6589 Commits

Author SHA1 Message Date
deadbeef
be0c96fd00 Add ice_candidate_pool_size to Obj-C and Java RTCConfiguration.
Review-Url: https://codereview.webrtc.org/1986073004
Cr-Commit-Position: refs/heads/master@{#12801}
2016-05-18 23:20:18 +00:00
honghaiz
b711f10d96 Revert of FrameBuffer for the new jitter buffer. (patchset #9 id:160001 of https://codereview.webrtc.org/1969403007/ )
Reason for revert:
Two tests added by this CL failed in Win DrMemory Full:
 TestFrameBuffer2.OneLayerStreamReordered - TestFrameBuffer2.WaitForFrame

See the link here:
https://build.chromium.org/p/client.webrtc/waterfall?builder=Win%20DrMemory%20Full

Original issue's description:
> FrameBuffer for the new jitter buffer.
>
> BUG=webrtc:5514
> R=danilchap@webrtc.org, mflodman@webrtc.org
>
> Committed: https://crrev.com/a376e70cf9d0df3c35d53533b454da542661775b
> Cr-Commit-Position: refs/heads/master@{#12798}

TBR=mflodman@webrtc.org,danilchap@webrtc.org,philipel@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/1991513004
Cr-Commit-Position: refs/heads/master@{#12800}
2016-05-18 22:52:36 +00:00
philipel
a376e70cf9 FrameBuffer for the new jitter buffer.
BUG=webrtc:5514
R=danilchap@webrtc.org, mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1969403007 .

Cr-Commit-Position: refs/heads/master@{#12798}
2016-05-18 16:10:14 +00:00
henrik.lundin
a89ab965f2 Enable muted state by default in VoE
This change turns muted state on by default in VoiceEngine, but not
for NetEq or AudioCodingModule when used stand-alone.

The expected effect is that voice channels that have not received any
packets for some time should reduce their CPU usage. This should have
a noticeable effect on endpoints with many incoming streams, but where
only a few have packets incoming at any given time (i.e., where an
intermediate server filters out the majority of the streams).

BUG=webrtc:5606
NOTRY=True

Review-Url: https://codereview.webrtc.org/1987143003
Cr-Commit-Position: refs/heads/master@{#12797}
2016-05-18 15:52:52 +00:00
Stefan Holmer
a3c2c3e606 Remove unneccessary log in PacedSender.
R=perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1990883002 .

Cr-Commit-Position: refs/heads/master@{#12796}
2016-05-18 14:06:02 +00:00
henrik.lundin
83e8286c6b AEC: Add UMA logging of buffer re-alignment
This change adds a UMA log that will be written to when a non-zero delay
correction is done in the AEC. The number of elements moved (positive or
negative) will be logged to
"WebRTC.Audio.AecDelayAdjustmentAgnosticValue" or
"WebRTC.Audio.AecDelayAdjustmentSystemValue", depending on whether
delay-agnostic AEC is used or not, respectively.

BUG=webrtc:5903

Review-Url: https://codereview.webrtc.org/1991723002
Cr-Commit-Position: refs/heads/master@{#12795}
2016-05-18 12:43:05 +00:00
henrik.lundin
42dda50860 Propagate muted info from VoE Channel to AudioConferenceMixer
Required updating of a few related classes and tests.

BUG=webrtc:5609
NOTRY=True

Review-Url: https://codereview.webrtc.org/1986093002
Cr-Commit-Position: refs/heads/master@{#12794}
2016-05-18 12:36:07 +00:00
kjellander
84f8df71af Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
Reason for revert:
This breaks our Chromium WebRTC FYI bots:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/6173

/mnt/data/b/build/slave/Linux_Builder/build/src/buildtools/linux64/gn gen //out/Release --check
  -> returned 1
ERROR at //content/renderer/media/rtc_video_decoder.cc:24:11: Can't include this header from here.
#include "third_party/webrtc/video_frame.h"
          ^-------------------------------
The target:
  //content/renderer:renderer
is including a file from the target:
  //third_party/webrtc:webrtc_common

The //content/renderer:renderer target should probably be updated to depend on   //third_party/webrtc:webrtc_common before relanding this.

