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2014-12-04Update makefiles after merge of Chromium at 40.0.2214.27Ben Murdoch
This commit was generated by merge_from_chromium.py. Change-Id: I231a22a5117516d0892018e313b21fab26b1f615
2014-11-06Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at b831a9e3d5f9f0563d249b726cffa8a070e58aee This commit was generated by merge_from_chromium.py. Change-Id: I6d6255972e3c34e7797e9b46fbc3c0fe7e552d43
2014-11-05Update makefiles after merge of Chromium at 5a645aa13b82Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I038f8684aa804c94ca2c5175bdeaf605bf0611c5
2014-11-05Remove uses of build date/time.pbos@webrtc.org
Uses of __DATE__ and __TIME__ are blocking deterministic Chromium builds. We're not really making use of these, and if anything they're likely to be misleading as it's impossible to distinguish between a new revision and a freshly-built old branch. R=mflodman@webrtc.org, tnakamura@webrtc.org BUG=3983 Review URL: https://webrtc-codereview.appspot.com/27039004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7635 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05Wire up bandwidth stats to the new API and webrtcvideoengine2.stefan@webrtc.org
Adds stats to verify bandwidth and pacer stats. BUG=1788 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24969004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05Update makefiles after merge of Chromium at 2d0da5605d75Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I2b5db589b04e302cb1067fe730b81f3fb21b06bb
2014-11-04Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 9a8c28f10f329c5ce91e77057933e60224000627 This commit was generated by merge_from_chromium.py. Change-Id: If9e805f5024e1fcdd99127626811f9650e109b1d
2014-11-04Update makefiles after merge of Chromium at a99b7ad25d02Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I172fda810eb6cb37d17ba35571733f9eaeb9b230
2014-11-04Reworked paced sender queuesprang@webrtc.org
Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage. Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these. Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Update makefiles after merge of Chromium at 30ec995cdb2dAndroid Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I77678e9f2e5044a6457f21cada6ee13b75fbfb0c
2014-11-04Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at a1fd19c12e4efdf4b8a5f92323070443d50dc34e This commit was generated by merge_from_chromium.py. Change-Id: Ibd48eca2d93e6324a2e886e451f27307aab45e9b
2014-11-04Enables AIMD control by default.stefan@webrtc.org
BUG=1788 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27019004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7604 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Roll chromium_revision: 28d1981..d3db2ffmarpan@webrtc.org
Pick up the libvpx roll: https://codereview.chromium.org/674753002 Summary of changes (https://chromium.googlesource.com/chromium/src/+/28d1981..d3db2ff/DEPS): * third_party/android_tools 36bf7ac..ea50ccc * third_party/boringssl 7ea8481..751e889 * third_party/icu 8ac906f..d8b2a9d * third_party/libvpx efe9712..2e5ced5 * third_party/usrsctp/usrsctplib * tools/gyp 1990:1991 * tools/swarming_client a57d7db..bcb3bc3 Clang is not updated in this roll. Made the change getchar() --> getc(stdin) as seems like getchar() isn't supported on android anymore. (getchar() was causing the error: undefined reference to '__srget') Update rate control parameter in vp9 test. R=andrew@webrtc.org TBR=ajm@google.com Review URL: https://webrtc-codereview.appspot.com/23229004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7598 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Add stats for video:asapersson@webrtc.org
- number of sent/received RTCP NACK/FIR/PLI per minute - percentage of unique sent/received NACK requests - percentage of discarded/duplicated packets by the jitter buffer - permille of sent/received key frames BUG=crbug/419657 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7592 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01Add VP9 codec to VCM and vie_auto_test.marpan@webrtc.org
Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Update makefiles after merge of Chromium at a41c404b1c7fAndroid Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: Ifcab5d7c5bd698b1a0a72100960585183048352d
2014-10-31Update makefiles after merge of Chromium at b210e2d62956Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I3b99d06f861694a90ee0f32a97380e1c99cfaa07
2014-10-31Update all .isolate files for the new format.kjellander@webrtc.org
R=kjellander@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/27809004 Patch from Marc-Antoine Ruel <maruel@chromium.org>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29Remove unused code in overuse detector.asapersson@webrtc.org
R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31579004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7557 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 4f2aa0829e4e69972202efb7de2f53cc8858e2c9 This commit was generated by merge_from_chromium.py. Change-Id: I5142c5b2111742e7eabf5c5a7ca1541ce639d7d6
