# Finetune your own AM based on FastSpeech2 with AISHELL-3. This example shows how to finetune your own AM based on FastSpeech2 with AISHELL-3. We use part of csmsc's data (top 200) as finetune data in this example. The example is implemented according to this [discussion](https://github.com/PaddlePaddle/PaddleSpeech/discussions/1842). Thanks to the developer for the idea. We use AISHELL-3 to train a multi-speaker fastspeech2 model. You can refer [examples/aishell3/tts3](https://github.com/lym0302/PaddleSpeech/tree/develop/examples/aishell3/tts3) to train multi-speaker fastspeech2 from scratch. ## Prepare ### Download Pretrained Fastspeech2 model Assume the path to the model is `./pretrained_models`. Download pretrained fastspeech2 model with aishell3: [fastspeech2_aishell3_ckpt_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_aishell3_ckpt_1.1.0.zip). ```bash mkdir -p pretrained_models && cd pretrained_models wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_aishell3_ckpt_1.1.0.zip unzip fastspeech2_aishell3_ckpt_1.1.0.zip cd ../ ``` ### Download MFA tools and pretrained model Assume the path to the MFA tool is `./tools`. Download [MFA](https://github.com/MontrealCorpusTools/Montreal-Forced-Aligner/releases/download/v1.0.1/montreal-forced-aligner_linux.tar.gz) and pretrained MFA models with aishell3: [aishell3_model.zip](https://paddlespeech.bj.bcebos.com/MFA/ernie_sat/aishell3_model.zip). ```bash mkdir -p tools && cd tools # mfa tool wget https://github.com/MontrealCorpusTools/Montreal-Forced-Aligner/releases/download/v1.0.1/montreal-forced-aligner_linux.tar.gz tar xvf montreal-forced-aligner_linux.tar.gz cp montreal-forced-aligner/lib/libpython3.6m.so.1.0 montreal-forced-aligner/lib/libpython3.6m.so # pretrained mfa model mkdir -p aligner && cd aligner wget https://paddlespeech.bj.bcebos.com/MFA/ernie_sat/aishell3_model.zip unzip aishell3_model.zip wget https://paddlespeech.bj.bcebos.com/MFA/AISHELL-3/with_tone/simple.lexicon cd ../../ ``` ### Prepare your data Assume the path to the dataset is `./input`. This directory contains audio files (*.wav) and label file (labels.txt). The audio file is in wav format. The format of the label file is: utt_id|pinyin. Here is an example of the first 200 data of csmsc. ```bash mkdir -p input && cd input wget https://paddlespeech.bj.bcebos.com/datasets/csmsc_mini.zip unzip csmsc_mini.zip cd ../ ``` When "Prepare" done. The structure of the current directory is listed below. ```text ├── input │ ├── csmsc_mini │ │ ├── 000001.wav │ │ ├── 000002.wav │ │ ├── 000003.wav │ │ ├── ... │ │ ├── 000200.wav │ │ ├── labels.txt │ └── csmsc_mini.zip ├── pretrained_models │ ├── fastspeech2_aishell3_ckpt_1.1.0 │ │ ├── default.yaml │ │ ├── energy_stats.npy │ │ ├── phone_id_map.txt │ │ ├── pitch_stats.npy │ │ ├── snapshot_iter_96400.pdz │ │ ├── speaker_id_map.txt │ │ └── speech_stats.npy │ └── fastspeech2_aishell3_ckpt_1.1.0.zip └── tools ├── aligner │ ├── aishell3_model │ ├── aishell3_model.zip │ └── simple.lexicon ├── montreal-forced-aligner │ ├── bin │ ├── lib │ └── pretrained_models └── montreal-forced-aligner_linux.tar.gz ... ``` ## Get Started Run the command below to 1. **source path**. 2. finetune the model. 3. synthesize wavs. - synthesize waveform from text file. ```bash ./run.sh ``` You can choose a range of stages you want to run, or set `stage` equal to `stop-stage` to run only one stage. ### Model Finetune Finetune a FastSpeech2 model. ```bash ./run.sh --stage 0 --stop-stage 0 ``` `stage 0` of `run.sh` calls `finetune.py`, here's the complete help message. ```text usage: finetune.py [-h] [--input_dir INPUT_DIR] [--pretrained_model_dir PRETRAINED_MODEL_DIR] [--mfa_dir MFA_DIR] [--dump_dir DUMP_DIR] [--output_dir OUTPUT_DIR] [--lang LANG] [--ngpu NGPU] optional arguments: -h, --help show this help message and exit --input_dir INPUT_DIR directory containing audio and label file --pretrained_model_dir PRETRAINED_MODEL_DIR Path to pretrained model --mfa_dir MFA_DIR directory to save aligned files --dump_dir DUMP_DIR directory to save feature files and metadata --output_dir OUTPUT_DIR directory to save finetune model --lang LANG Choose input audio language, zh or en --ngpu NGPU if ngpu=0, use cpu --epoch EPOCH the epoch of finetune --batch_size BATCH_SIZE the batch size of finetune, default -1 means same as pretrained model ``` 1. `--input_dir` is the directory containing audio and label file. 2. `--pretrained_model_dir` is the directory incluing pretrained fastspeech2_aishell3 model. 