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r257713 | dhubbard | 2010-04-16 16:22:30 -0500 (Fri, 16 Apr 2010) | 28 lines
Merged revisions 257686 via svnmerge from
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r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010) | 21 lines
Make the mixmonitor thread process audio frames faster
Mantis issue 17078 reports MixMonitor recordings have shorter durations than
the call duration. This was because the mixmonitor thread was not processing
frames from the audiohook fast enough. The mixmonitor thread would slowly fall
behind the most recent audio frame and when the channel hangs up, the mixmonitor
thread would exit without processing the same number of frames as the channel;
leaving the mixmonitor recording shorter than actual call duration.
This revision fixes this issue by moving the ast_audiohook_trigger_wait() and
the subsequent audiohook.status check into the block where the
ast_audiohook_read_frame() function returns NULL.
(closes issue #17078)
Reported by: geoff2010
Patches:
dw-M17078.patch uploaded by dhubbard (license 733)
Tested by: dhubbard, geoff2010
Review: https://reviewboard.asterisk.org/r/611/
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r237656 | mvanbaak | 2010-01-05 17:08:12 +0100 (Tue, 05 Jan 2010) | 6 lines
Make CLI command 'mixmonitor start|stop <channel> work again.
(closes issue #16534)
Reported by: jlaguilar
Fix as suggested by jlaguilar in the bugreport
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r230509 | dvossel | 2009-11-19 15:26:21 -0600 (Thu, 19 Nov 2009) | 17 lines
Merged revisions 230508 via svnmerge from
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r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009) | 10 lines
fixes MixMonitor thread not exiting when StopMixMonitor is used
(closes issue #16152)
Reported by: AlexMS
Patches:
stopmixmonitor_1.4.diff uploaded by dvossel (license 671)
Tested by: dvossel, AlexMS
Review: https://reviewboard.asterisk.org/r/424/
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r213113 | dvossel | 2009-08-19 16:21:00 -0500 (Wed, 19 Aug 2009) | 14 lines
Merged revisions 213103 via svnmerge from
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r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19 Aug 2009) | 8 lines
Fixes memory leak caused by incorrectly freeing mixmonitor
(closes issue #15699)
Reported by: edantie
Patches:
mixmonitor.patch uploaded by edantie (license 862)
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https://origsvn.digium.com/svn/asterisk/trunk
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r201445 | dvossel | 2009-06-17 14:45:35 -0500 (Wed, 17 Jun 2009) | 25 lines
Merged revisions 201423 via svnmerge from
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r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009) | 19 lines
StopMixMonitor race condition (not giving up file immediately)
StopMixMonitor only indicates to the MixMonitor thread to stop
writing to the file. It does not guarantee that the recording's
file handle is available to the dialplan immediately after execution.
This results in a race condition. To resolve this, the filestream
pointer is placed in a datastore on the channel. When StopMixMonitor
is called, the datastore is retrieved from the channel and the
filestream is closed immediately before returning to the dialplan.
Documentation indicating the use of StopMixMonitor to free files
has been updated as well.
(closes issue #15259)
Reported by: travisghansen
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/283/
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r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines
Merged revisions 200991 via svnmerge from
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r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines
Improve support for media paths that can generate multiple frames at once.
There are various media paths in Asterisk (codec translators and UDPTL, primarily)
that can generate more than one frame to be generated when the application calling
them expects only a single frame. This patch addresses a number of those cases,
at least the primary ones to solve the known problems. In addition it removes the
broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API
functions, and cleans up various code paths affected by these changes.
https://reviewboard.asterisk.org/r/175/
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r197828 | lmadsen | 2009-05-28 18:04:00 -0400 (Thu, 28 May 2009) | 8 lines
Update documentation in MixMonitor.
Updated the MixMonitor documentation for the 'b' option so that
it is more obvious that you must not optimize away the Local
channel when using this option.
