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Anomadarshi, simply implement a 1:2 upsampler and 2:1 downsampler this will provide your required functionality. Sampling a speech signal at only 4kHz, that is a nyquist freq of 2kHz, likely leading to a audio input bandwidth of perhaps 1.7kHz will provide very very poor quality speech. A minimum of 2.4kHz audio bandwidth is suggestion. The use of 16 bits would seem to be a waste of time. I would suggest changing your hardware to 8kHz sample rate and 8 bits. Or even 6 bits. Use of an AGC ahead of codec2 might improve results. regards glen english vk1xx On 12/06/2016 7:08 PM, Anomadarshi
Barua Shuvro wrote:
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