platform-external-webrtc

Files

minyue-webrtc adb58b88a1 Renable some Opus tests after Opus 1.2.1 update.

Bug: webrtc:8024
Change-Id: Ia7b9de70ef85e4ac32a7b84088b79cc6a260cc69
Reviewed-on: https://chromium-review.googlesource.com/586867
Reviewed-by: Felicia Lim <flim@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19164}

2017-07-27 07:40:14 +00:00

api

Add "max_ipv6_networks" field to RTCConfiguration.

2017-07-26 23:50:11 +00:00

audio

Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"

2017-07-26 14:28:51 +00:00

base

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

call

Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.

2017-07-26 09:09:44 +00:00

common_audio

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

common_video

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

examples

Update native plugin dll for turn servers and video.

2017-07-25 23:04:31 +00:00

logging

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

media

Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.

2017-07-26 09:09:44 +00:00

modules

Renable some Opus tests after Opus 1.2.1 update.

2017-07-27 07:40:14 +00:00

ortc

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

p2p

Relanding: Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting.

2017-07-26 23:09:33 +00:00

Add "max_ipv6_networks" field to RTCConfiguration.

2017-07-26 23:50:11 +00:00

rtc_base

Relanding: Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting.

2017-07-26 23:09:33 +00:00

rtc_tools

Print general usage information for event_log_analyzer

2017-07-20 15:05:09 +00:00

sdk

Adding Obj-C binding for RTCConfiguration::max_ipv6_networks.

2017-07-27 01:25:43 +00:00

stats

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

system_wrappers

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

test

Rename isolated_output to test_output and add a method to get the test_output directory.

2017-07-25 14:31:18 +00:00

video

Report minimum PSNR in VideoQualityTest and save corresponding frame to file

2017-07-25 12:45:03 +00:00

voice_engine

Move total audio energy and duration tracking to AudioLevel and protect with existing critial section.

2017-07-20 16:57:42 +00:00

.gitignore

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BUILD.gn

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

codereview.settings

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common_types.cc

Rename webrtc/base -> webrtc/rtc_base

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

common_types.h

Add parsing/serializing for MID RTP header extension.

2017-07-21 17:33:25 +00:00

config_unittest.cc

Rename webrtc/base -> webrtc/rtc_base

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

config.cc

Rename webrtc/base -> webrtc/rtc_base

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

config.h

Rename webrtc/base -> webrtc/rtc_base

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

DEPS

Move webrtc/{tools => rtc_tools}

2017-06-30 10:04:59 +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

Report interframe delay sum in old GetStats

2017-07-07 08:26:24 +00:00

video_send_stream.h

Move RTP keep-alive config from VideoSendStream::Config to Call::Config

2017-07-11 10:44:17 +00:00

webrtc.gni

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

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

whitespace.txt

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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.