Original issue's description:
> Add missing headers and fix some missing dependencies
>
> This is the first CL in a series of major cleanup and dependency
> corrections needed in order to satisfy 'gn check'.
>
> BUG=webrtc:4243, webrtc:5589
> NOTRY=True
>
> Committed: https://crrev.com/7bb6e75723eb64af079446cc6e3ff08c74fe02e4
> Cr-Commit-Position: refs/heads/master@{#12790}

TBR=pbos@webrtc.org,henrika@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4243, webrtc:5589

Review-Url: https://codereview.webrtc.org/1989823002
Cr-Commit-Position: refs/heads/master@{#12793}
2016-05-18 12:00:55 +00:00
kjellander
7bb6e75723 Add missing headers and fix some missing dependencies
This is the first CL in a series of major cleanup and dependency
corrections needed in order to satisfy 'gn check'.

BUG=webrtc:4243, webrtc:5589
NOTRY=True

Review-Url: https://codereview.webrtc.org/1990593002
Cr-Commit-Position: refs/heads/master@{#12790}
2016-05-18 11:06:22 +00:00
sakal
299ccdee0c Direct IP connect functionality for AppRTC Android demo.
This allows connecting between clients without using external servers, which is useful to OEMs if they are working in a network without internet connection. Implementation uses custom AppRTCClient that replaces WebSocketRTCClient if roomId looks like an IP. Instead of a web socket, this class uses direct TCP connection between peers as a signaling channel.

Review-Url: https://codereview.webrtc.org/1963053002
Cr-Commit-Position: refs/heads/master@{#12789}
2016-05-18 10:36:50 +00:00
henrik.lundin
5a216d0489 Add muted parameter to audio_frame_manipulator methods
BUG=webrtc:5609
NOTRY=True

Review-Url: https://codereview.webrtc.org/1983023002
Cr-Commit-Position: refs/heads/master@{#12788}
2016-05-18 09:34:24 +00:00
kwiberg
89f237cedc Fix UBSan errors (left shift of negative value, left shift overflows int)
BUG=chromium:603498

Review-Url: https://codereview.webrtc.org/1979973003
Cr-Commit-Position: refs/heads/master@{#12787}
2016-05-18 09:20:35 +00:00
henrik.lundin
60200d1094 Add FrameAndMuteInfo to AudioConferenceMixer
BUG=webrtc:5609

Review-Url: https://codereview.webrtc.org/1980343002
Cr-Commit-Position: refs/heads/master@{#12786}
2016-05-18 08:48:46 +00:00
tkchin
3c5a294b60 Fix Info.plist path in build_ios_libs.sh
NOTRY=True
BUG=

Review-Url: https://codereview.webrtc.org/1988003002
Cr-Commit-Position: refs/heads/master@{#12784}
2016-05-18 06:08:32 +00:00
Sergey Ulanov
837dde9f1d Remove DesktopFrame::shape().
Frame shape is no longer used and can be removed.

R=jamiewalch@chromium.org

Review URL: https://codereview.webrtc.org/1986963002 .

Cr-Commit-Position: refs/heads/master@{#12783}
2016-05-18 05:03:47 +00:00
skvlad
6c87a67b63 Do not create a temporary transport channel when using max-bundle
With this change, when max-bundle and rtcp-mux are both enabled, we no
longer create and destroy a temporary transport channel when a media
channel gets added. Instead, the media channel uses the correct bundled
transport channel from the start.

This fixes a bug where adding a media type would cause the ICE state to
briefly become Disconnected and then immediately recover. The temporary
channel was created in a non-writable state, which caused the
TransportController to declare the ICE state to be Disconnected (as not
all transport channels were writable). Right after creation, the
temporary channel was then destroyed and the ICE state went back to the
correct one.

BUG=webrtc:5856

Review-Url: https://codereview.webrtc.org/1972493002
Cr-Commit-Position: refs/heads/master@{#12781}
2016-05-18 00:49:58 +00:00
henrik.lundin
3a0a0f4b0d AudioConfMixer: Add muted variable to ParticipantFramePair
Renaming the ParticipantFramePair to ParticipantFrameStruct. The muted
variable is not yet used.

BUG=webrtc:5609

Review-Url: https://codereview.webrtc.org/1981243002
Cr-Commit-Position: refs/heads/master@{#12780}
2016-05-18 00:16:05 +00:00
henrik.lundin
d4ccb00b9e Propagate muted parameter to VoE::Channel
Deleted the temporary ACM method without the muted parameter, and had
to modify several tests for this. The muted parameter is not yet propagated to the AudioConferenceMixer; this is the next step.

