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minyue 25f6a39181 Relanding of "Adding debug dump to audio network adaptor." 
The original CL was https://codereview.webrtc.org/2356763002

but got reverted https://codereview.webrtc.org/2362003002/.

The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests.

BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2365723002
Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-23 05:23:28 +00:00
..
androidjunit
Make android_junit_tests use LocalRobolectricTestRunner
2016-09-18 19:28:15 +00:00
api
Support more QCOM specific color formats for Android HW decoder.
2016-09-22 17:44:34 +00:00
audio
Set the event log in Channel from AudioSendStream. This will re-enable logging of outgoing audio packets.
2016-09-21 13:51:52 +00:00
base
Revert of Allow the DTLS fingerprint verification to occur after the handshake. (patchset #11 id:200001 of https://codereview.webrtc.org/2163683003/ )
2016-09-20 00:21:00 +00:00
build
Roll chromium_revision cf9457edb7..cede888c27 (416297:419407)
2016-09-20 20:23:12 +00:00
call
Set the event log in Channel from AudioSendStream. This will re-enable logging of outgoing audio packets.
2016-09-21 13:51:52 +00:00
common_audio
Compensate for the IntelligibilityEnhancer processing delay in high bands
2016-09-20 21:52:08 +00:00
common_video
Revert of Move MutableDataY{,U,V} methods to I420Buffer only. (patchset #14 id:260001 of https://codereview.webrtc.org/2278883002/ )
2016-09-21 10:52:21 +00:00
examples
Revert of Rename AppRTCDemo on Android and iOS to AppRTCMobile (patchset #2 id:20001 of https://codereview.webrtc.org/2343403002/ )
2016-09-22 17:15:43 +00:00
libjingle
GN: Change group deps to public_deps.
2016-09-15 07:53:34 +00:00
media
GN: Remove non-existing header file: media/base/videorenderer.h
2016-09-22 11:51:57 +00:00
modules
Relanding of "Adding debug dump to audio network adaptor."
2016-09-23 05:23:28 +00:00
p2p
Fix a stun attribute leak.
2016-09-22 16:52:30 +00:00
pc
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
sdk
GN: Enable rtc_common_config for more targets.
2016-09-18 15:12:36 +00:00
stats
RTCTestStats moved into webrtc/stats/test (rtc_stats_test_utils)
2016-09-19 13:06:05 +00:00
system_wrappers
Roll chromium_revision cf9457edb7..cede888c27 (416297:419407)
2016-09-20 20:23:12 +00:00
test
Add custom info.plist to modules_unittests
2016-09-22 15:00:57 +00:00
tools
Added graph for plotting the audio level from an Rtc event log.
2016-09-22 14:01:55 +00:00
video
Revert of Move MutableDataY{,U,V} methods to I420Buffer only. (patchset #14 id:260001 of https://codereview.webrtc.org/2278883002/ )
2016-09-21 10:52:21 +00:00
voice_engine
Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ )
2016-09-22 10:36:34 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Revert of Rename AppRTCDemo on Android and iOS to AppRTCMobile (patchset #2 id:20001 of https://codereview.webrtc.org/2343403002/ )
2016-09-22 17:15:43 +00:00
call.h
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
2016-09-01 08:17:43 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
common_types.h
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
common.gyp
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
DEPS
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
engine_configurations.h
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Add codec name to CodecSpecificInfo and get the codec name stats from there instead.
2016-09-01 07:21:18 +00:00
video_frame.h
Introduce webrtc::VideoFrame::timestamp_us, and corresponding constructor.
2016-09-05 07:51:25 +00:00
video_receive_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
video_send_stream.h
Replace VideoCapturerInput with VideoSinkInterface.
2016-09-16 14:53:48 +00:00
webrtc_examples.gyp
Revert of Rename AppRTCDemo on Android and iOS to AppRTCMobile (patchset #2 id:20001 of https://codereview.webrtc.org/2343403002/ )
2016-09-22 17:15:43 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00

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.