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Dimitrii Voronin
2020-12-22 11:43:31 +02:00
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@@ -25,7 +25,7 @@
# Silero VAD
![image](https://user-images.githubusercontent.com/36505480/102233150-9f476580-3ef8-11eb-87fb-ae6f1edfe10f.png)
![image](https://user-images.githubusercontent.com/36505480/102872739-ce099280-4448-11eb-967b-724440165eb5.png)
**Silero VAD: pre-trained enterprise-grade Voice Activity Detector (VAD), Number Detector and Language Classifier.**
Enterprise-grade Speech Products made refreshingly simple (see our [STT](https://github.com/snakers4/silero-models) models).
@@ -153,7 +153,16 @@ Streaming latency depends on 2 variables:
So **batch size** for streaming is **num_steps * number of audio streams**. Time between receiving new audio chunks and getting results is shown in picture:
![image](https://user-images.githubusercontent.com/36505480/102475710-e18cb600-4062-11eb-8c34-da6e6ec5385d.png)
| Batch size | Pytorch model time, ms | Onnx model time, ms |
| :-------------: |:-------------:| :-----:|
| **2** | 9 | 2 |
| **4** | 11 | 4 |
| **8** | 14 | 7 |
| **16** | 19 | 12 |
| **40** | 36 | 29 |
| **80** | 64 | 55 |
| **120** | 96 | 85 |
| **200** | 157 | 137 |
We are working on lifting this 250 ms constraint.
@@ -161,7 +170,16 @@ We are working on lifting this 250 ms constraint.
**RTS** (seconds of audio processed per second, real time speed, or 1 / RTF) for full audio processing depends on **num_steps** (see previous paragraph) and **batch size** (bigger is better).
![image](https://user-images.githubusercontent.com/36505480/102475751-f2d5c280-4062-11eb-9791-3ec1632547bc.png)
| Batch size | num_steps | Pytorch model RTS | Onnx model RTS |
| :-------------: |:-------: | :-------------:| :-----:|
| **40** | **4** | 68 | 86 |
| **40** | **8** | 34 | 43 |
| **80** | **4** | 78 | 91 |
| **80** | **8** | 39 | 45 |
| **120** | **4** | 78 | 88 |
| **120** | **8** | 39 | 44 |
| **200** | **4** | 80 | 91 |
| **200** | **8** | 40 | 46 |
### VAD Quality Metrics
@@ -171,7 +189,7 @@ Since our VAD (only VAD, other networks are more flexible) was trained on chunks
[Webrtc](https://github.com/wiseman/py-webrtcvad) splits audio into frames, each frame has corresponding number (0 **or** 1). We use 30ms frames for webrtc, so each 250 ms chunk is split into 8 frames, their **mean** value is used as a treshold for plot.
![image](https://user-images.githubusercontent.com/36505480/102233150-9f476580-3ef8-11eb-87fb-ae6f1edfe10f.png)
![image](https://user-images.githubusercontent.com/36505480/102872739-ce099280-4448-11eb-967b-724440165eb5.png)
## FAQ