Log - cb4fdd1fdaaf2c11d40d3aeb3d5f62127230a6cc - fp2-dev/platform/external/chromium_org/third_party/webrtc

cb4fdd1 Update makefiles after merge of Chromium at 277428 by Android Chromium Automerger · 10 years ago
c7fcada Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 5fcef2b6df45ceab39ee96a616ab0a4d3c63b83a by Android Chromium Automerger · 10 years ago
5fcef2b Revert 6411 "Revert 6407 "Revert 6405 "Update generated asm offs..." by kjellander@webrtc.org · 10 years ago
4150d6e Revert 6407 "Revert 6405 "Update generated asm offsets scripts."" by minyue@webrtc.org · 10 years ago
4121fcd Revert 6405 "Update generated asm offsets scripts." by henrike@webrtc.org · 10 years ago
88417a9 Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
6298c29 Re-land "Create a joint encoder/decoder wrapper for iSAC in ACM" by henrik.lundin@webrtc.org · 10 years ago
3acaa1f Reland: Making WebRTC able to play and record audio to files for tests. by phoglund@webrtc.org · 10 years ago
3fe0d4f Revert 6395 "Making WebRTC able to play and record audio to file..." by minyue@webrtc.org · 10 years ago
994f778 Making WebRTC able to play and record audio to files for tests. by phoglund@webrtc.org · 10 years ago
caf328c Revert r6377 "Create a joint encoder/decoder wrapper for iSAC in ACM" by henrik.lundin@webrtc.org · 10 years ago
ae4a452 common_audio/signal_processing: Removes macro WEBRTC_SPL_RSHIFT_U16 by bjornv@webrtc.org · 10 years ago
6e6b951 common_audio/signal_processing: Moves WEBRTC_SPL_UMUL_16_16_RSFT16 to iSAC fix by bjornv@webrtc.org · 10 years ago
b96d9c7 modules/audio_processing: Adds a config for reported delays by bjornv@webrtc.org · 10 years ago
adda09e Update makefiles after merge of Chromium at 276202 by Android Chromium Automerger · 10 years ago
604ba6f Delete last file in neteq4 folder by henrik.lundin@webrtc.org · 10 years ago
bc9c195 MIPS optimizations for ISAC (patch #1) by andrew@webrtc.org · 10 years ago
d70e23e Noise suppression: Change signature to work on floats instead of ints by kwiberg@webrtc.org · 10 years ago
f006e8d Add kjellander@webrtc.org as OWNER for *.isolate by kjellander@webrtc.org · 10 years ago
eaaba5a Create a joint encoder/decoder wrapper for iSAC in ACM by henrik.lundin@webrtc.org · 10 years ago
6da16b3 Add thread annotations to AcmReceiver by henrik.lundin@webrtc.org · 10 years ago
8097a46 Update makefiles after merge of Chromium at 275833 by Android Chromium Automerger · 10 years ago
377e7fd Enables DelayCorrection tests by bjornv@webrtc.org · 10 years ago
6d7c6e6 Multi-threaded unit test for Audio Coding Module using iSAC by henrik.lundin@webrtc.org · 10 years ago
4b50adf audio_processing: Forces extended filter to be used in splitting filter test. by bjornv@webrtc.org · 10 years ago
e5abc85 Rename neteq4 folder to neteq by henrik.lundin@webrtc.org · 10 years ago
c50f06d Re-enable AudioCodingModuleMtTest again by henrik.lundin@webrtc.org · 10 years ago
20d9f00 Update makefiles after merge of Chromium at 275661 by Android Chromium Automerger · 10 years ago
9cfa46c Revert r6358 "AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera." by fischman@webrtc.org · 10 years ago
00035af Use XErrorTrap in MouseCursorMonitorX11 to catch the error if the shared window has been closed. by jiayl@webrtc.org · 10 years ago
fea960e AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera. by fischman@webrtc.org · 10 years ago
d3d2598 Unbreak NDEBUG compile by RTC_UNUSED()ing an assert()d variable. by fischman@webrtc.org · 10 years ago
