platform-external-webrtc

Files

Alex Loiko 300ec8c8db Remove WEBRTC_TRACE from webrtc/modules/audio_coding

We'd like to remove all occurrences of WEBRTC_TRACE and delete the
macro! One logging mechanism is enough.


NOTRY=True

Bug: webrtc:5118
Change-Id: Ic226318e0aebe3a71785fcb4ce07371872ab7128
Reviewed-on: https://chromium-review.googlesource.com/518133
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18712}

2017-06-22 10:05:51 +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

Enable more unittests on iOS, and disable those that fail on simulator

2017-06-22 08:47:20 +00:00

modules

Remove WEBRTC_TRACE from webrtc/modules/audio_coding

2017-06-22 10:05:51 +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

Fix the binary size regression on Chromium Windows.

2017-06-22 08:28:59 +00:00

rtc_base

Add webrtc/rtc_base skeleton.

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

sdk

Enable more unittests on iOS, and disable those that fail on simulator

2017-06-22 08:47:20 +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

Enable more unittests on iOS, and disable those that fail on simulator

2017-06-22 08:47:20 +00:00

voice_engine

Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.

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

.gitignore

…

BUILD.gn

Add webrtc/rtc_base skeleton.

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

codereview.settings

…

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

…

LICENSE_THIRD_PARTY

…

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

…

PRESUBMIT.py

…

README.chromium

…

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.