platform-external-webrtc

Files

stefan a86f57e624 Move GetOutstandingBytes() call inside field trial check to avoid taking an unneccessary lock.

BUG=webrtc:7926

Review-Url: https://codereview.webrtc.org/3009673002
Cr-Commit-Position: refs/heads/master@{#19540}

2017-08-26 13:33:43 +00:00

api

Remove dead code

2017-08-26 05:22:42 +00:00

audio

Add stats totalSamplesReceived and concealedSamples

2017-08-25 00:50:42 +00:00

call

Reverse |rtx_payload_types| map, and rename.

2017-08-25 11:44:25 +00:00

common_audio

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

common_video

Change logging method from WEBRTC_TRACE macro to LOG macro.

2017-08-22 07:29:13 +00:00

examples

Roll chromium_revision d323a482ee..7114a66134 (494468:497367) manually

2017-08-25 13:21:52 +00:00

logging

Move video send/receive stream headers to webrtc/call.

2017-08-22 12:43:23 +00:00

media

Reverse |rtx_payload_types| map, and rename.

2017-08-25 11:44:25 +00:00

modules

Move GetOutstandingBytes() call inside field trial check to avoid taking an unneccessary lock.

2017-08-26 13:33:43 +00:00

ortc

Wire up RTP keep-alive in ortc api.

2017-08-09 13:42:32 +00:00

p2p

Renamed dtlstransportchannel.h/.cc/_unittest.cc.

2017-08-24 01:05:50 +00:00

Add stats totalSamplesReceived and concealedSamples

2017-08-25 00:50:42 +00:00

rtc_base

Roll chromium_revision d323a482ee..7114a66134 (494468:497367) manually

2017-08-25 13:21:52 +00:00

rtc_tools

Move PacedSender ownership to RtpTransportControllerSend.

2017-08-22 14:16:49 +00:00

sdk

Android: Don't check ref count when releasing PeerConnectionInterface

2017-08-26 12:57:20 +00:00

stats

Add stats totalSamplesReceived and concealedSamples

2017-08-25 00:50:42 +00:00

system_wrappers

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

test

Roll chromium_revision d323a482ee..7114a66134 (494468:497367) manually

2017-08-25 13:21:52 +00:00

video

Update video_replay tool to be able to dump .jpg files.

2017-08-25 14:24:21 +00:00

voice_engine

Uncomment thread-checkers in ChannelProxy

2017-08-25 08:50:58 +00:00

.gitignore

…

BUILD.gn

Removing EXPAT_RELATIVE_PATH and rtc_relative_path.

2017-08-25 15:14:04 +00:00

codereview.settings

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common_types.cc

Make CodecType conversion functions non-optional.

2017-08-24 10:52:48 +00:00

common_types.h

Add stats totalSamplesReceived and concealedSamples

2017-08-25 00:50:42 +00:00

config_unittest.cc

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

config.cc

Audit of kConstants missing the const qualifier

2017-08-08 17:48:15 +00:00

config.h

Audit of kConstants missing the const qualifier

2017-08-08 17:48:15 +00:00

DEPS

Move video send/receive stream headers to webrtc/call.

2017-08-22 12:43:23 +00:00

LICENSE

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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

Move video send/receive stream headers to webrtc/call.

2017-08-22 12:43:23 +00:00

video_send_stream.h

Move video send/receive stream headers to webrtc/call.

2017-08-22 12:43:23 +00:00

webrtc.gni

Removing EXPAT_RELATIVE_PATH and rtc_relative_path.

2017-08-25 15:14:04 +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.