Commit Graph

  • 838cdb3db6 Revert of Fix chromium-style warnings. (patchset #1 id:1 of https://codereview.webrtc.org/2400993002/ ) terelius 2016-10-24 09:38:22 -07:00
  • 5d79a7cb1f rtcstats_objects.h updated with TODOs about stats not being collected or not being collected correctly. hbos 2016-10-24 09:27:10 -07:00
  • a6f495c7c2 Simplifying audio network adaptor by moving receiver frame length range to ctor. minyue 2016-10-24 09:19:14 -07:00
  • a73f6c9726 NetEq now works with packets as values, rather than pointers. ossu 2016-10-24 08:25:28 -07:00
  • d312713e61 Roll chromium_revision 1362287708..9b5bb47fa0 (426760:426837) + roll Android SDK to N ehmaldonado 2016-10-24 07:53:33 -07:00
  • 86b92e05f9 Drop VP8 frames older than the last sync frame in the RtpFrameReferenceFinder. philipel 2016-10-24 07:11:53 -07:00
  • 1655e45d85 Elimiteted race condition in the AudioMixer. aleloi 2016-10-24 06:56:56 -07:00
  • 2206c959f1 Revert of Fix some chromium style warnings in remote_bitrate_estimator.h (patchset #1 id:1 of https://codereview.webrtc.org/2387113008/ ) terelius 2016-10-24 06:43:25 -07:00
  • 01404084aa Add tests and fix thread annotations danilchap 2016-10-24 06:07:25 -07:00
  • b60d1962d8 Eliminate left shift of negative value by using multiplication instead kwiberg 2016-10-24 04:18:46 -07:00
  • 2fa7c67675 RTCTransportStats[1] added, supporting all members. hbos 2016-10-24 04:00:05 -07:00
  • 5de3a7e556 Remove unused variable from delay based BWE. terelius 2016-10-24 03:43:23 -07:00
  • 509eadd554 Fix chromium-style warnings. terelius 2016-10-24 03:24:17 -07:00
  • c22bcf4f4b Fix some chromium style warnings in remote_bitrate_estimator.h terelius 2016-10-24 02:57:06 -07:00
  • d7ce668287 Roll chromium_revision f9e01d4887..1362287708 (426685:426760) buildbot 2016-10-24 02:16:46 -07:00
  • a3cac0514a GN: move webrtc/video/ targets from webrtc/BUILD.gn into webrtc/video/BUILD.gn kjellander 2016-10-24 01:52:39 -07:00
  • 43536c3d6a Implement framesEncoded stat in video send ssrc stats. sakal 2016-10-24 01:46:43 -07:00
  • 267527459b Remove cricket::VideoCodec with, height and framerate properties Since WebRtcVideoSendStream have reconfigures the send codec to match the incoming captured frames widht and height they have not been used. Framerate has just been set when parsing sdp to 60fps and not changed elsewhere. perkj 2016-10-24 01:21:16 -07:00
  • d3c40089fa Delete always-zero ByteBufferWriter::start_. nisse 2016-10-24 01:06:14 -07:00
  • 61c053e329 Reland of Delete webrtc::VideoFrame::CopyFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ ) nisse 2016-10-24 00:44:13 -07:00
  • ebf524007f Allow using Java classes that don't require JNI in Chromium. sakal 2016-10-24 00:28:01 -07:00
  • 66712b024f Revert of Add method cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource. (patchset #5 id:80001 of https://codereview.webrtc.org/2334683002/ ) nisse 2016-10-24 00:22:31 -07:00
  • 151572ba05 Delete unused class AudioSourceWithMixStatus. nisse 2016-10-24 00:11:51 -07:00
  • 25445d3d4b Integrate FlexfecReceiveStream with Call. brandtr 2016-10-23 23:37:14 -07:00
  • 764e364933 Several subcomponents inside APM copy render audio from the render side to the capture side using the same pattern. Currently this is done independently for the submodules. peah 2016-10-22 05:04:30 -07:00
  • 897497361e Added the missing ReadQueuedRenderData() call to the AECM bitexactness test peah 2016-10-22 04:00:29 -07:00
  • 12986c4534 Added the missing ReadQueuedRenderData() call to the gain controller bitexactness test peah 2016-10-22 02:38:32 -07:00
  • da38293e51 Added the missing ReadQueuedRenderData() call to the AEC bitexactness test peah 2016-10-22 01:08:37 -07:00
  • 894c400c61 Android VideoFileRenderer: Wait for posted frames in release() Magnus Jedvert 2016-10-21 15:05:01 +02:00
  • 8c63a82bf5 Add a placeholder stat for logging the estimated residual echo likelihood. ivoc 2016-10-21 04:10:03 -07:00
  • 3355f6d6f5 Avoids invalid copy of audio buffer to task queue. henrika 2016-10-21 12:45:25 +02:00
  • c4d2dc4e02 Delete DataLog abstraction, which was almost unused. nisse 2016-10-21 01:52:55 -07:00
