webrtc_m130

History

mbonadei d66072b71b Moving asm code out of common_audio_c sources list

BUG=webrtc:7743

Review-Url: https://codereview.webrtc.org/2966173002
Cr-Commit-Position: refs/heads/master@{#18916}

2017-07-06 14:44:14 +00:00

api

Report timing frames info in GetStats.

2017-07-06 10:06:50 +00:00

audio

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

base

Delete SignalThread class.

2017-06-29 13:21:20 +00:00

call

Reland of Add received audio/video call duration metrics based on packets.

2017-07-04 11:05:06 +00:00

common_audio

Moving asm code out of common_audio_c sources list

2017-07-06 14:44:14 +00:00

common_video

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

examples

Fixes build issue based on usage of Android O specific API

2017-07-04 14:02:35 +00:00

logging

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

media

Update screen simulcast config

2017-07-06 12:05:50 +00:00

modules

Let NetEq reset the AudioFrame during muted state

2017-07-06 12:23:53 +00:00

ortc

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

p2p

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

Report timing frames info in GetStats.

2017-07-06 10:06:50 +00:00

rtc_base

Reland "Update includes for webrtc/{base => rtc_base} rename (3/3)"

2017-06-30 17:45:21 +00:00

rtc_tools

fix comment length

2017-07-03 07:47:44 +00:00

sdk

Fix null ref in NetworkMonitorAutoDetect if Connectivity Manager service is unavailable

2017-07-05 16:55:09 +00:00

stats

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

system_wrappers

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

test

Remove MAIN_NIB_FILE from Info.plist because the substitution is broken

2017-07-06 14:09:57 +00:00

video

Implement RTP keepalive in native stack.

2017-07-06 11:38:06 +00:00

voice_engine

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

.gitignore

…

BUILD.gn

Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ )

2017-06-30 15:11:44 +00:00

codereview.settings

…

common_types.cc

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

common_types.h

Implement RTP keepalive in native stack.

2017-07-06 11:38:06 +00:00

config_unittest.cc

Support encrypted RTP extensions (RFC 6904)

2017-06-29 19:31:36 +00:00

config.cc

Support encrypted RTP extensions (RFC 6904)

2017-06-29 19:31:36 +00:00

config.h

Support encrypted RTP extensions (RFC 6904)

2017-06-29 19:31:36 +00:00

DEPS

Move webrtc/{tools => rtc_tools}

2017-06-30 10:04:59 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

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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

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PRESUBMIT.py

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README.chromium

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typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

2017-04-06 11:32:27 +00:00

video_receive_stream.h

Report timing frames info in GetStats.

2017-07-06 10:06:50 +00:00

video_send_stream.h

Implement RTP keepalive in native stack.

2017-07-06 11:38:06 +00:00

webrtc.gni

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

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

whitespace.txt

…

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.