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  • a6e8cebbd5 Fix false positive DHECK in event_posix.cc sprang@webrtc.org 2015-02-11 15:19:08 +00:00
  • 11426dc719 Don't rely on webrtc/base/scoped_ptr.h to include stuff for you kwiberg@webrtc.org 2015-02-11 14:30:34 +00:00
  • fbcb5ceb16 Remove VideoSendStreamTest.ProducesStats. pbos@webrtc.org 2015-02-11 14:24:44 +00:00
  • 9d94a0c736 Switch to QueueUserAPC for shutting down the thread (no event needed). Also actually specifying the reserve stack size. tommi@webrtc.org 2015-02-11 14:16:08 +00:00
  • fddeaf5daa Switch to using AudioEncoderG722 instead of ACMG722 henrik.lundin@webrtc.org 2015-02-11 13:28:20 +00:00
  • 83bc721c7e Add Android specific VideoCapturer. The Java implementation of VideoCapturer is losely based on the the work in webrtc/modules/videocapturer. perkj@webrtc.org 2015-02-11 11:26:56 +00:00
  • c18957e877 Make Git ignore in resources more fine-grained kjellander@webrtc.org 2015-02-11 09:58:45 +00:00
  • 354becf4df Remove Git ignore exclusion of .sha1 files kjellander@webrtc.org 2015-02-11 09:44:01 +00:00
  • 7cc92aaf37 Use WebRtcVideoRenderFrame for texture frames. pbos@webrtc.org 2015-02-11 09:03:15 +00:00
  • 62f6e75673 Refactoring WebRTC Java/JNI audio recording in C++ and Java. henrika@webrtc.org 2015-02-11 08:38:35 +00:00
  • c2d0473320 Switch to using AudioEncoderPcm16B instead of ACMPCM16B henrik.lundin@webrtc.org 2015-02-11 08:25:12 +00:00
  • f58fe0ab2b Rename GYP and GN targets for video capture+render. kjellander@webrtc.org 2015-02-11 07:47:00 +00:00
  • 2c29c2eae2 C++ readability review for ajm. andrew@webrtc.org 2015-02-11 01:09:50 +00:00
  • 5d608955cf Fix bug when there are no blocks in a chunk in Beamformer aluebs@webrtc.org 2015-02-11 00:48:10 +00:00
  • bc35703694 Add a method to remove an existing renderer from the internal list of Android renderers. glaznev@webrtc.org 2015-02-10 23:23:11 +00:00
  • bc40324d9c Merge fixes and changed for Android AppRTCDemo from internal repo. glaznev@webrtc.org 2015-02-10 23:04:13 +00:00
  • d35a5c3506 Make ChannelBuffer aware of frequency bands aluebs@webrtc.org 2015-02-10 22:52:15 +00:00
  • d7472b52d6 base/arraysize.h: We use size_t, so need to include stddef.h kwiberg@webrtc.org 2015-02-10 22:46:42 +00:00
  • 91ba79ae3f Make sure that the norms are positive in Beamformer aluebs@webrtc.org 2015-02-10 22:38:05 +00:00
  • b6856d2823 Apply mask smoothing in Beamformer aluebs@webrtc.org 2015-02-10 18:23:06 +00:00
  • 8da96ac0f6 Switch to using AudioEncoderIlbc instead of ACMILBC henrik.lundin@webrtc.org 2015-02-10 15:33:21 +00:00
  • 1a072f93eb Address comments from previous review round for rtc::Event. tommi@webrtc.org 2015-02-10 12:27:48 +00:00
  • f4c10d24dc Always use DeliverI420Frame in WebRtcVideoEngine. pbos@webrtc.org 2015-02-10 10:19:32 +00:00
  • 027e113209 Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig. stefan@webrtc.org 2015-02-10 09:48:35 +00:00
  • 30015e3180 Fix bug in EventPosix where we'd miss a set event. In cases of timeout or error, we could change the state of the event to 'down' (unset) and subsequently never satisfy a Wait() for a given Set(). tommi@webrtc.org 2015-02-10 09:33:28 +00:00
  • 648f5d6dc7 pcm16b: Make input arrays const and use uint8_t[] for byte arrays kwiberg@webrtc.org 2015-02-10 09:18:28 +00:00
  • 948d61724c Create a separate thread for pacing. mflodman@webrtc.org 2015-02-10 08:58:16 +00:00
  • c11348b5d7 Fixing a bug in expand_rate calculation for stereo signal. minyue@webrtc.org 2015-02-10 08:35:38 +00:00
