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*DeepSpeech on PaddlePaddle* is an open-source implementation of end-to-end Automatic Speech Recognition (ASR) engine, with [PaddlePaddle](https://github.com/PaddlePaddle/Paddle) platform. Our vision is to empower both industrial application and academic research on speech recognition, via an easy-to-use, efficient and scalable implementation, including training, inference & testing module, and demo deployment.
For more information, please docs under `doc`.
For more information, please see below
[Install](docs/install.md)
[Getting Started](docs/geting_stared.md)
[Data Prepration](docs/data_preparation.md)
[Data Augmentation](docs/augmentation.md)
[Ngram LM](docs/ngram_lm.md)
[Server Demo](docs/server.md)
[Benchmark](docs/benchmark.md)
[Relased Model](docs/released_model.md)
[FAQ](docs/faq.md)
## Models
* [Baidu's Deep Speech2](http://proceedings.mlr.press/v48/amodei16.pdf)
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Please see [Getting Started](docs/geting_started.md) and [tiny egs](examples/tiny/README.md).
## Questions and Help
You are welcome to submit questions and bug reports in [Github Issues](https://github.com/PaddlePaddle/DeepSpeech/issues). You are also welcome to contribute to this project.

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*DeepSpeech on PaddlePaddle*是一个采用[PaddlePaddle](https://github.com/PaddlePaddle/Paddle)平台的端到端自动语音识别ASR引擎的开源项目
我们的愿景是为语音识别在工业应用和学术研究上,提供易于使用、高效和可扩展的工具,包括训练,推理,测试模块,以及 demo 部署。同时,我们还将发布一些预训练好的英语和普通话模型。
更多信息如下:
[安装](docs/install.md)
[开始](docs/geting_stared.md)
[数据处理](docs/data_preparation.md)
[数据增强](docs/augmentation.md)
[语言模型](docs/ngram_lm.md)
[服务部署](docs/server.md)
[Benchmark](docs/benchmark.md)
[Relased Model](docs/released_model.md)
[FAQ](docs/faq.md)
## 模型
* [Baidu's Deep Speech2](http://proceedings.mlr.press/v48/amodei16.pdf)

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We compare the training time with 1, 2, 4, 8 Tesla V100 GPUs (with a subset of LibriSpeech samples whose audio durations are between 6.0 and 7.0 seconds). And it shows that a **near-linear** acceleration with multiple GPUs has been achieved. In the following figure, the time (in seconds) cost for training is printed on the blue bars.
<img src="docs/images/multi_gpu_speedup.png" width=450><br/>
<img src="images/multi_gpu_speedup.png" width=450><br/>
| # of GPU | Acceleration Rate |
| -------- | --------------: |

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# FAQ
1. 音频变速快慢到达什么晨读会影响识别率?
变速会提升识别效果一般用0.9 1.0 1.1 的变速。
2. 音量大小到什么程度会影响识别率?
一般训练会固定音量到一个范围内波动过大会影响训练估计在10dB ~ 20dB吧。
3. 语音模型训练数据的最小时长要求时多少?
Aishell-1大约178h的数据数据越多越好。
4. 那些噪声或背景生会影响识别率?
主要是人生干扰和低信噪比会影响识别率。
5. 单条语音数据的长度限制是多少?
一般训练的语音长度会限制在1s~6s之间和训练配置有关。
6. 背景声在识别前是否需要分离出来,或做降噪处理?
需要分离的,需要结合具体场景考虑。
7. 模型是否带有VAD人生激活识别能力
VAD是单独的模型或模块模型不包含此能力。
8. 是否支持长语音识别?
一般过VAD后识别。
9. Mandarin LM Large语言模型需要的硬件配置时怎样的
内存能放得下LM即可。

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The grid search will print the WER (word error rate) or CER (character error rate) at each point in the hyper-parameters space, and draw the error surface optionally. A proper hyper-parameters range should include the global minima of the error surface for WER/CER, as illustrated in the following figure.
<p align="center">
<img src="docs/images/tuning_error_surface.png" width=550>
<img src="images/tuning_error_surface.png" width=550>
<br/>An example error surface for tuning on the dev-clean set of LibriSpeech
</p>

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# Aishell-1
## CTC
| Model | Config | Test set | CER |
| --- | --- | --- | --- |
| DeepSpeech2 | conf/deepspeech2.yaml | test | 0.078977 |
| DeepSpeech2 | release 1.8.5 | test | 0.080447 |

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# LibriSpeech
## CTC
| Model | Config | Test set | CER |
| --- | --- | --- | --- |
| DeepSpeech2 | conf/deepspeech2.yaml | test-clean | 0.073973 |
| DeepSpeech2 | release 1.8.5 | test-clean | 0.074939 |
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