deepspeech 0.6.0a7

A library for running inference on a DeepSpeech model

Project DeepSpeech

DeepSpeech is an open source Speech-To-Text engine, using a model trained by machine learning techniques based on Baidu's Deep Speech research paper. Project DeepSpeech uses Google's TensorFlow to make the implementation easier.

To install and use deepspeech all you have to do is:

# Create and activate a virtualenv
virtualenv -p python3 $HOME/tmp/deepspeech-venv/
source $HOME/tmp/deepspeech-venv/bin/activate

# Install DeepSpeech
pip3 install deepspeech

# Download pre-trained English model and extract
curl -LO https://github.com/mozilla/DeepSpeech/releases/download/v0.5.1/deepspeech-0.5.1-models.tar.gz
tar xvf deepspeech-0.5.1-models.tar.gz

# Download example audio files
curl -LO https://github.com/mozilla/DeepSpeech/releases/download/v0.5.1/audio-0.5.1.tar.gz
tar xvf audio-0.5.1.tar.gz

# Transcribe an audio file
deepspeech --model deepspeech-0.5.1-models/output_graph.pbmm --alphabet deepspeech-0.5.1-models/alphabet.txt --lm deepspeech-0.5.1-models/lm.binary --trie deepspeech-0.5.1-models/trie --audio audio/2830-3980-0043.wav

A pre-trained English model is available for use and can be downloaded using the instructions below. Currently, only 16-bit, 16 kHz, mono-channel WAVE audio files are supported in the Python client. A package with some example audio files is available for download in our release notes.

Quicker inference can be performed using a supported NVIDIA GPU on Linux. See the release notes to find which GPUs are supported. To run deepspeech on a GPU, install the GPU specific package:

# Create and activate a virtualenv
virtualenv -p python3 $HOME/tmp/deepspeech-gpu-venv/
source $HOME/tmp/deepspeech-gpu-venv/bin/activate

# Install DeepSpeech CUDA enabled package
pip3 install deepspeech-gpu

# Transcribe an audio file.
deepspeech --model deepspeech-0.5.1-models/output_graph.pbmm --alphabet deepspeech-0.5.1-models/alphabet.txt --lm deepspeech-0.5.1-models/lm.binary --trie deepspeech-0.5.1-models/trie --audio audio/2830-3980-0043.wav

Please ensure you have the required CUDA dependencies.

See the output of deepspeech -h for more information on the use of deepspeech. (If you experience problems running deepspeech, please check required runtime dependencies).

---

Table of Contents

• Using a Pre-trained Model
  • CUDA dependency
  • Getting the pre-trained model
  • Model compatibility
  • Using the Python package
  • Using the Node.JS package
  • Using the Command Line client
  • Installing bindings from source
  • Third party bindings
• Training your own Model
  • Prerequisites for training a model
  • Getting the training code
  • Installing Python dependencies
  • Recommendations
  • Common Voice training data
  • Training a model
  • Checkpointing
  • Exporting a model for inference
  • Exporting a model for TFLite
  • Making a mmap-able model for inference
  • Continuing training from a release model
  • Training with Augmentation
• Contribution guidelines
• Contact/Getting Help

Using a Pre-trained Model

Inference using a DeepSpeech pre-trained model can be done with a client/language binding package. We have four clients/language bindings in this repository, listed below, and also a few community-maintained clients/language bindings in other repositories, listed further down in this README.

• The Python package/language binding
• The Node.JS package/language binding
• The Command-Line client
• The .NET client/language binding

Running deepspeech might, see below, require some runtime dependencies to be already installed on your system:

• sox - The Python and Node.JS clients use SoX to resample files to 16kHz.
• libgomp1 - libsox (statically linked into the clients) depends on OpenMP. Some people have had to install this manually.
• libstdc++ - Standard C++ Library implementation. Some people have had to install this manually.
• libpthread - On Linux, some people have had to install libpthread manually.

Please refer to your system's documentation on how to install these dependencies.

CUDA dependency

The GPU capable builds (Python, NodeJS, C++, etc) depend on the same CUDA runtime as upstream TensorFlow. Currently with TensorFlow 1.14 it depends on CUDA 10.0 and CuDNN v7.5. See the TensorFlow documentation.

Getting the pre-trained model

If you want to use the pre-trained English model for performing speech-to-text, you can download it (along with other important inference material) from the DeepSpeech releases page. Alternatively, you can run the following command to download and unzip the model files in your current directory:

wget https://github.com/mozilla/DeepSpeech/releases/download/v0.5.1/deepspeech-0.5.1-models.tar.gz
tar xvfz deepspeech-0.5.1-models.tar.gz

Model compatibility

DeepSpeech models are versioned to keep you from trying to use an incompatible graph with a newer client after a breaking change was made to the code. If you get an error saying your model file version is too old for the client, you should either upgrade to a newer model release, re-export your model from the checkpoint using a newer version of the code, or downgrade your client if you need to use the old model and can't re-export it.

Using the Python package

Pre-built binaries which can be used for performing inference with a trained model can be installed with pip3. You can then use the deepspeech binary to do speech-to-text on an audio file:

For the Python bindings, it is highly recommended that you perform the installation within a Python 3.5 or later virtual environment. You can find more information about those in this documentation.

We will continue under the assumption that you already have your system properly setup to create new virtual environments.

Create a DeepSpeech 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-venv. You can create it using this command:

$ virtualenv -p python3 $HOME/tmp/deepspeech-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-venv/bin/activate

Installing DeepSpeech Python bindings

Once your environment has been set-up and loaded, you can use pip3 to manage packages locally. On a fresh setup of the virtualenv, you will have to install the DeepSpeech wheel. You can check if deepspeech is already installed with pip3 list.

