generic-library/webrtc
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  • 858dbbced2 cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. magjed@webrtc.org 2014-11-16 18:21:51 +00:00
  • 6a782c2a46 Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases. henrike@webrtc.org 2014-11-14 22:33:13 +00:00
  • be05c74ec8 Wrap the splitting filter in its own class aluebs@webrtc.org 2014-11-14 22:18:10 +00:00
  • 67c22478a4 Disable EndToEnd.GetStats test. pbos@webrtc.org 2014-11-14 17:42:51 +00:00
  • a73d746562 Revert 7702 "cricket::VideoAdapter: Drop frames before spending ..." magjed@webrtc.org 2014-11-14 13:25:25 +00:00
  • bbd8cad21f cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. magjed@webrtc.org 2014-11-14 12:10:46 +00:00
  • ece3890d3a Report total bitrate for all streams in GetStats. pbos@webrtc.org 2014-11-14 11:52:04 +00:00
  • 35c1ace185 Revert 7698 "WebRtcVideoMediaChannel::SetSendParams: Don't cap r..." magjed@webrtc.org 2014-11-13 16:21:49 +00:00
  • a1f5b96351 Remove unnecessary copying of libjingle resource files. kjellander@webrtc.org 2014-11-13 15:53:08 +00:00
  • 52da44b7e6 WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution magjed@webrtc.org 2014-11-13 15:43:11 +00:00
  • 49ff40e32e Make SetREMBData accept vector of SSRCs. pbos@webrtc.org 2014-11-13 14:42:37 +00:00
  • a9c2d454bd Fix and enable CanReceiveFec test. pbos@webrtc.org 2014-11-13 14:40:15 +00:00
  • ee30082af8 Set correct sample rate in far_frame in audioproc tool. bjornv@webrtc.org 2014-11-13 11:00:10 +00:00
  • 52bb521b47 Update isolate files for Android APK tests. kjellander@webrtc.org 2014-11-13 08:35:05 +00:00
  • 312614a438 Add jmi field for packets discarded due to network error guoweis@webrtc.org 2014-11-13 03:38:05 +00:00
  • 90b9b08332 Fix a platform check to use WEBRTC_WIN instead of OS_WIN. jiayl@webrtc.org 2014-11-12 20:53:00 +00:00
  • 6ca6190be2 Fix a SCTP message reordering issue in datachannel.cc. Previously DataChannel::SendQueuedDataMessages continues the loop of sending queued messages if the channel is blocked, which will cause message reordering if the channel becomes unblocked during the loop, i.e. messages attempted after the unblocking will be sent earlier than the older messages attempted before the unblocking. jiayl@webrtc.org 2014-11-12 17:28:40 +00:00
  • ea73ff7267 webrtc::Scaler: Preserve aspect ratio magjed@webrtc.org 2014-11-12 09:52:03 +00:00
  • 0b3d89b500 VideoSendStreamTest.SwapsI420VideoFrames: Initialize frame memory to avoid drmemory errors magjed@webrtc.org 2014-11-12 08:58:49 +00:00
  • 14ea50a8e3 Change the static_library("webrtc") to a source set in the GN build. kjellander@webrtc.org 2014-11-12 07:56:21 +00:00
  • 0e37b898f0 replace inline assembly WebRtcAecm_CalcLinearEnergiesNeon by intrinsics. andrew@webrtc.org 2014-11-11 19:34:14 +00:00
  • e497be3de1 replace inline assembly WebRtcAecm_StoreAdaptiveChannelNeon by intrinsics. andrew@webrtc.org 2014-11-11 19:32:33 +00:00
  • 0e71070207 Use ScreenCapturer to capture the whole and clip to the window rect when the shared window is on the top. jiayl@webrtc.org 2014-11-11 18:15:55 +00:00
  • a367aeab82 Bump to version 40 tnakamura@webrtc.org 2014-11-11 16:23:15 +00:00
