Commit Graph

2162 Commits

Author SHA1 Message Date
495d2fdd65 Reland of "Android GlRectDrawer: Add test for RGB rendering"
Reland of https://codereview.webrtc.org/1367923002/.

The bug was that not all platforms support glReadPixels() with GL_RGB. This CL uses GL_RGBA instead.

BUG=webrtc:4742
R=hbos@webrtc.org

Review URL: https://codereview.webrtc.org/1370653002 .

Cr-Commit-Position: refs/heads/master@{#10070}
2015-09-25 08:55:42 +00:00
6979b024d7 Adding stub files for RtpSender/RtpReceiver.
This will allow Chromium's build files to be updated, so that when the
real RtpSender CL is submitted, it doesn't break the FYI bots.

Review URL: https://codereview.webrtc.org/1364813004

Cr-Commit-Position: refs/heads/master@{#10065}
2015-09-24 23:47:59 +00:00
ea70d77fd5 VideoCapturerAndroid: Add test for making calls on stopped camera
BUG=webrtc:4978
R=perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1350663002 .

Cr-Commit-Position: refs/heads/master@{#10062}
2015-09-24 19:41:08 +00:00
34fbfff068 Remove VideoMediaChannel::SetRender().
Was a no-op in current implementation.

BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1334793003 .

Cr-Commit-Position: refs/heads/master@{#10059}
2015-09-24 17:20:36 +00:00
5e9a1bc790 Revert of Android GlRectDrawer: Add test for RGB rendering (patchset #3 id:40001 of https://codereview.webrtc.org/1367923002/ )
Reason for revert:
The test fails on Nexus 9.

Original issue's description:
> Android GlRectDrawer: Add test for RGB rendering
>
> BUG=webrtc:4742
> R=hbos@webrtc.org
>
> Committed: https://crrev.com/6b20ad99e04f594a9a131bea5d80940698e6e8fd
> Cr-Commit-Position: refs/heads/master@{#10050}

TBR=hbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4742

Review URL: https://codereview.webrtc.org/1363613003

Cr-Commit-Position: refs/heads/master@{#10058}
2015-09-24 16:16:20 +00:00
2bc68c731d Wire up QualityScaler for H.264 on Android.
BUG=webrtc:4968
R=magjed@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1365063002 .

Cr-Commit-Position: refs/heads/master@{#10055}
2015-09-24 14:22:37 +00:00
6b20ad99e0 Android GlRectDrawer: Add test for RGB rendering
BUG=webrtc:4742
R=hbos@webrtc.org

Review URL: https://codereview.webrtc.org/1367923002 .

Cr-Commit-Position: refs/heads/master@{#10050}
2015-09-24 13:05:33 +00:00
2efe58b189 VideoCapturerAndroidTest: Dispose PeerConnectionFactory with pending frames
Partial revert of change in testReturnBufferLateEndToEnd from https://codereview.webrtc.org/1350863002/. It is ok to dispose PeerConnectionFactory with pending frames after all.

BUG=webrtc:4909
R=perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1363303002 .

Cr-Commit-Position: refs/heads/master@{#10049}
2015-09-24 12:16:40 +00:00
4a3ccad29e Remove SetAudioDelayOffset() and friends.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1364093002

Cr-Commit-Position: refs/heads/master@{#10047}
2015-09-24 10:53:14 +00:00
61e933eac7 Remove ChannelManager::GetCapabilities()
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1364083002

Cr-Commit-Position: refs/heads/master@{#10045}
2015-09-24 08:45:41 +00:00
facbbecb51 Remove use of DeviceManager from ChannelManager.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1346153002

Cr-Commit-Position: refs/heads/master@{#10042}
2015-09-24 07:41:59 +00:00
7603c76ab0 Revert of Adding PeerConnectionInterface::SetConfiguration method. (patchset #4 id:60001 of https://codereview.webrtc.org/1317353005/ )
Reason for revert:
Broke FYI bots because SetConfiguration is pure virtual and MockPeerConnectionImpl doesn't implement it. Need to reland with SetConfiguration not pure virtual.