  • 8e612aba60 Remove voice_engine_ member variable and GetVoiceEngine() from ViEChannelManager. tommi@webrtc.org 2015-02-10 08:15:28 +00:00
  • 5b8f3e0206 Roll chromium_revision 598c3e9..601e6f3 kjellander@webrtc.org 2015-02-10 07:34:07 +00:00
  • 44ae4c8b07 Support using VP9 video codec in AppRTCDemo. glaznev@webrtc.org 2015-02-09 23:25:58 +00:00
  • f7e6cfd3a0 Add CHECK to EventWrapper to see if there's a subtle bug there or not. tommi@webrtc.org 2015-02-09 18:25:38 +00:00
  • 669bc7ee43 Modify default field trial implementation to allow WebRTC client to turn on feature code. glaznev@webrtc.org 2015-02-09 18:17:46 +00:00
  • 11c5db01af Revert 8273 "Temporarily change ThreadPosix to CHECK (crash) if ..." tommi@webrtc.org 2015-02-09 16:31:31 +00:00
  • 0d852d5c27 Use VideoReceiveStream as an ExternalRenderer. pbos@webrtc.org 2015-02-09 15:14:36 +00:00
  • d6e25a5b27 Revert r8297 "Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig." stefan@webrtc.org 2015-02-09 15:06:16 +00:00
  • 03c1c103e4 Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig. stefan@webrtc.org 2015-02-09 14:46:58 +00:00
  • 53d9012faf Clean kForever from basictypes and move it to the interfaces that actually have it. andresp@webrtc.org 2015-02-09 14:19:09 +00:00
  • e01bae24a5 Fixing a nit henrik.lundin@webrtc.org 2015-02-09 13:21:19 +00:00
  • 1c6239a3b6 G711: Make input arrays const and use uint8_t[] for byte arrays kwiberg@webrtc.org 2015-02-09 12:55:48 +00:00
  • d0165c62b5 Use a manual reset event in PosixThread. This fixes occasional hangs we've been seeing in the past few days. I'm using rtc::Event instead of the EventWrapper, so I'll wait with landing this cl until I've made that change in a separate cl. tommi@webrtc.org 2015-02-09 11:47:57 +00:00
  • 4c0fd965ce Move rtc::Event to rtc_base_approved. We need an event implementation in WebRTC that allows us to specify whether it's manually reset or automatically. EventWrapper currently doesn't support it and it adds a heap allocation + vtable, so rtc::Event is the lighter of the two. tommi@webrtc.org 2015-02-09 10:23:27 +00:00
  • 8cf9bdb3fa Remove USE_WEBRTC_DEV_BRANCH. pbos@webrtc.org 2015-02-09 10:16:33 +00:00
  • 2b69eab077 Restructure GYP for vp9, opus and direct trace kjellander@webrtc.org 2015-02-09 10:01:17 +00:00
  • f31f56d8d4 Remove default arguments in EncodedImageCallback. changbin.shao@webrtc.org 2015-02-09 09:14:03 +00:00
  • 6c930c7183 Cleanup: unify rotation to be enum based instead of int for degree. guoweis@webrtc.org 2015-02-09 01:28:12 +00:00
  • 7a57f8f101 Reland 8203 "Reducing locking in OveruseFrameDetect..." The issue that was causing the thread checker to report error, turned out to be unrelated. tommi@webrtc.org 2015-02-08 18:27:46 +00:00
  • 103f3289b5 Fix the binary layout of ProcessThreadImpl. We apparently hit an obscure problem on mac where seemingly an unaligned mutex causes memory corruption. The effect was that the |modules_| list became corrupt and we crashed. At this point I'm not exactly sure what the alignment requirements are but for now, I've fixed up the layout in a way that doesn't cause these same issues. tommi@webrtc.org 2015-02-08 00:48:10 +00:00
  • ec499beaf5 Increase testclient timeout from 1 to 5 seconds jlmiller@webrtc.org 2015-02-07 22:37:59 +00:00
  • fe19699a20 Revert 8260 "Base RWLockWrapper on rtc::SharedExclusiveLock." Unfortunately this caused channel teardown to hang. More details in email(s). tommi@webrtc.org 2015-02-07 22:35:54 +00:00
