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  • 21632124dd Include gflags properly and X11 include order in VideoEngine. pbos@webrtc.org 2013-05-17 14:25:02 +00:00
  • f5d4cb1958 Include files from webrtc/.. paths in video_engine/ pbos@webrtc.org 2013-05-17 13:44:48 +00:00
  • 9f557c140e Improve wraparound handling in the render time extrapolator. stefan@webrtc.org 2013-05-17 12:55:07 +00:00
  • 14d7700d00 Moved command line parsing to internal tools and moved back the mic volume thingie. phoglund@webrtc.org 2013-05-17 11:52:08 +00:00
  • e874a8f24b Enable WebRTC demo application on x86 Android fischman@webrtc.org 2013-05-17 05:41:07 +00:00
  • 8630cfe016 Guarding certain operations, e.g. bandwidth estimation, RTCP statistics update etc., not to be run on sync RTPS. turaj@webrtc.org 2013-05-16 23:54:54 +00:00
  • fe307e1332 Add one unit test for NACKing a key frame hclam@chromium.org 2013-05-16 21:19:59 +00:00
  • b3e5acfb66 Cleanup traces in WebRTC hclam@chromium.org 2013-05-16 21:13:02 +00:00
  • b9bb3d1e7d Avoid resetting encoder on identical settings. pbos@webrtc.org 2013-05-16 18:40:48 +00:00
  • 890f6092e6 Bugfix: VCM would report wrong sentBitrate marpan@webrtc.org 2013-05-16 15:38:44 +00:00
  • 9919ad5caf Formatted FEC stuff. phoglund@webrtc.org 2013-05-16 15:06:28 +00:00
  • 5c1948dfaf Moved force_volume_max to its own gyp file to avoid a circular dependency. phoglund@webrtc.org 2013-05-16 13:59:19 +00:00
  • 61d3c552a1 Wrote a small portable tool for forcing the mic volume to 100%. phoglund@webrtc.org 2013-05-16 13:10:00 +00:00
  • 29d5839233 New VideoEngine API implementation on top of old one, first steps. pbos@webrtc.org 2013-05-16 12:08:03 +00:00
  • 2038214c77 Log too long non-decodable duration events. stefan@webrtc.org 2013-05-16 11:39:06 +00:00
  • 4dee30927a Remove SetOverUseDetectorOptions and cleaned ViESharedData. mflodman@webrtc.org 2013-05-16 11:13:18 +00:00
  • 7ebbea14a9 Add handling of the absolute send time header extension to the rtp_rtcp module. solenberg@webrtc.org 2013-05-16 11:10:31 +00:00
  • 59a06670b5 Updated apprtc demo to interop with firefox. vikasmarwaha@webrtc.org 2013-05-16 01:05:19 +00:00
  • 40298d452c Added webaudio-and-webtrc.html to the demos index.html. vikasmarwaha@webrtc.org 2013-05-16 00:50:38 +00:00
  • 8c2e78b2de Roll chromium_revision 193311:199267 fischman@webrtc.org 2013-05-15 22:50:23 +00:00
  • 6cfa3907c8 Updating NACK RTX test mikhal@webrtc.org 2013-05-15 20:17:43 +00:00
  • cb20a5b2d7 VCM/JB: Bug fix in ExtractAndSetDecode BUG=1771 R=stefan@webrtc.org mikhal@webrtc.org 2013-05-15 17:10:44 +00:00
  • 5add4ad09c RemoteBitrateEstimatorTest::TestRateIncreaseReordering sent in arrival timestamps in non monotonically increasing order. Fixed. solenberg@webrtc.org 2013-05-15 13:49:57 +00:00
  • c93b1d038d CoreAudio Win: release resources safely under certain rare circumstance in GTalkplugin braveyao@webrtc.org 2013-05-15 10:14:56 +00:00
  • e2a800644c Linux support for typing detection niklas.enbom@webrtc.org 2013-05-14 21:33:11 +00:00
