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118 lines
5.4 KiB
118 lines
5.4 KiB
2 years ago
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# FastSpeech2 with Cantonese language
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## Dataset
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### Download and Extract
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If you don't have the Cantonese datasets mentioned above, please download and unzip [Guangzhou_Cantonese_Scripted_Speech_Corpus_Daily_Use_Sentence](https://magichub.com/datasets/guangzhou-cantonese-scripted-speech-corpus-daily-use-sentence/) and [Guangzhou_Cantonese_Scripted_Speech_Corpus_in_Vehicle](https://magichub.com/datasets/guangzhou-cantonese-scripted-speech-corpus-in-the-vehicle/) under `~/datasets/`.
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To obtain better performance, please combine these two datasets together as follows:
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```bash
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mkdir -p ~/datasets/canton_all/WAV
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cp -r ~/datasets/Guangzhou_Cantonese_Scripted_Speech_Corpus_Daily_Use_Sentence/WAV/* ~/datasets/canton_all/WAV
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cp -r ~/datasets/Guangzhou_Cantonese_Scripted_Speech_Corpus_in_Vehicle/WAV/* ~/datasets/canton_all/WAV
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```
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After that, it should be look like:
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```
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~/datasets/canton_all
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│ └── WAV
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│ └──G0001
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│ └──G0002
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│ ...
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│ └──G0071
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│ └──G0072
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```
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### Get MFA Result and Extract
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We use [MFA1.x](https://github.com/MontrealCorpusTools/Montreal-Forced-Aligner) to get durations for canton_fastspeech2.
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You can train your MFA model reference to [canton_mfa example](https://github.com/PaddlePaddle/PaddleSpeech/tree/develop/examples/other/mfa) (use MFA1.x now) of our repo.
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We here provide the MFA results of these two datasets. [canton_alignment.zip](https://paddlespeech.bj.bcebos.com/MFA/Canton/canton_alignment.zip)
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## Get Started
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Assume the path to the Cantonese MFA result of the two datsets mentioned above is `./canton_alignment`.
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Run the command below to
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1. **source path**.
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2. preprocess the dataset.
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3. train the model.
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4. synthesize wavs.
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- synthesize waveform from `metadata.jsonl`.
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- synthesize waveform from text file.
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```bash
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./run.sh
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```
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You can choose a range of stages you want to run, or set `stage` equal to `stop-stage` to use only one stage, for example, running the following command will only preprocess the dataset.
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```bash
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./run.sh --stage 0 --stop-stage 0
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```
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### Data Preprocessing
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```bash
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./local/preprocess.sh ${conf_path}
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```
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When it is done. A `dump` folder is created in the current directory. The structure of the dump folder is listed below.
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```text
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dump
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├── dev
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│ ├── norm
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│ └── raw
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├── phone_id_map.txt
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├── speaker_id_map.txt
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├── test
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│ ├── norm
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│ └── raw
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└── train
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├── energy_stats.npy
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├── norm
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├── pitch_stats.npy
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├── raw
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└── speech_stats.npy
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```
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The dataset is split into 3 parts, namely `train`, `dev`, and` test`, each of which contains a `norm` and `raw` subfolder. The raw folder contains speech、pitch and energy features of each utterance, while the norm folder contains normalized ones. The statistics used to normalize features are computed from the training set, which is located in `dump/train/*_stats.npy`.
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Also, there is a `metadata.jsonl` in each subfolder. It is a table-like file that contains phones, text_lengths, speech_lengths, durations, the path of speech features, the path of pitch features, a path of energy features, speaker, and id of each utterance.
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### Training details can refer to the script of [examples/aishell3/tts3](../../aishell3/tts3).
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## Pretrained Model(Waiting========)
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Pretrained FastSpeech2 model with no silence in the edge of audios:
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- [fastspeech2_aishell3_ckpt_1.1.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_aishell3_ckpt_1.1.0.zip)
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- [fastspeech2_conformer_aishell3_ckpt_0.2.0.zip](https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_conformer_aishell3_ckpt_0.2.0.zip) (Thanks for [@awmmmm](https://github.com/awmmmm)'s contribution)
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FastSpeech2 checkpoint contains files listed below.
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```text
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fastspeech2_aishell3_ckpt_1.1.0
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├── default.yaml # default config used to train fastspeech2
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├── energy_stats.npy # statistics used to normalize energy when training fastspeech2
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├── phone_id_map.txt # phone vocabulary file when training fastspeech2
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├── pitch_stats.npy # statistics used to normalize pitch when training fastspeech2
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├── snapshot_iter_96400.pdz # model parameters and optimizer states
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├── speaker_id_map.txt # speaker id map file when training a multi-speaker fastspeech2
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└── speech_stats.npy # statistics used to normalize spectrogram when training fastspeech2
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```
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You can use the following scripts to synthesize for `${BIN_DIR}/../sentences.txt` using pretrained fastspeech2 and parallel wavegan models.
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```bash
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source path.sh
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FLAGS_allocator_strategy=naive_best_fit \
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FLAGS_fraction_of_gpu_memory_to_use=0.01 \
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python3 ${BIN_DIR}/../synthesize_e2e.py \
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--am=fastspeech2_aishell3 \
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--am_config=fastspeech2_aishell3_ckpt_1.1.0/default.yaml \
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--am_ckpt=fastspeech2_aishell3_ckpt_1.1.0/snapshot_iter_96400.pdz \
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--am_stat=fastspeech2_aishell3_ckpt_1.1.0/speech_stats.npy \
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--voc=pwgan_aishell3 \
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--voc_config=pwg_aishell3_ckpt_0.5/default.yaml \
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--voc_ckpt=pwg_aishell3_ckpt_0.5/snapshot_iter_1000000.pdz \
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--voc_stat=pwg_aishell3_ckpt_0.5/feats_stats.npy \
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--lang=zh \
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--text=${BIN_DIR}/../sentences.txt \
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--output_dir=exp/default/test_e2e \
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--phones_dict=fastspeech2_aishell3_ckpt_1.1.0/phone_id_map.txt \
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--speaker_dict=fastspeech2_aishell3_ckpt_1.1.0/speaker_id_map.txt \
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--spk_id=0 \
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--inference_dir=exp/default/inference
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```
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