Log - 5ec8feeb2b3bebe7d5e06262e4d3efd76b63d356 - fp2-dev/platform/external/chromium_org/third_party/webrtc

5ec8fee Remove the use of AudioFrame::energy_ from AudioProcessing and VoE. by andrew@webrtc.org · 10 years ago
2a8fb71 Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe. by wu@webrtc.org · 10 years ago
11fa357 Add webrtc field trials API. by andresp@webrtc.org · 10 years ago
1a99e7a Re-enable NetEqExternalDecoderTest for Android by henrik.lundin@webrtc.org · 10 years ago
8334eda Re-enable NetEQ DecoderDatabase test for Android by henrik.lundin@webrtc.org · 10 years ago
409cf2a Revert "Audio processing: Feed each processing step its choice of int or float data" by mflodman@webrtc.org · 10 years ago
654bd9e Re-enable the BitrateEstimatorTest cases for the Call API. by solenberg@webrtc.org · 10 years ago
2248763 Remove all use of AudioFrame::energy_ from AudioCodingModule by henrik.lundin@webrtc.org · 10 years ago
6047281 Audio processing: Feed each processing step its choice of int or float data by kwiberg@webrtc.org · 10 years ago
903ce9e Remove WEBRTC_TRACE use in video_capture/ by pbos@webrtc.org · 10 years ago
2efcf70 Deleting all NetEq3 files by henrik.lundin@webrtc.org · 10 years ago
cf1f0b0 The webrtc::AudioFrame struct contains a variable energy_. Since the energy isn't always calculated when the frame is created, this change makes the CalculateEnergy method in Audio Conference Mixer always calculate the energy. by henrik.lundin@webrtc.org · 10 years ago
4a792f0 Deleting all ACM1 files by henrik.lundin@webrtc.org · 10 years ago
9c8f347 Echo cancellation functions docs: Follow style guide w.r.t. placement of * by kwiberg@webrtc.org · 10 years ago
aa169d2 One of the NetEq methods needs to be virtual. by turaj@webrtc.org · 10 years ago
da9b404 Modifying neteq.gyp by turaj@webrtc.org · 10 years ago
8b4f539 AudioBuffer: Eliminate data_was_mixed_, and document what's left of data_ by kwiberg@webrtc.org · 10 years ago
60f1422 Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine. by wu@webrtc.org · 10 years ago
8539c4a Fix odd codes in video_capture on Mac. by braveyao@webrtc.org · 10 years ago
4fb1a55 video_render.gypi: clean up some libraries directives to be more specific. by fischman@webrtc.org · 10 years ago
73c2412 Remove timestamp_extrapolator's dependency to Clock and vcm defines. by wu@webrtc.org · 10 years ago
8ec46c6 Allow the RTP level indicator computation to work at any sample rate. by andrew@webrtc.org · 10 years ago
ebb4b94 Implement the Windows screen capturer using the Magnification API. by jiayl@webrtc.org · 10 years ago
7c434be Revert 6048 "Implement the Windows screen capturer using the Mag..." by tina.legrand@webrtc.org · 10 years ago
c2b27b5 Implement the Windows screen capturer using the Magnification API. by jiayl@webrtc.org · 10 years ago
616cbcd Only clamp to 16 kHz when AECM is enabled. by andrew@webrtc.org · 10 years ago
c2e6438 Fix constness of AudioBuffer accessors. by andrew@webrtc.org · 10 years ago
0638464 Fix a data race in ACM1 when audio is pulled. by turaj@webrtc.org · 10 years ago
976ce98 Added include of assert.h for files calling assert but missing the include. by henrike@webrtc.org · 10 years ago
40b200b Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord() by henrike@webrtc.org · 10 years ago
3d5905b Disable failing GoogleWifiTrace3Mbps. by pbos@webrtc.org · 10 years ago
d592231 Disable GoogleWifiTrace3Mbps. by pbos@webrtc.org · 10 years ago
3848107 Adding BweFeedbackTest which tracks BWE performance over a set of simulated scenarios. by stefan@webrtc.org · 10 years ago
c78232f Fix iOS assembly compile error. by kjellander@webrtc.org · 10 years ago
ad4cce6 Roll chromium_revision 260462:266514 by kjellander@webrtc.org · 10 years ago
3cbb2df Remove Version method from ACM1 by henrik.lundin@webrtc.org · 10 years ago
dc37088 Remove ACM1 and NetEq3 related targets from modules.gyp by henrik.lundin@webrtc.org · 10 years ago
68a95e1 Remove AudioCodingModuleFactory by henrik.lundin@webrtc.org · 10 years ago
a48f3c2 Add clock to ACM config struct by henrik.lundin@webrtc.org · 10 years ago
db395e4 AEC: Startup phase only runs if reported_delay_enabled by bjornv@webrtc.org · 10 years ago
b4945d1 APM: limit native sample rate to 16kHz on mobile. by fischman@webrtc.org · 10 years ago
93d270f Using realpath instead of android_src in Android webview by michaelbai@google.com · 10 years ago
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
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
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
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
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
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
68bd1f3 Create ACM2 instance when calling AudioCodingModule::Create 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
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
722cd19 Removing AudioCoding duplicate tests 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
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
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
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
bd0a216 Remove usage of webrtc trace in video processing modules. by asapersson@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
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
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