aleloi a485dabc78 Revert of Made MixerAudioSource a pure interface. (patchset #7 id:350001 of https://codereview.webrtc.org/2396483002/ ) ...

Reason for revert:
breaks chromium FYI

Original issue's description:
> Made MixerAudioSource a pure interface.
>
> This required quite a few small changes in the mixing algorithm
> structure, the mixer interface and the mixer unit tests.
>
> BUG=webrtc:6346
>
> Committed: https://crrev.com/2ae5fdff86b784545cbd724de54bb5ffedde1adf
> Cr-Commit-Position: refs/heads/master@{#14567}

TBR=ivoc@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6346

Review-Url: https://codereview.webrtc.org/2394253003
Cr-Commit-Position: refs/heads/master@{#14568}

2016-10-07 12:04:58 +00:00

..

audio_coding

Delete old video defines in engine config.

2016-10-07 05:07:36 +00:00

audio_conference_mixer

Delete old video defines in engine config.

2016-10-07 05:07:36 +00:00

audio_device

Use std::abs instead of C-style abs.

2016-10-06 02:19:30 +00:00

audio_mixer

Revert of Made MixerAudioSource a pure interface. (patchset #7 id:350001 of https://codereview.webrtc.org/2396483002/ )

2016-10-07 12:04:58 +00:00

audio_processing

The FFT functionality in aec_rdft* is based on legacy C

2016-10-06 13:46:27 +00:00

bitrate_controller

Moved RtcEventLog files from call/ to logging/

2016-10-04 01:31:32 +00:00

congestion_controller

Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/

2016-10-01 05:29:53 +00:00

desktop_capture

Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/

2016-10-01 05:29:53 +00:00

include

Reland of Add pps id and sps id parsing to the h.264 depacketizer. (patchset #1 id:1 of https://codereview.webrtc.org/2265023002/ )

2016-09-02 11:07:35 +00:00

media_file

Delete old video defines in engine config.

2016-10-07 05:07:36 +00:00

pacing

Add AlrDetector

2016-10-04 15:43:15 +00:00

remote_bitrate_estimator

Make bwe_simulator a separate test target that doesnt run on the bots

2016-10-07 07:40:03 +00:00

rtp_rtcp

Use NtpTime in RtcpReceiver instead of pair of uints

2016-10-06 16:24:51 +00:00

utility

Delete old video defines in engine config.

2016-10-07 05:07:36 +00:00

video_capture

Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/

2016-10-01 05:29:53 +00:00

video_coding

Delete old video defines in engine config.

2016-10-07 05:07:36 +00:00

video_processing

Revert of Delete webrtc::VideoFrame::CopyFrame. (patchset #2 id:20001 of https://codereview.webrtc.org/2371363003/ )

2016-10-06 13:29:34 +00:00

BUILD.gn

Make bwe_simulator a separate test target that doesnt run on the bots

2016-10-07 07:40:03 +00:00

module_common_types_unittest.cc

Enable the -Wundef warning for clang

2016-09-29 00:42:08 +00:00

modules.gyp

GYP: Remove targets inside include_tests==1 that are converted to GN.

2016-09-15 11:57:39 +00:00

OWNERS

OWNERS: Make everyone able to change *.gn,*.gni files.

2016-09-09 12:51:48 +00:00