Commit Graph

10060 Commits

Author SHA1 Message Date
9397d84659 Roll chromium_revision 625f6c8..657e8d9 (356202:356260)
Change log: 625f6c8..657e8d9
Full diff: 625f6c8..657e8d9

No dependencies changed.
No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1412883004

Cr-Commit-Position: refs/heads/master@{#10428}
2015-10-27 12:03:55 +00:00
27f6fd346a Remove noparent from talk/OWNERS.
Lets webrtc root OWNERS approve talk/ code as well.

BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1413773005

Cr-Commit-Position: refs/heads/master@{#10427}
2015-10-27 11:08:19 +00:00
5ddee021dd Landmine: clobber to remove out/{Debug,Release}/args.gn
Landmine support was added back in
https://codereview.webrtc.org/1402923003/

BUG=webrtc:5070,webrtc:5123
TBR=phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/1415453006 .

Cr-Commit-Position: refs/heads/master@{#10426}
2015-10-27 11:00:00 +00:00
4f847da5a0 Use webrtc/base/checks.h in desktop_capture.
Collided with CHECKs included in logging headers.

BUG=webrtc:5118
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1418423003

Cr-Commit-Position: refs/heads/master@{#10425}
2015-10-27 10:43:11 +00:00
85a0496b8c Implement AudioSendStream::GetStats().
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1414743004

Cr-Commit-Position: refs/heads/master@{#10424}
2015-10-27 10:35:30 +00:00
2a0a2a410f Add stats for used video codec type for a sent video stream:
- "WebRTC.Video.Encoder.CodecType"

BUG=chromium:512752

Review URL: https://codereview.webrtc.org/1426673002

Cr-Commit-Position: refs/heads/master@{#10423}
2015-10-27 08:32:06 +00:00
18ba3e263c Roll chromium_revision faa5502..625f6c8 (356073:356202)
Change log: faa5502..625f6c8
Full diff: faa5502..625f6c8

Changed dependencies:
* src/third_party/libvpx_new/source/libvpx: b44c5cf..f4af1a9
* src/third_party/libyuv: ad36ba5..2844662
DEPS diff: faa5502..625f6c8/DEPS

No update to Clang.

TBR=marpan@webrtc.org, stefan@webrtc.org,
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1414033006

Cr-Commit-Position: refs/heads/master@{#10422}
2015-10-27 02:55:43 +00:00
18a944bf0a Revert of Adding the ability to change ICE servers through SetConfiguration. (patchset #7 id:120001 of https://codereview.webrtc.org/1391013007/ )
Reason for revert:
Caused compiler warning, breaking Chrome FYI bots.

Original issue's description:
> Adding the ability to change ICE servers through SetConfiguration.
>
> Added a SetIceServers method to PortAllocator. Also added a new
> PeerConnection Initialize method that takes a PortAllocator, in the
> hope that we can get rid of PortAllocatorFactoryInterface, since the
> only substantial thing a factory does is convert the webrtc:: ICE
> servers to cricket:: versions.
>
> Committed: https://crrev.com/d3b26d94399ff539db375a9b84010ee75479d4cf
> Cr-Commit-Position: refs/heads/master@{#10420}

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

Review URL: https://codereview.webrtc.org/1424803004

Cr-Commit-Position: refs/heads/master@{#10421}
2015-10-27 02:21:45 +00:00
d3b26d9439 Adding the ability to change ICE servers through SetConfiguration.
Added a SetIceServers method to PortAllocator. Also added a new
PeerConnection Initialize method that takes a PortAllocator, in the
hope that we can get rid of PortAllocatorFactoryInterface, since the
only substantial thing a factory does is convert the webrtc:: ICE
servers to cricket:: versions.

Review URL: https://codereview.webrtc.org/1391013007

Cr-Commit-Position: refs/heads/master@{#10420}
2015-10-27 00:55:27 +00:00
2b5586774c Exposing DTLS transport state from TransportChannel.
This is necessary in order to support the RTCPeerConnectionState enum in
the future, as well as a correct RTCIceConnectionState (which isn't a
combination ICE and DTLS state).

