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2014-04-10Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 55bc2810c06fe624311518c4502af5ca8a5c085c This commit was generated by merge_from_chromium.py. Change-Id: I9ccff119d81dc5f08bc0a54253907449806630ca
2014-04-09Update makefiles after merge of Chromium at 262754Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I354903dccf62f6b7d2bc87116fb6b5a126fe7613
2014-04-08Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.andresp@webrtc.org
Clean some logs and add asserts in the way. BUG=3153 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5861 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-07Update makefiles after merge of Chromium at 262110Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I16520e5550a81c5a16a4c7a907bedd1f481e6135
2014-04-04Update makefiles after merge of Chromium at 261622Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: Ib42e3532b721e1ba8f5a8e00711f582337238590
2014-04-01Update makefiles after merge of Chromium at 260927Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I60dc39e695f917c3b07ba006183d3d2a714b378e
2014-03-21Fix "unreachable code" warnings (MSVC warning 4702) in webrtc.pbos@webrtc.org
BUG=chromium:346399 TEST=none R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10139004 Patch from Peter Kasting <pkasting@chromium.org>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5747 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-20Use codec width/height as the encoded_image width/height.wu@webrtc.org
The raw_->w and raw_->h which are the stored image width/height may not be the encoded image size in the case when the incoming frame has a odd size. R=marpan@google.com, marpan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10289004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5739 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18Add #include <cstdlib> for std::abs.pbos@webrtc.org
IWYU violation. Fixes a breakage in the libc++ build of Chromium. BUG= R=earthdok@chromium.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10109004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5715 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13Replace labs with std::abs.pbos@webrtc.org
Resolves clang 3.5 warnings on OS X for -Wabsolute-value. BUG=chromium:351479 R=andrew@webrtc.org, thakis@chromium.org Review URL: https://webrtc-codereview.appspot.com/9869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5692 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-07Remove std:: prefixes from C functions in webrtc/.pbos@webrtc.org
std::memcpy -> memcpy for instance. This change was motivated by a compile report complaining that std::rand() was used instead of rand(), probably with a stdlib.h include instead of cstdlib. Use of C functions without the std:: prefix is a lot more common, so removing std:: to address this. BUG= R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9549004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5658 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-05Remove upper check for number of cores in VCM, I didn't find any good ↵mflodman@webrtc.org
reasons for checking this. BUG=2990 TEST=Manually adding a high number without any noticable change. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9399004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5645 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-03Fix compilation errors under clang 3.5.pbos@webrtc.org
Enables building tip-of-tree clang which introduces new warnings that cause compilation errors in our code base (-Werror). BUG= R=andrew@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9319004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5630 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-24Split the implementation of VP8Encoder|Decoder::Create into a seperated filewu@webrtc.org
(vp8_factory.cc). R=fischman@webrtc.org, marpan@google.com, marpan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5606 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-31Add BWE tools for parsing RTP files.stefan@webrtc.org
bwe_rtp_play feeds packets from an RTP file into the BWE and prints the estimates. bwe_rtp_to_text parses an RTP file and outputs the result to a text file. R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7689006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5466 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29Fix deadlock in video_receiver.cc.stefan@webrtc.org
In webrtc::vcm::VideoReceiver::ResetDecoder(), the lock order is: 1. take _receiveCritSect, 2. take process_crit_sect_ This conflicts with the follow code path: 1. webrtc::vcm::VideoReceiver::Process(), take process_crit_sect_ call -> webrtc::vcm::VideoReceiver::NackList(), 2. with nackStats=kNackKeyFrameRequest, take _receiveCritSect BUG=2861 TEST=trybots R=sprang@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5456 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27Fix "field '_testNo' is uninitialized" warnings.pbos@webrtc.org
BUG=2849 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5434 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-14Remove empty VideoCodecGeneric struct.pbos@webrtc.org
Struct was added prematurely and triggers a warning with -Wextern-c-compat in latest clang. R=henrika@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/7119004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5383 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-09Make code simpler on VCMEncodedCallback.andresp@webrtc.org
R=marpan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6689004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5358 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-09Isolate register post encode callback in video coding module to simplify ↵andresp@webrtc.org
code and critical sections. R=marpan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6659004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5357 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-08Isolate debug recording from video sender into a thread safe small class.andresp@webrtc.org
R=marpan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5353 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-19Refactoring MediaOptimization so it can easily be turned into a thread-safe ↵andresp@webrtc.org
class. BUG=2732 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5322 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11Add SwapFrame() to VideoSendStreamInput.pbos@webrtc.org
Optionally prevents doing a frame copy when putting frames into a VideoSendStream. PutFrame() is still there, which copies the frame. Also removes time_since_capture_ms as a parameter, since I420VideoFrame::render_time_ms() denotes when the frame was captured. BUG=2657 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5119004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5265 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04Add send frame rate statistics callbacksprang@webrtc.org
BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4479005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5213 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-26Implement and test EncodedImageCallback in new ViE API.sprang@webrtc.org
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5179 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18Rename AutoMute to SuspendBelowMinBitratehenrik.lundin@webrtc.org
Changes all instances throughout the WebRTC stack. BUG=2436 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-11Remove ".." from include_dirs in build/common.pbos@webrtc.org
BUG=1662 TEST=compile on trybots R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2332004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5107 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-06Fix for video_processor_intergration_tests to run in parallel.marpan@webrtc.org
