Log - 4abadab708d7102e0c69f2648e0f4b4acfe2b7fb - platform/external/webrtc

4abadab Simplify bwe tests. by stefan@webrtc.org · 10 years ago
2dc6f31 Adapting bitrate according to maxplaybackrate for Opus. by minyue@webrtc.org · 10 years ago
8328e7c Revert "Revert part of r7561, "Refactor audio conversion functions."" by andrew@webrtc.org · 10 years ago
14146e4 arm64 iOS build. by tkchin@webrtc.org · 10 years ago
50ca986 Improve the logging when a TCP connection is deleted. by jiayl@webrtc.org · 10 years ago
d0cf68e Add 15 fps support for Android devices with missing 15 fps camera mode. by glaznev@webrtc.org · 10 years ago
8aa4d2d Creating a C++ wrapper class for VAD by henrik.lundin@webrtc.org · 10 years ago
bcfb4d0 Revert part of r7561, "Refactor audio conversion functions." by kwiberg@webrtc.org · 10 years ago
8219529 Cleaning up r7562-7567. by minyue@webrtc.org · 10 years ago
879fac8 (Auto)update libjingle 78822708-> 78823675 by buildbot@webrtc.org · 10 years ago
5f73a37 Revert 7563 "before rebase" due to wrong submission by minyue@webrtc.org · 10 years ago
c11cc8d Revert 7564 "to submit" due to wrong submission by minyue@webrtc.org · 10 years ago
de386bf to submit by minyue@webrtc.org · 10 years ago
c673bb9 before rebase by minyue@webrtc.org · 10 years ago
0b62672 adding default rates by minyue@webrtc.org · 10 years ago
4fc4add Refactor audio conversion functions. by andrew@webrtc.org · 10 years ago
776e6f2 Use external VideoDecoders in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
2dd3134 Add stats for duplicate sent and received NACK requests. by asapersson@webrtc.org · 10 years ago
f567095 common_audio: Removed macro WEBRTC_SPL_RSHIFT_W32 by bjornv@webrtc.org · 10 years ago
7f10513 Remove unused code in overuse detector. by asapersson@webrtc.org · 10 years ago
decd930 AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket by kwiberg@webrtc.org · 10 years ago
cfe3845 Enable G.722 for Chromium builds by henrik.lundin@webrtc.org · 10 years ago
1abc146 (Auto)update libjingle 78738075-> 78738103 by buildbot@webrtc.org · 10 years ago
7998089 ApprtDemo Android: Switch between front and back camera. by perkj@webrtc.org · 10 years ago
663fdd0 Make an AudioEncoder subclass for Opus by kwiberg@webrtc.org · 10 years ago
2623695 Renaming bandwidth to bitrate in webrtcvoiceengine. by minyue@webrtc.org · 10 years ago
ffeaeed Make NSinst_t* const and rename to self in ns_core by aluebs@webrtc.org · 10 years ago
269fb4b move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
8b1b23f Make local functions static and dropWebRtcNs_ in ns_core by aluebs@webrtc.org · 10 years ago
28b5467 Make all comments whole sentences in ns_core by aluebs@webrtc.org · 10 years ago
bd6bdca scoped_ptr.h: Renames function and change namespace scope to fix conflicts with Chromium not detected by the FYI bots. by henrike@webrtc.org · 10 years ago
ae694ef (Auto)update libjingle 78642371-> 78680406 by buildbot@webrtc.org · 10 years ago
a296725 audio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>" by bjornv@webrtc.org · 10 years ago
67ca26e common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16 by bjornv@webrtc.org · 10 years ago
ff8a98e Use neteq_unittest_tools in audio_decoder_unittests by henrik.lundin@webrtc.org · 10 years ago
820efd5 Fix double backslashes in incoming_video_stream.cc by perkj@webrtc.org · 10 years ago
fbd55cb (Auto)update libjingle 78616359-> 78642371 by buildbot@webrtc.org · 10 years ago
f15dee6 Check if a datachannel in the current local description is an sctp channel before assuming rtp. by tommi@webrtc.org · 10 years ago
aada86b Add a simple AudioConverter class. by andrew@webrtc.org · 10 years ago
33a0e2d Only configure the SSL library in one place. by henrike@webrtc.org · 10 years ago
aca5803 Move (test) RtpFileReader to a lightweight target. by pbos@webrtc.org · 10 years ago
b787f4c Move scoped_ptr "free" functions into the webrtc namespace. by andrew@webrtc.org · 10 years ago
243eb8e Adding setting screen to AppRTCDemo. by glaznev@webrtc.org · 10 years ago
068b529 (Auto)update libjingle 78583324-> 78583691 by buildbot@webrtc.org · 10 years ago
df42988 Upgrade our scoped_ptr copy to match Chromium's latest. by andrew@webrtc.org · 10 years ago
2e7ee4b Fix the SrtpFilter crash caused by two local offers. by pthatcher@webrtc.org · 10 years ago
efc82c2 Implement screencast settings for WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
a37f1dd Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile by henrik.lundin@webrtc.org · 10 years ago
0552356 isacfix: Refactor big-endian reading and writing by kwiberg@webrtc.org · 10 years ago
9fed099 Increase max trace message size to 1024 characters. by pbos@webrtc.org · 10 years ago
c86ec3e Fix ::~LogMessage to print as a string. by pbos@webrtc.org · 10 years ago
1732df6 Use flags set by the port allocator. by braveyao@webrtc.org · 10 years ago
