Svmp 2.x#1

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Albertontheway wants to merge 656 commits intosvmp-1.4.xfrom

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Commits on May 30, 2014

Fixing a bug regarding VOE packet loss rate feedback to ACM
minyue@webrtc.org
committed
Fix bug where RTP headers in the packet history were replaced with the RTX wrapped headers.
stefan@webrtc.org
committed
Add native_test dependency to webrtc_perf_tests.
pbos@webrtc.org
committed
Multi-threaded test for Audio Coding Module
henrik.lundin@webrtc.org
committed
(Auto)update libjingle 68230113-> 68244456
buildbot@webrtc.org
committed
Disable AudioCodingModuleMtTest due to memcheck and tsan failures.
andrew@webrtc.org
committed
Add a Reset() method to AudioFrame.
andrew@webrtc.org
committed
Fix two bugs in DataChannel state transition.
jiayl@webrtc.org
committed
Implement mac version of AppRTCDemo.
tkchin@webrtc.org
committed
Fix AppRTC target configuration in libjingle_examples.gyp.
tkchin@webrtc.org
committed
Add a TSAN suppression for a benign TRACE_EVENT race.
andrew@webrtc.org
committed
Make OpenSSLStreamAdapter verify the leaf certificate digest for chained certificates.
jiayl@webrtc.org
committed
VideoCaptureAndroid: don't synchronized on camera thread.
fischman@webrtc.org
committed
Fix the build error from OpenSSLStreamAdapter::SSLVerifyCallback
jiayl@webrtc.org
committed

Commits on Jun 2, 2014

Enable -Wall, -Wextra and -Wunused-variable for talk/ on clang.
pbos@webrtc.org
committed
Re-enables CommonFormats test for Android.
bjornv@webrtc.org
committed
Adding thread annotations to parts of Audio Coding Module
henrik.lundin@webrtc.org
committed
Disable MouseCursorMonitorTest.FromScreen
henrik.lundin@webrtc.org
committed
Disable MouseCursorMonitorTest
henrik.lundin@webrtc.org
committed
Remove old perf_expectations no longer used.
kjellander@webrtc.org
committed
- Get rid of 'using' from .h
solenberg@webrtc.org
committed
(Auto)update libjingle 68275107-> 68379861
buildbot@webrtc.org
committed

Commits on Jun 3, 2014

Remove an optimization that's no longer worth the extra complexity it causes
kwiberg@webrtc.org
committed
Remove kMaxWaitForStatsMs from tsanv2 compilation.
pbos@webrtc.org
committed
(Auto)update libjingle 68379861-> 68445177
buildbot@webrtc.org
committed
Add kwiberg@webrtc.org to watchlist for audio_coding and audio_processing
kwiberg@webrtc.org
committed
Reformat integer accessors to look like their float counterparts
kwiberg@webrtc.org
committed
Re-enable AudioCodingModuleMtTest
henrik.lundin@webrtc.org
committed
Add empty webrtcmediaengine.cc.
pbos@webrtc.org
committed
Revert 6312 "Re-enable AudioCodingModuleMtTest"
turaj@webrtc.org
committed
PeerConnection(java): disable wait for flaky ICEConnection.COMPLETED.
fischman@webrtc.org
committed
Enable videoprocessor_integrationtest tests on android.
marpan@webrtc.org
committed
(Auto)update libjingle 68465410-> 68487517
buildbot@webrtc.org
committed
talk/ios: Fixes source after corrupt sync in r6305 (which corrupted r6291).
henrike@webrtc.org
committed
(Auto)update libjingle 68495561-> 68499439
buildbot@webrtc.org
committed

