• guidou's avatar
    Roll WebRTC 15456:15513 (40 commits) · 9397420a
    guidou authored
    Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/666ab2c..f23e926
    
    $ git log 666ab2c..f23e926 --date=short --no-merges --format=%ad %ae %s
    2016-12-09 pbos@webrtc.org Remove extra uses of basictypes.h.
    2016-12-09 brandtr@webrtc.org Only store sequence numbers for media stream in FlexFEC end-to-end test.
    2016-12-09 kthelgason@webrtc.org Disable failing perf test on Android.
    2016-12-09 minyue@webrtc.org Modify JavaToStdString to allow ISO-8859-1 encoded strings.
    2016-12-09 terelius@webrtc.org Implement Theil-Sen's method for fitting a line to noisy data (used in bandwidth estimation).
    2016-12-09 hbos@webrtc.org RTCInboundRTPStreamStats.packetsLost set by RTCStatsCollector.
    2016-12-09 hbos@webrtc.org RTCIceCandidatePairStats.requestsReceived defined by RTCStatsCollector.
    2016-12-09 kthelgason@webrtc.org Reland of Bump up scaling limit for MediaCodec. (patchset #1 id:1 of https://codereview.webrtc.org/2562963002/ )
    2016-12-09 peah@webrtc.org During AEC development, it is handy to be able to simulate different orders of the ProcessStream and ProcessReverseStream API calls.
    2016-12-09 peah@webrtc.org When recreating a call based on an aecdump recording the nearend used is the one stored in the aecdump.
    2016-12-09 asapersson@webrtc.org Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )
    2016-12-09 kthelgason@webrtc.org Revert of Bump up scaling limit for MediaCodec. (patchset #3 id:40001 of https://codereview.webrtc.org/2566533002/ )
    2016-12-09 kthelgason@webrtc.org Bump up scaling limit for MediaCodec.
    2016-12-08 stefan@webrtc.org Fix issue with deprecated CongestionController interface not working.
    2016-12-08 zijiehe@chromium.org Enable screen capturer tests for Linux / DirectX capturer / magnifier capturer
    2016-12-08 ehmaldonado@webrtc.org Refactor webrtc/modules/video_{capture,coding} for GN check
    2016-12-08 brandtr@webrtc.org Rename RtpStreamReceiver::IsFecEnabled to RtpStreamReceiver::IsUlpfecEnabled.
    2016-12-08 asapersson@webrtc.org Do not update OnReceivedRtcpReceiverReport if report block list is empty (and rtt zero).
    2016-12-08 kthelgason@webrtc.org Reland of Add ability to scale to arbitrary factors (patchset #1 id:1 of https://codereview.webrtc.org/2557323002/ )
    2016-12-08 nisse@webrtc.org Delete deprecated CongestionController constructor and packet_router method.
    2016-12-08 kjellander@webrtc.org Re-enable disabled VideoProcessorIntegrationTest tests
    2016-12-08 brandtr@webrtc.org Add FlexFEC settings toggle in Android AppRTCMobile.
    2016-12-08 nisse@webrtc.org Simplify an always true condition.
    2016-12-08 sakal@webrtc.org Fix error in VideoFileRenderer_nativeI420Scale.
    2016-12-08 brandtr@webrtc.org Generalize FlexfecReceiveStream::Config.
    2016-12-08 brandtr@webrtc.org Clean up FlexfecReceiveStream ctor signatures.
    2016-12-08 ehmaldonado@webrtc.org Refactor webrtc/modules/audio_processing for GN check
    2016-12-08 johan@webrtc.org Decode h264 fmtp sprop-parameter-sets to binary.
    2016-12-08 aleloi@webrtc.org Injectable output rate calculater for AudioMixer.
    2016-12-08 asapersson@webrtc.org Remove unused arguments and variable in MediaOptimization.
    2016-12-08 kthelgason@webrtc.org Revert of Add ability to scale to arbitrary factors (patchset #7 id:120001 of https://codereview.webrtc.org/2555483005/ )
    2016-12-08 kthelgason@webrtc.org Add ability to scale to arbitrary factors
    2016-12-08 hta@webrtc.org Refactoring: Declare cricket::Codec constructors protected.
    2016-12-08 magjed@webrtc.org Android classreferenceholder.h: Reorder function declaration keywords
    2016-12-07 zijiehe@chromium.org Log BitBlt failure
    2016-12-07 zhihuang@webrtc.org Create the Java Wrapper of RtpReceiverObserverInterface.
    2016-12-07 kjellander@webrtc.org Refactor webrtc/{api,audio} and modules/audio_coding for GN check
    2016-12-07 ossu@webrtc.org Moved call.h and most of api/call/* into call/
    2016-12-07 ehmaldonado@webrtc.org Clean up redundant include of ../webrtc_overrides
    2016-12-07 minyue@webrtc.org Adding OnReceivedOverhead to AudioEncoder.
    
    TBR=
    CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_archive_rel_ng;master.tryserver.chromium.mac:mac_chromium_archive_rel_ng
    BUG=
    
    Review-Url: https://codereview.chromium.org/2567643002
    Cr-Commit-Position: refs/heads/master@{#437564}
    9397420a
DEPS 41.3 KB