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2014-06-16Importing ThreadChecker class from Chromiumhenrik.lundin@webrtc.org
The ThreadChecker class is imported/re-implemented from Chromium. The implementation is changed to depend on WebRTC primitives. R=andrew@webrtc.org, henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6446 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16Adds aluebs@webrtc.org as owner to audio_processingbjornv@webrtc.org
BUG=N/A TESTED=trybots R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17709004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6445 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16bjornv@webrtc.org
We should avoid macros in general (see style guide) and the shift ones are particular dangerous since they assume that the user apply a non-negative shift. Related CL: https://webrtc-codereview.appspot.com/16669004 BUG=3348,3353 TESTED=trybots and manually on linux R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6444 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16Implements start bitrate for new video API.mflodman@webrtc.org
Added a new rampup test. BUG=2879 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6443 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16Add thread annotations to parts of ACMGenericCodechenrik.lundin@webrtc.org
This change adds annotations to all member variables that could be annotated without acquiring any new locks, or changing the lock structure in any other way. BUG=3041 R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18539004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6441 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16Add missing sources to webrtc/base/base.gypkjellander@webrtc.org
During my work on setting up a GN build for WebRTC, I discovered that the base.gyp is trying to remove source files (for the Chromium build) that are not added in the initial set. I assume these files should be listed and that GYP just doesn't complain when it's trying to remove a file that is not present in the sources list. natserver_main.cc is also removed, since it's not used anywhere. There are also a couple of other header files that are used in other code that probably also should be listed in base.gyp (please do this in another CL): * compile_assert.h * dscp.h * move.h * template_util.h BUG=None TEST=Trybots passing clobber compile step. R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13659004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6438 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-13Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc.glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20659004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6436 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-13Neon version of OverdriveAndSuppress()bjornv@webrtc.org
audioproc reports the average frame time going from 279us to 255us with the test data used. the output does not match the c version, but the difference seen is +-1. Performance gain on Nexus7: 8.8% BUG=3131 TESTED=trybots and manually R=bjornv@webrtc.org, cd@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19539004 Patch from Scott LaVarnway <slavarnw@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6433 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-13Pass GYP DEPTH variable to isolate.kjellander@webrtc.org
Similar change to https://codereview.chromium.org/322403003/ This will make it possible to handle different directory levels for special builds of WebRTC, without breaking GYP when the .isolate files are processed and their contents is verified. Also update all our .isolate files to use the <(DEPTH) variable. BUG=343106 TEST=Successful compile+test on Linux using: ninja -C out/Release tools/swarming_client/isolate.py run -s out/Release/tools_unittests.isolated Also trybots passing all tests. R=pbos@webrtc.org TBR=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13679004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6427 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12Revert 6415 "Update generated asm offsets scripts."wu@webrtc.org
> Update generated asm offsets scripts. > > This is the same CL as https://webrtc-codereview.appspot.com/16629004/ > Relanding and TBR from previous lgtm. > > Libvpx updated the unpack scripts to fix building dependencies. > > Roll libvpx 269083:275816 > See https://codereview.chromium.org/295313002/ > https://codereview.chromium.org/298063002/ > https://codereview.chromium.org/305533008/ > https://codereview.chromium.org/305703002/ > https://codereview.chromium.org/298383003/ > https://codereview.chromium.org/302863004/ > https://codereview.chromium.org/320923003/ > for the libvpx changes. > > See https://codereview.chromium.org/313243004/ > for the WebView changes. > > *NOTE* This CL will break the Android bots as they are built in a > Chromium checkout, which will pull in old libvpx DEPS. They will > cycle to green when we roll libvpx into Chromium. We must do the > rolls in this order becuase we have to land webrtc and libvpx at > the same time into Chromium. > > BUG=377062 > TBR=andrew@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/17689004 TBR=fgalligan@google.com Review URL: https://webrtc-codereview.appspot.com/13709004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6423 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12json.h include different header files depending on WEBRTC_CHROMIUM_BUILD ↵henrike@webrtc.org
