admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Commit Graph

  • fffc1e5578 Add functionality for parsing H264 profile-level-id magjed 2016-10-31 05:55:57 -07:00
  • f0a7c5ac16 Delete deprecated method VideoFrame::CreateFrame. nisse 2016-10-31 05:48:07 -07:00
  • 626bc952aa Reland of "Separating video settings in VideoQualityTest". minyue 2016-10-31 05:47:02 -07:00
  • 869e7cd8e7 Rename ProducerFec to UlpfecGenerator. brandtr 2016-10-31 05:27:07 -07:00
  • d55c3f68c8 Rename FecReceiver to UlpfecReceiver. brandtr 2016-10-31 04:51:33 -07:00
  • 6b825df37e Using AudioOption to enable audio network adaptor. minyue 2016-10-31 04:08:32 -07:00
  • 4ee7046998 Add unit tests for bandwidth limited resolution stats in SendStatisticsProxy. asapersson 2016-10-31 04:05:12 -07:00
  • 535830ec2d Rename Fec to Ulpfec in EndToEndTests. brandtr 2016-10-31 03:45:58 -07:00
  • ca27f9d5b9 It seems that if encoder_params.sSpatialLayers[0].sSliceArgument.uiSliceNum is configured to number of cores as determined by openh264 (or any number > 1 in my local tests), frame rate statistics will be mucked up (apparently thousands of frames per second) and quality will bottom out because bits per frame is then very low. sprang 2016-10-31 03:43:39 -07:00
  • e602f0ab08 Rename Fec to Ulpfec in VideoSendStreamTest. brandtr 2016-10-31 03:40:49 -07:00
  • 42ca68ab72 Ensure one does not register same rtp header extension with different id Added assert to RtpHeaderExtensionMap Altered tests that did. danilchap 2016-10-31 03:34:40 -07:00
  • 051f678808 Add a NeededFrequency() method to the AudioMixer::Source interface. aleloi 2016-10-31 03:26:40 -07:00
  • 9aa78832f9 Revert of "Separating video settings in VideoQualityTest". (patchset #4 id:60001 of https://codereview.webrtc.org/2314403007/ ) minyue 2016-10-31 03:23:01 -07:00
  • bc8074474d Eliminate left shift of negative value by using multiplication instead kwiberg 2016-10-31 02:26:10 -07:00
  • c5b435dcec Re-enable PostDelayed test for TaskQueue on Windows. The requirements have been relaxed a little bit which should allow the test to pass on our VMs that run the tests. tommi 2016-10-31 02:17:11 -07:00
  • a144be384f Delete videorendererfactory.h and cricket::GdiVideoRenderer. nisse 2016-10-31 00:59:47 -07:00
  • 18b8774792 Setting PATH so that the 'plistbuddy' utility can be found, in a typical OS X environment VladimirTechMan 2016-10-30 23:09:44 -07:00
  • 21d45d2ab6 Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known. Per 2016-10-30 21:37:57 +01:00
  • b68d655f36 Add DesktopCapturer GetSourceList SelectSource FocusOnSelectedSource functions zijiehe 2016-10-28 17:35:11 -07:00
  • fcab7d62d5 Revert of Add DesktopCapturer GetSourceList SelectSource FocusOnSelectedSource functions (patchset #3 id:120001 of https://codereview.chromium.org/2452263003/ ) zijiehe 2016-10-28 17:14:08 -07:00
  • 9cb0b3b4ac Add DesktopCapturer GetSourceList SelectSource FocusOnSelectedSource functions zijiehe 2016-10-28 17:06:50 -07:00
  • 05a55b500d Revert of Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known. (patchset #2 id:20001 of https://codereview.webrtc.org/2455963004/ ) emircan 2016-10-28 14:06:29 -07:00
  • e7fc7d5c02 Fixing flaky DtmfSenderTest by using fake clock. deadbeef 2016-10-28 13:53:08 -07:00