Review URL: https://codereview.webrtc.org/1414363002

Cr-Commit-Position: refs/heads/master@{#10419}
2015-10-27 00:23:34 +00:00
b0bb77fd61 Reland of Add experiment on weak ping delay during call set up time (patchset #1 id:1 of https://codereview.webrtc.org/1416773003/ )
Reason for revert:
https://codereview.chromium.org/1419253002 is landed to address this linker issue. Keep my fingers crossed.

Original issue's description:
> Revert of Add experiment on weak ping delay during call set up time (patchset #4 id:60001 of https://codereview.webrtc.org/1411883002/ )
>
> Reason for revert:
> This CL breaks Chromium, undefined reference link error to webrtc::field_trial::FindFullName. Adding the dependency system_wrappers to the rtc_p2p target is not enough to fix this.
>
> Looking at field_trial.h (in system_wrappers/interface/, not to be confused with the one in test/) the documentation says "WebRTC clients MUST provide an implementation of: ...FindFullName... Or link with a default one provided in: ...system_wrappers.gyp:field_trial_default).
>
> So maybe just depend on field_trial_default? Not sure what should be done in case there are implications to adding this dependency, I'm reverting this. Sorry :)
>
> Original issue's description:
> > Add experiment on weak ping delay during call set up time
> >
> > BUG=
> > R=pthatcher@webrtc.org
> >
> > Committed: https://crrev.com/3bf69b15f4c0c0ca4ab17c237084684a37bb8279
> > Cr-Commit-Position: refs/heads/master@{#10343}
>
> TBR=pthatcher@webrtc.org,juberti@webrtc.org,guoweis@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>

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

Review URL: https://codereview.webrtc.org/1413603005

Cr-Commit-Position: refs/heads/master@{#10418}
2015-10-26 22:10:06 +00:00
8f46c63f6f Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
Reason for revert:
Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail.

Original issue's description:
> Adding the ability to create an RtpSender without a track.
>
> This CL also changes AddStream to immediately create a sender, rather
> than waiting until the track is seen in SDP. And the PeerConnection now
> builds the list of "send streams" from the list of senders, rather than
> the collection of local media streams.
>
> Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> Cr-Commit-Position: refs/heads/master@{#10414}

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

Review URL: https://codereview.webrtc.org/1426443007

Cr-Commit-Position: refs/heads/master@{#10417}
2015-10-26 21:11:25 +00:00
aed571f6fb Roll chromium_revision 27af50f..faa5502 (356022:356073)
Change log: 27af50f..faa5502
Full diff: 27af50f..faa5502

No dependencies changed.
No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1422073002

Cr-Commit-Position: refs/heads/master@{#10416}
2015-10-26 19:22:12 +00:00
e2a83eee73 Introduce rtc::ArrayView<T>, which keeps track of an array that it doesn't own
The main intended use case is as a function argument, replacing the
harder-to-read and harder-to-use separate pointer and size arguments.
It's easier to read because it's just one argument instead of two, and
with clearly defined semantics; it's easier to use because it has
iterators, and will automatically figure out the size of arrays.

BUG=webrtc:5028
R=andrew@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1408403002 .

Cr-Commit-Position: refs/heads/master@{#10415}
2015-10-26 18:51:42 +00:00
ac9d92ccbe Adding the ability to create an RtpSender without a track.
This CL also changes AddStream to immediately create a sender, rather
than waiting until the track is seen in SDP. And the PeerConnection now
builds the list of "send streams" from the list of senders, rather than
the collection of local media streams.

