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| author | grunell <grunell@chromium.org> | 2016-09-21 09:16:17 (GMT) |
|---|---|---|
| committer | Commit bot <commit-bot@chromium.org> | 2016-09-21 09:19:41 (GMT) |
| commit | 652d1e0fc872ef81a7d6c9037f92be489fa04800 (patch) | |
| tree | 08fa1b271701e0e37c12b0511f3ebaa699ab38e3 | |
| parent | 4fcc2d6bfddd9592403148e752620a987d1c15ae (diff) | |
| download | chromium-652d1e0fc872ef81a7d6c9037f92be489fa04800.tar.gz chromium-652d1e0fc872ef81a7d6c9037f92be489fa04800.tar.xz | |
Roll WebRTC 14298:14316 (18 commits)
Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/779690e..28cfdc3
$ git log 779690e..28cfdc3 --date=short --no-merges --format=%ad %ae %s
2016-09-21 brandtr@webrtc.org Generalize FEC header formatting.
2016-09-20 nisse@webrtc.org Delete VideoFrameFactory, CapturedFrame, and related code.
2016-09-20 brandtr@webrtc.org Generalize FEC unit tests and rename GenerateFec.
2016-09-20 minyue@webrtc.org Adding SmoothingFilter to audio network adaptor.
2016-09-20 kwiberg@webrtc.org AcmReceiver::DecoderByPayloadType: Ask NetEq for decoder
2016-09-20 aluebs@webrtc.org Compensate for the IntelligibilityEnhancer processing delay in high bands
2016-09-20 kjellander@webrtc.org Roll chromium_revision cf9457edb7..cede888c27 (416297:419407)
2016-09-20 hbos@webrtc.org Removed api/rtcstats[report].h pseudonyms of api/stats/rtcstats[report].h
2016-09-20 henrik.lundin@webrtc.org NetEq: New test for muted state during CNG
2016-09-20 danilchap@webrtc.org Minor cleanups in RTPSender::UpdateRtpStats
2016-09-20 stefan@webrtc.org Stop increasing loss-based BWE if no feedback is received.
2016-09-20 henrika@webrtc.org Improves resolution when logging rate in the ADB class.
2016-09-20 kwiberg@webrtc.org AcmReceiver: Ask NetEq to delete all decoders at once instead of one by one
2016-09-20 kwiberg@webrtc.org AcmReceiver: Look up last decoder in NetEq's table of decoders
2016-09-20 henrik.lundin@webrtc.org Add new decoding statistics for muted output
2016-09-20 danilchap@webrtc.org Replace rtcp packet parsing in the RtcpReceiver.
2016-09-20 ossu@webrtc.org Added a ParsePayload method to AudioDecoder.
2016-09-20 johan@webrtc.org Remove dead code branches from P2PtransportChannel unittest.
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/2353253002
Cr-Commit-Position: refs/heads/master@{#420016}
| -rw-r--r-- | DEPS | 2 |
1 files changed, 1 insertions, 1 deletions
@@ -219,7 +219,7 @@ deps = { Var('chromium_git') + '/native_client/src/third_party/scons-2.0.1.git' + '@' + '1c1550e17fc26355d08627fbdec13d8291227067', 'src/third_party/webrtc': - Var('chromium_git') + '/external/webrtc/trunk/webrtc.git' + '@' + '779690ee89f2ca171aa55efa261e1bf9ed7326e2', # commit position 14298 + Var('chromium_git') + '/external/webrtc/trunk/webrtc.git' + '@' + '28cfdc349fbe32189a51e7cac00ae0fef07775ad', # commit position 14316 'src/third_party/openmax_dl': Var('chromium_git') + '/external/webrtc/deps/third_party/openmax.git' + '@' + Var('openmax_dl_revision'), |