Original issue's description:
> Adding PeerConnectionInterface::SetConfiguration method.
>
> Also updated the JNI and Objective-C bindings. Later, will have a CL to
> remove UpdateIce, which this method effectively replaces.
>
> BUG=webrtc:4945
>
> Committed: https://crrev.com/70702afbcb8418fe93747e7ed63bcbf5e56b90e9
> Cr-Commit-Position: refs/heads/master@{#10040}

TBR=guoweis@webrtc.org,pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4945

Review URL: https://codereview.webrtc.org/1361263002

Cr-Commit-Position: refs/heads/master@{#10041}
2015-09-24 00:37:20 +00:00
70702afbcb Adding PeerConnectionInterface::SetConfiguration method.
Also updated the JNI and Objective-C bindings. Later, will have a CL to
remove UpdateIce, which this method effectively replaces.

BUG=webrtc:4945

Review URL: https://codereview.webrtc.org/1317353005

Cr-Commit-Position: refs/heads/master@{#10040}
2015-09-23 23:32:38 +00:00
cbecd358e0 Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ )
Reason for revert:
This CL just landed: https://codereview.chromium.org/1323243006/

Which fixes the FYI bots for the original CL, and breaks them for this revert.

Original issue's description:
> Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
>
> Reason for revert:
> This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
>
> Original issue's description:
> > TransportController refactoring.
> >
> > Getting rid of TransportProxy, and in its place adding a
> > TransportController class which will facilitate access to and manage
> > the lifetimes of Transports. These Transports will now be accessed
> > solely from the worker thread, simplifying their implementation.
> >
> > This refactoring also pulls Transport-related code out of BaseSession.
> > Which means that BaseChannels will now rely on the TransportController
> > interface to create channels, rather than BaseSession.
> >
> > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> > Cr-Commit-Position: refs/heads/master@{#10022}
>
> TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c
> Cr-Commit-Position: refs/heads/master@{#10024}

TBR=pthatcher@webrtc.org,torbjorng@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1361773005

Cr-Commit-Position: refs/heads/master@{#10036}
2015-09-23 18:50:31 +00:00
d0b5b091e4 Add myself as OWNER of webrtc/voice_engine and talk/media/webrtc.
BUG=webrtc:4690
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1359983003 .

Cr-Commit-Position: refs/heads/master@{#10035}
2015-09-23 15:15:29 +00:00
c14f5ff60f Improving support for Android Audio Effects in WebRTC.
Now also supports AGC and NS effects and adds the possibility
to override default settings.

R=magjed@webrtc.org, pbos@webrtc.org, sophiechang@chromium.org
TBR=perkj
BUG=NONE

Review URL: https://codereview.webrtc.org/1344563002 .

Cr-Commit-Position: refs/heads/master@{#10030}
2015-09-23 12:09:40 +00:00
d5c75b1a0b Reduce LS_INFO spam from voice_engine/.
Removes ShouldIgnoreTrace from WebRtcVoiceEngine and removes the spammy
log instances instead. Also removes trace-style logging from getters
(::GetLocalSSRC() for instance would print what SSRC it got, spamming
the log).

BUG=
R=henrika@webrtc.org

Review URL: https://codereview.webrtc.org/1347353004 .

Cr-Commit-Position: refs/heads/master@{#10028}
2015-09-23 11:24:43 +00:00
7d173362d0 Remove the [Un]RegisterVoiceProcessor() API.
BUG=webrtc:4690
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1361633002 .

Cr-Commit-Position: refs/heads/master@{#10027}
2015-09-23 10:23:31 +00:00
09677342ae Remove VoEFile from VoeWrapper and the remaining places in libjingle where it was being used.
BUG=webrtc:4690
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1360773002 .

Cr-Commit-Position: refs/heads/master@{#10026}
2015-09-23 10:05:47 +00:00
f706c8ae91 VideoCapturerAndroid: Fix threading issues
This CL makes the following changes:
 * Instead of creating a new thread per startCapture()/stopCapture() cycle, VideoCapturerAndroid has a single thread that is initialized in the constructor and kept during the lifetime of the instance. This is more convenient because then it is always possible to post runnables without if-checks. This way, a lot of synchronize statements can be removed. Also, when the camera thread is preserved after stopCapture() it is possible to post late returnBuffer() calls to the correct thread.
 * FramePool now enforces single thread use and returnBuffer() calls are posted to the camera thread. This is important because the camera should only be used from one thread, and we call camera.addCallbackBuffer() in returnBuffer().
 * switchCamera() no longer returns false on failure, but instead signals the result via the callback.
 * Update the test testCaptureAndAsyncRender() - it's not a valid use case to have outstanding frames when calling PeerConnectionFactory.dispose(). Instead, the renderer implementations should have release() functions that block until all frames are returned. The release() functions need to be called in the correct order with PeerConnectionFactory.dispose() last.

