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

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
0725df6 Extends max sample rate from 96kHz to 192kHz on the input side. by henrika@webrtc.org · 10 years ago
a0acb1f sink_filter_ds.cc: add lock to Receive procedure to Pause(). by braveyao@webrtc.org · 10 years ago
5ae01bf Make ACM2 the default in voe_cmd_test. by andrew@webrtc.org · 10 years ago
15f109e Added simulations of capacity variations and wifi recordings. by stefan@webrtc.org · 10 years ago
53b062b Roll chromium_revision 255773:260462 by kjellander@webrtc.org · 10 years ago
7a8dee4 Fix ARM64 detection. by andrew@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
7c3f468 VoE Channel: Don't register codecs when stopping receiver by henrik.lundin@webrtc.org · 10 years ago
9136607 Restore support for code coverage in WebRTC by kjellander@webrtc.org · 10 years ago
ad239fe Add arm64 to typedefs.h by andrew@webrtc.org · 10 years ago
4c6d59a Allow loopback tests to do TURN when served from webrtc.googlecode.com. by andresp@webrtc.org · 10 years ago
66f5371 Add svn mime-type properties to loopback_test files so they can be served from: by andresp@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