BUG=webrtc:5609
TBR=perkj@webrtc.org

Review-Url: https://codereview.webrtc.org/1985743002
Cr-Commit-Position: refs/heads/master@{#12779}
2016-05-17 19:22:03 +00:00
pasko
e305d956c0 Remove runtime NEON detection
Chrome does not detect NEON instruction set at runtime in WebRTC code starting
with M50, which is now in Stable. Remove support for runtime detection for
simplicity.

The only remaining piece of Chrome that will continue to depend on runtime
detection is /net, where devices with _broken_ neon support are also detected,
and it is not configurable via GYP/GN.

BUG=522035
NOPRESUBMIT=true

Review-Url: https://codereview.webrtc.org/1955413003
Cr-Commit-Position: refs/heads/master@{#12778}
2016-05-17 17:56:48 +00:00
magjed
de8739c120 Disable libyuv jpeg support on Android
MJPEG capture is not used on Android. Therefore, disable jpeg support to
reduce libjingle_peerconnection_so file size by removing dependency to
libjpeg_turbo.

Also, remove unused build_libjpeg and rtc_build_libjpeg variables.

Review-Url: https://codereview.webrtc.org/1978243002
Cr-Commit-Position: refs/heads/master@{#12777}
2016-05-17 15:39:15 +00:00
Stefan Holmer
2b1f651d15 Potential fix for rtx/red issue where red is removed only from the remote description.
BUG=5675
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1964473002 .

Cr-Commit-Position: refs/heads/master@{#12776}
2016-05-17 14:33:41 +00:00
kwiberg
9b2228fdfc Fix UBSan errors (left shift of negative value)
BUG=chromium:603501

Review-Url: https://codereview.webrtc.org/1988723002
Cr-Commit-Position: refs/heads/master@{#12775}
2016-05-17 13:40:48 +00:00
nisse
c9c142f170 Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1983583002/ )
Reason for revert:
Should work after cl https://codereview.webrtc.org/1985693002/ is landed, which initializes the frames used by FakeWebRtcVideoCaptureModule. So intend to reland after that, with no changes.

Original issue's description:
> Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #2 id:290001 of https://codereview.webrtc.org/1963413004/ )
>
> Reason for revert:
> Speculative revert to see if failures on the DrMemory bot are related to this cl.  See e.g. here:
> https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/4243
>
> UNINITIALIZED READ: reading 0x04980040-0x04980060 32 byte(s) within 0x04980040-0x04980060
> # 0 CopyRow_AVX
> # 1 CopyPlane
> # 2 I420Copy
> # 3 webrtc::ExtractBuffer
> # 4 cricket::WebRtcVideoCapturer::SignalFrameCapturedOnStartThread
> # 5 cricket::WebRtcVideoCapturer::OnIncomingCapturedFrame
> # 6 FakeWebRtcVideoCaptureModule::SendFrame
> # 7 WebRtcVideoCapturerTest_TestCaptureVcm_Test::TestBody
> # 8 testing::internal::HandleSehExceptionsInMethodIfSupported<>
>
> Original issue's description:
> > Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1935443002/ )
> >
> > Reason for revert:
> > I plan to reland this change in a week or two, after downstream users are updated.
> >
> > Original issue's description:
> > > Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
> > >
> > > Reason for revert:
> > > Breaks chrome FYI bots.
> > >
> > > Original issue's description:
> > > > Delete webrtc::VideoFrame methods buffer and stride.
> > > >
> > > > To make the HasOneRef/IsMutable hack work, also had to change the
> > > > video_frame_buffer method to return a const ref to a scoped_ref_ptr,
> > > > to not imply an AddRef.
> > > >
> > > > BUG=webrtc:5682
> > >
> > > TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:5682
> > >
> > > Committed: https://crrev.com/5b3c443d301f2c2f18dac5b02652c08b91ea3828
> > > Cr-Commit-Position: refs/heads/master@{#12558}
> >
> > TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/d0dc66e0ea30c8614001e425a4ae0aa7dd56c2a7
> > Cr-Commit-Position: refs/heads/master@{#12721}
>
> TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/d49c30cd2fe442f2b5b4ecec8d5cbaa430464725
> Cr-Commit-Position: refs/heads/master@{#12745}

TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org,tommi@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/1979193003
Cr-Commit-Position: refs/heads/master@{#12773}
2016-05-17 11:05:51 +00:00
phoglund
ff274394fe Separate building and enabling libevent.
We're now ready https://codereview.webrtc.org/1984503002/ downstream,
so make sure we can enable libevent but still choose which libevent
implementation to use. This follows the common pattern where an enable_
flag controls whether we should use the feature at all, whereas build_
controls if we should use the dependency from our DEPS file or
something else.

NOTRY=True

Review-Url: https://codereview.webrtc.org/1980003002
Cr-Commit-Position: refs/heads/master@{#12772}
2016-05-17 10:44:36 +00:00
Henrik Boström
d98f6e000a Fixed typo. KT_DEFAULT different based on WEBRTC_CHROMIUM_BUILD
instead of WEBRTC_BUILD_CHROMIUM.

BUG=chromium:611698, webrtc:5795
R=perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1986053005 .

Cr-Commit-Position: refs/heads/master@{#12771}
2016-05-17 10:36:19 +00:00
hbos
a73ca5668e Polishing code to handle certificate generation failure in .mm files.
This is a follow-up to https://codereview.webrtc.org/1965313002/ which
was TBR-landed.

Minor code clean-up/corrections:

Property nativeConfiguration -> - method createNativeConfiguration.
RTCLogWarning -> RTCLogError.
setConfiguration returning NO instead of false.
initWithFactory returning nil instead of nullptr.
Braces around ifs.

Review-Url: https://codereview.webrtc.org/1978233002
Cr-Commit-Position: refs/heads/master@{#12770}
2016-05-17 10:29:07 +00:00
sakal
ee3732622c JUnit test framework for AppRTC Android demo.
This allows creating tests for AppRTC Android demo that will be run on
the host machine instead of a device. These tests can mock Android APIs
through Robolectric. Because the tests are run on the host machine,
they run much faster.

BUG=webrtc:5896
NOTRY=True

Review-Url: https://codereview.webrtc.org/1985663002
Cr-Commit-Position: refs/heads/master@{#12769}
2016-05-17 10:22:38 +00:00
Danil Chapovalov
2ccfbdf011 Undeprecate CreatePeerConnectionFactory which do not use network thread.
function was deprecated in https://codereview.webrtc.org/1968393002/
in favor of splitting network thread from worker thread.

TBR=pthatcher1
BUG=webrtc:5645

Review URL: https://codereview.webrtc.org/1981273002 .

Cr-Commit-Position: refs/heads/master@{#12768}
2016-05-17 09:38:48 +00:00
danilchap
e9021a3601 Propogate network-worker thread split to api
BUG=webrtc:5645

Review-Url: https://codereview.webrtc.org/1968393002
Cr-Commit-Position: refs/heads/master@{#12767}
2016-05-17 08:52:06 +00:00
nisse
744494f451 Make the FakeWebRtcVideoCaptureModule class initialize frame data.
Needed to avoid DrMemory warnings, if the frame is passed to libyuv
AVX assembly functions.

BUG=libyuv:377

Review-Url: https://codereview.webrtc.org/1985693002
Cr-Commit-Position: refs/heads/master@{#12765}
2016-05-17 06:51:11 +00:00
Alejandro Luebs
c9b0c26e0c Surface the IntelligibilityEnhancer on MediaConstraints
R=henrika@webrtc.org, peah@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1952123003 .

Cr-Commit-Position: refs/heads/master@{#12763}
2016-05-16 22:32:45 +00:00
Taylor Brandstetter
2abe427417 Revert of Increase the stun ping interval. (patchset #5 id:80001 of https://codereview.webrtc.org/1944003002/ )
Reason for revert:
This will take longer time for the RTT to converge.
Need to update the RTT calculation algorithm if doing this.

