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  • 0224c20fa6 * Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter. * Re-enable test CaptureNtpTimeWithNetworkJitter. * Use 100ms as the threadhold as a FYI since this is a performance test. wu@webrtc.org 2014-05-05 17:42:43 +00:00
  • 4220434d37 Implement the Windows screen capturer using the Magnification API. The original ScreenCapturerWin is renamed ScreenCapturerWinGdi. jiayl@webrtc.org 2014-05-05 16:08:47 +00:00
  • 7dccce3948 Revert 6048 "Implement the Windows screen capturer using the Mag..." tina.legrand@webrtc.org 2014-05-05 11:17:26 +00:00
  • 633aff6bd0 WebRTCDemo: correct set trace filter operation. braveyao@webrtc.org 2014-05-05 04:24:47 +00:00
  • 9f453b1a1b Add ALLOW_UNUSED and update COMPILE_ASSERT to Chromium's latest. andrew@webrtc.org 2014-05-04 03:04:26 +00:00
  • 41451d4e55 (Auto)update libjingle 66106643-> 66138442 buildbot@webrtc.org 2014-05-03 05:39:45 +00:00
  • b235c56017 Implement the Windows screen capturer using the Magnification API. The original ScreenCapturerWin is renamed ScreenCapturerWinGdi. jiayl@webrtc.org 2014-05-03 00:16:29 +00:00
  • cdaf2b9b73 Add svn:ignore to resources kjellander@webrtc.org 2014-05-02 19:40:15 +00:00
  • cc06c75f28 (Auto)update libjingle 66100938-> 66106643 buildbot@webrtc.org 2014-05-02 18:51:11 +00:00
  • 13d6776c46 (Auto)update libjingle 66098243-> 66100938 buildbot@webrtc.org 2014-05-02 17:33:29 +00:00
  • 0d34f1446a (Auto)update libjingle 66033941-> 66098243 buildbot@webrtc.org 2014-05-02 16:54:25 +00:00
  • 7f3a041d23 Removed NetworkTest.CanSwitchToExternalTransport since it tests an unsupported case and we should not maintain such a test. henrika@webrtc.org 2014-05-02 13:59:58 +00:00
  • 9205c87820 Pointers were not dereferenced in GetRtpStatistics. asapersson@webrtc.org 2014-05-02 13:24:42 +00:00
  • 24bd364d3e Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels. stefan@webrtc.org 2014-05-02 12:35:37 +00:00
  • e3a628997f Roll libvpx 264320:267596 fischman@webrtc.org 2014-05-02 00:04:20 +00:00
  • 14ea7e8922 AppRTCDemo(android): added a Heads-Up Display for bandwidth estimation. - tap display to toggle visibility - increased getStats frequency to 1hz. fischman@webrtc.org 2014-05-01 20:57:55 +00:00
  • dd92feb6dd AppRTCDemo(android): send the created SDP, not the local description after setting it fischman@webrtc.org 2014-05-01 19:06:18 +00:00
  • 560dce5d48 Pull openmax into third_party. turaj@webrtc.org 2014-05-01 18:53:51 +00:00
  • 9c16c39e61 Sets the SCTP port codec in the native SessionDescription. Previously it's only set when a SDP string is parsed into SessionDescription, causing failuring for native client. jiayl@webrtc.org 2014-05-01 18:30:30 +00:00
  • 53d82350c5 Ignore identical remote fingerprint in DtlsTransportChannelWrapper::SetRemoteFingerprint. Trying to set the same remote fingerprint could happen during renegotiation and should not fail. jiayl@webrtc.org 2014-05-01 00:00:19 +00:00
  • e44a84d851 Only clamp to 16 kHz when AECM is enabled. andrew@webrtc.org 2014-04-30 18:58:23 +00:00
  • ff2733204d Implement ObjC DataChannel wrapper tkchin@webrtc.org 2014-04-30 18:32:33 +00:00