3. `--mfa_dir` is the directory to save the results of aligning from pretrained MFA_aishell3 model. 4. `--dump_dir` is the directory including audio feature and metadata. 5. `--output_dir` is the directory to save finetune model. 6. `--lang` is the language of input audio, zh or en. 7. `--ngpu` is the number of gpu. 8. `--epoch` is the epoch of finetune. 9. `--batch_size` is the batch size of finetune. ### Synthesizing We use [HiFiGAN](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/aishell3/voc5) as the neural vocoder. Assume the path to the hifigan model is `./pretrained_models`. Download the pretrained HiFiGAN model from [hifigan_aishell3_ckpt_0.2.0](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/hifigan/hifigan_aishell3_ckpt_0.2.0.zip) and unzip it. ```bash cd pretrained_models wget https://paddlespeech.bj.bcebos.com/Parakeet/released_models/hifigan/hifigan_aishell3_ckpt_0.2.0.zip unzip hifigan_aishell3_ckpt_0.2.0.zip cd ../ ``` HiFiGAN checkpoint contains files listed below. ```text hifigan_aishell3_ckpt_0.2.0 ├── default.yaml # default config used to train HiFiGAN ├── feats_stats.npy # statistics used to normalize spectrogram when training HiFiGAN └── snapshot_iter_2500000.pdz # generator parameters of HiFiGAN ``` Modify `ckpt` in `run.sh` to the final model in `exp/default/checkpoints`. ```bash ./run.sh --stage 1 --stop-stage 1 ``` `stage 1` of `run.sh` calls `${BIN_DIR}/../synthesize_e2e.py`, which can synthesize waveform from text file. ```text usage: synthesize_e2e.py [-h] [--am {speedyspeech_csmsc,speedyspeech_aishell3,fastspeech2_csmsc,fastspeech2_ljspeech,fastspeech2_aishell3,fastspeech2_vctk,tacotron2_csmsc,tacotron2_ljspeech}] [--am_config AM_CONFIG] [--am_ckpt AM_CKPT] [--am_stat AM_STAT] [--phones_dict PHONES_DICT] [--tones_dict TONES_DICT] [--speaker_dict SPEAKER_DICT] [--spk_id SPK_ID] [--voc {pwgan_csmsc,pwgan_ljspeech,pwgan_aishell3,pwgan_vctk,mb_melgan_csmsc,style_melgan_csmsc,hifigan_csmsc,hifigan_ljspeech,hifigan_aishell3,hifigan_vctk,wavernn_csmsc}] [--voc_config VOC_CONFIG] [--voc_ckpt VOC_CKPT] [--voc_stat VOC_STAT] [--lang LANG] [--inference_dir INFERENCE_DIR] [--ngpu NGPU] [--text TEXT] [--output_dir OUTPUT_DIR] Synthesize with acoustic model & vocoder optional arguments: -h, --help show this help message and exit --am {speedyspeech_csmsc,speedyspeech_aishell3,fastspeech2_csmsc,fastspeech2_ljspeech,fastspeech2_aishell3,fastspeech2_vctk,tacotron2_csmsc,tacotron2_ljspeech} Choose acoustic model type of tts task. --am_config AM_CONFIG Config of acoustic model. --am_ckpt AM_CKPT Checkpoint file of acoustic model. --am_stat AM_STAT mean and standard deviation used to normalize spectrogram when training acoustic model. --phones_dict PHONES_DICT phone vocabulary file. --tones_dict TONES_DICT tone vocabulary file. --speaker_dict SPEAKER_DICT speaker id map file. --spk_id SPK_ID spk id for multi speaker acoustic model --voc {pwgan_csmsc,pwgan_ljspeech,pwgan_aishell3,pwgan_vctk,mb_melgan_csmsc,style_melgan_csmsc,hifigan_csmsc,hifigan_ljspeech,hifigan_aishell3,hifigan_vctk,wavernn_csmsc} Choose vocoder type of tts task. --voc_config VOC_CONFIG Config of voc. --voc_ckpt VOC_CKPT Checkpoint file of voc. --voc_stat VOC_STAT mean and standard deviation used to normalize spectrogram when training voc. --lang LANG Choose model language. zh or en --inference_dir INFERENCE_DIR dir to save inference models --ngpu NGPU if ngpu == 0, use cpu. --text TEXT text to synthesize, a 'utt_id sentence' pair per line. --output_dir OUTPUT_DIR output dir. ``` 1. `--am` is acoustic model type with the format {model_name}_{dataset} 2. `--am_config`, `--am_ckpt`, `--am_stat`, `--phones_dict` `--speaker_dict` are arguments for acoustic model, which correspond to the 5 files in the fastspeech2 pretrained model. 3. `--voc` is vocoder type with the format {model_name}_{dataset} 4. `--voc_config`, `--voc_ckpt`, `--voc_stat` are arguments for vocoder, which correspond to the 3 files in the parallel wavegan pretrained model. 5. `--lang` is the model language, which can be `zh` or `en`. 6. `--text` is the text file, which contains sentences to synthesize. 7. `--output_dir` is the directory to save synthesized audio files. 8. `--ngpu` is the number of gpus to use, if ngpu == 0, use cpu. ### Tips If you want to get better audio quality, you can use more audios to finetune.