  • 3e9a537601 Original CL: https://codereview.webrtc.org/2433153003/, commit 8b8d3e4c30e8ea3846b58dfd36d1fd35a7799df4. Revert CL: https://codereview.webrtc.org/2456333002/, commit 48dfab5c58119a4e65c52506ed55f8de79725bcf. ivoc 2016-10-28 07:55:33 -07:00
  • 1e45cc6ee0 Replace WebRtcVideoEncoderFactory::VideoCodec with cricket::VideoCodec magjed 2016-10-28 07:43:45 -07:00
  • e2a0177255 Style cleanups in rtp header extension traits: renamed kName to kUri and make it more const. remove IsSupportedBy to reduce header dependency. danilchap 2016-10-28 07:08:58 -07:00
  • af27ed01d7 Add algorithm for Residual Echo Detector. ivoc 2016-10-28 07:04:03 -07:00
  • 5f1b05129e Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known. perkj 2016-10-28 06:58:38 -07:00
  • c4b9b9419e Revert of Replace FileWrapper with File (in audio_device) (patchset #3 id:40001 of https://codereview.webrtc.org/2386963003/ ) terelius 2016-10-28 06:51:59 -07:00
  • 6ceab08322 GN: New conventions, default target and refactorings kjellander 2016-10-28 05:44:03 -07:00
  • 9f4a4a096b Add empty residual echo detector. ivoc 2016-10-28 05:39:16 -07:00
  • a9a1ac2ab8 Moved rtc::Base64 to base approved. philipel 2016-10-28 14:23:14 +02:00
  • 02ba211a9f Move RTCStatsCollector helper functions to anonymous namespace. hbos 2016-10-28 05:14:53 -07:00
  • f005a00094 Added calling of the stream_analog_level api in audioproc_f peah 2016-10-28 05:02:34 -07:00
  • 6d6762cb0e Add UINavigationController and settings bar button to AppRTCMobile. denicija 2016-10-28 04:53:16 -07:00
  • 5da65f241c Revert of Change ViEEncoder to not reconfigure the encoder until the video resolution is known. (patchset #4 id:60001 of https://codereview.webrtc.org/2455063002/ ) perkj 2016-10-28 04:48:47 -07:00
  • 48dfab5c58 Revert of New statistics interface for APM (patchset #11 id:200001 of https://codereview.webrtc.org/2433153003/ ) ivoc 2016-10-28 03:29:31 -07:00
  • 461c29e436 Change ViEEncoder to not reconfigure the encoder until the video resolution is known. perkj 2016-10-28 03:13:50 -07:00
  • 135259ac8f In order to be able to analyze the AGC behavior on aecdump recordings in an efficient manner, it is important to be able to use a standardized analysis script. For this to be feasible, data log points should be present. peah 2016-10-28 03:12:11 -07:00
  • 04055e95bf Removes all uses of FileWrapper in audio_device. palmkvist 2016-10-28 02:08:23 -07:00
  • 8b8d3e4c30 New statistics interface for APM ivoc 2016-10-28 01:32:20 -07:00
  • 37abf53116 Delete FrameObject::size member. nisse 2016-10-28 00:37:29 -07:00
  • 11f72b1945 Fix compile error for non Intel platforms Gordana.Cmiljanovic 2016-10-27 23:44:09 -07:00
  • 9922016ee4 Fix "IsLoopbackIp" to cover all loopback addresses; not just 127.0.0.1. deadbeef 2016-10-27 18:30:23 -07:00
  • 6be0a657c5 Move ScreenCapturer 'real' tests out of screen_capturer_unittest.cc. zijiehe 2016-10-27 16:50:35 -07:00
  • 32bcaf61f5 Improve RTC_DCHECK_op so that it won't trigger useless compiler warnings kwiberg 2016-10-27 13:35:59 -07:00
  • a73df559d6 Do not rely on specific ordering on generated candidates in TestGetAllPortsPortRange honghaiz 2016-10-27 11:44:17 -07:00
  • 45d18eb034 Re-enable the PostDelayed TaskQueue test on all platforms except Windows. tommi 2016-10-27 09:04:17 -07:00