  • 4ce838934c Address sanitizer out of bounds read in iSAC turaj@webrtc.org 2013-05-14 17:42:22 +00:00
  • 6bee05a4aa Remove const for plain data types in common_video/ pbos@webrtc.org 2013-05-14 14:27:15 +00:00
  • 29b2219914 Adding a factory to remote bitrate estimator and allow it to be set via config. andresp@webrtc.org 2013-05-14 12:10:58 +00:00
  • 1673481ed7 Fixes a bug where the render buffer size (and indirectly the non-continuous duration) was computed incorrectly. stefan@webrtc.org 2013-05-14 12:00:47 +00:00
  • 736c6f775e Fixed more perf expectations. phoglund@webrtc.org 2013-05-14 11:26:14 +00:00
  • 80c7e3b606 Adjusted perf expectations for mac large tests. phoglund@webrtc.org 2013-05-14 10:51:13 +00:00
  • bb984f516e Removed Mac capture crash and memory leak. mflodman@webrtc.org 2013-05-14 10:47:19 +00:00
  • a6ff84503e Add script for comparing video quality kjellander@webrtc.org 2013-05-14 09:43:04 +00:00
  • 6d07ad9ccc Added protoc_wrapper to blacklist, fixed tools/PRESUBMIT.py which was passing in the wrong args to CheckLongLines. phoglund@webrtc.org 2013-05-14 09:42:39 +00:00
  • 527f6c62fc Reformatted FEC tables. phoglund@webrtc.org 2013-05-14 09:25:01 +00:00
  • 8e3b594831 Remove const for plain data types in common_audio/ pbos@webrtc.org 2013-05-14 09:24:49 +00:00
  • 9213521ea9 Remove const for plain data types in voice_engine/ pbos@webrtc.org 2013-05-14 08:31:39 +00:00
  • 185bae4b6f Replace ExtraCodecOptions with new Config class that supports multiple settings at once. andresp@webrtc.org 2013-05-14 08:02:25 +00:00
  • c9cb4fffac Fix typo in log statement. witdh should be width. BUG=none TESTED=try bots Review URL: https://webrtc-codereview.appspot.com/1466004 fbarchard@google.com 2013-05-14 05:02:08 +00:00
  • 7bfb3a3227 Add more tracing for key frames. justinlin@chromium.org 2013-05-13 22:59:00 +00:00
  • 941fcc5841 Increased the limit for KViEMaxCaptureDevices from 10 to 256. See issue 1343. TBR=juberti@google.com vikasmarwaha@webrtc.org 2013-05-13 20:28:23 +00:00
  • 1993a559e8 Added Stereo url paramter to apprtc demo. vikasmarwaha@webrtc.org 2013-05-13 18:48:09 +00:00
  • 52b3905ec8 Updated WebRTC version to 3.31 TBR=wu@webrtc.org elham@webrtc.org 2013-05-13 17:00:56 +00:00
  • 43bf6ce322 Revert 4008 "Avoid resetting video encoder for similar configs." phoglund@webrtc.org 2013-05-13 15:39:26 +00:00
  • c53480fbcf Disabled flaky codec test (RunsCodecTestWithoutErrors) phoglund@webrtc.org 2013-05-13 15:10:02 +00:00
  • aa4efd1535 Avoid resetting video encoder for similar configs. pbos@webrtc.org 2013-05-13 11:27:16 +00:00
  • 7707d060bb Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. andresp@webrtc.org 2013-05-13 10:50:50 +00:00
  • 7a5615bc84 New WebAudio-WebRTC demo. henrika@webrtc.org 2013-05-13 09:29:13 +00:00
  • 7ee822805d Remove TEXT(x) for BUILDINFO macros. pbos@webrtc.org 2013-05-13 09:29:03 +00:00
  • 6b68c28cb1 Added a config class to ease passing a set of options across webrtc. Its main design reason is to expose control of experimental webrtc features. andresp@webrtc.org 2013-05-13 08:06:36 +00:00
