generic-library/webrtc
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  • 82462aade0 Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate. stefan@webrtc.org 2014-10-23 11:57:05 +00:00
  • 2192701135 Using the Unused turn configuration in two way test houssainy@google.com 2014-10-23 08:40:53 +00:00
  • ad553a2731 Let video_loopback use internal VCM capturers. pbos@webrtc.org 2014-10-23 08:24:02 +00:00
  • 15c717beee Add a memcheck exclusion for EndToEndTest.CanSwitchToUseAllSsrcs. andrew@webrtc.org 2014-10-23 05:37:37 +00:00
  • a9f0898e7d (Auto)update libjingle 78273470-> 78296920 buildbot@webrtc.org 2014-10-22 22:02:00 +00:00
  • 7bb4a9881d Merging Henrik's and Peter's changes for AppRTCDemo from https://github.com/hkjellander/AppRTCDemo. glaznev@webrtc.org 2014-10-22 17:43:37 +00:00
  • fce8f5d319 NOTE: This code review based on the running issue: https://webrtc-codereview.appspot.com/24939004/ houssainy@google.com 2014-10-22 17:24:20 +00:00
  • 3382059e55 Adding Two way video and audio streaming test to RtcBot houssainy@google.com 2014-10-22 17:17:15 +00:00
  • e9b7d03db6 HTTPS Server used instead of HTTP for loading the bots to avoid the media permission pop-up clicks every time running the test. houssainy@google.com 2014-10-22 16:34:25 +00:00
  • fb5410a8b7 (Auto)update libjingle 78262388-> 78262615 buildbot@webrtc.org 2014-10-22 15:45:17 +00:00
  • eacc6e4657 Remove some disabled tests in WebRtcVideoEngine2. pbos@webrtc.org 2014-10-22 15:36:54 +00:00
  • 82e430c316 Suppress libyuv uninitialized read in CopyRow_AVX kjellander@webrtc.org 2014-10-22 13:51:49 +00:00
  • 32452b20b8 Make ReconfigureVideoEncoder use current bitrate. pbos@webrtc.org 2014-10-22 12:15:24 +00:00
  • 860ccc9407 Tighten up MSan blacklist.txt owners. kjellander@webrtc.org 2014-10-22 11:20:07 +00:00
  • 3f8f5554a0 Disable TestVp8Impl.BaseUnitTest on MSan. pbos@webrtc.org 2014-10-22 10:30:30 +00:00
  • 76960d5f74 For FIR packet, payload length is zero, so SendToNetwork function is failing. stefan@webrtc.org 2014-10-22 09:47:14 +00:00
  • 1d9af96c06 Roll chromium_revision de13cf4..28d1981 (299488:300483) kjellander@webrtc.org 2014-10-22 06:43:29 +00:00
  • 67cf1d742b Break out WebRtcNs_Windowing function in ns_core aluebs@webrtc.org 2014-10-21 22:35:40 +00:00
  • 0e7099244c Break out WebRtcNs_Energy function in ns_core aluebs@webrtc.org 2014-10-21 22:14:10 +00:00
  • 7634c09406 Break out WebRtcNs_IFFT function in ns_core aluebs@webrtc.org 2014-10-21 21:27:00 +00:00
  • a5c36b397a (Auto)update libjingle 78193292-> 78199328 buildbot@webrtc.org 2014-10-21 20:44:16 +00:00
  • b6173abe59 Fix local address leakage when IceTransportsType is relay guoweis@webrtc.org 2014-10-21 20:40:21 +00:00
  • 333e2556ed Break out WebRtcNs_UpdateBuffer function in ns_core aluebs@webrtc.org 2014-10-21 20:33:09 +00:00
  • 1288cbb704 (Auto)update libjingle 78106439-> 78193292 buildbot@webrtc.org 2014-10-21 19:29:16 +00:00
  • def1e97ed2 Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests henrik.lundin@webrtc.org 2014-10-21 12:48:29 +00:00
  • 78ea06dd34 audio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> bjornv@webrtc.org 2014-10-21 07:17:24 +00:00
