Log - 53c1d3c561c92948cada27409bab1c5b9747ae8a - fp2-dev/platform/external/chromium_org/third_party/webrtc

feb904c Replace flooding logs in rtp_sender.cc with a comment. by andrew@webrtc.org · 10 years ago
8fb9156 iOS: baby steps to being able to include_tests=1 by fischman@webrtc.org · 10 years ago
bbea098 Moved voe_neteq_stats_unittest to audio_coding_module_unittest by henrik.lundin@webrtc.org · 10 years ago
9d10769 Propagate capture ntp timestamp from rtp to renderer. by wu@webrtc.org · 10 years ago
d581e2c Check if a header extension is registered before updating it and fail silently if it's not. by stefan@webrtc.org · 10 years ago
336f24b Make WebRTC Android examples build without sourcing envsetup.sh by kjellander@webrtc.org · 10 years ago
98f8320 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
4087215 Adding a config struct to NetEq by henrik.lundin@webrtc.org · 10 years ago
e7d9de3 New Packet and PacketSource classes for NetEq tests by henrik.lundin@webrtc.org · 10 years ago
a48d53b Fix gyp for video_capture/ensure_initialized.cc. by primiano@chromium.org · 10 years ago
4ff0eda 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
91d88e1 Added a new OnMoreData() interface which will not feed the playout data to APM. by xians@webrtc.org · 10 years ago
26103b9 Fix the captured screen rect conversion. by jiayl@webrtc.org · 10 years ago
c1caa69 NetEq changes. by turaj@webrtc.org · 10 years ago
8edccce Cleaned up logging in video_coding. by stefan@webrtc.org · 10 years ago
f16b605 Convert WEBRTC_TRACE to LOG in utility. by asapersson@webrtc.org · 10 years ago
55bc281 Disable UsesTraceCallback by pbos@webrtc.org · 10 years ago
5b3c956 Fix loopback test for case where no constraint is given. by andresp@webrtc.org · 10 years ago
9402619 Remove usage of webrtc trace in video processing modules. by asapersson@webrtc.org · 10 years ago
ccef356 Add ability to control peer connection constraints for the loopback test. by andresp@webrtc.org · 10 years ago
01f4fb6 Remove self-assignment hacks that were added to avoid unused variable warnings. by fischman@webrtc.org · 10 years ago
b8f935f Move a chatty creation log in neteq to LS_VERBOSE. by andrew@webrtc.org · 10 years ago
ca9bff6 Make Android-APK compile in release again. by solenberg@webrtc.org · 10 years ago
2e4c621 (landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds by henrika@webrtc.org · 10 years ago
f03e467 Unbreak android APK buildbots by emptying the video_capture_tests_apk target. by fischman@webrtc.org · 10 years ago
b515322 VideoCaptureAndroid: support multiple frame-rates per resolution. by fischman@webrtc.org · 10 years ago
d6e5cf9 Fix DesktopSize::is_empty() for the case when only width or only height is 0. by sergeyu@chromium.org · 10 years ago
01f4592 Move output_mixer_unittest.cc to utility_unittest.cc. by andrew@webrtc.org · 10 years ago
bdeb1d8 VideoCaptureAndroid: stop referencing ViERenderer by fischman@webrtc.org · 10 years ago
24224fc video_capture(iOS): move stopCapture to background thread by fischman@webrtc.org · 10 years ago
ea15f8d Implement FEC support in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
9968131 Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 10 years ago
cef07f9 New NetEq test to verify correct timestamp propagation by henrik.lundin@webrtc.org · 10 years ago
5260350 Removed the disabling of include_tests from r2729. by henrike@webrtc.org · 10 years ago
a32583c Updated WebRTC version to 3.52 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
8d93b11 Clean up traces and logs in RemoteBitrateEstimator. by stefan@webrtc.org · 10 years ago
022615b Log Fixit for parts of video_engine folder. by mflodman@webrtc.org · 10 years ago
a6948e2 Fix logging calls in bitrate_controller module. by andresp@webrtc.org · 10 years ago
168a51c Remove WEBRTC_TRACE use in common_video/ by pbos@webrtc.org · 10 years ago
a538def Fix a crash in WindowCapturereMac when capture() fails. by jiayl@webrtc.org · 10 years ago
4d17e20 Fix the library path for android 64-bit build by michaelbai@google.com · 10 years ago
f7c73b5 Consolidate audio conversion from Channel and TransmitMixer. by andrew@webrtc.org · 10 years ago
b5a182a Delay Estimator: Minor refactoring and added a setter function. by bjornv@webrtc.org · 10 years ago
a38c76b Rename RTPanalyze to rtp_analyze and remove old version by henrik.lundin@webrtc.org · 10 years ago
cceb392 Remove AudioDevice::{Microphone,Speaker}IsAvailable. by andrew@webrtc.org · 10 years ago
ffcd844 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
df9fa2b Add format specification to output file names by henrik.lundin@webrtc.org · 10 years ago
120c725 Extends max sample rate from 96kHz to 192kHz on the input side. by henrika@webrtc.org · 10 years ago
53da760 sink_filter_ds.cc: add lock to Receive procedure to Pause(). by braveyao@webrtc.org · 10 years ago
56aeb0e Make ACM2 the default in voe_cmd_test. by andrew@webrtc.org · 10 years ago
