platform-external-webrtc

Files

Emad Omara c6de0c98af Upgrade to (D)TLS1.2 using the new BoringSSL (D)TLSv1_2_method functions

Bug: webrtc:7865
Change-Id: I39344f385181132fe2e0f832eec1cf8fe0736dfe
Reviewed-on: https://chromium-review.googlesource.com/543795
Commit-Queue: Emad Omara <emadomara@google.com>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18706}

2017-06-22 00:30:04 +00:00

api

Implement timing frames.

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

audio

Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.

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

base

Upgrade to (D)TLS1.2 using the new BoringSSL (D)TLSv1_2_method functions

2017-06-22 00:30:04 +00:00

call

Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.

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

common_audio

Moving compile_assert_c.h to webrtc/base

2017-05-15 15:06:47 +00:00

common_video

Reland of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2949953003/ )

2017-06-21 08:02:59 +00:00

examples

Use information about blacklisted devices in video_quality_loopback_test

2017-06-19 08:16:45 +00:00

logging

Support building WebRTC without audio and video.

2017-06-15 19:52:32 +00:00

media

Reland of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2949953003/ )

2017-06-21 08:02:59 +00:00

modules

Fix timing frames and FEC conflict

2017-06-21 15:23:19 +00:00

ortc

Remove VirtualSocketServer's dependency on PhysicalSocketServer.

2017-05-17 01:00:06 +00:00

p2p

Enable SNI in ssl adapter.

2017-06-16 23:30:48 +00:00

Reland of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2949953003/ )

2017-06-21 08:02:59 +00:00

rtc_base

Add webrtc/rtc_base skeleton.

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

sdk

Fix AVFoundation framework import

2017-06-21 10:44:05 +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

Added new AudioProcessing fuzzer

2017-06-20 12:26:55 +00:00

tools

Add NetEq delay plotting to event_log_visualizer

2017-06-14 13:09:58 +00:00

video

What can't loopback test be more like full stack test?

2017-06-21 16:00:17 +00:00

voice_engine

Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.

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

.gitignore

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BUILD.gn

Add webrtc/rtc_base skeleton.

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

codereview.settings

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common_types.cc

Implement timing frames.

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

common_types.h

Implement timing frames.

2017-06-19 14:18:55 +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

Delete old include file webrtc/video_frame.h.

2017-06-13 13:05:05 +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

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.