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2014-09-11Mark all virtual overrides in the hierarchies of RtpDump andhenrike@webrtc.org
VCMPacketizationCallback as such. This will make further changes to these classes safer by ensuring that the compile breaks if the base class changes and not all overrides are fixed. This also marks all other such overrides in the affected files. BUG=none TEST=none R=henrike@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19319004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7161 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11Update makefiles after merge of Chromium at b62471bd5180Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I03e28d2901e702a21f5ad8f0aba69055baff2d94
2014-09-11Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 61576f53599cf7840d3c4ebab82802b90031adcd This commit was generated by merge_from_chromium.py. Change-Id: Ia64db11ba8bae14d075c94b5ec153e6c1bea9589
2014-09-11Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.henrik.lundin@webrtc.org
This will make a subsequent change I intend to do safer, where I'll change the return type of one of the base Module functions, by breaking the compile if I miss any overrides. This also highlighted a number of unused functions (in many cases apparently virtual "overrides" of no-longer-existent base functions). I've removed some of these. This also highlighted several cases where "virtual" was used unnecessarily to mark a function that was only defined in one class. Removed "virtual" in those cases. BUG=none TEST=none R=andrew@webrtc.org, henrik.lundin@webrtc.org, mallinath@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7146 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10Update makefiles after merge of Chromium at a301aef21f9eAndroid Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I44df4fe3b2e608a292d12afc442eb8e98952bd4e
2014-09-10Update makefiles after merge of Chromium at d0b993bb2548Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I54601e2515cfffcca8e93ac943d9f4f25e35bac9
2014-09-09Update makefiles after merge of Chromium at facf66e09bf8Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I2575ec46c0cdef25c91211bafbe906833dc16496
2014-09-09Update makefiles after merge of Chromium at 457b0a1c9412Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: Ied0bc9fe472e73f303e492f5e0dcb1044fbea4a4
2014-09-08Update makefiles after merge of Chromium at 041843cbf814Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: Ib50e00ed5e94f0daf8eefe82b049a93bc3f416ef
2014-09-08Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 69370488385c14d73e6ae8a3d5001c42884f9275 This commit was generated by merge_from_chromium.py. Change-Id: Icd984259a9896fb874700b1e7a2e42bbabfb204b
2014-09-08Fix RTT calculations for send-only channels.stefan@webrtc.org
As we don't know the SSRC of the other end in a send-only channel since we haven't received packets from that end, we are required to assume that the SSRC of the first report block is the correct SSRC to use for RTT calculations. BUG=3781 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7097 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04Change return value for number of discarded packets to be int.asapersson@webrtc.org
R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14209004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7054 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04Fix audio/video sync when FEC is enabled.stefan@webrtc.org
Also improves the tests by adding a test case for FEC, and running the a/v sync tests with NACK and simulated packet loss. BUG=crbug/374104 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19209004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7053 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03Partial revert of r7014 (Android APK refactor)kjellander@webrtc.org
This reverts selected parts of r7014 to enable rolling WebRTC in Chromium DEPS. This works around the problem with GYP includes being processed in the first pass (i.e. variables cannot be used for paths). Using a dependency with a path using a variable that is conditioned for build_with_chromium being 0 or 1 solves the Chromium build. These changes will be restored once I've finished a major GYP refactoring that will break out all test related code (at least the parts that includes the Android APK targets) into a separate chain of GYP targets that are not processed when generating projects for Chromium (which is why r7014 is breaking the Chromium build). BUG=3741 TESTED=Passing compilation of standalone using: GYP_DEFINES="OS=android component=static_library fastbuild=1 target_arch=arm" webrtc/build/gyp_webrtc ninja -C out/Debug Then verified the *_apk targets are generated and compiled. Passing compilation from a Chromium checkout with third_party/webrtc directory removed and a new empty third_party/webrtc mapped to the standalone checkout using: sudo mount --bind /path/to/trunk/webrtc third_party/webrtc Then running build/gyp_chromium I also verified WebRTC GYP targets exist and are able to compile. R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20299004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7040 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at f8698ce1dacfdcf804809638483adb702760469c This commit was generated by merge_from_chromium.py. Change-Id: If6328cd0f1a485164c204accf41b23892a162235
2014-09-01Android APK tests built from a normal WebRTC checkout.kjellander@webrtc.org
