Add speech recognition C++ example

.gitmodules

@@ -2,3 +2,6 @@
 	path = third_party/kaldi/submodule
 	url = https://github.com/kaldi-asr/kaldi
 	ignore = dirty
+[submodule "examples/libtorchaudio/simplectc"]
+	path = examples/libtorchaudio/simplectc
+	url = https://github.com/mthrok/ctcdecode

examples/libtorchaudio/.gitignore

@@ -1,3 +1,4 @@
 build
 data/output.wav
-data/pipeline.zip
+*.zip
+output

examples/libtorchaudio/CMakeLists.txt

@@ -14,4 +14,6 @@ message("libtorchaudio CMakeLists: ${TORCH_CXX_FLAGS}")
 set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} ${TORCH_CXX_FLAGS}")
 
 add_subdirectory(../.. libtorchaudio)
+add_subdirectory(simplectc)
 add_subdirectory(augmentation)
+add_subdirectory(speech_recognition)

examples/libtorchaudio/README.md

@@ -1,6 +1,7 @@
 # Libtorchaudio Examples
 
 * [Augmentation](./augmentation)
+* [Speech Recognition with wav2vec2.0](./speech_recognition)
 
 ## Build
 
@@ -14,6 +15,7 @@ It is currently not distributed, and it will be built alongside with the applica
 The following commands will build `libtorchaudio` and applications.
 
 ```bash
+git submodule update
 mkdir build
 cd build
 cmake -GNinja \

examples/libtorchaudio/build.sh

@@ -8,6 +8,7 @@ build_dir="${this_dir}/build"
 mkdir -p "${build_dir}"
 cd "${build_dir}"
 
+git submodule update
 cmake -GNinja \
       -DCMAKE_PREFIX_PATH="$(python -c 'import torch;print(torch.utils.cmake_prefix_path)')" \
       -DBUILD_SOX=ON \

examples/libtorchaudio/speech_recognition/CMakeLists.txt

@@ -0,0 +1,6 @@
+add_executable(transcribe transcribe.cpp)
+add_executable(transcribe_list transcribe_list.cpp)
+target_link_libraries(transcribe "${TORCH_LIBRARIES}" "${TORCHAUDIO_LIBRARY}" "${CTCDECODE_LIBRARY}")
+target_link_libraries(transcribe_list "${TORCH_LIBRARIES}" "${TORCHAUDIO_LIBRARY}" "${CTCDECODE_LIBRARY}")
+set_property(TARGET transcribe PROPERTY CXX_STANDARD 14)
+set_property(TARGET transcribe_list PROPERTY CXX_STANDARD 14)

examples/libtorchaudio/speech_recognition/README.md

@@ -0,0 +1,187 @@
+# Speech Recognition with wav2vec2.0
+
+This example demonstarates how you can use torchaudio's I/O features and models to run speech recognition in C++ application.
+
+**NOTE**
+This example uses `"sox_io"` backend for loading audio, which does not work on Windows. To make it work on
+Windows, you need to replace the part of loading audio and converting it to Tensor object.
+
+## 1. Create a transcription pipeline TorchScript file
+
+We will create a TorchScript that performs the following processes;
+
+1. Load audio from a file.
+1. Pass audio to encoder which produces the sequence of probability distribution on labels.
+1. Pass the encoder output to decoder which generates transcripts.
+
+For building decoder, we borrow the pre-trained weights published by `fairseq` and/or Hugging Face Transformers, then convert it `torchaudio`'s format, which supports TorchScript.
+
+### 1.1. From `fairseq`
+
+For `fairseq` models, you can download pre-trained weights
+You can download a model from [`fairseq` repository](https://github.com/pytorch/fairseq/tree/master/examples/wav2vec). Here, we will use `Base / 960h` model. You also need to download [the letter dictionary file](https://github.com/pytorch/fairseq/tree/master/examples/wav2vec#evaluating-a-ctc-model).
+
+For the decoder part, we use [simple_ctc](https://github.com/mthrok/ctcdecode), which also supports TorchScript.
+
+```bash
+mkdir -p pipeline-fairseq
+python build_pipeline_from_fairseq.py \
+    --model-file "wav2vec_small_960.pt" \
+    --dict-dir <DIRECTORY_WHERE_dict.ltr.txt_IS_FOUND> \
+    --output-path "./pipeline-fairseq/"
+```
+
+The above command should create the following TorchScript object files in the output directory.
+
+```
+decoder.zip  encoder.zip  loader.zip
+```
+
+* `loader.zip` loads audio file and generate waveform Tensor.
+* `encoder.zip` receives waveform Tensor and generates the sequence of probability distribution over the label.
+* `decoder.zip` receives the probability distribution over the label and generates a transcript.
+
+### 1.2. From Hugging Face Transformers
+
+
+[Hugging Face Transformers](https://huggingface.co/transformers/index.html) and [Hugging Face Model Hub](https://huggingface.co/models) provides `wav2vec2.0` models fine-tuned on variety of datasets and languages.
+
+We can also import the model published on Hugging Face Hub and run it in our C++ application.
+In the following example, we will try the Geremeny model, ([facebook/wav2vec2-large-xlsr-53-german](https://huggingface.co/facebook/wav2vec2-large-xlsr-53-german/tree/main)) on [VoxForge Germany dataset](http://www.voxforge.org/de/downloads).
