summaryrefslogtreecommitdiff
AgeCommit message (Collapse)Author
2013-11-20Added ViE API for getting overuse measure.asapersson@webrtc.org
R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3129005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5141 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20Deliver I420VideoFrames from VideoRender module.pbos@webrtc.org
Performance issue and simplicity, this implementation skips conversion to VideoEngine's frame format and then back again to I420VideoFrame. BUG=2526 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3989004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5140 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20Add possibility to get the last processed RTT from the call stats class (to ↵asapersson@webrtc.org
be used by RTP/RTCP module). R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2383004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5139 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20Rename newapi::Transport::SendRTP()->SendRtp().pbos@webrtc.org
Also fit rampup_tests.cc to use internal::TransportAdapter instead of implementing its own. BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5138 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20Rename RTP-extension constants.pbos@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3969004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5137 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20Rename video streams' start/stop methods.pbos@webrtc.org
{Start,Stop}{Send,Receive}() -> {Start,Stop}{Sending,Receiving}(). BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3609005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5136 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20Rename Call::Create{Receive,Send}Stream().pbos@webrtc.org
Renaming the methods to include Video. Long-term there will hopefully be AudioSendStream/AudioReceiveStreams as well. BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3439004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5135 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-19Add experimental noise suppression dummy API.aluebs@webrtc.org
Add this flag to the voe_cmd_test. R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3879004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5134 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-19Fix DesktopAndCursorComposer to restore frames to the original state.sergeyu@chromium.org
Screen capturers may reuse frame buffers and they expect that the frame content isn't changed by the frame consumer. DesktopAndCursorComposer draws mouse cursor on generated frames and it was releasing the frames with the mouse cursor on them. Fixed it to restore frame content erasing mouse cursor before returning desktop frames. BUG=crbug.com/316297 R=wez@chromium.org Review URL: https://webrtc-codereview.appspot.com/3899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5133 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18Adding back main() to the test. Now it is possible to choose between ACM1 ↵turaj@webrtc.org
and ACM2, furthermore, the test can simulate a channel with packet loss and FEC can be activated. Packet loss pattern is based on channel implementation in Channel{.cc,.h}, which currently is a determenistic pattern with 1 every 3rd packet is discarded. The main() was deleted in r4731. BUG= R=andrew@webrtc.org, minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2370004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5132 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18Rename AutoMute to SuspendBelowMinBitratehenrik.lundin@webrtc.org
Changes all instances throughout the WebRTC stack. BUG=2436 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18Fix test broken with r5128.stefan@webrtc.org
TBR=pbos@webrtc.org BUG=2530 Review URL: https://webrtc-codereview.appspot.com/3979004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5129 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18Hook up audio/video sync to Call.stefan@webrtc.org
Adds an end-to-end audio/video sync test. BUG=2530, 2608 TEST=trybots R=henrika@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3699004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5128 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15Fix breakage after introducing new test.stefan@webrtc.org
TBR=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3899005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5127 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15Improve Call tests for RTX.stefan@webrtc.org
Also does some refactoring to reuse RtpRtcpObserver. BUG=1811 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5126 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15Increment RTP timestamps for padding packetshenrik.lundin@webrtc.org
This CL makes the padding packets get their own RTP timestamps, rather than having the same timestamp as the last sent video packet. The purpose is to solve Issue 2611, where the overuse- detector does not react to padding packets. A test was implemented to verify that the padding packets do get their own timestamps. BUG=2611 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5125 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-14Implement VideoSendStream::SetCodec().pbos@webrtc.org
Removing assertion that SSRC count should be the same as the number of streams in the codec. It makes sense that you don't always use the same number of streams under one call. Dropping resolution due to CPU overuse for instance can require less streams, but the SSRCs should stay allocated so that operations can resume when not overusing any more. This change also means we can get rid of the ugly SendStreamState whose content wasn't defined. Instead we use SetCodec to change resolution etc. on the fly. Should something else have to be replaced on the fly then that functionality simply has to be implemented. BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3499005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5123 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13Disable all vie_auto_tests on Linux for now (take 2)kjellander@webrtc.org
Turns out OS_LINUX is not working in this context (see http://review.webrtc.org/3539005/) WEBRTC_LINUX is the right define to use. TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5119 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13Disable all automated vie_auto_tests on Linux for nowkjellander@webrtc.org
