (is_character_based_) {
words = split_utf8_str(s);
} else {
words = split_str(s, " ");
}
return words;
}
void Scorer::set_char_map(const std::vector<std::string>& char_list) {
char_list_ = char_list;
char_map_.clear();
for (size_t i = 0; i < char_list_.size(); i++) {
if (char_list_[i] == " ") {
SPACE_ID_ = i;
char_map_[' '] = i;
} else if (char_list_[i].size() == 1) {
char_map_[char_list_[i][0]] = i;
}
}
}
std::vector<std::string> Scorer::make_ngram(PathTrie* prefix) {
std::vector<std::string> ngram;
PathTrie* current_node = prefix;
PathTrie* new_node = nullptr;
for (int order = 0; order < max_order_; order++) {
std::vector<int> prefix_vec;
if (is_character_based_) {
new_node = current_node->get_path_vec(prefix_vec, SPACE_ID_, 1);
current_node = new_node;
} else {
new_node = current_node->get_path_vec(prefix_vec, SPACE_ID_);
current_node = new_node->parent; // Skipping spaces
}
// reconstruct word
std::string word = vec2str(prefix_vec);
ngram.push_back(word);
if (new_node->character == -1) {
// No more spaces, but still need order
for (int i = 0; i < max_order_ - order - 1; i++) {
ngram.push_back(START_TOKEN);
}
break;
}
}
std::reverse(ngram.begin(), ngram.end());
return ngram;
}
void Scorer::fill_dictionary(bool add_space) {
fst::StdVectorFst dictionary;
// First reverse char_list so ints can be accessed by chars
std::unordered_map<std::string, int> char_map;
for (size_t i = 0; i < char_list_.size(); i++) {
char_map[char_list_[i]] = i;
}
// For each unigram convert to ints and put in trie
int dict_size = 0;
for (const auto& word : vocabulary_) {
bool added = add_word_to_dictionary(
word, char_map, add_space, SPACE_ID_, &dictionary);
dict_size += added ? 1 : 0;
}
dict_size_ = dict_size;
/* Simplify FST
* This gets rid of "epsilon" transitions in the FST.
* These are transitions that don't require a string input to be taken.
* Getting rid of them is necessary to make the FST determinisitc, but
* can greatly increase the size of the FST
*/
fst::RmEpsilon(&dictionary);
fst::StdVectorFst* new_dict = new fst::StdVectorFst;
/* This makes the FST deterministic, meaning for any string input there's
* only one possible state the FST could be in. It is assumed our
* dictionary is deterministic when using it.
* (lest we'd have to check for multiple transitions at each state)
*/
fst::Determinize(dictionary, new_dict);
/* Finds the simplest equivalent fst. This is unnecessary but decreases
* memory usage of the dictionary
*/
fst::Minimize(new_dict);
this->dictionary = new_dict;
}
deep_speech_2/decoders/swig/scorer.h 已删除 100644 → 0
浏览文件 @ 788f7011
#ifndef SCORER_H_
#define SCORER_H_
#include <memory>
#include <string>
#include <unordered_map>
#include <vector>
#include "lm/enumerate_vocab.hh"
#include "lm/virtual_interface.hh"
#include "lm/word_index.hh"
#include "util/string_piece.hh"
#include "path_trie.h"
const double OOV_SCORE = -1000.0;
const std::string START_TOKEN = "<s>";
const std::string UNK_TOKEN = "<unk>";
const std::string END_TOKEN = "</s>";
// Implement a callback to retrive the dictionary of language model.
class RetriveStrEnumerateVocab : public lm::EnumerateVocab {
public:
RetriveStrEnumerateVocab() {}
void Add(lm::WordIndex index, const StringPiece &str) {
vocabulary.push_back(std::string(str.data(), str.length()));
}
std::vector<std::string> vocabulary;
};
/* External scorer to query score for n-gram or sentence, including language
* model scoring and word insertion.
*
* Example:
* Scorer scorer(alpha, beta, "path_of_language_model");
* scorer.get_log_cond_prob({ "WORD1", "WORD2", "WORD3" });
* scorer.get_sent_log_prob({ "WORD1", "WORD2", "WORD3" });
*/
class Scorer {
public:
Scorer(double alpha,
double beta,
const std::string &lm_path,
const std::vector<std::string> &vocabulary);
~Scorer();
double get_log_cond_prob(const std::vector<std::string> &words);
double get_sent_log_prob(const std::vector<std::string> &words);
// return the max order
size_t get_max_order() const { return max_order_; }
// return the dictionary size of language model
size_t get_dict_size() const { return dict_size_; }
// retrun true if the language model is character based
bool is_character_based() const { return is_character_based_; }
// reset params alpha & beta
void reset_params(float alpha, float beta);
// make ngram for a given prefix
std::vector<std::string> make_ngram(PathTrie *prefix);
// trransform the labels in index to the vector of words (word based lm) or
// the vector of characters (character based lm)
std::vector<std::string> split_labels(const std::vector<int> &labels);
// language model weight
double alpha;
// word insertion weight
double beta;
// pointer to the dictionary of FST
void *dictionary;
protected:
// necessary setup: load language model, set char map, fill FST's dictionary
void setup(const std::string &lm_path,
const std::vector<std::string> &vocab_list);
// load language model from given path
void load_lm(const std::string &lm_path);
// fill dictionary for FST
void fill_dictionary(bool add_space);
// set char map
void set_char_map(const std::vector<std::string> &char_list);
double get_log_prob(const std::vector<std::string> &words);
// translate the vector in index to string
std::string vec2str(const std::vector<int> &input);
private:
void *language_model_;
bool is_character_based_;
size_t max_order_;
size_t dict_size_;
int SPACE_ID_;
std::vector<std::string> char_list_;
std::unordered_map<char, int> char_map_;
std::vector<std::string> vocabulary_;
};
#endif // SCORER_H_
deep_speech_2/decoders/swig/setup.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Script to build and install decoder package."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
from setuptools import setup, Extension, distutils
import glob
import platform
import os, sys
import multiprocessing.pool
import argparse
parser = argparse.ArgumentParser(description=__doc__)
parser.add_argument(
"--num_processes",
default=1,
type=int,
help="Number of cpu processes to build package. (default: %(default)d)")
args = parser.parse_known_args()
# reconstruct sys.argv to pass to setup below
sys.argv = [sys.argv[0]] + args[1]
# monkey-patch for parallel compilation
# See: https://stackoverflow.com/a/13176803
def parallelCCompile(self,
sources,
output_dir=None,
macros=None,
include_dirs=None,
debug=0,
extra_preargs=None,
extra_postargs=None,
depends=None):
# those lines are copied from distutils.ccompiler.CCompiler directly
macros, objects, extra_postargs, pp_opts, build = self._setup_compile(
output_dir, macros, include_dirs, sources, depends, extra_postargs)
cc_args = self._get_cc_args(pp_opts, debug, extra_preargs)
# parallel code
def _single_compile(obj):
try:
src, ext = build[obj]
except KeyError:
return
self._compile(obj, src, ext, cc_args, extra_postargs, pp_opts)
# convert to list, imap is evaluated on-demand
thread_pool = multiprocessing.pool.ThreadPool(args[0].num_processes)
list(thread_pool.imap(_single_compile, objects))
return objects
def compile_test(header, library):
dummy_path = os.path.join(os.path.dirname(__file__), "dummy")
command = "bash -c \"g++ -include " + header \
+ " -l" + library + " -x c++ - <<<'int main() {}' -o " \
+ dummy_path + " >/dev/null 2>/dev/null && rm " \
+ dummy_path + " 2>/dev/null\""
return os.system(command) == 0
# hack compile to support parallel compiling
distutils.ccompiler.CCompiler.compile = parallelCCompile
FILES = glob.glob('kenlm/util/*.cc') \
+ glob.glob('kenlm/lm/*.cc') \
+ glob.glob('kenlm/util/double-conversion/*.cc')
FILES += glob.glob('openfst-1.6.3/src/lib/*.cc')
FILES = [
fn for fn in FILES
if not (fn.endswith('main.cc') or fn.endswith('test.cc') or fn.endswith(
'unittest.cc'))
]
LIBS = ['stdc++']
if platform.system() != 'Darwin':
LIBS.append('rt')
ARGS = ['-O3', '-DNDEBUG', '-DKENLM_MAX_ORDER=6', '-std=c++11']
if compile_test('zlib.h', 'z'):
ARGS.append('-DHAVE_ZLIB')
LIBS.append('z')
if compile_test('bzlib.h', 'bz2'):
ARGS.append('-DHAVE_BZLIB')
LIBS.append('bz2')
if compile_test('lzma.h', 'lzma'):
ARGS.append('-DHAVE_XZLIB')
LIBS.append('lzma')
os.system('swig -python -c++ ./decoders.i')
decoders_module = [
Extension(
name='_swig_decoders',
sources=FILES + glob.glob('*.cxx') + glob.glob('*.cpp'),
language='c++',
include_dirs=[
'.',
'kenlm',
'openfst-1.6.3/src/include',
'ThreadPool',
],
libraries=LIBS,
extra_compile_args=ARGS)
]
setup(
name='swig_decoders',
version='1.0',
description="""CTC decoders""",
ext_modules=decoders_module,
py_modules=['swig_decoders'], )
deep_speech_2/decoders/swig/setup.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#!/usr/bin/env bash
if [ ! -d kenlm ]; then
git clone https://github.com/luotao1/kenlm.git
echo -e "\n"
fi
if [ ! -d openfst-1.6.3 ]; then
echo "Download and extract openfst ..."
wget http://www.openfst.org/twiki/pub/FST/FstDownload/openfst-1.6.3.tar.gz
tar -xzvf openfst-1.6.3.tar.gz
echo -e "\n"
fi
if [ ! -d ThreadPool ]; then
git clone https://github.com/progschj/ThreadPool.git
echo -e "\n"
fi
echo "Install decoders ..."
python setup.py install --num_processes 4
deep_speech_2/decoders/swig_wrapper.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Wrapper for various CTC decoders in SWIG."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import swig_decoders
class Scorer(swig_decoders.Scorer):
"""Wrapper for Scorer.
:param alpha: Parameter associated with language model. Don't use
language model when alpha = 0.
:type alpha: float
:param beta: Parameter associated with word count. Don't use word
count when beta = 0.
:type beta: float
:model_path: Path to load language model.
:type model_path: basestring
"""
def __init__(self, alpha, beta, model_path, vocabulary):
swig_decoders.Scorer.__init__(self, alpha, beta, model_path, vocabulary)
def ctc_greedy_decoder(probs_seq, vocabulary):
"""Wrapper for ctc best path decoder in swig.
:param probs_seq: 2-D list of probability distributions over each time
step, with each element being a list of normalized
probabilities over vocabulary and blank.
:type probs_seq: 2-D list
:param vocabulary: Vocabulary list.
:type vocabulary: list
:return: Decoding result string.
:rtype: basestring
"""
result = swig_decoders.ctc_greedy_decoder(probs_seq.tolist(), vocabulary)
return result.decode('utf-8')
def ctc_beam_search_decoder(probs_seq,
vocabulary,
beam_size,
cutoff_prob=1.0,
cutoff_top_n=40,
ext_scoring_func=None):
"""Wrapper for the CTC Beam Search Decoder.
:param probs_seq: 2-D list of probability distributions over each time
step, with each element being a list of normalized
probabilities over vocabulary and blank.
:type probs_seq: 2-D list
:param vocabulary: Vocabulary list.
:type vocabulary: list
:param beam_size: Width for beam search.
:type beam_size: int
:param cutoff_prob: Cutoff probability in pruning,
default 1.0, no pruning.
:type cutoff_prob: float
:param cutoff_top_n: Cutoff number in pruning, only top cutoff_top_n
characters with highest probs in vocabulary will be
used in beam search, default 40.
:type cutoff_top_n: int
:param ext_scoring_func: External scoring function for
partially decoded sentence, e.g. word count
or language model.
:type external_scoring_func: callable
:return: List of tuples of log probability and sentence as decoding
results, in descending order of the probability.
:rtype: list
"""
beam_results = swig_decoders.ctc_beam_search_decoder(
probs_seq.tolist(), vocabulary, beam_size, cutoff_prob, cutoff_top_n,
ext_scoring_func)
beam_results = [(res[0], res[1].decode('utf-8')) for res in beam_results]
return beam_results
def ctc_beam_search_decoder_batch(probs_split,
vocabulary,
beam_size,
num_processes,
cutoff_prob=1.0,
cutoff_top_n=40,
ext_scoring_func=None):
"""Wrapper for the batched CTC beam search decoder.
:param probs_seq: 3-D list with each element as an instance of 2-D list
of probabilities used by ctc_beam_search_decoder().
:type probs_seq: 3-D list
:param vocabulary: Vocabulary list.
:type vocabulary: list
:param beam_size: Width for beam search.
:type beam_size: int
:param num_processes: Number of parallel processes.
:type num_processes: int
:param cutoff_prob: Cutoff probability in vocabulary pruning,
default 1.0, no pruning.
:type cutoff_prob: float
:param cutoff_top_n: Cutoff number in pruning, only top cutoff_top_n
characters with highest probs in vocabulary will be
used in beam search, default 40.
:type cutoff_top_n: int
:param num_processes: Number of parallel processes.
:type num_processes: int
:param ext_scoring_func: External scoring function for
partially decoded sentence, e.g. word count
or language model.
:type external_scoring_function: callable
:return: List of tuples of log probability and sentence as decoding
results, in descending order of the probability.
