- 4ecea3e Downmix before resampling in capture and render paths. by andrew@webrtc.org · 13 years ago
- 7a281a5 Fix Android build after test/ -> src/test/ by andrew@webrtc.org · 13 years ago
- 81cf5e4 Move test to src/test. by andrew@webrtc.org · 13 years ago
- 253912c Disable a few features to save CPU cycles on android by leozwang@webrtc.org · 13 years ago
- 5567ebf VPM: Assign correct required size for odd size destination frame. by marpan@webrtc.org · 13 years ago
- bd7aeba Expose a set of options to the OveruseDetector supporting experiments by astor@webrtc.org · 13 years ago
- f494fd0 Use system-independent sleep in video_capture_unittest. by hta@webrtc.org · 13 years ago
- 626dccc Use one OS-independent sleep function in a video test by hta@webrtc.org · 13 years ago
- 643be71 Adds variable for third party directory. by henrike@webrtc.org · 13 years ago
- b9c1833 Add channel info to the Actions->Codec Changes menu in the VoE test app. by tnakamura@webrtc.org · 13 years ago
- 77e1812 Fix the flakiness in FileBeforeStreamingTest by braveyao@webrtc.org · 13 years ago
- 7450896 Roll Chromium 140240:143348. by andrew@webrtc.org · 13 years ago
- 64f86fb Fix test app render bug. by mflodman@webrtc.org · 13 years ago
- 8baed51 This CL is part of enabling cpplint check for video_engine uploads. by mflodman@webrtc.org · 13 years ago
- 9ba151b Removed cpplint warnings from all impl-files to be able to add this check as presubmit step. I don't want to change the API right now, will come later, so there are several NOLINT comments added to get around this for now. by mflodman@webrtc.org · 13 years ago
- 2bd8d62 Sleep using no compile flags by hta@webrtc.org · 13 years ago
- 67f98ec Removed flaky REMB test. This test is now covered by: by mflodman@webrtc.org · 13 years ago
- 173538f Refactored function WebRtcIsacfix_GetLpcCoef() in iSAC-fix. by kma@webrtc.org · 13 years ago
- ba5a97d Moving buildbot to internal repository. by kjellander@webrtc.org · 13 years ago
- 3168e53 Working unit test for critical sections. by hta@webrtc.org · 13 years ago
- 5608fe9 Disabling FileBeforeStreamingTest due to flakiness. by kjellander@webrtc.org · 13 years ago
- 2259f85 Remove unused member variables found by clang's -Wunused-private-field. by wu@webrtc.org · 13 years ago
- 72e3a89 Created a wrapper class for condition_variable that lets me write (hopefully) reliable tests for some of its properties. by hta@webrtc.org · 13 years ago
- b38fca1 VAD Refactoring: API change of return value from int16_t to int. by bjornv@webrtc.org · 13 years ago
- f477aac Removed gflags header from vie_auto_test. by vspasova@webrtc.org · 13 years ago
- dfa6b69 Refine the error handling made in rev2373 by braveyao@webrtc.org · 13 years ago
- 67f256f Use 32 as the alignment if possible in VP8 wrapper. by wu@webrtc.org · 13 years ago
- df596ae VAD Refactoring of GMM test section by bjornv@webrtc.org · 13 years ago
- 50d5ca5 Refactoring of TestAllCodecs by tina.legrand@webrtc.org · 13 years ago
- db2f6cf Added missing define guard to sleep.h by hta@webrtc.org · 13 years ago
- 86a6aac Unittest utilities - starting out with an encapsulated trace-to-screen. by hta@webrtc.org · 13 years ago
- e3a0712 Deregister RTP module before deleting it. by mflodman@webrtc.org · 13 years ago
- 41adcdb An OS-independent sleep function, and one usage thereof. by hta@webrtc.org · 13 years ago
