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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
2014-07-24Make sure padding is sent on the first sending RTP module.mflodman@webrtc.org
R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13989004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6774 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-22The lastest commit on this file was inminyue@webrtc.org
https://webrtc-codereview.appspot.com/15529004/ The final patch set should have included this, but was missed. R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18839004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6755 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-22Remove remains of WEBRTC_NO_STL.andresp@webrtc.org
R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6752 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21MIPS optimizations for ISAC (patch #2)andrew@webrtc.org
Implemented functions: - WebRtcIsacfix_CalculateResidualEnergy - WebRtcIsacfix_Spec2Time - WebRtcIsacfix_Time2Spec - WebRtcIsacfix_HighpassFilterFixDec32 - WebRtcIsacfix_PCorr2Q32 Gain achieved: aprox. further 5% on top of patch#1 on ISAC encoding path. The optimizations are bit-exact to the C code, with the excception of the MIPS DSPr2 variant of the WebRtcIsacfix_Time2Spec function (the accuracy of the WebRtcIsacfix_Time2Spec MIPS DSPr2 variant is same or better than C variant). Code verification and improvement achieved have been determined using the iSACFixtest application. R=andrew@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19749004 Patch from Ljubomir Papuga <lpapuga@mips.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6749 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18This is to re-open an earlier CLminyue@webrtc.org
https://webrtc-codereview.appspot.com/16619005/ which is reverted due to an issue in audio conference mixer. This issue has been solved in https://webrtc-codereview.appspot.com/20779004/ BUG=webrtc:3155 R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18819005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6736 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18Runtime guard for iOS7 property.tkchin@webrtc.org
BUG=3487 R=glaznev@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19989004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6733 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18Fix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS.tkchin@webrtc.org
BUG=3581 R=glaznev@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16109004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6732 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18This is related to an earlier CL of enabling Opus 48 kHz.minyue@webrtc.org
https://webrtc-codereview.appspot.com/16619005/ It was reverted due to a build bot error, which this CL is to fix. The problem was that when audio conference mixer receives audio frames all at 48 kHz and mixed them, it uses Audio Processing Module (APM) to do a post-processing. However the APM cannot handle 48 kHz input. The current solution is not to allow the mixer to output 48 kHz. TEST=locally solved https://webrtc-codereview.appspot.com/16619005/ BUG= R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20779004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6730 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->floatkwiberg@webrtc.org
Specifically, when someone asks for a const pointer to the int16 version of the array, there's no need to invalidate the float version of that array, and vice versa. (But obviously, invalidation still has to happen when someone asks for a non-const pointer.) R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6725 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17Reduce runtime of RingBufferTest by a factor of 100.andrew@webrtc.org
This test was needlessly long. TBR=pbos Review URL: https://webrtc-codereview.appspot.com/15029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6724 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17Use _numMixedParticipants instead of audioFrameList->size() to determine if ↵wu@webrtc.org
there're more than one participants. There are two audioFrameLists. The previous check wouldn't work correctly if each list had a single member. TEST=chrome https://apprtc.appspot.com/?debug=loopback&video=false and verify e2e delay stats BUG= R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15019004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6723 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17Fix issue where padding is sent before media with undefined timestamps if ↵stefan@webrtc.org
not abs-send-time is enabled. This broke bandwidth estimation for calls without abs-send-time is enabled, but where RTX was. BUG= R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16929004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6719 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17Remove unused ExperimentalNS API in AudioProcessingaluebs@webrtc.org
R=andrew@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6718 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17AudioBuffer: Eliminate the SplitChannelBuffer classkwiberg@webrtc.org
It's just a container for two IFChannelBuffers, and doesn't earn its keep. The main problem is that the number of methods it needs that just forward calls to either of its two IFChannelBuffers was already large, and was about to grow. R=aluebs@webrtc.org, minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6717 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17Simplify AudioBuffer::mixed_low_pass_data APIaluebs@webrtc.org
R=andrew@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6715 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameterkwiberg@webrtc.org
R=aluebs@webrtc.org, minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13019004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6714 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17Add unit test for MediaFile WAV file writingkwiberg@webrtc.org
R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6713 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17Fixes up rtc so that it compiles on iOS 8 SDK.tkchin@webrtc.org
Adds support for UIInterfaceOrientationUnknown (new with in SDK) and makes it the same as UIInterfaceOrientationPortrait. R=noahric@google.com, tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13029004 Patch from David Maclachlan <dmaclach@chromium.org>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6712 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16r6709 lacks a change in BUILD.gnminyue@webrtc.org
BUG= R=marpan@google.com, marpan@webrtc.org, pbos@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6710 4adac7df-926f-26a2-2b94-8c16560cd09d
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
2014-07-16Compile-time guard for iOS7 specific property.tkchin@webrtc.org
BUG=3487 R=glaznev@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17969004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6706 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
2014-07-16Remove old padding path in RTPSender.pbos@webrtc.org
