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  • 37c2976511 Samples, add IPv6 supporting into Apprtc demo. braveyao@webrtc.org 2014-01-27 03:08:16 +00:00
  • e84978f3d8 Add a Config parameter to AudioProcessing::Create(). andrew@webrtc.org 2014-01-25 02:09:06 +00:00
  • 256d0ada35 Remove the check for audio codec num in WebRtcVoiceEngineTest.HasCorrectCodecs. wu@webrtc.org 2014-01-24 18:58:51 +00:00
  • 57f6c10d00 Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera. henrike@webrtc.org 2014-01-24 18:42:12 +00:00
  • 98aefcd8fe Update tsan suppressions for libjingle_media_unittest. wu@webrtc.org 2014-01-24 18:32:43 +00:00
  • ca5ff9972e Re-enable webrtcvoice/videoengine unittests. wu@webrtc.org 2014-01-24 17:37:46 +00:00
  • 871d949299 Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtcp modules. asapersson@webrtc.org 2014-01-24 13:23:49 +00:00
  • 24999d44c2 Allow ?audio=false&video=false to be used in combination to instantiate a recv-only client. andresp@webrtc.org 2014-01-24 12:25:50 +00:00
  • fd0f267bb1 Add new API (webrtc.gyp:webrtc) to merge_libs.gyp. pbos@webrtc.org 2014-01-24 10:43:47 +00:00
  • 99a8c7e039 Add trace-based delivery filter to BWE test framework. stefan@webrtc.org 2014-01-24 10:00:27 +00:00
  • c279a5d72c Wire up RTX in VideoReceiveStream. pbos@webrtc.org 2014-01-24 09:30:53 +00:00
  • 8d375c95b7 Fix deadlock on register/unregister observer while there is a an going callback. andresp@webrtc.org 2014-01-23 23:09:25 +00:00
  • a8910d2f88 Update talk to 60094938. wu@webrtc.org 2014-01-23 22:12:45 +00:00
  • 754de528b7 Fix array declarations in aec_rdft.h. andrew@webrtc.org 2014-01-23 20:55:14 +00:00
  • e7223e7795 Set NACKed packet to -1 in TestNackRetransmission. pbos@webrtc.org 2014-01-23 16:14:34 +00:00
  • 0e93257cee Add callbacks for receive channel RTP statistics sprang@webrtc.org 2014-01-23 10:00:39 +00:00
  • 91db93d24f Android, fixes crash on devices with only front cameras. henrike@webrtc.org 2014-01-22 21:31:24 +00:00
  • 570bc3d792 Make new baremetal trybots receive tryjobs by default. kjellander@webrtc.org 2014-01-22 19:00:01 +00:00
  • 0d92ef67c4 Libjingle source code has some spelling mistakes and one of them is "renegotation", which should be "renegotiation". mallinath@webrtc.org 2014-01-22 02:21:22 +00:00
  • 68cbd01216 enabling disabled data channels tests on win32. The real culprit was that ice candidates not included in SDP when there were failure causing transport channels never becoming writable. mallinath@webrtc.org 2014-01-22 00:16:46 +00:00
  • 7de3bb9df9 Output logs to stderr from voe_cmd_test by default. andrew@webrtc.org 2014-01-21 22:17:43 +00:00
  • 28da47c52f Android example apps: fixes issue where useful failure information was suppressed. henrike@webrtc.org 2014-01-21 19:03:51 +00:00
  • 1d2c034861 libyuv use extern c around jpeg includes. includes fixes to gyp build for intel/mips android, cros arm, ios, and pnacl. BUG=none TESTED=try bots R=andrew@webrtc.org, jzern@chromium.org fbarchard@google.com 2014-01-21 18:27:05 +00:00
  • 7dba27c740 Potential dead lock in receive statistics sprang@webrtc.org 2014-01-21 16:33:37 +00:00
