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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
2014-09-11Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."henrikg@webrtc.org
Speculative revert, seems to be reason for flaky Win FYI bot compile break. > Expose VideoEncoders with webrtc/video_encoder.h. > > Exposes VideoEncoders as part of the public API and provides a factory > method for creating them. > > BUG=3070 > R=mflodman@webrtc.org, stefan@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/21929004 TBR=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19329004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7151 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11Restore webrtc_base target until r7140 is rolled into Chromium.kjellander@webrtc.org
In r7140 the webrtc_base target was renamed to rtc_base. This breaks our FYI bots for rolling WebRTC in Chromium's DEPS. By re-adding a None target named webrtc_base, this transition should be smoother. TBR=henrikg@webrtc.org, TESTED=Passed build/gyp_chromium on a Chromium checkout with src/third_party/webrtc replaced by a mount like this: cd /path/to/chromium/src sudo mount --bind /path/to/webrtc/trunk/webrtc third_party/webrtc Review URL: https://webrtc-codereview.appspot.com/23589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7150 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11audio_processing_unittests: Enabled ApmTest.Process for all platforms but ↵bjornv@webrtc.org
Android During porting some neon optimizations to sse2 the ApmTest.Process failed despite bit exact outputs. The reason is that with float data between component, as to previously truncating to int, we get small deviations in logged metrics. This affected a noise probability to a small fraction, which is not a particular bug. This CL change the comparison from EXPECT_EQ() to EXPECT_NEAR() which then as a result makes the test run on Mac and Windows as well. For int values a deviation of 1 is acceptable, which would include any rounding errors. For float values a deviation of 0.0005 is chosen by looking at current test stats for the affected platforms/optimizations. BUG=114 TESTED=locally on linux with and without sse2 optimizations and trybots R=aluebs@webrtc.org, andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20289004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7149 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11Calculating round-trip-time in send-only channel in VoE.minyue@webrtc.org
TESTS=built chromium and tested with 1:1 hangout call BUG= R=stefan@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23489004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7147 4adac7df-926f-26a2-2b94-8c16560cd09d
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
2014-09-10Mark all virtual overrides in the hierarchy of AudioPacketizationCallback,henrike@webrtc.org
RTPStream, and NetEq as such. Also mark all other virtual overrides in the same files. 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 deletes ACMTest.cc, which existed solely to define ~ACMTest(), which was marked pure virtual in the header. (Pure virtual destructors still need a definition.) Because there is another pure virtual method in this class, the class is already abstract, so there's no benefit to making the desturctor pure. Making it non-pure allows removing the separate source file. BUG=none TEST=none R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29389004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7144 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10Fix MSVC warnings about value truncations, webrtc/base/ edition.henrike@webrtc.org
BUG=chromium:81439 TEST=none R=henrike@webrtc.org, marpan@google.com Review URL: https://webrtc-codereview.appspot.com/20249004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7143 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10Fix frame rate selection for Android camera.glaznev@webrtc.org
- Android camera supports multiple fps values for a single video resolution - change video source default video format selection to pick up best available fps. - Change fps range calculation to better match target fps value. BUG=2622 R=tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15339004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7142 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10Add schannel webrtc_base build using a new use_schannel gyp variable.tpsiaki@google.com
R=henrike@webrtc.org, thorcarpenter@google.com Review URL: https://webrtc-codereview.appspot.com/28409004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7141 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10Put base tests in webrtc_tests.gyphenrike@webrtc.org
BUG=N/A R=andrew@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14249004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7140 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10Convert GN visibility to be lists.brettw@chromium.org
This is a followup to my previous patch that missed this case. R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7137 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10Simplify gyp rules on video_render_module.andresp@webrtc.org
R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28439004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7135 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10Fix printing of error stack in rtcbot when a test fails via test.fail().houssainy@google.com
R=andresp@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7134 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10Fix compile error on JDK 1.7.kjellander@webrtc.org
