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2015-05-27Merge WebView M40 build 1832189 into AOSP.HEADandroid-m-preview-2android-m-preview-1mastermainTorne (Richard Coles)
2015-03-12Update mac makefiles.android-m-previewmaster-soongTorne (Richard Coles)
Some MIPS changes got cherrypicked here but only the linux makefiles were updated. Update the mac makefiles to match. Change-Id: I517d4e6cf18c3c9ff39dc949e48687cc44663c7e
2015-01-05MIPS: Update Chromium WebView makefiles (webrtc).Paul Lind
Generated by: android_webview/tools/gyp_webview linux-mips after change I10406245f9708b9eacc40af83549a79f021944c7 is applied. Change-Id: I941175ced296aeec3976a270c699669bbd281957
2015-01-05Set default 'mips_arch_variant%' to 'r6'Gordana Cmiljanovic
With this change webrtc optimizations are excluded for all mips32 arch variants. This needs to be done as we have one gyp_webview target for all mips32 arch variants and r6 doesn't support a number of instructions which are supported in r1/r2. This is a temp workaround until a better solution is found. Change-Id: I10406245f9708b9eacc40af83549a79f021944c7
2014-12-05Temporarily disable -Werror in Chromium.Torne (Richard Coles)
Temporarily disable the use of -Werror in the AOSP copy of Chromium so that the system-wide default warnings can be changed without breaking Chromium. We'll re-enable it once the warnings are settled and we've fixed any issues. Bug: 18632512 Change-Id: Ie4231811e8ba5487b946bebfb515f01982ef2f66
2014-12-04Merge from Chromium at DEPS revision 40.0.2214.27webview-m40_r4webview-m40_r3webview-m40_r2webview-m40_r1ub-webview-m40-releaseBen Murdoch
This commit was generated by merge_to_master.py. Change-Id: Id333a9c9629dab78507c3f81a708fe07ca55fd7b
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
2014-12-04Merge third_party/webrtc from ↵Ben Murdoch
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 71616dbb8cdd3ea13e0a964d18456ca3fe002dab This commit was generated by merge_from_chromium.py. Change-Id: I40a02c5091b60b596c558bfc0f03c06c63d7fb7a
2014-11-27Merge r7729 into M40 branch.kjellander@webrtc.org
BUG=chromium:436400 TBR=phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27259004 git-svn-id: http://webrtc.googlecode.com/svn/branches/40/webrtc@7759 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-12Create a 40 branch from trunk@7660tnakamura@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/branches/40/webrtc@7692 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-12Merge from Chromium at DEPS revision 03655fd3f6d7Torne (Richard Coles)
This commit was generated by merge_to_master.py. Change-Id: I659bdd3b127bb83ee982b3e6117a93ad4fa68d54
2014-11-12Merge third_party/webrtc from ↵Torne (Richard Coles)
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 60ab669c4c545b328b5c8b0453eb2cdecf851651 This commit was generated by merge_from_chromium.py. Change-Id: I7efe0f9a490c88eb3ec522902a351f9662e8753e
2014-11-07Remove partially defined WebRtcRTPHeader from Parse().pbos@webrtc.org
It' bit ugly that RtpDepacketizer::ParsedPayload partially defines WebRtcRTPHeader, and then sent to Parse() function for internal change. To make it clearer, the CL gets rid of using partially-defined WebRtcRTPHeader. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28919004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7660 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Use uint16s for port numbers in webrtc/p2p/base.pkasting@chromium.org
This is a necessary precursor to using uint16s for port numbers more consistently in Chromium code. This also makes some minor formatting changes in surrounding code (function declaration wrapping, virtual -> override). BUG=chromium:81439 TEST=none R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32379004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7656 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Fix WebRTC Win64 + BoringSSL build.henrike@webrtc.org
There were many size_t to int conversions. RAND_poll and RAND_seed no longer do anything in BoringSSL, so fix that one by removing it. Use a checked_cast for the remaining ones. BUG=chromium:429039 R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7655 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Volume buttons in AppRTCDemo should affect output audio volume (part II).henrika@webrtc.org
See https://webrtc-codereview.appspot.com/32399004/ for part I. BUG=3279 TEST=AppRTC demo R=perkj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7654 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Merge from Chromium at DEPS revision db3f05efe0f9Torne (Richard Coles)
This commit was generated by merge_to_master.py. Change-Id: Ibb07e7633f0f96e925c9bd5cdcb91747ad656b6e
2014-11-06Log formatting fix for VideoEncoderConfig.pbos@webrtc.org
R=mflodman@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/30889004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7648 4adac7df-926f-26a2-2b94-8c16560cd09d
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
2014-11-06Remove the state_ member from AudioDecoderkwiberg@webrtc.org
The subclasses that need a state pointer should declare them---with the right type, not void*, to get rid of all those casts. Two small but not quite trivial cleanups are included because they blocked the state_ removal: - AudioDecoderG722Stereo now inherits directly from AudioDecoder instead of being a subclass of AudioDecoderG722. - AudioDecoder now has a CngDecoderInstance member function, which is implemented only by AudioDecoderCng. This replaces the previous practice of calling AudioDecoder::state() and casting the result to a CNG_dec_inst*. It still isn't pretty, but now the blemish is plainly visible in the AudioDecoder class declaration. R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24169005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7644 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Adjust parameter in vp9 rate control test.marpan@webrtc.org
TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25099004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7638 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test.marpan@webrtc.org
TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28949004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7637 4adac7df-926f-26a2-2b94-8c16560cd09d
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
2014-11-05Restore old behavior for Android in fileutils.cckjellander@webrtc.org
From r7014 the Android APK tests are designed to be build from a standalone WebRTC checkout instead of a Chromium checkout. Because of that, the special handling for both cases can be removed. I also don't think we need to use the base::android::GetExternalStorageDirectory() method since all devices has a symlink at /sdcard that points to /storage/emulated/legacy on the Android device. This essentially reverts the changes in https://webrtc-codereview.appspot.com/1754005/ plus some minor changes. BUG=webrtc:3741 TEST=Locally running test_support_unittests APK test on an Android device using: CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py gtest -s test_support_unittests --verbose --isolate-file-path=webrtc/test/test_support_unittests.isolate R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15329004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7632 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05Fix android_clang build.glaznev@webrtc.org