  • 32c3247418 Fix for libtalkmobile build error bug=b/12549061 elham@webrtc.org 2014-01-21 16:16:58 +00:00
  • 7ef7df57d8 Removes script for generating supplement.gypi also adds git ignore for tools/gn. henrike@webrtc.org 2014-01-21 15:54:56 +00:00
  • e02d47515f Set up receiver RTX config using a std::map. pbos@webrtc.org 2014-01-20 14:43:55 +00:00
  • efaeda0c76 Add configuration and test for extended RTCP reference time reports to new video api. asapersson@webrtc.org 2014-01-20 08:34:49 +00:00
  • 32c26eb90b Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback henrike@webrtc.org 2014-01-17 23:12:51 +00:00
  • 4985927d36 Implement screen enumeration and individual screen capturing for Windows. jiayl@webrtc.org 2014-01-17 17:19:16 +00:00
  • ead202b973 Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started. henrike@webrtc.org 2014-01-16 23:26:37 +00:00
  • 2ce9a64b75 Talk: Removes deprecated example apps and moves the server apps to trunk/talk/examples. henrike@webrtc.org 2014-01-16 16:49:53 +00:00
  • 0af1ffa84d Android, WebRTCDemo: fix issue where changing remote IP was not working properly. henrike@webrtc.org 2014-01-16 16:45:15 +00:00
  • 4ffd9c7423 Add full path to headers aluebs@webrtc.org 2014-01-16 09:01:39 +00:00
  • 6a94734d4d Adds back set_sample_rate_hz() when Init is called in recordings. bjornv@webrtc.org 2014-01-16 08:41:09 +00:00
  • ea9392d5eb MIPS optimizations for NS audio processing module andrew@webrtc.org 2014-01-16 07:22:01 +00:00
  • fb4e256d49 Fix crash in MouseCursor::CopyOf() sergeyu@chromium.org 2014-01-16 04:45:35 +00:00
  • 8f35afab8c Exclude protoc objects from merge_libs.py. andrew@webrtc.org 2014-01-16 00:31:57 +00:00
  • 4b26e2eee3 Update libjingle to 59676287 sergeyu@chromium.org 2014-01-15 23:15:54 +00:00
  • 7a2ca7c621 Update needed to MockScreenCapturer after new methods addition to webrtc::ScreenCapturer. This change is also must for rolling webrtc in chrome. mallinath@webrtc.org 2014-01-15 19:00:13 +00:00
  • 8f19cb9fbc Revert 5387 "Re-enable webrtcvoice/videoengine unittests." wu@webrtc.org 2014-01-14 22:31:11 +00:00
  • eda6823397 Re-enable webrtcvoice/videoengine unittests. wu@webrtc.org 2014-01-14 22:15:09 +00:00
  • 017b619010 Extends the ScreenCapturer interface for individual display screen cast. Real implementations for each platform will be added in future CLs. jiayl@webrtc.org 2014-01-14 18:26:37 +00:00
  • 03cfde2d10 Roll Chromium 238260 -> 243863 wjia@webrtc.org 2014-01-14 17:48:34 +00:00
  • 8c5b27de9a Allow to skip turn by passing ts=false to apprtc. andresp@webrtc.org 2014-01-14 17:00:23 +00:00
  • 39fcfd78ae Remove empty VideoCodecGeneric struct. pbos@webrtc.org 2014-01-14 12:55:59 +00:00
  • d9faa46d57 Changing to using factory methods for some classes in NetEq henrik.lundin@webrtc.org 2014-01-14 10:18:45 +00:00
  • aebb1ade9d pRevert 5371 "Revert 5367 "Update talk to 59410372."" henrika@webrtc.org 2014-01-14 10:00:58 +00:00
  • 4371d4650a Temporarily disabling some more audio processing tests. aluebs@webrtc.org 2014-01-14 08:57:22 +00:00
