- 9e605b2 Fix Windows x64 errors in video_codecs_test_framework by kjellander@webrtc.org · 11 years ago
- 894a543 Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM. by turaj@webrtc.org · 11 years ago
- 33c6e92 Sync libvpx and its gyp wrapper from Chromium. by andrew@webrtc.org · 11 years ago
- 1fb8372 Increase maximum resolution to 4k x 3k. by fbarchard@google.com · 11 years ago
- 28166a5 VCM: Removing frame drop enable from Reset call BUG = 1387 by mikhal@webrtc.org · 11 years ago
- 9c4707e Android NDK build tools by kjellander@webrtc.org · 11 years ago
- 9cd6011 Fix perf output for audioproc and iSAC fixed-point tests by kjellander@webrtc.org · 11 years ago
- 4da62e0 Set SingleStream BWE in unittests. by stefan@webrtc.org · 11 years ago
- e3664d5 Set qpMax to 56 in for all VP8 tests. Fixes buildbot breakage. by stefan@webrtc.org · 11 years ago
- 6cd34e5 Updates to send side streaming mode: by mikhal@webrtc.org · 11 years ago
- 6bcf2ab Update version number to 3.23 by tnakamura@webrtc.org · 11 years ago
- 8b5ff39 Fix Win64 build breakage by henrikg@webrtc.org · 11 years ago
- 75e6669 Made it possible to render custom call output to file. by phoglund@webrtc.org · 11 years ago
- 05a655b Fixed a bug in iSAC transform functions on ARM-Neon platform. Performance unchanged. by kma@webrtc.org · 11 years ago
- 89c3de3 Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 11 years ago
- 34d1110 Enable indefinitely running vie_auto_test option by kjellander@webrtc.org · 11 years ago
- 4484b83 Use LOG_F interface for unsupported functions. by andrew@webrtc.org · 11 years ago
- f9ca8e1 Added labels in transform_neon.S in iSAC-fix, so the tables be shared with other files in iOS build. Also, moved several code lines in the same file, in case register values cannot be preserved after a function call which could cause a crash in some platforms (e.g. iOS etc.). by kma@webrtc.org · 11 years ago
- 313e6b5 Lint-cleaned video and audio receivers. by phoglund@webrtc.org · 11 years ago
- db325e2 Updated version number to 3.22 by elham@webrtc.org · 11 years ago
- 228e708 Moved almost all payload-related stuff to the payload registry. by phoglund@webrtc.org · 11 years ago
- cc895d1 Fixing/disabling Windows x64 warnings by kjellander@webrtc.org · 11 years ago
- eae59c5 Exchange TRY by enumerating image formats in Linux video capture by braveyao@webrtc.org · 11 years ago
- 9320328 Fix MaxChannels test; 32 -> 100. by andrew@webrtc.org · 11 years ago
- 48bfaa8 Remove (in practice) the voice engine channel limit. by andrew@webrtc.org · 11 years ago
- d6739c8 Adding a send side API for streaming by mikhal@webrtc.org · 11 years ago
- a7761c7 Fix mismatch between different NACK list lengths and packet buffers. by stefan@webrtc.org · 11 years ago
- 3442158 Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. by stefan@webrtc.org · 11 years ago
- 7e11001 Adding three frame sizes to Opus by tina.legrand@webrtc.org · 11 years ago
- 4d693f9 Implementing stereo support for G.722 by henrik.lundin@webrtc.org · 11 years ago
- fd2dd1a Set frame length for frame converting in external renderer by braveyao@webrtc.org · 11 years ago
- de55d0c Replaced relative path to reference from <(webrtc_root). by bjornv@webrtc.org · 11 years ago
- 2569ab5 Fix propagating RED paylaod-type to ACM. by turaj@webrtc.org · 11 years ago
- 57c45c2 Removing a codec from NetEq database has a bug. |funcDurationEst| is not updated. by turaj@webrtc.org · 11 years ago
- 6637489 fix for issue 281. by turaj@webrtc.org · 11 years ago
- 1f1321c fix issue 1322, accept -1 as default payload-type for redundant coding (FEC). by turaj@webrtc.org · 11 years ago
- 62564f1 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 · 11 years ago
- 8d759af VP8: Making key frame interval a tunnable parameter by mikhal@webrtc.org · 11 years ago
- 5f8b39f Fix NetEq4 unit tests for VS2012 by henrik.lundin@webrtc.org · 11 years ago
- ea85f98 Removing a hack for CNG by henrik.lundin@webrtc.org · 11 years ago
- 1e52bc2 Adding iSAC-fb support by henrik.lundin@webrtc.org · 11 years ago
- 3824adf Fix audio_e2e_test command line arguments by kjellander@webrtc.org · 11 years ago
- f8dc257 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 · 11 years ago
- 0bfd5f0 Re-committing r3428 by bjornv@webrtc.org · 11 years ago
- d8f84db Fixing problems in audio_decoder_unittests by henrik.lundin@webrtc.org · 11 years ago
- b51ee74 Disable iSAC fix test in audio_decoder_unittests by henrik.lundin@webrtc.org · 11 years ago
- a5b65e0 Re-enabling NetEqDecodingTest.TestBitExactness and .TestNetworkStatistics by henrik.lundin@webrtc.org · 11 years ago
- 6bf1c81 Enabling unit tests for NetEq4 in the bots by henrik.lundin@webrtc.org · 11 years ago
