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<a href="http://bamboo.asterisk.org/browse/ASTTRUNK-FREEBSD81-I386/log" style="font-family: Arial, sans-serif; font-size: 15px; font-weight:bold; color:#000">ASTTRUNK-FREEBSD81-I386-168</a>
<span style="font-family: Arial, sans-serif; font-size: 14px;"> has been queued, but there's no agent capable of building it.</span>
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<a href="http://bamboo.asterisk.org/browse/ASTTRUNK-FREEBSD81-I386/log" style="text-decoration: none; font-family: Arial, sans-serif;text-align:left;font-size:16px;font-weight:bold;color:#036" >Code Changes</a>
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<a href="http://bamboo.asterisk.org/browse/author/dvossel" style="font-family: Arial, sans-serif; font-size: 13px; font-weight:bold; color:#000">
dvossel</a><br>
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff<br/>
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-Functional changes<br/>
1. Dynamic global format list build by codecs defined in codecs.conf<br/>
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf<br/>
3. Negotiation of SILK attributes in chan_sip.<br/>
4. SPEEX 32khz with translation<br/>
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation<br/>
using codec_resample.c<br/>
6. Various changes to RTP code required to properly handle the dynamic format list<br/>
and formats with attributes.<br/>
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows<br/>
for conferences to take advantage of HD audio (Which sounds awesome)<br/>
8. Audiohooks are no longer limited to 8khz audio, and most effects have been<br/>
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.<br/>
9. codec_resample now uses its own code rather than depending on libresample.<br/>
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-Organizational changes<br/>
Global format list is moved from frame.c to format.c<br/>
Various format specific functions moved from frame.c to format.c<br/>
<br/>
Review: <a href="https://reviewboard.asterisk.org/r/1104/">https://reviewboard.asterisk.org/r/1104/</a><br/>
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(308582)
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<a href="http://bamboo.asterisk.org/browse/author/rmudgett" style="font-family: Arial, sans-serif; font-size: 13px; font-weight:bold; color:#000">
rmudgett</a><br>
Merged revisions 308622 via svnmerge from <br/>
<a href="https://origsvn.digium.com/svn/asterisk/branches/1.8">https://origsvn.digium.com/svn/asterisk/branches/1.8</a><br/>
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........<br/>
r308622 | rmudgett | 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011) | 9 lines<br/>
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sig_pri_new_ast_channel() should return NULL when new_ast_channel() fails.<br/>
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(closes issue #18874)<br/>
Reported by: cmaj<br/>
Patches:<br/>
patch-sig_pri-crash-possible-null-channel-pointer.diff.txt uploaded by cmaj (license 830)<br/>
<br/>
JIRA SWP-3172<br/>
........<br/>
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(308623)
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<a href="http://bamboo.asterisk.org/browse/author/rmudgett" style="font-family: Arial, sans-serif; font-size: 13px; font-weight:bold; color:#000">
rmudgett</a><br>
Fix compiler warning.
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(308624)
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