  • eb31b45aaf Fix MouseCursorMonitorMac to return correct hotspot position. sergeyu@chromium.org 2014-01-13 23:25:17 +00:00
  • 3907c2e7e5 Removes the remaining uses of the list wrapper class and the list wrapper class. henrike@webrtc.org 2014-01-13 22:41:34 +00:00
  • dde7aee40f WebRTCDemo: fix out-of-bounds array read. fischman@webrtc.org 2014-01-13 22:15:38 +00:00
  • d7568a08c3 PeerConnection(java): Add OnRenegotiationNeeded support fischman@webrtc.org 2014-01-13 22:04:12 +00:00
  • ad1863de74 Updated Webrtc version to 3.49 elham@webrtc.org 2014-01-13 17:49:49 +00:00
  • 79cf3acc79 Removes usage of ListWrapper from several files. henrike@webrtc.org 2014-01-13 15:21:30 +00:00
  • d0b436a935 Revert "Activate ACM test for Android in modules_tests." (rev5364). andresp@webrtc.org 2014-01-13 13:15:59 +00:00
  • 44461fa5cb Revert 5367 "Update talk to 59410372." henrika@webrtc.org 2014-01-13 09:35:02 +00:00
  • 8bc4fcfeb6 Temporarily disabling audio processing tests. aluebs@webrtc.org 2014-01-13 09:14:47 +00:00
  • 2c03bf1641 Increasing simulation time for NetEqPerformanceTest henrik.lundin@webrtc.org 2014-01-13 09:04:23 +00:00
  • bbd47fc5b5 Enables robust delay validation in AEC delay logging. bjornv@webrtc.org 2014-01-13 08:54:34 +00:00
  • 0f3356e20b Update talk to 59410372. mallinath@webrtc.org 2014-01-11 01:26:23 +00:00
  • 023cc5abc7 Minor voice engine improvements around AGC. andrew@webrtc.org 2014-01-11 01:25:53 +00:00
  • 573a1b45b5 Android: Fixes crash when exiting WebRTCDemo. henrike@webrtc.org 2014-01-10 22:58:06 +00:00
  • 7cc64b3747 Activate ACM test for Android in modules_tests. turaj@webrtc.org 2014-01-10 22:35:09 +00:00
  • f777cf2547 Permitting double start/stopping of streams. pbos@webrtc.org 2014-01-10 18:47:32 +00:00
  • a366e810a9 Adding NetEq performance test to webrtc_perf_tests henrik.lundin@webrtc.org 2014-01-10 08:24:04 +00:00
  • fa8d534e09 Delay Estimator: Adds unittests for robust validation. bjornv@webrtc.org 2014-01-10 07:42:07 +00:00
  • 4625df3e3e Fix NaCl compilation sergeyu@chromium.org 2014-01-09 21:26:50 +00:00
  • e7ce437333 Fixing lint errors in NetEq4 henrik.lundin@webrtc.org 2014-01-09 14:01:55 +00:00
  • c5aeb2aa15 Make code simpler on VCMEncodedCallback. andresp@webrtc.org 2014-01-09 08:04:32 +00:00
  • 1df9dc3957 Isolate register post encode callback in video coding module to simplify code and critical sections. andresp@webrtc.org 2014-01-09 08:01:57 +00:00
  • bb0de3ca9f Updated Demos so they work on FF, changed the third argument in CreateOffer to null as it doesnot really require sdpConstraints. vikasmarwaha@webrtc.org 2014-01-09 00:51:19 +00:00
  • 4177615e87 PeerConnection(java): replace ScopedLocalRef with ScopedLocalRefFrame and fix a local reference leak in OnMessage. fischman@webrtc.org 2014-01-09 00:31:17 +00:00
  • 1794693ec8 AppRTCDemo(android): close() the throw-away DataChannel. fischman@webrtc.org 2014-01-08 18:29:34 +00:00
  • b08a12d6e8 Isolate debug recording from video sender into a thread safe small class. andresp@webrtc.org 2014-01-08 12:38:22 +00:00
  • ab2405164a Add another test case for AST/TOF switching. solenberg@webrtc.org 2014-01-08 08:59:44 +00:00
