external/chromium_org/third_party/webrtc.git -

Age	Commit message (Expand)	Author
2014-04-04	Update makefiles after merge of Chromium at 261622	Android Chromium Automerger
2014-04-02	Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...	Android Chromium Automerger
2014-04-02	Remove AudioDevice::{Microphone,Speaker}IsAvailable.	andrew@webrtc.org
2014-04-02	This is to get rid of a bug relating to the return of NULL in calling GetDeco...	minyue@webrtc.org
2014-04-02	Add format specification to output file names	henrik.lundin@webrtc.org
2014-04-02	sink_filter_ds.cc: add lock to Receive procedure to Pause().	braveyao@webrtc.org
2014-04-01	Update makefiles after merge of Chromium at 260927	Android Chromium Automerger
2014-04-01	Added simulations of capacity variations and wifi recordings.	stefan@webrtc.org
2014-04-01	VoiceEngine(iOS & Android): removed NOT_SUPPORTED	fischman@webrtc.org
2014-03-31	Add tests for the RBE RemoveStream() API.	solenberg@webrtc.org
2014-03-27	Don't disable experimental AGC in audioproc.	andrew@webrtc.org
2014-03-26	Re-submit: rev5775	andresp@webrtc.org
2014-03-26	Makes ScreenCapturerMac exclude the window specified in DesktopCapturer::SetE...	jiayl@webrtc.org
2014-03-26	Add API to allow deducting bitrate from incoming estimates before the capacit...	solenberg@webrtc.org
2014-03-26	Protect write of send_target_bitrate.	andresp@webrtc.org
2014-03-25	Make RTPHeaderParser skip over unknown RTP header extensions rather than bail...	solenberg@webrtc.org
2014-03-25	Revert 5775 "Modify bitrate controller to update bitrate based o..."	andrew@webrtc.org
2014-03-25	iOS video_capture: move @private vars to impl.	fischman@webrtc.org
2014-03-25	Fix race condition in RTPSEnder.	sprang@webrtc.org
2014-03-25	Modify bitrate controller to update bitrate based on process call and not	andresp@webrtc.org
2014-03-25	iOS video_capture: start camera in the background.	fischman@webrtc.org
2014-03-25	iOS VideoEngine: move video_{capture,render} to ARC.	fischman@webrtc.org
2014-03-24	Have changes to REMB trigger RTCP to be sent immediately.	stefan@webrtc.org
2014-03-24	DelayEstimator: Updates delay_quality and adds soft reset.	bjornv@webrtc.org
2014-03-24	Run Opus with lower complexity setting on Android, iOS and/or ARM	tina.legrand@webrtc.org
2014-03-24	Disabled some of the remote bitrate estimator baseline tests.	stefan@webrtc.org
2014-03-24	VoE changes to allow forwarding of packets from VoE to ViE BWE.	solenberg@webrtc.org
2014-03-24	Add AIMD option to BWE API.	stefan@webrtc.org
2014-03-23	ACM2/NetEq4 did not decode Opus in stereo	tina.legrand@webrtc.org
2014-03-21	Refactor in BitrateController module.	andresp@webrtc.org
2014-03-21	Fixing crash in video_render_tests in release mode.	henrikg@webrtc.org
2014-03-21	Remove locks in SendSideBandwidthEstimation since those are only accessed whi...	andresp@webrtc.org
2014-03-21	Adding FEC support in NetEq 4.	minyue@webrtc.org
2014-03-21	Fix "unreachable code" warnings (MSVC warning 4702) in webrtc.	pbos@webrtc.org
2014-03-20	Use codec width/height as the encoded_image width/height.	wu@webrtc.org
2014-03-20	Changing the buffer size (slots) to 1.5 seconds @ 30 ms packets	henrik.lundin@webrtc.org
2014-03-20	Adjust the captured window rect when the window is maximized.	jiayl@webrtc.org
2014-03-19	Properly account for retransmitted packets when not using the pacer.	stefan@webrtc.org
2014-03-19	Fixes RTX related bugs.	stefan@webrtc.org
2014-03-19	Revert "Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets"	henrik.lundin@webrtc.org
2014-03-19	Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets	henrik.lundin@webrtc.org
2014-03-19	Simplify pacer interface.	pbos@webrtc.org
2014-03-18	Adds a method to WindowCapturer to bring a window to the front.	jiayl@webrtc.org
2014-03-18	Adding thread annotations to NetEq4	henrik.lundin@webrtc.org
2014-03-18	Add #include <cstdlib> for std::abs.	pbos@webrtc.org
2014-03-17	Small refactor on send_side_bandwidth_estimation.	andresp@webrtc.org
2014-03-14	References to includes in third_party should be relative, not absolute.	sprang@webrtc.org
2014-03-14	Fix a bug where network freeze during CNG causes delay	henrik.lundin@webrtc.org
2014-03-14	Remove legacy weirdness in Merge::Downsample	henrik.lundin@webrtc.org
2014-03-13	Race condition in RTPSender	sprang@webrtc.org