Log - 0e098e0b0340e3bb08c12cbc98c741d179cb2a63 - fp2-dev/platform/external/chromium_org/third_party/webrtc

0e098e0 AEC: Moved delay buffer size enums from aec_core.h to aec_core_internal.h by bjornv@webrtc.org · 10 years ago
676638c Disable capture test for FrameRate on Windows. by pbos@webrtc.org · 10 years ago
bb62a93 Introduce a config struct for AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
c298835 Disabling flaky CanReceiveFec. by pbos@webrtc.org · 10 years ago
73e1a8b Disable flaky RunsRtpRtcpTestWIthoutErrors. by pbos@webrtc.org · 10 years ago
abf78cc Fix the NetEq build by henrik.lundin@webrtc.org · 10 years ago
75d1487 Include buffer size limits in NetEq config struct by henrik.lundin@webrtc.org · 10 years ago
a714643 Add henrik.lundin as owner in AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
b0079ed Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
4820f6b Fix leak in remote bitrate estimator tests introduced in r5980 by stefan@webrtc.org · 10 years ago
8c4135e Support for simulating multiple independent flows in a network. by stefan@webrtc.org · 10 years ago
0a5fd54 Casting char to int in logs. by asapersson@webrtc.org · 10 years ago
85d90de Returns a NULL frame on all platforms if the captured window is closed. by jiayl@webrtc.org · 10 years ago
b991cd0 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
0061d86 * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. by wu@webrtc.org · 10 years ago
ee6695b Add an output capacity parameter to ACMResampler::Resample10Msec() by henrik.lundin@webrtc.org · 10 years ago
fbf2568 Add keyboard channel support to AudioBuffer. by andrew@webrtc.org · 10 years ago
86e3fa8 Fix the Android compilation (better structure for NetEq test libs) by henrik.lundin@webrtc.org · 10 years ago
dbebc39 Remove TraceCallback use from Call. by pbos@webrtc.org · 10 years ago
9d0f79f Rename Start/Stop in Video{Send,Receive}Streams. by pbos@webrtc.org · 10 years ago
e846663 Fixing a bug in ACM2 where the output frame energy was incorrectly set by henrik.lundin@webrtc.org · 10 years ago
757a92f Use unique filenames in AudioProcessingTests for parallelization. by andrew@webrtc.org · 10 years ago
e1b0595 AEC: Adds a reported_delay_enabled_ flag by bjornv@webrtc.org · 10 years ago
110a2d2 Remove ACM1/ACM2 switching from VoiceEngine tests by henrik.lundin@webrtc.org · 10 years ago
3ab5093 Restore sample_rate_hz() until Chromium is updated to not use it. by andrew@webrtc.org · 10 years ago
2e24460 Support arbitrary input/output rates and downmixing in AudioProcessing. by andrew@webrtc.org · 10 years ago
8b4811b Reland "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 10 years ago
79a6030 Remove 44.1 kHz workaround from the iOS AudioDevice. by andrew@webrtc.org · 10 years ago
0f437b0 Fix a bug in AcmReceiver::NetworkStatistics by henrik.lundin@webrtc.org · 10 years ago
a19bee3 Revert "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 10 years ago
a61127d Stop using ACM factory in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
69b14d5 Let A/V sync test use default AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
68bd1f3 Create ACM2 instance when calling AudioCodingModule::Create by henrik.lundin@webrtc.org · 10 years ago
13f9d37 Reland "Make VoiceEngine choose ACM2 by default"" by henrik.lundin@webrtc.org · 10 years ago
17d096a audio_processing: DestroyHandle() now returns void by bjornv@webrtc.org · 10 years ago
fb54df6 common_audio: VADFree() now returns void by bjornv@webrtc.org · 10 years ago
cf526f7 Resampler modifications in preparation for arbitrary audioproc rates. by andrew@webrtc.org · 10 years ago
11720c2 Fix multi-monitor support in the screen capturer for Mac. by sergeyu@chromium.org · 10 years ago
5fd5020 Revert r5937 "Fix multi-monitor support in the screen capturer for Mac." by sergeyu@chromium.org · 10 years ago
a4fbfd9 Add Chromium's ScopedVector. by andrew@webrtc.org · 10 years ago
a73081a Fix multi-monitor support in the screen capturer for Mac. by sergeyu@chromium.org · 10 years ago
bc6b15d Fix iSAC/48000 issue with ACM2. by turaj@webrtc.org · 10 years ago
499ee5e WebRtcAecm_Process: Reduce code duplication by kwiberg@webrtc.org · 10 years ago
2991a30 StereoToMono: Remove useless call to WebRtcSpl_SatW32ToW16 by kwiberg@webrtc.org · 10 years ago
514abde 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 · 10 years ago
3ea24b2 Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
14c5e8a Revert "Make VoiceEngine choose ACM2 by default" by henrik.lundin@webrtc.org · 10 years ago
722cd19 Removing AudioCoding duplicate tests by henrik.lundin@webrtc.org · 10 years ago
4f9c08f Make VoiceEngine choose ACM2 by default by henrik.lundin@webrtc.org · 10 years ago
db4b867 Fix crashes due to dangling external decoder pointer. by fischman@webrtc.org · 10 years ago
