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platform-external-webrtc/webrtc
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henrika 447dba9586 Add debuggable=true to AppRTCMobile manifest. 
Improves possibility to debug the demo application using adb.
As an example, now allows 'adb run as' which is useful when profiling CPU. It enables us to do profiling on non-rooted devices (excluding details).

BUG=NONE

Review-Url: https://codereview.webrtc.org/2676983003
Cr-Commit-Position: refs/heads/master@{#16446}
2017-02-06 12:58:01 +00:00
..
api
Adding C++ versions of currently spec'd "RtpParameters" structs.
2017-02-04 20:09:01 +00:00
audio
Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ )
2017-02-06 10:23:00 +00:00
base
Pick the DTLS handshake timeout based on the ICE RTT estimate
2017-02-03 01:18:37 +00:00
call
Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ )
2017-02-06 10:23:00 +00:00
common_audio
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
2017-01-24 14:58:22 +00:00
common_video
Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common.
2017-02-06 10:21:11 +00:00
examples
Add debuggable=true to AppRTCMobile manifest.
2017-02-06 12:58:01 +00:00
logging
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ )
2017-01-27 12:53:07 +00:00
media
Adding C++ versions of currently spec'd "RtpParameters" structs.
2017-02-04 20:09:01 +00:00
modules
Ensure that AEC3 is not run in tandem with AEC2
2017-02-06 11:39:42 +00:00
p2p
Fix Chromium FYI bot
2017-02-03 02:22:46 +00:00
pc
Adding C++ versions of currently spec'd "RtpParameters" structs.
2017-02-04 20:09:01 +00:00
sdk
Camera2Session: Add return statements after reportError where needed.
2017-02-06 12:55:21 +00:00
stats
RTCRTPStreamStats.mediaTrackId renamed to trackId.
2017-01-27 14:35:16 +00:00
system_wrappers
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
2017-01-24 14:58:22 +00:00
test
Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common.
2017-02-06 10:21:11 +00:00
tools
Remove all occurrences of "using std::string".
2017-02-02 16:10:00 +00:00
video
Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ )
2017-02-06 10:23:00 +00:00
voice_engine
Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common.
2017-02-06 10:21:11 +00:00
.gitignore
…
BUILD.gn
Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common.
2017-02-06 10:21:11 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
config.cc
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
config.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Remove temporary AddRtxInfo member function.
2017-02-06 10:34:25 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Adding build switch for Opus that supports 120ms ptime.
2017-02-02 01:31:11 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.