  • 913f7b8d5e Fix for glitches in ACM when switching desired output sample rate henrik.lundin@webrtc.org 2014-10-21 06:54:23 +00:00
  • a8c0edd29f Avoid using EGLContext class for Android 4.1 and below. glaznev@webrtc.org 2014-10-20 19:08:05 +00:00
  • b69ea9a35a common_audio: Replaced invalid operand in min_max_operations_neon.S" bjornv@webrtc.org 2014-10-20 14:08:35 +00:00
  • fa553ef605 Set up start bitrate in WebRtcVideoEngine2. pbos@webrtc.org 2014-10-20 11:07:07 +00:00
  • b35b136480 Make avg_{psnr,ssim}_threshold_ const. pbos@webrtc.org 2014-10-20 09:14:38 +00:00
  • 2abebe7baf audio_coding/codecs/isac/main: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> bjornv@webrtc.org 2014-10-20 08:26:41 +00:00
  • a5ce7bbe17 audio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> bjornv@webrtc.org 2014-10-20 08:24:54 +00:00
  • 28100cb388 Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." henrike@webrtc.org 2014-10-17 22:03:39 +00:00
  • 7992b40994 (Auto)update libjingle 77953038-> 77970462 buildbot@webrtc.org 2014-10-17 21:20:28 +00:00
  • b1dac33cac Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..." henrike@webrtc.org 2014-10-17 18:54:46 +00:00
  • 58202946a7 Cleaning up Android AppRTCDemo. glaznev@webrtc.org 2014-10-17 17:42:38 +00:00
  • 0371a37f85 Moving creating TURN configration to the host machine instead of the bots - rtcBot houssainy@google.com 2014-10-17 16:43:50 +00:00
  • f7030d4ed7 Query Android device orientation on every camera frame received. glaznev@webrtc.org 2014-10-17 16:25:06 +00:00
  • 9c58ea8d56 rtc_unittest: copied gtest excludes from libjingle_p2p_unittest since its tests have move to rtc_unittests. henrike@webrtc.org 2014-10-17 16:12:33 +00:00
  • c221db6165 Test names changed from e.g) testOneWayVideo/chrome=>chrome to testOneWayVideo/chrome-chrome. houssainy@google.com 2014-10-17 09:13:43 +00:00
  • 264e66f7a5 Add encoded_timestamp to AudioEncoder base class henrik.lundin@webrtc.org 2014-10-16 21:16:07 +00:00
  • 9ea6f8a84d New interface class AudioEncoder henrik.lundin@webrtc.org 2014-10-16 11:26:24 +00:00
  • 8efaa270d8 Disable a bunch of Nat and Ice tests when running under DrMemory. stefan@webrtc.org 2014-10-16 11:21:42 +00:00
  • 458c2c3b06 Improve rtcbot to load all test files at start and allow them to registerTests via: registerBotTest. After loading all tests main.js starts running the requested one on the command arguments. andresp@webrtc.org 2014-10-16 07:36:37 +00:00
  • 9aed002090 Add ability to include a larger time span (in addition to encode time) for measuring the processing time of a frame. Controlled by setting enable_extended_processing_usage. Enabled by default. asapersson@webrtc.org 2014-10-16 06:57:12 +00:00
  • d1ba6d9cbf Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. henrike@webrtc.org 2014-10-15 17:30:28 +00:00
  • 3e2f8ff36c Selecting bot_type changed to be specified in the test file houssainy@google.com 2014-10-15 15:01:11 +00:00
  • e93cbd13d5 Fix data races in ThreadTest.ThreeThreadsInvoke. pbos@webrtc.org 2014-10-15 14:54:56 +00:00
