Log - c18957e8773e558de869a809df97ec6ca924794b - platform/external/webrtc

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
273fbbb Update StreamDataCounter with FEC bytes. by asapersson@webrtc.org · 10 years ago
70117a8 AEC: Implements a new function for calculating delay metrics by bjornv@webrtc.org · 10 years ago
fc5ad95 Reland of: "Implement elapsed time and capture start NTP time estimation." revision @8139 by magjed@webrtc.org · 10 years ago
8501ee6 Support VP8 HW decoding on devices with Exynos codec. by glaznev@webrtc.org · 10 years ago
df9a41d Fix bug in GetREDStatus(): it doesn't actually return the current status. by pkasting@chromium.org · 10 years ago
82415e3 Update AppRTCDemo to use renamed GAE messages. by glaznev@webrtc.org · 10 years ago
041035b Add an AudioRingBuffer class wrapper for the ring_buffer.h C interface. by andrew@webrtc.org · 10 years ago
4dba2e9 Consolidate anonymous namespace content and file-static methods to all be in the by pkasting@chromium.org · 10 years ago
d7e34e1 Make it easier to use external libyuv + cleanup GYP files. by kjellander@webrtc.org · 10 years ago
d25c034 Refactor common_audio/vad: Removed usage of macro WEBRTC_SPL_MUL_16_16() by bjornv@webrtc.org · 10 years ago
04cd466 Move ThreadChecker into rtc_base_approved. by tommi@webrtc.org · 10 years ago
38d11b8 Enable encoder multi-threading for VP9. by marpan@webrtc.org · 10 years ago
6f200b5 Temporarily revert r8147 ("Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h") by kwiberg@webrtc.org · 10 years ago
b6fab2b Introduce rtc::CheckedDivExact by henrik.lundin@webrtc.org · 10 years ago
19eb4e4 Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h by kwiberg@webrtc.org · 10 years ago
995b4c9 Remove win_asan trybot from PRESUBMIT.py by kjellander@webrtc.org · 10 years ago
acb8085 Roll chromium_revision c086b4e..4664fe0 (312108:312733) by kjellander@webrtc.org · 10 years ago
7519de5 Revert 8136 "Remove frame copy in ViEExternalRendererImpl::Rende..." by tkchin@webrtc.org · 10 years ago
0f98844 Revert 8139 "Implement elapsed time and capture start NTP time e..." by tkchin@webrtc.org · 10 years ago
dacdd94 Reland r7980: by jiayl@webrtc.org · 10 years ago
8919cfe Change a GYP reference to cpufeatures.gypi by fdegans@chromium.org · 10 years ago
ad3ee2c Implement elapsed time and capture start NTP time estimation. by pbos@webrtc.org · 10 years ago
a02d768 Disable DtmfSenderTest.InsertDtmfWithCommaAsDelay due to flakiness by kjellander@webrtc.org · 10 years ago
456f014 Re-allowing RED in voice engine. by minyue@webrtc.org · 10 years ago
182ea46 Remove frame copy in ViEExternalRendererImpl::RenderFrame by magjed@webrtc.org · 10 years ago
73ee453 Switch to use range based loops in the BWE simulation framework. by stefan@webrtc.org · 10 years ago
36d5c3c Leave BIO_METHOD non-const. by davidben@webrtc.org · 10 years ago
586f2ed Change GetStreamBySsrc to not copy StreamParams. by tommi@webrtc.org · 10 years ago
7e5b380 Fix a crash in AllocationSequence. Internal bug 19074679. by jiayl@webrtc.org · 10 years ago
ff108fe Revert 8125 "Modify some tests to never use DTX disable mode" by kjellander@webrtc.org · 10 years ago
b40c7bb Change sprintf use in talk samples to snprintf by jlmiller@webrtc.org · 10 years ago
ea1c842 Correct GetDriveType error handling. by jlmiller@webrtc.org · 10 years ago
043db24 Modify some tests to never use DTX disable mode by henrik.lundin@webrtc.org · 10 years ago
e5251ad Integrate send-side BWE into simulation framework. by stefan@webrtc.org · 10 years ago
cfd82df Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. by asapersson@webrtc.org · 10 years ago
3dd33a6 Fix bug in thresholds for bitrate probing and adjust thresholds to allow a larger dispersion and concentration for successful probes. by stefan@webrtc.org · 10 years ago
fbd37bd Make iSAC SWB own its decoder by henrik.lundin@webrtc.org · 10 years ago
cceb166 Fix a use-after-free when sending queued messages is aborted for blocked channel. by jiayl@webrtc.org · 10 years ago
e65d9d9 Fix an unitialized variable warning. by andrew@webrtc.org · 10 years ago
c429b82 GN: Prepare to remove webrtc_base target by kjellander@webrtc.org · 10 years ago