- 15acf41 Added buildbot benchmarking in iSAC and APM into Android platform build. by kma@webrtc.org · 12 years ago
- 07d98c8 vp8 test: Updating creation of enc/dec by mikhal@webrtc.org · 12 years ago
- 867e025 Updating vp8 test structure by mikhal@webrtc.org · 12 years ago
- 9dda8f5 Updating Vp8 unit tests - Initiating the switch to gtest-based tests, and adding a stride test. by mikhal@webrtc.org · 12 years ago
- 4f2f31c Fixing path to ptypes.txt in NetEqRTPplay by henrik.lundin@webrtc.org · 12 years ago
- 455aeef Use different cpufeatures library when building with chrome. by wjia@webrtc.org · 12 years ago
- 82eefff Port Chromium's atomicops to WebRTC by hclam@chromium.org · 12 years ago
- 0eb2b97 Replace the last occurrence of .s with .h by leozwang@webrtc.org · 12 years ago
- 15e35cc Expose Set and Get Recording/Playout sample rate apis by leozwang@webrtc.org · 12 years ago
- eeb98fe Added auto-call feature to WebRTCDemo. by fischman@webrtc.org · 12 years ago
- c65ae4b Revert 3231 - VoE Changes to enable dual_streaming. by perkj@webrtc.org · 12 years ago
- fb2953b Adds two full stack performance metrics for end-to-end delay. by stefan@webrtc.org · 12 years ago
- 7c32ab5 First pass of MediaCodecDecoder which uses Android MediaCodec API. by dwkang@webrtc.org · 12 years ago
- d6f028b VoE Changes to enable dual_streaming. by turaj@webrtc.org · 12 years ago
- 1952f25 Dual-stream implementation, not including VoE APIs. by turaj@webrtc.org · 12 years ago
- 8927ea0 Fix a bug when iSAC-48kHz was added. by turaj@webrtc.org · 12 years ago
- 51566f7 Revert 3227 by mikhal@webrtc.org · 12 years ago
- e7c2a9e vp8 unittest: Adding qcif stride test by mikhal@webrtc.org · 12 years ago
- ad70aa4 48 kHz extension to iSAC. by turaj@webrtc.org · 12 years ago
- a619a4c Removed stale version of fuzzer; it's now internal. by phoglund@webrtc.org · 12 years ago
- cf4441c Add test to verify that padding only frames are passing through the RTP module. by stefan@webrtc.org · 12 years ago
- a82843b Changing default bitrate to 64000 bps for Opus. by tina.legrand@webrtc.org · 12 years ago
- fdbaf4a Removing redundant codec unittest targets. by kjellander@webrtc.org · 12 years ago
- 3c5fc84 Reformatted data_log. by phoglund@webrtc.org · 12 years ago
- 259cce5 Fix OOB read in padding tests. by stefan@webrtc.org · 12 years ago
- e88a485 Fixes chromium build bots. by henrike@webrtc.org · 12 years ago
- 557655f Fixed bug that caused frame_cutter_unittest to fail when built with MVS2008. by brykt@google.com · 12 years ago
- 216c5f6 Improved the conformance test: it will now show video tags and better verify that we set up a call. by phoglund@webrtc.org · 12 years ago
- 328820f Reformatted critical_section wrappers. by phoglund@webrtc.org · 12 years ago
- 7dcf36e Delete bad mergeinfo from webrtc/modules/video_capture/windows by andrew@webrtc.org · 12 years ago
- ad2a55a Use <(webrtc_root) to point to webrtc files in tools.gyp. by andrew@webrtc.org · 12 years ago
- 4184485 Delete {start,stop}CPULoad() since they're broken. by fischman@webrtc.org · 12 years ago
- e6c51f4 Enable building WebRTCDemo apk using Release webrtc libs, take 2. by fischman@webrtc.org · 12 years ago
- c541182 Fixes two bugs related to padding in the jitter buffer. by stefan@webrtc.org · 12 years ago
- 1a22213 Fixing neteq_unittests for VS 2012 by henrik.lundin@webrtc.org · 12 years ago
- 006b929 Corrected .h path. by phoglund@webrtc.org · 12 years ago