:rtype: list
"""
probs_split = [probs_seq.tolist() for probs_seq in probs_split]
batch_beam_results = swig_decoders.ctc_beam_search_decoder_batch(
probs_split, vocabulary, beam_size, num_processes, cutoff_prob,
cutoff_top_n, ext_scoring_func)
batch_beam_results = [
[(res[0], res[1].decode("utf-8")) for res in beam_results]
for beam_results in batch_beam_results
]
return batch_beam_results
deep_speech_2/decoders/tests/test_decoders.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Test decoders."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import unittest
from decoders import decoders_deprecated as decoder
class TestDecoders(unittest.TestCase):
def setUp(self):
self.vocab_list = ["\'", ' ', 'a', 'b', 'c', 'd']
self.beam_size = 20
self.probs_seq1 = [[
0.06390443, 0.21124858, 0.27323887, 0.06870235, 0.0361254,
0.18184413, 0.16493624
], [
0.03309247, 0.22866108, 0.24390638, 0.09699597, 0.31895462,
0.0094893, 0.06890021
], [
0.218104, 0.19992557, 0.18245131, 0.08503348, 0.14903535,
0.08424043, 0.08120984
], [
0.12094152, 0.19162472, 0.01473646, 0.28045061, 0.24246305,
0.05206269, 0.09772094
], [
0.1333387, 0.00550838, 0.00301669, 0.21745861, 0.20803985,
0.41317442, 0.01946335
], [
0.16468227, 0.1980699, 0.1906545, 0.18963251, 0.19860937,
0.04377724, 0.01457421
]]
self.probs_seq2 = [[
0.08034842, 0.22671944, 0.05799633, 0.36814645, 0.11307441,
0.04468023, 0.10903471
], [
0.09742457, 0.12959763, 0.09435383, 0.21889204, 0.15113123,
0.10219457, 0.20640612
], [
0.45033529, 0.09091417, 0.15333208, 0.07939558, 0.08649316,
0.12298585, 0.01654384
], [
0.02512238, 0.22079203, 0.19664364, 0.11906379, 0.07816055,
0.22538587, 0.13483174
], [
0.17928453, 0.06065261, 0.41153005, 0.1172041, 0.11880313,
0.07113197, 0.04139363
], [
0.15882358, 0.1235788, 0.23376776, 0.20510435, 0.00279306,
0.05294827, 0.22298418
]]
self.greedy_result = ["ac'bdc", "b'da"]
self.beam_search_result = ['acdc', "b'a"]
def test_greedy_decoder_1(self):
bst_result = decoder.ctc_greedy_decoder(self.probs_seq1,
self.vocab_list)
self.assertEqual(bst_result, self.greedy_result[0])
def test_greedy_decoder_2(self):
bst_result = decoder.ctc_greedy_decoder(self.probs_seq2,
self.vocab_list)
self.assertEqual(bst_result, self.greedy_result[1])
def test_beam_search_decoder_1(self):
beam_result = decoder.ctc_beam_search_decoder(
probs_seq=self.probs_seq1,
beam_size=self.beam_size,
vocabulary=self.vocab_list)
self.assertEqual(beam_result[0][1], self.beam_search_result[0])
def test_beam_search_decoder_2(self):
beam_result = decoder.ctc_beam_search_decoder(
probs_seq=self.probs_seq2,
beam_size=self.beam_size,
vocabulary=self.vocab_list)
self.assertEqual(beam_result[0][1], self.beam_search_result[1])
def test_beam_search_decoder_batch(self):
beam_results = decoder.ctc_beam_search_decoder_batch(
probs_split=[self.probs_seq1, self.probs_seq2],
beam_size=self.beam_size,
vocabulary=self.vocab_list,
num_processes=24)
self.assertEqual(beam_results[0][0][1], self.beam_search_result[0])
self.assertEqual(beam_results[1][0][1], self.beam_search_result[1])
if __name__ == '__main__':
unittest.main()
deep_speech_2/deploy/_init_paths.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Set up paths for DS2"""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import os.path
import sys
def add_path(path):
if path not in sys.path:
sys.path.insert(0, path)
this_dir = os.path.dirname(__file__)
# Add project path to PYTHONPATH
proj_path = os.path.join(this_dir, '..')
add_path(proj_path)
deep_speech_2/deploy/demo_client.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Client-end for the ASR demo."""
from pynput import keyboard
import struct
import socket
import sys
import argparse
import pyaudio
parser = argparse.ArgumentParser(description=__doc__)
parser.add_argument(
"--host_ip",
default="localhost",
type=str,
help="Server IP address. (default: %(default)s)")
parser.add_argument(
"--host_port",
default=8086,
type=int,
help="Server Port. (default: %(default)s)")
args = parser.parse_args()
is_recording = False
enable_trigger_record = True
def on_press(key):
"""On-press keyboard callback function."""
global is_recording, enable_trigger_record
if key == keyboard.Key.space:
if (not is_recording) and enable_trigger_record:
sys.stdout.write("Start Recording ... ")
sys.stdout.flush()
is_recording = True
def on_release(key):
"""On-release keyboard callback function."""
global is_recording, enable_trigger_record
if key == keyboard.Key.esc:
return False
elif key == keyboard.Key.space:
if is_recording == True:
is_recording = False
data_list = []
def callback(in_data, frame_count, time_info, status):
"""Audio recorder's stream callback function."""
global data_list, is_recording, enable_trigger_record
if is_recording:
data_list.append(in_data)
enable_trigger_record = False
elif len(data_list) > 0:
# Connect to server and send data
sock = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
sock.connect((args.host_ip, args.host_port))
sent = ''.join(data_list)
sock.sendall(struct.pack('>i', len(sent)) + sent)
print('Speech[length=%d] Sent.' % len(sent))
# Receive data from the server and shut down
received = sock.recv(1024)
print "Recognition Results: {}".format(received)
sock.close()
data_list = []
enable_trigger_record = True
return (in_data, pyaudio.paContinue)
def main():
# prepare audio recorder
p = pyaudio.PyAudio()
stream = p.open(
format=pyaudio.paInt32,
channels=1,
rate=16000,
input=True,
stream_callback=callback)
stream.start_stream()
# prepare keyboard listener
with keyboard.Listener(
on_press=on_press, on_release=on_release) as listener:
listener.join()
# close up
stream.stop_stream()
stream.close()
p.terminate()
if __name__ == "__main__":
main()
deep_speech_2/deploy/demo_server.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Server-end for the ASR demo."""
import os
import time
import random
import argparse
import functools
from time import gmtime, strftime
import SocketServer
import struct
import wave
import paddle.v2 as paddle
import _init_paths
from data_utils.data import DataGenerator
from model_utils.model import DeepSpeech2Model
from data_utils.utility import read_manifest
from utils.utility import add_arguments, print_arguments
parser = argparse.ArgumentParser(description=__doc__)
add_arg = functools.partial(add_arguments, argparser=parser)
# yapf: disable
add_arg('host_port', int, 8086, "Server's IP port.")
add_arg('beam_size', int, 500, "Beam search width.")
add_arg('num_conv_layers', int, 2, "# of convolution layers.")
add_arg('num_rnn_layers', int, 3, "# of recurrent layers.")
add_arg('rnn_layer_size', int, 2048, "# of recurrent cells per layer.")
add_arg('alpha', float, 2.15, "Coef of LM for beam search.")
add_arg('beta', float, 0.35, "Coef of WC for beam search.")
add_arg('cutoff_prob', float, 1.0, "Cutoff probability for pruning.")
add_arg('cutoff_top_n', int, 40, "Cutoff number for pruning.")
add_arg('use_gru', bool, False, "Use GRUs instead of simple RNNs.")
add_arg('use_gpu', bool, True, "Use GPU or not.")
add_arg('share_rnn_weights',bool, True, "Share input-hidden weights across "
"bi-directional RNNs. Not for GRU.")
add_arg('host_ip', str,
'localhost',
"Server's IP address.")
add_arg('speech_save_dir', str,
'demo_cache',
"Directory to save demo audios.")
add_arg('warmup_manifest', str,
'data/librispeech/manifest.test-clean',
"Filepath of manifest to warm up.")
add_arg('mean_std_path', str,
'data/librispeech/mean_std.npz',
"Filepath of normalizer's mean & std.")
add_arg('vocab_path', str,
'data/librispeech/eng_vocab.txt',
"Filepath of vocabulary.")
add_arg('model_path', str,
'./checkpoints/libri/params.latest.tar.gz',
"If None, the training starts from scratch, "
"otherwise, it resumes from the pre-trained model.")
add_arg('lang_model_path', str,
'lm/data/common_crawl_00.prune01111.trie.klm',
"Filepath for language model.")
add_arg('decoding_method', str,
'ctc_beam_search',
"Decoding method. Options: ctc_beam_search, ctc_greedy",
choices = ['ctc_beam_search', 'ctc_greedy'])
add_arg('specgram_type', str,
'linear',
"Audio feature type. Options: linear, mfcc.",
choices=['linear', 'mfcc'])
# yapf: disable
args = parser.parse_args()
class AsrTCPServer(SocketServer.TCPServer):
"""The ASR TCP Server."""
def __init__(self,
server_address,
RequestHandlerClass,
speech_save_dir,
audio_process_handler,
bind_and_activate=True):
self.speech_save_dir = speech_save_dir
self.audio_process_handler = audio_process_handler
SocketServer.TCPServer.__init__(
self, server_address, RequestHandlerClass, bind_and_activate=True)
class AsrRequestHandler(SocketServer.BaseRequestHandler):
"""The ASR request handler."""
def handle(self):
# receive data through TCP socket
chunk = self.request.recv(1024)
target_len = struct.unpack('>i', chunk[:4])[0]
data = chunk[4:]
while len(data) < target_len:
chunk = self.request.recv(1024)
data += chunk
# write to file
filename = self._write_to_file(data)
print("Received utterance[length=%d] from %s, saved to %s." %
(len(data), self.client_address[0], filename))
start_time = time.time()
transcript = self.server.audio_process_handler(filename)
finish_time = time.time()
print("Response Time: %f, Transcript: %s" %
(finish_time - start_time, transcript))
self.request.sendall(transcript.encode('utf-8'))
def _write_to_file(self, data):
# prepare save dir and filename
if not os.path.exists(self.server.speech_save_dir):
os.mkdir(self.server.speech_save_dir)
timestamp = strftime("%Y%m%d%H%M%S", gmtime())
out_filename = os.path.join(
self.server.speech_save_dir,
timestamp + "_" + self.client_address[0] + ".wav")
# write to wav file
file = wave.open(out_filename, 'wb')
file.setnchannels(1)
file.setsampwidth(4)
file.setframerate(16000)
file.writeframes(data)
file.close()
return out_filename
def warm_up_test(audio_process_handler,
manifest_path,
num_test_cases,
random_seed=0):
"""Warming-up test."""
manifest = read_manifest(manifest_path)
rng = random.Random(random_seed)
samples = rng.sample(manifest, num_test_cases)
for idx, sample in enumerate(samples):
print("Warm-up Test Case %d: %s", idx, sample['audio_filepath'])
start_time = time.time()
transcript = audio_process_handler(sample['audio_filepath'])
finish_time = time.time()
print("Response Time: %f, Transcript: %s" %
(finish_time - start_time, transcript))
def start_server():
"""Start the ASR server"""
# prepare data generator
data_generator = DataGenerator(
vocab_filepath=args.vocab_path,
mean_std_filepath=args.mean_std_path,
augmentation_config='{}',
specgram_type=args.specgram_type,
num_threads=1,
keep_transcription_text=True)
# prepare ASR model
ds2_model = DeepSpeech2Model(
vocab_size=data_generator.vocab_size,
num_conv_layers=args.num_conv_layers,
num_rnn_layers=args.num_rnn_layers,
rnn_layer_size=args.rnn_layer_size,
use_gru=args.use_gru,
pretrained_model_path=args.model_path,
share_rnn_weights=args.share_rnn_weights)
vocab_list = [chars.encode("utf-8") for chars in data_generator.vocab_list]
# prepare ASR inference handler
def file_to_transcript(filename):
feature = data_generator.process_utterance(filename, "")
result_transcript = ds2_model.infer_batch(
infer_data=[feature],
decoding_method=args.decoding_method,
beam_alpha=args.alpha,
beam_beta=args.beta,
beam_size=args.beam_size,
cutoff_prob=args.cutoff_prob,
cutoff_top_n=args.cutoff_top_n,
vocab_list=vocab_list,
language_model_path=args.lang_model_path,
num_processes=1)
return result_transcript[0]
# warming up with utterrances sampled from Librispeech
print('-----------------------------------------------------------')
print('Warming up ...')
warm_up_test(
audio_process_handler=file_to_transcript,
manifest_path=args.warmup_manifest,
num_test_cases=3)
print('-----------------------------------------------------------')
# start the server
server = AsrTCPServer(
server_address=(args.host_ip, args.host_port),
RequestHandlerClass=AsrRequestHandler,
speech_save_dir=args.speech_save_dir,
audio_process_handler=file_to_transcript)
print("ASR Server Started.")
server.serve_forever()
def main():
print_arguments(args)
paddle.init(use_gpu=args.use_gpu, trainer_count=1)
start_server()
if __name__ == "__main__":
main()
deep_speech_2/examples/aishell/run_data.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
cd ../.. > /dev/null
# download data, generate manifests
PYTHONPATH=.:$PYTHONPATH python data/aishell/aishell.py \
--manifest_prefix='data/aishell/manifest' \
--target_dir='~/.cache/paddle/dataset/speech/Aishell'
if [ $? -ne 0 ]; then
echo "Prepare Aishell failed. Terminated."
exit 1
fi
# build vocabulary
python tools/build_vocab.py \
--count_threshold=0 \
--vocab_path='data/aishell/vocab.txt' \
--manifest_paths 'data/aishell/manifest.train' 'data/aishell/manifest.dev'
if [ $? -ne 0 ]; then
echo "Build vocabulary failed. Terminated."
exit 1
fi
# compute mean and stddev for normalizer
python tools/compute_mean_std.py \
--manifest_path='data/aishell/manifest.train' \
--num_samples=2000 \
--specgram_type='linear' \
--output_path='data/aishell/mean_std.npz'
if [ $? -ne 0 ]; then
echo "Compute mean and stddev failed. Terminated."
exit 1
fi
echo "Aishell data preparation done."
exit 0
deep_speech_2/examples/aishell/run_infer.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
cd ../.. > /dev/null
# download language model
cd models/lm > /dev/null
sh download_lm_ch.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# infer
CUDA_VISIBLE_DEVICES=0 \
python -u infer.py \
--num_samples=10 \
--trainer_count=1 \
--beam_size=300 \
--num_proc_bsearch=8 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=1024 \
--alpha=1.4 \
--beta=2.4 \
--cutoff_prob=0.99 \
--cutoff_top_n=40 \
--use_gru=False \
--use_gpu=True \
--share_rnn_weights=False \
--infer_manifest='data/aishell/manifest.test' \
--mean_std_path='data/aishell/mean_std.npz' \
--vocab_path='data/aishell/vocab.txt' \
--model_path='checkpoints/aishell/params.latest.tar.gz' \
--lang_model_path='models/lm/zh_giga.no_cna_cmn.prune01244.klm' \
--decoding_method='ctc_beam_search' \
--error_rate_type='cer' \
--specgram_type='linear'
if [ $? -ne 0 ]; then
echo "Failed in inference!"