Restructure how the Android APK tests are compiled now that we have a Chromium checkout available (since r6938). This removes the need of several hacks that were needed when building these targets from inside a Chromium checkout. By creating a symlink to Chromium's base we can compile the required targets. This also removes the need of the previously precompiled binaries we keep in /deps/tools/android at Google code. All the user needs to do is to add the target_os = ["android"] entry to his .gclient as described at https://code.google.com/p/chromium/wiki/AndroidBuildInstructions Before committing this CL, the Android APK buildbots will need to be updated. This also solves http://crbug.com/402594 since the apply_svn_patch.py usage will be similar to the other standalone bots. It also solves http://crbug.com/399297 BUG=chromium:399297, chromium:402594 TESTED=Locally compiled all APK targets by running: GYP_DEFINES="OS=android include_tests=1 enable_tracing=1" gclient runhooks ninja -C out/Release checkdeps R=henrike@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7014 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28GN: Implement video_engine, video_capture and video_render.kjellander@webrtc.org
Also add more from common.gypi to webrtc.gni. These GN configs are based on GYP files in r6997. BUG=3441 TEST=Trybots and local compile using: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default Passed compile from a Chromium checkout with src/third_party/webrtc linked to the webrtc/ dir of a checkout with this patch applied. R=brettw@chromium.org, glaznev@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18709004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6999 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-27Update makefiles after merge of Chromium at b241671f0248Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I1c4f94279ad561b4f0f5d73edca8ae1e9e86e36c
2014-08-26Remove Android.mk build files.pbos@webrtc.org
These files are generally not maintained and break, some contain files that don't exist anymore and do not build anymore. If we need to add some of these back we should really set up a bot for them. R=andrew@webrtc.org, glaznev@webrtc.org, henrike@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/15249004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26Remove former team members from OWNERS and WATCHLISTSkjellander@webrtc.org
Remove the following (CCed) former team members from all OWNERS files and the WATCHLISTS file: * fischman@ * leozwang@ * mikhal@ * pwestin@ * wu@ BUG= R=henrike@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6973 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 67afd1fc176021f625e064f20ae747e23d87d727 This commit was generated by merge_from_chromium.py. Change-Id: I6c68c68562042b290acef201bb3d998bee1dc9be
2014-08-20Bump WebRTC version number. Starting now, we will be setting WebRTC major ↵tnakamura@webrtc.org
version numbers to align with Chrome. R=niklas.enbom@webrtc.org TBR=niklas.embom Review URL: https://webrtc-codereview.appspot.com/15219004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6942 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-19Merge third_party/webrtc from ↵Torne (Richard Coles)
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c1696da9a74c7ed4ed793ce993352bd370cfc414 This commit was generated by merge_from_chromium.py. Change-Id: Iffb1ebee1a492d44a944cc8571605a2b40061d3e
2014-08-14Small refactor on ViE to remove redudant conditions and long ifdefs.andresp@webrtc.org
BUG=3694 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22069004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-14Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c2ef523233552340785557abce1129a0f61537eb This commit was generated by merge_from_chromium.py. Change-Id: If1f6b3b63673dba2f4d77d2ddea983f77d0eaab9
2014-08-14Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics().stefan@webrtc.org
Fixes issues where statistics only was reported for the first stream if configured with simulcast, and in case of RTX the reported statistics was depending on the order of the report blocks. Also fixes issues with multiple report blocks in the SendStatisticsProxy and the RtcpStatisticsCallback. SendStatisticsProxy is now aware of RTX ssrcs, and the RTCPReceiver is calling the RtcpStatisticsCallback with the correct SSRCs, and not only the primary stream SSRC. R=mflodman@webrtc.org, sprang@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13Decreased kMaxOverusesBeforeApplyRampupDelay (from 7 to 4).asapersson@webrtc.org
Increased kStandardRampUpDelayMs (30 to 40s). BUG=1577 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6886 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-12Update makefiles after merge of Chromium at 288938Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: Ic0141d5da4e1ab26e23ffa3c860ea955981acdb0
2014-08-11Remove more dependencies on openssl, add dependency on boringssl. Continues ↵henrike@webrtc.org
on r6798 R=andrew@webrtc.org, fbarchard@chromium.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6867 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-07Remove the old H264 code now that a new H.264 packetizer has been implemented.stefan@webrtc.org
R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15109004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6847 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04Update makefiles after merge of Chromium at 287308Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I85bf9681e9c3bbef9f67d93b3d275289e6911e3c
2014-07-29Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 4a1b3e3a69d349b0d3e91f607f24e02d8b975688 This commit was generated by merge_from_chromium.py. Change-Id: Iada7abd78f123301a98db982a6272cd9487de72f
2014-07-16Raw packet loss rate reported by RTP_RTCP module may vary too drastically ↵minyue@webrtc.org
