Well, let´s see if I can help you some ...

"The question is ....
1. For real time streaming operation how to configure the
write(byte[], int offsetInBytes, int sizeInBytes) methods??? "

For real time or offline streaming, it works the same way: as you
receive your packets, call write so you can fill AudioTrack´s internal
buffer with the voice data. As the buffer is read, your audio will be
played by the system by itself. Parameters for this function are well
described on Android´s documentation.

"2. What is the function and purpose of
getPlaybackHeadPosition()???...
What is playback head position kindly explain! "

Playback head position can show you the actual status of AudioTrack´s
internal buffer: the position returned is the last avaliable sample in
the buffer. This way you know how full/empty it is and then take
actions as needed. To tell you the truth, unless there´s something
really unique/special in your code you won´t use it.

The playback will occur just like in any audio device: fill the buffer
and the data will be played. The write call blocks if the buffer doesn
´t have enough space for your data so you don´t have to syncronize
anything (like a producer/consumer algorithm). What you will probably
need is a separate thread to garantee you don´t lose packets when
blocket by write().
One last thing: all android audio applications have a considerable
audio delay because of the way audio is handled by the system. In it´s
actual state it´s not possible to develop low latency audio
applications.

Hope it helps,
Gabriel

On 18 maio, 13:16, BobG <[email protected]> wrote:
> 8khz sample rate sends 8 samples per ms. If you bundle up 50ms worth
> of samples in a packet (400 samples, 800 bytes) and send them the 50
> ms lag is almost undetectable (unless you can hear the original too...
> 2 phones at the same time for example)
>
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