pbos@webrtc.org a7f77720cb Merge in AGC manager and AGC tools. ...

R=bjornv@webrtc.org
BUG=4098

Review URL: https://webrtc-codereview.appspot.com/37379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7902 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-12-15 16:33:16 +00:00

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audio_coding

Add WebRtcIsacfix_FilterMaLoopNeon's intrinsics version.

2014-12-15 07:23:49 +00:00

audio_conference_mixer

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audio_device

Attempt to fix FYI bots.

2014-12-09 19:59:27 +00:00

audio_processing

Merge in AGC manager and AGC tools.

2014-12-15 16:33:16 +00:00

bitrate_controller

Change all system clock types to int64_t in bitrate_controller.

2014-12-08 19:46:23 +00:00

desktop_capture

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interface

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

media_file

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

pacing

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

remote_bitrate_estimator

Add support for parsing header only RTP dumps with bwe_rtp_play.

2014-12-04 15:43:49 +00:00

rtp_rtcp

Add video send bitrates to histogram stats:

2014-12-09 09:47:53 +00:00

utility

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

video_capture

OWNERS: Remove tomasl@ and mallinath@

2014-11-24 10:05:05 +00:00

video_coding

Remove jitter_estimate_test.h

2014-12-10 21:08:39 +00:00

video_processing

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

video_render

OWNERS: Remove tomasl@ and mallinath@

2014-11-24 10:05:05 +00:00

module_common_types_unittest.cc

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modules_java_chromium.gyp

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modules_java.gyp

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modules_tests.isolate

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modules_unittests.isolate

Removes unused test files by audio_processing/transient

2014-12-15 16:13:05 +00:00

modules.gyp

Merge audio_processing changes.

2014-12-15 09:41:24 +00:00

OWNERS

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