5101f84 AppRTCDemo(android): support app (UI) & capture rotation. by fischman@webrtc.org · 10 years ago
5befd8b VideoCaptureImpl::IncomingFrame(): avoid deadlock by acquiring _apiCs. by fischman@webrtc.org · 10 years ago
431772f Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 81f8df9af96c6b4bf43234f2a0162146a5da6112 by Android Chromium Automerger · 10 years ago
6e6292d Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 00d9c49cb076626f711988332749a0ebe8d2a32f by Android Chromium Automerger · 10 years ago
81f8df9 Fix the chain that propagates the audio frame's rtp and ntp timestamp including: by wu@webrtc.org · 10 years ago
553b68f Opus send rate overflows if over 65 kbps by tina.legrand@webrtc.org · 10 years ago
0e7d6a6 Revert 6341 "Fixes and enables SystemDelayTests." by bjornv@webrtc.org · 10 years ago
52e8925 Fixes and enables SystemDelayTests. by bjornv@webrtc.org · 10 years ago
14ac552 NetEq: Add thread annotation to const scoped_ptrs by henrik.lundin@webrtc.org · 10 years ago
39dec2c The correct fix of workaround in r6261. by bjornv@webrtc.org · 10 years ago
c806290 common_audio/signal_processing: Removed macro WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND by bjornv@webrtc.org · 10 years ago
903e746 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. by stefan@webrtc.org · 10 years ago
00d9c49 Android: cleanup gtest_target_type conditions. by henrike@webrtc.org · 10 years ago
fc94634 Make it possible to build webrtc for arm64. by solenberg@webrtc.org · 10 years ago
2b51241 Disables SystemDelayTest.CorrectDelayDuringDrift on Android by bjornv@webrtc.org · 10 years ago
625c309 Disables some modules_unittests on Android. by bjornv@webrtc.org · 10 years ago
d442cf9 Moved verbose logging in rtcp_receiver.cc to LS_VERBOSE. by andresp@webrtc.org · 10 years ago
fef1e23 Adding missing break in media_file_utility.cc. by mflodman@webrtc.org · 10 years ago
bb2058b Enable videoprocessor_integrationtest tests on android. by marpan@webrtc.org · 10 years ago
7115998 Revert 6312 "Re-enable AudioCodingModuleMtTest" by turaj@webrtc.org · 10 years ago
6e83888 Re-enable AudioCodingModuleMtTest by henrik.lundin@webrtc.org · 10 years ago
6038f4c Update makefiles after merge of Chromium at 274467 by Android Chromium Automerger · 10 years ago
baec5e7 Reformat integer accessors to look like their float counterparts by kwiberg@webrtc.org · 10 years ago
78dec3f Remove an optimization that's no longer worth the extra complexity it causes by kwiberg@webrtc.org · 10 years ago
cdbeb1d - Get rid of 'using' from .h by solenberg@webrtc.org · 10 years ago
bb57de4 Disable MouseCursorMonitorTest by henrik.lundin@webrtc.org · 10 years ago
1a3e45b Disable MouseCursorMonitorTest.FromScreen by henrik.lundin@webrtc.org · 10 years ago
81000de Adding thread annotations to parts of Audio Coding Module by henrik.lundin@webrtc.org · 10 years ago
f4d3760 Re-enables CommonFormats test for Android. by bjornv@webrtc.org · 10 years ago
57019f2 VideoCaptureAndroid: don't synchronized on camera thread. by fischman@webrtc.org · 10 years ago
35af59e Add a Reset() method to AudioFrame. by andrew@webrtc.org · 10 years ago
ab4f5eb Disable AudioCodingModuleMtTest due to memcheck and tsan failures. by andrew@webrtc.org · 10 years ago
58f48bb Multi-threaded test for Audio Coding Module by henrik.lundin@webrtc.org · 10 years ago
e0beaf4 Fix bug where RTP headers in the packet history were replaced with the RTX wrapped headers. by stefan@webrtc.org · 10 years ago
