Minyue Li ce433fafc1 Revert "Adding ANA config event to debug dump." ...

This reverts commit 652abc9a472426367e149db5a101b894179687aa.

Reason for revert: break upstream bots

Original change's description:
> Adding ANA config event to debug dump.
> 
> BUG=webrtc:7854
> 
> Change-Id: I12c33b8558fd49374a55282c391b87fde9e13a28
> Reviewed-on: https://chromium-review.googlesource.com/535554
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Michael T <tschumim@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18661}

TBR=minyue@webrtc.org,ossu@webrtc.org,tschumim@webrtc.org

Change-Id: Id1f93338e431c9cd8dade722be7edd16a648d044
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7854
Reviewed-on: https://chromium-review.googlesource.com/539737
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18663}

2017-06-19 15:23:02 +00:00

..

audio_coding

Revert "Adding ANA config event to debug dump."

2017-06-19 15:23:02 +00:00

audio_conference_mixer

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

audio_device

Reduces sensitivity in audio-glitch detector for iOS

2017-06-16 12:09:10 +00:00

audio_mixer

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

audio_processing

This CL finalizes the Conversational Speech tool.

2017-06-15 10:49:57 +00:00

bitrate_controller

Tune loss-based BWE to be more compatible with the low frequency loss reports of audio streams.

2017-06-16 14:47:00 +00:00

congestion_controller

Remove redundant std::min from ProbeBitrateEstimator.

2017-06-19 13:07:30 +00:00

desktop_capture

desktopCapture: scale the cursor image according to screen scale factor on OSX

2017-06-01 21:27:41 +00:00

include

Implement timing frames.

2017-06-19 14:18:55 +00:00

media_file

Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ )

2017-04-26 10:38:35 +00:00

pacing

Use RaceChecker instead of ThreadChecker in a few places.

2017-06-16 12:02:05 +00:00

remote_bitrate_estimator

Plot acknowledged bitrate when compiled with rtc_enable_bwe_test_logging.

2017-06-14 09:41:59 +00:00

rtp_rtcp

Implement timing frames.

2017-06-19 14:18:55 +00:00

utility

Remove deprecation warning from JVM::Initialize with the context parameter.

2017-06-01 11:59:52 +00:00

video_capture

Update webrtc/media and webrtc/modules to new VideoFrameBuffer interface

2017-06-14 13:46:38 +00:00

video_coding

Implement timing frames.

2017-06-19 14:18:55 +00:00

video_processing

Update webrtc/media and webrtc/modules to new VideoFrameBuffer interface

2017-06-14 13:46:38 +00:00

BUILD.gn

This is an initial cl, which contains small amount of implemented functions, and large amount of unimplemented ones.

2017-06-05 13:01:26 +00:00

module_common_types_unittest.cc

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

OWNERS

Remove all references to GYP

2016-11-16 19:11:38 +00:00