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2013-06-26Adding a first simple version of overuse detection, but not hooked up.mflodman@webrtc.org
BUG= R=asapersson@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1717004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4268 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-26Removed ViE file API.mflodman@webrtc.org
R=asapersson@webrtc.org, niklas.enbom@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1723004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4267 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-26Do basic parsing of RTCP headers in PcapFileReader to enable log filtering.solenberg@webrtc.org
BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1697004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4266 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-25Remove unused multi stream bandwidth estimator.solenberg@webrtc.org
BUG= R=mflodman@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1712004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4264 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-25Make sure padding packets are sent.stefan@webrtc.org
BUG=1837 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1717006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4260 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-21mac: Mark kCGLPFAFullScreen as allowed in a 10.6-only function.sergeyu@chromium.org
kCGLPFAFullScreen is marked deprecated starting with 10.6 in the 10.9 SDK, but it's functional on 10.6 and this code only runs on 10.6 and will go away when support for 10.6 is dropped. BUG=webrtc:1958 R=sergeyu@chromium.org Review URL: https://webrtc-codereview.appspot.com/1710004 Patch from Nico Weber <thakis@chromium.org>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4255 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-20Fix memory bot failurehclam@chromium.org
Exit the method with critical setting held. This should make the memory bot happy. TBR=pwestin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1704005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4251 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-20Enqueue packet in pacer if sending failshclam@chromium.org
If a packet cannot be sent while pacer is in use it should be queued. This avoid packet loss due to congestion. BUG=1930 R=pwestin@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1693004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4250 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-20VCM: removing max jitter estimatemikhal@webrtc.org
BUG= 1921 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1690004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4249 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-19Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing.andrew@webrtc.org
* Remove ANDROID_NOT_SUPPORTED from a bunch of echo metrics calls where it actually is supported. * No error to call GetTypingDetectionStatus. * Consolidate typing detection disablement to reduce boilerplate. R=niklas.enbom@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1683004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4247 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-19Fixes some pacer/padding issues found while testing.stefan@webrtc.org
- A bug was introduced in r4234 causing no paced packets to be sent. - Only update the sequence number counter if a padding packet is actually going to be sent, to avoid packet loss. - Have all packets go through the pacer if pacing is enabled to avoid reordering. - Fix race condition on reading capture_time_ms_/timestamp_ in rtp_sender.cc. BUG=1837 TEST=trybots and vie_auto_test --automated R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1682004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4246 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-18Use 3 threads for higher than 720p resolutionsfbarchard@google.com
BUG=1893 TEST=untested R=ajm@google.com, andrew@webrtc.org, dingkai@google.com, marpan@google.com, marpan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1684004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4243 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-18Add a log message to see video delay break downhclam@chromium.org
Shows video delay in terms of: 1. Min playout delay 2. Jitter delay 3. Max decode time 4. Render delay R=mikhal@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1674004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4242 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17Make ScreenCapturerMac work in versions of OSX before Lion.sergeyu@chromium.org
The screen capturer was broken when moving code to webrtc: width and height parameters for glReadPixels were swapped by mistkake. BUG=crbug.com/244102 R=alexeypa@chromium.org Review URL: https://webrtc-codereview.appspot.com/1678005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4239 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17Enable ScreenCapturer unittestssergeyu@chromium.org
previously ScreenCapturer unittests were disabled by mistake TBR=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1679004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4238 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17Use intptr_t to represent window IDs on all platforms.sergeyu@chromium.org
Previously void* was used on windows which makes it harder to work with the IDs in cross-platform code. R=alexeypa@chromium.org Review URL: https://webrtc-codereview.appspot.com/1672004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4237 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17Wire up pacer-based padding.stefan@webrtc.org
This connects the pacer-based padding with the RTP modules, which will generate padding packets roughly according to what the pacer suggests. It will only generate padding packets of maximum size to keep the number off padding packets as small as possible. This also sets a limit of how much padding + media bitrate which the pacer is allowed to "request" from the RTP modules. Padding will for now only be generated by the first sending RTP module. BUG=1837 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1612005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4234 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17Revert r4145 "Revert 4127 "Switch frame list implementation to std::map.""stefan@webrtc.org
