Log - 7d825e9b2c26e92e7e866c61f5e2bd6f68d7f904 - platform/external/webrtc

7d825e9 Revert "libjingle_unittest now compiles and passes on iOS!" by kjellander@webrtc.org · 11 years ago
c891577 Disable flaky CaptureNtpTimeWithNetworkJitter. by pbos@webrtc.org · 11 years ago
494aa0e AEC: Moved delay buffer size enums from aec_core.h to aec_core_internal.h by bjornv@webrtc.org · 11 years ago
8dfe8ff Disable capture test for FrameRate on Windows. by pbos@webrtc.org · 11 years ago
e772c71 Introduce a config struct for AudioCoding module by henrik.lundin@webrtc.org · 11 years ago
f043f79 Disabling flaky CanReceiveFec. by pbos@webrtc.org · 11 years ago
69e9950 Disable flaky RunsRtpRtcpTestWIthoutErrors. by pbos@webrtc.org · 11 years ago
12a3424 Fix the NetEq build by henrik.lundin@webrtc.org · 11 years ago
116ed1d Include buffer size limits in NetEq config struct by henrik.lundin@webrtc.org · 11 years ago
b08bbf5 Add henrik.lundin as owner in AudioCoding module by henrik.lundin@webrtc.org · 11 years ago
a0d3067 Use CreatePeerConnection method which accepts port_allocator. by mallinath@webrtc.org · 11 years ago
95cd155 libjingle_unittest now compiles and passes on iOS! by fischman@webrtc.org · 11 years ago
8f69330 Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 11 years ago
658a945 (Auto)update libjingle 65619249-> 65622932 by buildbot@webrtc.org · 11 years ago
ff90ed6 (Auto)update libjingle 65561104-> 65619249 by buildbot@webrtc.org · 11 years ago
2eceb8e Roll third_party/opus 258909:262302 by andrew@webrtc.org · 11 years ago
0175d76 Fix leak in remote bitrate estimator tests introduced in r5980 by stefan@webrtc.org · 11 years ago
4f616a0 Support for simulating multiple independent flows in a network. by stefan@webrtc.org · 11 years ago
46106f2 Casting char to int in logs. by asapersson@webrtc.org · 11 years ago
2b93402 (Auto)update libjingle 65484212-> 65561104 by buildbot@webrtc.org · 11 years ago
cc1ba15 Returns a NULL frame on all platforms if the captured window is closed. by jiayl@webrtc.org · 11 years ago
cd70119 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 11 years ago
93fd25c * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. by wu@webrtc.org · 11 years ago
439a4c4 Add an output capacity parameter to ACMResampler::Resample10Msec() by henrik.lundin@webrtc.org · 11 years ago
103657b Add keyboard channel support to AudioBuffer. by andrew@webrtc.org · 11 years ago
d57b814 Fix the Android compilation (better structure for NetEq test libs) by henrik.lundin@webrtc.org · 11 years ago
5ca6a53 Remove TraceCallback use from Call. by pbos@webrtc.org · 11 years ago
a5c8d2c Rename Start/Stop in Video{Send,Receive}Streams. by pbos@webrtc.org · 11 years ago
0a22774 Fixing a bug in ACM2 where the output frame energy was incorrectly set by henrik.lundin@webrtc.org · 11 years ago
f26c9e8 Use unique filenames in AudioProcessingTests for parallelization. by andrew@webrtc.org · 11 years ago
3f1aa24 (Auto)update libjingle 65469804-> 65484212 by buildbot@webrtc.org · 11 years ago
0d915ff Fix the return value of DtlsTransportChannelWrapper::SendPacket in the case of invalid RTP packet. by jiayl@webrtc.org · 11 years ago
e9d3760 AEC: Adds a reported_delay_enabled_ flag by bjornv@webrtc.org · 11 years ago
26e2b68 Remove ACM1/ACM2 switching from VoiceEngine tests by henrik.lundin@webrtc.org · 11 years ago
db14442 Exclude the new AudioProcessingTest from some sanitizer bots. by andrew@webrtc.org · 11 years ago
46b31b1 Restore sample_rate_hz() until Chromium is updated to not use it. by andrew@webrtc.org · 11 years ago
504fc89 (Auto)update libjingle 65394435-> 65417850 by buildbot@webrtc.org · 11 years ago
19b1be1 Provide GetStats method in RTCPeerConnection by tkchin@webrtc.org · 11 years ago
ddbb8a2 Support arbitrary input/output rates and downmixing in AudioProcessing. by andrew@webrtc.org · 11 years ago
34fe015 Reland "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 11 years ago
d59359a Remove 44.1 kHz workaround from the iOS AudioDevice. by andrew@webrtc.org · 11 years ago
20c71fd Fix a bug in AcmReceiver::NetworkStatistics by henrik.lundin@webrtc.org · 11 years ago
0c108d0 Revert "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 11 years ago
139706e Stop using ACM factory in VoiceEngine by henrik.lundin@webrtc.org · 11 years ago
d144bb6 Let A/V sync test use default AudioCoding module by henrik.lundin@webrtc.org · 11 years ago
0c1444c Create ACM2 instance when calling AudioCodingModule::Create by henrik.lundin@webrtc.org · 11 years ago
0a035c8 Disable tests in common_video_unittests for Dr Memory. by kjellander@webrtc.org · 11 years ago
372ae83 Reland "Make VoiceEngine choose ACM2 by default"" by henrik.lundin@webrtc.org · 11 years ago
5964fe0 audio_processing: DestroyHandle() now returns void by bjornv@webrtc.org · 11 years ago
2a79672 common_audio: VADFree() now returns void by bjornv@webrtc.org · 11 years ago
