platform-external-webrtc

Files

deadbeef b5388d7748 Move rtc_api_unittests into rtc_unittests.

This avoids adding an additional test target. Plus, everything in
rtc_api_unittests is (and likely will be) simple utility classes akin to
what's already being tested in rtc_unittests.

BUG=None
TBR=kjellander@webrtc.org

Review-Url: https://codereview.webrtc.org/2709573003
Cr-Commit-Position: refs/heads/master@{#16819}

2017-02-24 09:17:43 +00:00

api

Move rtc_api_unittests into rtc_unittests.

2017-02-24 09:17:43 +00:00

audio

Replace AudioReceiveStream::DeliverRtp with OnRtpPacket.

2017-02-21 14:28:10 +00:00

base

Making AsyncInvoker destructor thread-safe.

2017-02-24 01:10:07 +00:00

call

Introduce a new constructor to PlatformThread.

2017-02-22 19:22:05 +00:00

common_audio

Fix for left shift of potentially negative value.

2017-02-09 11:05:59 +00:00

common_video

Introduce a new constructor to PlatformThread.

2017-02-22 19:22:05 +00:00

examples

Remove non-ARC code from the codebase.

2017-02-22 08:42:11 +00:00

logging

Introduce a new constructor to PlatformThread.

2017-02-22 19:22:05 +00:00

media

Fix issue where video scaling gets stuck at low resolution

2017-02-22 17:30:27 +00:00

modules

Skips the first frame in DxgiDuplicatorController

2017-02-24 05:43:40 +00:00

ortc

Adding placeholder ortc_unittests target.

2017-02-23 07:45:38 +00:00

p2p

Accept SDP with TRANSPORT attributes missing from bundled m= sections.

2017-02-23 03:35:18 +00:00

Accept SDP with TRANSPORT attributes missing from bundled m= sections.

2017-02-23 03:35:18 +00:00

sdk

Added support for changing the volume of RTCAudioSource as discussed in BUG=webrtc:6533

2017-02-23 21:57:00 +00:00

stats

RTCRTPStreamStats.mediaTrackId renamed to trackId.

2017-01-27 14:35:16 +00:00

system_wrappers

Introduce a new constructor to PlatformThread.

2017-02-22 19:22:05 +00:00

test

Reland of Add optional visualization file writers to VideoProcessor tests. (patchset #1 id:1 of https://codereview.webrtc.org/2708103002/ )

2017-02-22 09:26:59 +00:00

tools

Propagate packet pacing information to SendTimeHistory.

2017-02-23 10:56:13 +00:00

video

Fixed Full stack tests to correctly process selected TL and SL while

2017-02-23 13:07:56 +00:00

voice_engine

Propagate packet pacing information to SendTimeHistory.

2017-02-23 10:56:13 +00:00

.gitignore

…

BUILD.gn

Move rtc_api_unittests into rtc_unittests.

2017-02-24 09:17:43 +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

Remove the Windows Wave audio device implementation.

2017-02-17 22:48:07 +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

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PRESUBMIT.py

…

README.chromium

…

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.