[asterisk-dev] More awesomeness in trunk for Asterisk 1.10

Terry Wilson twilson at digium.com
Wed Feb 23 21:11:03 CST 2011

> On Tue, 2011-02-22 at 23:04 +0000, SVN commits to the Asterisk project
> wrote:
>> Author: dvossel
>> Date: Tue Feb 22 17:04:49 2011
>> New Revision: 308582
>> URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=308582
>> Log:
>> Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
>> -Functional changes
>> 1. Dynamic global format list build by codecs defined in codecs.conf
>> 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
>> 3. Negotiation of SILK attributes in chan_sip.
>> 4. SPEEX 32khz with translation
>> 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
>>   using codec_resample.c
>> 6. Various changes to RTP code required to properly handle the dynamic format list
>>   and formats with attributes.
>> 7. ConfBridge now dynamically jumps to the best possible sample rate.  This allows
>>   for conferences to take advantage of HD audio (Which sounds awesome)
>> 8. Audiohooks are no longer limited to 8khz audio, and most effects have been
>>   updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
>> 9. codec_resample now uses its own code rather than depending on libresample.
>> -Organizational changes
>> Global format list is moved from frame.c to format.c
>> Various format specific functions moved from frame.c to format.c

Awesome. David's been busy...

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