2014-10-28Update makefiles after merge of Chromium at 82ca3b654cdaAndroid Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I74f8ff67d68c1bb764fd2a96bbd03a8f7713475f
2014-10-23Update makefiles after merge of Chromium at 9ef958e74e13Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I9f8bef9f285b9b8b005c64271d0c3b0911623223
2014-10-23Add macros and APIs for webrtc histograms.asapersson@webrtc.org
BUG=crbug/419657 Code that links system_wrappers.gyp:system_wrappers should either: - provide implementations for the APIs, or - link with default implementations in system_wrappers.gyp:system_wrappers_default. R=andresp@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7508 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23Adds support for sending first set of packets at increasingly higher ↵stefan@webrtc.org
bitrates to probe the link and faster ramp up to a high bitrate. Also wires up a finch experiment to control this. R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7505 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 4415bacc16e429ee132e9759ba6880043b61cbdd This commit was generated by merge_from_chromium.py. Change-Id: Ie6289bc4dd0cfeaac35bf8fe39e6bb03ce057148
2014-10-19Update makefiles after merge of Chromium at 89b463ddd92bAndroid Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: Ib4f1a99ca57b985be94ad5094d09418dc048e66a
2014-10-17Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."henrike@webrtc.org
BUG=3932 R=marpan@google.com Review URL: https://webrtc-codereview.appspot.com/27779004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7470 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-16Add ability to include a larger time span (in addition to encode time) for ↵asapersson@webrtc.org
measuring the processing time of a frame. Controlled by setting enable_extended_processing_usage. Enabled by default. R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13289004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7460 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 7da30673d742075b52c63bb96b78f2c35cc93991 This commit was generated by merge_from_chromium.py. Change-Id: I6ae37fb65fc9bcff451322ab8e1f52b667dc5d10
2014-10-15Update makefiles after merge of Chromium at 6e9c84566c9fAndroid Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: Ia99600712fa5534613907d4426663655656d2285
2014-10-14Add periodic logging of received RTP headers and estimated clock offsets for ↵stefan@webrtc.org
e2e delay. R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25789004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7444 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14Add a packet loss full stack test to the new API.stefan@webrtc.org
Remove all full stack tests for the old API. BUG=3750 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7442 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10Add VP9 codec to VCM and vie_auto_test.marpan@webrtc.org
Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. Passes trybots. R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7422 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. xians@webrtc.org
This also marks all virtual overrides of other classes in the same files. This will make a subsequent change I intend to do safer, where I'll change the argument types of the base Transport functions, by breaking the compile if I miss any overrides. This also highlighted a number of unused functions. I've removed some of these. TBR=mflodman@webrtc.org, pkasting@chromium.org BUG=none TEST=none Review URL: https://webrtc-codereview.appspot.com/28709004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09Estimating NTP time with a given RTT.minyue@webrtc.org
RemoteNtpTimeEstimator needed user to give a remote SSRC and it intended to call RtpRtcp module to obtain RTT, to be able to calculate Ntp time. When RTT cannot be directly obtained from the RtpRtcp module with the specified SSRC, RemoteNtpTimeEstimator would fail. This change allows RemoteNtpTimeEstimator to calculate NTP with an external RTT estimate. An immediate benefit is that capture_start_ntp_time_ms_ can be obtained in a Google hangout call. BUG= TEST=chromium + hangout call R=stefan@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24879004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7407 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Update makefiles after merge of Chromium at 9c6ac85c45faAndroid Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: Ib27ef7e6411396bac83926bf0b251668be9e6988
2014-09-30Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 296871bd3d804dfcd3c16c59a83fb173f6dfd438 This commit was generated by merge_from_chromium.py. Change-Id: Ibe75d8eba4230c87f7a0155047afcb845e0b93de
2014-09-28GN: Add common configs to all targets.kjellander@webrtc.org