BUG=webrtc:4909
R=hbos@webrtc.org, perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1350863002 .

Cr-Commit-Position: refs/heads/master@{#10025}
2015-09-23 10:01:40 +00:00
a81a42f584 Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
Reason for revert:
This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.

Original issue's description:
> TransportController refactoring.
>
> Getting rid of TransportProxy, and in its place adding a
> TransportController class which will facilitate access to and manage
> the lifetimes of Transports. These Transports will now be accessed
> solely from the worker thread, simplifying their implementation.
>
> This refactoring also pulls Transport-related code out of BaseSession.
> Which means that BaseChannels will now rely on the TransportController
> interface to create channels, rather than BaseSession.
>
> Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> Cr-Commit-Position: refs/heads/master@{#10022}

TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1358413003

Cr-Commit-Position: refs/heads/master@{#10024}
2015-09-23 09:24:27 +00:00
47ee2f3b9f TransportController refactoring.
Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.

This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.

Review URL: https://codereview.webrtc.org/1350523003

Cr-Commit-Position: refs/heads/master@{#10022}
2015-09-22 22:08:31 +00:00
c1a1b353ec Remove the SetLocalMonitor() API.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1344083004

Cr-Commit-Position: refs/heads/master@{#10020}
2015-09-22 20:31:28 +00:00
07d09364b0 Purge nss files and dependencies.
This replaces https://codereview.webrtc.org/1313233005
which was reverted after triggering Chromium issues.
The only difference is that we're cleaned up dependencies
on use_openssl from the gyp file.

Since https://codereview.chromium.org/1358913003 landed,
this CL should cause no Chromium issues.

BUG=webrtc:4497

Review URL: https://codereview.webrtc.org/1351503004

Cr-Commit-Position: refs/heads/master@{#10019}
2015-09-22 18:58:13 +00:00
8212265053 Android: Add class ThreadUtils with helper function joinUninterruptibly()
R=hbos@webrtc.org

Review URL: https://codereview.webrtc.org/1356293002 .

Cr-Commit-Position: refs/heads/master@{#10016}
2015-09-22 13:17:23 +00:00
22011c1b54 Remove Channel::SetRingbackTone() and Channel::PlayRingbackTone(), and the code beneath it (within libjingle).
BUG=webrtc:4690
TBR=juberti

Review URL: https://codereview.webrtc.org/1325023005

Cr-Commit-Position: refs/heads/master@{#10011}
2015-09-22 10:12:49 +00:00
ef5d5e45cb Add field trial for automic resize in MediaCodecVideoEncoder.
BUG=webrtc:4968

Review URL: https://codereview.webrtc.org/1351573002

Cr-Commit-Position: refs/heads/master@{#10009}
2015-09-22 08:40:46 +00:00
04ac81f2fd Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet).
BUG=4937
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1345913004 .

Cr-Commit-Position: refs/heads/master@{#10004}
2015-09-21 18:48:37 +00:00
275a2f16fd Revert of Replace readable with receiving where receiving means receiving anything (stun ping, response or da… (patchset #7 id:340001 of https://codereview.webrtc.org/1351673003/ )
Reason for revert:
Broke the Windows build:

[226/365] LINK_EMBED cc_perftests.exe
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\remoting\protocol\remoting_unittests.channel_socket_adapter_unittest.obj.rsp /c ..\..\remoting\protocol\channel_socket_adapter_unittest.cc /Foobj\remoting\protocol\remoting_unittests.channel_socket_adapter_unittest.obj /Fdobj\remoting\remoting_unittests.cc.pdb
e:\b\build\slave\win\build\src\remoting\protocol\channel_socket_adapter_unittest.cc(36) : error C3861: 'set_readable': identifier not found
ninja: build stopped: subcommand failed.

