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Guo-wei Shieh a6c86b23fe Revert "Enable IPv6 temporary address filtering on iOS." 
This reverts commit 29488c23644721c10a45eba74c67602b0262e582.

This broke chromium.fyi bot.

TBR=pthatcher@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1547883002 .

Cr-Commit-Position: refs/heads/master@{#11115}
2015-12-23 04:17:37 +00:00
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api
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audio
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base
Revert "Enable IPv6 temporary address filtering on iOS."
2015-12-23 04:17:37 +00:00
build
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call
Step 1 to prepare call_test.* for combined audio/video tests.
2015-12-21 11:14:05 +00:00
common_audio
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common_video
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examples
Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
2015-12-19 18:14:18 +00:00
libjingle
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modules
Adding bit exactness test for Opus decoding in NetEq.
2015-12-22 17:57:47 +00:00
p2p
Roll chromium_revision 1b6c421..db567a8 (365999:366304)
2015-12-20 20:25:17 +00:00
sound
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system_wrappers
Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that these are for infrequent updates.
2015-12-21 09:46:25 +00:00
test
Step 1 to prepare call_test.* for combined audio/video tests.
2015-12-21 11:14:05 +00:00
tools
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video
Disable VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly
2015-12-21 21:35:00 +00:00
voice_engine
Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
2015-12-19 18:14:18 +00:00
.gitignore
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audio_receive_stream.h
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audio_send_stream.h
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audio_state.h
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BUILD.gn
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call.h
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codereview.settings
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common_types.cc
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common_types.h
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common.gyp
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common.h
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config.cc
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config.h
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engine_configurations.h
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frame_callback.h
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LICENSE
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LICENSE_THIRD_PARTY
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OWNERS
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PATENTS
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PRESUBMIT.py
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README.chromium
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rtc_unittests.isolate
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stream.h
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supplement.gypi
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transport.h
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typedefs.h
Defines for ARM and MIPS CPU types.
2015-12-18 12:28:49 +00:00
video_decoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_encoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_engine_tests.isolate
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video_frame.h
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video_receive_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_renderer.h
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video_send_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
webrtc_examples.gyp
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webrtc_nonparallel_tests.isolate
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webrtc_perf_tests.isolate
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webrtc_tests.gypi
Step 1 to prepare call_test.* for combined audio/video tests.
2015-12-21 11:14:05 +00:00
webrtc.gyp
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README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.