webrtc_m130

History

ilnik e4350197ec Don't disable FEC if timing frames are disabled.

Don't disable fec for packets without timing frames extension
even if they are marked as belonging to timing frames.

BUG=webrtc:7894

Review-Url: https://codereview.webrtc.org/2956263002
Cr-Commit-Position: refs/heads/master@{#18826}

2017-06-29 09:27:48 +00:00

api

Expose ILBC codec in webrtc/api/audio_codecs/

2017-06-28 09:05:04 +00:00

audio

Disable AudioBweIntegrationTest.NoBandwidthDropAfterDtx - it's flaky

2017-06-26 12:04:12 +00:00

base

Reland "Move webrtc/{base => rtc_base} (stub headers)

2017-06-29 06:21:49 +00:00

call

nit: Rename RtpDemuxer::sink_ to RtpDemuxer::ssrc_sinks_

2017-06-28 15:18:51 +00:00

common_audio

Put attribute before function name instead of after, as required by GCC

2017-06-26 08:32:40 +00:00

common_video

Return WrappedI444Buffer in VP9Impl

2017-06-23 03:28:06 +00:00

examples

Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002 )

2017-06-29 06:04:25 +00:00

logging

Limit the number of simultaneous event logs.

2017-06-28 13:40:51 +00:00

media

Reland of Periodically update codec bit/frame rate settings.

2017-06-27 14:06:52 +00:00

modules

Don't disable FEC if timing frames are disabled.

2017-06-29 09:27:48 +00:00

ortc

Fixing bad use of std::sort in test method.

2017-06-27 14:52:50 +00:00

p2p

Fixing RTCIceCandidatePairStats.nominated for ICE controlling agent.

2017-06-28 16:37:23 +00:00

Fixing RTCIceCandidatePairStats.nominated for ICE controlling agent.

2017-06-28 16:37:23 +00:00

rtc_base

Reland "Move webrtc/{base => rtc_base} (stub headers)

2017-06-29 06:21:49 +00:00

sdk

Do not copy I420 frames in the decoder when not necessary.

2017-06-29 08:10:16 +00:00

stats

Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

2017-03-01 09:02:45 +00:00

system_wrappers

Prevent data race in GetStaticInstance

2017-05-29 07:01:14 +00:00

test

Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002 )

2017-06-29 06:04:25 +00:00

tools

Adding newline at the end of .proto files

2017-06-29 07:09:12 +00:00

video

Fix deadlock in webrtc_perf_tests

2017-06-27 14:21:01 +00:00

voice_engine

Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.

2017-06-21 08:05:22 +00:00

.gitignore

Flesh out webrtc/.gitignore

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

BUILD.gn

Add webrtc/rtc_base skeleton.

2017-06-21 12:50:22 +00:00

codereview.settings

Give a more specific URL for creating WebRTC checkout

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

common_types.cc

Allow parsing empty RTCP TargetBitrate messages, but stop sending them.

2017-06-22 12:40:25 +00:00

common_types.h

Allow parsing empty RTCP TargetBitrate messages, but stop sending them.

2017-06-22 12:40:25 +00:00

config.cc

Implement timing frames.

2017-06-19 14:18:55 +00:00

config.h

Implement timing frames.

2017-06-19 14:18:55 +00:00

DEPS

Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002 )

2017-06-29 06:04:25 +00:00

LICENSE

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

LICENSE_THIRD_PARTY

Delete unused spreadsort implementation.

2016-12-05 11:03:26 +00:00

OWNERS

Don't add or rename files in webrtc/ and webrtc/api/ without a proper review

2017-05-08 13:01:54 +00:00

PATENTS

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

PRESUBMIT.py

PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.

2014-05-23 17:27:18 +00:00

README.chromium

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

2017-04-06 11:32:27 +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 field trial for balanced degradation preference.

2017-06-14 12:52:21 +00:00

webrtc.gni

Adds detection of audio glitches for playout on iOS (reland)

2017-06-13 16:00:18 +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.