5ae9f5ed6cAdding logs in RTPSender::ReSendToNetwork.
mflodman@webrtc.org
2011-11-07 20:03:00 +00:00
bf483844afRestructuring and removing neteq_tests.gypi according to project structure discussed with Andrew. We want to flatten out the hierarchy and minimize the number of GYP files.
kjellander@webrtc.org
2011-11-07 16:05:19 +00:00
36e1ad9b5dRestructuring and removing ilbc_test.gypi. According to project structure discussed with Andrew. We want to flatten out the hierarchy and minimize the number of GYP files.
kjellander@webrtc.org
2011-11-07 15:27:11 +00:00
689cb300b5First version of PythonCharts. The reason why it is so simple is that I wanted to get something into the project that people can use to compare different test runs easily. More functionality will come later.
kjellander@webrtc.org
2011-11-07 15:25:47 +00:00
b353d21560...and now fix the Debug build.
andrew@webrtc.org
2011-11-05 00:57:33 +00:00
a5c4c1f1d4Fix for WebRTC issue 64, removed the screenupdate thread and events from start render as they are already created in the ctor. Review URL: http://webrtc-codereview.appspot.com/253008
vikasmarwaha@webrtc.org
2011-11-04 23:22:51 +00:00
20a370e875Changing the namespace of TestSuite to webrtc::test. Adding gmock initialization into main test runner class
kjellander@webrtc.org
2011-11-04 01:19:16 +00:00
1a8d08ad76Changing usage of gtest_main target, to use test_support_main instead.
kjellander@webrtc.org
2011-11-03 23:28:47 +00:00
89088b963eFix the path to protoc.gypi.
andrew@webrtc.org
2011-11-03 20:43:45 +00:00
2475a1953aCommitting a file that was part of CL 175002, but for wome reason weren't uploaded correctly.
tina.legrand@webrtc.org
2011-11-03 17:54:27 +00:00
fb389e3b02This CL is divided in several patches, to make review easier. Patch Set 1: Removing blanks at end of lines. Patch Set 2: Removing tabs. Patch Set 3: Fixing include-guards. Patch Set 4-7: Formatting files in the list. Patch Set 8: Formatting CNG.
tina.legrand@webrtc.org
2011-11-03 17:20:10 +00:00
a4b9660372Add mistakenly removed VAD enabling function.
andrew@webrtc.org
2011-11-03 01:36:27 +00:00
e203de7ba2jitter_buffer updates: 1. Determining continuity based on pictureId and not seq. numbers when available. 2. Hybrid bug fix: Don't set to decodable when the nack list is empty. Review URL: http://webrtc-codereview.appspot.com/255001
mikhal@webrtc.org
2011-11-03 00:42:52 +00:00
9116cf7c9bHave a guard on computing nrg to avoid wrap-around. This is discovered in a release test. During entropy coding of spectrum the value of "nrg" was too large and after shifting it became negative, resulting in decoder error. Review URL: http://webrtc-codereview.appspot.com/239016
turaj@webrtc.org
2011-11-01 17:29:34 +00:00
0ab521f754Resolving a crash related to strncopy followed by a strcat call. strncopy will not explicity copy or add a "\0" therefore strcat did not know where to append the "\n" which was causing an out of bounds crash. Because we are checking the length, strcpy should be good enough as it also copies the "\0". Please note that that I am pre-emptively adding 2 instead of 1 to the length to take into account of the \n that will be added later. Review URL: http://webrtc-codereview.appspot.com/253004
punyabrata@webrtc.org
2011-11-01 15:19:44 +00:00
36a992b030Merge streamparams and mediasession from libjingle and made necessary changes in peerconnection. -Removed ssrc from tracks. -Updated PeerConnectionMessage parsing and serialization.
perkj@webrtc.org
2011-11-01 11:46:56 +00:00
d6837709cfFixing VPMUnitTest compilation error on Windows. It tried to include Visual Leak Detector which is not a tool that is installed/configured by default in the build.
kjellander@webrtc.org
2011-11-01 01:51:10 +00:00
1e10bb32b9Remove global std::strings from fileutils.
andrew@webrtc.org
2011-10-31 20:22:02 +00:00
2c74bab8b9Remove unneeded assert and tracing.
andrew@webrtc.org
2011-10-31 19:54:20 +00:00
299e2c9ea4vie_autotest_custom_call.cc - fixed VieAutotestDevcoderObserver to use const int for videoChannel for IncomingCodecChanged, RequestNewKeyFrame
amyfong@webrtc.org
2011-10-31 19:10:26 +00:00
4d8c81878eThe implementation before this change list keeps the ownership of memory that is used by peer connection instances in the peer connection manager. This means that if the peer connection manager is deleted before all peer connections it has created, these peer connections will be pointing to invalid memory.