Removing RTPSender::SendPaddingAccordingToBitrate() as well as a couple of arguments from SendPadData(). BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14989004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6703 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16nrsh1 is written before tmp321 is read, so needs to be earlyclobberkwiberg@webrtc.org
Otherwise, the compiler is allowed to put them in the same register under the assumption that all inputs are read before any (non-earlyclobber) output is written, which in this case would result in nrsh2 being corrupted. BUG=3439 R=aluebs@webrtc.org, ljubomir.papuga@gmail.com Review URL: https://webrtc-codereview.appspot.com/16089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6700 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15Fix an invalid memory access due to typo in win/cursor.cc.jiayl@webrtc.org
BUG=crbug/391468 R=sergeyu@chromium.org Review URL: https://webrtc-codereview.appspot.com/19949004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6698 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15After an audio interruption the audio unit no longer invokes its render ↵tkchin@webrtc.org
callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue. CL also replaces deprecated AudioSession calls with equivalent AVAudioSession ones. BUG=3487 R=glaznev@webrtc.org, noahric@chromium.org Review URL: https://webrtc-codereview.appspot.com/21769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6697 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15Improvements to the pacer where it lost some budget due to truncation errors.stefan@webrtc.org
With this CL the resolution is increased to microseconds and proper rounding is done in the Process() function. This means that we will be allowed to send more than prior to r6664 as we previously truncated away parts of our budget. We will also not lose budget due to inaccurate calculations in TimeUntilNextProcess(), which was a regression in r6664. BUG=cr/393950 TEST=out/Debug/webrtc_perf_tests --gtest_filter=RampUpTest.Simulcast R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20949004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6694 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15Fix breakage introduced by r6691.pbos@webrtc.org
ModuleRtpRtcpImpl returned incorrectly on RemoteNTP as the RTCPReceiver::NTP changed return type. BUG= TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18799004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6693 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15Make RTCP sender report send media bytes.pbos@webrtc.org
r6654 changed RtpSender::Bytes() to return the number of bytes sent instead of number of media bytes. This is used by VideoEngine for stats. This change broke RTCP which sends this same count as the number of payload bytes sent (excluding headers and padding). BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6691 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
2014-07-14Fix deadlock in Android stopCapture() call.glaznev@webrtc.org
BUG=3467 R=braveyao@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18789004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6673 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-13GN: Fix include paths for WebRTC in Chromium build.kjellander@webrtc.org
Most WebRTC source files are using full paths for includes which requires the root to be in the include path. This is currently handled in the common_inherited_config config in webrtc/BUILD.gn: the .. include_dir. However, when built from Chromium, the include paths are not inherited in the same way when building the all target. Building the 'webrtc' target of Chrome works without the changes in this CL, but the default target fails. BUG=3441 TEST=Built the default target from a Chromium checkout with https://codereview.chromium.org/321313006/ applied and src/third_party/webrtc linked to the webrtc folder of the WebRTC workspace. R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15989004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6670 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 .tommi@webrtc.org
A few places were relying on temporalIdx being signed. Fix to explicitly check for kNoTemporalIdx. TBR=pbos,stefan Review URL: https://webrtc-codereview.appspot.com/13939005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6669 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11Remove always-true expression.tommi@webrtc.org
TBR=pbos Review URL: https://webrtc-codereview.appspot.com/16059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6668 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
2014-07-11Thread annotate RTCPSender.pbos@webrtc.org
Also fixes data races in RTCPSender::SetCSRCStatus() and RTCPSender::SetStartTimestamp(). BUG= R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6666 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
2014-07-11Document that channels are stored contiguously in AudioBufferaluebs@webrtc.org
R=andrew@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6661 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11Remove unnecessary build message.tommi@webrtc.org
R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6660 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
2014-07-10Count total bytes sent in RTPSender::Bytes().pbos@webrtc.org
Previously only media bytes were included, this adds header bytes and padding bytes to the calculation. BUG= R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19939004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6654 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10Fix data race in VCMTiming::ResetDecodeTime.pbos@webrtc.org
Also thread annotating class. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17939004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6653 4adac7df-926f-26a2-2b94-8c16560cd09d
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
2014-07-10audio_processing: Updates aec_core_sse2.c with changes made to aec_common.hbjornv@webrtc.org
The change of definitions moved to aec_common.h was done in CL17839005. BUG=3131 TBR=kwiberg@webrtc.org TESTED=builds locally Review URL: https://webrtc-codereview.appspot.com/16859004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6648 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10Neon version of SubbandCoherence()bjornv@webrtc.org
The performance gain on a Nexus 7 reported by audioproc is ~1.4% The output is NOT bit exact. Any difference seen is +-1. BUG=3131 R=bjornv@webrtc.org, cd@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17839005 Patch from Scott LaVarnway <slavarnw@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6647 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10Neon version of rftbsub_128()bjornv@webrtc.org
The performance gain on a Nexus 7 reported by audioproc is ~4.5% The output is bit exact. BUG=3131 TESTED=trybots and manually R=bjornv@webrtc.org, cd@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19919005 Patch from Scott LaVarnway <slavarnw@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6646 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10Revert "Remove remains of WEBRTC_NO_STL." (rev 6641).andresp@webrtc.org
Reason breaks linux_memcheck. TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6645 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-05 23:52:55 +0000