Original issue's description:
> Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet).
> If a connection does not receive for 30 seconds, it will be deleted.
> BUG=
>
> Committed: https://crrev.com/ae16f8547d3b447f62f6660f13688585c6c3de15
> Cr-Commit-Position: refs/heads/master@{#10001}

TBR=pthatcher@webrtc.org,honghaiz@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review URL: https://codereview.webrtc.org/1356103002

Cr-Commit-Position: refs/heads/master@{#10002}
2015-09-21 14:20:43 +00:00
ae16f8547d Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet).
If a connection does not receive for 30 seconds, it will be deleted.
BUG=

Review URL: https://codereview.webrtc.org/1351673003

Cr-Commit-Position: refs/heads/master@{#10001}
2015-09-21 13:54:19 +00:00
c19922c181 Android SurfaceViewRenderer: Block in release() until frames are returned and cleanup is done
BUG=webrtc:4742,webrtc:4909
R=hbos@webrtc.org

Review URL: https://codereview.webrtc.org/1354393002 .

Cr-Commit-Position: refs/heads/master@{#10000}
2015-09-21 12:51:20 +00:00
Per
e6d3adab08 Re-add SurfaceTexture as member for setLocalPreview in VideoCapturerAndroid.
The Android camera api requires a surface to be set in order work. In https://codereview.webrtc.org/1354683004/ this surfaceTexture was removed as a member but it turns out that can lead to camera freezes when the device is rotated. This cl re-adds the surface as a member.

BUG= webrtc:5021, webrtc:5003
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1349433003 .

Cr-Commit-Position: refs/heads/master@{#9999}
2015-09-21 11:57:54 +00:00
780be75190 Make PeerConnectionTest.doTest wait for ice candidates
This change the PeerConnectionTest.doTest wait for at least one ice candidate and also make sure the list of candidates in gotIceCandidates is synchronized.

BUG=webrtc:5010

Review URL: https://codereview.webrtc.org/1354913002

Cr-Commit-Position: refs/heads/master@{#9997}
2015-09-21 10:26:19 +00:00
35d1767cc3 Remove the video capture module on Android.
Video capture for android is now implemented in talk/app/webrtc/androidvideocapturer.h

BUG=webrtc:4475

Review URL: https://codereview.webrtc.org/1347083003

Cr-Commit-Position: refs/heads/master@{#9995}
2015-09-21 08:46:37 +00:00
8902433a43 Revert "TransportController refactoring."
This reverts commit 9af63f473e1d0d6c47a741a046c41642dfc1c178.

Cr-Commit-Position: refs/heads/master@{#9994}
2015-09-18 20:50:31 +00:00
9af63f473e TransportController refactoring.
Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.

This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.

This CL also adds some unit tests, and does some renaming.
For example, from "CandidateReady" to "CandidateGathered".

Review URL: https://codereview.webrtc.org/1246913005

Cr-Commit-Position: refs/heads/master@{#9993}
2015-09-18 19:56:02 +00:00
4a783086b6 Android: Add helper class GlTextureFrameBuffer
BUG=webrtc:4993
R=hbos@webrtc.org

Review URL: https://codereview.webrtc.org/1348513003 .

Cr-Commit-Position: refs/heads/master@{#9991}
2015-09-18 14:32:24 +00:00
Per
3520f9e049 Removes camera.setPreviewTexture in doStopCaptureOnCameraThread and removes the try catch statement since the only method throwing an exception was setPreviewTexture.
BUG=webrtc:5003
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1354683004 .

Cr-Commit-Position: refs/heads/master@{#9985}
2015-09-18 09:52:58 +00:00
586b19bdb6 Enable probing with repeated payload packets by default.
To make this possible padding only packets will have the same timestamp
as the previously sent media packet, as long as RTX is not enabled. This
has the side effect that if we send only padding for a long time without
sending media, a receive-side jitter buffer could potentially overflow.

In practice this shouldn't be an issue, partly because RTX is recommended and
used by default, but also because padding typically is terminated before being
received by a client. It is also not an issue for bandwidth estimation as long
as abs-send-time is used instead of toffset.

BUG=chromium:425925
R=mflodman@webrtc.org, sprang@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1327933003 .

Cr-Commit-Position: refs/heads/master@{#9984}
2015-09-18 09:14:42 +00:00
71df77bba0 Remove overridden basictypes.h.
* Use (u)intxx_t for (u)intxx typedefs for all platforms.
* Always include stdint.h.
* Add RTC_ prefix to ARCH_XXX macros.

