Log - f89ce467d1977ea1d867be4e5be2ccd9a206113c - fp2-dev/platform/external/chromium_org/third_party/webrtc

f89ce46 Implements start bitrate for new video API. by mflodman@webrtc.org · 10 years ago
0c14539 Add thread annotations to parts of ACMGenericCodec by henrik.lundin@webrtc.org · 10 years ago
d05de74 Add missing sources to webrtc/base/base.gyp by kjellander@webrtc.org · 10 years ago
bd98cef Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc. by glaznev@webrtc.org · 10 years ago
9257c64 Neon version of OverdriveAndSuppress() by bjornv@webrtc.org · 10 years ago
00dffd7 Pass GYP DEPTH variable to isolate. by kjellander@webrtc.org · 10 years ago
dd32ef8 Revert 6415 "Update generated asm offsets scripts." by wu@webrtc.org · 10 years ago
555f957 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 define. Also added OWNERS for base which is a copy of system_wrappers owners as the two folders are being merged. by henrike@webrtc.org · 10 years ago
19f89a1 Enable pacing by default and remove the option to disable it from the new API. by stefan@webrtc.org · 10 years ago
0e43e6f Update generated asm offsets scripts. by fgalligan@google.com · 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
f6eaabf Increased kMaxRampUpDelayMs (120 to 240s). by asapersson@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
6845de7 Add APIs to enable padding with redundant payloads. by stefan@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
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
9cd8281 Add additional metric (relative standard deviation of encode time) for overuse detection. by asapersson@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
b999e11 Make some methods in Clock class const declared by henrik.lundin@webrtc.org · 10 years ago
c05ab94 Remove unused test_env.py from isolate files + fix nss path. by kjellander@webrtc.org · 10 years ago
377e7fd Enables DelayCorrection tests by bjornv@webrtc.org · 10 years ago
f03a4a6 Updated conformance tests and w3c-ified them. by phoglund@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
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
daf186d ViEAutoTestAndroid: Unbreak compile by casting void* to jobject. 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
bdfcddf Make VideoSendStream/VideoReceiveStream configs const. by pbos@webrtc.org · 10 years ago
64027c5 Rebase webrtc/base with r6345 version of talk/base: by henrike@webrtc.org · 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
59a001f Adding back platform specific renderer to video loopback test. by mflodman@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
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
858611f Add native_test dependency to webrtc_perf_tests. by pbos@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
6f3ce73 * Revert clock.cc changes made in 6178, but keep the changes to the test. by wu@webrtc.org · 10 years ago
09c3605 Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
2b54919 Rebase webrtc/base with r6250: by henrike@webrtc.org · 10 years ago
1ef9cee Increase the threshold for CallPerfTest.CaptureNtpTimeWithNetworkDelay to avoid flaky. by wu@webrtc.org · 10 years ago
6537a08 VideoCaptureAndroid: quit & join the camera thread on stopCapture. by fischman@webrtc.org · 10 years ago
54dfb38 Echo canceler: Saturate output to guarantee it'll be in the allowed range by kwiberg@webrtc.org · 10 years ago
dd671de This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate. by minyue@webrtc.org · 10 years ago
ff6b4a8 common_audio/signal_processing: Fixes arm compilation issues with gcc 4.8 by bjornv@webrtc.org · 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
487dfb0 video_engine_tests_apk: enable running by adding nativeRunTests dependency. by fischman@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
17d5ac4 Adding R/W lock to SimulatedClock by henrik.lundin@webrtc.org · 10 years ago
4e95436 Added api for getting cpu measures using a struct. by asapersson@webrtc.org · 10 years ago