Commits on Jun 4, 2014

(Auto)update libjingle 68499439-> 68501302
buildbot@webrtc.org
committed
(Auto)update libjingle 68501302-> 68506654
buildbot@webrtc.org
committed
Adding missing break in media_file_utility.cc.
mflodman@webrtc.org
committed
Suppress memcheck error in VideoProcessorIntegrationTest
henrik.lundin@webrtc.org
committed
Moved verbose logging in rtcp_receiver.cc to LS_VERBOSE.
andresp@webrtc.org
committed
Disables some modules_unittests on Android.
bjornv@webrtc.org
committed
Disables SystemDelayTest.CorrectDelayDuringDrift on Android
bjornv@webrtc.org
committed
Disable P2PTransportChannelMultihomedTest.TestFailover under Memcheck
henrik.lundin@webrtc.org
committed
(Auto)update libjingle 68506654-> 68507189
buildbot@webrtc.org
committed
(Auto)update libjingle 68507189-> 68543735
buildbot@webrtc.org
committed
Make it possible to build webrtc for arm64.
solenberg@webrtc.org
committed
Fix retain cycle in RTCEAGLVideoView.
tkchin@webrtc.org
committed
Android: cleanup gtest_target_type conditions.
henrike@webrtc.org
committed
(Auto)update libjingle 68562943-> 68571194
buildbot@webrtc.org
committed

Commits on Jun 5, 2014

Disable a test in libjingle_peerconnection_unittest for DrMemory
henrik.lundin@webrtc.org
committed
Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC.
stefan@webrtc.org
committed
common_audio/signal_processing: Removed macro WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND
bjornv@webrtc.org
committed
The correct fix of workaround in r6261.
bjornv@webrtc.org
committed
Remove static initializer from WebRtcVideoEngine2.
pbos@webrtc.org
committed
Adding back platform specific renderer to video loopback test.
mflodman@webrtc.org
committed
NetEq: Add thread annotation to const scoped_ptrs
henrik.lundin@webrtc.org
committed
Fixes and enables SystemDelayTests.
bjornv@webrtc.org
committed
Remove remaining samples (AppRTC) since moved to Github
kjellander@webrtc.org
committed
Revert 6341 "Fixes and enables SystemDelayTests."
bjornv@webrtc.org
committed
Opus send rate overflows if over 65 kbps
tina.legrand@webrtc.org
committed
AppRTCDemo(android): remove HTML/regex hackery in favor of JSON struct.
fischman@webrtc.org
committed
Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
wu@webrtc.org
committed
Rebase webrtc/base with r6345 version of talk/base:
henrike@webrtc.org
committed
(Auto)update libjingle 68646004-> 68648993
buildbot@webrtc.org
committed

Commits on Jun 6, 2014

Make VideoSendStream/VideoReceiveStream configs const.
pbos@webrtc.org
committed
(Auto)update libjingle 68689052-> 68689059
buildbot@webrtc.org
committed
Fix C++11 -Wnarrowing in channel_unittest.cc.
pbos@webrtc.org
committed
(Auto)update libjingle 68701339-> 68703656
buildbot@webrtc.org
committed
VideoCaptureImpl::IncomingFrame(): avoid deadlock by acquiring _apiCs.
fischman@webrtc.org
committed
AppRTCDemo(android): support app (UI) & capture rotation.
fischman@webrtc.org
committed
ViEAutoTestAndroid: Unbreak compile by casting void* to jobject.
fischman@webrtc.org
committed
Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio.
jiayl@webrtc.org
committed
Unbreak NDEBUG compile by RTC_UNUSED()ing an assert()d variable.
fischman@webrtc.org
committed
AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera.
fischman@webrtc.org
committed
Use XErrorTrap in MouseCursorMonitorX11 to catch the error if the shared window has been closed.
jiayl@webrtc.org
committed
Add OpenGL Android video renderer which can display multiple
glaznev@webrtc.org
committed
Revert r6358 "AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera."
fischman@webrtc.org
committed
APPRTCDemo(objc): Remove regex parsing in favor of JSON struct.
tkchin@webrtc.org
committed
Revert "Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio."
jiayl@webrtc.org
committed

Commits on Jun 8, 2014

PRESUBMIT: Add Android ARM64 and remove Linux TSan
kjellander@webrtc.org
committed
Re-enable AudioCodingModuleMtTest again
henrik.lundin@webrtc.org
committed

