- b875349 fixing a bug in GetPlayoutDeviceName, previously it returns name as guid. by xians@google.com · 13 years ago
- ea72c34 Temporary add dummy implementation to RefCountModule. The reason is so that ADM and VideoCapture implementations can change to refcounted versions before forcing them. by perkj@google.com · 13 years ago
- 1e53166 Fix VP8 tests by henrik.lundin@webrtc.org · 13 years ago
- 9d23ba0 Make test app work on android by leozwang@google.com · 13 years ago
- fb298d3 Modified path on speex lib by zakkhoyt@google.com · 13 years ago
- 413b993 Put some table size information in one place. by andrew@webrtc.org · 13 years ago
- d7a4177 header included twice. Review URL: http://webrtc-codereview.appspot.com/139013 by turajs@google.com · 13 years ago
- eb74a37 Matlab scripts useful for parsing the output from DataLog by stefan@webrtc.org · 13 years ago
- 88a0da8 Add ref_count.h to gyp file. by perkj@google.com · 13 years ago
- 9de5917 Add an implementation of reference count to webrtc. by perkj@google.com · 13 years ago
- 2641fd1 Remove warnings in vp8_test by henrik.lundin@webrtc.org · 13 years ago
- ef04cf4 Adding reference counted version of the module interface. by perkj@google.com · 13 years ago
- 563f658 Adding to wathclist. Review URL: http://webrtc-codereview.appspot.com/139010 by mflodman@webrtc.org · 13 years ago
- 5a15ab9 Move the WebRtcDeviceManager and WebRtcMediaEngine to libjingle. by wu@webrtc.org · 13 years ago
- 4d905f8 Fix clang warnings in rtp. by andrew@webrtc.org · 13 years ago
- f1f93d8 Remove warning settings more stringent than Chromium's common.gypi. by andrew@webrtc.org · 13 years ago
- a80d026 Fix clang warnings in voice engine. by andrew@webrtc.org · 13 years ago
- bbd8908 Fix clang warnings in video coding. by andrew@webrtc.org · 13 years ago
- 49e58da Fix release mode "unused variable" warnings in peerconnection. by andrew@webrtc.org · 13 years ago
- 20f7428 Temporarily switch to Chrome's hosted libvpx copy. by andrew@webrtc.org · 13 years ago
- 87c546e Remove peerconnectionimpl_callbacks.h from libjingle.gyp. by tommi@webrtc.org · 13 years ago
- fac55d5 I've added two watchlist definitions (NetEQ and video codecs), and added myself to be notified when something changes. by henrik.lundin@webrtc.org · 13 years ago
- c6e54a9 Update to the peerconnection sample app. by tommi@webrtc.org · 13 years ago
- 84519ec Fixing some inconsistencies in WebRTC audio coding module. I've added setup information for all codecs which are not part of WebRTC, but possible to hook in. by tina.legrand@webrtc.org · 13 years ago
- d9e11b4 by zakkhoyt@google.com · 13 years ago
- 777ef59 Fix clang warnings in video engine. by andrew@webrtc.org · 13 years ago
- 243db12 media_opt_util: Fixed an assert and some code cleanup for AvgRecoveryFEC function. by marpan@google.com · 13 years ago
- b15bfd3 * Add the time_stamp as one parameter to the ViE ExternalRenderer interface. by wu@webrtc.org · 13 years ago
- ebb2744 To fix warning for unused variable. And fix some warning in test. by turajs@google.com · 13 years ago
- eaf3185 Take care of unused variable. by turajs@google.com · 13 years ago
- 9562a36 Last fixes to build with gcc 4.6. by andrew@webrtc.org · 13 years ago
- cdefd42 Adding code review watchlist to automatically CC e-mail addresses when new CLs are created. by mflodman@webrtc.org · 13 years ago
- 830099e Add a gyp flag to disable video functionality from dependencies shared by voice and video engine. by andrew@webrtc.org · 13 years ago
- e9f0e2e Moved _rtpReceiver to protected by pwestin@webrtc.org · 13 years ago
