webrtc_m130

History

stefan e9ad271db4 Increase the send-time history to 60 seconds.

This helps us avoid time-outs on really bad networks with long queues.
Also adding periodic logging of the fake network pipe's queue in milliseconds.

BUG=webrtc:5079

Review-Url: https://codereview.webrtc.org/2687013005
Cr-Commit-Position: refs/heads/master@{#16532}

2017-02-10 14:09:28 +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

Getting rid of "benign blocking error" log spam.

2017-02-09 23:09:22 +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

Allow residual echo detector to be enabled/disabled using AudioOptions, and no longer disable it on mobile platforms.

2017-02-10 13:11:09 +00:00

modules

Increase the send-time history to 60 seconds.

2017-02-10 14:09:28 +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 baaeb3f30e..ba3e8dd8fa (449272:449304)

2017-02-09 19:19:31 +00:00

stats

RTCRTPStreamStats.mediaTrackId renamed to trackId.

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

system_wrappers

replace NtpTime->Clock with Clock->NtpTime dependency

2017-02-09 19:15:25 +00:00

test

Increase the send-time history to 60 seconds.

2017-02-10 14:09:28 +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

…

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

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PRESUBMIT.py

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README.chromium

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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

Check for use_x11 before runnig desktop_capture_modules_tests on linux.

2017-02-10 09:38:23 +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.