[Speex-dev] Speex and Builder

[SteveK]

On Sep 21, 2005, at 5:59 AM, Jean-Marc Valin wrote:

>> 1) May I know how Speex compared with GIPS codec?  It seems that  
>> Google,
>> Yahoo, and Skype are licensing from GIPS.  Are there any good  
>> benchmarking
>> or fair comparisons?
>>
>
> I think these two emails sum up my opinion about Speex vs. iLBC:
> http://lists.xiph.org/pipermail/speex-dev/2005-June/003410.html
> http://lists.xiph.org/pipermail/speex-dev/2005-September/003652.html
>
>
>> 2) In particular, how is the jitter buffer control for Speex in  
>> response to
>> intermitent poor connection hiccups?  Is it robust enough to  
>> smooth out lost
>> packets (error concealment) and quick enough to recover the live
>> conversation smoothly once the packets received are back to normalcy?
>>
>
> There is packet loss concealment in Speex even without using the  
> jitter
> buffer (no, error concealment wasn't invented by GIPS). As for the
> jitter buffer itself, it's adaptive, so it just keeps the lowest
> possible delay that doesn't cause losses.
>
>
>> 3) Has anyone used Speex for live VoIP application?  Any good  
>> references or
>> examples to start with?
>>
>
> I've written a demo application for Linux that uses wideband, echo
> cancellation, denoising and the jitter buffer. I'll post it soon.

There's also asterisk, softphones based on the iaxclient library, and  
other softphones out there that use speech.  Asterisk now includes  
the speex preprocessor when you use the speex codec (in CVS), and  
iaxclient has used the speex preprocessor for quite some time.  They  
both now support PLC for speex (as well as other codecs).

>
>
>> 4) Since we are focusing on the user application development, may  
>> I know
>> what is the highest-level API for Speex that a developer can use  
>> such that
>> all lower-level issues related to buffering, error concealment,  
>> codec, etc.
>> are automatically taken care of?
>>
>
> There's only the Speex API, but I think it's simple enough that you
> don't need a higher level API (at least not for VoIP).
>
>     Jean-Marc
>
> -- 
> Jean-Marc Valin <Jean-Marc.Valin at USherbrooke.ca>
> Université de Sherbrooke
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