- 5600f6e Fixed standard PSNR/SSIM test. by phoglund@webrtc.org · 12 years ago
- c0539d9 Properly remove the bitrate observer when ViEEncoder is destructed. by stefan@webrtc.org · 12 years ago
- ff38bd8 Disable denoise filter for Arm, as it is not optimized enough yet. by fbarchard@google.com · 12 years ago
- c752958 Disabled some more flaky tests. Memcheck vie_auto_test should be very stable after this. by phoglund@webrtc.org · 12 years ago
- 12372c3 Fixing a bug related to RCU in NetEQ by henrik.lundin@webrtc.org · 12 years ago
- fdea552 Enable java soundcard impl as the default by leozwang@webrtc.org · 12 years ago
- 9b1f0ac Revert 3190 - Enable building WebRTCDemo apk using Release webrtc libs. by fischman@webrtc.org · 12 years ago
- eeb9c92 Enable building WebRTCDemo apk using Release webrtc libs. by fischman@webrtc.org · 12 years ago
- cf7cef4 Added metrics test code for the FEC packet masks. by marpan@webrtc.org · 12 years ago
- 543f68f Allow for 1 layer case to be set in temporal_layers. by marpan@webrtc.org · 12 years ago
- 8196e09 Revert 3183 - Fixes two bugs related to padding in the jitter buffer. by henrike@webrtc.org · 12 years ago
- 0ecdc23 Reverting r3185 by marpan@webrtc.org · 12 years ago
- 7ee1c72 Added metrics test code for the FEC packet masks. by marpan@webrtc.org · 12 years ago
- 6784b3c Remove ringtone from test app by leozwang@webrtc.org · 12 years ago
- 7bca2a3 Fixes two bugs related to padding in the jitter buffer. by stefan@webrtc.org · 12 years ago
- 0b3d60c Revert 3181 - Fixes two bugs related to padding in the jitter buffer. by henrike@webrtc.org · 12 years ago
- 9f651d9 Fixes two bugs related to padding in the jitter buffer. by stefan@webrtc.org · 12 years ago
- 617ecfd Added last (?) suppressions for known issues. by phoglund@webrtc.org · 12 years ago
- 8cf767f Added conformance tests. by phoglund@webrtc.org · 12 years ago
- d166091 Disabled flaky test on Linux, added disable-on-platform macros, fixed \n's by phoglund@webrtc.org · 12 years ago
- dbf7ca6 Opus mono/stereo on the same payloadtype, and fix of memory bug by tina.legrand@webrtc.org · 12 years ago
- e9c2556 Adding video_coding_integrationtests test. by kjellander@webrtc.org · 12 years ago
- 045abb3 VP8 wrapper: updating raw image allocation. by mikhal@webrtc.org · 12 years ago
- d9b18e9 Tool for editing of yuv-files. Specify a path to the clip that should be edited, the height and width of the clip, one set of frames that should be removed from the clip, and a path to where the result should be written. There is a executable created that make use of the library where the functionality is implemented. There is also a unittest added for the library. by brykt@google.com · 12 years ago
- 9ec89f4 Fixing vie and voe auto test project paths for test execution. by kjellander@webrtc.org · 12 years ago
- ea852b1 Revert 3170 - Added performance benchmarking in APM and iSAC-fix for Buildbots. by andrew@webrtc.org · 12 years ago
- f5f6740 Added performance benchmarking in APM and iSAC-fix for Buildbots. by kma@webrtc.org · 12 years ago
- 64ea357 Updated version number to 3.18 by elham@webrtc.org · 12 years ago
- aa242d2 Will now correctly identify the first-ever received packet as the first packet in its frame. by phoglund@webrtc.org · 12 years ago
- 78696d3 Wire up CallStats to provide modules with correct RTT. by mflodman@webrtc.org · 12 years ago
- d6f6ff0 Ensures that we can build using VS 2012 on Windows. by henrika@webrtc.org · 12 years ago
