Log - cf85f1cf3cd5ea2127cf318888a147d2afe1d985 - platform/external/webrtc

cf85f1c Reorganize libjingle path variables. by kjellander@webrtc.org · 11 years ago
9f4d212 adding sha1 files for audio classifier test by jan.skoglund@webrtc.org · 11 years ago
3e0b60f Switch to correct interpretation of int and float input data in audio_processing_unittest by bjornv@webrtc.org · 11 years ago
17e4064 Add a deinterleaved float interface to AudioProcessing. by andrew@webrtc.org · 11 years ago
b90991d Update libjingle 62472237->62550414 by henrike@webrtc.org · 11 years ago
7bd4a27 VideoCaptureAndroid: don't deliver frames after stopCapture(). by fischman@webrtc.org · 11 years ago
be50ab6 Including algorithm header to avoid VS2013 breakage by henrik.lundin@webrtc.org · 11 years ago
52e898d Add .bin and .rx files to svn:ignore in resources by kjellander@webrtc.org · 11 years ago
24dae94 Add pthatcher@webrtc.org to talk/OWNERS. by pbos@webrtc.org · 11 years ago
a25a92e Add third_party dependencies to svn:ignore by kjellander@webrtc.org · 11 years ago
db41b4d Remove the deprecated GetStats method from PeerConnectionInterface. by jiayl@webrtc.org · 11 years ago
80bbf4c Enable test SSLStreamAdapterTestDTLS.TestDTLSConnectWithSmallMtu since it does not fail anymore. by jiayl@webrtc.org · 11 years ago
40b3b68 Update libjingle 62364298->62472237 by henrike@webrtc.org · 11 years ago
1bbfb57 Rollback of r5629 "(Auto)update libjingle 62364298-> 62368661". by henrike@webrtc.org · 11 years ago
0117d1c Fix compilation errors under clang 3.5. by pbos@webrtc.org · 11 years ago
31413dc (Auto)update libjingle 62364298-> 62368661 by henrike@webrtc.org · 11 years ago
10adbef Exclude /out* instead of just /out from pylint checks. by fischman@webrtc.org · 11 years ago
2bd5944 Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed. by fischman@webrtc.org · 11 years ago
d3dc424 Remove posting of ICE messages from WebRTCSession in PeerConnection to signaling thread. by mallinath@webrtc.org · 11 years ago
bcfc167 AppRTCDemo(android): don't send local SDP until it's set. by fischman@webrtc.org · 11 years ago
b898ce9 Revert of r5622 "disable unit tests" as it should be fixed in r5623. by henrike@webrtc.org · 11 years ago
b8395eb (Auto)update libjingle 62293974-> 62364298 by henrike@webrtc.org · 11 years ago
eec3843 TSAN only disable of two of libjingle's tests for atomic ops as they are failing for TSAN-bot. by henrike@webrtc.org · 11 years ago
9fd8d87 Adds APIs for reporting pacer queuing delay. by jiayl@webrtc.org · 11 years ago
56e4a05 Remove ProcessingComponent's dependence on AudioProcessingImpl. by andrew@webrtc.org · 11 years ago
806768a (Auto)update libjingle 62281784-> 62293974 by henrike@webrtc.org · 11 years ago
704bf9e (Auto)update libjingle 62063505-> 62278774 by henrike@webrtc.org · 11 years ago
f0fc72f Call PrintWindow for the first time of capturing to capture the window frames correctly. by jiayl@webrtc.org · 11 years ago
00073aa Clean up CPU detection defines in SincResampler a little. by andrew@webrtc.org · 11 years ago
0231e80 Invalidate the whole screen when the frame size is changed. by jiayl@webrtc.org · 11 years ago
2038920 Use scoped_ptr<T[]> in SincResampler to avoid .get()[] weirdness. by andrew@webrtc.org · 11 years ago
c0e9aeb Add SetConfig method to FakeNetworkPipe and to DirectTransport by henrik.lundin@webrtc.org · 11 years ago
eaadeca iOS, AppRTCDemo: Fixes exception due to JSON for ice using "urls" instead of "url", which is introduced by r5599. by braveyao@webrtc.org · 11 years ago
90173e1 Roll libvpx 248011:251850 by marpan@webrtc.org · 11 years ago
bc1d224 Add experimental noise suppression flag to audioproc test by aluebs@webrtc.org · 11 years ago
050892a Missing include in experiments.h by sprang@webrtc.org · 11 years ago
7f52a6e Split the implementation of VP8Encoder|Decoder::Create into a seperated file by wu@webrtc.org · 11 years ago
79a1cff Android, AppRTCDemo: Fixes java exception due to JSON for ice using "urls" instead of "url". by henrike@webrtc.org · 11 years ago
bf88ecc Added turn-prober.sh: a super-simple prober for TURN servers & candidates. by fischman@webrtc.org · 11 years ago
78ea3d5 Check pcConfig (which can be null) before use. by wu@webrtc.org · 11 years ago
91cbaa4 (Auto)update libjingle 61966318-> 62063505 by henrike@webrtc.org · 11 years ago
23caa2d Fix to get total number of sent and received rtcp packets. by asapersson@webrtc.org · 11 years ago
4f0801b AviRecorder is missing a critical section. by braveyao@webrtc.org · 11 years ago
bc0470f AppRTC Sample: Switch AppRTC to use RTCIceServer.urls. by braveyao@webrtc.org · 11 years ago
55fcd71 Disable libjingle_peerconnection_java_unittest by kjellander@webrtc.org · 11 years ago
33af96c Removed unused mock methods in audio_processing by bjornv@webrtc.org · 11 years ago
d43aa9d Update libjingle 61901702->61966318 by henrike@webrtc.org · 11 years ago
a7b9818 Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702). by henrike@webrtc.org · 11 years ago