  • 9ecd6861eb Add svn:eol-style back which is lost in r3993 mistakenly. braveyao@webrtc.org 2013-05-13 05:38:13 +00:00
  • a404d1d8de Change watchlist. leozwang@webrtc.org 2013-05-10 22:46:55 +00:00
  • 7311083ccc Revert 3977 BUG=webrtc:1749 tnakamura@webrtc.org 2013-05-10 22:33:50 +00:00
  • 05ea12f12e Reverting r3978 elham@webrtc.org 2013-05-10 17:04:59 +00:00
  • d6ed000585 This is the first step to convert building the Android WebRTC demo to a proper GYP target, android ndk toolchains is being used to build the jni cpp files instead of using ndk-build. fischman@webrtc.org 2013-05-10 16:34:01 +00:00
  • 571b3369e7 Updating perf mikhal@webrtc.org 2013-05-09 20:03:47 +00:00
  • 1e3c794688 Use 2 threads for HD, or 1 for VGA or less. BUG=1739 TEST=try bots Review URL: https://webrtc-codereview.appspot.com/1438005 fbarchard@google.com 2013-05-09 18:43:38 +00:00
  • 06806701f0 Updating perf mikhal@webrtc.org 2013-05-09 17:42:58 +00:00
  • 6a36f0e46f Since the layout of the Android WebRTC demo application is fixed, if we start the demo application in portrait postion, the activity will be destroyed and then created again, force the demo application to start in landscape position to avoid activity re-creation. fischman@webrtc.org 2013-05-09 17:40:33 +00:00
  • e525309004 WebRTCDemo Android doesn't hangle activity recreation correctly. Also optimize Statsview a little bit. braveyao@webrtc.org 2013-05-09 08:52:50 +00:00
  • 219762a68a Drop Virtual webcam check script as moved into buildbot scripts. kjellander@webrtc.org 2013-05-09 07:53:08 +00:00
  • ebdfa8dcba Add fischman into OWNERS of WebRTCDemo Android. braveyao@webrtc.org 2013-05-09 07:30:38 +00:00
  • d72262dc01 Fix compile errors in ViE with latest clang. andrew@webrtc.org 2013-05-09 02:12:07 +00:00
  • c6a3755ada Update SincResampler with the latest Chromium code. andrew@webrtc.org 2013-05-08 20:35:43 +00:00
  • 44272739c2 Clean creation of VideoEngine: - clean a static variable just used to debug and not so necessary IMO. - clean a really ugly reinterpret cast - clean a extern "C" code and loading of dlls which is no longer in use. andresp@webrtc.org 2013-05-08 19:20:23 +00:00
  • 6155be2c61 Add /tools/protoc_wrappers to .gitignore. andrew@webrtc.org 2013-05-08 18:51:07 +00:00
  • aeb7d8757d Tweaked webrtc_reformat. phoglund@webrtc.org 2013-05-08 13:56:23 +00:00
  • 315d39866e Formatted dtmf_queue. phoglund@webrtc.org 2013-05-08 10:04:06 +00:00
  • 73a4d5ab12 Add script to ensure virtual webcam is running. kjellander@webrtc.org 2013-05-08 09:20:41 +00:00
  • f6d67ae21f Disable clang C++11 warnings to permit OVERRIDE keyword. pbos@webrtc.org 2013-05-08 08:34:34 +00:00
  • d98e784f5f Fix VCMProcessTimer::TimeUntilProcess() unsigned-integer underflow problem. stefan@webrtc.org 2013-05-08 06:38:53 +00:00
  • b55a12ad32 Enable protobuf use in Chromium. andrew@webrtc.org 2013-05-08 00:03:30 +00:00
  • e53084f837 Update protoc.gypi to match Chromium's latest. andrew@webrtc.org 2013-05-07 23:19:58 +00:00