exit 1
fi
exit 0
deep_speech_2/examples/aishell/run_infer_golden.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
cd ../.. > /dev/null
# download language model
cd models/lm > /dev/null
sh download_lm_ch.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# download well-trained model
cd models/aishell > /dev/null
sh download_model.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# infer
CUDA_VISIBLE_DEVICES=0 \
python -u infer.py \
--num_samples=10 \
--trainer_count=1 \
--beam_size=300 \
--num_proc_bsearch=8 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=1024 \
--alpha=1.4 \
--beta=2.4 \
--cutoff_prob=0.99 \
--cutoff_top_n=40 \
--use_gru=False \
--use_gpu=True \
--share_rnn_weights=False \
--infer_manifest='data/aishell/manifest.test' \
--mean_std_path='models/aishell/mean_std.npz' \
--vocab_path='models/aishell/vocab.txt' \
--model_path='models/aishell/params.tar.gz' \
--lang_model_path='models/lm/zh_giga.no_cna_cmn.prune01244.klm' \
--decoding_method='ctc_beam_search' \
--error_rate_type='cer' \
--specgram_type='linear'
if [ $? -ne 0 ]; then
echo "Failed in inference!"
exit 1
fi
exit 0
deep_speech_2/examples/aishell/run_test.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
cd ../.. > /dev/null
# download language model
cd models/lm > /dev/null
sh download_lm_ch.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# evaluate model
CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 \
python -u test.py \
--batch_size=128 \
--trainer_count=8 \
--beam_size=300 \
--num_proc_bsearch=8 \
--num_proc_data=8 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=1024 \
--alpha=1.4 \
--beta=2.4 \
--cutoff_prob=0.99 \
--cutoff_top_n=40 \
--use_gru=False \
--use_gpu=True \
--share_rnn_weights=False \
--test_manifest='data/aishell/manifest.test' \
--mean_std_path='data/aishell/mean_std.npz' \
--vocab_path='data/aishell/vocab.txt' \
--model_path='checkpoints/aishell/params.latest.tar.gz' \
--lang_model_path='models/lm/zh_giga.no_cna_cmn.prune01244.klm' \
--decoding_method='ctc_beam_search' \
--error_rate_type='cer' \
--specgram_type='linear'
if [ $? -ne 0 ]; then
echo "Failed in evaluation!"
exit 1
fi
exit 0
deep_speech_2/examples/aishell/run_test_golden.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
cd ../.. > /dev/null
# download language model
cd models/lm > /dev/null
sh download_lm_ch.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# download well-trained model
cd models/aishell > /dev/null
sh download_model.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# evaluate model
CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 \
python -u test.py \
--batch_size=128 \
--trainer_count=8 \
--beam_size=300 \
--num_proc_bsearch=8 \
--num_proc_data=8 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=1024 \
--alpha=1.4 \
--beta=2.4 \
--cutoff_prob=0.99 \
--cutoff_top_n=40 \
--use_gru=False \
--use_gpu=True \
--share_rnn_weights=False \
--test_manifest='data/aishell/manifest.test' \
--mean_std_path='models/aishell/mean_std.npz' \
--vocab_path='models/aishell/vocab.txt' \
--model_path='models/aishell/params.tar.gz' \
--lang_model_path='models/lm/zh_giga.no_cna_cmn.prune01244.klm' \
--decoding_method='ctc_beam_search' \
--error_rate_type='cer' \
--specgram_type='linear'
if [ $? -ne 0 ]; then
echo "Failed in evaluation!"
exit 1
fi
exit 0
deep_speech_2/examples/aishell/run_train.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
cd ../.. > /dev/null
# train model
# if you wish to resume from an exists model, uncomment --init_model_path
CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 \
python -u train.py \
--batch_size=64 \
--trainer_count=8 \
--num_passes=50 \
--num_proc_data=16 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=1024 \
--num_iter_print=100 \
--learning_rate=5e-4 \
--max_duration=27.0 \
--min_duration=0.0 \
--test_off=False \
--use_sortagrad=True \
--use_gru=False \
--use_gpu=True \
--is_local=True \
--share_rnn_weights=False \
--train_manifest='data/aishell/manifest.train' \
--dev_manifest='data/aishell/manifest.dev' \
--mean_std_path='data/aishell/mean_std.npz' \
--vocab_path='data/aishell/vocab.txt' \
--output_model_dir='./checkpoints/aishell' \
--augment_conf_path='conf/augmentation.config' \
--specgram_type='linear' \
--shuffle_method='batch_shuffle_clipped'
if [ $? -ne 0 ]; then
echo "Failed in training!"
exit 1
fi
exit 0
deep_speech_2/examples/deploy_demo/run_demo_client.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
cd ../.. > /dev/null
# start demo client
CUDA_VISIBLE_DEVICES=0 \
python -u deploy/demo_client.py \
--host_ip='localhost' \
--host_port=8086 \
if [ $? -ne 0 ]; then
echo "Failed in starting demo client!"
exit 1
fi
exit 0
deep_speech_2/examples/deploy_demo/run_english_demo_server.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
# TODO: replace the model with a mandarin model
cd ../.. > /dev/null
# download language model
cd models/lm > /dev/null
sh download_lm_en.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# download well-trained model
cd models/baidu_en8k > /dev/null
sh download_model.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# start demo server
CUDA_VISIBLE_DEVICES=0 \
python -u deploy/demo_server.py \
--host_ip='localhost' \
--host_port=8086 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=1024 \
--alpha=1.15 \
--beta=0.15 \
--cutoff_prob=1.0 \
--cutoff_top_n=40 \
--use_gru=True \
--use_gpu=True \
--share_rnn_weights=False \
--speech_save_dir='demo_cache' \
--warmup_manifest='data/tiny/manifest.test-clean' \
--mean_std_path='models/baidu_en8k/mean_std.npz' \
--vocab_path='models/baidu_en8k/vocab.txt' \
--model_path='models/baidu_en8k/params.tar.gz' \
--lang_model_path='models/lm/common_crawl_00.prune01111.trie.klm' \
--decoding_method='ctc_beam_search' \
--specgram_type='linear'
if [ $? -ne 0 ]; then
echo "Failed in starting demo server!"
exit 1
fi
exit 0
deep_speech_2/examples/librispeech/run_data.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
cd ../.. > /dev/null
# download data, generate manifests
PYTHONPATH=.:$PYTHONPATH python data/librispeech/librispeech.py \
--manifest_prefix='data/librispeech/manifest' \
--target_dir='~/.cache/paddle/dataset/speech/Libri' \
--full_download='True'
if [ $? -ne 0 ]; then
echo "Prepare LibriSpeech failed. Terminated."
exit 1
fi
cat data/librispeech/manifest.train-* | shuf > data/librispeech/manifest.train
# build vocabulary
python tools/build_vocab.py \
--count_threshold=0 \
--vocab_path='data/librispeech/vocab.txt' \
--manifest_paths='data/librispeech/manifest.train'
if [ $? -ne 0 ]; then
echo "Build vocabulary failed. Terminated."
exit 1
fi
# compute mean and stddev for normalizer
python tools/compute_mean_std.py \
--manifest_path='data/librispeech/manifest.train' \
--num_samples=2000 \
--specgram_type='linear' \
--output_path='data/librispeech/mean_std.npz'
if [ $? -ne 0 ]; then
echo "Compute mean and stddev failed. Terminated."
exit 1
fi
echo "LibriSpeech Data preparation done."
exit 0
deep_speech_2/examples/librispeech/run_infer.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
cd ../.. > /dev/null
# download language model
cd models/lm > /dev/null
sh download_lm_en.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# infer
CUDA_VISIBLE_DEVICES=0 \
python -u infer.py \
--num_samples=10 \
--trainer_count=1 \
--beam_size=500 \
--num_proc_bsearch=8 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=2048 \
--alpha=2.15 \
--beta=0.35 \
--cutoff_prob=1.0 \
--cutoff_top_n=40 \
--use_gru=False \
--use_gpu=True \
--share_rnn_weights=True \
--infer_manifest='data/librispeech/manifest.test-clean' \
--mean_std_path='data/librispeech/mean_std.npz' \
--vocab_path='data/librispeech/vocab.txt' \
--model_path='checkpoints/libri/params.latest.tar.gz' \
--lang_model_path='models/lm/common_crawl_00.prune01111.trie.klm' \
--decoding_method='ctc_beam_search' \
--error_rate_type='wer' \
--specgram_type='linear'
if [ $? -ne 0 ]; then
echo "Failed in inference!"
exit 1
fi
exit 0
deep_speech_2/examples/librispeech/run_infer_golden.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
cd ../.. > /dev/null
# download language model
cd models/lm > /dev/null
sh download_lm_en.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# download well-trained model
cd models/librispeech > /dev/null
sh download_model.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# infer
CUDA_VISIBLE_DEVICES=0 \
python -u infer.py \
--num_samples=10 \
--trainer_count=1 \
--beam_size=500 \
--num_proc_bsearch=8 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=2048 \
--alpha=2.15 \
--beta=0.35 \
--cutoff_prob=1.0 \
--cutoff_top_n=40 \
--use_gru=False \
--use_gpu=True \
--share_rnn_weights=True \
--infer_manifest='data/librispeech/manifest.test-clean' \
--mean_std_path='models/librispeech/mean_std.npz' \
--vocab_path='models/librispeech/vocab.txt' \
--model_path='models/librispeech/params.tar.gz' \
--lang_model_path='models/lm/common_crawl_00.prune01111.trie.klm' \
--decoding_method='ctc_beam_search' \
--error_rate_type='wer' \
--specgram_type='linear'
if [ $? -ne 0 ]; then
echo "Failed in inference!"
exit 1
fi
exit 0
deep_speech_2/examples/librispeech/run_test.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
cd ../.. > /dev/null
# download language model
cd models/lm > /dev/null
sh download_lm_en.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# evaluate model
CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 \
python -u test.py \
--batch_size=128 \
--trainer_count=8 \
--beam_size=500 \
--num_proc_bsearch=8 \
--num_proc_data=8 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=2048 \
--alpha=2.15 \
--beta=0.35 \
--cutoff_prob=1.0 \
--cutoff_top_n=40 \
--use_gru=False \
--use_gpu=True \
--share_rnn_weights=True \
--test_manifest='data/librispeech/manifest.test-clean' \
--mean_std_path='data/librispeech/mean_std.npz' \
--vocab_path='data/librispeech/vocab.txt' \
--model_path='checkpoints/libri/params.latest.tar.gz' \
--lang_model_path='models/lm/common_crawl_00.prune01111.trie.klm' \
--decoding_method='ctc_beam_search' \
--error_rate_type='wer' \
--specgram_type='linear'
if [ $? -ne 0 ]; then
echo "Failed in evaluation!"
exit 1
fi
exit 0
deep_speech_2/examples/librispeech/run_test_golden.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
cd ../.. > /dev/null
# download language model
cd models/lm > /dev/null
sh download_lm_en.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# download well-trained model
cd models/librispeech > /dev/null
sh download_model.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# evaluate model
CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 \
python -u test.py \
--batch_size=128 \
--trainer_count=8 \
--beam_size=500 \
--num_proc_bsearch=8 \
--num_proc_data=8 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=2048 \
--alpha=2.15 \
--beta=0.35 \
--cutoff_prob=1.0 \
--cutoff_top_n=40 \
--use_gru=False \
--use_gpu=True \
--share_rnn_weights=True \
--test_manifest='data/librispeech/manifest.test-clean' \
--mean_std_path='models/librispeech/mean_std.npz' \
--vocab_path='models/librispeech/vocab.txt' \
--model_path='models/librispeech/params.tar.gz' \
--lang_model_path='models/lm/common_crawl_00.prune01111.trie.klm' \
--decoding_method='ctc_beam_search' \
--error_rate_type='wer' \
--specgram_type='linear'
if [ $? -ne 0 ]; then
echo "Failed in evaluation!"
exit 1
fi
exit 0
deep_speech_2/examples/librispeech/run_train.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
cd ../.. > /dev/null
# train model
# if you wish to resume from an exists model, uncomment --init_model_path
CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 \
python -u train.py \
--batch_size=160 \
--trainer_count=8 \
--num_passes=50 \
--num_proc_data=16 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=2048 \
--num_iter_print=100 \
--learning_rate=5e-4 \
--max_duration=27.0 \
--min_duration=0.0 \
--test_off=False \
--use_sortagrad=True \
--use_gru=False \
--use_gpu=True \
--is_local=True \
--share_rnn_weights=True \
--train_manifest='data/librispeech/manifest.train' \
--dev_manifest='data/librispeech/manifest.dev-clean' \
--mean_std_path='data/librispeech/mean_std.npz' \
--vocab_path='data/librispeech/vocab.txt' \
--output_model_dir='./checkpoints/libri' \
--augment_conf_path='conf/augmentation.config' \
--specgram_type='linear' \
--shuffle_method='batch_shuffle_clipped'
if [ $? -ne 0 ]; then
echo "Failed in training!"
exit 1
fi
exit 0
deep_speech_2/examples/librispeech/run_tune.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
cd ../.. > /dev/null
# grid-search for hyper-parameters in language model
CUDA_VISIBLE_DEVICES=0,1,2,3 \
python -u tools/tune.py \
--num_batches=-1 \
--batch_size=128 \
--trainer_count=8 \
--beam_size=500 \
--num_proc_bsearch=12 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=2048 \
--num_alphas=45 \
--num_betas=8 \
--alpha_from=1.0 \
--alpha_to=3.2 \
--beta_from=0.1 \
--beta_to=0.45 \
--cutoff_prob=1.0 \
--cutoff_top_n=40 \
--use_gru=False \
--use_gpu=True \
--share_rnn_weights=True \
--tune_manifest='data/librispeech/manifest.dev-clean' \
--mean_std_path='data/librispeech/mean_std.npz' \
--vocab_path='models/librispeech/vocab.txt' \
--model_path='models/librispeech/params.tar.gz' \
--lang_model_path='models/lm/common_crawl_00.prune01111.trie.klm' \
--error_rate_type='wer' \
--specgram_type='linear'
if [ $? -ne 0 ]; then
echo "Failed in tuning!"
exit 1
fi
exit 0
deep_speech_2/examples/tiny/run_data.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
cd ../.. > /dev/null
# prepare folder
if [ ! -e data/tiny ]; then
mkdir data/tiny
fi
# download data, generate manifests
PYTHONPATH=.:$PYTHONPATH python data/librispeech/librispeech.py \
--manifest_prefix='data/tiny/manifest' \
--target_dir='~/.cache/paddle/dataset/speech/libri' \
--full_download='False'
if [ $? -ne 0 ]; then
echo "Prepare LibriSpeech failed. Terminated."
