- 3bb42ef Made e2e audio quality test write its results to perf. by phoglund@webrtc.org · 12 years ago
- 72feb0b Not to enum NOTPRESENT audio devices with CoreAudio on Win by braveyao@webrtc.org · 12 years ago
- 8e49b02 Add more audio codec information into codec list by leozwang@webrtc.org · 12 years ago
- 451aa5d Adding vp8 sequence coder: simple command line encode and decode. by mikhal@webrtc.org · 12 years ago
- 3a5a8a8 Properly zero out unmixed frames. by andrew@webrtc.org · 12 years ago
- 0e73950 Added buildbot benchmarking in iSAC and APM into Android platform build. by kma@webrtc.org · 12 years ago
- b968213 vp8 test: Updating creation of enc/dec by mikhal@webrtc.org · 12 years ago
- 251f64e Updating vp8 test structure by mikhal@webrtc.org · 12 years ago
- 60d25f9 Updating Vp8 unit tests - Initiating the switch to gtest-based tests, and adding a stride test. by mikhal@webrtc.org · 12 years ago
- 75f8c78 Fixing path to ptypes.txt in NetEqRTPplay by henrik.lundin@webrtc.org · 12 years ago
- df94329 Use different cpufeatures library when building with chrome. by wjia@webrtc.org · 12 years ago
- 81cffd1 Port Chromium's atomicops to WebRTC by hclam@chromium.org · 12 years ago
- 63a243a Replace the last occurrence of .s with .h by leozwang@webrtc.org · 12 years ago
- 96bcac8 Expose Set and Get Recording/Playout sample rate apis by leozwang@webrtc.org · 12 years ago
- f4e070e Added auto-call feature to WebRTCDemo. by fischman@webrtc.org · 12 years ago
- 2cf22a6 Revert 3231 - VoE Changes to enable dual_streaming. by perkj@webrtc.org · 12 years ago
- e861359 Adds two full stack performance metrics for end-to-end delay. by stefan@webrtc.org · 12 years ago
- 6bd737a First pass of MediaCodecDecoder which uses Android MediaCodec API. by dwkang@webrtc.org · 12 years ago
- 781cf06 libyuv r508 with scaler fix for overread horizontally that was caught by valgrind. by fbarchard@google.com · 12 years ago
- 767d87c VoE Changes to enable dual_streaming. by turaj@webrtc.org · 12 years ago
- 226db89 Dual-stream implementation, not including VoE APIs. by turaj@webrtc.org · 12 years ago
- 277ec8e Fix a bug when iSAC-48kHz was added. by turaj@webrtc.org · 12 years ago
- f18de86 Revert 3227 by mikhal@webrtc.org · 12 years ago
- ab83bb3 vp8 unittest: Adding qcif stride test by mikhal@webrtc.org · 12 years ago
- b0dff12 48 kHz extension to iSAC. by turaj@webrtc.org · 12 years ago
- 0bacb63 Removed stale version of fuzzer; it's now internal. by phoglund@webrtc.org · 12 years ago
- 8d0cd07 Add test to verify that padding only frames are passing through the RTP module. by stefan@webrtc.org · 12 years ago
- 5b4fe49 Changing default bitrate to 64000 bps for Opus. by tina.legrand@webrtc.org · 12 years ago
- ad0f3ba Removing redundant codec unittest targets. by kjellander@webrtc.org · 12 years ago
- ba21c95 Reformatted data_log. by phoglund@webrtc.org · 12 years ago
- c94f8d4 Fix OOB read in padding tests. by stefan@webrtc.org · 12 years ago
- 78bec2d Fixed bug where we would rewrite *deref_ptr = ...; to // deref_ptr = ...; by phoglund@webrtc.org · 12 years ago
- fc4a7ee Fixes chromium build bots. by henrike@webrtc.org · 12 years ago
- c7896df Fixed bug that caused frame_cutter_unittest to fail when built with MVS2008. by brykt@google.com · 12 years ago
- 53034fb 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
- 99f7c91 Reformatted critical_section wrappers. by phoglund@webrtc.org · 12 years ago
- 219df91 Delete bad mergeinfo from webrtc/modules/video_capture/windows by andrew@webrtc.org · 12 years ago
- dddc02b Use <(webrtc_root) to point to webrtc files in tools.gyp. by andrew@webrtc.org · 12 years ago
- d814d71 Delete {start,stop}CPULoad() since they're broken. by fischman@webrtc.org · 12 years ago
- be5b5ba Enable building WebRTCDemo apk using Release webrtc libs, take 2. by fischman@webrtc.org · 12 years ago
- bd941d3 Fixes two bugs related to padding in the jitter buffer. by stefan@webrtc.org · 12 years ago
- 8552c71 Fixing neteq_unittests for VS 2012 by henrik.lundin@webrtc.org · 12 years ago
- 34dab50 Corrected .h path. by phoglund@webrtc.org · 12 years ago
- 273ccad Fixed standard PSNR/SSIM test. by phoglund@webrtc.org · 12 years ago
- bf41508 Properly remove the bitrate observer when ViEEncoder is destructed. by stefan@webrtc.org · 12 years ago
- 662651a Disable denoise filter for Arm, as it is not optimized enough yet. by fbarchard@google.com · 12 years ago
- cde46fa Disabled some more flaky tests. Memcheck vie_auto_test should be very stable after this. by phoglund@webrtc.org · 12 years ago
- f826bb6 Fixing a bug related to RCU in NetEQ by henrik.lundin@webrtc.org · 12 years ago
- 56a1c2c Enable java soundcard impl as the default by leozwang@webrtc.org · 12 years ago
- de6f8fb Revert 3190 - Enable building WebRTCDemo apk using Release webrtc libs. by fischman@webrtc.org · 12 years ago