  • 65f933899b Fix constness of AudioBuffer accessors. andrew@webrtc.org 2014-04-30 16:44:13 +00:00
  • 740e6b339a (Auto)update libjingle 65843899-> 65880186 buildbot@webrtc.org 2014-04-30 15:33:45 +00:00
  • 7c82adae61 AppRTCDemo was blocking the main thread for network requests. This fixes it by making the background queue serial instead of using @synchronize to make the background operations serial. fischman@webrtc.org 2014-04-30 00:17:47 +00:00
  • 9bd49becc1 Fix a data race in ACM1 when audio is pulled. turaj@webrtc.org 2014-04-29 20:27:45 +00:00
  • a86c42c424 libjingle_unittest now compiles and passes on iOS! (reland of r5986) fischman@webrtc.org 2014-04-29 18:37:29 +00:00
  • 681f787cc4 (Auto)update libjingle 65752960-> 65813736 buildbot@webrtc.org 2014-04-29 17:55:26 +00:00
  • f2aafe4355 Added include of assert.h for files calling assert but missing the include. henrike@webrtc.org 2014-04-29 17:54:17 +00:00
  • f04a6ea733 MediaCodecVideoEncoder: limit MediaCodec bitrate to 95% of requested to avoid overshoot. fischman@webrtc.org 2014-04-29 17:53:30 +00:00
  • 82d3cb68cd Made common_types.h PacketTime declaration match https://code.google.com/p/webrtc/source/browse/trunk/talk/base/asyncpacketsocket.h#65 henrike@webrtc.org 2014-04-29 17:50:47 +00:00
  • ceffdbc371 Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord() henrike@webrtc.org 2014-04-29 17:11:07 +00:00
  • 0300939484 Disable failing GoogleWifiTrace3Mbps. pbos@webrtc.org 2014-04-29 15:25:59 +00:00
  • 9353e6bc55 Disable GoogleWifiTrace3Mbps. pbos@webrtc.org 2014-04-29 14:49:56 +00:00
  • dfe2a1c995 Adding BweFeedbackTest which tracks BWE performance over a set of simulated scenarios. stefan@webrtc.org 2014-04-29 14:21:42 +00:00
  • f223746521 Upping start bitrate to min, if set to a lower value i SetSendCodec. mflodman@webrtc.org 2014-04-29 12:38:42 +00:00
  • f5433753b8 Suppress DrMemory allocator mismatch errors. pbos@webrtc.org 2014-04-29 12:24:45 +00:00
  • 97e67cb476 Fix iOS assembly compile error. kjellander@webrtc.org 2014-04-29 10:25:30 +00:00
  • 060b84b3bb Remove neteq_unittests from Android builds henrik.lundin@webrtc.org 2014-04-29 09:49:51 +00:00
  • 59343ee3d8 Roll chromium_revision 260462:266514 kjellander@webrtc.org 2014-04-29 09:36:40 +00:00
  • acf15dc90f Remove Version method from ACM1 henrik.lundin@webrtc.org 2014-04-29 09:25:21 +00:00
  • 70e53fa34d Remove ACM1 and NetEq3 related targets from modules.gyp henrik.lundin@webrtc.org 2014-04-29 08:58:46 +00:00
  • fdf2053787 Remove AudioCodingModuleFactory henrik.lundin@webrtc.org 2014-04-29 08:22:14 +00:00
  • 0bc9b5a5a7 Add clock to ACM config struct henrik.lundin@webrtc.org 2014-04-29 08:09:31 +00:00
  • 059488f2ea AEC: Startup phase only runs if reported_delay_enabled bjornv@webrtc.org 2014-04-29 08:09:26 +00:00
  • af6640fce7 (Auto)update libjingle 65729829-> 65752960 buildbot@webrtc.org 2014-04-28 21:31:51 +00:00
  • 874920f22e Disable WebRtcSpl_ScaleAndAddVectorsWithRoundNeon due to crash. fischman@webrtc.org 2014-04-28 20:09:09 +00:00
  • 82a045aae0 APM: limit native sample rate to 16kHz on mobile. fischman@webrtc.org 2014-04-28 17:26:32 +00:00