exit 1
fi
head -n 64 data/tiny/manifest.dev-clean > data/tiny/manifest.tiny
# build vocabulary
python tools/build_vocab.py \
--count_threshold=0 \
--vocab_path='data/tiny/vocab.txt' \
--manifest_paths='data/tiny/manifest.dev-clean'
if [ $? -ne 0 ]; then
echo "Build vocabulary failed. Terminated."
exit 1
fi
# compute mean and stddev for normalizer
python tools/compute_mean_std.py \
--manifest_path='data/tiny/manifest.tiny' \
--num_samples=64 \
--specgram_type='linear' \
--output_path='data/tiny/mean_std.npz'
if [ $? -ne 0 ]; then
echo "Compute mean and stddev failed. Terminated."
exit 1
fi
echo "Tiny data preparation done."
exit 0
deep_speech_2/examples/tiny/run_infer.sh 已删除 100644 → 0
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#! /usr/bin/env bash
cd ../.. > /dev/null
# download language model
cd models/lm > /dev/null
sh download_lm_en.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# infer
CUDA_VISIBLE_DEVICES=0 \
python -u infer.py \
--num_samples=10 \
--trainer_count=1 \
--beam_size=500 \
--num_proc_bsearch=8 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=2048 \
--alpha=2.15 \
--beta=0.35 \
--cutoff_prob=1.0 \
--cutoff_top_n=40 \
--use_gru=False \
--use_gpu=True \
--share_rnn_weights=True \
--infer_manifest='data/tiny/manifest.tiny' \
--mean_std_path='data/tiny/mean_std.npz' \
--vocab_path='data/tiny/vocab.txt' \
--model_path='checkpoints/tiny/params.pass-19.tar.gz' \
--lang_model_path='models/lm/common_crawl_00.prune01111.trie.klm' \
--decoding_method='ctc_beam_search' \
--error_rate_type='wer' \
--specgram_type='linear'
if [ $? -ne 0 ]; then
echo "Failed in inference!"
exit 1
fi
exit 0
deep_speech_2/examples/tiny/run_infer_golden.sh 已删除 100644 → 0
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#! /usr/bin/env bash
cd ../.. > /dev/null
# download language model
cd models/lm > /dev/null
sh download_lm_en.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# download well-trained model
cd models/librispeech > /dev/null
sh download_model.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# infer
CUDA_VISIBLE_DEVICES=0 \
python -u infer.py \
--num_samples=10 \
--trainer_count=1 \
--beam_size=500 \
--num_proc_bsearch=8 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=2048 \
--alpha=2.15 \
--beta=0.35 \
--cutoff_prob=1.0 \
--cutoff_top_n=40 \
--use_gru=False \
--use_gpu=True \
--share_rnn_weights=True \
--infer_manifest='data/tiny/manifest.test-clean' \
--mean_std_path='models/librispeech/mean_std.npz' \
--vocab_path='models/librispeech/vocab.txt' \
--model_path='models/librispeech/params.tar.gz' \
--lang_model_path='models/lm/common_crawl_00.prune01111.trie.klm' \
--decoding_method='ctc_beam_search' \
--error_rate_type='wer' \
--specgram_type='linear'
if [ $? -ne 0 ]; then
echo "Failed in inference!"
exit 1
fi
exit 0
deep_speech_2/examples/tiny/run_test.sh 已删除 100644 → 0
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#! /usr/bin/env bash
cd ../.. > /dev/null
# download language model
cd models/lm > /dev/null
sh download_lm_en.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# evaluate model
CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 \
python -u test.py \
--batch_size=16 \
--trainer_count=8 \
--beam_size=500 \
--num_proc_bsearch=8 \
--num_proc_data=8 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=2048 \
--alpha=2.15 \
--beta=0.35 \
--cutoff_prob=1.0 \
--cutoff_top_n=40 \
--use_gru=False \
--use_gpu=True \
--share_rnn_weights=True \
--test_manifest='data/tiny/manifest.tiny' \
--mean_std_path='data/tiny/mean_std.npz' \
--vocab_path='data/tiny/vocab.txt' \
--model_path='checkpoints/params.pass-19.tar.gz' \
--lang_model_path='models/lm/common_crawl_00.prune01111.trie.klm' \
--decoding_method='ctc_beam_search' \
--error_rate_type='wer' \
--specgram_type='linear'
if [ $? -ne 0 ]; then
echo "Failed in evaluation!"
exit 1
fi
exit 0
deep_speech_2/examples/tiny/run_test_golden.sh 已删除 100644 → 0
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#! /usr/bin/env bash
cd ../.. > /dev/null
# download language model
cd models/lm > /dev/null
sh download_lm_en.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# download well-trained model
cd models/librispeech > /dev/null
sh download_model.sh
if [ $? -ne 0 ]; then
exit 1
fi
cd - > /dev/null
# evaluate model
CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 \
python -u test.py \
--batch_size=128 \
--trainer_count=8 \
--beam_size=500 \
--num_proc_bsearch=8 \
--num_proc_data=8 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=2048 \
--alpha=2.15 \
--beta=0.35 \
--cutoff_prob=1.0 \
--cutoff_top_n=40 \
--use_gru=False \
--use_gpu=True \
--share_rnn_weights=True \
--test_manifest='data/tiny/manifest.test-clean' \
--mean_std_path='models/librispeech/mean_std.npz' \
--vocab_path='models/librispeech/vocab.txt' \
--model_path='models/librispeech/params.tar.gz' \
--lang_model_path='models/lm/common_crawl_00.prune01111.trie.klm' \
--decoding_method='ctc_beam_search' \
--error_rate_type='wer' \
--specgram_type='linear'
if [ $? -ne 0 ]; then
echo "Failed in evaluation!"
exit 1
fi
exit 0
deep_speech_2/examples/tiny/run_train.sh 已删除 100644 → 0
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#! /usr/bin/env bash
cd ../.. > /dev/null
# train model
# if you wish to resume from an exists model, uncomment --init_model_path
CUDA_VISIBLE_DEVICES=0,1,2,3 \
python -u train.py \
--batch_size=16 \
--trainer_count=4 \
--num_passes=20 \
--num_proc_data=1 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=2048 \
--num_iter_print=100 \
--learning_rate=1e-5 \
--max_duration=27.0 \
--min_duration=0.0 \
--test_off=False \
--use_sortagrad=True \
--use_gru=False \
--use_gpu=True \
--is_local=True \
--share_rnn_weights=True \
--train_manifest='data/tiny/manifest.tiny' \
--dev_manifest='data/tiny/manifest.tiny' \
--mean_std_path='data/tiny/mean_std.npz' \
--vocab_path='data/tiny/vocab.txt' \
--output_model_dir='./checkpoints/tiny' \
--augment_conf_path='conf/augmentation.config' \
--specgram_type='linear' \
--shuffle_method='batch_shuffle_clipped'
if [ $? -ne 0 ]; then
echo "Fail in training!"
exit 1
fi
exit 0
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#! /usr/bin/env bash
cd ../.. > /dev/null
# grid-search for hyper-parameters in language model
CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 \
python -u tools/tune.py \
--num_batches=1 \
--batch_size=24 \
--trainer_count=8 \
--beam_size=500 \
--num_proc_bsearch=12 \
--num_conv_layers=2 \
--num_rnn_layers=3 \
--rnn_layer_size=2048 \
--num_alphas=45 \
--num_betas=8 \
--alpha_from=1.0 \
--alpha_to=3.2 \
--beta_from=0.1 \
--beta_to=0.45 \
--cutoff_prob=1.0 \
--cutoff_top_n=40 \
--use_gru=False \
--use_gpu=True \
--share_rnn_weights=True \
--tune_manifest='data/tiny/manifest.tiny' \
--mean_std_path='data/tiny/mean_std.npz' \
--vocab_path='data/tiny/vocab.txt' \
--model_path='checkpoints/params.pass-9.tar.gz' \
--lang_model_path='models/lm/common_crawl_00.prune01111.trie.klm' \
--error_rate_type='wer' \
--specgram_type='linear'
if [ $? -ne 0 ]; then
echo "Failed in tuning!"
exit 1
fi
exit 0
deep_speech_2/infer.py 已删除 100644 → 0
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"""Inferer for DeepSpeech2 model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import argparse
import functools
import paddle.v2 as paddle
from data_utils.data import DataGenerator
from model_utils.model import DeepSpeech2Model
from utils.error_rate import wer, cer
from utils.utility import add_arguments, print_arguments
parser = argparse.ArgumentParser(description=__doc__)
add_arg = functools.partial(add_arguments, argparser=parser)
# yapf: disable
add_arg('num_samples', int, 10, "# of samples to infer.")
add_arg('trainer_count', int, 8, "# of Trainers (CPUs or GPUs).")
add_arg('beam_size', int, 500, "Beam search width.")
add_arg('num_proc_bsearch', int, 8, "# of CPUs for beam search.")
add_arg('num_conv_layers', int, 2, "# of convolution layers.")
add_arg('num_rnn_layers', int, 3, "# of recurrent layers.")
add_arg('rnn_layer_size', int, 2048, "# of recurrent cells per layer.")
add_arg('alpha', float, 2.15, "Coef of LM for beam search.")
add_arg('beta', float, 0.35, "Coef of WC for beam search.")
add_arg('cutoff_prob', float, 1.0, "Cutoff probability for pruning.")
add_arg('cutoff_top_n', int, 40, "Cutoff number for pruning.")
add_arg('use_gru', bool, False, "Use GRUs instead of simple RNNs.")
add_arg('use_gpu', bool, True, "Use GPU or not.")
add_arg('share_rnn_weights',bool, True, "Share input-hidden weights across "
"bi-directional RNNs. Not for GRU.")
add_arg('infer_manifest', str,
'data/librispeech/manifest.dev-clean',
"Filepath of manifest to infer.")
add_arg('mean_std_path', str,
'data/librispeech/mean_std.npz',
"Filepath of normalizer's mean & std.")
add_arg('vocab_path', str,
'data/librispeech/vocab.txt',
"Filepath of vocabulary.")
add_arg('lang_model_path', str,
'models/lm/common_crawl_00.prune01111.trie.klm',
"Filepath for language model.")
add_arg('model_path', str,
'./checkpoints/libri/params.latest.tar.gz',
"If None, the training starts from scratch, "
"otherwise, it resumes from the pre-trained model.")
add_arg('decoding_method', str,
'ctc_beam_search',
"Decoding method. Options: ctc_beam_search, ctc_greedy",
choices = ['ctc_beam_search', 'ctc_greedy'])
add_arg('error_rate_type', str,
'wer',
"Error rate type for evaluation.",
choices=['wer', 'cer'])
add_arg('specgram_type', str,
'linear',
"Audio feature type. Options: linear, mfcc.",
choices=['linear', 'mfcc'])
# yapf: disable
args = parser.parse_args()
def infer():
"""Inference for DeepSpeech2."""
data_generator = DataGenerator(
vocab_filepath=args.vocab_path,
mean_std_filepath=args.mean_std_path,
augmentation_config='{}',
specgram_type=args.specgram_type,
num_threads=1,
keep_transcription_text=True,
num_conv_layers=args.num_conv_layers)
batch_reader = data_generator.batch_reader_creator(
manifest_path=args.infer_manifest,
batch_size=args.num_samples,
min_batch_size=1,
sortagrad=False,
shuffle_method=None)
infer_data = batch_reader().next()
ds2_model = DeepSpeech2Model(
vocab_size=data_generator.vocab_size,
num_conv_layers=args.num_conv_layers,
num_rnn_layers=args.num_rnn_layers,
rnn_layer_size=args.rnn_layer_size,
use_gru=args.use_gru,
pretrained_model_path=args.model_path,
share_rnn_weights=args.share_rnn_weights)
# decoders only accept string encoded in utf-8
vocab_list = [chars.encode("utf-8") for chars in data_generator.vocab_list]
result_transcripts = ds2_model.infer_batch(
infer_data=infer_data,
decoding_method=args.decoding_method,
beam_alpha=args.alpha,
beam_beta=args.beta,
beam_size=args.beam_size,
cutoff_prob=args.cutoff_prob,
cutoff_top_n=args.cutoff_top_n,
vocab_list=vocab_list,
language_model_path=args.lang_model_path,
num_processes=args.num_proc_bsearch,
feeding_dict=data_generator.feeding)
error_rate_func = cer if args.error_rate_type == 'cer' else wer
target_transcripts = [data[1] for data in infer_data]
for target, result in zip(target_transcripts, result_transcripts):
print("\nTarget Transcription: %s\nOutput Transcription: %s" %
(target, result))
print("Current error rate [%s] = %f" %
(args.error_rate_type, error_rate_func(target, result)))
ds2_model.logger.info("finish inference")
def main():
print_arguments(args)
paddle.init(use_gpu=args.use_gpu,
rnn_use_batch=True,
trainer_count=args.trainer_count)
infer()
if __name__ == '__main__':
main()
deep_speech_2/model_utils/model.py 已删除 100644 → 0
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"""Contains DeepSpeech2 model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import sys
import os
import time
import logging
import gzip
from distutils.dir_util import mkpath
import paddle.v2 as paddle
from decoders.swig_wrapper import Scorer
from decoders.swig_wrapper import ctc_greedy_decoder
from decoders.swig_wrapper import ctc_beam_search_decoder_batch
from model_utils.network import deep_speech_v2_network
logging.basicConfig(
format='[%(levelname)s %(asctime)s %(filename)s:%(lineno)d] %(message)s')
class DeepSpeech2Model(object):
"""DeepSpeech2Model class.
:param vocab_size: Decoding vocabulary size.
:type vocab_size: int
:param num_conv_layers: Number of stacking convolution layers.
:type num_conv_layers: int
:param num_rnn_layers: Number of stacking RNN layers.
:type num_rnn_layers: int
:param rnn_layer_size: RNN layer size (number of RNN cells).
:type rnn_layer_size: int
:param pretrained_model_path: Pretrained model path. If None, will train
from stratch.
:type pretrained_model_path: basestring|None
:param share_rnn_weights: Whether to share input-hidden weights between
forward and backward directional RNNs.Notice that
for GRU, weight sharing is not supported.