being defined or not. Since json.h/cc is not even used in chromium it is the wrong flag to use. Instead add WEBRTC_EXTERNAL define. Also added OWNERS for base which is a copy of system_wrappers owners as the two folders are being merged. BUG=3379 R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15779004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6421 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12Enable pacing by default and remove the option to disable it from the new API.stefan@webrtc.org
BUG=1672 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17659004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6416 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12Update generated asm offsets scripts.fgalligan@google.com
This is the same CL as https://webrtc-codereview.appspot.com/16629004/ Relanding and TBR from previous lgtm. Libvpx updated the unpack scripts to fix building dependencies. Roll libvpx 269083:275816 See https://codereview.chromium.org/295313002/ https://codereview.chromium.org/298063002/ https://codereview.chromium.org/305533008/ https://codereview.chromium.org/305703002/ https://codereview.chromium.org/298383003/ https://codereview.chromium.org/302863004/ https://codereview.chromium.org/320923003/ for the libvpx changes. See https://codereview.chromium.org/313243004/ for the WebView changes. *NOTE* This CL will break the Android bots as they are built in a Chromium checkout, which will pull in old libvpx DEPS. They will cycle to green when we roll libvpx into Chromium. We must do the rolls in this order becuase we have to land webrtc and libvpx at the same time into Chromium. BUG=377062 TBR=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17689004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6415 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12Revert 6411 "Revert 6407 "Revert 6405 "Update generated asm offs..."kjellander@webrtc.org
Turns out the previous revert was based on invalid assumptions. The libvpx in Chromium was reverted in http://chromegw.corp.google.com/viewvc/chrome?view=rev&revision=271259 which ends up with libvpx r269083. Therefore we should restore that same libvpx revision for WebRTC, which this revert will do. > Revert 6407 "Revert 6405 "Update generated asm offsets scripts."" > > > Revert 6405 "Update generated asm offsets scripts." > > > > TBR=fgalligan@google.com > > BUG=N/A > > > > Review URL: https://webrtc-codereview.appspot.com/20639004 > > TBR=henrike@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/15739004 TBR=minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6413 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12Revert 6407 "Revert 6405 "Update generated asm offsets scripts.""minyue@webrtc.org
> Revert 6405 "Update generated asm offsets scripts." > > TBR=fgalligan@google.com > BUG=N/A > > Review URL: https://webrtc-codereview.appspot.com/20639004 TBR=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15739004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6411 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12Increased kMaxRampUpDelayMs (120 to 240s).asapersson@webrtc.org
Add support for triggering on encode rsd metric if its thresholds are configured. Added unit tests. BUG=1577 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6410 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12Revert 6405 "Update generated asm offsets scripts."henrike@webrtc.org
TBR=fgalligan@google.com BUG=N/A Review URL: https://webrtc-codereview.appspot.com/20639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6407 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11Update generated asm offsets scripts.fgalligan@google.com
Libvpx updated the unpack scripts to fix building dependencies. Roll libvpx 269083:275816 See https://codereview.chromium.org/295313002/ https://codereview.chromium.org/298063002/ https://codereview.chromium.org/305533008/ https://codereview.chromium.org/305703002/ https://codereview.chromium.org/298383003/ https://codereview.chromium.org/302863004/ https://codereview.chromium.org/320923003/ for the libvpx changes. See https://codereview.chromium.org/313243004/ for the WebView changes. BUG=377062 R=andrew@webrtc.org, michaelbai@chromium.org Review URL: https://webrtc-codereview.appspot.com/16629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6405 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11Re-land "Create a joint encoder/decoder wrapper for iSAC in ACM"henrik.lundin@webrtc.org
The change was reverted since it was thought to cause a flaky test. But the test kept flaking after the change was reverted. This effectively reverts r6394, relanding r6377. BUG=3496 TBR=minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6404 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11Reland: Making WebRTC able to play and record audio to files for tests.phoglund@webrtc.org
By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to play out audio to a file and feed audio in from a file. We want to do so we can better test WebRTC-using applications by recording what the audio stack outputs and feeding known audio in for quality tests. R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6403 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11Add APIs to enable padding with redundant payloads.stefan@webrtc.org
Also makes a small change to the tests to remove flakiness. We can't do BWE only based on rtp timestamps if we preemptively resend packets instead of sending padding packets. BUG=1812,2992 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6400 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11Revert 6395 "Making WebRTC able to play and record audio to file..."minyue@webrtc.org