  • 3be565b502 Refactoring for typing detection niklas.enbom@webrtc.org 2013-05-07 21:04:24 +00:00
  • ef14488d03 Trigger a PLI if the duration of non-decodable frames exceeds a threshold. stefan@webrtc.org 2013-05-07 19:16:33 +00:00
  • 8f86cc8712 VCM/Receiver: Return null when can't extract frame. mikhal@webrtc.org 2013-05-07 18:05:21 +00:00
  • 474e915272 Relanding 3962: VCM/JB: Porting jitter_buffer_test to gtest TBR=stefan@webrtc.org mikhal@webrtc.org 2013-05-07 16:55:03 +00:00
  • 759b041019 Relanding r3952: VCM: Updating receiver logic BUG=r1734 R=stefan@webrtc.org mikhal@webrtc.org 2013-05-07 16:36:00 +00:00
  • 9c7685f9a6 VCM/JB: Break and skip to key if possible mikhal@webrtc.org 2013-05-07 16:07:52 +00:00
  • 3004c79c6a Fix clang errors in non-GYP_DEFINES=clang=1 build pbos@webrtc.org 2013-05-07 12:36:21 +00:00
  • d3a1959678 Fix jitter buffer unittest. stefan@webrtc.org 2013-05-07 12:35:58 +00:00
  • a5dee33639 Correctly add packets to nack list when sequence number wraps. stefan@webrtc.org 2013-05-07 11:11:17 +00:00
  • 0f29810288 Fix crash in pacer. pwestin@webrtc.org 2013-05-06 16:37:22 +00:00
  • 4ce19b1664 Revert r3952 "VCM: Updating receiver logic" stefan@webrtc.org 2013-05-06 13:16:51 +00:00
  • 273759048c Revert r3956 "VCM/JB: Porting jitter_buffer_test to gtest." stefan@webrtc.org 2013-05-06 13:12:58 +00:00
  • 233c58de47 Landing 1399004, Minor clean up on the un-used _measureDelay code Those code is/will never used, removing it makes the code better. xians@webrtc.org 2013-05-06 11:52:47 +00:00
  • 59aaebc3cd Add an option to override the TestToStderr trace printout time. andrew@webrtc.org 2013-05-05 19:44:19 +00:00
  • f9c289bafe Consolidate all third party licenses in LICENSE_THIRD_PARTY. andrew@webrtc.org 2013-05-05 18:54:10 +00:00
  • df3da84ec8 Updated WebRTC version number to 3.30 elham@webrtc.org 2013-05-03 23:11:37 +00:00
  • 45f2da0920 VCM/JB: Porting jitter_buffer_test to gtest. Tests were not modified, but ported as is. mikhal@webrtc.org 2013-05-03 22:22:46 +00:00
  • a31c428307 Remove 44.1 kHz workaround from AudioDevice on PulseAudio. andrew@webrtc.org 2013-05-03 19:01:46 +00:00
  • 7cb766b016 Remove 44.1 kHz workaround from AudioDevice on WASAPI. andrew@webrtc.org 2013-05-03 18:56:38 +00:00
  • bd4a2feddb Fix off-by-one buffer overflow in WebRtcNetEQ_PacketBufferInsert(). sergeyu@chromium.org 2013-05-03 18:11:36 +00:00
  • d3cd565ecf VCM: Updating receiver logic mikhal@webrtc.org 2013-05-03 17:54:18 +00:00
  • d293a58eaf Correct and update dir name leozwang@webrtc.org 2013-05-03 17:16:40 +00:00
  • 77f6b2175e Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..." pbos@webrtc.org 2013-05-03 12:02:11 +00:00
  • 2580bc4c30 Get rid of some unnecessary copying when sending REMBs. solenberg@webrtc.org 2013-05-03 09:22:14 +00:00
  • d5726a1286 Formatting ACM tests tina.legrand@webrtc.org 2013-05-03 07:34:12 +00:00
  • 03efc89151 Fix when SetMinimumPlayoutDelay is configured to 0 pwestin@webrtc.org 2013-05-02 21:19:12 +00:00
  • 42636e82d0 Removing bad code resulting in flaky test. pwestin@webrtc.org 2013-05-02 21:02:04 +00:00