125a66a Memory and Tsan tests: Turn off the new-ACM tests by tina.legrand@webrtc.org · 11 years ago
ef22151 Revert 5590 "description" by xians@webrtc.org · 11 years ago
0f2809a Add RTCP packet class. Adds packet types: sr, rr, bye, fir. by asapersson@webrtc.org · 11 years ago
c0907ef MIPS optimizations for AEC audio processing module by andrew@webrtc.org · 11 years ago
2643805 description by henrike@webrtc.org · 11 years ago
3f170dd Updated WebRTC version to 3.50 TBR= wu@webrtc.org by elham@webrtc.org · 11 years ago
d617a44 Add an AlignedFreeDeleter and remove scoped_ptr_malloc. by andrew@webrtc.org · 11 years ago
d4d5be8 Minor improvement in RoundToInt16 implementation. by turaj@webrtc.org · 11 years ago
a0a6df3 Modified overuse detection thresholds. by asapersson@webrtc.org · 11 years ago
04a691a Removing a variable that was never read by henrik.lundin@webrtc.org · 11 years ago
6606199 ifdef the alsa code based on macro USE_X11 by fbarchard@google.com · 11 years ago
056176b Presubmit script that prohibits cls to both trunk/webrtc and trunk/talk. by henrike@webrtc.org · 11 years ago
78f0db4 Fix the break caused by r5579. by turaj@webrtc.org · 11 years ago
571df2d Update libjingle 61759961->61834300 by henrike@webrtc.org · 11 years ago
c2d69d3 Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable. by turaj@webrtc.org · 11 years ago
97e7a64 Make WindowCapturerLinux handling window resize events. by jiayl@webrtc.org · 11 years ago
2421025 Added architecture for compiling under chrome NaCl. by andresp@webrtc.org · 11 years ago
056287e 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. by tina.legrand@webrtc.org · 11 years ago
8098e07 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 11 years ago
b7a91fa Removes VoERTP_RTCP::InsertExtraRTPPacket. by henrika@webrtc.org · 11 years ago
e384104 Fix DesktopAndCursorComposer not to crash by sergeyu@chromium.org · 11 years ago
5cf3e8f (Auto)update libjingle $LAST_P10_REVISION-> $NEW_P10_REVISION by henrike@webrtc.org · 11 years ago
27c6980 Move the volume quantization workaround from VoE to AGC. by andrew@webrtc.org · 11 years ago
00844d7 Remove obsolete voe_unit_test. by solenberg@webrtc.org · 11 years ago
358e336 PeerConnection(java): enable HW encoder on N5 for standalone build. by fischman@webrtc.org · 11 years ago
c2d75e0 PeerConnection(java): account for thread shutdown vagaries. by fischman@webrtc.org · 11 years ago
c320027 Don't print a warning if RTPPacketHistory::SetStorePacketStatus is called by mflodman@webrtc.org · 11 years ago
2086e0f Remove unnecessary warnings. by turaj@webrtc.org · 11 years ago
a079233 Remove external encryption API for VoE. by solenberg@webrtc.org · 11 years ago
0a9d822 Change mime type to text/html for multiple-relay.html by kjellander@webrtc.org · 11 years ago
346094c Incorrect overhead calculation when using FEC + RTP extension headers. by sprang@webrtc.org · 11 years ago
b60346e Reset estimate if no frame has been seen for a certain time (to avoid large jitter if stop sending). by asapersson@webrtc.org · 11 years ago
92fdfeb Update talk to 61699344. by mallinath@webrtc.org · 11 years ago
e384289 Adding tsan suppression for error introduced in r5555, causing libjingle_unittest to fail on TSan bot. by mflodman@webrtc.org · 11 years ago
340746a Misc small nits in NetEq by henrik.lundin@webrtc.org · 11 years ago
1009798 Demo of multi-pass encode - used for testing limits. by hta@webrtc.org · 11 years ago
f92aaff AudioProcessing is not a Module. by andrew@webrtc.org · 11 years ago
b8c254a (Auto)update libjingle 61549749-> 61608469 by henrike@webrtc.org · 11 years ago
e2fc13e Refactoring common_audio/signal_processing: Removed two macros used by isac only. by bjornv@webrtc.org · 11 years ago
c5d506a AppRTCDemo(android): clarified README on how to launch app using adb. by fischman@webrtc.org · 11 years ago
505f2a0 Disabling WebRtcSessionTest.TestIceStatesBundle under memcheck. by stefan@webrtc.org · 11 years ago
9075d51 Adding a critical section missing in r5543. by stefan@webrtc.org · 11 years ago
a3708ec PeerConnectionTest(java): unbreak following 61460797-p10 by fischman@webrtc.org · 11 years ago
385857d Update talk to 61549749. by mallinath@webrtc.org · 11 years ago
b9a088b Update talk to 61538839. by wu@webrtc.org · 11 years ago
0de2950 Revert 5545 "Update libjingle to 61514460" by wu@webrtc.org · 11 years ago
38bf249 Initialize output_will_be_muted_. by andrew@webrtc.org · 11 years ago
e749c9e Update libjingle to 61514460 by xians@webrtc.org · 11 years ago
8f690bc Increase overuse and normal use thresholds for Mac. by asapersson@webrtc.org · 11 years ago
ae2563a Fixes a race when writing to send_padding_. by stefan@webrtc.org · 11 years ago
12cb88c Add check to verify tree is open to PRESUBMIT.py. by kjellander@webrtc.org · 11 years ago
fcfc6a9 Small refactoring of NetEq unittest for CNG with clock drift by henrik.lundin@webrtc.org · 11 years ago