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henrika bc9ffad966 Adds support for dynamic buffer size handling on recording side for iOS. 
Will also ensure that full-duplex audio now works on iOS simulators.

Bug: b/37580746
Change-Id: Iab1af39b0e6e6c124435814558caf77c474bd612
Reviewed-on: https://chromium-review.googlesource.com/519246
Commit-Queue: Henrik Andreasson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18381}
2017-06-01 13:05:59 +00:00
..
api
Add separate base classes for I420 and I444 buffers
2017-05-30 08:21:59 +00:00
audio
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.
2017-06-01 11:02:35 +00:00
base
Support epoll in PhysicalSocketServer.
2017-05-31 20:09:18 +00:00
call
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.
2017-06-01 11:02:35 +00:00
common_audio
Moving compile_assert_c.h to webrtc/base
2017-05-15 15:06:47 +00:00
common_video
Add separate base classes for I420 and I444 buffers
2017-05-30 08:21:59 +00:00
examples
Switch from ScheduledExecutorService to ExecutorService.
2017-06-01 12:05:53 +00:00
logging
Print configured header extensions and codecs in rtc_event_log2text.
2017-06-01 12:23:03 +00:00
media
Recreate FlexfecReceiveStream separately from VideoReceiveStream.
2017-05-30 08:31:37 +00:00
modules
Adds support for dynamic buffer size handling on recording side for iOS.
2017-06-01 13:05:59 +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
pc
Reland of Removes usage of native base::android::GetApplicationContext()
2017-05-26 08:51:53 +00:00
sdk
Remove native method VideoTrack.free which doesn't exist.
2017-06-01 11:24:49 +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
Delete unneeded includes of system_wrappers/include/sleep.h
2017-06-01 12:34:08 +00:00
tools
Fix indexing error in event log analyzer.
2017-06-01 10:50:31 +00:00
video
Delete unneeded includes of system_wrappers/include/sleep.h
2017-06-01 12:34:08 +00:00
voice_engine
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.
2017-06-01 11:02:35 +00:00
.gitignore
…
BUILD.gn
Create unit tests for RtpDemuxer
2017-05-31 16:12:25 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
common_types.h
Delete RawVideoType enum, use the VideoType enum instead.
2017-04-28 14:18:05 +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
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +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
…
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.