  • dda1e60153 Reland of Delete unused file mediacommon.h. (patchset #1 id:1 of https://codereview.webrtc.org/2441493003/ ) nisse 2016-10-20 23:57:15 -07:00
  • bd163417d6 Roll chromium_revision 4c4977aa05..f9e01d4887 (426117:426685) buildbot 2016-10-20 19:51:36 -07:00
  • 84fbf9ee38 SUCCEEDED macro is misused zijiehe 2016-10-20 17:00:37 -07:00
  • bdb8df895a BringSelectedWindowToFront should bring the window to front instead of only focusing it zijiehe 2016-10-20 16:44:17 -07:00
  • 97abf244ca Use variadic templates instead of pump for RefCountedObject danilchap 2016-10-20 15:19:28 -07:00
  • 6c278491ad Move audio frame memory handling inside AudioMixer. aleloi 2016-10-20 14:24:39 -07:00
  • 920d30bc74 Replaced thread checker with race checker in AudioMixer. aleloi 2016-10-20 14:23:24 -07:00
  • 161a586b45 Fix some flaky tests by using longer timeout and/or fake clock. Also use const variables for timeout values. Honghai Zhang 2016-10-20 11:47:02 -07:00
  • b9eaeba71f Return nullptr from RTCCertificate::FromPEM on failure. jbroman 2016-10-20 10:27:21 -07:00
  • 58000a0c3d Move shared_desktop_frame.cc to webrtc/modules/desktop_capture:primitives Sergey Ulanov 2016-10-20 09:33:52 -07:00
  • 142f019d87 Append second nack list in same compound rtcp packet instead of replace danilchap 2016-10-20 08:22:42 -07:00
  • aed581a4f3 Made AudioReceiveStream a mixer participant. aleloi 2016-10-20 06:32:39 -07:00
  • 5f70d3b559 Fix org.mockito.Matchers deprecation warnings in DirectRTCClientTest. sakal 2016-10-20 05:49:10 -07:00
  • 201dfe90a7 Split audio mixer into interface and implementation. aleloi 2016-10-20 05:06:39 -07:00
  • 76648da8dc Add FlexfecReceiveStream. brandtr 2016-10-20 04:54:48 -07:00
  • 057b8d91a6 Remove all traces of Dr Memory. Henrik Kjellander 2016-10-20 13:17:14 +02:00
  • 69034df0b0 Make GN build screenshare_loopback palmkvist 2016-10-20 04:03:25 -07:00
  • 5a8724564c iOS: Optimize video scaling and cropping magjed 2016-10-20 03:34:29 -07:00
  • 7a973447eb Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. minyue 2016-10-20 03:27:12 -07:00
  • 1cb48232ac Android YuvConverter: Use OpenGL Framebuffer instead of EGL pixel buffer magjed 2016-10-20 03:19:16 -07:00
  • 9ab8a1884d Android: Extend functionality of EglRenderer magjed 2016-10-20 03:18:09 -07:00
  • ca20e7cfed Revert of Delete unused file mediacommon.h. (patchset #1 id:1 of https://codereview.webrtc.org/2437703002/ ) nisse 2016-10-20 02:52:58 -07:00
  • c1f8ecb661 Remove check for numberOfCameras from AppRTC Mobile PeerConnectionClient. sakal 2016-10-20 02:34:46 -07:00
  • be4aff76a6 Suppress deprecation warning in CallFragment. sakal 2016-10-20 02:32:47 -07:00
  • aff9ff0259 Create .git-blame-ignore-revs and add Java format CL to it. sakal 2016-10-20 02:20:34 -07:00
  • e33c5d918a Added a level controller initialization value to MediaConstraints. aleloi 2016-10-20 01:53:27 -07:00
  • 647915fb95 Add loopback option and improve UX of AppRTCMobile for Mac. denicija 2016-10-20 01:46:41 -07:00
  • 725e2121ad Prevent stripping of C interfaces in framework kthelgason 2016-10-20 01:31:42 -07:00
  • e037060161 Add to rtc::Optional equality/unequality comparisions with object danilchap 2016-10-20 00:58:15 -07:00
  • a34e796a1b Delete unused file mediacommon.h. nisse 2016-10-20 00:44:56 -07:00
  • 55928fef1e QualityScaler reset bugfix kthelgason 2016-10-20 00:42:56 -07:00
  • 0489e498eb Change RefCountedObject to use perfect forwarding. The main reason for doing this is to allow refcounted objects to accept rvalue references in ctor and be able to std::move ctor rvalue arguments. Also, refcounted.h is now generated using pump.py instead of manually creating each ctor version. perkj 2016-10-20 00:24:01 -07:00
  • 79f0bf3ab5 A variable in ScreenCapturerWinDirectx has a bad name zijiehe 2016-10-19 23:40:36 -07:00
  • f04f14e772 Revert of Move bitstream parser to more appropriate directory. (patchset #4 id:60001 of https://codereview.webrtc.org/2370853005/ ) kthelgason 2016-10-19 10:34:34 -07:00
  • cc6817e9ce Move current bitstream parser to more appropriate directory. kthelgason 2016-10-19 09:31:09 -07:00