BUG= R=kjellander@webrtc.org, tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30879004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7630 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05Revert 7623 "Remove the state_ member from AudioDecoder"niklas.enbom@webrtc.org
Breaks Chrome compile: e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3861: 'LOG_FERR1': identifier not found e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3861: 'LOG_FERR1': identifier not found e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(169) : error C3867: 'webrtc::NetEqImpl::GetAudioInternal': function call missing argument list; use '&webrtc::NetEqImpl::GetAudioInternal' to create a pointer to member ... > Remove the state_ member from AudioDecoder > > The subclasses that need a state pointer should declare them---with > the right type, not void*, to get rid of all those casts. > > Two small but not quite trivial cleanups are included because they > blocked the state_ removal: > > - AudioDecoderG722Stereo now inherits directly from AudioDecoder > instead of being a subclass of AudioDecoderG722. > > - AudioDecoder now has a CngDecoderInstance member function, which > is implemented only by AudioDecoderCng. This replaces the previous > practice of calling AudioDecoder::state() and casting the result > to a CNG_dec_inst*. It still isn't pretty, but now the blemish is > plainly visible in the AudioDecoder class declaration. > > R=henrik.lundin@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/24169005 TBR=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30879005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7629 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05Revert 7625 "Don't use DCHECK when you need the side effects..."niklas.enbom@webrtc.org
Reverting since 7623 might depend on this one > Don't use DCHECK when you need the side effects... > > R=pbos@webrtc.org > TBR=henrik.lundin@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/32369004 TBR=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7628 4adac7df-926f-26a2-2b94-8c16560cd09d
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
2014-11-04Don't use DCHECK when you need the side effects...kwiberg@webrtc.org
R=pbos@webrtc.org TBR=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32369004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7625 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Remove the state_ member from AudioDecoderkwiberg@webrtc.org
The subclasses that need a state pointer should declare them---with the right type, not void*, to get rid of all those casts. Two small but not quite trivial cleanups are included because they blocked the state_ removal: - AudioDecoderG722Stereo now inherits directly from AudioDecoder instead of being a subclass of AudioDecoderG722. - AudioDecoder now has a CngDecoderInstance member function, which is implemented only by AudioDecoderCng. This replaces the previous practice of calling AudioDecoder::state() and casting the result to a CNG_dec_inst*. It still isn't pretty, but now the blemish is plainly visible in the AudioDecoder class declaration. R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24169005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Add support for VP9 in webrtc::Call and video_loopback.stefan@webrtc.org
R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7622 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Reduce to 2 probes when probing for initial bandwidth.stefan@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23359005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7621 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Add UMA for measuring the diff between the BWE at 2 seconds compared to the ↵stefan@webrtc.org
BWE at 20 seconds when the BWE should have converged. BUG=crbug/425925 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30819005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7620 4adac7df-926f-26a2-2b94-8c16560cd09d
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
2014-11-04Adds support for finch experiments to video_loopback.stefan@webrtc.org
Adds support for logging to stderr via -logs. Enables abs-send-time by default. R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23329004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7613 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Fix problem with late packets in NetEqhenrik.lundin@webrtc.org
Since r7255, it could happen that an old packet would block the decoding process until enough packet was received for the buffer to flush. This CL fixes that by: - Partially reverting r7255; - Remove recent old packets before taking a decision for GetAudio; - Remove all old packets after a packet has been extracted for decoding; - Adding tests for reordered packets. BUG=chrome:423985 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25079004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7612 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Delete VideoReceiveStream channels in destructor.pbos@webrtc.org
R=stefan@webrtc.org BUG=1667 Review URL: https://webrtc-codereview.appspot.com/31909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16kwiberg@webrtc.org
R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26039004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7610 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Remove the useless dummy state parameter to WebRtcG711_*kwiberg@webrtc.org
R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7609 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Remove the codec_type_ member from AudioDecoderkwiberg@webrtc.org
It isn't actually required, as evidenced by the comparative ease with which it can be removed. R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31939004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7606 4adac7df-926f-26a2-2b94-8c16560cd09d
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
2014-11-04Improving error message from neteq_rtpplayhenrik.lundin@webrtc.org
If a packet with unknown RTP payload type is inserted, this CL will make sure that the error message is a little more detailed and gives a better understadning of what to do. BUG=2692 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7603 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Add format members to AudioConverter for DCHECKing.andrew@webrtc.org
And use a std::min. Post-commit fixes after: https://review.webrtc.org/30779004/ TBR=kwiberg Review URL: https://webrtc-codereview.appspot.com/25059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7600 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Update rate control parameter in vp9 test.marpan@webrtc.org
TBR=hellner@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7599 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-03 07:38:39 +0000