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  • cfd82dfc11 Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. Prepares for adding FEC bytes to the StreamDataCounter. asapersson@webrtc.org 2015-01-22 09:39:59 +00:00
  • 3dd33a6787 Fix bug in thresholds for bitrate probing and adjust thresholds to allow a larger dispersion and concentration for successful probes. stefan@webrtc.org 2015-01-22 09:12:23 +00:00
  • fbd37bd737 Make iSAC SWB own its decoder henrik.lundin@webrtc.org 2015-01-22 08:16:29 +00:00
  • cceb166a3f Fix a use-after-free when sending queued messages is aborted for blocked channel. jiayl@webrtc.org 2015-01-22 00:55:10 +00:00
  • e65d9d974c Fix an unitialized variable warning. andrew@webrtc.org 2015-01-21 22:05:12 +00:00
  • c429b824b3 GN: Prepare to remove webrtc_base target kjellander@webrtc.org 2015-01-21 20:22:33 +00:00
  • c78d81ae89 Re-land "Support 48kHz in AEC" aluebs@webrtc.org 2015-01-21 19:10:55 +00:00
  • e81c5d6d7e Fix TransientDetectorTest in modules_unittests on Android ARM64 aluebs@webrtc.org 2015-01-21 18:01:28 +00:00
  • 11af039590 Disable AcmSenderBitExactnessOldApi.Opus_stereo_20ms_voip on ARM64. minyue@webrtc.org 2015-01-21 14:22:39 +00:00
  • df7b65ba01 Change CreateOrGetReportBlockInformation to have one return path. asapersson@webrtc.org 2015-01-21 13:07:04 +00:00
  • f938922c5c Simplify and guard access to WindowsRealTimeClock. pbos@webrtc.org 2015-01-21 12:51:13 +00:00
  • 4fb7e25843 Update StatsReport and by extension StatsCollector to reduce data copying. tommi@webrtc.org 2015-01-21 11:36:18 +00:00
  • f66a6b2a00 Remove unnecessary dependencies from webrtc_all target. kjellander@webrtc.org 2015-01-21 10:06:55 +00:00
  • e7358eabbc Only report fraction of lost packets if report_block_stats has been updated. asapersson@webrtc.org 2015-01-21 09:00:19 +00:00
  • 9ffd8fe96b Indentation changes. asapersson@webrtc.org 2015-01-21 08:22:50 +00:00
  • fedb9ea6bc Correct the class name in peerconnection_jni.cc. braveyao@webrtc.org 2015-01-21 07:57:06 +00:00
  • 5f93d0a140 Update libjingle license statements at top of talk files for consistency jlmiller@webrtc.org 2015-01-20 21:36:13 +00:00
  • cbacd9e3bf Bump to version 41. tnakamura@webrtc.org 2015-01-20 18:52:01 +00:00
  • 7dba7860c7 Setting Opus target application. minyue@webrtc.org 2015-01-20 16:01:50 +00:00
  • 853049fa30 Move internal capture+render to build_with_chromium==0 condition kjellander@webrtc.org 2015-01-20 11:40:45 +00:00
  • 511ab3e7c8 Roll chromium_revision a6eafec..c086b4e kjellander@webrtc.org 2015-01-20 11:39:27 +00:00
  • ee0c100d54 Revert 8080 "Support 48kHz in AEC" tina.legrand@webrtc.org 2015-01-20 10:22:49 +00:00
  • f88f88edde Remove webrtc/base/compile_assert.h kwiberg@webrtc.org 2015-01-20 08:46:55 +00:00
  • 9691b36995 Cleanup for Rtp Rtcp API test. changbin.shao@intel.com 2015-01-20 05:42:52 +00:00
  • 8e327c45d0 Update StatsCollector's interface in preparation of more changes. tommi@webrtc.org 2015-01-19 20:41:26 +00:00