Commits on Jun 9, 2014

Fix Dr Memory download
kjellander@webrtc.org
committed
Rename neteq4 folder to neteq
henrik.lundin@webrtc.org
committed
audio_processing: Forces extended filter to be used in splitting filter test.
bjornv@webrtc.org
committed
Multi-threaded unit test for Audio Coding Module using iSAC
henrik.lundin@webrtc.org
committed
Updated conformance tests and w3c-ified them.
phoglund@webrtc.org
committed
Enables DelayCorrection tests
bjornv@webrtc.org
committed
Adds support for the "apt" format parameter and turns on the RTX feature.
stefan@webrtc.org
committed
Remove unused test_env.py from isolate files + fix nss path.
kjellander@webrtc.org
committed
Make some methods in Clock class const declared
henrik.lundin@webrtc.org
committed
Add thread annotations to AcmReceiver
henrik.lundin@webrtc.org
committed
Create a joint encoder/decoder wrapper for iSAC in ACM
henrik.lundin@webrtc.org
committed

Commits on Jun 10, 2014

Add kjellander@webrtc.org as OWNER for *.isolate
kjellander@webrtc.org
committed
Replace libjingle_root with talk_root variable.
kjellander@webrtc.org
committed
Add additional metric (relative standard deviation of encode time) for overuse detection.
asapersson@webrtc.org
committed
Move WebRtcVideoEngine2 fakes to unittest header.
pbos@webrtc.org
committed
(Auto)update libjingle 68891947-> 68893961
buildbot@webrtc.org
committed
Revert 6380 "Replace libjingle_root with talk_root variable."
kjellander@webrtc.org
committed
Noise suppression: Change signature to work on floats instead of ints
kwiberg@webrtc.org
committed
Adding the new video folder and pacer to the wathclist.
mflodman@webrtc.org
committed
MIPS optimizations for ISAC (patch #1)
andrew@webrtc.org
committed
Delete last file in neteq4 folder
henrik.lundin@webrtc.org
committed
Add support for NVidia VP8 HW encoder.
glaznev@webrtc.org
committed
Adds end to end DataChannel tests.
jiayl@webrtc.org
committed

Commits on Jun 11, 2014

modules/audio_processing: Adds a config for reported delays
bjornv@webrtc.org
committed
common_audio/signal_processing: Moves WEBRTC_SPL_UMUL_16_16_RSFT16 to iSAC fix
bjornv@webrtc.org
committed
common_audio/signal_processing: Removes macro WEBRTC_SPL_RSHIFT_U16
bjornv@webrtc.org
committed
Revert r6377 "Create a joint encoder/decoder wrapper for iSAC in ACM"
henrik.lundin@webrtc.org
committed
Making WebRTC able to play and record audio to files for tests.
phoglund@webrtc.org
committed
Revert 6395 "Making WebRTC able to play and record audio to file..."
minyue@webrtc.org
committed
(Auto)update libjingle 68949184-> 68982444
buildbot@webrtc.org
committed
Revert 6397 "(Auto)update libjingle 68949184-> 68982444"
minyue@webrtc.org
committed
(Auto)update libjingle 68982444-> 68983526
buildbot@webrtc.org
committed
Add APIs to enable padding with redundant payloads.
stefan@webrtc.org
committed
Re-land webrtcmediaengine.cc part of r6397.
pbos@webrtc.org
committed
Remove unused resource
henrik.lundin@webrtc.org
committed
Reland: Making WebRTC able to play and record audio to files for tests.
phoglund@webrtc.org
committed
Re-land "Create a joint encoder/decoder wrapper for iSAC in ACM"
henrik.lundin@webrtc.org
committed
Update generated asm offsets scripts.
fgalligan@google.com
committed
(Auto)update libjingle 68985065-> 69005149
buildbot@webrtc.org
committed