- 9243982 Fix a few small nits in NetEq4 by henrik.lundin@webrtc.org · 11 years ago
- 1cd0f31 Remove codereview.settings by henrik.lundin@webrtc.org · 11 years ago
- 1dd36c8 Revert 3428 by bjornv@webrtc.org · 11 years ago
- 7bf5944 Delay estimator wrapper API changes. This should finalize the changes to delay estimator making it work for multi-probe. by bjornv@webrtc.org · 11 years ago
- 11f64d3 Mac 64-bit compatibility for WebRTC. by henrike@webrtc.org · 11 years ago
- 54958f4 Initial upload of NetEq4 by henrik.lundin@webrtc.org · 11 years ago
- b4575c1 Fix webrtc compilation errors for Chrome Win64 by andrew@webrtc.org · 11 years ago
- 184b91c Set working dir for test run script + update resources by kjellander@webrtc.org · 12 years ago
- 437f62b Add <(DEPTH) to global includes by kjellander@webrtc.org · 12 years ago
- a13470d Optimize NACK list creation. by stefan@webrtc.org · 12 years ago
- 294f055 Fix Win64 warnings by kjellander@webrtc.org · 12 years ago
- 534c1ce Added tests for multiple near-end support. by bjornv@webrtc.org · 12 years ago
- aa3af37 Short CL: only name change. by bjornv@webrtc.org · 12 years ago
- 16f79ea Separated far-end handling in BinaryDelayEstimator. by bjornv@webrtc.org · 12 years ago
- ceca869 Moving ViE test files and deleting files no longer used. by mflodman@webrtc.org · 12 years ago
- 1de9d16 Fix path to perf Python scripts in test.gyp by kjellander@webrtc.org · 12 years ago
- d32e047 Reformatted rtp_sender: made lint clean. by phoglund@webrtc.org · 12 years ago
- d1f6087 Test launching script by kjellander@webrtc.org · 12 years ago
- b29af0e Moved several function pointer declarations in iSAC to isac initialization file. by kma@webrtc.org · 12 years ago
- 0664d36 Fixed text relocation code related to ARM assembly code. by kma@webrtc.org · 12 years ago
- ad89c14 Revert 3406 by kma@webrtc.org · 12 years ago
- 5cd9878 Revert 3405 by niklas.enbom@webrtc.org · 12 years ago
- 4e3c377 Moved all function pointer declarations in iSAC to a single place. by kma@webrtc.org · 12 years ago
- 3ffc265 Mainly hlundin's patch. by niklas.enbom@webrtc.org · 12 years ago
- 3165a5b Optimized WebRtcIsacfix_Time2Spec() for iSAC-Fix in ARM Neon processor. by kma@webrtc.org · 12 years ago
- 6d29497 Bug fix in WebRtcOpus_DurationEst by henrik.lundin@webrtc.org · 12 years ago
- 6caf203 Fix frame_editing_unittest.cc by kjellander@webrtc.org · 12 years ago
- 3d7848b Updated version number to 3.21 by elham@webrtc.org · 12 years ago
- db901d7 Fixes payload spelling error. by henrike@webrtc.org · 12 years ago
- 50e9567 RTP Receiver is now only deals with a receiver strategy. Cleaned up dependencies. by phoglund@webrtc.org · 12 years ago
- 4e629ff Replace AudioFrame's operator= with CopyFrom(). by andrew@webrtc.org · 12 years ago
- 0806dcf Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests. by stefan@webrtc.org · 12 years ago
- 81cfcb5 Remove '<(library)' in gyp files. by wjia@webrtc.org · 12 years ago
- 3da689c 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
- 16c9f55 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
- 3f6593a Remove <(library) from gyp file. by wjia@webrtc.org · 12 years ago
- 2d1a85d Posix Thread: Removes the setting of the run function to NULL which could cause data race. by henrike@webrtc.org · 12 years ago
- 99e89cd Make VoE handle longer delays by niklas.enbom@webrtc.org · 12 years ago
- 3d60f01 Adding timeEndPeriod to Synchronize function, see bug for details. by mflodman@webrtc.org · 12 years ago
- a09409a Extracted rtp receiver payload management to its own class, made video receiver depend on that instead. by phoglund@webrtc.org · 12 years ago
- d7debff Break out RtpClock to system_wrappers and make it more generic. by stefan@webrtc.org · 12 years ago
- fc37398 Convert psnr and ssim to strings before printing them. by stefan@webrtc.org · 12 years ago
- b40cae3 Add a counter to the video rtp play output filename. by stefan@webrtc.org · 12 years ago
- 087c593 Removing outdated comment by mikhal@webrtc.org · 12 years ago
- 70b88de Reformatted rtp_rtcp_impl*. by phoglund@webrtc.org · 12 years ago
- 32ad4a4 Made ViEToFileRenderer use a separate thread for rendering frames to file. by stefan@webrtc.org · 12 years ago
- 8ed17bb Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional. by phoglund@webrtc.org · 12 years ago
- 1b23416 logical 'and' of mutually exclusive tests is always false in ViECodecImpl::CodecValid() by braveyao@webrtc.org · 12 years ago
- 3d016e1 Fix android clang build. by wjia@webrtc.org · 12 years ago
- a3b8638 Fix android clang build. by wjia@webrtc.org · 12 years ago
- c1dd3c3 Fix simulated analog gain in audioproc. by andrew@webrtc.org · 12 years ago
- c6dee58 Remove extra line. by andrew@webrtc.org · 12 years ago