:type share_rnn_weights: bool
"""
def __init__(self, vocab_size, num_conv_layers, num_rnn_layers,
rnn_layer_size, use_gru, pretrained_model_path,
share_rnn_weights):
self._create_network(vocab_size, num_conv_layers, num_rnn_layers,
rnn_layer_size, use_gru, share_rnn_weights)
self._create_parameters(pretrained_model_path)
self._inferer = None
self._loss_inferer = None
self._ext_scorer = None
self.logger = logging.getLogger("")
self.logger.setLevel(level=logging.INFO)
def train(self,
train_batch_reader,
dev_batch_reader,
feeding_dict,
learning_rate,
gradient_clipping,
num_passes,
output_model_dir,
is_local=True,
num_iterations_print=100,
test_off=False):
"""Train the model.
:param train_batch_reader: Train data reader.
:type train_batch_reader: callable
:param dev_batch_reader: Validation data reader.
:type dev_batch_reader: callable
:param feeding_dict: Feeding is a map of field name and tuple index
of the data that reader returns.
:type feeding_dict: dict|list
:param learning_rate: Learning rate for ADAM optimizer.
:type learning_rate: float
:param gradient_clipping: Gradient clipping threshold.
:type gradient_clipping: float
:param num_passes: Number of training epochs.
:type num_passes: int
:param num_iterations_print: Number of training iterations for printing
a training loss.
:type rnn_iteratons_print: int
:param is_local: Set to False if running with pserver with multi-nodes.
:type is_local: bool
:param output_model_dir: Directory for saving the model (every pass).
:type output_model_dir: basestring
:param test_off: Turn off testing.
:type test_off: bool
"""
# prepare model output directory
if not os.path.exists(output_model_dir):
mkpath(output_model_dir)
# prepare optimizer and trainer
optimizer = paddle.optimizer.Adam(
learning_rate=learning_rate,
gradient_clipping_threshold=gradient_clipping)
trainer = paddle.trainer.SGD(
cost=self._loss,
parameters=self._parameters,
update_equation=optimizer,
is_local=is_local)
# create event handler
def event_handler(event):
global start_time, cost_sum, cost_counter
if isinstance(event, paddle.event.EndIteration):
cost_sum += event.cost
cost_counter += 1
if (event.batch_id + 1) % num_iterations_print == 0:
output_model_path = os.path.join(output_model_dir,
"params.latest.tar.gz")
with gzip.open(output_model_path, 'w') as f:
trainer.save_parameter_to_tar(f)
print("\nPass: %d, Batch: %d, TrainCost: %f" %
(event.pass_id, event.batch_id + 1,
cost_sum / cost_counter))
cost_sum, cost_counter = 0.0, 0
else:
sys.stdout.write('.')
sys.stdout.flush()
if isinstance(event, paddle.event.BeginPass):
start_time = time.time()
cost_sum, cost_counter = 0.0, 0
if isinstance(event, paddle.event.EndPass):
if test_off:
print("\n------- Time: %d sec, Pass: %d" %
(time.time() - start_time, event.pass_id))
else:
result = trainer.test(
reader=dev_batch_reader, feeding=feeding_dict)
print(
"\n------- Time: %d sec, Pass: %d, "
"ValidationCost: %s" %
(time.time() - start_time, event.pass_id, result.cost))
output_model_path = os.path.join(
output_model_dir, "params.pass-%d.tar.gz" % event.pass_id)
with gzip.open(output_model_path, 'w') as f:
trainer.save_parameter_to_tar(f)
# run train
trainer.train(
reader=train_batch_reader,
event_handler=event_handler,
num_passes=num_passes,
feeding=feeding_dict)
def infer_loss_batch(self, infer_data):
"""Model inference. Infer the ctc loss for a batch of speech
utterances.
:param infer_data: List of utterances to infer, with each utterance a
tuple of audio features and transcription text (empty
string).
:type infer_data: list
:return: List of ctc loss.
:rtype: List of float
"""
# define inferer
if self._loss_inferer == None:
self._loss_inferer = paddle.inference.Inference(
output_layer=self._loss, parameters=self._parameters)
# run inference
return self._loss_inferer.infer(input=infer_data)
def infer_batch(self, infer_data, decoding_method, beam_alpha, beam_beta,
beam_size, cutoff_prob, cutoff_top_n, vocab_list,
language_model_path, num_processes, feeding_dict):
"""Model inference. Infer the transcription for a batch of speech
utterances.
:param infer_data: List of utterances to infer, with each utterance
consisting of a tuple of audio features and
transcription text (empty string).
:type infer_data: list
:param decoding_method: Decoding method name, 'ctc_greedy' or
'ctc_beam_search'.
:param decoding_method: string
:param beam_alpha: Parameter associated with language model.
:type beam_alpha: float
:param beam_beta: Parameter associated with word count.
:type beam_beta: float
:param beam_size: Width for Beam search.
:type beam_size: int
:param cutoff_prob: Cutoff probability in pruning,
default 1.0, no pruning.
:type cutoff_prob: float
:param cutoff_top_n: Cutoff number in pruning, only top cutoff_top_n
characters with highest probs in vocabulary will be
used in beam search, default 40.
:type cutoff_top_n: int
:param vocab_list: List of tokens in the vocabulary, for decoding.
:type vocab_list: list
:param language_model_path: Filepath for language model.
:type language_model_path: basestring|None
:param num_processes: Number of processes (CPU) for decoder.
:type num_processes: int
:param feeding_dict: Feeding is a map of field name and tuple index
of the data that reader returns.
:type feeding_dict: dict|list
:return: List of transcription texts.
:rtype: List of basestring
"""
# define inferer
if self._inferer == None:
self._inferer = paddle.inference.Inference(
output_layer=self._log_probs, parameters=self._parameters)
# run inference
infer_results = self._inferer.infer(
input=infer_data, feeding=feeding_dict)
start_pos = [0] * (len(infer_data) + 1)
for i in xrange(len(infer_data)):
start_pos[i + 1] = start_pos[i] + infer_data[i][3][0]
probs_split = [
infer_results[start_pos[i]:start_pos[i + 1]]
for i in xrange(0, len(infer_data))
]
# run decoder
results = []
if decoding_method == "ctc_greedy":
# best path decode
for i, probs in enumerate(probs_split):
output_transcription = ctc_greedy_decoder(
probs_seq=probs, vocabulary=vocab_list)
results.append(output_transcription)
elif decoding_method == "ctc_beam_search":
# initialize external scorer
if self._ext_scorer == None:
self._loaded_lm_path = language_model_path
self.logger.info("begin to initialize the external scorer "
"for decoding")
self._ext_scorer = Scorer(beam_alpha, beam_beta,
language_model_path, vocab_list)
lm_char_based = self._ext_scorer.is_character_based()
lm_max_order = self._ext_scorer.get_max_order()
lm_dict_size = self._ext_scorer.get_dict_size()
self.logger.info("language model: "
"is_character_based = %d," % lm_char_based +
" max_order = %d," % lm_max_order +
" dict_size = %d" % lm_dict_size)
self.logger.info("end initializing scorer. Start decoding ...")
else:
self._ext_scorer.reset_params(beam_alpha, beam_beta)
assert self._loaded_lm_path == language_model_path
# beam search decode
num_processes = min(num_processes, len(probs_split))
beam_search_results = ctc_beam_search_decoder_batch(
probs_split=probs_split,
vocabulary=vocab_list,
beam_size=beam_size,
num_processes=num_processes,
ext_scoring_func=self._ext_scorer,
cutoff_prob=cutoff_prob,
cutoff_top_n=cutoff_top_n)
results = [result[0][1] for result in beam_search_results]
else:
raise ValueError("Decoding method [%s] is not supported." %
decoding_method)
return results
def _create_parameters(self, model_path=None):
"""Load or create model parameters."""
if model_path is None:
self._parameters = paddle.parameters.create(self._loss)
else:
self._parameters = paddle.parameters.Parameters.from_tar(
gzip.open(model_path))
def _create_network(self, vocab_size, num_conv_layers, num_rnn_layers,
rnn_layer_size, use_gru, share_rnn_weights):
"""Create data layers and model network."""
# paddle.data_type.dense_array is used for variable batch input.
# The size 161 * 161 is only an placeholder value and the real shape
# of input batch data will be induced during training.
audio_data = paddle.layer.data(
name="audio_spectrogram",
type=paddle.data_type.dense_array(161 * 161))
text_data = paddle.layer.data(
name="transcript_text",
type=paddle.data_type.integer_value_sequence(vocab_size))
seq_offset_data = paddle.layer.data(
name='sequence_offset',
type=paddle.data_type.integer_value_sequence(1))
seq_len_data = paddle.layer.data(
name='sequence_length',
type=paddle.data_type.integer_value_sequence(1))
index_range_datas = []
for i in xrange(num_rnn_layers):
index_range_datas.append(
paddle.layer.data(
name='conv%d_index_range' % i,
type=paddle.data_type.dense_vector(6)))
self._log_probs, self._loss = deep_speech_v2_network(
audio_data=audio_data,
text_data=text_data,
seq_offset_data=seq_offset_data,
seq_len_data=seq_len_data,
index_range_datas=index_range_datas,
dict_size=vocab_size,
num_conv_layers=num_conv_layers,
num_rnn_layers=num_rnn_layers,
rnn_size=rnn_layer_size,
use_gru=use_gru,
share_rnn_weights=share_rnn_weights)
deep_speech_2/model_utils/network.py 已删除 100644 → 0
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"""Contains DeepSpeech2 layers and networks."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import paddle.v2 as paddle
def conv_bn_layer(input, filter_size, num_channels_in, num_channels_out, stride,
padding, act, index_range_data):
"""Convolution layer with batch normalization.
:param input: Input layer.
:type input: LayerOutput
:param filter_size: The x dimension of a filter kernel. Or input a tuple for
two image dimension.
:type filter_size: int|tuple|list
:param num_channels_in: Number of input channels.
:type num_channels_in: int
:type num_channels_out: Number of output channels.
:type num_channels_in: out
:param padding: The x dimension of the padding. Or input a tuple for two
image dimension.
:type padding: int|tuple|list
:param act: Activation type.
:type act: BaseActivation
:param index_range_data: Index range to indicate sub region.
:type index_range_data: LayerOutput
:return: Batch norm layer after convolution layer.
:rtype: LayerOutput
"""
conv_layer = paddle.layer.img_conv(
input=input,
filter_size=filter_size,
num_channels=num_channels_in,
num_filters=num_channels_out,
stride=stride,
padding=padding,
act=paddle.activation.Linear(),
bias_attr=False)
batch_norm = paddle.layer.batch_norm(input=conv_layer, act=act)
# reset padding part to 0
scale_sub_region = paddle.layer.scale_sub_region(
batch_norm, index_range_data, value=0.0)
return scale_sub_region
def bidirectional_simple_rnn_bn_layer(name, input, size, act, share_weights):
"""Bidirectonal simple rnn layer with sequence-wise batch normalization.
The batch normalization is only performed on input-state weights.
:param name: Name of the layer.
:type name: string
:param input: Input layer.
:type input: LayerOutput
:param size: Number of RNN cells.
:type size: int
:param act: Activation type.
:type act: BaseActivation
:param share_weights: Whether to share input-hidden weights between
forward and backward directional RNNs.
:type share_weights: bool
:return: Bidirectional simple rnn layer.
:rtype: LayerOutput
"""
if share_weights:
# input-hidden weights shared between bi-direcitonal rnn.
input_proj = paddle.layer.fc(
input=input,
size=size,
act=paddle.activation.Linear(),
bias_attr=False)
# batch norm is only performed on input-state projection
input_proj_bn = paddle.layer.batch_norm(
input=input_proj, act=paddle.activation.Linear())
# forward and backward in time
forward_simple_rnn = paddle.layer.recurrent(
input=input_proj_bn, act=act, reverse=False)
backward_simple_rnn = paddle.layer.recurrent(
input=input_proj_bn, act=act, reverse=True)
else:
input_proj_forward = paddle.layer.fc(
input=input,
size=size,
act=paddle.activation.Linear(),
bias_attr=False)
input_proj_backward = paddle.layer.fc(
input=input,
size=size,
act=paddle.activation.Linear(),
bias_attr=False)
# batch norm is only performed on input-state projection
input_proj_bn_forward = paddle.layer.batch_norm(
input=input_proj_forward, act=paddle.activation.Linear())
input_proj_bn_backward = paddle.layer.batch_norm(
input=input_proj_backward, act=paddle.activation.Linear())
# forward and backward in time
forward_simple_rnn = paddle.layer.recurrent(
input=input_proj_bn_forward, act=act, reverse=False)
backward_simple_rnn = paddle.layer.recurrent(
input=input_proj_bn_backward, act=act, reverse=True)
return paddle.layer.concat(input=[forward_simple_rnn, backward_simple_rnn])
def bidirectional_gru_bn_layer(name, input, size, act):
"""Bidirectonal gru layer with sequence-wise batch normalization.
The batch normalization is only performed on input-state weights.
:param name: Name of the layer.
:type name: string
:param input: Input layer.
:type input: LayerOutput
:param size: Number of RNN cells.
:type size: int
:param act: Activation type.
:type act: BaseActivation
:return: Bidirectional simple rnn layer.
:rtype: LayerOutput
"""
input_proj_forward = paddle.layer.fc(
input=input,
size=size * 3,
act=paddle.activation.Linear(),
bias_attr=False)
input_proj_backward = paddle.layer.fc(
input=input,
size=size * 3,
act=paddle.activation.Linear(),
bias_attr=False)
# batch norm is only performed on input-related projections
input_proj_bn_forward = paddle.layer.batch_norm(
input=input_proj_forward, act=paddle.activation.Linear())
input_proj_bn_backward = paddle.layer.batch_norm(
input=input_proj_backward, act=paddle.activation.Linear())
# forward and backward in time
forward_gru = paddle.layer.grumemory(
input=input_proj_bn_forward, act=act, reverse=False)
backward_gru = paddle.layer.grumemory(
input=input_proj_bn_backward, act=act, reverse=True)
return paddle.layer.concat(input=[forward_gru, backward_gru])
def conv_group(input, num_stacks, index_range_datas):
"""Convolution group with stacked convolution layers.
:param input: Input layer.
:type input: LayerOutput
:param num_stacks: Number of stacked convolution layers.
:type num_stacks: int
:param index_range_datas: Index ranges for each convolution layer.
:type index_range_datas: tuple|list
:return: Output layer of the convolution group.