Commits on Jun 12, 2014

Revert 6405 "Update generated asm offsets scripts."
henrike@webrtc.org
committed
(Auto)update libjingle 69005149-> 69049090
buildbot@webrtc.org
committed
Disable flaky test on DrMemory Full.
pbos@webrtc.org
committed
Increased kMaxRampUpDelayMs (120 to 240s).
asapersson@webrtc.org
committed
Revert 6407 "Revert 6405 "Update generated asm offsets scripts.""
minyue@webrtc.org
committed
(Auto)update libjingle 69049090-> 69054765
buildbot@webrtc.org
committed
Revert 6411 "Revert 6407 "Revert 6405 "Update generated asm offs..."
kjellander@webrtc.org
committed
Fixed GetStats when local and remote track are using the same ssrc.
xians@webrtc.org
committed
Update generated asm offsets scripts.
fgalligan@google.com
committed
Enable pacing by default and remove the option to disable it from the new API.
stefan@webrtc.org
committed
(Auto)update libjingle 69069003-> 69082899
buildbot@webrtc.org
committed
Fix the flaky RTP DataChannel test.
jiayl@webrtc.org
committed
(Auto)update libjingle 69097619-> 69099564
buildbot@webrtc.org
committed
Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck
jiayl@webrtc.org
committed
json.h include different header files depending on WEBRTC_CHROMIUM_BUILD 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 …
henrike@webrtc.org
committed
Revert r6420 'Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck'
jiayl@webrtc.org
committed
Revert 6415 "Update generated asm offsets scripts."
wu@webrtc.org
committed

Commits on Jun 13, 2014

(Auto)update libjingle 69102234-> 69116997
buildbot@webrtc.org
committed
Initial owners file for talk/media/webrtc/.
pbos@webrtc.org
committed
(Auto)update libjingle 69131548-> 69132244
buildbot@webrtc.org
committed
Pass GYP DEPTH variable to isolate.
kjellander@webrtc.org
committed
Fix GYP DEPTH for libjingle isolate files
kjellander@webrtc.org
committed
Remove engine_codecs_ cache from unittests.
pbos@webrtc.org
committed
Implement RTX tests+fixes in WebRtcVideoEngine2.
pbos@webrtc.org
committed
Add NACK feedback parameter to WebRtcVideoEngine2.
pbos@webrtc.org
committed
(Auto)update libjingle 69143161-> 69144530
buildbot@webrtc.org
committed
Neon version of OverdriveAndSuppress()
bjornv@webrtc.org
committed
(Auto)update libjingle 69144530-> 69164179
buildbot@webrtc.org
committed
Add extra logging and latency restriction to VP8 HW encoder.
glaznev@webrtc.org
committed
Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc.
glaznev@webrtc.org
committed

Commits on Jun 16, 2014

(Auto)update libjingle 69188577-> 69260070
buildbot@webrtc.org
committed
Add missing sources to webrtc/base/base.gyp
kjellander@webrtc.org
committed
(Auto)update libjingle 69260070-> 69276003
buildbot@webrtc.org
committed
Disable flaky test (WebRtcVideoMediaChannelTest.GetStats) on DrMemory Full.
asapersson@webrtc.org
committed
Add thread annotations to parts of ACMGenericCodec
henrik.lundin@webrtc.org
committed
(Auto)update libjingle 69276003-> 69278008
buildbot@webrtc.org
committed
Implements start bitrate for new video API.
mflodman@webrtc.org
committed
common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16
bjornv@webrtc.org
committed
Adds aluebs@webrtc.org as owner to audio_processing
bjornv@webrtc.org
committed
Importing ThreadChecker class from Chromium
henrik.lundin@webrtc.org
committed
Remove ivinnichenko from webrtc/test/OWNERS
kjellander@webrtc.org
committed
(Auto)update libjingle 69278008-> 69291002
buildbot@webrtc.org
committed
Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class.
asapersson@webrtc.org
committed
(Auto)update libjingle 69291002-> 69292418
buildbot@webrtc.org
committed
Add max limit of number for overuses. When limit is reached always apply the rampup delay.
asapersson@webrtc.org
committed
(Auto)update libjingle 69292418-> 69293749
buildbot@webrtc.org
committed
Implement RTP extension support in WebRtcVideoEngine2.
pbos@webrtc.org
committed
(Auto)update libjingle 69306183-> 69323802
buildbot@webrtc.org
committed
Makes it possible to prevent some third party libraries (jsoncpp and openssl) from being linked. This makes it possible to link webrtc with external implementations of those libraries in case the p…
henrike@webrtc.org
committed