  • 43e54e36bf Revert 8095 "Update StatsCollector's interface in preparation of..." tommi@webrtc.org 2015-01-19 17:34:23 +00:00
  • 5b76fd79df Update StatsCollector's interface in preparation of more changes. tommi@webrtc.org 2015-01-19 16:49:33 +00:00
  • 474e36e623 Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps. stefan@webrtc.org 2015-01-19 15:44:47 +00:00
  • f9d3555ec3 Fixing LD_LIBRARY_PATH, improving safety for libjingle java unit test. phoglund@webrtc.org 2015-01-19 13:57:59 +00:00
  • ce3ac53757 Adding TRYSERVER_PROJECT to codereview.settings. kjellander@webrtc.org 2015-01-19 13:51:59 +00:00
  • 018c087a6d Add /talk/examples/androidtests/{bin,gen} to .gitignore. kjellander@webrtc.org 2015-01-19 12:52:43 +00:00
  • a32d15448d Disable tests failing on Android ARM64 (Nexus9). kjellander@webrtc.org 2015-01-19 12:46:01 +00:00
  • ff9462eb54 Disable WebRtcVideoMediaChannelSimulcastTest::SimulcastSend_* on tsan. sprang@webrtc.org 2015-01-19 12:06:35 +00:00
  • 2624b1ed23 Remove unused private data member engine_id_ tommi@webrtc.org 2015-01-19 07:54:29 +00:00
  • fe672e3839 release the turn allocation by sending a refresh request with lifetime 0 pthatcher@webrtc.org 2015-01-17 00:58:15 +00:00
  • d7de1209ae Re-enable the messagequeue unittests. These were commented out at one point but never reenabled. decurtis@webrtc.org 2015-01-16 17:52:53 +00:00
  • a1aea10af2 Revert r8076 "Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps." stefan@webrtc.org 2015-01-16 13:52:52 +00:00
  • 4ba1e44ff0 Remove unnecessary remote bitrate estimator build rule which serves no purpose. andresp@webrtc.org 2015-01-16 07:50:17 +00:00
  • 487a444215 Add stats collection for the data channel. decurtis@webrtc.org 2015-01-15 22:55:07 +00:00
  • 357469da5a Fixes reference counting problem when a TransportProxy points to a Transport prior to creating channels. decurtis@webrtc.org 2015-01-15 22:53:49 +00:00
  • ef2a5dd398 Update AppRTCDemo UI. tkchin@webrtc.org 2015-01-15 22:38:21 +00:00
  • 64d3c4b9ac Support 48kHz in AEC aluebs@webrtc.org 2015-01-15 19:52:05 +00:00
  • 89aa276e2e Fix a case where empty candidate id is used guoweis@webrtc.org 2015-01-15 18:52:36 +00:00
  • d82f55d2a7 Only adapt AGC when the desired signal is present aluebs@webrtc.org 2015-01-15 18:07:21 +00:00
  • 3e42a8a56a Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps. stefan@webrtc.org 2015-01-15 14:45:27 +00:00
  • 32e8528581 Log configs when creating video streams in Call. pbos@webrtc.org 2015-01-15 10:09:39 +00:00
  • 1f67b53c88 Remove dual stream functionality in ACM henrik.lundin@webrtc.org 2015-01-15 09:36:30 +00:00
  • 9ce01e6416 Clean unnecessary workaround for chromium import. andresp@webrtc.org 2015-01-15 09:12:45 +00:00
  • 0800db74b9 Add percentage of fec packets and recovered media packets to histogram stats: - "WebRTC.Video.ReceivedFecPacketsInPercent" - "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec" asapersson@webrtc.org 2015-01-15 07:40:20 +00:00