Chromium did the (u)intxx_t change in
https://codereview.chromium.org/117323010 and
https://codereview.chromium.org/639293007

BUG=chromium:468375
TBR=perkj@webrtc.org (for trivial talk/* changes)
NOTRY=true
NOPRESUBMIT=true

Review URL: https://codereview.webrtc.org/1349213003

Cr-Commit-Position: refs/heads/master@{#9983}
2015-09-18 08:48:39 +00:00
7cbd188c5e Remove GICE (again).
R=guoweis@webrtc.org

Review URL: https://codereview.webrtc.org/1353713002 .

Cr-Commit-Position: refs/heads/master@{#9979}
2015-09-18 01:55:03 +00:00
ac547a6538 Remove channel ids from various interfaces.
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.

IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately

BUG=webrtc:1695
R=henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1335353005 .

Cr-Commit-Position: refs/heads/master@{#9978}
2015-09-17 21:06:02 +00:00
e1c5ec72c6 Fixing bad merge (CHECK is now RTC_CHECK)
BUG=None
TBR=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1346013006 .

Cr-Commit-Position: refs/heads/master@{#9975}
2015-09-17 15:20:43 +00:00
fdd1b9a58e Reland: Bailing out if pc factory fails to get created.
This was reverted, but it turned out GOMA was down.

This prevents us from continuing if we fail initialization.
The failure will happen closer to its source, rather than
when we try to create the first peer connection.

BUG=None
R=glaznev@webrtc.org

Committed: https://crrev.com/6eb75d9e67f02c256436eb96f3c77026486561a1
Cr-Commit-Position: refs/heads/master@{#9948}

Review URL: https://codereview.webrtc.org/1339923004 .

Cr-Commit-Position: refs/heads/master@{#9974}
2015-09-17 14:45:56 +00:00
b071a19019 Full use of NnChannel::SetSendParameters and NnChannel::SetRecvParameters.
SetOptions(), SetMaxBandwidth(), Set[Send|Recv]RtpHeaderExtensions(), Set[Send|Recv]Codecs() are now private.

BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1327933002 .

Cr-Commit-Position: refs/heads/master@{#9973}
2015-09-17 14:43:06 +00:00
f2bfc2b8ef Remove some dead code.
WebRtcPassthroughRender has been dead since webrtcvideoengine.cc was
removed, FakeExternalTransport has probably been unused for a long time.

BUG=webrtc:1695
R=henrika@webrtc.org

Review URL: https://codereview.webrtc.org/1343393003 .

Cr-Commit-Position: refs/heads/master@{#9968}
2015-09-17 11:04:21 +00:00
91d6edef35 Add RTC_ prefix to (D)CHECKs and related 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.

Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1335923002

Cr-Commit-Position: refs/heads/master@{#9964}
2015-09-17 07:24:51 +00:00
7754285f7c Log to the webrtc log stream from webrtc/modules java code.
The purpose is to gather all webrtc logging in a single place and allow the app to redirect all webrtc logging to a single stream for offline debugging.

Moved Logging.java to webrtc/base to be shared by talk/ and modules/.

R=glaznev@webrtc.org, henrika@webrtc.org, magjed@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1335103004 .

Cr-Commit-Position: refs/heads/master@{#9959}
2015-09-16 23:20:48 +00:00
5975b3c5be Log to webrtc logging stream from java code.
Future log messages should all be sent to org.webrtc.Logging as well.

BUG=

Committed: https://crrev.com/66f0da2197974dcc1008f25df2bb4e1d463ad506
Cr-Commit-Position: refs/heads/master@{#9936}

R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1338033003 .

Cr-Commit-Position: refs/heads/master@{#9957}
2015-09-16 20:41:02 +00:00
384194369b Consolidate constructormagic macros with Chromium version and remove Chromium override.
Part of work removing dependency on Chromium's base.

Only adds "= delete". From https://codereview.chromium.org/1151443003 :
"This will guarantee the error to be at compile time, and not rely on the call visibility (private)."

In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.

Depends on https://codereview.webrtc.org/1345433002/

BUG=chromium:468375
(in particular comment #37)
NOTRY=true

Review URL: https://codereview.webrtc.org/1342543004

Cr-Commit-Position: refs/heads/master@{#9954}
2015-09-16 13:33:25 +00:00