Log - 02270cd718fd2047bbbf99fbe344e3d988480b57 - platform/external/webrtc

02270cd Implementing a packet router class, used to route RTP packets to the by mflodman@webrtc.org · 10 years ago
10a9e92 Fix delete of stack allocated object causing test crashes. by stefan@webrtc.org · 10 years ago
4b320cf Revert "Cleanup: unify rotation to be enum based instead of int for degree." by magjed@webrtc.org · 10 years ago
fb609a1 Wire up new feedback format by introducing a FeedbackPacket type. by stefan@webrtc.org · 10 years ago
353c8b8 audio_processing/agc: Changed to correct include path in agc_unittests by bjornv@webrtc.org · 10 years ago
bc3241a Update ProcessCallAfterXms to better match the performance of our faster bots. Previously I had made sure these tests didn't flake out on our slow trybots, but apparently I need to do the same for the fast bots :) by tommi@webrtc.org · 10 years ago
0c3e12b Revamp the ProcessThreadImpl implementation. by tommi@webrtc.org · 10 years ago
7502543 Base RWLockWrapper on rtc::SharedExclusiveLock. by pbos@webrtc.org · 10 years ago
5e05731 Roll chromium_revision cd35af6..598c3e9 by kjellander@webrtc.org · 10 years ago
57ac2c8 Default destination used by c line should be IPv4 only to avoid parsing error in legacy client. by guoweis@webrtc.org · 10 years ago
3e733a4 Cleanup: unify rotation to be enum based instead of int for degree. by guoweis@webrtc.org · 10 years ago
74d2788 Remove defined(__cplusplus) tests in C++ code. by jan.skoglund@webrtc.org · 10 years ago
f45c8ca Reland r8248 "Introduce ACMGenericCodecWrapper" by henrik.lundin@webrtc.org · 10 years ago
ec4521c Clean up Beamformer initialization by aluebs@webrtc.org · 10 years ago
e69220c Fix the value of the first byte of nal unit generated by fake H.264 encoder. by glaznev@webrtc.org · 10 years ago
f693229 Fix Android video renderer to support video frames with stride > width. by glaznev@webrtc.org · 10 years ago
cc64a9c voice_engine: Updates GetEcDelayMetrics() w.r.t. new metric by bjornv@webrtc.org · 10 years ago
4b9622f Roll gtest-parallel. by pbos@webrtc.org · 10 years ago
3a87630 Revert r8248 "Introduce ACMGenericCodecWrapper" by henrik.lundin@webrtc.org · 10 years ago
af8c13f Introduce ACMGenericCodecWrapper by henrik.lundin@webrtc.org · 10 years ago
5d32f43 Disable CondVarTest.InitFunctionsWork. by tommi@webrtc.org · 10 years ago
877ac76 Cleanup and prepare for bundling. by pthatcher@webrtc.org · 10 years ago
cf7efeb Add new AudioEncoderOpusTest by henrik.lundin@webrtc.org · 10 years ago
520a69e Revert 8238 "Add RefCounting for TransportProxies" by bjornv@webrtc.org · 10 years ago
875c97e Remove SetNotAlive method from the thread class. by tommi@webrtc.org · 10 years ago
c5f6971 Revert 8237 "Cleanup and prepare for bundling." by bjornv@webrtc.org · 10 years ago
dc096f2 system_wrappers: Disabled flaky test CondVarTest.PassBatonMultipleTimes by bjornv@webrtc.org · 10 years ago
4414939 Add method for incrementing RtpPacketCounter. Removes duplicate code. by asapersson@webrtc.org · 10 years ago
e250667 Add RefCounting for TransportProxies by decurtis@webrtc.org · 10 years ago
af01d93 Cleanup and prepare for bundling. by pthatcher@webrtc.org · 10 years ago
322a564 Fix datachannel stats id and timestamp. by decurtis@webrtc.org · 10 years ago
d43bdf5 Rewrite ThreadPosix. by tommi@webrtc.org · 10 years ago
bfdee69 Roll chromium_revision 9070a80..cd35af6 (313233:314322) by kjellander@webrtc.org · 10 years ago
