Log - 6b1114aea320f26f4dbe784305708139464dfd2c - fp2-dev/platform/external/chromium_org/third_party/webrtc

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
765ea72 Revert 5775 "Modify bitrate controller to update bitrate based o..." by andrew@webrtc.org · 10 years ago
f50914a Removing VideoCodecDerived and moving methods inside VideoCodec. by mallinath@webrtc.org · 10 years ago
0ac0bca Updated WebRTC version to 3.51 by elham@webrtc.org · 10 years ago
a090cc7 iOS video_capture: move @private vars to impl. by fischman@webrtc.org · 10 years ago
09fb237 Fix race condition in RTPSEnder. by sprang@webrtc.org · 10 years ago
0b11715 Change sprintf format string from %zu to %i by henrik.lundin@webrtc.org · 10 years ago
539bbde Modify bitrate controller to update bitrate based on process call and not by andresp@webrtc.org · 10 years ago
1f49208 Adding API for setting bandwidth estimation configurations. by stefan@webrtc.org · 10 years ago
892cd1f iOS video_capture: start camera in the background. by fischman@webrtc.org · 10 years ago
1dd9fb5 iOS VideoEngine: move video_{capture,render} to ARC. by fischman@webrtc.org · 10 years ago
1a19092 Add configuration for ability to use the encode usage measure for triggering overuse/underuse. by asapersson@webrtc.org · 10 years ago
50ac4d6 Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. by solenberg@webrtc.org · 10 years ago
85101db Have changes to REMB trigger RTCP to be sent immediately. by stefan@webrtc.org · 10 years ago
bb1d4c7 DelayEstimator: Updates delay_quality and adds soft reset. by bjornv@webrtc.org · 10 years ago
b45cf1e Run Opus with lower complexity setting on Android, iOS and/or ARM by tina.legrand@webrtc.org · 10 years ago
5ca38d1 Add targetBitrate to VideoCodec struct. by pbos@webrtc.org · 10 years ago
825acb1 Disabled some of the remote bitrate estimator baseline tests. by stefan@webrtc.org · 10 years ago
5f804f8 VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
61e9201 Add fir_filter to common_audio by aluebs@webrtc.org · 10 years ago
9b2b8ec Add AIMD option to BWE API. by stefan@webrtc.org · 10 years ago
95b5fde ACM2/NetEq4 did not decode Opus in stereo by tina.legrand@webrtc.org · 10 years ago
209791d Refactor in BitrateController module. by andresp@webrtc.org · 10 years ago
61e72f0 Fixing crash in video_render_tests in release mode. by henrikg@webrtc.org · 10 years ago
23e07d8 Remove locks in SendSideBandwidthEstimation since those are only accessed while owning locks in by andresp@webrtc.org · 10 years ago
97d92ed Adding FEC support in NetEq 4. by minyue@webrtc.org · 10 years ago
d327be4 Fix "unreachable code" warnings (MSVC warning 4702) in webrtc. by pbos@webrtc.org · 10 years ago
40fee00 Adding operator== and != methods for CodecInst and VideoCodec structures. by mallinath@webrtc.org · 10 years ago
bcee6b7 Use codec width/height as the encoded_image width/height. by wu@webrtc.org · 10 years ago
88fa18b Changing the buffer size (slots) to 1.5 seconds @ 30 ms packets by henrik.lundin@webrtc.org · 10 years ago
27bd3be Add ability to configure cpu overuse options via an API. by asapersson@webrtc.org · 10 years ago
55f4fe8 Prevent playout delay wrap-around in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
4d9df07 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
a183edc Extend perf tests to perform rampup on single stream. by andresp@webrtc.org · 10 years ago
b903292 Adjust the captured window rect when the window is maximized. by jiayl@webrtc.org · 10 years ago
5e18933 Properly account for retransmitted packets when not using the pacer. by stefan@webrtc.org · 10 years ago
f9d5709 Fixes RTX related bugs. by stefan@webrtc.org · 10 years ago
292e7f6 Disabling SendsSetSimulcastSsrcs. by pbos@webrtc.org · 10 years ago
ce06c77 Revert "Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets" by henrik.lundin@webrtc.org · 10 years ago
e8db41f Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets by henrik.lundin@webrtc.org · 10 years ago
16c3dcc Disable flaky CanSwitchToUseAllSsrcs. by pbos@webrtc.org · 10 years ago