Commit Graph

1564 Commits

Author SHA1 Message Date
46550cd947 * #27196: audiofile: fix memory leak 2013-07-11 17:00:52 -04:00
c560e82f1d * #26839: iax: fix warnings 2013-07-11 15:16:25 -04:00
6d7080d59f Merge branch 'master' into multichannel 2013-07-11 14:59:14 -04:00
df05f0880f sdp: fix warning if compiling without video support 2013-07-11 14:58:45 -04:00
9c661cdc06 * #26839: audio: fix formatting 2013-07-11 14:48:32 -04:00
c7fe1262c4 * #26839: audio: fix warnings 2013-07-11 14:48:10 -04:00
d2252d26cc * #26839: audiobuffer: fix warnings 2013-07-11 14:46:44 -04:00
478049e97c * #26839: audio: fix warning 2013-07-11 14:39:43 -04:00
0ae2105bdf * #26839: pulse: fix TODO
You don't need C++11 for uint32_t. C++11 just guarantees a cstdint header
instead of stdint.h
2013-07-11 14:31:08 -04:00
4c995137b6 * #26839: pulselayer: fix formatting with astyle 2013-07-11 14:29:45 -04:00
bb1e4873c5 * #26839: ringbuffer: fix formatting 2013-07-11 14:28:53 -04:00
111b3a15e5 * #26839: ringbuffer: fix warnings 2013-07-11 14:27:53 -04:00
f8c3dba551 * #26839: nameComparator: don't inherit from deprecated unary_function
We are using neither nameComparator::argument_type nor
nameComparator::result_type so there's no need to inherit. This
fixes -Weffc++ warnings.
2013-07-11 14:22:39 -04:00
34b93aee19 * #26839: audiofile: fix formatting, delete commented code, add FIXME 2013-07-11 13:48:43 -04:00
f93bb13142 * #26839: audiofile: fix warnings 2013-07-11 13:31:58 -04:00
2873e76c8e * #26839: audiortp: fix warnings, formatting 2013-07-11 13:29:36 -04:00
5bb9d4cbf7 * #26839: opus: fix warnings, formatting 2013-07-11 12:00:19 -04:00
af10d86b7c * #26839: audiobuffer: fix formatting with astyle 2013-07-11 11:53:47 -04:00
ccacbff784 Merge branch 'master' into multichannel 2013-07-11 10:46:06 -04:00
6c8fc9e955 iax: downgrade false positive errors to debug messages
When dialing with an IAX account, we get a lot of "No SIP CALL" error
messages being logged. This is clearly wrong and until we avoid
looking for SIPCalls from an IAX account, we should present this
message as debug info.
2013-07-10 16:34:12 -04:00
9fdbf217bb * #27096: iax: use union instead of dereferencing type-punned pointer
Patch sent upstream at https://sourceforge.net/p/iaxclient/bugs/43/
2013-07-10 16:30:26 -04:00
1a4a7f37ca * #27095: iax: zero out IAXContext correctly 2013-07-10 16:29:53 -04:00
6ebf64134c Merge branch 'master' into multichannel 2013-07-10 14:38:58 -04:00
236e999b15 * #26839: speexcodec.h must include global.h
Fixes regression from 868bf2cc1a
2013-07-10 14:21:53 -04:00
37cc77e3ec Merge branch 'master' into multichannel
Resolved Conflicts:
	daemon/src/sfl_types.h
2013-07-10 14:20:50 -04:00
868bf2cc1a audio: define SFLDataFormat as int16_t and use SFLDataFormat instead of short
A short is not guaranteed to be 16 bits.