Since the switch from icewm to openbox window manager on Linux in Chrome infra, causes the test to hang when creating Windows. TEST=trybots compile step BUG=chromium:318760 TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3539005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5118 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13Fix for RTX in combination with pacing.stefan@webrtc.org
Retransmissions didn't get sent over RTX when pacing was enabled since the pacer didn't keep track of whether a packet was a retransmit or not. BUG=1811 TEST=trybots R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3779004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5117 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13Inject config when creating channels to override the existing one.turaj@webrtc.org
BUG= R=xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3239004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5116 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-12Reimplementing NetEq4's AudioVectorhenrik.lundin@webrtc.org
The current implementation using std::vector is too slow. This CL introduces a new implementation, using a regular array as data container. In AudioMultiVector::ReadInterleavedFromIndex, a special case for 1 channel was implemented, to further reduce runtime. Finally, AudioMultiVector::Channels was reimplemented. The changes in this CL reduces the runtime of neteq4_speed_test by 33%. BUG=1363 R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5115 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-12Parse next RTCP XR report block after an unsupported block type.asapersson@webrtc.org
R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5114 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-11Reducing opus_test runtime to pass Android testminyue@webrtc.org
BUG=2609 R=solenberg@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5111 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-11MIPS optimizations for AECM audio processing moduleandrew@webrtc.org
R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2279005 Patch from Ljubomir Papuga <lpapuga@mips.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5110 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-11Move audio_processing dependencies to a variable.andrew@webrtc.org
R=aluebs@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3679004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5108 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-11Remove ".." from include_dirs in build/common.pbos@webrtc.org
BUG=1662 TEST=compile on trybots R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2332004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5107 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08Remove unnecessary include_dirs from audio_processing.andrew@webrtc.org
TBR=aluebs TESTED=trybots Review URL: https://webrtc-codereview.appspot.com/3659004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5106 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08Remove unneeded includes from trace_posix.cc.andrew@webrtc.org
TESTED=trybots R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5103 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08Fix for making sure that the packet in order checks are done prior to ↵stefan@webrtc.org
updating the last received packet state. Without this fix all packets are considered out-of-order by the rtp receiver, causing the last received state in the rtp receiver to never get valid. Also makes sure that only valid timestamps and receive times are used for audio/video sync. BUG=2608 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3609004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5102 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08Fix log build error for Chromium builds.henrikg@webrtc.org
This only happens when building in Chromium. Can't roll due to this. ../../third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc: In function 'Window {anonymous}::GetTopLevelWindow(Display*, Window)': ../../third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc:39:7: error: 'LS_INFO' was not declared in this scope ../../third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc:39:7: note: suggested alternative: ../../third_party/webrtc/system_wrappers/interface/logging.h:71:29: note: 'webrtc::LS_INFO' See for example http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20%5Blatest%20WebRTC%2Blibjingle%5D/builds/3039/steps/compile/logs/stdio R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3619004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5100 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08Remove update_resources.py as it's no longer used.kjellander@webrtc.org
After http://review.webrtc.org/2095004/ has been landed for normal WebRTC builds, and https://codereview.chromium.org/62273004/ and https://codereview.chromium.org/60513012/ for our Android APK builds with a Chromium checkout, we should be fine to remove this script. I have verified that the runhooks step on the Android testers is using the download_from_google_storage.py script to pull the resources from Google Storage. BUG=webrtc:2294 TEST=a few trybots passing compile step. R=phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3579004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5099 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-07Replace disabled logging with a restricted logging mode.andrew@webrtc.org
This will enable some low-level webrtc logging in a Chromium build, while limiting the binary size impact. For a Mac Release build, it results in an increase to Chrome.app of 37k and libpeerconnection.so of 25k. For comparison, enabling full logs costs 230k and 218k respectively. BUG=b/11470432 TESTED=voe_cmd_test produces logs of the appropriate severity. R=fischman@webrtc.org, henrikg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3479004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5097 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-06Updated WebRTC version to 3.46elham@webrtc.org
TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3539004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5093 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-06Fix for video_processor_intergration_tests to run in parallel.marpan@webrtc.org
BUG=2601. R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3519004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5091 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-06Update getUserMedia W3C conformance tests.kjellander@webrtc.org
