Log - 5cc0d0b1f1c0b684223419696347624da7bb22d8 - fp2-dev/platform/external/chromium_org/third_party/webrtc

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
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
1982636 Clean up traces and logs in RemoteBitrateEstimator. by stefan@webrtc.org · 10 years ago
4fe54a8 Fix logging calls in bitrate_controller module. by andresp@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
a0acb1f sink_filter_ds.cc: add lock to Receive procedure to Pause(). by braveyao@webrtc.org · 10 years ago
15f109e Added simulations of capacity variations and wifi recordings. by stefan@webrtc.org · 10 years ago
8f5ab19 VoiceEngine(iOS & Android): removed NOT_SUPPORTED by fischman@webrtc.org · 10 years ago
24532e0 Add tests for the RBE RemoveStream() API. by solenberg@webrtc.org · 10 years ago
bae92ab Don't disable experimental AGC in audioproc. by andrew@webrtc.org · 10 years ago
6c57efd Re-submit: rev5775 by andresp@webrtc.org · 10 years ago
0ab635c Makes ScreenCapturerMac exclude the window specified in DesktopCapturer::SetExcludedWindow. by jiayl@webrtc.org · 10 years ago
37f807f Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams. by solenberg@webrtc.org · 10 years ago
1e05528 Protect write of send_target_bitrate. by andresp@webrtc.org · 10 years ago
0027f0a Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out. by solenberg@webrtc.org · 10 years ago
765ea72 Revert 5775 "Modify bitrate controller to update bitrate based o..." by andrew@webrtc.org · 10 years ago
a090cc7 iOS video_capture: move @private vars to impl. by fischman@webrtc.org · 10 years ago