webrtc_m130

History

stefan b4ab381ce3 Use the configured remote ssrc instead of relying on the first received packet RtpStreamReceiver.

This solves an issue where if the first packet happens to be an RTX packet, it is recovered with an incorrect SSRC.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2928123002
Cr-Commit-Position: refs/heads/master@{#18512}

2017-06-09 13:12:11 +00:00

api

Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.

2017-06-02 21:37:37 +00:00

audio

New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.

2017-06-01 11:02:35 +00:00

base

Create a UIApplication when running tests on iOS.

2017-06-09 10:24:33 +00:00

call

Add RSID-based demuxing to RtpDemuxer

2017-06-08 11:19:13 +00:00

common_audio

Moving compile_assert_c.h to webrtc/base

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

common_video

Update I420Buffer to new VideoFrameBuffer interface

2017-06-01 17:02:26 +00:00

examples

Remove webrtc deps from AppRTCMobile.

2017-06-07 11:13:48 +00:00

logging

Delete rtc_event_log/ringbuffer.h

2017-06-07 08:53:30 +00:00

media

Revert of Periodically update codec bit/frame rate settings. (patchset #2 id:160001 of https://codereview.webrtc.org/2924023002/ )

2017-06-08 14:12:17 +00:00

modules

Rename class RtpStreamReceiver --> RtpVideoStreamReceiver.

2017-06-09 11:01:55 +00:00

ortc

Remove VirtualSocketServer's dependency on PhysicalSocketServer.

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

p2p

Fix issue with send-side bandwidth estimation over TURN TCP connections.

2017-05-27 01:40:05 +00:00

2017-06-02 21:37:37 +00:00

sdk

Ensure the openGLContext is current before trying to reshape the viewport

2017-06-09 12:16:10 +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

Create a UIApplication when running tests on iOS.

2017-06-09 10:24:33 +00:00

tools

Revert of Refactored incoming bitrate estimator. (patchset #8 id:140001 of https://codereview.webrtc.org/2917873002/ )

2017-06-08 07:10:31 +00:00

video

Use the configured remote ssrc instead of relying on the first received packet RtpStreamReceiver.

2017-06-09 13:12:11 +00:00

voice_engine

New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.

2017-06-01 11:02:35 +00:00

.gitignore

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BUILD.gn

Delete webrtc/call.h (replaced with webrtc/call/call.h).

2017-06-08 09:21:17 +00:00

codereview.settings

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common_types.cc

Add RSID-based demuxing to RtpDemuxer

2017-06-08 11:19:13 +00:00

common_types.h

Add RSID-based demuxing to RtpDemuxer

2017-06-08 11:19:13 +00:00

config.cc

Fix default extension id for video content type extension

2017-04-26 09:35:31 +00:00

config.h

Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )

2017-04-11 17:34:31 +00:00

DEPS

Delete webrtc/call.h (replaced with webrtc/call/call.h).

2017-06-08 09:21:17 +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_frame.h

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +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

Update adaptation stats to support degradations in both resolution and framerate.

2017-05-16 06:40:18 +00:00

webrtc.gni

Reland of Removes usage of native base::android::GetApplicationContext()

2017-05-26 08:51:53 +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.