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  • be50ab645a Including algorithm header to avoid VS2013 breakage henrik.lundin@webrtc.org 2014-03-04 15:10:03 +00:00
  • 52e898d7b9 Add .bin and .rx files to svn:ignore in resources kjellander@webrtc.org 2014-03-04 06:49:52 +00:00
  • 24dae9419a Add pthatcher@webrtc.org to talk/OWNERS. pbos@webrtc.org 2014-03-04 05:58:25 +00:00
  • a25a92e107 Add third_party dependencies to svn:ignore kjellander@webrtc.org 2014-03-04 02:12:28 +00:00
  • db41b4dbcd Remove the deprecated GetStats method from PeerConnectionInterface. jiayl@webrtc.org 2014-03-03 21:30:06 +00:00
  • 80bbf4c312 Enable test SSLStreamAdapterTestDTLS.TestDTLSConnectWithSmallMtu since it does not fail anymore. jiayl@webrtc.org 2014-03-03 18:41:27 +00:00
  • 40b3b68cdf Update libjingle 62364298->62472237 henrike@webrtc.org 2014-03-03 18:30:11 +00:00
  • 1bbfb57d71 Rollback of r5629 "(Auto)update libjingle 62364298-> 62368661". henrike@webrtc.org 2014-03-03 17:37:52 +00:00
  • 0117d1c48c Fix compilation errors under clang 3.5. pbos@webrtc.org 2014-03-03 16:47:03 +00:00
  • 31413dc635 (Auto)update libjingle 62364298-> 62368661 henrike@webrtc.org 2014-03-03 16:47:01 +00:00
  • 10adbeff78 Exclude /out* instead of just /out from pylint checks. fischman@webrtc.org 2014-03-01 02:09:36 +00:00
  • 2bd5944144 Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed. fischman@webrtc.org 2014-03-01 00:07:08 +00:00
  • d3dc424fe5 Remove posting of ICE messages from WebRTCSession in PeerConnection to signaling thread. These callbacks are called from signal thread already. There is no point in posting messages on the same thread again. mallinath@webrtc.org 2014-03-01 00:05:52 +00:00
  • bcfc1670d6 AppRTCDemo(android): don't send local SDP until it's set. fischman@webrtc.org 2014-03-01 00:02:27 +00:00
  • b898ce9227 Revert of r5622 "disable unit tests" as it should be fixed in r5623. henrike@webrtc.org 2014-02-28 23:18:44 +00:00
  • b8395ebe14 (Auto)update libjingle 62293974-> 62364298 henrike@webrtc.org 2014-02-28 21:57:22 +00:00
  • eec3843596 TSAN only disable of two of libjingle's tests for atomic ops as they are failing for TSAN-bot. henrike@webrtc.org 2014-02-28 18:06:42 +00:00
  • 9fd8d87ff5 Adds APIs for reporting pacer queuing delay. jiayl@webrtc.org 2014-02-27 22:32:40 +00:00
  • 56e4a05053 Remove ProcessingComponent's dependence on AudioProcessingImpl. andrew@webrtc.org 2014-02-27 22:23:17 +00:00
  • 806768a6ca (Auto)update libjingle 62281784-> 62293974 henrike@webrtc.org 2014-02-27 21:03:09 +00:00
  • 704bf9ebec (Auto)update libjingle 62063505-> 62278774 henrike@webrtc.org 2014-02-27 17:52:04 +00:00
  • f0fc72f70e Call PrintWindow for the first time of capturing to capture the window frames correctly. This will fix artifacts on the captured window frames, especially for cmd, which sometimes leaks glimpss of other window's content. jiayl@webrtc.org 2014-02-27 16:43:12 +00:00
  • 00073aafa8 Clean up CPU detection defines in SincResampler a little. andrew@webrtc.org 2014-02-27 04:12:34 +00:00