BUG=2601. R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3519004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5091 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28Adding tl0idx consideration for continuitymikhal@webrtc.org
R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2879004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5046 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28Removing the threshold from the auto-mute APIshenrik.lundin@webrtc.org
The threshold is now set equal to the minimum bitrate of the encoder. The test is also changed to have the REMB values depend on the minimum bitrate from the encoder. BUG=2436 R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5040 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23Expose VideoCodingModule's decoder stats up the stack from VCMTiming to ↵fischman@webrtc.org
chrome://webrtc-internals. R=juberti@google.com, mikhal@webrtc.org, stefan@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5027 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-22Upgrade scoped_ptr to Chromium's latest version.andrew@webrtc.org
Analogous to the recent libjingle change: http://cl/54929753-p10. This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather than scoped_array and scoped_ptr_malloc respectively. - Add Chromium's template-based COMPILE_ASSERT. We didn't have this previously in order to support the macro in C. Instead, move the existing macro to compile_assert_c.h. - Additionally copy the move.h and template_util.h depedencies and add the WARN_UNUSED_RESULT macro. - Leave scoped_array and scoped_ptr_malloc for now, but mark as deprecated. - Remove scoped_ptr foo(NULL) use. The default constructor handles it. - Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc. - Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove some repeated code. TESTED=trybots R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2449005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-22Add CurrentLayerId() to temporal layers.marpan@webrtc.org
same patch as: https://webrtc-codereview.appspot.com/2427004/ TBR=holmer@google.com Review URL: https://webrtc-codereview.appspot.com/2729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5012 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-04Revert r4913 that reverts r4911. Original CL description:andresp@webrtc.org
"Adding temporal layer strategy that keeps base layer framerate at an acceptable value." R=holmer@google.com, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2351006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4920 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03Revert 4911 "Adding temporal layer strategy that keeps base laye..."turaj@webrtc.org
> Adding temporal layer strategy that keeps base layer framerate at an acceptable value. > > R=stefan@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/2272005 TBR=andresp@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4913 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03Reformatting VPM: First step - No functional changes.mikhal@webrtc.org
R=marpan@google.com, marpan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2333004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4912 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03Adding temporal layer strategy that keeps base layer framerate at an ↵andresp@webrtc.org
acceptable value. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2272005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4911 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-30Propagate AutoMuter interface out to VideoCodingModulehenrik.lundin@webrtc.org
BUG=2436 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2311004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4878 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-28Change the parameters of calculating maximum decode time.wuchengli@chromium.org
- Reduce the window size from 20 to 10 seconds. If there is any spurious high decode time, it will be faster to pass it. - Ignore more samples at first because HW decoder has higher initialization latency. BUG=crbug.com/298176 TEST=Run apprtc loopback on Chromebook Daisy. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2315005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4874 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26Implemented AutoMuter in MediaOptimizationhenrik.lundin@webrtc.org
Also added a unittest. This is the first step towards creating an AutoMuter function in WebRTC. R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2294005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4857 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26Remove include_dirs from video_coding.pbos@webrtc.org
BUG=1662 TEST=compile on trybots R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2294007 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4853 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24Disable some VP8 tests on Android.andrew@webrtc.org
DecodeWithACompleteKeyFrame and FixedTemporalLayersStrategy. TBR=andresp BUG=2037 Review URL: https://webrtc-codereview.appspot.com/2283004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4829 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24MediaOptimization: Converting a few members to scoped_ptrshenrik.lundin@webrtc.org
For consistency with other parts of the code. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2275006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4822 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-23Reformatting media_optimization.cc and .hhenrik.lundin@webrtc.org
Ran both tools/refactoring/webrtc_reformat.py and clang-format. Changing VCMMediaOptimization -> MediaOptimization and VCMEncodedFrameSample -> EncodedFrameSample. Aligning the order of methods in .h and .cc files and fixing comments. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2265007 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4816 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-23Adding unit tests for default temporal layer strategy.andresp@webrtc.org
R=mflodman@webrtc.org, stefan@webrtc.org, stefan Review URL: https://webrtc-codereview.appspot.com/2235005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4810 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-20Avoid recursively taking critical section.stefan@webrtc.org
TEST=trybots BUG=2261 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2258006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4800 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-18Fix races in vcm::Process().stefan@webrtc.org
R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2241004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4775 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-17Fix typo in r4765.pbos@webrtc.org
Fixes compile error on all platforms. BUG= TEST=compile on tryboys R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2231004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4766 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-17Fix dangling pointer _encoder in video_sender.cc.pbos@webrtc.org
When _codecDataBase.SetSendCodec() fails, the encoder may be deleted. This is however not reflected in _encoder, which then becomes a dangling pointer to the deleted object. BUG=2384 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2229004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4765 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-16Split video coding module unit tests into sender and receiver unit tests.andresp@webrtc.org
R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2199005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4753 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-10 18:47:16 +0000