webrtc_m130

History

philipel 2c9f9f2bc9 Only create H264 frames if there are no gaps in the packet sequence number.

In the case of H264 we can't know which packet that is the fist packet of a
frame. In order to avoid creating incomplete frames we keep track of which
packets that we haven't received, and if there is a gap in the packet sequence
number leading up to this frame then a frame wont be created.

BUG=chromium:716558

Review-Url: https://codereview.webrtc.org/2926083002
Cr-Commit-Position: refs/heads/master@{#18559}

2017-06-13 09:47:28 +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

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

base

Delete ApplicationName and OrganizationName.

2017-06-13 09:04:51 +00:00

call

Use rtp_header_extension_map.h instead of rtp_header_extension.h

2017-06-12 14:01:20 +00:00

common_audio

Moving compile_assert_c.h to webrtc/base

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

common_video

Update webrtc/test to new VideoFrameBuffer interface

2017-06-11 16:10:16 +00:00

examples

Fall-back to OpenGL renderer if mac hardware doesn't support Metal

2017-06-12 14:32:02 +00:00

logging

Change rtp header extension AbsoluteSendTime::Write to take time in 24bit format

2017-06-13 09:08:14 +00:00

media

Increase number of unsignaled audio streams we handle to 4.

2017-06-13 07:38:27 +00:00

modules

Only create H264 frames if there are no gaps in the packet sequence number.

2017-06-13 09:47:28 +00:00

ortc

Remove VirtualSocketServer's dependency on PhysicalSocketServer.

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

p2p

Add SafeClamp(), which accepts args of different types

2017-06-12 18:40:47 +00:00

s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine

2017-06-12 08:16:46 +00:00

sdk

Access UIApplication on main thread

2017-06-13 09:37:47 +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

Use rtp_header_extension_map.h instead of rtp_header_extension.h

2017-06-12 14:01:20 +00:00

tools

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

video

Only create H264 frames if there are no gaps in the packet sequence number.

2017-06-13 09:47:28 +00:00

voice_engine

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

.gitignore

Flesh out webrtc/.gitignore

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

BUILD.gn

Split rtc_task_queue target. Add separate target for sequenced_task_checker and weak_ptr.

2017-06-12 12:53:35 +00:00

codereview.settings

Give a more specific URL for creating WebRTC checkout

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

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

…

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

…

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

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

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.