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  • 1faef7d084 Use codec width/height as the encoded_image width/height. wu@webrtc.org 2014-03-20 17:00:46 +00:00
  • 3ab57c514c Changing the buffer size (slots) to 1.5 seconds @ 30 ms packets henrik.lundin@webrtc.org 2014-03-20 15:09:38 +00:00
  • 979f1f8235 Add system wrapper dependency to libjingle targets. This is necessary to handle usage of STR_CASE_CMP in common_types.h ( as in https://webrtc-codereview.appspot.com/10099005/) mallinath@webrtc.org 2014-03-20 15:09:09 +00:00
  • 8a8c3ef2ae Add ability to configure cpu overuse options via an API. asapersson@webrtc.org 2014-03-20 13:15:01 +00:00
  • d66929995f Prevent playout delay wrap-around in VoiceEngine henrik.lundin@webrtc.org 2014-03-20 12:04:09 +00:00
  • 800b8dbda6 Removes error printout in voe_cmd_test which was caused by attempts to transmit RTCP packets even if a transport object was not registered. henrika@webrtc.org 2014-03-20 08:07:41 +00:00
  • c14807959b Extend perf tests to perform rampup on single stream. andresp@webrtc.org 2014-03-20 03:23:55 +00:00
  • c8ac17ca04 Adjust the captured window rect when the window is maximized. GetWindowRect includes the window frames for maximized window even they are off screen, causing content outside the window being captured falsely. The fix is to remove the left/right/bottom window frame from the captured rect. Mouse capturing is adjusted accordingly as well. jiayl@webrtc.org 2014-03-20 00:06:41 +00:00
  • ffe2620c97 (Auto)update libjingle 63352036-> 63363208 henrike@webrtc.org 2014-03-19 22:20:10 +00:00
  • 16395228f5 Properly account for retransmitted packets when not using the pacer. stefan@webrtc.org 2014-03-19 19:34:07 +00:00
  • 7c6ff2da26 Fixes RTX related bugs. stefan@webrtc.org 2014-03-19 18:14:52 +00:00
  • 9af85c4ac2 Disabling SendsSetSimulcastSsrcs. pbos@webrtc.org 2014-03-19 15:49:18 +00:00
  • 1e98a15adb Revert "Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets" henrik.lundin@webrtc.org 2014-03-19 14:16:52 +00:00
  • e5be877476 Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets henrik.lundin@webrtc.org 2014-03-19 13:36:58 +00:00
  • add4073593 Disable flaky CanSwitchToUseAllSsrcs. pbos@webrtc.org 2014-03-19 12:57:35 +00:00
  • 709e29742e Simplify pacer interface. pbos@webrtc.org 2014-03-19 10:59:52 +00:00
  • f577ae9eac Remove internal codecs from VideoSendStream. pbos@webrtc.org 2014-03-19 08:43:57 +00:00
  • ac4b87c258 Fix a deadlock in ViEEncoder::DeliverFrame. wuchengli@chromium.org 2014-03-19 03:44:20 +00:00
  • 8b61011b6f (Auto)update libjingle 63293120-> 63352036 henrike@webrtc.org 2014-03-18 21:39:10 +00:00
  • 08e2dd80a3 Exclude WebRtcVideoMediaChannelTest.AddRemoveCapturerMultipleSources for Valgrind on Mac kjellander@webrtc.org 2014-03-18 21:37:28 +00:00
  • 886c94f07c Adds a method to WindowCapturer to bring a window to the front. jiayl@webrtc.org 2014-03-18 17:10:36 +00:00
  • e9793ab8b8 (Auto)update libjingle 63111035-> 63293120 henrike@webrtc.org 2014-03-18 14:36:23 +00:00
  • dcc301be07 Adding thread annotations to NetEq4 henrik.lundin@webrtc.org 2014-03-18 11:49:22 +00:00
