Log - 63da1dd9720533990842f1e0aa94bf8269c86ae4 - platform/external/webrtc

63da1dd audio_processing: Now records mic volume level also when using new AGC by bjornv@webrtc.org · 10 years ago
ccd7e99 Temporarily change ThreadPosix to CHECK (crash) if we ever spend more than 30 seconds waiting for thread shutdown. There are cases on build bots where it looks like we're hitting this problem, but reproducing locally has been a struggle. by tommi@webrtc.org · 10 years ago
13a0e18 Temporarily disable a couple of ThreadChecker tests on Mac. by tommi@webrtc.org · 10 years ago
4770437 VirtualSocketServer out-of-order issue with closing TCP sockets by pthatcher@webrtc.org · 10 years ago
9baa9ca Add libjingle_peerconnection_so.so to Java test dependencies. by perkj@webrtc.org · 10 years ago
b5a1252 Hack to work around the current issues with rolling WebRTC into chromium. by tommi@webrtc.org · 10 years ago
751a365 Switch to using AudioEncoderPcmU/A instead of ACMPCMU/A by henrik.lundin@webrtc.org · 10 years ago
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