  • 0231e801d6 Invalidate the whole screen when the frame size is changed. Otherwise we'll compare frames of different sizes and read into invalid memory. jiayl@webrtc.org 2014-02-26 18:54:57 +00:00
  • 2038920a2b Use scoped_ptr<T[]> in SincResampler to avoid .get()[] weirdness. andrew@webrtc.org 2014-02-26 18:14:54 +00:00
  • c0e9aebe8f Add SetConfig method to FakeNetworkPipe and to DirectTransport henrik.lundin@webrtc.org 2014-02-26 13:34:52 +00:00
  • eaadecaf98 iOS, AppRTCDemo: Fixes exception due to JSON for ice using "urls" instead of "url", which is introduced by r5599. braveyao@webrtc.org 2014-02-26 04:16:02 +00:00
  • 90173e188f Roll libvpx 248011:251850 marpan@webrtc.org 2014-02-25 22:09:02 +00:00
  • bc1d22461b Add experimental noise suppression flag to audioproc test aluebs@webrtc.org 2014-02-25 16:50:22 +00:00
  • 050892a95b Missing include in experiments.h sprang@webrtc.org 2014-02-25 09:17:43 +00:00
  • 7f52a6ef2b Split the implementation of VP8Encoder|Decoder::Create into a seperated file (vp8_factory.cc). wu@webrtc.org 2014-02-24 23:56:39 +00:00
  • 79a1cff65a Android, AppRTCDemo: Fixes java exception due to JSON for ice using "urls" instead of "url". henrike@webrtc.org 2014-02-24 23:22:18 +00:00
  • bf88eccf33 Added turn-prober.sh: a super-simple prober for TURN servers & candidates. fischman@webrtc.org 2014-02-24 21:52:59 +00:00
  • 78ea3d50e0 Check pcConfig (which can be null) before use. wu@webrtc.org 2014-02-24 21:51:58 +00:00
  • 91cbaa477c (Auto)update libjingle 61966318-> 62063505 henrike@webrtc.org 2014-02-24 16:51:32 +00:00
  • 23caa2d8d6 Fix to get total number of sent and received rtcp packets. asapersson@webrtc.org 2014-02-24 09:27:38 +00:00
  • 4f0801bd39 AviRecorder is missing a critical section. braveyao@webrtc.org 2014-02-24 09:19:36 +00:00
  • bc0470f559 AppRTC Sample: Switch AppRTC to use RTCIceServer.urls. braveyao@webrtc.org 2014-02-24 03:43:03 +00:00
  • 55fcd716f3 Disable libjingle_peerconnection_java_unittest kjellander@webrtc.org 2014-02-23 18:47:27 +00:00
  • 33af96c5c2 Removed unused mock methods in audio_processing bjornv@webrtc.org 2014-02-21 23:56:05 +00:00
  • d43aa9de7a Update libjingle 61901702->61966318 henrike@webrtc.org 2014-02-21 23:43:24 +00:00
  • a7b981843f Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702). henrike@webrtc.org 2014-02-21 15:51:43 +00:00
  • 125a66adc2 Memory and Tsan tests: Turn off the new-ACM tests tina.legrand@webrtc.org 2014-02-21 14:12:00 +00:00
  • ef2215110c Revert 5590 "description" xians@webrtc.org 2014-02-21 10:31:29 +00:00
  • 0f2809a5ac Add RTCP packet class. Adds packet types: sr, rr, bye, fir. asapersson@webrtc.org 2014-02-21 08:14:45 +00:00
  • c0907eff42 MIPS optimizations for AEC audio processing module andrew@webrtc.org 2014-02-21 00:13:31 +00:00
  • 2643805a20 description henrike@webrtc.org 2014-02-20 22:32:53 +00:00
  • 3f170dd309 Updated WebRTC version to 3.50 TBR= wu@webrtc.org elham@webrtc.org 2014-02-20 22:31:07 +00:00
  • d617a44a4f Add an AlignedFreeDeleter and remove scoped_ptr_malloc. andrew@webrtc.org 2014-02-20 21:08:36 +00:00
  • d4d5be8781 Minor improvement in RoundToInt16 implementation. turaj@webrtc.org 2014-02-20 20:55:21 +00:00