  • 3f655aa5f7 Add #include <cstdlib> for std::abs. pbos@webrtc.org 2014-03-18 11:10:11 +00:00
  • 944cbeb292 Resolves TSan v2 warnings in voe_auto_test. henrika@webrtc.org 2014-03-18 10:32:33 +00:00
  • ed8b281265 Re-comitting r5711: "Fixing a flaky test in video_engine_tests" henrik.lundin@webrtc.org 2014-03-18 08:43:29 +00:00
  • 12499ff20b Revert 5711 "Fixing a flaky test in video_engine_tests" turaj@webrtc.org 2014-03-18 00:23:55 +00:00
  • d0f0c76cd9 Fixing a flaky test in video_engine_tests henrik.lundin@webrtc.org 2014-03-17 21:09:10 +00:00
  • 4e69f782b0 Small refactor on send_side_bandwidth_estimation. andresp@webrtc.org 2014-03-17 17:07:48 +00:00
  • ccb33a67b9 turn-prober: enable running headlessly and only emit output on error. fischman@webrtc.org 2014-03-17 16:27:41 +00:00
  • a714eafb83 Refactor rampup tests: - Cleanup test done condition (should be the same but with less code). - Split up functions blocks inside methods that were large. andresp@webrtc.org 2014-03-17 15:34:57 +00:00
  • 44eb87e6dd Tool to establish a loopback call via apprtc turn server. andresp@webrtc.org 2014-03-17 14:23:22 +00:00
  • 26caf0eb53 Suppresses/disables tsan/memcheck issues due to sync of 63111035. henrike@webrtc.org 2014-03-14 22:47:07 +00:00
  • 18e5911d92 (Auto)update libjingle 63089643-> 63111035 henrike@webrtc.org 2014-03-14 17:19:38 +00:00
  • cf6f46d716 References to includes in third_party should be relative, not absolute. sprang@webrtc.org 2014-03-14 16:22:18 +00:00
  • 4375e1acbb Add support for YUV4MPEG file reading to tools files. (Minor fix). mcasas@webrtc.org 2014-03-14 12:51:02 +00:00
  • 6e2d012b58 Add support for YUV4MPEG file reading to tools files. mcasas@webrtc.org 2014-03-14 12:45:45 +00:00
  • 24779fe7cc Fix a bug where network freeze during CNG causes delay henrik.lundin@webrtc.org 2014-03-14 12:40:05 +00:00
  • 367000fa8d Remove legacy weirdness in Merge::Downsample henrik.lundin@webrtc.org 2014-03-14 12:28:39 +00:00
  • f45a55083f (Auto)update libjingle 63019975-> 63089643 henrike@webrtc.org 2014-03-13 18:51:34 +00:00
  • 54464e6f49 Stopping network threads before tearing down test henrik.lundin@webrtc.org 2014-03-13 15:39:27 +00:00
  • 5a320fb06f Race condition in RTPSender sprang@webrtc.org 2014-03-13 15:12:37 +00:00
  • 41689018a0 Add max delay to trace based filters and enhances drop tail queues with delay statistics. stefan@webrtc.org 2014-03-13 15:01:32 +00:00
  • b10363f3b6 Re-landing "Routing SuspendChange to VideoSendStream::Stats" henrik.lundin@webrtc.org 2014-03-13 13:31:21 +00:00
  • 3349ae0cdc Implement minimum transmit bitrate. pbos@webrtc.org 2014-03-13 12:52:27 +00:00
  • 6ea4f6397e Enable all RampUpTest.UpDownUp* tests henrik.lundin@webrtc.org 2014-03-13 09:21:26 +00:00
  • b5f3029302 Replace labs with std::abs. pbos@webrtc.org 2014-03-13 08:53:39 +00:00
  • 827faae0ec Fixing incorrect memset. mallinath@webrtc.org 2014-03-13 02:15:12 +00:00
  • dd5d804efb Disable all protobuf dependent targets when enable_protobuf=0. andrew@webrtc.org 2014-03-13 00:57:52 +00:00
