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AlexR  
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 More options Jul 18 2012, 9:55 am
From: AlexR <alex.ru...@googlemail.com>
Date: Wed, 18 Jul 2012 06:55:11 -0700 (PDT)
Local: Wed, Jul 18 2012 9:55 am
Subject: Re: [discuss-webrtc] getusermedia audio
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@Tommy are you suggesting that the server should act as a peer instead of
normally having a peer2peer connection between two real users?
@Guillem I've already implemented Wami. Its fine for small amounts of data
(like 30sec of recording) but its not reliable for longer sessions that
last minutes. The bottleneck is that no matter how you do it (XMLHTTP or
other POST) there are large chunks of files that have to be transmitted
after the recording (wave is ~1.5MB per 30sec). Therefore a direct stream
to the server would be much more desirable. I actually implemented 'fake
streaming' by chunking the recordings and sending small bits to the server
every few seconds, but Flash isn't very helpful with this as it takes some
time before it saves the data as localStream. Also I've worked with a wami
clone called laurentsavaete-wami-for-ductus which uses XMLHTTP and base64
encoding.

On Wednesday, 18 July 2012 15:46:13 UTC+2, Guillem Perez wrote:

> See Wami-Recorder it's not perfect but maybe would work for your problem.

> On Wed, Jul 18, 2012 at 3:26 PM, Tommy Widenflycht (ᛏᚮᛘᛘᚤ) <
> tom...@google.com> wrote:

>> Right now it is entirely unclear how the recording API will look like...
>> I would suggest that you implement a PeerConnection client in your server.
>> That way you can use pure HTML in your clients.

>> /Tommy

>> On Wed, Jul 18, 2012 at 2:21 PM, AlexR <alex.ru...@googlemail.com> wrote:

>>> Hey everyone,

>>> I am currently developing an app that should use the webrtc (especially
>>> getUserMedia) to grab microphone user input and then send it to a server.
>>> The idea is to eliminate the need for streaming via FMS or other using the
>>> RMTP protocol as it also limits the app to Flash. I know that webrtc is
>>> made for real time communication, but I also see the need for this feature
>>> in the future.

>>> According to the updated draft found here
>>> http://dev.w3.org/2011/webrtc/editor/getusermedia.html#stream-api this
>>> can be accomplished by getUserMedia and setting MediaStreamConstraints to
>>> {audio: true}. As the MediaStreamRecorder was removed on 04/23/12 which
>>> leads me to believe that this kind of recording mechanism is deprecated. I
>>> have found an example using the {audio:true} to get the microphone
>>> http://pastie.org/3985501 but It is linked to Chrome Issue 112367
>>> http://code.google.com/p/chromium/issues/detail?id=112367 which has
>>> been opened a long time ago but did not yet get a clear fix.

>>> To my understanding webrtc/getUserMedia is not (yet) able to do what I
>>> intend to do. My question is now, if the current draft will incorporate any
>>> of these feature I would need to accomplish this as the draft isn't very
>>> clear about this. Also has anyone tried the above and ran into less
>>> problems?

>>> Thanks

>>> //AlexR

>>> --

>> --
>> Tommy Widenflycht, Senior Software Engineer
>> Google Sweden AB, Kungsbron 2, SE-11122 Stockholm, Sweden
>> Org. nr. 556656-6880
>> And yes, I have to include the above in every outgoing email according to
>> EU law.

>> --


 
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