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Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: September 02 2008 at 12:21pm | IP Logged
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Ajay,
Unfortunately we can’t let that source code go. I think it being used is some other project here.
However, its very easy to get the RTP media for a call on a per phone line basis and buffer it up in order to perform the proper FFT on the incoming audio data. This is done using each phone line’s receive IVR channels. The dual line IVR server sample app does exactly this function.
For example:
You need to set up in your VOIP app code the ability to obtain “receive IVR” sample block data for each phone line. You can tell the media engine to pass you 20Ms sample blocks of incoming audio data using a common PCM format/rate regardless of the actual format/rate used for the RTP media for the call.
You then take this sample block data from the media engine’s phone line Rx IVR channel and buffer it up until you have enough to perform your required FFT length. There should many sources for simple FFT code out there o the net. Possibly at CodeProject.
You sound like you are building a pretty cool VOIP app. Don’t forget that we offer paid enhanced support and contract services to customers. We can then help you get done whatever you may need. Something to think about.
Thanks Ajay,
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Technical Support
Topic: Any chance of getting source to Spectrum?