  • a0a6df3910 Modified overuse detection thresholds. asapersson@webrtc.org 2014-02-20 17:37:37 +00:00
  • 04a691adac Removing a variable that was never read henrik.lundin@webrtc.org 2014-02-20 15:27:00 +00:00
  • 66061992fb ifdef the alsa code based on macro USE_X11 BUG=none TEST=try bots R=andrew@webrtc.org fbarchard@google.com 2014-02-20 03:05:05 +00:00
  • 056176b962 Presubmit script that prohibits cls to both trunk/webrtc and trunk/talk. henrike@webrtc.org 2014-02-19 23:18:19 +00:00
  • 78f0db4710 Fix the break caused by r5579. turaj@webrtc.org 2014-02-19 23:07:31 +00:00
  • 571df2dca9 Update libjingle 61759961->61834300 henrike@webrtc.org 2014-02-19 23:04:26 +00:00
  • c2d69d3229 Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable. turaj@webrtc.org 2014-02-19 20:31:17 +00:00
  • 97e7a640d8 Make WindowCapturerLinux handling window resize events. We need to re-initialize the XServerPixelBuffer to the new size when a window resize event is received. jiayl@webrtc.org 2014-02-19 17:28:41 +00:00
  • 242102517d Added architecture for compiling under chrome NaCl. andresp@webrtc.org 2014-02-19 13:55:02 +00:00
  • 056287eee0 This CL separate all ACM tests with new and old implementation of ACM and NetEq. The reason is to debug an issue with failure on Android try bots. We need to see if the error only occurs with the new ACM/NetEq, or if it is a flakiness that affects both. tina.legrand@webrtc.org 2014-02-19 13:45:54 +00:00
  • 8098e07478 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). Add counter to RTCP sender and RTCP receiver. Add video api GetRtcpPacketTypes(). asapersson@webrtc.org 2014-02-19 11:59:02 +00:00
  • b7a91fa95a Removes VoERTP_RTCP::InsertExtraRTPPacket. henrika@webrtc.org 2014-02-19 08:58:08 +00:00
  • e384104166 Fix DesktopAndCursorComposer not to crash sergeyu@chromium.org 2014-02-18 23:26:34 +00:00
  • 5cf3e8f0f0 (Auto)update libjingle $LAST_P10_REVISION-> $NEW_P10_REVISION henrike@webrtc.org 2014-02-18 22:28:52 +00:00
  • 27c6980239 Move the volume quantization workaround from VoE to AGC. andrew@webrtc.org 2014-02-18 20:24:56 +00:00
  • 00844d7bef Remove obsolete voe_unit_test. solenberg@webrtc.org 2014-02-18 18:50:50 +00:00
  • 358e3367a3 PeerConnection(java): enable HW encoder on N5 for standalone build. fischman@webrtc.org 2014-02-18 17:29:37 +00:00
  • c2d75e0708 PeerConnection(java): account for thread shutdown vagaries. fischman@webrtc.org 2014-02-18 16:57:36 +00:00
  • c320027d6a Don't print a warning if RTPPacketHistory::SetStorePacketStatus is called twice with the same settings. mflodman@webrtc.org 2014-02-18 14:51:00 +00:00
  • 2086e0fbf3 Remove unnecessary warnings. turaj@webrtc.org 2014-02-18 14:22:20 +00:00
  • a07923339b Remove external encryption API for VoE. solenberg@webrtc.org 2014-02-18 11:27:22 +00:00
  • 0a9d822812 Change mime type to text/html for multiple-relay.html kjellander@webrtc.org 2014-02-18 08:45:13 +00:00
  • 346094cb01 Incorrect overhead calculation when using FEC + RTP extension headers. sprang@webrtc.org 2014-02-18 08:40:33 +00:00