Original issue's description:
> Increase the stun ping interval.
>
> Writable connections are pinged at a slower rate.
> The function IsPingable will filter out the writable connections.
> The interval for slower ping rate by default is WRITABLE_CONNECTION_PING_INTERVAL(2500ms) and can be set with the configuration.
>
> BUG=webrtc:1161
>
> Committed: https://crrev.com/8f7a5aad55a64f0d81b6436a22ffbdfcdcde91e0
> Cr-Commit-Position: refs/heads/master@{#12736}

TBR=honghaiz@webrtc.org,pthatcher@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:1161

Review URL: https://codereview.webrtc.org/1982713002 .

Cr-Commit-Position: refs/heads/master@{#12762}
2016-05-16 21:12:22 +00:00
Taylor Brandstetter
db0cd9e774 Adding getParameters/setParameters APIs to RtpReceiver.
This is similar to how a "receive" method is used to apply
RtpParameters to an RtpReceiver in ORTC. Currently, SetParameters
doesn't allow changing the parameters, so the main use of the API is
to retrieve the set of configured codecs. But other uses will likely
be made possible in the future.

R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1917193008 .

Cr-Commit-Position: refs/heads/master@{#12761}
2016-05-16 18:40:38 +00:00
aleloi
8bce67b745 Added UMA logging for audio codec usage. A histogram statistic named "WebRTC.Audio.Encoder.CodecType" has been created.
Review-Url: https://codereview.webrtc.org/1967503002
Cr-Commit-Position: refs/heads/master@{#12760}
2016-05-16 14:34:31 +00:00
Sami Kalliomäki
970567cab2 Fixes a bug where AppRTC Android Demo crashes with empty roomId.
BUG=
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1975743002 .

Cr-Commit-Position: refs/heads/master@{#12759}
2016-05-16 14:24:23 +00:00
Sami Kalliomäki
57f95dcf9d New UI for AppRTC Android Demo that is easier to use and better follows
Android design guidelines.

BUG=
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1970783002 .

Cr-Commit-Position: refs/heads/master@{#12758}
2016-05-16 14:21:45 +00:00
hbos
8ae8ab4e23 Makes ECDSA the default certificate to use (generated if no other certificates
are specified when constructing a peer connection, at:
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/webrtc/api/webrtcsessiondescriptionfactory.cc&q=webrtcsessiondescriptionfactory&sq=package:chromium&type=cs&l=191).

This does not affect WEBRTC_BUILD_CHROMIUM builds whose ECDSA launch is handled
separately: https://crbug.com/601850 (req. @chromium acc).

BUG=chromium:611698, webrtc:5795

Review-Url: https://codereview.webrtc.org/1972383002
Cr-Commit-Position: refs/heads/master@{#12756}
2016-05-16 09:45:45 +00:00
Tommi
0bcbbd345e Fix component build in chrome after recent TaskQueue cl
TBR=phoglund

Review URL: https://codereview.webrtc.org/1985533002 .

Cr-Commit-Position: refs/heads/master@{#12753}
2016-05-15 11:42:49 +00:00
kwiberg
fd8be3468a Remove webrtc/base/scoped_ptr.h
This is a re-land of https://codereview.webrtc.org/1942823002

TBR=tommi@webrtc.org
BUG=webrtc:5520

Review-Url: https://codereview.webrtc.org/1966423002
Cr-Commit-Position: refs/heads/master@{#12750}
2016-05-15 02:44:18 +00:00
tommi
c06b133b29 Reland of New task queueing primitive for async tasks: TaskQueue.
New task queueing primitive for async tasks: TaskQueue.
TaskQueue is a new way to asynchronously execute tasks sequentially
in a thread safe manner with minimal locking.  The implementation
uses OS supported APIs to do this that are compatible with async IO
notifications from things like sockets and files.
This class is a part of rtc_base_approved, so can be used by both
the webrtc and libjingle parts of the WebRTC library.  Moving forward,
we can replace rtc::Thread and webrtc::ProcessThread with this implementation.
NOTE: It should not be assumed that all tasks that execute on a TaskQueue,
run on the same thread.  E.g. on Mac and iOS, we use GCD dispatch queues
which means that tasks might execute on different threads depending on
what's the most efficient thing to do.

TBR=perkj@webrtc.org,phoglund@webrtc.org

Review-Url: https://codereview.webrtc.org/1984503002
Cr-Commit-Position: refs/heads/master@{#12749}
2016-05-14 18:31:49 +00:00
tommi
5ce1a2a629 Reland of Allow the localhost IP address even if it does not match the tcp port address (patchset #1 id:1 of https://codereview.webrtc.org/1979463003/ )
Reason for revert:
Relanding this change since the revert didn't make a difference.

