[asterisk-dev] Extend MixMonitor to record stereo files
Alex.Barnes at completeautomotivesolutions.co.uk
Thu Nov 14 10:18:50 CST 2013
Thanks for your reply Scott.
We hadn't noticed the new MixMonitor options 'r' and 't'; I must stop using the voip-info wiki rather than the new docs :)
We are currently testing to see if setting both options will produce two separate (or three maybe looking at the app_monitor.c file) wav files that we can merge together like we do when using Monitor. If this does work then extending FreePBX to pass these extra options should be trivial in comparison to adding stereo WAV support to Asterisk itself.
I'll continue reading around the Asterisk source in the meantime though so thanks again for pointing out the WAV handling issue.
From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Scott Griepentrog
Sent: 14 November 2013 15:16
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Extend MixMonitor to record stereo files
It's definitely possible. Many moons ago (before mixmonitor) I once hacked the mixing script in FreePBX to use sox to build a stero wav file from the two recordings - so I understand the usefulness of what you're trying to do.
Be aware that both the Asterisk project and the FreePBX project are open source, and you can contribute to either or both.
If I'm not mistaken (willing to be corrected on the internet if I'm wrong ;-) the WAV handling code is fixed in mono and 8khz (or at least used to be) and it may be tricky to add in stereo support. That being said, it's certainly possible to do it within Asterisk. It would require changes to both MixMonitor() or creation of StereoMixMonitor and the WAV handling. The part that I'm fuzzy on is how best to pass the two channels between them.
On Thu, Nov 14, 2013 at 3:57 AM, Alex Barnes <Alex.Barnes at completeautomotivesolutions.co.uk<mailto:Alex.Barnes at completeautomotivesolutions.co.uk>> wrote:
I was hoping somebody knowledgeable about the inner workings of app_mixmonitor could give me an idea of how feasible our change might be?
As a FreePBX user I would like MixMonitor to save recordings in stereo wav format where inbound calls are one channel and outbound are another.
This will allow me to more clearly hear who is speaking as well as more easily extract the audio of just one speaker.
The "FreePBX" part is important as we cannot make it use Monitor and an external bash script without hacking apart a lot of the dial plan, hence we are stuck using MixMonitor.
We are a dev company but with little C experience and zero Asterisk application development knowledge HOWEVER we're very happy to give it a whirl.
I'm literally just starting on researching this and no doubt have lots of reading to do regarding general Asterisk app development, research WAV format issues in C and MixMonitor, how to submit changes to the Asterisk project etc etc. I was hoping somebody could let me know if they think this will definitely not work or if they happen to know of some of the issues we may face and possibly where to start looking.
Thanks in advance
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