:rtype: LayerOutput
"""
conv = conv_bn_layer(
input=input,
filter_size=(11, 41),
num_channels_in=1,
num_channels_out=32,
stride=(3, 2),
padding=(5, 20),
act=paddle.activation.BRelu(),
index_range_data=index_range_datas[0])
for i in xrange(num_stacks - 1):
conv = conv_bn_layer(
input=conv,
filter_size=(11, 21),
num_channels_in=32,
num_channels_out=32,
stride=(1, 2),
padding=(5, 10),
act=paddle.activation.BRelu(),
index_range_data=index_range_datas[i + 1])
output_num_channels = 32
output_height = 160 // pow(2, num_stacks) + 1
return conv, output_num_channels, output_height
def rnn_group(input, size, num_stacks, use_gru, share_rnn_weights):
"""RNN group with stacked bidirectional simple RNN layers.
:param input: Input layer.
:type input: LayerOutput
:param size: Number of RNN cells in each layer.
:type size: int
:param num_stacks: Number of stacked rnn layers.
:type num_stacks: int
:param use_gru: Use gru if set True. Use simple rnn if set False.
:type use_gru: bool
:param share_rnn_weights: Whether to share input-hidden weights between
forward and backward directional RNNs.
It is only available when use_gru=False.
:type share_weights: bool
:return: Output layer of the RNN group.
:rtype: LayerOutput
"""
output = input
for i in xrange(num_stacks):
if use_gru:
output = bidirectional_gru_bn_layer(
name=str(i),
input=output,
size=size,
act=paddle.activation.Relu())
# BRelu does not support hppl, need to add later. Use Relu instead.
else:
output = bidirectional_simple_rnn_bn_layer(
name=str(i),
input=output,
size=size,
act=paddle.activation.BRelu(),
share_weights=share_rnn_weights)
return output
def deep_speech_v2_network(audio_data,
text_data,
seq_offset_data,
seq_len_data,
index_range_datas,
dict_size,
num_conv_layers=2,
num_rnn_layers=3,
rnn_size=256,
use_gru=False,
share_rnn_weights=True):
"""The DeepSpeech2 network structure.
:param audio_data: Audio spectrogram data layer.
:type audio_data: LayerOutput
:param text_data: Transcription text data layer.
:type text_data: LayerOutput
:param seq_offset_data: Sequence offset data layer.
:type seq_offset_data: LayerOutput
:param seq_len_data: Valid sequence length data layer.
:type seq_len_data: LayerOutput
:param index_range_datas: Index ranges data layers.
:type index_range_datas: tuple|list
:param dict_size: Dictionary size for tokenized transcription.
:type dict_size: int
:param num_conv_layers: Number of stacking convolution layers.
:type num_conv_layers: int
:param num_rnn_layers: Number of stacking RNN layers.
:type num_rnn_layers: int
:param rnn_size: RNN layer size (number of RNN cells).
:type rnn_size: int
:param use_gru: Use gru if set True. Use simple rnn if set False.
:type use_gru: bool
:param share_rnn_weights: Whether to share input-hidden weights between
forward and backward direction RNNs.
It is only available when use_gru=False.
:type share_weights: bool
:return: A tuple of an output unnormalized log probability layer (
before softmax) and a ctc cost layer.
:rtype: tuple of LayerOutput
"""
# convolution group
conv_group_output, conv_group_num_channels, conv_group_height = conv_group(
input=audio_data,
num_stacks=num_conv_layers,
index_range_datas=index_range_datas)
# convert data form convolution feature map to sequence of vectors
conv2seq = paddle.layer.block_expand(
input=conv_group_output,
num_channels=conv_group_num_channels,
stride_x=1,
stride_y=1,
block_x=1,
block_y=conv_group_height)
# remove padding part
remove_padding_data = paddle.layer.sub_seq(
input=conv2seq,
offsets=seq_offset_data,
sizes=seq_len_data,
act=paddle.activation.Linear(),
bias_attr=False)
# rnn group
rnn_group_output = rnn_group(
input=remove_padding_data,
size=rnn_size,
num_stacks=num_rnn_layers,
use_gru=use_gru,
share_rnn_weights=share_rnn_weights)
fc = paddle.layer.fc(
input=rnn_group_output,
size=dict_size + 1,
act=paddle.activation.Linear(),
bias_attr=True)
# probability distribution with softmax
log_probs = paddle.layer.mixed(
input=paddle.layer.identity_projection(input=fc),
act=paddle.activation.Softmax())
# ctc cost
ctc_loss = paddle.layer.warp_ctc(
input=fc,
label=text_data,
size=dict_size + 1,
blank=dict_size,
norm_by_times=True)
return log_probs, ctc_loss
deep_speech_2/models/aishell/download_model.sh 已删除 100644 → 0
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#! /usr/bin/env bash
. ../../utils/utility.sh
URL='http://cloud.dlnel.org/filepub/?uuid=6c83b9d8-3255-4adf-9726-0fe0be3d0274'
MD5=28521a58552885a81cf92a1e9b133a71
TARGET=./aishell_model.tar.gz
echo "Download Aishell model ..."
download $URL $MD5 $TARGET
if [ $? -ne 0 ]; then
echo "Fail to download Aishell model!"
exit 1
fi
tar -zxvf $TARGET
exit 0
deep_speech_2/models/baidu_en8k/download_model.sh 已删除 100644 → 0
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#! /usr/bin/env bash
. ../../utils/utility.sh
URL='To-be-added'
MD5=a19d40cb3b558eb696c44d883f32cfda
TARGET=./baidu_en8k_model.tar.gz
echo "Download BaiduEn8k model ..."
download $URL $MD5 $TARGET
if [ $? -ne 0 ]; then
echo "Fail to download BaiduEn8k model!"
exit 1
fi
tar -zxvf $TARGET
exit 0
deep_speech_2/models/librispeech/download_model.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
. ../../utils/utility.sh
URL='http://cloud.dlnel.org/filepub/?uuid=6020a634-5399-4423-b021-c5ed32680fff'
MD5=2ef08f8b608a7c555592161fc14d81a6
TARGET=./librispeech_model.tar.gz
echo "Download LibriSpeech model ..."
download $URL $MD5 $TARGET
if [ $? -ne 0 ]; then
echo "Fail to download LibriSpeech model!"
exit 1
fi
tar -zxvf $TARGET
exit 0
deep_speech_2/models/lm/download_lm_ch.sh 已删除 100644 → 0
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#! /usr/bin/env bash
. ../../utils/utility.sh
URL=http://cloud.dlnel.org/filepub/?uuid=d21861e4-4ed6-45bb-ad8e-ae417a43195e
MD5="29e02312deb2e59b3c8686c7966d4fe3"
TARGET=./zh_giga.no_cna_cmn.prune01244.klm
echo "Download language model ..."
download $URL $MD5 $TARGET
if [ $? -ne 0 ]; then
echo "Fail to download the language model!"
exit 1
fi
exit 0
deep_speech_2/models/lm/download_lm_en.sh 已删除 100644 → 0
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#! /usr/bin/env bash
. ../../utils/utility.sh
URL=http://paddlepaddle.bj.bcebos.com/model_zoo/speech/common_crawl_00.prune01111.trie.klm
MD5="099a601759d467cd0a8523ff939819c5"
TARGET=./common_crawl_00.prune01111.trie.klm
echo "Download language model ..."
download $URL $MD5 $TARGET
if [ $? -ne 0 ]; then
echo "Fail to download the language model!"
exit 1
fi
exit 0
deep_speech_2/requirements.txt 已删除 100644 → 0
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scipy==0.13.1
resampy==0.1.5
SoundFile==0.9.0.post1
python_speech_features
deep_speech_2/setup.sh 已删除 100644 → 0
浏览文件 @ 788f7011
#! /usr/bin/env bash
# install python dependencies
if [ -f "requirements.txt" ]; then
pip install -r requirements.txt
fi
if [ $? != 0 ]; then
echo "Install python dependencies failed !!!"
exit 1
fi
# install package libsndfile
python -c "import soundfile"
if [ $? != 0 ]; then
echo "Install package libsndfile into default system path."
wget "http://www.mega-nerd.com/libsndfile/files/libsndfile-1.0.28.tar.gz"
if [ $? != 0 ]; then
echo "Download libsndfile-1.0.28.tar.gz failed !!!"
exit 1
fi
tar -zxvf libsndfile-1.0.28.tar.gz
cd libsndfile-1.0.28
./configure > /dev/null && make > /dev/null && make install > /dev/null
cd ..
rm -rf libsndfile-1.0.28
rm libsndfile-1.0.28.tar.gz
fi
# install decoders
python -c "import swig_decoders"
if [ $? != 0 ]; then
cd decoders/swig > /dev/null
sh setup.sh
cd - > /dev/null
fi
echo "Install all dependencies successfully."
deep_speech_2/test.py 已删除 100644 → 0
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"""Evaluation for DeepSpeech2 model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import argparse
import functools
import paddle.v2 as paddle
from data_utils.data import DataGenerator
from model_utils.model import DeepSpeech2Model
from utils.error_rate import wer, cer
from utils.utility import add_arguments, print_arguments
parser = argparse.ArgumentParser(description=__doc__)
add_arg = functools.partial(add_arguments, argparser=parser)
# yapf: disable
add_arg('batch_size', int, 128, "Minibatch size.")
add_arg('trainer_count', int, 8, "# of Trainers (CPUs or GPUs).")
add_arg('beam_size', int, 500, "Beam search width.")
add_arg('num_proc_bsearch', int, 8, "# of CPUs for beam search.")
add_arg('num_proc_data', int, 8, "# of CPUs for data preprocessing.")
add_arg('num_conv_layers', int, 2, "# of convolution layers.")
add_arg('num_rnn_layers', int, 3, "# of recurrent layers.")
add_arg('rnn_layer_size', int, 2048, "# of recurrent cells per layer.")
add_arg('alpha', float, 2.15, "Coef of LM for beam search.")
add_arg('beta', float, 0.35, "Coef of WC for beam search.")
add_arg('cutoff_prob', float, 1.0, "Cutoff probability for pruning.")
add_arg('cutoff_top_n', int, 40, "Cutoff number for pruning.")
add_arg('use_gru', bool, False, "Use GRUs instead of simple RNNs.")
add_arg('use_gpu', bool, True, "Use GPU or not.")
add_arg('share_rnn_weights',bool, True, "Share input-hidden weights across "
"bi-directional RNNs. Not for GRU.")
add_arg('test_manifest', str,
'data/librispeech/manifest.test-clean',
"Filepath of manifest to evaluate.")
add_arg('mean_std_path', str,
'data/librispeech/mean_std.npz',
"Filepath of normalizer's mean & std.")
add_arg('vocab_path', str,
'data/librispeech/vocab.txt',
"Filepath of vocabulary.")
add_arg('model_path', str,
'./checkpoints/libri/params.latest.tar.gz',
"If None, the training starts from scratch, "
"otherwise, it resumes from the pre-trained model.")
add_arg('lang_model_path', str,
'models/lm/common_crawl_00.prune01111.trie.klm',
"Filepath for language model.")
add_arg('decoding_method', str,
'ctc_beam_search',
"Decoding method. Options: ctc_beam_search, ctc_greedy",
choices = ['ctc_beam_search', 'ctc_greedy'])
add_arg('error_rate_type', str,
'wer',
"Error rate type for evaluation.",
choices=['wer', 'cer'])
add_arg('specgram_type', str,
'linear',
"Audio feature type. Options: linear, mfcc.",
choices=['linear', 'mfcc'])
# yapf: disable
args = parser.parse_args()
def evaluate():
"""Evaluate on whole test data for DeepSpeech2."""
data_generator = DataGenerator(
vocab_filepath=args.vocab_path,
mean_std_filepath=args.mean_std_path,
augmentation_config='{}',
specgram_type=args.specgram_type,
num_threads=args.num_proc_data,
keep_transcription_text=True,
num_conv_layers=args.num_conv_layers)
batch_reader = data_generator.batch_reader_creator(
manifest_path=args.test_manifest,
batch_size=args.batch_size,
min_batch_size=1,
sortagrad=False,
shuffle_method=None)
ds2_model = DeepSpeech2Model(
vocab_size=data_generator.vocab_size,
num_conv_layers=args.num_conv_layers,
num_rnn_layers=args.num_rnn_layers,
rnn_layer_size=args.rnn_layer_size,
use_gru=args.use_gru,
pretrained_model_path=args.model_path,
share_rnn_weights=args.share_rnn_weights)
# decoders only accept string encoded in utf-8
vocab_list = [chars.encode("utf-8") for chars in data_generator.vocab_list]
error_rate_func = cer if args.error_rate_type == 'cer' else wer
error_sum, num_ins = 0.0, 0
for infer_data in batch_reader():
result_transcripts = ds2_model.infer_batch(
infer_data=infer_data,
decoding_method=args.decoding_method,
beam_alpha=args.alpha,
beam_beta=args.beta,
beam_size=args.beam_size,
cutoff_prob=args.cutoff_prob,
cutoff_top_n=args.cutoff_top_n,
vocab_list=vocab_list,
language_model_path=args.lang_model_path,
num_processes=args.num_proc_bsearch,
feeding_dict=data_generator.feeding)
target_transcripts = [data[1] for data in infer_data]
for target, result in zip(target_transcripts, result_transcripts):
error_sum += error_rate_func(target, result)
num_ins += 1
print("Error rate [%s] (%d/?) = %f" %
(args.error_rate_type, num_ins, error_sum / num_ins))
print("Final error rate [%s] (%d/%d) = %f" %
(args.error_rate_type, num_ins, num_ins, error_sum / num_ins))
ds2_model.logger.info("finish evaluation")
def main():
print_arguments(args)
paddle.init(use_gpu=args.use_gpu,
rnn_use_batch=True,
trainer_count=args.trainer_count)
evaluate()
if __name__ == '__main__':
main()
deep_speech_2/tools/_init_paths.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Set up paths for DS2"""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import os.path
import sys
def add_path(path):
if path not in sys.path:
sys.path.insert(0, path)
this_dir = os.path.dirname(__file__)
# Add project path to PYTHONPATH
proj_path = os.path.join(this_dir, '..')
add_path(proj_path)
deep_speech_2/tools/build_vocab.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Build vocabulary from manifest files.
Each item in vocabulary file is a character.
