webrtc_m130

History

sergeyu 6dbbd89f18 Fix BitrateProber to match the requested bitrate more precisely

Previously BirateProber was calculating delay between probes based on
the size of the previous probe. Because of that the actual sent bitrate
can deviate greatly from the target value. With this change it uses
total number of bytes in the cluster to estimate delay before each
probe.

BUG=webrtc:6952

Review-Url: https://codereview.webrtc.org/2613543003
Cr-Original-Commit-Position: refs/heads/master@{#15971}
Committed: 599c5011e7
Review-Url: https://codereview.webrtc.org/2613543003
Cr-Commit-Position: refs/heads/master@{#16127}

2017-01-17 23:07:59 +00:00

androidjunit

Format all Java in WebRTC.

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

api

RTCMediaStreamTrackStats.kind added and collected.

2017-01-17 10:53:23 +00:00

audio

Fix for bwe with overhead on audio only calls.

2017-01-17 07:55:07 +00:00

base

Remove dependency to Chromium code from WebRTC Java code.

2017-01-17 14:46:34 +00:00

build

Replace default locale with US locale on Android.

2017-01-16 12:57:32 +00:00

call

Let FlexfecReceiveStreamImpl send RTCP RRs.

2017-01-17 09:33:54 +00:00

common_audio

Replace RTC_DCHECK(false) with RTC_NOTREACHED().

2017-01-12 10:24:27 +00:00

common_video

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

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

examples

Android: AppRTCMobile: Don't leak CallActivity.

2017-01-17 15:25:53 +00:00

logging

GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false.

2017-01-17 10:37:34 +00:00

media

Revert of Fix flaky WebRtcVideoChannel2BaseTest.GetStats T (patchset #1 id:1 of https://codereview.webrtc.org/2634273002/ )

2017-01-17 13:56:53 +00:00

modules

Fix BitrateProber to match the requested bitrate more precisely

2017-01-17 23:07:59 +00:00

p2p

Remove unused lambda capture to unbreak downstream code.

2017-01-17 14:18:10 +00:00

Provide better message for when RTCP mux "require" policy is triggered.

2017-01-13 19:34:39 +00:00

sdk

Move file capturer/renderer tests to the correct location.

2017-01-17 11:32:06 +00:00

stats

RTCMediaStreamTrackStats.kind added and collected.

2017-01-17 10:53:23 +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

Let FlexfecReceiveStreamImpl send RTCP RRs.

2017-01-17 09:33:54 +00:00

tools

Comparison of videos with reference frame not starting from zero

2017-01-17 11:24:57 +00:00

video

Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )

2017-01-17 12:03:53 +00:00

voice_engine

UdpTransport:IsIpAddressValid: Added extra :: check for ipv6

2017-01-17 10:31:37 +00:00

.gitignore

…

BUILD.gn

Rename base/analytics/ to base/numerics/

2017-01-13 17:14:33 +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

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

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

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +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.