Original issue's description:
> Revert of Allow the localhost IP address even if it does not match the tcp port address (patchset #4 id:120001 of https://codereview.webrtc.org/1914803002/ )
>
> Reason for revert:
> Speculatively reverting due to failures on the memcheck bot (and possibly other bots):
>
> https://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/5910/steps/video_engine_tests/logs/EndToEndTest.SendsAndReceivesH264
>
> Original issue's description:
> > This fixes an issue similar to
> > https://bugs.chromium.org/p/webrtc/issues/detail?id=3927
> > where the localhost IP does not match the turn port address.
> > The issue here is in the TCP port.
> >
> > BUG=
> > R=pthatcher@webrtc.org
> >
> > Committed: https://crrev.com/6705012904e6cbbefce6fbce0a3f615b7aeafd8f
> > Cr-Commit-Position: refs/heads/master@{#12707}
>
> TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,honghaiz@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/1cbf0a73eb4b475e8beb878ea3a4d650191f0c08
> Cr-Commit-Position: refs/heads/master@{#12728}

TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,honghaiz@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review-Url: https://codereview.webrtc.org/1979073002
Cr-Commit-Position: refs/heads/master@{#12746}
2016-05-14 10:19:39 +00:00
tommi
d49c30cd2f Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #2 id:290001 of https://codereview.webrtc.org/1963413004/ )
Reason for revert:
Speculative revert to see if failures on the DrMemory bot are related to this cl.  See e.g. here:
https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/4243

UNINITIALIZED READ: reading 0x04980040-0x04980060 32 byte(s) within 0x04980040-0x04980060
# 0 CopyRow_AVX
# 1 CopyPlane
# 2 I420Copy
# 3 webrtc::ExtractBuffer
# 4 cricket::WebRtcVideoCapturer::SignalFrameCapturedOnStartThread
# 5 cricket::WebRtcVideoCapturer::OnIncomingCapturedFrame
# 6 FakeWebRtcVideoCaptureModule::SendFrame
# 7 WebRtcVideoCapturerTest_TestCaptureVcm_Test::TestBody
# 8 testing::internal::HandleSehExceptionsInMethodIfSupported<>

Original issue's description:
> Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1935443002/ )
>
> Reason for revert:
> I plan to reland this change in a week or two, after downstream users are updated.
>
> Original issue's description:
> > Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
> >
> > Reason for revert:
> > Breaks chrome FYI bots.
> >
> > Original issue's description:
> > > Delete webrtc::VideoFrame methods buffer and stride.
> > >
> > > To make the HasOneRef/IsMutable hack work, also had to change the
> > > video_frame_buffer method to return a const ref to a scoped_ref_ptr,
> > > to not imply an AddRef.
> > >
> > > BUG=webrtc:5682
> >
> > TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/5b3c443d301f2c2f18dac5b02652c08b91ea3828
> > Cr-Commit-Position: refs/heads/master@{#12558}
>
> TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5682
>
> Committed: https://crrev.com/d0dc66e0ea30c8614001e425a4ae0aa7dd56c2a7
> Cr-Commit-Position: refs/heads/master@{#12721}

TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/1983583002
Cr-Commit-Position: refs/heads/master@{#12745}
2016-05-14 10:18:13 +00:00
pbos
a1025073a9 Use generic CPU-overuse thresholds for iOS.
Prevents overly-aggressive CPU adaptation on iOS where input frames get
downscaled after ~40% CPU usage.

BUG=
R=tommi@webrtc.org

Review-Url: https://codereview.webrtc.org/1973293003
Cr-Commit-Position: refs/heads/master@{#12744}
2016-05-14 10:04:25 +00:00
perkj
fea93099f0 This reland https://codereview.webrtc.org/1932683002/.
Remove ViEEncoder::SetNetworkStatus.

Original cl description:
This cl removed ViEEncoder::SetNetworkStatus. Instead the PacedSender will report that frames can not be sent when the network is down and the BitrateController will report an estimated available bandwidth of 0 bps.

Patchset #1 is a pure reland.
Patchset #2 change the bitrate allocator to always return an initial bitrate >0 regardless of the estimates. The observer will be notified though if the network is down.

BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/1972183004
Cr-Commit-Position: refs/heads/master@{#12743}
2016-05-14 07:58:54 +00:00
Peter Boström
1299615838 Make sure WebRTC works without libvpx VP9 support.
Wires up existing libvpx_build_vp9==0 GYP flag into WebRTC and makes VP9
optional. Change is GYP only for now since libvpx's GN files build VP9
unconditionally.

BUG=webrtc:5884
R=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1970343002 .

Cr-Commit-Position: refs/heads/master@{#12741}
2016-05-14 00:03:28 +00:00
Danil Chapovalov
dae07bae82 Fix BaseChannel destructor when network thread differ from worker thread
BUG=webrtc:5645
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1970223002 .

Cr-Commit-Position: refs/heads/master@{#12740}
2016-05-13 23:44:02 +00:00
zijiehe
99f8cd0ae2 CURSORINFO.flags should be checked before capturing its bitmap
BUG=566178, 428886

Review-Url: https://codereview.webrtc.org/1959863002
Cr-Commit-Position: refs/heads/master@{#12739}
2016-05-13 23:41:59 +00:00
tommi
3f90087ce8 Revert of New task queueing primitive for async tasks: TaskQueue. (patchset #8 id:330001 of https://codereview.webrtc.org/1927133004/ )
Reason for revert:
sigh.  Have to revert again as there seems to have have been some change made for pnacl and CrOS.

Original issue's description:
> Reland of New task queueing primitive for async tasks: TaskQueue. (patchset #1 id:1 of https://codereview.webrtc.org/1935483002/ )
>
> New task queueing primitive for async tasks: TaskQueue.
> TaskQueue is a new way to asynchronously execute tasks sequentially
> in a thread safe manner with minimal locking.  The implementation
> uses OS supported APIs to do this that are compatible with async IO
> notifications from things like sockets and files.
>
> This class is a part of rtc_base_approved, so can be used by both
> the webrtc and libjingle parts of the WebRTC library.  Moving forward,
> we can replace rtc::Thread and webrtc::ProcessThread with this implementation.
>
> NOTE: It should not be assumed that all tasks that execute on a TaskQueue,
> run on the same thread.  E.g. on Mac and iOS, we use GCD dispatch queues
> which means that tasks might execute on different threads depending on
> what's the most efficient thing to do.
>
> TBR=perkj@webrtc.org
>
> Committed: https://crrev.com/65d1f2aba216d077c6d22488f03e56984aef1c68
> Cr-Commit-Position: refs/heads/master@{#12737}

TBR=perkj@webrtc.org,phoglund@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/1981573002
Cr-Commit-Position: refs/heads/master@{#12738}
2016-05-13 21:33:39 +00:00
tommi
65d1f2aba2 Reland of New task queueing primitive for async tasks: TaskQueue. (patchset #1 id:1 of https://codereview.webrtc.org/1935483002/ )
New task queueing primitive for async tasks: TaskQueue.
TaskQueue is a new way to asynchronously execute tasks sequentially
in a thread safe manner with minimal locking.  The implementation
uses OS supported APIs to do this that are compatible with async IO
notifications from things like sockets and files.

This class is a part of rtc_base_approved, so can be used by both
the webrtc and libjingle parts of the WebRTC library.  Moving forward,
we can replace rtc::Thread and webrtc::ProcessThread with this implementation.

NOTE: It should not be assumed that all tasks that execute on a TaskQueue,
run on the same thread.  E.g. on Mac and iOS, we use GCD dispatch queues
which means that tasks might execute on different threads depending on
what's the most efficient thing to do.

TBR=perkj@webrtc.org

Review-Url: https://codereview.webrtc.org/1927133004
Cr-Commit-Position: refs/heads/master@{#12737}
2016-05-13 20:05:05 +00:00
zhihuang
8f7a5aad55 Increase the stun ping interval.
Writable connections are pinged at a slower rate.
The function IsPingable will filter out the writable connections.
The interval for slower ping rate by default is WRITABLE_CONNECTION_PING_INTERVAL(2500ms) and can be set with the configuration.

BUG=webrtc:1161

Review-Url: https://codereview.webrtc.org/1944003002
Cr-Commit-Position: refs/heads/master@{#12736}
2016-05-13 19:23:07 +00:00