Log - 2740f66d22013edfc82062ab23545db801a5a7b5 - platform/external/webrtc

2740f66 Remove output files from .gitignore. by andrew@webrtc.org · 13 years ago
26b9777 Only trigger one call to OnNetworkChanged for each incoming RTCP packet. by mflodman@webrtc.org · 13 years ago
471e83e Refactored ViESharedData. by mflodman@webrtc.org · 13 years ago
9af365d Fixing VP8 RTP parser bug by henrik.lundin@webrtc.org · 13 years ago
6f2c016 Updating to VP8 RTP spec rev -02 by henrik.lundin@webrtc.org · 13 years ago
517dcb7 Migrate to using the same gyp files and gypi generator script as Chromium. by stefan@webrtc.org · 13 years ago
6d26ef7 Refactored ViESender. by mflodman@webrtc.org · 13 years ago
d492f72 Added empty unit tests to get code coverage measured. by kjellander@webrtc.org · 13 years ago
55d81ea ViE Custom Call observer now using pointers, fixed protection method and miscellaneous TODO cleanup by amyfong@webrtc.org · 13 years ago
ba028a3 Fix sample rate printout in process_test. by andrew@webrtc.org · 13 years ago
c4ac357 Fix spelling errors in DEPS. by andrew@webrtc.org · 13 years ago
f3d10d3 Fixed release compilation error-warnings. by phoglund@webrtc.org · 13 years ago
3379b0c Fixing resources script on Windows by kjellander@webrtc.org · 13 years ago
c4c56ed Rewrote vie_auto_test to use googletest macros. by phoglund@webrtc.org · 13 years ago
0c80c7f Script for automatic download of resources. by kjellander@webrtc.org · 13 years ago
48b68c0 Added support for 96 kHz sampling frequency. by bjornv@webrtc.org · 13 years ago
4257790 NetEQ-related bug in ACM by henrik.lundin@webrtc.org · 13 years ago
543c3ea Fixing Release compilation errors by kjellander@webrtc.org · 13 years ago
89ab652 Cleaning up NetEQ statistics by henrik.lundin@webrtc.org · 13 years ago
df10de4 Removing statistics API from NetEQ by henrik.lundin@webrtc.org · 13 years ago
7d3e949 This CL is to support certain audio devices which don't offer volume control. Try to be more compatible to those rare cases. by braveyao@webrtc.org · 13 years ago
2b838b4 video_coding: updating the session info unit test following recent changes by mikhal@webrtc.org · 13 years ago
425b377 video_coding: Updating internal_defines to resolve latest build error. Refers to JB flush update. by mikhal@webrtc.org · 13 years ago
f13388f video_coding: Requesting a key frame after a JB flush by mikhal@webrtc.org · 13 years ago
6b9a7f8 video_coding: Allowing for a decodable state independent of selective nacking by mikhal@webrtc.org · 13 years ago
828af1b Add lookahead to the delay estimator. by andrew@webrtc.org · 13 years ago
5a52939 Make DMO init safe when not supported. by andrew@webrtc.org · 13 years ago
91617ff by henrikg@webrtc.org · 13 years ago
dfe89e3 Move ViE main/test/AutoTest to test/auto_test. by mflodman@webrtc.org · 13 years ago
8594f76 Add a gyp variable for AEC debug dumps. by andrew@webrtc.org · 13 years ago
a249f35 Correct several makefile errors for Android build. by kma@webrtc.org · 13 years ago
6830bdd Fix xcode build. by mflodman@webrtc.org · 13 years ago
94ea32e Move video_engine/source* to video_engine/. No code changes except paths in gyp-files. by mflodman@webrtc.org · 13 years ago
274c2ef Adding empty test method required to get code coverage by kjellander@webrtc.org · 13 years ago
3caa327 VP8 wrapper: Turn on some mild amount of deblocking in post-processing. by marpan@webrtc.org · 13 years ago
ce9d89d Fixes linux build error introduced in r980. by henrike@webrtc.org · 13 years ago
ad98a3e Fixes TEST crash triggered by webrtc-codereview.appspot.com/268014. by henrike@webrtc.org · 13 years ago
31d3070 Addressed review comments from http://webrtc-codereview.appspot.com/256004/ by henrike@webrtc.org · 13 years ago
ced1186 Changed keyword __restrict__ to __restrict. by kma@webrtc.org · 13 years ago
3798ecb Made CPU initialization on Windows lazy to prevent long startup time. by henrike@webrtc.org · 13 years ago
543611a Reverting r972 due to compilation error on Windows Release build. by kjellander@webrtc.org · 13 years ago
2f047cc Removed unnecessary variable to avoid compiler error on Win. by bjornv@webrtc.org · 13 years ago
ba74924 Remove use of exceptions in NetEQ test code by henrik.lundin@webrtc.org · 13 years ago
6a9835d Delay estimator structural changes. by bjornv@webrtc.org · 13 years ago
fa9b016 Optimized WebRtcIsacfix_AutocorrFix() function for iSAC fix. by kma@webrtc.org · 13 years ago
f556b9d This modification is supposed to fix the webrtc issue 144/145. With this fix, people could set/get mic volume before StartSend(). by braveyao@webrtc.org · 13 years ago
3041c62 Updating libvpx to revision bdd35c13: This revision includes a fix for frame rate adaptation - avoid resetting the frame rate to 30 on each update. by mikhal@webrtc.org · 13 years ago
917fa6b ViE Custom Call added SetImageScaleStatus toggle option and other changes. by amyfong@webrtc.org · 13 years ago
cd7b57e Fixing release compilation error by kjellander@webrtc.org · 13 years ago
3f1cb8e Restructuring and adding dummy unit test target. by kjellander@webrtc.org · 13 years ago