"""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import argparse
import functools
import codecs
import json
from collections import Counter
import os.path
import _init_paths
from data_utils.utility import read_manifest
from utils.utility import add_arguments, print_arguments
parser = argparse.ArgumentParser(description=__doc__)
add_arg = functools.partial(add_arguments, argparser=parser)
# yapf: disable
add_arg('count_threshold', int, 0, "Truncation threshold for char counts.")
add_arg('vocab_path', str,
'data/librispeech/vocab.txt',
"Filepath to write the vocabulary.")
add_arg('manifest_paths', str,
None,
"Filepaths of manifests for building vocabulary. "
"You can provide multiple manifest files.",
nargs='+',
required=True)
# yapf: disable
args = parser.parse_args()
def count_manifest(counter, manifest_path):
manifest_jsons = read_manifest(manifest_path)
for line_json in manifest_jsons:
for char in line_json['text']:
counter.update(char)
def main():
print_arguments(args)
counter = Counter()
for manifest_path in args.manifest_paths:
count_manifest(counter, manifest_path)
count_sorted = sorted(counter.items(), key=lambda x: x[1], reverse=True)
with codecs.open(args.vocab_path, 'w', 'utf-8') as fout:
for char, count in count_sorted:
if count < args.count_threshold: break
fout.write(char + '\n')
if __name__ == '__main__':
main()
deep_speech_2/tools/compute_mean_std.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Compute mean and std for feature normalizer, and save to file."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import argparse
import functools
import _init_paths
from data_utils.normalizer import FeatureNormalizer
from data_utils.augmentor.augmentation import AugmentationPipeline
from data_utils.featurizer.audio_featurizer import AudioFeaturizer
from utils.utility import add_arguments, print_arguments
parser = argparse.ArgumentParser(description=__doc__)
add_arg = functools.partial(add_arguments, argparser=parser)
# yapf: disable
add_arg('num_samples', int, 2000, "# of samples to for statistics.")
add_arg('specgram_type', str,
'linear',
"Audio feature type. Options: linear, mfcc.",
choices=['linear', 'mfcc'])
add_arg('manifest_path', str,
'data/librispeech/manifest.train',
"Filepath of manifest to compute normalizer's mean and stddev.")
add_arg('output_path', str,
'data/librispeech/mean_std.npz',
"Filepath of write mean and stddev to (.npz).")
# yapf: disable
args = parser.parse_args()
def main():
print_arguments(args)
augmentation_pipeline = AugmentationPipeline('{}')
audio_featurizer = AudioFeaturizer(specgram_type=args.specgram_type)
def augment_and_featurize(audio_segment):
augmentation_pipeline.transform_audio(audio_segment)
return audio_featurizer.featurize(audio_segment)
normalizer = FeatureNormalizer(
mean_std_filepath=None,
manifest_path=args.manifest_path,
featurize_func=augment_and_featurize,
num_samples=args.num_samples)
normalizer.write_to_file(args.output_path)
if __name__ == '__main__':
main()
deep_speech_2/tools/profile.sh 已删除 100644 → 0
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#! /usr/bin/env bash
BATCH_SIZE_PER_GPU=64
MIN_DURATION=6.0
MAX_DURATION=7.0
function join_by { local IFS="$1"; shift; echo "$*"; }
for NUM_GPUS in 16 8 4 2 1
do
DEVICES=$(join_by , $(seq 0 $(($NUM_GPUS-1))))
BATCH_SIZE=$(($BATCH_SIZE_PER_GPU * $NUM_GPUS))
CUDA_VISIBLE_DEVICES=$DEVICES \
python train.py \
--batch_size=$BATCH_SIZE \
--num_passes=1 \
--test_off=True \
--trainer_count=$NUM_GPUS \
--min_duration=$MIN_DURATION \
--max_duration=$MAX_DURATION > tmp.log 2>&1
if [ $? -ne 0 ];then
exit 1
fi
cat tmp.log | grep "Time" | awk '{print "GPU Num: " "'"$NUM_GPUS"'" " Time: "$3}'
rm tmp.log
done
deep_speech_2/tools/tune.py 已删除 100644 → 0
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"""Beam search parameters tuning for DeepSpeech2 model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import sys
import os
import numpy as np
import argparse
import functools
import gzip
import logging
import paddle.v2 as paddle
import _init_paths
from data_utils.data import DataGenerator
from decoders.swig_wrapper import Scorer
from decoders.swig_wrapper import ctc_beam_search_decoder_batch
from model_utils.model import deep_speech_v2_network
from utils.error_rate import wer, cer
from utils.utility import add_arguments, print_arguments
parser = argparse.ArgumentParser(description=__doc__)
add_arg = functools.partial(add_arguments, argparser=parser)
# yapf: disable
add_arg('num_batches', int, -1, "# of batches tuning on. "
"Default -1, on whole dev set.")
add_arg('batch_size', int, 256, "# of samples per batch.")
add_arg('trainer_count', int, 8, "# of Trainers (CPUs or GPUs).")
add_arg('beam_size', int, 500, "Beam search width.")
add_arg('num_proc_bsearch', int, 8, "# of CPUs for beam search.")
add_arg('num_proc_data', int, 8, "# of CPUs for data preprocessing.")
add_arg('num_conv_layers', int, 2, "# of convolution layers.")
add_arg('num_rnn_layers', int, 3, "# of recurrent layers.")
add_arg('rnn_layer_size', int, 2048, "# of recurrent cells per layer.")
add_arg('num_alphas', int, 45, "# of alpha candidates for tuning.")
add_arg('num_betas', int, 8, "# of beta candidates for tuning.")
add_arg('alpha_from', float, 1.0, "Where alpha starts tuning from.")
add_arg('alpha_to', float, 3.2, "Where alpha ends tuning with.")
add_arg('beta_from', float, 0.1, "Where beta starts tuning from.")
add_arg('beta_to', float, 0.45, "Where beta ends tuning with.")
add_arg('cutoff_prob', float, 1.0, "Cutoff probability for pruning.")
add_arg('cutoff_top_n', int, 40, "Cutoff number for pruning.")
add_arg('use_gru', bool, False, "Use GRUs instead of simple RNNs.")
add_arg('use_gpu', bool, True, "Use GPU or not.")
add_arg('share_rnn_weights',bool, True, "Share input-hidden weights across "
"bi-directional RNNs. Not for GRU.")
add_arg('tune_manifest', str,
'data/librispeech/manifest.dev-clean',
"Filepath of manifest to tune.")
add_arg('mean_std_path', str,
'data/librispeech/mean_std.npz',
"Filepath of normalizer's mean & std.")
add_arg('vocab_path', str,
'data/librispeech/vocab.txt',
"Filepath of vocabulary.")
add_arg('lang_model_path', str,
'models/lm/common_crawl_00.prune01111.trie.klm',
"Filepath for language model.")
add_arg('model_path', str,
'./checkpoints/libri/params.latest.tar.gz',
"If None, the training starts from scratch, "
"otherwise, it resumes from the pre-trained model.")
add_arg('error_rate_type', str,
'wer',
"Error rate type for evaluation.",
choices=['wer', 'cer'])
add_arg('specgram_type', str,
'linear',
"Audio feature type. Options: linear, mfcc.",
choices=['linear', 'mfcc'])
# yapf: disable
args = parser.parse_args()
logging.basicConfig(
format='[%(levelname)s %(asctime)s %(filename)s:%(lineno)d] %(message)s')
def tune():
"""Tune parameters alpha and beta incrementally."""
if not args.num_alphas >= 0:
raise ValueError("num_alphas must be non-negative!")
if not args.num_betas >= 0:
raise ValueError("num_betas must be non-negative!")
data_generator = DataGenerator(
vocab_filepath=args.vocab_path,
mean_std_filepath=args.mean_std_path,
augmentation_config='{}',
specgram_type=args.specgram_type,
num_threads=args.num_proc_data,
keep_transcription_text=True,
num_conv_layers=args.num_conv_layers)
audio_data = paddle.layer.data(
name="audio_spectrogram",
type=paddle.data_type.dense_array(161 * 161))
text_data = paddle.layer.data(
name="transcript_text",
type=paddle.data_type.integer_value_sequence(data_generator.vocab_size))
seq_offset_data = paddle.layer.data(
name='sequence_offset',
type=paddle.data_type.integer_value_sequence(1))
seq_len_data = paddle.layer.data(
name='sequence_length',
type=paddle.data_type.integer_value_sequence(1))
index_range_datas = []
for i in xrange(args.num_rnn_layers):
index_range_datas.append(
paddle.layer.data(
name='conv%d_index_range' % i,
type=paddle.data_type.dense_vector(6)))
output_probs, _ = deep_speech_v2_network(
audio_data=audio_data,
text_data=text_data,
seq_offset_data=seq_offset_data,
seq_len_data=seq_len_data,
index_range_datas=index_range_datas,
dict_size=data_generator.vocab_size,
num_conv_layers=args.num_conv_layers,
num_rnn_layers=args.num_rnn_layers,
rnn_size=args.rnn_layer_size,
use_gru=args.use_gru,
share_rnn_weights=args.share_rnn_weights)
batch_reader = data_generator.batch_reader_creator(
manifest_path=args.tune_manifest,
batch_size=args.batch_size,
sortagrad=False,
shuffle_method=None)
# load parameters
if not os.path.isfile(args.model_path):
raise IOError("Invaid model path: %s" % args.model_path)
parameters = paddle.parameters.Parameters.from_tar(
gzip.open(args.model_path))
inferer = paddle.inference.Inference(
output_layer=output_probs, parameters=parameters)
# decoders only accept string encoded in utf-8
vocab_list = [chars.encode("utf-8") for chars in data_generator.vocab_list]
# init logger
logger = logging.getLogger("")
logger.setLevel(level=logging.INFO)
# init external scorer
logger.info("begin to initialize the external scorer for tuning")
if not os.path.isfile(args.lang_model_path):
raise IOError("Invaid language model path: %s" % args.lang_model_path)
ext_scorer = Scorer(
alpha=args.alpha_from,
beta=args.beta_from,
model_path=args.lang_model_path,
vocabulary=vocab_list)
logger.info("language model: "
"is_character_based = %d," % ext_scorer.is_character_based() +
" max_order = %d," % ext_scorer.get_max_order() +
" dict_size = %d" % ext_scorer.get_dict_size())
logger.info("end initializing scorer. Start tuning ...")
error_rate_func = cer if args.error_rate_type == 'cer' else wer
# create grid for search
cand_alphas = np.linspace(args.alpha_from, args.alpha_to, args.num_alphas)
cand_betas = np.linspace(args.beta_from, args.beta_to, args.num_betas)
params_grid = [(alpha, beta) for alpha in cand_alphas
for beta in cand_betas]
err_sum = [0.0 for i in xrange(len(params_grid))]
err_ave = [0.0 for i in xrange(len(params_grid))]
num_ins, cur_batch = 0, 0
## incremental tuning parameters over multiple batches
for infer_data in batch_reader():
if (args.num_batches >= 0) and (cur_batch >= args.num_batches):
break
infer_results = inferer.infer(input=infer_data,
feeding=data_generator.feeding)
start_pos = [0] * (len(infer_data) + 1)
for i in xrange(len(infer_data)):
start_pos[i + 1] = start_pos[i] + infer_data[i][3][0]
probs_split = [
infer_results[start_pos[i]:start_pos[i + 1]]
for i in xrange(0, len(infer_data))
]
target_transcripts = [ data[1] for data in infer_data ]
num_ins += len(target_transcripts)
# grid search
for index, (alpha, beta) in enumerate(params_grid):
# reset alpha & beta
ext_scorer.reset_params(alpha, beta)
beam_search_results = ctc_beam_search_decoder_batch(
probs_split=probs_split,
vocabulary=vocab_list,
beam_size=args.beam_size,
num_processes=args.num_proc_bsearch,
cutoff_prob=args.cutoff_prob,
cutoff_top_n=args.cutoff_top_n,
ext_scoring_func=ext_scorer, )
result_transcripts = [res[0][1] for res in beam_search_results]
for target, result in zip(target_transcripts, result_transcripts):
err_sum[index] += error_rate_func(target, result)
err_ave[index] = err_sum[index] / num_ins
if index % 2 == 0:
sys.stdout.write('.')
sys.stdout.flush()
# output on-line tuning result at the end of current batch
err_ave_min = min(err_ave)
min_index = err_ave.index(err_ave_min)
print("\nBatch %d [%d/?], current opt (alpha, beta) = (%s, %s), "
" min [%s] = %f" %(cur_batch, num_ins,
"%.3f" % params_grid[min_index][0],
"%.3f" % params_grid[min_index][1],
args.error_rate_type, err_ave_min))
cur_batch += 1
# output WER/CER at every (alpha, beta)
print("\nFinal %s:\n" % args.error_rate_type)
for index in xrange(len(params_grid)):
print("(alpha, beta) = (%s, %s), [%s] = %f"
% ("%.3f" % params_grid[index][0], "%.3f" % params_grid[index][1],
args.error_rate_type, err_ave[index]))
err_ave_min = min(err_ave)
min_index = err_ave.index(err_ave_min)
print("\nFinish tuning on %d batches, final opt (alpha, beta) = (%s, %s)"
% (args.num_batches, "%.3f" % params_grid[min_index][0],
"%.3f" % params_grid[min_index][1]))
logger.info("finish tuning")
def main():
print_arguments(args)
paddle.init(use_gpu=args.use_gpu,
rnn_use_batch=True,
trainer_count=args.trainer_count)
tune()
if __name__ == '__main__':
main()
deep_speech_2/train.py 已删除 100644 → 0
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"""Trainer for DeepSpeech2 model."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import argparse
import functools
import paddle.v2 as paddle
from model_utils.model import DeepSpeech2Model
from data_utils.data import DataGenerator
from utils.utility import add_arguments, print_arguments
parser = argparse.ArgumentParser(description=__doc__)
add_arg = functools.partial(add_arguments, argparser=parser)
# yapf: disable
add_arg('batch_size', int, 256, "Minibatch size.")
add_arg('trainer_count', int, 8, "# of Trainers (CPUs or GPUs).")
add_arg('num_passes', int, 200, "# of training epochs.")
add_arg('num_proc_data', int, 16, "# of CPUs for data preprocessing.")
add_arg('num_conv_layers', int, 2, "# of convolution layers.")
add_arg('num_rnn_layers', int, 3, "# of recurrent layers.")
add_arg('rnn_layer_size', int, 2048, "# of recurrent cells per layer.")
add_arg('num_iter_print', int, 100, "Every # iterations for printing "
"train cost.")
add_arg('learning_rate', float, 5e-4, "Learning rate.")
add_arg('max_duration', float, 27.0, "Longest audio duration allowed.")
add_arg('min_duration', float, 0.0, "Shortest audio duration allowed.")
add_arg('test_off', bool, False, "Turn off testing.")
add_arg('use_sortagrad', bool, True, "Use SortaGrad or not.")
add_arg('use_gpu', bool, True, "Use GPU or not.")
add_arg('use_gru', bool, False, "Use GRUs instead of simple RNNs.")
add_arg('is_local', bool, True, "Use pserver or not.")
add_arg('share_rnn_weights',bool, True, "Share input-hidden weights across "
"bi-directional RNNs. Not for GRU.")
add_arg('train_manifest', str,
'data/librispeech/manifest.train',
"Filepath of train manifest.")
add_arg('dev_manifest', str,
'data/librispeech/manifest.dev-clean',
"Filepath of validation manifest.")
add_arg('mean_std_path', str,
'data/librispeech/mean_std.npz',
"Filepath of normalizer's mean & std.")
add_arg('vocab_path', str,
'data/librispeech/vocab.txt',
"Filepath of vocabulary.")
add_arg('init_model_path', str,
None,
"If None, the training starts from scratch, "
"otherwise, it resumes from the pre-trained model.")
add_arg('output_model_dir', str,
"./checkpoints/libri",
"Directory for saving checkpoints.")
add_arg('augment_conf_path',str,
'conf/augmentation.config',
"Filepath of augmentation configuration file (json-format).")
add_arg('specgram_type', str,
'linear',
"Audio feature type. Options: linear, mfcc.",
choices=['linear', 'mfcc'])
add_arg('shuffle_method', str,
'batch_shuffle_clipped',
"Shuffle method.",
choices=['instance_shuffle', 'batch_shuffle', 'batch_shuffle_clipped'])
# yapf: disable
args = parser.parse_args()
def train():
"""DeepSpeech2 training."""
train_generator = DataGenerator(
vocab_filepath=args.vocab_path,
mean_std_filepath=args.mean_std_path,
augmentation_config=open(args.augment_conf_path, 'r').read(),
max_duration=args.max_duration,
min_duration=args.min_duration,
specgram_type=args.specgram_type,
num_threads=args.num_proc_data,
num_conv_layers=args.num_conv_layers)
dev_generator = DataGenerator(
vocab_filepath=args.vocab_path,
mean_std_filepath=args.mean_std_path,
augmentation_config="{}",
specgram_type=args.specgram_type,
num_threads=args.num_proc_data,
num_conv_layers=args.num_conv_layers)
train_batch_reader = train_generator.batch_reader_creator(
manifest_path=args.train_manifest,
batch_size=args.batch_size,
min_batch_size=args.trainer_count,
sortagrad=args.use_sortagrad if args.init_model_path is None else False,
shuffle_method=args.shuffle_method)
dev_batch_reader = dev_generator.batch_reader_creator(
manifest_path=args.dev_manifest,
batch_size=args.batch_size,
min_batch_size=1, # must be 1, but will have errors.
sortagrad=False,
shuffle_method=None)
ds2_model = DeepSpeech2Model(
vocab_size=train_generator.vocab_size,
num_conv_layers=args.num_conv_layers,
num_rnn_layers=args.num_rnn_layers,
rnn_layer_size=args.rnn_layer_size,
use_gru=args.use_gru,
pretrained_model_path=args.init_model_path,
share_rnn_weights=args.share_rnn_weights)
ds2_model.train(
train_batch_reader=train_batch_reader,
dev_batch_reader=dev_batch_reader,
feeding_dict=train_generator.feeding,
learning_rate=args.learning_rate,
gradient_clipping=400,
num_passes=args.num_passes,
num_iterations_print=args.num_iter_print,
output_model_dir=args.output_model_dir,
is_local=args.is_local,
test_off=args.test_off)
def main():
print_arguments(args)
paddle.init(use_gpu=args.use_gpu,
rnn_use_batch=True,
trainer_count=args.trainer_count,
log_clipping=True)
train()
if __name__ == '__main__':
main()
deep_speech_2/utils/error_rate.py 已删除 100644 → 0
浏览文件 @ 788f7011
# -*- coding: utf-8 -*-
"""This module provides functions to calculate error rate in different level.
e.g. wer for word-level, cer for char-level.
"""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import numpy as np
def _levenshtein_distance(ref, hyp):
"""Levenshtein distance is a string metric for measuring the difference
between two sequences. Informally, the levenshtein disctance is defined as
the minimum number of single-character edits (substitutions, insertions or
deletions) required to change one word into the other. We can naturally
extend the edits to word level when calculate levenshtein disctance for
two sentences.
"""
m = len(ref)
n = len(hyp)
# special case
if ref == hyp:
return 0
if m == 0:
return n
if n == 0:
return m
if m < n:
ref, hyp = hyp, ref
m, n = n, m
# use O(min(m, n)) space
distance = np.zeros((2, n + 1), dtype=np.int32)
# initialize distance matrix
for j in xrange(n + 1):
distance[0][j] = j
# calculate levenshtein distance
for i in xrange(1, m + 1):
prev_row_idx = (i - 1) % 2
cur_row_idx = i % 2
distance[cur_row_idx][0] = i
for j in xrange(1, n + 1):
if ref[i - 1] == hyp[j - 1]:
distance[cur_row_idx][j] = distance[prev_row_idx][j - 1]
else:
s_num = distance[prev_row_idx][j - 1] + 1
i_num = distance[cur_row_idx][j - 1] + 1
d_num = distance[prev_row_idx][j] + 1
distance[cur_row_idx][j] = min(s_num, i_num, d_num)
return distance[m % 2][n]
def wer(reference, hypothesis, ignore_case=False, delimiter=' '):
"""Calculate word error rate (WER). WER compares reference text and
hypothesis text in word-level. WER is defined as:
.. math::
WER = (Sw + Dw + Iw) / Nw
where
.. code-block:: text
Sw is the number of words subsituted,
Dw is the number of words deleted,
Iw is the number of words inserted,
Nw is the number of words in the reference
We can use levenshtein distance to calculate WER. Please draw an attention
that empty items will be removed when splitting sentences by delimiter.
:param reference: The reference sentence.
:type reference: basestring
:param hypothesis: The hypothesis sentence.
:type hypothesis: basestring
:param ignore_case: Whether case-sensitive or not.
:type ignore_case: bool
:param delimiter: Delimiter of input sentences.
:type delimiter: char
:return: Word error rate.
:rtype: float
:raises ValueError: If the reference length is zero.
"""
if ignore_case == True:
reference = reference.lower()
hypothesis = hypothesis.lower()
ref_words = filter(None, reference.split(delimiter))
hyp_words = filter(None, hypothesis.split(delimiter))
if len(ref_words) == 0:
raise ValueError("Reference's word number should be greater than 0.")
edit_distance = _levenshtein_distance(ref_words, hyp_words)
wer = float(edit_distance) / len(ref_words)
return wer
def cer(reference, hypothesis, ignore_case=False, remove_space=False):
"""Calculate charactor error rate (CER). CER compares reference text and
hypothesis text in char-level. CER is defined as:
.. math::
CER = (Sc + Dc + Ic) / Nc
where
.. code-block:: text
Sc is the number of characters substituted,
Dc is the number of characters deleted,
Ic is the number of characters inserted
Nc is the number of characters in the reference
We can use levenshtein distance to calculate CER. Chinese input should be
encoded to unicode. Please draw an attention that the leading and tailing
space characters will be truncated and multiple consecutive space
characters in a sentence will be replaced by one space character.
:param reference: The reference sentence.
:type reference: basestring
:param hypothesis: The hypothesis sentence.
:type hypothesis: basestring
:param ignore_case: Whether case-sensitive or not.
:type ignore_case: bool
:param remove_space: Whether remove internal space characters
:type remove_space: bool
:return: Character error rate.
:rtype: float
:raises ValueError: If the reference length is zero.
"""
if ignore_case == True:
reference = reference.lower()
hypothesis = hypothesis.lower()
join_char = ' '
if remove_space == True:
join_char = ''
reference = join_char.join(filter(None, reference.split(' ')))
hypothesis = join_char.join(filter(None, hypothesis.split(' ')))
if len(reference) == 0:
raise ValueError("Length of reference should be greater than 0.")
edit_distance = _levenshtein_distance(reference, hypothesis)
cer = float(edit_distance) / len(reference)
return cer
deep_speech_2/utils/tests/test_error_rate.py 已删除 100644 → 0
浏览文件 @ 788f7011
# -*- coding: utf-8 -*-
"""Test error rate."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import unittest
from utils import error_rate
class TestParse(unittest.TestCase):
def test_wer_1(self):
ref = 'i UM the PHONE IS i LEFT THE portable PHONE UPSTAIRS last night'
hyp = 'i GOT IT TO the FULLEST i LOVE TO portable FROM OF STORES last '\
'night'
word_error_rate = error_rate.wer(ref, hyp)
self.assertTrue(abs(word_error_rate - 0.769230769231) < 1e-6)
def test_wer_2(self):
ref = 'as any in england i would say said gamewell proudly that is '\
'in his day'
hyp = 'as any in england i would say said came well proudly that is '\
'in his day'
word_error_rate = error_rate.wer(ref, hyp)
self.assertTrue(abs(word_error_rate - 0.1333333) < 1e-6)
def test_wer_3(self):
ref = 'the lieutenant governor lilburn w boggs afterward governor '\
'was a pronounced mormon hater and throughout the period of '\
'the troubles he manifested sympathy with the persecutors'
hyp = 'the lieutenant governor little bit how bags afterward '\
'governor was a pronounced warman hater and throughout the '\
'period of th troubles he manifests sympathy with the '\
'persecutors'
word_error_rate = error_rate.wer(ref, hyp)
self.assertTrue(abs(word_error_rate - 0.2692307692) < 1e-6)
def test_wer_4(self):
ref = 'the wood flamed up splendidly under the large brewing copper '\
'and it sighed so deeply'
hyp = 'the wood flame do splendidly under the large brewing copper '\
'and its side so deeply'
word_error_rate = error_rate.wer(ref, hyp)
self.assertTrue(abs(word_error_rate - 0.2666666667) < 1e-6)
def test_wer_5(self):
ref = 'all the morning they trudged up the mountain path and at noon '\
'unc and ojo sat on a fallen tree trunk and ate the last of '\
'the bread which the old munchkin had placed in his pocket'
hyp = 'all the morning they trudged up the mountain path and at noon '\
'unc in ojo sat on a fallen tree trunk and ate the last of '\
'the bread which the old munchkin had placed in his pocket'
word_error_rate = error_rate.wer(ref, hyp)
self.assertTrue(abs(word_error_rate - 0.027027027) < 1e-6)
def test_wer_6(self):
ref = 'i UM the PHONE IS i LEFT THE portable PHONE UPSTAIRS last night'
word_error_rate = error_rate.wer(ref, ref)
self.assertEqual(word_error_rate, 0.0)
def test_wer_7(self):
ref = ' '
hyp = 'Hypothesis sentence'
with self.assertRaises(ValueError):
word_error_rate = error_rate.wer(ref, hyp)
def test_cer_1(self):
ref = 'werewolf'
hyp = 'weae wolf'
char_error_rate = error_rate.cer(ref, hyp)
self.assertTrue(abs(char_error_rate - 0.25) < 1e-6)
def test_cer_2(self):
ref = 'werewolf'
hyp = 'weae wolf'
char_error_rate = error_rate.cer(ref, hyp, remove_space=True)
self.assertTrue(abs(char_error_rate - 0.125) < 1e-6)
def test_cer_3(self):
ref = 'were wolf'
hyp = 'were wolf'
char_error_rate = error_rate.cer(ref, hyp)
self.assertTrue(abs(char_error_rate - 0.0) < 1e-6)
def test_cer_4(self):
ref = 'werewolf'
char_error_rate = error_rate.cer(ref, ref)
self.assertEqual(char_error_rate, 0.0)
def test_cer_5(self):
ref = u'我是中国人'
hyp = u'我是 美洲人'
char_error_rate = error_rate.cer(ref, hyp)
self.assertTrue(abs(char_error_rate - 0.6) < 1e-6)
def test_cer_6(self):
ref = u'我 是 中 国 人'
hyp = u'我 是 美 洲 人'
char_error_rate = error_rate.cer(ref, hyp, remove_space=True)
self.assertTrue(abs(char_error_rate - 0.4) < 1e-6)
def test_cer_7(self):
ref = u'我是中国人'
char_error_rate = error_rate.cer(ref, ref)
self.assertFalse(char_error_rate, 0.0)
def test_cer_8(self):
ref = ''
hyp = 'Hypothesis'
with self.assertRaises(ValueError):
char_error_rate = error_rate.cer(ref, hyp)
if __name__ == '__main__':
unittest.main()
deep_speech_2/utils/utility.py 已删除 100644 → 0
浏览文件 @ 788f7011
"""Contains common utility functions."""
from __future__ import absolute_import
from __future__ import division
from __future__ import print_function
import distutils.util
def print_arguments(args):
"""Print argparse's arguments.
Usage:
.. code-block:: python
parser = argparse.ArgumentParser()
parser.add_argument("name", default="Jonh", type=str, help="User name.")
args = parser.parse_args()
print_arguments(args)
:param args: Input argparse.Namespace for printing.
:type args: argparse.Namespace
"""
print("----------- Configuration Arguments -----------")
for arg, value in sorted(vars(args).iteritems()):
print("%s: %s" % (arg, value))
print("------------------------------------------------")
def add_arguments(argname, type, default, help, argparser, **kwargs):
"""Add argparse's argument.
Usage:
.. code-block:: python
parser = argparse.ArgumentParser()
add_argument("name", str, "Jonh", "User name.", parser)
args = parser.parse_args()
"""
type = distutils.util.strtobool if type == bool else type
argparser.add_argument(
"--" + argname,
default=default,
type=type,
help=help + ' Default: %(default)s.',
**kwargs)
deep_speech_2/utils/utility.sh 已删除 100644 → 0
浏览文件 @ 788f7011
download() {
URL=$1
MD5=$2
TARGET=$3
if [ -e $TARGET ]; then
md5_result=`md5sum $TARGET | awk -F[' '] '{print $1}'`
if [ $MD5 == $md5_result ]; then
echo "$TARGET already exists, download skipped."
return 0
fi
fi
wget -c $URL -O "$TARGET"
if [ $? -ne 0 ]; then
return 1
fi
md5_result=`md5sum $TARGET | awk -F[' '] '{print $1}'`
if [ ! $MD5 == $md5_result ]; then
return 1
fi
}
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