JDK 1.7 gives an error like this: warning: [static] static method should be qualified by type name R=pbos@webrtc.org TBR=henrike@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/29399004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7133 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10Remove DestructEncoderInst and its codec-specific implementations.henrik.lundin@webrtc.org
This method is seemingly never called. BUG=none TEST=none R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24539004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7131 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
2014-09-10include cstdlib for free() and abort()andrew@webrtc.org
This previous CL added uses of free() and abort() without including cstdlib: https://webrtc-codereview.appspot.com/22449004 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23559004 Patch from Mostyn Bramley-Moore <mostynb@opera.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7127 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10Update makefiles after merge of Chromium at d0b993bb2548Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I54601e2515cfffcca8e93ac943d9f4f25e35bac9
2014-09-09Add a new class InterfaceAddress inherited from IPAddress to keep track of ↵guoweis@webrtc.org
IPv6 Address flags. Skeleton put in place in Network::GetFilterIPs() which will be used to filter addresses BUG=3773 R=jiayl@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23439004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7126 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09Fix up configs applying to GN build.brettw@chromium.org
The audio_processing target didn't have the build configs applying to it which led to some logging errors. TBR=kjellander Review URL: https://webrtc-codereview.appspot.com/22339004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7125 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09Change explicit static cast from int to uint16_t to implicit cast of 0u.fbarchard@google.com
BUG=3663 TESTED=local windows build with VS2013. R=harryjin@google.com, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28389004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7123 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09Fix the RTC+Chromium GN build.brettw@chromium.org
LOGGING_INSIDE_WEBRTC was being set in the inherited config, whereas in the GYP build this define is not inherited. This caused duplicate logging macros to be defined in Chrome files dependening on WebRTC targets. Move LOGGING_INSIDE_WEBRTC to the common config (non-inherited). TBR=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7122 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09TurnPort should retry allocation with a new address on error ↵jiayl@webrtc.org
STUN_ERROR_ALLOCATION_MISMATCH. BUG=3570 R=juberti@webrtc.org, mallinath@webrtc.org Committed: https://code.google.com/p/webrtc/source/detail?r=7070 Review URL: https://webrtc-codereview.appspot.com/20999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7120 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09Bot Browser files moved to /bot/browser/houssainy@google.com
because android files will be a different and will need to add more files for Android. There was a CL to move the browser files form bot/browser/ to /bot/ as i thought it will be the same files used for Android. R=andresp@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7119 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09Update makefiles after merge of Chromium at facf66e09bf8Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I2575ec46c0cdef25c91211bafbe906833dc16496
2014-09-09fix a bug in the logic when new Networks are merged. This happens whenguoweis@webrtc.org
we have 2 networks with the same key BUG=410554 in chromium http://code.google.com/p/chromium/issues/detail?id=410554 Corresponding change in chromium is https://codereview.chromium.org/536133003/ R=jiayl@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19249005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7117 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09Expose VideoEncoders with webrtc/video_encoder.h.pbos@webrtc.org
Exposes VideoEncoders as part of the public API and provides a factory method for creating them. BUG=3070 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21929004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7114 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09Update makefiles after merge of Chromium at 457b0a1c9412Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: Ied0bc9fe472e73f303e492f5e0dcb1044fbea4a4
2014-09-08Initialize ChannelBuffer's memory to avoid uninitialized reads.andrew@webrtc.org
Removed the zero out memset in this change: https://review.webrtc.org/24469004/ assuming it was unneeded. Dr. Memory taught me that assupmtion was invalid. linux_memcheck try runs might have caught this, if they weren't flaking out on unrelated stuff. TBR=claguna@google.com Review URL: https://webrtc-codereview.appspot.com/28429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7113 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08Convert GN visibility to be a list.brettw@chromium.org
GN visibility currently allows either string or list types, but this is causing some problems for some templates. I'm going to require it to be lists, so am changing all callers before pushing the new binary. R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25439004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7111 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08Add ctors to ChannelBuffer to enable copying on construction.andrew@webrtc.org
