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webrtc_m130/webrtc
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henrika 90ee13ce1c Changes construction of AudioTrack for API 21 and higher. 
If we are on API level 21 or higher, it is possible to use a special AudioTrack
constructor that uses AudioAttributes and AudioFormat as input. It allows us to
supersede the notion of stream types for defining the behavior of audio playback,
and to allow certain platforms or routing policies to use this information for more
refined volume or routing decisions.

BUG=b/30018586

Review-Url: https://codereview.webrtc.org/2804013002
Cr-Commit-Position: refs/heads/master@{#17737}
2017-04-18 12:07:52 +00:00
..
api
Modified the rtp_receiver_unittests.
2017-04-11 18:28:10 +00:00
audio
Fix lint errors to enable stricter PyLint rules
2017-04-12 19:06:13 +00:00
base
base/testutils.h: Use namespace webrtc::testing instead of plain testing
2017-04-18 10:18:22 +00:00
call
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
common_audio
Replace Clock with timeutils in AudioEncoder.
2017-04-18 07:11:48 +00:00
common_video
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
examples
Address denicija's comments for AppRTCMobile video codec setting.
2017-04-11 12:36:43 +00:00
logging
Delay based logging.
2017-04-11 08:50:23 +00:00
media
Fix SDP stream ID mismatch issue when a track's stream changes.
2017-04-14 17:41:32 +00:00
modules
Changes construction of AudioTrack for API 21 and higher.
2017-04-18 12:07:52 +00:00
ortc
Enabling 'gn check' on //webrtc/ortc.
2017-04-11 09:52:49 +00:00
p2p
Fix compilation issues of std::unique_ptr
2017-04-09 16:09:06 +00:00
pc
Fix SDP stream ID mismatch issue when a track's stream changes.
2017-04-14 17:41:32 +00:00
sdk
Fix poor seed for Android HW encoder picture ID randomization.
2017-04-13 19:25:56 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Rename COMPILE_ASSERT macro to RTC_COMPILE_ASSERT
2017-04-11 06:21:43 +00:00
test
Revert of Enabling 'gn check' on //webrtc/test. (patchset #10 id:180001 of https://codereview.webrtc.org/2796363003/ )
2017-04-13 09:13:09 +00:00
tools
Revert of "Add first part of the network_tester functionality" (patchset #1 id:1 of https://codereview.webrtc.org/2811253005/ )
2017-04-13 10:03:51 +00:00
video
Disable EndToEndTest.ContentTypeSwitches test on Windows due to flakiness
2017-04-13 21:09:57 +00:00
voice_engine
Replace Clock with timeutils in AudioEncoder.
2017-04-18 07:11:48 +00:00
.gitignore
…
BUILD.gn
Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ )
2017-04-07 07:59:12 +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
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
common_types.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
config.cc
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
config.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
DEPS
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +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
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_decoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
video_encoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
video_frame.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
video_receive_stream.h
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +00:00
video_send_stream.h
Add number of quality adapt changes to VideoSendStream stats.
2017-04-04 12:51:49 +00:00
webrtc.gni
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
2017-03-04 03:41:59 +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.