3dfabf9 libyuv r1000 roll for DEPS update to new chromium moving location of gold linker on linux. by fbarchard@google.com · 11 years ago
ec3d8ec Fix typo by renaming RTCSessionDescriptonDelegate -> RTCSessionsDescriptionDelegate by tkchin@webrtc.org · 11 years ago
229e16e Add resource audio for audio processing tests. by andrew@webrtc.org · 11 years ago
54fd700 Remove ASSERT in TransportChannelProxy::SetImplementation, when by mallinath@webrtc.org · 11 years ago
f5a33f1 Resampler modifications in preparation for arbitrary audioproc rates. by andrew@webrtc.org · 11 years ago
3d9ec1f Fix multi-monitor support in the screen capturer for Mac. by sergeyu@chromium.org · 11 years ago
8e5ec52 (Auto)update libjingle 65152644-> 65219629 by buildbot@webrtc.org · 11 years ago
7d055a6 Revert r5937 "Fix multi-monitor support in the screen capturer for Mac." by sergeyu@chromium.org · 11 years ago
0daa8be Add Chromium's ScopedVector. by andrew@webrtc.org · 11 years ago
be7585b Fix multi-monitor support in the screen capturer for Mac. by sergeyu@chromium.org · 11 years ago
a596a38 Fix iSAC/48000 issue with ACM2. by turaj@webrtc.org · 11 years ago
29540b1 Revert "PeerConnectionFactory: delay deletion of owned threads." by fischman@webrtc.org · 11 years ago
1a87f52 (Auto)update libjingle 65151416-> 65151642 by buildbot@webrtc.org · 11 years ago
cea024d PeerConnectionFactory: delay deletion of owned threads. by fischman@webrtc.org · 11 years ago
b476d36 Roll libvpx 259973:264320 by marpan@webrtc.org · 11 years ago
aeb0c28 Update PRESUBMIT.py's list of "DO_NOT_SUBMIT_FILES". by henrike@webrtc.org · 11 years ago
e57ae02 WebRtcAecm_Process: Reduce code duplication by kwiberg@webrtc.org · 11 years ago
d2f366f StereoToMono: Remove useless call to WebRtcSpl_SatW32ToW16 by kwiberg@webrtc.org · 11 years ago
6680348 Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago
0f73755 Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago
e2e9abb Revert "Make VoiceEngine choose ACM2 by default" by henrik.lundin@webrtc.org · 11 years ago
0b3c6c3 (Auto)update libjingle 65086785-> 65104022 by buildbot@webrtc.org · 11 years ago
adaf809 Removing AudioCoding duplicate tests by henrik.lundin@webrtc.org · 11 years ago
6cec07f Make VoiceEngine choose ACM2 by default by henrik.lundin@webrtc.org · 11 years ago
c0a15b7 Fix crashes due to dangling external decoder pointer. by fischman@webrtc.org · 11 years ago
39b868b (Auto)update libjingle 65055925-> 65086785 by buildbot@webrtc.org · 11 years ago
8f88f20 Expand the test max wait time from 1000ms to 2000ms. by jiayl@webrtc.org · 11 years ago
c187291 Set include_internal_video_capture=1 for video_capture_tests by kjellander@webrtc.org · 11 years ago
f927fd6 Re-enable AGC tests: by aluebs@webrtc.org · 11 years ago
7de47bc Remove use of tmpnam. by kjellander@webrtc.org · 11 years ago
2c3f1ab Replace flooding logs in rtp_sender.cc with a comment. by andrew@webrtc.org · 11 years ago
36eda7c Workaround for https://bugzilla.mozilla.org/show_bug.cgi?id=996329, where the m line from firefox have a space at the end. by wu@webrtc.org · 11 years ago
ca539bb iOS: baby steps to being able to include_tests=1 by fischman@webrtc.org · 11 years ago
7c6e3d1 Moved voe_neteq_stats_unittest to audio_coding_module_unittest by henrik.lundin@webrtc.org · 11 years ago
6c75c98 Propagate capture ntp timestamp from rtp to renderer. by wu@webrtc.org · 11 years ago
1fd5b45 (Auto)update libjingle 64956819-> 64982143 by buildbot@webrtc.org · 11 years ago
2f8d5f3 Check if a header extension is registered before updating it and fail silently if it's not. by stefan@webrtc.org · 11 years ago
190b72a Make libjingle Android example build without sourcing envsetup.sh by kjellander@webrtc.org · 11 years ago
6e105ed Make WebRTC Android examples build without sourcing envsetup.sh by kjellander@webrtc.org · 11 years ago
ad4440a In shared socket mode, use udp port as default receiver even if by mallinath@webrtc.org · 11 years ago
505f400 (Auto)update libjingle 64909599-> 64919255 by buildbot@webrtc.org · 11 years ago
e98598d Make everyone an OWNER for .gyp/.gypi add/delete purposes, talk/ edition. by fischman@webrtc.org · 11 years ago
2c89b5c Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 11 years ago
35ead38 Adding a config struct to NetEq by henrik.lundin@webrtc.org · 11 years ago
810acbc New Packet and PacketSource classes for NetEq tests by henrik.lundin@webrtc.org · 11 years ago
1da6047 (Auto)update libjingle 64813990-> 64909599 by buildbot@webrtc.org · 11 years ago
cf0b46c iosdeviceinfo.cc: remove unnecessary file by fischman@webrtc.org · 11 years ago
5cf7396 Fix gyp for video_capture/ensure_initialized.cc. by primiano@chromium.org · 11 years ago
f875f15 (Auto)update libjingle 64709629-> 64813990 by buildbot@webrtc.org · 11 years ago
b9309be Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago