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Training Your Own Model

Prerequisites for training a model

Getting the training code

Install Git Large File Storage either manually or through a package-manager if available on your system. Then clone the DeepSpeech repository normally:

git clone https://github.com/mozilla/DeepSpeech

Creating a virtual environment

In creating a virtual environment you will create a directory containing a python3 binary and everything needed to run deepspeech. You can use whatever directory you want. For the purpose of the documentation, we will rely on $HOME/tmp/deepspeech-train-venv. You can create it using this command:

$ virtualenv -p python3 $HOME/tmp/deepspeech-train-venv/

Once this command completes successfully, the environment will be ready to be activated.

Activating the environment

Each time you need to work with DeepSpeech, you have to activate this virtual environment. This is done with this simple command:

$ source $HOME/tmp/deepspeech-train-venv/bin/activate

Installing Python dependencies

Install the required dependencies using pip3:

cd DeepSpeech
pip3 install -r requirements.txt

You'll also need to install the ds_ctcdecoder Python package. ds_ctcdecoder is required for decoding the outputs of the deepspeech acoustic model into text. You can use util/taskcluster.py with the --decoder flag to get a URL to a binary of the decoder package appropriate for your platform and Python version:

pip3 install $(python3 util/taskcluster.py --decoder)

This command will download and install the ds_ctcdecoder package. You can override the platform with --arch if you want the package for ARM7 (--arch arm) or ARM64 (--arch arm64). If you prefer building the ds_ctcdecoder package from source, see the native_client README file.

Recommendations

If you have a capable (NVIDIA, at least 8GB of VRAM) GPU, it is highly recommended to install TensorFlow with GPU support. Training will be significantly faster than using the CPU. To enable GPU support, you can do:

pip3 uninstall tensorflow
pip3 install 'tensorflow-gpu==1.14.0'

Please ensure you have the required CUDA dependency.

It has been reported for some people failure at training:

tensorflow.python.framework.errors_impl.UnknownError: Failed to get convolution algorithm. This is probably because cuDNN failed to initialize, so try looking to see if a warning log message was printed above.
     [[{{node tower_0/conv1d/Conv2D}}]]

Setting the TF_FORCE_GPU_ALLOW_GROWTH environment variable to true seems to help in such cases. This could also be due to an incorrect version of libcudnn. Double check your versions with the TensorFlow 1.14 documentation.

Common Voice training data

The Common Voice corpus consists of voice samples that were donated through Mozilla's Common Voice Initiative. You can download individual CommonVoice v2.0 language data sets from here. After extraction of such a data set, you'll find the following contents:

  • the *.tsv files output by CorporaCreator for the downloaded language
  • the mp3 audio files they reference in a clips sub-directory.

For bringing this data into a form that DeepSpeech understands, you have to run the CommonVoice v2.0 importer (bin/import_cv2.py):

bin/import_cv2.py --filter_alphabet path/to/some/alphabet.txt /path/to/extracted/language/archive

Providing a filter alphabet is optional. It will exclude all samples whose transcripts contain characters not in the specified alphabet. Running the importer with -h will show you some additional options.

Once the import is done, the clips sub-directory will contain for each required .mp3 an additional .wav file. It will also add the following .csv files:

  • clips/train.csv
  • clips/dev.csv
  • clips/test.csv

All entries in these CSV files refer to their samples by absolute paths. So moving this sub-directory would require another import or tweaking the CSV files accordingly.

To use Common Voice data during training, validation and testing, you pass (comma separated combinations of) their filenames into --train_files, --dev_files, --test_files parameters of DeepSpeech.py.

If, for example, Common Voice language en was extracted to ../data/CV/en/, DeepSpeech.py could be called like this:

./DeepSpeech.py --train_files ../data/CV/en/clips/train.csv --dev_files ../data/CV/en/clips/dev.csv --test_files ../data/CV/en/clips/test.csv

Training a model

The central (Python) script is DeepSpeech.py in the project's root directory. For its list of command line options, you can call:

./DeepSpeech.py --helpfull

To get the output of this in a slightly better-formatted way, you can also look up the option definitions in ``util/flags.py` <util/flags.py>`_.

For executing pre-configured training scenarios, there is a collection of convenience scripts in the bin folder. Most of them are named after the corpora they are configured for. Keep in mind that most speech corpora are very large, on the order of tens of gigabytes, and some aren't free. Downloading and preprocessing them can take a very long time, and training on them without a fast GPU (GTX 10 series or newer recommended) takes even longer.

If you experience GPU OOM errors while training, try reducing the batch size with the ``--train_batch_size``, ``--dev_batch_size`` and ``--test_batch_size`` parameters.

As a simple first example you can open a terminal, change to the directory of the DeepSpeech checkout, activate the virtualenv created above, and run:

./bin/run-ldc93s1.sh

This script will train on a small sample dataset composed of just a single audio file, the sample file for the TIMIT Acoustic-Phonetic Continuous Speech Corpus, which can be overfitted on a GPU in a few minutes for demonstration purposes. From here, you can alter any variables with regards to what dataset is used, how many training iterations are run and the default values of the network parameters.

Feel also free to pass additional (or overriding) DeepSpeech.py parameters to these scripts. Then, just run the script to train the modified network.

Each dataset has a corresponding importer script in bin/ that can be used to download (if it's freely available) and preprocess the dataset. See bin/import_librivox.py for an example of how to import and preprocess a large dataset for training with DeepSpeech.

