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  • 0b8636a783 In AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(), continue enumerating if one individual device failed. braveyao@webrtc.org 2013-07-04 07:24:12 +00:00
  • 1303af31d6 Makes it possible to build ndk-apks of native unit tests if the workspace is inside a chromium checkout. henrike@webrtc.org 2013-07-03 21:50:33 +00:00
  • d900e8bea8 Proper spacing for end-of-namespace comments. pbos@webrtc.org 2013-07-03 15:12:26 +00:00
  • 45426eadf5 In call to Opus decoder: frame length too large tina.legrand@webrtc.org 2013-07-03 13:32:04 +00:00
  • f6f033f8bd Possible divide by 0 in ACM. tina.legrand@webrtc.org 2013-07-03 12:00:14 +00:00
  • b1698ab827 Error in update of read index in ACM tina.legrand@webrtc.org 2013-07-03 09:25:34 +00:00
  • ecd3c800c4 Add Magnus to root owners. tommi@webrtc.org 2013-07-03 08:21:41 +00:00
  • c66aaaf921 Rename unit_test.{cc,h} under module_unittest. pbos@webrtc.org 2013-07-03 07:56:33 +00:00
  • 510dfad636 Update myself in webrtc watchlist yujie.mao@webrtc.org 2013-07-03 01:13:18 +00:00
  • 65a1f2cb2b Remove log of undefined input values in GetCodec. pbos@webrtc.org 2013-07-02 13:02:14 +00:00
  • 504af45a6f Diff NTP and internal once in VideoCaptureImpl. pbos@webrtc.org 2013-07-02 10:15:43 +00:00
  • 546c91dc2e Build all java files into jar for each module on Android fischman@webrtc.org 2013-07-01 17:52:39 +00:00
  • d4803ced60 WebRTCViEDemo: Use global reference when passing variables across different threads yujie.mao@webrtc.org 2013-07-01 14:55:37 +00:00
  • 90cc3b95b7 Android opengles renderer: add thread sync to swap frame and draw native. braveyao@webrtc.org 2013-06-28 23:53:11 +00:00
  • 5616abadf5 Suppress excessive logging in video_coding hclam@chromium.org 2013-06-27 19:47:40 +00:00
  • 2a7fd5355d Moves tools/update.py to trunk/webrtc/tools and updates it so that it no longer pulls any information from the DEPS file. henrike@webrtc.org 2013-06-27 18:36:28 +00:00
  • 83cebb25d7 Removes unused main function that is poluting the build. henrike@webrtc.org 2013-06-27 18:31:13 +00:00
  • 0021632f40 Re-add WebRTCDemo dependencies as dependencies (not just inputs) because they also need to be built for this target! fischman@webrtc.org 2013-06-27 17:35:32 +00:00
  • 1d4a2d5daf Move TickTime::QueryOsForTicks out-of-line fischman@webrtc.org 2013-06-27 17:15:20 +00:00
  • 4cf1a8af69 Removes kStateFree and kStateDecoding, added a free_frames_ list which simplifies finding a free frame. stefan@webrtc.org 2013-06-27 15:20:14 +00:00
  • 7bcc7e3b43 Fixed bad parameter passing in compare_videos.py phoglund@webrtc.org 2013-06-27 14:05:26 +00:00
  • 2de80ddc72 Fix unnamed-type-template-args warnings on clang. pbos@webrtc.org 2013-06-27 10:18:09 +00:00
  • 3145a642b7 Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes. fischman@webrtc.org 2013-06-26 20:20:05 +00:00
  • e6168f5f41 Adding a first simple version of overuse detection, but not hooked up. mflodman@webrtc.org 2013-06-26 11:23:01 +00:00
  • 1c986e7c89 Removed ViE file API. mflodman@webrtc.org 2013-06-26 09:12:49 +00:00
