- 23e3559 Updating Perf numbers for Win Large Test. by tina.legrand@webrtc.org · 12 years ago
- 244251a Moved almost all payload-related stuff to the payload registry. by phoglund@webrtc.org · 12 years ago
- fa53d87 Fixing/disabling Windows x64 warnings by kjellander@webrtc.org · 12 years ago
- 254d85a Exchange TRY by enumerating image formats in Linux video capture by braveyao@webrtc.org · 12 years ago
- 6ed8ebc Fix MaxChannels test; 32 -> 100. by andrew@webrtc.org · 12 years ago
- 4a6f62d Remove (in practice) the voice engine channel limit. by andrew@webrtc.org · 12 years ago
- dbe97d2 Adding a send side API for streaming by mikhal@webrtc.org · 12 years ago
- becf9c8 Fix mismatch between different NACK list lengths and packet buffers. by stefan@webrtc.org · 12 years ago
- b586507 Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. by stefan@webrtc.org · 12 years ago
- 46d90dc Adding three frame sizes to Opus by tina.legrand@webrtc.org · 12 years ago
- d087789 Adjusted net_50_5_plr_5 on Linux, removed all gilbert_elliot metrics (too flaky), added mac expectations. by phoglund@webrtc.org · 12 years ago
- aaad613 Implementing stereo support for G.722 by henrik.lundin@webrtc.org · 12 years ago
- 7050f96 Set frame length for frame converting in external renderer by braveyao@webrtc.org · 12 years ago
- ac46c6d Replaced relative path to reference from <(webrtc_root). by bjornv@webrtc.org · 12 years ago
- 9d532fd Fix propagating RED paylaod-type to ACM. by turaj@webrtc.org · 12 years ago
- 763faea Removing a codec from NetEq database has a bug. |funcDurationEst| is not updated. by turaj@webrtc.org · 12 years ago
- c0ada86 fix for issue 281. by turaj@webrtc.org · 12 years ago
- 8c8ad85 fix issue 1322, accept -1 as default payload-type for redundant coding (FEC). by turaj@webrtc.org · 12 years ago
- 119c67d Adding a max jitter filter to the JB estimate - allowing two modes, one will return the last estimate (current setting), and another will return the max value seen, and allow setting an initial value. by mikhal@webrtc.org · 12 years ago
- e07c661 VP8: Making key frame interval a tunnable parameter by mikhal@webrtc.org · 12 years ago
- 6e3968f Fix NetEq4 unit tests for VS2012 by henrik.lundin@webrtc.org · 12 years ago
- 73deaad Removing a hack for CNG by henrik.lundin@webrtc.org · 12 years ago
- 96a08ce Fixed stale regression values and calibrated some vie_auto_test values. by phoglund@webrtc.org · 12 years ago
- ac59dba Adding iSAC-fb support by henrik.lundin@webrtc.org · 12 years ago
- 3d13d9f Fix audio_e2e_test command line arguments by kjellander@webrtc.org · 12 years ago
- 73a702c This is a change in the iOS audio device to use VoiceProcessingIO API instead of RemoteIO. This way we don't need to use WebRTC EC and NS because it happens on the device hardware. by andrew@webrtc.org · 12 years ago
- 7ded92b Re-committing r3428 by bjornv@webrtc.org · 12 years ago
- 51f11eb Fixing problems in audio_decoder_unittests by henrik.lundin@webrtc.org · 12 years ago
- ddf981c Disable iSAC fix test in audio_decoder_unittests by henrik.lundin@webrtc.org · 12 years ago
- 4892448 Re-enabling NetEqDecodingTest.TestBitExactness and .TestNetworkStatistics by henrik.lundin@webrtc.org · 12 years ago
- 63464a9 Enabling unit tests for NetEq4 in the bots by henrik.lundin@webrtc.org · 12 years ago
- e1d468c Fix a few small nits in NetEq4 by henrik.lundin@webrtc.org · 12 years ago
- c21988f Remove codereview.settings by henrik.lundin@webrtc.org · 12 years ago
- e12b1b5 Revert 3428 by bjornv@webrtc.org · 12 years ago
- 61ec7da Delay estimator wrapper API changes. This should finalize the changes to delay estimator making it work for multi-probe. by bjornv@webrtc.org · 12 years ago
- 57e6b81 Mac 64-bit compatibility for WebRTC. by henrike@webrtc.org · 12 years ago
- d94659d Initial upload of NetEq4 by henrik.lundin@webrtc.org · 12 years ago
- 63e0964 Fix webrtc compilation errors for Chrome Win64 by andrew@webrtc.org · 12 years ago
- 9ae4c66 Set working dir for test run script + update resources by kjellander@webrtc.org · 12 years ago
- e1888af Add <(DEPTH) to global includes by kjellander@webrtc.org · 12 years ago
- bf535b9 Optimize NACK list creation. by stefan@webrtc.org · 12 years ago
- b2d7497 Fix Win64 warnings by kjellander@webrtc.org · 12 years ago
- 8526459 Added tests for multiple near-end support. by bjornv@webrtc.org · 12 years ago
- 57f3a11 Short CL: only name change. by bjornv@webrtc.org · 12 years ago
- 94c213a Separated far-end handling in BinaryDelayEstimator. by bjornv@webrtc.org · 12 years ago
- 59d2095 Moving ViE test files and deleting files no longer used. by mflodman@webrtc.org · 12 years ago
- d3ecb61 Fix path to perf Python scripts in test.gyp by kjellander@webrtc.org · 12 years ago
- 43da54a Reformatted rtp_sender: made lint clean. by phoglund@webrtc.org · 12 years ago
- 3e47a0a Test launching script by kjellander@webrtc.org · 12 years ago
- c4373bc Moved several function pointer declarations in iSAC to isac initialization file. by kma@webrtc.org · 12 years ago
- 16d540e Fixed text relocation code related to ARM assembly code. by kma@webrtc.org · 12 years ago
- e8482f0 Revert 3406 by kma@webrtc.org · 12 years ago
- cd2f135 Revert 3405 by niklas.enbom@webrtc.org · 12 years ago
- ebef7e4 Moved all function pointer declarations in iSAC to a single place. by kma@webrtc.org · 12 years ago
- 05e7bfe Mainly hlundin's patch. by niklas.enbom@webrtc.org · 12 years ago
- 4782911 Optimized WebRtcIsacfix_Time2Spec() for iSAC-Fix in ARM Neon processor. by kma@webrtc.org · 12 years ago
- 5dfb1f2 Bug fix in WebRtcOpus_DurationEst by henrik.lundin@webrtc.org · 12 years ago
- 8126602 Fix frame_editing_unittest.cc by kjellander@webrtc.org · 12 years ago
- a812a3a Updated version number to 3.21 by elham@webrtc.org · 12 years ago
- 0973861 Fixes payload spelling error. by henrike@webrtc.org · 12 years ago
- 5accd37 RTP Receiver is now only deals with a receiver strategy. Cleaned up dependencies. by phoglund@webrtc.org · 12 years ago
- 8382ad5 Added perf expectations for stack tests. by phoglund@webrtc.org · 12 years ago
- ae1a58b Replace AudioFrame's operator= with CopyFrom(). by andrew@webrtc.org · 12 years ago
- 899699e Enabled full lint checking for ALL WebRTC changes. by phoglund@webrtc.org · 12 years ago
- a678a3b Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests. by stefan@webrtc.org · 12 years ago
- a3c82bf Remove '<(library)' in gyp files. by wjia@webrtc.org · 12 years ago
- bb599b7 This CL includes part of changes in a larger one. The final goal is to allow multiple delay estimators using the same reference (far-end) to save computational complexity. by bjornv@webrtc.org · 12 years ago