  • b60346e951 Reset estimate if no frame has been seen for a certain time (to avoid large jitter if stop sending). Add delay before start processing after a reset. asapersson@webrtc.org 2014-02-17 19:02:15 +00:00
  • 92fdfebedd Update talk to 61699344. mallinath@webrtc.org 2014-02-17 18:49:41 +00:00
  • e3842897e2 Adding tsan suppression for error introduced in r5555, causing libjingle_unittest to fail on TSan bot. mflodman@webrtc.org 2014-02-17 15:09:39 +00:00
  • 340746aa13 Misc small nits in NetEq henrik.lundin@webrtc.org 2014-02-17 11:37:16 +00:00
  • 1009798b31 Demo of multi-pass encode - used for testing limits. hta@webrtc.org 2014-02-15 06:13:41 +00:00
  • f92aaff104 AudioProcessing is not a Module. andrew@webrtc.org 2014-02-15 04:22:49 +00:00
  • b8c254abd6 (Auto)update libjingle 61549749-> 61608469 henrike@webrtc.org 2014-02-14 23:38:45 +00:00
  • e2fc13e42f Refactoring common_audio/signal_processing: Removed two macros used by isac only. bjornv@webrtc.org 2014-02-14 23:12:34 +00:00
  • c5d506a106 AppRTCDemo(android): clarified README on how to launch app using adb. fischman@webrtc.org 2014-02-14 17:55:13 +00:00
  • 505f2a0348 Disabling WebRtcSessionTest.TestIceStatesBundle under memcheck. stefan@webrtc.org 2014-02-14 12:38:06 +00:00
  • 9075d519a2 Adding a critical section missing in r5543. stefan@webrtc.org 2014-02-14 09:45:58 +00:00
  • a3708ecdfe PeerConnectionTest(java): unbreak following 61460797-p10 fischman@webrtc.org 2014-02-14 01:51:33 +00:00
  • 385857dfd4 Update talk to 61549749. mallinath@webrtc.org 2014-02-14 00:56:12 +00:00
  • b9a088b920 Update talk to 61538839. wu@webrtc.org 2014-02-13 23:18:49 +00:00
  • 0de29504ab Revert 5545 "Update libjingle to 61514460" wu@webrtc.org 2014-02-13 19:54:28 +00:00
  • 38bf249049 Initialize output_will_be_muted_. andrew@webrtc.org 2014-02-13 17:43:44 +00:00
  • e749c9ebdb Update libjingle to 61514460 xians@webrtc.org 2014-02-13 15:09:40 +00:00
  • 8f690bc222 Increase overuse and normal use thresholds for Mac. asapersson@webrtc.org 2014-02-13 14:43:18 +00:00
  • ae2563ae2f Fixes a race when writing to send_padding_. stefan@webrtc.org 2014-02-13 13:48:38 +00:00
  • 12cb88cab9 Add check to verify tree is open to PRESUBMIT.py. kjellander@webrtc.org 2014-02-13 11:53:43 +00:00
  • fcfc6a990e Small refactoring of NetEq unittest for CNG with clock drift henrik.lundin@webrtc.org 2014-02-13 11:42:28 +00:00
  • 3eda643a91 PeerConnection(java): added MediaConstraints support to AudioSource, now fed to AudioTrack. fischman@webrtc.org 2014-02-13 04:01:04 +00:00
  • 540acde5b3 PeerConnection(java): use MediaCodec for HW-accelerated video encode where available. fischman@webrtc.org 2014-02-13 03:56:14 +00:00
  • 17342e5092 Add a method to inform AudioProcessing that its output will be muted. andrew@webrtc.org 2014-02-12 22:28:31 +00:00
  • de782180b0 Change the type of propagation delta from int64 to int. The delta value never exceeds the range of int. Changing it to int will save memory and copying cost. jiayl@webrtc.org 2014-02-12 19:19:23 +00:00
  • 07b5950c12 Initialize key_pressed_. andrew@webrtc.org 2014-02-12 16:41:13 +00:00
  • ce8e077cf0 Add a keypress field to the audioproc debug proto. andrew@webrtc.org 2014-02-12 15:28:30 +00:00