platform-external-webrtc

Files

sprang 3ebabf1c29 Screen content simulcast layers should not be downscaled.

Fix config so, size isn't downscaled, add unit test coverage.

BUG=webrtc:7171, webrtc:4172

Review-Url: https://codereview.webrtc.org/2692343007
Cr-Commit-Position: refs/heads/master@{#16648}

2017-02-16 15:35:22 +00:00

api

Re-land of RTCInboundRTPStreamStats.qpSum collected.

2017-02-16 13:34:48 +00:00

audio

Remove VoEVideoSync interface.

2017-02-15 08:42:31 +00:00

base

Delete unused files macconversion.h and .cc.

2017-02-15 09:07:57 +00:00

call

Rename flexfec AddAndProcessReceivedPacket --> OnRtpPacket.

2017-02-16 14:52:32 +00:00

common_audio

Fix for left shift of potentially negative value.

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

common_video

Added Vp9 simulcast tests.

2017-02-15 10:23:28 +00:00

examples

Modify android video_quality_loopback_test to run commands from the src dir.

2017-02-16 15:20:26 +00:00

logging

Move AudioDecoder and related stuff to the api/ directory

2017-02-10 16:15:44 +00:00

media

Screen content simulcast layers should not be downscaled.

2017-02-16 15:35:22 +00:00

modules

Rename flexfec AddAndProcessReceivedPacket --> OnRtpPacket.

2017-02-16 14:52:32 +00:00

p2p

Fix the build break by keeping the old Port::AddAddress method since the

2017-02-14 01:33:27 +00:00

Re-land of RTCInboundRTPStreamStats.qpSum collected.

2017-02-16 13:34:48 +00:00

sdk

Roll chromium_revision 69e724195b..3dd2a5021d (450712:450867)

2017-02-16 09:06:04 +00:00

stats

RTCRTPStreamStats.mediaTrackId renamed to trackId.

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

system_wrappers

Change NtpTime representation to single uint64_t

2017-02-15 09:18:15 +00:00

test

Rename flexfec AddAndProcessReceivedPacket --> OnRtpPacket.

2017-02-16 14:52:32 +00:00

tools

Better comparison of videos with barcode errors

2017-02-16 09:36:43 +00:00

video

Added Vp9 simulcast tests.

2017-02-15 10:23:28 +00:00

voice_engine

Remove VoEVideoSync interface.

2017-02-15 08:42:31 +00:00

.gitignore

…

BUILD.gn

Added gn target for rtc_event_log2rtp_dump.

2017-02-08 08:28:09 +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 VoEExternalMedia interface.

2017-02-14 12:55:00 +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

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.