4260aa2 Fixing a bug regarding VOE packet loss rate feedback to ACM by minyue@webrtc.org · 10 years ago
27de386 Revert 6272 "Update generated asm offsets scripts." by sprang@webrtc.org · 10 years ago
b15df23 Increase VPMVideoDecimator's initial max_frame_rate_ to 60, which allow us potentially do 60fps. by wu@webrtc.org · 10 years ago
09c3605 Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
bf2bd58 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at ff6b4a8eddca609ad2691b54f443b6f1e9342579 by Android Chromium Automerger · 10 years ago
6537a08 VideoCaptureAndroid: quit & join the camera thread on stopCapture. by fischman@webrtc.org · 10 years ago
52dfe97 Update makefiles after merge of Chromium at 273259 by Android Chromium Automerger · 10 years ago
24e7aa7 Update makefiles after merge of Chromium at 273199 by Android Chromium Automerger · 10 years ago
54dfb38 Echo canceler: Saturate output to guarantee it'll be in the allowed range by kwiberg@webrtc.org · 10 years ago
47475b8 Update makefiles after merge of Chromium at 273188 by Android Chromium Automerger · 10 years ago
b552ce6 Better buffer size estimation in NetEq for redundant packets by minyue@webrtc.org · 10 years ago
81f6488 Revert 6257 "Rename neteq4 folder to neteq" by henrik.lundin@webrtc.org · 10 years ago
1bdf186 Add support of texture frames for video capturer. by wuchengli@chromium.org · 10 years ago
605e4d0 Rename neteq4 folder to neteq by henrik.lundin@webrtc.org · 10 years ago
86101b9 Disable MouseCursorMonitorTest due to flake on Windows. by andrew@webrtc.org · 10 years ago
5424828 Revert "Add support of texture frames for video capturer." by wuchengli@chromium.org · 10 years ago
a3b8c85 Add support of texture frames for video capturer. by wuchengli@chromium.org · 10 years ago
98e1ef1 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at e066d34bb747f730084f1726408ca8348ff25da7 by Android Chromium Automerger · 10 years ago
e066d34 Fix a bug preventing FilePlayer from playing encoded wav files by henrik.lundin@webrtc.org · 10 years ago
f290b35 Update makefiles after merge of Chromium at 272740 by Android Chromium Automerger · 10 years ago
271bf09 Update makefiles after merge of Chromium at 272566 by Android Chromium Automerger · 10 years ago
13978fc Avoid reading uninitialized values (outside baundary) in DFT arithmatic decoder of iSAC-fix. by turaj@webrtc.org · 10 years ago
91c0a25 1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED by minyue@webrtc.org · 10 years ago
e38123c Adds missing include of assert header. by henrike@webrtc.org · 10 years ago
8d385d8 WebRTCDemo: move the deletion of CritSect to end of the dtor to fix a crash in Android video renderer. by braveyao@webrtc.org · 10 years ago
774b3d3 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
b45df1c Initializes WINDOWPLACEMENT::length in GetCroppedWindowRect. by jiayl@webrtc.org · 10 years ago
0a9ed7c Revert 6202 "Switch to using base/constructormagic.h and remove ..." by mcasas@webrtc.org · 10 years ago
8520b33 Revert 6208 "Patch from henrike@webrtc.org" by mcasas@webrtc.org · 10 years ago
1528532 Patch from henrike@webrtc.org by mcasas@webrtc.org · 10 years ago
da3266d Enabling NetEq bit-exactness test for Win x64 by henrik.lundin@webrtc.org · 10 years ago
28b7c07 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
b5e74d6 Revert r6198 "Expose the original packet length in in the RTP play tools." by stefan@webrtc.org · 10 years ago
2ffdd60 Expose the original packet length in in the RTP play tools. by stefan@webrtc.org · 10 years ago