(closes issue #14829)
Reported by: licedey
Tested by: mmichelson, licedey, lmadsen
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r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) | 3 lines
Add some missing cleanup to app_mixmonitor
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r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines
Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up.
app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed
audio to a file. Since this thread runs independently of the channel, it is possible that
the mixmonitor thread's channel pointer will point to freed memory when the channel either
is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the
cases slightly differently).
The solution for this is to employ a datastore, which has the nice benefit of allowing us
to hook into channel masquerades and hangups and update our pointer as necessary. If this
looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more
involved since it does a lot more operations on the channel that is being spied upon.
app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there
is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em-
ploy a condition-and-boolean combination to ensure that the channel thread finishes with
our structure before the mixmonitor thread attempts to free it. No crashes!
(closes issue #14374)
Reported by: aragon
Patches:
14374.patch uploaded by putnopvut (license 60)
Tested by: aragon, putnopvut
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(closes issue #13990)
Reported by: eliel
Patches:
array_len.diff uploaded by eliel (license 64)
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This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format. Currently, a new format is available for
applications and dialplan functions. A good number of conversions to the new format
are also included.
For more information, see the following message to asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html
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administrator to make the decision of what permissions will actually be given,
through the use of the process umask.
(Closes issue# 13751)
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This work is done by lmadsen, junky and mvanbaak
during AstriDevCon.
This is the second audit the CLI got, and
this time lmadsen made sure he had _ALL_ modules
loaded that have CLI commands in them.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 lines
Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait).
(closes issue #11945)
Reported by: xheliox
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Reported by: snuffy
Patch by: snuffy
(Closes issue #11547)
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formatting fixes.
Reported by: eliel
Patch by: eliel,tilghman
(Closes issue #11209)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov 2007) | 6 lines
Close the audio file before sending it to the post processing application.
(closes issue #11357)
Reported by: reformed
Patches:
mixmonitor.patch uploaded by reformed (license 330)
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In this commit:
- move the ast_register/unregister_app functions to module.h
to avoid the need to include pbx.h for the simpler apps;
- move the ast_group structure to channel.h to remove the
dependency of app.h on linkedlists.h
Note, this is a long process that I am doing in small steps.
The main difficulty is that now for each subsystem we
have a single header (e.g. channel.h) included by the subsystem
provider (usually one file, e.g. channel.c) and by its clients
(dozens of them, e.g. we have some 70+ apps and 30+ functions).
This requires the clients to include all the extra headers
required by the provider (eg. lock.h, linkedlists.h, definitions
of substructures...) even though many of the clients would be
just happy with opaque struct declarations and function prototypes.
The long term plan is to eventually rectify this structure
so that the compilation can become faster, and also APIs
are more stable.
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who really need it.
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were included almost everywhere.
Remove some of the instances.
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build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
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r89241 | mmichelson | 2007-11-13 10:02:02 -0600 (Tue, 13 Nov 2007) | 5 lines
Reverting commit made in revision 89205 since it is unnecessary.
Thanks to Kevin for pointing this out
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mixmonitor->post_process
string. This fix prevents that.
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r89205 | mmichelson | 2007-11-12 18:56:46 -0600 (Mon, 12 Nov 2007) | 5 lines
Some sanity checking for MixMonitor. If only 1 argument is given, then the args.options
and args.post_process strings are uninitialized and could contain garbage. This change
handles this situation properly by only using arguments that we have parsed.
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agressive in not initializing arrays passed to pbx_substitute_variables_xxxx; I reviewed the code (again) and hopefully found every possible spot where substitute_variables is called conditionally, and made sure the char array involved was set to a null string.
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(closes issue #11171, reported and patched by blitzrage)
Many thanks!
