platform-external-webrtc

Files

kwiberg 09f090c40e Remove workaround for bug 6986

BUG=webrtc:6986

Review-Url: https://codereview.webrtc.org/2681733002
Cr-Commit-Position: refs/heads/master@{#16933}

2017-03-01 09:57:11 +00:00

api

Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

2017-03-01 09:02:45 +00:00

audio

Replace NULL with nullptr or null in webrtc/audio/ and common_audio/.

2017-02-26 12:18:12 +00:00

base

Revert of Use sched_yield on all POSIX platforms in PlatformThread. (patchset #1 id:1 of https://codereview.webrtc.org/2725573002/ )

2017-02-28 14:20:38 +00:00

call

Reland Move fake_audio_device to its own target.

2017-02-28 22:41:05 +00:00

common_audio

Replace NULL with nullptr or null in webrtc/audio/ and common_audio/.

2017-02-26 12:18:12 +00:00

common_video

Enable GN check for webrtc/common_video

2017-02-28 00:37:54 +00:00

examples

iOS AppRTCMobile: Fix SocketRocket bug

2017-02-27 13:20:18 +00:00

logging

Remove unused include from rtc_event_log_parser.cc

2017-02-28 09:46:19 +00:00

media

Split FlexFEC field trial in two.

2017-02-28 23:43:10 +00:00

modules

Disable AudioCoding tests on ios

2017-03-01 09:16:51 +00:00

ortc

Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

2017-02-26 02:15:09 +00:00

p2p

RTCIceCandidatePairStats.[total/current]RoundTripTime collected.

2017-02-28 14:34:47 +00:00

Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

2017-03-01 09:02:45 +00:00

sdk

Split FlexFEC field trial in two.

2017-02-28 23:43:10 +00:00

stats

Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

2017-03-01 09:02:45 +00:00

system_wrappers

Provide an inline implementation of IsEnabled.

2017-02-28 17:26:22 +00:00

test

Reland Move fake_audio_device to its own target.

2017-02-28 22:41:05 +00:00

tools

Add probe logging to RtcEventLog.

2017-02-27 10:18:46 +00:00

video

Recreate WebrtcVideoSendStream if screen content setting is changed.

2017-02-28 21:23:26 +00:00

voice_engine

Remove workaround for bug 6986

2017-03-01 09:57:11 +00:00

.gitignore

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BUILD.gn

Revert of Removing unused audio device tests (patchset #1 id:1 of https://codereview.webrtc.org/2716843002/ )

2017-02-26 15:12:50 +00:00

call.h

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

codereview.settings

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common_types.cc

Reject XR TargetBitrate items with unsupported layer indices

2016-12-06 14:09:00 +00:00

common_types.h

Remove the Windows Wave audio device implementation.

2017-02-17 22:48:07 +00:00

config.cc

Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

2017-02-26 02:15:09 +00:00

config.h

Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

2017-02-26 02:15:09 +00:00

DEPS

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

LICENSE

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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

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PRESUBMIT.py

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README.chromium

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typedefs.h

Only define NO_RETURN if undefined

2017-01-30 08:54:19 +00:00

video_decoder.h

Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )

2017-02-09 12:53:45 +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

Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )

2017-02-09 12:53:45 +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

Move iOS tests to XCTest from gtest.

2017-02-14 12:58:56 +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.