platform-external-webrtc

Files

asapersson f4e44af724 Do not report cpu limited resolution stats when degradation preference is disabled and no scaling is done.

When degradation preference is kDegradationDisabled, do not update WebRTC.Video.CpuLimitedResolutionInPercent.

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2807133002
Cr-Commit-Position: refs/heads/master@{#17757}

2017-04-19 09:01:06 +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

Delete VieRemb class, move functionality to PacketRouter.

2017-04-19 06:38:35 +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

Add error codes for Android audio recording start errors.

2017-04-18 17:30:29 +00:00

logging

Delay based logging.

2017-04-11 08:50:23 +00:00

media

Relanding: Remove rtc_p2p_unittests from ortc_unittests and rtc_media_unittests

2017-04-18 22:49:09 +00:00

modules

Delete VieRemb class, move functionality to PacketRouter.

2017-04-19 06:38:35 +00:00

ortc

Relanding: Remove rtc_p2p_unittests from ortc_unittests and rtc_media_unittests

2017-04-18 22:49:09 +00:00

p2p

Relanding: Remove rtc_p2p_unittests from ortc_unittests and rtc_media_unittests

2017-04-18 22:49:09 +00:00

Adding integration test for unsignaled inbound RTP stream stats.

2017-04-18 23:01:17 +00:00

sdk

Add Java binding for new getStats implementation.

2017-04-18 17:27:51 +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

Reland "Add first part of the network_tester functionality".

2017-04-19 08:58:38 +00:00

video

Do not report cpu limited resolution stats when degradation preference is disabled and no scaling is done.

2017-04-19 09:01:06 +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

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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

Delete VieRemb class, move functionality to PacketRouter.

2017-04-19 06:38:35 +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.