over time. This CL is to add a filter to the value in VoE before lending it to audio coding module. The filter is an exponential filter borrowed from video coding module. The method is written in a new class called PacketLossProtector (not sure if the name is nice), which can be used in the future for more sophisticated logic. BUG= R=henrika@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6709 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 82383d9b14ff8e5fedf5a70229eb0ac6b512909a This commit was generated by merge_from_chromium.py. Change-Id: I8c578be801fa38420e875a4a8cef17e7522252e2
2014-07-16Print an info log instead of return an error if an external encoder is ↵stefan@webrtc.org
de-registered, but no corresponding internal encoder can be registered automatically. This is not an error case if for instance an external h.264 encoder is registered, but no internal implementation exists. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6704 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15Remove the VPM denoiser.pbos@webrtc.org
The VPM denoiser give bad results, is slow and has not been used in practice. Instead we use the VP8 denoiser. Testing this denoiser takes up a lot of runtime on linux_memcheck (about 4 minutes) which we can do without. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16069004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6688 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.tommi@webrtc.org
--- Fixes for re-enabling more MSVC level 4 warnings: webrtc/ edition This contains fixes for the following sorts of issues: * Possibly-uninitialized local variable * Signedness mismatch * Assignment inside conditional This also contains a small number of other cleanups to nearby code. In particular several warning-disables for MSVC are removed because they don't seem to be necessary (either that warning is not enabled or the code does not trigger it). BUG=crbug.com/81439 TEST=none R=henrika@webrtc.org, pkasting@chromium.org Review URL: https://webrtc-codereview.appspot.com/18769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6667 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.stefan@webrtc.org
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems. R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21869005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11Cast payload types to int for logging.pbos@webrtc.org
uint8_t gets interpreted as char and printed as such, instead of being printed in decimal, casting them to int allows us to read what payload types are actually used without converting them from ASCII first. BUG=chromium:390874 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6662 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11Remove the send-side cname getter APIs from voice and video engine.stefan@webrtc.org
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname. R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10Update makefiles after merge of Chromium at 282385Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I85d3e5fb3d9291809471c199df114e462a4739d6
2014-07-10Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module ↵andresp@webrtc.org
into vie_channel. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16839004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6649 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 138adbb0bcdab60afda25a8727e5a071abc4ae36 This commit was generated by merge_from_chromium.py. Change-Id: Iffa5413ebfb78de36b84b4e85d94adc093f912df
2014-07-09Thread annotations for vie_encoder.cc/.hstefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8739005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6638 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp ↵andresp@webrtc.org
module into vie_channel. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20879004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6626 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08Some refactoring inside rtp_rtcp/.pbos@webrtc.org
Renaming ModuleRTPUtility -> RtpUtility. Renaming RTPHeaderParser -> RtpHeaderParser. Making RtpHeaderParser accept size_t instead of int for packet length. Making RtpUtility::RtpHeaderParser accept size_t for packet length. BUG= R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07Preserve RTP states for restarted VideoSendStreams.pbos@webrtc.org
A restarted VideoSendStream would previously be completely reset, causing gaps in sequence numbers and potentially RTP timestamps as well. This broke SRTP which requires fairly sequential sequence numbers. Presumably, were this sent without SRTP, we'd still have problems on the receiving end as the corresponding receiver is unaware of this reset. Also adding annotation to RTPSender and addressing some unlocked access to ssrc_, ssrc_rtx_ and rtx_. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04Add boilerplate code for H.264.stefan@webrtc.org
R=mflodman@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17849005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04Configure RTX send status on new modules.pbos@webrtc.org
Fixes bug where newly-allocated modules wouldn't send payload-based padding (or probably not send over RTX at all). As the newly-added test exposed lock-inversions shown on tsan in VideoReceiver, VideoReceiver was thread-annotated and locks taken less. BUG=chromium:391085 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6601 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04Introduces PacedVideoSender to test framework and moves the Pacer to use Clock.stefan@webrtc.org
R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6600 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-26 02:57:59 +0000