  • f27fdeb9c9 AppRTCDemo(android): don't initialize process-globals more than once. fischman@webrtc.org 2014-04-28 16:32:38 +00:00
  • 497ff21fad Using realpath instead of android_src in Android webview michaelbai@google.com 2014-04-28 15:46:46 +00:00
  • 1152fe2247 Only download the VS toolchain if DEPOT_TOOLS_WIN_TOOLCHAIN=1. andrew@webrtc.org 2014-04-28 15:46:32 +00:00
  • de1429e9ad Add thread annotations to Call API. pbos@webrtc.org 2014-04-28 13:00:21 +00:00
  • 7d825e9b2c Revert "libjingle_unittest now compiles and passes on iOS!" kjellander@webrtc.org 2014-04-28 12:50:47 +00:00
  • c891577e6d Disable flaky CaptureNtpTimeWithNetworkJitter. pbos@webrtc.org 2014-04-28 11:57:32 +00:00
  • 494aa0e93d AEC: Moved delay buffer size enums from aec_core.h to aec_core_internal.h bjornv@webrtc.org 2014-04-28 11:42:27 +00:00
  • 8dfe8ff590 Disable capture test for FrameRate on Windows. pbos@webrtc.org 2014-04-28 11:27:36 +00:00
  • e772c71743 Introduce a config struct for AudioCoding module henrik.lundin@webrtc.org 2014-04-28 10:16:57 +00:00
  • f043f79711 Disabling flaky CanReceiveFec. pbos@webrtc.org 2014-04-28 09:00:50 +00:00
  • 69e9950469 Disable flaky RunsRtpRtcpTestWIthoutErrors. pbos@webrtc.org 2014-04-28 08:49:07 +00:00
  • 12a34247a4 Fix the NetEq build henrik.lundin@webrtc.org 2014-04-28 08:36:35 +00:00
  • 116ed1d4f0 Include buffer size limits in NetEq config struct henrik.lundin@webrtc.org 2014-04-28 08:20:04 +00:00
  • b08bbf57a6 Add henrik.lundin as owner in AudioCoding module henrik.lundin@webrtc.org 2014-04-28 08:15:35 +00:00
  • a0d3067575 Use CreatePeerConnection method which accepts port_allocator. mallinath@webrtc.org 2014-04-26 00:00:15 +00:00
  • 95cd1551f8 libjingle_unittest now compiles and passes on iOS! fischman@webrtc.org 2014-04-25 23:59:56 +00:00
  • 8f69330310 Replace scoped_array<T> with scoped_ptr<T[]>. andrew@webrtc.org 2014-04-25 23:10:28 +00:00
  • 658a94595d (Auto)update libjingle 65619249-> 65622932 buildbot@webrtc.org 2014-04-25 22:03:30 +00:00
  • ff90ed6e96 (Auto)update libjingle 65561104-> 65619249 buildbot@webrtc.org 2014-04-25 21:12:10 +00:00
  • 2eceb8ef46 Roll third_party/opus 258909:262302 andrew@webrtc.org 2014-04-25 15:29:33 +00:00
  • 0175d76c72 Fix leak in remote bitrate estimator tests introduced in r5980 stefan@webrtc.org 2014-04-25 11:38:57 +00:00
  • 4f616a02bd Support for simulating multiple independent flows in a network. stefan@webrtc.org 2014-04-25 10:59:24 +00:00
  • 46106f2a05 Casting char to int in logs. asapersson@webrtc.org 2014-04-25 07:02:52 +00:00
  • 2b93402e36 (Auto)update libjingle 65484212-> 65561104 buildbot@webrtc.org 2014-04-25 00:18:27 +00:00
  • cc1ba15fe7 Returns a NULL frame on all platforms if the captured window is closed. Part of the fix for crbug/360181. On Mac/Linux, it previously continues capturing even if the window is closed. Now it stops by returning a NULL frame. On Windows, it used to stop capturing when the window is minimized. Now fixed to match other platforms. Note: the crbug still needs a chrome side fix to close the notification bar. This fix only stops the stream (i.e. stream onended event fired). jiayl@webrtc.org 2014-04-24 23:45:56 +00:00