TBR=tnakamura@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1678004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4233 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for ↵stefan@webrtc.org
de..."" TBR=tnakamura@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1677004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4232 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-15Fix AV sync issuehclam@chromium.org
r4229 introduced an AV sync issue due to an error. This is a one linear fix and provides the correct current video delay for synchronization. TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1675004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4231 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-14Log current and target AV delay in ViESyncModulehclam@chromium.org
R=mikhal@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1668006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4229 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-14Merge more tests into modules_{unit,integration}tests.kjellander@webrtc.org
A new test target named 'modules_integrationtests' is created and the following test targets were merged into it: * audio_coding_module_test * test_fec * video_coding_integrationtests * vp8_integrationtests A couple of other targets were merged into modules_unittests: * audio_coding_unittests * audioproc_unittest * common_unittests * video_coding_unittests * video_processing_unittests * vp8_unittests I wasn't able to merge audio_decoder_unittests and neteq_unittests due to conflicts with different defines in these tests. Some tests that have special requirements aren't merged into modules_integrationtests yet. I took the opportunity to rename them since the bot configs will need to be update anyway: * audio_device_test_api -> audio_device_integrationtests * video_capture_module_test -> video_capture_integrationtests * video_render_module_test -> video_render_integrationtests Exclude files were added for modules_integrationtests to make sure the memcheck and tsan bots doesn't tests that are too slow (audio_coding_module_test and vp8_integrationtests were previously disabled on those bots). Suppressions for AudioCodingModuleTest needed to be added to get modules_integrationtests to pass memcheck (even if the test is excluded from execution). BUG=1843 TEST=local execution on Linux and trybots (passing except the merged tests of course) R=andrew@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1656004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4228 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-14WebRTCDemo: ensures that using front and back camera work as expected.henrike@webrtc.org
I.e. egress: Real world up is stream up. Ingress: stream up is app up. Local (preview): Real world up is app up. BUG=1763 R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1642004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4227 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-13Fixes linker issue with no op trace.henrike@webrtc.org
BUG=N/A R=henrikg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1669004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4226 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12Risk of division by zero.turaj@webrtc.org
bug=b9338699 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1634004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4223 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12Revert 4211 "Build all java files into jar for each module on An..."fischman@webrtc.org
Reason for revert: behold the meltdown of the "trunk" bots on http://build.chromium.org/p/chromium.webrtc.fyi/waterfall Turns out that include in gyp is fraught with peril: https://code.google.com/p/gyp/wiki/InputFormatReference#Including_Other_Files > Build all java files into jar for each module on Android > > BUG= > R=fischman@webrtc.org, niklas.enbom@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/1636004 > > Patch from Jeremy Mao <yujie.mao@intel.com>. TBR=fischman@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/1660005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4222 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test.kjellander@webrtc.org
Take two of http://review.webrtc.org/1657004/ This time with execution on trybots. BUG=1925 TEST=win,win_rel,mac,mac_rel,linux,linux_rel trybots passing. R=mflodman TBR=mflodman Review URL: https://webrtc-codereview.appspot.com/1658004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4221 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test.kjellander@webrtc.org
Disable on Windows due to failures on bots. BUG=1925 TEST=compile on Linux and Windows. R=mflodman TBR=mflodman Review URL: https://webrtc-codereview.appspot.com/1657004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4220 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12Fix breakage due to test_fec conversion to gtest.kjellander@webrtc.org
In my attempt to commit a subset of http://review.webrtc.org/1647005/ instead of all of it, I forgot to add the gtest dependency to the test_fec.gypi. This CL fixes that. TEST=local compile + win_rel,mac_rel,linux_rel trybots BUG=1916 R=marpan TBR=marpan Review URL: https://webrtc-codereview.appspot.com/1655004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4219 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12Convert test_fec to gtestkjellander@webrtc.org
All tests needs to be gtest tests in order to be executed with the upcoming isolate/swarm framework. TEST=trybots passing BUG=1916 R=andrew@webrtc.org, marpan@google.com Review URL: https://webrtc-codereview.appspot.com/1647005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4218 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test.kjellander@webrtc.org
BUG=1790 TEST=Just local compilation. R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1654004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4217 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12G722_1/G722_1C codecs won't instantiatetina.legrand@webrtc.org
BUG=issue1890 R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1650004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4215 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-11Reorganize test targets in WebRTCkjellander@webrtc.org