Review URL: https://codereview.webrtc.org/1413713003

Cr-Commit-Position: refs/heads/master@{#10414}
2015-10-26 18:48:26 +00:00
4cba4eba59 Disable denoising for VP9 by default.
BUG=webrtc:5108
R=marpan@webrtc.org

Review URL: https://codereview.webrtc.org/1418133012

Cr-Commit-Position: refs/heads/master@{#10413}
2015-10-26 18:18:24 +00:00
65e7d4cf20 Remove CanCreateAndDestroyManyVideoStreams.
This test was used to verify that VideoEngine handles were handed back
correctly. This is no longer applicable.

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

Review URL: https://codereview.webrtc.org/1425673002 .

Cr-Commit-Position: refs/heads/master@{#10412}
2015-10-26 16:39:43 +00:00
c4ef1439f6 Revert "Add GN Build file for rtc_sound target."
This reverts commit f054819e257a4f9cbb7fa82ba51dc2335f4359ec,
2d3747de9b7c3014e106d3766dc07cf5da3e1881 and
7ef0553c85c5b373535d7f6161e9a6d3b5b9a826.
It seems harder than expected to get a GN build for rtc_sound
and we lack sufficient trybot support for the case where
WebRTC is built as part of Chromium.

The Debug builds failed like this:
[6939/7454] SOLINK ./libcontent.so
FAILED: ../../third_party/llvm-build/Release+Asserts/bin/clang++ -shared -Wl,--fatal-warnings -fPIC -Wl,-z,noexecstack -Wl,-z,now -Wl,-z,relro -Wl,-z,defs -B../../third_party/binutils/Linux_x64/Release/bin -fuse-ld=gold -Wl,--icf=all -pthread -m64 -Wl,--export-dynamic -o ./libcontent.so -Wl,-soname=libcontent.so @./libcontent.so.rsp && { readelf -d ./libcontent.so | grep SONAME ; nm -gD -f p ./libcontent.so | cut -f1-2 -d' '; } > ./libcontent.so.tmp && if ! cmp -s ./libcontent.so.tmp ./libcontent.so.TOC; then mv ./libcontent.so.tmp ./libcontent.so.TOC; fi
../../third_party/webrtc/sound/alsasoundsystem.cc:453: error: undefined reference to 'rtc::LateBindingSymbolTable::Load()'
../../third_party/webrtc/base/latebindingsymboltable.h.def:62: error: undefined reference to 'rtc::LateBindingSymbolTable::IsLoaded() const'
../../third_party/webrtc/base/latebindingsymboltable.h.def:62: error: undefined reference to 'rtc::LateBindingSymbolTable::IsLoaded() const'
../../third_party/webrtc/base/latebindingsymboltable.h.def:62: error: undefined reference to 'rtc::LateBindingSymbolTable::IsLoaded() const'
../../third_party/webrtc/base/latebindingsymboltable.h.def:62: error: undefined reference to 'rtc::LateBindingSymbolTable::IsLoaded() const'
../../third_party/webrtc/base/latebindingsymboltable.cc.def:63: error: undefined reference to 'rtc::LateBindingSymbolTable::LateBindingSymbolTable(rtc::LateBindingSymbolTable::TableInfo const*, void**)'
../../third_party/webrtc/base/latebindingsymboltable.cc.def:65: error: undefined reference to 'rtc::LateBindingSymbolTable::~LateBindingSymbolTable()'
clang: error: linker command failed with exit code 1 (use -v to see invocation)
ninja: build stopped: subcommand failed.

BUG=webrtc:4160
TBR=tfarina@chromium.org

Review URL: https://codereview.webrtc.org/1407893005 .

Cr-Commit-Position: refs/heads/master@{#10411}
2015-10-26 16:39:21 +00:00
717432f130 Remove network_enabled_crit_ in call.cc.
After #10321 (5a289393928c18af580c6339ba77600fb67006e2) I don't see that
we still need this lock.

R=pbos@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1409193003 .

Cr-Commit-Position: refs/heads/master@{#10410}
2015-10-26 15:34:58 +00:00
09b38f3ca0 Re-enable VP9 resize test.
TBR=stefan@webrtc.org
BUG=webrtc:5097

Review URL: https://codereview.webrtc.org/1409143005 .

