Log - 365ca407f428f03d0882531073974263787201e6 - fp2-dev/platform/external/chromium_org/third_party/webrtc

365ca40 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
0c0795e Fix broken audio. by leozwang@webrtc.org · 11 years ago
a0bba27 G722-stereo has been missing when creating AudioDecoder. by turaj@webrtc.org · 11 years ago
88a7940 NetEq4 fails if the first packets inserted in are out-of-band DTMFs. by turaj@webrtc.org · 11 years ago
88f12ab Fix flakiness in network up/down event tests when running under memcheck. by stefan@webrtc.org · 11 years ago
6fc5215 WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14. by fischman@webrtc.org · 11 years ago
dca71b2 Add interface to signal a network down event. by stefan@webrtc.org · 11 years ago
6e34ceb Split condition_variable_win.cc into native (for Vista and newer OS versions) and generic implementation (based on events). by henrike@webrtc.org · 11 years ago
f386e2b Remove VoE's default call in Trace::SetLevelFilter. by andrew@webrtc.org · 11 years ago
aa0fcd7 Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust. by solenberg@webrtc.org · 11 years ago
31b4448 Alphabetize include order in fake_voe_external_media.h. by andrew@webrtc.org · 11 years ago
a7a643e Restart Android capture after orientation change. Also prevent an NPE on exit. by fischman@webrtc.org · 11 years ago
13f66d1 Add some VoE and AudioProcessing mocks. by andrew@webrtc.org · 11 years ago
8826e34 Refactor unittest trace printouts to a separate class. by andrew@webrtc.org · 11 years ago
0c1f10b Enable the below APIs for iOS. by sjlee@webrtc.org · 11 years ago
9522792 Introduced pause and resume to the pacer by pwestin@webrtc.org · 11 years ago
f49577f Updated WebRTC version to 3.27 by elham@webrtc.org · 11 years ago
7fd368f Bugfix for extended RTP/RTCP test by pwestin@webrtc.org · 11 years ago
c075e25 Move the VIE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 11 years ago
4c27c03 Add trace printouts to all unit tests. by andrew@webrtc.org · 11 years ago
e1198e6 Add min and target bitrate to VideoCodec. by marpan@webrtc.org · 11 years ago
aa922de Move the VoE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 11 years ago
af6aa7b Creating a copy of Udp transport under webrtc/test by pwestin@webrtc.org · 11 years ago
495c563 Cleanup nanosleep -> SleepMs Remove some leftover stuff by hta@webrtc.org · 11 years ago
4a48fd6 WebRtc_Word -> stdint in audio_coding/g711/ by pbos@webrtc.org · 11 years ago
a2df078 Remove incorrect asserts. by stefan@webrtc.org · 11 years ago
e49f252 WebRtc_Word -> stdint in audio_coding/cng/ by pbos@webrtc.org · 11 years ago
0f2782f Fix -Wstring-conversion warnings. by wu@webrtc.org · 11 years ago
5815b7c Thread safety issue fix in incoming_video_stream.cc. See issue 1465. by vikasmarwaha@webrtc.org · 11 years ago
757bf0f Account for header inside I420Encoder::InitEncode. by pbos@webrtc.org · 11 years ago
6313692 Follow-up fix for r3681. by stefan@webrtc.org · 11 years ago
46618db Fixed initialization of SPL in echo_control_mobile. by kma@webrtc.org · 11 years ago
9cd73ed Android: rename android_build_type gyp variable. by wjia@webrtc.org · 11 years ago
ffe2ec6 Updated WebRTC version number to 3.26 by elham@webrtc.org · 11 years ago
692c69d Fix framerate sent to account for actually sent frames. by stefan@webrtc.org · 11 years ago
72e204a Change VCM interface to take target bitrate in bits per second. by stefan@webrtc.org · 11 years ago
e3339fc Generic video-codec support. by pbos@webrtc.org · 11 years ago
0b5c7f1 Revert the deletion of test_api_nack.cc in r3674. by stefan@webrtc.org · 11 years ago
87ef38e Truncated delay quality to avoid negative return values by bjornv@webrtc.org · 11 years ago
946d240 Adding RTX on source by mikhal@webrtc.org · 11 years ago
cf9ab12 Adding Opus frame length test by tina.legrand@webrtc.org · 11 years ago
aecc559 Fixed a crash issue in NSX module. by kma@webrtc.org · 11 years ago
912b7f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
40749c1 Added destructors for tests to control destruct order by pwestin@webrtc.org · 11 years ago
b793abe Increasing size of nack list in buffered mode. by mikhal@webrtc.org · 11 years ago
2daec4c 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
2637d61 Refactor webrtc specific Event implementation to an EventFactory. by stefan@webrtc.org · 11 years ago
15a03fd Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 11 years ago
df00856 Tool found: pass by value when pass by reference is better in system wrapper unit test. by henrike@webrtc.org · 11 years ago
0c46af6 Change intrinsic code in isac fix to let it pass chrome clang compiler. by kma@webrtc.org · 11 years ago
160b327 Fixes issue detected by tool. by henrike@webrtc.org · 11 years ago