henrike@webrtc.org
2011-10-31 18:00:10 +00:00
066f9e5a2fRay, please verify that this cl fixes the issue. Once the verification has been made, please review:
henrike@webrtc.org
2011-10-28 23:15:47 +00:00
1f6d740571This CL is about to manually reset the ShutdownRenderEvent at StopPlayout(). It could happen that if you want to restart playout, the new sponsored Render thread would catch this event if the previous Render thread quits before this event is set. With this modification, the device plugging out/in during talking would be supported well. Review URL: http://webrtc-codereview.appspot.com/248002
braveyao@webrtc.org
2011-10-28 21:30:30 +00:00
8129752c3bAdd refcount and scoped_refptr.
perkj@webrtc.org
2011-10-28 15:08:54 +00:00
94cfde7c66Removed scoped_refptr from libjingle.gyp
perkj@webrtc.org
2011-10-28 14:26:41 +00:00
7e08613bdaMove refcount and scoped_refptr to merge with libjingle. Deleted scoped_refptr_msg.h.
perkj@webrtc.org
2011-10-28 14:26:25 +00:00
250cd6f41bAdded a VAD unit test to common_audio. At this stage it runs through the API calls, but should later be complemented with tests on a file. Review URL: http://webrtc-codereview.appspot.com/243002
bjornv@webrtc.org
2011-10-28 12:45:58 +00:00
eb65860720Reverts the workaround in r823 and solves a macro bug.
stefan@webrtc.org
2011-10-28 12:25:34 +00:00
dfbebb916cAdd a documented_interfaces watchlist.
andrew@webrtc.org
2011-10-27 22:33:27 +00:00
ca4666b75cvie wintest added hybrid protection mode also fixed Max Framerate to reflect its actually the min framerate Review URL: http://webrtc-codereview.appspot.com/244010
amyfong@webrtc.org
2011-10-27 21:16:40 +00:00
1e7e60b739Fixed issue build failling due to vie_autotest_custom_call calling GetBandwidthUsage, which was changed in r822. Review URL: http://webrtc-codereview.appspot.com/240014
amyfong@webrtc.org
2011-10-27 20:53:30 +00:00
2d28aff785Workaround for an issue where frames are grabbed for decoding prematurely.
stefan@webrtc.org
2011-10-27 16:13:18 +00:00
fbea4e555dSolves two bandwidth estimation issues and measures the sent video bitrate.
stefan@webrtc.org
2011-10-27 16:08:29 +00:00
7e4269e9eeChanged VP8 qp min and added noise reduction.
mflodman@webrtc.org
2011-10-27 12:59:47 +00:00
8fc663b3aeDon't trigger false ViE SetReceiveCodec warning.
mflodman@webrtc.org
2011-10-26 11:30:52 +00:00
6b7799021cFixing build errors on Windows platform. Minor changes...
kjellander@webrtc.org
2011-10-26 02:38:09 +00:00
fdde8b3fb7Add references to src/ copies for LICENSE etc.
andrew@webrtc.org
2011-10-26 01:05:07 +00:00
cb18121990Add an unpacker tool for audioproc debug files.
andrew@webrtc.org
2011-10-26 00:27:17 +00:00
fc9bcef8c7Data alignment fix for SSIM.
frkoenig@google.com
2011-10-26 00:07:32 +00:00
78c767f9baRewrote codec test to use fake camera.
phoglund@webrtc.org
2011-10-25 12:54:38 +00:00
d855c1a4e8Reverts r807 and fixes the real issue in the VCM.
stefan@webrtc.org
2011-10-25 11:52:48 +00:00
bdb55c806fThis CL is an attempt to remove a crash we can see when closing down VoiceEgine. It can happen that the capture thread tries to access an invalid object after StopPlayout has been called.
henrika@webrtc.org
2011-10-25 11:03:28 +00:00
f0cd394a2ePut fwrite calls under corresponding macros since they shouldn't show up in release build. This also make chromeos build happy. BUG=none TEST=compile Review URL: http://webrtc-codereview.appspot.com/247006
wjia@webrtc.org
2011-10-25 00:40:43 +00:00
913644b92dFor commiting changes in CL 277002, due to file structure changes introduced during the review of the code. Review URL: http://webrtc-codereview.appspot.com/246005
kma@webrtc.org
2011-10-24 21:36:33 +00:00
0d0037c2fdReturn cached data instead of sleeping in CpuWrapperMac (shaves 2s off WebrtcMediaEngine creation time on Mac). Review URL: http://webrtc-codereview.appspot.com/226005
henrike@webrtc.org
2011-10-24 15:48:14 +00:00
0a9c318c9fThe fread result is no longer ignored.
phoglund@webrtc.org
2011-10-24 15:33:07 +00:00