  • f7c5d4fac7 Revert 7679 "webrtc::Scaler: Preserve aspect ratio" magjed@webrtc.org 2014-11-11 13:12:09 +00:00
  • 525baea03f Add PROJECT to codereview.settings kjellander@webrtc.org 2014-11-11 10:00:47 +00:00
  • 944fb57140 Roll chromium_revision 375f736..91f1781 kjellander@webrtc.org 2014-11-11 09:57:19 +00:00
  • 809986b95f webrtc::Scaler: Preserve aspect ratio magjed@webrtc.org 2014-11-11 09:51:30 +00:00
  • cd621a8657 Add thread annotations to overuse_frame_detector class. asapersson@webrtc.org 2014-11-11 09:40:19 +00:00
  • 8038d42749 Follow-up fixes for G722 henrik.lundin@webrtc.org 2014-11-11 08:38:24 +00:00
  • 1431e4dd1c Revert 7675 "Make an AudioEncoder subclass for iSAC" turaj@webrtc.org 2014-11-11 01:44:13 +00:00
  • 05feff013e Make an AudioEncoder subclass for iSAC kwiberg@webrtc.org 2014-11-10 23:53:08 +00:00
  • 33045ab2c1 Change from talk/p2p (r7664) "(Auto)update libjingle 79414100-> 79428003". henrike@webrtc.org 2014-11-10 19:43:11 +00:00
  • 43e033e778 Change from talk/p2p (r7572): "Improve the logging when a TCP connection is deleted." henrike@webrtc.org 2014-11-10 19:40:29 +00:00
  • 4ffc7341ca replace inline assembly WebRtcAecm_ResetAdaptiveChannelNeon by intrinsics. andrew@webrtc.org 2014-11-10 17:27:53 +00:00
  • d024f759a8 clear asm code and unused functions in audio processing module andrew@webrtc.org 2014-11-10 17:19:57 +00:00
  • c4922316b4 Removes talk/xmllite, talk/xmpp and talk/p2p as they are no longer used by gyp/gn builds. henrike@webrtc.org 2014-11-10 15:31:24 +00:00
  • d819803d45 Wire up DSCP support in WebRtcVideoEngine2. pbos@webrtc.org 2014-11-10 14:41:43 +00:00
  • 83d4804a50 Put send-side bwe probing under finch experiment. stefan@webrtc.org 2014-11-10 13:55:16 +00:00
  • 957e802fe0 Refactor SetDefaultEncoderConfig to work on existing codecs. pbos@webrtc.org 2014-11-10 12:36:11 +00:00
  • a5d29fcd59 Add unit to dropped frames. pbos@webrtc.org 2014-11-10 09:54:19 +00:00
  • bd495fab27 .gitignore updates kjellander@webrtc.org 2014-11-10 06:51:34 +00:00
  • 3c1970f9f3 (Auto)update libjingle 79414100-> 79428003 buildbot@webrtc.org 2014-11-07 17:58:41 +00:00
  • 188d3b2245 Enable VP9 video codec support on webrtcvideoengine behind a field trial. andresp@webrtc.org 2014-11-07 13:21:04 +00:00
  • f85dbce041 Reapply "Advertise G722 as 8 kHz rather than 16 kHz"" henrik.lundin@webrtc.org 2014-11-07 12:25:00 +00:00
  • d105cc81dc Change dummy address to use 0.0.0.0 instead of :: This is to not break compatiblity with FF. perkj@webrtc.org 2014-11-07 11:22:06 +00:00
  • d42a3adf42 Remove partially defined WebRtcRTPHeader from Parse(). pbos@webrtc.org 2014-11-07 11:02:12 +00:00
  • a2ef4fe9c3 Prevent a lot of VideoSendStream reconfigures. pbos@webrtc.org 2014-11-07 10:54:43 +00:00
  • 82775b1396 Refactor webrtcvideoengines to have the default list of supported codecs being generated in runtime. This will allow to plugin VP9 based on a field trial. andresp@webrtc.org 2014-11-07 09:37:54 +00:00
  • 5e160660a6 Reland Volume buttons in AppRTCDemo should affect output audio volume (part I). henrika@webrtc.org 2014-11-06 20:35:13 +00:00
  • 332331fb01 Use uint16s for port numbers in webrtc/p2p/base. pkasting@chromium.org 2014-11-06 20:19:22 +00:00
  • d89b69aade Fix WebRTC Win64 + BoringSSL build. henrike@webrtc.org 2014-11-06 17:23:09 +00:00
  • dd43bbed8f Volume buttons in AppRTCDemo should affect output audio volume (part II). henrika@webrtc.org 2014-11-06 15:48:05 +00:00
  • dced5d7835 Revert "Advertise G722 as 8 kHz rather than 16 kHz" henrik.lundin@webrtc.org 2014-11-06 15:27:43 +00:00
  • 34bda43fa6 (Auto)update libjingle 79326895-> 79329222 buildbot@webrtc.org 2014-11-06 12:44:55 +00:00
  • e5421e9602 Volume buttons in AppRTCDemo should affect output audio volume. henrika@webrtc.org 2014-11-06 12:19:19 +00:00
  • fd0efb694a Remove deprecated PeerConnection APIs. Removes PeerConnectionObserver::OnError. Removes MediaConstraints argument to PeerConnection::AddStream. None of these have ever been implemented and have been removed from the spec. perkj@webrtc.org 2014-11-06 12:16:36 +00:00
  • 19b4741004 Removing unused method GetDefaultVideoEncoderConfig. andresp@webrtc.org 2014-11-06 11:16:32 +00:00
  • 931e3da8f2 Log formatting fix for VideoEncoderConfig. pbos@webrtc.org 2014-11-06 09:35:08 +00:00
  • 0ef890a4ba (Auto)update libjingle 79285346-> 79320771 buildbot@webrtc.org 2014-11-06 09:22:08 +00:00
  • 6340acde68 AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. mcasas@webrtc.org 2014-11-06 09:05:48 +00:00
  • 1dcca4028f Advertise G722 as 8 kHz rather than 16 kHz henrik.lundin@webrtc.org 2014-11-06 08:55:01 +00:00
  • 8b2058e733 Remove the state_ member from AudioDecoder kwiberg@webrtc.org 2014-11-06 07:54:31 +00:00
  • 32022c6fb1 Revert 7642 "Fix memcheck and dr memory after flakiness dashboar..." kjellander@webrtc.org 2014-11-06 06:26:34 +00:00
  • 724fbaf473 Fix memcheck and dr memory after flakiness dashboard deployment. kjellander@webrtc.org 2014-11-06 06:04:09 +00:00
  • 7e4a05ec29 Exclude SendsAndReceivesVP9 for linux-memcheck. marpan@webrtc.org 2014-11-06 05:47:59 +00:00
  • 53bed75104 Change DrMemory exclusion to match changed test name. andrew@webrtc.org 2014-11-06 05:33:01 +00:00
  • f6b7c7e6a6 Exclude SendsAndReceivesVP9 for WinDrMemory. marpan@webrtc.org 2014-11-06 05:09:26 +00:00
  • e1745cbb7c Adjust parameter in vp9 rate control test. marpan@webrtc.org 2014-11-06 02:55:53 +00:00
  • 5f1e2e42a8 Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test. marpan@webrtc.org 2014-11-06 02:02:28 +00:00
  • ee9d61ce45 This fixes a small memory leak (found using Xcode/Instruments on iOS) in the ObjC bindings of PeerConnection. The generated session description has to be released by the recipient tkchin@webrtc.org 2014-11-05 22:01:53 +00:00
  • 6a364fe11b Remove uses of build date/time. pbos@webrtc.org 2014-11-05 17:40:28 +00:00
  • 0bae1fab4a Wire up bandwidth stats to the new API and webrtcvideoengine2. stefan@webrtc.org 2014-11-05 14:05:29 +00:00
  • a22a628356 (Auto)update libjingle 79205306-> 79244016 buildbot@webrtc.org 2014-11-05 13:25:48 +00:00
  • 72fd339352 Restore old behavior for Android in fileutils.cc kjellander@webrtc.org 2014-11-05 06:28:50 +00:00
  • f6e1600a7d Roll chromium_revision d3db2ff..375f736 kjellander@webrtc.org 2014-11-05 02:09:20 +00:00
