- 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
- ad0f05b Remove empty directories. by andrew@webrtc.org · 13 years ago
- 2e84c11 Updating bitrate controller tests to test naming conventions. by kjellander@webrtc.org · 13 years ago
- baaf243 Extracted a method for sending padded data. by phoglund@webrtc.org · 13 years ago
- bb24b14 Libvpx waterfall additional changes. The CL http://review.webrtc.org/595005/ was not complete since it didn't put the libvpx waterfall at its own port. by kjellander@webrtc.org · 13 years ago
- 7d3b07a Update to chromium r139469. by wu@webrtc.org · 13 years ago
- 36ccce4 Remove documentation folders. by andrew@webrtc.org · 13 years ago
- 16fcb24 Disable flaky VolumeTests only on Linux. by andrew@webrtc.org · 13 years ago
- e7e64e3 Fix compilation errors on ChromeOS by leozwang@webrtc.org · 13 years ago
- 0cb79cc Fixing gyp bug in https://webrtc-codereview.appspot.com/599006 by niklas.enbom@webrtc.org · 13 years ago
- dc257b5 Add option to configure error concealment and disable by default. by stefan@webrtc.org · 13 years ago
- 327ada1 Refactored IncomingVideoStream and VideoRenderFrame, to get code in better shape when hunting BUG=481. by mflodman@webrtc.org · 13 years ago
- 9259e7b Added a step for restarting pulseaudio. by phoglund@webrtc.org · 13 years ago
- 281b798 Resolved Coverity warnings. by bjornv@webrtc.org · 13 years ago
- b5ea03a Add print out stats summary to integrationtest.cc by leozwang@webrtc.org · 13 years ago
- 459955f Move audio_frame_operations to the utility module. by andrew@webrtc.org · 13 years ago
- aafa49b Disable flaky VolumeTest.DefaultSpeakerVolumeIsAtMost255. by andrew@webrtc.org · 13 years ago
- 5f23d64 Set the stream delay to zero if too low. by andrew@webrtc.org · 13 years ago
- 2fc6e38 Check return value of fwrite. [Video Module] by leozwang@webrtc.org · 13 years ago
- 8a7a019 Syncing tests on try master with build master. by kjellander@webrtc.org · 13 years ago
- 1eef9c1 Bitrate bugfixes Review URL: https://webrtc-codereview.appspot.com/609005 by pwestin@webrtc.org · 13 years ago
- 5abab0b Revert 2311 - Disable error concealment. by stefan@webrtc.org · 13 years ago
- 3348b29 Disable error concealment. by stefan@webrtc.org · 13 years ago
- fe87f8c Libvpx buildbot waterfall. by kjellander@webrtc.org · 13 years ago
- ca8d788 Fix a bug where a RAII object was created for just one line instead of a block. by mflodman@webrtc.org · 13 years ago
- dbaa893 Completed rewrite of APM extended test. by phoglund@webrtc.org · 13 years ago
- 1747427 VAD Refactoring: Replaced pointer operation with array index by bjornv@webrtc.org · 13 years ago
- 4e12d30 VAD Refactoring: Removed assign calls by bjornv@webrtc.org · 13 years ago
- 0de1ee3 NetEQ: Remove an unnecessary condition, to fix a clang warning by tina.legrand@webrtc.org · 13 years ago
- b6e4cc7 Valgrind memcheck and TSAN script now uses Chrome+WebRTC suppression files. by kjellander@webrtc.org · 13 years ago
- 0d321da Refactored ARM specific code in Noise Suppression. Bit exact. by kma@webrtc.org · 13 years ago
- 1755a57 Check return result of fwrite, [APM] by leozwang@webrtc.org · 13 years ago
- f14575f Dynamically load codec list by leozwang@webrtc.org · 13 years ago
- 351fb6d Exclude code that don't work on android in voe_cmd_test by leozwang@webrtc.org · 13 years ago
- 15cd04a Enabling vp8_integrationtests on all platforms by kjellander@webrtc.org · 13 years ago
- 21c5bda Enabling udp_transport_unittests for TSAN by kjellander@webrtc.org · 13 years ago
- 10d3b52 I haven't done any refactoring here. by turaj@webrtc.org · 13 years ago
- f45d47a Remove mixing tests from voe_extended_test.cc by andrew@webrtc.org · 13 years ago
- 51b4f3e Try to fix MixingTest on the Win bots. by andrew@webrtc.org · 13 years ago