webrtc_m130

History

terelius c8e0552c07 Limit the number of simultaneous event logs.

BUG=webrtc:7887

Review-Url: https://codereview.webrtc.org/2956003003
Cr-Commit-Position: refs/heads/master@{#18808}

2017-06-28 13:40:51 +00:00

api

Expose ILBC codec in webrtc/api/audio_codecs/

2017-06-28 09:05:04 +00:00

audio

Disable AudioBweIntegrationTest.NoBandwidthDropAfterDtx - it's flaky

2017-06-26 12:04:12 +00:00

base

Rename safe_cmp::{Eq,Ne,Lt,Le,Ge,Gt} to Safe{Eq,Ne,Lt,Le,Ge,Gt}

2017-06-26 08:31:31 +00:00

call

Reland of Periodically update codec bit/frame rate settings.

2017-06-27 14:06:52 +00:00

common_audio

Put attribute before function name instead of after, as required by GCC

2017-06-26 08:32:40 +00:00

common_video

Return WrappedI444Buffer in VP9Impl

2017-06-23 03:28:06 +00:00

examples

Notify delegates about audio glitches in real time

2017-06-27 09:43:27 +00:00

logging

Limit the number of simultaneous event logs.

2017-06-28 13:40:51 +00:00

media

Reland of Periodically update codec bit/frame rate settings.

2017-06-27 14:06:52 +00:00

modules

Workaround for hardware encoders crashing timing frames processing

2017-06-28 10:53:19 +00:00

ortc

Fixing bad use of std::sort in test method.

2017-06-27 14:52:50 +00:00

p2p

Handle the PROTO_TSL when getting the protocol priority.

2017-06-27 22:11:24 +00:00

Support getting external HMAC auth context with libsrtp 2.1.0.

2017-06-28 10:35:57 +00:00

rtc_base

Add webrtc/rtc_base skeleton.

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

sdk

Fix samplingMatrix for I420Frames converted from VideoFrame.

2017-06-28 07:58:42 +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

Expose ILBC codec in webrtc/api/audio_codecs/

2017-06-28 09:05:04 +00:00

tools

Add NetEq delay plotting to event_log_visualizer

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

video

Fix deadlock in webrtc_perf_tests

2017-06-27 14:21:01 +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

Allow parsing empty RTCP TargetBitrate messages, but stop sending them.

2017-06-22 12:40:25 +00:00

common_types.h

Allow parsing empty RTCP TargetBitrate messages, but stop sending them.

2017-06-22 12:40:25 +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

Delete unused spreadsort implementation.

2016-12-05 11:03:26 +00:00

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

…

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.