asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	76b379c1c8	Fix error that caused truncate operations to fail Another very inappropriate placement of a ')' (again introduced in r362151) caused the various truncate operations to attempt to truncate the sound file at a position of '0'. (issue ASTERISK-19655) Reported by: Matt Jordan (issue ASTERISK-19810) Reported by: colbec ........ Merged revisions 364578 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@364579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	380734fd7b	Fix error that caused seek format operations to set max file size to '1' or '0' A very inappropriate placement of a ')' (introduced in r362151) caused the maximum size of a file to be set as the result of a comparison operation, as opposed to the result of the ftello operation. This resulted in seeking being restricted to the beginning of the file, or 1 byte into the file. Thanks to the Asterisk Test Suite for properly freaking out about this on at least one test. (issue ASTERISK-19655) Reported by: Matt Jordan ........ Merged revisions 362304 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@362305 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	61fb4c0692	Check for IO stream failures in various format's truncate/seek operations For the formats that support seek and/or truncate operations, many of the C library calls used to determine or set the current position indicator in the file stream were not being checked. In some situations, if an error occurred, a negative value would be returned from the library call. This could then be interpreted inappropriately as positional data. This patch checks the return values from these library calls before using them in subsequent operations. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362151 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@362152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	455c5c8c66	Add missing newlines to CLI logging ........ Merged revisions 361471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@361472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	04a28ea1e2	Fix compile problem when old version of libvorbisfile v1.1.2 is used. The principle difference between libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition of the predefined callbacks OV_CALLBACKS_xxx in vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the configure script to detect if libvorbisfile.h declares OV_CALLBACKS_NOCLOSE. * Copied the declaration of OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile. (closes issue ASTERISK-19370) Reported by: Jonn Taylor ........ Merged revisions 355608 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	d7368aabe8	Fix voicemail problems when using ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file format because it did not implement the seek and tell format callbacks among other problems. Since we were already using the libvorbis and libvorbisenc libraries we can use libvorbisfile as it is also part of the vorbis library package. * Made use the libvorbisfile to handle the ogg/vorbis file stream. The format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile. (closes issue ASTERISK-16926) Reported by: sque Patches: ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded by sque ........ Merged revisions 355365 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355375 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Nicholson	cf5149dd31	don't call ltohl() twice on the same value ASTERISK-18739 Patch by: pawel (modified) ........ Merged revisions 344048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	a3f15d7ea9	Merged revisions 330213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330213 \| seanbright \| 2011-07-29 13:18:56 -0400 (Fri, 29 Jul 2011) \| 2 lines Correct the check for O_RDONLY. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@330217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	cc838d5474	Merged revisions 330203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330203 \| seanbright \| 2011-07-29 12:58:08 -0400 (Fri, 29 Jul 2011) \| 2 lines Only write to wav files that were opened to be written to. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@330204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	7caa2349af	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/branches/1.10@328247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	331759b64c	Support for writing and reading raw slin files 8khz-192khz. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	a86c1d68e9	Moves celt and silk format attribute files into res folder. It was inconsistent to have the silk and celt format attribute modules in the format file interpreter folder. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	7fc2a184cd	Adds the format_attr_celt file which was also missing from the CELT pass through patch. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	9c73ece406	Merged revisions 319083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319083 \| dvossel \| 2011-05-16 09:26:33 -0500 (Mon, 16 May 2011) \| 5 lines Fixes Big Endian build issue. (closes issue #19298) Reported by: tzafrir ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	37aa52fd78	Merged revisions 316265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 \| russell \| 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) \| 5 lines Fix a bunch of compiler warnings generated by gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	afc8368dea	Merged revisions 315259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315259 \| russell \| 2011-04-25 14:37:32 -0500 (Mon, 25 Apr 2011) \| 24 lines Merged revisions 315258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315258 \| russell \| 2011-04-25 14:31:44 -0500 (Mon, 25 Apr 2011) \| 17 lines Merged revisions 315257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315257 \| russell \| 2011-04-25 14:28:41 -0500 (Mon, 25 Apr 2011) \| 10 lines Be more flexible with unknown chunks in wav files. This patch makes format_wav ignore unknown chunks instead of erroring out on them. (closes issue #18306) Reported by: jhirsch Patches: wav_skip_unknown_blocks.diff uploaded by jhirsch (license 1156) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	d760e81f37	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 Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 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	14 years ago