0ec50be Changing include guard in frame_callback.h. by mflodman@webrtc.org · 10 years ago
200ac00 Remove temp files in audio_processing_unittest.cc. by pbos@webrtc.org · 10 years ago
0e8bf6c Enable bitrate probing by default. by stefan@webrtc.org · 10 years ago
b1786db audio_processing: Added a new AEC delay metric value that gives the amount of poor delays by bjornv@webrtc.org · 10 years ago
0e81fdf Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting. by pkasting@chromium.org · 10 years ago
19f3f71 Fix apparent typo: int -> char. by pkasting@chromium.org · 10 years ago
946ad76 Switched lists of packets to lists of packet pointers. Allows Packet polymorphism. by stefan@webrtc.org · 10 years ago
c957ffc Fixed potential crash if rtp packet history is completely full. by sprang@webrtc.org · 10 years ago
c420a86 Change name for local CriticalSectionScoped variable by henrik.lundin@webrtc.org · 10 years ago
a1dfbf1 WebRtcG722_Decode: Input array should be const uint8_t[] by kwiberg@webrtc.org · 10 years ago
026b892 Using << on an int8_t or uint8_t will output a character rather than a number. by pkasting@chromium.org · 10 years ago
005b6ff Convert some EXPECTs to ASSERTs to avoid crashes when object creation fails. by pkasting@chromium.org · 10 years ago
5e16161 Remove CPU monitor from WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
aef0779 Rewrite ThreadWindows. by tommi@webrtc.org · 10 years ago
f2ec814 Move use of DEPTH into build_with_chromium==1. by kjellander@webrtc.org · 10 years ago
f88bee6 Refactor senders into senders and sources in the simulation framework. by stefan@webrtc.org · 10 years ago
a671f4b Fixing a VoE test to set correct rate for iSAC by henrik.lundin@webrtc.org · 10 years ago
05db352 Fix a bug in ACM test channel by henrik.lundin@webrtc.org · 10 years ago
3154a1c Reland r8210 "Add a new parameter to ACMGenericCodec constructor"" by henrik.lundin@webrtc.org · 10 years ago
4455f62 WebRtcIsacfix_Time2SpecNeon and _Spec2TimeNeon: Fix stack alignment by henrik.lundin@webrtc.org · 10 years ago
8820ac7 peerconnectin_server: missing comma in sprintfn() in r8128 by braveyao@webrtc.org · 10 years ago
2bbc35d Remove unused method, SetAffinity, from the ThreadWrapper class. by tommi@webrtc.org · 10 years ago
6752b85 Revert r8210 "Add a new parameter to ACMGenericCodec constructor" by henrik.lundin@webrtc.org · 10 years ago
c3643f2 Add a new parameter to ACMGenericCodec constructor by henrik.lundin@webrtc.org · 10 years ago
2444d96 Control the max IPv6 Networks used by WebRTC. by guoweis@webrtc.org · 10 years ago
4ddde2e Add arbitrary microphone geometry input to audioproc_f test utility. by mgraczyk@chromium.org · 10 years ago
1398025 Add new members to AudioEncoderOpus::Config by henrik.lundin@webrtc.org · 10 years ago
7a37bfc Revert 8203 "Reducing locking in OveruseFrameDetector and increa..." by tommi@webrtc.org · 10 years ago
a33f05e Re-land "Remove <(webrtc_root) from source file entries." by kjellander@webrtc.org · 10 years ago
bdebccf Fix a number of things in AudioEncoderDecoderIsac* by henrik.lundin@webrtc.org · 10 years ago
18e7585 Reducing locking in OveruseFrameDetector and increasing constness. by tommi@webrtc.org · 10 years ago
50fe359 Add tracing for slow paths in new video API. by pbos@webrtc.org · 10 years ago
4161715 Remove ChangeUniqueID. by tommi@webrtc.org · 10 years ago
