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Is it possible to change the Audio Codec used? #10
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Sorry, I really don't know how to change the codec. You should check the On Fri, Aug 1, 2014 at 12:17 PM, sarinrohan notifications@github.com
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I tried to change stuff. It's there in webrtcvoiceengine.cc. Though can you let me know what changes you made to the WebRTC Java libraries before compiling them? After I recompile, a few methods such as InitializeAndroidGlobals gives me an error in AndroidRTC. |
I haven't made any changes to the WebRTC Java libraries, but it's an old On Mon, Aug 4, 2014 at 7:26 AM, sarinrohan notifications@github.com wrote:
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Found a workaround. Passing other 2 parameters as True to initialiseAndroidGlobals works. Its something to do with initialising audio and video streams. It was updated in the newer releases. |
Good to know thanks ! On Mon, Aug 11, 2014 at 8:10 AM, sarinrohan notifications@github.com
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Hey, |
Hi, I've updated the libjingle and you should now be able to change audio and video codecs (as well as some other parameters). |
While running the app on my phone, when I check the logs on Eclipse, I find that it sets the audio codec as Opus at 32kbits. Since the internet connection I am using is pretty poor, is it possible to adjust the SendCodec bitrate to a lower value in your app for better audio?
Also, would it help to use the compiled library for Google's WebRTCDemo and create a VoiceEngine and simply edit the sendcodec?
Thanks.
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