  • 492ee28b73 Use bayesian estimate of acked bitrate. Stefan Holmer 2016-10-27 17:19:20 +02:00
  • 9890a5861f Testing of FileVideoCapturer. mandermo 2016-10-27 07:26:36 -07:00
  • da35f3e3ce Delete unused features of rtc::FilesystemInterface and related classes. nisse 2016-10-27 04:44:39 -07:00
  • a101e560b9 Remove LOGGING=1 define. kjellander 2016-10-27 02:34:31 -07:00
  • fe90b4176c Improves audio logs of native audio layers on Android henrika 2016-10-27 01:42:16 -07:00
  • 68e6bdd970 Remove use of VoECodec in video/call tests. solenberg 2016-10-27 00:23:06 -07:00
  • 5e49c2f09e Restore symbol level for Android builds. kjellander 2016-10-27 00:00:14 -07:00
  • bc59b061a7 Add gn_isolate_map.pyl file for WebRTC stand-alone tests kjellander 2016-10-26 23:55:24 -07:00
  • b1125685a0 Roll chromium_revision 9b5bb47fa0..04e7c673d9 (426837:427632) kjellander 2016-10-26 13:38:03 -07:00
  • e183121657 Enable clang style plugin in webrtc/modules/desktop_capture sergeyu 2016-10-26 13:15:42 -07:00
  • 54b0acb432 Change destruction order to fix potential invalid pointer dereference. erikchen 2016-10-26 11:10:24 -07:00
  • 489c0d4832 Decrease threshold for key frame generation. glaznev 2016-10-26 10:52:56 -07:00
  • 91c2d43ced Disable TaskQueueTest.PostDelayed because of flakiness terelius 2016-10-26 09:04:30 -07:00
  • e5ba44eab1 Implement framesDecoded stat in video receive ssrc stats. sakal 2016-10-26 07:09:24 -07:00
  • 784a83122b Check that stats_proxy_ is non-NULL before use. nisse 2016-10-26 07:02:23 -07:00
  • 5819660f9d MB: Add Linux swarming bots with memory sanitizers on the FYI waterfall. ehmaldonado 2016-10-26 07:00:24 -07:00
  • 059fb4480b - Replace FakeAudioProcessing in WVoE unittest with MockAudioProcessing. - Update MockAudioProcessing to current APM interface. - Replace calls to VoEAudioProcessing::Start/StopAecDump with direct calls on APM. - Add AudioProcessing* in WVoE, get it from VoE, so we can call directly on APM. solenberg 2016-10-26 05:12:24 -07:00
  • 16b6d6dc5b Reland of "Separating video settings in VideoQualityTest". minyue 2016-10-26 05:04:06 -07:00
  • c1600c5695 Follow standard sending CVO rtp header extension Include CVO in key frame. Include CVO in delta frame when rotation changes. Include CVO when it is non zero to support current receiver implementation. danilchap 2016-10-26 03:33:11 -07:00
  • b906172e02 Reland of Move bitstream parser to more appropriate directory. (patchset #1 id:1 of https://codereview.webrtc.org/2430353004/ ) kthelgason 2016-10-26 02:48:16 -07:00
  • 12ba1867a2 Move parsing from tests to Transport helper in RTPSenderTests making tests cleaner danilchap 2016-10-26 02:41:55 -07:00
  • a8bec8d8e7 Testing of VideoFileRenderer with byte frames mandermo 2016-10-26 01:47:07 -07:00
  • 940b6d648f Clean up logging in AudioSendStream::SetupSendCodec(). solenberg 2016-10-25 11:19:07 -07:00
  • da389e3518 PrintTo functions for RTCStats added in rtcstatscollector_unittest.cc hbos 2016-10-25 10:55:08 -07:00
  • d89ab145cd Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit. johan 2016-10-25 10:50:32 -07:00
  • 57e13defc7 Minor cleanup of rtc::BasicPacketSocketFactory implementation. johan 2016-10-25 10:15:06 -07:00
  • 0d8ade543d Remove remnants of libsrtp1 mattdr 2016-10-25 09:47:26 -07:00
