asterisk

Commit Graph

Author	SHA1	Message	Date
Leif Madsen	a525edea59	Merged revisions 328247 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 \| lmadsen \| 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) \| 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 \| lmadsen \| 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) \| 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	c26c190711	Asterisk media architecture conversion - no more format bitfields This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	f0190a1092	Fix reading samples from format_mp3 after ast_seekstream/ast_tellstream. There is a bug when using ast_seekstream/ast_tellstream with format_mp3 in that the file read position is not reset before attempting to read samples. So when we seek to determine the maximum size of the file (as in res_agi's STREAM FILE) we weren't then resetting the file pointer so that we could properly read samples. This patch addresses that (in a similar manner to format_wav.c). (closes issue #15224) Reported by: rbd Patches: 20091230_addons_1.4_issue15224.diff uploaded by seanbright (license 71) Tested by: rbd, seanbright Review: https://reviewboard.asterisk.org/r/453 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238014 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	8f336e3d01	Fix trunk building git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227614 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	cea6ee4143	Fix audio problems with format_mp3. This problem was introduced when the AST_FRIENDLY_OFFSET patch was merged. I'm surprised that nobody noticed any trouble when testing that patch, but this fixes the code that fills in the buffer to start filling in after the offset portion of the buffer. (closes issue #15850) Reported by: 99gixxer Patches: issue15850.diff1.txt uploaded by russell (license 2) Tested by: 99gixxer git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	078b0b0eff	Fix memory corruption caused by format_mp3. format_mp3 claimed that it provided AST_FRIENDLY_OFFSET in frames returned by read(). However, it lied. This means that other parts of the code that attempted to make use of the offset buffer would end up corrupting the fields in the ast_filestream structure. This resulted in quite a few crashes due to unexpected values for fields in ast_filestream. This patch closes out quite a few bugs. However, some of these bugs have been open for a while and have been an area where more than one bug has been discussed. So with that said, anyone that is following one of the issues closed here, if you still have a problem, please open a new bug report for the specific problem you are still having. If you do, please ensure that the bug report is based on the newest version of Asterisk, and that this patch is applied if format_mp3 is in use. Thanks! (closes issue #15109) Reported by: jvandal Tested by: aragon, russell, zerohalo, marhbere, rgj (closes issue #14958) Reported by: aragon (closes issue #15123) Reported by: axisinternet (closes issue #15041) Reported by: maxnuv (closes issue #15396) Reported by: aragon (closes issue #15195) Reported by: amorsen Tested by: amorsen (closes issue #15781) Reported by: jensvb (closes issue #15735) Reported by: thom4fun (closes issue #15460) Reported by: marhbere git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c511a26749	Move Asterisk-addons modules into the main Asterisk source tree. Someone asked yesterday, "is there a good reason why we can't just put these modules in Asterisk?". After a brief discussion, as long as the modules are clearly set aside in their own directory and not enabled by default, it is perfectly fine. For more information about why a module goes in addons, see README-addons.txt. chan_ooh323 does not currently compile as it is behind some trunk API updates. However, it will not build by default, so it should be okay for now. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

7 Commits (9988918829c3fd00fc35074f94de12902ebf2052)