  • 61c1247224 Fix a case where empty candidate id is used guoweis@webrtc.org 2015-01-15 06:53:07 +00:00
  • 6c3855258d Add WebRtcIsacfix_AllpassFilter2FixDec16Neon()'s intrinsics version. andrew@webrtc.org 2015-01-15 02:56:06 +00:00
  • 5a92b78e86 Add beamforming to audioproc_float utility. mgraczyk@chromium.org 2015-01-15 01:28:36 +00:00
  • 6b6301588e Move ring_buffer to common_audio. andrew@webrtc.org 2015-01-15 00:09:53 +00:00
  • fd630a50d2 Add BundlePolicy to RTCConfiguration. Don't change any behavior. Just make it possible to make progress in Chromium while we work on the behavior. pthatcher@webrtc.org 2015-01-14 23:19:06 +00:00
  • 693e01c910 Fix searching for DirectX SDK during GN build. kjellander@webrtc.org 2015-01-14 21:25:25 +00:00
  • f1c8b90520 Remove WebRtcVideoEncoderFactory2. pbos@webrtc.org 2015-01-14 17:29:27 +00:00
  • e5a31e1bf5 Revert removing of compile_assert.h. turaj@webrtc.org 2015-01-14 17:17:11 +00:00
  • 85fa94dff5 Exclude EndToEndTest.SendsAndReceivesH264 for Dr Memory. kjellander@webrtc.org 2015-01-14 17:00:15 +00:00
  • 387841ac5c Improved fairness simulation by starting the flows 20 seconds apart. stefan@webrtc.org 2015-01-14 16:45:29 +00:00
  • f18fba2f7b Implement SimulcastEncoderAdapter support. pbos@webrtc.org 2015-01-14 16:26:23 +00:00
  • 8315d7de85 Remove dual stream functionality in VoiceEngine henrik.lundin@webrtc.org 2015-01-14 16:07:26 +00:00
  • b4e5d1b34e Remove RTX SSRC when deleting the default receive stream. mflodman@webrtc.org 2015-01-14 15:07:07 +00:00
  • 2ebfac5649 Remove COMPILE_ASSERT and use static_assert everywhere kwiberg@webrtc.org 2015-01-14 10:51:54 +00:00
  • 86e1e487e7 Move system_wrappers.gyp files to the proper directory. andresp@webrtc.org 2015-01-14 09:30:52 +00:00
  • a35f741bb0 Add .classpath + talk/app/webrtc/androidtests to .gitignore kjellander@webrtc.org 2015-01-14 09:05:38 +00:00
  • f7a5893f80 Combine RegKeyTests to prevent parallel execution. pbos@webrtc.org 2015-01-14 09:03:16 +00:00
  • ef090927f4 No longer asserting in mocks, split first test case in two methods. phoglund@webrtc.org 2015-01-14 08:56:06 +00:00
  • 69f47381fb Roll chromium_revision 3dd2edf..a6eafec (310717:311223) kjellander@webrtc.org 2015-01-14 06:06:04 +00:00
  • d6e84d9d13 Always copy processed audio to output buffer in ProcessStream. mgraczyk@chromium.org 2015-01-14 01:33:54 +00:00
  • c0da63c707 Optimize minimum delay in blocker aluebs@webrtc.org 2015-01-13 22:28:35 +00:00
  • af9d56f38c Unify the two copies of template_util.h kwiberg@webrtc.org 2015-01-13 20:32:04 +00:00
  • 0b0c24177b Only return Rtx mode in RTXSendStatus(). pbos@webrtc.org 2015-01-13 14:15:15 +00:00
  • 3df38b442f Unify the two copies of compile_assert.h kwiberg@webrtc.org 2015-01-13 11:37:48 +00:00
  • 58a1ba6ffc Roll chromium_revision 271c6cc..3dd2edf (309333:310717) kjellander@webrtc.org 2015-01-13 10:59:21 +00:00
  • 46323b3786 Remove useless AudioProcessing::Create() overload. andrew@webrtc.org 2015-01-13 06:48:06 +00:00