- c7d5f62 Fix build errors on Windows. by tommi@webrtc.org · 13 years ago
- 74c640a fix build break Review URL: http://webrtc-codereview.appspot.com/132004 by turajs@google.com · 13 years ago
- 7796c02 Wrap encode, decode, PLC NB functions in #define to avoid warnings. by turajs@google.com · 13 years ago
- 8ecd0e8 Remove Clang warning for PCM16B. by turajs@google.com · 13 years ago
- f990eb3 Hi, by mallinath@webrtc.org · 13 years ago
- eba8c32 Resolving a race condition issue related to using shared devices by punyabrata@google.com · 13 years ago
- 8811e5a Switch to a smoother stretch algorithm on Windows and delete buffers from previous conversations on linux when switching back to peer list. by tommi@webrtc.org · 13 years ago
- 3266d8d have the voe_cmd_test compiled with external transport enabled. by xians@google.com · 13 years ago
- e74a9ea AudioDeviceUtility::WaitForKey() pulls two characters if the first one is a newline, but discards the final value. by xians@google.com · 13 years ago
- 3fcabbe Modified include path after after moving files to webrtc_dev. by perkj@google.com · 13 years ago
- 932096c Porting gtalk alsa impl from depot to webrtc by xians@google.com · 13 years ago
- 46171cf video coding tests: Adding a Normal distribution to simulate packet arrival times by mikhal@webrtc.org · 13 years ago
- 8571af7 Updating to new VP8 rtp format by henrik.lundin@webrtc.org · 13 years ago
- 0973408 Fixes build issue in http://code.google.com/p/webrtc/issues/detail?id=56. by hellner@google.com · 13 years ago
- 81fd2bf New ACM codec database, created at compile time. by tina.legrand@webrtc.org · 13 years ago
- af931bd Update of iLBC reference files for version 1.1.1, new SQRT. by tina.legrand@webrtc.org · 13 years ago
- a41b4ce Changing iLBC to use the new improved SQRT, WebRtcSpl_SqrtFloor(). by tina.legrand@webrtc.org · 13 years ago
- c9cff24 Adding classes to be used for logging data within the engines and the by stefan@webrtc.org · 13 years ago
- 4094c49 Temporarily use digital AGC in WebRTC since Chromium can't support analog AGC. by perkj@google.com · 13 years ago
- c9b75e0 removing the warnings from the voe tests. by xians@google.com · 13 years ago
- 2aa5d50 Issue reported in WebRTC. A variable is defined and set, but never used. by tina.legrand@webrtc.org · 13 years ago
- 36450af Removing unsupported codecs from ptypes file by henrik.lundin@webrtc.org · 13 years ago
- 92bace1 Hi, by mallinath@webrtc.org · 13 years ago
- bd4494c Remove the divide-by-2 when mixing. by andrew@webrtc.org · 13 years ago
- b7ac56d video coding tests: updating quality tests following r466 by mikhal@webrtc.org · 13 years ago
- d24a97f video coding test: deleting unused file(resampler_test.cc) by mikhal@webrtc.org · 13 years ago
- 2c3b1fb video_coding tests: removing unused functionality from test_util by mikhal@webrtc.org · 13 years ago
- a057a95 video_coding: Updating protection logic in media optimization utility: by mikhal@webrtc.org · 13 years ago
- 552f173 video_coding: Moving video metrics computation to a designated file. by mikhal@webrtc.org · 13 years ago
- e46d69f Fix gcc 4.6 set but unused warnings in AEC. by andrew@webrtc.org · 13 years ago
- b62c776 moving all new version related files to webrtc_dev and removed from webrtc. by mallinath@webrtc.org · 13 years ago
- ffbe7a7 Cast away the unused state argument value to silence gcc 4.6 warnings. by andrew@webrtc.org · 13 years ago
- 7f2bbbb To remove all calls involving scratch-memory by turajs@google.com · 13 years ago
- ac55f7b by turajs@google.com · 13 years ago
