# DeepSpeech2 on PaddlePaddle: Design Doc We are planning to build Deep Speech 2 (DS2) \[[1](#references)\], a powerful Automatic Speech Recognition (ASR) engine, on PaddlePaddle. For the first-stage plan, we have the following short-term goals: - Release a basic distributed implementation of DS2 on PaddlePaddle. - Contribute a chapter of Deep Speech to PaddlePaddle Book. Intensive system optimization and low-latency inference library (details in \[[1](#references)\]) are not yet covered in this first-stage plan. ## Table of Contents - [Tasks](#tasks) - [Task Dependency](#task-dependency) - [Design Details](#design-details) - [Overview](#overview) - [Row Convolution](#row-convolution) - [Beam Search With CTC and LM](#beam-search-with-ctc-and-lm) - [Future Work](#future-work) - [References](#references) ## Tasks We roughly break down the project into 14 tasks: 1. Develop an **audio data provider**: - Json filelist generator. - Audio file format transformer. - Spectrogram feature extraction, power normalization etc. - Batch data reader with SortaGrad. - Data augmentation (optional). - Prepare (one or more) public English data sets & baseline. 2. Create a **simplified DS2 model configuration**: - With only fixed-length (by padding) audio sequences (otherwise need *Task 3*). - With only bidirectional-GRU (otherwise need *Task 4*). - With only greedy decoder (otherwise need *Task 5, 6*). 3. Develop to support **variable-shaped** dense-vector (image) batches of input data. - Update `DenseScanner` in `dataprovider_converter.py`, etc. 4. Develop a new **lookahead-row-convolution layer** (See \[[1](#references)\] for details): - Lookahead convolution windows. - Within-row convolution, without kernels shared across rows. 5. Build KenLM **language model** (5-gram) for beam search decoder: - Use KenLM toolkit. - Prepare the corpus & train the model. - Create infererence interfaces (for Task 6). 6. Develop a **beam search decoder** with CTC + LM + WORDCOUNT: - Beam search with CTC. - Beam search with external custom scorer (e.g. LM). - Try to design a more general beam search interface. 7. Develop a **Word Error Rate evaluator**: - update `ctc_error_evaluator`(CER) to support WER. 8. Prepare internal dataset for Mandarin (optional): - Dataset, baseline, evaluation details. - Particular data preprocessing for Mandarin. - Might need cooperating with the Speech Department. 9. Create **standard DS2 model configuration**: - With variable-length audio sequences (need *Task 3*). - With unidirectional-GRU + row-convolution (need *Task 4*). - With CTC-LM beam search decoder (need *Task 5, 6*). 10. Make it run perfectly on **clusters**. 11. Experiments and **benchmarking** (for accuracy, not efficiency): - With public English dataset. - With internal (Baidu) Mandarin dataset (optional). 12. Time **profiling** and optimization. 13. Prepare **docs**. 14. Prepare PaddlePaddle **Book** chapter with a simplified version. ## Task Dependency Tasks parallelizable within phases: Roadmap | Description | Parallelizable Tasks ----------- | :------------------------------------ | :-------------------- Phase I | Simplified model & components | *Task 1* ~ *Task 8* Phase II | Standard model & benchmarking & profiling | *Task 9* ~ *Task 12* Phase III | Documentations | *Task13* ~ *Task14* Issue for each task will be created later. Contributions, discussions and comments are all highly appreciated and welcomed! ## Design Details ### Overview Traditional **ASR** (Automatic Speech Recognition) pipelines require great human efforts devoted to elaborately tuning multiple hand-engineered components (e.g. audio feature design, accoustic model, pronuncation model and language model etc.). **Deep Speech 2** (**DS2**) \[[1](#references)\], however, trains such ASR models in an end-to-end manner, replacing most intermediate modules with only a single deep network architecture. With scaling up both the data and model sizes, DS2 achieves a very significant performance boost. Please read Deep Speech 2 \[[1](#references),[2](#references)\] paper for more background knowledge. The classical DS2 network contains 15 layers (from bottom to top): - **Two** data layers (audio spectrogram, transcription text) - **Three** 2D convolution layers - **Seven** uni-directional simple-RNN layers - **One** lookahead row convolution layers - **One** fully-connected layers - **One** CTC-loss layer

Figure 1. Archetecture of Deep Speech 2 Network.
