zhihuang 95a64ca8aa Revert of Verify sender ssrc when receiving rtcp target bitrate (patchset #3 id:40001 of https://codereview.webrtc.org/3000373002/ ) ...

Reason for revert:
Might be the root cause of an internal test failure.

Original issue's description:
> Verify sender ssrc when receiving rtcp target bitrate
>
> BUG=webrtc:8137
>
> Review-Url: https://codereview.webrtc.org/3000373002
> Cr-Commit-Position: refs/heads/master@{#19524}
> Committed: a7a4beb419

TBR=danilchap@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:8137

Review-Url: https://codereview.webrtc.org/3005633002
Cr-Commit-Position: refs/heads/master@{#19529}

2017-08-25 19:59:42 +00:00

..

audio_coding

Reimplement the builtin audio codec factories using the new stuff in api/

2017-08-25 10:10:50 +00:00

audio_conference_mixer

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

audio_device

Removes usage audio_device_test_api.

2017-08-25 14:47:31 +00:00

audio_mixer

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

audio_processing

Decrease the default reverberation in AEC3

2017-08-25 14:07:30 +00:00

bitrate_controller

Final version of BBR, with tweaks made for WebRTC, major changes:

2017-08-20 16:19:58 +00:00

congestion_controller

Use small BWE period when there is a true network degradation.

2017-08-23 17:22:28 +00:00

desktop_capture

Check whether on_screen is null before performing CFBooleanGetValue()

2017-08-25 18:14:14 +00:00

include

Add AudioFrame::ResetWithoutMuting() to address performance regression.

2017-07-08 23:36:26 +00:00

media_file

Replace WEBRTC_TRACE logging in modules/media_file/

2017-08-23 17:30:38 +00:00

pacing

Move PacedSender ownership to RtpTransportControllerSend.

2017-08-22 14:16:49 +00:00

remote_bitrate_estimator

Use small BWE period when there is a true network degradation.

2017-08-23 17:22:28 +00:00

rtp_rtcp

Revert of Verify sender ssrc when receiving rtcp target bitrate (patchset #3 id:40001 of https://codereview.webrtc.org/3000373002/ )

2017-08-25 19:59:42 +00:00

utility

Tracking mock_process_thread with a GN target

2017-08-03 12:57:11 +00:00

video_capture

Replace old logging mechanism in webrtc/modules/video_capture/linux

2017-08-16 13:51:27 +00:00

video_coding

Delete batch mode from VideoProcessorIntegrationTest.

2017-08-25 14:11:19 +00:00

video_processing

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

BUILD.gn

Remove remains of webrtc/base

2017-07-19 17:40:47 +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