  • 257dc39841 Refactoring: Hide VideoCodec.codecSpecific as "private" hta 2016-10-25 09:05:06 -07:00
  • 189f9b1b65 Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ ) terelius 2016-10-25 07:56:37 -07:00
  • d0af5c6fd4 Fix a deadlock in EglRenderer.releaseEglSurface. sakal 2016-10-25 07:20:52 -07:00
  • 2d81eb33f5 Fix BWE simulations so that it uses the delay based BWE. terelius 2016-10-25 07:04:37 -07:00
  • 1836fd6257 Clean up logging in AudioSendStream::SetupSendCodec(). solenberg 2016-10-25 06:44:45 -07:00
  • 701d628f5f Moved the AGC render sample queue into the audio processing module peah 2016-10-25 05:42:20 -07:00
  • d8872c5907 Removed the file resources/audioproc.aecdump.sha1 file which is no longer used. peah 2016-10-25 04:56:23 -07:00
  • a062460a68 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 2016-10-25 04:45:24 -07:00
  • 67c8bc4bf2 RTCStats equality operator added. hbos 2016-10-25 04:31:23 -07:00
  • 01bbc3c074 Reland of Deflake ChangingNetworkRoute test. stefan 2016-10-25 04:19:48 -07:00
  • 77f5953672 Revert of Deflake ChangingNetworkRoute test. (patchset #1 id:1 of https://codereview.webrtc.org/2426073002/ ) ehmaldonado 2016-10-25 04:02:21 -07:00
  • 67118201fb Deflake ChangingNetworkRoute test. stefan 2016-10-25 03:39:30 -07:00
  • cc34833809 Remove now unused code in RtpHeaderExtensionMap Remove functions to enumerate all extensions, Remove concept of the inactive extension. Decision if extension should be included into rtp header is done by rtp_sender GetTotalLengthInBytes now calculates all extension, included or not. That is used only for calculating how much space to reserve for fec. Since extension might suddenly be included in the next packet (which still might belong to same fec group), it is safer to calculate all registered extension. danilchap 2016-10-25 03:12:28 -07:00
  • 611f267370 Make WebRTC compatible with OpenH264 v1.6. sprang 2016-10-25 03:08:57 -07:00
  • af1ae310ef Remove dead dependencies on xmllite and xmpp. kjellander 2016-10-25 01:27:12 -07:00
  • dbf6705895 codec_unittest.cc: Fix TEST vs TEST_F mismatch Magnus Jedvert 2016-10-25 10:17:37 +02:00
  • 06c8e1eaa7 Revert of H264 codec: Check profile-level-id when matching (patchset #2 id:60001 of https://codereview.webrtc.org/2347863003/ ) Magnus Jedvert 2016-10-25 09:53:52 +02:00
  • fcba8feeb8 Delete left-over file profiler_unittest.cc. nisse 2016-10-25 00:33:36 -07:00
  • 74097fd3f5 Delete unused file screencastid.h. nisse 2016-10-25 00:17:43 -07:00
  • e58d73d23e Fix more swarming test failures by using the fake clock or longer timeout. In the swarming test, the machines sometimes were blocked for 1-2 seconds without processing anything. This CL makes sure that 1 second timeout is only used with fake clock. honghaiz 2016-10-24 16:38:26 -07:00
  • a6b8298b48 Use relative names in GN to make Chromium happy kwiberg 2016-10-24 16:31:17 -07:00
  • 4a18f16c62 Update XServerPixelBuffer to handle errors returned from XGetImage(). sergeyu 2016-10-24 15:45:45 -07:00
  • da2bf4e150 Stop using old AudioCodingModule::RegisterReceiveCodec overloads kwiberg 2016-10-24 13:47:09 -07:00
  • 88b7074745 Remove unused function implementations from FakeWebRtcVoiceEngine. solenberg 2016-10-24 13:46:01 -07:00
  • fb70b45030 Preventing TURN redirects to loopback addresses. deadbeef 2016-10-24 13:15:59 -07:00