We don't have to persist on this 2-3-7-1-1-1 depth \[[2](#references)\]. Similar networks with different depths might also work well. As in \[[1](#references)\], authors use a different depth (e.g. 2-2-3-1-1-1) for final experiments. Key ingredients about the layers: - **Data Layers**: - Frame sequences data of audio **spectrogram** (with FFT). - Token sequences data of **transcription** text (labels). - These two type of sequences do not have the same lengthes, thus a CTC-loss layer is required. - **2D Convolution Layers**: - Not only temporal convolution, but also **frequency convolution**. Like a 2D image convolution, but with a variable dimension (i.e. temporal dimension). - With striding for only the first convlution layer. - No pooling for all convolution layers. - **Uni-directional RNNs** - Uni-directional + row convolution: for low-latency inference. - Bi-direcitional + without row convolution: if we don't care about the inference latency. - **Row convolution**: - For looking only a few steps ahead into the feature, instead of looking into a whole sequence in bi-directional RNNs. - Not nessesary if with bi-direcitional RNNs. - "**Row**" means convolutions are done within each frequency dimension (row), and no convolution kernels shared across. - **Batch Normalization Layers**: - Added to all above layers (except for data and loss layer). - Sequence-wise normalization for RNNs: BatchNorm only performed on input-state projection and not state-state projection, for efficiency consideration. Required Components | PaddlePaddle Support | Need to Develop :------------------------------------- | :-------------------------------------- | :----------------------- Data Layer I (Spectrogram) | Not supported yet. | TBD (Task 3) Data Layer II (Transcription) | `paddle.data_type.integer_value_sequence` | - 2D Convolution Layer | `paddle.layer.image_conv_layer` | - DataType Converter (vec2seq) | `paddle.layer.block_expand` | - Bi-/Uni-directional RNNs | `paddle.layer.recurrent_group` | - Row Convolution Layer | Not supported yet. | TBD (Task 4) CTC-loss Layer | `paddle.layer.warp_ctc` | - Batch Normalization Layer | `paddle.layer.batch_norm` | - CTC-Beam search | Not supported yet. | TBD (Task 6) ### Row Convolution TODO by Assignees ### Beam Search with CTC and LM

Figure 2. Algorithm for CTC Beam Search Decoder.
- The **Beam Search Decoder** for DS2 CTC-trained network follows the similar approach in \[[3](#references)\] as shown in Figure 2, with two important modifications for the ambiguous parts: - 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. - An **external scorer** would be passed into the decoder to evaluate a candidate prefix during decoding whenever a white space appended in English decoding and any character appended in Mandarin decoding. - Such external scorer consists of language model, word count or any other customed scorers. - The **language model** is built from Task 5, with parameters should be carefully tuned to achieve minimum WER/CER (c.f. Task 7) - This decoder needs to perform with **high efficiency** for the convenience of parameters tuning and speech recognition in reality. ## Future Work - Efficiency Improvement - Accuracy Improvement - Low-latency Inference Library - Large-scale benchmarking ## References 1. Dario Amodei, etc., [Deep Speech 2 : End-to-End Speech Recognition in English and Mandarin](http://proceedings.mlr.press/v48/amodei16.pdf). ICML 2016. 2. Dario Amodei, etc., [Deep Speech 2 : End-to-End Speech Recognition in English and Mandarin](https://arxiv.org/abs/1512.02595). arXiv:1512.02595. 3. Awni Y. Hannun, etc. [First-Pass Large Vocabulary Continuous Speech Recognition using Bi-Directional Recurrent DNNs](https://arxiv.org/abs/1408.2873). arXiv:1408.2873