- 3719800 GetRecPayloadType now logs a warning instead of and error when the user asks for the payload type while no packets have been received. by henrika@webrtc.org · 13 years ago
- 1905415 Correct gypi files to match the actual filenames. by stefan@webrtc.org · 13 years ago
- d63d06a bump version to 3.8 Review URL: https://webrtc-codereview.appspot.com/657004 by niklas.enbom@webrtc.org · 13 years ago
- 4de777b Refine the error processing of StopRecordingMicrophone. by braveyao@webrtc.org · 13 years ago
- bdf7ee5 This simple change should adress issue 471. by turaj@webrtc.org · 13 years ago
- 352d09a Updates to videoprocessor_integration test: by marpan@webrtc.org · 13 years ago
- f088448 Libyuv Scalerunittest: Added PSNR check to some tests in scaler unittest: by marpan@webrtc.org · 13 years ago
- 139c467 Fixed a/v sync issue. by mflodman@webrtc.org · 13 years ago
- 46d83fa Use digital mode on mobile by leozwang@webrtc.org · 13 years ago
- c35f1d2 FEC: Fix to coverity issue 14448: unintended sign extension. by marpan@webrtc.org · 13 years ago
- f0d4696 Add support for SSE intrinsics on gcc in libvpx. by stefan@webrtc.org · 13 years ago
- d418514 Bumped version number to 3.7. by mflodman@webrtc.org · 13 years ago
- b1c3276 VAD Refactoring: WebRtcVad_Process() by bjornv@webrtc.org · 13 years ago
- 5f9f1db This change make PulseAudio only start for the tests on the LinuxLargeTests bot. by kjellander@webrtc.org · 13 years ago
- 5e7ca60 Use new fileutil functions for trace in ACM by tina.legrand@webrtc.org · 13 years ago
- 1c28473 Fix master's "Start PulseAudio" step. by andrew@webrtc.org · 13 years ago
- 0594916 Add audio_e2e_test to LinuxLargeTests. by andrew@webrtc.org · 13 years ago
- 9f6577b Restore default source in e2e test. by andrew@webrtc.org · 13 years ago
- 6724c42 Add VoiceEngine apm settings to test application by leozwang@webrtc.org · 13 years ago
- be58164 Add a variable for the libjpeg include directory. by andrew@webrtc.org · 13 years ago
- f08f52f Fixing issues with slaves.cfg on Windows. by kjellander@webrtc.org · 13 years ago
- eec739f VAD Refactoring: Changed pointer structure in WebRtcVad_FindMinimum(). by bjornv@webrtc.org · 13 years ago
- 78a3110 Disable multi_res_encoding in libvpx. by marpan@webrtc.org · 13 years ago
- fa7138f Change CriticalSectionScoped to use pointer constructor by tina.legrand@webrtc.org · 13 years ago
- 276dc18 Add libremote_bitrate_estimator to android makefile by leozwang@webrtc.org · 13 years ago
- f85b35a Refactored Neon code for AECM module, by using pure assembly code. by kma@webrtc.org · 13 years ago
- 38506ef Disable cpu detection on arm-neon by leozwang@webrtc.org · 13 years ago
- d81ab13 abs() was used instead of fabsf(), which returns int and not float and therefore truncated the return value. by stefan@webrtc.org · 13 years ago
- f7d0c77 Added the bitrate estimator test to the trybots. by phoglund@webrtc.org · 13 years ago
- 90af7f8 Changing Celt to run on 20 msec frames by tina.legrand@webrtc.org · 13 years ago
- d2956d8 Renamed test_bwe. by phoglund@webrtc.org · 13 years ago
- 9354cc9 Refactoring the receive-side bandwidth estimation into its own module. by stefan@webrtc.org · 13 years ago
- f4c6aa2 Improve the reliablity of the audio e2e test. by andrew@webrtc.org · 13 years ago
- b0bcf13 Trival fix to relative paths of audio files in voe_ui_win_test by braveyao@webrtc.org · 13 years ago