Commits on Jun 17, 2014

(Auto)update libjingle 69337301-> 69359922
buildbot@webrtc.org
committed
Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded …
minyue@webrtc.org
committed
Updated W3C getusermedia tests to the latest version of the spec.
phoglund@webrtc.org
committed
Restore ptypes.txt file
henrik.lundin@webrtc.org
committed
Initial GN work for WebRTC
kjellander@webrtc.org
committed
Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..."
minyue@webrtc.org
committed
(Auto)update libjingle 69359922-> 69365993
buildbot@webrtc.org
committed
Increasing tolerances quite a bit to fight flakes.
phoglund@webrtc.org
committed
Rolling new version of opus.gyp
tina.legrand@webrtc.org
committed
Rebase webrtc/base with r6464 version of talk/base:
henrike@webrtc.org
committed
base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/
henrike@webrtc.org
committed
Makes the sid of a closed DataChannel available to reuse per the spec.
jiayl@webrtc.org
committed
Add a check to all.gyp to respect the include_tests variable.
andrew@webrtc.org
committed
Add high perf mode to VP8
niklas.enbom@webrtc.org
committed
Add round-robin selection of send stream to pad on.
stefan@webrtc.org
committed
Update generated asm offsets scripts.
fgalligan@google.com
committed
Update usrsctp to r8875
jiayl@webrtc.org
committed
Revert 6473 "Update generated asm offsets scripts."
turaj@webrtc.org
committed

Commits on Jun 18, 2014

Revert "Restore ptypes.txt file"
henrik.lundin@webrtc.org
committed
Update PacketSource and RtpFileSource
henrik.lundin@webrtc.org
committed
Neon version of FilterAdaptation()
bjornv@webrtc.org
committed
Update generated asm offsets scripts.
fgalligan@google.com
committed
Update webrtc to fix unpack_lib expansion.
fgalligan@google.com
committed
Properly shut down the SCTP stack.
jiayl@webrtc.org
committed
Exclude SctpDataMediaChannelTest on Win DrMemory for third_party/usrsctp issues.
jiayl@webrtc.org
committed
Fix a memory leak in SctpDataMediaChannelTest.
jiayl@webrtc.org
committed

Commits on Jun 19, 2014

Exclude flaky test PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate on memcheck.
jiayl@webrtc.org
committed
(Auto)update libjingle 69506154-> 69515138
buildbot@webrtc.org
committed
Fixes and re-enables tests disabled on Android
bjornv@webrtc.org
committed
Adding test::AudioSink interface and derived classes
henrik.lundin@webrtc.org
committed
Ensure that the start bitrate can be set multiple times.
stefan@webrtc.org
committed
Changes to tests and tools in audio_processing.
bjornv@webrtc.org
committed
Adding an empty constructor implementation to the AudioSink class
henrik.lundin@webrtc.org
committed
Maintain constantness of the input to iSAC-fix decoder, and prevent heap-buffer overflow.
turaj@webrtc.org
committed
Revert 6481 and 6482
fgalligan@google.com
committed
(Auto)update libjingle 69543894-> 69555283
buildbot@webrtc.org
committed
(Auto)update libjingle 69555283-> 69567902
buildbot@webrtc.org
committed
(Auto)update libjingle 69567902-> 69568113
buildbot@webrtc.org
committed
Do not call CaptureCursor in ScreenCapturerWinGdi if no MouseShapeObserver.
jiayl@webrtc.org
committed
(Auto)update libjingle 69568113-> 69587333
buildbot@webrtc.org
committed