This CL updates these tests to the spec as of http://dev.w3.org/2011/webrtc/editor/archives/20130824/getusermedia.html There are still a lot of functionality that lacks testing. I've put a bunch of TODOs in there but I'm unlikely to get time to implement them all any time soon... TEST=local testing with Chrome Canary. BUG= R=phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5090 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-06Sending status fix for module.asapersson@webrtc.org
R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3339004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5089 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-05Add missing dependencies to .isolate fileskjellander@webrtc.org
Also fix invalid paths in video_engine_tests.isolate. TEST=trybots passing compile step (no .isolate use is deployed on them yet) BUG=chromium:300017 R=pbos@webrtc.org TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3399005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5084 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04Fix broken build on x86 Androidfischman@webrtc.org
BUG=2545 R=fischman@webrtc.org, henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3019004 Patch from Lu Quiang <qiang.lu@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5081 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04Removed unused code.asapersson@webrtc.org
R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3219004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5073 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-03Make video quality analysis unittests print to log instead of stdout.kjellander@webrtc.org
I think it's best to avoid printing these perf numbers since when we turn on perf measurements for Android, it will be for all tests as far as I understand it works today. TEST=trybots passing tools_unittests BUG=none R=phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3109005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5072 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"sheu@chromium.org
This reverts commit f4ca3808bd9ec2293ec205f2f4a7d9739ce1f2df. TBR=niklas.emblom@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/3269004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5071 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31Remove extra copy in VideoCaptureImpl::IncomingFrameI420sheu@chromium.org
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an existing memory buffer without taking ownership. Use this to remove an extra copy in VideoCaptureImpl::IncomingFrameI420. BUG=1128 BUG=chromium:310271 TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux TBR=fischman@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3239005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5070 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"sheu@chromium.org
This reverts commit 99f9743fe39066ba93b41f2b0a417696cbbd06fb. Revert while build breakage is fixed. BUG=None TBR=niklas.emblom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3249004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5069 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31Remove extra copy in VideoCaptureImpl::IncomingFrameI420sheu@chromium.org
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an existing memory buffer without taking ownership. Use this to remove an extra copy in VideoCaptureImpl::IncomingFrameI420. BUG=1128 TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux R=fischman@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5068 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31Address Clag Analyzer issues.turaj@webrtc.org
Following are the issues related to NetEq 4, discovered by Clang Analyzer. https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b44b95.html#EndPath Valid; perhaps unlikely, addressed. https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-6beef6.html#EndPath Valid, addressed. https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2e3883.html#EndPath Valid; Addressed https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-293659.html#EndPath Valid; Addressed. https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b875cd.html#EndPath Valid; Addressed. https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/index.html Not valid; https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-86f2ed.html#EndPath Not Valid; the assert statement will be short-circuited, however I also added a check of nullity of |packet|. https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-3a5669.html#EndPath Not Valid: |energy_input| and |energy_expand| are both non-negative, therefore if-statement condition on line 226 is not satisfied unless |energy_input| >= 1. Therefore |energy_input| cannot be zero after normalization to 14-bits, i.e. operations on lines 228 & 229. https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2f914f.html#EndPath Valid; addressed. https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2332b1.html#EndPath Valid; addressed. https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-de8dea.html#EndPath Not valid; |out_len| is set when Process() is called, however, it makes sense to initialize to zero when declaring |out_len|. https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b671a3.html#EndPath Valid; addressed. BUG= R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2729005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5064 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31Propagate estimated RTT from receivers to rtt observer.asapersson@webrtc.org
BUG=1613 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3119004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5063 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31Video bandwidth not reported correctlysprang@webrtc.org
ViEChannel::GetBandwidthUsage fails to aggregate video_bitrate_sent in the same way as the total, fec and nack. BUG=2579 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5062 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.ccsergeyu@chromium.org
Chromium issue: https://code.google.com/p/chromium/issues/detail?id=310146 BUG=2551 R=wez@chromium.org Review URL: https://webrtc-codereview.appspot.com/2759004 Patch from Daniel Nicoara <dnicoara@chromium.org>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5061 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-30Remove unused make_scoped_ptr which causes an "ambiguous" error with ↵wu@webrtc.org
chromium build. TEST=build R=andrew@webrtc.org, fischman@webrtc.org TBR=andrew Review URL: https://webrtc-codereview.appspot.com/3149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5059 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-10 21:06:51 +0000