[asterisk-dev] More awesomeness in trunk for Asterisk 1.10
russell at digium.com
Tue Feb 22 17:07:48 CST 2011
On Tue, 2011-02-22 at 23:04 +0000, SVN commits to the Asterisk project
> Author: dvossel
> Date: Tue Feb 22 17:04:49 2011
> New Revision: 308582
> URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=308582
> 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
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