- 66b0c5d Remove the error return on SetAGC failure introduced by r3605. by andrew@webrtc.org · 11 years ago
- ad3fd52 1. Updated test pages to include Chrome Frame meta tag by elham@webrtc.org · 11 years ago
- 333987b Adds new AEC API to audio_processing. by bjornv@webrtc.org · 11 years ago
- 08b9b59 Fix for build error on android introduced with r3609. by stefan@webrtc.org · 11 years ago
- 2654c43 Split the NACK list into multiple RTCPs if it's too big. by stefan@webrtc.org · 11 years ago
- df1cfd1 None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise. by turaj@webrtc.org · 11 years ago
- 6316d17 Expose the capture-side AudioProcessing object and allow it to be injected. by andrew@webrtc.org · 11 years ago
- 2c1f9d4 AEC Refactoring: Removes lint warning by bjornv@webrtc.org · 11 years ago
- 87d8f2d Updated version number to 3.25 by elham@webrtc.org · 11 years ago
- eeaacdb Refactor NACK list creation to build the NACK list as packets arrive. by stefan@webrtc.org · 11 years ago
- 552f230 compile fix for get_nprocs() with uClibc by phoglund@webrtc.org · 11 years ago
- 89cc166 Fixed coverity defects (CID 14657 and 14656). by phoglund@webrtc.org · 11 years ago
- 4aa2314 VideoCaptureAndroid can now capture just buffers without also rendering to a SurfaceView. by fischman@webrtc.org · 11 years ago
- 7d2689d Don't upsample the capture signal early. by andrew@webrtc.org · 11 years ago
- 3da576e Update integration tests for idempotent RTP header settings. by bemasc@google.com · 11 years ago
- 13a186f Refactored inline assembly code in complex_fft.c, by combining the individual __asm lines into a single block, to avoid potential register usage problems when building with different tools. by kma@webrtc.org · 11 years ago
- ad179ce Properly error check calls to AudioProcessing. by andrew@webrtc.org · 11 years ago
- 8e4340d Enable External MediaProcessing on Mobile by leozwang@webrtc.org · 11 years ago
- 8648aad Make RtpHeaderExtensionMap::Register and ::Deregister idempotent. by bemasc@google.com · 11 years ago
- bb2973a Return an error when greater than 16 kHz is used with AECM. by andrew@webrtc.org · 11 years ago
- 1dcba31 Destroy VCM and VPM instead of delete. by mflodman@webrtc.org · 11 years ago
- 22fc115 Limit ARM instruction "strheq" to Apple's clang compiler only. by kma@webrtc.org · 11 years ago
- fef10a3 Turn off error concealment in videoprocessor_integration tests. by marpan@webrtc.org · 11 years ago
- 51d5c6d Add supporting to V4L2_PIX_FMT_JPEG since it works same as MJPEG. by braveyao@webrtc.org · 11 years ago
- eeb8b8f Rewrite the jitter buffer statistics test and put make it robust under valgrind. by stefan@webrtc.org · 11 years ago
- 12509cf AEC Refactoring: by bjornv@webrtc.org · 11 years ago
- 6f93416 Fix to send a full NACK list at least roughly once every 1.5 x RTT. by stefan@webrtc.org · 11 years ago
- 9ea9696 Fixed a bug in WebRtcNsx_PrepareSpectrumNeon() for NS in ARM Neon platform. by kma@webrtc.org · 11 years ago
- 24e40df Refactor WebRtc_CreateBuffer to return the instance. by andrew@webrtc.org · 11 years ago
- 90aa2fe Use ninja with merge_libs.py on Windows and clean up. by andrew@webrtc.org · 11 years ago
- c60d33b Force a memcpy directly from the AEC ring buffer. by andrew@webrtc.org · 11 years ago
- 86e2e1f Remove unneeded libvpx path from vp8 include_dirs. by andrew@webrtc.org · 11 years ago
- 101eb2c Refactor ring_buffer interface, add a feature and a test. by andrew@webrtc.org · 11 years ago
- 4211e6a New attempt at fixing hard-coded libvpx source. by phoglund@webrtc.org · 11 years ago
- bf03dd4 Revert "Fixing hard-coded libvpx source path." by phoglund@webrtc.org · 11 years ago
- 7f53b4c Fixing hard-coded libvpx source path. by phoglund@webrtc.org · 11 years ago
- 933af52 Ported assembly coding in APM from Android to iOS. by kma@webrtc.org · 11 years ago
- ca65c51 Handle multiple calls to set initial delay by mikhal@webrtc.org · 11 years ago
- c1f0f68 Remove WEBRTC_TRACE completely when tracing is disabled. by wjia@webrtc.org · 11 years ago
- fa9a633 Minor bug fix in maxFPS parameter declaration. by vikasmarwaha@webrtc.org · 11 years ago
- c1c5aad Fix for WebRTC Issue 1384. Some cameras return 0 fps for all capabilities which causes divide-by-zero. by vikasmarwaha@webrtc.org · 11 years ago
