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Commits on Sep 24, 2014

• Enable render downmixing to mono in AudioProcessing.

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  andrew@webrtc.org

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• Add missing DesktopConfigurationMonitor Unlock in webrtc::ScreenCapturerMac

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  jiayl@webrtc.org

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• Fix a problem in Thread::Send.

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  jiayl@webrtc.org

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• Call NS AnalyzeCaptureAudio before AEC

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  aluebs@webrtc.org

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• Reduce jitter delay for low fps streams.

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• Moved the filter calculation from analyze to process in ns_core

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• audioproc: Now also writes to output file in simulation mode

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  bjornv@webrtc.org

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• WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t

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  kwiberg@webrtc.org

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• Thread annotation of rtc::CriticalSection.

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  pbos@webrtc.org

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• Move thread_annotations.h to webrtc/base/.

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Commits on Sep 23, 2014

• Change Android video renderer to maintain video aspect

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  glaznev@webrtc.org

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• Switch HW video decoder to output byte buffers if video

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• (Auto)update libjingle 76169599-> 76176062

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  buildbot@webrtc.org

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• Use VPX_IMG_FMT_*/VPX_PLANE_* defines

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  johannkoenig@google.com

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• Enable ipv6 by default for webrtc under a Finch experiment.

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  guoweis@webrtc.org

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• Revert "Remove DTMF status methods from Voice Engine" r7276

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  henrik.lundin@webrtc.org

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• Remove DTMF status methods from Voice Engine

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  henrik.lundin@webrtc.org

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• Revert "Set minimum SDK level to 10.7 for Mac and iOS" (r7175)

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  kjellander@webrtc.org

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• gn: Hide modules/video_capture:video_capture_internal_impl behind an arg

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  pbos@webrtc.org

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• Reland "Converting five tests to use new AudioCoding interface" (r7258)

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  henrik.lundin@webrtc.org

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• Reland (rev 7259) "Convert AcmReceiverTest to new AudioCoding interface"

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• audio_processing/agc: Solved building with AGC_DEBUG + few style changes

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  bjornv@webrtc.org

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• Skeleton for registering external encoders/decoders.

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  pbos@webrtc.org

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• Unit tests for SSLAdapter

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• modules_unittests: Turned on ApmTest.Process test for Android

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  bjornv@webrtc.org

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• Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..."

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  andrew@webrtc.org

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Commits on Sep 22, 2014

• WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t

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  kwiberg@webrtc.org

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• Remove engine-level SetOptions.

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  pbos@webrtc.org

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• Revert "Converting five tests to use new AudioCoding interface" (rev 7258).

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  andresusanopinto

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• Adding test file path as argument of the rtcBot run command's arguments.

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  houssainy@google.com

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• Remove Get/SetNetEQPlayoutMode APIs

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  henrik.lundin@webrtc.org

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• Adding webrtc_video_streaming test

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  houssainy@google.com

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• Revert "Convert AcmReceiverTest to new AudioCoding interface" (rev 7258).

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• Convert AcmReceiverTest to new AudioCoding interface

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  henrik.lundin@webrtc.org

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• Converting five tests to use new AudioCoding interface

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  henrik.lundin@webrtc.org

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