This is needed to ensure we have the same build with GN as with GYP, since GYP includes the common.gypi on a global level. Several fixes has been needed in the past because some code have been built without the right defines. BUG=3441 R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/28589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7317 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-26Mark virtual overrides of ViENetwork and VoENetwork as such.henrikg@webrtc.org
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. BUG=none TEST=none R=henrikg@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26439004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7305 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25Bump to version 39tnakamura@webrtc.org
TBR=niklas.enbom Review URL: https://webrtc-codereview.appspot.com/24749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7299 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 291035ed1d8ec308ffbc81e9cd119e2f53f92f86 This commit was generated by merge_from_chromium.py. Change-Id: I8de5d3b4724dd14ebda167e51683b554ddb5e024
2014-09-24Move thread_annotations.h to webrtc/base/.pbos@webrtc.org
R=andresp@webrtc.org, mflodman@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/27579004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22Update makefiles after merge of Chromium at fb34b348eeadAndroid Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I247d3aafc57df5f4ac99db567f5b7c2313fb7b7a
2014-09-19Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 70861e04b3580c1350c5479c9ee26469f38ff782 This commit was generated by merge_from_chromium.py. Change-Id: Ib21179e777d68b7a3deb3e39851d6f26ffd72b39
2014-09-18Split video_render_module implementation into default and internal ↵andresp@webrtc.org
implementation. Targets must now link with implementation of their choice instead of at "gyp"-time. Targets linking with libjingle_media: - internal implementation when build_with_chromium=0, default otherwise. Targets linking with default render implementation: - video_engine_tests - video_loopback - video_replay - anything dependent on webrtc_test_common Targets linking with internal render implementation: - vie_auto_test - video_render_tests - libwebrtcdemo-jni - video_engine_core_unittests GN changes: - Not many since there is almost no test definitions. Work-around for chromium: - Until chromium has updated libpeerconnection to link with video_capture_impl and video_render_impl, webrtc target automatically depends on it. This should fix the FYI bots and not require a webrtc roll to fix. Re-enable android tests by reverting 7026 (some tests left disabled). TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in. BUG=3770 R=kjellander@webrtc.org, pbos@webrtc.org TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7217 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17Split video_capture_module specific implementation (external vs internal ↵andresp@webrtc.org
capture) into its own targets. Dependencies must link directly with the desired one. Targets linking with libjingle_media: - internal implementation when build_with_chromium=0, default otherwise. Targets linking with default/external capture implementation: - anything dependent on webrtc_test_common - anything dependent on video_engine_core Targets linking with internal capture implementation: - vie_auto_test - anything dependent on webrtc_test_renderer GN changes: - Not many since there is almost no test definitions. TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in. BUG=3768 R=glaznev@webrtc.org TBR=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7209 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17Split video engine android initialization into each internal module ↵andresp@webrtc.org
initialization. This is to later on allow targets to pick at link time if to include the external or internal implementation. In order to do that the video_engine cannot compile different based on which option is picked later on. BUG=3768,3770 R=glaznev@webrtc.org, stefan@webrtc.org TBR=henrike@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7208 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-15Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 9a52f504918a75e4c864ac661bce1e934adba7b1 This commit was generated by merge_from_chromium.py. Change-Id: Id63d69bfd855c182c6c1ca27874bc11fbd880cc7
2014-09-13Update makefiles after merge of Chromium at 6a4d455b8650Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I76592b034e4c868de1bd5ebafa48efc39ee50f0c
2014-09-12Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such.stefan@webrtc.org
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 highlighted a number of unused functions which I've removed. -- This is was reviewed in https://webrtc-codereview.appspot.com/19309004/, but -- a new cl was needed to resolve a small conflict before committing. BUG=none TEST=none TBR=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7162 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-10 04:02:50 +0000