  • a5fd2f1348 Do basic parsing of RTCP headers in PcapFileReader to enable log filtering. solenberg@webrtc.org 2013-06-26 08:36:07 +00:00
  • 892d750ba6 Add *.DS_Store to .gitignore so that ._.DS_Store is ignored too. solenberg@webrtc.org 2013-06-26 08:22:53 +00:00
  • 91811e2b04 Remove unused multi stream bandwidth estimator. solenberg@webrtc.org 2013-06-25 20:36:14 +00:00
  • a4c5abb52a Make sure padding packets are sent. stefan@webrtc.org 2013-06-25 15:46:16 +00:00
  • bb25256775 Added changes in apprtc demo to ignore turn address through query string for FF. Also made sure that the iceServers array doesnot include transport parameter in turn url for FF. Finally removed turn: from the turn_url when creating iceservers for pre-M28 chrome. vikasmarwaha@webrtc.org 2013-06-25 14:52:51 +00:00
  • 3348ae2b97 mac: Mark kCGLPFAFullScreen as allowed in a 10.6-only function. sergeyu@chromium.org 2013-06-21 23:33:10 +00:00
  • bb4f225a5b Roll libvpx to 207593. -pick up libvpx roll to c259af4f. marpan@webrtc.org 2013-06-21 22:19:34 +00:00
  • 6eb53f71d6 Fix memory bot failure hclam@chromium.org 2013-06-20 23:01:39 +00:00
  • 2e402ce873 Enqueue packet in pacer if sending fails hclam@chromium.org 2013-06-20 20:18:31 +00:00
  • 9ca7360b97 VCM: removing max jitter estimate mikhal@webrtc.org 2013-06-20 20:13:07 +00:00
  • 0851df8d60 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. andrew@webrtc.org 2013-06-19 17:03:47 +00:00
  • 8ccb9f9716 Fixes some pacer/padding issues found while testing. stefan@webrtc.org 2013-06-19 14:13:42 +00:00
  • 2d7617afce Add dummy Android test APK to be used for buildbot automation testing. kjellander@webrtc.org 2013-06-19 09:10:49 +00:00
  • d7148c86c5 Use 3 threads for higher than 720p resolutions BUG=1893 TEST=untested R=ajm@google.com, andrew@webrtc.org, dingkai@google.com, marpan@google.com, marpan@webrtc.org fbarchard@google.com 2013-06-18 22:06:42 +00:00
  • 30fb7b83d5 Add a log message to see video delay break down hclam@chromium.org 2013-06-18 21:37:09 +00:00
  • 6cfe178af2 Chromium Android tools for test execution. kjellander@webrtc.org 2013-06-18 07:14:33 +00:00
  • a20eb91154 Make ScreenCapturerMac work in versions of OSX before Lion. sergeyu@chromium.org 2013-06-17 22:22:40 +00:00
  • 9e182795a9 Enable ScreenCapturer unittests sergeyu@chromium.org 2013-06-17 21:14:36 +00:00
  • a590b41c9a Use intptr_t to represent window IDs on all platforms. sergeyu@chromium.org 2013-06-17 20:02:21 +00:00
  • 508a84b255 Wire up pacer-based padding. stefan@webrtc.org 2013-06-17 12:53:37 +00:00
  • 50fb4afade Revert r4145 "Revert 4127 "Switch frame list implementation to std::map."" stefan@webrtc.org 2013-06-17 07:33:58 +00:00
  • c8b29a2feb Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de..."" stefan@webrtc.org 2013-06-17 07:13:16 +00:00
  • 7262ad1385 Fix AV sync issue hclam@chromium.org 2013-06-15 06:51:27 +00:00
  • 9b23ecb939 Log current and target AV delay in ViESyncModule hclam@chromium.org 2013-06-14 23:30:58 +00:00
  • 63e988856e Merge more tests into modules_{unit,integration}tests. kjellander@webrtc.org 2013-06-14 20:09:44 +00:00
  • f27389ca9f WebRTCDemo: ensures that using front and back camera work as expected. henrike@webrtc.org 2013-06-14 05:37:13 +00:00