  • 2eb1660791 Switch ThreadCheckerImpl over to using PlatformThreadRef. Like PlatformThreadId, this type is borrowed from Chromium. The difference between the two is that PlatformThreadRef is pthread_t on posix platforms. On Windows PlatformThreadRef and PlatformThreadId are the same thing. tommi@webrtc.org 2015-02-07 19:17:47 +00:00
  • 2bf0e90c9d Revert 8275 "This CL adds an API to the SSL stream adapters and ..." tommi@webrtc.org 2015-02-07 11:12:19 +00:00
  • 1d4830a077 Disable ProcessThread tests that are dependent on timing. Some of the bots are too slow for the tests to make much sense as they are. tommi@webrtc.org 2015-02-07 08:44:28 +00:00
  • 95a32ec098 Revert 8271 "VirtualSocketServer out-of-order issue with closing..." bjornv@webrtc.org 2015-02-07 06:46:56 +00:00
  • 2a44be93e8 Normalize delay-and-sum mask in Beamformer aluebs@webrtc.org 2015-02-07 02:41:24 +00:00
  • 799e667e9f Add high frequency correction to Beamformer aluebs@webrtc.org 2015-02-07 01:07:09 +00:00
  • 0c7ec770ff Cleanup: unify rotation to be enum based instead of int for degree. guoweis@webrtc.org 2015-02-06 21:01:23 +00:00
  • 110443aaac Cleanup: unify rotation to be enum based instead of int for degree. guoweis@webrtc.org 2015-02-06 20:00:00 +00:00
  • 1d11c8202b This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer. pthatcher@webrtc.org 2015-02-06 19:46:53 +00:00
  • 63da1dd972 audio_processing: Now records mic volume level also when using new AGC bjornv@webrtc.org 2015-02-06 19:44:21 +00:00
  • ccd7e99f0a Temporarily change ThreadPosix to CHECK (crash) if we ever spend more than 30 seconds waiting for thread shutdown. There are cases on build bots where it looks like we're hitting this problem, but reproducing locally has been a struggle. tommi@webrtc.org 2015-02-06 19:26:42 +00:00
  • 13a0e184ee Temporarily disable a couple of ThreadChecker tests on Mac. tommi@webrtc.org 2015-02-06 16:38:58 +00:00
  • 4770437da9 VirtualSocketServer out-of-order issue with closing TCP sockets pthatcher@webrtc.org 2015-02-06 16:33:08 +00:00
  • 9baa9ca399 Add libjingle_peerconnection_so.so to Java test dependencies. This fix a problem where the Java test is not dependent on the so file. perkj@webrtc.org 2015-02-06 16:08:57 +00:00
  • b5a1252e66 Hack to work around the current issues with rolling WebRTC into chromium. In order to figure out the issue with the Mac 10.9 debug bot, this patch disables the ThreadChecker class on Mac in debug builds. For diagnostic purposes, it instead prints out when there's a thread mismatch. I'm also adding a DCHECK in case fetching the current thread id ever returns 0. tommi@webrtc.org 2015-02-06 15:39:05 +00:00
  • 751a36590a Switch to using AudioEncoderPcmU/A instead of ACMPCMU/A henrik.lundin@webrtc.org 2015-02-06 14:03:29 +00:00
  • 02270cd718 Implementing a packet router class, used to route RTP packets to the sending RTP module for the specified simulcast layer a frame belongs to. This CL also removes the corresponding functionality from the RTP RTCP module and fixes lint warnings in the files touched. mflodman@webrtc.org 2015-02-06 13:10:19 +00:00
  • 10a9e924eb Fix delete of stack allocated object causing test crashes. stefan@webrtc.org 2015-02-06 12:59:39 +00:00
  • 4b320cf214 Revert "Cleanup: unify rotation to be enum based instead of int for degree." magjed@webrtc.org 2015-02-06 12:58:09 +00:00
  • fb609a1f57 Wire up new feedback format by introducing a FeedbackPacket type. stefan@webrtc.org 2015-02-06 12:20:33 +00:00
  • 353c8b8c08 audio_processing/agc: Changed to correct include path in agc_unittests bjornv@webrtc.org 2015-02-06 12:02:38 +00:00