Cr-Commit-Position: refs/heads/master@{#10409}
2015-10-26 15:22:41 +00:00
7ef0553c85 Fix for Win GN Build.
This changes it to inherit common configuration, in order to LOG() macro
take effect (hopefully).

This should fix the following errors:
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc.exe "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86/cl.exe" /nologo /showIncludes /FC @obj/third_party/webrtc/sound/rtc_sound/nullsoundsystem.obj.rsp /c ../../third_party/webrtc/sound/nullsoundsystem.cc /Foobj/third_party/webrtc/sound/rtc_sound/nullsoundsystem.obj /Fdobj/third_party/webrtc/sound/rtc_sound_cc.pdb
e:\b\build\slave\win_gn\build\src\third_party\webrtc\sound\nullsoundsystem.cc(78) : error C3861: 'LOG': identifier not found
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc.exe "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86/cl.exe" /nologo /showIncludes /FC @obj/third_party/webrtc/sound/rtc_sound/platformsoundsystemfactory.obj.rsp /c ../../third_party/webrtc/sound/platformsoundsystemfactory.cc /Foobj/third_party/webrtc/sound/rtc_sound/platformsoundsystemfactory.obj /Fdobj/third_party/webrtc/sound/rtc_sound_cc.pdb
e:\b\build\slave\win_gn\build\src\third_party\webrtc\sound\platformsoundsystemfactory.cc(29) : error C3861: 'LOG': identifier not found
ninja: build stopped: subcommand failed.

BUG=webrtc:4160
R=kjellander@webrtc.org
NOTRY=True

Review URL: https://codereview.webrtc.org/1419413002

Cr-Commit-Position: refs/heads/master@{#10408}
2015-10-26 13:48:11 +00:00
2d3747de9b Fix for Mac GN BUILD.
It can't find //webrtc/base:rtc_base, which is weird, the fix is to use
a relative path.

This should fix the following error:

ERROR at //third_party/webrtc/sound/BUILD.gn:38:5: Can't load input
file.
    "//webrtc/base:rtc_base",
    ^-----------------------

NOTRY=true
BUG=webrtc:4160
TBR=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1419953003

Cr-Commit-Position: refs/heads/master@{#10407}
2015-10-26 12:47:41 +00:00
e9eca8f5ae Removing AudioCoding class, a.k.a the new ACM API
We have decided not to do a switch from old (AudioCodingModule) to new
(AudioCoding) API. Instead, we will gradually evolve the old API to
meet the new design goals.

As a consequence of this decision, the AudioCoding and AudioCodingImpl
classes are deleted. Also removing associated unit test sources. No
test coverage is lost with this operation, since the tests for the
"old" API are testing more than the deleted tests did.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1415163002

Cr-Commit-Position: refs/heads/master@{#10406}
2015-10-26 12:26:45 +00:00
f054819e25 Add GN Build file for rtc_sound target.
Tested on Linux with the following command lines:

$ gn gen out-gn/Release --args='is_debug=false target_cpu="x64"
build_with_chromium=false'
$ ninja -C out-gn/Release rtc_sound

BUG=webrtc:4160
R=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1425583002

Cr-Commit-Position: refs/heads/master@{#10405}
2015-10-26 12:15:33 +00:00
213b5987c9 Roll chromium_revision c86a4e2..27af50f (356002:356022)
Change log: c86a4e2..27af50f
Full diff: c86a4e2..27af50f

No dependencies changed.
No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1425623002

Cr-Commit-Position: refs/heads/master@{#10404}
2015-10-26 11:24:08 +00:00
415d2cd745 Use webrtc/base/logging.h for video.
BUG=webrtc:5118
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1415413004 .