- 7659b36 revert the file path in the voe_auto_test by xians@google.com · 13 years ago
- 350d091 Send the hangup message when asked to disconnect from a peer. by tommi@webrtc.org · 13 years ago
- c57f9c3 by xians@webrtc.org · 13 years ago
- 4fcb0ca Removing warning in video capture module for linux and auto test. by mflodman@webrtc.org · 13 years ago
- b55c988 Updated peerconnection_unittest slightly. Also added it to the build. by hellner@google.com · 13 years ago
- 23a8065 Fixed broken build due to r453. by hellner@google.com · 13 years ago
- b2801f3 Added the remaining test cases for the webrtcsession unittest also some minor refactoring. by hellner@google.com · 13 years ago
- 59af6f1 Porting Mac keypress detection from GIPS repository. by zakkhoyt@google.com · 13 years ago
- ba9bd69 video_coding_tests: Fix build error by mikhal@webrtc.org · 13 years ago
- aed0348 Roll gyp 985:1012 by andrew@webrtc.org · 13 years ago
- 40373cc Bugfix in unittest and some minor refactoring. by hellner@google.com · 13 years ago
- eb9572e Add the new peerconnection factory to the scons file. by wu@webrtc.org · 13 years ago
- e129ae9 by niklas.enbom@webrtc.org · 13 years ago
- 3227ed5 Fixed potential memory leak in unit test and removed an unnecessary copy. by hellner@google.com · 13 years ago
- 102b227 First version of the peerconnection client application for Linux. by tommi@webrtc.org · 13 years ago
- 137ece4 * Make GetReadyState accessible via the PeerConnection interface. by tommi@webrtc.org · 13 years ago
- 44d356d Fix unused variable warning in spatial_resampler.cc by stefan@webrtc.org · 13 years ago
- 1cdc6b5 This CL adding a factory class which has the responsibility of creating peerconnection objects. This is very basic class doesn't do any reference count, user has the responsibility to delete the object externally. by mallinath@webrtc.org · 13 years ago
- d1015fe Replaced regular sleep with a talk_base::Thread::ProcessMessages(..) call so that Posts get some execution time from the main thread. by hellner@google.com · 13 years ago
- 5cc9c68 Fixing a warning discovered while compiling with clang. by turajs@google.com · 13 years ago
- 057efc8 Removed unused variables and unnecessary assert: causing build error in vpm_test. by marpan@google.com · 13 years ago
- 4f39000 Fix warnings on Ubuntu 11.04 (gcc 4.5) by andrew@webrtc.org · 13 years ago
- 37fd004 Remove the X11 headers we don't need. by wu@webrtc.org · 13 years ago
- cf36b2a Match new[] / delete [] by frkoenig@google.com · 13 years ago
- accd686 Implementation of media streams. Work in progress. by perkj@google.com · 13 years ago
- 49cbc51 Fix unused variable warning in video_coding. by stefan@webrtc.org · 13 years ago
- 7f593c1 Fix gcc 4.6 unused variable warnings in audio_processing. by andrew@webrtc.org · 13 years ago
- 6724cf8 VP8: Adding a flag to indicate the libvpx version. When in Cayuga, additional API's will be used. by mikhal@webrtc.org · 13 years ago
- 9788e18 * Add PeerConnectionProxy to forward all the API calls to signaling thread. by wu@webrtc.org · 13 years ago
- 4482b04 revert r430 to keep webrtc always ready to roll in chromium. by wjia@google.com · 13 years ago
- f9f1deb Get ready for libvpx Cayuga (v0.9.7-p1). by wjia@google.com · 13 years ago
- dec6aa5 This CL will remove sending any signal after calling Close and RemoveStream. I am thinking to remove Close method at all, since application can directly delete the object if it wants to end the call with all active streams. Will send that change later in a different CL. by mallinath@webrtc.org · 13 years ago