If you've run the old importers (in util/importers/), they could have removed source files that are needed for the new importers to run. In that case, simply remove the extracted folders and let the importer extract and process the dataset from scratch, and things should work.

Checkpointing

During training of a model so-called checkpoints will get stored on disk. This takes place at a configurable time interval. The purpose of checkpoints is to allow interruption (also in the case of some unexpected failure) and later continuation of training without losing hours of training time. Resuming from checkpoints happens automatically by just (re)starting training with the same --checkpoint_dir of the former run.

Be aware however that checkpoints are only valid for the same model geometry they had been generated from. In other words: If there are error messages of certain Tensors having incompatible dimensions, this is most likely due to an incompatible model change. One usual way out would be to wipe all checkpoint files in the checkpoint directory or changing it before starting the training.

Exporting a model for inference

If the --export_dir parameter is provided, a model will have been exported to this directory during training. Refer to the corresponding README.md for information on building and running a client that can use the exported model.

Exporting a model for TFLite

If you want to experiment with the TF Lite engine, you need to export a model that is compatible with it, then use the --export_tflite flags. If you already have a trained model, you can re-export it for TFLite by running DeepSpeech.py again and specifying the same checkpoint_dir that you used for training, as well as passing --export_tflite --export_dir /model/export/destination.

Making a mmap-able model for inference

The output_graph.pb model file generated in the above step will be loaded in memory to be dealt with when running inference. This will result in extra loading time and memory consumption. One way to avoid this is to directly read data from the disk.

TensorFlow has tooling to achieve this: it requires building the target //tensorflow/contrib/util:convert_graphdef_memmapped_format (binaries are produced by our TaskCluster for some systems including Linux/amd64 and macOS/amd64), use util/taskcluster.py tool to download, specifying tensorflow as a source and convert_graphdef_memmapped_format as artifact.

Producing a mmap-able model is as simple as:

$ convert_graphdef_memmapped_format --in_graph=output_graph.pb --out_graph=output_graph.pbmm

Upon sucessfull run, it should report about conversion of a non-zero number of nodes. If it reports converting 0 nodes, something is wrong: make sure your model is a frozen one, and that you have not applied any incompatible changes (this includes quantize_weights).

Continuing training from a release model

If you'd like to use one of the pre-trained models released by Mozilla to bootstrap your training process (transfer learning, fine tuning), you can do so by using the --checkpoint_dir flag in DeepSpeech.py. Specify the path where you downloaded the checkpoint from the release, and training will resume from the pre-trained model.

For example, if you want to fine tune the entire graph using your own data in my-train.csv, my-dev.csv and my-test.csv, for three epochs, you can something like the following, tuning the hyperparameters as needed:

mkdir fine_tuning_checkpoints
python3 DeepSpeech.py --n_hidden 2048 --checkpoint_dir path/to/checkpoint/folder --epochs 3 --train_files my-train.csv --dev_files my-dev.csv --test_files my_dev.csv --learning_rate 0.0001

Note: the released models were trained with --n_hidden 2048, so you need to use that same value when initializing from the release models.

Training with augmentation

Augmentation is a useful technique for better generalization of machine learning models. Thus, a pre-processing pipeline with various augmentation techniques on raw pcm and spectrogram has been implemented and can be used while training the model. Following are the available augmentation techniques that can be enabled at training time by using the corresponding flags in the command line.

Audio Augmentation

  1. Standard deviation for Gaussian additive noise: --data_aug_features_additive
  2. Standard deviation for Normal distribution around 1 for multiplicative noise: --data_aug_features_multiplicative
  3. Standard deviation for speeding-up tempo. If Standard deviation is 0, this augmentation is not performed: --augmentation_speed_up_std

Spectrogram Augmentation

Inspired by Google Paper on SpecAugment: A Simple Data Augmentation Method for Automatic Speech Recognition

  1. Keep rate of dropout augmentation on a spectrogram (if 1, no dropout will be performed on the spectrogram):
    • Keep Rate : --augmentation_spec_dropout_keeprate value between range [0 - 1]
  2. Whether to use frequency and time masking augmentation:
    • Enable / Disable : --augmentation_freq_and_time_masking / --noaugmentation_freq_and_time_masking
    • Max range of masks in the frequency domain when performing freqtime-mask augmentation: --augmentation_freq_and_time_masking_freq_mask_range eg: 5
    • Number of masks in the frequency domain when performing freqtime-mask augmentation: --augmentation_freq_and_time_masking_number_freq_masks eg: 3
    • Max range of masks in the time domain when performing freqtime-mask augmentation: --augmentation_freq_and_time_masking_time_mask_rangee eg: 2
    • Number of masks in the time domain when performing freqtime-mask augmentation: augmentation_freq_and_time_masking_number_time_masks eg: 3
  3. Whether to use spectrogram speed and tempo scaling:
    • Enable / Disable : --augmentation_pitch_and_tempo_scaling / --noaugmentation_pitch_and_tempo_scaling.
    • Min value of pitch scaling: --augmentation_pitch_and_tempo_scaling_min_pitch eg:0.95
    • Max value of pitch scaling: --augmentation_pitch_and_tempo_scaling_max_pitch eg:1.2
    • Max value of tempo scaling: --augmentation_pitch_and_tempo_scaling_max_tempo eg:1.2