  • f87c0aff7f audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> bjornv@webrtc.org 2014-10-15 12:51:23 +00:00
  • f02ba9be54 audio_processing/agc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> bjornv@webrtc.org 2014-10-15 11:16:48 +00:00
  • 8dc00d76af audio_processing/ns: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> bjornv@webrtc.org 2014-10-15 09:31:40 +00:00
  • 99e561f6a6 Extend AcmSwitchingOutputFrequencyOldApi with more frequencies henrik.lundin@webrtc.org 2014-10-15 08:50:00 +00:00
  • 64f5611b3d Roll chromium_revision 2d714fa..de13cf4 (298667:299488) kjellander@webrtc.org 2014-10-15 05:59:42 +00:00
  • fab5439112 common_audio: Removed version API from signal_processing bjornv@webrtc.org 2014-10-15 04:38:42 +00:00
  • 81ddc78536 (Auto)update libjingle 77701902-> 77709729 buildbot@webrtc.org 2014-10-14 22:39:24 +00:00
  • 1ecbe45c7e (Auto)update libjingle 77689511-> 77696841 buildbot@webrtc.org 2014-10-14 20:29:28 +00:00
  • 43336b6b9f Remove unused (no-op) VideoOptions. pbos@webrtc.org 2014-10-14 19:12:06 +00:00
  • a4351a045d libjingle: use _stricmp instead of deprecated stricmp. henrike@webrtc.org 2014-10-14 17:07:41 +00:00
  • a73a678e25 Remove -1 from Call::Config::start_bitrate_bps. pbos@webrtc.org 2014-10-14 11:52:10 +00:00
  • eb24b04f16 Add periodic logging of received RTP headers and estimated clock offsets for e2e delay. stefan@webrtc.org 2014-10-14 11:40:13 +00:00
  • 81a78930ee New ACM test to trigger audio glitch when switching output sample rate henrik.lundin@webrtc.org 2014-10-14 10:49:58 +00:00
  • c216b9aeaf Add a packet loss full stack test to the new API. stefan@webrtc.org 2014-10-14 10:38:49 +00:00
  • a57678a70e Workarounds for a bug in VS2013.3 linker when PGO is turned on. kwiberg@webrtc.org 2014-10-14 09:40:04 +00:00
  • 7fe1e03dd6 Wire up external encoders. pbos@webrtc.org 2014-10-14 04:25:33 +00:00
  • f68cc0b0c3 (Auto)update libjingle 77554188-> 77629208 buildbot@webrtc.org 2014-10-14 01:17:42 +00:00
  • 82e6fa533c Move exlusion of VP9 integration tests for DrMemory from modules_unittests to modules_tests file. marpan@webrtc.org 2014-10-14 00:34:19 +00:00
  • b6af4283ca Adjust speech probability in NS when echo aluebs@webrtc.org 2014-10-13 20:48:05 +00:00
  • 1e6a5dd14e Removes xmllite from talk/xmllite since webrtc/xmllite is used instead. henrike@webrtc.org 2014-10-13 18:27:11 +00:00
  • 8bee130fa0 Disable VP9 integration tests on DrMemory. marpan@webrtc.org 2014-10-13 17:10:40 +00:00
  • bc1a4578e0 common_audio: Removed macro WEBRTC_SPL_RSHIFT_W16 bjornv@webrtc.org 2014-10-13 14:00:43 +00:00
  • a3722b643d iSAC tests: Type buffers as uint8_t[] to avoid casts kwiberg@webrtc.org 2014-10-13 13:29:04 +00:00
  • d4fe824862 audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >> bjornv@webrtc.org 2014-10-13 13:01:13 +00:00
  • 396a5e0001 WebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[] kwiberg@webrtc.org 2014-10-13 11:23:24 +00:00
  • 3f7f899a15 WebRtcIsac_UpdateBwEstimate et al.: Type byte streams as uint8, not uint16 kwiberg@webrtc.org 2014-10-13 11:07:06 +00:00
  • 1172988c79 Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[] kwiberg@webrtc.org 2014-10-13 10:53:42 +00:00