f0f1dc2 Removed redundant VP8 width/height and made sure the generic width/height is set. by stefan@webrtc.org · 11 years ago
23a7047 Revert "Internal clean up: removing unused include line." by dwkang@webrtc.org · 11 years ago
177ec87 Internal clean up: removing unused include line. by dwkang@webrtc.org · 11 years ago
d451969 Fixed issue 1497 in iSAC fixed point. by kma@webrtc.org · 11 years ago
9f4ae03 Fix frame_editing_unittest reference file handling. by kjellander@webrtc.org · 11 years ago
6b12f97 Optimized EstCodeLpcCoef() for iSAC with intrinsics in Android-Neon platform. by kma@webrtc.org · 11 years ago
8623e9b Lazy capture_device_info acquisition. by pbos@webrtc.org · 11 years ago
7b6cbc9 Refactor barcode decoder to use Zxing's C++ version by kjellander@webrtc.org · 11 years ago
04b8071 Splitting out video_coding_test executable again. by kjellander@webrtc.org · 11 years ago
e4a0d58 Fixed an assembly code error in AECM for ARMv7. by kma@webrtc.org · 11 years ago
02c9104 Disable frame dropper for screenshare mode. by stefan@webrtc.org · 11 years ago
e33b9de Move video_coding OWNERS to video_coding/. by stefan@webrtc.org · 11 years ago
2ce235e Fixed a ton of Python lint errors, enabled python lint checking. by phoglund@webrtc.org · 11 years ago
fe6b986 Fix debug file buffer bug introduced in r3574. by andrew@webrtc.org · 11 years ago
78e450f Enabling bufffering mode with no sync module or VoE by mikhal@webrtc.org · 11 years ago
0f919be Remove the error return on SetAGC failure introduced by r3605. by andrew@webrtc.org · 11 years ago
7f34a88 1. Updated test pages to include Chrome Frame meta tag by elham@webrtc.org · 11 years ago
4c2225f Adds new AEC API to audio_processing. by bjornv@webrtc.org · 11 years ago
3c365b2 Fix for build error on android introduced with r3609. by stefan@webrtc.org · 11 years ago
15fc445 Split the NACK list into multiple RTCPs if it's too big. by stefan@webrtc.org · 11 years ago
8665399 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
b79627b Expose the capture-side AudioProcessing object and allow it to be injected. by andrew@webrtc.org · 11 years ago
d432df4 AEC Refactoring: Removes lint warning by bjornv@webrtc.org · 11 years ago
640d1bb Updated version number to 3.25 by elham@webrtc.org · 11 years ago
bb78b2c Refactor NACK list creation to build the NACK list as packets arrive. by stefan@webrtc.org · 11 years ago
e294b5f compile fix for get_nprocs() with uClibc by phoglund@webrtc.org · 11 years ago
46e08da Fixed coverity defects (CID 14657 and 14656). by phoglund@webrtc.org · 11 years ago
3d1a637 VideoCaptureAndroid can now capture just buffers without also rendering to a SurfaceView. by fischman@webrtc.org · 11 years ago
4de0a10 Don't upsample the capture signal early. by andrew@webrtc.org · 11 years ago
03c41d7 Update integration tests for idempotent RTP header settings. by bemasc@google.com · 11 years ago
ea67fe0 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
b563e5e Properly error check calls to AudioProcessing. by andrew@webrtc.org · 11 years ago
1794994 Enable External MediaProcessing on Mobile by leozwang@webrtc.org · 11 years ago
d1932fc Make RtpHeaderExtensionMap::Register and ::Deregister idempotent. by bemasc@google.com · 11 years ago
f1de5e9 Return an error when greater than 16 kHz is used with AECM. by andrew@webrtc.org · 11 years ago
c6242c9 Destroy VCM and VPM instead of delete. by mflodman@webrtc.org · 11 years ago
f653f97 Limit ARM instruction "strheq" to Apple's clang compiler only. by kma@webrtc.org · 11 years ago
d836fe5 Turn off error concealment in videoprocessor_integration tests. by marpan@webrtc.org · 11 years ago
0e686e6 Add supporting to V4L2_PIX_FMT_JPEG since it works same as MJPEG. by braveyao@webrtc.org · 11 years ago
1827258 Rewrite the jitter buffer statistics test and put make it robust under valgrind. by stefan@webrtc.org · 11 years ago
840fa03 AEC Refactoring: by bjornv@webrtc.org · 11 years ago
6e7945f Fix to send a full NACK list at least roughly once every 1.5 x RTT. by stefan@webrtc.org · 11 years ago
5666f6c Fixed a bug in WebRtcNsx_PrepareSpectrumNeon() for NS in ARM Neon platform. by kma@webrtc.org · 11 years ago
db32d60 Refactor WebRtc_CreateBuffer to return the instance. by andrew@webrtc.org · 11 years ago
001d6ed Use ninja with merge_libs.py on Windows and clean up. by andrew@webrtc.org · 11 years ago
ef31f50 Force a memcpy directly from the AEC ring buffer. by andrew@webrtc.org · 11 years ago
479f9f4 Remove unneeded libvpx path from vp8 include_dirs. by andrew@webrtc.org · 11 years ago
b3e5a62 Refactor ring_buffer interface, add a feature and a test. by andrew@webrtc.org · 11 years ago
d9b6942 New attempt at fixing hard-coded libvpx source. by phoglund@webrtc.org · 11 years ago