- a2d8b75 An API to get the internal estimation quality in the delay estimator has been added. Unit tests have been updated. There is no impact to other parts in WebRTC. by bjornv@webrtc.org · 12 years ago
- 2e2a4cf Remove <(library) from gyp file. by wjia@webrtc.org · 12 years ago
- a3e6bec Posix Thread: Removes the setting of the run function to NULL which could cause data race. by henrike@webrtc.org · 12 years ago
- 4ad6445 Fixed URL unquoting in bot names. Added iOS Device. Removed unnecessary filter code. by phoglund@webrtc.org · 12 years ago
- c39962a Adding TRYSERVER_ROOT to codereview.settings by kjellander@webrtc.org · 12 years ago
- 218c542 Make VoE handle longer delays by niklas.enbom@webrtc.org · 12 years ago
- c7e935f Adding timeEndPeriod to Synchronize function, see bug for details. by mflodman@webrtc.org · 12 years ago
- efae5d5 Extracted rtp receiver payload management to its own class, made video receiver depend on that instead. by phoglund@webrtc.org · 12 years ago
- 20ed36d Break out RtpClock to system_wrappers and make it more generic. by stefan@webrtc.org · 12 years ago
- 3b7feb2 Convert psnr and ssim to strings before printing them. by stefan@webrtc.org · 12 years ago
- a4b5886 Add a counter to the video rtp play output filename. by stefan@webrtc.org · 12 years ago
- ebc6d8f libyuv r540 roll for valgrind tools update, optimized ARGBToI444_SSSE3 and I420Copy single memcpy per plane if contiguous. by fbarchard@google.com · 12 years ago
- 00c18db Fix libvpx for Android by hclam@chromium.org · 12 years ago
- 2fd947f Removing outdated comment by mikhal@webrtc.org · 12 years ago
- 14d1898 Removing arena_thread_freeres suppression by kjellander@webrtc.org · 12 years ago
- acfdd96 Reformatted rtp_rtcp_impl*. by phoglund@webrtc.org · 12 years ago
- 77a584b Made ViEToFileRenderer use a separate thread for rendering frames to file. by stefan@webrtc.org · 12 years ago
- a22a9bd Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional. by phoglund@webrtc.org · 12 years ago
- 49273ff logical 'and' of mutually exclusive tests is always false in ViECodecImpl::CodecValid() by braveyao@webrtc.org · 12 years ago
- b119369 Fix android clang build. by wjia@webrtc.org · 12 years ago
- 3f9db37 Fix android clang build. by wjia@webrtc.org · 12 years ago
- bafdae3 Fix simulated analog gain in audioproc. by andrew@webrtc.org · 12 years ago
- f908011 Remove extra line. by andrew@webrtc.org · 12 years ago
- 75ba519 Updating chromium_revision 169394:176094 by kjellander@webrtc.org · 12 years ago
- e7dc7f8 Disable full stack PSNR/SSIM triggers on Mac and Win for now due to flakiness. Adding plots of PSNR and SSIM. by stefan@webrtc.org · 12 years ago
- 26901c2 libyuv r534 for tools folder valgrind and endian fix for big endian platforms like s390x. by fbarchard@google.com · 12 years ago
- be86a6d Explicitly disable sincos optimization on Android. by leozwang@webrtc.org · 12 years ago
- e468f08 Disable PSNR/SSIM thresholds for the Gilber-Elliot test. by stefan@webrtc.org · 12 years ago
- 171ac59 Corrected TSAN suppression. by phoglund@webrtc.org · 12 years ago
- dc6fa02 Fixing error in argument parsing by kjellander@webrtc.org · 12 years ago
- 8f13810 Improved memory tool test wrapper script by kjellander@webrtc.org · 12 years ago
- 0af0d3d Address a build issue with Android-Clang compiler: by kma@webrtc.org · 12 years ago
- ef1a760 Rounding error fix in media_opt_util. by marpan@webrtc.org · 12 years ago