2013-07-10 13:47:07 -04:00
7841a3a04b Merge branch 'master' into multichannel
Resolved Conflicts:
	daemon/src/audio/alsa/alsalayer.cpp
	daemon/src/audio/audioloop.cpp
	daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
	daemon/src/audio/codecs/alaw.cpp
	daemon/src/audio/codecs/audiocodecfactory.cpp
	daemon/src/audio/codecs/g722.cpp
	daemon/src/audio/codecs/gsmcodec.cpp
	daemon/src/audio/codecs/ulaw.cpp
	daemon/src/audio/dcblocker.cpp
	daemon/src/audio/mainbuffer.cpp
	daemon/src/audio/pulseaudio/audiostream.cpp
	daemon/src/audio/pulseaudio/pulselayer.cpp
	daemon/src/audio/ringbuffer.cpp
	daemon/src/audio/samplerateconverter.cpp
	daemon/src/audio/sound/audiofile.cpp
2013-07-09 14:11:10 -04:00
0a48d2ba83 * #26839: alsa: fix comments 2013-07-09 13:38:35 -04:00
4d471ba014 * #26839: audio: fix formatting with astyle 2013-07-09 13:30:16 -04:00
b2eb2b75a1 Merge branch 'master' into multichannel 2013-07-09 11:34:52 -04:00
3209eb4784 * #26807: pulse: fix build for older pulseaudio (< 1.0.0) 2013-07-09 10:48:09 -04:00
9dc8964150 * #20661: opus: ensure that HAVE_OPUS is defined 2013-07-08 14:58:34 -04:00
bc787648c4 Merge branch 'master' into multichannel
Resolved Conflicts:
	daemon/src/audio/audioloop.cpp
	daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
	daemon/src/audio/codecs/audiocodec.h
	daemon/src/audio/pulseaudio/pulselayer.cpp
2013-07-08 14:06:50 -04:00
8e51ae977d * #26896: daemon: remove eclipse project files 2013-07-08 11:50:49 -04:00
9894e13979 * #25537: daemon/dbus renamed daemon/client 2013-07-05 16:04:37 -04:00
3ae58a98d0 * #25537: daemon: allocate bus dispatcher on heap
This avoids exposing dbus dependency in client.h, we can just forward
declare it.
2013-07-05 16:04:37 -04:00
ac11026595 * #25537: daemon: move dbusmanager.* to client.* 2013-07-05 16:04:37 -04:00
7360999f3a * #25537: daemon: rename DBusManager => Client 2013-07-05 16:04:36 -04:00
72b03c94ef * #25537: daemon: rename dbus_ to client_
Rationale: it represents interaction with the various clients, that
it uses D-Bus is an implementation detail.
2013-07-05 16:04:36 -04:00
c0eace9741 * #26807: pulse: create context with pa_context_new_with_proplist 2013-07-04 17:50:19 -04:00
e01146a9dc * #26796: daemon: remove managerimpl_registration 2013-07-04 11:25:10 -04:00
64caafe210 * #14255: dbus: added "started" signal for when daemon is running
Might be useful if daemon has been restarted without the client knowing it.
2013-07-03 15:31:21 -04:00
3b2142540b * #11942: dbus: always notify client when recording playback stopped 2013-07-03 14:18:04 -04:00
f3ef185c83 sipaccount/tls: cleanup 2013-07-03 14:02:13 -04:00
fe027880d3 * #26628: TLS: limit number of ciphers to avoid abort in pjsip 2013-07-03 13:34:31 -04:00
9dc5ca34fd * #7078: audio: suspend audio processing if peer hungup and no calls remain
checkAudio() was being called before the call was removed from the
call list.
2013-06-27 16:23:55 -04:00
53b2393f28 manager: no need to call .get() on shared_ptr 2013-06-27 15:54:08 -04:00
c5ed959a0d * #7037: manager: added comment explaining checkAudio() 2013-06-27 15:53:14 -04:00
dc3c38f6ef * #26544: dbus: improve recording API
Namely by telling the client who triggered recording whether or not
recording has started.
2013-06-27 15:06:42 -04:00
12ba4e340b * #7037: audio: stop audio stream if user starts then stops dialing
This allows other applications to resume audio.
2013-06-27 14:18:31 -04:00