26 Commits

Author SHA1 Message Date
solenberg
88499ecaca Moving/renaming webrtc/common.h.
This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.

- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.

BUG=webrtc:5879

Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
2016-09-07 14:34:45 +00:00
ossu
e1f5b4a7fe voice_engine: Removed old variants of Channel constructor and CreateChannel
These are no longer used internally and their interface is not to be
considered public. They were due to be changed in
https://codereview.webrtc.org/1993783002/ but remained due to a
misunderstanding.

Review-Url: https://codereview.webrtc.org/2082483003
Cr-Commit-Position: refs/heads/master@{#13816}
2016-08-18 11:23:04 +00:00
ivoc
14d5dbe5b3 Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
The breaking tests in Chromium have been temporarily disabled, they will be fixed and reenabled soon.

Original CLs: https://codereview.webrtc.org/1748403002/, https://codereview.webrtc.org/2107253002/ and https://codereview.webrtc.org/2106103003/.

TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org,tkchin@webrtc.org
BUG=webrtc:4741, webrtc:5603, chromium:609749

Review-Url: https://codereview.webrtc.org/2110113003
Cr-Commit-Position: refs/heads/master@{#13379}
2016-07-04 14:07:03 +00:00
ivoc
9e03c3b372 Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.

Original issue's description:
> Move RtcEventLog object from inside VoiceEngine to Call.
>
> In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
> The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
>
> BUG=webrtc:4741,webrtc:5603,chromium:609749
> R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016
> Cr-Commit-Position: refs/heads/master@{#13321}

TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741,webrtc:5603,chromium:609749

Review-Url: https://codereview.webrtc.org/2111813002
Cr-Commit-Position: refs/heads/master@{#13340}
2016-06-30 07:59:49 +00:00
Ivo Creusen
1895526c61 Move RtcEventLog object from inside VoiceEngine to Call.
In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.

BUG=webrtc:4741,webrtc:5603,chromium:609749
R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1748403002 .

Cr-Commit-Position: refs/heads/master@{#13321}
2016-06-29 11:57:01 +00:00
ossu
5f7cfa50e5 Moved CreateBuiltinDecoderFactory out to VoEBaseImpl.
VoEBase is plumbed to optionally take an AudioDecoderFactory, or create
a builtin factory if none is provided.

Retained the CreateChannel interfaces in Channel and ChannelManager
and added variants for injecting an AudioDecoderFactory. The
"old-style" variants call CreateBuiltinAudioDecoderFactory to get a
factory to use.

(Just realized this means each channel uses a separate factory with the
old-style calls. Probably ok.)

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/1993783002
Cr-Commit-Position: refs/heads/master@{#12961}
2016-05-30 15:11:36 +00:00
kwiberg
b7f89d6e66 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/
Also introduce a pair of scoped_ptr <-> unique_ptr conversion
functions. By using them judiciously, we can keep these CL:s small and
avoid having to convert enormous amounts of code at once.

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1702983002

Cr-Commit-Position: refs/heads/master@{#11658}
2016-02-17 18:04:26 +00:00
pbos
d8de1154c9 Remove mutable from rtc::CriticalSections.
A couple of mutables were added after last removal of mutables, so
removing those. rtc::CriticalSection is const-lockable.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1652983002

Cr-Commit-Position: refs/heads/master@{#11447}
2016-02-01 17:00:59 +00:00
tommi
31fc21f454 Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/
Also remove mischievous tab character!
This is a part of getting rid of CriticalSectionWrapper and makes the code slightly simpler.

BUG=

Review URL: https://codereview.webrtc.org/1607353002

Cr-Commit-Position: refs/heads/master@{#11346}
2016-01-21 18:37:44 +00:00
Henrik Kjellander
98f53510b2 system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1413333002 .

Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +00:00
Peter Boström
5c389d3e09 Split webrtc/video into webrtc/{audio,call,video}.
Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts
into webrtc/call, splitting out audio/shared components with separate
OWNERS files.

BUG=webrtc:4690
R=solenberg@webrtc.org, tina.legrand@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1227923005 .

Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 11:58:39 +00:00
henrikg
3c089d751e Add RTC_ prefix to contructormagic macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS

Related CL: https://codereview.webrtc.org/1335923002/

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1345433002

Cr-Commit-Position: refs/heads/master@{#9953}
2015-09-16 12:37:52 +00:00
ivoc
b04965ccf8 Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call.
An option was added to voe_cmd_test to make a RtcEventLog dump.

BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1267683002

Cr-Commit-Position: refs/heads/master@{#9901}
2015-09-09 07:09:49 +00:00
Minyue
2013aeced2 Propagating RTT from send-only channel to receive-only channel.
This is important for obtaining ntp time at receiver-only channel, which does not have RTT directly.

BUG=3978

TEST=chromium with hangout calls
R=henrika@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29989004

Cr-Commit-Position: refs/heads/master@{#9186}
2015-05-13 12:14:36 +00:00
kwiberg@webrtc.org
00b8f6b364 Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36229004

Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 14:43:50 +00:00
henrike@webrtc.org
88fbb2d86b Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
Same as https://webrtc-codereview.appspot.com/19519004. The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing...
(tested locally).

BUG=3380
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 21:18:46 +00:00
mcasas@webrtc.org
2fa7f79094 Revert 6202 "Switch to using base/constructormagic.h and remove ..."
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
> 
> BUG=N/A
> R=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/19519004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 11:07:29 +00:00
henrike@webrtc.org
125ffd709d Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 15:20:44 +00:00
turaj@webrtc.org
03f33709f8 Inject config when creating channels to override the existing one.
BUG=
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5116 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 00:02:48 +00:00
minyue@webrtc.org
e509f943ed This issue is related to
https://chromereviews.googleplex.com/9908014/

I was thinking about shipping ACM2 from the signal repository. There seems to be too many changes in one CL.

BUG=
R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2171004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4733 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 17:03:00 +00:00
pbos@webrtc.org
676ff1ed89 Ref-counted rewrite of ChannelManager.
The complexity of the last ChannelManager and potentially usage of it as well caused race conditions and deadlocks in loopback voe_auto_test. This ref-counted solution takes no long-term locks, uses less locks overall and is significantly easier to understand.

ScopedChannel has been split up into a ChannelOwner with a reference to a channel and an Iterator over ChannelManager. Previous code was really used for both things. ChannelOwner is used as a shared pointer to a channel object, while an Iterator should work as expected.

BUG=2081
R=tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1802004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4502 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-07 17:57:36 +00:00
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments.
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-03 15:12:26 +00:00
pbos@webrtc.org
956aa7e087 Include files from webrtc/.. paths in voice_engine/
BUG=1662
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1434005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4079 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-21 13:52:32 +00:00
pbos@webrtc.org
9213521ea9 Remove const for plain data types in voice_engine/
BUG=1644
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1463004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4018 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 08:31:39 +00:00
pbos@webrtc.org
6141e13873 WebRtc_Word32 -> int32_t in voice_engine/
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1305004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3792 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-09 10:09:10 +00:00
andrew@webrtc.org
14b43beb7c Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00