- 5f97232 Removing a TODO in the FEC: renaming the exisiting packets mask to indicate random mode, by marpan@webrtc.org · 13 years ago
- cac603f Fix for the alignment problems/mismatch in ViECapture and VP8Encoder. by wu@webrtc.org · 13 years ago
- f4c2de9 Added some tests to videoprocessor_integrationtest, for testing: by marpan@webrtc.org · 13 years ago
- 8866bb1 FEC: Added another set of packet masks for the FEC. by marpan@webrtc.org · 13 years ago
- 20e13ed New attempt to revert r2362, since drover failed. by bjornv@webrtc.org · 13 years ago
- cb89c6f Revert 2363 - Refactoring the receive-side bandwidth estimation into its own module. by bjornv@webrtc.org · 13 years ago
- df37398 Renamed test_bwe. by phoglund@webrtc.org · 13 years ago
- f728814 Refactoring the receive-side bandwidth estimation into its own module. by stefan@webrtc.org · 13 years ago
- d2acea6 Minor style changes by bjornv@webrtc.org · 13 years ago
- 3007b26 Roll Chromium 134666:140240. by andrew@webrtc.org · 13 years ago
- da7fdf4 Fix to scaler in libyuv for odd size frames. by marpan@webrtc.org · 13 years ago
- ba108ae This CL contains some refactoring. Spectrum coding is main place that is affected. Therefore, I have bit-exactness test, test_spectrum_ by turaj@webrtc.org · 13 years ago
- 2cc5509 Fix syntax error in jpeg.gypi. by andrew@webrtc.org · 13 years ago
- ad6083f Added condition for which jpeg lib to use. by mflodman@webrtc.org · 13 years ago
- 77fd39a ACM PCM16B, fixing a copy-and-paste error. by tina.legrand@webrtc.org · 13 years ago
- e6f235c Attempt to fix broken encoding. by phoglund@webrtc.org · 13 years ago
- 9cf4d72 by niklas.enbom@webrtc.org · 13 years ago
- 82bf033 by niklas.enbom@webrtc.org · 13 years ago
- 265e38c Fixing test gypi for bit rate controller by niklas.enbom@webrtc.org · 13 years ago
- f1d6e0a Removed the obsolete sanity check and added new test HTML files. by phoglund@webrtc.org · 13 years ago
- ab12990 In the past we support calling StartPlayingFileLocally() before StartPlayout(), then when playout is started, the file would be played out immediately. by braveyao@webrtc.org · 13 years ago
- 899baa8 Temporarily disable first partition packet counting to avoid a bug in ProducerFec which doesn't properly handle important packets. by marpan@webrtc.org · 13 years ago
- 354b0ed Check return result of fwrite [Audio Module] by leozwang@webrtc.org · 13 years ago
- c3b2683 Refactored the pitch filter function in iSAC-fix. One important purpose is to prepare the function for assembly optimization in ARM platforms. by kma@webrtc.org · 13 years ago
- 5b4f36d ACM: Too short char vector by tina.legrand@webrtc.org · 13 years ago
- 343301f Fixing release compilation on Linux and Mac trybots by kjellander@webrtc.org · 13 years ago
- c03df17 Enabling audio_coding_module_test on trybots by kjellander@webrtc.org · 13 years ago
- 4517585 Adding separate payload types for stereo modes by tina.legrand@webrtc.org · 13 years ago
- c2722a0 Fixed compiler warning Review URL: https://webrtc-codereview.appspot.com/624005 by pwestin@webrtc.org · 13 years ago
- 29c5a23 Renamed to Network Emulator and improved error handling. by kjellander@webrtc.org · 13 years ago
- f5d934d Upgrade libvpx to cab6ac16 (v. 1.1.1 pre-point-release). by stefan@webrtc.org · 13 years ago
- 7d8c567 Ignore return value of fwrites. by andrew@webrtc.org · 13 years ago
- 595749f Network simulation script based on Dummynet. by kjellander@webrtc.org · 13 years ago