HI,
I am using ezsdk_dm816x-evm_5_02_02_60_setuplinux Netra package with omx-ti816x-src_05_02_00_15
and linux-2.6.37-psp04.00.00.12.
When an audio call is made i observe a latency of around half a second(around 400ms).
In the application , realtime audio data is captured and rendered and the voice is heard with a delay of around half a second.
The sampling rate is 16Khz and samples per frame=320.So, each frame is of about 20 ms.
Just wanted to know from the alsa drivers perspective if there could be any delay.
Are there any ways to reduce the latency observed?
The words spoken are not in lip sync when a two way call is made.
Any ideas or suggestion on the above problem would be very helpful.
Regards,
Leena M