  • d4ed1a3e2c Fixes linker issue with no op trace. henrike@webrtc.org 2013-06-13 09:34:54 +00:00
  • a19333954d Apprtc CSS: Add flip to local view of FireFox and remove warning of Canary braveyao@webrtc.org 2013-06-13 03:49:03 +00:00
  • fee739c224 Risk of division by zero. turaj@webrtc.org 2013-06-12 20:10:06 +00:00
  • dd97ef4e28 Revert 4211 "Build all java files into jar for each module on An..." fischman@webrtc.org 2013-06-12 17:39:29 +00:00
  • 20a993f88a Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. kjellander@webrtc.org 2013-06-12 14:38:01 +00:00
  • 935d705370 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. kjellander@webrtc.org 2013-06-12 13:59:57 +00:00
  • 04996cd5e5 Fix breakage due to test_fec conversion to gtest. kjellander@webrtc.org 2013-06-12 12:15:33 +00:00
  • 22bbbdfa68 Convert test_fec to gtest kjellander@webrtc.org 2013-06-12 11:55:05 +00:00
  • 7124dd8561 Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test. kjellander@webrtc.org 2013-06-12 08:28:09 +00:00
  • 18275a8429 Update bots to make LKGR progress. kjellander@webrtc.org 2013-06-12 08:10:18 +00:00
  • b097670264 G722_1/G722_1C codecs won't instantiate tina.legrand@webrtc.org 2013-06-12 07:41:42 +00:00
  • 2ef9513916 libyuv r723 with convert util -attenuate feature used to fix transparent pixels used by Effects. By attenuating and then unattenuating, any transparent pixels will have RGB value of black, which will filter correctly when bilinear resized. BUG=none TEST=try bots R=andrew@webrtc.org fbarchard@google.com 2013-06-11 22:03:29 +00:00
  • 6c35e0b0f7 Reorganize test targets in WebRTC kjellander@webrtc.org 2013-06-11 08:29:17 +00:00
  • 6d6d95e2b8 Add support for test disable files in webrtc_tests.py kjellander@webrtc.org 2013-06-11 06:03:32 +00:00
  • 1374965680 Build all java files into jar for each module on Android fischman@webrtc.org 2013-06-10 23:34:27 +00:00
  • 4af0878e57 Allow the screen capturer to capture oversized cursors and cursors without alpha channel (Windows). alexeypa@chromium.org 2013-06-10 22:29:17 +00:00
  • 5e03f8ab67 Landing binary cursor image files to be used in a follow up CL. alexeypa@chromium.org 2013-06-10 21:07:31 +00:00
  • dfa1c4afc6 libyuv r722 for OWNERS file for chromium, white space fix for lint, unittests on scale use randomize to reduce overhead, and neon change from vld1.u8 to vld1.8 for better compiler portability. BUG=none TEST=none R=andrew@webrtc.org fbarchard@google.com 2013-06-10 19:35:17 +00:00
  • fe6b57187d AppRTCDemo: delete hosted android_channel.html now that it's no longer necessary. fischman@webrtc.org 2013-06-10 17:22:50 +00:00
  • 5137b9752f Updated WebRTC version to 3.33 elham@webrtc.org 2013-06-10 17:03:51 +00:00
  • 509754c4c9 Making no NACK mode work again in VideoEngine. mflodman@webrtc.org 2013-06-10 15:50:12 +00:00
  • 1819fd711a RW lock access to ssrc maps in VideoCall. pbos@webrtc.org 2013-06-10 13:48:26 +00:00
  • adb51f5709 Add back the WEBRTC_DIRECT_TRACE flag. solenberg@webrtc.org 2013-06-10 09:03:41 +00:00
  • 83a062cc5f AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout() braveyao@webrtc.org 2013-06-10 08:09:05 +00:00