  • 3c16d8bd1c (Auto)update libjingle 77414393-> 77554188 buildbot@webrtc.org 2014-10-13 06:35:10 +00:00
  • c502df54f8 Merge the supporting to UYVY on Linux video capture in crbug/410202 to webrtc standalone. braveyao@webrtc.org 2014-10-13 02:13:00 +00:00
  • 651c05e4fc Release _inputSendPin & _outputCapturePin before _captureFilter & _sinkFilter since they should depend on the filters. The previous steps work fine for all the webcam, but have problem on SplitCam driver as in the issue report. Anyway it's always good to de-initial with the reversing order to initialization. braveyao@webrtc.org 2014-10-13 02:11:55 +00:00
  • 7f7b0a1cdd Re-enable ThreadCheckerDeathTest.MethodNotAllowedOnDifferentThreadInDebug (missed when enabling other base tests). henrike@webrtc.org 2014-10-10 21:41:55 +00:00
  • 4ddbbed16e Disable SendsAndReceivesVP9 test for now. marpan@webrtc.org 2014-10-10 21:25:20 +00:00
  • c87b74717b Adjust/increase rate control thresold for a vp9 test. marpan@webrtc.org 2014-10-10 17:55:57 +00:00
  • 573c78e31c Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. Passes trybots. marpan@webrtc.org 2014-10-10 16:44:47 +00:00
  • 3cefbc99f4 Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. This also marks all virtual overrides of other classes in the same files. xians@webrtc.org 2014-10-10 09:42:53 +00:00
  • afede835ee Cleanup scripts and suppressions for TSan v1 kjellander@webrtc.org 2014-10-10 09:18:34 +00:00
  • fae6bc4106 Remove talk_base from suppressions. pbos@webrtc.org 2014-10-10 08:45:03 +00:00
  • e46bc77e94 Reland 28629004: adding new AEC dump start interface for chrome. xians@webrtc.org 2014-10-10 08:36:56 +00:00
  • c5593ef1aa Workaround deps2git issue with inline Python in DEPS. kjellander@webrtc.org 2014-10-10 07:16:05 +00:00
  • c732a3e511 Re-enable allmost all base tests. henrike@webrtc.org 2014-10-09 22:08:15 +00:00
  • 4a73519690 Re-enables a bunch of base unittests part II. henrike@webrtc.org 2014-10-09 20:27:13 +00:00
  • dae40dcde9 Change setting VP8 codec specific info values by HW VP8 encoder to follow SW implementation. glaznev@webrtc.org 2014-10-09 17:53:09 +00:00
  • e30dab77df base/thread_unittest: wrap test was setting current thread to NULL. henrike@webrtc.org 2014-10-09 15:41:40 +00:00
  • 17f8ddd6c4 Make pbos and kjellander only owners of tsan2 suppressions. henrike@webrtc.org 2014-10-09 15:40:18 +00:00
  • 8768f161cd Fix comments in common_types.h henrik.lundin@webrtc.org 2014-10-09 12:58:45 +00:00
  • 3ff788cf73 Increase timeout for AsyncWriteTest.TestWrite. pbos@webrtc.org 2014-10-09 12:47:15 +00:00
  • 4bd2db9a55 Opus wrapper: Use const for inputs and uint8[] for byte streams kwiberg@webrtc.org 2014-10-09 11:21:10 +00:00
  • 1bada48401 Make DEPS find check_root_dir.py in legacy checkouts. kjellander@webrtc.org 2014-10-09 10:53:02 +00:00
  • 2c0cdbce22 Estimating NTP time with a given RTT. minyue@webrtc.org 2014-10-09 10:52:43 +00:00
  • c803907d87 Removing useless packets when inserting them (NetEq) minyue@webrtc.org 2014-10-09 10:49:54 +00:00
  • 0b0ac8236b Remove root_dir variable from DEPS + enforce rename. kjellander@webrtc.org 2014-10-09 09:11:27 +00:00