Commits on Jun 20, 2014

(Auto)update libjingle 69587333-> 69588608
buildbot@webrtc.org
committed
(Auto)update libjingle 69588608-> 69588980
buildbot@webrtc.org
committed
(Auto)update libjingle 69588980-> 69589535
buildbot@webrtc.org
committed
(Auto)update libjingle 69589535-> 69600065
buildbot@webrtc.org
committed
Remove unused ALLOCATE_DELAY constant.
pbos@webrtc.org
committed
Add tests of texture frames in video_send_stream_test.
wuchengli@chromium.org
committed
(Auto)update libjingle 69600065-> 69617317
buildbot@webrtc.org
committed
(Auto)update libjingle 69617317-> 69623266
buildbot@webrtc.org
committed
Do not hold the critical section in VideoCaptureAndroid::SetCaptureRotation since it would case possible deadlock with OS Camear thread.
braveyao@webrtc.org
committed
Re-evalutes the ICE role on ICE restart.
jiayl@webrtc.org
committed
Limits the send and receive buffer by bytes, not by packets.
jiayl@webrtc.org
committed
(Auto)update libjingle 69634309-> 69640360
buildbot@webrtc.org
committed
- Exit from a camera thread lopper loop() method only after all camera release calls are completed. This fixes camera exceptions observed from time to time when calling camera functions on a termin…
glaznev@webrtc.org
committed
When creating an answer, takes the codec preference from the offer.
wu@webrtc.org
committed

Commits on Jun 21, 2014

GN: BUILD.gn for system_wrappers
kjellander@webrtc.org
committed
Roll chromium_revision 272489:277350 + fix sanitizer options
kjellander@webrtc.org
committed
Add third_party/colorama to DEPS
kjellander@webrtc.org
committed

Commits on Jun 22, 2014

Exclude flaky libjingle_peerconnection_unittest test for Memcheck.
kjellander@webrtc.org
committed
TSan v2 deadlock suppressions.
kjellander@webrtc.org
committed

Commits on Jun 23, 2014

Disables tests that breaks Android bots
bjornv@webrtc.org
committed
Roll libvpx 269083:278497
fgalligan@google.com
committed
Rebase webrtc/base with r6521 version of talk/base:
henrike@webrtc.org
committed
GN: Add BUILD.gn files + kjellander to OWNERS
kjellander@webrtc.org
committed
Annotating the rest of AudioCodingModuleImpl
henrik.lundin@webrtc.org
committed
Fix array declarations in aec_core.c
andrew@webrtc.org
committed

Commits on Jun 24, 2014

Annotating the rest of AcmGenericCodec
henrik.lundin@webrtc.org
committed
(Auto)update libjingle 69648312-> 69830415
buildbot@webrtc.org
committed
Disable PhysicalSocketTest.TestUdpReadyToSendIPv4 for TSAN2
aluebs@webrtc.org
committed
Neon version of ScaleErrorSignal()
bjornv@webrtc.org
committed
Exclude AsyncWriteTest.TestWrite from Win DrMemory Full bot and suppress the reported errors
aluebs@webrtc.org
committed
Removing neteq decode lock and friends
henrik.lundin@webrtc.org
committed
talk/base and webrtc/base suppression had the same names for their suppressions which is not allowed. Renamed the talk/base ones as they are going away.
henrike@webrtc.org
committed
Change SdpSerializeCandidate to output candidate line without the "a=" and without the leading \r\n", i.e. candidate-attribute as defined in section 15.1 of [ICE].
wu@webrtc.org
committed

Commits on Jun 25, 2014

Remove payload duplication in AudioDecoderTest
henrik.lundin@webrtc.org
committed
Neon version of FilterFar()
bjornv@webrtc.org
committed
This is to compare NetEq with various codecs under a shared packet loss pattern.
minyue@webrtc.org
committed
Add RTCP packet types to packet builder:
asapersson@webrtc.org
committed
pkg-config-wrapper should not be run when build_nss is disabled (=0).
henrike@webrtc.org
committed
fix after r6472 in rtp_sender, comparison between signed and unsigned integer expressions.
henrike@webrtc.org
committed
Bump version number to 3.55
tnakamura@webrtc.org
committed

Commits on Jul 8, 2014

Merge upstream version 3.55
dkeppler
committed