- 28afee0 Enable building WebRTCDemo apk using Release webrtc libs. by fischman@webrtc.org · 12 years ago
- f3cefe1 Added metrics test code for the FEC packet masks. by marpan@webrtc.org · 12 years ago
- c09e779 Allow for 1 layer case to be set in temporal_layers. by marpan@webrtc.org · 12 years ago
- 7d5dacc Revert 3183 - Fixes two bugs related to padding in the jitter buffer. by henrike@webrtc.org · 12 years ago
- c244cef Reverting r3185 by marpan@webrtc.org · 12 years ago
- 9934947 Added metrics test code for the FEC packet masks. by marpan@webrtc.org · 12 years ago
- aa46ea0 Remove ringtone from test app by leozwang@webrtc.org · 12 years ago
- e4fb44c Fixes two bugs related to padding in the jitter buffer. by stefan@webrtc.org · 12 years ago
- 891d55e Revert 3181 - Fixes two bugs related to padding in the jitter buffer. by henrike@webrtc.org · 12 years ago
- d42e51c Fixes two bugs related to padding in the jitter buffer. by stefan@webrtc.org · 12 years ago
- 0f8286f Added last (?) suppressions for known issues. by phoglund@webrtc.org · 12 years ago
- 7d74bdb Added conformance tests. by phoglund@webrtc.org · 12 years ago
- 8d334d3 Disabled flaky test on Linux, added disable-on-platform macros, fixed \n's by phoglund@webrtc.org · 12 years ago
- c459058 Opus mono/stereo on the same payloadtype, and fix of memory bug by tina.legrand@webrtc.org · 12 years ago
- 81fb7bf Adding video_coding_integrationtests test. by kjellander@webrtc.org · 12 years ago
- 8049608 VP8 wrapper: updating raw image allocation. by mikhal@webrtc.org · 12 years ago
- 4de3dfe 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
- 52ec985 Fixing vie and voe auto test project paths for test execution. by kjellander@webrtc.org · 12 years ago
- b43502e Revert 3170 - Added performance benchmarking in APM and iSAC-fix for Buildbots. by andrew@webrtc.org · 12 years ago
- 4cd8f1f Added performance benchmarking in APM and iSAC-fix for Buildbots. by kma@webrtc.org · 12 years ago
- 6e46d5b Updated version number to 3.18 by elham@webrtc.org · 12 years ago
- 107d4ef Rolling chromium_revision 157509:169394 by kjellander@webrtc.org · 12 years ago
- ef90c32 Will now correctly identify the first-ever received packet as the first packet in its frame. by phoglund@webrtc.org · 12 years ago
- 7c894b7 Wire up CallStats to provide modules with correct RTT. by mflodman@webrtc.org · 12 years ago
- 5ba3dec Ensures that we can build using VS 2012 on Windows. by henrika@webrtc.org · 12 years ago
- 221b11a Pulling Opus version 1.0.1 from Chromium by tina.legrand@webrtc.org · 12 years ago
- c3e5d34 Add a logging_no_op.cc when enable_tracing==0. by andrew@webrtc.org · 12 years ago
- 418443c Remove operator overloading from RTPFragmentationHeader. by andrew@webrtc.org · 12 years ago
- ad7f1fe 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
- 6ba79a8 Condition for DirectX variable on Windows by kjellander@webrtc.org · 12 years ago
- 849fb8e Removed codec comparison test: it didn't work and probably never will. by phoglund@webrtc.org · 12 years ago
- e3b2bc6 Will now fix old src-relative paths so we go to webrtc/ paths. by phoglund@webrtc.org · 12 years ago
- 97dcf36 Adding Direct X SDK include directory. by kjellander@webrtc.org · 12 years ago
- 087723b Updated license path in LICENSE and LICENSE_THIRD_PARTY. by mflodman@webrtc.org · 12 years ago
- f89fb9d Remove ViE lint warnings that should have been caught at upload time. by mflodman@webrtc.org · 12 years ago
- 1c61196 Removed not used include. by mflodman@webrtc.org · 12 years ago
- 6e76ef4 Add third_party/winsdk_samples/src to gitignore. by andrew@webrtc.org · 12 years ago
- 4c4d01d Setting capture stride to width by mikhal@webrtc.org · 12 years ago
- 4b97793 Ensure opus_demo has a targets block. by andrew@webrtc.org · 12 years ago
- 8cd18c5 Add winsdk_samples to provide directshow_baseclasses. by andrew@webrtc.org · 12 years ago
- cfcadab Build opus_demo by leozwang@webrtc.org · 12 years ago
- 3ec52c0 Adding mflodman's reformat script with some fixes. by phoglund@webrtc.org · 12 years ago
- b15d285 Reformatted most of the CPU stuff in system_wrappers. by phoglund@webrtc.org · 12 years ago
- 5835adf Reorganize gyp for Android by leozwang@webrtc.org · 12 years ago
- 3263a7a Setting correct stride for VP8 encoder by mikhal@webrtc.org · 12 years ago
- 32b3f40 Adding an aligned stride test to LibYuv by mikhal@webrtc.org · 12 years ago
- 8187877 Reland 3135 - Previous failure was bot flakiness. ***** by tommi@webrtc.org · 12 years ago
- 951b6c4 Revert 3135 - This broke the Mac bots somehow. Here's the error: by tommi@webrtc.org · 12 years ago
- 704eb8f Restructure the video_capture code a bit to make room for a Media Foundation class implementation. by tommi@webrtc.org · 12 years ago
- 655d8f5 Add a kTraceTerseInfo level for non-verbose logging. by andrew@webrtc.org · 12 years ago