Cr-Commit-Position: refs/heads/master@{#10403}
2015-10-26 10:35:26 +00:00
f9af108e08 Roll chromium_revision c708f39..c86a4e2 (355993:356002)
Change log: c708f39..c86a4e2
Full diff: c708f39..c86a4e2

No dependencies changed.
No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1418413002

Cr-Commit-Position: refs/heads/master@{#10402}
2015-10-26 02:54:22 +00:00
484e5482f9 Roll chromium_revision bbfaf80..c708f39 (355989:355993)
Change log: bbfaf80..c708f39
Full diff: bbfaf80..c708f39

No dependencies changed.
No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1416873006

Cr-Commit-Position: refs/heads/master@{#10401}
2015-10-25 18:53:38 +00:00
eb2a91e28f Roll chromium_revision 5512fa0..bbfaf80 (355985:355989)
Change log: 5512fa0..bbfaf80
Full diff: 5512fa0..bbfaf80

No dependencies changed.
No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1405323005

Cr-Commit-Position: refs/heads/master@{#10400}
2015-10-25 10:55:04 +00:00
7542ed604c Roll chromium_revision da8662f..5512fa0 (355980:355985)
Change log: da8662f..5512fa0
Full diff: da8662f..5512fa0

No dependencies changed.
No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1423863002

Cr-Commit-Position: refs/heads/master@{#10399}
2015-10-25 05:27:52 +00:00
9ec27e14b7 Roll chromium_revision da9833c..da8662f (355969:355980)
Change log: da9833c..da8662f
Full diff: da9833c..da8662f

No dependencies changed.
No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1411083011

Cr-Commit-Position: refs/heads/master@{#10398}
2015-10-24 18:53:36 +00:00
5d9b92b53d Update Bind to match its comments and always capture by value. Also update the generated count to 9 args.
The existing comment is wrong, and the test even ensures it: Bind will capture reference values by reference. That makes it hard to use with AsyncInvoker, because you can't safely Bind to a function that takes (const) reference params.

The new version of this code strips references in the bound object, so it captures by value, but can bind against functions that take const references, they'll just be references to the copy.

As the class comment implies, actual by-reference args should be passed as pointers or things that safely share (e.g. scoped_refptr) and not references directly. A new test case ensures the pointer reference works. The new code will also give a compiler error if you try to bind
to a non-const reference.

BUG=

Review URL: https://codereview.webrtc.org/1291543006

Cr-Commit-Position: refs/heads/master@{#10397}
2015-10-24 18:14:52 +00:00
2dd8bf825a Roll chromium_revision 53f0e22..da9833c (355953:355969)
Change log: 53f0e22..da9833c
Full diff: 53f0e22..da9833c

No dependencies changed.
No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1404353009

Cr-Commit-Position: refs/heads/master@{#10396}
2015-10-24 10:57:15 +00:00
7d35afdcbf Roll chromium_revision bd99556..53f0e22 (355580:355953)
Change log: bd99556..53f0e22
Full diff: bd99556..53f0e22

Changed dependencies:
* src/third_party/libyuv: e6a54f2..ad36ba5
DEPS diff: bd99556..53f0e22/DEPS

No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1422883002

Cr-Commit-Position: refs/heads/master@{#10395}
2015-10-24 03:31:52 +00:00
401fb0648a SurfaceTextureHelper: Remove use of quitSafely() because it's API lvl 18
There is no reason to not just quit() in release().

Review URL: https://codereview.webrtc.org/1418563005

Cr-Commit-Position: refs/heads/master@{#10394}
2015-10-24 01:17:59 +00:00
238b15d543 SurfaceViewRenderer: Remove use of quitSafely() because it's API lvl 18
I replaced quitSafely() with a CountDownLatch. The reason for not using ThreadUtils.invokeUninterruptibly() is that I want to stop accepting frames asap, and invokeUninterruptibly() would still accept frames during the waiting time.

