重新采样音频缓冲区从44100到16000

I have audio data in format of data-uri, then I converted this data-uri into a buffer now I need this buffer data in new samplerate, currently audio data is in 44.1khz and I need data in 16khz, and If I recorded the audio using RecordRTC API and if I record audio in low sample rate then I got distorted audio voice, So I am not getting how to resample my audio buffer, I have audio data in format of data-uri, then I