- 804d552 Add a logging_no_op.cc when enable_tracing==0. by andrew@webrtc.org · 12 years ago
- 856edd5 Remove operator overloading from RTPFragmentationHeader. by andrew@webrtc.org · 12 years ago
- e3ada29 Fixes (or at least reduces) the flakiness in the full stack test by making sure the different frame monitors are registered and deregistered in the right order. Also makes sure only local preview frames which are actually transmitted are rendered by moving the local preview rendering to an effect filter. by stefan@webrtc.org · 12 years ago
- 95430b6 Condition for DirectX variable on Windows by kjellander@webrtc.org · 12 years ago
- fca1d1d Removed codec comparison test: it didn't work and probably never will. by phoglund@webrtc.org · 12 years ago
- 2bf4761 Adding Direct X SDK include directory. by kjellander@webrtc.org · 12 years ago
- 1d50745 Remove ViE lint warnings that should have been caught at upload time. by mflodman@webrtc.org · 12 years ago
- eb72e23 Removed not used include. by mflodman@webrtc.org · 12 years ago
- e905921 Setting capture stride to width by mikhal@webrtc.org · 12 years ago
- 551e488 Ensure opus_demo has a targets block. by andrew@webrtc.org · 12 years ago
- fb0c0d9 Add winsdk_samples to provide directshow_baseclasses. by andrew@webrtc.org · 12 years ago
- b13c5e2 Build opus_demo by leozwang@webrtc.org · 12 years ago
- 4c54650 Reformatted most of the CPU stuff in system_wrappers. by phoglund@webrtc.org · 12 years ago
- ab9aa45 Reorganize gyp for Android by leozwang@webrtc.org · 12 years ago
- 05eec40 Setting correct stride for VP8 encoder by mikhal@webrtc.org · 12 years ago
- 052382e Adding an aligned stride test to LibYuv by mikhal@webrtc.org · 12 years ago
- 0f224ff Reland 3135 - Previous failure was bot flakiness. ***** by tommi@webrtc.org · 12 years ago
- 3b7f2ab Revert 3135 - This broke the Mac bots somehow. Here's the error: by tommi@webrtc.org · 12 years ago
- a882006 Restructure the video_capture code a bit to make room for a Media Foundation class implementation. by tommi@webrtc.org · 12 years ago
- bc687c5 Add a kTraceTerseInfo level for non-verbose logging. by andrew@webrtc.org · 12 years ago
- d064f58 Add Chromium's perf_test to testsupport. by andrew@webrtc.org · 12 years ago
- 3082003 Updating Memory allocation for rotation and related tests. by mikhal@webrtc.org · 12 years ago
- 8e3d40c Fix possible race condition and access into an empty list. by stefan@webrtc.org · 12 years ago
- 7d32491 Move SSRC list to RemoteBitrateEstimator. by stefan@webrtc.org · 12 years ago
- c05b561 Allow NetEQ to use real packet durations. by tina.legrand@webrtc.org · 12 years ago
- 0739180 Use cpu_features library from ndk when built with chromium. by wjia@webrtc.org · 12 years ago
- 10b747a Define enable_android_opensl when built with chromium. by wjia@webrtc.org · 12 years ago
- 03a161e Fixes http://code.google.com/p/webrtc/issues/detail?id=941 by henrike@webrtc.org · 12 years ago
- b238aca Porting ARM optimization from Android to ios. by kma@webrtc.org · 12 years ago
- ece4890 Add warning comment Review URL: https://webrtc-codereview.appspot.com/933012 by niklas.enbom@webrtc.org · 12 years ago
- 641b4aa Fix ordered comparison warnings in the RTPtimeshift unit test by tina.legrand@webrtc.org · 12 years ago
- e296783 Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled. by mflodman@webrtc.org · 12 years ago
- e6527c1 Replaced remb unittest sleep with fake clock. by mflodman@webrtc.org · 12 years ago
