Otherwise, we may delete a useful connection because the current best connection may be failing.
BUG=
Review URL: https://codereview.webrtc.org/1364683002
Cr-Commit-Position: refs/heads/master@{#10063}
Was previously using the wrong preprocessor define (DEBUG vs _DEBUG).
Review URL: https://codereview.webrtc.org/1361173002
Cr-Commit-Position: refs/heads/master@{#10061}
It's good hygiene and just generally the right thing to do. And
apparently at least sometimes required by Microsoft's compiler.
Review URL: https://codereview.webrtc.org/1364233002
Cr-Commit-Position: refs/heads/master@{#10060}
This will paves the way for continuous nomination lite and multi-networking.
2. Combined checking and pinging to remove some redundant checking and to make it switch to more frequent ping mode earlier.
Review URL: https://codereview.webrtc.org/1311433009
Cr-Commit-Position: refs/heads/master@{#10057}
We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.
(This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means
(likely just the linker omitting compilation units with no incoming
references).)
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1349393003
Cr-Commit-Position: refs/heads/master@{#10046}
A couple of places were missing handling of UYVY video formats.
BUG=webrtc:4816
Review URL: https://codereview.webrtc.org/1317613003
Cr-Commit-Position: refs/heads/master@{#10044}
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}
No longer lock-order inverting since RTP/RTCP modules are instantiated
on construction and no longer guarded by a separate lock.
BUG=webrtc:2999
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1347283004 .
Cr-Commit-Position: refs/heads/master@{#10034}
The BWE expects arrival timestamps in ms, while the audio path delivered
them in us, causing the BWE to break down under the combined audio/video
BWE experiment. This was introduced in r9892 (68786d2040).
BUG=webrtc:4758
R=mflodman@webrtc.org, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1360913004 .
Cr-Commit-Position: refs/heads/master@{#10032}
Bug 4985 revealed two flaws
1. Opus duration estimate did not return correct length for DTX packets,
2. NetEq DoCodecInternalCng did not assign enough buffer.
P.S.
Generalizing problem 1, current NetEq decode function checks memory size by calling the duration estimate function. This is not ideal. A better way is to let codec's decode function to receive buffer size and return failure if it is not enough. This can be made in a separate CL.
BUG=webrtc:4985
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1334303005 .
Cr-Commit-Position: refs/heads/master@{#10031}
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}
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}
In the middle of refactoring, I replaced the VideoCapturer with
FrameGeneratorCapturer, to reuse the code, and with that disabled the camera.
Now adding capturer_ element to VideoQualityTest and ignoring
frame_generator_capturer_ from the parent class test::CallTest.
Review URL: https://codereview.webrtc.org/1356933005
Cr-Commit-Position: refs/heads/master@{#10023}
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}
* Make sure they're all final and don't allow copying or assignment.
* Get rid of the single-channel PCM decoder classes.
* Move some includes from .h to .cc files where possible.
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1353803002
Cr-Commit-Position: refs/heads/master@{#10021}
Currently, it's sitting in AudioEncoderIsac*'s files, which is less
than obvious. This CL puts the encoder and decoder in separate files
together with the C implementation; CLs are afoot to make it so for
the other built-in codecs as well.
BUG=webrtc:4557
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1339253003 .
Cr-Commit-Position: refs/heads/master@{#10018}
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1348613003
Cr-Commit-Position: refs/heads/master@{#10015}
In screensharing full stack tests, instead of using YuvFileGenerator by default
when no scrolling is used, I always used ScrollingImageFileGenerator.
That possibly slowed down the test a little bit, at least for the slowed
devices, as it unnecessarily copied few MBs per frame.
BUG=chromium:534220
Review URL: https://codereview.webrtc.org/1359783002
Cr-Commit-Position: refs/heads/master@{#10014}
Added to prevent Chromium from breaking if KeyType (now an enum) starts being used in Chromium before KeyType changes to a parameterizable class. When enum -> class change happens, IntKeyTypeFamilyToKeyType will be updated at the same time.
Once Chromium starts using class KeyType with parameters this function can be removed.
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1363543002 .
Cr-Commit-Position: refs/heads/master@{#10013}
While refactoring, I incorrectly set the target bitrate for one of the tests to
500k instead of 2000k.
This does not fix all the perf regressions.
BUG=534220
Review URL: https://codereview.webrtc.org/1356123002
Cr-Commit-Position: refs/heads/master@{#10008}
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}
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}
Reason for revert:
Breaks FYI bots.
ninja: error: '../../third_party/webrtc_overrides/webrtc/base/logging.cc', needed by 'obj/third_party/webrtc_overrides/webrtc/base/rtc_base.logging.o', missing and no known rule to make it
Original issue's description:
> Update build files to use webrtc_overrides in Chromium instead of overrides.
>
> This is a part of moving the overrides to Chromium. See bug comment #65 for all steps.
>
> BUG=chromium:468375
>
> Committed: https://crrev.com/baae0a8a6c873ddf812a5687b84638359b2e7e5b
> Cr-Commit-Position: refs/heads/master@{#9996}
TBR=kjellander@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:468375
Review URL: https://codereview.webrtc.org/1352423002
Cr-Commit-Position: refs/heads/master@{#9998}
This is a part of moving the overrides to Chromium. See bug comment #65 for all steps.
BUG=chromium:468375
Review URL: https://codereview.webrtc.org/1354933002
Cr-Commit-Position: refs/heads/master@{#9996}
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}
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}
