Log - 1b6da28047ccc8ac50a2e2b09c142bea7679761a - platform/external/webrtc

1b6da28 Bugfix for NACK behavior. Current code sends a number of duplicate NACK requests. by pwestin@webrtc.org · 12 years ago
f556890 Added possibility to repeat frames. Also added unittest for that feature. by brykt@google.com · 12 years ago
d73527c Changed assert to log. by mflodman@webrtc.org · 12 years ago
d0d4149 Adding AUDIO application as default for Opus stereo by tina.legrand@webrtc.org · 12 years ago
ad0ed58 Fixed a missed initialization (found by valgrind FYI bot). by phoglund@webrtc.org · 12 years ago
ac77084 Roll opus to 172355 and delete opus_demo from webrtc opus by leozwang@webrtc.org · 12 years ago
6bc5d4d Reformatted sort. by phoglund@webrtc.org · 12 years ago
1960219 Make protection method, filename and resolution configurable for FullStackTest. by stefan@webrtc.org · 12 years ago
4275ab1 Implement NetEq duration estimation for Opus. by tina.legrand@webrtc.org · 12 years ago
515ef24 Clean up variable after it gets deleted by leozwang@webrtc.org · 12 years ago
e239bf0 Making I420VideoFrame ref-counted by mikhal@webrtc.org · 12 years ago
b13dfbf Making barcode tools work on Windows + fixes. by kjellander@webrtc.org · 12 years ago
0b18fb3 vie auto test: Adding a constructor for NetworkParameters by mikhal@webrtc.org · 12 years ago
622c8bd ViE autotest: Adding loss models to the external transport by mikhal@webrtc.org · 12 years ago
6e0ce73 Reformatted map classes. by phoglund@webrtc.org · 12 years ago
61f39a3 Fixed bad header name. by phoglund@webrtc.org · 12 years ago
07bf43c Replaced the _audio parameter with a strategy. by phoglund@webrtc.org · 12 years ago
59ad541 Reformatted rw_lock classes. by phoglund@webrtc.org · 12 years ago
eaebeb3 Without specifying the input files the offsets will not automatically be regenerated when building for different architectures. That is very risky as it will cause crashes rather than build errors. by stefan@webrtc.org · 12 years ago
10abe25 Make audioproc output files be written to output dir by default. by kjellander@webrtc.org · 12 years ago
3c37354 Initialize 3 variables which are preventing VS2012 from building. by fbarchard@google.com · 12 years ago
4c32439 Roll libyuv to r520. Includes security fix to mark stack as not executable. by fbarchard@google.com · 12 years ago
ad6845f Updated version number to 3.19 by elham@webrtc.org · 12 years ago
c5fcb08 Update trace_event.h to match the one in Chromium by hclam@chromium.org · 12 years ago
dec09ee libyuv r515 ports matrix effects to Neon by fbarchard@google.com · 12 years ago
4aee6b6 Added API to get receive side video delay. by mflodman@webrtc.org · 12 years ago
1c75918 Disabled flaky test. by phoglund@webrtc.org · 12 years ago
7659d91 Decoupled video rtp receiver from rtp receiver. by phoglund@webrtc.org · 12 years ago
52d981f Reformatted list classes. by phoglund@webrtc.org · 12 years ago
3251939 Remove latency excl network and add render time diff stats. by stefan@webrtc.org · 12 years ago
b8ba4d8 Add number of inserted samples to NetEq statistics. by roosa@google.com · 12 years ago
c454fab Reformatting ACM. All changes are bit-exact in this CL. by turaj@webrtc.org · 12 years ago
ddebc17 Fix for buffer overflow, WebRTC issue 1196 by elham@webrtc.org · 12 years ago
96dc627 vpm unit test: Diasble frame dropping in tests by mikhal@webrtc.org · 12 years ago
4493db5 vpm: removing unnecessary memcpy by mikhal@webrtc.org · 12 years ago
7acb65a Added jitter to fake network pipe. by mflodman@webrtc.org · 12 years ago
91c91df Track the actual render time rather than the decode time. by stefan@webrtc.org · 12 years ago
e19b078 Changed so that frame_cutter takes and argument where one can specify in which interval the frames should be deleted between the first frame to cut and the last frame to cut. This can for example be used to decrease the frame rate. by brykt@google.com · 12 years ago
0240e8e Wider TSAN suppression for issue 300 by kjellander@webrtc.org · 12 years ago
92bb417 Decoupled RTP audio processor from RTP receiver. by phoglund@webrtc.org · 12 years ago
5b689ef Will now only require near-perfect PSNR and SSIM. by phoglund@webrtc.org · 12 years ago
86464ea ISAC_main_inst initialized to NULL to avoid potentially garbage pointer passed to WebRtcIsacfix_EncoderInit by fbarchard@google.com · 12 years ago
a8544ea Vp8 tests: Removing legacy unused tests and reorganization of existing ones. by mikhal@webrtc.org · 12 years ago
7877b0f Added noexecstack markers for assembly files (webrtc issue 1172). by kma@webrtc.org · 12 years ago
fa5b6bf Optimized WebRtcIsacfix_Spec2Time() for iSAC-Fix in ARM Neon processor. Speed doubled. by kma@webrtc.org · 12 years ago
1b60ceb Add GetAudioFrame API to VoiceEngine. by roosa@google.com · 12 years ago
b718619 Expose NetEq playout mode off through VoiceEngine. by roosa@google.com · 12 years ago
0870f02 Add API to retreive last received RTP timestamp to VoiceEngine. by roosa@google.com · 12 years ago
