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Commits on Jan 22, 2015

• Integrate send-side BWE into simulation framework.

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  stefan@webrtc.org

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• Split packets/bytes in StreamDataCounter into RtpPacketCounter struct.

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  asapersson@webrtc.org

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• Fix bug in thresholds for bitrate probing and adjust thresholds to allow a larger dispersion and concentration for successful probes.

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  stefan@webrtc.org

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• Make iSAC SWB own its decoder

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  henrik.lundin@webrtc.org

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• Fix a use-after-free when sending queued messages is aborted for blocked channel.

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  jiayl@webrtc.org

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Commits on Jan 21, 2015

• Fix an unitialized variable warning.

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  andrew@webrtc.org

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• GN: Prepare to remove webrtc_base target

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  kjellander@webrtc.org

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• Re-land "Support 48kHz in AEC"

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  aluebs@webrtc.org

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• Fix TransientDetectorTest in modules_unittests on Android ARM64

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  aluebs@webrtc.org

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• Disable AcmSenderBitExactnessOldApi.Opus_stereo_20ms_voip on ARM64.

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  minyue@webrtc.org

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• Change CreateOrGetReportBlockInformation to have one return path.

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  asapersson@webrtc.org

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• Simplify and guard access to WindowsRealTimeClock.

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  pbos@webrtc.org

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• Update StatsReport and by extension StatsCollector to reduce data copying.

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  tommi@webrtc.org

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• Remove unnecessary dependencies from webrtc_all target.

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  kjellander@webrtc.org

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• Only report fraction of lost packets if report_block_stats has been updated.

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  asapersson@webrtc.org

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• Indentation changes.

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  asapersson@webrtc.org

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• Correct the class name in peerconnection_jni.cc.

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  braveyao@webrtc.org

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Commits on Jan 20, 2015

• Update libjingle license statements at top of talk files for consistency

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  jlmiller@webrtc.org

  committed
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• Bump to version 41.

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  tnakamura@webrtc.org

  committed
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• Setting Opus target application.

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  minyue@webrtc.org

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• Move internal capture+render to build_with_chromium==0 condition

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  kjellander@webrtc.org

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• Roll chromium_revision a6eafec..c086b4e

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  kjellander@webrtc.org

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• Revert 8080 "Support 48kHz in AEC"

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  tina.legrand@webrtc.org

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• Remove webrtc/base/compile_assert.h

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  kwiberg@webrtc.org

  committed
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• Cleanup for Rtp Rtcp API test.

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  changbin.shao@intel.com

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Commits on Jan 19, 2015

• Update StatsCollector's interface in preparation of more changes.

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  tommi@webrtc.org

  committed
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• Revert 8095 "Update StatsCollector's interface in preparation of..."

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  tommi@webrtc.org

  committed
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• Update StatsCollector's interface in preparation of more changes.

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  tommi@webrtc.org

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• Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps.

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  stefan@webrtc.org

  committed
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• Fixing LD_LIBRARY_PATH, improving safety for libjingle java unit test.

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  phoglund@webrtc.org

  committed
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• Adding TRYSERVER_PROJECT to codereview.settings.

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  kjellander@webrtc.org

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• Add /talk/examples/androidtests/{bin,gen} to .gitignore.

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  kjellander@webrtc.org

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• Disable tests failing on Android ARM64 (Nexus9).

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  kjellander@webrtc.org

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• Disable WebRtcVideoMediaChannelSimulcastTest::SimulcastSend_* on tsan.

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  sprang@webrtc.org

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• Remove unused private data member engine_id_

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  tommi@webrtc.org

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