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in the pbx, a 25-30% speed boost. The two updates used, are, first, to merge the ast_exists_extension() and the ast_spawn_extension() where they are called sequentially in a loop in the code, into a slightly upgraded version of ast_spawn_extension(), with a few extra args; and, second, I modified the substitute_variables_helper_full, so it zeroes out the byte after the evaluated string instead of demanding you pre-zero the buffer; I also went thru the code and removed the code that zeroed this buffer before every call to the substitute_variables_helper_full. The first fix provides about a 9% speedup, and the second the rest. These figures come from the 'PIPS' benchmark I describe in blogs, conf. reports, etc.
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didn't make much sense
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Closes issue #11039, as suggested by seanbright.
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(closes issue #10724)
Reported by: eliel
Patches:
chan_skinny.c.patch uploaded by eliel (license 64)
chan_oss.c.patch uploaded by eliel (license 64)
chan_mgcp.c.patch2 uploaded by eliel (license 64)
pbx_config.c.patch uploaded by seanbright (license 71)
iax2-provision.c.patch uploaded by eliel (license 64)
chan_gtalk.c.patch uploaded by eliel (license 64)
pbx_ael.c.patch uploaded by seanbright (license 71)
file.c.patch uploaded by seanbright (license 71)
image.c.patch uploaded by seanbright (license 71)
cli.c.patch uploaded by moy (license 222)
astobj2.c.patch uploaded by moy (license 222)
asterisk.c.patch uploaded by moy (license 222)
res_limit.c.patch uploaded by seanbright (license 71)
res_convert.c.patch uploaded by seanbright (license 71)
res_crypto.c.patch uploaded by seanbright (license 71)
app_osplookup.c.patch uploaded by seanbright (license 71)
app_rpt.c.patch uploaded by seanbright (license 71)
app_mixmonitor.c.patch uploaded by seanbright (license 71)
channel.c.patch uploaded by seanbright (license 71)
translate.c.patch uploaded by seanbright (license 71)
udptl.c.patch uploaded by seanbright (license 71)
threadstorage.c.patch uploaded by seanbright (license 71)
db.c.patch uploaded by seanbright (license 71)
cdr.c.patch uploaded by moy (license 222)
pbd_dundi.c.patch uploaded by moy (license 222)
app_osplookup-rev83558.patch uploaded by moy (license 222)
res_clioriginate.c.patch uploaded by moy (license 222)
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listen and manipulate the audio going through a channel.
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few other formatting cleanups, too.
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universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
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run on peer. Had to upgrade ast_flag stuff to 64 bits to do this.
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ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application.
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ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it.
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r72381 | file | 2007-06-27 19:25:12 -0400 (Wed, 27 Jun 2007) | 10 lines
Merged revisions 72378 via svnmerge from
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r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun 2007) | 2 lines
Update documentation to clarify variable usage with MixMonitor. (issue #9494 reported by netoguy)
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guidelines changes
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command. (issue #9862 reported by eliel)
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places in the code where the same block of code for creating detached threads
was replicated. (patch from bbryant)
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r52717 | file | 2007-01-29 18:43:40 -0500 (Mon, 29 Jan 2007) | 10 lines
Merged revisions 52716 via svnmerge from
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r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan 2007) | 2 lines
Now that filename is part of the structure and since it comes before postprocess... we have to add it to our postprocess line. (reported on asterisk-dev by Boris Bakchiev)
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r52163 | file | 2007-01-24 20:51:35 -0500 (Wed, 24 Jan 2007) | 10 lines
Merged revisions 52162 via svnmerge from
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r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan 2007) | 2 lines
Add another note about audio files being played back to each bridged party. (issue #8718 reported by ppyy)
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r51407 | file | 2007-01-22 14:13:44 -0500 (Mon, 22 Jan 2007) | 10 lines
Merged revisions 51406 via svnmerge from
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r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan 2007) | 2 lines
Move filestream creation to Mixmonitor loop. This will prevent a blank file from being created if no frames ever pass through to be recorded. (issue #7589 reported by steve_mcneil)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51408 f38db490-d61c-443f-a65b-d21fe96a405b
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