Jason Parker	b5f2e2acef	Merged revisions 284701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284701 \| qwell \| 2010-09-02 11:43:09 -0500 (Thu, 02 Sep 2010) \| 8 lines Add slin16 support for format_wav (new wav16 file extension) (closes issue #15029) Reported by: andrew Patches: wav16.patch uploaded by andrew (license 240) Tested by: qwell, andrew ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	317429294c	Merged revisions 279472 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279472 \| tilghman \| 2010-07-25 22:27:06 -0500 (Sun, 25 Jul 2010) \| 2 lines Formats need to load before apps, because some apps call ast_format_str_reduce() at load time. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b4e18d5660	Add load priority order, such that preload becomes unnecessary in most cases git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	fcb055fb4e	addition of G.719 pass-through support (closes issue #16293) Reported by: malcolmd Patches: g719.passthrough.patch.7 uploaded by malcolmd (license 924) format_g719.c uploaded by malcolmd (license 924) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	910144937d	Update supported file extensions in doxygen. Updated the doxygen \arg line after looking at the file for some other Asterisk documentation and noticing they weren't up to date. Thanks to seanbright for looking at the code for me :) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257988 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	8ab22f5dd0	Set a module load priority for format modules. A recent change to app_voicemail made it such that the module now assumes that all format modules are available while processing voicemail configuration. However, when autoloading modules, it was possible that app_voicemail was loaded before the format modules. Since format modules don't depend on anything, set a module load priority on them to ensure that they get loaded first when autoloading. This fix applies to trunk, 1.6.1, and 1.6.2. The fix for 1.4 and 1.6.0 will require a different approach since the module load priority functionality is not present in the module API. (issue #16412) Reported by: jiddings git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d8e0c58437	Expand codec bitfield from 32 bits to 64 bits. Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	39e730b4e2	Remove useless debugging message. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	82fb56886e	More 'static' qualifiers on module global variables. The 'pglobal' tool is quite handy indeed :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	e6b2e9a750	Const-ify the world (or at least a good part of it) This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	89050de146	Merged revisions 186841 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186841 \| mmichelson \| 2009-04-07 19:09:04 -0500 (Tue, 07 Apr 2009) \| 8 lines Fix a few typos of the word "frequency." (closes issue #14842) Reported by: jvandal Patches: frequency-typo.diff uploaded by jvandal (license 413) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	cbe45a6e06	Merged revisions 175825 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175825 \| oej \| 2009-02-15 21:33:17 +0100 (SÃ¶n, 15 Feb 2009) \| 2 lines format_ilbc does not depend on codec libraries and can therefore always be made. My mistake. Ursäkta! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175827 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	798e83aecb	Merged revisions 175792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175792 \| oej \| 2009-02-15 21:20:21 +0100 (SÃ¶n, 15 Feb 2009) \| 2 lines Disable format_ilbc.so by default, like codec_ilbc.so ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	2a53f2ec98	Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14) This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported. Along the way, some related work was done: 1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way. 2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec. 3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip must decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result). Review: http://reviewboard.digium.com/r/158/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	ca38e81892	Add alw as a valid file extension for alaw and ulw as a valid file extension for ulaw. (closes issue #14001) Reported by: henrikw Patches: alw.diff uploaded by henrikw (license 627) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	bd4eb070f3	bring over all the fixes for the warnings found by gcc 4.3.x from the 1.4 branch, and add the ones needed for all the new code here too git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153616 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	8b14e5f493	Reverting format addition for now git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	9a56bae67a	Add native 16kHz format for wav file format. (Closes issue #13657) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	2b06ef2040	Merged revisions 143903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r143903 \| seanbright \| 2008-09-22 18:49:00 -0400 (Mon, 22 Sep 2008) \| 8 lines Use the advertised header size in .au files instead of just assuming they are 24 bytes (the minimum). (closes issue #13450) Reported by: jamessan Patches: pcm-header.diff uploaded by jamessan (license 246) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@143904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	6305009cfc	More merges from resolve-shadow warnings: utils/ codecs/ and a change I missed from formats/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136408 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	b7a4db0ced	Start moving in changes from my resolve-shadow-warnings branch. Going to do this in pieces so the diffs are a little bit smaller and more reviewable. pbx/ and formats/ first. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Brett Bryant	d185405755	Janitor project to convert sizeof to ARRAY_LEN macro. (closes issue #13002) Reported by: caio1982 Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129045 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	f7eb823a7a	Fix a few places where frame data was used directly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	f1e9371da8	- revert change to ast_queue_hangup and create ast_queue_hangup_with_cause - make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	0e6140c564	Use a 32k file buffer on recordings, which increases the efficiency of file recording. (closes issue #11962) Reported by: garlew Patches: recording.patch uploaded by garlew (license 376) bug-11962.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	8b31f34e78	Merged revisions 111658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111658 \| qwell \| 2008-03-28 11:19:56 -0500 (Fri, 28 Mar 2008) \| 8 lines The file size of WAV49 does not need to be an even number. (closes issue #12128) Reported by: mdu113 Patches: 12128-noevenlength.diff uploaded by qwell (license 4) Tested by: qwell, mdu113 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5ca5d97673	Merge changes from team/russell/g722-sillyness ... Fix a number of other places where the number of samples in a G722 frame was not properly handled because of various reasons. main/rtp.c: - When a G722 frame is read from the smoother, the number of samples in the frame must be divided by 2 before being sent out over the network. Even though G722 is 16 kHz, an error in some previous spec has made it so that we have to list the number of samples such as if it was 8 kHz. main/file.c: - When scheduling the next time to expect a frame, take into account that the format of the file we're reading from may not be 8 kHz. codecs/codec_g722.c: - When converting from G722 to slinear, g722_decode() expects its samples parameter to be in the silly (real samples / 2) format. Make it so. - When converting from slinear to G722, properly set the number of samples in the frame to be the number of bytes of output * 2. formats/format_pcm.c: - This format module handles G722, among a number of other formats. However, the read() and seek() functions did not account for the fact that G722 has 2 samples per byte. (closes issue #12130, reported by rickross, patched by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	a82ab5369e	minor formatting changes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	a20b146df5	add a file-format driver for 16KHz signed linear... which may or may not work git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	100ef27af9	Merged revisions 93180 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93180 \| kpfleming \| 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007) \| 23 lines In http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html, rizzo brought up some issues related to the way that the metadata required for menuselect and the rest of the build system is extracted from the source files. Since I had a few hours to kill on an airplane today, I decided to improve this situation... so now the system caches the extracted metadata and uses it to build the menuselect 'tree' as much as it can. The result of this is that when a single source file is changed, only the metadata for that file needs to be extracted again, and the rest is used from the cache files. I also reduced the number of forked processes required to do the metadata extraction; it was actually possible to do most of what we needed in the Makefiles themselves without using any shell scripts at all! On my laptop, these changes resulted in an 80% decrease in the time required for the 'menuselect.makeopts' automatic check to occur after editing a single source file. While doing this work I also cleaned up a few minor things in the Makefiles, adding a check for 'awk' to the configure script and changed all remaining places we use 'grep' or 'awk' to use the ones found by the configure script, and changed the 'prep_tarball' script to build the menuselect metadata so that tarballs of Asterisk will include it and won't require the user to wait while it is extracted after unpacking. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93184 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	5490889153	Put into Makefile.moddir_rules the common instructions used to generate loadable and embedded module lists. Individual Makefiles now are a lot simpler, possibly as simple as this: -include $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps MODULE_PREFIX=cdr_ all: _all include $(ASTTOPDIR)/Makefile.moddir_rules and also more flexible because in a single directory we can combine various types of modules (app_, cdr_, func_, ... ) by simply listing them in the MODULE_PREFIX variable. The individual Makefiles can also create list of modules to be excluded by listing them in the variablel MODULE_EXCLUDE (see an example in channels/Makefile). With this change it becomes trivial to integrate a directory with locally created/modified sources into the main build. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	d652be0930	normalize subdirs' Makefile by using ASTTOPDIR and not .. to reference the top level directory. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

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236 Commits (c64b86f90bc73d3c1ad5887754c2bb4eb2bba222)