> Making WebRTC able to play and record audio to files for tests. > > By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to > WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to > play out audio to a file and feed audio in from a file. We want to do > so we can better test WebRTC-using applications by recording what the > audio stack outputs and feeding known audio in for quality tests. > > R=henrika@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/20609004 TBR=phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6396 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11Making WebRTC able to play and record audio to files for tests.phoglund@webrtc.org
By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to play out audio to a file and feed audio in from a file. We want to do so we can better test WebRTC-using applications by recording what the audio stack outputs and feeding known audio in for quality tests. R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20609004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6395 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11Revert r6377 "Create a joint encoder/decoder wrapper for iSAC in ACM"henrik.lundin@webrtc.org
BUG=3469 TBR=minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16679004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6394 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11common_audio/signal_processing: Removes macro WEBRTC_SPL_RSHIFT_U16bjornv@webrtc.org
This macro is only used at a few places and implies a cast to uint16_t before right shifting. All places already use uint16_t. Further, the amount of shifts applied in the macro has no sanity check for negativity, makes the macro dangerous to use. BUG=3348,3353 TESTED=trybots and manually R=kwiberg@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16669004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6393 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11common_audio/signal_processing: Moves WEBRTC_SPL_UMUL_16_16_RSFT16 to iSAC fixbjornv@webrtc.org
This macro is only used by the fixed point version of iSAC. Replacing the (five) locations in arith_routines_logist.c, where it is used, with the actual operation. BUG=3348,3353 TESTED=trybots and manually R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14659004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6392 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11modules/audio_processing: Adds a config for reported delaysbjornv@webrtc.org
There are platforms and devices where the reported delays are untrusted and we currently solve that with an extended filter length and a slightly more conservative delay handling. With this change we give the user the possibility to turn off reported system delay values completely. - Includes new unit tests. TESTED=trybots and manual testing R=aluebs@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6391 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10Update makefiles after merge of Chromium at 276202Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I00be1885a20e1c8d4e5758fa281dca19d3ba4407
2014-06-10Delete last file in neteq4 folderhenrik.lundin@webrtc.org
The .isolate file can now be safely removed, since issue 3462 is resolved. BUG=2996 R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16659004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6388 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10MIPS optimizations for ISAC (patch #1)andrew@webrtc.org
Implemented functions: - WebRtcIsacfix_AutocorrMIPS - WebRtcIsacfix_FilterArLoop - WebRtcIsacfix_FilterMaLoopMIPS - WebRtcIsacfix_AllpassFilter2FixDec16MIPS (only MIPS DSP) - WebRtcIsacfix_PitchFilterCore (only MIPS DSPR2) Gain achieved: from aprox. 15% (MIPS32) up to aprox. 40% (MIPS DSPR2) R=andrew@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17559005 Patch from Ljubomir Papuga <lpapuga@mips.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6387 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10Noise suppression: Change signature to work on floats instead of intskwiberg@webrtc.org
Internally, it already worked on floats. This patch just changes the signature of a bunch of functions so that floats can be passed directly from the new and improved AudioBuffer without converting the data to int and back again first. (The reference data to the ApmTest.Process test had to be modified slightly; this is because the noise suppressor comes immediately after the echo canceller, which also works on floats. If I truncate to integers between the two steps, ApmTest.Process doesn't complain, but of course that's exactly the sort of thing the float conversion is supposed to let us avoid...) BUG= R=aluebs@webrtc.org, bjornv@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13519004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6385 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10Add additional metric (relative standard deviation of encode time) for ↵asapersson@webrtc.org
overuse detection. This code is currently only for testing. BUG=1577 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19619004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6381 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10Add kjellander@webrtc.org as OWNER for *.isolatekjellander@webrtc.org