  • 569fdef732 Revert some variables to uint32_t to fix compile errors on Mac gcc. andrew@webrtc.org 2013-06-08 00:43:25 +00:00
  • 6f69eb78dd Allow audio devices with up to 64 channels on Mac. andrew@webrtc.org 2013-06-07 17:56:50 +00:00
  • 1064cf06b0 Fixed Rtp/Rtcp tests pwestin@webrtc.org 2013-06-07 16:03:19 +00:00
  • 6367fe885a Fix relative path to .gitignore and other minor changes. andrew@webrtc.org 2013-06-07 15:43:04 +00:00
  • 3ba883f0fc Removing functionality for inserting pre-encoded frames instead of raw video frames. The functionality hasn't been used for a long time and should be done properly if used in the future. mflodman@webrtc.org 2013-06-07 13:57:57 +00:00
  • b69cc15467 Add script for appending entries to .gitignore. andrew@webrtc.org 2013-06-07 02:04:12 +00:00
  • da710448b2 Fix size_t to int conversion error on Win64. andrew@webrtc.org 2013-06-07 01:43:12 +00:00
  • 7e4ff354e3 Remove fake screen capturer because it's not used anywhere. sergeyu@chromium.org 2013-06-06 23:11:33 +00:00
  • 8d80fa83fc Fix for STL vector function data not available. pwestin@webrtc.org 2013-06-06 21:33:06 +00:00
  • d30859e58e Connect ACM with RTP module for audio NACK. Depends on http://review.webrtc.org/1507004/ pwestin@webrtc.org 2013-06-06 21:09:01 +00:00
  • a305e9612a Nack for audio. turaj@webrtc.org 2013-06-06 19:00:09 +00:00
  • d9c4658756 Fix leaks in DesktopRegion sergeyu@chromium.org 2013-06-05 19:24:42 +00:00
  • 2b3a29a1fa Implement DetectNumberOfCores on Android and make it consistent on Linux and Android fischman@webrtc.org 2013-06-05 16:37:42 +00:00
  • db24995680 Wire up Nack for Voe pwestin@webrtc.org 2013-06-05 15:33:20 +00:00
  • 7f1b0ae888 Fix init list for VideoSendStream::Config::Rtp. pbos@webrtc.org 2013-06-05 11:39:18 +00:00
  • 025f4f152b Stats+Config moved into VideoSend/ReceiveStreams. pbos@webrtc.org 2013-06-05 11:33:21 +00:00
  • fec34d7afa Merge webrtc_utility_unittests into modules_unittests. kjellander@webrtc.org 2013-06-05 08:58:46 +00:00
  • b2d29bd2fe Restore relative include paths to libyuv. andrew@webrtc.org 2013-06-04 23:53:48 +00:00
  • 3942f3a985 Issue 1847, memcopy is wrong and unnecessary, it is sufficient to store the pointer before clearing the instance, and write back the pointer. turaj@webrtc.org 2013-06-04 21:31:22 +00:00
  • 16d78bd307 Fix scale.cc build error with mingw64 -m32 gcc BUG=571 TESTED=gcc scale.cc R=andrew@webrtc.org fbarchard@google.com 2013-06-04 19:41:00 +00:00
  • 9238de9d49 resolve b9050210. Avoid pushing sync packet before any packet received. Do not turn on AV-sync if initial delay is zero. turaj@webrtc.org 2013-06-04 19:18:39 +00:00
  • 3d34f66292 Move screen capturers from chromium to webrtc. sergeyu@chromium.org 2013-06-04 18:51:23 +00:00
  • b7a8f43670 Roll chromium_revision in webrtc 199267:203806 fischman@webrtc.org 2013-06-04 17:10:24 +00:00
  • 430464c776 Add WebKit/Tools/Scripts to support Android test execution. kjellander@webrtc.org 2013-06-04 16:29:45 +00:00
  • a817962bab Refactor padding and rtp header functionality. stefan@webrtc.org 2013-06-04 13:47:36 +00:00