1ece0cb Revert "Remove <(webrtc_root) from source file entries." by kjellander@webrtc.org · 10 years ago
a26f511 Remove frame copy in ViEExternalRendererImpl::RenderFrame by magjed@webrtc.org · 10 years ago
a87c398 Move audio_codec_speed_tests into include_tests==1 condition. by kjellander@webrtc.org · 10 years ago
2d2a1f9 Remove <(webrtc_root) from source file entries. by kjellander@webrtc.org · 10 years ago
73ca194 Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h by kwiberg@webrtc.org · 10 years ago
43c8839 Allow rtp packet history to dynamically expand in size. by sprang@webrtc.org · 10 years ago
827d7e8 Change AsyncInvoker to store its closure in a scoped_refptr instead of using a raw pointer. by perkj@webrtc.org · 10 years ago
a742cb1 Enable DTLS for peerconnection example. If it's a loopback test, then we recreate another peerconnection with DTLS off. by braveyao@webrtc.org · 10 years ago
f17ee9c Add case to ApmTest.Process to test the extended filter mode by aluebs@webrtc.org · 10 years ago
e7a4a12 Add arraysize() macro from Chromium, and make use of it in a few places. by pkasting@chromium.org · 10 years ago
035e912 Move channel_buffer.{h,cc} to common_audio. by kjellander@webrtc.org · 10 years ago
a67ca1a Only report the first rtp packet because it indicates the media has started flowing. by honghaiz@google.com · 10 years ago
a094cac Add stats for network merge. by guoweis@webrtc.org · 10 years ago
7d2b6a9 Enable Clang warning implicit-fallthrough and annotate the code. by kjellander@webrtc.org · 10 years ago
a907e01 Adding constness. by tommi@webrtc.org · 10 years ago
664ccb7 Reland r8125: Modify some tests to never use DTX disable mode by henrik.lundin@webrtc.org · 10 years ago
37c0559 Notify jitter buffer about received FEC packets (to avoid sending NACK request for these packets). by asapersson@webrtc.org · 10 years ago
22c2f05 Add "score" unit to SSIM perf score output. by kjellander@webrtc.org · 10 years ago
4aecd00 Add support for 40 and 60 ms frames to AudioEncoderIlbc by henrik.lundin@webrtc.org · 10 years ago
2a6558c Make sure ByteReader<T>::Read* is properly constified. by sprang@webrtc.org · 10 years ago
7aef80c GN: Remove webrtc_base target in favor for rtc_base. by kjellander@webrtc.org · 10 years ago
9b64a6e Adjust parameter in videoprocessor_integrationtest for VP9. by marpan@webrtc.org · 10 years ago
dc8a9da Adjust qp-max settinhg in VP9 wrapper. by marpan@webrtc.org · 10 years ago
922cfcd Use non-zero data in AudioRingBufferTest. by andrew@webrtc.org · 10 years ago
36401ab Update GAE API paths for join/leave. by tkchin@webrtc.org · 10 years ago
8bb32d6 Minor updates to AudioEncoderCng by henrik.lundin@webrtc.org · 10 years ago
db1ebf6 Add jakehilton@gmail.com to AUTHORS by tnakamura@webrtc.org · 10 years ago
478cedc Add new methods to AudioEncoder interface by henrik.lundin@webrtc.org · 10 years ago
5614cf1 audio_processing: Use fixed aggregation window in delay metrics by bjornv@webrtc.org · 10 years ago
6e25182 Whitespace change after enabling gnumbd by kjellander@webrtc.org · 10 years ago
ccd608e Whitespace change for git updater by kjellander@webrtc.org · 10 years ago
0bc73a1 Whitespace change to trigger git updater by kjellander@webrtc.org · 10 years ago
f68ffca Add PRESUBMIT check for GYP files including source files above itself. by kjellander@webrtc.org · 10 years ago
76e5e20 Roll chromium_revision 4664fe0..9070a80 (312733:313233) by kjellander@webrtc.org · 10 years ago