3b839d0 PRESUBMIT: Add linux_msan to default trybots. by kjellander@webrtc.org · 10 years ago
3f7bcc1 (Auto)update libjingle 78430441-> 78445452 by buildbot@webrtc.org · 10 years ago
c7ed8db (Auto)update libjingle 78427027-> 78430441 by buildbot@webrtc.org · 10 years ago
4709887 Add HD support to Android if we detect a hardware video encoder that can be used. This Change the internal class MediaCodecVideoEncoder to have a one public method for checking if the platform is supported. It also adds &hd=true to the reqest url a hardware encoder is detected. by perkj@webrtc.org · 10 years ago
39b1743 Adding the subtool rtcBot report visualizer by houssainy@google.com · 10 years ago
ad3b5a5 Move min transmit bitrate to VideoEncoderConfig. by pbos@webrtc.org · 10 years ago
c9d6d14 patch from issue 25469004 by pthatcher@webrtc.org · 10 years ago
8fe75ee (Auto)update libjingle 78381351-> 78389679 by buildbot@webrtc.org · 10 years ago
fb5e9fc (Auto)update libjingle 78344087-> 78381351 by buildbot@webrtc.org · 10 years ago
7e19a11 Break out WebRtcNs_ComputeDdUpdate function in ns_core by aluebs@webrtc.org · 10 years ago
f8ea0d5 Break out WebRtcNs_UpdateNoise function in ns_core by aluebs@webrtc.org · 10 years ago
799e88a Break out FFT function in ns_core by aluebs@webrtc.org · 10 years ago
8454ad8 Break out ComputeSnr function in ns_core by aluebs@webrtc.org · 10 years ago
0d3e254 Adding three video conference bots test by houssainy@google.com · 10 years ago
0e19d0c Adding file from test.webrtc.org domain to be downloaded by houssainy@google.com · 10 years ago
580d367 Add macros and APIs for webrtc histograms. by asapersson@webrtc.org · 10 years ago
9d446f2 (Auto)update libjingle 78296920-> 78342456 by buildbot@webrtc.org · 10 years ago
8539bd0 Download full Chromium checkouts by default by kjellander@webrtc.org · 10 years ago
82462aa Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate. by stefan@webrtc.org · 10 years ago
2192701 Using the Unused turn configuration in two way test by houssainy@google.com · 10 years ago
ad553a2 Let video_loopback use internal VCM capturers. by pbos@webrtc.org · 10 years ago
15c717b Add a memcheck exclusion for EndToEndTest.CanSwitchToUseAllSsrcs. by andrew@webrtc.org · 10 years ago
a9f0898 (Auto)update libjingle 78273470-> 78296920 by buildbot@webrtc.org · 10 years ago
7bb4a98 Merging Henrik's and Peter's changes for AppRTCDemo by glaznev@webrtc.org · 10 years ago
fce8f5d NOTE: This code review based on the running issue: by houssainy@google.com · 10 years ago
3382059 Adding Two way video and audio streaming test to RtcBot by houssainy@google.com · 10 years ago
e9b7d03 HTTPS Server used instead of HTTP for loading the bots to avoid the media permission pop-up clicks every time running the test. by houssainy@google.com · 10 years ago
fb5410a (Auto)update libjingle 78262388-> 78262615 by buildbot@webrtc.org · 10 years ago
eacc6e4 Remove some disabled tests in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
82e430c Suppress libyuv uninitialized read in CopyRow_AVX by kjellander@webrtc.org · 10 years ago
32452b2 Make ReconfigureVideoEncoder use current bitrate. by pbos@webrtc.org · 10 years ago
860ccc9 Tighten up MSan blacklist.txt owners. by kjellander@webrtc.org · 10 years ago
3f8f555 Disable TestVp8Impl.BaseUnitTest on MSan. by pbos@webrtc.org · 10 years ago
76960d5 For FIR packet, payload length is zero, so SendToNetwork function is failing. by stefan@webrtc.org · 10 years ago
1d9af96 Roll chromium_revision de13cf4..28d1981 (299488:300483) by kjellander@webrtc.org · 10 years ago
67cf1d7 Break out WebRtcNs_Windowing function in ns_core by aluebs@webrtc.org · 10 years ago
0e70992 Break out WebRtcNs_Energy function in ns_core by aluebs@webrtc.org · 10 years ago
7634c09 Break out WebRtcNs_IFFT function in ns_core by aluebs@webrtc.org · 10 years ago
a5c36b3 (Auto)update libjingle 78193292-> 78199328 by buildbot@webrtc.org · 10 years ago
b6173ab Fix local address leakage when IceTransportsType is relay by guoweis@webrtc.org · 10 years ago
333e255 Break out WebRtcNs_UpdateBuffer function in ns_core by aluebs@webrtc.org · 10 years ago
1288cbb (Auto)update libjingle 78106439-> 78193292 by buildbot@webrtc.org · 10 years ago
def1e97 Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests by henrik.lundin@webrtc.org · 10 years ago
78ea06d audio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> by bjornv@webrtc.org · 10 years ago
913f7b8 Fix for glitches in ACM when switching desired output sample rate by henrik.lundin@webrtc.org · 10 years ago
a8c0edd Avoid using EGLContext class for Android 4.1 and below. by glaznev@webrtc.org · 10 years ago
b69ea9a common_audio: Replaced invalid operand in min_max_operations_neon.S" by bjornv@webrtc.org · 10 years ago
fa553ef Set up start bitrate in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago