02-17-2014 01:21 AM
I am working on UDP voice sender/rec and used it on network it is working good but i want to reduce bandwidth (present BW: 180 kbps meseared on wireshark) over channel to 32/64 kbps. please help me to do it.
02-17-2014 01:48 AM - edited 02-17-2014 01:50 AM
Hi prashantece,
at which university are you studying?
What kind of help do you need?
- When you want to have a programming job done you may advertize it in the "job offerings" forum.
- When you need help on specific problems you should attach your VI and pin-point the problem, some good example data preferred…
- When you need general ideas on your question topic you may read a good book or look up Wikipedia! You may look for abbreviations like FLAC, MP3, OPUS, AAC, CELT, SILK…
02-17-2014 03:48 AM
You can ofcourse change sampling parameters to 8-bit and 11kHz sampling, you can zip the stream before sending it, or use some compressed sound format. What are you doing now?
/Y
02-17-2014 04:09 AM
here i attched my vi
nd i want to make a voice transmission system using udp protocol.
02-17-2014 04:11 AM
at the given combination (8bit and 11k) there is very poor quality of audio(like donald duck) is coming so any other idea.....
regards
02-17-2014 05:07 AM
02-17-2014 05:09 AM
02-17-2014 06:16 AM - edited 02-17-2014 06:17 AM
Hi prasha,
present BW: 180 kbps meseared on wireshark over channel to 32/64 kbps
Well, right now you need a BW of [stereo, 16bit, 11,025kHz] 2×11025×8 = 176,4kbps.
To reach your goal of 48kbps you should use [mono, 8bit, 6000Hz] settings: 1×6000×8=48kbps.
When it comes to "listening quality" (which is not mentioned in your first post!) you really should read literature on the abbrevations mentioned above…
02-17-2014 04:14 PM
@prashantece wrote:
at the given combination (8bit and 11k) there is very poor quality of audio(like donald duck) is coming so any other idea.....
regards
In the VI you posted you're using the wrong output from the grab sound.
02-17-2014 08:03 PM
Hi, prashantece,
Do you want just to resample your voice data?