Also: - Fix the constness of some parameters. - Add more const overloads. - Use DCHECK in place of assert. - Removed an unnecessary memset. R=claguna@google.com Review URL: https://webrtc-codereview.appspot.com/24469004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7107 4adac7df-926f-26a2-2b94-8c16560cd09d
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
2014-09-08Set a default speech type in iSAC wrapperhenrik.lundin@webrtc.org
If the decoder encounters an error, it may leave the speech type unassigned, leading to a use-of-uninitialized-value in subsequent lines. BUG=crbug/411162 R=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23519004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7104 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08Starting to implement the new ACM APIhenrik.lundin@webrtc.org
The new implementation class is called AudioCodingImpl, and will in the end replace AudioCodingModuleImpl. This is work in progress. BUG=3520 R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7103 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08Adding the ability to test on Chrome for Android.houssainy@google.com
use "android-chrome" as type in rtcbot running command. Example: node test.js android-chrome R=andresp@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20329004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7102 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08audio_processing: Removed use of macro WEBRTC_SPL_UMUL_16_16bjornv@webrtc.org
The macro replaced is a trivial multiplication after explicit casts to uint16_t and uint32_t. This CL replaces its use with "*" and adds explicit casts if necessary. Affected components: * AECMobile * AGC * Noise Suppression (fixed point version) BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27389004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7101 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08video_processing: Removed usage of WEBRTC_SPL_UMUL_16_16bjornv@webrtc.org
The trivial macro WEBRTC_SPL_UMUL_16_16 is nothing but plain mutliplication of casted values. This CL explicitly use "*" at place and casts if necessary. BUG=3348,3353 TESTED=locally on linux and trybots R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23499004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7100 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08- Adding AndroidDeviceManager to botManager.js to help in selecting devices, ↵houssainy@google.com
in case running test on Android devices. - Select BotType using nodeJs terminal command. - ping_pong.js test added. R=andresp@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19159004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7099 4adac7df-926f-26a2-2b94-8c16560cd09d
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
2014-09-08Ignore FEC packet in stats, if it is first packet on ssrc.sprang@webrtc.org
BUG=chrome:410456 R=mflodman@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22309004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7096 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07GN: Prefix WebRTC specific variables with "rtc_"kjellander@webrtc.org
BUG=3441 TESTED=Trybots + Running GN in a Chromium checkout with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using: --args="os=\"android\" cpu_arch=\"arm\"" R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/27379004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07Add video_capture_tests_apk_targetkjellander@webrtc.org
In https://codereview.chromium.org/500423004/ the target that was previously used to build the Android APK tests was removed. When building these tests from a standalone checkout, the video_capture_tests_apk target was missing in the chain of targets that gets generated into the 'all' target. BUG=3764 TESTED=Trybots. TBR=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7094 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-06Fix rm command for class cleanup in r7091kjellander@webrtc.org
In https://webrtc-codereview.appspot.com/20339004 the rm command was missing 'r' for recursive mode. TBR=henrike@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/26379004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7092 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-06Cleanup temporary class files for OpenSlDemokjellander@webrtc.org
I've seen tryjobs failing when they shouldn't on the Android trybots and I suspect this might have something to do with it. TBR=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20339004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7091 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05Create a new interface for AudioCodingModulehenrik.lundin@webrtc.org
This is a first draft of the interface, and is work in progress. BUG=3520 R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17299004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7085 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05Drop buildbot_tests.py scriptkjellander@webrtc.org
This is no longer used since the buildbots have moved over to recipes (where these arguments are configured). See https://code.google.com/p/chromium/codesearch#chromium/tools/build/scripts/slave/recipe_modules/webrtc/api.py&l=73 for details. This is essentially a revert of https://webrtc-codereview.appspot.com/1021006 BUG=None TESTED=None R=phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26369004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7079 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05Modifying audio_coding/codecs/OWNERShenrik.lundin@webrtc.org
Adding myself. R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7077 4adac7df-926f-26a2-2b94-8c16560cd09d
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