BUG=webrtc:4742

Review URL: https://codereview.webrtc.org/1418223002

Cr-Commit-Position: refs/heads/master@{#10393}
2015-10-24 01:14:33 +00:00
c3402fc3ef EGL10.eglCreateWindowSurface(): Replace Surface input with SurfaceHolder
Sending a Surface as input to EGL10.eglCreateWindowSurface() is not supported everywhere. See this code as reference:
ae9610220b/opengl/tools/glgen/stubs/egl/eglCreateWindowSurface.java (42)

Sending a SurfaceHolder as input instead should hopefully be supported everywhere, and this is also what GlSurfaceView does:
http://grepcode.com/file/repository.grepcode.com/java/ext/com.google.android/android/5.1.1_r1/android/opengl/GLSurfaceView.java#1076

Review URL: https://codereview.webrtc.org/1416213004

Cr-Commit-Position: refs/heads/master@{#10392}
2015-10-24 01:13:20 +00:00
90d67ddc1d Remove two more deprecated methods from SocketAddress API.
This patch removes IPToString and IPToSensitiveString static helper
methods, since there are class methods that replace them already, and
they aren't used by anyone anymore.

BUG=None
R=pthacher@webrtc.org

Review URL: https://codereview.webrtc.org/1408873005

Cr-Commit-Position: refs/heads/master@{#10391}
2015-10-23 18:22:06 +00:00
49e196af40 Remove VideoFrameType aliases for FrameType.
No longer used in Chromium, so these can now be removed.

BUG=webrtc:5042
R=mflodman@webrtc.org
TBR=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1415693002 .

Cr-Commit-Position: refs/heads/master@{#10390}
2015-10-23 13:58:27 +00:00
a99069db63 Fix win32 header include order in rtp_utility.h.
Matches the include order in webrtc/base/criticalsection.h and makes use
of winsock2.h instead of winsock.h for consistency.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1407053008

Cr-Commit-Position: refs/heads/master@{#10389}
2015-10-23 13:32:44 +00:00
225789d067 Move logging CriticalSection into implementation.
Prevents including platform headers from all files that include logging.
Also removes warn_slow_logs_delay_ which adds contention to the logging
critical section.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1416373004 .

Cr-Commit-Position: refs/heads/master@{#10388}
2015-10-23 13:21:10 +00:00
aa0429928d Don't wait until distant future to shut down video app.
BUG=
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1415033005 .

Cr-Commit-Position: refs/heads/master@{#10387}
2015-10-23 13:10:05 +00:00
27dfe201a5 Remove final from rtc::Buffer.
With it removed, you can now use it with scoped_refptr by wrapping it in
an rtc::RefCountedObject<rtc::Buffer>.

BUG=

Review URL: https://codereview.webrtc.org/1414053003

Cr-Commit-Position: refs/heads/master@{#10386}
2015-10-23 13:01:14 +00:00
1e737c6f2c Fix thread safety in VcmCapturer.
Makes VcmCapturer::Stop() blocking so that no frames can be in delivery
while the camera has stopped.

BUG=
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1411813004 .

Cr-Commit-Position: refs/heads/master@{#10385}
2015-10-23 12:46:06 +00:00
bbe876f0d3 Set send times in send time history via OnSentPacket.
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1419503004

Cr-Commit-Position: refs/heads/master@{#10384}
2015-10-23 09:05:43 +00:00
9a4cd87640 Add support for handling reordered SS data on the receive-side for VP9.
BUG=chromium:500602

Review URL: https://codereview.webrtc.org/1386903002

Cr-Commit-Position: refs/heads/master@{#10383}
2015-10-23 07:27:22 +00:00
a3587fb779 clean up field_trial_default target, to be used by remoting_perftests.
TBR=tommi@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1415743005

Cr-Commit-Position: refs/heads/master@{#10382}
2015-10-23 02:33:22 +00:00
00507f8eb6 Separate StunProber::Start into Prepare and Run so we could create multiple of them and send out STUN pings at regular interval.
Also update the wake up logic to handle the case if <5 ms interval is requested.

