external/chromium_org/third_party/webrtc.git -

Age	Commit message (Expand)	Author
2014-10-27	Move (test) RtpFileReader to a lightweight target.	pbos@webrtc.org
2014-10-27	Move scoped_ptr "free" functions into the webrtc namespace.	andrew@webrtc.org
2014-10-27	Upgrade our scoped_ptr copy to match Chromium's latest.	andrew@webrtc.org
2014-10-27	Merge from Chromium at DEPS revision 614f7b807940	Torne (Richard Coles)
2014-10-27	Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile	henrik.lundin@webrtc.org
2014-10-27	isacfix: Refactor big-endian reading and writing	kwiberg@webrtc.org
2014-10-27	Increase max trace message size to 1024 characters.	pbos@webrtc.org
2014-10-27	Fix ::~LogMessage to print as a string.	pbos@webrtc.org
2014-10-24	Adding the subtool rtcBot report visualizer	houssainy@google.com
2014-10-24	Move min transmit bitrate to VideoEncoderConfig.	pbos@webrtc.org
2014-10-23	Break out WebRtcNs_ComputeDdUpdate function in ns_core	aluebs@webrtc.org
2014-10-23	Break out WebRtcNs_UpdateNoise function in ns_core	aluebs@webrtc.org
2014-10-23	Break out FFT function in ns_core	aluebs@webrtc.org
2014-10-23	Break out ComputeSnr function in ns_core	aluebs@webrtc.org
2014-10-23	Adding three video conference bots test	houssainy@google.com
2014-10-23	Update makefiles after merge of Chromium at 9ef958e74e13	Android Chromium Automerger
2014-10-23	Adding file from test.webrtc.org domain to be downloaded	houssainy@google.com
2014-10-23	Add macros and APIs for webrtc histograms.	asapersson@webrtc.org
2014-10-23	Adds support for sending first set of packets at increasingly higher bitrates...	stefan@webrtc.org
2014-10-23	Using the Unused turn configuration in two way test	houssainy@google.com
2014-10-23	Let video_loopback use internal VCM capturers.	pbos@webrtc.org
2014-10-22	NOTE: This code review based on the running issue:	houssainy@google.com
2014-10-22	Adding Two way video and audio streaming test to RtcBot	houssainy@google.com
2014-10-22	HTTPS Server used instead of HTTP for loading the bots to avoid the media per...	houssainy@google.com
2014-10-22	Make ReconfigureVideoEncoder use current bitrate.	pbos@webrtc.org
2014-10-22	Disable TestVp8Impl.BaseUnitTest on MSan.	pbos@webrtc.org
2014-10-22	For FIR packet, payload length is zero, so SendToNetwork function is failing.	stefan@webrtc.org
2014-10-21	Break out WebRtcNs_Windowing function in ns_core	aluebs@webrtc.org
2014-10-21	Break out WebRtcNs_Energy function in ns_core	aluebs@webrtc.org
2014-10-21	Break out WebRtcNs_IFFT function in ns_core	aluebs@webrtc.org
2014-10-21	Break out WebRtcNs_UpdateBuffer function in ns_core	aluebs@webrtc.org
2014-10-21	Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests	henrik.lundin@webrtc.org
2014-10-21	Update makefiles after merge of Chromium at b03027d23881	Android Chromium Automerger
2014-10-21	Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...	Android Chromium Automerger
2014-10-21	audio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>	bjornv@webrtc.org
2014-10-21	Fix for glitches in ACM when switching desired output sample rate	henrik.lundin@webrtc.org
2014-10-20	common_audio: Replaced invalid operand in min_max_operations_neon.S"	bjornv@webrtc.org
2014-10-20	Make avg_{psnr,ssim}_threshold_ const.	pbos@webrtc.org
2014-10-20	audio_coding/codecs/isac/main: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>	bjornv@webrtc.org
2014-10-20	audio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>	bjornv@webrtc.org
2014-10-19	Update makefiles after merge of Chromium at 89b463ddd92b	Android Chromium Automerger
2014-10-17	Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle...	henrike@webrtc.org
2014-10-17	Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."	henrike@webrtc.org
2014-10-17	Moving creating TURN configration to the host machine instead of the bots - r...	houssainy@google.com
2014-10-17	Query Android device orientation on every camera frame received.	glaznev@webrtc.org
2014-10-17	Test names changed from e.g) testOneWayVideo/chrome=>chrome to testOneWayVide...	houssainy@google.com
2014-10-16	Add encoded_timestamp to AudioEncoder base class	henrik.lundin@webrtc.org
2014-10-16	New interface class AudioEncoder	henrik.lundin@webrtc.org
2014-10-16	Improve rtcbot to load all test files at start and allow them to registerTests	andresp@webrtc.org
2014-10-16	Add ability to include a larger time span (in addition to encode time) for me...	asapersson@webrtc.org