  • dc8662435b Fix android_clang build. glaznev@webrtc.org 2014-11-05 01:15:10 +00:00
  • 368215dacb Revert 7623 "Remove the state_ member from AudioDecoder" niklas.enbom@webrtc.org 2014-11-05 00:45:58 +00:00
  • 8a232f65dd Revert 7625 "Don't use DCHECK when you need the side effects..." niklas.enbom@webrtc.org 2014-11-05 00:43:59 +00:00
  • 795d003770 (Auto)update libjingle 79200114-> 79205306 buildbot@webrtc.org 2014-11-05 00:14:02 +00:00
  • 8125744a5f Cleanup RTCVideoRenderer interface. tkchin@webrtc.org 2014-11-04 23:06:15 +00:00
  • b8425bc4f3 Don't use DCHECK when you need the side effects... kwiberg@webrtc.org 2014-11-04 22:10:18 +00:00
  • 45ecf4c092 (Auto)update libjingle 79169148-> 79192489 buildbot@webrtc.org 2014-11-04 21:48:54 +00:00
  • 9e525585fd Remove the state_ member from AudioDecoder kwiberg@webrtc.org 2014-11-04 21:18:47 +00:00
  • 7c29e8c2f3 Add support for VP9 in webrtc::Call and video_loopback. stefan@webrtc.org 2014-11-04 19:41:15 +00:00
  • d839e0ab52 Reduce to 2 probes when probing for initial bandwidth. stefan@webrtc.org 2014-11-04 19:33:55 +00:00
  • db26247a9b Add UMA for measuring the diff between the BWE at 2 seconds compared to the BWE at 20 seconds when the BWE should have converged. stefan@webrtc.org 2014-11-04 19:32:10 +00:00
  • 8944c9d08b AppRTCDemoActivity: use differnet Themes for different API levels mcasas@webrtc.org 2014-11-04 17:26:22 +00:00
  • d367321a3f Add kjellander as PRESUBMIT.py OWNER kjellander@webrtc.org 2014-11-04 17:06:31 +00:00
  • dcebf2daa7 Reworked paced sender queue sprang@webrtc.org 2014-11-04 16:27:16 +00:00
  • fad9aecbf5 Remove protected files from talk/PRESUBMIT.py. pbos@webrtc.org 2014-11-04 16:06:35 +00:00
  • 88ef632286 Falling back on single-stream on multiple SSRC. pbos@webrtc.org 2014-11-04 15:29:29 +00:00
  • 28af64105b Presubmit was not whitelisting libjingle_tests.gyp or sound.gyp due to a missing comma leading to a concatenation of the two strings in the whitelist. henrike@webrtc.org 2014-11-04 15:11:46 +00:00
  • b3265accd9 Adds support for finch experiments to video_loopback. stefan@webrtc.org 2014-11-04 14:57:14 +00:00
  • 52b42cb069 Fix problem with late packets in NetEq henrik.lundin@webrtc.org 2014-11-04 14:03:58 +00:00
  • 09cc686c8b Delete VideoReceiveStream channels in destructor. pbos@webrtc.org 2014-11-04 13:48:15 +00:00
  • 6de75ca3ed Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16 kwiberg@webrtc.org 2014-11-04 13:29:24 +00:00
  • c78cf97ecb Remove the useless dummy state parameter to WebRtcG711_* kwiberg@webrtc.org 2014-11-04 13:23:36 +00:00
  • b5d045e94d ReAdd PeerConnectionInterface::AddStream to fix Chrome build. AddStream(MediaStreamInterface* stream, const MediaConstraintsInterface* constraints); This will be removed once Chrome has been updated. perkj@webrtc.org 2014-11-04 13:01:33 +00:00
  • 18de6f9622 Change the PeerConnection proxy templates to use blocking method calls instead of using Thread::Send. The problem with Thread::Send is that it processes incoming pending messages and for the proxies, this can mean that multiple incoming calls can concurrently run on the same thread, resulting in unexpected behavior. tommi@webrtc.org 2014-11-04 12:08:48 +00:00