To perform the installation, just use pip3 as such:

$ pip3 install deepspeech

If deepspeech is already installed, you can update it as such:

$ pip3 install --upgrade deepspeech

Alternatively, if you have a supported NVIDIA GPU on Linux, you can install the GPU specific package as follows:

$ pip3 install deepspeech-gpu

See the release notes to find which GPUs are supported. Please ensure you have the required CUDA dependency.

You can update deepspeech-gpu as follows:

$ pip3 install --upgrade deepspeech-gpu

In both cases, pip3 should take care of installing all the required dependencies. After installation has finished, you should be able to call deepspeech from the command-line.

Note: the following command assumes you downloaded the pre-trained model.

deepspeech --model models/output_graph.pbmm --alphabet models/alphabet.txt --lm models/lm.binary --trie models/trie --audio my_audio_file.wav

The arguments --lm and --trie are optional, and represent a language model.

See client.py for an example of how to use the package programatically.

Using the Node.JS package

You can download the Node.JS bindings using npm:

npm install deepspeech

Please note that as of now, we only support Node.JS versions 4, 5 and 6. Once SWIG has support we can build for newer versions.

Alternatively, if you're using Linux and have a supported NVIDIA GPU, you can install the GPU specific package as follows:

npm install deepspeech-gpu

See the release notes to find which GPUs are supported. Please ensure you have the required CUDA dependency.

See client.js for an example of how to use the bindings. Or download the wav example.

Using the Command-Line client

To download the pre-built binaries for the deepspeech command-line (compiled C++) client, use util/taskcluster.py:

python3 util/taskcluster.py --target .

or if you're on macOS:

python3 util/taskcluster.py --arch osx --target .

also, if you need some binaries different than current master, like v0.2.0-alpha.6, you can use --branch:

python3 util/taskcluster.py --branch "v0.2.0-alpha.6" --target "."

The script taskcluster.py will download native_client.tar.xz (which includes the deepspeech binary, generate_trie and associated libraries) and extract it into the current folder. Also, taskcluster.py will download binaries for Linux/x86_64 by default, but you can override that behavior with the --arch parameter. See the help info with python util/taskcluster.py -h for more details. Specific branches of DeepSpeech or TensorFlow can be specified as well.

Note: the following command assumes you downloaded the pre-trained model.

./deepspeech --model models/output_graph.pbmm --alphabet models/alphabet.txt --lm models/lm.binary --trie models/trie --audio audio_input.wav

See the help output with ./deepspeech -h and the native client README for more details.

Installing bindings from source

If pre-built binaries aren't available for your system, you'll need to install them from scratch. Follow these native_client installation instructions.

Third party bindings

In addition to the bindings above, third party developers have started to provide bindings to other languages:

• Asticode provides Golang bindings in its go-astideepspeech repo.
• RustAudio provide a Rust binding, the installation and use of which is described in their deepspeech-rs repo.
• stes provides preliminary PKGBUILDs to install the client and python bindings on Arch Linux in the arch-deepspeech repo.
• gst-deepspeech provides a GStreamer plugin which can be used from any language with GStreamer bindings.

Training Your Own Model

Prerequisites for training a model

• Python 3.6
• Git Large File Storage
• Mac or Linux environment

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.

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

Contribution guidelines

This repository is governed by Mozilla's code of conduct and etiquette guidelines. For more details, please read the Mozilla Community Participation Guidelines.

Before making a Pull Request, check your changes for basic mistakes and style problems by using a linter. We have cardboardlinter setup in this repository, so for example, if you've made some changes and would like to run the linter on just the changed code, you can use the follow command:

pip install pylint cardboardlint
cardboardlinter --refspec master

This will compare the code against master and run the linter on all the changes. We plan to introduce more linter checks (e.g. for C++) in the future. To run it automatically as a git pre-commit hook, do the following:

cat <<\EOF > .git/hooks/pre-commit
#!/bin/bash
if [ ! -x "$(command -v cardboardlinter)" ]; then
    exit 0
fi

# First, stash index and work dir, keeping only the
# to-be-committed changes in the working directory.
echo "Stashing working tree changes..." 1>&2
old_stash=$(git rev-parse -q --verify refs/stash)
git stash save -q --keep-index
new_stash=$(git rev-parse -q --verify refs/stash)

# If there were no changes (e.g., `--amend` or `--allow-empty`)
# then nothing was stashed, and we should skip everything,
# including the tests themselves.  (Presumably the tests passed
# on the previous commit, so there is no need to re-run them.)
if [ "$old_stash" = "$new_stash" ]; then
    echo "No changes, skipping lint." 1>&2
    exit 0
fi

# Run tests
cardboardlinter --refspec HEAD -n auto
status=$?

# Restore changes
echo "Restoring working tree changes..." 1>&2
git reset --hard -q && git stash apply --index -q && git stash drop -q

# Exit with status from test-run: nonzero prevents commit
exit $status
EOF
chmod +x .git/hooks/pre-commit

This will run the linters on just the changes made in your commit.

Contact/Getting Help

There are several ways to contact us or to get help:

1. FAQ - We have a list of common questions, and their answers, in our FAQ. When just getting started, it's best to first check the FAQ to see if your question is addressed.

2. Discourse Forums - If your question is not addressed in the FAQ, the Discourse Forums is the next place to look. They contain conversations on General Topics, Using Deep Speech, and Deep Speech Development.

3. IRC - If your question is not addressed by either the FAQ or Discourse Forums, you can contact us on the #machinelearning channel on Mozilla IRC; people there can try to answer/help

4. Issues - Finally, if all else fails, you can open an issue in our repo.