webrtc_m130

History

BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2686033004
Cr-Commit-Position: refs/heads/master@{#16518}

2017-02-09 16:32:53 +00:00

api

Adding more comments to every header file in api/ subdirectory.

2017-02-08 09:38:21 +00:00

audio

Wire up audio packet loss to BWE.

2017-02-07 15:14:08 +00:00

base

Use fake clock in some more networks tests.

2017-02-08 21:18:00 +00:00

call

Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.

2017-02-07 09:18:43 +00:00

common_audio

Fix for left shift of potentially negative value.

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

common_video

Reland of Added VP8 simulcast tests. Fixed analyzer to correctly infer timestamps. (patchset #2 id:150001 of https://codereview.webrtc.org/2687073002/ )

2017-02-09 12:58:53 +00:00

examples

Move Android video quality loopback script.

2017-02-09 09:05:54 +00:00

logging

Add option to print information about configured SSRCs from RTC event logs.

2017-02-08 17:37:30 +00:00

media

Added a flag to AudioCodecSpec to indicate adaptive bitrate support.

2017-02-09 13:14:32 +00:00

modules

Remove rtcp_utility as mostly unused.

2017-02-09 13:21:42 +00:00

p2p

Increase STUN RTOs to work better on poor networks, such as 2G networks.

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

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

sdk

Roll chromium_revision cf2dce6a6d..2019b9e075 (448969:449230)

2017-02-09 07:56:43 +00:00

stats

RTCRTPStreamStats.mediaTrackId renamed to trackId.

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

system_wrappers

Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )

2017-01-24 14:58:22 +00:00

test

Add new target direct_transport and remove fake_network and direct_transport from test_common.

2017-02-08 13:55:51 +00:00

tools

Move Android video quality loopback script.

2017-02-09 09:05:54 +00:00

video

Fixing typo

2017-02-09 16:32:53 +00:00

voice_engine

Remove unused voe_stress_test.cc

2017-02-09 13:03:25 +00:00

.gitignore

Flesh out webrtc/.gitignore

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

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

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

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

Adding build switch for Opus that supports 120ms ptime.

2017-02-02 01:31:11 +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.