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