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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
2013-06-25Remove 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
2013-06-20Enqueue 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
2013-06-19Fixes 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
2013-06-17Wire 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
2013-06-15Fix 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
2013-06-14WebRTCDemo: 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
2013-06-12Revert 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
2013-06-12Disable 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
2013-06-11Reorganize 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
2013-06-10Updated 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
2013-06-07Removing 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
2013-06-05Fix init list for VideoSendStream::Config::Rtp.pbos@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1616004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4183 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-05Stats+Config moved into VideoSend/ReceiveStreams.pbos@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1561006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4182 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04Update the remote bitrate estimator before passing the packet to the RTP module.stefan@webrtc.org
This solves the problem of reconstructed packets biasing the bandwidth estimate. TEST=vie_auto_test --automated, trybots R=mflodman@webrtc.org, solenberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1594005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4171 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04Remove XvRenderer.pbos@webrtc.org
One test renderer per platform is sufficient, multiple code paths are bad. BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1612004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4170 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04Add support for padding in pacer.stefan@webrtc.org
This improves pacer-based padding by making sure it limits padding according to: - Never pad more than 800 kbps. - Padding + media should not go above a given target bitrate. Also adds appropriate unittests to make sure we reach the given targets. BUG=1837 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1582005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4168 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-03Setting SSRC in vie_loopback_testmikhal@webrtc.org
BUG=1822 R=pwestin@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1603004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4159 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29Use int for FPS instead of size_t.pbos@webrtc.org
BUG= TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1578005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4136 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29Correctly set SSRCs for extra send RTP modules.stefan@webrtc.org
Fixes a regression introduced in r4096. BUG=1845 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1585004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4134 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29Remove assert for aborting FrameGeneratorCapturer.pbos@webrtc.org
BUG= TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1586004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4133 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29Fake VideoCapturer based on FrameGeneratorpbos@webrtc.org
BUG=1793 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1539004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4132 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29Fix a return value mismatch introduced in r4129.stefan@webrtc.org
TBR=mflodman@webrtc.org TEST=vie_auto_test, trybots Review URL: https://webrtc-codereview.appspot.com/1584005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4131 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29Breaking out RTP header parsing from the RTP module.stefan@webrtc.org
This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video. Moving bandwidth estimation before the RTP module is also required for RTX. TEST=vie_auto_test, voe_auto_test, trybots. BUG=1811 R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1545004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29Break video_engine/new_include/common.h into smaller parts.pbos@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1571005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4128 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp.andrew@webrtc.org
R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1574004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4126 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-28Updated WebRTC version to 3.32elham@webrtc.org
TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1576004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4122 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-28Don't return an estimated receive BW for channels not receiving video.mflodman@webrtc.org
BUG=1834 TEST=ViE RTP autotest R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1572004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4121 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-28Include gflags with "gflags/gflags.h" instead of <>pbos@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1551004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4120 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-28Improve vie_autotest_rtp_rtcp by reenabling important tests and reducing ↵stefan@webrtc.org
flakiness. R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1567004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4118 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-28Default constructors for new VideoEngine structs.pbos@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1543004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4115 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-28Remove libvpx_intrinsics_sse4_1.a in Android.mk since this target is no ↵fischman@webrtc.org
longer generated in libvpx R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1569004 Patch from Jeremy Mao <yujie.mao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4114 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-27- Created RemoteBitrateEstimator wrapper for use internally in (ViE) ↵solenberg@webrtc.org
ChannelGroup. - Changed implementation of SetReceiveAbsoluteSendTimeStatus API so the RBE instance is changed when at least one channel in a group has the extension enabled. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1553005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4113 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-27Adding Mac test renderer, some test refactoring and made cpplint pass.mflodman@webrtc.org
BUG=1667 TEST=Rendered video in Mac loopback test. R=pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1554004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4112 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-27Disabling a flaky expectation in vie_autotest_rtp_rtcp.cc.stefan@webrtc.org
TBR=phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1566004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4106 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-24Make sure GlxRenderer frees its resources.pbos@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1544004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4098 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-23Adds integration test for RTX and fixes bugs found.stefan@webrtc.org
BUG=1811 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4096 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-23Fix regression where retransmission bitrate is no longer estimated.stefan@webrtc.org
BUG=1813 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1530004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4095 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-23CreateEmptyFrame casts from size_t to int.pbos@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1540004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4094 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-23FrameGenerator class for future fake capture device.pbos@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1511004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4093 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-23Control new VideoEngine tests with gflags.pbos@webrtc.org
BUG=1703 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1497005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4092 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-23Adds print out of incoming resolution.henrike@webrtc.org
BUG=N/A R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1532004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4091 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-22API to control target delay in NetEq jitter buffer. NetEq maintains the ↵turaj@webrtc.org
given delay unless channel conditions require a higher delay. TEST=unit-test, manual, trybots. R=henrik.lundin@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1384005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4087 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-26 01:13:59 +0000