cc2ecb3 Restructuring and adding dummy unit test target. by kjellander@webrtc.org · 13 years ago
b72268e Restructuring and adding dummy unit test target. by kjellander@webrtc.org · 13 years ago
64a897a Restructuring and adding dummy unit test target. by kjellander@webrtc.org · 13 years ago
1310e0b Adding third_party/cygwin to git ignore by kjellander@webrtc.org · 13 years ago
8f89f09 Note: this patch may seem intimidating but it mostly moves code around and renames things. There are quite few actual changes. by phoglund@webrtc.org · 13 years ago
c05b56a Fixing compilation error by kjellander@webrtc.org · 13 years ago
9f9af7d Remove Peerconnection Dev branch. by perkj@webrtc.org · 13 years ago
0403ef4 Restructuring and adding unit test targets on project level instead of in common_audio. by kjellander@webrtc.org · 13 years ago
7eba346 Remove module targets due to Linux-Release errors. by andrew@webrtc.org · 13 years ago
337dc68 Included modules in webrtc.gyp and fixed build errors. by phoglund@webrtc.org · 13 years ago
af26f64 Inband DTMF stereo support by niklas.enbom@webrtc.org · 13 years ago
e33a102 Resubmitting http://webrtc-codereview.appspot.com/269007/ by niklas.enbom@webrtc.org · 13 years ago
fcf33eb Limit number of send-side BWE increases to one per second. by stefan@webrtc.org · 13 years ago
3d1ef0c Add an OWNERS file for python_charts. by andrew@webrtc.org · 13 years ago
d0e5b96 Fix Amy's email address. by andrew@webrtc.org · 13 years ago
81d4499 Microphone volume on Mac not being printed properly due by punyabrata@webrtc.org · 13 years ago
755b04a Add RMS computation for the RTP level indicator. by andrew@webrtc.org · 13 years ago
6a85b17 Potential fix for crash after Mac sleep. by andrew@webrtc.org · 13 years ago
85596d5 Setting completeFrame to true for all created encoded images. by kjellander@webrtc.org · 13 years ago
cde1e7f Use a TraceNoop instance when tracing disabled (to be used in Chromium). by tommi@webrtc.org · 13 years ago
bc91d5a NetEQ tests by henrik.lundin@webrtc.org · 13 years ago
aedf3a7 Loosening makefile rule so it matches *.target.mk but not Android.mk files. by kjellander@webrtc.org · 13 years ago
a02ef1a Fix broken tree. by mflodman@webrtc.org · 13 years ago
1f69c03 Added size sanity check for copying app specific RTCP data. by mflodman@webrtc.org · 13 years ago
33df533 Change luminance of all pixels by a specified value. by henrik.lundin@webrtc.org · 13 years ago
7de0765 Disables a flaky metric test. by stefan@webrtc.org · 13 years ago
ded85f1 Enable WEBRTC_NO_TRACE for Chromium builds. by tommi@webrtc.org · 13 years ago
0db7dc6 Add file-playing channels to voe_cmd_test. by andrew@webrtc.org · 13 years ago
cd82438 Unpack the full set of audioproc data. by andrew@webrtc.org · 13 years ago
d71d480 Fixed a build error of audio conference mixer in Android. by kma@webrtc.org · 13 years ago
b351d6a Reverting rev 929 due to failing assert on Linux. by stefan@webrtc.org · 13 years ago
9b18ed6 Removed incorrect dependency. by phoglund@webrtc.org · 13 years ago
fd3a0ef RTP bw estimate fix. by mflodman@webrtc.org · 13 years ago
1144ba2 Base and codec tests now run verify output and render to file instead of to screen. by phoglund@webrtc.org · 13 years ago
62e48eb adding owners for test by niklas.enbom@webrtc.org · 13 years ago
50b3cbe First pass. You can now enable a stereo codec and send and receive. This does not include more advances use cases (DTMF etc), but I'd rather keep the CLs manageable. by niklas.enbom@webrtc.org · 13 years ago
b61c410 Fixed a couple of Android makefiles to let voe and vie build properly. by kma@webrtc.org · 13 years ago
13318ef (1) Corrected the makefile for testing iLBC in Android, and changed the location of the test makefile to make it consistent with audio_processing. by kma@webrtc.org · 13 years ago
7a4eb28 Calculate the available bandwidth before sending a TMMBR by mflodman@webrtc.org · 13 years ago
637a59e jitter buffer update: waiting for key frame when Nack is enabled and continuity cannot be determined. by mflodman@webrtc.org · 13 years ago
855a77c Audio Coding Module: Fixing a bug that prevented the encoder from being re-initialized when changing codec from mono to stereo. by tina.legrand@webrtc.org · 13 years ago
c4f129f Improve the mixing saturation protection scheme. by andrew@webrtc.org · 13 years ago
41f3855 Upgrade libvpx to b615a6d4. by marpan@webrtc.org · 13 years ago
d30b688 Remove TraceScan executable. by andrew@webrtc.org · 13 years ago
4b13fc9 Add delay modification to process_test. by andrew@webrtc.org · 13 years ago
2f32b5c Fixes an issue where file playing could happen at a lower sampling frequency than the file. by henrike@webrtc.org · 13 years ago
eb4ef17 Removing vplib include and VideoInterpolator when not needed by mikhal@webrtc.org · 13 years ago
488ed92 Removing exceptions since not used by kjellander@webrtc.org · 13 years ago
c3a4dcd Removing exceptions since not used by kjellander@webrtc.org · 13 years ago
ad79d6f Removing exceptions since not used by kjellander@webrtc.org · 13 years ago