  • 577bc19210 Android: Move YuvConverter to its own file Magnus Jedvert 2016-10-19 15:29:02 +02:00
  • b6f1fb5337 Delete RTPSender::BuildRtpHeader function and all dependencies danilchap 2016-10-19 06:11:39 -07:00
  • 061ea0df03 Remove VideoCodec resolution validation. This is needed to be able to land https://codereview.webrtc.org/2408153002/ "Remove cricket::VideoCodec with, height and framerate properties" without breaking upstream projects. Per 2016-10-19 13:57:59 +02:00
  • e3e411a1aa Removed perkj@ from video WATCHLIST Per 2016-10-19 13:52:08 +02:00
  • 73c5d4a083 Include ScreenCapturerAndroid in libjingle_peerconnection_java.jar sakal 2016-10-19 02:46:19 -07:00
  • 09347858f7 Reland of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2402853002/ ) nisse 2016-10-19 00:30:30 -07:00
  • 4e52386339 Reland of Add path for recovered packets from internal::Call to RtpStreamReceiver. (patchset #1 id:1 of https://codereview.webrtc.org/2427733002/ ) brandtr 2016-10-18 23:50:45 -07:00
  • 249beee124 Remove DesktopRegion parameter from DesktopCapturer::Capture zijiehe 2016-10-18 23:13:29 -07:00
  • e8295fcc48 Roll chromium_revision eb9b71b64b..4c4977aa05 (426008:426117) buildbot 2016-10-18 19:57:14 -07:00
  • 6a4607e100 Deflaky ScreenCapturerTest zijiehe 2016-10-18 18:22:18 -07:00
  • 1eb12934e7 Handle BW drop in ALR region and initiate probing Irfan Sheriff 2016-10-18 17:04:25 -07:00
  • a9c7cfac41 Prepare for introduction of rtc::PacketTransportInterface. johan 2016-10-18 15:38:38 -07:00
  • 1203066236 Compilerwarning possible loss of data in file port.h bertholdherrmann08 2016-10-18 14:00:02 -07:00
  • cc555c5019 RTCDataChannelStats[1] added, supporting all stats members. hbos 2016-10-18 12:48:31 -07:00
  • 1394c7b594 Fix for flaky test: EndToEndTest.VerifyHistogramStatsWithRtx asapersson 2016-10-18 11:50:50 -07:00
  • 0b7be9c272 Roll chromium_revision c8b7ee41e0..eb9b71b64b (425645:426008) buildbot 2016-10-18 11:45:17 -07:00
  • 9960bb1469 Call OnTransportFeedback just when feedback_observer exist. michaelt 2016-10-18 09:40:34 -07:00
  • 53fe19d6f3 Set min and max rate on caller and on callee side. michaelt 2016-10-18 09:39:22 -07:00
  • 64e1a32e2f Second try to get "Support for video file instead of camera and output video out to file" accepted mandermo 2016-10-18 08:47:51 -07:00
  • 67a8c986ab Revert of Support for video file instead of camera and output video out to file (patchset #17 id:320001 of https://codereview.webrtc.org/2273573003/ ) kjellander 2016-10-18 06:05:35 -07:00
  • f33970b15e Add unittest for I420Buffer::Rotate. nisse 2016-10-18 06:01:34 -07:00
  • 6ed592d0ab Rename variables to reflect that DelayBasedBwe lives on the send side rather than receive side. terelius 2016-10-18 05:55:30 -07:00
  • 5588a13fe7 Now uses rtc::Buffer in AudioDeviceBuffer. henrika 2016-10-18 05:14:30 -07:00
  • 44666997ca Support for video file instead of camera and output video out to file mandermo 2016-10-18 04:52:00 -07:00
  • 9e83c97e9f Add rtc::Optional::emplace danilchap 2016-10-18 04:07:18 -07:00
  • 7a3776102f Removed RTPHeader from NetEq's Packet struct. ossu 2016-10-18 04:06:13 -07:00
  • 553024ab34 During a fix of an unrelated issue, a bug was introduced in the rtp analyzer tool: when the number of data points was divisible by RTPStatitstics.PLOT_RESOLUTION_MS (which is 50), pyplot.plot was called with arrays of different lengths. One of the arrays could be one element larger. aleloi 2016-10-18 01:44:44 -07:00
  • e405d9b8df Add a fuzzer for FlexfecReceiver. brandtr 2016-10-18 01:18:04 -07:00
  • e6b58291a0 Extends how AppRTCMobile handles audio focus on Android henrika 2016-10-18 10:17:49 +02:00
  • 91718a1629 Roll script: Update after SVN support was dropped from depot tools kjellander 2016-10-18 01:14:56 -07:00
  • 5c639895fb Import build/config/clang/clang.gni in webrtc/base/BUILD.gn ehmaldonado 2016-10-17 12:28:44 -07:00
  • 862d74d017 Revert of Add path for recovered packets from internal::Call to RtpStreamReceiver. (patchset #2 id:60001 of https://codereview.webrtc.org/2390823009/ ) honghaiz 2016-10-17 09:42:31 -07:00
  • c4fd23cdda Add rtc::Optional::reset danilchap 2016-10-17 07:16:54 -07:00