This CL will lower the number of test targets in WebRTC by: Add common_audio_unittests and merge the following targets into it (copied from http://review.webrtc.org/1584006): * resampler_unittests * signal_processing_unittests * vad_unittests Merge into modules_unittests: * bitrate_controller_unittests * desktop_capture_unittests * media_file_unittests * remote_bitrate_estimator_unittests * rtp_rtcp_unittests * paced_sender_unittests Merge into test_support_unittests: * channel_transport_unittests channel_transport.gyp was also removed in favor for test.gyp. I had to remove a main method from rtcp_format_remb_unittest.cc since it caused the fileutils.h code to not be able to find the right project root path in ordrer to provide correct paths to test files. Buildbot configuration update will be synced with the commit of this CL. TEST=trybots BUG=1843 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4213 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10Build all java files into jar for each module on Androidfischman@webrtc.org
BUG= R=fischman@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1636004 Patch from Jeremy Mao <yujie.mao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4211 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10Allow the screen capturer to capture oversized cursors and cursors without ↵alexeypa@chromium.org
alpha channel (Windows). Changes in this CL: - CaptureCursor() scans the cursor to verify that it has alpha channel. - The AND mask of the cursor is used to reconstruct transparency if the cursor does not have alpha channel. - CaptureCursor() always outlines the cursor when a "screen reverse" pixel detected. Previously it was only done for black and while cursors. Added desktop_capture_unittest.MouseCursorShapeTest to test the cursor conversion code. BUG=chromium:223147 R=sergeyu@chromium.org Review URL: https://webrtc-codereview.appspot.com/1627004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4210 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10Landing binary cursor image files to be used in a follow up CL.alexeypa@chromium.org
See https://webrtc-codereview.appspot.com/1627004/ for more details. TBR since that CL has been reviewed and LGTMed. TBR=sergeyu@chromium.org BUG=chromium:223147 Review URL: https://webrtc-codereview.appspot.com/1647004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4209 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10Updated WebRTC version to 3.33elham@webrtc.org
R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1645004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4204 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10Making no NACK mode work again in VideoEngine.mflodman@webrtc.org
BUG=1910 TEST=ViE autotest loopback with no protection and some percent packet loss R=mikhal@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1631004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4203 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10RW lock access to ssrc maps in VideoCall.pbos@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1640004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4202 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10Add back the WEBRTC_DIRECT_TRACE flag.solenberg@webrtc.org
BUG= R=andresp@webrtc.org, andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1596004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4201 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout()braveyao@webrtc.org
BUG=1891 Test=ManualTest R=fischman@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1622004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4200 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-08Revert some variables to uint32_t to fix compile errors on Mac gcc.andrew@webrtc.org
TBR=xians Review URL: https://webrtc-codereview.appspot.com/1633004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4199 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-07Allow audio devices with up to 64 channels on Mac.andrew@webrtc.org
Does not increase memory requirements. Adds an additional check to ensure configurations requiring more memory per IO block than the input ring buffer contains are rejected. BUG=1904 TESTED=Using Soundflower (64 channels) at 48 kHz as input gives good quality. Selecting a higher sample rate (96 kHz), which would otherwise give choppy audio, instead results in an error. R=henrika@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1628004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4198 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-07Fixed Rtp/Rtcp testspwestin@webrtc.org
R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1627005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4196 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-07Fix relative path to .gitignore and other minor changes.andrew@webrtc.org
R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1624005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4195 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-07Removing functionality for inserting pre-encoded frames instead of rawmflodman@webrtc.org
video frames. The functionality hasn't been used for a long time and should be done properly if used in the future. This is a pre-step for implementing CPU overload control. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1630004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4194 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-07Add script for appending entries to .gitignore.andrew@webrtc.org
TBR=kjellander Review URL: https://webrtc-codereview.appspot.com/1629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4193 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-07Fix size_t to int conversion error on Win64.andrew@webrtc.org
TBR=pwestin Review URL: https://webrtc-codereview.appspot.com/1626005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4192 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-06Remove fake screen capturer because it's not used anywhere.sergeyu@chromium.org
R=alexeypa@chromium.org Review URL: https://webrtc-codereview.appspot.com/1625004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4191 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-06Fix for STL vector function data not available.pwestin@webrtc.org
R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1626004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4190 4adac7df-926f-26a2-2b94-8c16560cd09d
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