  • bccd53de57 Delay Estimator: Converts a constant into a configurable parameter. bjornv@webrtc.org 2014-01-08 08:18:15 +00:00
  • e00265ed49 Fix a compile error on Android on sctpdataengine.cc. wu@webrtc.org 2014-01-07 19:32:40 +00:00
  • d335094852 Init to 16 kHz in the fixed-point profile. andrew@webrtc.org 2014-01-07 18:57:10 +00:00
  • b6541ca3a1 Ensure capture_levels_ is sized correctly at init time. andrew@webrtc.org 2014-01-07 18:36:10 +00:00
  • cf9d364063 Now printing less output from compare_videos.py. phoglund@webrtc.org 2014-01-07 17:59:30 +00:00
  • 60730cfe3c Remove the requirement to call set_sample_rate_hz and friends. andrew@webrtc.org 2014-01-07 17:45:09 +00:00
  • 39669c5c8f Remove outdated DestroyVideoSendStream comment. pbos@webrtc.org 2014-01-07 12:27:22 +00:00
  • ccd42840bc Wire up statistics in video send stream of new video engine api sprang@webrtc.org 2014-01-07 09:54:34 +00:00
  • 0b7d8e6fcb AppRTC: Alert the user to failure to acquire TURN server. fischman@webrtc.org 2014-01-06 23:46:53 +00:00
  • acc05ac7d1 Roll libvpx 232686:241571 marpan@webrtc.org 2014-01-06 21:04:22 +00:00
  • a9bdee6690 Add Christophe Dumez to AUTHORS. andrew@webrtc.org 2014-01-06 19:43:21 +00:00
  • 7bdaf837d4 Updated PeerConnection samples so they run on FF. vikasmarwaha@webrtc.org 2014-01-03 23:13:01 +00:00
  • f6d6ed0c66 Update talk to 59039880. wu@webrtc.org 2014-01-03 22:08:47 +00:00
  • e667234ee2 libyuv r949 includes changes to allow any width, mainly relating to fixed point math overflows. BUG=none TEST=try bots R=ronghuawu@google.com fbarchard@google.com 2014-01-03 18:57:22 +00:00
  • a89d17d5b7 Delay Estimator: robust_validation should be stored over a reset bjornv@webrtc.org 2014-01-02 07:07:04 +00:00
  • 2240763ec2 libyuv r930 for RGB24ToUV_NEON improved color accuracy to avoid red tint, and use malloc with variable sized row buffers to avoid stack overflow and relax width restrictions. Previously was limited to 4k on x86 and 1080p on arm. In practice the new limitation is 32767 pixels wide. BUG=none TESTED=try bots R=tpsiaki@google.com, wjia@webrtc.org fbarchard@google.com 2013-12-28 07:00:18 +00:00
  • 2fb72cfeec Add include guards to forward_error_correction_internal.h braveyao@webrtc.org 2013-12-24 05:06:12 +00:00
  • 0062a6d099 Fix the include guard in transmit_mixer.h braveyao@webrtc.org 2013-12-24 03:58:51 +00:00
  • a7cfa6704a Fix the include guard in transmit_mixer.h braveyao@webrtc.org 2013-12-24 03:39:10 +00:00
  • 000dde99c8 Android build: make it quiet on success and not overly noisy on failure. fischman@webrtc.org 2013-12-20 22:49:35 +00:00
  • a63fc87139 Fix JS error in adapter.js for FF for the case when ?transport=xxx is missing in TURN url. BUG=2737 R=juberti@webrtc.org vikasmarwaha@webrtc.org 2013-12-20 22:10:17 +00:00
  • f6acf98a46 Fix the android clang bot for compiling with thread annotations. andresp@webrtc.org 2013-12-20 21:54:26 +00:00
  • cf2b3acc48 Update Android trybots in the default try job list. kjellander@webrtc.org 2013-12-20 21:20:42 +00:00
  • 7fb75ecbd4 Add thread_annotations for clang targets. andresp@webrtc.org 2013-12-20 20:20:50 +00:00