988e753 Set include_internal_video_capture=1 for video_capture_tests by kjellander@webrtc.org · 10 years ago
1857d7e Re-enable AGC tests: by aluebs@webrtc.org · 10 years ago
32e7755 Remove use of tmpnam. by kjellander@webrtc.org · 10 years ago
566af28 Replace flooding logs in rtp_sender.cc with a comment. by andrew@webrtc.org · 10 years ago
a738ae3 iOS: baby steps to being able to include_tests=1 by fischman@webrtc.org · 10 years ago
0b559b6 Moved voe_neteq_stats_unittest to audio_coding_module_unittest by henrik.lundin@webrtc.org · 10 years ago
a1626fe Propagate capture ntp timestamp from rtp to renderer. by wu@webrtc.org · 10 years ago
aee97d8 Check if a header extension is registered before updating it and fail silently if it's not. by stefan@webrtc.org · 10 years ago
f6d791d Make WebRTC Android examples build without sourcing envsetup.sh by kjellander@webrtc.org · 10 years ago
6b1114a Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
633c598 Adding a config struct to NetEq by henrik.lundin@webrtc.org · 10 years ago
fd59b22 New Packet and PacketSource classes for NetEq tests by henrik.lundin@webrtc.org · 10 years ago
966744e Fix gyp for video_capture/ensure_initialized.cc. by primiano@chromium.org · 10 years ago
290c5a5 Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
e338cc2 Added a new OnMoreData() interface which will not feed the playout data to APM. by xians@webrtc.org · 10 years ago
538aff6 Fix the captured screen rect conversion. by jiayl@webrtc.org · 10 years ago
d399a50 NetEq changes. by turaj@webrtc.org · 10 years ago
b18bff5 Cleaned up logging in video_coding. by stefan@webrtc.org · 10 years ago
28d1b61 Convert WEBRTC_TRACE to LOG in utility. by asapersson@webrtc.org · 10 years ago
19ca463 Disable UsesTraceCallback by pbos@webrtc.org · 10 years ago
3841668 Fix loopback test for case where no constraint is given. by andresp@webrtc.org · 10 years ago
bd0a216 Remove usage of webrtc trace in video processing modules. by asapersson@webrtc.org · 10 years ago
ea1b72d Add ability to control peer connection constraints for the loopback test. by andresp@webrtc.org · 10 years ago
284f401 Remove self-assignment hacks that were added to avoid unused variable warnings. by fischman@webrtc.org · 10 years ago
9c31dee Move a chatty creation log in neteq to LS_VERBOSE. by andrew@webrtc.org · 10 years ago
303f24f Make Android-APK compile in release again. by solenberg@webrtc.org · 10 years ago
7a06daa (landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds by henrika@webrtc.org · 10 years ago
365b4aa Unbreak android APK buildbots by emptying the video_capture_tests_apk target. by fischman@webrtc.org · 10 years ago
4e8afab VideoCaptureAndroid: support multiple frame-rates per resolution. by fischman@webrtc.org · 10 years ago
523753b Fix DesktopSize::is_empty() for the case when only width or only height is 0. by sergeyu@chromium.org · 10 years ago
eb90479 Move output_mixer_unittest.cc to utility_unittest.cc. by andrew@webrtc.org · 10 years ago
a67c9a4 VideoCaptureAndroid: stop referencing ViERenderer by fischman@webrtc.org · 10 years ago
fc0693b video_capture(iOS): move stopCapture to background thread by fischman@webrtc.org · 10 years ago
d8b4d0f Implement FEC support in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
4b0cd7f Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 10 years ago
5406963 New NetEq test to verify correct timestamp propagation by henrik.lundin@webrtc.org · 10 years ago
213590d Removed the disabling of include_tests from r2729. by henrike@webrtc.org · 10 years ago
ff46b81 Updated WebRTC version to 3.52 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
1982636 Clean up traces and logs in RemoteBitrateEstimator. by stefan@webrtc.org · 10 years ago
44c9b9a Log Fixit for parts of video_engine folder. by mflodman@webrtc.org · 10 years ago
4fe54a8 Fix logging calls in bitrate_controller module. by andresp@webrtc.org · 10 years ago
3aded9d Remove WEBRTC_TRACE use in common_video/ by pbos@webrtc.org · 10 years ago
7cb3251 Fix a crash in WindowCapturereMac when capture() fails. by jiayl@webrtc.org · 10 years ago
0115a83 Fix the library path for android 64-bit build by michaelbai@google.com · 10 years ago
bf4f232 Consolidate audio conversion from Channel and TransmitMixer. by andrew@webrtc.org · 10 years ago
0eb8ec6 Delay Estimator: Minor refactoring and added a setter function. by bjornv@webrtc.org · 10 years ago
3aa1ac2 Rename RTPanalyze to rtp_analyze and remove old version by henrik.lundin@webrtc.org · 10 years ago
c55faad Remove AudioDevice::{Microphone,Speaker}IsAvailable. by andrew@webrtc.org · 10 years ago
acb49e5 This is to get rid of a bug relating to the return of NULL in calling GetDecoder when there are DTMF packets. by minyue@webrtc.org · 10 years ago
71c9ebd Add format specification to output file names by henrik.lundin@webrtc.org · 10 years ago