  • c7bec8484b (Auto)update libjingle 62948689-> 63019975 henrike@webrtc.org 2014-03-12 19:53:43 +00:00
  • 9269ba14f0 (Git)ignore all of /net. Works around issue: gclient sync, git clean -df, gclient runhooks -> failure (regression in r4466). henrike@webrtc.org 2014-03-12 15:50:36 +00:00
  • c2313fb73c Enable VS2013 for Windows compilation by default. kjellander@webrtc.org 2014-03-12 14:41:58 +00:00
  • 95153cc4cd Remove platform-specific code from new-API tests. pbos@webrtc.org 2014-03-12 13:22:00 +00:00
  • ca8cb95364 Implement a test for an old corner-case in NetEq henrik.lundin@webrtc.org 2014-03-12 10:26:52 +00:00
  • 04ea23234a Developing NetEqImpl unit tests henrik.lundin@webrtc.org 2014-03-12 05:55:10 +00:00
  • 10bd88e2b5 (Auto)update libjingle 62871616-> 62948689 henrike@webrtc.org 2014-03-11 21:07:25 +00:00
  • 21df84711a Disable TestOpusNewACM on Android. andrew@webrtc.org 2014-03-11 20:40:59 +00:00
  • be39470203 Revert "Routing SuspendChange to VideoSendStream::Stats" henrik.lundin@webrtc.org 2014-03-11 17:13:14 +00:00
  • 12acd6ea8c Reorder includes in audio_processing_impl_unittest. andrew@webrtc.org 2014-03-11 16:55:14 +00:00
  • cdefc91ffc Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instead of _rtpRtcpModule now. braveyao@webrtc.org 2014-03-11 16:19:56 +00:00
  • 1598b80f52 Routing SuspendChange to VideoSendStream::Stats henrik.lundin@webrtc.org 2014-03-11 14:57:35 +00:00
  • c3d13d38f4 Classes and tests for audio an classifier. The class can be used to classify whether a frame of audio contains speech or music. The classifier uses the music/speech classifier in Opus. jan.skoglund@webrtc.org 2014-03-10 22:50:19 +00:00
  • a8b97373d5 Add tests and modify tools for new float deinterleaved interface. andrew@webrtc.org 2014-03-10 22:26:12 +00:00
  • 3046b843b2 Adding new data files for audio classifier unit testing on Android try bots jan.skoglund@webrtc.org 2014-03-10 20:52:46 +00:00
  • d3d6bce9ed (Auto)update libjingle 62865357-> 62871616 henrike@webrtc.org 2014-03-10 20:41:22 +00:00
  • d32797f853 Add a float interface to PushSincResampler. andrew@webrtc.org 2014-03-10 18:51:42 +00:00
  • bc206eadb8 iOS video_render: omit no-op setNeedsDisplay fischman@webrtc.org 2014-03-10 18:48:19 +00:00
  • f792d17870 AppRTCDemo(iOS): video support; part 1 of 2: webrtc/. fischman@webrtc.org 2014-03-10 17:12:08 +00:00
  • 0537634154 (Auto)update libjingle 62713454-> 62865357 henrike@webrtc.org 2014-03-10 15:53:12 +00:00
  • 4a47be0f52 Disable CallTest.ReceivesAndRetransmitsNack for TSan kjellander@webrtc.org 2014-03-10 12:50:29 +00:00
  • 36b6221cd4 Adding a link to issue henrik.lundin@webrtc.org 2014-03-10 10:24:35 +00:00
  • 6b0cbcba42 Roll chromium_revision 249215:255773 kjellander@webrtc.org 2014-03-10 09:51:17 +00:00
  • 9b5f4d8a84 Fix build breakage introduce with r5665. stefan@webrtc.org 2014-03-10 09:38:39 +00:00
  • f9e7c9d865 Add option to bwe_rtp_to_text to output arrival times only in nanoseconds. stefan@webrtc.org 2014-03-10 09:11:21 +00:00
  • a01daf0359 RTCPeerConnectionTest(objc): deflake by ignoring ICECompleted. fischman@webrtc.org 2014-03-08 03:17:55 +00:00
  • 13320ea3d3 PeerConnectionTest(objc): expect ICE Completed state post 61460797-p10 fischman@webrtc.org 2014-03-07 22:15:30 +00:00
  • 781146964e Roll libvpx 251850:254609 marpan@webrtc.org 2014-03-07 22:13:51 +00:00
  • 11aab0edc2 Populate VoiceReceiverInfo::delay_estimate_ms, jitter_buffer_ms, and jitter_buffer_preferred_ms to getStats. jiayl@webrtc.org 2014-03-07 18:56:26 +00:00
  • 64e0405552 Avoid crash in ViEEncoder::DeRegisterExternalEncoder(). fischman@webrtc.org 2014-03-07 18:00:05 +00:00
  • cc08e3f9b1 Moves WEBRTC_POSIX define from header file to gyp-settings. henrike@webrtc.org 2014-03-07 15:30:21 +00:00
  • 3ecc162d01 Remove std:: prefixes from C functions in webrtc/. pbos@webrtc.org 2014-03-07 15:23:34 +00:00
  • 371243dfa3 Remove std:: prefixes from C functions in talk/. pbos@webrtc.org 2014-03-07 15:22:04 +00:00
  • 46509c8d58 adding FEC support to WebRTC Opus wrapper and tests. minyue@webrtc.org 2014-03-07 11:49:11 +00:00
  • 04546884bf This CL is to add Opus complexity knob and to test it. minyue@webrtc.org 2014-03-07 08:55:48 +00:00
  • ebdb0e3ad0 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. wu@webrtc.org 2014-03-06 23:49:08 +00:00
  • 79047f99c1 (Auto)update libjingle 62691533-> 62713454 henrike@webrtc.org 2014-03-06 23:46:59 +00:00
  • 2d213e450c (Auto)update libjingle 62550414-> 62691533 henrike@webrtc.org 2014-03-06 18:51:21 +00:00
  • f714e7faea Remove abs() use in PseudoTcp::process. pbos@webrtc.org 2014-03-06 18:31:08 +00:00
  • 45846977f9 Fixes a bug in the simulation framework where the time offset is accumulating as the packet trace is repeated, causing increasingly large gaps with no packets being transmitted. stefan@webrtc.org 2014-03-06 15:46:46 +00:00
  • ed865b5d46 NetEq4: Changing the behavior of playout_timestamp_ update henrik.lundin@webrtc.org 2014-03-06 10:28:07 +00:00
  • 60ad5fdadf Potential deadlock in VideoSendStreamTest::ProducesStats sprang@webrtc.org 2014-03-06 10:03:36 +00:00
  • 998cb8fcd0 Use DISABLE_ instead of commenting out tests henrik.lundin@webrtc.org 2014-03-06 09:12:00 +00:00
  • 845862f279 Adding a new ramp-up-down-up test henrik.lundin@webrtc.org 2014-03-06 07:19:28 +00:00
  • a0d11da359 Remove upper check for number of cores in VCM, I didn't find any good reasons for checking this. mflodman@webrtc.org 2014-03-05 15:18:45 +00:00
  • cf85f1cf3c Reorganize libjingle path variables. kjellander@webrtc.org 2014-03-05 00:40:33 +00:00
  • 9f4d2125d7 adding sha1 files for audio classifier test jan.skoglund@webrtc.org 2014-03-05 00:27:24 +00:00
  • 3e0b60f465 Switch to correct interpretation of int and float input data in audio_processing_unittest bjornv@webrtc.org 2014-03-05 00:18:53 +00:00
  • 17e40641b3 Add a deinterleaved float interface to AudioProcessing. andrew@webrtc.org 2014-03-04 20:58:13 +00:00
  • b90991dade Update libjingle 62472237->62550414 henrike@webrtc.org 2014-03-04 19:54:57 +00:00
  • 7bd4a27502 VideoCaptureAndroid: don't deliver frames after stopCapture(). fischman@webrtc.org 2014-03-04 18:17:55 +00:00