e823012 Added simulations of capacity variations and wifi recordings. by stefan@webrtc.org · 10 years ago
38d4ad7 Roll chromium_revision 255773:260462 by kjellander@webrtc.org · 10 years ago
7830689 Fix ARM64 detection. by andrew@webrtc.org · 10 years ago
172f42a VoiceEngine(iOS & Android): removed NOT_SUPPORTED by fischman@webrtc.org · 10 years ago
3f2f440 Add tests for the RBE RemoveStream() API. by solenberg@webrtc.org · 10 years ago
c9dca91 VoE Channel: Don't register codecs when stopping receiver by henrik.lundin@webrtc.org · 10 years ago
4910a7f Restore support for code coverage in WebRTC by kjellander@webrtc.org · 10 years ago
d30a9ea Add arm64 to typedefs.h by andrew@webrtc.org · 10 years ago
1fb05fc Allow loopback tests to do TURN when served from webrtc.googlecode.com. by andresp@webrtc.org · 10 years ago
e438054 Add svn mime-type properties to loopback_test files so they can be served from: by andresp@webrtc.org · 10 years ago
5f8dfa0 Don't disable experimental AGC in audioproc. by andrew@webrtc.org · 10 years ago
d7aa228 Re-submit: rev5775 by andresp@webrtc.org · 10 years ago
ba75592 Makes ScreenCapturerMac exclude the window specified in DesktopCapturer::SetExcludedWindow. by jiayl@webrtc.org · 10 years ago
2f0c5f7 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
9511bbd Protect write of send_target_bitrate. by andresp@webrtc.org · 10 years ago
2bf87a2 Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out. by solenberg@webrtc.org · 10 years ago
ca28c29 Revert 5775 "Modify bitrate controller to update bitrate based o..." by andrew@webrtc.org · 10 years ago
2a0cbfc Removing VideoCodecDerived and moving methods inside VideoCodec. by mallinath@webrtc.org · 10 years ago
60ae794 Updated WebRTC version to 3.51 by elham@webrtc.org · 10 years ago
e965143 iOS video_capture: move @private vars to impl. by fischman@webrtc.org · 10 years ago
a4f259b Fix race condition in RTPSEnder. by sprang@webrtc.org · 10 years ago
9deb87b Change sprintf format string from %zu to %i by henrik.lundin@webrtc.org · 10 years ago
a0320c2 Modify bitrate controller to update bitrate based on process call and not by andresp@webrtc.org · 10 years ago
5d8c954 Adding API for setting bandwidth estimation configurations. by stefan@webrtc.org · 10 years ago
7fd6ac1 iOS video_capture: start camera in the background. by fischman@webrtc.org · 10 years ago
25bbc98 iOS VideoEngine: move video_{capture,render} to ARC. by fischman@webrtc.org · 10 years ago
154951d Add configuration for ability to use the encode usage measure for triggering overuse/underuse. by asapersson@webrtc.org · 10 years ago
2d3624c Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. by solenberg@webrtc.org · 10 years ago
5374dab Have changes to REMB trigger RTCP to be sent immediately. by stefan@webrtc.org · 10 years ago
1df7a5a DelayEstimator: Updates delay_quality and adds soft reset. by bjornv@webrtc.org · 10 years ago
f70d0b9 Run Opus with lower complexity setting on Android, iOS and/or ARM by tina.legrand@webrtc.org · 10 years ago
3d6910c Add targetBitrate to VideoCodec struct. by pbos@webrtc.org · 10 years ago
deac6f5 Disabled some of the remote bitrate estimator baseline tests. by stefan@webrtc.org · 10 years ago
fec6b6e VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
24be06c Add fir_filter to common_audio by aluebs@webrtc.org · 10 years ago
b9d0acb Add AIMD option to BWE API. by stefan@webrtc.org · 10 years ago
6531cb0 ACM2/NetEq4 did not decode Opus in stereo by tina.legrand@webrtc.org · 10 years ago
3b6c0e5 Refactor in BitrateController module. by andresp@webrtc.org · 10 years ago
0b43be5 Fixing crash in video_render_tests in release mode. by henrikg@webrtc.org · 10 years ago
f2b62a8 Remove locks in SendSideBandwidthEstimation since those are only accessed while owning locks in by andresp@webrtc.org · 10 years ago
835b016 Adding FEC support in NetEq 4. by minyue@webrtc.org · 10 years ago
8d49b3f Fix "unreachable code" warnings (MSVC warning 4702) in webrtc. by pbos@webrtc.org · 10 years ago
18c2945 Adding operator== and != methods for CodecInst and VideoCodec structures. by mallinath@webrtc.org · 10 years ago
47f52f2 Use codec width/height as the encoded_image width/height. by wu@webrtc.org · 10 years ago
44c6a36 Changing the buffer size (slots) to 1.5 seconds @ 30 ms packets by henrik.lundin@webrtc.org · 10 years ago
9da327c Add ability to configure cpu overuse options via an API. by asapersson@webrtc.org · 10 years ago
a5db8e3 Prevent playout delay wrap-around in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
650772a Removes error printout in voe_cmd_test which was caused by attempts to transmit RTCP packets even if a transport object was not registered. by henrika@webrtc.org · 10 years ago
700d14b Extend perf tests to perform rampup on single stream. by andresp@webrtc.org · 10 years ago
ac1fabd Adjust the captured window rect when the window is maximized. by jiayl@webrtc.org · 10 years ago