- 66ccc6e Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. by pwestin@webrtc.org · 11 years ago
- 9fe8f22 Refactor webrtc specific Event implementation to an EventFactory. by stefan@webrtc.org · 11 years ago
- d4caede Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 11 years ago
- b661eae Tool found: pass by value when pass by reference is better in system wrapper unit test. by henrike@webrtc.org · 11 years ago
- a284f1d Change intrinsic code in isac fix to let it pass chrome clang compiler. by kma@webrtc.org · 11 years ago
- 2c62fd9 Fixes issue detected by tool. by henrike@webrtc.org · 11 years ago
- 020e6ad Removed redundant VP8 width/height and made sure the generic width/height is set. by stefan@webrtc.org · 11 years ago
- 9b78141 Revert "Internal clean up: removing unused include line." by dwkang@webrtc.org · 11 years ago
- 1a8d06e Internal clean up: removing unused include line. by dwkang@webrtc.org · 11 years ago
- 2d2bfb0 Fixed issue 1497 in iSAC fixed point. by kma@webrtc.org · 11 years ago
- 38a5679 Fix frame_editing_unittest reference file handling. by kjellander@webrtc.org · 11 years ago
- c51b060 Optimized EstCodeLpcCoef() for iSAC with intrinsics in Android-Neon platform. by kma@webrtc.org · 11 years ago
- 2a3949f Lazy capture_device_info acquisition. by pbos@webrtc.org · 11 years ago
- 46672bb Refactor barcode decoder to use Zxing's C++ version by kjellander@webrtc.org · 11 years ago
- c96125a Splitting out video_coding_test executable again. by kjellander@webrtc.org · 11 years ago
- ad807de Fixed an assembly code error in AECM for ARMv7. by kma@webrtc.org · 11 years ago
- 2dbb66b Disable frame dropper for screenshare mode. by stefan@webrtc.org · 11 years ago
- 2a070a5 Move video_coding OWNERS to video_coding/. by stefan@webrtc.org · 11 years ago
- 2733e12 Fixed a ton of Python lint errors, enabled python lint checking. by phoglund@webrtc.org · 11 years ago
- 4621446 Fix debug file buffer bug introduced in r3574. by andrew@webrtc.org · 11 years ago
- ace0823 Enabling bufffering mode with no sync module or VoE by mikhal@webrtc.org · 11 years ago
- 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