  • cd70119a10 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. wu@webrtc.org 2014-04-24 22:10:24 +00:00
  • 93fd25c20c * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. * Cast rtp header extension to int in log in rtp_utility.cc. wu@webrtc.org 2014-04-24 20:33:08 +00:00
  • 439a4c49f9 Add an output capacity parameter to ACMResampler::Resample10Msec() henrik.lundin@webrtc.org 2014-04-24 19:05:33 +00:00
  • 103657b484 Add keyboard channel support to AudioBuffer. andrew@webrtc.org 2014-04-24 18:28:56 +00:00
  • d57b8149c2 Fix the Android compilation (better structure for NetEq test libs) henrik.lundin@webrtc.org 2014-04-24 13:19:04 +00:00
  • 5ca6a5387e Remove TraceCallback use from Call. pbos@webrtc.org 2014-04-24 11:35:33 +00:00
  • a5c8d2c9b3 Rename Start/Stop in Video{Send,Receive}Streams. pbos@webrtc.org 2014-04-24 11:13:21 +00:00
  • 0a2277448e Fixing a bug in ACM2 where the output frame energy was incorrectly set henrik.lundin@webrtc.org 2014-04-24 08:11:39 +00:00
  • f26c9e8369 Use unique filenames in AudioProcessingTests for parallelization. andrew@webrtc.org 2014-04-24 03:46:46 +00:00
  • 3f1aa24078 (Auto)update libjingle 65469804-> 65484212 buildbot@webrtc.org 2014-04-24 00:00:12 +00:00
  • 0d915ff603 Fix the return value of DtlsTransportChannelWrapper::SendPacket in the case of invalid RTP packet. jiayl@webrtc.org 2014-04-23 20:46:29 +00:00
  • e9d3760d5c AEC: Adds a reported_delay_enabled_ flag bjornv@webrtc.org 2014-04-23 13:20:07 +00:00
  • 26e2b687fc Remove ACM1/ACM2 switching from VoiceEngine tests henrik.lundin@webrtc.org 2014-04-23 08:39:41 +00:00
  • db144429b2 Exclude the new AudioProcessingTest from some sanitizer bots. andrew@webrtc.org 2014-04-23 04:16:24 +00:00
  • 46b31b17df Restore sample_rate_hz() until Chromium is updated to not use it. andrew@webrtc.org 2014-04-23 03:33:54 +00:00
  • 504fc89f36 (Auto)update libjingle 65394435-> 65417850 buildbot@webrtc.org 2014-04-23 03:23:19 +00:00
  • 19b1be159e Provide GetStats method in RTCPeerConnection tkchin@webrtc.org 2014-04-22 21:05:38 +00:00
  • ddbb8a2c24 Support arbitrary input/output rates and downmixing in AudioProcessing. andrew@webrtc.org 2014-04-22 21:00:04 +00:00
  • 34fe0153b9 Reland "Stop using ACM factory in VoiceEngine" henrik.lundin@webrtc.org 2014-04-22 19:04:34 +00:00
  • d59359af4d Remove 44.1 kHz workaround from the iOS AudioDevice. andrew@webrtc.org 2014-04-22 18:07:49 +00:00
  • 20c71fd1dc Fix a bug in AcmReceiver::NetworkStatistics henrik.lundin@webrtc.org 2014-04-22 10:11:21 +00:00
  • 0c108d0b4d Revert "Stop using ACM factory in VoiceEngine" henrik.lundin@webrtc.org 2014-04-22 09:44:00 +00:00
  • 139706ec0b Stop using ACM factory in VoiceEngine henrik.lundin@webrtc.org 2014-04-22 08:51:21 +00:00
  • d144bb6812 Let A/V sync test use default AudioCoding module henrik.lundin@webrtc.org 2014-04-22 08:36:33 +00:00
  • 0c1444c748 Create ACM2 instance when calling AudioCodingModule::Create henrik.lundin@webrtc.org 2014-04-22 08:18:42 +00:00
  • 0a035c8924 Disable tests in common_video_unittests for Dr Memory. kjellander@webrtc.org 2014-04-22 08:07:03 +00:00
  • 372ae83228 Reland "Make VoiceEngine choose ACM2 by default"" henrik.lundin@webrtc.org 2014-04-22 07:21:03 +00:00