- 85e32df MIPS optimizations for Signal Processing Library patch01 by andrew@webrtc.org · 11 years ago
- ed1c7f4 AEC refactoring: Moved typedefs to _internal.h by bjornv@webrtc.org · 11 years ago
- 4f33453 Changing non-const reference arguments to pointers, ACM by tina.legrand@webrtc.org · 11 years ago
- db1733f Misc cleanups to webrtc/android code: by fischman@webrtc.org · 11 years ago
- c01c6c3 Refactoring AEC: AecCore struct made private by bjornv@webrtc.org · 11 years ago
- 325931a Refactor AEC: PowerLevel by bjornv@webrtc.org · 11 years ago
- 798195e Added a pointer getter to the system_delay variable. by bjornv@webrtc.org · 11 years ago
- 191efa0 Refactoring AEC: Added a SetConfigCore function by bjornv@webrtc.org · 11 years ago
- e27e49b Moved out buffer handling to ProcessFrame() by bjornv@webrtc.org · 11 years ago
- 5efb2ee Removed unused get_config function. The configuration is already stored and handled in the audio processing module, so there is no need for this functionality. by bjornv@webrtc.org · 11 years ago
- 213217c Stop and restart fix. by mflodman@webrtc.org · 11 years ago
- e563de7 Revert 3543 by tina.legrand@webrtc.org · 11 years ago
- fec9a9f Changing non-const reference arguments to pointers, ACM by tina.legrand@webrtc.org · 11 years ago
- 2325284 Fixed typo in vie_autotest_loopback.cc. by pbos@webrtc.org · 11 years ago
- 83db9e9 Replace gtest_prod.h include with our own FRIEND_TEST macro. by andrew@webrtc.org · 11 years ago
- cb139b1 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 11 years ago
- 9c643ec Added getter for far_time_buf in AEC. Only used in AEC debug dump. by bjornv@webrtc.org · 11 years ago
- 38417a8 This refactoring CL moves the nlp_mode member value from aecpc_t to aec_t, since it it never used at that level. Further, I removed two suppression variables by depending on nlp_mode directly. by bjornv@webrtc.org · 11 years ago
- ef1346d * Name change * Removed WebRtcAec_ function name prepending on private function. by bjornv@webrtc.org · 11 years ago
- 24ba537 Update to codec unit test: by marpan@webrtc.org · 11 years ago
- 432bc1a fixing nack list size calculation by mikhal@webrtc.org · 11 years ago
- 39eb955 Updated version number to 3.24 by elham@webrtc.org · 11 years ago
- 5962e5e Remove the dependency on dxguid.lib. by tommi@webrtc.org · 11 years ago
- 9e3e8e5 Move directx_sdk_path definition variable into the video_render_module gyp file. by tommi@webrtc.org · 11 years ago
- 85e2e0e Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings. by stefan@webrtc.org · 11 years ago
- ce3f2ca Add VoE interface to VieRTP test by mikhal@webrtc.org · 11 years ago
- b115f2c Increase threshold in codec unit test. by marpan@webrtc.org · 11 years ago
- 4db69af Adding a receive side API for buffering mode. by mikhal@webrtc.org · 11 years ago
- 95f5fd9 Bug fix for webrtc issue 1391. Typo in sin_length for socket address. by vikasmarwaha@webrtc.org · 11 years ago
- e422fa5 This refactoring CL contains an API to get low level echo metrics stats. by bjornv@webrtc.org · 11 years ago
- 3942fd8 This Cl includes by bjornv@webrtc.org · 11 years ago
- d3eadf1 Moved the actual calculations to aec_core to avoid passing up low level members. by bjornv@webrtc.org · 11 years ago
- a974cea Make VoiceEngineImpl inherit from VoiceEngine. by tommi@webrtc.org · 11 years ago
- 1aa1eec Modify SincResampler to build in webrtc. by andrew@webrtc.org · 11 years ago
- 3e32dd1 Duplicated sampling frequency multiplier to aecpc_t struct to avoid a getter. by bjornv@webrtc.org · 11 years ago
- 64506e2 Roll Chromium revision 176094:182149 by kjellander@webrtc.org · 11 years ago
- 8504ad3 Moved debug file handling to aec_core from echo_cancellation.c. This removes dependency on low level member variables. by bjornv@webrtc.org · 11 years ago
- 4a0743e Added delay estimation test to audio processing unit tests. by bjornv@webrtc.org · 11 years ago
- e740a7b Remove MultiStreamMode from test. by stefan@webrtc.org · 11 years ago
- 4c6689a Reset ssrc when calling SetSendCodec. by mflodman@webrtc.org · 11 years ago
- abaff53 Fixing lint warnings from previous commit by tina.legrand@webrtc.org · 11 years ago
- 1368a6a Import stringize_macros from Chromium. by andrew@webrtc.org · 11 years ago
- 260bedc Import SincResampler from Chromium. by andrew@webrtc.org · 11 years ago
- 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