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xiangyuecn avatar xiangyuecn commented on July 18, 2024

支持,你的流程是正确的

你把录音数据buffers里面的新数据自己缓冲一下,比如缓冲1秒的数据,比较容易检测到是否开始说话,不需要太精确从哪个位置开始说话,然后从这1秒的数据开始 将后续的录音数据全部存起来

最后在检测停止说话,检测到停止后,已缓冲的数据就是语音数据,你直接用Recorder.SampleData+mock方法,直接进行转码就可以得到mp3了

内存管理很容易做的,参考【教程】【音频流】【上传】实时转码并上传-通用版,用unknown格式+清理一下buffers,不会占用内存

from recorder.

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