d8aeb30 Revert 3269 by andrew@webrtc.org · 12 years ago
735a6ce Will now only require near-perfect PSNR and SSIM. by phoglund@webrtc.org · 12 years ago
740be44 Reformatted file_* classes. by phoglund@webrtc.org · 12 years ago
4e16f25 Remove atomicops.h from WebRTC by hclam@chromium.org · 12 years ago
9f0fc97 Rolllibvpx to 7a09f6b89268 by marpan@webrtc.org · 12 years ago
770a01e Fix build by including trace_event_internal in webrtc namespace by hclam@chromium.org · 12 years ago
f222a00 Use TRACE_EVENT to track time spent in VP8 encoding by hclam@chromium.org · 12 years ago
d2bcde2 Suppressing TSan warnings for system_wrappers_unittests by kjellander@webrtc.org · 12 years ago
ad7efa6 Port Chromium's trace_event.h to WebKit and add by hclam@chromium.org · 12 years ago
02d9df4 Updated webrtc_resources_revision to 11, for adding two test files for APM and iSAC. by kma@webrtc.org · 12 years ago
71258c5 Add a third full stack test and support for random jitter in ext transport. by stefan@webrtc.org · 12 years ago
eaf7cf2 Adding a simple fake network pipe to use for testing. Next CL will contain an external transport implementation using this link and I'll follow up later making this more advanced. by mflodman@webrtc.org · 12 years ago
f98ffc6 Removing default trybot names by kjellander@webrtc.org · 12 years ago
42259e7 VoE Changes to enable dual_streaming. by turaj@webrtc.org · 12 years ago
36965b1 Bug fix for iSAC fixed-point. The bug was the result of changes in iSAC floating-point to add 48 kHz extension. by turaj@webrtc.org · 12 years ago
55edaec Revert r3254 due to bot failure on android. by marpan@webrtc.org · 12 years ago
1f3476d Roll libvpx to 000c8414b510. by marpan@webrtc.org · 12 years ago
5bbe069 Reformatted event* classes. by phoglund@webrtc.org · 12 years ago
3bb42ef Made e2e audio quality test write its results to perf. by phoglund@webrtc.org · 12 years ago
72feb0b Not to enum NOTPRESENT audio devices with CoreAudio on Win by braveyao@webrtc.org · 12 years ago
8e49b02 Add more audio codec information into codec list by leozwang@webrtc.org · 12 years ago
451aa5d Adding vp8 sequence coder: simple command line encode and decode. by mikhal@webrtc.org · 12 years ago
3a5a8a8 Properly zero out unmixed frames. by andrew@webrtc.org · 12 years ago
0e73950 Added buildbot benchmarking in iSAC and APM into Android platform build. by kma@webrtc.org · 12 years ago
b968213 vp8 test: Updating creation of enc/dec by mikhal@webrtc.org · 12 years ago
251f64e Updating vp8 test structure by mikhal@webrtc.org · 12 years ago
60d25f9 Updating Vp8 unit tests - Initiating the switch to gtest-based tests, and adding a stride test. by mikhal@webrtc.org · 12 years ago
75f8c78 Fixing path to ptypes.txt in NetEqRTPplay by henrik.lundin@webrtc.org · 12 years ago
df94329 Use different cpufeatures library when building with chrome. by wjia@webrtc.org · 12 years ago
81cffd1 Port Chromium's atomicops to WebRTC by hclam@chromium.org · 12 years ago
63a243a Replace the last occurrence of .s with .h by leozwang@webrtc.org · 12 years ago
96bcac8 Expose Set and Get Recording/Playout sample rate apis by leozwang@webrtc.org · 12 years ago
f4e070e Added auto-call feature to WebRTCDemo. by fischman@webrtc.org · 12 years ago
2cf22a6 Revert 3231 - VoE Changes to enable dual_streaming. by perkj@webrtc.org · 12 years ago
e861359 Adds two full stack performance metrics for end-to-end delay. by stefan@webrtc.org · 12 years ago
6bd737a First pass of MediaCodecDecoder which uses Android MediaCodec API. by dwkang@webrtc.org · 12 years ago
781cf06 libyuv r508 with scaler fix for overread horizontally that was caught by valgrind. by fbarchard@google.com · 12 years ago
767d87c VoE Changes to enable dual_streaming. by turaj@webrtc.org · 12 years ago
226db89 Dual-stream implementation, not including VoE APIs. by turaj@webrtc.org · 12 years ago
277ec8e Fix a bug when iSAC-48kHz was added. by turaj@webrtc.org · 12 years ago
f18de86 Revert 3227 by mikhal@webrtc.org · 12 years ago
ab83bb3 vp8 unittest: Adding qcif stride test by mikhal@webrtc.org · 12 years ago
b0dff12 48 kHz extension to iSAC. by turaj@webrtc.org · 12 years ago
0bacb63 Removed stale version of fuzzer; it's now internal. by phoglund@webrtc.org · 12 years ago
8d0cd07 Add test to verify that padding only frames are passing through the RTP module. by stefan@webrtc.org · 12 years ago
5b4fe49 Changing default bitrate to 64000 bps for Opus. by tina.legrand@webrtc.org · 12 years ago
ad0f3ba Removing redundant codec unittest targets. by kjellander@webrtc.org · 12 years ago
ba21c95 Reformatted data_log. by phoglund@webrtc.org · 12 years ago
c94f8d4 Fix OOB read in padding tests. by stefan@webrtc.org · 12 years ago
78bec2d Fixed bug where we would rewrite *deref_ptr = ...; to // deref_ptr = ...; by phoglund@webrtc.org · 12 years ago
fc4a7ee Fixes chromium build bots. by henrike@webrtc.org · 12 years ago
c7896df Fixed bug that caused frame_cutter_unittest to fail when built with MVS2008. by brykt@google.com · 12 years ago