This should make project-wide changes for isolate files easier and make it more obvious who's a suitable reviewer for them. BUG= R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19689004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6379 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at bdfcddf7091e92134143e9a2d9ccce908e43979e This commit was generated by merge_from_chromium.py. Change-Id: I3ad41d87e226143d8099f6e8c8c2907d059352e3
2014-06-09Create a joint encoder/decoder wrapper for iSAC in ACMhenrik.lundin@webrtc.org
This CL extends the ACMISAC wrapper class to inherit from AudioDecoder as well (the type of object that NetEq uses). The class has it's own lock protecting the iSAC instance. This way, we can remove the neteq_decode_lock_ (a.k.a. decoder_lock_) in a later CL. The old AcmAudioDecoderIsac class is deleted. R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6377 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09Add thread annotations to AcmReceiverhenrik.lundin@webrtc.org
This change adds thread annotations to AcmReceiver. These are the annotations that could be added without changing acquiring the locks in more locations, or changing the lock structure. BUG=3401 R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6376 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09Update makefiles after merge of Chromium at 275833Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: Iad0f8b40d3547d8d6337888a84071a951f8302d6
2014-06-09Make some methods in Clock class const declaredhenrik.lundin@webrtc.org
R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20579004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6375 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09Remove unused test_env.py from isolate files + fix nss path.kjellander@webrtc.org
This is not necessary for executing tests for WebRTC. It probably appeared in our .isolate files because of code copied from Chromium. BUG= TEST=All non-baremetal trybots passing. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6373 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09Merge from Chromium at DEPS revision 275586Torne (Richard Coles)
This commit was generated by merge_to_master.py. Change-Id: I8f394f473b73d82c4a79da05c2b85b0ad2a8beed
2014-06-09Enables DelayCorrection testsbjornv@webrtc.org
The fix has been done elsewhere and the test pass. BUG=3445 R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15679007 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6371 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09Updated conformance tests and w3c-ified them.phoglund@webrtc.org
I intend here to put these up for review on W3C. This moves the tests to use the W3C-style vendor prefix handling and updates the tests to the latest drafts. This yields 44 Pass 24 Fail and 13 pass 54 fail 1 timeout on Firefox. As far I can tell all failures are correct; in particular FF media media stream tracks do not adhere to the standard. Also I can't get FF to get a remote video up in the peerconnection test, just the local one. BUG=webrtc:3455 R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6370 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09Multi-threaded unit test for Audio Coding Module using iSAChenrik.lundin@webrtc.org
This test extends AudioCodingModuleTest and AudioCodingModuleMtTest to using iSAC as codec. R=kwiberg@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6369 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09audio_processing: Forces extended filter to be used in splitting filter test.bjornv@webrtc.org
The behavior differ between "normal" and "extended" modes when using AEC. In the extended filter mode nothing is processed until we have received a farend frame. This is exactly what is needed in this part of the splitting filter test. On Android, we do not use the normal mode, which made the test to fail. BUG=3445 R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12679004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6368 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09Rename neteq4 folder to neteqhenrik.lundin@webrtc.org
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for a hard-coded reference to change. This CL effectively reverts r6257 "Rename neteq4 folder to neteq". BUG=2996 TBR=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-08Re-enable AudioCodingModuleMtTest againhenrik.lundin@webrtc.org
Increase timeout and decrease test length. BUG=3426 R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15679006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6365 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-07Update makefiles after merge of Chromium at 275661Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I9f8d417e25bac16cf9f0cea277d28da37190aab2
2014-06-06Revert r6358 "AppRTCDemo(Android): only stop the cameraThread's looper after ↵fischman@webrtc.org
stopping the camera." Makes stopping flakier for some reason :/ BUG= R=glaznev@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21609004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6361 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06Use XErrorTrap in MouseCursorMonitorX11 to catch the error if the shared ↵jiayl@webrtc.org
window has been closed. BUG=crbug/374457 R=sergeyu@chromium.org Review URL: https://webrtc-codereview.appspot.com/13599004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6359 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06AppRTCDemo(Android): only stop the cameraThread's looper after stopping the ↵fischman@webrtc.org
camera. R=glaznev@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6358 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-11 02:13:41 +0000