webrtc_m130

History

skvlad 1d3c7e0b16 Move MetricsObserverInterface out of peerconnectioninterface.h

This change moves the definition of the UMA MetricsObserverInterface from api/peerconnectioninterface.h into api/umametrics.h. This allows us to remove the unwanted dependency on peerconnectioninterface.h from files in webrtc/p2p.

This is a simple refactoring with no functional changes.

BUG=None

Review-Url: https://codereview.webrtc.org/2627093005
Cr-Commit-Position: refs/heads/master@{#16020}

2017-01-12 01:50:30 +00:00

androidjunit

Format all Java in WebRTC.

2016-09-29 11:12:51 +00:00

api

Move MetricsObserverInterface out of peerconnectioninterface.h

2017-01-12 01:50:30 +00:00

audio

GN: Refactor webrtc_nonparallel_tests and audio_tests to avoid crossing package boundaries.

2017-01-05 14:03:24 +00:00

base

Replace ASSERT(false) by RTC_NOTREACHED().

2017-01-11 13:56:46 +00:00

build

Disable some filesystem tests that don't make sense on iOS.

2017-01-10 11:00:41 +00:00

call

Clean up ramp-up tests and make sure they all pass.

2017-01-09 12:19:24 +00:00

common_audio

Fix for left shift of potentially negative value.

2017-01-10 11:37:20 +00:00

common_video

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

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

examples

Initial implementation of Android audio recording error handling.

2017-01-11 21:00:01 +00:00

logging

Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros.

2016-12-20 13:03:58 +00:00

media

Reland of: Adding error output param to SetConfiguration, using new RTCError type.

2017-01-11 20:28:30 +00:00

modules

Initial implementation of Android audio recording error handling.

2017-01-11 21:00:01 +00:00

p2p

Move MetricsObserverInterface out of peerconnectioninterface.h

2017-01-12 01:50:30 +00:00

Replace ASSERT(false) by RTC_NOTREACHED().

2017-01-11 13:56:46 +00:00

sdk

Update Android instrumentation tests to use JUnit4.

2017-01-11 14:22:56 +00:00

stats

RTCMediaStreamTrackStats.ssrcIds removed.

2017-01-11 12:47:23 +00:00

system_wrappers

Reland of Delete unused code from systeminfo. (patchset #1 id:1 of https://codereview.webrtc.org/2584563004/ )

2017-01-09 09:22:16 +00:00

test

Don't detect a new frame if a previous packet is used in a previous frame.

2017-01-11 10:01:56 +00:00

tools

Add method needed to extract frame capture and arrival timestamps from rtc event logs.

2016-12-20 16:51:52 +00:00

video

Make OverheadObserver::OnOverheadChanged count RTP headers only

2017-01-10 16:58:32 +00:00

voice_engine

Make OverheadObserver::OnOverheadChanged count RTP headers only

2017-01-10 16:58:32 +00:00

.gitignore

Flesh out webrtc/.gitignore

2015-11-16 19:02:02 +00:00

BUILD.gn

Reland of "Remove webrtc/libjingle/{xmllite,xmpp}"

2017-01-09 07:02:01 +00:00

call.h

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

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

codereview.settings

Give a more specific URL for creating WebRTC checkout

2016-04-01 07:36:53 +00:00

common_types.cc

Reject XR TargetBitrate items with unsupported layer indices

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

common_types.h

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

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

config.cc

Keep all codec parameters in VideoReceiveStream::Decoder

2016-11-25 11:56:41 +00:00

config.h

Keep all codec parameters in VideoReceiveStream::Decoder

2016-11-25 11:56:41 +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

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

Remove extra uses of basictypes.h.

2016-12-09 14:54:08 +00:00

video_decoder.h

Move VideoDecoder::Create() logic to separate internal video decoder factory

2016-12-01 08:27:35 +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

Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.

2016-11-28 15:02:19 +00:00

video_send_stream.h

Implement periodic bandwidth probing in application-limited region.

2016-11-28 21:11:24 +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.