  • 16825b1a82 Use int64_t more consistently for times, in particular for RTT values. pkasting@chromium.org 2015-01-12 21:51:21 +00:00
  • a7add19cf4 audio_processing: Replaced macro WEBRTC_SPL_MUL_16_16 with * in high_pass_filter bjornv@webrtc.org 2015-01-12 21:12:29 +00:00
  • 2a26734f04 Partial revert of r7396 henrik.lundin@webrtc.org 2015-01-12 20:52:21 +00:00
  • be40eb0579 Allow 720x1280 frames encoding on Android. glaznev@webrtc.org 2015-01-12 17:55:47 +00:00
  • a525c98ca5 Fix parallelizability in ApmTests. pbos@webrtc.org 2015-01-12 17:31:18 +00:00
  • 45db7eefa2 Use Java based audio as default for WebRTC. henrika@webrtc.org 2015-01-12 14:27:23 +00:00
  • 81134d019d Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory. In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is specified in the call to CreatePeerConnectionFactory. perkj@webrtc.org 2015-01-12 08:30:16 +00:00
  • 88a4298234 common_audio: Made input vector const in WebRtcSpl_LevinsonDurbin() bjornv@webrtc.org 2015-01-12 05:53:43 +00:00
  • c14e3572c6 common_audio: Made input signal const in WebRtcSplFilterMAFastQ12() bjornv@webrtc.org 2015-01-12 05:50:52 +00:00
  • 19e4e8d751 Add support for trying alternate server (STUN 300 error message) on TCP guoweis@webrtc.org 2015-01-10 02:41:32 +00:00
  • 0ba1533fdb Added support for an Origin header in STUN messages. pthatcher@webrtc.org 2015-01-10 00:47:02 +00:00
  • 2693a54614 Add WEBRTC_BEAMFORMER define to BUILD.gn aluebs@webrtc.org 2015-01-09 23:26:13 +00:00
  • 8f27fcce79 Revert 8028 "Support associated payload type when registering Rt..." andrew@webrtc.org 2015-01-09 20:22:46 +00:00
  • 80452d70cb Sync Android AppRTCDemo with internal repo. glaznev@webrtc.org 2015-01-09 19:34:06 +00:00
  • 9657265f39 Revert "Accept incoming pings before remote answer is set to reduce connection latency." pthatcher@webrtc.org 2015-01-09 19:08:27 +00:00
  • f3fd8e7cdf Add NEON intrinsics version for transform_neon andrew@webrtc.org 2015-01-09 18:29:37 +00:00
  • 1592df78ef PRESUBMIT: Add GN trybots for Windows and Mac. kjellander@webrtc.org 2015-01-09 15:38:29 +00:00
  • 2a169640a3 Support associated payload type when registering Rtx payload type. pbos@webrtc.org 2015-01-09 15:16:10 +00:00
  • 8649fed1b8 GN: Fix Windows build. kjellander@webrtc.org 2015-01-08 21:22:01 +00:00
  • 2ead571fb6 Hard define the GUID for AudioEndpoint to avoid conflicts during compile. decurtis@webrtc.org 2015-01-08 19:18:01 +00:00
  • 758d6d431e audio_processing/aecm: Removed usage of macro WEBRTC_SPL_MUL_16_16 bjornv@webrtc.org 2015-01-08 17:52:56 +00:00
  • dec649cbab audio_processing/ns: Replaced WEBRTC_SPL_MUL_16_16 macro with * bjornv@webrtc.org 2015-01-08 17:34:33 +00:00
  • 5e5b32706a audio_processing/agc: Removed usage of macro WEBRTC_SPL_MUL_16_16 in legacy/agc bjornv@webrtc.org 2015-01-08 17:25:34 +00:00
  • 124b9c70f9 Suppress races in event tracing code. pbos@webrtc.org 2015-01-08 12:38:42 +00:00
  • 5f09564354 Suppress AsyncHttpRequestTest.TestCancel leak for LSan kjellander@webrtc.org 2015-01-08 10:45:59 +00:00