  • bc3241a8cc Update ProcessCallAfterXms to better match the performance of our faster bots. Previously I had made sure these tests didn't flake out on our slow trybots, but apparently I need to do the same for the fast bots :) tommi@webrtc.org 2015-02-06 11:28:11 +00:00
  • 0c3e12b7bf Revamp the ProcessThreadImpl implementation. tommi@webrtc.org 2015-02-06 09:44:12 +00:00
  • 75025434bf Base RWLockWrapper on rtc::SharedExclusiveLock. pbos@webrtc.org 2015-02-06 08:32:32 +00:00
  • 5e05731b0f Roll chromium_revision cd35af6..598c3e9 kjellander@webrtc.org 2015-02-06 07:25:02 +00:00
  • 57ac2c84dd Default destination used by c line should be IPv4 only to avoid parsing error in legacy client. guoweis@webrtc.org 2015-02-06 00:45:13 +00:00
  • 3e733a43f5 Cleanup: unify rotation to be enum based instead of int for degree. guoweis@webrtc.org 2015-02-05 23:40:19 +00:00
  • 74d27884af Remove defined(__cplusplus) tests in C++ code. jan.skoglund@webrtc.org 2015-02-05 19:17:44 +00:00
  • f45c8ca88b Reland r8248 "Introduce ACMGenericCodecWrapper" henrik.lundin@webrtc.org 2015-02-05 18:29:39 +00:00
  • ec4521cdb4 Clean up Beamformer initialization aluebs@webrtc.org 2015-02-05 18:16:36 +00:00
  • e69220ca84 Fix the value of the first byte of nal unit generated by fake H.264 encoder. glaznev@webrtc.org 2015-02-05 17:56:15 +00:00
  • f6932297e7 Fix Android video renderer to support video frames with stride > width. glaznev@webrtc.org 2015-02-05 17:29:59 +00:00
  • cc64a9cc4f voice_engine: Updates GetEcDelayMetrics() w.r.t. new metric bjornv@webrtc.org 2015-02-05 12:52:44 +00:00
  • 4b9622fb1e Roll gtest-parallel. pbos@webrtc.org 2015-02-05 12:37:24 +00:00
  • 3a87630629 Revert r8248 "Introduce ACMGenericCodecWrapper" henrik.lundin@webrtc.org 2015-02-05 09:36:58 +00:00
  • af8c13f2a1 Introduce ACMGenericCodecWrapper henrik.lundin@webrtc.org 2015-02-05 09:19:37 +00:00
  • 5d32f43219 Disable CondVarTest.InitFunctionsWork. The order of Sleep/Wake calls doesn't seem to be guaranteed, so this test is flaky. tommi@webrtc.org 2015-02-05 06:25:35 +00:00
  • 877ac765ad Cleanup and prepare for bundling. pthatcher@webrtc.org 2015-02-04 22:03:09 +00:00
  • cf7efeba37 Add new AudioEncoderOpusTest henrik.lundin@webrtc.org 2015-02-04 15:34:05 +00:00
  • 520a69e8ea Revert 8238 "Add RefCounting for TransportProxies" bjornv@webrtc.org 2015-02-04 12:45:44 +00:00
  • 875c97ed9d Remove SetNotAlive method from the thread class. Also cleaning up methods with the same name in other classes that are derived from the above method. tommi@webrtc.org 2015-02-04 11:11:53 +00:00
  • c5f697135e Revert 8237 "Cleanup and prepare for bundling." bjornv@webrtc.org 2015-02-04 10:22:14 +00:00
  • dc096f2c7e system_wrappers: Disabled flaky test CondVarTest.PassBatonMultipleTimes bjornv@webrtc.org 2015-02-04 09:14:14 +00:00
  • 4414939954 Add method for incrementing RtpPacketCounter. Removes duplicate code. asapersson@webrtc.org 2015-02-04 08:34:47 +00:00
  • e2506670a4 Add RefCounting for TransportProxies decurtis@webrtc.org 2015-02-03 23:18:39 +00:00
  • af01d93aa2 Cleanup and prepare for bundling. pthatcher@webrtc.org 2015-02-03 23:13:37 +00:00
  • 322a564f49 Fix datachannel stats id and timestamp. decurtis@webrtc.org 2015-02-03 22:09:37 +00:00
  • d43bdf50c5 Rewrite ThreadPosix. This is the same change as already made for Windows: https://webrtc-codereview.appspot.com/37069004/ tommi@webrtc.org 2015-02-03 16:29:57 +00:00
  • bfdee69d48 Roll chromium_revision 9070a80..cd35af6 (313233:314322) kjellander@webrtc.org 2015-02-03 15:23:34 +00:00
  • 0ec50be2f7 Changing include guard in frame_callback.h. mflodman@webrtc.org 2015-02-03 14:51:06 +00:00