BUG=

Review URL: https://codereview.webrtc.org/1422593002

Cr-Commit-Position: refs/heads/master@{#10381}
2015-10-23 02:16:02 +00:00
4f6a8b5f55 Revert of Add experiment on weak ping delay during call set up time (patchset #1 id:1 of https://codereview.webrtc.org/1406153005/ )
Reason for revert:
Still cause break on mac. reverting it again.

Original issue's description:
> Reland of Add experiment on weak ping delay during call set up time (patchset #1 id:1 of https://codereview.webrtc.org/1423443002/ )
>
> Reason for revert:
> This should be safe to land now.
>
> Original issue's description:
> > Revert of Add experiment on weak ping delay during call set up time (patchset #4 id:60001 of https://codereview.webrtc.org/1411883002/ )
> >
> > Reason for revert:
> > guoweis - Here's the target that's failing:
> > https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle_nacl.gyp&l=17
> >
> > This has unfortunately been causing problems repeatedly for us since libjingle_nacl is maintained separately from libjingle (I don't know the history).
> >
> > The way this works for Chrome in general is that the FindFullName method is implemented in init_webrtc.cc in the overrides folder in Chrome and that hooks WebRTC up with Chrome's implementation.  I'm not sure if that's the right thing to do for nacl, how webrtc is initialized there etc.  I'll ping the nacl team for some help too offline and include you.  Reverting this change for now.
> >
> > Original issue's description:
> > > Add experiment on weak ping delay during call set up time
> > >
> > > BUG=
> > > R=pthatcher@webrtc.org
> > >
> > > Committed: https://crrev.com/3bf69b15f4c0c0ca4ab17c237084684a37bb8279
> > > Cr-Commit-Position: refs/heads/master@{#10343}
> >
> > TBR=pthatcher@webrtc.org,juberti@webrtc.org,guoweis@webrtc.org
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=
> >
> > Committed: https://crrev.com/a01d44022355796d4fd86d00aae6d3263573b6f1
> > Cr-Commit-Position: refs/heads/master@{#10350}
>
> TBR=pthatcher@webrtc.org,juberti@webrtc.org,tommi@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/e26ce1b7a4644942b239ed788a737200762db3b3
> Cr-Commit-Position: refs/heads/master@{#10379}

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

Review URL: https://codereview.webrtc.org/1413843003

Cr-Commit-Position: refs/heads/master@{#10380}
2015-10-23 01:00:46 +00:00
e26ce1b7a4 Reland of Add experiment on weak ping delay during call set up time (patchset #1 id:1 of https://codereview.webrtc.org/1423443002/ )
Reason for revert:
This should be safe to land now.

Original issue's description:
> Revert of Add experiment on weak ping delay during call set up time (patchset #4 id:60001 of https://codereview.webrtc.org/1411883002/ )
>
> Reason for revert:
> guoweis - Here's the target that's failing:
> https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle_nacl.gyp&l=17
>
> This has unfortunately been causing problems repeatedly for us since libjingle_nacl is maintained separately from libjingle (I don't know the history).
>
> The way this works for Chrome in general is that the FindFullName method is implemented in init_webrtc.cc in the overrides folder in Chrome and that hooks WebRTC up with Chrome's implementation.  I'm not sure if that's the right thing to do for nacl, how webrtc is initialized there etc.  I'll ping the nacl team for some help too offline and include you.  Reverting this change for now.
>
> Original issue's description:
> > Add experiment on weak ping delay during call set up time
> >
> > BUG=
> > R=pthatcher@webrtc.org
> >
> > Committed: https://crrev.com/3bf69b15f4c0c0ca4ab17c237084684a37bb8279
> > Cr-Commit-Position: refs/heads/master@{#10343}
>
> TBR=pthatcher@webrtc.org,juberti@webrtc.org,guoweis@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/a01d44022355796d4fd86d00aae6d3263573b6f1
> Cr-Commit-Position: refs/heads/master@{#10350}

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

Review URL: https://codereview.webrtc.org/1406153005

Cr-Commit-Position: refs/heads/master@{#10379}
2015-10-23 00:49:34 +00:00