+
+```bash
+mkdir -p pipeline-hf
+python build_pipeline_from_huggingface_transformers.py \
+    --model facebook/wav2vec2-large-xlsr-53-german \
+    --output-path ./pipeline-hf/
+```
+
+The resulting TorchScript object files should be same as the `fairseq` example.
+
+## 2. Build the application
+
+Please refer to [the top level README.md](../README.md)
+
+## 3. Run the application
+
+Now we run the C++ applicaiton [`transcribe`](./transcribe.cpp), with the TorchScript object we created in Step.1.1. and an input audio file.
+
+```bash
+../build/speech_recognition/transcribe ./pipeline-fairseq ../data/input.wav
+```
+
+This will output something like the following.
+
+```
+Loading module from: ./pipeline/loader.zip
+Loading module from: ./pipeline/encoder.zip
+Loading module from: ./pipeline/decoder.zip
+Loading the audio
+Running inference
+Generating the transcription
+I HAD THAT CURIOSITY BESIDE ME AT THIS MOMENT
+Done.
+```
+
+## 4. Evaluate the pipeline on Librispeech dataset
+
+Let's evaluate this word error rate (WER) of this application using [Librispeech dataset](https://www.openslr.org/12).
+
+### 4.1. Create a list of audio paths
+
+For the sake of simplifying our C++ code, we will first parse the Librispeech dataset to get the list of audio path 
+
+```bash
+python parse_librispeech.py <PATH_TO_YOUR_DATASET>/LibriSpeech/test-clean ./flist.txt
+```
+
+The list should look like the following;
+
+```bash
+head flist.txt
+
+1089-134691-0000    /LibriSpeech/test-clean/1089/134691/1089-134691-0000.flac    HE COULD WAIT NO LONGER
+```
+
+### 4.2. Run the transcription
+
+[`transcribe_list`](./transcribe_list.cpp) processes the input flist list and feed the audio path one by one to the pipeline, then generate reference file and hypothesis file.
+
+```bash
+../build/speech_recognition/transcribe_list ./pipeline-fairseq ./flist.txt <OUTPUT_DIR>
+```
+
+### 4.3. Score WER
+
+You need `sclite` for this step. You can download the code from [SCTK repository](https://github.com/usnistgov/SCTK).
+
+```bash
+# in the output directory
+sclite -r ref.trn -h hyp.trn -i wsj -o pralign -o sum
+```
+
+WER can be found in the resulting `hyp.trn.sys`. Check out the column that starts with `Sum/Avg` the first column of the third block is `100 - WER`.
+
+In our test, we got the following results.
+
+|          model                            | Fine Tune | test-clean | test-other |
+|:-----------------------------------------:|----------:|:----------:|:----------:|
+| Base<br/>`wav2vec_small_960`              |      960h |        3.1 |        7.7 |
+| Large<br/>`wav2vec_big_960`               |      960h |        2.6 |        5.9 |
+| Large (LV-60)<br/>`wav2vec2_vox_960h_new` |      960h |        2.9 |        6.2 |
+| Large (LV-60) + Self Training<br/>`wav2vec_vox_960h_pl` | 960h | 1.9 |      4.5 |
+
+
+You can also check `hyp.trn.pra` file to see what errors were made.
+
+```
+id: (3528-168669-0005)
+Scores: (#C #S #D #I) 7 1 0 0
+REF:  there is a stone to be RAISED heavy 
+HYP:  there is a stone to be RACED  heavy 
+Eval:                        S            
+```
+
+## 5. Evaluate the pipeline on VoxForge dataset
+
+Now we use the pipeline we created in step 1.2. This time with German language dataset from VoxForge.
+
+### 5.1. Create a list of audio paths
+
+Download an archive from http://www.repository.voxforge1.org/downloads/de/Trunk/Audio/Main/16kHz_16bit/, and extract it to your local file system, then run the following to generate the file list.
+
+```bash
+python parse_voxforge.py <PATH_TO_YOUR_DATASET> > ./flist-de.txt
+```
+
+The list should look like
+
+```bash
+head flist-de.txt
+de5-001    /datasets/voxforge/de/guenter-20140214-afn/wav/de5-001.wav    ES SOLL ETWA FÜNFZIGTAUSEND VERSCHIEDENE SORTEN GEBEN
+```
+
+### 5.2. Run the application and score WER
+
+This process is same as the Librispeech example. We just use the pipeline with the Germany model and file list of Germany dataset. Refer to the corresponding ssection in Librispeech evaluation..
+
+```bash
+../build/speech_recognition/transcribe_list ./pipeline-hf ./flist-de.txt <OUTPUT_DIR>
+```
+
+Then
+
+```bash
+# in the output directory
+sclite -r ref.trn -h hyp.trn -i wsj -o pralign -o sum
+```
+
+You can find the defailt of evalauation result in PRA.
+
+```
+id: (guenter-20140214-afn/mfc/de5-012)
+Scores: (#C #S #D #I) 4 1 1 0
+REF:  die ausgaben kÖnnen gigantisch STEIGE N
+HYP:  die ausgaben kÖnnen gigantisch ****** STEIGEN
+Eval:                                 D      S
+```