webrtc_m130

History

danilchap 74e8df8feb Fix rtcp_sender to support sdes with 31 chunk

Previosly it supported up to only 15 chunks which is a limit for csrcs in an rtp packet.

BUG=None

Review-Url: https://codereview.webrtc.org/2758533002
Cr-Commit-Position: refs/heads/master@{#17274}

2017-03-16 15:04:08 +00:00

api

Change the type of session_id() from string to int64_t.

2017-03-13 18:00:54 +00:00

audio

Move delay_based_bwe_ into CongestionController

2017-03-09 14:40:08 +00:00

base

Better handling of error condition in MediaCodecVideoEncoder.

2017-03-15 12:53:14 +00:00

call

Reland of write frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2748643002/ )

2017-03-15 14:48:54 +00:00

common_audio

iSAC fix entropy coder: Recently added DCHECK could in fact trigger

2017-03-13 12:28:47 +00:00

common_video

Add FullStack test for simulcast screenshare mode.

2017-03-09 17:23:30 +00:00

examples

Try re-enabling two PeerConnectionClientTests:

2017-03-15 12:50:13 +00:00

logging

Enable GN check for webrtc/logging

2017-03-08 10:12:11 +00:00

media

(Re)move VoE specific enums from common_types.h.

2017-03-16 08:20:23 +00:00

modules

Fix rtcp_sender to support sdes with 31 chunk

2017-03-16 15:04:08 +00:00

ortc

Add the function ToRtpParameters.

2017-03-09 01:15:06 +00:00

p2p

Making candidate pool size behave as decided in JSEP.

2017-03-10 23:18:00 +00:00

Making candidate pool size behave as decided in JSEP.

2017-03-10 23:18:00 +00:00

sdk

Better handling of error condition in MediaCodecVideoEncoder.

2017-03-15 12:53:14 +00:00

stats

Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

2017-03-01 09:02:45 +00:00

system_wrappers

Delete utf_util_win.h.

2017-03-14 08:32:50 +00:00

test

Reland of write frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2748643002/ )

2017-03-15 14:48:54 +00:00

tools

Improve error handling for ffmpeg operations

2017-03-15 15:27:31 +00:00

video

De-parameterized EndToEndTests.

2017-03-16 12:19:49 +00:00

voice_engine

Remove VoEHardware interface.

2017-03-15 15:08:07 +00:00

.gitignore

Flesh out webrtc/.gitignore

2015-11-16 19:02:02 +00:00

BUILD.gn

Remove voe_cmd_test.

2017-03-07 09:43:21 +00:00

call.h

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

codereview.settings

Give a more specific URL for creating WebRTC checkout

2016-04-01 07:36:53 +00:00

common_types.cc

Reject XR TargetBitrate items with unsupported layer indices

2016-12-06 14:09:00 +00:00

common_types.h

(Re)move VoE specific enums from common_types.h.

2017-03-16 08:20:23 +00:00

config.cc

Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

2017-02-26 02:15:09 +00:00

config.h

Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

2017-02-26 02:15:09 +00:00

DEPS

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

Delete unused spreadsort implementation.

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

no_op_function.cc

Moving no_op_function.cc out of webrtc/build

2017-01-24 13:25:12 +00:00

OWNERS

Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it

2016-11-29 10:52:22 +00:00

PATENTS

…

PRESUBMIT.py

PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.

2014-05-23 17:27:18 +00:00

README.chromium

…

typedefs.h

Only define NO_RETURN if undefined

2017-01-30 08:54:19 +00:00

video_decoder.h

Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )

2017-02-09 12:53:45 +00:00

video_encoder.h

Move usage of QualityScaler to ViEEncoder.

2016-11-29 09:44:22 +00:00

video_frame.h

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

video_receive_stream.h

Add |protected_by_flexfec| flag to VideoReceiveStream::Config.

2017-03-09 14:13:20 +00:00

video_send_stream.h

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +00:00

webrtc.gni

Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )

2017-03-04 03:41:59 +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.